diff options
Diffstat (limited to 'sound')
382 files changed, 21403 insertions, 8442 deletions
diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index 1eddf8fa188f..8797d42e2b76 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -776,7 +776,7 @@ static int check_codec(struct aoa_codec *codec, struct codec_connection *cc; /* if the codec has a 'codec' node, we require a reference */ - if (codec->node && (strcmp(codec->node->name, "codec") == 0)) { + if (of_node_name_eq(codec->node, "codec")) { snprintf(propname, sizeof(propname), "platform-%s-codec-ref", codec->name); ref = of_get_property(ldev->sound, propname, NULL); @@ -1008,8 +1008,8 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev) return -ENODEV; /* by breaking out we keep a reference */ - while ((sound = of_get_next_child(sdev->ofdev.dev.of_node, sound))) { - if (sound->type && strcasecmp(sound->type, "soundchip") == 0) + for_each_child_of_node(sdev->ofdev.dev.of_node, sound) { + if (of_node_is_type(sound, "soundchip")) break; } if (!sound) diff --git a/sound/aoa/soundbus/core.c b/sound/aoa/soundbus/core.c index 70bcaa7f93dd..065d3a55725e 100644 --- a/sound/aoa/soundbus/core.c +++ b/sound/aoa/soundbus/core.c @@ -74,11 +74,11 @@ static int soundbus_uevent(struct device *dev, struct kobj_uevent_env *env) of = &soundbus_dev->ofdev; /* stuff we want to pass to /sbin/hotplug */ - retval = add_uevent_var(env, "OF_NAME=%s", of->dev.of_node->name); + retval = add_uevent_var(env, "OF_NAME=%pOFn", of->dev.of_node); if (retval) return retval; - retval = add_uevent_var(env, "OF_TYPE=%s", of->dev.of_node->type); + retval = add_uevent_var(env, "OF_TYPE=%s", of_node_get_device_type(of->dev.of_node)); if (retval) return retval; diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index 000b58522106..40ebde2e1ab1 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -47,8 +47,8 @@ static int alloc_dbdma_descriptor_ring(struct i2sbus_dev *i2sdev, /* We use the PCI APIs for now until the generic one gets fixed * enough or until we get some macio-specific versions */ - r->space = dma_zalloc_coherent(&macio_get_pci_dev(i2sdev->macio)->dev, - r->size, &r->bus_addr, GFP_KERNEL); + r->space = dma_alloc_coherent(&macio_get_pci_dev(i2sdev->macio)->dev, + r->size, &r->bus_addr, GFP_KERNEL); if (!r->space) return -ENOMEM; @@ -154,21 +154,22 @@ static int i2sbus_add_dev(struct macio_dev *macio, struct device_node *np) { struct i2sbus_dev *dev; - struct device_node *child = NULL, *sound = NULL; + struct device_node *child, *sound = NULL; struct resource *r; int i, layout = 0, rlen, ok = force; - static const char *rnames[] = { "i2sbus: %s (control)", - "i2sbus: %s (tx)", - "i2sbus: %s (rx)" }; + char node_name[6]; + static const char *rnames[] = { "i2sbus: %pOFn (control)", + "i2sbus: %pOFn (tx)", + "i2sbus: %pOFn (rx)" }; static irq_handler_t ints[] = { i2sbus_bus_intr, i2sbus_tx_intr, i2sbus_rx_intr }; - if (strlen(np->name) != 5) + if (snprintf(node_name, sizeof(node_name), "%pOFn", np) != 5) return 0; - if (strncmp(np->name, "i2s-", 4)) + if (strncmp(node_name, "i2s-", 4)) return 0; dev = kzalloc(sizeof(struct i2sbus_dev), GFP_KERNEL); @@ -176,8 +177,8 @@ static int i2sbus_add_dev(struct macio_dev *macio, return 0; i = 0; - while ((child = of_get_next_child(np, child))) { - if (strcmp(child->name, "sound") == 0) { + for_each_child_of_node(np, child) { + if (of_node_name_eq(child, "sound")) { i++; sound = child; } @@ -228,13 +229,13 @@ static int i2sbus_add_dev(struct macio_dev *macio, dev->sound.pcmid = -1; dev->macio = macio; dev->control = control; - dev->bus_number = np->name[4] - 'a'; + dev->bus_number = node_name[4] - 'a'; INIT_LIST_HEAD(&dev->sound.codec_list); for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) { dev->interrupts[i] = -1; snprintf(dev->rnames[i], sizeof(dev->rnames[i]), - rnames[i], np->name); + rnames[i], np); } for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) { int irq = irq_of_parse_and_map(np, i); diff --git a/sound/aoa/soundbus/sysfs.c b/sound/aoa/soundbus/sysfs.c index 81da020bddef..a2d55e15afbb 100644 --- a/sound/aoa/soundbus/sysfs.c +++ b/sound/aoa/soundbus/sysfs.c @@ -1,18 +1,10 @@ // SPDX-License-Identifier: GPL-2.0 #include <linux/kernel.h> +#include <linux/of.h> #include <linux/stat.h> /* FIX UP */ #include "soundbus.h" -#define soundbus_config_of_attr(field, format_string) \ -static ssize_t \ -field##_show (struct device *dev, struct device_attribute *attr, \ - char *buf) \ -{ \ - struct soundbus_dev *mdev = to_soundbus_device (dev); \ - return sprintf (buf, format_string, mdev->ofdev.dev.of_node->field); \ -} - static ssize_t modalias_show(struct device *dev, struct device_attribute *attr, char *buf) { @@ -25,17 +17,33 @@ static ssize_t modalias_show(struct device *dev, struct device_attribute *attr, strcat(buf, "\n"); length = strlen(buf); } else { - length = sprintf(buf, "of:N%sT%s\n", - of->dev.of_node->name, of->dev.of_node->type); + length = sprintf(buf, "of:N%pOFn%c%s\n", + of->dev.of_node, 'T', + of_node_get_device_type(of->dev.of_node)); } return length; } static DEVICE_ATTR_RO(modalias); -soundbus_config_of_attr (name, "%s\n"); +static ssize_t name_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct soundbus_dev *sdev = to_soundbus_device(dev); + struct platform_device *of = &sdev->ofdev; + + return sprintf(buf, "%pOFn\n", of->dev.of_node); +} static DEVICE_ATTR_RO(name); -soundbus_config_of_attr (type, "%s\n"); + +static ssize_t type_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct soundbus_dev *sdev = to_soundbus_device(dev); + struct platform_device *of = &sdev->ofdev; + + return sprintf(buf, "%s\n", of_node_get_device_type(of->dev.of_node)); +} static DEVICE_ATTR_RO(type); struct attribute *soundbus_dev_attrs[] = { diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index 5fbd47a9177e..28867732a318 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -31,7 +31,6 @@ endif # SND_ARM config SND_PXA2XX_LIB tristate - select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97 select SND_DMAENGINE_PCM config SND_PXA2XX_LIB_AC97 diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 26b5e245b074..f7d2b373da0a 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -171,7 +171,8 @@ static int snd_compr_free(struct inode *inode, struct file *f) } data->stream.ops->free(&data->stream); - kfree(data->stream.runtime->buffer); + if (!data->stream.runtime->dma_buffer_p) + kfree(data->stream.runtime->buffer); kfree(data->stream.runtime); kfree(data); return 0; @@ -505,7 +506,7 @@ static int snd_compr_allocate_buffer(struct snd_compr_stream *stream, struct snd_compr_params *params) { unsigned int buffer_size; - void *buffer; + void *buffer = NULL; buffer_size = params->buffer.fragment_size * params->buffer.fragments; if (stream->ops->copy) { @@ -514,7 +515,18 @@ static int snd_compr_allocate_buffer(struct snd_compr_stream *stream, * the data from core */ } else { - buffer = kmalloc(buffer_size, GFP_KERNEL); + if (stream->runtime->dma_buffer_p) { + + if (buffer_size > stream->runtime->dma_buffer_p->bytes) + dev_err(&stream->device->dev, + "Not enough DMA buffer"); + else + buffer = stream->runtime->dma_buffer_p->area; + + } else { + buffer = kmalloc(buffer_size, GFP_KERNEL); + } + if (!buffer) return -ENOMEM; } @@ -529,7 +541,8 @@ static int snd_compress_check_input(struct snd_compr_params *params) { /* first let's check the buffer parameter's */ if (params->buffer.fragment_size == 0 || - params->buffer.fragments > INT_MAX / params->buffer.fragment_size) + params->buffer.fragments > INT_MAX / params->buffer.fragment_size || + params->buffer.fragments == 0) return -EINVAL; /* now codec parameters */ diff --git a/sound/core/control.c b/sound/core/control.c index 9aa15bfc7936..fad7db402443 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -348,67 +348,100 @@ static int snd_ctl_find_hole(struct snd_card *card, unsigned int count) return 0; } -/** - * snd_ctl_add - add the control instance to the card - * @card: the card instance - * @kcontrol: the control instance to add - * - * Adds the control instance created via snd_ctl_new() or - * snd_ctl_new1() to the given card. Assigns also an unique - * numid used for fast search. - * - * It frees automatically the control which cannot be added. - * - * Return: Zero if successful, or a negative error code on failure. - * - */ -int snd_ctl_add(struct snd_card *card, struct snd_kcontrol *kcontrol) +enum snd_ctl_add_mode { + CTL_ADD_EXCLUSIVE, CTL_REPLACE, CTL_ADD_ON_REPLACE, +}; + +/* add/replace a new kcontrol object; call with card->controls_rwsem locked */ +static int __snd_ctl_add_replace(struct snd_card *card, + struct snd_kcontrol *kcontrol, + enum snd_ctl_add_mode mode) { struct snd_ctl_elem_id id; unsigned int idx; unsigned int count; - int err = -EINVAL; + struct snd_kcontrol *old; + int err; - if (! kcontrol) - return err; - if (snd_BUG_ON(!card || !kcontrol->info)) - goto error; id = kcontrol->id; if (id.index > UINT_MAX - kcontrol->count) - goto error; + return -EINVAL; - down_write(&card->controls_rwsem); - if (snd_ctl_find_id(card, &id)) { - up_write(&card->controls_rwsem); - dev_err(card->dev, "control %i:%i:%i:%s:%i is already present\n", - id.iface, - id.device, - id.subdevice, - id.name, - id.index); - err = -EBUSY; - goto error; - } - if (snd_ctl_find_hole(card, kcontrol->count) < 0) { - up_write(&card->controls_rwsem); - err = -ENOMEM; - goto error; + old = snd_ctl_find_id(card, &id); + if (!old) { + if (mode == CTL_REPLACE) + return -EINVAL; + } else { + if (mode == CTL_ADD_EXCLUSIVE) { + dev_err(card->dev, + "control %i:%i:%i:%s:%i is already present\n", + id.iface, id.device, id.subdevice, id.name, + id.index); + return -EBUSY; + } + + err = snd_ctl_remove(card, old); + if (err < 0) + return err; } + + if (snd_ctl_find_hole(card, kcontrol->count) < 0) + return -ENOMEM; + list_add_tail(&kcontrol->list, &card->controls); card->controls_count += kcontrol->count; kcontrol->id.numid = card->last_numid + 1; card->last_numid += kcontrol->count; + id = kcontrol->id; count = kcontrol->count; - up_write(&card->controls_rwsem); for (idx = 0; idx < count; idx++, id.index++, id.numid++) snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_ADD, &id); + + return 0; +} + +static int snd_ctl_add_replace(struct snd_card *card, + struct snd_kcontrol *kcontrol, + enum snd_ctl_add_mode mode) +{ + int err = -EINVAL; + + if (! kcontrol) + return err; + if (snd_BUG_ON(!card || !kcontrol->info)) + goto error; + + down_write(&card->controls_rwsem); + err = __snd_ctl_add_replace(card, kcontrol, mode); + up_write(&card->controls_rwsem); + if (err < 0) + goto error; return 0; error: snd_ctl_free_one(kcontrol); return err; } + +/** + * snd_ctl_add - add the control instance to the card + * @card: the card instance + * @kcontrol: the control instance to add + * + * Adds the control instance created via snd_ctl_new() or + * snd_ctl_new1() to the given card. Assigns also an unique + * numid used for fast search. + * + * It frees automatically the control which cannot be added. + * + * Return: Zero if successful, or a negative error code on failure. + * + */ +int snd_ctl_add(struct snd_card *card, struct snd_kcontrol *kcontrol) +{ + return snd_ctl_add_replace(card, kcontrol, CTL_ADD_EXCLUSIVE); +} EXPORT_SYMBOL(snd_ctl_add); /** @@ -428,53 +461,8 @@ EXPORT_SYMBOL(snd_ctl_add); int snd_ctl_replace(struct snd_card *card, struct snd_kcontrol *kcontrol, bool add_on_replace) { - struct snd_ctl_elem_id id; - unsigned int count; - unsigned int idx; - struct snd_kcontrol *old; - int ret; - - if (!kcontrol) - return -EINVAL; - if (snd_BUG_ON(!card || !kcontrol->info)) { - ret = -EINVAL; - goto error; - } - id = kcontrol->id; - down_write(&card->controls_rwsem); - old = snd_ctl_find_id(card, &id); - if (!old) { - if (add_on_replace) - goto add; - up_write(&card->controls_rwsem); - ret = -EINVAL; - goto error; - } - ret = snd_ctl_remove(card, old); - if (ret < 0) { - up_write(&card->controls_rwsem); - goto error; - } -add: - if (snd_ctl_find_hole(card, kcontrol->count) < 0) { - up_write(&card->controls_rwsem); - ret = -ENOMEM; - goto error; - } - list_add_tail(&kcontrol->list, &card->controls); - card->controls_count += kcontrol->count; - kcontrol->id.numid = card->last_numid + 1; - card->last_numid += kcontrol->count; - id = kcontrol->id; - count = kcontrol->count; - up_write(&card->controls_rwsem); - for (idx = 0; idx < count; idx++, id.index++, id.numid++) - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_ADD, &id); - return 0; - -error: - snd_ctl_free_one(kcontrol); - return ret; + return snd_ctl_add_replace(card, kcontrol, + add_on_replace ? CTL_ADD_ON_REPLACE : CTL_REPLACE); } EXPORT_SYMBOL(snd_ctl_replace); @@ -1361,9 +1349,12 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, kctl->tlv.c = snd_ctl_elem_user_tlv; /* This function manage to free the instance on failure. */ - err = snd_ctl_add(card, kctl); - if (err < 0) - return err; + down_write(&card->controls_rwsem); + err = __snd_ctl_add_replace(card, kctl, CTL_ADD_EXCLUSIVE); + if (err < 0) { + snd_ctl_free_one(kctl); + goto unlock; + } offset = snd_ctl_get_ioff(kctl, &info->id); snd_ctl_build_ioff(&info->id, kctl, offset); /* @@ -1374,10 +1365,10 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, * which locks the element. */ - down_write(&card->controls_rwsem); card->user_ctl_count++; - up_write(&card->controls_rwsem); + unlock: + up_write(&card->controls_rwsem); return 0; } diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 753d5fc4b284..59a4adc286ed 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -25,6 +25,9 @@ #include <linux/mm.h> #include <linux/dma-mapping.h> #include <linux/genalloc.h> +#ifdef CONFIG_X86 +#include <asm/set_memory.h> +#endif #include <sound/memalloc.h> /* @@ -82,31 +85,32 @@ EXPORT_SYMBOL(snd_free_pages); #ifdef CONFIG_HAS_DMA /* allocate the coherent DMA pages */ -static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *dma) +static void snd_malloc_dev_pages(struct snd_dma_buffer *dmab, size_t size) { - int pg; gfp_t gfp_flags; - if (WARN_ON(!dma)) - return NULL; - pg = get_order(size); gfp_flags = GFP_KERNEL | __GFP_COMP /* compound page lets parts be mapped */ | __GFP_NORETRY /* don't trigger OOM-killer */ | __GFP_NOWARN; /* no stack trace print - this call is non-critical */ - return dma_alloc_coherent(dev, PAGE_SIZE << pg, dma, gfp_flags); + dmab->area = dma_alloc_coherent(dmab->dev.dev, size, &dmab->addr, + gfp_flags); +#ifdef CONFIG_X86 + if (dmab->area && dmab->dev.type == SNDRV_DMA_TYPE_DEV_UC) + set_memory_wc((unsigned long)dmab->area, + PAGE_ALIGN(size) >> PAGE_SHIFT); +#endif } /* free the coherent DMA pages */ -static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, - dma_addr_t dma) +static void snd_free_dev_pages(struct snd_dma_buffer *dmab) { - int pg; - - if (ptr == NULL) - return; - pg = get_order(size); - dma_free_coherent(dev, PAGE_SIZE << pg, ptr, dma); +#ifdef CONFIG_X86 + if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_UC) + set_memory_wb((unsigned long)dmab->area, + PAGE_ALIGN(dmab->bytes) >> PAGE_SHIFT); +#endif + dma_free_coherent(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); } #ifdef CONFIG_GENERIC_ALLOCATOR @@ -199,12 +203,15 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, */ dmab->dev.type = SNDRV_DMA_TYPE_DEV; #endif /* CONFIG_GENERIC_ALLOCATOR */ + /* fall through */ case SNDRV_DMA_TYPE_DEV: - dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr); + case SNDRV_DMA_TYPE_DEV_UC: + snd_malloc_dev_pages(dmab, size); break; #endif #ifdef CONFIG_SND_DMA_SGBUF case SNDRV_DMA_TYPE_DEV_SG: + case SNDRV_DMA_TYPE_DEV_UC_SG: snd_malloc_sgbuf_pages(device, size, dmab, NULL); break; #endif @@ -275,11 +282,13 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab) break; #endif /* CONFIG_GENERIC_ALLOCATOR */ case SNDRV_DMA_TYPE_DEV: - snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); + case SNDRV_DMA_TYPE_DEV_UC: + snd_free_dev_pages(dmab); break; #endif #ifdef CONFIG_SND_DMA_SGBUF case SNDRV_DMA_TYPE_DEV_SG: + case SNDRV_DMA_TYPE_DEV_UC_SG: snd_free_sgbuf_pages(dmab); break; #endif diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index f8d4a419f3af..467039b342b5 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1062,8 +1062,8 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream) runtime->oss.channels = params_channels(params); runtime->oss.rate = params_rate(params); - vfree(runtime->oss.buffer); - runtime->oss.buffer = vmalloc(runtime->oss.period_bytes); + kvfree(runtime->oss.buffer); + runtime->oss.buffer = kvzalloc(runtime->oss.period_bytes, GFP_KERNEL); if (!runtime->oss.buffer) { err = -ENOMEM; goto failure; @@ -2328,7 +2328,7 @@ static void snd_pcm_oss_release_substream(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime; runtime = substream->runtime; - vfree(runtime->oss.buffer); + kvfree(runtime->oss.buffer); runtime->oss.buffer = NULL; #ifdef CONFIG_SND_PCM_OSS_PLUGINS snd_pcm_oss_plugin_clear(substream); diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 0391cb1a4f19..31cb2acf8afc 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -66,8 +66,8 @@ static int snd_pcm_plugin_alloc(struct snd_pcm_plugin *plugin, snd_pcm_uframes_t return -ENXIO; size /= 8; if (plugin->buf_frames < frames) { - vfree(plugin->buf); - plugin->buf = vmalloc(size); + kvfree(plugin->buf); + plugin->buf = kvzalloc(size, GFP_KERNEL); plugin->buf_frames = frames; } if (!plugin->buf) { @@ -111,7 +111,7 @@ int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames) while (plugin->next) { if (plugin->dst_frames) frames = plugin->dst_frames(plugin, frames); - if (snd_BUG_ON(frames <= 0)) + if (snd_BUG_ON((snd_pcm_sframes_t)frames <= 0)) return -ENXIO; plugin = plugin->next; err = snd_pcm_plugin_alloc(plugin, frames); @@ -123,7 +123,7 @@ int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames) while (plugin->prev) { if (plugin->src_frames) frames = plugin->src_frames(plugin, frames); - if (snd_BUG_ON(frames <= 0)) + if (snd_BUG_ON((snd_pcm_sframes_t)frames <= 0)) return -ENXIO; plugin = plugin->prev; err = snd_pcm_plugin_alloc(plugin, frames); @@ -191,7 +191,7 @@ int snd_pcm_plugin_free(struct snd_pcm_plugin *plugin) if (plugin->private_free) plugin->private_free(plugin); kfree(plugin->buf_channels); - vfree(plugin->buf); + kvfree(plugin->buf); kfree(plugin); return 0; } diff --git a/sound/core/pcm.c b/sound/core/pcm.c index fdb9b92fc8d6..01b9d62eef14 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -25,6 +25,7 @@ #include <linux/time.h> #include <linux/mutex.h> #include <linux/device.h> +#include <linux/nospec.h> #include <sound/core.h> #include <sound/minors.h> #include <sound/pcm.h> @@ -129,6 +130,7 @@ static int snd_pcm_control_ioctl(struct snd_card *card, return -EFAULT; if (stream < 0 || stream > 1) return -EINVAL; + stream = array_index_nospec(stream, 2); if (get_user(subdevice, &info->subdevice)) return -EFAULT; mutex_lock(®ister_mutex); diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 4e6110d778bd..6c0b30391ba9 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -2172,6 +2172,10 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, if (err < 0) goto _end_unlock; + runtime->twake = runtime->control->avail_min ? : 1; + if (runtime->status->state == SNDRV_PCM_STATE_RUNNING) + snd_pcm_update_hw_ptr(substream); + if (!is_playback && runtime->status->state == SNDRV_PCM_STATE_PREPARED && size >= runtime->start_threshold) { @@ -2180,10 +2184,8 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, goto _end_unlock; } - runtime->twake = runtime->control->avail_min ? : 1; - if (runtime->status->state == SNDRV_PCM_STATE_RUNNING) - snd_pcm_update_hw_ptr(substream); avail = snd_pcm_avail(substream); + while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t cont; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 66c90f486af9..818dff1de545 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -36,6 +36,7 @@ #include <sound/timer.h> #include <sound/minors.h> #include <linux/uio.h> +#include <linux/delay.h> #include "pcm_local.h" @@ -91,12 +92,12 @@ static DECLARE_RWSEM(snd_pcm_link_rwsem); * and this may lead to a deadlock when the code path takes read sem * twice (e.g. one in snd_pcm_action_nonatomic() and another in * snd_pcm_stream_lock()). As a (suboptimal) workaround, let writer to - * spin until it gets the lock. + * sleep until all the readers are completed without blocking by writer. */ -static inline void down_write_nonblock(struct rw_semaphore *lock) +static inline void down_write_nonfifo(struct rw_semaphore *lock) { while (!down_write_trylock(lock)) - cond_resched(); + msleep(1); } #define PCM_LOCK_DEFAULT 0 @@ -1967,7 +1968,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) res = -ENOMEM; goto _nolock; } - down_write_nonblock(&snd_pcm_link_rwsem); + down_write_nonfifo(&snd_pcm_link_rwsem); write_lock_irq(&snd_pcm_link_rwlock); if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN || substream->runtime->status->state != substream1->runtime->status->state || @@ -2014,7 +2015,7 @@ static int snd_pcm_unlink(struct snd_pcm_substream *substream) struct snd_pcm_substream *s; int res = 0; - down_write_nonblock(&snd_pcm_link_rwsem); + down_write_nonfifo(&snd_pcm_link_rwsem); write_lock_irq(&snd_pcm_link_rwlock); if (!snd_pcm_stream_linked(substream)) { res = -EALREADY; @@ -2369,7 +2370,8 @@ int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) static void pcm_release_private(struct snd_pcm_substream *substream) { - snd_pcm_unlink(substream); + if (snd_pcm_stream_linked(substream)) + snd_pcm_unlink(substream); } void snd_pcm_release_substream(struct snd_pcm_substream *substream) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 08d5662039e3..ee601d7f0926 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -1236,6 +1236,28 @@ int snd_rawmidi_transmit(struct snd_rawmidi_substream *substream, } EXPORT_SYMBOL(snd_rawmidi_transmit); +/** + * snd_rawmidi_proceed - Discard the all pending bytes and proceed + * @substream: rawmidi substream + * + * Return: the number of discarded bytes + */ +int snd_rawmidi_proceed(struct snd_rawmidi_substream *substream) +{ + struct snd_rawmidi_runtime *runtime = substream->runtime; + unsigned long flags; + int count = 0; + + spin_lock_irqsave(&runtime->lock, flags); + if (runtime->avail < runtime->buffer_size) { + count = runtime->buffer_size - runtime->avail; + __snd_rawmidi_transmit_ack(substream, count); + } + spin_unlock_irqrestore(&runtime->lock, flags); + return count; +} +EXPORT_SYMBOL(snd_rawmidi_proceed); + static long snd_rawmidi_kernel_write1(struct snd_rawmidi_substream *substream, const unsigned char __user *userbuf, const unsigned char *kernelbuf, diff --git a/sound/core/seq/oss/seq_oss_timer.c b/sound/core/seq/oss/seq_oss_timer.c index ba127c22539a..0778d28421da 100644 --- a/sound/core/seq/oss/seq_oss_timer.c +++ b/sound/core/seq/oss/seq_oss_timer.c @@ -92,7 +92,7 @@ snd_seq_oss_process_timer_event(struct seq_oss_timer *rec, union evrec *ev) case TMR_WAIT_REL: parm += rec->cur_tick; rec->realtime = 0; - /* fall through and continue to next */ + /* fall through */ case TMR_WAIT_ABS: if (parm == 0) { rec->realtime = 1; diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 92e6524a3a9d..7d4640d1fe9f 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -393,7 +393,7 @@ static ssize_t snd_seq_read(struct file *file, char __user *buf, size_t count, if (!(snd_seq_file_flags(file) & SNDRV_SEQ_LFLG_INPUT)) return -ENXIO; - if (!access_ok(VERIFY_WRITE, buf, count)) + if (!access_ok(buf, count)) return -EFAULT; /* check client structures are in place */ diff --git a/sound/core/seq/seq_system.c b/sound/core/seq/seq_system.c index 8ce1d0b40dce..0dc5d5a45ecc 100644 --- a/sound/core/seq/seq_system.c +++ b/sound/core/seq/seq_system.c @@ -123,6 +123,7 @@ int __init snd_seq_system_client_init(void) { struct snd_seq_port_callback pcallbacks; struct snd_seq_port_info *port; + int err; port = kzalloc(sizeof(*port), GFP_KERNEL); if (!port) @@ -134,6 +135,10 @@ int __init snd_seq_system_client_init(void) /* register client */ sysclient = snd_seq_create_kernel_client(NULL, 0, "System"); + if (sysclient < 0) { + kfree(port); + return sysclient; + } /* register timer */ strcpy(port->name, "Timer"); @@ -144,7 +149,10 @@ int __init snd_seq_system_client_init(void) port->flags = SNDRV_SEQ_PORT_FLG_GIVEN_PORT; port->addr.client = sysclient; port->addr.port = SNDRV_SEQ_PORT_SYSTEM_TIMER; - snd_seq_kernel_client_ctl(sysclient, SNDRV_SEQ_IOCTL_CREATE_PORT, port); + err = snd_seq_kernel_client_ctl(sysclient, SNDRV_SEQ_IOCTL_CREATE_PORT, + port); + if (err < 0) + goto error_port; /* register announcement port */ strcpy(port->name, "Announce"); @@ -154,16 +162,24 @@ int __init snd_seq_system_client_init(void) port->flags = SNDRV_SEQ_PORT_FLG_GIVEN_PORT; port->addr.client = sysclient; port->addr.port = SNDRV_SEQ_PORT_SYSTEM_ANNOUNCE; - snd_seq_kernel_client_ctl(sysclient, SNDRV_SEQ_IOCTL_CREATE_PORT, port); + err = snd_seq_kernel_client_ctl(sysclient, SNDRV_SEQ_IOCTL_CREATE_PORT, + port); + if (err < 0) + goto error_port; announce_port = port->addr.port; kfree(port); return 0; + + error_port: + snd_seq_system_client_done(); + kfree(port); + return err; } /* unregister our internal client */ -void __exit snd_seq_system_client_done(void) +void snd_seq_system_client_done(void) { int oldsysclient = sysclient; diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c index cb988efd1ed0..e5a40795914a 100644 --- a/sound/core/seq/seq_virmidi.c +++ b/sound/core/seq/seq_virmidi.c @@ -149,9 +149,7 @@ static void snd_vmidi_output_work(struct work_struct *work) /* discard the outputs in dispatch mode unless subscribed */ if (vmidi->seq_mode == SNDRV_VIRMIDI_SEQ_DISPATCH && !(vmidi->rdev->flags & SNDRV_VIRMIDI_SUBSCRIBE)) { - char buf[32]; - while (snd_rawmidi_transmit(substream, buf, sizeof(buf)) > 0) - ; /* ignored */ + snd_rawmidi_proceed(substream); return; } diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c index 84fffabdd129..c1cfaa01a5cb 100644 --- a/sound/core/sgbuf.c +++ b/sound/core/sgbuf.c @@ -23,6 +23,7 @@ #include <linux/mm.h> #include <linux/vmalloc.h> #include <linux/export.h> +#include <asm/pgtable.h> #include <sound/memalloc.h> @@ -43,6 +44,8 @@ int snd_free_sgbuf_pages(struct snd_dma_buffer *dmab) dmab->area = NULL; tmpb.dev.type = SNDRV_DMA_TYPE_DEV; + if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_UC_SG) + tmpb.dev.type = SNDRV_DMA_TYPE_DEV_UC; tmpb.dev.dev = sgbuf->dev; for (i = 0; i < sgbuf->pages; i++) { if (!(sgbuf->table[i].addr & ~PAGE_MASK)) @@ -72,12 +75,20 @@ void *snd_malloc_sgbuf_pages(struct device *device, struct snd_dma_buffer tmpb; struct snd_sg_page *table; struct page **pgtable; + int type = SNDRV_DMA_TYPE_DEV; + pgprot_t prot = PAGE_KERNEL; dmab->area = NULL; dmab->addr = 0; dmab->private_data = sgbuf = kzalloc(sizeof(*sgbuf), GFP_KERNEL); if (! sgbuf) return NULL; + if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_UC_SG) { + type = SNDRV_DMA_TYPE_DEV_UC; +#ifdef pgprot_noncached + prot = pgprot_noncached(PAGE_KERNEL); +#endif + } sgbuf->dev = device; pages = snd_sgbuf_aligned_pages(size); sgbuf->tblsize = sgbuf_align_table(pages); @@ -98,7 +109,7 @@ void *snd_malloc_sgbuf_pages(struct device *device, if (chunk > maxpages) chunk = maxpages; chunk <<= PAGE_SHIFT; - if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, device, + if (snd_dma_alloc_pages_fallback(type, device, chunk, &tmpb) < 0) { if (!sgbuf->pages) goto _failed; @@ -125,7 +136,7 @@ void *snd_malloc_sgbuf_pages(struct device *device, } sgbuf->size = size; - dmab->area = vmap(sgbuf->page_table, sgbuf->pages, VM_MAP, PAGE_KERNEL); + dmab->area = vmap(sgbuf->page_table, sgbuf->pages, VM_MAP, prot); if (! dmab->area) goto _failed; if (res_size) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index 529d9f405fa9..052e00590259 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -41,6 +41,7 @@ config SND_OXFW * Mackie(Loud) U.420/U.420d * TASCAM FireOne * Stanton Controllers & Systems 1 Deck/Mixer + * APOGEE duet FireWire To compile this driver as a module, choose M here: the module will be called snd-oxfw. @@ -147,7 +148,9 @@ config SND_FIREWIRE_MOTU help Say Y here to enable support for FireWire devices which MOTU produced: * 828mk2 + * Traveler * 828mk3 + * Audio Express To compile this driver as a module, choose M here: the module will be called snd-firewire-motu. @@ -159,5 +162,6 @@ config SND_FIREFACE help Say Y here to include support for RME fireface series. * Fireface 400 + * Fireface 800 endif # SND_FIREWIRE diff --git a/sound/firewire/amdtp-stream-trace.h b/sound/firewire/amdtp-stream-trace.h index 54cdd4ffa9ce..ac20acf48fc6 100644 --- a/sound/firewire/amdtp-stream-trace.h +++ b/sound/firewire/amdtp-stream-trace.h @@ -131,7 +131,7 @@ TRACE_EVENT(in_packet_without_header, __entry->index = index; ), TP_printk( - "%02u %04u %04x %04x %02d %03u %3u %3u %02u %01u %02u", + "%02u %04u %04x %04x %02d %03u %02u %03u %02u %01u %02u", __entry->second, __entry->cycle, __entry->src, @@ -169,7 +169,7 @@ TRACE_EVENT(out_packet_without_header, __entry->dest = fw_parent_device(s->unit)->node_id; __entry->payload_quadlets = payload_length / 4; __entry->data_blocks = data_blocks, - __entry->data_blocks = s->data_block_counter, + __entry->data_block_counter = s->data_block_counter, __entry->packet_index = s->packet_index; __entry->irq = !!in_interrupt(); __entry->index = index; diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index cb9acfe60f6a..3ada55ed5381 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -140,6 +140,28 @@ const unsigned int amdtp_rate_table[CIP_SFC_COUNT] = { }; EXPORT_SYMBOL(amdtp_rate_table); +static int apply_constraint_to_size(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *s = hw_param_interval(params, rule->var); + const struct snd_interval *r = + hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval t = {0}; + unsigned int step = 0; + int i; + + for (i = 0; i < CIP_SFC_COUNT; ++i) { + if (snd_interval_test(r, amdtp_rate_table[i])) + step = max(step, amdtp_syt_intervals[i]); + } + + t.min = roundup(s->min, step); + t.max = rounddown(s->max, step); + t.integer = 1; + + return snd_interval_refine(s, &t); +} + /** * amdtp_stream_add_pcm_hw_constraints - add hw constraints for PCM substream * @s: the AMDTP stream, which must be initialized. @@ -194,16 +216,19 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, * number equals to SYT_INTERVAL. So the number is 8, 16 or 32, * depending on its sampling rate. For accurate period interrupt, it's * preferrable to align period/buffer sizes to current SYT_INTERVAL. - * - * TODO: These constraints can be improved with proper rules. - * Currently apply LCM of SYT_INTERVALs. */ - err = snd_pcm_hw_constraint_step(runtime, 0, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 32); + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + apply_constraint_to_size, NULL, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + goto end; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + apply_constraint_to_size, NULL, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) goto end; - err = snd_pcm_hw_constraint_step(runtime, 0, - SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 32); end: return err; } @@ -629,15 +654,17 @@ end: } static int handle_in_packet_without_header(struct amdtp_stream *s, - unsigned int payload_quadlets, unsigned int cycle, + unsigned int payload_length, unsigned int cycle, unsigned int index) { __be32 *buffer; + unsigned int payload_quadlets; unsigned int data_blocks; struct snd_pcm_substream *pcm; unsigned int pcm_frames; buffer = s->buffer.packets[s->packet_index].buffer; + payload_quadlets = payload_length / 4; data_blocks = payload_quadlets / s->data_block_quadlets; trace_in_packet_without_header(s, cycle, payload_quadlets, data_blocks, diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 93676354f87f..d91874275d2c 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -126,23 +126,6 @@ end: return err; } -static void bebob_free(struct snd_bebob *bebob) -{ - snd_bebob_stream_destroy_duplex(bebob); - fw_unit_put(bebob->unit); - - kfree(bebob->maudio_special_quirk); - - mutex_destroy(&bebob->mutex); - kfree(bebob); -} - -/* - * This module releases the FireWire unit data after all ALSA character devices - * are released by applications. This is for releasing stream data or finishing - * transactions safely. Thus at returning from .remove(), this module still keep - * references for the unit. - */ static void bebob_card_free(struct snd_card *card) { @@ -152,7 +135,7 @@ bebob_card_free(struct snd_card *card) clear_bit(bebob->card_index, devices_used); mutex_unlock(&devices_mutex); - bebob_free(card->private_data); + snd_bebob_stream_destroy_duplex(bebob); } static const struct snd_bebob_spec * @@ -192,7 +175,6 @@ do_registration(struct work_struct *work) return; mutex_lock(&devices_mutex); - for (card_index = 0; card_index < SNDRV_CARDS; card_index++) { if (!test_bit(card_index, devices_used) && enable[card_index]) break; @@ -208,6 +190,11 @@ do_registration(struct work_struct *work) mutex_unlock(&devices_mutex); return; } + set_bit(card_index, devices_used); + mutex_unlock(&devices_mutex); + + bebob->card->private_free = bebob_card_free; + bebob->card->private_data = bebob; err = name_device(bebob); if (err < 0) @@ -248,23 +235,10 @@ do_registration(struct work_struct *work) if (err < 0) goto error; - set_bit(card_index, devices_used); - mutex_unlock(&devices_mutex); - - /* - * After registered, bebob instance can be released corresponding to - * releasing the sound card instance. - */ - bebob->card->private_free = bebob_card_free; - bebob->card->private_data = bebob; bebob->registered = true; return; error: - mutex_unlock(&devices_mutex); - snd_bebob_stream_destroy_duplex(bebob); - kfree(bebob->maudio_special_quirk); - bebob->maudio_special_quirk = NULL; snd_card_free(bebob->card); dev_info(&bebob->unit->device, "Sound card registration failed: %d\n", err); @@ -295,15 +269,15 @@ bebob_probe(struct fw_unit *unit, const struct ieee1394_device_id *entry) } /* Allocate this independent of sound card instance. */ - bebob = kzalloc(sizeof(struct snd_bebob), GFP_KERNEL); - if (bebob == NULL) + bebob = devm_kzalloc(&unit->device, sizeof(struct snd_bebob), + GFP_KERNEL); + if (!bebob) return -ENOMEM; - bebob->unit = fw_unit_get(unit); - bebob->entry = entry; - bebob->spec = spec; dev_set_drvdata(&unit->device, bebob); + bebob->entry = entry; + bebob->spec = spec; mutex_init(&bebob->mutex); spin_lock_init(&bebob->lock); init_waitqueue_head(&bebob->hwdep_wait); @@ -379,12 +353,12 @@ static void bebob_remove(struct fw_unit *unit) cancel_delayed_work_sync(&bebob->dwork); if (bebob->registered) { - /* No need to wait for releasing card object in this context. */ - snd_card_free_when_closed(bebob->card); - } else { - /* Don't forget this case. */ - bebob_free(bebob); + // Block till all of ALSA character devices are released. + snd_card_free(bebob->card); } + + mutex_destroy(&bebob->mutex); + fw_unit_put(bebob->unit); } static const struct snd_bebob_rate_spec normal_rate_spec = { @@ -434,7 +408,7 @@ static const struct ieee1394_device_id bebob_id_table[] = { /* Apogee Electronics, DA/AD/DD-16X (X-FireWire card) */ SND_BEBOB_DEV_ENTRY(VEN_APOGEE, 0x00010048, &spec_normal), /* Apogee Electronics, Ensemble */ - SND_BEBOB_DEV_ENTRY(VEN_APOGEE, 0x00001eee, &spec_normal), + SND_BEBOB_DEV_ENTRY(VEN_APOGEE, 0x01eeee, &spec_normal), /* ESI, Quatafire610 */ SND_BEBOB_DEV_ENTRY(VEN_ESI, 0x00010064, &spec_normal), /* AcousticReality, eARMasterOne */ diff --git a/sound/firewire/bebob/bebob_maudio.c b/sound/firewire/bebob/bebob_maudio.c index c266997ad299..51152ca4af57 100644 --- a/sound/firewire/bebob/bebob_maudio.c +++ b/sound/firewire/bebob/bebob_maudio.c @@ -261,8 +261,9 @@ snd_bebob_maudio_special_discover(struct snd_bebob *bebob, bool is1814) struct special_params *params; int err; - params = kzalloc(sizeof(struct special_params), GFP_KERNEL); - if (params == NULL) + params = devm_kzalloc(&bebob->card->card_dev, + sizeof(struct special_params), GFP_KERNEL); + if (!params) return -ENOMEM; mutex_lock(&bebob->mutex); diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 774eb2205668..ed50b222d36e 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -122,25 +122,12 @@ static void dice_card_strings(struct snd_dice *dice) strcpy(card->mixername, "DICE"); } -static void dice_free(struct snd_dice *dice) +static void dice_card_free(struct snd_card *card) { + struct snd_dice *dice = card->private_data; + snd_dice_stream_destroy_duplex(dice); snd_dice_transaction_destroy(dice); - fw_unit_put(dice->unit); - - mutex_destroy(&dice->mutex); - kfree(dice); -} - -/* - * This module releases the FireWire unit data after all ALSA character devices - * are released by applications. This is for releasing stream data or finishing - * transactions safely. Thus at returning from .remove(), this module still keep - * references for the unit. - */ -static void dice_card_free(struct snd_card *card) -{ - dice_free(card->private_data); } static void do_registration(struct work_struct *work) @@ -155,6 +142,8 @@ static void do_registration(struct work_struct *work) &dice->card); if (err < 0) return; + dice->card->private_free = dice_card_free; + dice->card->private_data = dice; err = snd_dice_transaction_init(dice); if (err < 0) @@ -192,19 +181,10 @@ static void do_registration(struct work_struct *work) if (err < 0) goto error; - /* - * After registered, dice instance can be released corresponding to - * releasing the sound card instance. - */ - dice->card->private_free = dice_card_free; - dice->card->private_data = dice; dice->registered = true; return; error: - snd_dice_stream_destroy_duplex(dice); - snd_dice_transaction_destroy(dice); - snd_dice_stream_destroy_duplex(dice); snd_card_free(dice->card); dev_info(&dice->unit->device, "Sound card registration failed: %d\n", err); @@ -223,10 +203,9 @@ static int dice_probe(struct fw_unit *unit, } /* Allocate this independent of sound card instance. */ - dice = kzalloc(sizeof(struct snd_dice), GFP_KERNEL); - if (dice == NULL) + dice = devm_kzalloc(&unit->device, sizeof(struct snd_dice), GFP_KERNEL); + if (!dice) return -ENOMEM; - dice->unit = fw_unit_get(unit); dev_set_drvdata(&unit->device, dice); @@ -261,12 +240,12 @@ static void dice_remove(struct fw_unit *unit) cancel_delayed_work_sync(&dice->dwork); if (dice->registered) { - /* No need to wait for releasing card object in this context. */ - snd_card_free_when_closed(dice->card); - } else { - /* Don't forget this case. */ - dice_free(dice); + // Block till all of ALSA character devices are released. + snd_card_free(dice->card); } + + mutex_destroy(&dice->mutex); + fw_unit_put(dice->unit); } static void dice_bus_reset(struct fw_unit *unit) diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c index ef689997d6a5..6c6ea149ef6b 100644 --- a/sound/firewire/digi00x/digi00x.c +++ b/sound/firewire/digi00x/digi00x.c @@ -41,20 +41,12 @@ static int name_card(struct snd_dg00x *dg00x) return 0; } -static void dg00x_free(struct snd_dg00x *dg00x) +static void dg00x_card_free(struct snd_card *card) { + struct snd_dg00x *dg00x = card->private_data; + snd_dg00x_stream_destroy_duplex(dg00x); snd_dg00x_transaction_unregister(dg00x); - - fw_unit_put(dg00x->unit); - - mutex_destroy(&dg00x->mutex); - kfree(dg00x); -} - -static void dg00x_card_free(struct snd_card *card) -{ - dg00x_free(card->private_data); } static void do_registration(struct work_struct *work) @@ -70,6 +62,8 @@ static void do_registration(struct work_struct *work) &dg00x->card); if (err < 0) return; + dg00x->card->private_free = dg00x_card_free; + dg00x->card->private_data = dg00x; err = name_card(dg00x); if (err < 0) @@ -101,14 +95,10 @@ static void do_registration(struct work_struct *work) if (err < 0) goto error; - dg00x->card->private_free = dg00x_card_free; - dg00x->card->private_data = dg00x; dg00x->registered = true; return; error: - snd_dg00x_transaction_unregister(dg00x); - snd_dg00x_stream_destroy_duplex(dg00x); snd_card_free(dg00x->card); dev_info(&dg00x->unit->device, "Sound card registration failed: %d\n", err); @@ -120,8 +110,9 @@ static int snd_dg00x_probe(struct fw_unit *unit, struct snd_dg00x *dg00x; /* Allocate this independent of sound card instance. */ - dg00x = kzalloc(sizeof(struct snd_dg00x), GFP_KERNEL); - if (dg00x == NULL) + dg00x = devm_kzalloc(&unit->device, sizeof(struct snd_dg00x), + GFP_KERNEL); + if (!dg00x) return -ENOMEM; dg00x->unit = fw_unit_get(unit); @@ -173,12 +164,12 @@ static void snd_dg00x_remove(struct fw_unit *unit) cancel_delayed_work_sync(&dg00x->dwork); if (dg00x->registered) { - /* No need to wait for releasing card object in this context. */ - snd_card_free_when_closed(dg00x->card); - } else { - /* Don't forget this case. */ - dg00x_free(dg00x); + // Block till all of ALSA character devices are released. + snd_card_free(dg00x->card); } + + mutex_destroy(&dg00x->mutex); + fw_unit_put(dg00x->unit); } static const struct ieee1394_device_id snd_dg00x_id_table[] = { diff --git a/sound/firewire/fireface/Makefile b/sound/firewire/fireface/Makefile index 8f807284ba54..79a7d6d99d72 100644 --- a/sound/firewire/fireface/Makefile +++ b/sound/firewire/fireface/Makefile @@ -1,3 +1,4 @@ snd-fireface-objs := ff.o ff-transaction.o ff-midi.o ff-proc.o amdtp-ff.o \ - ff-stream.o ff-pcm.o ff-hwdep.o ff-protocol-ff400.o + ff-stream.o ff-pcm.o ff-hwdep.o ff-protocol-ff400.o \ + ff-protocol-ff800.o obj-$(CONFIG_SND_FIREFACE) += snd-fireface.o diff --git a/sound/firewire/fireface/ff-pcm.c b/sound/firewire/fireface/ff-pcm.c index bf47f9ec8703..d0bc96b20a65 100644 --- a/sound/firewire/fireface/ff-pcm.c +++ b/sound/firewire/fireface/ff-pcm.c @@ -8,11 +8,6 @@ #include "ff.h" -static inline unsigned int get_multiplier_mode_with_index(unsigned int index) -{ - return ((int)index - 1) / 2; -} - static int hw_rule_rate(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { @@ -24,10 +19,16 @@ static int hw_rule_rate(struct snd_pcm_hw_params *params, struct snd_interval t = { .min = UINT_MAX, .max = 0, .integer = 1 }; - unsigned int i, mode; + unsigned int i; for (i = 0; i < ARRAY_SIZE(amdtp_rate_table); i++) { - mode = get_multiplier_mode_with_index(i); + enum snd_ff_stream_mode mode; + int err; + + err = snd_ff_stream_get_multiplier_mode(i, &mode); + if (err < 0) + continue; + if (!snd_interval_test(c, pcm_channels[mode])) continue; @@ -49,10 +50,16 @@ static int hw_rule_channels(struct snd_pcm_hw_params *params, struct snd_interval t = { .min = UINT_MAX, .max = 0, .integer = 1 }; - unsigned int i, mode; + unsigned int i; for (i = 0; i < ARRAY_SIZE(amdtp_rate_table); i++) { - mode = get_multiplier_mode_with_index(i); + enum snd_ff_stream_mode mode; + int err; + + err = snd_ff_stream_get_multiplier_mode(i, &mode); + if (err < 0) + continue; + if (!snd_interval_test(r, amdtp_rate_table[i])) continue; @@ -66,7 +73,6 @@ static int hw_rule_channels(struct snd_pcm_hw_params *params, static void limit_channels_and_rates(struct snd_pcm_hardware *hw, const unsigned int *pcm_channels) { - unsigned int mode; unsigned int rate, channels; int i; @@ -76,7 +82,12 @@ static void limit_channels_and_rates(struct snd_pcm_hardware *hw, hw->rate_max = 0; for (i = 0; i < ARRAY_SIZE(amdtp_rate_table); i++) { - mode = get_multiplier_mode_with_index(i); + enum snd_ff_stream_mode mode; + int err; + + err = snd_ff_stream_get_multiplier_mode(i, &mode); + if (err < 0) + continue; channels = pcm_channels[mode]; if (pcm_channels[mode] == 0) @@ -141,7 +152,7 @@ static int pcm_open(struct snd_pcm_substream *substream) if (err < 0) goto release_lock; - err = ff->spec->protocol->get_clock(ff, &rate, &src); + err = snd_ff_transaction_get_clock(ff, &rate, &src); if (err < 0) goto release_lock; diff --git a/sound/firewire/fireface/ff-proc.c b/sound/firewire/fireface/ff-proc.c index 40ccbfd8ef89..a0c550dabe9a 100644 --- a/sound/firewire/fireface/ff-proc.c +++ b/sound/firewire/fireface/ff-proc.c @@ -12,16 +12,205 @@ static void proc_dump_clock_config(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_ff *ff = entry->private_data; + __le32 reg; + u32 data; + unsigned int rate; + const char *src; + int err; - ff->spec->protocol->dump_clock_config(ff, buffer); + err = snd_fw_transaction(ff->unit, TCODE_READ_BLOCK_REQUEST, + SND_FF_REG_CLOCK_CONFIG, ®, sizeof(reg), 0); + if (err < 0) + return; + + data = le32_to_cpu(reg); + + snd_iprintf(buffer, "Output S/PDIF format: %s (Emphasis: %s)\n", + (data & 0x20) ? "Professional" : "Consumer", + (data & 0x40) ? "on" : "off"); + + snd_iprintf(buffer, "Optical output interface format: %s\n", + ((data >> 8) & 0x01) ? "S/PDIF" : "ADAT"); + + snd_iprintf(buffer, "Word output single speed: %s\n", + ((data >> 8) & 0x20) ? "on" : "off"); + + snd_iprintf(buffer, "S/PDIF input interface: %s\n", + ((data >> 8) & 0x02) ? "Optical" : "Coaxial"); + + switch ((data >> 1) & 0x03) { + case 0x01: + rate = 32000; + break; + case 0x00: + rate = 44100; + break; + case 0x03: + rate = 48000; + break; + case 0x02: + default: + return; + } + + if (data & 0x08) + rate *= 2; + else if (data & 0x10) + rate *= 4; + + snd_iprintf(buffer, "Sampling rate: %d\n", rate); + + if (data & 0x01) { + src = "Internal"; + } else { + switch ((data >> 10) & 0x07) { + case 0x00: + src = "ADAT1"; + break; + case 0x01: + src = "ADAT2"; + break; + case 0x03: + src = "S/PDIF"; + break; + case 0x04: + src = "Word"; + break; + case 0x05: + src = "LTC"; + break; + default: + return; + } + } + + snd_iprintf(buffer, "Sync to clock source: %s\n", src); } static void proc_dump_sync_status(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_ff *ff = entry->private_data; + __le32 reg; + u32 data; + int err; + + err = snd_fw_transaction(ff->unit, TCODE_READ_QUADLET_REQUEST, + SND_FF_REG_SYNC_STATUS, ®, sizeof(reg), 0); + if (err < 0) + return; + + data = le32_to_cpu(reg); + + snd_iprintf(buffer, "External source detection:\n"); + + snd_iprintf(buffer, "Word Clock:"); + if ((data >> 24) & 0x20) { + if ((data >> 24) & 0x40) + snd_iprintf(buffer, "sync\n"); + else + snd_iprintf(buffer, "lock\n"); + } else { + snd_iprintf(buffer, "none\n"); + } + + snd_iprintf(buffer, "S/PDIF:"); + if ((data >> 16) & 0x10) { + if ((data >> 16) & 0x04) + snd_iprintf(buffer, "sync\n"); + else + snd_iprintf(buffer, "lock\n"); + } else { + snd_iprintf(buffer, "none\n"); + } + + snd_iprintf(buffer, "ADAT1:"); + if ((data >> 8) & 0x04) { + if ((data >> 8) & 0x10) + snd_iprintf(buffer, "sync\n"); + else + snd_iprintf(buffer, "lock\n"); + } else { + snd_iprintf(buffer, "none\n"); + } + + snd_iprintf(buffer, "ADAT2:"); + if ((data >> 8) & 0x08) { + if ((data >> 8) & 0x20) + snd_iprintf(buffer, "sync\n"); + else + snd_iprintf(buffer, "lock\n"); + } else { + snd_iprintf(buffer, "none\n"); + } + + snd_iprintf(buffer, "\nUsed external source:\n"); + + if (((data >> 22) & 0x07) == 0x07) { + snd_iprintf(buffer, "None\n"); + } else { + switch ((data >> 22) & 0x07) { + case 0x00: + snd_iprintf(buffer, "ADAT1:"); + break; + case 0x01: + snd_iprintf(buffer, "ADAT2:"); + break; + case 0x03: + snd_iprintf(buffer, "S/PDIF:"); + break; + case 0x04: + snd_iprintf(buffer, "Word:"); + break; + case 0x07: + snd_iprintf(buffer, "Nothing:"); + break; + case 0x02: + case 0x05: + case 0x06: + default: + snd_iprintf(buffer, "unknown:"); + break; + } + + if ((data >> 25) & 0x07) { + switch ((data >> 25) & 0x07) { + case 0x01: + snd_iprintf(buffer, "32000\n"); + break; + case 0x02: + snd_iprintf(buffer, "44100\n"); + break; + case 0x03: + snd_iprintf(buffer, "48000\n"); + break; + case 0x04: + snd_iprintf(buffer, "64000\n"); + break; + case 0x05: + snd_iprintf(buffer, "88200\n"); + break; + case 0x06: + snd_iprintf(buffer, "96000\n"); + break; + case 0x07: + snd_iprintf(buffer, "128000\n"); + break; + case 0x08: + snd_iprintf(buffer, "176400\n"); + break; + case 0x09: + snd_iprintf(buffer, "192000\n"); + break; + case 0x00: + snd_iprintf(buffer, "unknown\n"); + break; + } + } + } - ff->spec->protocol->dump_sync_status(ff, buffer); + snd_iprintf(buffer, "Multiplied:"); + snd_iprintf(buffer, "%d\n", (data & 0x3ff) * 250); } static void add_node(struct snd_ff *ff, struct snd_info_entry *root, diff --git a/sound/firewire/fireface/ff-protocol-ff400.c b/sound/firewire/fireface/ff-protocol-ff400.c index 64c3cb0fb926..2280fab9b3c7 100644 --- a/sound/firewire/fireface/ff-protocol-ff400.c +++ b/sound/firewire/fireface/ff-protocol-ff400.c @@ -14,85 +14,60 @@ #define FF400_ISOC_COMM_START 0x000080100508ull #define FF400_TX_PACKET_FORMAT 0x00008010050cull #define FF400_ISOC_COMM_STOP 0x000080100510ull -#define FF400_SYNC_STATUS 0x0000801c0000ull -#define FF400_FETCH_PCM_FRAMES 0x0000801c0000ull /* For block request. */ -#define FF400_CLOCK_CONFIG 0x0000801c0004ull -#define FF400_MIDI_HIGH_ADDR 0x0000801003f4ull -#define FF400_MIDI_RX_PORT_0 0x000080180000ull -#define FF400_MIDI_RX_PORT_1 0x000080190000ull - -static int ff400_get_clock(struct snd_ff *ff, unsigned int *rate, - enum snd_ff_clock_src *src) +/* + * Fireface 400 manages isochronous channel number in 3 bit field. Therefore, + * we can allocate between 0 and 7 channel. + */ +static int keep_resources(struct snd_ff *ff, unsigned int rate) { - __le32 reg; - u32 data; + enum snd_ff_stream_mode mode; + int i; int err; - err = snd_fw_transaction(ff->unit, TCODE_READ_QUADLET_REQUEST, - FF400_SYNC_STATUS, ®, sizeof(reg), 0); + // Check whether the given value is supported or not. + for (i = 0; i < CIP_SFC_COUNT; i++) { + if (amdtp_rate_table[i] == rate) + break; + } + if (i >= CIP_SFC_COUNT) + return -EINVAL; + + err = snd_ff_stream_get_multiplier_mode(i, &mode); if (err < 0) return err; - data = le32_to_cpu(reg); - /* Calculate sampling rate. */ - switch ((data >> 1) & 0x03) { - case 0x01: - *rate = 32000; - break; - case 0x00: - *rate = 44100; - break; - case 0x03: - *rate = 48000; - break; - case 0x02: - default: - return -EIO; - } - - if (data & 0x08) - *rate *= 2; - else if (data & 0x10) - *rate *= 4; + /* Keep resources for in-stream. */ + ff->tx_resources.channels_mask = 0x00000000000000ffuLL; + err = fw_iso_resources_allocate(&ff->tx_resources, + amdtp_stream_get_max_payload(&ff->tx_stream), + fw_parent_device(ff->unit)->max_speed); + if (err < 0) + return err; - /* Calculate source of clock. */ - if (data & 0x01) { - *src = SND_FF_CLOCK_SRC_INTERNAL; - } else { - /* TODO: 0x00, 0x01, 0x02, 0x06, 0x07? */ - switch ((data >> 10) & 0x07) { - case 0x03: - *src = SND_FF_CLOCK_SRC_SPDIF; - break; - case 0x04: - *src = SND_FF_CLOCK_SRC_WORD; - break; - case 0x05: - *src = SND_FF_CLOCK_SRC_LTC; - break; - case 0x00: - default: - *src = SND_FF_CLOCK_SRC_ADAT; - break; - } - } + /* Keep resources for out-stream. */ + err = amdtp_ff_set_parameters(&ff->rx_stream, rate, + ff->spec->pcm_playback_channels[mode]); + if (err < 0) + return err; + ff->rx_resources.channels_mask = 0x00000000000000ffuLL; + err = fw_iso_resources_allocate(&ff->rx_resources, + amdtp_stream_get_max_payload(&ff->rx_stream), + fw_parent_device(ff->unit)->max_speed); + if (err < 0) + fw_iso_resources_free(&ff->tx_resources); - return 0; + return err; } static int ff400_begin_session(struct snd_ff *ff, unsigned int rate) { __le32 reg; - int i, err; + int err; - /* Check whether the given value is supported or not. */ - for (i = 0; i < CIP_SFC_COUNT; i++) { - if (amdtp_rate_table[i] == rate) - break; - } - if (i == CIP_SFC_COUNT) - return -EINVAL; + err = keep_resources(ff, rate); + if (err < 0) + return err; /* Set the number of data blocks transferred in a second. */ reg = cpu_to_le32(rate); @@ -142,233 +117,45 @@ static void ff400_finish_session(struct snd_ff *ff) FF400_ISOC_COMM_STOP, ®, sizeof(reg), 0); } -static int ff400_switch_fetching_mode(struct snd_ff *ff, bool enable) +static void ff400_handle_midi_msg(struct snd_ff *ff, __le32 *buf, size_t length) { - __le32 *reg; int i; - int err; - reg = kcalloc(18, sizeof(__le32), GFP_KERNEL); - if (reg == NULL) - return -ENOMEM; + for (i = 0; i < length / 4; i++) { + u32 quad = le32_to_cpu(buf[i]); + u8 byte; + unsigned int index; + struct snd_rawmidi_substream *substream; - if (enable) { + /* Message in first port. */ /* - * Each quadlet is corresponding to data channels in a data - * blocks in reverse order. Precisely, quadlets for available - * data channels should be enabled. Here, I take second best - * to fetch PCM frames from all of data channels regardless of - * stf. + * This value may represent the index of this unit when the same + * units are on the same IEEE 1394 bus. This driver doesn't use + * it. */ - for (i = 0; i < 18; ++i) - reg[i] = cpu_to_le32(0x00000001); - } - - err = snd_fw_transaction(ff->unit, TCODE_WRITE_BLOCK_REQUEST, - FF400_FETCH_PCM_FRAMES, reg, - sizeof(__le32) * 18, 0); - kfree(reg); - return err; -} - -static void ff400_dump_sync_status(struct snd_ff *ff, - struct snd_info_buffer *buffer) -{ - __le32 reg; - u32 data; - int err; - - err = snd_fw_transaction(ff->unit, TCODE_READ_QUADLET_REQUEST, - FF400_SYNC_STATUS, ®, sizeof(reg), 0); - if (err < 0) - return; - - data = le32_to_cpu(reg); - - snd_iprintf(buffer, "External source detection:\n"); - - snd_iprintf(buffer, "Word Clock:"); - if ((data >> 24) & 0x20) { - if ((data >> 24) & 0x40) - snd_iprintf(buffer, "sync\n"); - else - snd_iprintf(buffer, "lock\n"); - } else { - snd_iprintf(buffer, "none\n"); - } - - snd_iprintf(buffer, "S/PDIF:"); - if ((data >> 16) & 0x10) { - if ((data >> 16) & 0x04) - snd_iprintf(buffer, "sync\n"); - else - snd_iprintf(buffer, "lock\n"); - } else { - snd_iprintf(buffer, "none\n"); - } - - snd_iprintf(buffer, "ADAT:"); - if ((data >> 8) & 0x04) { - if ((data >> 8) & 0x10) - snd_iprintf(buffer, "sync\n"); - else - snd_iprintf(buffer, "lock\n"); - } else { - snd_iprintf(buffer, "none\n"); - } - - snd_iprintf(buffer, "\nUsed external source:\n"); - - if (((data >> 22) & 0x07) == 0x07) { - snd_iprintf(buffer, "None\n"); - } else { - switch ((data >> 22) & 0x07) { - case 0x00: - snd_iprintf(buffer, "ADAT:"); - break; - case 0x03: - snd_iprintf(buffer, "S/PDIF:"); - break; - case 0x04: - snd_iprintf(buffer, "Word:"); - break; - case 0x07: - snd_iprintf(buffer, "Nothing:"); - break; - case 0x01: - case 0x02: - case 0x05: - case 0x06: - default: - snd_iprintf(buffer, "unknown:"); - break; - } - - if ((data >> 25) & 0x07) { - switch ((data >> 25) & 0x07) { - case 0x01: - snd_iprintf(buffer, "32000\n"); - break; - case 0x02: - snd_iprintf(buffer, "44100\n"); - break; - case 0x03: - snd_iprintf(buffer, "48000\n"); - break; - case 0x04: - snd_iprintf(buffer, "64000\n"); - break; - case 0x05: - snd_iprintf(buffer, "88200\n"); - break; - case 0x06: - snd_iprintf(buffer, "96000\n"); - break; - case 0x07: - snd_iprintf(buffer, "128000\n"); - break; - case 0x08: - snd_iprintf(buffer, "176400\n"); - break; - case 0x09: - snd_iprintf(buffer, "192000\n"); - break; - case 0x00: - snd_iprintf(buffer, "unknown\n"); - break; + index = (quad >> 8) & 0xff; + if (index > 0) { + substream = READ_ONCE(ff->tx_midi_substreams[0]); + if (substream != NULL) { + byte = quad & 0xff; + snd_rawmidi_receive(substream, &byte, 1); } } - } - - snd_iprintf(buffer, "Multiplied:"); - snd_iprintf(buffer, "%d\n", (data & 0x3ff) * 250); -} -static void ff400_dump_clock_config(struct snd_ff *ff, - struct snd_info_buffer *buffer) -{ - __le32 reg; - u32 data; - unsigned int rate; - const char *src; - int err; - - err = snd_fw_transaction(ff->unit, TCODE_READ_BLOCK_REQUEST, - FF400_CLOCK_CONFIG, ®, sizeof(reg), 0); - if (err < 0) - return; - - data = le32_to_cpu(reg); - - snd_iprintf(buffer, "Output S/PDIF format: %s (Emphasis: %s)\n", - (data & 0x20) ? "Professional" : "Consumer", - (data & 0x40) ? "on" : "off"); - - snd_iprintf(buffer, "Optical output interface format: %s\n", - ((data >> 8) & 0x01) ? "S/PDIF" : "ADAT"); - - snd_iprintf(buffer, "Word output single speed: %s\n", - ((data >> 8) & 0x20) ? "on" : "off"); - - snd_iprintf(buffer, "S/PDIF input interface: %s\n", - ((data >> 8) & 0x02) ? "Optical" : "Coaxial"); - - switch ((data >> 1) & 0x03) { - case 0x01: - rate = 32000; - break; - case 0x00: - rate = 44100; - break; - case 0x03: - rate = 48000; - break; - case 0x02: - default: - return; - } - - if (data & 0x08) - rate *= 2; - else if (data & 0x10) - rate *= 4; - - snd_iprintf(buffer, "Sampling rate: %d\n", rate); - - if (data & 0x01) { - src = "Internal"; - } else { - switch ((data >> 10) & 0x07) { - case 0x00: - src = "ADAT"; - break; - case 0x03: - src = "S/PDIF"; - break; - case 0x04: - src = "Word"; - break; - case 0x05: - src = "LTC"; - break; - default: - return; + /* Message in second port. */ + index = (quad >> 24) & 0xff; + if (index > 0) { + substream = READ_ONCE(ff->tx_midi_substreams[1]); + if (substream != NULL) { + byte = (quad >> 16) & 0xff; + snd_rawmidi_receive(substream, &byte, 1); + } } } - - snd_iprintf(buffer, "Sync to clock source: %s\n", src); } const struct snd_ff_protocol snd_ff_protocol_ff400 = { - .get_clock = ff400_get_clock, + .handle_midi_msg = ff400_handle_midi_msg, .begin_session = ff400_begin_session, .finish_session = ff400_finish_session, - .switch_fetching_mode = ff400_switch_fetching_mode, - - .dump_sync_status = ff400_dump_sync_status, - .dump_clock_config = ff400_dump_clock_config, - - .midi_high_addr_reg = FF400_MIDI_HIGH_ADDR, - .midi_rx_port_0_reg = FF400_MIDI_RX_PORT_0, - .midi_rx_port_1_reg = FF400_MIDI_RX_PORT_1, }; diff --git a/sound/firewire/fireface/ff-protocol-ff800.c b/sound/firewire/fireface/ff-protocol-ff800.c new file mode 100644 index 000000000000..2acbf6039770 --- /dev/null +++ b/sound/firewire/fireface/ff-protocol-ff800.c @@ -0,0 +1,143 @@ +/* + * ff-protocol-ff800.c - a part of driver for RME Fireface series + * + * Copyright (c) 2018 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include <linux/delay.h> + +#include "ff.h" + +#define FF800_STF 0x0000fc88f000 +#define FF800_RX_PACKET_FORMAT 0x0000fc88f004 +#define FF800_ALLOC_TX_STREAM 0x0000fc88f008 +#define FF800_ISOC_COMM_START 0x0000fc88f00c +#define FF800_TX_S800_FLAG 0x00000800 +#define FF800_ISOC_COMM_STOP 0x0000fc88f010 + +#define FF800_TX_PACKET_ISOC_CH 0x0000801c0008 + +static int allocate_rx_resources(struct snd_ff *ff) +{ + u32 data; + __le32 reg; + int err; + + // Controllers should allocate isochronous resources for rx stream. + err = fw_iso_resources_allocate(&ff->rx_resources, + amdtp_stream_get_max_payload(&ff->rx_stream), + fw_parent_device(ff->unit)->max_speed); + if (err < 0) + return err; + + // Set isochronous channel and the number of quadlets of rx packets. + data = ff->rx_stream.data_block_quadlets << 3; + data = (data << 8) | ff->rx_resources.channel; + reg = cpu_to_le32(data); + return snd_fw_transaction(ff->unit, TCODE_WRITE_QUADLET_REQUEST, + FF800_RX_PACKET_FORMAT, ®, sizeof(reg), 0); +} + +static int allocate_tx_resources(struct snd_ff *ff) +{ + __le32 reg; + unsigned int count; + unsigned int tx_isoc_channel; + int err; + + reg = cpu_to_le32(ff->tx_stream.data_block_quadlets); + err = snd_fw_transaction(ff->unit, TCODE_WRITE_QUADLET_REQUEST, + FF800_ALLOC_TX_STREAM, ®, sizeof(reg), 0); + if (err < 0) + return err; + + // Wait till the format of tx packet is available. + count = 0; + while (count++ < 10) { + u32 data; + err = snd_fw_transaction(ff->unit, TCODE_READ_QUADLET_REQUEST, + FF800_TX_PACKET_ISOC_CH, ®, sizeof(reg), 0); + if (err < 0) + return err; + + data = le32_to_cpu(reg); + if (data != 0xffffffff) { + tx_isoc_channel = data; + break; + } + + msleep(50); + } + if (count >= 10) + return -ETIMEDOUT; + + // NOTE: this is a makeshift to start OHCI 1394 IR context in the + // channel. On the other hand, 'struct fw_iso_resources.allocated' is + // not true and it's not deallocated at stop. + ff->tx_resources.channel = tx_isoc_channel; + + return 0; +} + +static int ff800_begin_session(struct snd_ff *ff, unsigned int rate) +{ + __le32 reg; + int err; + + reg = cpu_to_le32(rate); + err = snd_fw_transaction(ff->unit, TCODE_WRITE_QUADLET_REQUEST, + FF800_STF, ®, sizeof(reg), 0); + if (err < 0) + return err; + + // If starting isochronous communication immediately, change of STF has + // no effect. In this case, the communication runs based on former STF. + // Let's sleep for a bit. + msleep(100); + + err = allocate_rx_resources(ff); + if (err < 0) + return err; + + err = allocate_tx_resources(ff); + if (err < 0) + return err; + + reg = cpu_to_le32(0x80000000); + reg |= cpu_to_le32(ff->tx_stream.data_block_quadlets); + if (fw_parent_device(ff->unit)->max_speed == SCODE_800) + reg |= cpu_to_le32(FF800_TX_S800_FLAG); + return snd_fw_transaction(ff->unit, TCODE_WRITE_QUADLET_REQUEST, + FF800_ISOC_COMM_START, ®, sizeof(reg), 0); +} + +static void ff800_finish_session(struct snd_ff *ff) +{ + __le32 reg; + + reg = cpu_to_le32(0x80000000); + snd_fw_transaction(ff->unit, TCODE_WRITE_QUADLET_REQUEST, + FF800_ISOC_COMM_STOP, ®, sizeof(reg), 0); +} + +static void ff800_handle_midi_msg(struct snd_ff *ff, __le32 *buf, size_t length) +{ + int i; + + for (i = 0; i < length / 4; i++) { + u8 byte = le32_to_cpu(buf[i]) & 0xff; + struct snd_rawmidi_substream *substream; + + substream = READ_ONCE(ff->tx_midi_substreams[0]); + if (substream) + snd_rawmidi_receive(substream, &byte, 1); + } +} + +const struct snd_ff_protocol snd_ff_protocol_ff800 = { + .handle_midi_msg = ff800_handle_midi_msg, + .begin_session = ff800_begin_session, + .finish_session = ff800_finish_session, +}; diff --git a/sound/firewire/fireface/ff-stream.c b/sound/firewire/fireface/ff-stream.c index 78880922120e..a490e4553721 100644 --- a/sound/firewire/fireface/ff-stream.c +++ b/sound/firewire/fireface/ff-stream.c @@ -10,73 +10,71 @@ #define CALLBACK_TIMEOUT_MS 200 -static int get_rate_mode(unsigned int rate, unsigned int *mode) +int snd_ff_stream_get_multiplier_mode(enum cip_sfc sfc, + enum snd_ff_stream_mode *mode) { - int i; - - for (i = 0; i < CIP_SFC_COUNT; i++) { - if (amdtp_rate_table[i] == rate) - break; - } - - if (i == CIP_SFC_COUNT) + static const enum snd_ff_stream_mode modes[] = { + [CIP_SFC_32000] = SND_FF_STREAM_MODE_LOW, + [CIP_SFC_44100] = SND_FF_STREAM_MODE_LOW, + [CIP_SFC_48000] = SND_FF_STREAM_MODE_LOW, + [CIP_SFC_88200] = SND_FF_STREAM_MODE_MID, + [CIP_SFC_96000] = SND_FF_STREAM_MODE_MID, + [CIP_SFC_176400] = SND_FF_STREAM_MODE_HIGH, + [CIP_SFC_192000] = SND_FF_STREAM_MODE_HIGH, + }; + + if (sfc >= CIP_SFC_COUNT) return -EINVAL; - *mode = ((int)i - 1) / 2; + *mode = modes[sfc]; return 0; } -/* - * Fireface 400 manages isochronous channel number in 3 bit field. Therefore, - * we can allocate between 0 and 7 channel. - */ -static int keep_resources(struct snd_ff *ff, unsigned int rate) +static void release_resources(struct snd_ff *ff) { - int mode; - int err; - - err = get_rate_mode(rate, &mode); - if (err < 0) - return err; + fw_iso_resources_free(&ff->tx_resources); + fw_iso_resources_free(&ff->rx_resources); +} - /* Keep resources for in-stream. */ - err = amdtp_ff_set_parameters(&ff->tx_stream, rate, - ff->spec->pcm_capture_channels[mode]); - if (err < 0) - return err; - ff->tx_resources.channels_mask = 0x00000000000000ffuLL; - err = fw_iso_resources_allocate(&ff->tx_resources, - amdtp_stream_get_max_payload(&ff->tx_stream), - fw_parent_device(ff->unit)->max_speed); - if (err < 0) - return err; +static int switch_fetching_mode(struct snd_ff *ff, bool enable) +{ + unsigned int count; + __le32 *reg; + int i; + int err; - /* Keep resources for out-stream. */ - err = amdtp_ff_set_parameters(&ff->rx_stream, rate, - ff->spec->pcm_playback_channels[mode]); - if (err < 0) - return err; - ff->rx_resources.channels_mask = 0x00000000000000ffuLL; - err = fw_iso_resources_allocate(&ff->rx_resources, - amdtp_stream_get_max_payload(&ff->rx_stream), - fw_parent_device(ff->unit)->max_speed); - if (err < 0) - fw_iso_resources_free(&ff->tx_resources); + count = 0; + for (i = 0; i < SND_FF_STREAM_MODE_COUNT; ++i) + count = max(count, ff->spec->pcm_playback_channels[i]); + + reg = kcalloc(count, sizeof(__le32), GFP_KERNEL); + if (!reg) + return -ENOMEM; + + if (!enable) { + /* + * Each quadlet is corresponding to data channels in a data + * blocks in reverse order. Precisely, quadlets for available + * data channels should be enabled. Here, I take second best + * to fetch PCM frames from all of data channels regardless of + * stf. + */ + for (i = 0; i < count; ++i) + reg[i] = cpu_to_le32(0x00000001); + } + err = snd_fw_transaction(ff->unit, TCODE_WRITE_BLOCK_REQUEST, + SND_FF_REG_FETCH_PCM_FRAMES, reg, + sizeof(__le32) * count, 0); + kfree(reg); return err; } -static void release_resources(struct snd_ff *ff) -{ - fw_iso_resources_free(&ff->tx_resources); - fw_iso_resources_free(&ff->rx_resources); -} - static inline void finish_session(struct snd_ff *ff) { ff->spec->protocol->finish_session(ff); - ff->spec->protocol->switch_fetching_mode(ff, false); + switch_fetching_mode(ff, false); } static int init_stream(struct snd_ff *ff, enum amdtp_stream_direction dir) @@ -149,7 +147,7 @@ int snd_ff_stream_start_duplex(struct snd_ff *ff, unsigned int rate) if (ff->substreams_counter == 0) return 0; - err = ff->spec->protocol->get_clock(ff, &curr_rate, &src); + err = snd_ff_transaction_get_clock(ff, &curr_rate, &src); if (err < 0) return err; if (curr_rate != rate || @@ -168,9 +166,29 @@ int snd_ff_stream_start_duplex(struct snd_ff *ff, unsigned int rate) * packets. Then, the device transfers packets. */ if (!amdtp_stream_running(&ff->rx_stream)) { - err = keep_resources(ff, rate); + enum snd_ff_stream_mode mode; + int i; + + for (i = 0; i < CIP_SFC_COUNT; ++i) { + if (amdtp_rate_table[i] == rate) + break; + } + if (i >= CIP_SFC_COUNT) + return -EINVAL; + + err = snd_ff_stream_get_multiplier_mode(i, &mode); if (err < 0) - goto error; + return err; + + err = amdtp_ff_set_parameters(&ff->tx_stream, rate, + ff->spec->pcm_capture_channels[mode]); + if (err < 0) + return err; + + err = amdtp_ff_set_parameters(&ff->rx_stream, rate, + ff->spec->pcm_playback_channels[mode]); + if (err < 0) + return err; err = ff->spec->protocol->begin_session(ff, rate); if (err < 0) @@ -188,7 +206,7 @@ int snd_ff_stream_start_duplex(struct snd_ff *ff, unsigned int rate) goto error; } - err = ff->spec->protocol->switch_fetching_mode(ff, true); + err = switch_fetching_mode(ff, true); if (err < 0) goto error; } diff --git a/sound/firewire/fireface/ff-transaction.c b/sound/firewire/fireface/ff-transaction.c index 332b29f8ed75..5f4ddfd55403 100644 --- a/sound/firewire/fireface/ff-transaction.c +++ b/sound/firewire/fireface/ff-transaction.c @@ -8,6 +8,72 @@ #include "ff.h" +#define SND_FF_REG_MIDI_RX_PORT_0 0x000080180000ull +#define SND_FF_REG_MIDI_RX_PORT_1 0x000080190000ull + +int snd_ff_transaction_get_clock(struct snd_ff *ff, unsigned int *rate, + enum snd_ff_clock_src *src) +{ + __le32 reg; + u32 data; + int err; + + err = snd_fw_transaction(ff->unit, TCODE_READ_QUADLET_REQUEST, + SND_FF_REG_CLOCK_CONFIG, ®, sizeof(reg), 0); + if (err < 0) + return err; + data = le32_to_cpu(reg); + + /* Calculate sampling rate. */ + switch ((data >> 1) & 0x03) { + case 0x01: + *rate = 32000; + break; + case 0x00: + *rate = 44100; + break; + case 0x03: + *rate = 48000; + break; + case 0x02: + default: + return -EIO; + } + + if (data & 0x08) + *rate *= 2; + else if (data & 0x10) + *rate *= 4; + + /* Calculate source of clock. */ + if (data & 0x01) { + *src = SND_FF_CLOCK_SRC_INTERNAL; + } else { + /* TODO: 0x02, 0x06, 0x07? */ + switch ((data >> 10) & 0x07) { + case 0x00: + *src = SND_FF_CLOCK_SRC_ADAT1; + break; + case 0x01: + *src = SND_FF_CLOCK_SRC_ADAT2; + break; + case 0x03: + *src = SND_FF_CLOCK_SRC_SPDIF; + break; + case 0x04: + *src = SND_FF_CLOCK_SRC_WORD; + break; + case 0x05: + *src = SND_FF_CLOCK_SRC_LTC; + break; + default: + return -EIO; + } + } + + return 0; +} + static void finish_transmit_midi_msg(struct snd_ff *ff, unsigned int port, int rcode) { @@ -90,10 +156,10 @@ static void transmit_midi_msg(struct snd_ff *ff, unsigned int port) fill_midi_buf(ff, port, i, buf[i]); if (port == 0) { - addr = ff->spec->protocol->midi_rx_port_0_reg; + addr = SND_FF_REG_MIDI_RX_PORT_0; callback = finish_transmit_midi0_msg; } else { - addr = ff->spec->protocol->midi_rx_port_1_reg; + addr = SND_FF_REG_MIDI_RX_PORT_1; callback = finish_transmit_midi1_msg; } @@ -140,42 +206,10 @@ static void handle_midi_msg(struct fw_card *card, struct fw_request *request, { struct snd_ff *ff = callback_data; __le32 *buf = data; - u32 quad; - u8 byte; - unsigned int index; - struct snd_rawmidi_substream *substream; - int i; fw_send_response(card, request, RCODE_COMPLETE); - for (i = 0; i < length / 4; i++) { - quad = le32_to_cpu(buf[i]); - - /* Message in first port. */ - /* - * This value may represent the index of this unit when the same - * units are on the same IEEE 1394 bus. This driver doesn't use - * it. - */ - index = (quad >> 8) & 0xff; - if (index > 0) { - substream = READ_ONCE(ff->tx_midi_substreams[0]); - if (substream != NULL) { - byte = quad & 0xff; - snd_rawmidi_receive(substream, &byte, 1); - } - } - - /* Message in second port. */ - index = (quad >> 24) & 0xff; - if (index > 0) { - substream = READ_ONCE(ff->tx_midi_substreams[1]); - if (substream != NULL) { - byte = (quad >> 16) & 0xff; - snd_rawmidi_receive(substream, &byte, 1); - } - } - } + ff->spec->protocol->handle_midi_msg(ff, buf, length); } static int allocate_own_address(struct snd_ff *ff, int i) @@ -203,36 +237,33 @@ static int allocate_own_address(struct snd_ff *ff, int i) } /* - * The configuration to start asynchronous transactions for MIDI messages is in - * 0x'0000'8010'051c. This register includes the other options, thus this driver - * doesn't touch it and leaves the decision to userspace. The userspace MUST add - * 0x04000000 to write transactions to the register to receive any MIDI - * messages. - * - * Here, I just describe MIDI-related offsets of the register, in little-endian - * order. - * * Controllers are allowed to register higher 4 bytes of address to receive - * the transactions. The register is 0x'0000'8010'03f4. On the other hand, the - * controllers are not allowed to register lower 4 bytes of the address. They - * are forced to select from 4 options by writing corresponding bits to - * 0x'0000'8010'051c. + * the transactions. Different models have different registers for this purpose; + * e.g. 0x'0000'8010'03f4 for Fireface 400. + * The controllers are not allowed to register lower 4 bytes of the address. + * They are forced to select one of 4 options for the part of address by writing + * corresponding bits to 0x'0000'8010'051f. + * + * The 3rd-6th bits of this register are flags to indicate lower 4 bytes of + * address to which the device transferrs the transactions. In short: + * - 0x20: 0x'....'....'0000'0180 + * - 0x10: 0x'....'....'0000'0100 + * - 0x08: 0x'....'....'0000'0080 + * - 0x04: 0x'....'....'0000'0000 * - * The 3rd-6th bits in MSB of this register are used to indicate lower 4 bytes - * of address to which the device transferrs the transactions. - * - 6th: 0x'....'....'0000'0180 - * - 5th: 0x'....'....'0000'0100 - * - 4th: 0x'....'....'0000'0080 - * - 3rd: 0x'....'....'0000'0000 + * This driver configure 0x'....'....'0000'0000 to receive MIDI messages from + * units. The 3rd bit of the register should be configured, however this driver + * deligates this task to userspace applications due to a restriction that this + * register is write-only and the other bits have own effects. * - * This driver configure 0x'....'....'0000'0000 for units to receive MIDI - * messages. 3rd bit of the register should be configured, however this driver - * deligates this task to user space applications due to a restriction that - * this register is write-only and the other bits have own effects. + * Unlike Fireface 800, Fireface 400 cancels transferring asynchronous + * transactions when the 1st and 2nd of the register stand. These two bits have + * the same effect. + * - 0x02, 0x01: cancel transferring * - * The 1st and 2nd bits in LSB of this register are used to cancel transferring - * asynchronous transactions. These two bits have the same effect. - * - 1st/2nd: cancel transferring + * On the other hand, the bits have no effect on Fireface 800. This model + * cancels asynchronous transactions when the higher 4 bytes of address is + * overwritten with zero. */ int snd_ff_transaction_reregister(struct snd_ff *ff) { @@ -247,7 +278,7 @@ int snd_ff_transaction_reregister(struct snd_ff *ff) addr = (fw_card->node_id << 16) | (ff->async_handler.offset >> 32); reg = cpu_to_le32(addr); return snd_fw_transaction(ff->unit, TCODE_WRITE_QUADLET_REQUEST, - ff->spec->protocol->midi_high_addr_reg, + ff->spec->midi_high_addr, ®, sizeof(reg), 0); } @@ -288,7 +319,7 @@ void snd_ff_transaction_unregister(struct snd_ff *ff) /* Release higher 4 bytes of address. */ reg = cpu_to_le32(0x00000000); snd_fw_transaction(ff->unit, TCODE_WRITE_QUADLET_REQUEST, - ff->spec->protocol->midi_high_addr_reg, + ff->spec->midi_high_addr, ®, sizeof(reg), 0); fw_core_remove_address_handler(&ff->async_handler); diff --git a/sound/firewire/fireface/ff.c b/sound/firewire/fireface/ff.c index 4974bc7980e9..36575f4159d1 100644 --- a/sound/firewire/fireface/ff.c +++ b/sound/firewire/fireface/ff.c @@ -27,20 +27,12 @@ static void name_card(struct snd_ff *ff) dev_name(&ff->unit->device), 100 << fw_dev->max_speed); } -static void ff_free(struct snd_ff *ff) +static void ff_card_free(struct snd_card *card) { + struct snd_ff *ff = card->private_data; + snd_ff_stream_destroy_duplex(ff); snd_ff_transaction_unregister(ff); - - fw_unit_put(ff->unit); - - mutex_destroy(&ff->mutex); - kfree(ff); -} - -static void ff_card_free(struct snd_card *card) -{ - ff_free(card->private_data); } static void do_registration(struct work_struct *work) @@ -55,6 +47,8 @@ static void do_registration(struct work_struct *work) &ff->card); if (err < 0) return; + ff->card->private_free = ff_card_free; + ff->card->private_data = ff; err = snd_ff_transaction_register(ff); if (err < 0) @@ -84,14 +78,10 @@ static void do_registration(struct work_struct *work) if (err < 0) goto error; - ff->card->private_free = ff_card_free; - ff->card->private_data = ff; ff->registered = true; return; error: - snd_ff_transaction_unregister(ff); - snd_ff_stream_destroy_duplex(ff); snd_card_free(ff->card); dev_info(&ff->unit->device, "Sound card registration failed: %d\n", err); @@ -102,11 +92,9 @@ static int snd_ff_probe(struct fw_unit *unit, { struct snd_ff *ff; - ff = kzalloc(sizeof(struct snd_ff), GFP_KERNEL); - if (ff == NULL) + ff = devm_kzalloc(&unit->device, sizeof(struct snd_ff), GFP_KERNEL); + if (!ff) return -ENOMEM; - - /* initialize myself */ ff->unit = fw_unit_get(unit); dev_set_drvdata(&unit->device, ff); @@ -149,14 +137,24 @@ static void snd_ff_remove(struct fw_unit *unit) cancel_work_sync(&ff->dwork.work); if (ff->registered) { - /* No need to wait for releasing card object in this context. */ - snd_card_free_when_closed(ff->card); - } else { - /* Don't forget this case. */ - ff_free(ff); + // Block till all of ALSA character devices are released. + snd_card_free(ff->card); } + + mutex_destroy(&ff->mutex); + fw_unit_put(ff->unit); } +static const struct snd_ff_spec spec_ff800 = { + .name = "Fireface800", + .pcm_capture_channels = {28, 20, 12}, + .pcm_playback_channels = {28, 20, 12}, + .midi_in_ports = 1, + .midi_out_ports = 1, + .protocol = &snd_ff_protocol_ff800, + .midi_high_addr = 0x000200000320ull, +}; + static const struct snd_ff_spec spec_ff400 = { .name = "Fireface400", .pcm_capture_channels = {18, 14, 10}, @@ -164,9 +162,22 @@ static const struct snd_ff_spec spec_ff400 = { .midi_in_ports = 2, .midi_out_ports = 2, .protocol = &snd_ff_protocol_ff400, + .midi_high_addr = 0x0000801003f4ull, }; static const struct ieee1394_device_id snd_ff_id_table[] = { + /* Fireface 800 */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_SPECIFIER_ID | + IEEE1394_MATCH_VERSION | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_RME, + .specifier_id = OUI_RME, + .version = 0x000001, + .model_id = 0x101800, + .driver_data = (kernel_ulong_t)&spec_ff800, + }, /* Fireface 400 */ { .match_flags = IEEE1394_MATCH_VENDOR_ID | @@ -174,7 +185,7 @@ static const struct ieee1394_device_id snd_ff_id_table[] = { IEEE1394_MATCH_VERSION | IEEE1394_MATCH_MODEL_ID, .vendor_id = OUI_RME, - .specifier_id = 0x000a35, + .specifier_id = OUI_RME, .version = 0x000002, .model_id = 0x101800, .driver_data = (kernel_ulong_t)&spec_ff400, diff --git a/sound/firewire/fireface/ff.h b/sound/firewire/fireface/ff.h index 64df44beb950..7dfc7745a914 100644 --- a/sound/firewire/fireface/ff.h +++ b/sound/firewire/fireface/ff.h @@ -31,23 +31,34 @@ #include "../amdtp-stream.h" #include "../iso-resources.h" -#define SND_FF_STREAM_MODES 3 - #define SND_FF_MAXIMIM_MIDI_QUADS 9 #define SND_FF_IN_MIDI_PORTS 2 #define SND_FF_OUT_MIDI_PORTS 2 +#define SND_FF_REG_SYNC_STATUS 0x0000801c0000ull +/* For block write request. */ +#define SND_FF_REG_FETCH_PCM_FRAMES 0x0000801c0000ull +#define SND_FF_REG_CLOCK_CONFIG 0x0000801c0004ull + +enum snd_ff_stream_mode { + SND_FF_STREAM_MODE_LOW = 0, + SND_FF_STREAM_MODE_MID, + SND_FF_STREAM_MODE_HIGH, + SND_FF_STREAM_MODE_COUNT, +}; + struct snd_ff_protocol; struct snd_ff_spec { const char *const name; - const unsigned int pcm_capture_channels[SND_FF_STREAM_MODES]; - const unsigned int pcm_playback_channels[SND_FF_STREAM_MODES]; + const unsigned int pcm_capture_channels[SND_FF_STREAM_MODE_COUNT]; + const unsigned int pcm_playback_channels[SND_FF_STREAM_MODE_COUNT]; unsigned int midi_in_ports; unsigned int midi_out_ports; const struct snd_ff_protocol *protocol; + u64 midi_high_addr; }; struct snd_ff { @@ -89,31 +100,24 @@ struct snd_ff { enum snd_ff_clock_src { SND_FF_CLOCK_SRC_INTERNAL, SND_FF_CLOCK_SRC_SPDIF, - SND_FF_CLOCK_SRC_ADAT, + SND_FF_CLOCK_SRC_ADAT1, + SND_FF_CLOCK_SRC_ADAT2, SND_FF_CLOCK_SRC_WORD, SND_FF_CLOCK_SRC_LTC, - /* TODO: perhaps ADAT2 and TCO exists. */ + /* TODO: perhaps TCO exists. */ }; struct snd_ff_protocol { - int (*get_clock)(struct snd_ff *ff, unsigned int *rate, - enum snd_ff_clock_src *src); + void (*handle_midi_msg)(struct snd_ff *ff, __le32 *buf, size_t length); int (*begin_session)(struct snd_ff *ff, unsigned int rate); void (*finish_session)(struct snd_ff *ff); - int (*switch_fetching_mode)(struct snd_ff *ff, bool enable); - - void (*dump_sync_status)(struct snd_ff *ff, - struct snd_info_buffer *buffer); - void (*dump_clock_config)(struct snd_ff *ff, - struct snd_info_buffer *buffer); - - u64 midi_high_addr_reg; - u64 midi_rx_port_0_reg; - u64 midi_rx_port_1_reg; }; +extern const struct snd_ff_protocol snd_ff_protocol_ff800; extern const struct snd_ff_protocol snd_ff_protocol_ff400; +int snd_ff_transaction_get_clock(struct snd_ff *ff, unsigned int *rate, + enum snd_ff_clock_src *src); int snd_ff_transaction_register(struct snd_ff *ff); int snd_ff_transaction_reregister(struct snd_ff *ff); void snd_ff_transaction_unregister(struct snd_ff *ff); @@ -125,6 +129,8 @@ int amdtp_ff_add_pcm_hw_constraints(struct amdtp_stream *s, int amdtp_ff_init(struct amdtp_stream *s, struct fw_unit *unit, enum amdtp_stream_direction dir); +int snd_ff_stream_get_multiplier_mode(enum cip_sfc sfc, + enum snd_ff_stream_mode *mode); int snd_ff_stream_init_duplex(struct snd_ff *ff); void snd_ff_stream_destroy_duplex(struct snd_ff *ff); int snd_ff_stream_start_duplex(struct snd_ff *ff, unsigned int rate); diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index f2d073365cf6..faf0e001c4c5 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -184,36 +184,17 @@ end: return err; } -static void efw_free(struct snd_efw *efw) -{ - snd_efw_stream_destroy_duplex(efw); - snd_efw_transaction_remove_instance(efw); - fw_unit_put(efw->unit); - - kfree(efw->resp_buf); - - mutex_destroy(&efw->mutex); - kfree(efw); -} - -/* - * This module releases the FireWire unit data after all ALSA character devices - * are released by applications. This is for releasing stream data or finishing - * transactions safely. Thus at returning from .remove(), this module still keep - * references for the unit. - */ static void efw_card_free(struct snd_card *card) { struct snd_efw *efw = card->private_data; - if (efw->card_index >= 0) { - mutex_lock(&devices_mutex); - clear_bit(efw->card_index, devices_used); - mutex_unlock(&devices_mutex); - } + mutex_lock(&devices_mutex); + clear_bit(efw->card_index, devices_used); + mutex_unlock(&devices_mutex); - efw_free(card->private_data); + snd_efw_stream_destroy_duplex(efw); + snd_efw_transaction_remove_instance(efw); } static void @@ -226,9 +207,8 @@ do_registration(struct work_struct *work) if (efw->registered) return; - mutex_lock(&devices_mutex); - /* check registered cards */ + mutex_lock(&devices_mutex); for (card_index = 0; card_index < SNDRV_CARDS; ++card_index) { if (!test_bit(card_index, devices_used) && enable[card_index]) break; @@ -244,12 +224,18 @@ do_registration(struct work_struct *work) mutex_unlock(&devices_mutex); return; } + set_bit(card_index, devices_used); + mutex_unlock(&devices_mutex); + + efw->card->private_free = efw_card_free; + efw->card->private_data = efw; /* prepare response buffer */ snd_efw_resp_buf_size = clamp(snd_efw_resp_buf_size, SND_EFW_RESPONSE_MAXIMUM_BYTES, 4096U); - efw->resp_buf = kzalloc(snd_efw_resp_buf_size, GFP_KERNEL); - if (efw->resp_buf == NULL) { + efw->resp_buf = devm_kzalloc(&efw->card->card_dev, + snd_efw_resp_buf_size, GFP_KERNEL); + if (!efw->resp_buf) { err = -ENOMEM; goto error; } @@ -284,25 +270,11 @@ do_registration(struct work_struct *work) if (err < 0) goto error; - set_bit(card_index, devices_used); - mutex_unlock(&devices_mutex); - - /* - * After registered, efw instance can be released corresponding to - * releasing the sound card instance. - */ - efw->card->private_free = efw_card_free; - efw->card->private_data = efw; efw->registered = true; return; error: - mutex_unlock(&devices_mutex); - snd_efw_transaction_remove_instance(efw); - snd_efw_stream_destroy_duplex(efw); snd_card_free(efw->card); - kfree(efw->resp_buf); - efw->resp_buf = NULL; dev_info(&efw->unit->device, "Sound card registration failed: %d\n", err); } @@ -312,10 +284,9 @@ efw_probe(struct fw_unit *unit, const struct ieee1394_device_id *entry) { struct snd_efw *efw; - efw = kzalloc(sizeof(struct snd_efw), GFP_KERNEL); + efw = devm_kzalloc(&unit->device, sizeof(struct snd_efw), GFP_KERNEL); if (efw == NULL) return -ENOMEM; - efw->unit = fw_unit_get(unit); dev_set_drvdata(&unit->device, efw); @@ -363,12 +334,12 @@ static void efw_remove(struct fw_unit *unit) cancel_delayed_work_sync(&efw->dwork); if (efw->registered) { - /* No need to wait for releasing card object in this context. */ - snd_card_free_when_closed(efw->card); - } else { - /* Don't forget this case. */ - efw_free(efw); + // Block till all of ALSA character devices are released. + snd_card_free(efw->card); } + + mutex_destroy(&efw->mutex); + fw_unit_put(efw->unit); } static const struct ieee1394_device_id efw_id_table[] = { diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c index 30957477e005..9ebe510ea26b 100644 --- a/sound/firewire/isight.c +++ b/sound/firewire/isight.c @@ -602,8 +602,6 @@ static void isight_card_free(struct snd_card *card) struct isight *isight = card->private_data; fw_iso_resources_destroy(&isight->resources); - fw_unit_put(isight->unit); - mutex_destroy(&isight->mutex); } static u64 get_unit_base(struct fw_unit *unit) @@ -640,7 +638,7 @@ static int isight_probe(struct fw_unit *unit, if (!isight->audio_base) { dev_err(&unit->device, "audio unit base not found\n"); err = -ENXIO; - goto err_unit; + goto error; } fw_iso_resources_init(&isight->resources, unit); @@ -669,12 +667,12 @@ static int isight_probe(struct fw_unit *unit, dev_set_drvdata(&unit->device, isight); return 0; - -err_unit: - fw_unit_put(isight->unit); - mutex_destroy(&isight->mutex); error: snd_card_free(card); + + mutex_destroy(&isight->mutex); + fw_unit_put(isight->unit); + return err; } @@ -703,7 +701,11 @@ static void isight_remove(struct fw_unit *unit) isight_stop_streaming(isight); mutex_unlock(&isight->mutex); - snd_card_free_when_closed(isight->card); + // Block till all of ALSA character devices are released. + snd_card_free(isight->card); + + mutex_destroy(&isight->mutex); + fw_unit_put(isight->unit); } static const struct ieee1394_device_id isight_id_table[] = { diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c index 300d31b6f191..220e61926ea4 100644 --- a/sound/firewire/motu/motu.c +++ b/sound/firewire/motu/motu.c @@ -52,26 +52,12 @@ static void name_card(struct snd_motu *motu) dev_name(&motu->unit->device), 100 << fw_dev->max_speed); } -static void motu_free(struct snd_motu *motu) +static void motu_card_free(struct snd_card *card) { - snd_motu_transaction_unregister(motu); + struct snd_motu *motu = card->private_data; + snd_motu_transaction_unregister(motu); snd_motu_stream_destroy_duplex(motu); - fw_unit_put(motu->unit); - - mutex_destroy(&motu->mutex); - kfree(motu); -} - -/* - * This module releases the FireWire unit data after all ALSA character devices - * are released by applications. This is for releasing stream data or finishing - * transactions safely. Thus at returning from .remove(), this module still keep - * references for the unit. - */ -static void motu_card_free(struct snd_card *card) -{ - motu_free(card->private_data); } static void do_registration(struct work_struct *work) @@ -86,6 +72,8 @@ static void do_registration(struct work_struct *work) &motu->card); if (err < 0) return; + motu->card->private_free = motu_card_free; + motu->card->private_data = motu; name_card(motu); @@ -120,18 +108,10 @@ static void do_registration(struct work_struct *work) if (err < 0) goto error; - /* - * After registered, motu instance can be released corresponding to - * releasing the sound card instance. - */ - motu->card->private_free = motu_card_free; - motu->card->private_data = motu; motu->registered = true; return; error: - snd_motu_transaction_unregister(motu); - snd_motu_stream_destroy_duplex(motu); snd_card_free(motu->card); dev_info(&motu->unit->device, "Sound card registration failed: %d\n", err); @@ -143,14 +123,13 @@ static int motu_probe(struct fw_unit *unit, struct snd_motu *motu; /* Allocate this independently of sound card instance. */ - motu = kzalloc(sizeof(struct snd_motu), GFP_KERNEL); - if (motu == NULL) + motu = devm_kzalloc(&unit->device, sizeof(struct snd_motu), GFP_KERNEL); + if (!motu) return -ENOMEM; - - motu->spec = (const struct snd_motu_spec *)entry->driver_data; motu->unit = fw_unit_get(unit); dev_set_drvdata(&unit->device, motu); + motu->spec = (const struct snd_motu_spec *)entry->driver_data; mutex_init(&motu->mutex); spin_lock_init(&motu->lock); init_waitqueue_head(&motu->hwdep_wait); @@ -174,12 +153,12 @@ static void motu_remove(struct fw_unit *unit) cancel_delayed_work_sync(&motu->dwork); if (motu->registered) { - /* No need to wait for releasing card object in this context. */ - snd_card_free_when_closed(motu->card); - } else { - /* Don't forget this case. */ - motu_free(motu); + // Block till all of ALSA character devices are released. + snd_card_free(motu->card); } + + mutex_destroy(&motu->mutex); + fw_unit_put(motu->unit); } static void motu_bus_update(struct fw_unit *unit) diff --git a/sound/firewire/oxfw/oxfw-scs1x.c b/sound/firewire/oxfw/oxfw-scs1x.c index f33497cdc706..9d9545880a28 100644 --- a/sound/firewire/oxfw/oxfw-scs1x.c +++ b/sound/firewire/oxfw/oxfw-scs1x.c @@ -372,8 +372,9 @@ int snd_oxfw_scs1x_add(struct snd_oxfw *oxfw) struct fw_scs1x *scs; int err; - scs = kzalloc(sizeof(struct fw_scs1x), GFP_KERNEL); - if (scs == NULL) + scs = devm_kzalloc(&oxfw->card->card_dev, sizeof(struct fw_scs1x), + GFP_KERNEL); + if (!scs) return -ENOMEM; scs->fw_dev = fw_parent_device(oxfw->unit); oxfw->spec = scs; diff --git a/sound/firewire/oxfw/oxfw-spkr.c b/sound/firewire/oxfw/oxfw-spkr.c index cb905af0660d..66d4b1f73f0f 100644 --- a/sound/firewire/oxfw/oxfw-spkr.c +++ b/sound/firewire/oxfw/oxfw-spkr.c @@ -270,8 +270,9 @@ int snd_oxfw_add_spkr(struct snd_oxfw *oxfw, bool is_lacie) unsigned int i, first_ch; int err; - spkr = kzalloc(sizeof(struct fw_spkr), GFP_KERNEL); - if (spkr == NULL) + spkr = devm_kzalloc(&oxfw->card->card_dev, sizeof(struct fw_spkr), + GFP_KERNEL); + if (!spkr) return -ENOMEM; oxfw->spec = spkr; diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index d9361f352133..f230a9e44c3c 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -517,8 +517,9 @@ assume_stream_formats(struct snd_oxfw *oxfw, enum avc_general_plug_dir dir, if (err < 0) goto end; - formats[eid] = kmemdup(buf, *len, GFP_KERNEL); - if (formats[eid] == NULL) { + formats[eid] = devm_kmemdup(&oxfw->card->card_dev, buf, *len, + GFP_KERNEL); + if (!formats[eid]) { err = -ENOMEM; goto end; } @@ -535,7 +536,8 @@ assume_stream_formats(struct snd_oxfw *oxfw, enum avc_general_plug_dir dir, continue; eid++; - formats[eid] = kmemdup(buf, *len, GFP_KERNEL); + formats[eid] = devm_kmemdup(&oxfw->card->card_dev, buf, *len, + GFP_KERNEL); if (formats[eid] == NULL) { err = -ENOMEM; goto end; @@ -597,8 +599,9 @@ static int fill_stream_formats(struct snd_oxfw *oxfw, if (err < 0) break; - formats[eid] = kmemdup(buf, len, GFP_KERNEL); - if (formats[eid] == NULL) { + formats[eid] = devm_kmemdup(&oxfw->card->card_dev, buf, len, + GFP_KERNEL); + if (!formats[eid]) { err = -ENOMEM; break; } diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 2ea8be6c8584..3d27f3378d5d 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -20,6 +20,7 @@ #define VENDOR_LACIE 0x00d04b #define VENDOR_TASCAM 0x00022e #define OUI_STANTON 0x001260 +#define OUI_APOGEE 0x0003db #define MODEL_SATELLITE 0x00200f @@ -113,35 +114,13 @@ end: return err; } -static void oxfw_free(struct snd_oxfw *oxfw) +static void oxfw_card_free(struct snd_card *card) { - unsigned int i; + struct snd_oxfw *oxfw = card->private_data; snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream); if (oxfw->has_output) snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->tx_stream); - - fw_unit_put(oxfw->unit); - - for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) { - kfree(oxfw->tx_stream_formats[i]); - kfree(oxfw->rx_stream_formats[i]); - } - - kfree(oxfw->spec); - mutex_destroy(&oxfw->mutex); - kfree(oxfw); -} - -/* - * This module releases the FireWire unit data after all ALSA character devices - * are released by applications. This is for releasing stream data or finishing - * transactions safely. Thus at returning from .remove(), this module still keep - * references for the unit. - */ -static void oxfw_card_free(struct snd_card *card) -{ - oxfw_free(card->private_data); } static int detect_quirks(struct snd_oxfw *oxfw) @@ -208,7 +187,6 @@ static int detect_quirks(struct snd_oxfw *oxfw) static void do_registration(struct work_struct *work) { struct snd_oxfw *oxfw = container_of(work, struct snd_oxfw, dwork.work); - int i; int err; if (oxfw->registered) @@ -218,6 +196,8 @@ static void do_registration(struct work_struct *work) &oxfw->card); if (err < 0) return; + oxfw->card->private_free = oxfw_card_free; + oxfw->card->private_data = oxfw; err = name_card(oxfw); if (err < 0) @@ -258,28 +238,11 @@ static void do_registration(struct work_struct *work) if (err < 0) goto error; - /* - * After registered, oxfw instance can be released corresponding to - * releasing the sound card instance. - */ - oxfw->card->private_free = oxfw_card_free; - oxfw->card->private_data = oxfw; oxfw->registered = true; return; error: - snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream); - if (oxfw->has_output) - snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->tx_stream); - for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; ++i) { - kfree(oxfw->tx_stream_formats[i]); - oxfw->tx_stream_formats[i] = NULL; - kfree(oxfw->rx_stream_formats[i]); - oxfw->rx_stream_formats[i] = NULL; - } snd_card_free(oxfw->card); - kfree(oxfw->spec); - oxfw->spec = NULL; dev_info(&oxfw->unit->device, "Sound card registration failed: %d\n", err); } @@ -293,14 +256,13 @@ static int oxfw_probe(struct fw_unit *unit, return -ENODEV; /* Allocate this independent of sound card instance. */ - oxfw = kzalloc(sizeof(struct snd_oxfw), GFP_KERNEL); - if (oxfw == NULL) + oxfw = devm_kzalloc(&unit->device, sizeof(struct snd_oxfw), GFP_KERNEL); + if (!oxfw) return -ENOMEM; - - oxfw->entry = entry; oxfw->unit = fw_unit_get(unit); dev_set_drvdata(&unit->device, oxfw); + oxfw->entry = entry; mutex_init(&oxfw->mutex); spin_lock_init(&oxfw->lock); init_waitqueue_head(&oxfw->hwdep_wait); @@ -347,12 +309,12 @@ static void oxfw_remove(struct fw_unit *unit) cancel_delayed_work_sync(&oxfw->dwork); if (oxfw->registered) { - /* No need to wait for releasing card object in this context. */ - snd_card_free_when_closed(oxfw->card); - } else { - /* Don't forget this case. */ - oxfw_free(oxfw); + // Block till all of ALSA character devices are released. + snd_card_free(oxfw->card); } + + mutex_destroy(&oxfw->mutex); + fw_unit_put(oxfw->unit); } static const struct compat_info griffin_firewave = { @@ -436,6 +398,13 @@ static const struct ieee1394_device_id oxfw_id_table[] = { .vendor_id = OUI_STANTON, .model_id = 0x002000, }, + // APOGEE, duet FireWire + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_APOGEE, + .model_id = 0x01dddd, + }, { } }; MODULE_DEVICE_TABLE(ieee1394, oxfw_id_table); diff --git a/sound/firewire/tascam/amdtp-tascam.c b/sound/firewire/tascam/amdtp-tascam.c index ab482423c165..a52d1f76c610 100644 --- a/sound/firewire/tascam/amdtp-tascam.c +++ b/sound/firewire/tascam/amdtp-tascam.c @@ -117,6 +117,55 @@ int amdtp_tscm_add_pcm_hw_constraints(struct amdtp_stream *s, return amdtp_stream_add_pcm_hw_constraints(s, runtime); } +static void read_status_messages(struct amdtp_stream *s, + __be32 *buffer, unsigned int data_blocks) +{ + struct snd_tscm *tscm = container_of(s, struct snd_tscm, tx_stream); + bool used = READ_ONCE(tscm->hwdep->used); + int i; + + for (i = 0; i < data_blocks; i++) { + unsigned int index; + __be32 before; + __be32 after; + + index = be32_to_cpu(buffer[0]) % SNDRV_FIREWIRE_TASCAM_STATE_COUNT; + before = tscm->state[index]; + after = buffer[s->data_block_quadlets - 1]; + + if (used && index > 4 && index < 16) { + __be32 mask; + + if (index == 5) + mask = cpu_to_be32(~0x0000ffff); + else if (index == 6) + mask = cpu_to_be32(~0x0000ffff); + else if (index == 8) + mask = cpu_to_be32(~0x000f0f00); + else + mask = cpu_to_be32(~0x00000000); + + if ((before ^ after) & mask) { + struct snd_firewire_tascam_change *entry = + &tscm->queue[tscm->push_pos]; + + spin_lock_irq(&tscm->lock); + entry->index = index; + entry->before = before; + entry->after = after; + if (++tscm->push_pos >= SND_TSCM_QUEUE_COUNT) + tscm->push_pos = 0; + spin_unlock_irq(&tscm->lock); + + wake_up(&tscm->hwdep_wait); + } + } + + tscm->state[index] = after; + buffer += s->data_block_quadlets; + } +} + static unsigned int process_tx_data_blocks(struct amdtp_stream *s, __be32 *buffer, unsigned int data_blocks, @@ -128,7 +177,7 @@ static unsigned int process_tx_data_blocks(struct amdtp_stream *s, if (data_blocks > 0 && pcm) read_pcm_s32(s, pcm, buffer, data_blocks); - /* A place holder for control messages. */ + read_status_messages(s, buffer, data_blocks); return data_blocks; } diff --git a/sound/firewire/tascam/tascam-hwdep.c b/sound/firewire/tascam/tascam-hwdep.c index 4e4c1e9020e8..0414abf5daa8 100644 --- a/sound/firewire/tascam/tascam-hwdep.c +++ b/sound/firewire/tascam/tascam-hwdep.c @@ -16,18 +16,93 @@ #include "tascam.h" +static long tscm_hwdep_read_locked(struct snd_tscm *tscm, char __user *buf, + long count, loff_t *offset) +{ + struct snd_firewire_event_lock_status event = { + .type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS, + }; + + event.status = (tscm->dev_lock_count > 0); + tscm->dev_lock_changed = false; + count = min_t(long, count, sizeof(event)); + + spin_unlock_irq(&tscm->lock); + + if (copy_to_user(buf, &event, count)) + return -EFAULT; + + return count; +} + +static long tscm_hwdep_read_queue(struct snd_tscm *tscm, char __user *buf, + long remained, loff_t *offset) +{ + char __user *pos = buf; + unsigned int type = SNDRV_FIREWIRE_EVENT_TASCAM_CONTROL; + struct snd_firewire_tascam_change *entries = tscm->queue; + long count; + + // At least, one control event can be copied. + if (remained < sizeof(type) + sizeof(*entries)) { + spin_unlock_irq(&tscm->lock); + return -EINVAL; + } + + // Copy the type field later. + count = sizeof(type); + remained -= sizeof(type); + pos += sizeof(type); + + while (true) { + unsigned int head_pos; + unsigned int tail_pos; + unsigned int length; + + if (tscm->pull_pos == tscm->push_pos) + break; + else if (tscm->pull_pos < tscm->push_pos) + tail_pos = tscm->push_pos; + else + tail_pos = SND_TSCM_QUEUE_COUNT; + head_pos = tscm->pull_pos; + + length = (tail_pos - head_pos) * sizeof(*entries); + if (remained < length) + length = rounddown(remained, sizeof(*entries)); + if (length == 0) + break; + + spin_unlock_irq(&tscm->lock); + if (copy_to_user(pos, &entries[head_pos], length)) + return -EFAULT; + + spin_lock_irq(&tscm->lock); + + tscm->pull_pos = tail_pos % SND_TSCM_QUEUE_COUNT; + + count += length; + remained -= length; + pos += length; + } + + spin_unlock_irq(&tscm->lock); + + if (copy_to_user(buf, &type, sizeof(type))) + return -EFAULT; + + return count; +} + static long hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count, loff_t *offset) { struct snd_tscm *tscm = hwdep->private_data; DEFINE_WAIT(wait); - union snd_firewire_event event = { - .lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS, - }; spin_lock_irq(&tscm->lock); - while (!tscm->dev_lock_changed) { + while (!tscm->dev_lock_changed && tscm->push_pos == tscm->pull_pos) { prepare_to_wait(&tscm->hwdep_wait, &wait, TASK_INTERRUPTIBLE); spin_unlock_irq(&tscm->lock); schedule(); @@ -37,15 +112,15 @@ static long hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count, spin_lock_irq(&tscm->lock); } - event.lock_status.status = (tscm->dev_lock_count > 0); - tscm->dev_lock_changed = false; - - spin_unlock_irq(&tscm->lock); - - count = min_t(long, count, sizeof(event.lock_status)); - - if (copy_to_user(buf, &event, count)) - return -EFAULT; + // NOTE: The acquired lock should be released in callee side. + if (tscm->dev_lock_changed) { + count = tscm_hwdep_read_locked(tscm, buf, count, offset); + } else if (tscm->push_pos != tscm->pull_pos) { + count = tscm_hwdep_read_queue(tscm, buf, count, offset); + } else { + spin_unlock_irq(&tscm->lock); + count = 0; + } return count; } @@ -59,7 +134,7 @@ static __poll_t hwdep_poll(struct snd_hwdep *hwdep, struct file *file, poll_wait(file, &tscm->hwdep_wait, wait); spin_lock_irq(&tscm->lock); - if (tscm->dev_lock_changed) + if (tscm->dev_lock_changed || tscm->push_pos != tscm->pull_pos) events = EPOLLIN | EPOLLRDNORM; else events = 0; @@ -123,6 +198,14 @@ static int hwdep_unlock(struct snd_tscm *tscm) return err; } +static int tscm_hwdep_state(struct snd_tscm *tscm, void __user *arg) +{ + if (copy_to_user(arg, tscm->state, sizeof(tscm->state))) + return -EFAULT; + + return 0; +} + static int hwdep_release(struct snd_hwdep *hwdep, struct file *file) { struct snd_tscm *tscm = hwdep->private_data; @@ -147,6 +230,8 @@ static int hwdep_ioctl(struct snd_hwdep *hwdep, struct file *file, return hwdep_lock(tscm); case SNDRV_FIREWIRE_IOCTL_UNLOCK: return hwdep_unlock(tscm); + case SNDRV_FIREWIRE_IOCTL_TASCAM_STATE: + return tscm_hwdep_state(tscm, (void __user *)arg); default: return -ENOIOCTLCMD; } @@ -185,5 +270,7 @@ int snd_tscm_create_hwdep_device(struct snd_tscm *tscm) hwdep->private_data = tscm; hwdep->exclusive = true; + tscm->hwdep = hwdep; + return err; } diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c index d3fdc463a884..ef57fa4db323 100644 --- a/sound/firewire/tascam/tascam.c +++ b/sound/firewire/tascam/tascam.c @@ -85,20 +85,12 @@ static int identify_model(struct snd_tscm *tscm) return 0; } -static void tscm_free(struct snd_tscm *tscm) +static void tscm_card_free(struct snd_card *card) { + struct snd_tscm *tscm = card->private_data; + snd_tscm_transaction_unregister(tscm); snd_tscm_stream_destroy_duplex(tscm); - - fw_unit_put(tscm->unit); - - mutex_destroy(&tscm->mutex); - kfree(tscm); -} - -static void tscm_card_free(struct snd_card *card) -{ - tscm_free(card->private_data); } static void do_registration(struct work_struct *work) @@ -110,6 +102,8 @@ static void do_registration(struct work_struct *work) &tscm->card); if (err < 0) return; + tscm->card->private_free = tscm_card_free; + tscm->card->private_data = tscm; err = identify_model(tscm); if (err < 0) @@ -141,18 +135,10 @@ static void do_registration(struct work_struct *work) if (err < 0) goto error; - /* - * After registered, tscm instance can be released corresponding to - * releasing the sound card instance. - */ - tscm->card->private_free = tscm_card_free; - tscm->card->private_data = tscm; tscm->registered = true; return; error: - snd_tscm_transaction_unregister(tscm); - snd_tscm_stream_destroy_duplex(tscm); snd_card_free(tscm->card); dev_info(&tscm->unit->device, "Sound card registration failed: %d\n", err); @@ -164,11 +150,9 @@ static int snd_tscm_probe(struct fw_unit *unit, struct snd_tscm *tscm; /* Allocate this independent of sound card instance. */ - tscm = kzalloc(sizeof(struct snd_tscm), GFP_KERNEL); - if (tscm == NULL) + tscm = devm_kzalloc(&unit->device, sizeof(struct snd_tscm), GFP_KERNEL); + if (!tscm) return -ENOMEM; - - /* initialize myself */ tscm->unit = fw_unit_get(unit); dev_set_drvdata(&unit->device, tscm); @@ -216,12 +200,12 @@ static void snd_tscm_remove(struct fw_unit *unit) cancel_delayed_work_sync(&tscm->dwork); if (tscm->registered) { - /* No need to wait for releasing card object in this context. */ - snd_card_free_when_closed(tscm->card); - } else { - /* Don't forget this case. */ - tscm_free(tscm); + // Block till all of ALSA character devices are released. + snd_card_free(tscm->card); } + + mutex_destroy(&tscm->mutex); + fw_unit_put(tscm->unit); } static const struct ieee1394_device_id snd_tscm_id_table[] = { diff --git a/sound/firewire/tascam/tascam.h b/sound/firewire/tascam/tascam.h index a5bd167eb5d9..6a411ee0dcf1 100644 --- a/sound/firewire/tascam/tascam.h +++ b/sound/firewire/tascam/tascam.h @@ -62,6 +62,8 @@ struct snd_fw_async_midi_port { int consume_bytes; }; +#define SND_TSCM_QUEUE_COUNT 16 + struct snd_tscm { struct snd_card *card; struct fw_unit *unit; @@ -89,6 +91,13 @@ struct snd_tscm { /* For MIDI message outgoing transactions. */ struct snd_fw_async_midi_port out_ports[TSCM_MIDI_OUT_PORT_MAX]; + + // A cache of status information in tx isoc packets. + __be32 state[SNDRV_FIREWIRE_TASCAM_STATE_COUNT]; + struct snd_hwdep *hwdep; + struct snd_firewire_tascam_change queue[SND_TSCM_QUEUE_COUNT]; + unsigned int pull_pos; + unsigned int push_pos; }; #define TSCM_ADDR_BASE 0xffff00000000ull diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index 5bc4a1d587d4..60cb00fd0c69 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -48,9 +48,11 @@ void snd_hdac_ext_bus_ppcap_enable(struct hdac_bus *bus, bool enable) } if (enable) - snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, 0, AZX_PPCTL_GPROCEN); + snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, + AZX_PPCTL_GPROCEN, AZX_PPCTL_GPROCEN); else - snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, AZX_PPCTL_GPROCEN, 0); + snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, + AZX_PPCTL_GPROCEN, 0); } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_ppcap_enable); @@ -68,9 +70,11 @@ void snd_hdac_ext_bus_ppcap_int_enable(struct hdac_bus *bus, bool enable) } if (enable) - snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, 0, AZX_PPCTL_PIE); + snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, + AZX_PPCTL_PIE, AZX_PPCTL_PIE); else - snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, AZX_PPCTL_PIE, 0); + snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, + AZX_PPCTL_PIE, 0); } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_ppcap_int_enable); @@ -194,7 +198,8 @@ static int check_hdac_link_power_active(struct hdac_ext_link *link, bool enable) */ int snd_hdac_ext_bus_link_power_up(struct hdac_ext_link *link) { - snd_hdac_updatel(link->ml_addr, AZX_REG_ML_LCTL, 0, AZX_MLCTL_SPA); + snd_hdac_updatel(link->ml_addr, AZX_REG_ML_LCTL, + AZX_MLCTL_SPA, AZX_MLCTL_SPA); return check_hdac_link_power_active(link, true); } @@ -222,8 +227,8 @@ int snd_hdac_ext_bus_link_power_up_all(struct hdac_bus *bus) int ret; list_for_each_entry(hlink, &bus->hlink_list, list) { - snd_hdac_updatel(hlink->ml_addr, - AZX_REG_ML_LCTL, 0, AZX_MLCTL_SPA); + snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, + AZX_MLCTL_SPA, AZX_MLCTL_SPA); ret = check_hdac_link_power_active(hlink, true); if (ret < 0) return ret; @@ -243,7 +248,8 @@ int snd_hdac_ext_bus_link_power_down_all(struct hdac_bus *bus) int ret; list_for_each_entry(hlink, &bus->hlink_list, list) { - snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, AZX_MLCTL_SPA, 0); + snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, + AZX_MLCTL_SPA, 0); ret = check_hdac_link_power_active(hlink, false); if (ret < 0) return ret; diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c index 714a51721a31..012305177f68 100644 --- a/sound/hda/hdac_bus.c +++ b/sound/hda/hdac_bus.c @@ -9,8 +9,6 @@ #include <sound/hdaudio.h> #include "trace.h" -static void process_unsol_events(struct work_struct *work); - static const struct hdac_bus_ops default_ops = { .command = snd_hdac_bus_send_cmd, .get_response = snd_hdac_bus_get_response, @@ -37,7 +35,7 @@ int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev, bus->io_ops = io_ops; INIT_LIST_HEAD(&bus->stream_list); INIT_LIST_HEAD(&bus->codec_list); - INIT_WORK(&bus->unsol_work, process_unsol_events); + INIT_WORK(&bus->unsol_work, snd_hdac_bus_process_unsol_events); spin_lock_init(&bus->reg_lock); mutex_init(&bus->cmd_mutex); bus->irq = -1; @@ -148,7 +146,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_bus_queue_event); /* * process queued unsolicited events */ -static void process_unsol_events(struct work_struct *work) +void snd_hdac_bus_process_unsol_events(struct work_struct *work) { struct hdac_bus *bus = container_of(work, struct hdac_bus, unsol_work); struct hdac_device *codec; @@ -171,6 +169,7 @@ static void process_unsol_events(struct work_struct *work) drv->unsol_event(codec, res); } } +EXPORT_SYMBOL_GPL(snd_hdac_bus_process_unsol_events); /** * snd_hdac_bus_add_device - Add a codec to bus diff --git a/sound/hda/hdac_component.c b/sound/hda/hdac_component.c index 6e46a9c73aed..a6d37b9d6413 100644 --- a/sound/hda/hdac_component.c +++ b/sound/hda/hdac_component.c @@ -54,41 +54,44 @@ EXPORT_SYMBOL_GPL(snd_hdac_set_codec_wakeup); /** * snd_hdac_display_power - Power up / down the power refcount * @bus: HDA core bus + * @idx: HDA codec address, pass HDA_CODEC_IDX_CONTROLLER for controller * @enable: power up or down * - * This function is supposed to be used only by a HD-audio controller - * driver that needs the interaction with graphics driver. + * This function is used by either HD-audio controller or codec driver that + * needs the interaction with graphics driver. * - * This function manages a refcount and calls the get_power() and + * This function updates the power status, and calls the get_power() and * put_power() ops accordingly, toggling the codec wakeup, too. - * - * Returns zero for success or a negative error code. */ -int snd_hdac_display_power(struct hdac_bus *bus, bool enable) +void snd_hdac_display_power(struct hdac_bus *bus, unsigned int idx, bool enable) { struct drm_audio_component *acomp = bus->audio_component; - if (!acomp || !acomp->ops) - return -ENODEV; - dev_dbg(bus->dev, "display power %s\n", enable ? "enable" : "disable"); + if (enable) + set_bit(idx, &bus->display_power_status); + else + clear_bit(idx, &bus->display_power_status); - if (enable) { - if (!bus->drm_power_refcount++) { + if (!acomp || !acomp->ops) + return; + + if (bus->display_power_status) { + if (!bus->display_power_active) { if (acomp->ops->get_power) acomp->ops->get_power(acomp->dev); snd_hdac_set_codec_wakeup(bus, true); snd_hdac_set_codec_wakeup(bus, false); + bus->display_power_active = true; } } else { - WARN_ON(!bus->drm_power_refcount); - if (!