diff options
Diffstat (limited to 'sound')
207 files changed, 11567 insertions, 7087 deletions
diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index ce431e6e07c..5066a3768b2 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; diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 823359ed95e..a61d7a9a995 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); @@ -164,8 +198,8 @@ 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); diff --git a/sound/arm/pxa2xx-pcm.c b/sound/arm/pxa2xx-pcm.c index 26422a3584e..69a2455b447 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 65f86b56ba4..2a8fc08d52a 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 c0c2f57a0d6..313f22e9d92 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 diff --git a/sound/core/Makefile b/sound/core/Makefile index 43d4117428a..5e890cfed42 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 99db892d729..98969541cbc 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/soc/soc-dmaengine-pcm.c b/sound/core/pcm_dmaengine.c index aa924d9b798..aa924d9b798 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 82bb029d441..6e03b465e44 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); diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 11048cc744d..915b4d7fbb2 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; diff --git a/sound/firewire/speakers.c b/sound/firewire/speakers.c index 2c638650394..fe9e6e2f2c5 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; } diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index 9942691cc0c..afef0d73807 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 */ diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 103b33373fd..6effe99bbb9 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", @@ -1167,7 +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/oss/dmabuf.c b/sound/oss/dmabuf.c index a59c88818f4..461d94cfecb 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/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 59c5e9c03d5..8de66ccd727 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -152,14 +152,9 @@ config SND_HDA_CODEC_HDMI This module is automatically loaded at probing. config SND_HDA_I915 - bool "Build Display HD-audio controller/codec power well support for i915 cards" + bool + default y depends on DRM_I915 - help - Say Y here to include full HDMI and DisplayPort HD-audio controller/codec - power-well support for Intel Haswell graphics cards based on the i915 driver. - - Note that this option must be enabled for Intel Haswell C+ stepping machines, otherwise - the GPU audio controller/codecs will not be initialized or damaged when exit from S3 mode. config SND_HDA_CODEC_CIRRUS bool "Build Cirrus Logic codec support" diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 7c11d46b84d..48a9d004d6d 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 8a005f0e5ca..5b6c4e3c92c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -666,6 +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 @@ -1216,11 +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); } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 701c2e069b1..7aa9870040c 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, @@ -885,6 +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 */ @@ -972,6 +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); diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 8e77cbbad87..ac41e9cdc97 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -142,6 +142,9 @@ static void parse_user_hints(struct hda_codec *codec) 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; @@ -522,7 +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) @@ -624,7 +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; } @@ -648,7 +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; } @@ -813,6 +816,8 @@ static void resume_path_from_idx(struct hda_codec *codec, int path_idx) 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, @@ -830,7 +835,13 @@ static const struct snd_kcontrol_new control_templates[] = { .put = hda_gen_mixer_mute_put, /* replaced */ .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0), }, - HDA_BIND_MUTE(NULL, 0, 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 */ @@ -937,8 +948,8 @@ static int add_stereo_sw(struct hda_codec *codec, const char *pfx, } /* playback mute control with the software mute bit check */ -static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +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; @@ -949,10 +960,22 @@ static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol, 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) { @@ -1541,7 +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) @@ -1554,7 +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); @@ -1582,7 +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; @@ -1600,7 +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 */ @@ -3724,7 +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]; @@ -3733,14 +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; @@ -3752,10 +3780,19 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, 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 << nid); + spec->mute_bits |= (1ULL << mute_nid); else - spec->mute_bits &= ~(1ULL << nid); + spec->mute_bits &= ~(1ULL << mute_nid); set_pin_eapd(codec, nid, !mute); continue; } @@ -3786,14 +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; @@ -3801,8 +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 */ @@ -3815,8 +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); @@ -3887,7 +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 e199a852388..48d44026705 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -220,6 +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 */ diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index ce67608734b..fe0bda19de1 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_intel.c b/sound/pci/hda/hda_intel.c index 8860dd52952..e54ebd53084 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -555,6 +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 @@ -1160,7 +1163,7 @@ static int azx_reset(struct azx *chip, int full_reset) goto __skip; /* clear STATESTS */ - azx_writeb(chip, STATESTS, STATESTS_INT_MASK); + azx_writew(chip, STATESTS, STATESTS_INT_MASK); /* reset controller */ azx_enter_link_reset(chip); @@ -1242,7 +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); @@ -1397,8 +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); @@ -1409,7 +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; } @@ -1451,8 +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); @@ -2971,6 +2975,16 @@ 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); @@ -2983,11 +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; } @@ -2996,6 +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; @@ -3078,13 +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); } @@ -3139,6 +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 @@ -3831,11 +3883,13 @@ static int azx_probe_continue(struct azx *chip) /* 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); } @@ -3887,7 +3941,7 @@ 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) + if ((chip->driver_caps & AZX_DCAPS_PM_RUNTIME) || chip->use_vga_switcheroo) pm_runtime_put_noidle(&pci->dev); return 0; diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 3fd2973183e..05b3e3e9108 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; diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index ec12abd4526..379420c44ee 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_proc.c b/sound/pci/hda/hda_proc.c index 9760f001916..a8cb22eec89 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -582,6 +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) { @@ -751,6 +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 d97f0d61a15..0cbdd87dde6 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; @@ -1681,432 +460,11 @@ 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) { @@ -2172,7 +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; @@ -2205,110 +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 @@ -2395,90 +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) { @@ -2509,569 +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, @@ -3220,7 +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; @@ -3234,12 +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: @@ -3247,169 +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 @@ -3423,167 +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 @@ -3599,14 +930,34 @@ 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}, + {}, + }; switch (action) { case HDA_FIXUP_ACT_PRE_PROBE: - spec->gen.vmaster_mute.hook = ad_vmaster_eapd_hook; + 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) @@ -3617,9 +968,18 @@ static void ad1884_fixup_hp_eapd(struct hda_codec *codec, } } +/* 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[] = { @@ -3633,15 +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; @@ -3674,1170 +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 * @@ -4850,299 +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; @@ -5169,110 +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_conexant.c b/sound/pci/hda/patch_conexant.c index de00ce16647..4edd2d0f9a3 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; @@ -3200,6 +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; } @@ -3224,6 +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, @@ -3246,6 +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) */ @@ -3302,6 +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[] = { @@ -3311,6 +3384,7 @@ 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), @@ -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 030ca8652a1..9a58893d52a 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -67,6 +67,8 @@ 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 +553,17 @@ 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 +881,19 @@ 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; + eld = &per_pin->sink_eld; if (!eld->monitor_present) return; @@ -959,6 +973,7 @@ 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 +982,8 @@ 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); @@ -1329,6 +1344,7 @@ 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, @@ -1338,6 +1354,18 @@ 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 (codec->vendor_id == 0x80862807 && + eld->eld_valid && !old_eld_valid && per_pin->setup) { + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_UNMUTE); + hdmi_setup_audio_infoframe(codec, per_pin, + per_pin->non_pcm); + } } mutex_unlock(&pin_eld->lock); @@ -1510,14 +1538,17 @@ 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); } @@ -1557,6 +1588,9 @@ 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; @@ -1692,8 +1726,7 @@ 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; } @@ -1781,6 +1814,9 @@ 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); @@ -1989,8 +2025,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; - if (codec->vendor_id == 0x80862807) + if (codec->vendor_id == 0x80862807) { codec->patch_ops.set_power_state = haswell_set_power_state; + codec->dp_mst = true; + } generic_hdmi_init_per_pins(codec); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8bd22614986..9e9378cde8f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -282,6 +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 */ @@ -826,7 +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 @@ -1031,6 +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, @@ -1089,6 +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[]) { @@ -1341,6 +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. @@ -1843,8 +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[] = { @@ -2523,6 +2537,7 @@ enum { ALC269_TYPE_ALC269VD, ALC269_TYPE_ALC280, ALC269_TYPE_ALC282, + ALC269_TYPE_ALC283, ALC269_TYPE_ALC284, ALC269_TYPE_ALC286, }; @@ -2548,6 +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; @@ -2573,15 +2589,81 @@ 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, @@ -2712,6 +2794,7 @@ static int alc269_resume(struct hda_codec *codec) hda_call_check_power_status(codec, 0x01); if (spec->has_alc5505_dsp) alc5505_dsp_resume(codec); + return 0; } #endif /* CONFIG_PM */ @@ -3251,6 +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) @@ -3258,11 +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); } } @@ -3298,6 +3404,45 @@ 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; + } +} + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -3334,6 +3479,7 @@ enum { ALC269_FIXUP_ACER_AC700, ALC269_FIXUP_LIMIT_INT_MIC_BOOST, ALC269VB_FIXUP_ORDISSIMO_EVE2, + ALC283_FIXUP_CHROME_BOOK, }; static const struct hda_fixup alc269_fixups[] = { @@ -3585,11 +3731,20 @@ static const struct hda_fixup alc269_fixups[] = { { } }, }, + [ALC283_FIXUP_CHROME_BOOK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc283_fixup_chromebook, + }, }; 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), @@ -3627,6 +3782,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { 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, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), @@ -3645,11 +3801,6 @@ 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), @@ -3660,8 +3811,16 @@ 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 */ @@ -3830,11 +3989,15 @@ 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; @@ -3862,7 +4025,8 @@ static int patch_alc269(struct hda_codec *codec) 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); @@ -4326,9 +4490,11 @@ 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 e2f83591161..fba0cef1c47 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); + } } } @@ -2817,6 +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), @@ -3228,10 +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 }, {} @@ -3239,6 +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[]) { @@ -3267,6 +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" }, {} }; @@ -3612,20 +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); @@ -3915,6 +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 e2481baddc7..0bc20ef5687 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; } } diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 2a8ad9d1a2a..bb9ebc5543d 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: @@ -2397,6 +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/hdspm.c b/sound/pci/rme9652/hdspm.c index bd501931ee2..3cde55b753e 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 @@ -461,7 +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) */ @@ -537,36 +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", @@ -629,7 +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[] = { @@ -638,14 +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[] = { @@ -653,14 +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[] = { @@ -668,7 +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[] = { @@ -745,8 +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, @@ -760,7 +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, @@ -773,7 +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, @@ -788,7 +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, @@ -802,7 +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, @@ -817,7 +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, @@ -856,11 +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 */ }; @@ -879,7 +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; @@ -941,7 +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; @@ -976,12 +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[] = { @@ -992,6 +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 */ @@ -1107,14 +1250,11 @@ static int hdspm_rate_multiplier(struct hdspm *hdspm, int rate) else if (hdspm->control_register & HDSPM_DoubleSpeed) return rate * 2; - }; + } return rate; } -static int hdspm_tco_sync_check(struct hdspm *hdspm); -static int hdspm_sync_in_sync_check(struct hdspm *hdspm); - -/* 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; @@ -1127,17 +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: @@ -2129,6 +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; } @@ -2152,6 +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; } @@ -2183,6 +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 @@ -2196,15 +2378,23 @@ 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) \ @@ -2270,7 +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; @@ -2289,28 +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; } @@ -2359,33 +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; } @@ -2809,16 +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; } @@ -2873,19 +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; @@ -2898,7 +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; } } @@ -2912,31 +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; } @@ -2964,7 +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; } @@ -3010,7 +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; @@ -3027,19 +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; } } @@ -3067,16 +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; } @@ -3141,7 +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; } @@ -3203,7 +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; } @@ -3276,7 +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; } @@ -3313,6 +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, \ @@ -3352,7 +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; } @@ -3587,7 +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; } @@ -3595,7 +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; } @@ -3745,9 +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 @@ -3807,7 +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; @@ -3975,7 +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; } @@ -4021,7 +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; } @@ -4066,7 +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; } @@ -4112,7 +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; @@ -4159,7 +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; } @@ -4284,7 +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), @@ -4298,7 +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), @@ -4335,7 +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[] = { @@ -4345,7 +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), @@ -4501,77 +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); @@ -4665,6 +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"); @@ -4768,6 +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"); } @@ -4909,11 +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"); } @@ -5097,7 +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 */ @@ -5123,9 +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); @@ -5371,6 +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)) @@ -6013,7 +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) { @@ -6479,10 +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; @@ -6490,6 +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; @@ -6558,6 +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), @@ -6565,7 +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; } @@ -6580,10 +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; diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 45eeaa9f7fe..5138b849305 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" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index bc0261476d7..61a64d28190 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 ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),) snd-soc-core-objs += soc-generic-dmaengine-pcm.o endif diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 3fdd87fa18a..e48d38a1b95 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -13,6 +13,7 @@ config SND_ATMEL_SOC_PDC config SND_ATMEL_SOC_DMA tristate depends on SND_ATMEL_SOC + select SND_SOC_GENERIC_DMAENGINE_PCM config SND_ATMEL_SOC_SSC tristate @@ -32,6 +33,26 @@ config SND_AT91_SOC_SAM9G20_WM8731 Say Y if you want to add support for SoC audio on WM8731-based AT91sam9g20 evaluation board. +config SND_ATMEL_SOC_WM8904 + tristate "Atmel ASoC driver for boards using WM8904 codec" + depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC + select SND_ATMEL_SOC_SSC + select SND_ATMEL_SOC_DMA + select SND_SOC_WM8904 + help + Say Y if you want to add support for Atmel ASoC driver for boards using + WM8904 codec. + +config SND_AT91_SOC_SAM9X5_WM8731 + tristate "SoC Audio support for WM8731-based at91sam9x5 board" + depends on ATMEL_SSC && SND_ATMEL_SOC && SOC_AT91SAM9X5 + select SND_ATMEL_SOC_SSC + select SND_ATMEL_SOC_DMA + select SND_SOC_WM8731 + help + Say Y if you want to add support for audio SoC on an + at91sam9x5 based board that is using WM8731 codec. + config SND_AT91_SOC_AFEB9260 