--bus->drm_power_refcount) + if (bus->display_power_active) { if (acomp->ops->put_power) acomp->ops->put_power(acomp->dev); + bus->display_power_active = false; + } } - - return 0; } EXPORT_SYMBOL_GPL(snd_hdac_display_power); @@ -321,10 +324,12 @@ int snd_hdac_acomp_exit(struct hdac_bus *bus) if (!acomp) return 0; - WARN_ON(bus->drm_power_refcount); - if (bus->drm_power_refcount > 0 && acomp->ops) + if (WARN_ON(bus->display_power_active) && acomp->ops) acomp->ops->put_power(acomp->dev); + bus->display_power_active = false; + bus->display_power_status = 0; + component_master_del(dev, &hdac_component_master_ops); bus->audio_component = NULL; diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index dbf02a3a8d2f..95b073ee4b32 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -622,23 +622,6 @@ int snd_hdac_power_down_pm(struct hdac_device *codec) EXPORT_SYMBOL_GPL(snd_hdac_power_down_pm); #endif -/** - * snd_hdac_link_power - Enable/disable the link power for a codec - * @codec: the codec object - * @bool: enable or disable the link power - */ -int snd_hdac_link_power(struct hdac_device *codec, bool enable) -{ - if (!codec->link_power_control) - return 0; - - if (codec->bus->ops->link_power) - return codec->bus->ops->link_power(codec->bus, enable); - else - return -EINVAL; -} -EXPORT_SYMBOL_GPL(snd_hdac_link_power); - /* codec vendor labels */ struct hda_vendor_id { unsigned int id; diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 419e285e0226..996dbc850224 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -359,7 +359,8 @@ static const struct regmap_config hda_regmap_cfg = { .cache_type = REGCACHE_RBTREE, .reg_read = hda_reg_read, .reg_write = hda_reg_write, - .use_single_rw = true, + .use_single_read = true, + .use_single_write = true, }; /** diff --git a/sound/i2c/cs8427.c b/sound/i2c/cs8427.c index 2647309bc675..8afa2f888466 100644 --- a/sound/i2c/cs8427.c +++ b/sound/i2c/cs8427.c @@ -118,7 +118,7 @@ static int snd_cs8427_send_corudata(struct snd_i2c_device *device, struct cs8427 *chip = device->private_data; char *hw_data = udata ? chip->playback.hw_udata : chip->playback.hw_status; - char data[32]; + unsigned char data[32]; int err, idx; if (!memcmp(hw_data, ndata, count)) diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index ac0ab6eb40f0..47e0b2820ace 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -389,7 +389,8 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip, case OPTi9XX_HW_82C931: /* disable 3D sound (set GPIO1 as output, low) */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(20), 0x04, 0x0c); - case OPTi9XX_HW_82C933: /* FALL THROUGH */ + /* fall through */ + case OPTi9XX_HW_82C933: /* * The BTC 1817DW has QS1000 wavetable which is connected * to the serial digital input of the OPTI931. @@ -400,7 +401,8 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip, * or digital input signal. */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(26), 0x01, 0x01); - case OPTi9XX_HW_82C930: /* FALL THROUGH */ + /* fall through */ + case OPTi9XX_HW_82C930: snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x03); snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0x00, 0xff); snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0x10 | diff --git a/sound/isa/sb/emu8000_patch.c b/sound/isa/sb/emu8000_patch.c index d45a6b9d6437..3d44c358c4b3 100644 --- a/sound/isa/sb/emu8000_patch.c +++ b/sound/isa/sb/emu8000_patch.c @@ -183,10 +183,10 @@ snd_emu8000_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, } if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS) { - if (!access_ok(VERIFY_READ, data, sp->v.size)) + if (!access_ok(data, sp->v.size)) return -EFAULT; } else { - if (!access_ok(VERIFY_READ, data, sp->v.size * 2)) + if (!access_ok(data, sp->v.size * 2)) return -EFAULT; } diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index 481797744b3c..8288fae90085 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -130,13 +130,13 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) chip->playback_format = SB_DSP_HI_OUTPUT_AUTO; break; } - /* fallthru */ + /* fall through */ case SB_HW_201: if (rate > 23000) { chip->playback_format = SB_DSP_HI_OUTPUT_AUTO; break; } - /* fallthru */ + /* fall through */ case SB_HW_20: chip->playback_format = SB_DSP_LO_OUTPUT_AUTO; break; @@ -287,7 +287,7 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream) chip->capture_format = SB_DSP_HI_INPUT_AUTO; break; } - /* fallthru */ + /* fall through */ case SB_HW_20: chip->capture_format = SB_DSP_LO_INPUT_AUTO; break; @@ -387,7 +387,7 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) case SB_MODE_PLAYBACK_16: /* ok.. playback is active */ if (chip->hardware != SB_HW_JAZZ16) break; - /* fallthru */ + /* fall through */ case SB_MODE_PLAYBACK_8: substream = chip->playback_substream; if (chip->playback_format == SB_DSP_OUTPUT) @@ -397,7 +397,7 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) case SB_MODE_CAPTURE_16: if (chip->hardware != SB_HW_JAZZ16) break; - /* fallthru */ + /* fall through */ case SB_MODE_CAPTURE_8: substream = chip->capture_substream; if (chip->capture_format == SB_DSP_INPUT) diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 32453f81b95a..3a5008837576 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -1531,7 +1531,6 @@ static int snd_wss_playback_open(struct snd_pcm_substream *substream) if (err < 0) { if (chip->release_dma) chip->release_dma(chip, chip->dma_private_data, chip->dma1); - snd_free_pages(runtime->dma_area, runtime->dma_bytes); return err; } chip->playback_substream = substream; @@ -1572,7 +1571,6 @@ static int snd_wss_capture_open(struct snd_pcm_substream *substream) if (err < 0) { if (chip->release_dma) chip->release_dma(chip, chip->dma_private_data, chip->dma2); - snd_free_pages(runtime->dma_area, runtime->dma_bytes); return err; } chip->capture_substream = substream; diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c index c8904e732aaa..a4ed54aeaf1d 100644 --- a/sound/mips/hal2.c +++ b/sound/mips/hal2.c @@ -500,7 +500,8 @@ static const struct snd_pcm_hardware hal2_pcm_hw = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER), + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_SYNC_APPLPTR), .formats = SNDRV_PCM_FMTBIT_S16_BE, .rates = SNDRV_PCM_RATE_8000_48000, .rate_min = 8000, @@ -563,6 +564,8 @@ static int hal2_playback_prepare(struct snd_pcm_substream *substream) dac->sample_rate = hal2_compute_rate(dac, runtime->rate); memset(&dac->pcm_indirect, 0, sizeof(dac->pcm_indirect)); dac->pcm_indirect.hw_buffer_size = H2_BUF_SIZE; + dac->pcm_indirect.hw_queue_size = H2_BUF_SIZE / 2; + dac->pcm_indirect.hw_io = dac->buffer_dma; dac->pcm_indirect.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream); dac->substream = substream; hal2_setup_dac(hal2); @@ -575,9 +578,6 @@ static int hal2_playback_trigger(struct snd_pcm_substream *substream, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_START: - hal2->dac.pcm_indirect.hw_io = hal2->dac.buffer_dma; - hal2->dac.pcm_indirect.hw_data = 0; - substream->ops->ack(substream); hal2_start_dac(hal2); break; case SNDRV_PCM_TRIGGER_STOP: @@ -615,7 +615,6 @@ static int hal2_playback_ack(struct snd_pcm_substream *substream) struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); struct hal2_codec *dac = &hal2->dac; - dac->pcm_indirect.hw_queue_size = H2_BUF_SIZE / 2; return snd_pcm_indirect_playback_transfer(substream, &dac->pcm_indirect, hal2_playback_transfer); @@ -655,6 +654,7 @@ static int hal2_capture_prepare(struct snd_pcm_substream *substream) memset(&adc->pcm_indirect, 0, sizeof(adc->pcm_indirect)); adc->pcm_indirect.hw_buffer_size = H2_BUF_SIZE; adc->pcm_indirect.hw_queue_size = H2_BUF_SIZE / 2; + adc->pcm_indirect.hw_io = adc->buffer_dma; adc->pcm_indirect.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream); adc->substream = substream; hal2_setup_adc(hal2); @@ -667,9 +667,6 @@ static int hal2_capture_trigger(struct snd_pcm_substream *substream, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_START: - hal2->adc.pcm_indirect.hw_io = hal2->adc.buffer_dma; - hal2->adc.pcm_indirect.hw_data = 0; - printk(KERN_DEBUG "buffer_dma %x\n", hal2->adc.buffer_dma); hal2_start_adc(hal2); break; case SNDRV_PCM_TRIGGER_STOP: diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index f4459d1a9d67..27b468f057dd 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -824,7 +824,7 @@ static int snd_ac97_put_spsa(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ { struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0xff; + int shift = (kcontrol->private_value >> 8) & 0x0f; int mask = (kcontrol->private_value >> 16) & 0xff; // int invert = (kcontrol->private_value >> 24) & 0xff; unsigned short value, old, new; diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index a31fe1550903..aad74e809797 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -1183,7 +1183,7 @@ static int snd_card_asihpi_capture_prepare(struct snd_pcm_substream *substream) static u64 snd_card_asihpi_capture_formats(struct snd_card_asihpi *asihpi, u32 h_stream) { - struct hpi_format hpi_format; + struct hpi_format hpi_format; u16 format; u16 err; u32 h_control; diff --git a/sound/pci/asihpi/hpios.c b/sound/pci/asihpi/hpios.c index 5ef4fe964366..7c91330af719 100644 --- a/sound/pci/asihpi/hpios.c +++ b/sound/pci/asihpi/hpios.c @@ -49,7 +49,7 @@ u16 hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size, /*?? any benefit in using managed dmam_alloc_coherent? */ p_mem_area->vaddr = dma_alloc_coherent(&pdev->dev, size, &p_mem_area->dma_handle, - GFP_DMA32 | GFP_KERNEL); + GFP_KERNEL); if (p_mem_area->vaddr) { HPI_DEBUG_LOG(DEBUG, "allocated %d bytes, dma 0x%x vma %p\n", diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index a1e4944dcfe8..1a41f8c80243 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -903,15 +903,15 @@ static int snd_atiixp_playback_prepare(struct snd_pcm_substream *substream) case 8: data |= ATI_REG_OUT_DMA_SLOT_BIT(10) | ATI_REG_OUT_DMA_SLOT_BIT(11); - /* fallthru */ + /* fall through */ case 6: data |= ATI_REG_OUT_DMA_SLOT_BIT(7) | ATI_REG_OUT_DMA_SLOT_BIT(8); - /* fallthru */ + /* fall through */ case 4: data |= ATI_REG_OUT_DMA_SLOT_BIT(6) | ATI_REG_OUT_DMA_SLOT_BIT(9); - /* fallthru */ + /* fall through */ default: data |= ATI_REG_OUT_DMA_SLOT_BIT(3) | ATI_REG_OUT_DMA_SLOT_BIT(4); diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index 2e5b460a847c..96ece1a71cf1 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -1115,6 +1115,7 @@ vortex_adbdma_setbuffers(vortex_t * vortex, int adbdma, hwwrite(vortex->mmio, VORTEX_ADBDMA_BUFBASE + (adbdma << 4) + 0xc, snd_pcm_sgbuf_get_addr(dma->substream, psize * 3)); + /* fall through */ /* 3 pages */ case 3: dma->cfg0 |= 0x12000000; @@ -1122,12 +1123,14 @@ vortex_adbdma_setbuffers(vortex_t * vortex, int adbdma, hwwrite(vortex->mmio, VORTEX_ADBDMA_BUFBASE + (adbdma << 4) + 0x8, snd_pcm_sgbuf_get_addr(dma->substream, psize * 2)); + /* fall through */ /* 2 pages */ case 2: dma->cfg0 |= 0x88000000 | 0x44000000 | 0x10000000 | (psize - 1); hwwrite(vortex->mmio, VORTEX_ADBDMA_BUFBASE + (adbdma << 4) + 0x4, snd_pcm_sgbuf_get_addr(dma->substream, psize)); + /* fall through */ /* 1 page */ case 1: dma->cfg0 |= 0x80000000 | 0x40000000 | ((psize - 1) << 0xc); @@ -1390,17 +1393,20 @@ vortex_wtdma_setbuffers(vortex_t * vortex, int wtdma, dma->cfg1 |= 0x88000000 | 0x44000000 | 0x30000000 | (psize-1); hwwrite(vortex->mmio, VORTEX_WTDMA_BUFBASE + (wtdma << 4) + 0xc, snd_pcm_sgbuf_get_addr(dma->substream, psize * 3)); + /* fall through */ /* 3 pages */ case 3: dma->cfg0 |= 0x12000000; dma->cfg1 |= 0x80000000 | 0x40000000 | ((psize-1) << 0xc); hwwrite(vortex->mmio, VORTEX_WTDMA_BUFBASE + (wtdma << 4) + 0x8, snd_pcm_sgbuf_get_addr(dma->substream, psize * 2)); + /* fall through */ /* 2 pages */ case 2: dma->cfg0 |= 0x88000000 | 0x44000000 | 0x10000000 | (psize-1); hwwrite(vortex->mmio, VORTEX_WTDMA_BUFBASE + (wtdma << 4) + 0x4, snd_pcm_sgbuf_get_addr(dma->substream, psize)); + /* fall through */ /* 1 page */ case 1: dma->cfg0 |= 0x80000000 | 0x40000000 | ((psize-1) << 0xc); diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 146e1a3498c7..750eec437a79 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -1443,7 +1443,8 @@ static const struct snd_pcm_hardware snd_cs46xx_playback = .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER /*|*/ - /*SNDRV_PCM_INFO_RESUME*/), + /*SNDRV_PCM_INFO_RESUME*/ | + SNDRV_PCM_INFO_SYNC_APPLPTR), .formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE), @@ -1465,7 +1466,8 @@ static const struct snd_pcm_hardware snd_cs46xx_capture = .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER /*|*/ - /*SNDRV_PCM_INFO_RESUME*/), + /*SNDRV_PCM_INFO_RESUME*/ | + SNDRV_PCM_INFO_SYNC_APPLPTR), .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000, .rate_min = 5500, diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index 598d140bb7cb..5fc497c6d738 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -903,6 +903,9 @@ int cs46xx_dsp_proc_done (struct snd_cs46xx *chip) struct dsp_spos_instance * ins = chip->dsp_spos_instance; int i; + if (!ins) + return 0; + snd_info_free_entry(ins->proc_sym_info_entry); ins->proc_sym_info_entry = NULL; diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 6ebe817801ea..1f25e6d029d8 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -36,6 +36,7 @@ #include <linux/init.h> #include <linux/mutex.h> #include <linux/moduleparam.h> +#include <linux/nospec.h> #include <sound/core.h> #include <sound/tlv.h> @@ -1026,6 +1027,8 @@ static int snd_emu10k1_ipcm_poke(struct snd_emu10k1 *emu, if (ipcm->substream >= EMU10K1_FX8010_PCM_COUNT) return -EINVAL; + ipcm->substream = array_index_nospec(ipcm->substream, + EMU10K1_FX8010_PCM_COUNT); if (ipcm->channels > 32) return -EINVAL; pcm = &emu->fx8010.pcm[ipcm->substream]; @@ -1072,6 +1075,8 @@ static int snd_emu10k1_ipcm_peek(struct snd_emu10k1 *emu, if (ipcm->substream >= EMU10K1_FX8010_PCM_COUNT) return -EINVAL; + ipcm->substream = array_index_nospec(ipcm->substream, + EMU10K1_FX8010_PCM_COUNT); pcm = &emu->fx8010.pcm[ipcm->substream]; mutex_lock(&emu->fx8010.lock); spin_lock_irq(&emu->reg_lock); diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 9f2b6097f486..30b3472d0b75 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -1753,7 +1753,8 @@ static const struct snd_pcm_hardware snd_emu10k1_fx8010_playback = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_RESUME | - /* SNDRV_PCM_INFO_MMAP_VALID | */ SNDRV_PCM_INFO_PAUSE), + /* SNDRV_PCM_INFO_MMAP_VALID | */ SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_SYNC_APPLPTR), .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_48000, .rate_min = 48000, diff --git a/sound/pci/hda/dell_wmi_helper.c b/sound/pci/hda/dell_wmi_helper.c deleted file mode 100644 index bbd6c87a4ed6..000000000000 --- a/sound/pci/hda/dell_wmi_helper.c +++ /dev/null @@ -1,48 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0 -/* Helper functions for Dell Mic Mute LED control; - * to be included from codec driver - */ - -#if IS_ENABLED(CONFIG_DELL_LAPTOP) -#include <linux/dell-led.h> - -static int (*dell_micmute_led_set_func)(int); - -static void dell_micmute_update(struct hda_codec *codec) -{ - struct hda_gen_spec *spec = codec->spec; - - dell_micmute_led_set_func(spec->micmute_led.led_value); -} - -static void alc_fixup_dell_wmi(struct hda_codec *codec, - const struct hda_fixup *fix, int action) -{ - bool removefunc = false; - - if (action == HDA_FIXUP_ACT_PROBE) { - if (!dell_micmute_led_set_func) - dell_micmute_led_set_func = symbol_request(dell_micmute_led_set); - if (!dell_micmute_led_set_func) { - codec_warn(codec, "Failed to find dell wmi symbol dell_micmute_led_set\n"); - return; - } - - removefunc = (dell_micmute_led_set_func(false) < 0) || - (snd_hda_gen_add_micmute_led(codec, - dell_micmute_update) < 0); - } - - if (dell_micmute_led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) { - symbol_put(dell_micmute_led_set); - dell_micmute_led_set_func = NULL; - } -} - -#else /* CONFIG_DELL_LAPTOP */ -static void alc_fixup_dell_wmi(struct hda_codec *codec, - const struct hda_fixup *fix, int action) -{ -} - -#endif /* CONFIG_DELL_LAPTOP */ diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index b9a6b66aeb0e..df0d636145f8 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -13,7 +13,7 @@ #include <linux/export.h> #include <linux/sort.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index d1a6a9c1329a..f1457c6b3969 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -9,7 +9,7 @@ #ifndef __SOUND_HDA_BEEP_H #define __SOUND_HDA_BEEP_H -#include "hda_codec.h" +#include <sound/hda_codec.h> #define HDA_BEEP_MODE_OFF 0 #define HDA_BEEP_MODE_ON 1 diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index d361bb77ca00..1ec706ced75c 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -11,7 +11,7 @@ #include <linux/pm.h> #include <linux/pm_runtime.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" /* @@ -81,6 +81,12 @@ static int hda_codec_driver_probe(struct device *dev) hda_codec_patch_t patch; int err; + if (codec->bus->core.ext_ops) { + if (WARN_ON(!codec->bus->core.ext_ops->hdev_attach)) + return -EINVAL; + return codec->bus->core.ext_ops->hdev_attach(&codec->core); + } + if (WARN_ON(!codec->preset)) return -EINVAL; @@ -109,7 +115,8 @@ static int hda_codec_driver_probe(struct device *dev) err = snd_hda_codec_build_controls(codec); if (err < 0) goto error_module; - if (codec->card->registered) { + /* only register after the bus probe finished; otherwise it's racy */ + if (!codec->bus->bus_probing && codec->card->registered) { err = snd_card_register(codec->card); if (err < 0) goto error_module; @@ -134,6 +141,12 @@ static int hda_codec_driver_remove(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); + if (codec->bus->core.ext_ops) { + if (WARN_ON(!codec->bus->core.ext_ops->hdev_detach)) + return -EINVAL; + return codec->bus->core.ext_ops->hdev_detach(&codec->core); + } + if (codec->patch_ops.free) codec->patch_ops.free(codec); snd_hda_codec_cleanup_for_unbind(codec); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 26d348b47867..9f8d59e7e89f 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -27,7 +27,7 @@ #include <linux/pm.h> #include <linux/pm_runtime.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include <sound/asoundef.h> #include <sound/tlv.h> #include <sound/initval.h> @@ -36,6 +36,7 @@ #include "hda_beep.h" #include "hda_jack.h" #include <sound/hda_hwdep.h> +#include <sound/hda_component.h> #define codec_in_pm(codec) snd_hdac_is_in_pm(&codec->core) #define hda_codec_is_power_on(codec) snd_hdac_is_power_on(&codec->core) @@ -799,6 +800,13 @@ void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec) static unsigned int hda_set_power_state(struct hda_codec *codec, unsigned int power_state); +/* enable/disable display power per codec */ +static void codec_display_power(struct hda_codec *codec, bool enable) +{ + if (codec->display_power_control) + snd_hdac_display_power(&codec->bus->core, codec->addr, enable); +} + /* also called from hda_bind.c */ void snd_hda_codec_register(struct hda_codec *codec) { @@ -806,7 +814,7 @@ void snd_hda_codec_register(struct hda_codec *codec) return; if (device_is_registered(hda_codec_dev(codec))) { snd_hda_register_beep_device(codec); - snd_hdac_link_power(&codec->core, true); + codec_display_power(codec, true); pm_runtime_enable(hda_codec_dev(codec)); /* it was powered up in snd_hda_codec_new(), now all done */ snd_hda_power_down(codec); @@ -834,7 +842,7 @@ static int snd_hda_codec_dev_free(struct snd_device *device) codec->in_freeing = 1; snd_hdac_device_unregister(&codec->core); - snd_hdac_link_power(&codec->core, false); + codec_display_power(codec, false); put_device(hda_codec_dev(codec)); return 0; } @@ -2926,7 +2934,7 @@ static int hda_codec_runtime_suspend(struct device *dev) (codec_has_clkstop(codec) && codec_has_epss(codec) && (state & AC_PWRST_CLK_STOP_OK))) snd_hdac_codec_link_down(&codec->core); - snd_hdac_link_power(&codec->core, false); + codec_display_power(codec, false); return 0; } @@ -2934,7 +2942,7 @@ static int hda_codec_runtime_resume(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); - snd_hdac_link_power(&codec->core, true); + codec_display_power(codec, true); snd_hdac_codec_link_up(&codec->core); hda_call_codec_resume(codec); pm_runtime_mark_last_busy(dev); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h deleted file mode 100644 index 0d98bb9068b1..000000000000 --- a/sound/pci/hda/hda_codec.h +++ /dev/null @@ -1,534 +0,0 @@ -/* - * Universal Interface for Intel High Definition Audio Codec - * - * Copyright (c) 2004 Takashi Iwai <tiwai@suse.de> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the Free - * Software Foundation; either version 2 of the License, or (at your option) - * any later version. - * - * This program is distributed in the hope that it will be useful, but WITHOUT - * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or - * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for - * more details. - * - * You should have received a copy of the GNU General Public License along with - * this program; if not, write to the Free Software Foundation, Inc., 59 - * Temple Place - Suite 330, Boston, MA 02111-1307, USA. - */ - -#ifndef __SOUND_HDA_CODEC_H -#define __SOUND_HDA_CODEC_H - -#include <linux/kref.h> -#include <linux/mod_devicetable.h> -#include <sound/info.h> -#include <sound/control.h> -#include <sound/pcm.h> -#include <sound/hwdep.h> -#include <sound/hdaudio.h> -#include <sound/hda_verbs.h> -#include <sound/hda_regmap.h> - -/* - * Structures - */ - -struct hda_bus; -struct hda_beep; -struct hda_codec; -struct hda_pcm; -struct hda_pcm_stream; - -/* - * codec bus - * - * each controller needs to creata a hda_bus to assign the accessor. - * A hda_bus contains several codecs in the list codec_list. - */ -struct hda_bus { - struct hdac_bus core; - - struct snd_card *card; - - struct pci_dev *pci; - const char *modelname; - - struct mutex prepare_mutex; - - /* assigned PCMs */ - DECLARE_BITMAP(pcm_dev_bits, SNDRV_PCM_DEVICES); - - /* misc op flags */ - unsigned int needs_damn_long_delay :1; - unsigned int allow_bus_reset:1; /* allow bus reset at fatal error */ - /* status for codec/controller */ - unsigned int shutdown :1; /* being unloaded */ - unsigned int response_reset:1; /* controller was reset */ - unsigned int in_reset:1; /* during reset operation */ - unsigned int no_response_fallback:1; /* don't fallback at RIRB error */ - - int primary_dig_out_type; /* primary digital out PCM type */ - unsigned int mixer_assigned; /* codec addr for mixer name */ -}; - -/* from hdac_bus to hda_bus */ -#define to_hda_bus(bus) container_of(bus, struct hda_bus, core) - -/* - * codec preset - * - * Known codecs have the patch to build and set up the controls/PCMs - * better than the generic parser. - */ -typedef int (*hda_codec_patch_t)(struct hda_codec *); - -#define HDA_CODEC_ID_SKIP_PROBE 0x00000001 -#define HDA_CODEC_ID_GENERIC_HDMI 0x00000101 -#define HDA_CODEC_ID_GENERIC 0x00000201 - -#define HDA_CODEC_REV_ENTRY(_vid, _rev, _name, _patch) \ - { .vendor_id = (_vid), .rev_id = (_rev), .name = (_name), \ - .api_version = HDA_DEV_LEGACY, \ - .driver_data = (unsigned long)(_patch) } -#define HDA_CODEC_ENTRY(_vid, _name, _patch) \ - HDA_CODEC_REV_ENTRY(_vid, 0, _name, _patch) - -struct hda_codec_driver { - struct hdac_driver core; - const struct hda_device_id *id; -}; - -int __hda_codec_driver_register(struct hda_codec_driver *drv, const char *name, - struct module *owner); -#define hda_codec_driver_register(drv) \ - __hda_codec_driver_register(drv, KBUILD_MODNAME, THIS_MODULE) -void hda_codec_driver_unregister(struct hda_codec_driver *drv); -#define module_hda_codec_driver(drv) \ - module_driver(drv, hda_codec_driver_register, \ - hda_codec_driver_unregister) - -/* ops set by the preset patch */ -struct hda_codec_ops { - int (*build_controls)(struct hda_codec *codec); - int (*build_pcms)(struct hda_codec *codec); - int (*init)(struct hda_codec *codec); - void (*free)(struct hda_codec *codec); - void (*unsol_event)(struct hda_codec *codec, unsigned int res); - void (*set_power_state)(struct hda_codec *codec, hda_nid_t fg, - unsigned int power_state); -#ifdef CONFIG_PM - int (*suspend)(struct hda_codec *codec); - int (*resume)(struct hda_codec *codec); - int (*check_power_status)(struct hda_codec *codec, hda_nid_t nid); -#endif - void (*reboot_notify)(struct hda_codec *codec); - void (*stream_pm)(struct hda_codec *codec, hda_nid_t nid, bool on); -}; - -/* PCM callbacks */ -struct hda_pcm_ops { - int (*open)(struct hda_pcm_stream *info, struct hda_codec *codec, - struct snd_pcm_substream *substream); - int (*close)(struct hda_pcm_stream *info, struct hda_codec *codec, - struct snd_pcm_substream *substream); - int (*prepare)(struct hda_pcm_stream *info, struct hda_codec *codec, - unsigned int stream_tag, unsigned int format, - struct snd_pcm_substream *substream); - int (*cleanup)(struct hda_pcm_stream *info, struct hda_codec *codec, - struct snd_pcm_substream *substream); - unsigned int (*get_delay)(struct hda_pcm_stream *info, - struct hda_codec *codec, - struct snd_pcm_substream *substream); -}; - -/* PCM information for each substream */ -struct hda_pcm_stream { - unsigned int substreams; /* number of substreams, 0 = not exist*/ - unsigned int channels_min; /* min. number of channels */ - unsigned int channels_max; /* max. number of channels */ - hda_nid_t nid; /* default NID to query rates/formats/bps, or set up */ - u32 rates; /* supported rates */ - u64 formats; /* supported formats (SNDRV_PCM_FMTBIT_) */ - unsigned int maxbps; /* supported max. bit per sample */ - const struct snd_pcm_chmap_elem *chmap; /* chmap to override */ - struct hda_pcm_ops ops; -}; - -/* PCM types */ -enum { - HDA_PCM_TYPE_AUDIO, - HDA_PCM_TYPE_SPDIF, - HDA_PCM_TYPE_HDMI, - HDA_PCM_TYPE_MODEM, - HDA_PCM_NTYPES -}; - -#define SNDRV_PCM_INVALID_DEVICE (-1) -/* for PCM creation */ -struct hda_pcm { - char *name; - struct hda_pcm_stream stream[2]; - unsigned int pcm_type; /* HDA_PCM_TYPE_XXX */ - int device; /* device number to assign */ - struct snd_pcm *pcm; /* assigned PCM instance */ - bool own_chmap; /* codec driver provides own channel maps */ - /* private: */ - struct hda_codec *codec; - struct kref kref; - struct list_head list; -}; - -/* codec information */ -struct hda_codec { - struct hdac_device core; - struct hda_bus *bus; - struct snd_card *card; - unsigned int addr; /* codec addr*/ - u32 probe_id; /* overridden id for probing */ - - /* detected preset */ - const struct hda_device_id *preset; - const char *modelname; /* model name for preset */ - - /* set by patch */ - struct hda_codec_ops patch_ops; - - /* PCM to create, set by patch_ops.build_pcms callback */ - struct list_head pcm_list_head; - - /* codec specific info */ - void *spec; - - /* beep device */ - struct hda_beep *beep; - unsigned int beep_mode; - - /* widget capabilities cache */ - u32 *wcaps; - - struct snd_array mixers; /* list of assigned mixer elements */ - struct snd_array nids; /* list of mapped mixer elements */ - - struct list_head conn_list; /* linked-list of connection-list */ - - struct mutex spdif_mutex; - struct mutex control_mutex; - struct snd_array spdif_out; - unsigned int spdif_in_enable; /* SPDIF input enable? */ - const hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */ - struct snd_array init_pins; /* initial (BIOS) pin configurations */ - struct snd_array driver_pins; /* pin configs set by codec parser */ - struct snd_array cvt_setups; /* audio convert setups */ - - struct mutex user_mutex; -#ifdef CONFIG_SND_HDA_RECONFIG - struct snd_array init_verbs; /* additional init verbs */ - struct snd_array hints; /* additional hints */ - struct snd_array user_pins; /* default pin configs to override */ -#endif - -#ifdef CONFIG_SND_HDA_HWDEP - struct snd_hwdep *hwdep; /* assigned hwdep device */ -#endif - - /* misc flags */ - unsigned int in_freeing:1; /* being released */ - unsigned int registered:1; /* codec was registered */ - unsigned int spdif_status_reset :1; /* needs to toggle SPDIF for each - * status change - * (e.g. Realtek codecs) - */ - unsigned int pin_amp_workaround:1; /* pin out-amp takes index - * (e.g. Conexant codecs) - */ - unsigned int single_adc_amp:1; /* adc in-amp takes no index - * (e.g. CX20549 codec) - */ - unsigned int no_sticky_stream:1; /* no sticky-PCM stream assignment */ - unsigned int pins_shutup:1; /* pins are shut up */ - unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */ - unsigned int no_jack_detect:1; /* Machine has no jack-detection */ - unsigned int inv_eapd:1; /* broken h/w: inverted EAPD control */ - unsigned int inv_jack_detect:1; /* broken h/w: inverted detection bit */ - unsigned int pcm_format_first:1; /* PCM format must be set first */ - unsigned int cached_write:1; /* write only to caches */ - unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */ - unsigned int dump_coef:1; /* dump processing coefs in codec proc file */ - unsigned int power_save_node:1; /* advanced PM for each widget */ - unsigned int auto_runtime_pm:1; /* enable automatic codec runtime pm */ - unsigned int force_pin_prefix:1; /* Add location prefix */ - unsigned int link_down_at_suspend:1; /* link down at runtime suspend */ -#ifdef CONFIG_PM - unsigned long power_on_acct; - unsigned long power_off_acct; - unsigned long power_jiffies; -#endif - - /* filter the requested power state per nid */ - unsigned int (*power_filter)(struct hda_codec *codec, hda_nid_t nid, - unsigned int power_state); - - /* codec-specific additional proc output */ - void (*proc_widget_hook)(struct snd_info_buffer *buffer, - struct hda_codec *codec, hda_nid_t nid); - - /* jack detection */ - struct snd_array jacktbl; - unsigned long jackpoll_interval; /* In jiffies. Zero means no poll, rely on unsol events */ - struct delayed_work jackpoll_work; - - /* jack detection */ - struct snd_array jacks; - - int depop_delay; /* depop delay in ms, -1 for default delay time */ - - /* fix-up list */ - int fixup_id; - const struct hda_fixup *fixup_list; - const char *fixup_name; - - /* additional init verbs */ - struct snd_array verbs; -}; - -#define dev_to_hda_codec(_dev) container_of(_dev, struct hda_codec, core.dev) -#define hda_codec_dev(_dev) (&(_dev)->core.dev) - -#define list_for_each_codec(c, bus) \ - list_for_each_entry(c, &(bus)->core.codec_list, core.list) -#define list_for_each_codec_safe(c, n, bus) \ - list_for_each_entry_safe(c, n, &(bus)->core.codec_list, core.list) - -/* snd_hda_codec_read/write optional flags */ -#define HDA_RW_NO_RESPONSE_FALLBACK (1 << 0) - -/* - * constructors - */ -int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, - unsigned int codec_addr, struct hda_codec **codecp); -int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card, - unsigned int codec_addr, struct hda_codec *codec); -int snd_hda_codec_configure(struct hda_codec *codec); -int snd_hda_codec_update_widgets(struct hda_codec *codec); - -/* - * low level functions - */ -static inline unsigned int -snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, - int flags, - unsigned int verb, unsigned int parm) -{ - return snd_hdac_codec_read(&codec->core, nid, flags, verb, parm); -} - -static inline int -snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags, - unsigned int verb, unsigned int parm) -{ - return snd_hdac_codec_write(&codec->core, nid, flags, verb, parm); -} - -#define snd_hda_param_read(codec, nid, param) \ - snd_hdac_read_parm(&(codec)->core, nid, param) -#define snd_hda_get_sub_nodes(codec, nid, start_nid) \ - snd_hdac_get_sub_nodes(&(codec)->core, nid, start_nid) -int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, - hda_nid_t *conn_list, int max_conns); -static inline int -snd_hda_get_num_conns(struct hda_codec *codec, hda_nid_t nid) -{ - return snd_hda_get_connections(codec, nid, NULL, 0); -} - -#define snd_hda_get_raw_connections(codec, nid, list, max_conns) \ - snd_hdac_get_connections(&(codec)->core, nid, list, max_conns) -#define snd_hda_get_num_raw_conns(codec, nid) \ - snd_hdac_get_connections(&(codec)->core, nid, NULL, 0); - -int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid, - const hda_nid_t **listp); -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); -unsigned int snd_hda_get_num_devices(struct hda_codec *codec, hda_nid_t nid); -int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid, - u8 *dev_list, int max_devices); -int snd_hda_get_dev_select(struct hda_codec *codec, hda_nid_t nid); -int snd_hda_set_dev_select(struct hda_codec *codec, hda_nid_t nid, int dev_id); - -struct hda_verb { - hda_nid_t nid; - u32 verb; - u32 param; -}; - -void snd_hda_sequence_write(struct hda_codec *codec, - const struct hda_verb *seq); - -/* unsolicited event */ -static inline void -snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex) -{ - snd_hdac_bus_queue_event(&bus->core, res, res_ex); -} - -/* cached write */ -static inline int -snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, - int flags, unsigned int verb, unsigned int parm) -{ - return snd_hdac_regmap_write(&codec->core, nid, verb, parm); -} - -/* the struct for codec->pin_configs */ -struct hda_pincfg { - hda_nid_t nid; - unsigned char ctrl; /* original pin control value */ - unsigned char target; /* target pin control value */ - unsigned int cfg; /* default configuration */ -}; - -unsigned int snd_hda_codec_get_pincfg(struct hda_codec *codec, hda_nid_t nid); -int snd_hda_codec_set_pincfg(struct hda_codec *codec, hda_nid_t nid, - unsigned int cfg); -int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list, - hda_nid_t nid, unsigned int cfg); /* for hwdep */ -void snd_hda_shutup_pins(struct hda_codec *codec); - -/* SPDIF controls */ -struct hda_spdif_out { - hda_nid_t nid; /* Converter nid values relate to */ - unsigned int status; /* IEC958 status bits */ - unsigned short ctls; /* SPDIF control bits */ -}; -struct hda_spdif_out *snd_hda_spdif_out_of_nid(struct hda_codec *codec, - hda_nid_t nid); -void snd_hda_spdif_ctls_unassign(struct hda_codec *codec, int idx); -void snd_hda_spdif_ctls_assign(struct hda_codec *codec, int idx, hda_nid_t nid); - -/* - * Mixer - */ -int snd_hda_codec_build_controls(struct hda_codec *codec); - -/* - * PCM - */ -int snd_hda_codec_parse_pcms(struct hda_codec *codec); -int snd_hda_codec_build_pcms(struct hda_codec *codec); - -__printf(2, 3) -struct hda_pcm *snd_hda_codec_pcm_new(struct hda_codec *codec, - const char *fmt, ...); - -static inline void snd_hda_codec_pcm_get(struct hda_pcm *pcm) -{ - kref_get(&pcm->kref); -} -void snd_hda_codec_pcm_put(struct hda_pcm *pcm); - -int snd_hda_codec_prepare(struct hda_codec *codec, - struct hda_pcm_stream *hinfo, - unsigned int stream, - unsigned int format, - struct snd_pcm_substream *substream); -void snd_hda_codec_cleanup(struct hda_codec *codec, - struct hda_pcm_stream *hinfo, - struct snd_pcm_substream *substream); - -void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, - u32 stream_tag, - int channel_id, int format); -void __snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid, - int do_now); -#define snd_hda_codec_cleanup_stream(codec, nid) \ - __snd_hda_codec_cleanup_stream(codec, nid, 0) - -#define snd_hda_query_supported_pcm(codec, nid, ratesp, fmtsp, bpsp) \ - snd_hdac_query_supported_pcm(&(codec)->core, nid, ratesp, fmtsp, bpsp) -#define snd_hda_is_supported_format(codec, nid, fmt) \ - snd_hdac_is_supported_format(&(codec)->core, nid, fmt) - -extern const struct snd_pcm_chmap_elem snd_pcm_2_1_chmaps[]; - -int snd_hda_attach_pcm_stream(struct hda_bus *_bus, struct hda_codec *codec, - struct hda_pcm *cpcm); - -/* - * Misc - */ -void snd_hda_get_codec_name(struct hda_codec *codec, char *name, int namelen); -void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg, - unsigned int power_state); - -int snd_hda_lock_devices(struct hda_bus *bus); -void snd_hda_unlock_devices(struct hda_bus *bus); -void snd_hda_bus_reset(struct hda_bus *bus); -void snd_hda_bus_reset_codecs(struct hda_bus *bus); - -int snd_hda_codec_set_name(struct hda_codec *codec, const char *name); - -/* - * power management - */ -extern const struct dev_pm_ops hda_codec_driver_pm; - -static inline -int hda_call_check_power_status(struct hda_codec *codec, hda_nid_t nid) -{ -#ifdef CONFIG_PM - if (codec->patch_ops.check_power_status) - return codec->patch_ops.check_power_status(codec, nid); -#endif - return 0; -} - -/* - * power saving - */ -#define snd_hda_power_up(codec) snd_hdac_power_up(&(codec)->core) -#define snd_hda_power_up_pm(codec) snd_hdac_power_up_pm(&(codec)->core) -#define snd_hda_power_down(codec) snd_hdac_power_down(&(codec)->core) -#define snd_hda_power_down_pm(codec) snd_hdac_power_down_pm(&(codec)->core) -#ifdef CONFIG_PM -void snd_hda_set_power_save(struct hda_bus *bus, int delay); -void snd_hda_update_power_acct(struct hda_codec *codec); -#else -static inline void snd_hda_set_power_save(struct hda_bus *bus, int delay) {} -#endif - -#ifdef CONFIG_SND_HDA_PATCH_LOADER -/* - * patch firmware - */ -int snd_hda_load_patch(struct hda_bus *bus, size_t size, const void *buf); -#endif - -#ifdef CONFIG_SND_HDA_DSP_LOADER -int snd_hda_codec_load_dsp_prepare(struct hda_codec *codec, unsigned int format, - unsigned int size, - struct snd_dma_buffer *bufp); -void snd_hda_codec_load_dsp_trigger(struct hda_codec *codec, bool start); -void snd_hda_codec_load_dsp_cleanup(struct hda_codec *codec, - struct snd_dma_buffer *dmab); -#else -static inline int -snd_hda_codec_load_dsp_prepare(struct hda_codec *codec, unsigned int format, - unsigned int size, - struct snd_dma_buffer *bufp) -{ - return -ENOSYS; -} -static inline void -snd_hda_codec_load_dsp_trigger(struct hda_codec *codec, bool start) {} -static inline void -snd_hda_codec_load_dsp_cleanup(struct hda_codec *codec, - struct snd_dma_buffer *dmab) {} -#endif - -#endif /* __SOUND_HDA_CODEC_H */ diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index a12e594d4e3b..532e081f8b8a 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -130,8 +130,9 @@ static int azx_pcm_hw_params(struct snd_pcm_substream *substream, azx_dev->core.bufsize = 0; azx_dev->core.period_bytes = 0; azx_dev->core.format_val = 0; - ret = chip->ops->substream_alloc_pages(chip, substream, - params_buffer_bytes(hw_params)); + ret = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + unlock: dsp_unlock(azx_dev); return ret; @@ -141,7 +142,6 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream) { struct azx_pcm *apcm = snd_pcm_substream_chip(substream); struct azx_dev *azx_dev = get_azx_dev(substream); - struct azx *chip = apcm->chip; struct hda_pcm_stream *hinfo = to_hda_pcm_stream(substream); int err; @@ -152,7 +152,7 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream) snd_hda_codec_cleanup(apcm->codec, hinfo, substream); - err = chip->ops->substream_free_pages(chip, substream); + err = snd_pcm_lib_free_pages(substream); azx_stream(azx_dev)->prepared = 0; dsp_unlock(azx_dev); return err; @@ -732,6 +732,7 @@ int snd_hda_attach_pcm_stream(struct hda_bus *_bus, struct hda_codec *codec, int pcm_dev = cpcm->device; unsigned int size; int s, err; + int type = SNDRV_DMA_TYPE_DEV_SG; list_for_each_entry(apcm, &chip->pcm_list, list) { if (apcm->pcm->device == pcm_dev) { @@ -770,7 +771,9 @@ int snd_hda_attach_pcm_stream(struct hda_bus *_bus, struct hda_codec *codec, size = CONFIG_SND_HDA_PREALLOC_SIZE * 1024; if (size > MAX_PREALLOC_SIZE) size = MAX_PREALLOC_SIZE; - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, + if (chip->uc_buffer) + type = SNDRV_DMA_TYPE_DEV_UC_SG; + snd_pcm_lib_preallocate_pages_for_all(pcm, type, chip->card->dev, size, MAX_PREALLOC_SIZE); return 0; @@ -986,20 +989,9 @@ static int azx_get_response(struct hdac_bus *bus, unsigned int addr, return azx_rirb_get_response(bus, addr, res); } -static int azx_link_power(struct hdac_bus *bus, bool enable) -{ - struct azx *chip = bus_to_azx(bus); - - if (chip->ops->link_power) - return chip->ops->link_power(chip, enable); - else - return -EINVAL; -} - static const struct hdac_bus_ops bus_core_ops = { .command = azx_send_cmd, .get_response = azx_get_response, - .link_power = azx_link_power, }; #ifdef CONFIG_SND_HDA_DSP_LOADER @@ -1220,27 +1212,6 @@ void snd_hda_bus_reset(struct hda_bus *bus) bus->in_reset = 0; } -static int get_jackpoll_interval(struct azx *chip) -{ - int i; - unsigned int j; - - if (!chip->jackpoll_ms) - return 0; - - i = chip->jackpoll_ms[chip->dev_index]; - if (i == 0) - return 0; - if (i < 50 || i > 60000) - j = 0; - else - j = msecs_to_jiffies(i); - if (j == 0) - dev_warn(chip->card->dev, - "jackpoll_ms value out of range: %d\n", i); - return j; -} - /* HD-audio bus initialization */ int azx_bus_init(struct azx *chip, const char *model, const struct hdac_io_ops *io_ops) @@ -1323,7 +1294,7 @@ int azx_probe_codecs(struct azx *chip, unsigned int max_slots) err = snd_hda_codec_new(&chip->bus, chip->card, c, &codec); if (err < 0) continue; - codec->jackpoll_interval = get_jackpoll_interval(chip); + codec->jackpoll_interval = chip->jackpoll_interval; codec->beep_mode = chip->beep_mode; codecs++; } diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index 53c3cd28bc99..7185ed574b41 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -20,7 +20,7 @@ #include <sound/core.h> #include <sound/pcm.h> #include <sound/initval.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include <sound/hda_register.h> #define AZX_MAX_CODECS HDA_MAX_CODECS @@ -50,11 +50,7 @@ /* 24 unused */ #define AZX_DCAPS_COUNT_LPIB_DELAY (1 << 25) /* Take LPIB as delay */ #define AZX_DCAPS_PM_RUNTIME (1 << 26) /* runtime PM support */ -#ifdef CONFIG_SND_HDA_I915 -#define AZX_DCAPS_I915_POWERWELL (1 << 27) /* HSW i915 powerwell support */ -#else -#define AZX_DCAPS_I915_POWERWELL 0 /* NOP */ -#endif +/* 27 unused */ #define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */ #define AZX_DCAPS_NO_MSI64 (1 << 29) /* Stick to 32-bit MSIs */ #define AZX_DCAPS_SEPARATE_STREAM_TAG (1 << 30) /* capture and playback use separate stream tag */ @@ -76,7 +72,6 @@ struct azx_dev { * when link position is not greater than FIFO size */ unsigned int insufficient:1; - unsigned int wc_marked:1; }; #define azx_stream(dev) (&(dev)->core) @@ -88,11 +83,6 @@ struct azx; struct hda_controller_ops { /* Disable msi if supported, PCI only */ int (*disable_msi_reset_irq)(struct azx *); - int (*substream_alloc_pages)(struct azx *chip, - struct snd_pcm_substream *substream, - size_t size); - int (*substream_free_pages)(struct azx *chip, - struct snd_pcm_substream *substream); void (*pcm_mmap_prepare)(struct snd_pcm_substream *substream, struct vm_area_struct *area); /* Check if current position is acceptable */ @@ -127,7 +117,7 @@ struct azx { int capture_streams; int capture_index_offset; int num_streams; - const int *jackpoll_ms; /* per-card jack poll interval */ + int jackpoll_interval; /* jack poll interval in jiffies */ /* Register interaction. */ const struct hda_controller_ops *ops; @@ -176,11 +166,10 @@ struct azx { #define azx_bus(chip) (&(chip)->bus.core) #define bus_to_azx(_bus) container_of(_bus, struct azx, bus.core) -#ifdef CONFIG_X86 -#define azx_snoop(chip) ((chip)->snoop) -#else -#define azx_snoop(chip) true -#endif +static inline bool azx_snoop(struct azx *chip) +{ + return !IS_ENABLED(CONFIG_X86) || chip->snoop; +} /* * macros for easy use diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index ba7fe9b6655c..806b12ed44a2 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -27,7 +27,7 @@ #include <sound/core.h> #include <asm/unaligned.h> #include <sound/hda_chmap.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" enum eld_versions { diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 579984ecdec3..4095cd7c56c6 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -29,10 +29,11 @@ #include <linux/string.h> #include <linux/bitops.h> #include <linux/module.h> +#include <linux/leds.h> #include <sound/core.h> #include <sound/jack.h> #include <sound/tlv.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" @@ -4035,6 +4036,36 @@ int snd_hda_gen_add_micmute_led(struct hda_codec *codec, } EXPORT_SYMBOL_GPL(snd_hda_gen_add_micmute_led); +#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO) +static void call_ledtrig_micmute(struct hda_codec *codec) +{ + struct hda_gen_spec *spec = codec->spec; + + ledtrig_audio_set(LED_AUDIO_MICMUTE, + spec->micmute_led.led_value ? LED_ON : LED_OFF); +} +#endif + +/** + * snd_hda_gen_fixup_micmute_led - A fixup for mic-mute LED trigger + * + * Pass this function to the quirk entry if another driver supports the + * audio mic-mute LED trigger. Then this will bind the mixer capture switch + * change with the LED. + * + * Note that this fixup has to be called after other fixup that sets + * cap_sync_hook. Otherwise the chaining wouldn't work. + */ +void snd_hda_gen_fixup_micmute_led(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ +#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO) + if (action == HDA_FIXUP_ACT_PROBE) + snd_hda_gen_add_micmute_led(codec, call_ledtrig_micmute); +#endif +} +EXPORT_SYMBOL_GPL(snd_hda_gen_fixup_micmute_led); + /* * parse digital I/Os and set up NIDs in BIOS auto-parse mode */ diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 10123664fa61..78d77042b05a 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -357,5 +357,7 @@ int snd_hda_gen_fix_pin_power(struct hda_codec *codec, hda_nid_t pin); int snd_hda_gen_add_micmute_led(struct hda_codec *codec, void (*hook)(struct hda_codec *)); +void snd_hda_gen_fixup_micmute_led(struct hda_codec *codec, + const struct hda_fixup *fix, int action); #endif /* __SOUND_HDA_GENERIC_H */ diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index cc009a4a3d1d..268bba6ec985 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -23,7 +23,7 @@ #include <linux/compat.h> #include <linux/nospec.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include <sound/hda_hwdep.h> #include <sound/minors.h> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 625cb6c7b7d6..e5c49003e75f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -63,7 +63,7 @@ #include <linux/vgaarb.h> #include <linux/vga_switcheroo.h> #include <linux/firmware.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_controller.h" #include "hda_intel.h" @@ -310,31 +310,28 @@ enum { #define AZX_DCAPS_INTEL_HASWELL \ (/*AZX_DCAPS_ALIGN_BUFSIZE |*/ AZX_DCAPS_COUNT_LPIB_DELAY |\ AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_COMPONENT |\ - AZX_DCAPS_I915_POWERWELL | AZX_DCAPS_SNOOP_TYPE(SCH)) + AZX_DCAPS_SNOOP_TYPE(SCH)) /* Broadwell HDMI can't use position buffer reliably, force to use LPIB */ #define AZX_DCAPS_INTEL_BROADWELL \ (/*AZX_DCAPS_ALIGN_BUFSIZE |*/ AZX_DCAPS_POSFIX_LPIB |\ AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_COMPONENT |\ - AZX_DCAPS_I915_POWERWELL | AZX_DCAPS_SNOOP_TYPE(SCH)) + AZX_DCAPS_SNOOP_TYPE(SCH)) #define AZX_DCAPS_INTEL_BAYTRAIL \ - (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_I915_COMPONENT |\ - AZX_DCAPS_I915_POWERWELL) + (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_I915_COMPONENT) #define AZX_DCAPS_INTEL_BRASWELL \ (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\ - AZX_DCAPS_I915_COMPONENT | AZX_DCAPS_I915_POWERWELL) + AZX_DCAPS_I915_COMPONENT) #define AZX_DCAPS_INTEL_SKYLAKE \ (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\ - AZX_DCAPS_SEPARATE_STREAM_TAG | AZX_DCAPS_I915_COMPONENT |\ - AZX_DCAPS_I915_POWERWELL) + AZX_DCAPS_SEPARATE_STREAM_TAG | AZX_DCAPS_I915_COMPONENT) #define AZX_DCAPS_INTEL_BROXTON \ (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\ - AZX_DCAPS_SEPARATE_STREAM_TAG | AZX_DCAPS_I915_COMPONENT |\ - AZX_DCAPS_I915_POWERWELL) + AZX_DCAPS_SEPARATE_STREAM_TAG | AZX_DCAPS_I915_COMPONENT) /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ @@ -399,61 +396,6 @@ static char *driver_short_names[] = { [AZX_DRIVER_GENERIC] = "HD-Audio Generic", }; -#ifdef CONFIG_X86 -static void __mark_pages_wc(struct azx *chip, struct snd_dma_buffer *dmab, bool on) -{ - int pages; - - if (azx_snoop(chip)) - return; - if (!dmab || !dmab->area || !dmab->bytes) - return; - -#ifdef CONFIG_SND_DMA_SGBUF - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_SG) { - struct snd_sg_buf *sgbuf = dmab->private_data; - if (!chip->uc_buffer) - return; /* deal with only CORB/RIRB buffers */ - if (on) - set_pages_array_wc(sgbuf->page_table, sgbuf->pages); - else - set_pages_array_wb(sgbuf->page_table, sgbuf->pages); - return; - } -#endif - - pages = (dmab->bytes + PAGE_SIZE - 1) >> PAGE_SHIFT; - if (on) - set_memory_wc((unsigned long)dmab->area, pages); - else - set_memory_wb((unsigned long)dmab->area, pages); -} - -static inline void mark_pages_wc(struct azx *chip, struct snd_dma_buffer *buf, - bool on) -{ - __mark_pages_wc(chip, buf, on); -} -static inline void mark_runtime_wc(struct azx *chip, struct azx_dev *azx_dev, - struct snd_pcm_substream *substream, bool on) -{ - if (azx_dev->wc_marked != on) { - __mark_pages_wc(chip, snd_pcm_get_dma_buf(substream), on); - azx_dev->wc_marked = on; - } -} -#else -/* NOP for other archs */ -static inline void mark_pages_wc(struct azx *chip, struct snd_dma_buffer *buf, - bool on) -{ -} -static inline void mark_runtime_wc(struct azx *chip, struct azx_dev *azx_dev, - struct snd_pcm_substream *substream, bool on) -{ -} -#endif - static int azx_acquire_irq(struct azx *chip, int do_disconnect); static void set_default_power_save(struct azx *chip); @@ -646,8 +588,7 @@ static void hda_intel_init_chip(struct azx *chip, bool full_reset) struct pci_dev *pci = chip->pci; u32 val; - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) - snd_hdac_set_codec_wakeup(bus, true); + snd_hdac_set_codec_wakeup(bus, true); if (chip->driver_type == AZX_DRIVER_SKL) { pci_read_config_dword(pci, INTEL_HDA_CGCTL, &val); val = val & ~INTEL_HDA_CGCTL_MISCBDCGE; @@ -659,8 +600,8 @@ static void hda_intel_init_chip(struct azx *chip, bool full_reset) val = val | INTEL_HDA_CGCTL_MISCBDCGE; pci_write_config_dword(pci, INTEL_HDA_CGCTL, val); } - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) - snd_hdac_set_codec_wakeup(bus, false); + + snd_hdac_set_codec_wakeup(bus, false); /* reduce dma latency to avoid noise */ if (IS_BXT(pci)) @@ -722,13 +663,8 @@ static int azx_position_check(struct azx *chip, struct azx_dev *azx_dev) return 0; } -/* Enable/disable i915 display power for the link */ -static int azx_intel_link_power(struct azx *chip, bool enable) -{ - struct hdac_bus *bus = azx_bus(chip); - - return snd_hdac_display_power(bus, enable); -} +#define display_power(chip, enable) \ + snd_hdac_display_power(azx_bus(chip), HDA_CODEC_IDX_CONTROLLER, enable) /* * Check whether the current DMA position is acceptable for updating @@ -985,35 +921,75 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) mutex_unlock(&card_list_lock); return 0; } -#else -#define azx_add_card_list(chip) /* NOP */ -#define azx_del_card_list(chip) /* NOP */ -#endif /* CONFIG_PM */ -#ifdef CONFIG_PM_SLEEP /* * power management */ -static int azx_suspend(struct device *dev) +static bool azx_is_pm_ready(struct snd_card *card) { - struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; struct hda_intel *hda; - struct hdac_bus *bus; if (!card) - return 0; - + return false; chip = card->private_data; hda = container_of(chip, struct hda_intel, chip); if (chip->disabled || hda->init_failed || !chip->running) + return false; + return true; +} + +static void __azx_runtime_suspend(struct azx *chip) +{ + azx_stop_chip(chip); + azx_enter_link_reset(chip); + azx_clear_irq_pending(chip); + display_power(chip, false); +} + +static void __azx_runtime_resume(struct azx *chip) +{ + struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + struct hdac_bus *bus = azx_bus(chip); + struct hda_codec *codec; + int status; + + display_power(chip, true); + if (hda->need_i915_power) + snd_hdac_i915_set_bclk(bus); + + /* Read STATESTS before controller reset */ + status = azx_readw(chip, STATESTS); + + azx_init_pci(chip); + hda_intel_init_chip(chip, true); + + if (status) { + list_for_each_codec(codec, &chip->bus) + if (status & (1 << codec->addr)) + schedule_delayed_work(&codec->jackpoll_work, + codec->jackpoll_interval); + } + + /* power down again for link-controlled chips */ + if (!hda->need_i915_power) + display_power(chip, false); +} + +#ifdef CONFIG_PM_SLEEP +static int azx_suspend(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip; + struct hdac_bus *bus; + + if (!azx_is_pm_ready(card)) return 0; + chip = card->private_data; bus = azx_bus(chip); snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - azx_clear_irq_pending(chip); - azx_stop_chip(chip); - azx_enter_link_reset(chip); + __azx_runtime_suspend(chip); if (bus->irq >= 0) { free_irq(bus->irq, chip); bus->irq = -1; @@ -1021,9 +997,6 @@ static int azx_suspend(struct device *dev) if (chip->msi) pci_disable_msi(chip->pci); - if ((chip->driver_caps & AZX_DCAPS_I915_POWERWELL) - && hda->need_i915_power) - snd_hdac_display_power(bus, false); trace_azx_suspend(chip); return 0; @@ -1031,41 +1004,19 @@ static int azx_suspend(struct device *dev) static int azx_resume(struct device *dev) { - struct pci_dev *pci = to_pci_dev(dev); struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; - struct hda_intel *hda; - struct hdac_bus *bus; - if (!card) + if (!azx_is_pm_ready(card)) return 0; chip = card->private_data; - hda = container_of(chip, struct hda_intel, chip); - bus = azx_bus(chip); - if (chip->disabled || hda->init_failed || !chip->running) - return 0; - - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { - snd_hdac_display_power(bus, true); - if (hda->need_i915_power) - snd_hdac_i915_set_bclk(bus); - } - if (chip->msi) - if (pci_enable_msi(pci) < 0) + if (pci_enable_msi(chip->pci) < 0) chip->msi = 0; if (azx_acquire_irq(chip, 1) < 0) return -EIO; - azx_init_pci(chip); - - hda_intel_init_chip(chip, true); - - /* power down again for link-controlled chips */ - if ((chip->driver_caps & AZX_DCAPS_I915_POWERWELL) && - !hda->need_i915_power) - snd_hdac_display_power(bus, false); - + __azx_runtime_resume(chip); snd_power_change_state(card, SNDRV_CTL_POWER_D0); trace_azx_resume(chip); @@ -1100,21 +1051,14 @@ static int azx_thaw_noirq(struct device *dev) } #endif /* CONFIG_PM_SLEEP */ -#ifdef CONFIG_PM static int azx_runtime_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; - struct hda_intel *hda; - if (!card) + if (!azx_is_pm_ready(card)) return 0; - chip = card->private_data; - hda = container_of(chip, struct hda_intel, chip); - if (chip->disabled || hda->init_failed) - return 0; - if (!azx_has_pm_runtime(chip)) return 0; @@ -1122,13 +1066,7 @@ static int azx_runtime_suspend(struct device *dev) azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | STATESTS_INT_MASK); - azx_stop_chip(chip); - azx_enter_link_reset(chip); - azx_clear_irq_pending(chip); - if ((chip->driver_caps & AZX_DCAPS_I915_POWERWELL) - && hda->need_i915_power) - snd_hdac_display_power(azx_bus(chip), false); - + __azx_runtime_suspend(chip); trace_azx_runtime_suspend(chip); return 0; } @@ -1137,51 +1075,18 @@ static int azx_runtime_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; - struct hda_intel *hda; - struct hdac_bus *bus; - struct hda_codec *codec; - int status; - if (!card) + if (!azx_is_pm_ready(card)) return 0; - chip = card->private_data; - hda = container_of(chip, struct hda_intel, chip); - bus = azx_bus(chip); - if (chip->disabled || hda->init_failed) - return 0; - if (!azx_has_pm_runtime(chip)) return 0; - - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { - snd_hdac_display_power(bus, true); - if (hda->need_i915_power) - snd_hdac_i915_set_bclk(bus); - } - - /* Read STATESTS before controller reset */ - status = azx_readw(chip, STATESTS); - - azx_init_pci(chip); - hda_intel_init_chip(chip, true); - - if (status) { - list_for_each_codec(codec, &chip->bus) - if (status & (1 << codec->addr)) - schedule_delayed_work(&codec->jackpoll_work, - codec->jackpoll_interval); - } + __azx_runtime_resume(chip); /* disable controller Wake Up event*/ azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & ~STATESTS_INT_MASK); - /* power down again for link-controlled chips */ - if ((chip->driver_caps & AZX_DCAPS_I915_POWERWELL) && - !hda->need_i915_power) - snd_hdac_display_power(bus, false); - trace_azx_runtime_resume(chip); return 0; } @@ -1222,6 +1127,8 @@ static const struct dev_pm_ops azx_pm = { #define AZX_PM_OPS &azx_pm #else +#define azx_add_card_list(chip) /* NOP */ +#define azx_del_card_list(chip) /* NOP */ #define AZX_PM_OPS NULL #endif /* CONFIG_PM */ @@ -1429,11 +1336,8 @@ static int azx_free(struct azx *chip) #ifdef CONFIG_SND_HDA_PATCH_LOADER release_firmware(chip->fw); #endif + display_power(chip, false); - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { - if (hda->need_i915_power) - snd_hdac_display_power(bus, false); - } if (chip->driver_caps & AZX_DCAPS_I915_COMPONENT) snd_hdac_i915_exit(bus); kfree(hda); @@ -1772,7 +1676,8 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, chip->driver_type = driver_caps & 0xff; check_msi(chip); chip->dev_index = dev; - chip->jackpoll_ms = jackpoll_ms; + if (jackpoll_ms[dev] >= 50 && jackpoll_ms[dev] <= 60000) + chip->jackpoll_interval = msecs_to_jiffies(jackpoll_ms[dev]); INIT_LIST_HEAD(&chip->pcm_list); INIT_WORK(&hda->irq_pending_work, azx_irq_pending_work); INIT_LIST_HEAD(&hda->list); @@ -1989,8 +1894,7 @@ static int azx_first_init(struct azx *chip) /* initialize chip */ azx_init_pci(chip); - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) - snd_hdac_i915_set_bclk(bus); + snd_hdac_i915_set_bclk(bus); hda_intel_init_chip(chip, (probe_only[dev] & 2) == 0); @@ -2095,48 +1999,17 @@ static int dma_alloc_pages(struct hdac_bus *bus, struct snd_dma_buffer *buf) { struct azx *chip = bus_to_azx(bus); - int err; - err = snd_dma_alloc_pages(type, - bus->dev, - size, buf); - if (err < 0) - return err; - mark_pages_wc(chip, buf, true); - return 0; + if (!azx_snoop(chip) && type == SNDRV_DMA_TYPE_DEV) + type = SNDRV_DMA_TYPE_DEV_UC; + return snd_dma_alloc_pages(type, bus->dev, size, buf); } static void dma_free_pages(struct hdac_bus *bus, struct snd_dma_buffer *buf) { - struct azx *chip = bus_to_azx(bus); - - mark_pages_wc(chip, buf, false); snd_dma_free_pages(buf); } -static int substream_alloc_pages(struct azx *chip, - struct snd_pcm_substream *substream, - size_t size) -{ - struct azx_dev *azx_dev = get_azx_dev(substream); - int ret; - - mark_runtime_wc(chip, azx_dev, substream, false); - ret = snd_pcm_lib_malloc_pages(substream, size); - if (ret < 0) - return ret; - mark_runtime_wc(chip, azx_dev, substream, true); - return 0; -} - -static int substream_free_pages(struct azx *chip, - struct snd_pcm_substream *substream) -{ - struct azx_dev *azx_dev = get_azx_dev(substream); - mark_runtime_wc(chip, azx_dev, substream, false); - return snd_pcm_lib_free_pages(substream); -} - static void pcm_mmap_prepare(struct snd_pcm_substream *substream, struct vm_area_struct *area) { @@ -2161,11 +2034,8 @@ static const struct hdac_io_ops pci_hda_io_ops = { static const struct hda_controller_ops pci_hda_ops = { .disable_msi_reset_irq = disable_msi_reset_irq, - .substream_alloc_pages = substream_alloc_pages, - .substream_free_pages = substream_free_pages, .pcm_mmap_prepare = pcm_mmap_prepare, .position_check = azx_position_check, - .link_power = azx_intel_link_power, }; static int azx_probe(struct pci_dev *pci, @@ -2256,6 +2126,8 @@ static struct snd_pci_quirk power_save_blacklist[] = { /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1849, 0xc892, "Asrock B85M-ITX", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ + SND_PCI_QUIRK(0x1849, 0x0397, "Asrock N68C-S UCC", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1849, 0x7662, "Asrock H81M-HDS", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1043, 0x8733, "Asus Prime X370-Pro", 0), @@ -2313,6 +2185,7 @@ static int azx_probe_continue(struct azx *chip) int dev = chip->dev_index; int err; + to_hda_bus(bus)->bus_probing = 1; hda->probe_continued = 1; /* bind with i915 if needed */ @@ -2330,10 +2203,13 @@ static int azx_probe_continue(struct azx *chip) goto out_free; } else { /* don't bother any longer */ - chip->driver_caps &= - ~(AZX_DCAPS_I915_COMPONENT | AZX_DCAPS_I915_POWERWELL); + chip->driver_caps &= ~AZX_DCAPS_I915_COMPONENT; } } + + /* HSW/BDW controllers need this power */ + if (CONTROLLER_IN_GPU(pci)) + hda->need_i915_power = 1; } /* Request display power well for the HDA controller or codec. For @@ -2341,18 +2217,7 @@ static int azx_probe_continue(struct azx *chip) * this power. For other platforms, like Baytrail/Braswell, only the * display codec needs the power and it can be released after probe. */ - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { - /* HSW/BDW controllers need this power */ - if (CONTROLLER_IN_GPU(pci)) - hda->need_i915_power = 1; - - err = snd_hdac_display_power(bus, true); - if (err < 0) { - dev_err(chip->card->dev, - "Cannot turn on display power on i915\n"); - goto i915_power_fail; - } - } + display_power(chip, true); err = azx_first_init(chip); if (err < 0) @@ -2400,14 +2265,12 @@ static int azx_probe_continue(struct azx *chip) pm_runtime_put_autosuspend(&pci->dev); out_free: - if ((chip->driver_caps & AZX_DCAPS_I915_POWERWELL) - && !hda->need_i915_power) - snd_hdac_display_power(bus, false); - -i915_power_fail: + if (err < 0 || !hda->need_i915_power) + display_power(chip, false); if (err < 0) hda->init_failed = 1; complete_all(&hda->probe_wait); + to_hda_bus(bus)->bus_probing = 0; return err; } @@ -2583,6 +2446,10 @@ static const struct pci_device_id azx_ids[] = { /* AMD Hudson */ { PCI_DEVICE(0x1022, 0x780d), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB }, + /* AMD Stoney */ + { PCI_DEVICE(0x1022, 0x157a), + .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB | + AZX_DCAPS_PM_RUNTIME }, /* AMD Raven */ { PCI_DEVICE(0x1022, 0x15e3), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB | diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index a33234e04d4f..74b46952fc98 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -15,7 +15,7 @@ #include <sound/core.h> #include <sound/control.h> #include <sound/jack.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" @@ -339,9 +339,15 @@ void snd_hda_jack_report_sync(struct hda_codec *codec) if (jack->nid) { if (!jack->jack || jack->block_report) continue; - state = get_jack_plug_state(jack->pin_sense); - snd_jack_report(jack->jack, - state ? jack->type : 0); + state = jack->button_state; + if (get_jack_plug_state(jack->pin_sense)) + state |= jack->type; + snd_jack_report(jack->jack, state); + if (jack->button_state) { + snd_jack_report(jack->jack, + state & ~jack->button_state); + jack->button_state = 0; /* button released */ + } } } EXPORT_SYMBOL_GPL(snd_hda_jack_report_sync); @@ -379,15 +385,19 @@ static void hda_free_jack_priv(struct snd_jack *jack) * @nid: pin NID to assign * @name: string name for the jack * @phantom_jack: flag to deal as a phantom jack + * @type: jack type bits to be reported, 0 for guessing from pincfg + * @keymap: optional jack / key mapping * * This assigns a jack-detection kctl to the given pin. The kcontrol * will have the given name and index. */ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, - const char *name, bool phantom_jack) + const char *name, bool phantom_jack, + int type, const struct hda_jack_keymap *keymap) { struct hda_jack_tbl *jack; - int err, state, type; + const struct hda_jack_keymap *map; + int err, state, buttons; jack = snd_hda_jack_tbl_new(codec, nid); if (!jack) @@ -395,16 +405,30 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, if (jack->jack) return 0; /* already created */ - type = get_input_jack_type(codec, nid); - err = snd_jack_new(codec->card, name, type, + if (!type) + type = get_input_jack_type(codec, nid); + + buttons = 0; + if (keymap) { + for (map = keymap; map->type; map++) + buttons |= map->type; + } + + err = snd_jack_new(codec->card, name, type | buttons, &jack->jack, true, phantom_jack); if (err < 0) return err; jack->phantom_jack = !!phantom_jack; jack->type = type; + jack->button_state = 0; jack->jack->private_data = jack; jack->jack->private_free = hda_free_jack_priv; + if (keymap) { + for (map = keymap; map->type; map++) + snd_jack_set_key(jack->jack, map->type, map->key); + } + state = snd_hda_jack_detect(codec, nid); snd_jack_report(jack->jack, state ? jack->type : 0); @@ -437,7 +461,7 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, if (phantom_jack) /* Example final name: "Internal Mic Phantom Jack" */ strncat(name, " Phantom", sizeof(name) - strlen(name) - 1); - err = snd_hda_jack_add_kctl(codec, nid, name, phantom_jack); + err = snd_hda_jack_add_kctl(codec, nid, name, phantom_jack, 0, NULL); if (err < 0) return err; @@ -508,19 +532,25 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, } EXPORT_SYMBOL_GPL(snd_hda_jack_add_kctls); -static void call_jack_callback(struct hda_codec *codec, +static void call_jack_callback(struct hda_codec *codec, unsigned int res, struct hda_jack_tbl *jack) { struct hda_jack_callback *cb; - for (cb = jack->callback; cb; cb = cb->next) + for (cb = jack->callback; cb; cb = cb->next) { + cb->jack = jack; + cb->unsol_res = res; cb->func(codec, cb); + } if (jack->gated_jack) { struct hda_jack_tbl *gated = snd_hda_jack_tbl_get(codec, jack->gated_jack); if (gated) { - for (cb = gated->callback; cb; cb = cb->next) + for (cb = gated->callback; cb; cb = cb->next) { + cb->jack = gated; + cb->unsol_res = res; cb->func(codec, cb); + } } } } @@ -540,7 +570,7 @@ void snd_hda_jack_unsol_event(struct hda_codec *codec, unsigned int res) return; event->jack_dirty = 1; - call_jack_callback(codec, event); + call_jack_callback(codec, res, event); snd_hda_jack_report_sync(codec); } EXPORT_SYMBOL_GPL(snd_hda_jack_unsol_event); @@ -566,7 +596,7 @@ void snd_hda_jack_poll_all(struct hda_codec *codec) if (old_sense == get_jack_plug_state(jack->pin_sense)) continue; changes = 1; - call_jack_callback(codec, jack); + call_jack_callback(codec, 0, jack); } if (changes) snd_hda_jack_report_sync(codec); diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index e9814c0168ea..1d713201c160 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -13,6 +13,7 @@ #define __SOUND_HDA_JACK_H #include <linux/err.h> +#include <sound/jack.h> struct auto_pin_cfg; struct hda_jack_tbl; @@ -24,6 +25,8 @@ struct hda_jack_callback { hda_nid_t nid; hda_jack_callback_fn func; unsigned int private_data; /* arbitrary data */ + unsigned int unsol_res; /* unsolicited event bits */ + struct hda_jack_tbl *jack; /* associated jack entry */ struct hda_jack_callback *next; }; @@ -40,9 +43,15 @@ struct hda_jack_tbl { hda_nid_t gating_jack; /* valid when gating jack plugged */ hda_nid_t gated_jack; /* gated is dependent on this jack */ int type; + int button_state; struct snd_jack *jack; }; +struct hda_jack_keymap { + enum snd_jack_types type; + int key; +}; + struct hda_jack_tbl * snd_hda_jack_tbl_get(struct hda_codec *codec, hda_nid_t nid); struct hda_jack_tbl * @@ -82,7 +91,8 @@ static inline bool snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid) bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid); int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, - const char *name, bool phantom_jack); + const char *name, bool phantom_jack, + int type, const struct hda_jack_keymap *keymap); int snd_hda_jack_add_kctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg); diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index c6b778b2580c..a65740419650 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -25,7 +25,7 @@ #include <linux/slab.h> #include <sound/core.h> #include <linux/module.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" static int dump_coef = -1; diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index 6ec79c58d48d..c154b19a0c45 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -14,7 +14,7 @@ #include <linux/string.h> #include <linux/export.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include <sound/hda_hwdep.h> #include <sound/minors.h> diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 0621920f7617..97a176d817a0 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -31,11 +31,12 @@ #include <linux/of_device.h> #include <linux/slab.h> #include <linux/time.h> +#include <linux/string.h> #include <sound/core.h> #include <sound/initval.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_controller.h" /* Defines for Nvidia Tegra HDA support */ @@ -99,19 +100,6 @@ static void dma_free_pages(struct hdac_bus *bus, struct snd_dma_buffer *buf) snd_dma_free_pages(buf); } -static int substream_alloc_pages(struct azx *chip, - struct snd_pcm_substream *substream, - size_t size) -{ - return snd_pcm_lib_malloc_pages(substream, size); -} - -static int substream_free_pages(struct azx *chip, - struct snd_pcm_substream *substream) -{ - return snd_pcm_lib_free_pages(substream); -} - /* * Register access ops. Tegra HDA register access is DWORD only. */ @@ -180,10 +168,7 @@ static const struct hdac_io_ops hda_tegra_io_ops = { .dma_free_pages = dma_free_pages, }; -static const struct hda_controller_ops hda_tegra_ops = { - .substream_alloc_pages = substream_alloc_pages, - .substream_free_pages = substream_free_pages, -}; +static const struct hda_controller_ops hda_tegra_ops; /* nothing special */ static void hda_tegra_init(struct hda_tegra *hda) { @@ -249,10 +234,12 @@ static int hda_tegra_suspend(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); + struct hdac_bus *bus = azx_bus(chip); snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); azx_stop_chip(chip); + synchronize_irq(bus->irq); azx_enter_link_reset(chip); hda_tegra_disable_clocks(hda); @@ -360,6 +347,8 @@ static int hda_tegra_first_init(struct azx *chip, struct platform_device *pdev) int err; unsigned short gcap; int irq_id = platform_get_irq(pdev, 0); + const char *sname; + struct device_node *root; err = hda_tegra_init_chip(chip, pdev); if (err) @@ -417,8 +406,23 @@ static int hda_tegra_first_init(struct azx *chip, struct platform_device *pdev) return -ENODEV; } + /* driver name */ strcpy(card->driver, "tegra-hda"); - strcpy(card->shortname, "tegra-hda"); + + root = of_find_node_by_path("/"); + sname = of_get_property(root, "compatible", NULL); + of_node_put(root); + if (!sname) { + dev_err(card->dev, + "failed to get compatible property from root node\n"); + return -ENODEV; + } + /* shortname for card */ + if (strlen(sname) > sizeof(card->shortname)) + dev_info(card->dev, "truncating shortname for card\n"); + strncpy(card->shortname, sname, sizeof(card->shortname)); + + /* longname for card */ snprintf(card->longname, sizeof(card->longname), "%s at 0x%lx irq %i", card->shortname, bus->addr, bus->irq); @@ -529,7 +533,7 @@ static void hda_tegra_probe_work(struct work_struct *work) goto out_free; /* create codec instances */ - err = azx_probe_codecs(chip, 0); + err = azx_probe_codecs(chip, 8); if (err < 0) goto out_free; diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index fd476fb40e1b..ebfd0be885b3 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -24,7 +24,7 @@ #include <linux/module.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_beep.h" diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index c2d9ee9cfdc0..21d0f0610913 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -22,7 +22,7 @@ #include <linux/slab.h> #include <linux/module.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index dffd60cebc31..29882bda7632 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -31,8 +31,9 @@ #include <linux/types.h> #include <linux/io.h> #include <linux/pci.h> +#include <asm/io.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" @@ -81,12 +82,12 @@ #define SCP_GET 1 #define EFX_FILE "ctefx.bin" -#define SBZ_EFX_FILE "ctefx-sbz.bin" +#define DESKTOP_EFX_FILE "ctefx-desktop.bin" #define R3DI_EFX_FILE "ctefx-r3di.bin" #ifdef CONFIG_SND_HDA_CODEC_CA0132_DSP MODULE_FIRMWARE(EFX_FILE); -MODULE_FIRMWARE(SBZ_EFX_FILE); +MODULE_FIRMWARE(DESKTOP_EFX_FILE); MODULE_FIRMWARE(R3DI_EFX_FILE); #endif @@ -152,7 +153,10 @@ enum { XBASS_XOVER, EQ_PRESET_ENUM, SMART_VOLUME_ENUM, - MIC_BOOST_ENUM + MIC_BOOST_ENUM, + AE5_HEADPHONE_GAIN_ENUM, + AE5_SOUND_FILTER_ENUM, + ZXR_HEADPHONE_GAIN #define EFFECTS_COUNT (EFFECT_END_NID - EFFECT_START_NID) }; @@ -666,6 +670,65 @@ static const struct ct_dsp_volume_ctl ca0132_alt_vol_ctls[] = { } }; +/* Values for ca0113_mmio_command_set for selecting output. */ +#define AE5_CA0113_OUT_SET_COMMANDS 6 +struct ae5_ca0113_output_set { + unsigned int group[AE5_CA0113_OUT_SET_COMMANDS]; + unsigned int target[AE5_CA0113_OUT_SET_COMMANDS]; + unsigned int vals[AE5_CA0113_OUT_SET_COMMANDS]; +}; + +static const struct ae5_ca0113_output_set ae5_ca0113_output_presets[] = { + { .group = { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 }, + .target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 }, + .vals = { 0x00, 0x00, 0x40, 0x00, 0x00, 0x3f } + }, + { .group = { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 }, + .target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 }, + .vals = { 0x3f, 0x3f, 0x00, 0x00, 0x00, 0x00 } + }, + { .group = { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 }, + .target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 }, + .vals = { 0x00, 0x00, 0x40, 0x00, 0x00, 0x3f } + } +}; + +/* ae5 ca0113 command sequences to set headphone gain levels. */ +#define AE5_HEADPHONE_GAIN_PRESET_MAX_COMMANDS 4 +struct ae5_headphone_gain_set { + char *name; + unsigned int vals[AE5_HEADPHONE_GAIN_PRESET_MAX_COMMANDS]; +}; + +static const struct ae5_headphone_gain_set ae5_headphone_gain_presets[] = { + { .name = "Low (16-31", + .vals = { 0xff, 0x2c, 0xf5, 0x32 } + }, + { .name = "Medium (32-149", + .vals = { 0x38, 0xa8, 0x3e, 0x4c } + }, + { .name = "High (150-600", + .vals = { 0xff, 0xff, 0xff, 0x7f } + } +}; + +struct ae5_filter_set { + char *name; + unsigned int val; +}; + +static const struct ae5_filter_set ae5_filter_presets[] = { + { .name = "Slow Roll Off", + .val = 0xa0 + }, + { .name = "Minimum Phase", + .val = 0xc0 + }, + { .name = "Fast Roll Off", + .val = 0x80 + } +}; + enum hda_cmd_vendor_io { /* for DspIO node */ VENDOR_DSPIO_SCP_WRITE_DATA_LOW = 0x000, @@ -685,6 +748,9 @@ enum hda_cmd_vendor_io { VENDOR_CHIPIO_DATA_LOW = 0x300, VENDOR_CHIPIO_DATA_HIGH = 0x400, + VENDOR_CHIPIO_8051_WRITE_DIRECT = 0x500, + VENDOR_CHIPIO_8051_READ_DIRECT = 0xD00, + VENDOR_CHIPIO_GET_PARAMETER = 0xF00, VENDOR_CHIPIO_STATUS = 0xF01, VENDOR_CHIPIO_HIC_POST_READ = 0x702, @@ -692,6 +758,9 @@ enum hda_cmd_vendor_io { VENDOR_CHIPIO_8051_DATA_WRITE = 0x707, VENDOR_CHIPIO_8051_DATA_READ = 0xF07, + VENDOR_CHIPIO_8051_PMEM_READ = 0xF08, + VENDOR_CHIPIO_8051_IRAM_WRITE = 0x709, + VENDOR_CHIPIO_8051_IRAM_READ = 0xF09, VENDOR_CHIPIO_CT_EXTENSIONS_ENABLE = 0x70A, VENDOR_CHIPIO_CT_EXTENSIONS_GET = 0xF0A, @@ -798,6 +867,12 @@ enum control_param_id { * impedance is selected*/ CONTROL_PARAM_PORTD_160OHM_GAIN = 10, + /* + * This control param name was found in the 8051 memory, and makes + * sense given the fact the AE-5 uses it and has the ASI flag set. + */ + CONTROL_PARAM_ASI = 23, + /* Stream Control */ /* Select stream with the given ID */ @@ -955,7 +1030,11 @@ struct ca0132_spec { long eq_preset_val; unsigned int tlv[4]; struct hda_vmaster_mute_hook vmaster_mute; - + /* AE-5 Control values */ + unsigned char ae5_headphone_gain_val; + unsigned char ae5_filter_val; + /* ZxR Control Values */ + unsigned char zxr_gain_set; struct hda_codec *codec; struct delayed_work unsol_hp_work; @@ -995,10 +1074,25 @@ enum { QUIRK_ALIENWARE, QUIRK_ALIENWARE_M17XR4, QUIRK_SBZ, + QUIRK_ZXR, + QUIRK_ZXR_DBPRO, QUIRK_R3DI, QUIRK_R3D, + QUIRK_AE5, }; +#ifdef CONFIG_PCI +#define ca0132_quirk(spec) ((spec)->quirk) +#define ca0132_use_pci_mmio(spec) ((spec)->use_pci_mmio) +#define ca0132_use_alt_functions(spec) ((spec)->use_alt_functions) +#define ca0132_use_alt_controls(spec) ((spec)->use_alt_controls) +#else +#define ca0132_quirk(spec) ({ (void)(spec); QUIRK_NONE; }) +#define ca0132_use_alt_functions(spec) ({ (void)(spec); false; }) +#define ca0132_use_pci_mmio(spec) ({ (void)(spec); false; }) +#define ca0132_use_alt_controls(spec) ({ (void)(spec); false; }) +#endif + static const struct hda_pintbl alienware_pincfgs[] = { { 0x0b, 0x90170110 }, /* Builtin Speaker */ { 0x0c, 0x411111f0 }, /* N/A */ @@ -1028,6 +1122,21 @@ static const struct hda_pintbl sbz_pincfgs[] = { {} }; +/* Sound Blaster ZxR pin configs taken from Windows Driver */ +static const struct hda_pintbl zxr_pincfgs[] = { + { 0x0b, 0x01047110 }, /* Port G -- Lineout FRONT L/R */ + { 0x0c, 0x414510f0 }, /* SPDIF Out 1 - Disabled*/ + { 0x0d, 0x014510f0 }, /* Digital Out */ + { 0x0e, 0x41c520f0 }, /* SPDIF In - Disabled*/ + { 0x0f, 0x0122711f }, /* Port A -- BackPanel HP */ + { 0x10, 0x01017111 }, /* Port D -- Center/LFE */ + { 0x11, 0x01017114 }, /* Port B -- LineMicIn2 / Rear L/R */ + { 0x12, 0x01a271f0 }, /* Port C -- LineIn1 */ + { 0x13, 0x908700f0 }, /* What U Hear In*/ + { 0x18, 0x50d000f0 }, /* N/A */ + {} +}; + /* Recon3D pin configs taken from Windows Driver */ static const struct hda_pintbl r3d_pincfgs[] = { { 0x0b, 0x01014110 }, /* Port G -- Lineout FRONT L/R */ @@ -1043,6 +1152,21 @@ static const struct hda_pintbl r3d_pincfgs[] = { {} }; +/* Sound Blaster AE-5 pin configs taken from Windows Driver */ +static const struct hda_pintbl ae5_pincfgs[] = { + { 0x0b, 0x01017010 }, /* Port G -- Lineout FRONT L/R */ + { 0x0c, 0x014510f0 }, /* SPDIF Out 1 */ + { 0x0d, 0x014510f0 }, /* Digital Out */ + { 0x0e, 0x01c510f0 }, /* SPDIF In */ + { 0x0f, 0x01017114 }, /* Port A -- Rear L/R. */ + { 0x10, 0x01017012 }, /* Port D -- Center/LFE or FP Hp */ + { 0x11, 0x01a170ff }, /* Port B -- LineMicIn2 / Rear Headphone */ + { 0x12, 0x01a170f0 }, /* Port C -- LineIn1 */ + { 0x13, 0x908700f0 }, /* What U Hear In*/ + { 0x18, 0x50d000f0 }, /* N/A */ + {} +}; + /* Recon3D integrated pin configs taken from Windows Driver */ static const struct hda_pintbl r3di_pincfgs[] = { { 0x0b, 0x01014110 }, /* Port G -- Lineout FRONT L/R */ @@ -1065,10 +1189,12 @@ static const struct snd_pci_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1028, 0x0708, "Alienware 15 R2 2016", QUIRK_ALIENWARE), SND_PCI_QUIRK(0x1102, 0x0010, "Sound Blaster Z", QUIRK_SBZ), SND_PCI_QUIRK(0x1102, 0x0023, "Sound Blaster Z", QUIRK_SBZ), + SND_PCI_QUIRK(0x1102, 0x0033, "Sound Blaster ZxR", QUIRK_SBZ), SND_PCI_QUIRK(0x1458, 0xA016, "Recon3Di", QUIRK_R3DI), SND_PCI_QUIRK(0x1458, 0xA026, "Gigabyte G1.Sniper Z97", QUIRK_R3DI), SND_PCI_QUIRK(0x1458, 0xA036, "Gigabyte GA-Z170X-Gaming 7", QUIRK_R3DI), SND_PCI_QUIRK(0x1102, 0x0013, "Recon3D", QUIRK_R3D), + SND_PCI_QUIRK(0x1102, 0x0051, "Sound Blaster AE-5", QUIRK_AE5), {} }; @@ -1454,6 +1580,20 @@ static void chipio_set_conn_rate(struct hda_codec *codec, } /* + * Writes to the 8051's internal address space directly instead of indirectly, + * giving access to the special function registers located at addresses + * 0x80-0xFF. + */ +static void chipio_8051_write_direct(struct hda_codec *codec, + unsigned int addr, unsigned int data) +{ + unsigned int verb; + + verb = VENDOR_CHIPIO_8051_WRITE_DIRECT | data; + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, verb, addr); +} + +/* * Enable clocks. */ static void chipio_enable_clocks(struct hda_codec *codec) @@ -2973,7 +3113,7 @@ static void dspload_post_setup(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; codec_dbg(codec, "---- dspload_post_setup ------\n"); - if (!spec->use_alt_functions) { + if (!ca0132_use_alt_functions(spec)) { /*set DSP speaker to 2.0 configuration*/ chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x18), 0x08080080); chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x19), 0x3f800000); @@ -3088,7 +3228,9 @@ static bool dspload_wait_loaded(struct hda_codec *codec) } /* - * Setup GPIO for the other variants of Core3D. + * ca0113 related functions. The ca0113 acts as the HDA bus for the pci-e + * based cards, and has a second mmio region, region2, that's used for special + * commands. */ /* @@ -3096,8 +3238,11 @@ static bool dspload_wait_loaded(struct hda_codec *codec) * the mmio address 0x320 is used to set GPIO pins. The format for the data * The first eight bits are just the number of the pin. So far, I've only seen * this number go to 7. + * AE-5 note: The AE-5 seems to use pins 2 and 3 to somehow set the color value + * of the on-card LED. It seems to use pin 2 for data, then toggles 3 to on and + * then off to send that bit. */ -static void ca0132_mmio_gpio_set(struct hda_codec *codec, unsigned int gpio_pin, +static void ca0113_mmio_gpio_set(struct hda_codec *codec, unsigned int gpio_pin, bool enable) { struct ca0132_spec *spec = codec->spec; @@ -3110,6 +3255,89 @@ static void ca0132_mmio_gpio_set(struct hda_codec *codec, unsigned int gpio_pin, } /* + * Special pci region2 commands that are only used by the AE-5. They follow + * a set format, and require reads at certain points to seemingly 'clear' + * the response data. My first tests didn't do these reads, and would cause + * the card to get locked up until the memory was read. These commands + * seem to work with three distinct values that I've taken to calling group, + * target-id, and value. + */ +static void ca0113_mmio_command_set(struct hda_codec *codec, unsigned int group, + unsigned int target, unsigned int value) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int write_val; + + writel(0x0000007e, spec->mem_base + 0x210); + readl(spec->mem_base + 0x210); + writel(0x0000005a, spec->mem_base + 0x210); + readl(spec->mem_base + 0x210); + readl(spec->mem_base + 0x210); + + writel(0x00800005, spec->mem_base + 0x20c); + writel(group, spec->mem_base + 0x804); + + writel(0x00800005, spec->mem_base + 0x20c); + write_val = (target & 0xff); + write_val |= (value << 8); + + + writel(write_val, spec->mem_base + 0x204); + /* + * Need delay here or else it goes too fast and works inconsistently. + */ + msleep(20); + + readl(spec->mem_base + 0x860); + readl(spec->mem_base + 0x854); + readl(spec->mem_base + 0x840); + + writel(0x00800004, spec->mem_base + 0x20c); + writel(0x00000000, spec->mem_base + 0x210); + readl(spec->mem_base + 0x210); + readl(spec->mem_base + 0x210); +} + +/* + * This second type of command is used for setting the sound filter type. + */ +static void ca0113_mmio_command_set_type2(struct hda_codec *codec, + unsigned int group, unsigned int target, unsigned int value) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int write_val; + + writel(0x0000007e, spec->mem_base + 0x210); + readl(spec->mem_base + 0x210); + writel(0x0000005a, spec->mem_base + 0x210); + readl(spec->mem_base + 0x210); + readl(spec->mem_base + 0x210); + + writel(0x00800003, spec->mem_base + 0x20c); + writel(group, spec->mem_base + 0x804); + + writel(0x00800005, spec->mem_base + 0x20c); + write_val = (target & 0xff); + write_val |= (value << 8); + + + writel(write_val, spec->mem_base + 0x204); + msleep(20); + readl(spec->mem_base + 0x860); + readl(spec->mem_base + 0x854); + readl(spec->mem_base + 0x840); + + writel(0x00800004, spec->mem_base + 0x20c); + writel(0x00000000, spec->mem_base + 0x210); + readl(spec->mem_base + 0x210); + readl(spec->mem_base + 0x210); +} + +/* + * Setup GPIO for the other variants of Core3D. + */ + +/* * Sets up the GPIO pins so that they are discoverable. If this isn't done, * the card shows as having no GPIO pins. */ @@ -3117,8 +3345,9 @@ static void ca0132_gpio_init(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: + case QUIRK_AE5: snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00); snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x53); snd_hda_codec_write(codec, 0x01, 0, 0x790, 0x23); @@ -3127,6 +3356,8 @@ static void ca0132_gpio_init(struct hda_codec *codec) snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00); snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x5B); break; + default: + break; } } @@ -3136,7 +3367,7 @@ static void ca0132_gpio_setup(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DIRECTION, 0x07); @@ -3155,6 +3386,8 @@ static void ca0132_gpio_setup(struct hda_codec *codec) snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0x0C); break; + default: + break; } } @@ -3928,6 +4161,144 @@ exit: return err < 0 ? err : 0; } +static int ae5_headphone_gain_set(struct hda_codec *codec, long val); +static int zxr_headphone_gain_set(struct hda_codec *codec, long val); +static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val); + +static void ae5_mmio_select_out(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int i; + + for (i = 0; i < AE5_CA0113_OUT_SET_COMMANDS; i++) + ca0113_mmio_command_set(codec, + ae5_ca0113_output_presets[spec->cur_out_type].group[i], + ae5_ca0113_output_presets[spec->cur_out_type].target[i], + ae5_ca0113_output_presets[spec->cur_out_type].vals[i]); +} + +/* + * These are the commands needed to setup output on each of the different card + * types. + */ +static void ca0132_alt_select_out_quirk_handler(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int tmp; + + switch (spec->cur_out_type) { + case SPEAKER_OUT: + switch (ca0132_quirk(spec)) { + case QUIRK_SBZ: + ca0113_mmio_gpio_set(codec, 7, false); + ca0113_mmio_gpio_set(codec, 4, true); + ca0113_mmio_gpio_set(codec, 1, true); + chipio_set_control_param(codec, 0x0d, 0x18); + break; + case QUIRK_ZXR: + ca0113_mmio_gpio_set(codec, 2, true); + ca0113_mmio_gpio_set(codec, 3, true); + ca0113_mmio_gpio_set(codec, 5, false); + zxr_headphone_gain_set(codec, 0); + chipio_set_control_param(codec, 0x0d, 0x24); + break; + case QUIRK_R3DI: + chipio_set_control_param(codec, 0x0d, 0x24); + r3di_gpio_out_set(codec, R3DI_LINE_OUT); + break; + case QUIRK_R3D: + chipio_set_control_param(codec, 0x0d, 0x24); + ca0113_mmio_gpio_set(codec, 1, true); + break; + case QUIRK_AE5: + ae5_mmio_select_out(codec); + ae5_headphone_gain_set(codec, 2); + tmp = FLOAT_ZERO; + dspio_set_uint_param(codec, 0x96, 0x29, tmp); + dspio_set_uint_param(codec, 0x96, 0x2a, tmp); + chipio_set_control_param(codec, 0x0d, 0xa4); + chipio_write(codec, 0x18b03c, 0x00000012); + break; + default: + break; + } + break; + case HEADPHONE_OUT: + switch (ca0132_quirk(spec)) { + case QUIRK_SBZ: + ca0113_mmio_gpio_set(codec, 7, true); + ca0113_mmio_gpio_set(codec, 4, true); + ca0113_mmio_gpio_set(codec, 1, false); + chipio_set_control_param(codec, 0x0d, 0x12); + break; + case QUIRK_ZXR: + ca0113_mmio_gpio_set(codec, 2, false); + ca0113_mmio_gpio_set(codec, 3, false); + ca0113_mmio_gpio_set(codec, 5, true); + zxr_headphone_gain_set(codec, spec->zxr_gain_set); + chipio_set_control_param(codec, 0x0d, 0x21); + break; + case QUIRK_R3DI: + chipio_set_control_param(codec, 0x0d, 0x21); + r3di_gpio_out_set(codec, R3DI_HEADPHONE_OUT); + break; + case QUIRK_R3D: + chipio_set_control_param(codec, 0x0d, 0x21); + ca0113_mmio_gpio_set(codec, 0x1, false); + break; + case QUIRK_AE5: + ae5_mmio_select_out(codec); + ae5_headphone_gain_set(codec, + spec->ae5_headphone_gain_val); + tmp = FLOAT_ONE; + dspio_set_uint_param(codec, 0x96, 0x29, tmp); + dspio_set_uint_param(codec, 0x96, 0x2a, tmp); + chipio_set_control_param(codec, 0x0d, 0xa1); + chipio_write(codec, 0x18b03c, 0x00000012); + break; + default: + break; + } + break; + case SURROUND_OUT: + switch (ca0132_quirk(spec)) { + case QUIRK_SBZ: + ca0113_mmio_gpio_set(codec, 7, false); + ca0113_mmio_gpio_set(codec, 4, true); + ca0113_mmio_gpio_set(codec, 1, true); + chipio_set_control_param(codec, 0x0d, 0x18); + break; + case QUIRK_ZXR: + ca0113_mmio_gpio_set(codec, 2, true); + ca0113_mmio_gpio_set(codec, 3, true); + ca0113_mmio_gpio_set(codec, 5, false); + zxr_headphone_gain_set(codec, 0); + chipio_set_control_param(codec, 0x0d, 0x24); + break; + case QUIRK_R3DI: + chipio_set_control_param(codec, 0x0d, 0x24); + r3di_gpio_out_set(codec, R3DI_LINE_OUT); + break; + case QUIRK_R3D: + ca0113_mmio_gpio_set(codec, 1, true); + chipio_set_control_param(codec, 0x0d, 0x24); + break; + case QUIRK_AE5: + ae5_mmio_select_out(codec); + ae5_headphone_gain_set(codec, 2); + tmp = FLOAT_ZERO; + dspio_set_uint_param(codec, 0x96, 0x29, tmp); + dspio_set_uint_param(codec, 0x96, 0x2a, tmp); + chipio_set_control_param(codec, 0x0d, 0xa4); + chipio_write(codec, 0x18b03c, 0x00000012); + break; + default: + break; + } + break; + } +} + /* * This function behaves similarly to the ca0132_select_out funciton above, * except with a few differences. It adds the ability to select the current @@ -3978,26 +4349,11 @@ static int ca0132_alt_select_out(struct hda_codec *codec) if (err < 0) goto exit; + ca0132_alt_select_out_quirk_handler(codec); + switch (spec->cur_out_type) { case SPEAKER_OUT: codec_dbg(codec, "%s speaker\n", __func__); - /*speaker out config*/ - switch (spec->quirk) { - case QUIRK_SBZ: - ca0132_mmio_gpio_set(codec, 7, false); - ca0132_mmio_gpio_set(codec, 4, true); - ca0132_mmio_gpio_set(codec, 1, true); - chipio_set_control_param(codec, 0x0D, 0x18); - break; - case QUIRK_R3DI: - chipio_set_control_param(codec, 0x0D, 0x24); - r3di_gpio_out_set(codec, R3DI_LINE_OUT); - break; - case QUIRK_R3D: - chipio_set_control_param(codec, 0x0D, 0x24); - ca0132_mmio_gpio_set(codec, 1, true); - break; - } /* disable headphone node */ pin_ctl = snd_hda_codec_read(codec, spec->out_pins[1], 0, @@ -4021,23 +4377,6 @@ static int ca0132_alt_select_out(struct hda_codec *codec) break; case HEADPHONE_OUT: codec_dbg(codec, "%s hp\n", __func__); - /* Headphone out config*/ - switch (spec->quirk) { - case QUIRK_SBZ: - ca0132_mmio_gpio_set(codec, 7, true); - ca0132_mmio_gpio_set(codec, 4, true); - ca0132_mmio_gpio_set(codec, 1, false); - chipio_set_control_param(codec, 0x0D, 0x12); - break; - case QUIRK_R3DI: - chipio_set_control_param(codec, 0x0D, 0x21); - r3di_gpio_out_set(codec, R3DI_HEADPHONE_OUT); - break; - case QUIRK_R3D: - chipio_set_control_param(codec, 0x0D, 0x21); - ca0132_mmio_gpio_set(codec, 0x1, false); - break; - } snd_hda_codec_write(codec, spec->out_pins[0], 0, AC_VERB_SET_EAPD_BTLENABLE, 0x00); @@ -4067,23 +4406,7 @@ static int ca0132_alt_select_out(struct hda_codec *codec) break; case SURROUND_OUT: codec_dbg(codec, "%s surround\n", __func__); - /* Surround out config*/ - switch (spec->quirk) { - case QUIRK_SBZ: - ca0132_mmio_gpio_set(codec, 7, false); - ca0132_mmio_gpio_set(codec, 4, true); - ca0132_mmio_gpio_set(codec, 1, true); - chipio_set_control_param(codec, 0x0D, 0x18); - break; - case QUIRK_R3DI: - chipio_set_control_param(codec, 0x0D, 0x24); - r3di_gpio_out_set(codec, R3DI_LINE_OUT); - break; - case QUIRK_R3D: - ca0132_mmio_gpio_set(codec, 1, true); - chipio_set_control_param(codec, 0x0D, 0x24); - break; - } + /* enable line out node */ pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); @@ -4108,14 +4431,21 @@ static int ca0132_alt_select_out(struct hda_codec *codec) snd_hda_set_pin_ctl(codec, spec->out_pins[3], pin_ctl | PIN_OUT); - if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]) - dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ONE); - else - dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_EIGHT); + dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_EIGHT); break; } + /* + * Surround always sets it's scp command to req 0x04 to FLOAT_EIGHT. + * With this set though, X_BASS cannot be enabled. So, if we have OutFX + * enabled, we need to make sure X_BASS is off, otherwise everything + * sounds all muffled. Running ca0132_effects_set with X_BASS as the + * effect should sort this out. + */ + if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]) + ca0132_effects_set(codec, X_BASS, + spec->effects_switch[X_BASS - EFFECT_START_NID]); - /* run through the output dsp commands for line-out */ + /* run through the output dsp commands for the selected output. */ for (i = 0; i < alt_out_presets[spec->cur_out_type].commands; i++) { err = dspio_set_uint_param(codec, alt_out_presets[spec->cur_out_type].mids[i], @@ -4138,7 +4468,7 @@ static void ca0132_unsol_hp_delayed(struct work_struct *work) to_delayed_work(work), struct ca0132_spec, unsol_hp_work); struct hda_jack_tbl *jack; - if (spec->use_alt_functions) + if (ca0132_use_alt_functions(spec)) ca0132_alt_select_out(spec->codec); else ca0132_select_out(spec->codec); @@ -4152,7 +4482,6 @@ static void ca0132_unsol_hp_delayed(struct work_struct *work) static void ca0132_set_dmic(struct hda_codec *codec, int enable); static int ca0132_mic_boost_set(struct hda_codec *codec, long val); -static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val); static void resume_mic1(struct hda_codec *codec, unsigned int oldval); static int stop_mic1(struct hda_codec *codec); static int ca0132_cvoice_switch_set(struct hda_codec *codec); @@ -4223,14 +4552,14 @@ static int ca0132_alt_set_vipsource(struct hda_codec *codec, int val) chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); - if (spec->quirk == QUIRK_R3DI) + if (ca0132_quirk(spec) == QUIRK_R3DI) chipio_set_conn_rate(codec, 0x0F, SR_96_000); if (spec->in_enum_val == REAR_LINE_IN) tmp = FLOAT_ZERO; else { - if (spec->quirk == QUIRK_SBZ) + if (ca0132_quirk(spec) == QUIRK_SBZ) tmp = FLOAT_THREE; else tmp = FLOAT_ONE; @@ -4242,7 +4571,7 @@ static int ca0132_alt_set_vipsource(struct hda_codec *codec, int val) codec_dbg(codec, "%s: on.", __func__); chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_16_000); chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_16_000); - if (spec->quirk == QUIRK_R3DI) + if (ca0132_quirk(spec) == QUIRK_R3DI) chipio_set_conn_rate(codec, 0x0F, SR_16_000); if (spec->effects_switch[VOICE_FOCUS - EFFECT_START_NID]) @@ -4338,16 +4667,23 @@ static int ca0132_alt_select_in(struct hda_codec *codec) switch (spec->cur_mic_type) { case REAR_MIC: - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: case QUIRK_R3D: - ca0132_mmio_gpio_set(codec, 0, false); + ca0113_mmio_gpio_set(codec, 0, false); + tmp = FLOAT_THREE; + break; + case QUIRK_ZXR: tmp = FLOAT_THREE; break; case QUIRK_R3DI: r3di_gpio_mic_set(codec, R3DI_REAR_MIC); tmp = FLOAT_ONE; break; + case QUIRK_AE5: + ca0113_mmio_command_set(codec, 0x48, 0x28, 0x00); + tmp = FLOAT_THREE; + break; default: tmp = FLOAT_ONE; break; @@ -4355,60 +4691,84 @@ static int ca0132_alt_select_in(struct hda_codec *codec) chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); - if (spec->quirk == QUIRK_R3DI) + if (ca0132_quirk(spec) == QUIRK_R3DI) chipio_set_conn_rate(codec, 0x0F, SR_96_000); dspio_set_uint_param(codec, 0x80, 0x00, tmp); chipio_set_stream_control(codec, 0x03, 1); chipio_set_stream_control(codec, 0x04, 1); - - if (spec->quirk == QUIRK_SBZ) { + switch (ca0132_quirk(spec)) { + case QUIRK_SBZ: chipio_write(codec, 0x18B098, 0x0000000C); chipio_write(codec, 0x18B09C, 0x0000000C); + break; + case QUIRK_ZXR: + chipio_write(codec, 0x18B098, 0x0000000C); + chipio_write(codec, 0x18B09C, 0x000000CC); + break; + case QUIRK_AE5: + chipio_write(codec, 0x18B098, 0x0000000C); + chipio_write(codec, 0x18B09C, 0x0000004C); + break; + default: + break; } ca0132_alt_mic_boost_set(codec, spec->mic_boost_enum_val); break; case REAR_LINE_IN: ca0132_mic_boost_set(codec, 0); - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: case QUIRK_R3D: - ca0132_mmio_gpio_set(codec, 0, false); + ca0113_mmio_gpio_set(codec, 0, false); break; case QUIRK_R3DI: r3di_gpio_mic_set(codec, R3DI_REAR_MIC); break; + case QUIRK_AE5: + ca0113_mmio_command_set(codec, 0x48, 0x28, 0x00); + break; + default: + break; } chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); - if (spec->quirk == QUIRK_R3DI) + if (ca0132_quirk(spec) == QUIRK_R3DI) chipio_set_conn_rate(codec, 0x0F, SR_96_000); tmp = FLOAT_ZERO; dspio_set_uint_param(codec, 0x80, 0x00, tmp); - if (spec->quirk == QUIRK_SBZ) { + switch (ca0132_quirk(spec)) { + case QUIRK_SBZ: + case QUIRK_AE5: chipio_write(codec, 0x18B098, 0x00000000); chipio_write(codec, 0x18B09C, 0x00000000); + break; + default: + break; } - chipio_set_stream_control(codec, 0x03, 1); chipio_set_stream_control(codec, 0x04, 1); break; case FRONT_MIC: - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: case QUIRK_R3D: - ca0132_mmio_gpio_set(codec, 0, true); - ca0132_mmio_gpio_set(codec, 5, false); + ca0113_mmio_gpio_set(codec, 0, true); + ca0113_mmio_gpio_set(codec, 5, false); tmp = FLOAT_THREE; break; case QUIRK_R3DI: r3di_gpio_mic_set(codec, R3DI_FRONT_MIC); tmp = FLOAT_ONE; break; + case QUIRK_AE5: + ca0113_mmio_command_set(codec, 0x48, 0x28, 0x3f); + tmp = FLOAT_THREE; + break; default: tmp = FLOAT_ONE; break; @@ -4416,7 +4776,7 @@ static int ca0132_alt_select_in(struct hda_codec *codec) chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); - if (spec->quirk == QUIRK_R3DI) + if (ca0132_quirk(spec) == QUIRK_R3DI) chipio_set_conn_rate(codec, 0x0F, SR_96_000); dspio_set_uint_param(codec, 0x80, 0x00, tmp); @@ -4424,9 +4784,17 @@ static int ca0132_alt_select_in(struct hda_codec *codec) chipio_set_stream_control(codec, 0x03, 1); chipio_set_stream_control(codec, 0x04, 1); - if (spec->quirk == QUIRK_SBZ) { + switch (ca0132_quirk(spec)) { + case QUIRK_SBZ: chipio_write(codec, 0x18B098, 0x0000000C); chipio_write(codec, 0x18B09C, 0x000000CC); + break; + case QUIRK_AE5: + chipio_write(codec, 0x18B098, 0x0000000C); + chipio_write(codec, 0x18B09C, 0x0000004C); + break; + default: + break; } ca0132_alt_mic_boost_set(codec, spec->mic_boost_enum_val); break; @@ -4435,7 +4803,6 @@ static int ca0132_alt_select_in(struct hda_codec *codec) snd_hda_power_down_pm(codec); return 0; - } /* @@ -4507,6 +4874,8 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val) /* if PE if off, turn off out effects. */ if (!spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]) val = 0; + if (spec->cur_out_type == SURROUND_OUT && nid == X_BASS) + val = 0; } /* for in effect, qualify with CrystalVoice */ @@ -4520,7 +4889,7 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val) val = 0; /* If Voice Focus on SBZ, set to two channel. */ - if ((nid == VOICE_FOCUS) && (spec->quirk == QUIRK_SBZ) + if ((nid == VOICE_FOCUS) && ca0132_use_pci_mmio(spec) && (spec->cur_mic_type != REAR_LINE_IN)) { if (spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID]) { @@ -4539,7 +4908,7 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val) * For SBZ noise reduction, there's an extra command * to module ID 0x47. No clue why. */ - if ((nid == NOISE_REDUCTION) && (spec->quirk == QUIRK_SBZ) + if ((nid == NOISE_REDUCTION) && ca0132_use_pci_mmio(spec) && (spec->cur_mic_type != REAR_LINE_IN)) { if (spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID]) { @@ -4555,7 +4924,7 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val) } /* If rear line in disable effects. */ - if (spec->use_alt_functions && + if (ca0132_use_alt_functions(spec) && spec->in_enum_val == REAR_LINE_IN) val = 0; } @@ -4585,7 +4954,7 @@ static int ca0132_pe_switch_set(struct hda_codec *codec) codec_dbg(codec, "ca0132_pe_switch_set: val=%ld\n", spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]); - if (spec->use_alt_functions) + if (ca0132_use_alt_functions(spec)) ca0132_alt_select_out(codec); i = OUT_EFFECT_START_NID - EFFECT_START_NID; @@ -4645,7 +5014,7 @@ static int ca0132_cvoice_switch_set(struct hda_codec *codec) /* set correct vipsource */ oldval = stop_mic1(codec); - if (spec->use_alt_functions) + if (ca0132_use_alt_functions(spec)) ret |= ca0132_alt_set_vipsource(codec, 1); else ret |= ca0132_set_vipsource(codec, 1); @@ -4678,6 +5047,27 @@ static int ca0132_alt_mic_boost_set(struct hda_codec *codec, long val) return ret; } +static int ae5_headphone_gain_set(struct hda_codec *codec, long val) +{ + unsigned int i; + + for (i = 0; i < 4; i++) + ca0113_mmio_command_set(codec, 0x48, 0x11 + i, + ae5_headphone_gain_presets[val].vals[i]); + return 0; +} + +/* + * gpio pin 1 is a relay that switches on/off, apparently setting the headphone + * amplifier to handle a 600 ohm load. + */ +static int zxr_headphone_gain_set(struct hda_codec *codec, long val) +{ + ca0113_mmio_gpio_set(codec, 1, val); + + return 0; +} + static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -4693,7 +5083,7 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol, auto_jack = spec->vnode_lswitch[VNID_HP_ASEL - VNODE_START_NID]; if (!auto_jack) { - if (spec->use_alt_functions) + if (ca0132_use_alt_functions(spec)) ca0132_alt_select_out(codec); else ca0132_select_out(codec); @@ -4710,7 +5100,7 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol, } if (nid == VNID_HP_ASEL) { - if (spec->use_alt_functions) + if (ca0132_use_alt_functions(spec)) ca0132_alt_select_out(codec); else ca0132_select_out(codec); @@ -4942,6 +5332,112 @@ static int ca0132_alt_mic_boost_put(struct snd_kcontrol *kcontrol, return 1; } +/* + * Sound BlasterX AE-5 Headphone Gain Controls. + */ +#define AE5_HEADPHONE_GAIN_MAX 3 +static int ae5_headphone_gain_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + char *sfx = " Ohms)"; + char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = AE5_HEADPHONE_GAIN_MAX; + if (uinfo->value.enumerated.item >= AE5_HEADPHONE_GAIN_MAX) + uinfo->value.enumerated.item = AE5_HEADPHONE_GAIN_MAX - 1; + sprintf(namestr, "%s %s", + ae5_headphone_gain_presets[uinfo->value.enumerated.item].name, + sfx); + strcpy(uinfo->value.enumerated.name, namestr); + return 0; +} + +static int ae5_headphone_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->ae5_headphone_gain_val; + return 0; +} + +static int ae5_headphone_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + int sel = ucontrol->value.enumerated.item[0]; + unsigned int items = AE5_HEADPHONE_GAIN_MAX; + + if (sel >= items) + return 0; + + codec_dbg(codec, "ae5_headphone_gain: boost=%d\n", + sel); + + spec->ae5_headphone_gain_val = sel; + + if (spec->out_enum_val == HEADPHONE_OUT) + ae5_headphone_gain_set(codec, spec->ae5_headphone_gain_val); + + return 1; +} + +/* + * Sound BlasterX AE-5 sound filter enumerated control. + */ +#define AE5_SOUND_FILTER_MAX 3 + +static int ae5_sound_filter_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = AE5_SOUND_FILTER_MAX; + if (uinfo->value.enumerated.item >= AE5_SOUND_FILTER_MAX) + uinfo->value.enumerated.item = AE5_SOUND_FILTER_MAX - 1; + sprintf(namestr, "%s", + ae5_filter_presets[uinfo->value.enumerated.item].name); + strcpy(uinfo->value.enumerated.name, namestr); + return 0; +} + +static int ae5_sound_filter_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->ae5_filter_val; + return 0; +} + +static int ae5_sound_filter_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + int sel = ucontrol->value.enumerated.item[0]; + unsigned int items = AE5_SOUND_FILTER_MAX; + + if (sel >= items) + return 0; + + codec_dbg(codec, "ae5_sound_filter: %s\n", + ae5_filter_presets[sel].name); + + spec->ae5_filter_val = sel; + + ca0113_mmio_command_set_type2(codec, 0x48, 0x07, + ae5_filter_presets[sel].val); + + return 1; +} /* * Input Select Control for alternative ca0132 codecs. This exists because @@ -5318,7 +5814,7 @@ static int ca0132_switch_put(struct snd_kcontrol *kcontrol, /* mic boost */ if (nid == spec->input_pins[0]) { spec->cur_mic_boost = *valp; - if (spec->use_alt_functions) { + if (ca0132_use_alt_functions(spec)) { if (spec->in_enum_val != REAR_LINE_IN) changed = ca0132_mic_boost_set(codec, *valp); } else { @@ -5330,6 +5826,16 @@ static int ca0132_switch_put(struct snd_kcontrol *kcontrol, goto exit; } + if (nid == ZXR_HEADPHONE_GAIN) { + spec->zxr_gain_set = *valp; + if (spec->cur_out_type == HEADPHONE_OUT) + changed = zxr_headphone_gain_set(codec, *valp); + else + changed = 0; + + goto exit; + } + exit: snd_hda_power_down(codec); return changed; @@ -5604,7 +6110,7 @@ static int add_fx_switch(struct hda_codec *codec, hda_nid_t nid, /* If using alt_controls, add FX: prefix. But, don't add FX: * prefix to OutFX or InFX enable controls. */ - if ((spec->use_alt_controls) && (nid <= IN_EFFECT_END_NID)) + if (ca0132_use_alt_controls(spec) && (nid <= IN_EFFECT_END_NID)) sprintf(namestr, "FX: %s %s Switch", pfx, dirstr[dir]); else sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]); @@ -5705,6 +6211,50 @@ static int ca0132_alt_add_mic_boost_enum(struct hda_codec *codec) } /* + * Add headphone gain enumerated control for the AE-5. This switches between + * three modes, low, medium, and high. When non-headphone outputs are selected, + * it is automatically set to high. This is the same behavior as Windows. + */ +static int ae5_add_headphone_gain_enum(struct hda_codec *codec) +{ + struct snd_kcontrol_new knew = + HDA_CODEC_MUTE_MONO("AE-5: Headphone Gain", + AE5_HEADPHONE_GAIN_ENUM, 1, 0, HDA_OUTPUT); + knew.info = ae5_headphone_gain_info; + knew.get = ae5_headphone_gain_get; + knew.put = ae5_headphone_gain_put; + return snd_hda_ctl_add(codec, AE5_HEADPHONE_GAIN_ENUM, + snd_ctl_new1(&knew, codec)); +} + +/* + * Add sound filter enumerated control for the AE-5. This adds three different + * settings: Slow Roll Off, Minimum Phase, and Fast Roll Off. From what I've + * read into it, it changes the DAC's interpolation filter. + */ +static int ae5_add_sound_filter_enum(struct hda_codec *codec) +{ + struct snd_kcontrol_new knew = + HDA_CODEC_MUTE_MONO("AE-5: Sound Filter", + AE5_SOUND_FILTER_ENUM, 1, 0, HDA_OUTPUT); + knew.info = ae5_sound_filter_info; + knew.get = ae5_sound_filter_get; + knew.put = ae5_sound_filter_put; + return snd_hda_ctl_add(codec, AE5_SOUND_FILTER_ENUM, + snd_ctl_new1(&knew, codec)); +} + +static int zxr_add_headphone_gain_switch(struct hda_codec *codec) +{ + struct snd_kcontrol_new knew = + CA0132_CODEC_MUTE_MONO("ZxR: 600 Ohm Gain", + ZXR_HEADPHONE_GAIN, 1, HDA_OUTPUT); + + return snd_hda_ctl_add(codec, ZXR_HEADPHONE_GAIN, + snd_ctl_new1(&knew, codec)); +} + +/* * Need to create slave controls for the alternate codecs that have surround * capabilities. */ @@ -5837,7 +6387,7 @@ static int ca0132_build_controls(struct hda_codec *codec) return err; } /* Setup vmaster with surround slaves for desktop ca0132 devices */ - if (spec->use_alt_functions) { + if (ca0132_use_alt_functions(spec)) { snd_hda_set_vmaster_tlv(codec, spec->dacs[0], HDA_OUTPUT, spec->tlv); snd_hda_add_vmaster(codec, "Master Playback Volume", @@ -5847,7 +6397,8 @@ static int ca0132_build_controls(struct hda_codec *codec) NULL, ca0132_alt_slave_pfxs, "Playback Switch", true, &spec->vmaster_mute.sw_kctl); - + if (err < 0) + return err; } /* Add in and out effects controls. @@ -5855,8 +6406,8 @@ static int ca0132_build_controls(struct hda_codec *codec) */ num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT; for (i = 0; i < num_fx; i++) { - /* SBZ and R3D break if Echo Cancellation is used. */ - if (spec->quirk == QUIRK_SBZ || spec->quirk == QUIRK_R3D) { + /* Desktop cards break if Echo Cancellation is used. */ + if (ca0132_use_pci_mmio(spec)) { if (i == (ECHO_CANCELLATION - IN_EFFECT_START_NID + OUT_EFFECTS_COUNT)) continue; @@ -5873,9 +6424,15 @@ static int ca0132_build_controls(struct hda_codec *codec) * EQ presets, and Smart Volume presets. Also, change names to add FX * prefix, and change PlayEnhancement and CrystalVoice to match. */ - if (spec->use_alt_controls) { - ca0132_alt_add_svm_enum(codec); - add_ca0132_alt_eq_presets(codec); + if (ca0132_use_alt_controls(spec)) { + err = ca0132_alt_add_svm_enum(codec); + if (err < 0) + return err; + + err = add_ca0132_alt_eq_presets(codec); + if (err < 0) + return err; + err = add_fx_switch(codec, PLAY_ENHANCEMENT, "Enable OutFX", 0); if (err < 0) @@ -5912,17 +6469,46 @@ static int ca0132_build_controls(struct hda_codec *codec) if (err < 0) return err; } - add_voicefx(codec); + err = add_voicefx(codec); + if (err < 0) + return err; /* * If the codec uses alt_functions, you need the enumerated controls * to select the new outputs and inputs, plus add the new mic boost * setting control. */ - if (spec->use_alt_functions) { - ca0132_alt_add_output_enum(codec); - ca0132_alt_add_input_enum(codec); - ca0132_alt_add_mic_boost_enum(codec); + if (ca0132_use_alt_functions(spec)) { + err = ca0132_alt_add_output_enum(codec); + if (err < 0) + return err; + err = ca0132_alt_add_mic_boost_enum(codec); + if (err < 0) + return err; + /* + * ZxR only has microphone input, there is no front panel + * header on the card, and aux-in is handled by the DBPro board. + */ + if (ca0132_quirk(spec) != QUIRK_ZXR) { + err = ca0132_alt_add_input_enum(codec); + if (err < 0) + return err; + } + } + + if (ca0132_quirk(spec) == QUIRK_AE5) { + err = ae5_add_headphone_gain_enum(codec); + if (err < 0) + return err; + err = ae5_add_sound_filter_enum(codec); + if (err < 0) + return err; + } + + if (ca0132_quirk(spec) == QUIRK_ZXR) { + err = zxr_add_headphone_gain_switch(codec); + if (err < 0) + return err; } #ifdef ENABLE_TUNING_CONTROLS add_tuning_ctls(codec); @@ -5949,12 +6535,33 @@ static int ca0132_build_controls(struct hda_codec *codec) return err; } - if (spec->use_alt_functions) + if (ca0132_use_alt_functions(spec)) ca0132_alt_add_chmap_ctls(codec); return 0; } +static int dbpro_build_controls(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + int err = 0; + + if (spec->dig_out) { + err = snd_hda_create_spdif_out_ctls(codec, spec->dig_out, + spec->dig_out); + if (err < 0) + return err; + } + + if (spec->dig_in) { + err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in); + if (err < 0) + return err; + } + + return 0; +} + /* * PCM */ @@ -6006,7 +6613,7 @@ static int ca0132_build_pcms(struct hda_codec *codec) info = snd_hda_codec_pcm_new(codec, "CA0132 Analog"); if (!info) return -ENOMEM; - if (spec->use_alt_functions) { + if (ca0132_use_alt_functions(spec)) { info->own_chmap = true; info->stream[SNDRV_PCM_STREAM_PLAYBACK].chmap = ca0132_alt_chmaps; @@ -6020,7 +6627,7 @@ static int ca0132_build_pcms(struct hda_codec *codec) info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[0]; /* With the DSP enabled, desktops don't use this ADC. */ - if (!spec->use_alt_functions) { + if (!ca0132_use_alt_functions(spec)) { info = snd_hda_codec_pcm_new(codec, "CA0132 Analog Mic-In2"); if (!info) return -ENOMEM; @@ -6058,6 +6665,40 @@ static int ca0132_build_pcms(struct hda_codec *codec) return 0; } +static int dbpro_build_pcms(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + struct hda_pcm *info; + + info = snd_hda_codec_pcm_new(codec, "CA0132 Alt Analog"); + if (!info) + return -ENOMEM; + info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0132_pcm_analog_capture; + info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1; + info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[0]; + + + if (!spec->dig_out && !spec->dig_in) + return 0; + + info = snd_hda_codec_pcm_new(codec, "CA0132 Digital"); + if (!info) + return -ENOMEM; + info->pcm_type = HDA_PCM_TYPE_SPDIF; + if (spec->dig_out) { + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = + ca0132_pcm_digital_playback; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dig_out; + } + if (spec->dig_in) { + info->stream[SNDRV_PCM_STREAM_CAPTURE] = + ca0132_pcm_digital_capture; + info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in; + } + + return 0; +} + static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac) { if (pin) { @@ -6184,7 +6825,7 @@ static void ca0132_init_dmic(struct hda_codec *codec) * Bit 6: set to select Data2, clear for Data1 * Bit 7: set to enable DMic, clear for AMic */ - if (spec->quirk == QUIRK_ALIENWARE_M17XR4) + if (ca0132_quirk(spec) == QUIRK_ALIENWARE_M17XR4) val = 0x33; else val = 0x23; @@ -6238,69 +6879,48 @@ static void ca0132_refresh_widget_caps(struct hda_codec *codec) } /* - * Recon3D r3d_setup_defaults sub functions. + * Creates a dummy stream to bind the output to. This seems to have to be done + * after changing the main outputs source and destination streams. */ - -static void r3d_dsp_scp_startup(struct hda_codec *codec) +static void ca0132_alt_create_dummy_stream(struct hda_codec *codec) { - unsigned int tmp; - - tmp = 0x00000000; - dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp); - - tmp = 0x00000001; - dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp); - - tmp = 0x00000004; - dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); - - tmp = 0x00000005; - dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); - - tmp = 0x00000000; - dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); - -} + struct ca0132_spec *spec = codec->spec; + unsigned int stream_format; -static void r3d_dsp_initial_mic_setup(struct hda_codec *codec) -{ - unsigned int tmp; + stream_format = snd_hdac_calc_stream_format(48000, 2, + SNDRV_PCM_FORMAT_S32_LE, 32, 0); - /* Mic 1 Setup */ - chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); - chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); - /* This ConnPointID is unique to Recon3Di. Haven't seen it elsewhere */ - chipio_set_conn_rate(codec, 0x0F, SR_96_000); - tmp = FLOAT_ONE; - dspio_set_uint_param(codec, 0x80, 0x00, tmp); + snd_hda_codec_setup_stream(codec, spec->dacs[0], spec->dsp_stream_id, + 0, stream_format); - /* Mic 2 Setup, even though it isn't connected on SBZ */ - chipio_set_conn_rate(codec, MEM_CONNID_MICIN2, SR_96_000); - chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2, SR_96_000); - chipio_set_conn_rate(codec, 0x0F, SR_96_000); - tmp = FLOAT_ZERO; - dspio_set_uint_param(codec, 0x80, 0x01, tmp); + snd_hda_codec_cleanup_stream(codec, spec->dacs[0]); } /* - * Initialize Sound Blaster Z analog microphones. + * Initialize mic for non-chromebook ca0132 implementations. */ -static void sbz_init_analog_mics(struct hda_codec *codec) +static void ca0132_alt_init_analog_mics(struct hda_codec *codec) { + struct ca0132_spec *spec = codec->spec; unsigned int tmp; /* Mic 1 Setup */ chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); - tmp = FLOAT_THREE; + if (ca0132_quirk(spec) == QUIRK_R3DI) { + chipio_set_conn_rate(codec, 0x0F, SR_96_000); + tmp = FLOAT_ONE; + } else + tmp = FLOAT_THREE; dspio_set_uint_param(codec, 0x80, 0x00, tmp); - /* Mic 2 Setup, even though it isn't connected on SBZ */ + /* Mic 2 setup (not present on desktop cards) */ chipio_set_conn_rate(codec, MEM_CONNID_MICIN2, SR_96_000); chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2, SR_96_000); + if (ca0132_quirk(spec) == QUIRK_R3DI) + chipio_set_conn_rate(codec, 0x0F, SR_96_000); tmp = FLOAT_ZERO; dspio_set_uint_param(codec, 0x80, 0x01, tmp); - } /* @@ -6333,7 +6953,6 @@ static void sbz_connect_streams(struct hda_codec *codec) codec_dbg(codec, "Connect Streams exited, mutex released.