tristate "SoC Audio support for AFEB9260 board" depends on ARCH_AT91 && ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index 41967ccb6f4..5baabc8bde3 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -11,6 +11,10 @@ obj-$(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 d1282652679..06082e5e5dc 100644 --- a/sound/soc/atmel/atmel-pcm-dma.c +++ b/sound/soc/atmel/atmel-pcm-dma.c @@ -91,138 +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 f3fdfa07fcb..0ecf356027f 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, IER, 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 00000000000..7222380131e --- /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/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c new file mode 100644 index 00000000000..992ae38d5a1 --- /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 d6f7694fcad..c8a2de103c5 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -341,7 +341,7 @@ static struct platform_driver au1xac97c_driver = { .remove = au1xac97c_drvremove, }; -module_platform_driver(&au1xac97c_driver); +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 a497a0cfeba..decba87a074 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 a822ab822bb..986dcec79fa 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -379,9 +379,6 @@ 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; - wd->mmio = devm_ioremap_resource(&pdev->dev, iores); if (IS_ERR(wd->mmio)) return PTR_ERR(wd->mmio); diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index efb1daecd0d..e82eb373a73 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -294,11 +294,12 @@ static int asoc_bfin_ac97_probe(struct platform_device *pdev) /* Request PB3 as reset pin */ ret = devm_gpio_request_one(&pdev->dev, CONFIG_SND_BF5XX_RESET_GPIO_NUM, - GPIOF_OUT_INIT_HIGH, "SND_AD198x RESET") { + 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); - goto gpio_err; + return ret; } #endif diff --git a/sound/soc/blackfin/bf5xx-ac97.h b/sound/soc/blackfin/bf5xx-ac97.h index 15c635e33f4..a680fdc9bb4 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/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index ac73c607410..efa75b5086a 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, }; @@ -363,9 +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); diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index 17ad70bca9f..a57643d6402 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -376,9 +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); @@ -411,7 +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); @@ -426,7 +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/codecs/Kconfig b/sound/soc/codecs/Kconfig index badb6fbacaa..15106c04547 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 @@ -20,6 +21,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD73311 select SND_SOC_ADAU1373 if I2C 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 @@ -54,6 +56,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C 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 @@ -122,6 +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 @@ -145,8 +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 @@ -198,6 +205,9 @@ config SND_SOC_AK4104 config SND_SOC_AK4535 tristate +config SND_SOC_AK4554 + tristate + config SND_SOC_AK4641 tristate @@ -292,6 +302,12 @@ config SND_SOC_MAX9850 config SND_SOC_HDMI_CODEC tristate +config SND_SOC_PCM1681 + tristate + +config SND_SOC_PCM1792A + tristate + config SND_SOC_PCM3008 tristate @@ -500,6 +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 70fd8066f54..bc126764a44 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 @@ -42,6 +43,8 @@ snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.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 @@ -114,6 +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 @@ -138,6 +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 @@ -171,6 +176,8 @@ 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_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-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o @@ -239,6 +246,7 @@ obj-$(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/ac97.c b/sound/soc/codecs/ac97.c index ec7351803c2..8d9ba4ba4bf 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) { @@ -117,6 +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 89fcf7d6e7b..7257a8885f4 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) { @@ -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 b1f2baf42b4..5fac8adbc13 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 d1124a5b347..ebff1128be5 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -91,7 +91,7 @@ #define ADAU1701_OSCIPOW_OPD 0x04 #define ADAU1701_DACSET_DACINIT 1 -#define ADAU1707_CLKDIV_UNSET (-1UL) +#define ADAU1707_CLKDIV_UNSET (-1U) #define ADAU1701_FIRMWARE "adau1701.bin" @@ -247,21 +247,21 @@ static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv) gpio_is_valid(adau1701->gpio_pll_mode[1])) { switch (clkdiv) { case 64: - gpio_set_value(adau1701->gpio_pll_mode[0], 0); - gpio_set_value(adau1701->gpio_pll_mode[1], 0); + 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(adau1701->gpio_pll_mode[0], 0); - gpio_set_value(adau1701->gpio_pll_mode[1], 1); + 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(adau1701->gpio_pll_mode[0], 1); - gpio_set_value(adau1701->gpio_pll_mode[1], 0); + 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(adau1701->gpio_pll_mode[0], 1); - gpio_set_value(adau1701->gpio_pll_mode[1], 1); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 1); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 1); break; } } @@ -269,10 +269,10 @@ static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv) adau1701->pll_clkdiv = clkdiv; if (gpio_is_valid(adau1701->gpio_nreset)) { - gpio_set_value(adau1701->gpio_nreset, 0); + gpio_set_value_cansleep(adau1701->gpio_nreset, 0); /* minimum reset time is 20ns */ udelay(1); - gpio_set_value(adau1701->gpio_nreset, 1); + gpio_set_value_cansleep(adau1701->gpio_nreset, 1); /* power-up time may be as long as 85ms */ mdelay(85); } @@ -734,7 +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); diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 3c839cc4e00..15b012d0f22 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 506d474c4d2..8f388edff58 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 c7cfdf957e4..71059c07ae7 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 00000000000..79e9555766c --- /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 1f303983ae0..72e953b2cb4 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 de625813c0e..657808ba141 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); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index b60b08ccc1d..9e81b639269 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -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, @@ -242,6 +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 index a081d9fcb16..c4cf0699e77 100644 --- a/sound/soc/codecs/bt-sco.c +++ b/sound/soc/codecs/bt-sco.c @@ -15,15 +15,27 @@ #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, @@ -31,7 +43,12 @@ static struct snd_soc_dai_driver bt_sco_dai = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_bt_sco; +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) { @@ -50,6 +67,9 @@ static struct platform_device_id bt_sco_driver_ids[] = { { .name = "dfbmcs320", }, + { + .name = "bt-sco", + }, {}, }; MODULE_DEVICE_TABLE(platform, bt_sco_driver_ids); diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 8e4779812b9..83c835d9fd8 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 03036b32673..a20f1bb8f07 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 987f728718c..be2ba1b6fe4 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/dmic.c b/sound/soc/codecs/dmic.c index 66967ba6f75..b2090b2a5e2 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/hdmi.c b/sound/soc/codecs/hdmi.c index 2bcae2b40c9..68342b121c9 100644 --- a/sound/soc/codecs/hdmi.c +++ b/sound/soc/codecs/hdmi.c @@ -23,11 +23,20 @@ #define DRV_NAME "hdmi-audio-codec" -static struct snd_soc_codec_driver 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 hdmi_codec_dai = { .name = "hdmi-hifi", .playback = { + .stream_name = "Playback", .channels_min = 2, .channels_max = 8, .rates = SNDRV_PCM_RATE_32000 | @@ -37,6 +46,25 @@ static struct snd_soc_dai_driver 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 hdmi_codec_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c index 9f9f59573f7..0e5743ea79d 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 a6ac2313047..31f91560e9f 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 3eeada57e87..566a367c94f 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 ad5313f98f2..0569a4c3ae0 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2084,8 +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) @@ -2132,8 +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; } diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c index 6b6c74cd83e..29549cdbf4c 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 5402dfbbb71..4d3c8fd8c5d 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; diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c new file mode 100644 index 00000000000..651ce092367 --- /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 00000000000..2a8eccf64c7 --- /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 00000000000..7a83d1fc102 --- /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 f2a6282b41f..b6618c4a759 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 index ce585e37e38..c26a8f814b1 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -50,8 +50,6 @@ static const struct regmap_range_cfg rt5640_ranges[] = { static struct reg_default init_list[] = { {RT5640_PR_BASE + 0x3d, 0x3600}, - {RT5640_PR_BASE + 0x1c, 0x0D21}, - {RT5640_PR_BASE + 0x1b, 0x0000}, {RT5640_PR_BASE + 0x12, 0x0aa8}, {RT5640_PR_BASE + 0x14, 0x0aaa}, {RT5640_PR_BASE + 0x20, 0x6110}, @@ -384,15 +382,11 @@ static const SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5640_snd_controls[] = { /* Speaker Output Volume */ - SOC_DOUBLE("Speaker Playback Switch", RT5640_SPK_VOL, - RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), 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 Playback Switch", RT5640_HP_VOL, - RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), 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, @@ -737,28 +731,21 @@ static const struct snd_kcontrol_new rt5640_mono_mix[] = { RT5640_M_BST1_MM_SFT, 1, 1), }; -/* INL/R source */ -static const char * const rt5640_inl_src[] = { - "IN2P", "MONOP" -}; +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 SOC_ENUM_SINGLE_DECL( - rt5640_inl_enum, RT5640_INL_INR_VOL, - RT5640_INL_SEL_SFT, rt5640_inl_src); +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 rt5640_inl_mux = - SOC_DAPM_ENUM("INL source", rt5640_inl_enum); +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 char * const rt5640_inr_src[] = { - "IN2N", "MONON" -}; - -static const SOC_ENUM_SINGLE_DECL( - rt5640_inr_enum, RT5640_INL_INR_VOL, - RT5640_INR_SEL_SFT, rt5640_inr_src); - -static const struct snd_kcontrol_new rt5640_inr_mux = - SOC_DAPM_ENUM("INR source", rt5640_inr_enum); +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[] = { @@ -891,33 +878,6 @@ static const SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5640_sdi_mux = SOC_DAPM_ENUM("SDI select", rt5640_sdi_sel_enum); -static int spk_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: - regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, - 0x0001, 0x0001); - regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, - 0xf000, 0xf000); - break; - - case SND_SOC_DAPM_PRE_PMD: - regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, - 0xf000, 0x0000); - regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, - 0x0001, 0x0000); - break; - - default: - return 0; - } - return 0; -} - static int rt5640_set_dmic1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -966,6 +926,117 @@ static int rt5640_set_dmic2_event(struct snd_soc_dapm_widget *w, 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), @@ -1005,9 +1076,6 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { 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), - /* IN Mux */ - SND_SOC_DAPM_MUX("INL Mux", SND_SOC_NOPM, 0, 0, &rt5640_inl_mux), - SND_SOC_DAPM_MUX("INR Mux", SND_SOC_NOPM, 0, 0, &rt5640_inr_mux), /* 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)), @@ -1158,15 +1226,28 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { 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("Improve HP Amp Drv", RT5640_PWR_ANLG1, - SND_SOC_NOPM, 0, NULL, 0), - SND_SOC_DAPM_PGA("HP L Amp", RT5640_PWR_ANLG1, + 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_PGA("HP R Amp", RT5640_PWR_ANLG1, + 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, - SND_SOC_NOPM, 0, spk_event, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + 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"), @@ -1407,9 +1488,11 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"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"}, @@ -1422,13 +1505,15 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"Mono MIX", "OUTVOL L Switch", "OUTVOL L"}, {"Mono MIX", "BST1 Switch", "BST1"}, - {"HP L Amp", NULL, "HPO MIX L"}, - {"HP R Amp", NULL, "HPO MIX R"}, + {"HP Amp", NULL, "HPO MIX L"}, + {"HP Amp", NULL, "HPO MIX R"}, - {"SPOLP", NULL, "SPOL MIX"}, - {"SPOLN", NULL, "SPOL MIX"}, - {"SPORP", NULL, "SPOR MIX"}, - {"SPORN", NULL, "SPOR MIX"}, + {"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"}, @@ -1438,8 +1523,10 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"HPOL", NULL, "Improve HP Amp Drv"}, {"HPOR", NULL, "Improve HP Amp Drv"}, - {"HPOL", NULL, "HP L Amp"}, - {"HPOR", NULL, "HP R Amp"}, + {"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"}, @@ -1818,17 +1905,13 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec, RT5640_PWR_BG | RT5640_PWR_VREF2, RT5640_PWR_VREF1 | RT5640_PWR_MB | RT5640_PWR_BG | RT5640_PWR_VREF2); - mdelay(10); + 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_DEPOP_M1, - 0x001d, 0x0019); - snd_soc_update_bits(codec, RT5640_DEPOP_M2, - 0x2000, 0x2000); snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); } @@ -1872,8 +1955,6 @@ static int rt5640_probe(struct snd_soc_codec *codec) rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); - snd_soc_update_bits(codec, RT5640_DEPOP_M1, 0x001d, 0x0019); - snd_soc_update_bits(codec, RT5640_DEPOP_M2, 0x2000, 0x2000); snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); snd_soc_update_bits(codec, RT5640_DSP_PATH2, 0xfc00, 0x0c00); @@ -2095,6 +2176,8 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, 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) diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index c48286d7118..5e8df25a13f 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -145,6 +145,8 @@ /* 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 @@ -153,6 +155,7 @@ #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 @@ -1201,6 +1204,14 @@ #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) @@ -2087,6 +2098,7 @@ struct rt5640_priv { 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 d659d3adcfb..1f4093f3f3a 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -153,6 +153,8 @@ 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_POST_PMU: snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, @@ -160,9 +162,17 @@ static int power_vag_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_PRE_PMD: - snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, - SGTL5000_VAG_POWERUP, 0); - msleep(400); + /* + * 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; @@ -388,7 +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", @@ -644,16 +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; } @@ -1470,6 +1483,7 @@ static struct snd_soc_codec_driver sgtl5000_driver = { 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, @@ -1527,6 +1541,9 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, 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; } diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 73e205c892a..38f3b105c17 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -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/spdif_receiver.c b/sound/soc/codecs/spdif_receiver.c index e9d7881ed2c..e3501f40c7b 100644 --- a/sound/soc/codecs/spdif_receiver.c +++ b/sound/soc/codecs/spdif_receiver.c @@ -23,11 +23,26 @@ #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", diff --git a/sound/soc/codecs/spdif_transmitter.c b/sound/soc/codecs/spdif_transmitter.c index 18280499fd5..a078aa31052 100644 --- a/sound/soc/codecs/spdif_transmitter.c +++ b/sound/soc/codecs/spdif_transmitter.c @@ -25,10 +25,24 @@ #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", diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index f8d30e5f637..492644e67ac 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 cfb55fe35e9..06edb396e73 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/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index b1f6982c7c9..7b8f3d965f4 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 17df4e32fea..2ed57d4aa44 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 e5b92688313..6e3f269243e 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -138,8 +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; @@ -147,10 +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); @@ -158,42 +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; } /* @@ -1492,6 +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); @@ -1582,6 +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 8e6e5b01602..1e3884d6b3f 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 44621ddc332..3c79dbb6c32 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -429,7 +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; } @@ -437,9 +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; diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 6d0aa44c375..c94d4c1e3da 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 54cd3da09ab..b7ab2ef567c 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 f5e835662cd..d5ebcb00019 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -410,39 +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; @@ -523,14 +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; @@ -670,14 +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"); diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 282fd232cdf..8bbddc151aa 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -998,6 +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 | @@ -1421,9 +1423,6 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "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" }, @@ -1499,23 +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"), @@ -1567,22 +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"), @@ -1614,6 +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, @@ -1781,6 +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 2e7cb4ba161..bbd64384ca1 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), @@ -432,6 +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 | @@ -842,9 +841,6 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "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" }, @@ -979,10 +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" }, @@ -1006,6 +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, @@ -1170,6 +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 0e8b3aaf6c8..af1318ddb06 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/wm8727.c b/sound/soc/codecs/wm8727.c index 462f5e4d5c0..7b1a6d5c11c 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 5276062d6c7..456bb8c6d75 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 0a4ab4c423d..d96ebf52d95 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 f1fdbf63abb..8092495605c 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 fa24cedee68..eebcb1da3b7 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; diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 4c9fb142cb2..4dfa8dceeab 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1012,7 +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" @@ -1202,7 +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 0a4ffdd1d2a..f156010e52b 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 e2de9ecfd64..11d80f3b613 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2621,8 +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; } @@ -3046,8 +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; @@ -3175,7 +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; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index ba832b77c54..86426a117b0 