\n"); mutex_unlock(&spec->chipio_mutex); - } /* @@ -6360,19 +6979,29 @@ static void sbz_chipio_startup_data(struct hda_codec *codec) chipio_set_stream_channels(codec, 0x0C, 6); chipio_set_stream_control(codec, 0x0C, 1); /* No clue what these control */ - chipio_write_no_mutex(codec, 0x190030, 0x0001e0c0); - chipio_write_no_mutex(codec, 0x190034, 0x0001e1c1); - chipio_write_no_mutex(codec, 0x190038, 0x0001e4c2); - chipio_write_no_mutex(codec, 0x19003c, 0x0001e5c3); - chipio_write_no_mutex(codec, 0x190040, 0x0001e2c4); - chipio_write_no_mutex(codec, 0x190044, 0x0001e3c5); - chipio_write_no_mutex(codec, 0x190048, 0x0001e8c6); - chipio_write_no_mutex(codec, 0x19004c, 0x0001e9c7); - chipio_write_no_mutex(codec, 0x190050, 0x0001ecc8); - chipio_write_no_mutex(codec, 0x190054, 0x0001edc9); - chipio_write_no_mutex(codec, 0x190058, 0x0001eaca); - chipio_write_no_mutex(codec, 0x19005c, 0x0001ebcb); - + if (ca0132_quirk(spec) == QUIRK_SBZ) { + chipio_write_no_mutex(codec, 0x190030, 0x0001e0c0); + chipio_write_no_mutex(codec, 0x190034, 0x0001e1c1); + chipio_write_no_mutex(codec, 0x190038, 0x0001e4c2); + chipio_write_no_mutex(codec, 0x19003c, 0x0001e5c3); + chipio_write_no_mutex(codec, 0x190040, 0x0001e2c4); + chipio_write_no_mutex(codec, 0x190044, 0x0001e3c5); + chipio_write_no_mutex(codec, 0x190048, 0x0001e8c6); + chipio_write_no_mutex(codec, 0x19004c, 0x0001e9c7); + chipio_write_no_mutex(codec, 0x190050, 0x0001ecc8); + chipio_write_no_mutex(codec, 0x190054, 0x0001edc9); + chipio_write_no_mutex(codec, 0x190058, 0x0001eaca); + chipio_write_no_mutex(codec, 0x19005c, 0x0001ebcb); + } else if (ca0132_quirk(spec) == QUIRK_ZXR) { + chipio_write_no_mutex(codec, 0x190038, 0x000140c2); + chipio_write_no_mutex(codec, 0x19003c, 0x000141c3); + chipio_write_no_mutex(codec, 0x190040, 0x000150c4); + chipio_write_no_mutex(codec, 0x190044, 0x000151c5); + chipio_write_no_mutex(codec, 0x190050, 0x000142c8); + chipio_write_no_mutex(codec, 0x190054, 0x000143c9); + chipio_write_no_mutex(codec, 0x190058, 0x000152ca); + chipio_write_no_mutex(codec, 0x19005c, 0x000153cb); + } chipio_write_no_mutex(codec, 0x19042c, 0x00000001); codec_dbg(codec, "Startup Data exited, mutex released.\n"); @@ -6380,35 +7009,58 @@ static void sbz_chipio_startup_data(struct hda_codec *codec) } /* - * Sound Blaster Z uses these after DSP is loaded. Weird SCP commands - * without a 0x20 source like normal. + * Custom DSP SCP commands where the src value is 0x00 instead of 0x20. This is + * done after the DSP is loaded. */ -static void sbz_dsp_scp_startup(struct hda_codec *codec) +static void ca0132_alt_dsp_scp_startup(struct hda_codec *codec) { - unsigned int tmp; - - tmp = 0x00000003; - dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); - - tmp = 0x00000000; - dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp); - - tmp = 0x00000001; - dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp); - - tmp = 0x00000004; - dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); - - tmp = 0x00000005; - dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); - - tmp = 0x00000000; - dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + struct ca0132_spec *spec = codec->spec; + unsigned int tmp, i; + /* + * Gotta run these twice, or else mic works inconsistently. Not clear + * why this is, but multiple tests have confirmed it. + */ + for (i = 0; i < 2; i++) { + switch (ca0132_quirk(spec)) { + case QUIRK_SBZ: + case QUIRK_AE5: + tmp = 0x00000003; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + tmp = 0x00000000; + dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp); + tmp = 0x00000001; + dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp); + tmp = 0x00000004; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + tmp = 0x00000005; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + tmp = 0x00000000; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + break; + case QUIRK_R3D: + case QUIRK_R3DI: + tmp = 0x00000000; + dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp); + tmp = 0x00000001; + dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp); + tmp = 0x00000004; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + tmp = 0x00000005; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + tmp = 0x00000000; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + break; + default: + break; + } + msleep(100); + } } -static void sbz_dsp_initial_mic_setup(struct hda_codec *codec) +static void ca0132_alt_dsp_initial_mic_setup(struct hda_codec *codec) { + struct ca0132_spec *spec = codec->spec; unsigned int tmp; chipio_set_stream_control(codec, 0x03, 0); @@ -6423,8 +7075,163 @@ static void sbz_dsp_initial_mic_setup(struct hda_codec *codec) chipio_set_stream_control(codec, 0x03, 1); chipio_set_stream_control(codec, 0x04, 1); - chipio_write(codec, 0x18b098, 0x0000000c); - chipio_write(codec, 0x18b09C, 0x0000000c); + switch (ca0132_quirk(spec)) { + case QUIRK_SBZ: + chipio_write(codec, 0x18b098, 0x0000000c); + chipio_write(codec, 0x18b09C, 0x0000000c); + break; + case QUIRK_AE5: + chipio_write(codec, 0x18b098, 0x0000000c); + chipio_write(codec, 0x18b09c, 0x0000004c); + break; + default: + break; + } +} + +static void ae5_post_dsp_register_set(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + chipio_8051_write_direct(codec, 0x93, 0x10); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x44); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc2); + + writeb(0xff, spec->mem_base + 0x304); + writeb(0xff, spec->mem_base + 0x304); + writeb(0xff, spec->mem_base + 0x304); + writeb(0xff, spec->mem_base + 0x304); + writeb(0x00, spec->mem_base + 0x100); + writeb(0xff, spec->mem_base + 0x304); + writeb(0x00, spec->mem_base + 0x100); + writeb(0xff, spec->mem_base + 0x304); + writeb(0x00, spec->mem_base + 0x100); + writeb(0xff, spec->mem_base + 0x304); + writeb(0x00, spec->mem_base + 0x100); + writeb(0xff, spec->mem_base + 0x304); + + ca0113_mmio_command_set(codec, 0x30, 0x2b, 0x3f); + ca0113_mmio_command_set(codec, 0x30, 0x2d, 0x3f); + ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83); +} + +static void ae5_post_dsp_param_setup(struct hda_codec *codec) +{ + /* + * Param3 in the 8051's memory is represented by the ascii string 'mch' + * which seems to be 'multichannel'. This is also mentioned in the + * AE-5's registry values in Windows. + */ + chipio_set_control_param(codec, 3, 0); + /* + * I believe ASI is 'audio serial interface' and that it's used to + * change colors on the external LED strip connected to the AE-5. + */ + chipio_set_control_flag(codec, CONTROL_FLAG_ASI_96KHZ, 1); + + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, 0x724, 0x83); + chipio_set_control_param(codec, CONTROL_PARAM_ASI, 0); + + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x92); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_HIGH, 0xfa); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_DATA_WRITE, 0x22); +} + +static void ae5_post_dsp_pll_setup(struct hda_codec *codec) +{ + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x41); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc8); + + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x45); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PLL_PMU_WRITE, 0xcc); + + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x40); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PLL_PMU_WRITE, 0xcb); + + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x43); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc7); + + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x51); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PLL_PMU_WRITE, 0x8d); +} + +static void ae5_post_dsp_stream_setup(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + mutex_lock(&spec->chipio_mutex); + + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, 0x725, 0x81); + + chipio_set_conn_rate_no_mutex(codec, 0x70, SR_96_000); + + chipio_set_stream_channels(codec, 0x0C, 6); + chipio_set_stream_control(codec, 0x0C, 1); + + chipio_set_stream_source_dest(codec, 0x5, 0x43, 0x0); + + chipio_set_stream_source_dest(codec, 0x18, 0x9, 0xd0); + chipio_set_conn_rate_no_mutex(codec, 0xd0, SR_96_000); + chipio_set_stream_channels(codec, 0x18, 6); + chipio_set_stream_control(codec, 0x18, 1); + + chipio_set_control_param_no_mutex(codec, CONTROL_PARAM_ASI, 4); + + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x43); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc7); + + ca0113_mmio_command_set(codec, 0x48, 0x01, 0x80); + + mutex_unlock(&spec->chipio_mutex); +} + +static void ae5_post_dsp_startup_data(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + mutex_lock(&spec->chipio_mutex); + + chipio_write_no_mutex(codec, 0x189000, 0x0001f101); + chipio_write_no_mutex(codec, 0x189004, 0x0001f101); + chipio_write_no_mutex(codec, 0x189024, 0x00014004); + chipio_write_no_mutex(codec, 0x189028, 0x0002000f); + + ca0113_mmio_command_set(codec, 0x48, 0x0a, 0x05); + chipio_set_control_param_no_mutex(codec, CONTROL_PARAM_ASI, 7); + ca0113_mmio_command_set(codec, 0x48, 0x0b, 0x12); + ca0113_mmio_command_set(codec, 0x48, 0x04, 0x00); + ca0113_mmio_command_set(codec, 0x48, 0x06, 0x48); + ca0113_mmio_command_set(codec, 0x48, 0x0a, 0x05); + ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83); + ca0113_mmio_command_set(codec, 0x48, 0x0f, 0x00); + ca0113_mmio_command_set(codec, 0x48, 0x10, 0x00); + ca0113_mmio_gpio_set(codec, 0, true); + ca0113_mmio_gpio_set(codec, 1, true); + ca0113_mmio_command_set(codec, 0x48, 0x07, 0x80); + + chipio_write_no_mutex(codec, 0x18b03c, 0x00000012); + + ca0113_mmio_command_set(codec, 0x48, 0x0f, 0x00); + ca0113_mmio_command_set(codec, 0x48, 0x10, 0x00); + + mutex_unlock(&spec->chipio_mutex); } /* @@ -6485,9 +7292,8 @@ static void r3d_setup_defaults(struct hda_codec *codec) if (spec->dsp_state != DSP_DOWNLOADED) return; - r3d_dsp_scp_startup(codec); - - r3d_dsp_initial_mic_setup(codec); + ca0132_alt_dsp_scp_startup(codec); + ca0132_alt_init_analog_mics(codec); /*remove DSP headroom*/ tmp = FLOAT_ZERO; @@ -6501,7 +7307,7 @@ static void r3d_setup_defaults(struct hda_codec *codec) /* Set speaker source? */ dspio_set_uint_param(codec, 0x32, 0x00, tmp); - if (spec->quirk == QUIRK_R3DI) + if (ca0132_quirk(spec) == QUIRK_R3DI) r3di_gpio_dsp_status_set(codec, R3DI_DSP_DOWNLOADED); /* Setup effect defaults */ @@ -6523,19 +7329,16 @@ static void r3d_setup_defaults(struct hda_codec *codec) static void sbz_setup_defaults(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; - unsigned int tmp, stream_format; + unsigned int tmp; int num_fx; int idx, i; if (spec->dsp_state != DSP_DOWNLOADED) return; - sbz_dsp_scp_startup(codec); - - sbz_init_analog_mics(codec); - + ca0132_alt_dsp_scp_startup(codec); + ca0132_alt_init_analog_mics(codec); sbz_connect_streams(codec); - sbz_chipio_startup_data(codec); chipio_set_stream_control(codec, 0x03, 1); @@ -6561,8 +7364,7 @@ static void sbz_setup_defaults(struct hda_codec *codec) /* Set speaker source? */ dspio_set_uint_param(codec, 0x32, 0x00, tmp); - sbz_dsp_initial_mic_setup(codec); - + ca0132_alt_dsp_initial_mic_setup(codec); /* out, in effects + voicefx */ num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT + 1; @@ -6575,23 +7377,74 @@ static void sbz_setup_defaults(struct hda_codec *codec) } } - /* - * Have to make a stream to bind the sound output to, otherwise - * you'll get dead audio. Before I did this, it would bind to an - * audio input, and would never work - */ - stream_format = snd_hdac_calc_stream_format(48000, 2, - SNDRV_PCM_FORMAT_S32_LE, 32, 0); + ca0132_alt_create_dummy_stream(codec); +} - snd_hda_codec_setup_stream(codec, spec->dacs[0], spec->dsp_stream_id, - 0, stream_format); +/* + * Setup default parameters for the Sound BlasterX AE-5 DSP. + */ +static void ae5_setup_defaults(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int tmp; + int num_fx; + int idx, i; - snd_hda_codec_cleanup_stream(codec, spec->dacs[0]); + if (spec->dsp_state != DSP_DOWNLOADED) + return; - snd_hda_codec_setup_stream(codec, spec->dacs[0], spec->dsp_stream_id, - 0, stream_format); + ca0132_alt_dsp_scp_startup(codec); + ca0132_alt_init_analog_mics(codec); + chipio_set_stream_control(codec, 0x03, 1); + chipio_set_stream_control(codec, 0x04, 1); - snd_hda_codec_cleanup_stream(codec, spec->dacs[0]); + /* New, unknown SCP req's */ + tmp = FLOAT_ZERO; + dspio_set_uint_param(codec, 0x96, 0x29, tmp); + dspio_set_uint_param(codec, 0x96, 0x2a, tmp); + dspio_set_uint_param(codec, 0x80, 0x0d, tmp); + dspio_set_uint_param(codec, 0x80, 0x0e, tmp); + + ca0113_mmio_command_set(codec, 0x30, 0x2e, 0x3f); + ca0113_mmio_gpio_set(codec, 0, false); + ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00); + + /* Internal loopback off */ + tmp = FLOAT_ONE; + dspio_set_uint_param(codec, 0x37, 0x08, tmp); + dspio_set_uint_param(codec, 0x37, 0x10, tmp); + + /*remove DSP headroom*/ + tmp = FLOAT_ZERO; + dspio_set_uint_param(codec, 0x96, 0x3C, tmp); + + /* set WUH source */ + tmp = FLOAT_TWO; + dspio_set_uint_param(codec, 0x31, 0x00, tmp); + chipio_set_conn_rate(codec, MEM_CONNID_WUH, SR_48_000); + + /* Set speaker source? */ + dspio_set_uint_param(codec, 0x32, 0x00, tmp); + + ca0132_alt_dsp_initial_mic_setup(codec); + ae5_post_dsp_register_set(codec); + ae5_post_dsp_param_setup(codec); + ae5_post_dsp_pll_setup(codec); + ae5_post_dsp_stream_setup(codec); + ae5_post_dsp_startup_data(codec); + + /* out, in effects + voicefx */ + num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT + 1; + for (idx = 0; idx < num_fx; idx++) { + for (i = 0; i <= ca0132_effects[idx].params; i++) { + dspio_set_uint_param(codec, + ca0132_effects[idx].mid, + ca0132_effects[idx].reqs[i], + ca0132_effects[idx].def_vals[i]); + } + } + + ca0132_alt_create_dummy_stream(codec); } /* @@ -6601,7 +7454,7 @@ static void ca0132_init_flags(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; - if (spec->use_alt_functions) { + if (ca0132_use_alt_functions(spec)) { chipio_set_control_flag(codec, CONTROL_FLAG_DSP_96KHZ, 1); chipio_set_control_flag(codec, CONTROL_FLAG_DAC_96KHZ, 1); chipio_set_control_flag(codec, CONTROL_FLAG_ADC_B_96KHZ, 1); @@ -6634,7 +7487,7 @@ static void ca0132_init_params(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; - if (spec->use_alt_functions) { + if (ca0132_use_alt_functions(spec)) { chipio_set_conn_rate(codec, MEM_CONNID_WUH, SR_48_000); chipio_set_conn_rate(codec, 0x0B, SR_48_000); chipio_set_control_param(codec, CONTROL_PARAM_SPDIF1_SOURCE, 0); @@ -6671,14 +7524,16 @@ static bool ca0132_download_dsp_images(struct hda_codec *codec) * can use the default firmware, but I'll leave the option in case * it needs it again. */ - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: - if (request_firmware(&fw_entry, SBZ_EFX_FILE, + case QUIRK_R3D: + case QUIRK_AE5: + if (request_firmware(&fw_entry, DESKTOP_EFX_FILE, codec->card->dev) != 0) { - codec_dbg(codec, "SBZ alt firmware not detected. "); + codec_dbg(codec, "Desktop firmware not found."); spec->alt_firmware_present = false; } else { - codec_dbg(codec, "Sound Blaster Z firmware selected."); + codec_dbg(codec, "Desktop firmware selected."); spec->alt_firmware_present = true; } break; @@ -6743,7 +7598,7 @@ static void ca0132_download_dsp(struct hda_codec *codec) } /* For codecs using alt functions, this is already done earlier */ - if (spec->dsp_state == DSP_DOWNLOADED && (!spec->use_alt_functions)) + if (spec->dsp_state == DSP_DOWNLOADED && !ca0132_use_alt_functions(spec)) ca0132_set_dsp_msr(codec, true); } @@ -6780,7 +7635,7 @@ static void amic_callback(struct hda_codec *codec, struct hda_jack_callback *cb) { struct ca0132_spec *spec = codec->spec; - if (spec->use_alt_functions) + if (ca0132_use_alt_functions(spec)) ca0132_alt_select_in(codec); else ca0132_select_mic(codec); @@ -6795,7 +7650,7 @@ static void ca0132_init_unsol(struct hda_codec *codec) snd_hda_jack_detect_enable_callback(codec, UNSOL_TAG_DSP, ca0132_process_dsp_response); /* Front headphone jack detection */ - if (spec->use_alt_functions) + if (ca0132_use_alt_functions(spec)) snd_hda_jack_detect_enable_callback(codec, spec->unsol_tag_front_hp, hp_callback); } @@ -6885,7 +7740,7 @@ static void ca0132_init_chip(struct hda_codec *codec) mutex_init(&spec->chipio_mutex); spec->cur_out_type = SPEAKER_OUT; - if (!spec->use_alt_functions) + if (!ca0132_use_alt_functions(spec)) spec->cur_mic_type = DIGITAL_MIC; else spec->cur_mic_type = REAR_MIC; @@ -6911,7 +7766,7 @@ static void ca0132_init_chip(struct hda_codec *codec) * Sets defaults for the effect slider controls, only for alternative * ca0132 codecs. Also sets x-bass crossover frequency to 80hz. */ - if (spec->use_alt_controls) { + if (ca0132_use_alt_controls(spec)) { spec->xbass_xover_freq = 8; for (i = 0; i < EFFECT_LEVEL_SLIDERS; i++) spec->fx_ctl_val[i] = effect_slider_defaults[i]; @@ -6921,6 +7776,14 @@ static void ca0132_init_chip(struct hda_codec *codec) spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID] = 1; spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID] = 0; + /* + * The ZxR doesn't have a front panel header, and it's line-in is on + * the daughter board. So, there is no input enum control, and we need + * to make sure that spec->in_enum_val is set properly. + */ + if (ca0132_quirk(spec) == QUIRK_ZXR) + spec->in_enum_val = REAR_MIC; + #ifdef ENABLE_TUNING_CONTROLS ca0132_init_tuning_defaults(codec); #endif @@ -6948,11 +7811,11 @@ static void sbz_region2_exit(struct hda_codec *codec) for (i = 0; i < 8; i++) writeb(0xb3, spec->mem_base + 0x304); - ca0132_mmio_gpio_set(codec, 0, false); - ca0132_mmio_gpio_set(codec, 1, false); - ca0132_mmio_gpio_set(codec, 4, true); - ca0132_mmio_gpio_set(codec, 5, false); - ca0132_mmio_gpio_set(codec, 7, false); + ca0113_mmio_gpio_set(codec, 0, false); + ca0113_mmio_gpio_set(codec, 1, false); + ca0113_mmio_gpio_set(codec, 4, true); + ca0113_mmio_gpio_set(codec, 5, false); + ca0113_mmio_gpio_set(codec, 7, false); } static void sbz_set_pin_ctl_default(struct hda_codec *codec) @@ -6995,6 +7858,16 @@ static void sbz_gpio_shutdown_commands(struct hda_codec *codec, int dir, AC_VERB_SET_GPIO_DATA, data); } +static void zxr_dbpro_power_state_shutdown(struct hda_codec *codec) +{ + hda_nid_t pins[7] = {0x05, 0x0c, 0x09, 0x0e, 0x08, 0x11, 0x01}; + unsigned int i; + + for (i = 0; i < 7; i++) + snd_hda_codec_write(codec, pins[i], 0, + AC_VERB_SET_POWER_STATE, 0x03); +} + static void sbz_exit_chip(struct hda_codec *codec) { chipio_set_stream_control(codec, 0x03, 0); @@ -7037,6 +7910,61 @@ static void r3d_exit_chip(struct hda_codec *codec) snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x5b); } +static void ae5_exit_chip(struct hda_codec *codec) +{ + chipio_set_stream_control(codec, 0x03, 0); + chipio_set_stream_control(codec, 0x04, 0); + + ca0113_mmio_command_set(codec, 0x30, 0x32, 0x3f); + ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83); + ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83); + ca0113_mmio_command_set(codec, 0x30, 0x30, 0x00); + ca0113_mmio_command_set(codec, 0x30, 0x2b, 0x00); + ca0113_mmio_command_set(codec, 0x30, 0x2d, 0x00); + ca0113_mmio_gpio_set(codec, 0, false); + ca0113_mmio_gpio_set(codec, 1, false); + + snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00); + snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x53); + + chipio_set_control_param(codec, CONTROL_PARAM_ASI, 0); + + chipio_set_stream_control(codec, 0x18, 0); + chipio_set_stream_control(codec, 0x0c, 0); + + snd_hda_codec_write(codec, 0x01, 0, 0x724, 0x83); +} + +static void zxr_exit_chip(struct hda_codec *codec) +{ + chipio_set_stream_control(codec, 0x03, 0); + chipio_set_stream_control(codec, 0x04, 0); + chipio_set_stream_control(codec, 0x14, 0); + chipio_set_stream_control(codec, 0x0C, 0); + + chipio_set_conn_rate(codec, 0x41, SR_192_000); + chipio_set_conn_rate(codec, 0x91, SR_192_000); + + chipio_write(codec, 0x18a020, 0x00000083); + + snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00); + snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x53); + + ca0132_clear_unsolicited(codec); + sbz_set_pin_ctl_default(codec); + snd_hda_codec_write(codec, 0x0B, 0, AC_VERB_SET_EAPD_BTLENABLE, 0x00); + + ca0113_mmio_gpio_set(codec, 5, false); + ca0113_mmio_gpio_set(codec, 2, false); + ca0113_mmio_gpio_set(codec, 3, false); + ca0113_mmio_gpio_set(codec, 0, false); + ca0113_mmio_gpio_set(codec, 4, true); + ca0113_mmio_gpio_set(codec, 0, true); + ca0113_mmio_gpio_set(codec, 5, true); + ca0113_mmio_gpio_set(codec, 2, false); + ca0113_mmio_gpio_set(codec, 3, false); +} + static void ca0132_exit_chip(struct hda_codec *codec) { /* put any chip cleanup stuffs here. */ @@ -7140,11 +8068,6 @@ static void sbz_pre_dsp_setup(struct hda_codec *codec) writel(0x00820680, spec->mem_base + 0x01C); writel(0x00820680, spec->mem_base + 0x01C); - snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfc); - snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfd); - snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfe); - snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xff); - chipio_write(codec, 0x18b0a4, 0x000000c2); snd_hda_codec_write(codec, 0x11, 0, @@ -7153,12 +8076,6 @@ static void sbz_pre_dsp_setup(struct hda_codec *codec) static void r3d_pre_dsp_setup(struct hda_codec *codec) { - - snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfc); - snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfd); - snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfe); - snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xff); - chipio_write(codec, 0x18b0a4, 0x000000c2); snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, @@ -7205,23 +8122,116 @@ static void ca0132_mmio_init(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; - writel(0x00000000, spec->mem_base + 0x400); - writel(0x00000000, spec->mem_base + 0x408); - writel(0x00000000, spec->mem_base + 0x40C); - writel(0x00880680, spec->mem_base + 0x01C); - writel(0x00000083, spec->mem_base + 0xC0C); + if (ca0132_quirk(spec) == QUIRK_AE5) + writel(0x00000001, spec->mem_base + 0x400); + else + writel(0x00000000, spec->mem_base + 0x400); + + if (ca0132_quirk(spec) == QUIRK_AE5) + writel(0x00000001, spec->mem_base + 0x408); + else + writel(0x00000000, spec->mem_base + 0x408); + + if (ca0132_quirk(spec) == QUIRK_AE5) + writel(0x00000001, spec->mem_base + 0x40c); + else + writel(0x00000000, spec->mem_base + 0x40C); + + if (ca0132_quirk(spec) == QUIRK_ZXR) + writel(0x00880640, spec->mem_base + 0x01C); + else + writel(0x00880680, spec->mem_base + 0x01C); + + if (ca0132_quirk(spec) == QUIRK_AE5) + writel(0x00000080, spec->mem_base + 0xC0C); + else + writel(0x00000083, spec->mem_base + 0xC0C); + writel(0x00000030, spec->mem_base + 0xC00); writel(0x00000000, spec->mem_base + 0xC04); + + if (ca0132_quirk(spec) == QUIRK_AE5) + writel(0x00000000, spec->mem_base + 0xC0C); + else + writel(0x00000003, spec->mem_base + 0xC0C); + writel(0x00000003, spec->mem_base + 0xC0C); writel(0x00000003, spec->mem_base + 0xC0C); writel(0x00000003, spec->mem_base + 0xC0C); - writel(0x00000003, spec->mem_base + 0xC0C); - writel(0x000000C1, spec->mem_base + 0xC08); + + if (ca0132_quirk(spec) == QUIRK_AE5) + writel(0x00000001, spec->mem_base + 0xC08); + else + writel(0x000000C1, spec->mem_base + 0xC08); + writel(0x000000F1, spec->mem_base + 0xC08); writel(0x00000001, spec->mem_base + 0xC08); writel(0x000000C7, spec->mem_base + 0xC08); writel(0x000000C1, spec->mem_base + 0xC08); writel(0x00000080, spec->mem_base + 0xC04); + + if (ca0132_quirk(spec) == QUIRK_AE5) { + writel(0x00000000, spec->mem_base + 0x42c); + writel(0x00000000, spec->mem_base + 0x46c); + writel(0x00000000, spec->mem_base + 0x4ac); + writel(0x00000000, spec->mem_base + 0x4ec); + writel(0x00000000, spec->mem_base + 0x43c); + writel(0x00000000, spec->mem_base + 0x47c); + writel(0x00000000, spec->mem_base + 0x4bc); + writel(0x00000000, spec->mem_base + 0x4fc); + writel(0x00000600, spec->mem_base + 0x100); + writel(0x00000014, spec->mem_base + 0x410); + writel(0x0000060f, spec->mem_base + 0x100); + writel(0x0000070f, spec->mem_base + 0x100); + writel(0x00000aff, spec->mem_base + 0x830); + writel(0x00000000, spec->mem_base + 0x86c); + writel(0x0000006b, spec->mem_base + 0x800); + writel(0x00000001, spec->mem_base + 0x86c); + writel(0x0000006b, spec->mem_base + 0x800); + writel(0x00000057, spec->mem_base + 0x804); + writel(0x00800000, spec->mem_base + 0x20c); + } +} + +/* + * This function writes to some SFR's, does some region2 writes, and then + * eventually resets the codec with the 0x7ff verb. Not quite sure why it does + * what it does. + */ +static void ae5_register_set(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + chipio_8051_write_direct(codec, 0x93, 0x10); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x44); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc2); + + writeb(0x0f, spec->mem_base + 0x304); + writeb(0x0f, spec->mem_base + 0x304); + writeb(0x0f, spec->mem_base + 0x304); + writeb(0x0f, spec->mem_base + 0x304); + writeb(0x0e, spec->mem_base + 0x100); + writeb(0x1f, spec->mem_base + 0x304); + writeb(0x0c, spec->mem_base + 0x100); + writeb(0x3f, spec->mem_base + 0x304); + writeb(0x08, spec->mem_base + 0x100); + writeb(0x7f, spec->mem_base + 0x304); + writeb(0x00, spec->mem_base + 0x100); + writeb(0xff, spec->mem_base + 0x304); + + ca0113_mmio_command_set(codec, 0x30, 0x2d, 0x3f); + + chipio_8051_write_direct(codec, 0x90, 0x00); + chipio_8051_write_direct(codec, 0x90, 0x10); + + ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83); + + chipio_write(codec, 0x18b0a4, 0x000000c2); + + snd_hda_codec_write(codec, 0x01, 0, 0x7ff, 0x00); + snd_hda_codec_write(codec, 0x01, 0, 0x7ff, 0x00); } /* @@ -7235,7 +8245,7 @@ static void ca0132_alt_init(struct hda_codec *codec) ca0132_alt_vol_setup(codec); - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: codec_dbg(codec, "SBZ alt_init"); ca0132_gpio_init(codec); @@ -7257,6 +8267,23 @@ static void ca0132_alt_init(struct hda_codec *codec) snd_hda_sequence_write(codec, spec->chip_init_verbs); snd_hda_sequence_write(codec, spec->desktop_init_verbs); break; + case QUIRK_AE5: + ca0132_gpio_init(codec); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x49); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PLL_PMU_WRITE, 0x88); + chipio_write(codec, 0x18b030, 0x00000020); + snd_hda_sequence_write(codec, spec->chip_init_verbs); + snd_hda_sequence_write(codec, spec->desktop_init_verbs); + ca0113_mmio_command_set(codec, 0x30, 0x32, 0x3f); + break; + case QUIRK_ZXR: + snd_hda_sequence_write(codec, spec->chip_init_verbs); + snd_hda_sequence_write(codec, spec->desktop_init_verbs); + break; + default: + break; } } @@ -7283,7 +8310,7 @@ static int ca0132_init(struct hda_codec *codec) spec->dsp_reload = true; spec->dsp_state = DSP_DOWNLOAD_INIT; } else { - if (spec->quirk == QUIRK_SBZ) + if (ca0132_quirk(spec) == QUIRK_SBZ) sbz_dsp_startup_check(codec); return 0; } @@ -7293,32 +8320,39 @@ static int ca0132_init(struct hda_codec *codec) spec->dsp_state = DSP_DOWNLOAD_INIT; spec->curr_chip_addx = INVALID_CHIP_ADDRESS; - if (spec->use_pci_mmio) + if (ca0132_use_pci_mmio(spec)) ca0132_mmio_init(codec); snd_hda_power_up_pm(codec); + if (ca0132_quirk(spec) == QUIRK_AE5) + ae5_register_set(codec); + ca0132_init_unsol(codec); ca0132_init_params(codec); ca0132_init_flags(codec); snd_hda_sequence_write(codec, spec->base_init_verbs); - if (spec->use_alt_functions) + if (ca0132_use_alt_functions(spec)) ca0132_alt_init(codec); ca0132_download_dsp(codec); ca0132_refresh_widget_caps(codec); - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_R3DI: case QUIRK_R3D: r3d_setup_defaults(codec); break; case QUIRK_SBZ: + case QUIRK_ZXR: sbz_setup_defaults(codec); break; + case QUIRK_AE5: + ae5_setup_defaults(codec); + break; default: ca0132_setup_defaults(codec); ca0132_init_analog_mic2(codec); @@ -7336,7 +8370,7 @@ static int ca0132_init(struct hda_codec *codec) init_input(codec, cfg->dig_in_pin, spec->dig_in); - if (!spec->use_alt_functions) { + if (!ca0132_use_alt_functions(spec)) { snd_hda_sequence_write(codec, spec->chip_init_verbs); snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, VENDOR_CHIPIO_PARAM_EX_ID_SET, 0x0D); @@ -7344,11 +8378,11 @@ static int ca0132_init(struct hda_codec *codec) VENDOR_CHIPIO_PARAM_EX_VALUE_SET, 0x20); } - if (spec->quirk == QUIRK_SBZ) + if (ca0132_quirk(spec) == QUIRK_SBZ) ca0132_gpio_setup(codec); snd_hda_sequence_write(codec, spec->spec_init_verbs); - if (spec->use_alt_functions) { + if (ca0132_use_alt_functions(spec)) { ca0132_alt_select_out(codec); ca0132_alt_select_in(codec); } else { @@ -7372,30 +8406,65 @@ static int ca0132_init(struct hda_codec *codec) return 0; } +static int dbpro_init(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + unsigned int i; + + init_output(codec, cfg->dig_out_pins[0], spec->dig_out); + init_input(codec, cfg->dig_in_pin, spec->dig_in); + + for (i = 0; i < spec->num_inputs; i++) + init_input(codec, spec->input_pins[i], spec->adcs[i]); + + return 0; +} + static void ca0132_free(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; cancel_delayed_work_sync(&spec->unsol_hp_work); snd_hda_power_up(codec); - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: sbz_exit_chip(codec); break; + case QUIRK_ZXR: + zxr_exit_chip(codec); + break; case QUIRK_R3D: r3d_exit_chip(codec); break; + case QUIRK_AE5: + ae5_exit_chip(codec); + break; case QUIRK_R3DI: r3di_gpio_shutdown(codec); break; + default: + break; } snd_hda_sequence_write(codec, spec->base_exit_verbs); ca0132_exit_chip(codec); snd_hda_power_down(codec); +#ifdef CONFIG_PCI if (spec->mem_base) pci_iounmap(codec->bus->pci, spec->mem_base); +#endif + kfree(spec->spec_init_verbs); + kfree(codec->spec); +} + +static void dbpro_free(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + zxr_dbpro_power_state_shutdown(codec); + kfree(spec->spec_init_verbs); kfree(codec->spec); } @@ -7414,6 +8483,13 @@ static const struct hda_codec_ops ca0132_patch_ops = { .reboot_notify = ca0132_reboot_notify, }; +static const struct hda_codec_ops dbpro_patch_ops = { + .build_controls = dbpro_build_controls, + .build_pcms = dbpro_build_pcms, + .init = dbpro_init, + .free = dbpro_free, +}; + static void ca0132_config(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; @@ -7425,16 +8501,42 @@ static void ca0132_config(struct hda_codec *codec) spec->multiout.dac_nids = spec->dacs; spec->multiout.num_dacs = 3; - if (!spec->use_alt_functions) + if (!ca0132_use_alt_functions(spec)) spec->multiout.max_channels = 2; else spec->multiout.max_channels = 6; - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_ALIENWARE: - codec_dbg(codec, "ca0132_config: QUIRK_ALIENWARE applied.\n"); + codec_dbg(codec, "%s: QUIRK_ALIENWARE applied.\n", __func__); snd_hda_apply_pincfgs(codec, alienware_pincfgs); + break; + case QUIRK_SBZ: + codec_dbg(codec, "%s: QUIRK_SBZ applied.\n", __func__); + snd_hda_apply_pincfgs(codec, sbz_pincfgs); + break; + case QUIRK_ZXR: + codec_dbg(codec, "%s: QUIRK_ZXR applied.\n", __func__); + snd_hda_apply_pincfgs(codec, zxr_pincfgs); + break; + case QUIRK_R3D: + codec_dbg(codec, "%s: QUIRK_R3D applied.\n", __func__); + snd_hda_apply_pincfgs(codec, r3d_pincfgs); + break; + case QUIRK_R3DI: + codec_dbg(codec, "%s: QUIRK_R3DI applied.\n", __func__); + snd_hda_apply_pincfgs(codec, r3di_pincfgs); + break; + case QUIRK_AE5: + codec_dbg(codec, "%s: QUIRK_AE5 applied.\n", __func__); + snd_hda_apply_pincfgs(codec, ae5_pincfgs); + break; + default: + break; + } + switch (ca0132_quirk(spec)) { + case QUIRK_ALIENWARE: spec->num_outputs = 2; spec->out_pins[0] = 0x0b; /* speaker out */ spec->out_pins[1] = 0x0f; @@ -7454,15 +8556,6 @@ static void ca0132_config(struct hda_codec *codec) break; case QUIRK_SBZ: case QUIRK_R3D: - if (spec->quirk == QUIRK_SBZ) { - codec_dbg(codec, "%s: QUIRK_SBZ applied.\n", __func__); - snd_hda_apply_pincfgs(codec, sbz_pincfgs); - } - if (spec->quirk == QUIRK_R3D) { - codec_dbg(codec, "%s: QUIRK_R3D applied.\n", __func__); - snd_hda_apply_pincfgs(codec, r3d_pincfgs); - } - spec->num_outputs = 2; spec->out_pins[0] = 0x0B; /* Line out */ spec->out_pins[1] = 0x0F; /* Rear headphone out */ @@ -7487,10 +8580,62 @@ static void ca0132_config(struct hda_codec *codec) spec->multiout.dig_out_nid = spec->dig_out; spec->dig_in = 0x09; break; - case QUIRK_R3DI: - codec_dbg(codec, "%s: QUIRK_R3DI applied.\n", __func__); - snd_hda_apply_pincfgs(codec, r3di_pincfgs); + case QUIRK_ZXR: + spec->num_outputs = 2; + spec->out_pins[0] = 0x0B; /* Line out */ + spec->out_pins[1] = 0x0F; /* Rear headphone out */ + spec->out_pins[2] = 0x10; /* Center/LFE */ + spec->out_pins[3] = 0x11; /* Rear surround */ + spec->shared_out_nid = 0x2; + spec->unsol_tag_hp = spec->out_pins[1]; + spec->unsol_tag_front_hp = spec->out_pins[2]; + + spec->adcs[0] = 0x7; /* Rear Mic / Line-in */ + spec->adcs[1] = 0x8; /* Not connected, no front mic */ + spec->adcs[2] = 0xa; /* what u hear */ + spec->num_inputs = 2; + spec->input_pins[0] = 0x12; /* Rear Mic / Line-in */ + spec->input_pins[1] = 0x13; /* What U Hear */ + spec->shared_mic_nid = 0x7; + spec->unsol_tag_amic1 = spec->input_pins[0]; + break; + case QUIRK_ZXR_DBPRO: + spec->adcs[0] = 0x8; /* ZxR DBPro Aux In */ + + spec->num_inputs = 1; + spec->input_pins[0] = 0x11; /* RCA Line-in */ + + spec->dig_out = 0x05; + spec->multiout.dig_out_nid = spec->dig_out; + + spec->dig_in = 0x09; + break; + case QUIRK_AE5: + spec->num_outputs = 2; + spec->out_pins[0] = 0x0B; /* Line out */ + spec->out_pins[1] = 0x11; /* Rear headphone out */ + spec->out_pins[2] = 0x10; /* Front Headphone / Center/LFE*/ + spec->out_pins[3] = 0x0F; /* Rear surround */ + spec->shared_out_nid = 0x2; + spec->unsol_tag_hp = spec->out_pins[1]; + spec->unsol_tag_front_hp = spec->out_pins[2]; + + spec->adcs[0] = 0x7; /* Rear Mic / Line-in */ + spec->adcs[1] = 0x8; /* Front Mic, but only if no DSP */ + spec->adcs[2] = 0xa; /* what u hear */ + + spec->num_inputs = 2; + spec->input_pins[0] = 0x12; /* Rear Mic / Line-in */ + spec->input_pins[1] = 0x13; /* What U Hear */ + spec->shared_mic_nid = 0x7; + spec->unsol_tag_amic1 = spec->input_pins[0]; + + /* SPDIF I/O */ + spec->dig_out = 0x05; + spec->multiout.dig_out_nid = spec->dig_out; + break; + case QUIRK_R3DI: spec->num_outputs = 2; spec->out_pins[0] = 0x0B; /* Line out */ spec->out_pins[1] = 0x0F; /* Rear headphone out */ @@ -7547,7 +8692,11 @@ static int ca0132_prepare_verbs(struct hda_codec *codec) struct ca0132_spec *spec = codec->spec; spec->chip_init_verbs = ca0132_init_verbs0; - if (spec->quirk == QUIRK_SBZ || spec->quirk == QUIRK_R3D) + /* + * Since desktop cards use pci_mmio, this can be used to determine + * whether or not to use these verbs instead of a separate bool. + */ + if (ca0132_use_pci_mmio(spec)) spec->desktop_init_verbs = ca0132_init_verbs1; spec->spec_init_verbs = kcalloc(NUM_SPEC_VERBS, sizeof(struct hda_verb), @@ -7579,6 +8728,29 @@ static int ca0132_prepare_verbs(struct hda_codec *codec) return 0; } +/* + * The Sound Blaster ZxR shares the same PCI subsystem ID as some regular + * Sound Blaster Z cards. However, they have different HDA codec subsystem + * ID's. So, we check for the ZxR's subsystem ID, as well as the DBPro + * daughter boards ID. + */ +static void sbz_detect_quirk(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + switch (codec->core.subsystem_id) { + case 0x11020033: + spec->quirk = QUIRK_ZXR; + break; + case 0x1102003f: + spec->quirk = QUIRK_ZXR_DBPRO; + break; + default: + spec->quirk = QUIRK_SBZ; + break; + } +} + static int patch_ca0132(struct hda_codec *codec) { struct ca0132_spec *spec; @@ -7593,26 +8765,39 @@ static int patch_ca0132(struct hda_codec *codec) codec->spec = spec; spec->codec = codec; - codec->patch_ops = ca0132_patch_ops; - codec->pcm_format_first = 1; - codec->no_sticky_stream = 1; - /* Detect codec quirk */ quirk = snd_pci_quirk_lookup(codec->bus->pci, ca0132_quirks); if (quirk) spec->quirk = quirk->value; else spec->quirk = QUIRK_NONE; + if (ca0132_quirk(spec) == QUIRK_SBZ) + sbz_detect_quirk(codec); + + if (ca0132_quirk(spec) == QUIRK_ZXR_DBPRO) + codec->patch_ops = dbpro_patch_ops; + else + codec->patch_ops = ca0132_patch_ops; + + codec->pcm_format_first = 1; + codec->no_sticky_stream = 1; + spec->dsp_state = DSP_DOWNLOAD_INIT; spec->num_mixers = 1; /* Set which mixers each quirk uses. */ - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: spec->mixers[0] = desktop_mixer; snd_hda_codec_set_name(codec, "Sound Blaster Z"); break; + case QUIRK_ZXR: + spec->mixers[0] = desktop_mixer; + snd_hda_codec_set_name(codec, "Sound Blaster ZxR"); + break; + case QUIRK_ZXR_DBPRO: + break; case QUIRK_R3D: spec->mixers[0] = desktop_mixer; snd_hda_codec_set_name(codec, "Recon3D"); @@ -7621,15 +8806,21 @@ static int patch_ca0132(struct hda_codec *codec) spec->mixers[0] = r3di_mixer; snd_hda_codec_set_name(codec, "Recon3Di"); break; + case QUIRK_AE5: + spec->mixers[0] = desktop_mixer; + snd_hda_codec_set_name(codec, "Sound BlasterX AE-5"); + break; default: spec->mixers[0] = ca0132_mixer; break; } /* Setup whether or not to use alt functions/controls/pci_mmio */ - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: case QUIRK_R3D: + case QUIRK_AE5: + case QUIRK_ZXR: spec->use_alt_controls = true; spec->use_alt_functions = true; spec->use_pci_mmio = true; @@ -7646,6 +8837,7 @@ static int patch_ca0132(struct hda_codec *codec) break; } +#ifdef CONFIG_PCI if (spec->use_pci_mmio) { spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); if (spec->mem_base == NULL) { @@ -7653,6 +8845,7 @@ static int patch_ca0132(struct hda_codec *codec) spec->quirk = QUIRK_NONE; } } +#endif spec->base_init_verbs = ca0132_base_init_verbs; spec->base_exit_verbs = ca0132_base_exit_verbs; diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index a7f91be45194..64fa5a82bb9f 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -23,7 +23,7 @@ #include <linux/module.h> #include <sound/core.h> #include <sound/tlv.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 1b2195dd2b26..52642ba3e2c0 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -25,7 +25,7 @@ #include <linux/slab.h> #include <linux/module.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 3c5f2a603754..a4ee7656d9ee 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -27,7 +27,7 @@ #include <sound/core.h> #include <sound/jack.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_beep.h" @@ -923,6 +923,8 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x8079, "HP EliteBook 840 G3", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x807C, "HP EliteBook 820 G3", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x80FD, "HP ProBook 640 G2", CXT_FIXUP_HP_DOCK), + SND_PCI_QUIRK(0x103c, 0x828c, "HP EliteBook 840 G4", CXT_FIXUP_HP_DOCK), + SND_PCI_QUIRK(0x103c, 0x83b2, "HP EliteBook 840 G5", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x83b3, "HP EliteBook 830 G5", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x83d3, "HP ProBook 640 G4", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), @@ -930,6 +932,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x822e, "HP ProBook 440 G4", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x836e, "HP ProBook 455 G5", CXT_FIXUP_MUTE_LED_GPIO), + SND_PCI_QUIRK(0x103c, 0x837f, "HP ProBook 470 G5", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x8455, "HP Z2 G4", CXT_FIXUP_HP_MIC_NO_PRESENCE), diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index cb587dce67a9..46f88dc7b7e8 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -41,7 +41,7 @@ #include <sound/hdaudio.h> #include <sound/hda_i915.h> #include <sound/hda_chmap.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_jack.h" @@ -2142,7 +2142,7 @@ static int generic_hdmi_build_jack(struct hda_codec *codec, int pcm_idx) strncat(hdmi_str, " Phantom", sizeof(hdmi_str) - strlen(hdmi_str) - 1); ret = snd_hda_jack_add_kctl(codec, per_pin->pin_nid, hdmi_str, - phantom_jack); + phantom_jack, 0, NULL); if (ret < 0) return ret; jack = snd_hda_jack_tbl_get(codec, per_pin->pin_nid); @@ -2616,11 +2616,7 @@ static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid) intel_haswell_enable_all_pins(codec, true); intel_haswell_fixup_enable_dp12(codec); - /* For Haswell/Broadwell, the controller is also in the power well and - * can cover the codec power request, and so need not set this flag. - */ - if (!is_haswell(codec) && !is_broadwell(codec)) - codec->core.link_power_control = 1; + codec->display_power_control = 1; codec->patch_ops.set_power_state = haswell_set_power_state; codec->depop_delay = 0; @@ -2656,7 +2652,7 @@ static int patch_i915_byt_hdmi(struct hda_codec *codec) /* For Valleyview/Cherryview, only the display codec is in the display * power well and can use link_power ops to request/release the power. */ - codec->core.link_power_control = 1; + codec->display_power_control = 1; codec->depop_delay = 0; codec->auto_runtime_pm = 1; @@ -3834,6 +3830,10 @@ HDA_CODEC_ENTRY(0x10de0020, "Tegra30 HDMI", patch_tegra_hdmi), HDA_CODEC_ENTRY(0x10de0022, "Tegra114 HDMI", patch_tegra_hdmi), HDA_CODEC_ENTRY(0x10de0028, "Tegra124 HDMI", patch_tegra_hdmi), HDA_CODEC_ENTRY(0x10de0029, "Tegra210 HDMI/DP", patch_tegra_hdmi), +HDA_CODEC_ENTRY(0x10de002d, "Tegra186 HDMI/DP0", patch_tegra_hdmi), +HDA_CODEC_ENTRY(0x10de002e, "Tegra186 HDMI/DP1", patch_tegra_hdmi), +HDA_CODEC_ENTRY(0x10de002f, "Tegra194 HDMI/DP2", patch_tegra_hdmi), +HDA_CODEC_ENTRY(0x10de0030, "Tegra194 HDMI/DP3", patch_tegra_hdmi), HDA_CODEC_ENTRY(0x10de0040, "GPU 40 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0041, "GPU 41 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0042, "GPU 42 HDMI/DP", patch_nvhdmi), diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e58537e13ad3..1ffa36e987b4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -32,7 +32,7 @@ #include <linux/input.h> #include <sound/core.h> #include <sound/jack.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" @@ -117,6 +117,7 @@ struct alc_spec { int codec_variant; /* flag for other variants */ unsigned int has_alc5505_dsp:1; unsigned int no_depop_delay:1; + unsigned int done_hp_init:1; /* for PLL fix */ hda_nid_t pll_nid; @@ -388,6 +389,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0285: case 0x10ec0298: case 0x10ec0289: + case 0x10ec0300: alc_update_coef_idx(codec, 0x10, 1<<9, 0); break; case 0x10ec0275: @@ -513,6 +515,15 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type) } } +/* get a primary headphone pin if available */ +static hda_nid_t alc_get_hp_pin(struct alc_spec *spec) +{ + if (spec->gen.autocfg.hp_pins[0]) + return spec->gen.autocfg.hp_pins[0]; + if (spec->gen.autocfg.line_out_type == AC_JACK_HP_OUT) + return spec->gen.autocfg.line_out_pins[0]; + return 0; +} /* * Realtek SSID verification @@ -723,9 +734,7 @@ do_sku: * 15 : 1 --> enable the function "Mute internal speaker * when the external headphone out jack is plugged" */ - if (!spec->gen.autocfg.hp_pins[0] && - !(spec->gen.autocfg.line_out_pins[0] && - spec->gen.autocfg.line_out_type == AUTO_PIN_HP_OUT)) { + if (!alc_get_hp_pin(spec)) { hda_nid_t nid; tmp = (ass >> 11) & 0x3; /* HP to chassis */ nid = ports[tmp]; @@ -1846,6 +1855,8 @@ enum { ALC887_FIXUP_BASS_CHMAP, ALC1220_FIXUP_GB_DUAL_CODECS, ALC1220_FIXUP_CLEVO_P950, + ALC1220_FIXUP_SYSTEM76_ORYP5, + ALC1220_FIXUP_SYSTEM76_ORYP5_PINS, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -2047,6 +2058,17 @@ static void alc1220_fixup_clevo_p950(struct hda_codec *codec, snd_hda_override_conn_list(codec, 0x1b, 1, conn1); } +static void alc_fixup_headset_mode_no_hp_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action); + +static void alc1220_fixup_system76_oryp5(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + alc1220_fixup_clevo_p950(codec, fix, action); + alc_fixup_headset_mode_no_hp_mic(codec, fix, action); +} + static const struct hda_fixup alc882_fixups[] = { [ALC882_FIXUP_ABIT_AW9D_MAX] = { .type = HDA_FIXUP_PINS, @@ -2291,6 +2313,19 @@ static const struct hda_fixup alc882_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc1220_fixup_clevo_p950, }, + [ALC1220_FIXUP_SYSTEM76_ORYP5] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc1220_fixup_system76_oryp5, + }, + [ALC1220_FIXUP_SYSTEM76_ORYP5_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + {} + }, + .chained = true, + .chain_id = ALC1220_FIXUP_SYSTEM76_ORYP5, + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -2367,6 +2402,8 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x9501, "Clevo P950HR", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1558, 0x95e1, "Clevo P95xER", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1558, 0x95e2, "Clevo P950ER", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1558, 0x96e1, "System76 Oryx Pro (oryp5)", ALC1220_FIXUP_SYSTEM76_ORYP5_PINS), + SND_PCI_QUIRK(0x1558, 0x97e1, "System76 Oryx Pro (oryp5)", ALC1220_FIXUP_SYSTEM76_ORYP5_PINS), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", ALC882_FIXUP_LENOVO_Y530), @@ -2830,6 +2867,7 @@ enum { ALC269_TYPE_ALC215, ALC269_TYPE_ALC225, ALC269_TYPE_ALC294, + ALC269_TYPE_ALC300, ALC269_TYPE_ALC700, }; @@ -2864,6 +2902,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) case ALC269_TYPE_ALC215: case ALC269_TYPE_ALC225: case ALC269_TYPE_ALC294: + case ALC269_TYPE_ALC300: case ALC269_TYPE_ALC700: ssids = alc269_ssids; break; @@ -2955,7 +2994,7 @@ static void alc282_restore_default_value(struct hda_codec *codec) static void alc282_init(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp_pin_sense; int coef78; @@ -2992,7 +3031,7 @@ static void alc282_init(struct hda_codec *codec) static void alc282_shutup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp_pin_sense; int coef78; @@ -3070,14 +3109,9 @@ static void alc283_restore_default_value(struct hda_codec *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]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp_pin_sense; - if (!spec->gen.autocfg.hp_outs) { - if (spec->gen.autocfg.line_out_type == AC_JACK_HP_OUT) - hp_pin = spec->gen.autocfg.line_out_pins[0]; - } - alc283_restore_default_value(codec); if (!hp_pin) @@ -3111,14 +3145,9 @@ static void alc283_init(struct hda_codec *codec) 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]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp_pin_sense; - if (!spec->gen.autocfg.hp_outs) { - if (spec->gen.autocfg.line_out_type == AC_JACK_HP_OUT) - hp_pin = spec->gen.autocfg.line_out_pins[0]; - } - if (!hp_pin) { alc269_shutup(codec); return; @@ -3152,7 +3181,7 @@ static void alc283_shutup(struct hda_codec *codec) static void alc256_init(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp_pin_sense; if (!hp_pin) @@ -3188,7 +3217,7 @@ static void alc256_init(struct hda_codec *codec) static void alc256_shutup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp_pin_sense; if (!hp_pin) { @@ -3224,7 +3253,7 @@ static void alc256_shutup(struct hda_codec *codec) static void alc225_init(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp1_pin_sense, hp2_pin_sense; if (!hp_pin) @@ -3267,7 +3296,7 @@ static void alc225_init(struct hda_codec *codec) static void alc225_shutup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp1_pin_sense, hp2_pin_sense; if (!hp_pin) { @@ -3311,7 +3340,7 @@ static void alc225_shutup(struct hda_codec *codec) static void alc_default_init(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp_pin_sense; if (!hp_pin) @@ -3340,7 +3369,7 @@ static void alc_default_init(struct hda_codec *codec) static void alc_default_shutup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp_pin_sense; if (!hp_pin) { @@ -3369,6 +3398,48 @@ static void alc_default_shutup(struct hda_codec *codec) snd_hda_shutup_pins(codec); } +static void alc294_hp_init(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t hp_pin = alc_get_hp_pin(spec); + int i, val; + + if (!hp_pin) + return; + + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + + msleep(100); + + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + + alc_update_coef_idx(codec, 0x6f, 0x000f, 0);/* Set HP depop to manual mode */ + alc_update_coefex_idx(codec, 0x58, 0x00, 0x8000, 0x8000); /* HP depop procedure start */ + + /* Wait for depop procedure finish */ + val = alc_read_coefex_idx(codec, 0x58, 0x01); + for (i = 0; i < 20 && val & 0x0080; i++) { + msleep(50); + val = alc_read_coefex_idx(codec, 0x58, 0x01); + } + /* Set HP depop to auto mode */ + alc_update_coef_idx(codec, 0x6f, 0x000f, 0x000b); + msleep(50); +} + +static void alc294_init(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->done_hp_init) { + alc294_hp_init(codec); + spec->done_hp_init = true; + } + alc_default_init(codec); +} + static void alc5505_coef_set(struct hda_codec *codec, unsigned int index_reg, unsigned int val) { @@ -4099,6 +4170,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0295: case 0x10ec0289: case 0x10ec0299: + alc_process_coef_fw(codec, alc225_pre_hsmode); alc_process_coef_fw(codec, coef0225); break; case 0x10ec0867: @@ -4733,7 +4805,7 @@ static void alc_update_headset_mode(struct hda_codec *codec) struct alc_spec *spec = codec->spec; hda_nid_t mux_pin = spec->gen.imux_pins[spec->gen.cur_mux[0]]; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); int new_headset_mode; @@ -4985,9 +5057,18 @@ static void alc_fixup_tpt470_dock(struct hda_codec *codec, { 0x19, 0x21a11010 }, /* dock mic */ { } }; + /* Assure the speaker pin to be coupled with DAC NID 0x03; otherwise + * the speaker output becomes too low by some reason on Thinkpads with + * ALC298 codec + */ + static hda_nid_t preferred_pairs[] = { + 0x14, 0x03, 0x17, 0x02, 0x21, 0x02, + 0 + }; struct alc_spec *spec = codec->spec; if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->gen.preferred_dacs = preferred_pairs; spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; snd_hda_apply_pincfgs(codec, pincfgs); } else if (action == HDA_FIXUP_ACT_INIT) { @@ -5003,7 +5084,7 @@ static void alc_fixup_tpt470_dock(struct hda_codec *codec, static void alc_shutup_dell_xps13(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - int hp_pin = spec->gen.autocfg.hp_pins[0]; + int hp_pin = alc_get_hp_pin(spec); /* Prevent pop noises when headphones are plugged in */ snd_hda_codec_write(codec, hp_pin, 0, @@ -5096,7 +5177,7 @@ static void alc271_hp_gate_mic_jack(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PROBE) { int mic_pin = find_ext_mic_pin(codec); - int hp_pin = spec->gen.autocfg.hp_pins[0]; + int hp_pin = alc_get_hp_pin(spec); if (snd_BUG_ON(!mic_pin || !hp_pin)) return; @@ -5358,6 +5439,83 @@ static void alc274_fixup_bind_dacs(struct hda_codec *codec, spec->gen.preferred_dacs = preferred_pairs; } +/* The DAC of NID 0x3 will introduce click/pop noise on headphones, so invalidate it */ +static void alc285_fixup_invalidate_dacs(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + snd_hda_override_wcaps(codec, 0x03, 0); +} + +static const struct hda_jack_keymap alc_headset_btn_keymap[] = { + { SND_JACK_BTN_0, KEY_PLAYPAUSE }, + { SND_JACK_BTN_1, KEY_VOICECOMMAND }, + { SND_JACK_BTN_2, KEY_VOLUMEUP }, + { SND_JACK_BTN_3, KEY_VOLUMEDOWN }, + {} +}; + +static void alc_headset_btn_callback(struct hda_codec *codec, + struct hda_jack_callback *jack) +{ + int report = 0; + + if (jack->unsol_res & (7 << 13)) + report |= SND_JACK_BTN_0; + + if (jack->unsol_res & (1 << 16 | 3 << 8)) + report |= SND_JACK_BTN_1; + + /* Volume up key */ + if (jack->unsol_res & (7 << 23)) + report |= SND_JACK_BTN_2; + + /* Volume down key */ + if (jack->unsol_res & (7 << 10)) + report |= SND_JACK_BTN_3; + + jack->jack->button_state = report; +} + +static void alc_fixup_headset_jack(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_jack_detect_enable_callback(codec, 0x55, + alc_headset_btn_callback); + snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false, + SND_JACK_HEADSET, alc_headset_btn_keymap); + break; + case HDA_FIXUP_ACT_INIT: + switch (codec->core.vendor_id) { + case 0x10ec0225: + case 0x10ec0295: + case 0x10ec0299: + alc_write_coef_idx(codec, 0x48, 0xd011); + alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); + alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8); + break; + case 0x10ec0236: + case 0x10ec0256: + alc_write_coef_idx(codec, 0x48, 0xd011); + alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); + break; + } + break; + } +} + +static void alc_fixup_disable_mic_vref(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) + snd_hda_codec_set_pin_target(codec, 0x19, PIN_VREFHIZ); +} + /* for hda_fixup_thinkpad_acpi() */ #include "thinkpad_helper.c" @@ -5368,9 +5526,6 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, hda_fixup_thinkpad_acpi(codec, fix, action); } -/* for dell wmi mic mute led */ -#include "dell_wmi_helper.c" - /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" @@ -5448,7 +5603,7 @@ enum { ALC292_FIXUP_TPT440_DOCK, ALC292_FIXUP_TPT440, ALC283_FIXUP_HEADSET_MIC, - ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED, + ALC255_FIXUP_MIC_MUTE_LED, ALC282_FIXUP_ASPIRE_V5_PINS, ALC280_FIXUP_HP_GPIO4, ALC286_FIXUP_HP_GPIO_LED, @@ -5470,6 +5625,7 @@ enum { ALC293_FIXUP_LENOVO_SPK_NOISE, ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, ALC255_FIXUP_DELL_SPK_NOISE, + ALC225_FIXUP_DISABLE_MIC_VREF, ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC295_FIXUP_DISABLE_DAC3, ALC280_FIXUP_HP_HEADSET_MIC, @@ -5493,8 +5649,18 @@ enum { ALC298_FIXUP_TPT470_DOCK, ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, + ALC256_FIXUP_HUAWEI_MBXP_PINS, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, + ALC285_FIXUP_LENOVO_HEADPHONE_NOISE, + ALC295_FIXUP_HP_AUTO_MUTE, + ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE, + ALC294_FIXUP_ASUS_MIC, + ALC294_FIXUP_ASUS_HEADSET_MIC, + ALC294_FIXUP_ASUS_SPK, + ALC225_FIXUP_HEADSET_JACK, + ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE, + ALC285_FIXUP_LENOVO_PC_BEEP_IN_NOISE, }; static const struct hda_fixup alc269_fixups[] = { @@ -5659,6 +5825,8 @@ static const struct hda_fixup alc269_fixups[] = { [ALC269_FIXUP_HP_MUTE_LED_MIC3] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_hp_mute_led_mic3, + .chained = true, + .chain_id = ALC295_FIXUP_HP_AUTO_MUTE }, [ALC269_FIXUP_HP_GPIO_LED] = { .type = HDA_FIXUP_FUNC, @@ -5740,7 +5908,7 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode, .chained = true, - .chain_id = ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED + .chain_id = ALC255_FIXUP_MIC_MUTE_LED }, [ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC] = { .type = HDA_FIXUP_FUNC, @@ -5764,6 +5932,24 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC256_FIXUP_HUAWEI_MBXP_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + {0x12, 0x90a60130}, + {0x13, 0x40000000}, + {0x14, 0x90170110}, + {0x18, 0x411111f0}, + {0x19, 0x04a11040}, + {0x1a, 0x411111f0}, + {0x1b, 0x90170112}, + {0x1d, 0x40759a05}, + {0x1e, 0x411111f0}, + {0x21, 0x04211020}, + { } + }, + .chained = true, + .chain_id = ALC255_FIXUP_MIC_MUTE_LED + }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_x101_headset_mic, @@ -5966,7 +6152,7 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode_alc255, .chained = true, - .chain_id = ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED + .chain_id = ALC255_FIXUP_MIC_MUTE_LED }, [ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC] = { .type = HDA_FIXUP_FUNC, @@ -6001,9 +6187,9 @@ static const struct hda_fixup alc269_fixups[] = { { }, }, }, - [ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED] = { + [ALC255_FIXUP_MIC_MUTE_LED] = { .type = HDA_FIXUP_FUNC, - .v.func = alc_fixup_dell_wmi, + .v.func = snd_hda_gen_fixup_micmute_led, }, [ALC282_FIXUP_ASPIRE_V5_PINS] = { .type = HDA_FIXUP_PINS, @@ -6062,7 +6248,7 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode_dell_alc288, .chained = true, - .chain_id = ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED + .chain_id = ALC255_FIXUP_MIC_MUTE_LED }, [ALC288_FIXUP_DELL1_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, @@ -6161,6 +6347,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE }, + [ALC225_FIXUP_DISABLE_MIC_VREF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_mic_vref, + .chained = true, + .chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE + }, [ALC225_FIXUP_DELL1_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -6170,7 +6362,7 @@ static const struct hda_fixup alc269_fixups[] = { {} }, .chained = true, - .chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE + .chain_id = ALC225_FIXUP_DISABLE_MIC_VREF }, [ALC280_FIXUP_HP_HEADSET_MIC] = { .type = HDA_FIXUP_FUNC, @@ -6362,6 +6554,79 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC285_FIXUP_LENOVO_HEADPHONE_NOISE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_invalidate_dacs, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI + }, + [ALC295_FIXUP_HP_AUTO_MUTE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_auto_mute_via_amp, + }, + [ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MIC + }, + [ALC294_FIXUP_ASUS_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x13, 0x90a60160 }, /* use as internal mic */ + { 0x19, 0x04a11120 }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, + [ALC294_FIXUP_ASUS_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1103c }, /* use as headset mic */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, + [ALC294_FIXUP_ASUS_SPK] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* Set EAPD high */ + { 0x20, AC_VERB_SET_COEF_INDEX, 0x40 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x8800 }, + { } + }, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC + }, + [ALC225_FIXUP_HEADSET_JACK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_headset_jack, + }, + [ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, + [ALC285_FIXUP_LENOVO_PC_BEEP_IN_NOISE] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* Disable PCBEEP-IN passthrough */ + { 0x20, AC_VERB_SET_COEF_INDEX, 0x36 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x57d7 }, + { } + }, + .chained = true, + .chain_id = ALC285_FIXUP_LENOVO_HEADPHONE_NOISE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -6376,7 +6641,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0762, "Acer Aspire E1-472", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS), + SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK), + SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS), SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X), @@ -6420,6 +6689,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0871, "Dell Precision 3630", ALC255_FIXUP_DELL_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0872, "Dell Precision 3630", ALC255_FIXUP_DELL_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0873, "Dell Precision 3930", ALC255_FIXUP_DUMMY_LINEOUT_VERB), + SND_PCI_QUIRK(0x1028, 0x0935, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), @@ -6490,6 +6760,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x1043, 0x10a1, "ASUS UX391UA", ALC294_FIXUP_ASUS_SPK), SND_PCI_QUIRK(0x1043, 0x10c0, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x10d0, "ASUS X540LA/X540LJ", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), @@ -6500,6 +6771,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x12e0, "ASUS X541SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), + SND_PCI_QUIRK(0x1043, 0x14a1, "ASUS UX533FD", ALC294_FIXUP_ASUS_SPK), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), @@ -6532,6 +6804,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1558, 0x1325, "System76 Darter Pro (darp5)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC233_FIXUP_LENOVO_MULTI_CODECS), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), @@ -6595,6 +6869,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), 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(0x19e5, 0x3200, "Huawei MBX", ALC255_FIXUP_MIC_MUTE_LED), + SND_PCI_QUIRK(0x19e5, 0x3201, "Huawei MBX", ALC255_FIXUP_MIC_MUTE_LED), + SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MBXP", ALC256_FIXUP_HUAWEI_MBXP_PINS), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ #if 0 @@ -6720,7 +6997,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC255_FIXUP_DELL2_MIC_NO_PRESENCE, .name = "alc255-dell2"}, {.id = ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc293-dell1"}, {.id = ALC283_FIXUP_HEADSET_MIC, .name = "alc283-headset"}, - {.id = ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED, .name = "alc255-dell-mute"}, + {.id = ALC255_FIXUP_MIC_MUTE_LED, .name = "alc255-dell-mute"}, {.id = ALC282_FIXUP_ASPIRE_V5_PINS, .name = "aspire-v5"}, {.id = ALC280_FIXUP_HP_GPIO4, .name = "hp-gpio4"}, {.id = ALC286_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"}, @@ -6740,7 +7017,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC293_FIXUP_LENOVO_SPK_NOISE, .name = "lenovo-spk-noise"}, {.id = ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, .name = "lenovo-hotkey"}, {.id = ALC255_FIXUP_DELL_SPK_NOISE, .name = "dell-spk-noise"}, - {.id = ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc255-dell1"}, + {.id = ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc225-dell1"}, {.id = ALC295_FIXUP_DISABLE_DAC3, .name = "alc295-disable-dac3"}, {.id = ALC280_FIXUP_HP_HEADSET_MIC, .name = "alc280-hp-headset"}, {.id = ALC221_FIXUP_HP_FRONT_MIC, .name = "alc221-hp-mic"}, @@ -6759,6 +7036,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC255_FIXUP_DUMMY_LINEOUT_VERB, .name = "alc255-dummy-lineout"}, {.id = ALC255_FIXUP_DELL_HEADSET_MIC, .name = "alc255-dell-headset"}, {.id = ALC295_FIXUP_HP_X360, .name = "alc295-hp-x360"}, + {.id = ALC225_FIXUP_HEADSET_JACK, .name = "alc-sense-combo"}, {} }; #define ALC225_STANDARD_PINS \ @@ -7034,6 +7312,15 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x12, 0x90a60130}, {0x19, 0x03a11020}, {0x21, 0x0321101f}), + SND_HDA_PIN_QUIRK(0x10ec0285, 0x17aa, "Lenovo", ALC285_FIXUP_LENOVO_PC_BEEP_IN_NOISE, + {0x12, 0x90a60130}, + {0x14, 0x90170110}, + {0x19, 0x04a11040}, + {0x21, 0x04211020}), + SND_HDA_PIN_QUIRK(0x10ec0286, 0x1025, "Acer", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE, + {0x12, 0x90a60130}, + {0x17, 0x90170110}, + {0x21, 0x02211020}), SND_HDA_PIN_QUIRK(0x10ec0288, 0x1028, "Dell", ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60120}, {0x14, 0x90170110}, @@ -7097,6 +7384,14 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0293, 0x1028, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_STANDARD_PINS, {0x13, 0x90a60140}), + SND_HDA_PIN_QUIRK(0x10ec0294, 0x1043, "ASUS", ALC294_FIXUP_ASUS_MIC, + {0x14, 0x90170110}, + {0x1b, 0x90a70130}, + {0x21, 0x04211020}), + SND_HDA_PIN_QUIRK(0x10ec0294, 0x1043, "ASUS", ALC294_FIXUP_ASUS_SPK, + {0x12, 0x90a60130}, + {0x17, 0x90170110}, + {0x21, 0x04211020}), SND_HDA_PIN_QUIRK(0x10ec0295, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, ALC295_STANDARD_PINS, {0x17, 0x21014020}, @@ -7294,6 +7589,11 @@ static int patch_alc269(struct hda_codec *codec) spec->codec_variant = ALC269_TYPE_ALC294; spec->gen.mixer_nid = 0; /* ALC2x4 does not have any loopback mixer path */ alc_update_coef_idx(codec, 0x6b, 0x0018, (1<<4) | (1<<3)); /* UAJ MIC Vref control by verb */ + spec->init_hook = alc294_init; + break; + case 0x10ec0300: + spec->codec_variant = ALC269_TYPE_ALC300; + spec->gen.mixer_nid = 0; /* no loopback on ALC300 */ break; case 0x10ec0700: case 0x10ec0701: @@ -7301,6 +7601,7 @@ static int patch_alc269(struct hda_codec *codec) spec->codec_variant = ALC269_TYPE_ALC700; spec->gen.mixer_nid = 0; /* ALC700 does not have any loopback mixer path */ alc_update_coef_idx(codec, 0x4a, 1 << 15, 0); /* Combo jack auto trigger control */ + spec->init_hook = alc294_init; break; } @@ -8211,6 +8512,7 @@ static const struct hda_model_fixup alc662_fixup_models[] = { {.id = ALC668_FIXUP_DELL_XPS13, .name = "dell-xps13"}, {.id = ALC662_FIXUP_ASUS_Nx50, .name = "asus-nx50"}, {.id = ALC668_FIXUP_ASUS_Nx51, .name = "asus-nx51"}, + {.id = ALC668_FIXUP_ASUS_G751, .name = "asus-g751"}, {.id = ALC891_FIXUP_HEADSET_MODE, .name = "alc891-headset"}, {.id = ALC891_FIXUP_DELL_MIC_NO_PRESENCE, .name = "alc891-headset-multi"}, {.id = ALC662_FIXUP_ACER_VERITON, .name = "acer-veriton"}, @@ -8404,6 +8706,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0295, "ALC295", patch_alc269), HDA_CODEC_ENTRY(0x10ec0298, "ALC298", patch_alc269), HDA_CODEC_ENTRY(0x10ec0299, "ALC299", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0300, "ALC300", patch_alc269), HDA_CODEC_REV_ENTRY(0x10ec0861, 0x100340, "ALC660", patch_alc861), HDA_CODEC_ENTRY(0x10ec0660, "ALC660-VD", patch_alc861vd), HDA_CODEC_ENTRY(0x10ec0861, "ALC861", patch_alc861), diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index f63acb1b965c..c49d25bcd7f2 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -27,7 +27,7 @@ #include <linux/slab.h> #include <linux/module.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" /* si3054 verbs */ diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 046705b4691a..1b6ecfb01759 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -32,7 +32,7 @@ #include <linux/module.h> #include <sound/core.h> #include <sound/jack.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_beep.h" @@ -77,6 +77,7 @@ enum { STAC_DELL_M6_BOTH, STAC_DELL_EQ, STAC_ALIENWARE_M17X, + STAC_ELO_VUPOINT_15MX, STAC_92HD89XX_HP_FRONT_JACK, STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK, STAC_92HD73XX_ASUS_MOBO, @@ -1879,6 +1880,18 @@ static void stac92hd73xx_fixup_no_jd(struct hda_codec *codec, codec->no_jack_detect = 1; } + +static void stac92hd73xx_disable_automute(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct sigmatel_spec *spec = codec->spec; + + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + spec->gen.suppress_auto_mute = 1; +} + static const struct hda_fixup stac92hd73xx_fixups[] = { [STAC_92HD73XX_REF] = { .type = HDA_FIXUP_FUNC, @@ -1904,6 +1917,10 @@ static const struct hda_fixup stac92hd73xx_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = stac92hd73xx_fixup_alienware_m17x, }, + [STAC_ELO_VUPOINT_15MX] = { + .type = HDA_FIXUP_FUNC, + .v.func = stac92hd73xx_disable_automute, + }, [STAC_92HD73XX_INTEL] = { .type = HDA_FIXUP_PINS, .v.pins = intel_dg45id_pin_configs, @@ -1942,6 +1959,7 @@ static const struct hda_model_fixup stac92hd73xx_models[] = { { .id = STAC_DELL_M6_BOTH, .name = "dell-m6" }, { .id = STAC_DELL_EQ, .name = "dell-eq" }, { .id = STAC_ALIENWARE_M17X, .name = "alienware" }, + { .id = STAC_ELO_VUPOINT_15MX, .name = "elo-vupoint-15mx" }, { .id = STAC_92HD73XX_ASUS_MOBO, .name = "asus-mobo" }, {} }; @@ -1991,6 +2009,8 @@ static const struct snd_pci_quirk stac92hd73xx_fixup_tbl[] = { "Alienware M17x", STAC_ALIENWARE_M17X), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0490, "Alienware M17x R3", STAC_DELL_EQ), + SND_PCI_QUIRK(0x1059, 0x1011, + "ELO VuPoint 15MX", STAC_ELO_VUPOINT_15MX), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1927, "HP Z1 G2", STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x2b17, diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 6b9617aee0e6..9f6f13e25145 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -52,7 +52,7 @@ #include <linux/module.h> #include <sound/core.h> #include <sound/asoundef.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c index 568575b72f2f..4089feb8c68e 100644 --- a/sound/pci/hda/thinkpad_helper.c +++ b/sound/pci/hda/thinkpad_helper.c @@ -3,12 +3,11 @@ * to be included from codec driver */ -#if IS_ENABLED(CONFIG_THINKPAD_ACPI) +#if IS_ENABLED(CONFIG_THINKPAD_ACPI) && IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO) #include <linux/acpi.h> -#include <linux/thinkpad_acpi.h> +#include <linux/leds.h> -static int (*led_set_func)(int, bool); static void (*old_vmaster_hook)(void *, int); static bool is_thinkpad(struct hda_codec *codec) @@ -23,50 +22,20 @@ static void update_tpacpi_mute_led(void *private_data, int enabled) if (old_vmaster_hook) old_vmaster_hook(private_data, enabled); - if (led_set_func) - led_set_func(TPACPI_LED_MUTE, !enabled); -} - -static void update_tpacpi_micmute(struct hda_codec *codec) -{ - struct hda_gen_spec *spec = codec->spec; - - led_set_func(TPACPI_LED_MICMUTE, spec->micmute_led.led_value); + ledtrig_audio_set(LED_AUDIO_MUTE, enabled ? LED_OFF : LED_ON); } static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct hda_gen_spec *spec = codec->spec; - bool removefunc = false; if (action == HDA_FIXUP_ACT_PROBE) { if (!is_thinkpad(codec)) return; - if (!led_set_func) - led_set_func = symbol_request(tpacpi_led_set); - if (!led_set_func) { - codec_warn(codec, - "Failed to find thinkpad-acpi symbol tpacpi_led_set\n"); - return; - } - - removefunc = true; - if (led_set_func(TPACPI_LED_MUTE, false) >= 0) { - old_vmaster_hook = spec->vmaster_mute.hook; - spec->vmaster_mute.hook = update_tpacpi_mute_led; - removefunc = false; - } - if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0 && - !snd_hda_gen_add_micmute_led(codec, - update_tpacpi_micmute)) - removefunc = false; - } - - if (led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) { - symbol_put(tpacpi_led_set); - led_set_func = NULL; - old_vmaster_hook = NULL; + old_vmaster_hook = spec->vmaster_mute.hook; + spec->vmaster_mute.hook = update_tpacpi_mute_led; + snd_hda_gen_fixup_micmute_led(codec, fix, action); } } diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 5ee468d1aefe..ffddcdfe0c66 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -38,11 +38,6 @@ #include <sound/ac97_codec.h> #include <sound/info.h> #include <sound/initval.h> -/* for 440MX workaround */ -#include <asm/pgtable.h> -#ifdef CONFIG_X86 -#include <asm/set_memory.h> -#endif MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Intel 82801AA,82901AB,i810,i820,i830,i840,i845,MX440; SiS 7012; Ali 5455"); @@ -374,7 +369,6 @@ struct ichdev { unsigned int ali_slot; /* ALI DMA slot */ struct ac97_pcm *pcm; int pcm_open_flag; - unsigned int page_attr_changed: 1; unsigned int suspended: 1; }; @@ -724,25 +718,6 @@ static void snd_intel8x0_setup_periods(struct intel8x0 *chip, struct ichdev *ich iputbyte(chip, port + ichdev->roff_sr, ICH_FIFOE | ICH_BCIS | ICH_LVBCI); } -#ifdef __i386__ -/* - * Intel 82443MX running a 100MHz processor system bus has a hardware bug, - * which aborts PCI busmaster for audio transfer. A workaround is to set - * the pages as non-cached. For details, see the errata in - * http://download.intel.com/design/chipsets/specupdt/24505108.pdf - */ -static void fill_nocache(void *buf, int size, int nocache) -{ - size = (size + PAGE_SIZE - 1) >> PAGE_SHIFT; - if (nocache) - set_pages_uc(virt_to_page(buf), size); - else - set_pages_wb(virt_to_page(buf), size); -} -#else -#define fill_nocache(buf, size, nocache) do { ; } while (0) -#endif - /* * Interrupt handler */ @@ -850,7 +825,7 @@ static int snd_intel8x0_pcm_trigger(struct snd_pcm_substream *substream, int cmd switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: ichdev->suspended = 0; - /* fallthru */ + /* fall through */ case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: val = ICH_IOCE | ICH_STARTBM; @@ -858,7 +833,7 @@ static int snd_intel8x0_pcm_trigger(struct snd_pcm_substream *substream, int cmd break; case SNDRV_PCM_TRIGGER_SUSPEND: ichdev->suspended = 1; - /* fallthru */ + /* fall through */ case SNDRV_PCM_TRIGGER_STOP: val = 0; break; @@ -892,7 +867,7 @@ static int snd_intel8x0_ali_trigger(struct snd_pcm_substream *substream, int cmd switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: ichdev->suspended = 0; - /* fallthru */ + /* fall through */ case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -909,7 +884,7 @@ static int snd_intel8x0_ali_trigger(struct snd_pcm_substream *substream, int cmd break; case SNDRV_PCM_TRIGGER_SUSPEND: ichdev->suspended = 1; - /* fallthru */ + /* fall through */ case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* pause */ @@ -938,23 +913,12 @@ static int snd_intel8x0_hw_params(struct snd_pcm_substream *substream, { struct intel8x0 *chip = snd_pcm_substream_chip(substream); struct ichdev *ichdev = get_ichdev(substream); - struct snd_pcm_runtime *runtime = substream->runtime; int dbl = params_rate(hw_params) > 48000; int err; - if (chip->fix_nocache && ichdev->page_attr_changed) { - fill_nocache(runtime->dma_area, runtime->dma_bytes, 0); /* clear */ - ichdev->page_attr_changed = 0; - } err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); if (err < 0) return err; - if (chip->fix_nocache) { - if (runtime->dma_area && ! ichdev->page_attr_changed) { - fill_nocache(runtime->dma_area, runtime->dma_bytes, 1); - ichdev->page_attr_changed = 1; - } - } if (ichdev->pcm_open_flag) { snd_ac97_pcm_close(ichdev->pcm); ichdev->pcm_open_flag = 0; @@ -974,17 +938,12 @@ static int snd_intel8x0_hw_params(struct snd_pcm_substream *substream, static int snd_intel8x0_hw_free(struct snd_pcm_substream *substream) { - struct intel8x0 *chip = snd_pcm_substream_chip(substream); struct ichdev *ichdev = get_ichdev(substream); if (ichdev->pcm_open_flag) { snd_ac97_pcm_close(ichdev->pcm); ichdev->pcm_open_flag = 0; } - if (chip->fix_nocache && ichdev->page_attr_changed) { - fill_nocache(substream->runtime->dma_area, substream->runtime->dma_bytes, 0); - ichdev->page_attr_changed = 0; - } return snd_pcm_lib_free_pages(substream); } @@ -1510,6 +1469,9 @@ struct ich_pcm_table { int ac97_idx; }; +#define intel8x0_dma_type(chip) \ + ((chip)->fix_nocache ? SNDRV_DMA_TYPE_DEV_UC : SNDRV_DMA_TYPE_DEV) + static int snd_intel8x0_pcm1(struct intel8x0 *chip, int device, struct ich_pcm_table *rec) { @@ -1540,7 +1502,7 @@ static int snd_intel8x0_pcm1(struct intel8x0 *chip, int device, strcpy(pcm->name, chip->card->shortname); chip->pcm[device] = pcm; - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_pcm_lib_preallocate_pages_for_all(pcm, intel8x0_dma_type(chip), snd_dma_pci_data(chip->pci), rec->prealloc_size, rec->prealloc_max_size); @@ -2629,11 +2591,8 @@ static int snd_intel8x0_free(struct intel8x0 *chip) __hw_end: if (chip->irq >= 0) free_irq(chip->irq, chip); - if (chip->bdbars.area) { - if (chip->fix_nocache) - fill_nocache(chip->bdbars.area, chip->bdbars.bytes, 0); + if (chip->bdbars.area) snd_dma_free_pages(&chip->bdbars); - } if (chip->addr) pci_iounmap(chip->pci, chip->addr); if (chip->bmaddr) @@ -2657,17 +2616,6 @@ static int intel8x0_suspend(struct device *dev) snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); for (i = 0; i < chip->pcm_devs; i++) snd_pcm_suspend_all(chip->pcm[i]); - /* clear nocache */ - if (chip->fix_nocache) { - for (i = 0; i < chip->bdbars_count; i++) { - struct ichdev *ichdev = &chip->ichd[i]; - if (ichdev->substream && ichdev->page_attr_changed) { - struct snd_pcm_runtime *runtime = ichdev->substream->runtime; - if (runtime->dma_area) - fill_nocache(runtime->dma_area, runtime->dma_bytes, 0); - } - } - } for (i = 0; i < chip->ncodecs; i++) snd_ac97_suspend(chip->ac97[i]); if (chip->device_type == DEVICE_INTEL_ICH4) @@ -2708,25 +2656,9 @@ static int intel8x0_resume(struct device *dev) ICH_PCM_SPDIF_1011); } - /* refill nocache */ - if (chip->fix_nocache) - fill_nocache(chip->bdbars.area, chip->bdbars.bytes, 1); - for (i = 0; i < chip->ncodecs; i++) snd_ac97_resume(chip->ac97[i]); - /* refill nocache */ - if (chip->fix_nocache) { - for (i = 0; i < chip->bdbars_count; i++) { - struct ichdev *ichdev = &chip->ichd[i]; - if (ichdev->substream && ichdev->page_attr_changed) { - struct snd_pcm_runtime *runtime = ichdev->substream->runtime; - if (runtime->dma_area) - fill_nocache(runtime->dma_area, runtime->dma_bytes, 1); - } - } - } - /* resume status */ for (i = 0; i < chip->bdbars_count; i++) { struct ichdev *ichdev = &chip->ichd[i]; @@ -3057,6 +2989,12 @@ static int snd_intel8x0_create(struct snd_card *card, chip->inside_vm = snd_intel8x0_inside_vm(pci); + /* + * Intel 82443MX running a 100MHz processor system bus has a hardware + * bug, which aborts PCI busmaster for audio transfer. A workaround + * is to set the pages as non-cached. For details, see the errata in + * http://download.intel.com/design/chipsets/specupdt/24505108.pdf + */ if (pci->vendor == PCI_VENDOR_ID_INTEL && pci->device == PCI_DEVICE_ID_INTEL_440MX) chip->fix_nocache = 1; /* enable workaround */ @@ -3128,7 +3066,7 @@ static int snd_intel8x0_create(struct snd_card *card, /* allocate buffer descriptor lists */ /* the start of each lists must be aligned to 8 bytes */ - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), + if (snd_dma_alloc_pages(intel8x0_dma_type(chip), snd_dma_pci_data(pci), chip->bdbars_count * sizeof(u32) * ICH_MAX_FRAGS * 2, &chip->bdbars) < 0) { snd_intel8x0_free(chip); @@ -3137,9 +3075,6 @@ static int snd_intel8x0_create(struct snd_card *card, } /* tables must be aligned to 8 bytes here, but the kernel pages are much bigger, so we don't care (on i386) */ - /* workaround for 440MX */ - if (chip->fix_nocache) - fill_nocache(chip->bdbars.area, chip->bdbars.bytes, 1); int_sta_masks = 0; for (i = 0; i < chip->bdbars_count; i++) { ichdev = &chip->ichd[i]; diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 943a726b1c1b..c84629190cba 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -1171,16 +1171,6 @@ static int snd_intel8x0m_create(struct snd_card *card, } port_inited: - if (request_irq(pci->irq, snd_intel8x0m_interrupt, IRQF_SHARED, - KBUILD_MODNAME, chip)) { - dev_err(card->dev, "unable to grab IRQ %d\n", pci->irq); - snd_intel8x0m_free(chip); - return -EBUSY; - } - chip->irq = pci->irq; - pci_set_master(pci); - synchronize_irq(chip->irq); - /* initialize offsets */ chip->bdbars_count = 2; tbl = intel_regs; @@ -1224,11 +1214,21 @@ static int snd_intel8x0m_create(struct snd_card *card, chip->int_sta_reg = ICH_REG_GLOB_STA; chip->int_sta_mask = int_sta_masks; + pci_set_master(pci); + if ((err = snd_intel8x0m_chip_init(chip, 1)) < 0) { snd_intel8x0m_free(chip); return err; } + if (request_irq(pci->irq, snd_intel8x0m_interrupt, IRQF_SHARED, + KBUILD_MODNAME, chip)) { + dev_err(card->dev, "unable to grab IRQ %d\n", pci->irq); + snd_intel8x0m_free(chip); + return -EBUSY; + } + chip->irq = pci->irq; + if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { snd_intel8x0m_free(chip); return err; diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index f0906ba416d4..3ac8c71d567c 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -319,7 +319,8 @@ static const struct snd_pcm_hardware snd_rme32_spdif_info = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_SYNC_START), + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_SYNC_APPLPTR), .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE), .rates = (SNDRV_PCM_RATE_32000 | @@ -346,7 +347,8 @@ static const struct snd_pcm_hardware snd_rme32_adat_info = SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_SYNC_START), + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_SYNC_APPLPTR), .formats= SNDRV_PCM_FMTBIT_S16_LE, .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000), @@ -370,7 +372,8 @@ static const struct snd_pcm_hardware snd_rme32_spdif_fd_info = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_SYNC_START), + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_SYNC_APPLPTR), .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE), .rates = (SNDRV_PCM_RATE_32000 | @@ -397,7 +400,8 @@ static const struct snd_pcm_hardware snd_rme32_adat_fd_info = SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_SYNC_START), + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_SYNC_APPLPTR), .formats= SNDRV_PCM_FMTBIT_S16_LE, .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000), @@ -1104,16 +1108,6 @@ snd_rme32_pcm_trigger(struct snd_pcm_substream *substream, int cmd) snd_pcm_trigger_done(s, substream); } - /* prefill playback buffer */ - if (cmd == SNDRV_PCM_TRIGGER_START && rme32->fullduplex_mode) { - snd_pcm_group_for_each_entry(s, substream) { - if (s == rme32->playback_substream) { - s->ops->ack(s); - break; - } - } - } - switch (cmd) { case SNDRV_PCM_TRIGGER_START: if (rme32->running && ! RME32_ISWORKING(rme32)) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 1bff4b1b39cd..ba99ff0e93e0 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -30,6 +30,7 @@ #include <linux/math64.h> #include <linux/vmalloc.h> #include <linux/io.h> +#include <linux/nospec.h> #include <sound/core.h> #include <sound/control.h> @@ -4092,15 +4093,16 @@ static int snd_hdsp_channel_info(struct snd_pcm_substream *substream, struct snd_pcm_channel_info *info) { struct hdsp *hdsp = snd_pcm_substream_chip(substream); - int mapped_channel; + unsigned int channel = info->channel; - if (snd_BUG_ON(info->channel >= hdsp->max_channels)) + if (snd_BUG_ON(channel >= hdsp->max_channels)) return -EINVAL; + channel = array_index_nospec(channel, hdsp->max_channels); - if ((mapped_channel = hdsp->channel_map[info->channel]) < 0) + if (hdsp->channel_map[channel] < 0) return -EINVAL; - info->offset = mapped_channel * HDSP_CHANNEL_BUFFER_BYTES; + info->offset = hdsp->channel_map[channel] * HDSP_CHANNEL_BUFFER_BYTES; info->first = 0; info->step = 32; return 0; diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 11b5b5e0e058..679ad0415e3b 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6534,7 +6534,7 @@ static int snd_hdspm_create_alsa_devices(struct snd_card *card, dev_dbg(card->dev, "Update mixer controls...\n"); hdspm_update_simple_mixer_controls(hdspm); - dev_dbg(card->dev, "Initializeing complete ???\n"); + dev_dbg(card->dev, "Initializing complete?\n"); err = snd_card_register(card); if (err < 0) { diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c index 48dd44f8e914..d692e4070167 100644 --- a/sound/ppc/pmac.c +++ b/sound/ppc/pmac.c @@ -908,7 +908,7 @@ static void detect_byte_swap(struct snd_pmac *chip) /* if seems that Keylargo can't byte-swap */ for (mio = chip->node->parent; mio; mio = mio->parent) { - if (strcmp(mio->name, "mac-io") == 0) { + if (of_node_name_eq(mio, "mac-io")) { if (of_device_is_compatible(mio, "Keylargo")) chip->can_byte_swap = 0; break; @@ -1313,7 +1313,7 @@ int snd_pmac_new(struct snd_card *card, struct snd_pmac **chip_return) } else if (chip->is_pbook_G3) { struct device_node* mio; for (mio = chip->node->parent; mio; mio = mio->parent) { - if (strcmp(mio->name, "mac-io") == 0) { + if (of_node_name_eq(mio, "mac-io")) { struct resource r; if (of_address_to_resource(mio, 0, &r) == 0) chip->macio_base = diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index 0779a2912237..6d7ffffcce95 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -1365,8 +1365,8 @@ int snd_pmac_tumbler_init(struct snd_pmac *chip) mix->anded_reset = 0; mix->reset_on_sleep = 1; - for (np = chip->node->child; np; np = np->sibling) { - if (!strcmp(np->name, "sound")) { + for_each_child_of_node(chip->node, np) { + if (of_node_name_eq(np, "sound")) { if (of_get_property(np, "has-anded-reset", NULL)) mix->anded_reset = 1; if (of_get_property(np, "layout-id", NULL)) diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 1cf11cf51e1d..6592a422a047 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -46,13 +46,11 @@ source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" source "sound/soc/bcm/Kconfig" source "sound/soc/cirrus/Kconfig" -source "sound/soc/davinci/Kconfig" source "sound/soc/dwc/Kconfig" source "sound/soc/fsl/Kconfig" source "sound/soc/hisilicon/Kconfig" source "sound/soc/jz4740/Kconfig" source "sound/soc/nuc900/Kconfig" -source "sound/soc/omap/Kconfig" source "sound/soc/kirkwood/Kconfig" source "sound/soc/img/Kconfig" source "sound/soc/intel/Kconfig" @@ -70,9 +68,11 @@ source "sound/soc/sti/Kconfig" source "sound/soc/stm/Kconfig" source "sound/soc/sunxi/Kconfig" source "sound/soc/tegra/Kconfig" +source "sound/soc/ti/Kconfig" source "sound/soc/txx9/Kconfig" source "sound/soc/uniphier/Kconfig" source "sound/soc/ux500/Kconfig" +source "sound/soc/xilinx/Kconfig" source "sound/soc/xtensa/Kconfig" source "sound/soc/zte/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 62a5f87c3cfc..48c48c1c893c 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -30,7 +30,6 @@ obj-$(CONFIG_SND_SOC) += atmel/ obj-$(CONFIG_SND_SOC) += au1x/ obj-$(CONFIG_SND_SOC) += bcm/ obj-$(CONFIG_SND_SOC) += cirrus/ -obj-$(CONFIG_SND_SOC) += davinci/ obj-$(CONFIG_SND_SOC) += dwc/ obj-$(CONFIG_SND_SOC) += fsl/ obj-$(CONFIG_SND_SOC) += hisilicon/ @@ -41,7 +40,6 @@ obj-$(CONFIG_SND_SOC) += mediatek/ obj-$(CONFIG_SND_SOC) += meson/ obj-$(CONFIG_SND_SOC) += mxs/ obj-$(CONFIG_SND_SOC) += nuc900/ -obj-$(CONFIG_SND_SOC) += omap/ obj-$(CONFIG_SND_SOC) += kirkwood/ obj-$(CONFIG_SND_SOC) += pxa/ obj-$(CONFIG_SND_SOC) += qcom/ @@ -54,8 +52,10 @@ obj-$(CONFIG_SND_SOC) += sti/ obj-$(CONFIG_SND_SOC) += stm/ obj-$(CONFIG_SND_SOC) += sunxi/ obj-$(CONFIG_SND_SOC) += tegra/ +obj-$(CONFIG_SND_SOC) += ti/ obj-$(CONFIG_SND_SOC) += txx9/ obj-$(CONFIG_SND_SOC) += uniphier/ obj-$(CONFIG_SND_SOC) += ux500/ +obj-$(CONFIG_SND_SOC) += xilinx/ obj-$(CONFIG_SND_SOC) += xtensa/ obj-$(CONFIG_SND_SOC) += zte/ diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 58c1dcb4d255..33ebec990c2f 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -19,3 +19,9 @@ config SND_SOC_AMD_CZ_RT5645_MACH depends on SND_SOC_AMD_ACP && I2C help This option enables machine driver for rt5645. + +config SND_SOC_AMD_ACP3x + tristate "AMD Audio Coprocessor-v3.x support" + depends on X86 && PCI + help + This option enables ACP v3.x I2S support on AMD platform diff --git a/sound/soc/amd/Makefile b/sound/soc/amd/Makefile index 79b0622fa5d3..8e1c571c3161 100644 --- a/sound/soc/amd/Makefile +++ b/sound/soc/amd/Makefile @@ -5,3 +5,4 @@ snd-soc-acp-rt5645-mach-objs := acp-rt5645.o obj-$(CONFIG_SND_SOC_AMD_ACP) += acp_audio_dma.o obj-$(CONFIG_SND_SOC_AMD_CZ_DA7219MX98357_MACH) += snd-soc-acp-da7219mx98357-mach.o obj-$(CONFIG_SND_SOC_AMD_CZ_RT5645_MACH) += snd-soc-acp-rt5645-mach.o +obj-$(CONFIG_SND_SOC_AMD_ACP3x) += raven/ diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 8e3275a96a82..a5daad973ce5 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -42,7 +42,7 @@ #include "../codecs/da7219.h" #include "../codecs/da7219-aad.h" -#define CZ_PLAT_CLK 25000000 +#define CZ_PLAT_CLK 48000000 #define DUAL_CHANNEL 2 static struct snd_soc_jack cz_jack; @@ -75,7 +75,7 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) da7219_dai_clk = clk_get(component->dev, "da7219-dai-clks"); ret = snd_soc_card_jack_new(card, "Headset Jack", - SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | + SND_JACK_HEADSET | SND_JACK_LINEOUT | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, &cz_jack, NULL, 0); @@ -133,7 +133,7 @@ static const struct snd_pcm_hw_constraint_list constraints_channels = { .mask = 0, }; -static int cz_da7219_startup(struct snd_pcm_substream *substream) +static int cz_da7219_play_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -150,7 +150,28 @@ static int cz_da7219_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_SP_INSTANCE; + machine->play_i2s_instance = I2S_SP_INSTANCE; + return da7219_clk_enable(substream); +} + +static int cz_da7219_cap_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL1; return da7219_clk_enable(substream); } @@ -162,11 +183,22 @@ static void cz_da7219_shutdown(struct snd_pcm_substream *substream) static int cz_max_startup(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_card *card = rtd->card; struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); - machine->i2s_instance = I2S_BT_INSTANCE; + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->play_i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } @@ -177,21 +209,43 @@ static void cz_max_shutdown(struct snd_pcm_substream *substream) static int cz_dmic0_startup(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_card *card = rtd->card; struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); - machine->i2s_instance = I2S_BT_INSTANCE; + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->cap_i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } static int cz_dmic1_startup(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_card *card = rtd->card; struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); - machine->i2s_instance = I2S_SP_INSTANCE; + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL0; return da7219_clk_enable(substream); } @@ -201,8 +255,13 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } +static const struct snd_soc_ops cz_da7219_play_ops = { + .startup = cz_da7219_play_startup, + .shutdown = cz_da7219_shutdown, +}; + static const struct snd_soc_ops cz_da7219_cap_ops = { - .startup = cz_da7219_startup, + .startup = cz_da7219_cap_startup, .shutdown = cz_da7219_shutdown, }; @@ -233,7 +292,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { | SND_SOC_DAIFMT_CBM_CFM, .init = cz_da7219_init, .dpcm_playback = 1, - .ops = &cz_da7219_cap_ops, + .ops = &cz_da7219_play_ops, }, { .name = "amd-da7219-cap", @@ -344,7 +403,7 @@ static struct regulator_config acp_da7219_cfg = { static struct regulator_ops acp_da7219_ops = { }; -static struct regulator_desc acp_da7219_desc = { +static const struct regulator_desc acp_da7219_desc = { .name = "reg-fixed-1.8V", .type = REGULATOR_VOLTAGE, .owner = THIS_MODULE, diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 3135e9eafd18..f4011bebc7ec 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -303,11 +303,10 @@ static void set_acp_to_i2s_dma_descriptors(void __iomem *acp_mmio, u32 size, } /* Create page table entries in ACP SRAM for the allocated memory */ -static void acp_pte_config(void __iomem *acp_mmio, struct page *pg, +static void acp_pte_config(void __iomem *acp_mmio, dma_addr_t addr, u16 num_of_pages, u32 pte_offset) { u16 page_idx; - u64 addr; u32 low; u32 high; u32 offset; @@ -317,7 +316,6 @@ static void acp_pte_config(void __iomem *acp_mmio, struct page *pg, /* Load the low address of page int ACP SRAM through SRBM */ acp_reg_write((offset + (page_idx * 8)), acp_mmio, mmACP_SRBM_Targ_Idx_Addr); - addr = page_to_phys(pg); low = lower_32_bits(addr); high = upper_32_bits(addr); @@ -333,7 +331,7 @@ static void acp_pte_config(void __iomem *acp_mmio, struct page *pg, acp_reg_write(high, acp_mmio, mmACP_SRBM_Targ_Idx_Data); /* Move to next physically contiguos page */ - pg++; + addr += PAGE_SIZE; } } @@ -343,7 +341,7 @@ static void config_acp_dma(void __iomem *acp_mmio, { u16 ch_acp_sysmem, ch_acp_i2s; - acp_pte_config(acp_mmio, rtd->pg, rtd->num_of_pages, + acp_pte_config(acp_mmio, rtd->dma_addr, rtd->num_of_pages, rtd->pte_offset); if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) { @@ -850,7 +848,6 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, int status; uint64_t size; u32 val = 0; - struct page *pg; struct snd_pcm_runtime *runtime; struct audio_substream_data *rtd; struct snd_soc_pcm_runtime *prtd = substream->private_data; @@ -867,8 +864,12 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, return -EINVAL; if (pinfo) { - rtd->i2s_instance = pinfo->i2s_instance; - rtd->capture_channel = pinfo->capture_channel; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + rtd->i2s_instance = pinfo->play_i2s_instance; + } else { + rtd->i2s_instance = pinfo->cap_i2s_instance; + rtd->capture_channel = pinfo->capture_channel; + } } if (adata->asic_type == CHIP_STONEY) { val = acp_reg_read(adata->acp_mmio, @@ -982,16 +983,14 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, return status; memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); - pg = virt_to_page(substream->dma_buffer.area); - if (pg) { + if (substream->dma_buffer.area) { acp_set_sram_bank_state(rtd->acp_mmio, 0, true); /* Save for runtime private data */ - rtd->pg = pg; + rtd->dma_addr = substream->dma_buffer.addr; rtd->order = get_order(size); /* Fill the page table entries in ACP SRAM */ - rtd->pg = pg; rtd->size = size; rtd->num_of_pages = PAGE_ALIGN(size) >> PAGE_SHIFT; rtd->direction = substream->stream; @@ -1147,18 +1146,21 @@ static int acp_dma_new(struct snd_soc_pcm_runtime *rtd) struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, DRV_NAME); struct audio_drv_data *adata = dev_get_drvdata(component->dev); + struct device *parent = component->dev->parent; switch (adata->asic_type) { case CHIP_STONEY: ret = snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, SNDRV_DMA_TYPE_DEV, - NULL, ST_MIN_BUFFER, + parent, + ST_MIN_BUFFER, ST_MAX_BUFFER); break; default: ret = snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, SNDRV_DMA_TYPE_DEV, - NULL, MIN_BUFFER, + parent, + MIN_BUFFER, MAX_BUFFER); break; } diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index be3963e8f4fa..e5ab6c6040a6 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -123,7 +123,7 @@ enum acp_dma_priority_level { }; struct audio_substream_data { - struct page *pg; + dma_addr_t dma_addr; unsigned int order; u16 num_of_pages; u16 i2s_instance; @@ -158,7 +158,8 @@ struct audio_drv_data { * and dma driver */ struct acp_platform_info { - u16 i2s_instance; + u16 play_i2s_instance; + u16 cap_i2s_instance; u16 capture_channel; }; diff --git a/sound/soc/amd/raven/Makefile b/sound/soc/amd/raven/Makefile new file mode 100644 index 000000000000..108d1acf189b --- /dev/null +++ b/sound/soc/amd/raven/Makefile @@ -0,0 +1,6 @@ +# SPDX-License-Identifier: GPL-2.0+ +# Raven Ridge platform Support +snd-pci-acp3x-objs := pci-acp3x.o +snd-acp3x-pcm-dma-objs := acp3x-pcm-dma.o +obj-$(CONFIG_SND_SOC_AMD_ACP3x) += snd-pci-acp3x.o +obj-$(CONFIG_SND_SOC_AMD_ACP3x) += snd-acp3x-pcm-dma.o diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c new file mode 100644 index 000000000000..3d58338fa3cf --- /dev/null +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -0,0 +1,779 @@ +// SPDX-License-Identifier: GPL-2.0+ +// +// AMD ALSA SoC PCM Driver +// +//Copyright 2016 Advanced Micro Devices, Inc. + +#include <linux/platform_device.h> +#include <linux/module.h> +#include <linux/err.h> +#include <linux/io.h> +#include <linux/pm_runtime.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "acp3x.h" + +#define DRV_NAME "acp3x-i2s-audio" + +struct i2s_dev_data { + bool tdm_mode; + unsigned int i2s_irq; + u32 tdm_fmt; + void __iomem *acp3x_base; + struct snd_pcm_substream *play_stream; + struct snd_pcm_substream *capture_stream; +}; + +struct i2s_stream_instance { + u16 num_pages; + u16 channels; + u32 xfer_resolution; + struct page *pg; + void __iomem *acp3x_base; +}; + +static const struct snd_pcm_hardware acp3x_pcm_hardware_playback = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_96000, + .rate_min = 8000, + .rate_max = 96000, + .buffer_bytes_max = PLAYBACK_MAX_NUM_PERIODS * PLAYBACK_MAX_PERIOD_SIZE, + .period_bytes_min = PLAYBACK_MIN_PERIOD_SIZE, + .period_bytes_max = PLAYBACK_MAX_PERIOD_SIZE, + .periods_min = PLAYBACK_MIN_NUM_PERIODS, + .periods_max = PLAYBACK_MAX_NUM_PERIODS, +}; + +static const struct snd_pcm_hardware acp3x_pcm_hardware_capture = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .buffer_bytes_max = CAPTURE_MAX_NUM_PERIODS * CAPTURE_MAX_PERIOD_SIZE, + .period_bytes_min = CAPTURE_MIN_PERIOD_SIZE, + .period_bytes_max = CAPTURE_MAX_PERIOD_SIZE, + .periods_min = CAPTURE_MIN_NUM_PERIODS, + .periods_max = CAPTURE_MAX_NUM_PERIODS, +}; + +static int acp3x_power_on(void __iomem *acp3x_base, bool on) +{ + u16 val, mask; + u32 timeout; + + if (on == true) { + val = 1; + mask = ACP3x_POWER_ON; + } else { + val = 0; + mask = ACP3x_POWER_OFF; + } + + rv_writel(val, acp3x_base + mmACP_PGFSM_CONTROL); + timeout = 0; + while (true) { + val = rv_readl(acp3x_base + mmACP_PGFSM_STATUS); + if ((val & ACP3x_POWER_OFF_IN_PROGRESS) == mask) + break; + if (timeout > 100) { + pr_err("ACP3x power state change failure\n"); + return -ENODEV; + } + timeout++; + cpu_relax(); + } + return 0; +} + +static int acp3x_reset(void __iomem *acp3x_base) +{ + u32 val, timeout; + + rv_writel(1, acp3x_base + mmACP_SOFT_RESET); + timeout = 0; + while (true) { + val = rv_readl(acp3x_base + mmACP_SOFT_RESET); + if ((val & ACP3x_SOFT_RESET__SoftResetAudDone_MASK) || + timeout > 100) { + if (val & ACP3x_SOFT_RESET__SoftResetAudDone_MASK) + break; + return -ENODEV; + } + timeout++; + cpu_relax(); + } + + rv_writel(0, acp3x_base + mmACP_SOFT_RESET); + timeout = 0; + while (true) { + val = rv_readl(acp3x_base + mmACP_SOFT_RESET); + if (!val || timeout > 100) { + if (!val) + break; + return -ENODEV; + } + timeout++; + cpu_relax(); + } + return 0; +} + +static int acp3x_init(void __iomem *acp3x_base) +{ + int ret; + + /* power on */ + ret = acp3x_power_on(acp3x_base, true); + if (ret) { + pr_err("ACP3x power on failed\n"); + return ret; + } + /* Reset */ + ret = acp3x_reset(acp3x_base); + if (ret) { + pr_err("ACP3x reset failed\n"); + return ret; + } + return 0; +} + +static int acp3x_deinit(void __iomem *acp3x_base) +{ + int ret; + + /* Reset */ + ret = acp3x_reset(acp3x_base); + if (ret) { + pr_err("ACP3x reset failed\n"); + return ret; + } + /* power off */ + ret = acp3x_power_on(acp3x_base, false); + if (ret) { + pr_err("ACP3x power off failed\n"); + return ret; + } + return 0; +} + +static irqreturn_t i2s_irq_handler(int irq, void *dev_id) +{ + u16 play_flag, cap_flag; + u32 val; + struct i2s_dev_data *rv_i2s_data = dev_id; + + if (!rv_i2s_data) + return IRQ_NONE; + + play_flag = 0; + cap_flag = 0; + val = rv_readl(rv_i2s_data->acp3x_base + mmACP_EXTERNAL_INTR_STAT); + if ((val & BIT(BT_TX_THRESHOLD)) && rv_i2s_data->play_stream) { + rv_writel(BIT(BT_TX_THRESHOLD), rv_i2s_data->acp3x_base + + mmACP_EXTERNAL_INTR_STAT); + snd_pcm_period_elapsed(rv_i2s_data->play_stream); + play_flag = 1; + } + + if ((val & BIT(BT_RX_THRESHOLD)) && rv_i2s_data->capture_stream) { + rv_writel(BIT(BT_RX_THRESHOLD), rv_i2s_data->acp3x_base + + mmACP_EXTERNAL_INTR_STAT); + snd_pcm_period_elapsed(rv_i2s_data->capture_stream); + cap_flag = 1; + } + + if (play_flag | cap_flag) + return IRQ_HANDLED; + else + return IRQ_NONE; +} + +static void config_acp3x_dma(struct i2s_stream_instance *rtd, int direction) +{ + u16 page_idx; + u64 addr; + u32 low, high, val, acp_fifo_addr; + struct page *pg = rtd->pg; + + /* 8 scratch registers used to map one 64 bit address */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + val = 0; + else + val = rtd->num_pages * 8; + + /* Group Enable */ + rv_writel(ACP_SRAM_PTE_OFFSET | BIT(31), rtd->acp3x_base + + mmACPAXI2AXI_ATU_BASE_ADDR_GRP_1); + rv_writel(PAGE_SIZE_4K_ENABLE, rtd->acp3x_base + + mmACPAXI2AXI_ATU_PAGE_SIZE_GRP_1); + + for (page_idx = 0; page_idx < rtd->num_pages; page_idx++) { + /* Load the low address of page int ACP SRAM through SRBM */ + addr = page_to_phys(pg); + low = lower_32_bits(addr); + high = upper_32_bits(addr); + + rv_writel(low, rtd->acp3x_base + mmACP_SCRATCH_REG_0 + val); + high |= BIT(31); + rv_writel(high, rtd->acp3x_base + mmACP_SCRATCH_REG_0 + val + + 4); + /* Move to next physically contiguos page */ + val += 8; + pg++; + } + + if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + /* Config ringbuffer */ + rv_writel(MEM_WINDOW_START, rtd->acp3x_base + + mmACP_BT_TX_RINGBUFADDR); + rv_writel(MAX_BUFFER, rtd->acp3x_base + + mmACP_BT_TX_RINGBUFSIZE); + rv_writel(DMA_SIZE, rtd->acp3x_base + mmACP_BT_TX_DMA_SIZE); + + /* Config audio fifo */ + acp_fifo_addr = ACP_SRAM_PTE_OFFSET + (rtd->num_pages * 8) + + PLAYBACK_FIFO_ADDR_OFFSET; + rv_writel(acp_fifo_addr, rtd->acp3x_base + + mmACP_BT_TX_FIFOADDR); + rv_writel(FIFO_SIZE, rtd->acp3x_base + mmACP_BT_TX_FIFOSIZE); + } else { + /* Config ringbuffer */ + rv_writel(MEM_WINDOW_START + MAX_BUFFER, rtd->acp3x_base + + mmACP_BT_RX_RINGBUFADDR); + rv_writel(MAX_BUFFER, rtd->acp3x_base + + mmACP_BT_RX_RINGBUFSIZE); + rv_writel(DMA_SIZE, rtd->acp3x_base + mmACP_BT_RX_DMA_SIZE); + + /* Config audio fifo */ + acp_fifo_addr = ACP_SRAM_PTE_OFFSET + + (rtd->num_pages * 8) + CAPTURE_FIFO_ADDR_OFFSET; + rv_writel(acp_fifo_addr, rtd->acp3x_base + + mmACP_BT_RX_FIFOADDR); + rv_writel(FIFO_SIZE, rtd->acp3x_base + mmACP_BT_RX_FIFOSIZE); + } + + /* Enable watermark/period interrupt to host */ + rv_writel(BIT(BT_TX_THRESHOLD) | BIT(BT_RX_THRESHOLD), + rtd->acp3x_base + mmACP_EXTERNAL_INTR_CNTL); +} + +static int acp3x_dma_open(struct snd_pcm_substream *substream) +{ + int ret = 0; + + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *prtd = substream->private_data; + struct snd_soc_component *component = snd_soc_rtdcom_lookup(prtd, + DRV_NAME); + struct i2s_dev_data *adata = dev_get_drvdata(component->dev); + + struct i2s_stream_instance *i2s_data = kzalloc(sizeof(struct i2s_stream_instance), + GFP_KERNEL); + if (!i2s_data) + return -EINVAL; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + runtime->hw = acp3x_pcm_hardware_playback; + else + runtime->hw = acp3x_pcm_hardware_capture; + + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) { + dev_err(component->dev, "set integer constraint failed\n"); + kfree(i2s_data); + return ret; + } + + if (!adata->play_stream && !adata->capture_stream) + rv_writel(1, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + adata->play_stream = substream; + else + adata->capture_stream = substream; + + i2s_data->acp3x_base = adata->acp3x_base; + runtime->private_data = i2s_data; + return 0; +} + +static int acp3x_dma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int status; + u64 size; + struct page *pg; + struct snd_pcm_runtime *runtime = substream->runtime; + struct i2s_stream_instance *rtd = runtime->private_data; + + if (!rtd) + return -EINVAL; + + size = params_buffer_bytes(params); + status = snd_pcm_lib_malloc_pages(substream, size); + if (status < 0) + return status; + + memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); + pg = virt_to_page(substream->dma_buffer.area); + if (pg) { + rtd->pg = pg; + rtd->num_pages = (PAGE_ALIGN(size) >> PAGE_SHIFT); + config_acp3x_dma(rtd, substream->stream); + status = 0; + } else { + status = -ENOMEM; + } + return status; +} + +static snd_pcm_uframes_t acp3x_dma_pointer(struct snd_pcm_substream *substream) +{ + u32 pos = 0; + struct i2s_stream_instance *rtd = substream->runtime->private_data; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + pos = rv_readl(rtd->acp3x_base + + mmACP_BT_TX_LINKPOSITIONCNTR); + else + pos = rv_readl(rtd->acp3x_base + + mmACP_BT_RX_LINKPOSITIONCNTR); + + if (pos >= MAX_BUFFER) + pos = 0; + + return bytes_to_frames(substream->runtime, pos); +} + +static int acp3x_dma_new(struct snd_soc_pcm_runtime *rtd) +{ + return snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, + SNDRV_DMA_TYPE_DEV, + NULL, MIN_BUFFER, + MAX_BUFFER); +} + +static int acp3x_dma_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int acp3x_dma_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return snd_pcm_lib_default_mmap(substream, vma); +} + +static int acp3x_dma_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *prtd = substream->private_data; + struct i2s_stream_instance *rtd = substream->runtime->private_data; + struct snd_soc_component *component = snd_soc_rtdcom_lookup(prtd, + DRV_NAME); + struct i2s_dev_data *adata = dev_get_drvdata(component->dev); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + adata->play_stream = NULL; + else + adata->capture_stream = NULL; + + /* Disable ACP irq, when the current stream is being closed and + * another stream is also not active. + */ + if (!adata->play_stream && !adata->capture_stream) + rv_writel(0, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); + kfree(rtd); + return 0; +} + +static struct snd_pcm_ops acp3x_dma_ops = { + .open = acp3x_dma_open, + .close = acp3x_dma_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = acp3x_dma_hw_params, + .hw_free = acp3x_dma_hw_free, + .pointer = acp3x_dma_pointer, + .mmap = acp3x_dma_mmap, +}; + + +static int acp3x_dai_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + + struct i2s_dev_data *adata = snd_soc_dai_get_drvdata(cpu_dai); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + adata->tdm_mode = false; + break; + case SND_SOC_DAIFMT_DSP_A: + adata->tdm_mode = true; + break; + default: + return -EINVAL; + } + + return 0; +} + +static int acp3x_dai_set_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, + u32 rx_mask, int slots, int slot_width) +{ + u32 val = 0; + u16 slot_len; + + struct i2s_dev_data *adata = snd_soc_dai_get_drvdata(cpu_dai); + + switch (slot_width) { + case SLOT_WIDTH_8: + slot_len = 8; + break; + case SLOT_WIDTH_16: + slot_len = 16; + break; + case SLOT_WIDTH_24: + slot_len = 24; + break; + case SLOT_WIDTH_32: + slot_len = 0; + break; + default: + return -EINVAL; + } + + val = rv_readl(adata->acp3x_base + mmACP_BTTDM_ITER); + rv_writel((val | 0x2), adata->acp3x_base + mmACP_BTTDM_ITER); + val = rv_readl(adata->acp3x_base + mmACP_BTTDM_IRER); + rv_writel((val | 0x2), adata->acp3x_base + mmACP_BTTDM_IRER); + + val = (FRM_LEN | (slots << 15) | (slot_len << 18)); + rv_writel(val, adata->acp3x_base + mmACP_BTTDM_TXFRMT); + rv_writel(val, adata->acp3x_base + mmACP_BTTDM_RXFRMT); + + adata->tdm_fmt = val; + return 0; +} + +static int acp3x_dai_i2s_hwparams(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + u32 val = 0; + struct i2s_stream_instance *rtd = substream->runtime->private_data; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_S8: + rtd->xfer_resolution = 0x0; + break; + case SNDRV_PCM_FORMAT_S16_LE: + rtd->xfer_resolution = 0x02; + break; + case SNDRV_PCM_FORMAT_S24_LE: + rtd->xfer_resolution = 0x04; + break; + case SNDRV_PCM_FORMAT_S32_LE: + rtd->xfer_resolution = 0x05; + break; + default: + return -EINVAL; + } + val = rv_readl(rtd->acp3x_base + mmACP_BTTDM_ITER); + val = val | (rtd->xfer_resolution << 3); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + rv_writel(val, rtd->acp3x_base + mmACP_BTTDM_ITER); + else + rv_writel(val, rtd->acp3x_base + mmACP_BTTDM_IRER); + + return 0; +} + +static int acp3x_dai_i2s_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + int ret = 0; + struct i2s_stream_instance *rtd = substream->runtime->private_data; + u32 val, period_bytes; + + period_bytes = frames_to_bytes(substream->runtime, + substream->runtime->period_size); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + rv_writel(period_bytes, rtd->acp3x_base + + mmACP_BT_TX_INTR_WATERMARK_SIZE); + val = rv_readl(rtd->acp3x_base + mmACP_BTTDM_ITER); + val = val | BIT(0); + rv_writel(val, rtd->acp3x_base + mmACP_BTTDM_ITER); + } else { + rv_writel(period_bytes, rtd->acp3x_base + + mmACP_BT_RX_INTR_WATERMARK_SIZE); + val = rv_readl(rtd->acp3x_base + mmACP_BTTDM_IRER); + val = val | BIT(0); + rv_writel(val, rtd->acp3x_base + mmACP_BTTDM_IRER); + } + rv_writel(1, rtd->acp3x_base + mmACP_BTTDM_IER); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + val = rv_readl(rtd->acp3x_base + mmACP_BTTDM_ITER); + val = val & ~BIT(0); + rv_writel(val, rtd->acp3x_base + mmACP_BTTDM_ITER); + } else { + val = rv_readl(rtd->acp3x_base + mmACP_BTTDM_IRER); + val = val & ~BIT(0); + rv_writel(val, rtd->acp3x_base + mmACP_BTTDM_IRER); + } + rv_writel(0, rtd->acp3x_base + mmACP_BTTDM_IER); + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + +struct snd_soc_dai_ops acp3x_dai_i2s_ops = { + .hw_params = acp3x_dai_i2s_hwparams, + .trigger = acp3x_dai_i2s_trigger, + .set_fmt = acp3x_dai_i2s_set_fmt, + .set_tdm_slot = acp3x_dai_set_tdm_slot, +}; + +static struct snd_soc_dai_driver acp3x_i2s_dai_driver = { + .playback = { + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 8, + + .rate_min = 8000, + .rate_max = 96000, + }, + .capture = { + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + }, + .ops = &acp3x_dai_i2s_ops, +}; + +static const struct snd_soc_component_driver acp3x_i2s_component = { + .name = DRV_NAME, + .ops = &acp3x_dma_ops, + .pcm_new = acp3x_dma_new, +}; + +static int acp3x_audio_probe(struct platform_device *pdev) +{ + int status; + struct resource *res; + struct i2s_dev_data *adata; + unsigned int irqflags; + + if (!pdev->dev.platform_data) { + dev_err(&pdev->dev, "platform_data not retrieved\n"); + return -ENODEV; + } + irqflags = *((unsigned int *)(pdev->dev.platform_data)); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); + return -ENODEV; + } + + adata = devm_kzalloc(&pdev->dev, sizeof(*adata), GFP_KERNEL); + if (!adata) + return -ENOMEM; + + adata->acp3x_base = devm_ioremap(&pdev->dev, res->start, + resource_size(res)); + + res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); + if (!res) { + dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); + return -ENODEV; + } + + adata->i2s_irq = res->start; + adata->play_stream = NULL; + adata->capture_stream = NULL; + + dev_set_drvdata(&pdev->dev, adata); + /* Initialize ACP */ + status = acp3x_init(adata->acp3x_base); + if (status) + return -ENODEV; + status = devm_snd_soc_register_component(&pdev->dev, + &acp3x_i2s_component, + &acp3x_i2s_dai_driver, 1); + if (status) { + dev_err(&pdev->dev, "Fail to register acp i2s dai\n"); + goto dev_err; + } + status = devm_request_irq(&pdev->dev, adata->i2s_irq, i2s_irq_handler, + irqflags, "ACP3x_I2S_IRQ", adata); + if (status) { + dev_err(&pdev->dev, "ACP3x I2S IRQ request failed\n"); + goto dev_err; + } + + pm_runtime_set_autosuspend_delay(&pdev->dev, 10000); + pm_runtime_use_autosuspend(&pdev->dev); + pm_runtime_enable(&pdev->dev); + return 0; +dev_err: + status = acp3x_deinit(adata->acp3x_base); + if (status) + dev_err(&pdev->dev, "ACP de-init failed\n"); + else + dev_info(&pdev->dev, "ACP de-initialized\n"); + /*ignore device status and return driver probe error*/ + return -ENODEV; +} + +static int acp3x_audio_remove(struct platform_device *pdev) +{ + int ret; + struct i2s_dev_data *adata = dev_get_drvdata(&pdev->dev); + + ret = acp3x_deinit(adata->acp3x_base); + if (ret) + dev_err(&pdev->dev, "ACP de-init failed\n"); + else + dev_info(&pdev->dev, "ACP de-initialized\n"); + + pm_runtime_disable(&pdev->dev); + return 0; +} + +static int acp3x_resume(struct device *dev) +{ + int status; + u32 val; + struct i2s_dev_data *adata = dev_get_drvdata(dev); + + status = acp3x_init(adata->acp3x_base); + if (status) + return -ENODEV; + + if (adata->play_stream && adata->play_stream->runtime) { + struct i2s_stream_instance *rtd = + adata->play_stream->runtime->private_data; + config_acp3x_dma(rtd, SNDRV_PCM_STREAM_PLAYBACK); + rv_writel((rtd->xfer_resolution << 3), + rtd->acp3x_base + mmACP_BTTDM_ITER); + if (adata->tdm_mode == true) { + rv_writel(adata->tdm_fmt, adata->acp3x_base + + mmACP_BTTDM_TXFRMT); + val = rv_readl(adata->acp3x_base + mmACP_BTTDM_ITER); + rv_writel((val | 0x2), adata->acp3x_base + + mmACP_BTTDM_ITER); + } + } + + if (adata->capture_stream && adata->capture_stream->runtime) { + struct i2s_stream_instance *rtd = + adata->capture_stream->runtime->private_data; + config_acp3x_dma(rtd, SNDRV_PCM_STREAM_CAPTURE); + rv_writel((rtd->xfer_resolution << 3), + rtd->acp3x_base + mmACP_BTTDM_IRER); + if (adata->tdm_mode == true) { + rv_writel(adata->tdm_fmt, adata->acp3x_base + + mmACP_BTTDM_RXFRMT); + val = rv_readl(adata->acp3x_base + mmACP_BTTDM_IRER); + rv_writel((val | 0x2), adata->acp3x_base + + mmACP_BTTDM_IRER); + } + } + + rv_writel(1, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); + return 0; +} + + +static int acp3x_pcm_runtime_suspend(struct device *dev) +{ + int status; + struct i2s_dev_data *adata = dev_get_drvdata(dev); + + status = acp3x_deinit(adata->acp3x_base); + if (status) + dev_err(dev, "ACP de-init failed\n"); + else + dev_info(dev, "ACP de-initialized\n"); + + rv_writel(0, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); + + return 0; +} + +static int acp3x_pcm_runtime_resume(struct device *dev) +{ + int status; + struct i2s_dev_data *adata = dev_get_drvdata(dev); + + status = acp3x_init(adata->acp3x_base); + if (status) + return -ENODEV; + rv_writel(1, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); + return 0; +} + +static const struct dev_pm_ops acp3x_pm_ops = { + .runtime_suspend = acp3x_pcm_runtime_suspend, + .runtime_resume = acp3x_pcm_runtime_resume, + .resume = acp3x_resume, +}; + +static struct platform_driver acp3x_dma_driver = { + .probe = acp3x_audio_probe, + .remove = acp3x_audio_remove, + .driver = { + .name = "acp3x_rv_i2s", + .pm = &acp3x_pm_ops, + }, +}; + +module_platform_driver(acp3x_dma_driver); + +MODULE_AUTHOR("Maruthi.Bayyavarapu@amd.com"); +MODULE_AUTHOR("Vijendar.Mukunda@amd.com"); +MODULE_DESCRIPTION("AMD ACP 3.x PCM Driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h new file mode 100644 index 000000000000..4f2cadd90a87 --- /dev/null +++ b/sound/soc/amd/raven/acp3x.h @@ -0,0 +1,58 @@ +/* SPDX-License-Identifier: GPL-2.0+ */ +/* + * AMD ALSA SoC PCM Driver + * + * Copyright 2016 Advanced Micro Devices, Inc. + */ + +#include "chip_offset_byte.h" + +#define ACP3x_PHY_BASE_ADDRESS 0x1240000 +#define ACP3x_I2S_MODE 0 +#define ACP3x_REG_START 0x1240000 +#define ACP3x_REG_END 0x1250200 +#define I2S_MODE 0x04 +#define BT_TX_THRESHOLD 26 +#define BT_RX_THRESHOLD 25 +#define ACP3x_POWER_ON 0x00 +#define ACP3x_POWER_ON_IN_PROGRESS 0x01 +#define ACP3x_POWER_OFF 0x02 +#define ACP3x_POWER_OFF_IN_PROGRESS 0x03 +#define ACP3x_SOFT_RESET__SoftResetAudDone_MASK 0x00010001 + +#define ACP_SRAM_PTE_OFFSET 0x02050000 +#define PAGE_SIZE_4K_ENABLE 0x2 +#define MEM_WINDOW_START 0x4000000 +#define PLAYBACK_FIFO_ADDR_OFFSET 0x400 +#define CAPTURE_FIFO_ADDR_OFFSET 0x500 + +#define PLAYBACK_MIN_NUM_PERIODS 2 +#define PLAYBACK_MAX_NUM_PERIODS 8 +#define PLAYBACK_MAX_PERIOD_SIZE 16384 +#define PLAYBACK_MIN_PERIOD_SIZE 4096 +#define CAPTURE_MIN_NUM_PERIODS 2 +#define CAPTURE_MAX_NUM_PERIODS 8 +#define CAPTURE_MAX_PERIOD_SIZE 16384 +#define CAPTURE_MIN_PERIOD_SIZE 4096 + +#define MAX_BUFFER (PLAYBACK_MAX_PERIOD_SIZE * PLAYBACK_MAX_NUM_PERIODS) +#define MIN_BUFFER MAX_BUFFER +#define FIFO_SIZE 0x100 +#define DMA_SIZE 0x40 +#define FRM_LEN 0x100 + +#define SLOT_WIDTH_8 0x08 +#define SLOT_WIDTH_16 0x10 +#define SLOT_WIDTH_24 0x18 +#define SLOT_WIDTH_32 0x20 + + +static inline u32 rv_readl(void __iomem *base_addr) +{ + return readl(base_addr - ACP3x_PHY_BASE_ADDRESS); +} + +static inline void rv_writel(u32 val, void __iomem *base_addr) +{ + writel(val, base_addr - ACP3x_PHY_BASE_ADDRESS); +} diff --git a/sound/soc/amd/raven/chip_offset_byte.h b/sound/soc/amd/raven/chip_offset_byte.h new file mode 100644 index 000000000000..9c1fac58fb2a --- /dev/null +++ b/sound/soc/amd/raven/chip_offset_byte.h @@ -0,0 +1,639 @@ +/* SPDX-License-Identifier: GPL-2.0+ */ +/* + * AMD ACP 3.0 Register Documentation + * + * Copyright 2016 Advanced Micro Devices, Inc. + */ + +#ifndef _acp_ip_OFFSET_HEADER +#define _acp_ip_OFFSET_HEADER +// Registers from ACP_DMA block + +#define mmACP_DMA_CNTL_0 0x1240000 +#define mmACP_DMA_CNTL_1 0x1240004 +#define mmACP_DMA_CNTL_2 0x1240008 +#define mmACP_DMA_CNTL_3 0x124000C +#define mmACP_DMA_CNTL_4 0x1240010 +#define mmACP_DMA_CNTL_5 0x1240014 +#define mmACP_DMA_CNTL_6 0x1240018 +#define mmACP_DMA_CNTL_7 0x124001C +#define mmACP_DMA_DSCR_STRT_IDX_0 0x1240020 +#define mmACP_DMA_DSCR_STRT_IDX_1 0x1240024 +#define mmACP_DMA_DSCR_STRT_IDX_2 0x1240028 +#define mmACP_DMA_DSCR_STRT_IDX_3 0x124002C +#define mmACP_DMA_DSCR_STRT_IDX_4 0x1240030 +#define mmACP_DMA_DSCR_STRT_IDX_5 0x1240034 +#define mmACP_DMA_DSCR_STRT_IDX_6 0x1240038 +#define mmACP_DMA_DSCR_STRT_IDX_7 0x124003C +#define mmACP_DMA_DSCR_CNT_0 0x1240040 +#define mmACP_DMA_DSCR_CNT_1 0x1240044 +#define mmACP_DMA_DSCR_CNT_2 0x1240048 +#define mmACP_DMA_DSCR_CNT_3 0x124004C +#define mmACP_DMA_DSCR_CNT_4 0x1240050 +#define mmACP_DMA_DSCR_CNT_5 0x1240054 +#define mmACP_DMA_DSCR_CNT_6 0x1240058 +#define mmACP_DMA_DSCR_CNT_7 0x124005C +#define mmACP_DMA_PRIO_0 0x1240060 +#define mmACP_DMA_PRIO_1 0x1240064 +#define mmACP_DMA_PRIO_2 0x1240068 +#define mmACP_DMA_PRIO_3 0x124006C +#define mmACP_DMA_PRIO_4 0x1240070 +#define mmACP_DMA_PRIO_5 0x1240074 +#define mmACP_DMA_PRIO_6 0x1240078 +#define mmACP_DMA_PRIO_7 0x124007C +#define mmACP_DMA_CUR_DSCR_0 0x1240080 +#define mmACP_DMA_CUR_DSCR_1 0x1240084 +#define mmACP_DMA_CUR_DSCR_2 0x1240088 +#define mmACP_DMA_CUR_DSCR_3 0x124008C +#define mmACP_DMA_CUR_DSCR_4 0x1240090 +#define mmACP_DMA_CUR_DSCR_5 0x1240094 +#define mmACP_DMA_CUR_DSCR_6 0x1240098 +#define mmACP_DMA_CUR_DSCR_7 0x124009C +#define mmACP_DMA_CUR_TRANS_CNT_0 0x12400A0 +#define mmACP_DMA_CUR_TRANS_CNT_1 0x12400A4 +#define mmACP_DMA_CUR_TRANS_CNT_2 0x12400A8 +#define mmACP_DMA_CUR_TRANS_CNT_3 0x12400AC +#define mmACP_DMA_CUR_TRANS_CNT_4 0x12400B0 +#define mmACP_DMA_CUR_TRANS_CNT_5 0x12400B4 +#define mmACP_DMA_CUR_TRANS_CNT_6 0x12400B8 +#define mmACP_DMA_CUR_TRANS_CNT_7 0x12400BC +#define mmACP_DMA_ERR_STS_0 0x12400C0 +#define mmACP_DMA_ERR_STS_1 0x12400C4 +#define mmACP_DMA_ERR_STS_2 0x12400C8 +#define mmACP_DMA_ERR_STS_3 0x12400CC +#define mmACP_DMA_ERR_STS_4 0x12400D0 +#define mmACP_DMA_ERR_STS_5 0x12400D4 +#define mmACP_DMA_ERR_STS_6 0x12400D8 +#define mmACP_DMA_ERR_STS_7 0x12400DC +#define mmACP_DMA_DESC_BASE_ADDR 0x12400E0 +#define mmACP_DMA_DESC_MAX_NUM_DSCR 0x12400E4 +#define mmACP_DMA_CH_STS 0x12400E8 +#define mmACP_DMA_CH_GROUP 0x12400EC +#define mmACP_DMA_CH_RST_STS 0x12400F0 + + +// Registers from ACP_AXI2AXIATU block + +#define mmACPAXI2AXI_ATU_PAGE_SIZE_GRP_1 0x1240C00 +#define mmACPAXI2AXI_ATU_BASE_ADDR_GRP_1 0x1240C04 +#define mmACPAXI2AXI_ATU_PAGE_SIZE_GRP_2 0x1240C08 +#define mmACPAXI2AXI_ATU_BASE_ADDR_GRP_2 0x1240C0C +#define mmACPAXI2AXI_ATU_PAGE_SIZE_GRP_3 0x1240C10 +#define mmACPAXI2AXI_ATU_BASE_ADDR_GRP_3 0x1240C14 +#define mmACPAXI2AXI_ATU_PAGE_SIZE_GRP_4 0x1240C18 +#define mmACPAXI2AXI_ATU_BASE_ADDR_GRP_4 0x1240C1C +#define mmACPAXI2AXI_ATU_PAGE_SIZE_GRP_5 0x1240C20 +#define mmACPAXI2AXI_ATU_BASE_ADDR_GRP_5 0x1240C24 +#define mmACPAXI2AXI_ATU_PAGE_SIZE_GRP_6 0x1240C28 +#define mmACPAXI2AXI_ATU_BASE_ADDR_GRP_6 0x1240C2C +#define mmACPAXI2AXI_ATU_PAGE_SIZE_GRP_7 0x1240C30 +#define mmACPAXI2AXI_ATU_BASE_ADDR_GRP_7 0x1240C34 +#define mmACPAXI2AXI_ATU_PAGE_SIZE_GRP_8 0x1240C38 +#define mmACPAXI2AXI_ATU_BASE_ADDR_GRP_8 0x1240C3C +#define mmACPAXI2AXI_ATU_CTRL 0x1240C40 + + +// Registers from ACP_CLKRST block + +#define mmACP_SOFT_RESET 0x1241000 +#define mmACP_CONTROL 0x1241004 +#define mmACP_STATUS 0x1241008 +#define mmACP_DSP0_OCD_HALT_ON_RST 0x124100C +#define mmACP_DYNAMIC_CG_MASTER_CONTROL 0x1241010 + + +// Registers from ACP_MISC block + +#define mmACP_EXTERNAL_INTR_ENB 0x1241800 +#define mmACP_EXTERNAL_INTR_CNTL 0x1241804 +#define mmACP_EXTERNAL_INTR_STAT 0x1241808 +#define mmACP_DSP0_INTR_CNTL 0x124180C +#define mmACP_DSP0_INTR_STAT 0x1241810 +#define mmACP_DSP_SW_INTR_CNTL 0x1241814 +#define mmACP_DSP_SW_INTR_STAT 0x1241818 +#define mmACP_SW_INTR_TRIG 0x124181C +#define mmACP_SMU_MAILBOX 0x1241820 +#define mmDSP_INTERRUPT_ROUTING_CTRL 0x1241824 +#define mmACP_DSP0_WATCHDOG_TIMER_CNTL 0x1241828 +#define mmACP_DSP0_EXT_TIMER1_CNTL 0x124182C +#define mmACP_DSP0_EXT_TIMER2_CNTL 0x1241830 +#define mmACP_DSP0_EXT_TIMER3_CNTL 0x1241834 +#define mmACP_DSP0_EXT_TIMER4_CNTL 0x1241838 +#define mmACP_DSP0_EXT_TIMER5_CNTL 0x124183C +#define mmACP_DSP0_EXT_TIMER6_CNTL 0x1241840 +#define mmACP_DSP0_EXT_TIMER1_CURR_VALUE 0x1241844 +#define mmACP_DSP0_EXT_TIMER2_CURR_VALUE 0x1241848 +#define mmACP_DSP0_EXT_TIMER3_CURR_VALUE 0x124184C +#define mmACP_DSP0_EXT_TIMER4_CURR_VALUE 0x1241850 +#define mmACP_DSP0_EXT_TIMER5_CURR_VALUE 0x1241854 +#define mmACP_DSP0_EXT_TIMER6_CURR_VALUE 0x1241858 +#define mmACP_FW_STATUS 0x124185C +#define mmACP_TIMER 0x1241874 +#define mmACP_TIMER_CNTL 0x1241878 +#define mmACP_PGMEM_CTRL 0x12418C0 +#define mmACP_ERROR_STATUS 0x12418C4 +#define mmACP_SW_I2S_ERROR_REASON 0x12418C8 +#define mmACP_MEM_PG_STS 0x12418CC + + +// Registers from ACP_PGFSM block + +#define mmACP_I2S_PIN_CONFIG 0x1241400 +#define mmACP_PAD_PULLUP_PULLDOWN_CTRL 0x1241404 +#define mmACP_PAD_DRIVE_STRENGTH_CTRL 0x1241408 +#define mmACP_SW_PAD_KEEPER_EN 0x124140C +#define mmACP_SW_WAKE_EN 0x1241410 +#define mmACP_I2S_WAKE_EN 0x1241414 +#define mmACP_PME_EN 0x1241418 +#define mmACP_PGFSM_CONTROL 0x124141C +#define mmACP_PGFSM_STATUS 0x1241420 + + +// Registers from ACP_SCRATCH block + +#define mmACP_SCRATCH_REG_0 0x1250000 +#define mmACP_SCRATCH_REG_1 0x1250004 +#define mmACP_SCRATCH_REG_2 0x1250008 +#define mmACP_SCRATCH_REG_3 0x125000C +#define mmACP_SCRATCH_REG_4 0x1250010 +#define mmACP_SCRATCH_REG_5 0x1250014 +#define mmACP_SCRATCH_REG_6 0x1250018 +#define mmACP_SCRATCH_REG_7 0x125001C +#define mmACP_SCRATCH_REG_8 0x1250020 +#define mmACP_SCRATCH_REG_9 0x1250024 +#define mmACP_SCRATCH_REG_10 0x1250028 +#define mmACP_SCRATCH_REG_11 0x125002C +#define mmACP_SCRATCH_REG_12 0x1250030 +#define mmACP_SCRATCH_REG_13 0x1250034 +#define mmACP_SCRATCH_REG_14 0x1250038 +#define mmACP_SCRATCH_REG_15 0x125003C +#define mmACP_SCRATCH_REG_16 0x1250040 +#define mmACP_SCRATCH_REG_17 0x1250044 +#define mmACP_SCRATCH_REG_18 0x1250048 +#define mmACP_SCRATCH_REG_19 0x125004C +#define mmACP_SCRATCH_REG_20 0x1250050 +#define mmACP_SCRATCH_REG_21 0x1250054 +#define mmACP_SCRATCH_REG_22 0x1250058 +#define mmACP_SCRATCH_REG_23 0x125005C +#define mmACP_SCRATCH_REG_24 0x1250060 +#define mmACP_SCRATCH_REG_25 0x1250064 +#define mmACP_SCRATCH_REG_26 0x1250068 +#define mmACP_SCRATCH_REG_27 0x125006C +#define mmACP_SCRATCH_REG_28 0x1250070 +#define mmACP_SCRATCH_REG_29 0x1250074 +#define mmACP_SCRATCH_REG_30 0x1250078 +#define mmACP_SCRATCH_REG_31 0x125007C +#define mmACP_SCRATCH_REG_32 0x1250080 +#define mmACP_SCRATCH_REG_33 0x1250084 +#define mmACP_SCRATCH_REG_34 0x1250088 +#define mmACP_SCRATCH_REG_35 0x125008C +#define mmACP_SCRATCH_REG_36 0x1250090 +#define mmACP_SCRATCH_REG_37 0x1250094 +#define mmACP_SCRATCH_REG_38 0x1250098 +#define mmACP_SCRATCH_REG_39 0x125009C +#define mmACP_SCRATCH_REG_40 0x12500A0 +#define mmACP_SCRATCH_REG_41 0x12500A4 +#define mmACP_SCRATCH_REG_42 0x12500A8 +#define mmACP_SCRATCH_REG_43 0x12500AC +#define mmACP_SCRATCH_REG_44 0x12500B0 +#define mmACP_SCRATCH_REG_45 0x12500B4 +#define mmACP_SCRATCH_REG_46 0x12500B8 +#define mmACP_SCRATCH_REG_47 0x12500BC +#define mmACP_SCRATCH_REG_48 0x12500C0 +#define mmACP_SCRATCH_REG_49 0x12500C4 +#define mmACP_SCRATCH_REG_50 0x12500C8 +#define mmACP_SCRATCH_REG_51 0x12500CC +#define mmACP_SCRATCH_REG_52 0x12500D0 +#define mmACP_SCRATCH_REG_53 0x12500D4 +#define mmACP_SCRATCH_REG_54 0x12500D8 +#define mmACP_SCRATCH_REG_55 0x12500DC +#define mmACP_SCRATCH_REG_56 0x12500E0 +#define mmACP_SCRATCH_REG_57 0x12500E4 +#define mmACP_SCRATCH_REG_58 0x12500E8 +#define mmACP_SCRATCH_REG_59 0x12500EC +#define mmACP_SCRATCH_REG_60 0x12500F0 +#define mmACP_SCRATCH_REG_61 0x12500F4 +#define mmACP_SCRATCH_REG_62 0x12500F8 +#define mmACP_SCRATCH_REG_63 0x12500FC +#define mmACP_SCRATCH_REG_64 0x1250100 +#define mmACP_SCRATCH_REG_65 0x1250104 +#define mmACP_SCRATCH_REG_66 0x1250108 +#define mmACP_SCRATCH_REG_67 0x125010C +#define mmACP_SCRATCH_REG_68 0x1250110 +#define mmACP_SCRATCH_REG_69 0x1250114 +#define mmACP_SCRATCH_REG_70 0x1250118 +#define mmACP_SCRATCH_REG_71 0x125011C +#define mmACP_SCRATCH_REG_72 0x1250120 +#define mmACP_SCRATCH_REG_73 0x1250124 +#define mmACP_SCRATCH_REG_74 0x1250128 +#define mmACP_SCRATCH_REG_75 0x125012C +#define mmACP_SCRATCH_REG_76 0x1250130 +#define mmACP_SCRATCH_REG_77 0x1250134 +#define mmACP_SCRATCH_REG_78 0x1250138 +#define mmACP_SCRATCH_REG_79 0x125013C +#define mmACP_SCRATCH_REG_80 0x1250140 +#define mmACP_SCRATCH_REG_81 0x1250144 +#define mmACP_SCRATCH_REG_82 0x1250148 +#define mmACP_SCRATCH_REG_83 0x125014C +#define mmACP_SCRATCH_REG_84 0x1250150 +#define mmACP_SCRATCH_REG_85 0x1250154 +#define mmACP_SCRATCH_REG_86 0x1250158 +#define mmACP_SCRATCH_REG_87 0x125015C +#define mmACP_SCRATCH_REG_88 0x1250160 +#define mmACP_SCRATCH_REG_89 0x1250164 +#define mmACP_SCRATCH_REG_90 0x1250168 +#define mmACP_SCRATCH_REG_91 0x125016C +#define mmACP_SCRATCH_REG_92 0x1250170 +#define mmACP_SCRATCH_REG_93 0x1250174 +#define mmACP_SCRATCH_REG_94 0x1250178 +#define mmACP_SCRATCH_REG_95 0x125017C +#define mmACP_SCRATCH_REG_96 0x1250180 +#define mmACP_SCRATCH_REG_97 0x1250184 +#define mmACP_SCRATCH_REG_98 0x1250188 +#define mmACP_SCRATCH_REG_99 0x125018C +#define mmACP_SCRATCH_REG_100 0x1250190 +#define mmACP_SCRATCH_REG_101 0x1250194 +#define mmACP_SCRATCH_REG_102 0x1250198 +#define mmACP_SCRATCH_REG_103 0x125019C +#define mmACP_SCRATCH_REG_104 0x12501A0 +#define mmACP_SCRATCH_REG_105 0x12501A4 +#define mmACP_SCRATCH_REG_106 0x12501A8 +#define mmACP_SCRATCH_REG_107 0x12501AC +#define mmACP_SCRATCH_REG_108 0x12501B0 +#define mmACP_SCRATCH_REG_109 0x12501B4 +#define mmACP_SCRATCH_REG_110 0x12501B8 +#define mmACP_SCRATCH_REG_111 0x12501BC +#define mmACP_SCRATCH_REG_112 0x12501C0 +#define mmACP_SCRATCH_REG_113 0x12501C4 +#define mmACP_SCRATCH_REG_114 0x12501C8 +#define mmACP_SCRATCH_REG_115 0x12501CC +#define mmACP_SCRATCH_REG_116 0x12501D0 +#define mmACP_SCRATCH_REG_117 0x12501D4 +#define mmACP_SCRATCH_REG_118 0x12501D8 +#define mmACP_SCRATCH_REG_119 0x12501DC +#define mmACP_SCRATCH_REG_120 0x12501E0 +#define mmACP_SCRATCH_REG_121 0x12501E4 +#define mmACP_SCRATCH_REG_122 0x12501E8 +#define mmACP_SCRATCH_REG_123 0x12501EC +#define mmACP_SCRATCH_REG_124 0x12501F0 +#define mmACP_SCRATCH_REG_125 0x12501F4 +#define mmACP_SCRATCH_REG_126 0x12501F8 +#define mmACP_SCRATCH_REG_127 0x12501FC +#define mmACP_SCRATCH_REG_128 0x1250200 + + +// Registers from ACP_SW_ACLK block + +#define mmSW_CORB_Base_Address 0x1243200 +#define mmSW_CORB_Write_Pointer 0x1243204 +#define mmSW_CORB_Read_Pointer 0x1243208 +#define mmSW_CORB_Control 0x124320C +#define mmSW_CORB_Size 0x1243214 +#define mmSW_RIRB_Base_Address 0x1243218 +#define mmSW_RIRB_Write_Pointer 0x124321C +#define mmSW_RIRB_Response_Interrupt_Count 0x1243220 +#define mmSW_RIRB_Control 0x1243224 +#define mmSW_RIRB_Size 0x1243228 +#define mmSW_RIRB_FIFO_MIN_THDL 0x124322C +#define mmSW_imm_cmd_UPPER_WORD 0x1243230 +#define mmSW_imm_cmd_LOWER_QWORD 0x1243234 +#define mmSW_imm_resp_UPPER_WORD 0x1243238 +#define mmSW_imm_resp_LOWER_QWORD 0x124323C +#define mmSW_imm_cmd_sts 0x1243240 +#define mmSW_BRA_BASE_ADDRESS 0x1243244 +#define mmSW_BRA_TRANSFER_SIZE 0x1243248 +#define mmSW_BRA_DMA_BUSY 0x124324C +#define mmSW_BRA_RESP 0x1243250 +#define mmSW_BRA_RESP_FRAME_ADDR 0x1243254 +#define mmSW_BRA_CURRENT_TRANSFER_SIZE 0x1243258 +#define mmSW_STATE_CHANGE_STATUS_0TO7 0x124325C +#define mmSW_STATE_CHANGE_STATUS_8TO11 0x1243260 +#define mmSW_STATE_CHANGE_STATUS_MASK_0to7 0x1243264 +#define mmSW_STATE_CHANGE_STATUS_MASK_8to11 0x1243268 +#define mmSW_CLK_FREQUENCY_CTRL 0x124326C +#define mmSW_ERROR_INTR_MASK 0x1243270 +#define mmSW_PHY_TEST_MODE_DATA_OFF 0x1243274 + + +// Registers from ACP_SW_SWCLK block + +#define mmACP_SW_EN 0x1243000 +#define mmACP_SW_EN_STATUS 0x1243004 +#define mmACP_SW_FRAMESIZE 0x1243008 +#define mmACP_SW_SSP_Counter 0x124300C +#define mmACP_SW_Audio_TX_EN 0x1243010 +#define mmACP_SW_Audio_TX_EN_STATUS 0x1243014 +#define mmACP_SW_Audio_TX_Frame_Format 0x1243018 +#define mmACP_SW_Audio_TX_SampleInterval 0x124301C +#define mmACP_SW_Audio_TX_Hctrl_DP0 0x1243020 +#define mmACP_SW_Audio_TX_Hctrl_DP1 0x1243024 +#define mmACP_SW_Audio_TX_Hctrl_DP2 0x1243028 +#define mmACP_SW_Audio_TX_Hctrl_DP3 0x124302C +#define mmACP_SW_Audio_TX_offset_DP0 0x1243030 +#define mmACP_SW_Audio_TX_offset_DP1 0x1243034 +#define mmACP_SW_Audio_TX_offset_DP2 0x1243038 +#define mmACP_SW_Audio_TX_offset_DP3 0x124303C +#define mmACP_SW_Audio_TX_Channel_Enable_DP0 0x1243040 +#define mmACP_SW_Audio_TX_Channel_Enable_DP1 0x1243044 +#define mmACP_SW_Audio_TX_Channel_Enable_DP2 0x1243048 +#define mmACP_SW_Audio_TX_Channel_Enable_DP3 0x124304C +#define mmACP_SW_BT_TX_EN 0x1243050 +#define mmACP_SW_BT_TX_EN_STATUS 0x1243054 +#define mmACP_SW_BT_TX_Frame_Format 0x1243058 +#define mmACP_SW_BT_TX_SampleInterval 0x124305C +#define mmACP_SW_BT_TX_Hctrl 0x1243060 +#define mmACP_SW_BT_TX_offset 0x1243064 +#define mmACP_SW_BT_TX_Channel_Enable_DP0 0x1243068 +#define mmACP_SW_Headset_TX_EN 0x124306C +#define mmACP_SW_Headset_TX_EN_STATUS 0x1243070 +#define mmACP_SW_Headset_TX_Frame_Format 0x1243074 +#define mmACP_SW_Headset_TX_SampleInterval 0x1243078 +#define mmACP_SW_Headset_TX_Hctrl 0x124307C +#define mmACP_SW_Headset_TX_offset 0x1243080 +#define mmACP_SW_Headset_TX_Channel_Enable_DP0 0x1243084 +#define mmACP_SW_Audio_RX_EN 0x1243088 +#define mmACP_SW_Audio_RX_EN_STATUS 0x124308C +#define mmACP_SW_Audio_RX_Frame_Format 0x1243090 +#define mmACP_SW_Audio_RX_SampleInterval 0x1243094 +#define mmACP_SW_Audio_RX_Hctrl_DP0 0x1243098 +#define mmACP_SW_Audio_RX_Hctrl_DP1 0x124309C +#define mmACP_SW_Audio_RX_Hctrl_DP2 0x1243100 +#define mmACP_SW_Audio_RX_Hctrl_DP3 0x1243104 +#define mmACP_SW_Audio_RX_offset_DP0 0x1243108 +#define mmACP_SW_Audio_RX_offset_DP1 0x124310C +#define mmACP_SW_Audio_RX_offset_DP2 0x1243110 +#define mmACP_SW_Audio_RX_offset_DP3 0x1243114 +#define mmACP_SW_Audio_RX_Channel_Enable_DP0 0x1243118 +#define mmACP_SW_Audio_RX_Channel_Enable_DP1 0x124311C +#define mmACP_SW_Audio_RX_Channel_Enable_DP2 0x1243120 +#define mmACP_SW_Audio_RX_Channel_Enable_DP3 0x1243124 +#define mmACP_SW_BT_RX_EN 0x1243128 +#define mmACP_SW_BT_RX_EN_STATUS 0x124312C +#define mmACP_SW_BT_RX_Frame_Format 0x1243130 +#define mmACP_SW_BT_RX_SampleInterval 0x1243134 +#define mmACP_SW_BT_RX_Hctrl 0x1243138 +#define mmACP_SW_BT_RX_offset 0x124313C +#define mmACP_SW_BT_RX_Channel_Enable_DP0 0x1243140 +#define mmACP_SW_Headset_RX_EN 0x1243144 +#define mmACP_SW_Headset_RX_EN_STATUS 0x1243148 +#define mmACP_SW_Headset_RX_Frame_Format 0x124314C +#define mmACP_SW_Headset_RX_SampleInterval 0x1243150 +#define mmACP_SW_Headset_RX_Hctrl 0x1243154 +#define mmACP_SW_Headset_RX_offset 0x1243158 +#define mmACP_SW_Headset_RX_Channel_Enable_DP0 0x124315C +#define mmACP_SW_BPT_PORT_EN 0x1243160 +#define mmACP_SW_BPT_PORT_EN_STATUS 0x1243164 +#define mmACP_SW_BPT_PORT_Frame_Format 0x1243168 +#define mmACP_SW_BPT_PORT_SampleInterval 0x124316C +#define mmACP_SW_BPT_PORT_Hctrl 0x1243170 +#define mmACP_SW_BPT_PORT_offset 0x1243174 +#define mmACP_SW_BPT_PORT_Channel_Enable 0x1243178 +#define mmACP_SW_BPT_PORT_First_byte_addr 0x124317C +#define mmACP_SW_CLK_RESUME_CTRL 0x1243180 +#define mmACP_SW_CLK_RESUME_Delay_Cntr 0x1243184 +#define mmACP_SW_BUS_RESET_CTRL 0x1243188 +#define mmACP_SW_PRBS_ERR_STATUS 0x124318C + + +// Registers from ACP_AUDIO_BUFFERS block + +#define mmACP_I2S_RX_RINGBUFADDR 0x1242000 +#define mmACP_I2S_RX_RINGBUFSIZE 0x1242004 +#define mmACP_I2S_RX_LINKPOSITIONCNTR 0x1242008 +#define mmACP_I2S_RX_FIFOADDR 0x124200C +#define mmACP_I2S_RX_FIFOSIZE 0x1242010 +#define mmACP_I2S_RX_DMA_SIZE 0x1242014 +#define mmACP_I2S_RX_LINEARPOSITIONCNTR_HIGH 0x1242018 +#define mmACP_I2S_RX_LINEARPOSITIONCNTR_LOW 0x124201C +#define mmACP_I2S_RX_INTR_WATERMARK_SIZE 0x1242020 +#define mmACP_I2S_TX_RINGBUFADDR 0x1242024 +#define mmACP_I2S_TX_RINGBUFSIZE 0x1242028 +#define mmACP_I2S_TX_LINKPOSITIONCNTR 0x124202C +#define mmACP_I2S_TX_FIFOADDR 0x1242030 +#define mmACP_I2S_TX_FIFOSIZE 0x1242034 +#define mmACP_I2S_TX_DMA_SIZE 0x1242038 +#define mmACP_I2S_TX_LINEARPOSITIONCNTR_HIGH 0x124203C +#define mmACP_I2S_TX_LINEARPOSITIONCNTR_LOW 0x1242040 +#define mmACP_I2S_TX_INTR_WATERMARK_SIZE 0x1242044 +#define mmACP_BT_RX_RINGBUFADDR 0x1242048 +#define mmACP_BT_RX_RINGBUFSIZE 0x124204C +#define mmACP_BT_RX_LINKPOSITIONCNTR 0x1242050 +#define mmACP_BT_RX_FIFOADDR 0x1242054 +#define mmACP_BT_RX_FIFOSIZE 0x1242058 +#define mmACP_BT_RX_DMA_SIZE 0x124205C +#define mmACP_BT_RX_LINEARPOSITIONCNTR_HIGH 0x1242060 +#define mmACP_BT_RX_LINEARPOSITIONCNTR_LOW 0x1242064 +#define mmACP_BT_RX_INTR_WATERMARK_SIZE 0x1242068 +#define mmACP_BT_TX_RINGBUFADDR 0x124206C +#define mmACP_BT_TX_RINGBUFSIZE 0x1242070 +#define mmACP_BT_TX_LINKPOSITIONCNTR 0x1242074 +#define mmACP_BT_TX_FIFOADDR 0x1242078 +#define mmACP_BT_TX_FIFOSIZE 0x124207C +#define mmACP_BT_TX_DMA_SIZE 0x1242080 +#define mmACP_BT_TX_LINEARPOSITIONCNTR_HIGH 0x1242084 +#define mmACP_BT_TX_LINEARPOSITIONCNTR_LOW 0x1242088 +#define mmACP_BT_TX_INTR_WATERMARK_SIZE 0x124208C +#define mmACP_HS_RX_RINGBUFADDR 0x1242090 +#define mmACP_HS_RX_RINGBUFSIZE 0x1242094 +#define mmACP_HS_RX_LINKPOSITIONCNTR 0x1242098 +#define mmACP_HS_RX_FIFOADDR 0x124209C +#define mmACP_HS_RX_FIFOSIZE 0x12420A0 +#define mmACP_HS_RX_DMA_SIZE 0x12420A4 +#define mmACP_HS_RX_LINEARPOSITIONCNTR_HIGH 0x12420A8 +#define mmACP_HS_RX_LINEARPOSITIONCNTR_LOW 0x12420AC +#define mmACP_HS_RX_INTR_WATERMARK_SIZE 0x12420B0 +#define mmACP_HS_TX_RINGBUFADDR 0x12420B4 +#define mmACP_HS_TX_RINGBUFSIZE 0x12420B8 +#define mmACP_HS_TX_LINKPOSITIONCNTR 0x12420BC +#define mmACP_HS_TX_FIFOADDR 0x12420C0 +#define mmACP_HS_TX_FIFOSIZE 0x12420C4 +#define mmACP_HS_TX_DMA_SIZE 0x12420C8 +#define mmACP_HS_TX_LINEARPOSITIONCNTR_HIGH 0x12420CC +#define mmACP_HS_TX_LINEARPOSITIONCNTR_LOW 0x12420D0 +#define mmACP_HS_TX_INTR_WATERMARK_SIZE 0x12420D4 + + +// Registers from ACP_I2S_TDM block + +#define mmACP_I2STDM_IER 0x1242400 +#define mmACP_I2STDM_IRER 0x1242404 +#define mmACP_I2STDM_RXFRMT 0x1242408 +#define mmACP_I2STDM_ITER 0x124240C +#define mmACP_I2STDM_TXFRMT 0x1242410 + + +// Registers from ACP_BT_TDM block + +#define mmACP_BTTDM_IER 0x1242800 +#define mmACP_BTTDM_IRER 0x1242804 +#define mmACP_BTTDM_RXFRMT 0x1242808 +#define mmACP_BTTDM_ITER 0x124280C +#define mmACP_BTTDM_TXFRMT 0x1242810 + + +// Registers from AZALIA_IP block + +#define mmAudio_Az_Global_Capabilities 0x1200000 +#define mmAudio_Az_Minor_Version 0x1200002 +#define mmAudio_Az_Major_Version 0x1200003 +#define mmAudio_Az_Output_Payload_Capability 0x1200004 +#define mmAudio_Az_Input_Payload_Capability 0x1200006 +#define mmAudio_Az_Global_Control 0x1200008 +#define mmAudio_Az_Wake_Enable 0x120000C +#define mmAudio_Az_State_Change_Status 0x120000E +#define mmAudio_Az_Global_Status 0x1200010 +#define mmAudio_Az_Linked_List_Capability_Header 0x1200014 +#define mmAudio_Az_Output_Stream_Payload_Capability 0x1200018 +#define mmAudio_Az_Input_Stream_Payload_Capability 0x120001A +#define mmAudio_Az_Interrupt_Control 0x1200020 +#define mmAudio_Az_Interrupt_Status 0x1200024 +#define mmAudio_Az_Wall_Clock_Counter 0x1200030 +#define mmAudio_Az_Stream_Synchronization 0x1200038 +#define mmAudio_Az_CORB_Lower_Base_Address 0x1200040 +#define mmAudio_Az_CORB_Upper_Base_Address 0x1200044 +#define mmAudio_Az_CORB_Write_Pointer 0x1200048 +#define mmAudio_Az_CORB_Read_Pointer 0x120004A +#define mmAudio_Az_CORB_Control 0x120004C +#define mmAudio_Az_CORB_Status 0x120004D +#define mmAudio_Az_CORB_Size 0x120004E +#define mmAudio_Az_RIRB_Lower_Base_Address 0x1200050 +#define mmAudio_Az_RIRB_Upper_Base_Address 0x1200054 +#define mmAudio_Az_RIRB_Write_Pointer 0x1200058 +#define mmAudio_Az_RIRB_Response_Interrupt_Count 0x120005A +#define mmAudio_Az_RIRB_Control 0x120005C +#define mmAudio_Az_RIRB_Status 0x120005D +#define mmAudio_Az_RIRB_Size 0x120005E +#define mmAudio_Az_Immediate_Command_Output_Interface 0x1200060 +#define mmAudio_Az_Immediate_Response_Input_Interface 0x1200064 +#define mmAudio_Az_Immediate_Command_Status 0x1200068 +#define mmAudio_Az_DPLBASE 0x1200070 +#define mmAudio_Az_DPUBASE 0x1200074 +#define mmAudio_Az_Input_SD0CTL_and_STS 0x1200080 +#define mmAudio_Az_Input_SD0LPIB 0x1200084 +#define mmAudio_Az_Input_SD0CBL 0x1200088 +#define mmAudio_Az_Input_SD0LVI 0x120008C +#define mmAudio_Az_Input_SD0FIFOS 0x1200090 +#define mmAudio_Az_Input_SD0FMT 0x1200092 +#define mmAudio_Az_Input_SD0BDPL 0x1200098 +#define mmAudio_Az_Input_SD0BDPU 0x120009C +#define mmAudio_Az_Input_SD1CTL_and_STS 0x12000A0 +#define mmAudio_Az_Input_SD1LPIB 0x12000A4 +#define mmAudio_Az_Input_SD1CBL 0x12000A8 +#define mmAudio_Az_Input_SD1LVI 0x12000AC +#define mmAudio_Az_Input_SD1FIFOS 0x12000B0 +#define mmAudio_Az_Input_SD1FMT 0x12000B2 +#define mmAudio_Az_Input_SD1BDPL 0x12000B8 +#define mmAudio_Az_Input_SD1BDPU 0x12000BC +#define mmAudio_Az_Input_SD2CTL_and_STS 0x12000C0 +#define mmAudio_Az_Input_SD2LPIB 0x12000C4 +#define mmAudio_Az_Input_SD2CBL 0x12000C8 +#define mmAudio_Az_Input_SD2LVI 0x12000CC +#define mmAudio_Az_Input_SD2FIFOS 0x12000D0 +#define mmAudio_Az_Input_SD2FMT 0x12000D2 +#define mmAudio_Az_Input_SD2BDPL 0x12000D8 +#define mmAudio_Az_Input_SD2BDPU 0x12000DC +#define mmAudio_Az_Input_SD3CTL_and_STS 0x12000E0 +#define mmAudio_Az_Input_SD3LPIB 0x12000E4 +#define mmAudio_Az_Input_SD3CBL 0x12000E8 +#define mmAudio_Az_Input_SD3LVI 0x12000EC +#define mmAudio_Az_Input_SD3FIFOS 0x12000F0 +#define mmAudio_Az_Input_SD3FMT 0x12000F2 +#define mmAudio_Az_Input_SD3BDPL 0x12000F8 +#define mmAudio_Az_Input_SD3BDPU 0x12000FC +#define mmAudio_Az_Output_SD0CTL_and_STS 0x1200100 +#define mmAudio_Az_Output_SD0LPIB 0x1200104 +#define mmAudio_Az_Output_SD0CBL 0x1200108 +#define mmAudio_Az_Output_SD0LVI 0x120010C +#define mmAudio_Az_Output_SD0FIFOS 0x1200110 +#define mmAudio_Az_Output_SD0FMT 0x1200112 +#define mmAudio_Az_Output_SD0BDPL 0x1200118 +#define mmAudio_Az_Output_SD0BDPU 0x120011C +#define mmAudio_Az_Output_SD1CTL_and_STS 0x1200120 +#define mmAudio_Az_Output_SD1LPIB 0x1200124 +#define mmAudio_Az_Output_SD1CBL 0x1200128 +#define mmAudio_Az_Output_SD1LVI 0x120012C +#define mmAudio_Az_Output_SD1FIFOS 0x1200130 +#define mmAudio_Az_Output_SD1FMT 0x1200132 +#define mmAudio_Az_Output_SD1BDPL 0x1200138 +#define mmAudio_Az_Output_SD1BDPU 0x120013C +#define mmAudio_Az_Output_SD2CTL_and_STS 0x1200140 +#define mmAudio_Az_Output_SD2LPIB 0x1200144 +#define mmAudio_Az_Output_SD2CBL 0x1200148 +#define mmAudio_Az_Output_SD2LVI 0x120014C +#define mmAudio_Az_Output_SD2FIFOS 0x1200150 +#define mmAudio_Az_Output_SD2FMT 0x1200152 +#define mmAudio_Az_Output_SD2BDPL 0x1200158 +#define mmAudio_Az_Output_SD2BDPU 0x120015C +#define mmAudio_Az_Output_SD3CTL_and_STS 0x1200160 +#define mmAudio_Az_Output_SD3LPIB 0x1200164 +#define mmAudio_Az_Output_SD3CBL 0x1200168 +#define mmAudio_Az_Output_SD3LVI 0x120016C +#define mmAudio_Az_Output_SD3FIFOS 0x1200170 +#define mmAudio_Az_Output_SD3FMT 0x1200172 +#define mmAudio_Az_Output_SD3BDPL 0x1200178 +#define mmAudio_Az_Output_SD3BDPU 0x120017C +#define mmAudioAZ_Misc_Control_Register_1 0x1200180 +#define mmAudioAZ_Misc_Control_Register_2 0x1200182 +#define mmAudioAZ_Misc_Control_Register_3 0x1200183 +#define mmAudio_AZ_Multiple_Links_Capability_Header 0x1200200 +#define mmAudio_AZ_Multiple_Links_Capability_Declaration 0x1200204 +#define mmAudio_AZ_Link0_Capabilities 0x1200240 +#define mmAudio_AZ_Link0_Control 0x1200244 +#define mmAudio_AZ_Link0_Output_Stream_ID 0x1200248 +#define mmAudio_AZ_Link0_SDI_Identifier 0x120024C +#define mmAudio_AZ_Link0_Per_Stream_Overhead 0x1200250 +#define mmAudio_AZ_Link0_Wall_Frame_Counter 0x1200258 +#define mmAudio_AZ_Link0_Output_Payload_Capability_L 0x1200260 +#define mmAudio_AZ_Link0_Output_Payload_Capability_U 0x1200264 +#define mmAudio_AZ_Link0_Input_Payload_Capability_L 0x1200270 +#define mmAudio_AZ_Link0_Input_Payload_Capability_U 0x1200274 +#define mmAudio_Az_Input_SD0LICBA 0x1202084 +#define mmAudio_Az_Input_SD1LICBA 0x12020A4 +#define mmAudio_Az_Input_SD2LICBA 0x12020C4 +#define mmAudio_Az_Input_SD3LICBA 0x12020E4 +#define mmAudio_Az_Output_SD0LICBA 0x1202104 +#define mmAudio_Az_Output_SD1LICBA 0x1202124 +#define mmAudio_Az_Output_SD2LICBA 0x1202144 +#define mmAudio_Az_Output_SD3LICBA 0x1202164 +#define mmAUDIO_AZ_POWER_MANAGEMENT_CONTROL 0x1204000 +#define mmAUDIO_AZ_IOC_SOFTRST_CONTROL 0x1204004 +#define mmAUDIO_AZ_IOC_CLKGATE_CONTROL 0x1204008 + + +// Registers from ACP_AZALIA block + +#define mmACP_AZ_PAGE0_LBASE_ADDR 0x1243800 +#define mmACP_AZ_PAGE0_UBASE_ADDR 0x1243804 +#define mmACP_AZ_PAGE0_PGEN_SIZE 0x1243808 +#define mmACP_AZ_PAGE0_OFFSET 0x124380C +#define mmACP_AZ_PAGE1_LBASE_ADDR 0x1243810 +#define mmACP_AZ_PAGE1_UBASE_ADDR 0x1243814 +#define mmACP_AZ_PAGE1_PGEN_SIZE 0x1243818 +#define mmACP_AZ_PAGE1_OFFSET 0x124381C +#define mmACP_AZ_PAGE2_LBASE_ADDR 0x1243820 +#define mmACP_AZ_PAGE2_UBASE_ADDR 0x1243824 +#define mmACP_AZ_PAGE2_PGEN_SIZE 0x1243828 +#define mmACP_AZ_PAGE2_OFFSET 0x124382C +#define mmACP_AZ_PAGE3_LBASE_ADDR 0x1243830 +#define mmACP_AZ_PAGE3_UBASE_ADDR 0x1243834 +#define mmACP_AZ_PAGE3_PGEN_SIZE 0x1243838 +#define mmACP_AZ_PAGE3_OFFSET 0x124383C +#define mmACP_AZ_PAGE4_LBASE_ADDR 0x1243840 +#define mmACP_AZ_PAGE4_UBASE_ADDR 0x1243844 +#define mmACP_AZ_PAGE4_PGEN_SIZE 0x1243848 +#define mmACP_AZ_PAGE4_OFFSET 0x124384C +#define mmACP_AZ_PAGE5_LBASE_ADDR 0x1243850 +#define mmACP_AZ_PAGE5_UBASE_ADDR 0x1243854 +#define mmACP_AZ_PAGE5_PGEN_SIZE 0x1243858 +#define mmACP_AZ_PAGE5_OFFSET 0x124385C +#define mmACP_AZ_PAGE6_LBASE_ADDR 0x1243860 +#define mmACP_AZ_PAGE6_UBASE_ADDR 0x1243864 +#define mmACP_AZ_PAGE6_PGEN_SIZE 0x1243868 +#define mmACP_AZ_PAGE6_OFFSET 0x124386C +#define mmACP_AZ_PAGE7_LBASE_ADDR 0x1243870 +#define mmACP_AZ_PAGE7_UBASE_ADDR 0x1243874 +#define mmACP_AZ_PAGE7_PGEN_SIZE 0x1243878 +#define mmACP_AZ_PAGE7_OFFSET 0x124387C + + +#endif diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c new file mode 100644 index 000000000000..facec2472b34 --- /dev/null +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -0,0 +1,156 @@ +// SPDX-License-Identifier: GPL-2.0+ +// +// AMD ACP PCI Driver +// +//Copyright 2016 Advanced Micro Devices, Inc. + +#include <linux/pci.h> +#include <linux/module.h> +#include <linux/io.h> +#include <linux/platform_device.h> +#include <linux/interrupt.h> + +#include "acp3x.h" + +struct acp3x_dev_data { + void __iomem *acp3x_base; + bool acp3x_audio_mode; + struct resource *res; + struct platform_device *pdev; +}; + +static int snd_acp3x_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + int ret; + u32 addr, val; + struct acp3x_dev_data *adata; + struct platform_device_info pdevinfo; + unsigned int irqflags; + + if (pci_enable_device(pci)) { + dev_err(&pci->dev, "pci_enable_device failed\n"); + return -ENODEV; + } + + ret = pci_request_regions(pci, "AMD ACP3x audio"); + if (ret < 0) { + dev_err(&pci->dev, "pci_request_regions failed\n"); + goto disable_pci; + } + + adata = devm_kzalloc(&pci->dev, sizeof(struct acp3x_dev_data), + GFP_KERNEL); + if (!adata) { + ret = -ENOMEM; + goto release_regions; + } + + /* check for msi interrupt support */ + ret = pci_enable_msi(pci); + if (ret) + /* msi is not enabled */ + irqflags = IRQF_SHARED; + else + /* msi is enabled */ + irqflags = 0; + + addr = pci_resource_start(pci, 0); + adata->acp3x_base = ioremap(addr, pci_resource_len(pci, 0)); + if (!adata->acp3x_base) { + ret = -ENOMEM; + goto release_regions; + } + pci_set_master(pci); + pci_set_drvdata(pci, adata); + + val = rv_readl(adata->acp3x_base + mmACP_I2S_PIN_CONFIG); + switch (val) { + case I2S_MODE: + adata->res = devm_kzalloc(&pci->dev, + sizeof(struct resource) * 2, + GFP_KERNEL); + if (!adata->res) { + ret = -ENOMEM; + goto unmap_mmio; + } + + adata->res[0].name = "acp3x_i2s_iomem"; + adata->res[0].flags = IORESOURCE_MEM; + adata->res[0].start = addr; + adata->res[0].end = addr + (ACP3x_REG_END - ACP3x_REG_START); + + adata->res[1].name = "acp3x_i2s_irq"; + adata->res[1].flags = IORESOURCE_IRQ; + adata->res[1].start = pci->irq; + adata->res[1].end = pci->irq; + + adata->acp3x_audio_mode = ACP3x_I2S_MODE; + + memset(&pdevinfo, 0, sizeof(pdevinfo)); + pdevinfo.name = "acp3x_rv_i2s"; + pdevinfo.id = 0; + pdevinfo.parent = &pci->dev; + pdevinfo.num_res = 2; + pdevinfo.res = adata->res; + pdevinfo.data = &irqflags; + pdevinfo.size_data = sizeof(irqflags); + + adata->pdev = platform_device_register_full(&pdevinfo); + if (IS_ERR(adata->pdev)) { + dev_err(&pci->dev, "cannot register %s device\n", + pdevinfo.name); + ret = PTR_ERR(adata->pdev); + goto unmap_mmio; + } + break; + default: + dev_err(&pci->dev, "Invalid ACP audio mode : %d\n", val); + ret = -ENODEV; + goto unmap_mmio; + } + return 0; + +unmap_mmio: + pci_disable_msi(pci); + iounmap(adata->acp3x_base); +release_regions: + pci_release_regions(pci); +disable_pci: + pci_disable_device(pci); + + return ret; +} + +static void snd_acp3x_remove(struct pci_dev *pci) +{ + struct acp3x_dev_data *adata = pci_get_drvdata(pci); + + platform_device_unregister(adata->pdev); + iounmap(adata->acp3x_base); + + pci_disable_msi(pci); + pci_release_regions(pci); + pci_disable_device(pci); +} + +static const struct pci_device_id snd_acp3x_ids[] = { + { PCI_DEVICE(PCI_VENDOR_ID_AMD, 0x15e2), + .class = PCI_CLASS_MULTIMEDIA_OTHER << 8, + .class_mask = 0xffffff }, + { 0, }, +}; +MODULE_DEVICE_TABLE(pci, snd_acp3x_ids); + +static struct pci_driver acp3x_driver = { + .name = KBUILD_MODNAME, + .id_table = snd_acp3x_ids, + .probe = snd_acp3x_probe, + .remove = snd_acp3x_remove, +}; + +module_pci_driver(acp3x_driver); + +MODULE_AUTHOR("Maruthi.Bayyavarapu@amd.com"); +MODULE_DESCRIPTION("AMD ACP3x PCI driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 64b784e96f84..64f86f0b87e5 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -97,4 +97,16 @@ config SND_ATMEL_SOC_I2S help Say Y or M if you want to add support for Atmel ASoc driver for boards using I2S. + +config SND_SOC_MIKROE_PROTO + tristate "Support for Mikroe-PROTO board" + depends on OF + depends on SND_SOC_I2C_AND_SPI + select SND_SOC_WM8731 + help + Say Y or M if you want to add support for MikroElektronika PROTO Audio + Board. This board contains the WM8731 codec, which can be configured + using I2C over SDA (MPU Data Input) and SCL (MPU Clock Input) pins. + Both playback and capture are supported. + endif diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index cd87cb4bcff5..9f41bfa0fea3 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -17,6 +17,7 @@ snd-soc-sam9x5-wm8731-objs := sam9x5_wm8731.o snd-atmel-soc-classd-objs := atmel-classd.o snd-atmel-soc-pdmic-objs := atmel-pdmic.o snd-atmel-soc-tse850-pcm5142-objs := tse850-pcm5142.o +snd-soc-mikroe-proto-objs := mikroe-proto.o obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o @@ -24,3 +25,4 @@ obj-$(CONFIG_SND_AT91_SOC_SAM9X5_WM8731) += snd-soc-sam9x5-wm8731.o obj-$(CONFIG_SND_ATMEL_SOC_CLASSD) += snd-atmel-soc-classd.o obj-$(CONFIG_SND_ATMEL_SOC_PDMIC) += snd-atmel-soc-pdmic.o obj-$(CONFIG_SND_ATMEL_SOC_TSE850_PCM5142) += snd-atmel-soc-tse850-pcm5142.o +obj-$(CONFIG_SND_SOC_MIKROE_PROTO) += snd-soc-mikroe-proto.o diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index d3b69682d9c2..6291ec7f9dd6 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -1005,11 +1005,11 @@ static int asoc_ssc_init(struct device *dev) struct ssc_device *ssc = dev_get_drvdata(dev); int ret; - ret = snd_soc_register_component(dev, &atmel_ssc_component, + ret = devm_snd_soc_register_component(dev, &atmel_ssc_component, &atmel_ssc_dai, 1); if (ret) { dev_err(dev, "Could not register DAI: %d\n", ret); - goto err; + return ret; } if (ssc->pdata->use_dma) @@ -1019,15 +1019,10 @@ static int asoc_ssc_init(struct device *dev) if (ret) { dev_err(dev, "Could not register PCM: %d\n", ret); - goto err_unregister_dai; + return ret; } return 0; - -err_unregister_dai: - snd_soc_unregister_component(dev); -err: - return ret; } static void asoc_ssc_exit(struct device *dev) @@ -1038,8 +1033,6 @@ static void asoc_ssc_exit(struct device *dev) atmel_pcm_dma_platform_unregister(dev); else atmel_pcm_pdc_platform_unregister(dev); - - snd_soc_unregister_component(dev); } /** diff --git a/sound/soc/atmel/mikroe-proto.c b/sound/soc/atmel/mikroe-proto.c new file mode 100644 index 000000000000..d47aaa5bf75a --- /dev/null +++ b/sound/soc/atmel/mikroe-proto.c @@ -0,0 +1,165 @@ +/* + * ASoC driver for PROTO AudioCODEC (with a WM8731) + * + * Author: Florian Meier, <koalo@koalo.de> + * Copyright 2013 + * + * 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/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/jack.h> + +#include "../codecs/wm8731.h" + +#define XTAL_RATE 12288000 /* This is fixed on this board */ + +static int snd_proto_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + + /* Set proto sysclk */ + int ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, + XTAL_RATE, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "Failed to set WM8731 SYSCLK: %d\n", + ret); + return ret; + } + + return 0; +} + +static const struct snd_soc_dapm_widget snd_proto_widget[] = { + SND_SOC_DAPM_MIC("Microphone Jack", NULL), + SND_SOC_DAPM_HP("Headphone Jack", NULL), +}; + +static const struct snd_soc_dapm_route snd_proto_route[] = { + /* speaker connected to LHPOUT/RHPOUT */ + {"Headphone Jack", NULL, "LHPOUT"}, + {"Headphone Jack", NULL, "RHPOUT"}, + + /* mic is connected to Mic Jack, with WM8731 Mic Bias */ + {"MICIN", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Microphone Jack"}, +}; + +/* audio machine driver */ +static struct snd_soc_card snd_proto = { + .name = "snd_mikroe_proto", + .owner = THIS_MODULE, + .dapm_widgets = snd_proto_widget, + .num_dapm_widgets = ARRAY_SIZE(snd_proto_widget), + .dapm_routes = snd_proto_route, + .num_dapm_routes = ARRAY_SIZE(snd_proto_route), +}; + +static int snd_proto_probe(struct platform_device *pdev) +{ + struct snd_soc_dai_link *dai; + struct device_node *np = pdev->dev.of_node; + struct device_node *codec_np, *cpu_np; + struct device_node *bitclkmaster = NULL; + struct device_node *framemaster = NULL; + unsigned int dai_fmt; + int ret = 0; + + if (!np) { + dev_err(&pdev->dev, "No device node supplied\n"); + return -EINVAL; + } + + snd_proto.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&snd_proto, "model"); + if (ret) + return ret; + + dai = devm_kzalloc(&pdev->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; + + snd_proto.dai_link = dai; + snd_proto.num_links = 1; + + dai->name = "WM8731"; + dai->stream_name = "WM8731 HiFi"; + dai->codec_dai_name = "wm8731-hifi"; + dai->init = &snd_proto_init; + + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (!codec_np) { + dev_err(&pdev->dev, "audio-codec node missing\n"); + return -EINVAL; + } + dai->codec_of_node = codec_np; + + cpu_np = of_parse_phandle(np, "i2s-controller", 0); + if (!cpu_np) { + dev_err(&pdev->dev, "i2s-controller missing\n"); + return -EINVAL; + } + dai->cpu_of_node = cpu_np; + dai->platform_of_node = cpu_np; + + dai_fmt = snd_soc_of_parse_daifmt(np, NULL, + &bitclkmaster, &framemaster); + if (bitclkmaster != framemaster) { + dev_err(&pdev->dev, "Must be the same bitclock and frame master\n"); + return -EINVAL; + } + if (bitclkmaster) { + dai_fmt &= ~SND_SOC_DAIFMT_MASTER_MASK; + if (codec_np == bitclkmaster) + dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + else + dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + } + of_node_put(bitclkmaster); + of_node_put(framemaster); + dai->dai_fmt = dai_fmt; + + of_node_put(codec_np); + of_node_put(cpu_np); + + ret = snd_soc_register_card(&snd_proto); + if (ret && ret != -EPROBE_DEFER) + dev_err(&pdev->dev, + "snd_soc_register_card() failed: %d\n", ret); + + return ret; +} + +static int snd_proto_remove(struct platform_device *pdev) +{ + return snd_soc_unregister_card(&snd_proto); +} + +static const struct of_device_id snd_proto_of_match[] = { + { .compatible = "mikroe,mikroe-proto", }, + {}, +}; +MODULE_DEVICE_TABLE(of, snd_proto_of_match); + +static struct platform_driver snd_proto_driver = { + .driver = { + .name = "snd-mikroe-proto", + .of_match_table = snd_proto_of_match, + }, + .probe = snd_proto_probe, + .remove = snd_proto_remove, +}; + +module_platform_driver(snd_proto_driver); + +MODULE_AUTHOR("Florian Meier"); +MODULE_DESCRIPTION("ASoC Driver for PROTO board (WM8731)"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/tse850-pcm5142.c b/sound/soc/atmel/tse850-pcm5142.c index 3a1393283156..214adcad5419 100644 --- a/sound/soc/atmel/tse850-pcm5142.c +++ b/sound/soc/atmel/tse850-pcm5142.c @@ -1,44 +1,38 @@ -/* - * TSE-850 audio - ASoC driver for the Axentia TSE-850 with a PCM5142 codec - * - * Copyright (C) 2016 Axentia Technologies AB - * - * Author: Peter Rosin <peda@axentia.se> - * - * 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. - */ - -/* - * loop1 relays - * IN1 +---o +------------+ o---+ OUT1 - * \ / - * + + - * | / | - * +--o +--. | - * | add | | - * | V | - * | .---. | - * DAC +----------->|Sum|---+ - * | '---' | - * | | - * + + - * - * IN2 +---o--+------------+--o---+ OUT2 - * loop2 relays - * - * The 'loop1' gpio pin controlls two relays, which are either in loop - * position, meaning that input and output are directly connected, or - * they are in mixer position, meaning that the signal is passed through - * the 'Sum' mixer. Similarly for 'loop2'. - * - * In the above, the 'loop1' relays are inactive, thus feeding IN1 to the - * mixer (if 'add' is active) and feeding the mixer output to OUT1. The - * 'loop2' relays are active, short-cutting the TSE-850 from channel 2. - * IN1, IN2, OUT1 and OUT2 are TSE-850 connectors and DAC is the PCB name - * of the (filtered) output from the PCM5142 codec. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// TSE-850 audio - ASoC driver for the Axentia TSE-850 with a PCM5142 codec +// +// Copyright (C) 2016 Axentia Technologies AB +// +// Author: Peter Rosin <peda@axentia.se> +// +// loop1 relays +// IN1 +---o +------------+ o---+ OUT1 +// \ / +// + + +// | / | +// +--o +--. | +// | add | | +// | V | +// | .---. | +// DAC +----------->|Sum|---+ +// | '---' | +// | | +// + + +// +// IN2 +---o--+------------+--o---+ OUT2 +// loop2 relays +// +// The 'loop1' gpio pin controlls two relays, which are either in loop +// position, meaning that input and output are directly connected, or +// they are in mixer position, meaning that the signal is passed through +// the 'Sum' mixer. Similarly for 'loop2'. +// +// In the above, the 'loop1' relays are inactive, thus feeding IN1 to the +// mixer (if 'add' is active) and feeding the mixer output to OUT1. The +// 'loop2' relays are active, short-cutting the TSE-850 from channel 2. +// IN1, IN2, OUT1 and OUT2 are TSE-850 connectors and DAC is the PCB name +// of the (filtered) output from the PCM5142 codec. #include <linux/clk.h> #include <linux/gpio.h> @@ -452,4 +446,4 @@ module_platform_driver(tse850_driver); /* Module information */ MODULE_AUTHOR("Peter Rosin <peda@axentia.se>"); MODULE_DESCRIPTION("ALSA SoC driver for TSE-850 with PCM5142 codec"); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c index b733f1446353..b7c358b48d8d 100644 --- a/sound/soc/bcm/cygnus-ssp.c +++ b/sound/soc/bcm/cygnus-ssp.c @@ -1334,7 +1334,7 @@ static int cygnus_ssp_probe(struct platform_device *pdev) cygaud->active_ports = 0; dev_dbg(dev, "Registering %d DAIs\n", active_port_count); - err = snd_soc_register_component(dev, &cygnus_ssp_component, + err = devm_snd_soc_register_component(dev, &cygnus_ssp_component, cygnus_ssp_dai, active_port_count); if (err) { dev_err(dev, "snd_soc_register_dai failed\n"); @@ -1345,32 +1345,27 @@ static int cygnus_ssp_probe(struct platform_device *pdev) if (cygaud->irq_num <= 0) { dev_err(dev, "platform_get_irq failed\n"); err = cygaud->irq_num; - goto err_irq; + return err; } err = audio_clk_init(pdev, cygaud); if (err) { dev_err(dev, "audio clock initialization failed\n"); - goto err_irq; + return err; } err = cygnus_soc_platform_register(dev, cygaud); if (err) { dev_err(dev, "platform reg error %d\n", err); - goto err_irq; + return err; } return 0; - -err_irq: - snd_soc_unregister_component(dev); - return err; } static int cygnus_ssp_remove(struct platform_device *pdev) { cygnus_soc_platform_unregister(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); return 0; } diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index efb095dbcd71..62bdb7e333b8 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -35,6 +35,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ADAU7002 select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER + select SND_SOC_AK4118 if I2C select SND_SOC_AK4458 if I2C select SND_SOC_AK4535 if I2C select SND_SOC_AK4554 @@ -82,6 +83,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ES7241 select SND_SOC_GTM601 select SND_SOC_HDAC_HDMI + select SND_SOC_HDAC_HDA select SND_SOC_ICS43432 select SND_SOC_INNO_RK3036 select SND_SOC_ISABELLE if I2C @@ -109,6 +111,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MT6351 if MTK_PMIC_WRAP select SND_SOC_NAU8540 if I2C select SND_SOC_NAU8810 if I2C + select SND_SOC_NAU8822 if I2C select SND_SOC_NAU8824 if I2C select SND_SOC_NAU8825 if I2C select SND_SOC_HDMI_CODEC @@ -119,6 +122,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM186X_I2C if I2C select SND_SOC_PCM186X_SPI if SPI_MASTER select SND_SOC_PCM3008 + select SND_SOC_PCM3060_I2C if I2C + select SND_SOC_PCM3060_SPI if SPI_MASTER select SND_SOC_PCM3168A_I2C if I2C select SND_SOC_PCM3168A_SPI if SPI_MASTER select SND_SOC_PCM5102A @@ -388,6 +393,11 @@ config SND_SOC_AK4104 tristate "AKM AK4104 CODEC" depends on SPI_MASTER +config SND_SOC_AK4118 + tristate "AKM AK4118 CODEC" + depends on I2C + select REGMAP_I2C + config SND_SOC_AK4458 tristate "AKM AK4458 CODEC" depends on I2C @@ -575,7 +585,11 @@ config SND_SOC_DA9055 tristate config SND_SOC_DMIC - tristate + tristate "Generic Digital Microphone CODEC" + depends on GPIOLIB + help + Enable support for the Generic Digital Microphone CODEC. + Select this if your sound card has DMICs. config SND_SOC_HDMI_CODEC tristate @@ -615,6 +629,10 @@ config SND_SOC_HDAC_HDMI select SND_PCM_ELD select HDMI +config SND_SOC_HDAC_HDA + tristate + select SND_HDA + config SND_SOC_ICS43432 tristate @@ -629,7 +647,8 @@ config SND_SOC_LM49453 tristate config SND_SOC_MAX98088 - tristate + tristate "Maxim MAX98088/9 Low-Power, Stereo Audio Codec" + depends on I2C config SND_SOC_MAX98090 tristate @@ -732,6 +751,21 @@ config SND_SOC_PCM186X_SPI config SND_SOC_PCM3008 tristate +config SND_SOC_PCM3060 + tristate + +config SND_SOC_PCM3060_I2C + tristate "Texas Instruments PCM3060 CODEC - I2C" + depends on I2C + select SND_SOC_PCM3060 + select REGMAP_I2C + +config SND_SOC_PCM3060_SPI + tristate "Texas Instruments PCM3060 CODEC - SPI" + depends on SPI_MASTER + select SND_SOC_PCM3060 + select REGMAP_SPI + config SND_SOC_PCM3168A tristate @@ -1299,6 +1333,10 @@ config SND_SOC_NAU8810 tristate "Nuvoton Technology Corporation NAU88C10 CODEC" depends on I2C +config SND_SOC_NAU8822 + tristate "Nuvoton Technology Corporation NAU88C22 CODEC" + depends on I2C + config SND_SOC_NAU8824 tristate "Nuvoton Technology Corporation NAU88L24 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 7ae7c85e8219..66f55d185620 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -27,6 +27,7 @@ snd-soc-adav801-objs := adav801.o snd-soc-adav803-objs := adav803.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o +snd-soc-ak4118-objs := ak4118.o snd-soc-ak4458-objs := ak4458.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4554-objs := ak4554.o @@ -78,6 +79,7 @@ snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o snd-soc-gtm601-objs := gtm601.o snd-soc-hdac-hdmi-objs := hdac_hdmi.o +snd-soc-hdac-hda-objs := hdac_hda.o snd-soc-ics43432-objs := ics43432.o snd-soc-inno-rk3036-objs := inno_rk3036.o snd-soc-isabelle-objs := isabelle.o @@ -106,6 +108,7 @@ snd-soc-msm8916-digital-objs := msm8916-wcd-digital.o snd-soc-mt6351-objs := mt6351.o snd-soc-nau8540-objs := nau8540.o snd-soc-nau8810-objs := nau8810.o +snd-soc-nau8822-objs := nau8822.o snd-soc-nau8824-objs := nau8824.o snd-soc-nau8825-objs := nau8825.o snd-soc-hdmi-codec-objs := hdmi-codec.o @@ -119,6 +122,9 @@ snd-soc-pcm186x-objs := pcm186x.o snd-soc-pcm186x-i2c-objs := pcm186x-i2c.o snd-soc-pcm186x-spi-objs := pcm186x-spi.o snd-soc-pcm3008-objs := pcm3008.o +snd-soc-pcm3060-objs := pcm3060.o +snd-soc-pcm3060-i2c-objs := pcm3060-i2c.o +snd-soc-pcm3060-spi-objs := pcm3060-spi.o snd-soc-pcm3168a-objs := pcm3168a.o snd-soc-pcm3168a-i2c-objs := pcm3168a-i2c.o snd-soc-pcm3168a-spi-objs := pcm3168a-spi.o @@ -285,6 +291,7 @@ obj-$(CONFIG_SND_SOC_ADAV801) += snd-soc-adav801.o obj-$(CONFIG_SND_SOC_ADAV803) += snd-soc-adav803.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o +obj-$(CONFIG_SND_SOC_AK4118) += snd-soc-ak4118.o obj-$(CONFIG_SND_SOC_AK4458) += snd-soc-ak4458.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4554) += snd-soc-ak4554.o @@ -338,6 +345,7 @@ obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o obj-$(CONFIG_SND_SOC_GTM601) += snd-soc-gtm601.o obj-$(CONFIG_SND_SOC_HDAC_HDMI) += snd-soc-hdac-hdmi.o +obj-$(CONFIG_SND_SOC_HDAC_HDA) += snd-soc-hdac-hda.o obj-$(CONFIG_SND_SOC_ICS43432) += snd-soc-ics43432.o obj-$(CONFIG_SND_SOC_INNO_RK3036) += snd-soc-inno-rk3036.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o @@ -366,6 +374,7 @@ obj-$(CONFIG_SND_SOC_MSM8916_WCD_DIGITAL) +=snd-soc-msm8916-digital.o obj-$(CONFIG_SND_SOC_MT6351) += snd-soc-mt6351.o obj-$(CONFIG_SND_SOC_NAU8540) += snd-soc-nau8540.o obj-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o +obj-$(CONFIG_SND_SOC_NAU8822) += snd-soc-nau8822.o obj-$(CONFIG_SND_SOC_NAU8824) += snd-soc-nau8824.o obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o @@ -379,6 +388,9 @@ obj-$(CONFIG_SND_SOC_PCM186X) += snd-soc-pcm186x.o obj-$(CONFIG_SND_SOC_PCM186X_I2C) += snd-soc-pcm186x-i2c.o obj-$(CONFIG_SND_SOC_PCM186X_SPI) += snd-soc-pcm186x-spi.