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -819,8 +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; @@ -1432,14 +1433,12 @@ SOC_DAPM_SINGLE("AIF1.1 Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING, #define WM8994_CLASS_W_SWITCH(xname, reg, shift, max, invert) \ SOC_SINGLE_EXT(xname, reg, shift, max, invert, \ - snd_soc_get_volsw, wm8994_put_class_w) + 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); @@ -3487,7 +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; } @@ -3575,8 +3575,9 @@ static void wm8958_mic_id(void *data, u16 status) /* If nothing present then clear our statuses */ dev_dbg(codec->dev, "Detected open circuit\n"); - schedule_delayed_work(&wm8994->open_circuit_work, - msecs_to_jiffies(2500)); + queue_delayed_work(system_power_efficient_wq, + &wm8994->open_circuit_work, + msecs_to_jiffies(2500)); return; } @@ -3690,8 +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"); @@ -3936,8 +3938,9 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) id_delay = wm8994->wm8994->pdata.mic_id_delay; if (wm8994->mic_detecting) - schedule_delayed_work(&wm8994->mic_complete_work, - msecs_to_jiffies(id_delay)); + queue_delayed_work(system_power_efficient_wq, + &wm8994->mic_complete_work, + msecs_to_jiffies(id_delay)); else wm8958_button_det(codec, reg); @@ -4010,9 +4013,6 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) 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; @@ -4385,8 +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]); @@ -4445,6 +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)); } @@ -4452,6 +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/wm8995.c b/sound/soc/codecs/wm8995.c index 90a65c42754..da2899e6c40 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/wm8997.c b/sound/soc/codecs/wm8997.c new file mode 100644 index 00000000000..6ec3de3efa4 --- /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 00000000000..5e91c6a7d56 --- /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/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 05252ac936a..b38f3506418 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -225,15 +225,8 @@ struct wm_coeff_ctl_ops { struct snd_ctl_elem_info *uinfo); }; -struct wm_coeff { - struct device *dev; - struct list_head ctl_list; - struct regmap *regmap; -}; - struct wm_coeff_ctl { const char *name; - struct snd_card *card; struct wm_adsp_alg_region region; struct wm_coeff_ctl_ops ops; struct wm_adsp *adsp; @@ -378,7 +371,6 @@ static int wm_coeff_info(struct snd_kcontrol *kcontrol, static int wm_coeff_write_control(struct snd_kcontrol *kcontrol, const void *buf, size_t len) { - struct wm_coeff *wm_coeff= snd_kcontrol_chip(kcontrol); 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; @@ -401,7 +393,7 @@ static int wm_coeff_write_control(struct snd_kcontrol *kcontrol, if (!scratch) return -ENOMEM; - ret = regmap_raw_write(wm_coeff->regmap, reg, scratch, + ret = regmap_raw_write(adsp->regmap, reg, scratch, ctl->len); if (ret) { adsp_err(adsp, "Failed to write %zu bytes to %x\n", @@ -434,7 +426,6 @@ static int wm_coeff_put(struct snd_kcontrol *kcontrol, static int wm_coeff_read_control(struct snd_kcontrol *kcontrol, void *buf, size_t len) { - struct wm_coeff *wm_coeff= snd_kcontrol_chip(kcontrol); 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; @@ -457,7 +448,7 @@ static int wm_coeff_read_control(struct snd_kcontrol *kcontrol, if (!scratch) return -ENOMEM; - ret = regmap_raw_read(wm_coeff->regmap, reg, scratch, ctl->len); + 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); @@ -481,37 +472,18 @@ static int wm_coeff_get(struct snd_kcontrol *kcontrol, return 0; } -static int wm_coeff_add_kcontrol(struct wm_coeff *wm_coeff, - struct wm_coeff_ctl *ctl, - const struct snd_kcontrol_new *kctl) -{ - int ret; - struct snd_kcontrol *kcontrol; - - kcontrol = snd_ctl_new1(kctl, wm_coeff); - ret = snd_ctl_add(ctl->card, kcontrol); - if (ret < 0) { - dev_err(wm_coeff->dev, "Failed to add %s: %d\n", - kctl->name, ret); - return ret; - } - ctl->kcontrol = kcontrol; - return 0; -} - struct wmfw_ctl_work { - struct wm_coeff *wm_coeff; + struct wm_adsp *adsp; struct wm_coeff_ctl *ctl; struct work_struct work; }; -static int wmfw_add_ctl(struct wm_coeff *wm_coeff, - struct wm_coeff_ctl *ctl) +static int wmfw_add_ctl(struct wm_adsp *adsp, struct wm_coeff_ctl *ctl) { struct snd_kcontrol_new *kcontrol; int ret; - if (!wm_coeff || !ctl || !ctl->name || !ctl->card) + if (!ctl || !ctl->name) return -EINVAL; kcontrol = kzalloc(sizeof(*kcontrol), GFP_KERNEL); @@ -525,14 +497,17 @@ static int wmfw_add_ctl(struct wm_coeff *wm_coeff, kcontrol->put = wm_coeff_put; kcontrol->private_value = (unsigned long)ctl; - ret = wm_coeff_add_kcontrol(wm_coeff, - ctl, kcontrol); + ret = snd_soc_add_card_controls(adsp->card, + kcontrol, 1); if (ret < 0) goto err_kcontrol; kfree(kcontrol); - list_add(&ctl->list, &wm_coeff->ctl_list); + ctl->kcontrol = snd_soc_card_get_kcontrol(adsp->card, + ctl->name); + + list_add(&ctl->list, &adsp->ctl_list); return 0; err_kcontrol: @@ -753,13 +728,12 @@ out: return ret; } -static int wm_coeff_init_control_caches(struct wm_coeff *wm_coeff) +static int wm_coeff_init_control_caches(struct wm_adsp *adsp) { struct wm_coeff_ctl *ctl; int ret; - list_for_each_entry(ctl, &wm_coeff->ctl_list, - list) { + list_for_each_entry(ctl, &adsp->ctl_list, list) { if (!ctl->enabled || ctl->set) continue; ret = wm_coeff_read_control(ctl->kcontrol, @@ -772,13 +746,12 @@ static int wm_coeff_init_control_caches(struct wm_coeff *wm_coeff) return 0; } -static int wm_coeff_sync_controls(struct wm_coeff *wm_coeff) +static int wm_coeff_sync_controls(struct wm_adsp *adsp) { struct wm_coeff_ctl *ctl; int ret; - list_for_each_entry(ctl, &wm_coeff->ctl_list, - list) { + list_for_each_entry(ctl, &adsp->ctl_list, list) { if (!ctl->enabled) continue; if (ctl->set) { @@ -799,15 +772,14 @@ static void wm_adsp_ctl_work(struct work_struct *work) struct wmfw_ctl_work, work); - wmfw_add_ctl(ctl_work->wm_coeff, ctl_work->ctl); + wmfw_add_ctl(ctl_work->adsp, ctl_work->ctl); kfree(ctl_work); } -static int wm_adsp_create_control(struct snd_soc_codec *codec, +static int wm_adsp_create_control(struct wm_adsp *dsp, const struct wm_adsp_alg_region *region) { - struct wm_adsp *dsp = snd_soc_codec_get_drvdata(codec); struct wm_coeff_ctl *ctl; struct wmfw_ctl_work *ctl_work; char *name; @@ -842,7 +814,7 @@ static int wm_adsp_create_control(struct snd_soc_codec *codec, snprintf(name, PAGE_SIZE, "DSP%d %s %x", dsp->num, region_name, region->alg); - list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list, + list_for_each_entry(ctl, &dsp->ctl_list, list) { if (!strcmp(ctl->name, name)) { if (!ctl->enabled) @@ -866,7 +838,6 @@ static int wm_adsp_create_control(struct snd_soc_codec *codec, ctl->set = 0; ctl->ops.xget = wm_coeff_get; ctl->ops.xput = wm_coeff_put; - ctl->card = codec->card->snd_card; ctl->adsp = dsp; ctl->len = region->len; @@ -882,7 +853,7 @@ static int wm_adsp_create_control(struct snd_soc_codec *codec, goto err_ctl_cache; } - ctl_work->wm_coeff = dsp->wm_coeff; + ctl_work->adsp = dsp; ctl_work->ctl = ctl; INIT_WORK(&ctl_work->work, wm_adsp_ctl_work); schedule_work(&ctl_work->work); @@ -903,7 +874,7 @@ err_name: return ret; } -static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) +static int wm_adsp_setup_algs(struct wm_adsp *dsp) { struct regmap *regmap = dsp->regmap; struct wmfw_adsp1_id_hdr adsp1_id; @@ -1091,7 +1062,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) 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(codec, region); + 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)); @@ -1108,7 +1079,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) 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(codec, region); + 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)); @@ -1137,7 +1108,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) 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(codec, region); + 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)); @@ -1154,7 +1125,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) 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(codec, region); + 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)); @@ -1171,7 +1142,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) 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(codec, region); + 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)); @@ -1391,6 +1362,8 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, 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, @@ -1425,7 +1398,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; - ret = wm_adsp_setup_algs(dsp, codec); + ret = wm_adsp_setup_algs(dsp); if (ret != 0) goto err; @@ -1434,12 +1407,12 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, goto err; /* Initialize caches for enabled and unset controls */ - ret = wm_coeff_init_control_caches(dsp->wm_coeff); + ret = wm_coeff_init_control_caches(dsp); if (ret != 0) goto err; /* Sync set controls */ - ret = wm_coeff_sync_controls(dsp->wm_coeff); + ret = wm_coeff_sync_controls(dsp); if (ret != 0) goto err; @@ -1460,10 +1433,8 @@ 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->wm_coeff->ctl_list, - list) { + list_for_each_entry(ctl, &dsp->ctl_list, list) ctl->enabled = 0; - } break; default: @@ -1520,6 +1491,8 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, unsigned int val; int ret; + dsp->card = codec->card; + switch (event) { case SND_SOC_DAPM_POST_PMU: /* @@ -1582,7 +1555,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; - ret = wm_adsp_setup_algs(dsp, codec); + ret = wm_adsp_setup_algs(dsp); if (ret != 0) goto err; @@ -1591,12 +1564,12 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, goto err; /* Initialize caches for enabled and unset controls */ - ret = wm_coeff_init_control_caches(dsp->wm_coeff); + ret = wm_coeff_init_control_caches(dsp); if (ret != 0) goto err; /* Sync set controls */ - ret = wm_coeff_sync_controls(dsp->wm_coeff); + ret = wm_coeff_sync_controls(dsp); if (ret != 0) goto err; @@ -1637,10 +1610,8 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, ret); } - list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list, - list) { + 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, @@ -1679,49 +1650,38 @@ int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs) } INIT_LIST_HEAD(&adsp->alg_regions); - - adsp->wm_coeff = kzalloc(sizeof(*adsp->wm_coeff), - GFP_KERNEL); - if (!adsp->wm_coeff) - return -ENOMEM; - adsp->wm_coeff->regmap = adsp->regmap; - adsp->wm_coeff->dev = adsp->dev; - INIT_LIST_HEAD(&adsp->wm_coeff->ctl_list); + INIT_LIST_HEAD(&adsp->ctl_list); if (dvfs) { adsp->dvfs = devm_regulator_get(adsp->dev, "DCVDD"); if (IS_ERR(adsp->dvfs)) { ret = PTR_ERR(adsp->dvfs); dev_err(adsp->dev, "Failed to get DCVDD: %d\n", ret); - goto out_coeff; + return ret; } ret = regulator_enable(adsp->dvfs); if (ret != 0) { dev_err(adsp->dev, "Failed to enable DCVDD: %d\n", ret); - goto out_coeff; + return ret; } ret = regulator_set_voltage(adsp->dvfs, 1200000, 1800000); if (ret != 0) { dev_err(adsp->dev, "Failed to initialise DVFS: %d\n", ret); - goto out_coeff; + return ret; } ret = regulator_disable(adsp->dvfs); if (ret != 0) { dev_err(adsp->dev, "Failed to disable DCVDD: %d\n", ret); - goto out_coeff; + return ret; } } return 0; - -out_coeff: - kfree(adsp->wm_coeff); - return ret; } EXPORT_SYMBOL_GPL(wm_adsp2_init); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 9f922c82536..d018dea6254 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -39,6 +39,7 @@ struct wm_adsp { int type; struct device *dev; struct regmap *regmap; + struct snd_soc_card *card; int base; int sysclk_reg; @@ -57,7 +58,7 @@ struct wm_adsp { struct regulator *dvfs; - struct wm_coeff *wm_coeff; + struct list_head ctl_list; }; #define WM_ADSP1(wname, num) \ diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 2d9e099415a..8b50e5958de 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -699,9 +699,7 @@ EXPORT_SYMBOL_GPL(wm_hubs_update_class_w); 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); @@ -721,9 +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/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 70eb37a5dd1..25c31f1655f 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -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 aa438546c91..704e246f5b1 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. @@ -109,11 +112,11 @@ config SND_SOC_IMX_SSI tristate config SND_SOC_IMX_PCM_FIQ - bool + tristate select FIQ config SND_SOC_IMX_PCM_DMA - bool + tristate select SND_SOC_GENERIC_DMAENGINE_PCM config SND_SOC_IMX_AUDMUX @@ -175,7 +178,6 @@ config SND_SOC_IMX_WM8962 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 wm8962 codec. @@ -187,14 +189,23 @@ 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 + 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 d4b4aa8b564..8db705b0fdf 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 @@ -43,6 +45,7 @@ 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-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o @@ -51,4 +54,5 @@ obj-$(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/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c new file mode 100644 index 00000000000..3920c3e849c --- /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 00000000000..b1266790d11 --- /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 2f2d837df07..c6b743978d5 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) /* @@ -121,11 +141,14 @@ struct fsl_ssi_private { bool new_binding; bool ssi_on_imx; + bool imx_ac97; + bool use_dma; struct clk *clk; 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; @@ -298,6 +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 * @@ -319,70 +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 = @@ -492,6 +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: @@ -510,12 +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; } @@ -534,22 +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; } @@ -587,6 +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. @@ -663,6 +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" @@ -673,14 +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"); @@ -689,38 +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? */ @@ -739,13 +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 @@ -763,24 +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 */ @@ -794,7 +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 */ @@ -808,9 +1045,30 @@ static int fsl_ssi_probe(struct platform_device *pdev) } if (ssi_private->ssi_on_imx) { - ret = imx_pcm_dma_init(pdev); - if (ret) - 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; + } } /* @@ -845,6 +1103,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) } done: + if (ssi_private->imx_ac97) + fsl_ssi_ac97_init(); + return 0; error_dai: @@ -853,27 +1114,15 @@ error_dai: 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; } @@ -883,20 +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) { + if (ssi_private->ssi_on_imx) imx_pcm_dma_exit(pdev); - clk_disable_unprepare(ssi_private->clk); - clk_put(ssi_private->clk); - } 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; } @@ -919,6 +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 e260f1f899d..ab17381cc98 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -73,8 +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)); @@ -224,14 +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)); @@ -243,6 +251,66 @@ 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; @@ -267,6 +335,8 @@ static int imx_audmux_probe(struct platform_device *pdev) if (audmux_type == IMX31_AUDMUX) audmux_debugfs_init(); + 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 b8ff44b9daf..38a4209af7c 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 9df173c091a..a3d60d4bea4 100644 --- a/sound/soc/fsl/imx-mc13783.c +++ b/sound/soc/fsl/imx-mc13783.c @@ -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 fde4d2ea68c..4dc1296688e 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> @@ -64,7 +65,6 @@ 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); @@ -74,3 +74,5 @@ 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 310d9029032..34043c55f2a 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; @@ -366,9 +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); @@ -377,15 +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) @@ -406,3 +408,5 @@ 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.h b/sound/soc/fsl/imx-pcm.h index 67f656c7c32..5d5b73303e1 100644 --- a/sound/soc/fsl/imx-pcm.h +++ b/sound/soc/fsl/imx-pcm.h @@ -22,17 +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; } -#ifdef CONFIG_SND_SOC_IMX_PCM_DMA +struct imx_pcm_fiq_params { + int irq; + void __iomem *base; + + /* 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 @@ -46,11 +52,13 @@ 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; } diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 3f726e4f88d..46c5b4fdfc5 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> @@ -129,8 +129,10 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) } data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); - if (IS_ERR(data->codec_clk)) + if (IS_ERR(data->codec_clk)) { + ret = PTR_ERR(data->codec_clk); goto fail; + } data->clk_frequency = clk_get_rate(data->codec_clk); diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c new file mode 100644 index 00000000000..816013b0ebb --- /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 51be3772cba..f58bcd85c07 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -571,13 +571,13 @@ 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); @@ -595,7 +595,12 @@ static int imx_ssi_probe(struct platform_device *pdev) goto failed_register; } - ret = imx_pcm_fiq_init(pdev); + 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; + + ret = imx_pcm_fiq_init(pdev, &ssi->fiq_params); if (ret) goto failed_pcm_fiq; diff --git a/sound/soc/fsl/imx-ssi.h b/sound/soc/fsl/imx-ssi.h index d5003cefca8..fb1616ba8c5 100644 --- a/sound/soc/fsl/imx-ssi.h +++ b/sound/soc/fsl/imx-ssi.h @@ -209,6 +209,7 @@ 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; }; diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 52a36a90f4f..722afe69169 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -15,7 +15,7 @@ #include <linux/module.h> #include <linux/of_platform.h> -#include <linux/of_i2c.h> +#include <linux/i2c.h> #include <linux/slab.h> #include <linux/clk.h> #include <sound/soc.h> @@ -217,7 +217,8 @@ static int imx_wm8962_probe(struct platform_device *pdev) 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"); - return -EINVAL; + ret = -EINVAL; + goto fail; } data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 6cf8355a854..8c49147db84 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/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index c62d715235e..78ed4a42ad2 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 3e62ae9e7bb..9e781385cb8 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 a9f14530c3d..b238434f92b 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)); @@ -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 4c9dad3263c..7fce340ab3e 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 +547,31 @@ 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,mvebu-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 +581,4 @@ 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 b979c715471..025be0e9716 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 1d0ed6f8add..27545b0c485 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 4d92637ddb3..f8e1ccc1c58 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/mxs/Kconfig b/sound/soc/mxs/Kconfig index 78d321cbe8b..219235c0221 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-saif.c b/sound/soc/mxs/mxs-saif.c index 54511c5e6a7..b56b8a0e8de 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -31,7 +31,6 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> -#include <asm/mach-types.h> #include "mxs-saif.h" diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 1b134d72f12..4bb273786ff 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; } @@ -91,11 +105,13 @@ static struct snd_soc_dai_link mxs_sgtl5000_dai[] = { .stream_name = "HiFi Playback", .codec_dai_name = "sgtl5000", .ops = &mxs_sgtl5000_hifi_ops, + .playback_only = true, }, { .name = "HiFi Rx", .stream_name = "HiFi Capture", .codec_dai_name = "sgtl5000", .ops = &mxs_sgtl5000_hifi_ops, + .capture_only = true, }, }; @@ -154,8 +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 f4c2417a873..8987bf987e5 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -333,9 +333,6 @@ static int nuc900_ac97_drvprobe(struct platform_device *pdev) spin_lock_init(&nuc900_audio->lock); nuc900_audio->res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!nuc900_audio->res) - return ret; - nuc900_audio->mmio = devm_ioremap_resource(&pdev->dev, nuc900_audio->res); if (IS_ERR(nuc900_audio->mmio)) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 9f5d55e6b17..daa78a0095f 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 diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 361e4c03646..83433fdea32 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; \ \ diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index 70cd5c7b2e1..ebb13906b3a 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 4db1f8e6e17..12e566be379 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -480,15 +480,12 @@ static int asoc_dmic_probe(struct platform_device *pdev) 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-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 7483efb6dc6..6c19bba2357 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; diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index a49dc52f8ab..90d2a7cd256 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -480,9 +480,6 @@ static int asoc_mcpdm_probe(struct platform_device *pdev) 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/pxa/Kconfig b/sound/soc/pxa/Kconfig index b3580946754..4db74a083db 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 diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c index 4ad76099dd4..5b7d969f89a 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 97b711e1282..bbea7780eac 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 5d57e071cdf..8235e231d89 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; } diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index 62142ce367c..41752a5fe3b 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); @@ -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 6f4dd7543e8..a3119a00d8f 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 1475515712e..f1059d999de 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" @@ -48,44 +49,44 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_cold_reset, }; -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 @@ -119,7 +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; @@ -135,7 +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; diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index f7ca7166411..d5340a08885 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 ecff116cb7b..806da27b8b6 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/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c index f4ea4f6663a..13c9ee0cb83 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/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c index 58cfb1eb7dd..945e8abdc10 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/ac97.c b/sound/soc/samsung/ac97.c index 2dd623fa388..2acf987844e 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -404,18 +404,13 @@ 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; } + 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); @@ -462,7 +457,7 @@ static int s3c_ac97_probe(struct platform_device *pdev) 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; @@ -485,7 +480,7 @@ static int s3c_ac97_remove(struct platform_device *pdev) { 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); diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index 21b79262010..9338d11e921 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 189a7a6d502..0e86315a3ea 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 c0e6d9a19ef..821a5023100 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 959c702235c..b302f3b7a58 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); @@ -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) @@ -1018,18 +1089,18 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) 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 @@ -1060,13 +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"); @@ -1075,7 +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; } @@ -1115,15 +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) { @@ -1200,7 +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: @@ -1230,33 +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/pcm.c b/sound/soc/samsung/pcm.c index 1566afe9ef5..e54256fc4b2 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/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 47e23864ea7..ea885cb9f76 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 8b3414551a6..9c8ebd872fa 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_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index 581ea4a06fc..5fd7a05a9b9 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/spdif.c b/sound/soc/samsung/spdif.c index 2e5ebb2f198..28487dcc453 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 6bcb1164d59..56d8ff6a402 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 849b387d17d..aaf3dcd1ee2 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 30390260bb6..b33ca7cd085 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 { @@ -1289,6 +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; @@ -1305,11 +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) @@ -1321,7 +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; @@ -1347,7 +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)) @@ -1357,26 +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 diff --git a/sound/soc/sh/rcar/Makefile b/sound/soc/sh/rcar/Makefile new file mode 100644 index 00000000000..0ff492df792 --- /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 00000000000..d80deb7ccf1 --- /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 00000000000..a3570602851 --- /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 00000000000..babb203b43b --- /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 00000000000..9cc6986a8cf --- /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 00000000000..184d9008cec --- /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_USB_HPBIF flags + */ + if (!(flags & RSND_SCU_USB_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 00000000000..fae26d3f79d --- /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 06a8000aa07..53c9ecdd119 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 0ec070cf723..4d0561312f3 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); @@ -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) @@ -1223,9 +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; @@ -1428,6 +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; @@ -1718,8 +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; @@ -1798,12 +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", @@ -2080,6 +2100,117 @@ 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); @@ -2098,6 +2229,35 @@ int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops) 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 * @@ -2299,6 +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 @@ -2541,59 +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 @@ -3908,10 +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; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b94190820e8..c17c14c394d 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, @@ -73,16 +82,18 @@ 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, @@ -174,36 +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; @@ -211,6 +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; @@ -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 */ @@ -507,17 +662,12 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm, return 0; } -static void dapm_kcontrol_free(struct snd_kcontrol *kctl) -{ - kfree(kctl->private_data); -} - /* * Determine if a kcontrol is shared. If it is, look it up. If it isn't, * 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; @@ -525,9 +675,6 @@ 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; char *long_name; const char *name; @@ -546,25 +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; @@ -587,7 +715,6 @@ 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; } } @@ -602,10 +729,8 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, long_name = kasprintf(GFP_KERNEL, "%s %s", w->name + prefix_len, w->kcontrol_news[kci].name); - if (long_name == NULL) { - kfree(wlist); + if (long_name == NULL) return -ENOMEM; - } name = long_name; } else if (wname_in_long_name) { @@ -616,23 +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); - kcontrol->private_free = dapm_kcontrol_free; 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); return ret; } } - kcontrol->private_data = wlist; + ret = dapm_kcontrol_add_widget(kcontrol, w); + if (ret) + return ret; + w->kcontrols[kci] = kcontrol; - path->kcontrol = kcontrol; return 0; } @@ -652,13 +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); } } @@ -679,19 +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; } @@ -812,6 +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; @@ -907,6 +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; @@ -1061,7 +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, @@ -1071,7 +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, @@ -1243,10 +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; @@ -1280,55 +1416,50 @@ static void dapm_seq_check_event(struct snd_soc_dapm_context *dapm, 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; - - mask |= cur_mask; - if (power) - value |= cur_mask; + value |= w->off_val << w->shift; - 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) { @@ -1338,7 +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); @@ -1346,8 +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); } } @@ -1359,8 +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); @@ -1383,7 +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++) @@ -1443,7 +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++) @@ -1453,37 +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); + } } } @@ -1595,6 +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: @@ -1611,8 +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, @@ -1646,9 +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); @@ -1688,7 +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 @@ -1730,29 +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(dapm, w, SND_SOC_DAPM_WILL_PMD); + dapm_seq_check_event(card, w, SND_SOC_DAPM_WILL_PMD); } list_for_each_entry(w, &up_list, power_list) { - dapm_seq_check_event(dapm, w, SND_SOC_DAPM_WILL_PMU); + 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); @@ -1763,7 +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); @@ -1798,8 +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"); @@ -1936,22 +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; @@ -1966,73 +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); @@ -2111,6 +2238,7 @@ 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); } @@ -2205,70 +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) @@ -2276,8 +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); @@ -2297,6 +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); @@ -2327,6 +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); @@ -2358,15 +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; } @@ -2568,14 +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; @@ -2585,7 +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; } } @@ -2611,12 +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; } @@ -2626,8 +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); @@ -2644,8 +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; @@ -2653,17 +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; } @@ -2681,9 +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; @@ -2695,10 +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); @@ -2707,33 +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); + + mask = mask << shift; + val = val << shift; - widget->value = val; + change = snd_soc_test_bits(codec, reg, mask, val); + if (change) { + update.kcontrol = kcontrol; + update.reg = reg; + update.mask = mask; + update.val = val; - update.kcontrol = kcontrol; - update.widget = widget; - update.reg = reg; - update.mask = mask; - update.val = val; - widget->dapm->update = &update; + card->update = &update; - soc_dapm_mixer_update_power(widget, kcontrol, connect); + soc_dapm_mixer_update_power(card, kcontrol, connect); - widget->dapm->update = NULL; - } + card->update = NULL; } mutex_unlock(&card->dapm_mutex); - return 0; + return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw); @@ -2749,12 +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] = @@ -2776,15 +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; @@ -2800,24 +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]; - - widget->value = val; + update.kcontrol = kcontrol; + update.reg = e->reg; + update.mask = mask; + update.val = val; + card->update = &update; - 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(card, kcontrol, mux, e); - soc_dapm_mux_update_power(widget, kcontrol, mux, e); - - widget->dapm->update = NULL; - } + card->update = NULL; } mutex_unlock(&card->dapm_mutex); @@ -2835,11 +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); @@ -2854,34 +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); @@ -2901,12 +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]) @@ -2942,15 +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; @@ -2966,24 +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]; - - widget->value = val; + update.kcontrol = kcontrol; + update.reg = e->reg; + update.mask = mask; + update.val = val; + card->update = &update; - 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(card, kcontrol, mux, e); - soc_dapm_mux_update_power(widget, kcontrol, mux, e); - - widget->dapm->update = NULL; - } + card->update = NULL; } mutex_unlock(&card->dapm_mutex); @@ -3080,7 +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, @@ -3127,16 +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: @@ -3152,6 +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: @@ -3416,9 +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) { @@ -3445,29 +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); } } } @@ -3529,7 +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); } /** @@ -3798,7 +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 8ca9ecc5ac5..122c0c18b9d 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 0bb5cccd776..71358e3b54d 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 b6c640332a1..330c9a6b5cb 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -411,8 +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 */ @@ -1832,18 +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; @@ -2027,6 +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/spear/Kconfig b/sound/soc/spear/Kconfig index 3567d73b218..0a53053495f 100644 --- a/sound/soc/spear/Kconfig +++ b/sound/soc/spear/Kconfig @@ -1,6 +1,6 @@ config SND_SPEAR_SOC tristate - select SND_SOC_DMAENGINE_PCM + select SND_DMAENGINE_PCM config SND_SPEAR_SPDIF_OUT tristate diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 995b120c2cd..8fc653ca3ab 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. @@ -61,7 +61,7 @@ config SND_SOC_TEGRA30_I2S config SND_SOC_TEGRA_RT5640 tristate "SoC Audio support for Tegra boards using an RT5640 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_RT5640 @@ -71,7 +71,7 @@ config SND_SOC_TEGRA_RT5640 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 @@ -81,7 +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 @@ -92,7 +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 @@ -110,7 +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/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index e58233f7df6..ae27bcd586d 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -334,12 +334,6 @@ 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; - } - regs = devm_ioremap_resource(&pdev->dev, mem); if (IS_ERR(regs)) { ret = PTR_ERR(regs); @@ -389,9 +383,9 @@ 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]; + 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) @@ -432,8 +426,6 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) return 0; -err_unregister_pcm: - tegra_pcm_platform_unregister(&pdev->dev); err_unregister_component: snd_soc_unregister_component(&pdev->dev); err_asoc_utils_fini: diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 5eaa12cdc6e..551b3c93ce9 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_i2s.c b/sound/soc/tegra/tegra30_i2s.c index d04146cad61..47565fd0450 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); diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 48d05d9e100..c61ea3a1030 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_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index 08794f915a9..4511c5a875e 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -99,6 +99,7 @@ static struct snd_soc_jack_gpio tegra_rt5640_hp_jack_gpio = { 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[] = { diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c index f87fc53e9b8..8e774d1a243 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 05c68aab5cf..734bfcd2114 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 4bcce8a3cde..e0305a14856 100644 --- a/sound/soc/txx9/txx9aclc-ac97.c +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -184,9 +184,6 @@ 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; - drvdata->base = devm_ioremap_resource(&pdev->dev, r); if (IS_ERR(drvdata->base)) return PTR_ERR(drvdata->base); diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index 8f5cd00a6e4..178d1bad625 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -52,6 +52,7 @@ static 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/usb/6fire/comm.c b/sound/usb/6fire/comm.c index 9e6e3ffd86b..23452ee617e 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 6a0840b0dcf..780d5ed8e5d 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 b9defcdeb7e..780bf3f62d2 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 26722423330..f3dd7266c39 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 c321006e543..84851b9f555 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 2aa4e13063a..b5eb97fdc84 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -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; @@ -582,6 +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; @@ -593,6 +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); @@ -614,6 +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; @@ -625,6 +660,7 @@ int usb6fire_pcm_init(struct sfire_chip *chip) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &pcm_ops); if (ret) { + usb6fire_pcm_buffers_destroy(rt); kfree(rt); snd_printk(KERN_ERR PREFIX "error preallocating pcm buffers.\n"); @@ -669,6 +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 9b01133ee3f..f5779d6182c 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/endpoint.c b/sound/usb/endpoint.c index 7a444b5501d..93e970f2b3c 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/hiface/pcm.c b/sound/usb/hiface/pcm.c index 6430ed2a9f6..c21a3df9a0d 100644 --- a/sound/usb/hiface/pcm.c +++ b/sound/usb/hiface/pcm.c @@ -503,7 +503,7 @@ static snd_pcm_uframes_t hiface_pcm_pointer(struct snd_pcm_substream *alsa_sub) snd_pcm_uframes_t dma_offset; if (rt->panic || !sub) - return SNDRV_PCM_STATE_XRUN; + return SNDRV_PCM_POS_XRUN; spin_lock_irqsave(&sub->lock, flags); dma_offset = sub->dma_off; diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index d5438083fd6..95558ef4a7a 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/pcm.c b/sound/usb/pcm.c index 15b151ed489..b375d58871e 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -327,6 +327,137 @@ static int search_roland_implicit_fb(struct usb_device *dev, int ifnum, 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 */ @@ -336,9 +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)) @@ -383,118 +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 && - 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) { - implicit_fb = 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.c b/sound/usb/quirks.c index 1bc45e71f1f..0df9ede99df 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -319,19 +319,19 @@ static int create_auto_midi_quirk(struct snd_usb_audio *chip, if (altsd->bNumEndpoints < 1) return -ENODEV; epd = get_endpoint(alts, 0); - if (!usb_endpoint_xfer_bulk(epd) || + 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 < 0 && err != -ENODEV) + if (err != -ENODEV) return err; break; case 0x0582: /* Roland */ err = create_roland_midi_quirk(chip, iface, driver, alts); - if (err < 0 && err != -ENODEV) + if (err != -ENODEV) return err; break; } diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index 1f9bbd55553..5a51b18c50f 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -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); } |