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o +obj-$(CONFIG_SND_SOC_PCM3060) += snd-soc-pcm3060.o +obj-$(CONFIG_SND_SOC_PCM3060_I2C) += snd-soc-pcm3060-i2c.o +obj-$(CONFIG_SND_SOC_PCM3060_SPI) += snd-soc-pcm3060-spi.o obj-$(CONFIG_SND_SOC_PCM3168A) += snd-soc-pcm3168a.o obj-$(CONFIG_SND_SOC_PCM3168A_I2C) += snd-soc-pcm3168a-i2c.o obj-$(CONFIG_SND_SOC_PCM3168A_SPI) += snd-soc-pcm3168a-spi.o diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index be136e981653..bef3e9e74c26 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -518,7 +518,8 @@ static int adau1761_setup_digmic_jackdetect(struct snd_soc_component *component) ARRAY_SIZE(adau1761_jack_detect_controls)); if (ret) return ret; - case ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE: /* fallthrough */ + /* fall through */ + case ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE: ret = snd_soc_dapm_add_routes(dapm, adau1761_no_dmic_routes, ARRAY_SIZE(adau1761_no_dmic_routes)); if (ret) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 57169b8ff14e..3959e6ad113d 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -21,11 +21,18 @@ #include <linux/i2c.h> #include <linux/spi/spi.h> #include <linux/regmap.h> +#include <asm/unaligned.h> #include "sigmadsp.h" #include "adau17x1.h" #include "adau-utils.h" +#define ADAU17X1_SAFELOAD_TARGET_ADDRESS 0x0006 +#define ADAU17X1_SAFELOAD_TRIGGER 0x0007 +#define ADAU17X1_SAFELOAD_DATA 0x0001 +#define ADAU17X1_SAFELOAD_DATA_SIZE 20 +#define ADAU17X1_WORD_SIZE 4 + static const char * const adau17x1_capture_mixer_boost_text[] = { "Normal operation", "Boost Level 1", "Boost Level 2", "Boost Level 3", }; @@ -60,6 +67,9 @@ static const struct snd_kcontrol_new adau17x1_controls[] = { SOC_ENUM("Mic Bias Mode", adau17x1_mic_bias_mode_enum), }; +static int adau17x1_setup_firmware(struct snd_soc_component *component, + unsigned int rate); + static int adau17x1_pll_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -313,7 +323,7 @@ static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = { { "Capture", NULL, "Right Decimator" }, }; -bool adau17x1_has_dsp(struct adau *adau) +static bool adau17x1_has_dsp(struct adau *adau) { switch (adau->type) { case ADAU1761: @@ -324,7 +334,17 @@ bool adau17x1_has_dsp(struct adau *adau) return false; } } -EXPORT_SYMBOL_GPL(adau17x1_has_dsp); + +static bool adau17x1_has_safeload(struct adau *adau) +{ + switch (adau->type) { + case ADAU1761: + case ADAU1781: + return true; + default: + return false; + } +} static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) @@ -836,7 +856,7 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg) } EXPORT_SYMBOL_GPL(adau17x1_volatile_register); -int adau17x1_setup_firmware(struct snd_soc_component *component, +static int adau17x1_setup_firmware(struct snd_soc_component *component, unsigned int rate) { int ret; @@ -880,7 +900,6 @@ err: return ret; } -EXPORT_SYMBOL_GPL(adau17x1_setup_firmware); int adau17x1_add_widgets(struct snd_soc_component *component) { @@ -957,6 +976,56 @@ int adau17x1_resume(struct snd_soc_component *component) } EXPORT_SYMBOL_GPL(adau17x1_resume); +static int adau17x1_safeload(struct sigmadsp *sigmadsp, unsigned int addr, + const uint8_t bytes[], size_t len) +{ + uint8_t buf[ADAU17X1_WORD_SIZE]; + uint8_t data[ADAU17X1_SAFELOAD_DATA_SIZE]; + unsigned int addr_offset; + unsigned int nbr_words; + int ret; + + /* write data to safeload addresses. Check if len is not a multiple of + * 4 bytes, if so we need to zero pad. + */ + nbr_words = len / ADAU17X1_WORD_SIZE; + if ((len - nbr_words * ADAU17X1_WORD_SIZE) == 0) { + ret = regmap_raw_write(sigmadsp->control_data, + ADAU17X1_SAFELOAD_DATA, bytes, len); + } else { + nbr_words++; + memset(data, 0, ADAU17X1_SAFELOAD_DATA_SIZE); + memcpy(data, bytes, len); + ret = regmap_raw_write(sigmadsp->control_data, + ADAU17X1_SAFELOAD_DATA, data, + nbr_words * ADAU17X1_WORD_SIZE); + } + + if (ret < 0) + return ret; + + /* Write target address, target address is offset by 1 */ + addr_offset = addr - 1; + put_unaligned_be32(addr_offset, buf); + ret = regmap_raw_write(sigmadsp->control_data, + ADAU17X1_SAFELOAD_TARGET_ADDRESS, buf, ADAU17X1_WORD_SIZE); + if (ret < 0) + return ret; + + /* write nbr of words to trigger address */ + put_unaligned_be32(nbr_words, buf); + ret = regmap_raw_write(sigmadsp->control_data, + ADAU17X1_SAFELOAD_TRIGGER, buf, ADAU17X1_WORD_SIZE); + if (ret < 0) + return ret; + + return 0; +} + +static const struct sigmadsp_ops adau17x1_sigmadsp_ops = { + .safeload = adau17x1_safeload, +}; + int adau17x1_probe(struct device *dev, struct regmap *regmap, enum adau17x1_type type, void (*switch_mode)(struct device *dev), const char *firmware_name) @@ -1002,8 +1071,13 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap, dev_set_drvdata(dev, adau); if (firmware_name) { - adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap, NULL, - firmware_name); + if (adau17x1_has_safeload(adau)) { + adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap, + &adau17x1_sigmadsp_ops, firmware_name); + } else { + adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap, + NULL, firmware_name); + } if (IS_ERR(adau->sigmadsp)) { dev_warn(dev, "Could not find firmware file: %ld\n", PTR_ERR(adau->sigmadsp)); diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index e6fe87beec07..98a3b6f5bc96 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -68,10 +68,6 @@ int adau17x1_resume(struct snd_soc_component *component); extern const struct snd_soc_dai_ops adau17x1_dai_ops; -int adau17x1_setup_firmware(struct snd_soc_component *component, - unsigned int rate); -bool adau17x1_has_dsp(struct adau *adau); - #define ADAU17X1_CLOCK_CONTROL 0x4000 #define ADAU17X1_PLL_CONTROL 0x4002 #define ADAU17X1_REC_POWER_MGMT 0x4009 diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 32bc545c19cf..6dec8a65eafc 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -13,7 +13,7 @@ #include <linux/slab.h> #include <linux/spi/spi.h> #include <linux/of_device.h> -#include <linux/of_gpio.h> +#include <linux/gpio/consumer.h> #include <linux/regulator/consumer.h> #include <sound/asoundef.h> #include <sound/core.h> @@ -268,8 +268,8 @@ static const struct regmap_config ak4104_regmap = { static int ak4104_spi_probe(struct spi_device *spi) { - struct device_node *np = spi->dev.of_node; struct ak4104_private *ak4104; + struct gpio_desc *reset_gpiod; unsigned int val; int ret; @@ -297,19 +297,11 @@ static int ak4104_spi_probe(struct spi_device *spi) return ret; } - if (np) { - enum of_gpio_flags flags; - int gpio = of_get_named_gpio_flags(np, "reset-gpio", 0, &flags); - - if (gpio_is_valid(gpio)) { - ret = devm_gpio_request_one(&spi->dev, gpio, - flags & OF_GPIO_ACTIVE_LOW ? - GPIOF_OUT_INIT_LOW : GPIOF_OUT_INIT_HIGH, - "ak4104 reset"); - if (ret < 0) - return ret; - } - } + reset_gpiod = devm_gpiod_get_optional(&spi->dev, "reset", + GPIOD_OUT_HIGH); + if (IS_ERR(reset_gpiod) && + PTR_ERR(reset_gpiod) == -EPROBE_DEFER) + return -EPROBE_DEFER; /* read the 'reserved' register - according to the datasheet, it * should contain 0x5b. Not a good way to verify the presence of diff --git a/sound/soc/codecs/ak4118.c b/sound/soc/codecs/ak4118.c new file mode 100644 index 000000000000..238ab29f2bf4 --- /dev/null +++ b/sound/soc/codecs/ak4118.c @@ -0,0 +1,438 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * ak4118.c -- Asahi Kasei ALSA Soc Audio driver + * + * Copyright 2018 DEVIALET + */ + +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <linux/slab.h> + +#include <sound/asoundef.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#define AK4118_REG_CLK_PWR_CTL 0x00 +#define AK4118_REG_FORMAT_CTL 0x01 +#define AK4118_REG_IO_CTL0 0x02 +#define AK4118_REG_IO_CTL1 0x03 +#define AK4118_REG_INT0_MASK 0x04 +#define AK4118_REG_INT1_MASK 0x05 +#define AK4118_REG_RCV_STATUS0 0x06 +#define AK4118_REG_RCV_STATUS1 0x07 +#define AK4118_REG_RXCHAN_STATUS0 0x08 +#define AK4118_REG_RXCHAN_STATUS1 0x09 +#define AK4118_REG_RXCHAN_STATUS2 0x0a +#define AK4118_REG_RXCHAN_STATUS3 0x0b +#define AK4118_REG_RXCHAN_STATUS4 0x0c +#define AK4118_REG_TXCHAN_STATUS0 0x0d +#define AK4118_REG_TXCHAN_STATUS1 0x0e +#define AK4118_REG_TXCHAN_STATUS2 0x0f +#define AK4118_REG_TXCHAN_STATUS3 0x10 +#define AK4118_REG_TXCHAN_STATUS4 0x11 +#define AK4118_REG_BURST_PREAMB_PC0 0x12 +#define AK4118_REG_BURST_PREAMB_PC1 0x13 +#define AK4118_REG_BURST_PREAMB_PD0 0x14 +#define AK4118_REG_BURST_PREAMB_PD1 0x15 +#define AK4118_REG_QSUB_CTL 0x16 +#define AK4118_REG_QSUB_TRACK 0x17 +#define AK4118_REG_QSUB_INDEX 0x18 +#define AK4118_REG_QSUB_MIN 0x19 +#define AK4118_REG_QSUB_SEC 0x1a +#define AK4118_REG_QSUB_FRAME 0x1b +#define AK4118_REG_QSUB_ZERO 0x1c +#define AK4118_REG_QSUB_ABS_MIN 0x1d +#define AK4118_REG_QSUB_ABS_SEC 0x1e +#define AK4118_REG_QSUB_ABS_FRAME 0x1f +#define AK4118_REG_GPE 0x20 +#define AK4118_REG_GPDR 0x21 +#define AK4118_REG_GPSCR 0x22 +#define AK4118_REG_GPLR 0x23 +#define AK4118_REG_DAT_MASK_DTS 0x24 +#define AK4118_REG_RX_DETECT 0x25 +#define AK4118_REG_STC_DAT_DETECT 0x26 +#define AK4118_REG_RXCHAN_STATUS5 0x27 +#define AK4118_REG_TXCHAN_STATUS5 0x28 +#define AK4118_REG_MAX 0x29 + +#define AK4118_REG_FORMAT_CTL_DIF0 (1 << 4) +#define AK4118_REG_FORMAT_CTL_DIF1 (1 << 5) +#define AK4118_REG_FORMAT_CTL_DIF2 (1 << 6) + +struct ak4118_priv { + struct regmap *regmap; + struct gpio_desc *reset; + struct gpio_desc *irq; + struct snd_soc_component *component; +}; + +static const struct reg_default ak4118_reg_defaults[] = { + {AK4118_REG_CLK_PWR_CTL, 0x43}, + {AK4118_REG_FORMAT_CTL, 0x6a}, + {AK4118_REG_IO_CTL0, 0x88}, + {AK4118_REG_IO_CTL1, 0x48}, + {AK4118_REG_INT0_MASK, 0xee}, + {AK4118_REG_INT1_MASK, 0xb5}, + {AK4118_REG_RCV_STATUS0, 0x00}, + {AK4118_REG_RCV_STATUS1, 0x10}, + {AK4118_REG_TXCHAN_STATUS0, 0x00}, + {AK4118_REG_TXCHAN_STATUS1, 0x00}, + {AK4118_REG_TXCHAN_STATUS2, 0x00}, + {AK4118_REG_TXCHAN_STATUS3, 0x00}, + {AK4118_REG_TXCHAN_STATUS4, 0x00}, + {AK4118_REG_GPE, 0x77}, + {AK4118_REG_GPDR, 0x00}, + {AK4118_REG_GPSCR, 0x00}, + {AK4118_REG_GPLR, 0x00}, + {AK4118_REG_DAT_MASK_DTS, 0x3f}, + {AK4118_REG_RX_DETECT, 0x00}, + {AK4118_REG_STC_DAT_DETECT, 0x00}, + {AK4118_REG_TXCHAN_STATUS5, 0x00}, +}; + +static const char * const ak4118_input_select_txt[] = { + "RX0", "RX1", "RX2", "RX3", "RX4", "RX5", "RX6", "RX7", +}; +static SOC_ENUM_SINGLE_DECL(ak4118_insel_enum, AK4118_REG_IO_CTL1, 0x0, + ak4118_input_select_txt); + +static const struct snd_kcontrol_new ak4118_input_mux_controls = + SOC_DAPM_ENUM("Input Select", ak4118_insel_enum); + +static const char * const ak4118_iec958_fs_txt[] = { + "44100", "48000", "32000", "22050", "11025", "24000", "16000", "88200", + "8000", "96000", "64000", "176400", "192000", +}; + +static const int ak4118_iec958_fs_val[] = { + 0x0, 0x2, 0x3, 0x4, 0x5, 0x6, 0x7, 0x8, 0x9, 0xA, 0xB, 0xC, 0xE, +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(ak4118_iec958_fs_enum, AK4118_REG_RCV_STATUS1, + 0x4, 0x4, ak4118_iec958_fs_txt, + ak4118_iec958_fs_val); + +static struct snd_kcontrol_new ak4118_iec958_controls[] = { + SOC_SINGLE("IEC958 Parity Errors", AK4118_REG_RCV_STATUS0, 0, 1, 0), + SOC_SINGLE("IEC958 No Audio", AK4118_REG_RCV_STATUS0, 1, 1, 0), + SOC_SINGLE("IEC958 PLL Lock", AK4118_REG_RCV_STATUS0, 4, 1, 1), + SOC_SINGLE("IEC958 Non PCM", AK4118_REG_RCV_STATUS0, 6, 1, 0), + SOC_ENUM("IEC958 Sampling Freq", ak4118_iec958_fs_enum), +}; + +static const struct snd_soc_dapm_widget ak4118_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("INRX0"), + SND_SOC_DAPM_INPUT("INRX1"), + SND_SOC_DAPM_INPUT("INRX2"), + SND_SOC_DAPM_INPUT("INRX3"), + SND_SOC_DAPM_INPUT("INRX4"), + SND_SOC_DAPM_INPUT("INRX5"), + SND_SOC_DAPM_INPUT("INRX6"), + SND_SOC_DAPM_INPUT("INRX7"), + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, + &ak4118_input_mux_controls), +}; + +static const struct snd_soc_dapm_route ak4118_dapm_routes[] = { + {"Input Mux", "RX0", "INRX0"}, + {"Input Mux", "RX1", "INRX1"}, + {"Input Mux", "RX2", "INRX2"}, + {"Input Mux", "RX3", "INRX3"}, + {"Input Mux", "RX4", "INRX4"}, + {"Input Mux", "RX5", "INRX5"}, + {"Input Mux", "RX6", "INRX6"}, + {"Input Mux", "RX7", "INRX7"}, +}; + + +static int ak4118_set_dai_fmt_master(struct ak4118_priv *ak4118, + unsigned int format) +{ + int dif; + + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + dif = AK4118_REG_FORMAT_CTL_DIF0 | AK4118_REG_FORMAT_CTL_DIF2; + break; + case SND_SOC_DAIFMT_RIGHT_J: + dif = AK4118_REG_FORMAT_CTL_DIF0 | AK4118_REG_FORMAT_CTL_DIF1; + break; + case SND_SOC_DAIFMT_LEFT_J: + dif = AK4118_REG_FORMAT_CTL_DIF2; + break; + default: + return -ENOTSUPP; + } + + return dif; +} + +static int ak4118_set_dai_fmt_slave(struct ak4118_priv *ak4118, + unsigned int format) +{ + int dif; + + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + dif = AK4118_REG_FORMAT_CTL_DIF0 | AK4118_REG_FORMAT_CTL_DIF1 | + AK4118_REG_FORMAT_CTL_DIF2; + break; + case SND_SOC_DAIFMT_LEFT_J: + dif = AK4118_REG_FORMAT_CTL_DIF1 | AK4118_REG_FORMAT_CTL_DIF2; + break; + default: + return -ENOTSUPP; + } + + return dif; +} + +static int ak4118_set_dai_fmt(struct snd_soc_dai *dai, + unsigned int format) +{ + struct snd_soc_component *component = dai->component; + struct ak4118_priv *ak4118 = snd_soc_component_get_drvdata(component); + int dif; + int ret = 0; + + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + /* component is master */ + dif = ak4118_set_dai_fmt_master(ak4118, format); + break; + case SND_SOC_DAIFMT_CBS_CFS: + /*component is slave */ + dif = ak4118_set_dai_fmt_slave(ak4118, format); + break; + default: + ret = -ENOTSUPP; + goto exit; + } + + /* format not supported */ + if (dif < 0) { + ret = dif; + goto exit; + } + + ret = regmap_update_bits(ak4118->regmap, AK4118_REG_FORMAT_CTL, + AK4118_REG_FORMAT_CTL_DIF0 | + AK4118_REG_FORMAT_CTL_DIF1 | + AK4118_REG_FORMAT_CTL_DIF2, dif); + if (ret < 0) + goto exit; + +exit: + return ret; +} + +static int ak4118_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + return 0; +} + +static const struct snd_soc_dai_ops ak4118_dai_ops = { + .hw_params = ak4118_hw_params, + .set_fmt = ak4118_set_dai_fmt, +}; + +static struct snd_soc_dai_driver ak4118_dai = { + .name = "ak4118-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_22050 | 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_3LE | + SNDRV_PCM_FMTBIT_S24_LE + }, + .ops = &ak4118_dai_ops, +}; + +static irqreturn_t ak4118_irq_handler(int irq, void *data) +{ + struct ak4118_priv *ak4118 = data; + struct snd_soc_component *component = ak4118->component; + struct snd_kcontrol_new *kctl_new; + struct snd_kcontrol *kctl; + struct snd_ctl_elem_id *id; + unsigned int i; + + if (!component) + return IRQ_NONE; + + for (i = 0; i < ARRAY_SIZE(ak4118_iec958_controls); i++) { + kctl_new = &ak4118_iec958_controls[i]; + kctl = snd_soc_card_get_kcontrol(component->card, + kctl_new->name); + if (!kctl) + continue; + id = &kctl->id; + snd_ctl_notify(component->card->snd_card, + SNDRV_CTL_EVENT_MASK_VALUE, id); + } + + return IRQ_HANDLED; +} + +static int ak4118_probe(struct snd_soc_component *component) +{ + struct ak4118_priv *ak4118 = snd_soc_component_get_drvdata(component); + int ret = 0; + + ak4118->component = component; + + /* release reset */ + gpiod_set_value(ak4118->reset, 0); + + /* unmask all int1 sources */ + ret = regmap_write(ak4118->regmap, AK4118_REG_INT1_MASK, 0x00); + if (ret < 0) { + dev_err(component->dev, + "failed to write regmap 0x%x 0x%x: %d\n", + AK4118_REG_INT1_MASK, 0x00, ret); + return ret; + } + + /* rx detect enable on all channels */ + ret = regmap_write(ak4118->regmap, AK4118_REG_RX_DETECT, 0xff); + if (ret < 0) { + dev_err(component->dev, + "failed to write regmap 0x%x 0x%x: %d\n", + AK4118_REG_RX_DETECT, 0xff, ret); + return ret; + } + + ret = snd_soc_add_component_controls(component, ak4118_iec958_controls, + ARRAY_SIZE(ak4118_iec958_controls)); + if (ret) { + dev_err(component->dev, + "failed to add component kcontrols: %d\n", ret); + return ret; + } + + return 0; +} + +static void ak4118_remove(struct snd_soc_component *component) +{ + struct ak4118_priv *ak4118 = snd_soc_component_get_drvdata(component); + + /* hold reset */ + gpiod_set_value(ak4118->reset, 1); +} + +static const struct snd_soc_component_driver soc_component_drv_ak4118 = { + .probe = ak4118_probe, + .remove = ak4118_remove, + .dapm_widgets = ak4118_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4118_dapm_widgets), + .dapm_routes = ak4118_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ak4118_dapm_routes), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct regmap_config ak4118_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .reg_defaults = ak4118_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ak4118_reg_defaults), + + .cache_type = REGCACHE_NONE, + .max_register = AK4118_REG_MAX - 1, +}; + +static int ak4118_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ak4118_priv *ak4118; + int ret; + + ak4118 = devm_kzalloc(&i2c->dev, sizeof(struct ak4118_priv), + GFP_KERNEL); + if (ak4118 == NULL) + return -ENOMEM; + + ak4118->regmap = devm_regmap_init_i2c(i2c, &ak4118_regmap); + if (IS_ERR(ak4118->regmap)) + return PTR_ERR(ak4118->regmap); + + i2c_set_clientdata(i2c, ak4118); + + ak4118->reset = devm_gpiod_get(&i2c->dev, "reset", GPIOD_OUT_HIGH); + if (IS_ERR(ak4118->reset)) { + ret = PTR_ERR(ak4118->reset); + if (ret != -EPROBE_DEFER) + dev_err(&i2c->dev, "Failed to get reset: %d\n", ret); + return ret; + } + + ak4118->irq = devm_gpiod_get(&i2c->dev, "irq", GPIOD_IN); + if (IS_ERR(ak4118->irq)) { + ret = PTR_ERR(ak4118->irq); + if (ret != -EPROBE_DEFER) + dev_err(&i2c->dev, "Failed to get IRQ: %d\n", ret); + return ret; + } + + ret = devm_request_threaded_irq(&i2c->dev, gpiod_to_irq(ak4118->irq), + NULL, ak4118_irq_handler, + IRQF_TRIGGER_RISING | IRQF_ONESHOT, + "ak4118-irq", ak4118); + if (ret < 0) { + dev_err(&i2c->dev, "Fail to request_irq: %d\n", ret); + return ret; + } + + return snd_soc_register_component(&i2c->dev, &soc_component_drv_ak4118, + &ak4118_dai, 1); +} + +static int ak4118_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_component(&i2c->dev); + return 0; +} + +static const struct of_device_id ak4118_of_match[] = { + { .compatible = "asahi-kasei,ak4118", }, + {} +}; +MODULE_DEVICE_TABLE(of, ak4118_of_match); + +static const struct i2c_device_id ak4118_id_table[] = { + { "ak4118", 0 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, ak4118_id_table); + +static struct i2c_driver ak4118_i2c_driver = { + .driver = { + .name = "ak4118", + .of_match_table = of_match_ptr(ak4118_of_match), + }, + .id_table = ak4118_id_table, + .probe = ak4118_i2c_probe, + .remove = ak4118_i2c_remove, +}; + +module_i2c_driver(ak4118_i2c_driver); + +MODULE_DESCRIPTION("Asahi Kasei AK4118 ALSA SoC driver"); +MODULE_AUTHOR("Adrien Charruel <adrien.charruel@devialet.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index 299ada4dfaa0..70d4c89bd6fc 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -456,7 +456,7 @@ static int ak4458_startup(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_dai_ops ak4458_dai_ops = { +static const struct snd_soc_dai_ops ak4458_dai_ops = { .startup = ak4458_startup, .hw_params = ak4458_hw_params, .set_fmt = ak4458_set_dai_fmt, diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c index 448bb90c9c8e..8179512129d3 100644 --- a/sound/soc/codecs/ak5558.c +++ b/sound/soc/codecs/ak5558.c @@ -130,16 +130,12 @@ static int ak5558_hw_params(struct snd_pcm_substream *substream, u8 bits; int pcm_width = max(params_physical_width(params), ak5558->slot_width); - /* set master/slave audio interface */ - bits = snd_soc_component_read32(component, AK5558_02_CONTROL1); - bits &= ~AK5558_BITS; - switch (pcm_width) { case 16: - bits |= AK5558_DIF_24BIT_MODE; + bits = AK5558_DIF_24BIT_MODE; break; case 32: - bits |= AK5558_DIF_32BIT_MODE; + bits = AK5558_DIF_32BIT_MODE; break; default: return -EINVAL; @@ -168,18 +164,15 @@ static int ak5558_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) } /* set master/slave audio interface */ - format = snd_soc_component_read32(component, AK5558_02_CONTROL1); - format &= ~AK5558_DIF; - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - format |= AK5558_DIF_I2S_MODE; + format = AK5558_DIF_I2S_MODE; break; case SND_SOC_DAIFMT_LEFT_J: - format |= AK5558_DIF_MSB_MODE; + format = AK5558_DIF_MSB_MODE; break; case SND_SOC_DAIFMT_DSP_B: - format |= AK5558_DIF_MSB_MODE; + format = AK5558_DIF_MSB_MODE; break; default: return -EINVAL; @@ -246,7 +239,7 @@ static int ak5558_startup(struct snd_pcm_substream *substream, &ak5558_rate_constraints); } -static struct snd_soc_dai_ops ak5558_dai_ops = { +static const struct snd_soc_dai_ops ak5558_dai_ops = { .startup = ak5558_startup, .hw_params = ak5558_hw_params, diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c index 668cd3754209..e9b7f72d880b 100644 --- a/sound/soc/codecs/cs35l33.c +++ b/sound/soc/codecs/cs35l33.c @@ -857,7 +857,8 @@ static const struct regmap_config cs35l33_regmap = { .readable_reg = cs35l33_readable_register, .writeable_reg = cs35l33_writeable_register, .cache_type = REGCACHE_RBTREE, - .use_single_rw = true, + .use_single_read = true, + .use_single_write = true, }; static int __maybe_unused cs35l33_runtime_resume(struct device *dev) diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index bd6226bde45f..9f4a59871cee 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -1105,7 +1105,8 @@ static struct regmap_config cs35l35_regmap = { .readable_reg = cs35l35_readable_register, .precious_reg = cs35l35_precious_register, .cache_type = REGCACHE_RBTREE, - .use_single_rw = true, + .use_single_read = true, + .use_single_write = true, }; static irqreturn_t cs35l35_irq(int irq, void *data) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 407554175282..ab27d2b94d02 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -154,11 +154,11 @@ static const struct snd_kcontrol_new cs4265_snd_controls[] = { SOC_SINGLE("E to F Buffer Disable Switch", CS4265_SPDIF_CTL1, 6, 1, 0), SOC_ENUM("C Data Access", cam_mode_enum), + SOC_SINGLE("SPDIF Switch", CS4265_SPDIF_CTL2, 5, 1, 1), SOC_SINGLE("Validity Bit Control Switch", CS4265_SPDIF_CTL2, 3, 1, 0), SOC_ENUM("SPDIF Mono/Stereo", spdif_mono_stereo_enum), - SOC_SINGLE("MMTLR Data Switch", CS4265_SPDIF_CTL2, - 0, 1, 0), + SOC_SINGLE("MMTLR Data Switch", CS4265_SPDIF_CTL2, 0, 1, 0), SOC_ENUM("Mono Channel Select", spdif_mono_select_enum), SND_SOC_BYTES("C Data Buffer", CS4265_C_DATA_BUFF, 24), }; @@ -221,10 +221,11 @@ static const struct snd_soc_dapm_route cs4265_audio_map[] = { {"LINEOUTR", NULL, "DAC"}, {"SPDIFOUT", NULL, "SPDIF"}, + {"Pre-amp MIC", NULL, "MICL"}, + {"Pre-amp MIC", NULL, "MICR"}, + {"ADC Mux", "MIC", "Pre-amp MIC"}, {"ADC Mux", "LINEIN", "LINEINL"}, {"ADC Mux", "LINEIN", "LINEINR"}, - {"ADC Mux", "MIC", "MICL"}, - {"ADC Mux", "MIC", "MICR"}, {"ADC", NULL, "ADC Mux"}, {"DOUT", NULL, "ADC"}, {"DAI1 Capture", NULL, "DOUT"}, @@ -496,7 +497,8 @@ static int cs4265_set_bias_level(struct snd_soc_component *component, SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000) #define CS4265_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \ - SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE) + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE) static const struct snd_soc_dai_ops cs4265_ops = { .hw_params = cs4265_pcm_hw_params, diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 3c266eeb89bf..33d74f163bd7 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -29,8 +29,8 @@ #include <linux/i2c.h> #include <linux/delay.h> #include <linux/regulator/consumer.h> +#include <linux/gpio/consumer.h> #include <linux/of_device.h> -#include <linux/of_gpio.h> /* * The codec isn't really big-endian or little-endian, since the I2S @@ -658,8 +658,8 @@ static const struct regmap_config cs4270_regmap = { static int cs4270_i2c_probe(struct i2c_client *i2c_client, const struct i2c_device_id *id) { - struct device_node *np = i2c_client->dev.of_node; struct cs4270_private *cs4270; + struct gpio_desc *reset_gpiod; unsigned int val; int ret, i; @@ -678,20 +678,11 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, if (ret < 0) return ret; - /* See if we have a way to bring the codec out of reset */ - if (np) { - enum of_gpio_flags flags; - int gpio = of_get_named_gpio_flags(np, "reset-gpio", 0, &flags); - - if (gpio_is_valid(gpio)) { - ret = devm_gpio_request_one(&i2c_client->dev, gpio, - flags & OF_GPIO_ACTIVE_LOW ? - GPIOF_OUT_INIT_LOW : GPIOF_OUT_INIT_HIGH, - "cs4270 reset"); - if (ret < 0) - return ret; - } - } + reset_gpiod = devm_gpiod_get_optional(&i2c_client->dev, "reset", + GPIOD_OUT_HIGH); + if (IS_ERR(reset_gpiod) && + PTR_ERR(reset_gpiod) == -EPROBE_DEFER) + return -EPROBE_DEFER; cs4270->regmap = devm_regmap_init_i2c(i2c_client, &cs4270_regmap); if (IS_ERR(cs4270->regmap)) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 5080d7a3c279..fd2bd74024c1 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -21,6 +21,7 @@ * - master mode *NOT* supported */ +#include <linux/clk.h> #include <linux/module.h> #include <linux/slab.h> #include <sound/core.h> @@ -41,6 +42,7 @@ enum master_slave_mode { struct cs42l51_private { unsigned int mclk; + struct clk *mclk_handle; unsigned int audio_mode; /* The mode (I2S or left-justified) */ enum master_slave_mode func; }; @@ -237,6 +239,10 @@ static const struct snd_soc_dapm_widget cs42l51_dapm_widgets[] = { &cs42l51_adcr_mux_controls), }; +static const struct snd_soc_dapm_widget cs42l51_dapm_mclk_widgets[] = { + SND_SOC_DAPM_CLOCK_SUPPLY("MCLK") +}; + static const struct snd_soc_dapm_route cs42l51_routes[] = { {"HPL", NULL, "Left DAC"}, {"HPR", NULL, "Right DAC"}, @@ -487,6 +493,14 @@ static struct snd_soc_dai_driver cs42l51_dai = { static int cs42l51_component_probe(struct snd_soc_component *component) { int ret, reg; + struct snd_soc_dapm_context *dapm; + struct cs42l51_private *cs42l51; + + cs42l51 = snd_soc_component_get_drvdata(component); + dapm = snd_soc_component_get_dapm(component); + + if (cs42l51->mclk_handle) + snd_soc_dapm_new_controls(dapm, cs42l51_dapm_mclk_widgets, 1); /* * DAC configuration @@ -540,6 +554,13 @@ int cs42l51_probe(struct device *dev, struct regmap *regmap) dev_set_drvdata(dev, cs42l51); + cs42l51->mclk_handle = devm_clk_get(dev, "MCLK"); + if (IS_ERR(cs42l51->mclk_handle)) { + if (PTR_ERR(cs42l51->mclk_handle) != -ENOENT) + return PTR_ERR(cs42l51->mclk_handle); + cs42l51->mclk_handle = NULL; + } + /* Verify that we have a CS42L51 */ ret = regmap_read(regmap, CS42L51_CHIP_REV_ID, &val); if (ret < 0) { diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index 80dc42197154..3f7b255587e6 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -2362,7 +2362,9 @@ static const struct regmap_config cs43130_regmap = { .precious_reg = cs43130_precious_register, .volatile_reg = cs43130_volatile_register, .cache_type = REGCACHE_RBTREE, - .use_single_rw = true, /* needed for regcache_sync */ + /* needed for regcache_sync */ + .use_single_read = true, + .use_single_write = true, }; static u16 const cs43130_dc_threshold[CS43130_DC_THRESHOLD] = { diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index 8c4926df9286..da921da50ef0 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -30,9 +30,39 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> +#define MAX_MODESWITCH_DELAY 70 +static int modeswitch_delay; +module_param(modeswitch_delay, uint, 0644); + +static int wakeup_delay; +module_param(wakeup_delay, uint, 0644); + struct dmic { struct gpio_desc *gpio_en; int wakeup_delay; + /* Delay after DMIC mode switch */ + int modeswitch_delay; +}; + +int dmic_daiops_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct dmic *dmic = snd_soc_component_get_drvdata(component); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + if (dmic->modeswitch_delay) + mdelay(dmic->modeswitch_delay); + + break; + } + + return 0; +} + +static const struct snd_soc_dai_ops dmic_dai_ops = { + .trigger = dmic_daiops_trigger, }; static int dmic_aif_event(struct snd_soc_dapm_widget *w, @@ -68,6 +98,7 @@ static struct snd_soc_dai_driver dmic_dai = { | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, + .ops = &dmic_dai_ops, }; static int dmic_component_probe(struct snd_soc_component *component) @@ -85,6 +116,15 @@ static int dmic_component_probe(struct snd_soc_component *component) device_property_read_u32(component->dev, "wakeup-delay-ms", &dmic->wakeup_delay); + device_property_read_u32(component->dev, "modeswitch-delay-ms", + &dmic->modeswitch_delay); + if (wakeup_delay) + dmic->wakeup_delay = wakeup_delay; + if (modeswitch_delay) + dmic->modeswitch_delay = modeswitch_delay; + + if (dmic->modeswitch_delay > MAX_MODESWITCH_DELAY) + dmic->modeswitch_delay = MAX_MODESWITCH_DELAY; snd_soc_component_set_drvdata(component, dmic); @@ -148,6 +188,7 @@ static const struct of_device_id dmic_dev_match[] = { {.compatible = "dmic-codec"}, {} }; +MODULE_DEVICE_TABLE(of, dmic_dev_match); static struct platform_driver dmic_driver = { .driver = { diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index e9fc2fd97d2f..04a3aa770722 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -566,14 +566,14 @@ static int es8328_set_sysclk(struct snd_soc_dai *codec_dai, break; case 22579200: mclkdiv2 = 1; - /* fallthru */ + /* fall through */ case 11289600: es8328->sysclk_constraints = &constraints_11289; es8328->mclk_ratios = ratios_11289; break; case 24576000: mclkdiv2 = 1; - /* fallthru */ + /* fall through */ case 12288000: es8328->sysclk_constraints = &constraints_12288; es8328->mclk_ratios = ratios_12288; @@ -824,7 +824,8 @@ const struct regmap_config es8328_regmap_config = { .val_bits = 8, .max_register = ES8328_REG_MAX, .cache_type = REGCACHE_RBTREE, - .use_single_rw = true, + .use_single_read = true, + .use_single_write = true, }; EXPORT_SYMBOL_GPL(es8328_regmap_config); diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c new file mode 100644 index 000000000000..ffecdaaa8cf2 --- /dev/null +++ b/sound/soc/codecs/hdac_hda.c @@ -0,0 +1,483 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2015-18 Intel Corporation. + +/* + * hdac_hda.c - ASoC extensions to reuse the legacy HDA codec drivers + * with ASoC platform drivers. These APIs are called by the legacy HDA + * codec drivers using hdac_ext_bus_ops ops. + */ + +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/module.h> +#include <linux/pm_runtime.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/hdaudio_ext.h> +#include <sound/hda_codec.h> +#include <sound/hda_register.h> +#include "hdac_hda.h" + +#define HDAC_ANALOG_DAI_ID 0 +#define HDAC_DIGITAL_DAI_ID 1 +#define HDAC_ALT_ANALOG_DAI_ID 2 + +#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_U24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE | \ + SNDRV_PCM_FMTBIT_U32_LE | \ + SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE) + +static int hdac_hda_dai_open(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai); +static void hdac_hda_dai_close(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai); +static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai); +static int hdac_hda_dai_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai); +static int hdac_hda_dai_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width); +static struct hda_pcm *snd_soc_find_pcm_from_dai(struct hdac_hda_priv *hda_pvt, + struct snd_soc_dai *dai); + +static const struct snd_soc_dai_ops hdac_hda_dai_ops = { + .startup = hdac_hda_dai_open, + .shutdown = hdac_hda_dai_close, + .prepare = hdac_hda_dai_prepare, + .hw_free = hdac_hda_dai_hw_free, + .set_tdm_slot = hdac_hda_dai_set_tdm_slot, +}; + +static struct snd_soc_dai_driver hdac_hda_dais[] = { +{ + .id = HDAC_ANALOG_DAI_ID, + .name = "Analog Codec DAI", + .ops = &hdac_hda_dai_ops, + .playback = { + .stream_name = "Analog Codec Playback", + .channels_min = 1, + .channels_max = 16, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = STUB_FORMATS, + .sig_bits = 24, + }, + .capture = { + .stream_name = "Analog Codec Capture", + .channels_min = 1, + .channels_max = 16, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = STUB_FORMATS, + .sig_bits = 24, + }, +}, +{ + .id = HDAC_DIGITAL_DAI_ID, + .name = "Digital Codec DAI", + .ops = &hdac_hda_dai_ops, + .playback = { + .stream_name = "Digital Codec Playback", + .channels_min = 1, + .channels_max = 16, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = STUB_FORMATS, + .sig_bits = 24, + }, + .capture = { + .stream_name = "Digital Codec Capture", + .channels_min = 1, + .channels_max = 16, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = STUB_FORMATS, + .sig_bits = 24, + }, +}, +{ + .id = HDAC_ALT_ANALOG_DAI_ID, + .name = "Alt Analog Codec DAI", + .ops = &hdac_hda_dai_ops, + .playback = { + .stream_name = "Alt Analog Codec Playback", + .channels_min = 1, + .channels_max = 16, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = STUB_FORMATS, + .sig_bits = 24, + }, + .capture = { + .stream_name = "Alt Analog Codec Capture", + .channels_min = 1, + .channels_max = 16, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = STUB_FORMATS, + .sig_bits = 24, + }, +} + +}; + +static int hdac_hda_dai_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + struct hdac_hda_priv *hda_pvt; + struct hdac_hda_pcm *pcm; + + hda_pvt = snd_soc_component_get_drvdata(component); + pcm = &hda_pvt->pcm[dai->id]; + if (tx_mask) + pcm[dai->id].stream_tag[SNDRV_PCM_STREAM_PLAYBACK] = tx_mask; + else + pcm[dai->id].stream_tag[SNDRV_PCM_STREAM_CAPTURE] = rx_mask; + + return 0; +} + +static int hdac_hda_dai_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct hdac_hda_priv *hda_pvt; + struct hda_pcm_stream *hda_stream; + struct hda_pcm *pcm; + + hda_pvt = snd_soc_component_get_drvdata(component); + pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai); + if (!pcm) + return -EINVAL; + + hda_stream = &pcm->stream[substream->stream]; + snd_hda_codec_cleanup(&hda_pvt->codec, hda_stream, substream); + + return 0; +} + +static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct hdac_hda_priv *hda_pvt; + struct snd_pcm_runtime *runtime = substream->runtime; + struct hdac_device *hdev; + struct hda_pcm_stream *hda_stream; + unsigned int format_val; + struct hda_pcm *pcm; + unsigned int stream; + int ret = 0; + + hda_pvt = snd_soc_component_get_drvdata(component); + hdev = &hda_pvt->codec.core; + pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai); + if (!pcm) + return -EINVAL; + + hda_stream = &pcm->stream[substream->stream]; + + format_val = snd_hdac_calc_stream_format(runtime->rate, + runtime->channels, + runtime->format, + hda_stream->maxbps, + 0); + if (!format_val) { + dev_err(&hdev->dev, + "invalid format_val, rate=%d, ch=%d, format=%d\n", + runtime->rate, runtime->channels, runtime->format); + return -EINVAL; + } + + stream = hda_pvt->pcm[dai->id].stream_tag[substream->stream]; + + ret = snd_hda_codec_prepare(&hda_pvt->codec, hda_stream, + stream, format_val, substream); + if (ret < 0) + dev_err(&hdev->dev, "codec prepare failed %d\n", ret); + + return ret; +} + +static int hdac_hda_dai_open(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct hdac_hda_priv *hda_pvt; + struct hda_pcm_stream *hda_stream; + struct hda_pcm *pcm; + int ret; + + hda_pvt = snd_soc_component_get_drvdata(component); + pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai); + if (!pcm) + return -EINVAL; + + snd_hda_codec_pcm_get(pcm); + + hda_stream = &pcm->stream[substream->stream]; + + ret = hda_stream->ops.open(hda_stream, &hda_pvt->codec, substream); + if (ret < 0) + snd_hda_codec_pcm_put(pcm); + + return ret; +} + +static void hdac_hda_dai_close(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct hdac_hda_priv *hda_pvt; + struct hda_pcm_stream *hda_stream; + struct hda_pcm *pcm; + + hda_pvt = snd_soc_component_get_drvdata(component); + pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai); + if (!pcm) + return; + + hda_stream = &pcm->stream[substream->stream]; + + hda_stream->ops.close(hda_stream, &hda_pvt->codec, substream); + + snd_hda_codec_pcm_put(pcm); +} + +static struct hda_pcm *snd_soc_find_pcm_from_dai(struct hdac_hda_priv *hda_pvt, + struct snd_soc_dai *dai) +{ + struct hda_codec *hcodec = &hda_pvt->codec; + struct hda_pcm *cpcm; + const char *pcm_name; + + switch (dai->id) { + case HDAC_ANALOG_DAI_ID: + pcm_name = "Analog"; + break; + case HDAC_DIGITAL_DAI_ID: + pcm_name = "Digital"; + break; + case HDAC_ALT_ANALOG_DAI_ID: + pcm_name = "Alt Analog"; + break; + default: + dev_err(&hcodec->core.dev, "invalid dai id %d\n", dai->id); + return NULL; + } + + list_for_each_entry(cpcm, &hcodec->pcm_list_head, list) { + if (strpbrk(cpcm->name, pcm_name)) + return cpcm; + } + + dev_err(&hcodec->core.dev, "didn't find PCM for DAI %s\n", dai->name); + return NULL; +} + +static int hdac_hda_codec_probe(struct snd_soc_component *component) +{ + struct hdac_hda_priv *hda_pvt = + snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + struct hdac_device *hdev = &hda_pvt->codec.core; + struct hda_codec *hcodec = &hda_pvt->codec; + struct hdac_ext_link *hlink; + hda_codec_patch_t patch; + int ret; + + hlink = snd_hdac_ext_bus_get_link(hdev->bus, dev_name(&hdev->dev)); + if (!hlink) { + dev_err(&hdev->dev, "hdac link not found\n"); + return -EIO; + } + + snd_hdac_ext_bus_link_get(hdev->bus, hlink); + + ret = snd_hda_codec_device_new(hcodec->bus, component->card->snd_card, + hdev->addr, hcodec); + if (ret < 0) { + dev_err(&hdev->dev, "failed to create hda codec %d\n", ret); + goto error_no_pm; + } + + /* + * snd_hda_codec_device_new decrements the usage count so call get pm + * else the device will be powered off + */ + pm_runtime_get_noresume(&hdev->dev); + + hcodec->bus->card = dapm->card->snd_card; + + ret = snd_hda_codec_set_name(hcodec, hcodec->preset->name); + if (ret < 0) { + dev_err(&hdev->dev, "name failed %s\n", hcodec->preset->name); + goto error; + } + + ret = snd_hdac_regmap_init(&hcodec->core); + if (ret < 0) { + dev_err(&hdev->dev, "regmap init failed\n"); + goto error; + } + + patch = (hda_codec_patch_t)hcodec->preset->driver_data; + if (patch) { + ret = patch(hcodec); + if (ret < 0) { + dev_err(&hdev->dev, "patch failed %d\n", ret); + goto error; + } + } else { + dev_dbg(&hdev->dev, "no patch file found\n"); + } + + ret = snd_hda_codec_parse_pcms(hcodec); + if (ret < 0) { + dev_err(&hdev->dev, "unable to map pcms to dai %d\n", ret); + goto error; + } + + ret = snd_hda_codec_build_controls(hcodec); + if (ret < 0) { + dev_err(&hdev->dev, "unable to create controls %d\n", ret); + goto error; + } + + hcodec->core.lazy_cache = true; + + /* + * hdac_device core already sets the state to active and calls + * get_noresume. So enable runtime and set the device to suspend. + * pm_runtime_enable is also called during codec registeration + */ + pm_runtime_put(&hdev->dev); + pm_runtime_suspend(&hdev->dev); + + return 0; + +error: + pm_runtime_put(&hdev->dev); +error_no_pm: + snd_hdac_ext_bus_link_put(hdev->bus, hlink); + return ret; +} + +static void hdac_hda_codec_remove(struct snd_soc_component *component) +{ + struct hdac_hda_priv *hda_pvt = + snd_soc_component_get_drvdata(component); + struct hdac_device *hdev = &hda_pvt->codec.core; + struct hdac_ext_link *hlink = NULL; + + hlink = snd_hdac_ext_bus_get_link(hdev->bus, dev_name(&hdev->dev)); + if (!hlink) { + dev_err(&hdev->dev, "hdac link not found\n"); + return; + } + + snd_hdac_ext_bus_link_put(hdev->bus, hlink); + pm_runtime_disable(&hdev->dev); +} + +static const struct snd_soc_dapm_route hdac_hda_dapm_routes[] = { + {"AIF1TX", NULL, "Codec Input Pin1"}, + {"AIF2TX", NULL, "Codec Input Pin2"}, + {"AIF3TX", NULL, "Codec Input Pin3"}, + + {"Codec Output Pin1", NULL, "AIF1RX"}, + {"Codec Output Pin2", NULL, "AIF2RX"}, + {"Codec Output Pin3", NULL, "AIF3RX"}, +}; + +static const struct snd_soc_dapm_widget hdac_hda_dapm_widgets[] = { + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "Analog Codec Playback", 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "Digital Codec Playback", 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF3RX", "Alt Analog Codec Playback", 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "Analog Codec Capture", 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "Digital Codec Capture", 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF3TX", "Alt Analog Codec Capture", 0, + SND_SOC_NOPM, 0, 0), + + /* Input Pins */ + SND_SOC_DAPM_INPUT("Codec Input Pin1"), + SND_SOC_DAPM_INPUT("Codec Input Pin2"), + SND_SOC_DAPM_INPUT("Codec Input Pin3"), + + /* Output Pins */ + SND_SOC_DAPM_OUTPUT("Codec Output Pin1"), + SND_SOC_DAPM_OUTPUT("Codec Output Pin2"), + SND_SOC_DAPM_OUTPUT("Codec Output Pin3"), +}; + +static const struct snd_soc_component_driver hdac_hda_codec = { + .probe = hdac_hda_codec_probe, + .remove = hdac_hda_codec_remove, + .idle_bias_on = false, + .dapm_widgets = hdac_hda_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(hdac_hda_dapm_widgets), + .dapm_routes = hdac_hda_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(hdac_hda_dapm_routes), +}; + +static int hdac_hda_dev_probe(struct hdac_device *hdev) +{ + struct hdac_ext_link *hlink; + struct hdac_hda_priv *hda_pvt; + int ret; + + /* hold the ref while we probe */ + hlink = snd_hdac_ext_bus_get_link(hdev->bus, dev_name(&hdev->dev)); + if (!hlink) { + dev_err(&hdev->dev, "hdac link not found\n"); + return -EIO; + } + snd_hdac_ext_bus_link_get(hdev->bus, hlink); + + hda_pvt = hdac_to_hda_priv(hdev); + if (!hda_pvt) + return -ENOMEM; + + /* ASoC specific initialization */ + ret = devm_snd_soc_register_component(&hdev->dev, + &hdac_hda_codec, hdac_hda_dais, + ARRAY_SIZE(hdac_hda_dais)); + if (ret < 0) { + dev_err(&hdev->dev, "failed to register HDA codec %d\n", ret); + return ret; + } + + dev_set_drvdata(&hdev->dev, hda_pvt); + snd_hdac_ext_bus_link_put(hdev->bus, hlink); + + return ret; +} + +static int hdac_hda_dev_remove(struct hdac_device *hdev) +{ + return 0; +} + +static struct hdac_ext_bus_ops hdac_ops = { + .hdev_attach = hdac_hda_dev_probe, + .hdev_detach = hdac_hda_dev_remove, +}; + +struct hdac_ext_bus_ops *snd_soc_hdac_hda_get_ops(void) +{ + return &hdac_ops; +} +EXPORT_SYMBOL_GPL(snd_soc_hdac_hda_get_ops); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("ASoC Extensions for legacy HDA Drivers"); +MODULE_AUTHOR("Rakesh Ughreja<rakesh.a.ughreja@intel.com>"); diff --git a/sound/soc/codecs/hdac_hda.h b/sound/soc/codecs/hdac_hda.h new file mode 100644 index 000000000000..e444ef593360 --- /dev/null +++ b/sound/soc/codecs/hdac_hda.h @@ -0,0 +1,24 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright(c) 2015-18 Intel Corporation. + */ + +#ifndef __HDAC_HDA_H__ +#define __HDAC_HDA_H__ + +struct hdac_hda_pcm { + int stream_tag[2]; +}; + +struct hdac_hda_priv { + struct hda_codec codec; + struct hdac_hda_pcm pcm[2]; +}; + +#define hdac_to_hda_priv(_hdac) \ + container_of(_hdac, struct hdac_hda_priv, codec.core) +#define hdac_to_hda_codec(_hdac) container_of(_hdac, struct hda_codec, core) + +struct hdac_ext_bus_ops *snd_soc_hdac_hda_get_ops(void); + +#endif /* __HDAC_HDA_H__ */ diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 7b8533abf637..b19d7a3e7a2c 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -121,8 +121,16 @@ struct hdac_hdmi_dai_port_map { struct hdac_hdmi_cvt *cvt; }; +/* + * pin to port mapping table where the value indicate the pin number and + * the index indicate the port number with 1 base. + */ +static const int icl_pin2port_map[] = {0x4, 0x6, 0x8, 0xa, 0xb}; + struct hdac_hdmi_drv_data { unsigned int vendor_nid; + const int *port_map; /* pin to port mapping table */ + int port_num; }; struct hdac_hdmi_priv { @@ -1329,11 +1337,12 @@ static int hdac_hdmi_add_pin(struct hdac_device *hdev, hda_nid_t nid) return 0; } -#define INTEL_VENDOR_NID 0x08 -#define INTEL_GLK_VENDOR_NID 0x0b +#define INTEL_VENDOR_NID_0x2 0x02 +#define INTEL_VENDOR_NID_0x8 0x08 +#define INTEL_VENDOR_NID_0xb 0x0b #define INTEL_GET_VENDOR_VERB 0xf81 #define INTEL_SET_VENDOR_VERB 0x781 -#define INTEL_EN_DP12 0x02 /* enable DP 1.2 features */ +#define INTEL_EN_DP12 0x02 /* enable DP 1.2 features */ #define INTEL_EN_ALL_PIN_CVTS 0x01 /* enable 2nd & 3rd pins and convertors */ static void hdac_hdmi_skl_enable_all_pins(struct hdac_device *hdev) @@ -1410,6 +1419,12 @@ static int hdac_hdmi_create_dais(struct hdac_device *hdev, if (ret) return ret; + /* Filter out 44.1, 88.2 and 176.4Khz */ + rates &= ~(SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_176400); + if (!rates) + return -EINVAL; + sprintf(dai_name, "intel-hdmi-hifi%d", i+1); hdmi_dais[i].name = devm_kstrdup(&hdev->dev, dai_name, GFP_KERNEL); @@ -1532,7 +1547,26 @@ free_widgets: static int hdac_hdmi_pin2port(void *aptr, int pin) { - return pin - 4; /* map NID 0x05 -> port #1 */ + struct hdac_device *hdev = aptr; + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); + const int *map = hdmi->drv_data->port_map; + int i; + + if (!hdmi->drv_data->port_num) + return pin - 4; /* map NID 0x05 -> port #1 */ + + /* + * looking for the pin number in the mapping table and return + * the index which indicate the port number + */ + for (i = 0; i < hdmi->drv_data->port_num; i++) { + if (pin == map[i]) + return i + 1; + } + + /* return -1 if pin number exceeds our expectation */ + dev_err(&hdev->dev, "Can't find the port for pin %d\n", pin); + return -1; } static void hdac_hdmi_eld_notify_cb(void *aptr, int port, int pipe) @@ -1543,9 +1577,18 @@ static void hdac_hdmi_eld_notify_cb(void *aptr, int port, int pipe) struct hdac_hdmi_port *hport = NULL; struct snd_soc_component *component = hdmi->component; int i; - - /* Don't know how this mapping is derived */ - hda_nid_t pin_nid = port + 0x04; + hda_nid_t pin_nid; + + if (!hdmi->drv_data->port_num) { + /* for legacy platforms */ + pin_nid = port + 0x04; + } else if (port < hdmi->drv_data->port_num) { + /* get pin number from the pin2port mapping table */ + pin_nid = hdmi->drv_data->port_map[port - 1]; + } else { + dev_err(&hdev->dev, "Can't find the pin for port %d\n", port); + return; + } dev_dbg(&hdev->dev, "%s: for pin:%d port=%d\n", __func__, pin_nid, pipe); @@ -1598,7 +1641,7 @@ static struct snd_pcm *hdac_hdmi_get_pcm_from_id(struct snd_soc_card *card, { struct snd_soc_pcm_runtime *rtd; - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { if (rtd->pcm && (rtd->pcm->device == device)) return rtd->pcm; } @@ -1847,51 +1890,31 @@ static void hdmi_codec_remove(struct snd_soc_component *component) pm_runtime_disable(&hdev->dev); } -#ifdef CONFIG_PM -static int hdmi_codec_prepare(struct device *dev) -{ - struct hdac_device *hdev = dev_to_hdac_dev(dev); - - pm_runtime_get_sync(&hdev->dev); - - /* - * Power down afg. - * codec_read is preferred over codec_write to set the power state. - * This way verb is send to set the power state and response - * is received. So setting power state is ensured without using loop - * to read the state. - */ - snd_hdac_codec_read(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - - return 0; -} - -static void hdmi_codec_complete(struct device *dev) +#ifdef CONFIG_PM_SLEEP +static int hdmi_codec_resume(struct device *dev) { struct hdac_device *hdev = dev_to_hdac_dev(dev); struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); + int ret; - /* Power up afg */ - snd_hdac_codec_read(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - - hdac_hdmi_skl_enable_all_pins(hdev); - hdac_hdmi_skl_enable_dp12(hdev); - + ret = pm_runtime_force_resume(dev); + if (ret < 0) + return ret; /* * As the ELD notify callback request is not entertained while the * device is in suspend state. Need to manually check detection of * all pins here. pin capablity change is not support, so use the * already set pin caps. + * + * NOTE: this is safe to call even if the codec doesn't actually resume. + * The pin check involves only with DRM audio component hooks, so it + * works even if the HD-audio side is still dreaming peacefully. */ hdac_hdmi_present_sense_all_pins(hdev, hdmi, false); - - pm_runtime_put_sync(&hdev->dev); + return 0; } #else -#define hdmi_codec_prepare NULL -#define hdmi_codec_complete NULL +#define hdmi_codec_resume NULL #endif static const struct snd_soc_component_driver hdmi_hda_codec = { @@ -1961,21 +1984,24 @@ static int hdac_hdmi_get_spk_alloc(struct hdac_device *hdev, int pcm_idx) port = list_first_entry(&pcm->port_list, struct hdac_hdmi_port, head); - if (!port) - return 0; - if (!port || !port->eld.eld_valid) return 0; return port->eld.info.spk_alloc; } +static struct hdac_hdmi_drv_data intel_icl_drv_data = { + .vendor_nid = INTEL_VENDOR_NID_0x2, + .port_map = icl_pin2port_map, + .port_num = ARRAY_SIZE(icl_pin2port_map), +}; + static struct hdac_hdmi_drv_data intel_glk_drv_data = { - .vendor_nid = INTEL_GLK_VENDOR_NID, + .vendor_nid = INTEL_VENDOR_NID_0xb, }; static struct hdac_hdmi_drv_data intel_drv_data = { - .vendor_nid = INTEL_VENDOR_NID, + .vendor_nid = INTEL_VENDOR_NID_0x8, }; static int hdac_hdmi_dev_probe(struct hdac_device *hdev) @@ -2028,13 +2054,7 @@ static int hdac_hdmi_dev_probe(struct hdac_device *hdev) * Turned off in the runtime_suspend during the first explicit * pm_runtime_suspend call. */ - ret = snd_hdac_display_power(hdev->bus, true); - if (ret < 0) { - dev_err(&hdev->dev, - "Cannot turn on display power on i915 err: %d\n", - ret); - return ret; - } + snd_hdac_display_power(hdev->bus, hdev->addr, true); ret = hdac_hdmi_parse_and_map_nid(hdev, &hdmi_dais, &num_dais); if (ret < 0) { @@ -2062,6 +2082,8 @@ static int hdac_hdmi_dev_remove(struct hdac_device *hdev) struct hdac_hdmi_port *port, *port_next; int i; + snd_hdac_display_power(hdev->bus, hdev->addr, false); + list_for_each_entry_safe(pcm, pcm_next, &hdmi->pcm_list, head) { pcm->cvt = NULL; if (list_empty(&pcm->port_list)) @@ -2093,81 +2115,11 @@ static int hdac_hdmi_dev_remove(struct hdac_device *hdev) } #ifdef CONFIG_PM -/* - * Power management sequences - * ========================== - * - * The following explains the PM handling of HDAC HDMI with its parent - * device SKL and display power usage - * - * Probe - * ----- - * In SKL probe, - * 1. skl_probe_work() powers up the display (refcount++ -> 1) - * 2. enumerates the codecs on the link - * 3. powers down the display (refcount-- -> 0) - * - * In HDAC HDMI probe, - * 1. hdac_hdmi_dev_probe() powers up the display (refcount++ -> 1) - * 2. probe the codec - * 3. put the HDAC HDMI device to runtime suspend - * 4. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0) - * - * Once children are runtime suspended, SKL device also goes to runtime - * suspend - * - * HDMI Playback - * ------------- - * Open HDMI device, - * 1. skl_runtime_resume() invoked - * 2. hdac_hdmi_runtime_resume() powers up the display (refcount++ -> 1) - * - * Close HDMI device, - * 1. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0) - * 2. skl_runtime_suspend() invoked - * - * S0/S3 Cycle with playback in progress - * ------------------------------------- - * When the device is opened for playback, the device is runtime active - * already and the display refcount is 1 as explained above. - * - * Entering to S3, - * 1. hdmi_codec_prepare() invoke the runtime resume of codec which just - * increments the PM runtime usage count of the codec since the device - * is in use already - * 2. skl_suspend() powers down the display (refcount-- -> 0) - * - * Wakeup from S3, - * 1. skl_resume() powers up the display (refcount++ -> 1) - * 2. hdmi_codec_complete() invokes the runtime suspend of codec which just - * decrements the PM runtime usage count of the codec since the device - * is in use already - * - * Once playback is stopped, the display refcount is set to 0 as explained - * above in the HDMI playback sequence. The PM handlings are designed in - * such way that to balance the refcount of display power when the codec - * device put to S3 while playback is going on. - * - * S0/S3 Cycle without playback in progress - * ---------------------------------------- - * Entering to S3, - * 1. hdmi_codec_prepare() invoke the runtime resume of codec - * 2. skl_runtime_resume() invoked - * 3. hdac_hdmi_runtime_resume() powers up the display (refcount++ -> 1) - * 4. skl_suspend() powers down the display (refcount-- -> 0) - * - * Wakeup from S3, - * 1. skl_resume() powers up the display (refcount++ -> 1) - * 2. hdmi_codec_complete() invokes the runtime suspend of codec - * 3. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0) - * 4. skl_runtime_suspend() invoked - */ static int hdac_hdmi_runtime_suspend(struct device *dev) { struct hdac_device *hdev = dev_to_hdac_dev(dev); struct hdac_bus *bus = hdev->bus; struct hdac_ext_link *hlink = NULL; - int err; dev_dbg(dev, "Enter: %s\n", __func__); @@ -2184,11 +2136,6 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) */ snd_hdac_codec_read(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - err = snd_hdac_display_power(bus, false); - if (err < 0) { - dev_err(dev, "Cannot turn on display power on i915\n"); - return err; - } hlink = snd_hdac_ext_bus_get_link(bus, dev_name(dev)); if (!hlink) { @@ -2198,6 +2145,8 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) snd_hdac_ext_bus_link_put(bus, hlink); + snd_hdac_display_power(bus, hdev->addr, false); + return 0; } @@ -2206,7 +2155,6 @@ static int hdac_hdmi_runtime_resume(struct device *dev) struct hdac_device *hdev = dev_to_hdac_dev(dev); struct hdac_bus *bus = hdev->bus; struct hdac_ext_link *hlink = NULL; - int err; dev_dbg(dev, "Enter: %s\n", __func__); @@ -2222,11 +2170,7 @@ static int hdac_hdmi_runtime_resume(struct device *dev) snd_hdac_ext_bus_link_get(bus, hlink); - err = snd_hdac_display_power(bus, true); - if (err < 0) { - dev_err(dev, "Cannot turn on display power on i915\n"); - return err; - } + snd_hdac_display_power(bus, hdev->addr, true); hdac_hdmi_skl_enable_all_pins(hdev); hdac_hdmi_skl_enable_dp12(hdev); @@ -2244,8 +2188,7 @@ static int hdac_hdmi_runtime_resume(struct device *dev) static const struct dev_pm_ops hdac_hdmi_pm = { SET_RUNTIME_PM_OPS(hdac_hdmi_runtime_suspend, hdac_hdmi_runtime_resume, NULL) - .prepare = hdmi_codec_prepare, - .complete = hdmi_codec_complete, + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, hdmi_codec_resume) }; static const struct hda_device_id hdmi_list[] = { @@ -2256,6 +2199,8 @@ static const struct hda_device_id hdmi_list[] = { &intel_glk_drv_data), HDA_CODEC_EXT_ENTRY(0x8086280d, 0x100000, "Geminilake HDMI", &intel_glk_drv_data), + HDA_CODEC_EXT_ENTRY(0x8086280f, 0x100000, "Icelake HDMI", + &intel_icl_drv_data), {} }; diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index d00734d31e04..e5b6769b9797 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -795,6 +795,8 @@ static int hdmi_codec_probe(struct platform_device *pdev) if (hcd->spdif) hcp->daidrv[i] = hdmi_spdif_dai; + dev_set_drvdata(dev, hcp); + ret = devm_snd_soc_register_component(dev, &hdmi_driver, hcp->daidrv, dai_count); if (ret) { @@ -802,8 +804,6 @@ static int hdmi_codec_probe(struct platform_device *pdev) __func__, ret); return ret; } - - dev_set_drvdata(dev, hcp); return 0; } diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index fb515aaa54fc..ca172a4b6849 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -16,6 +16,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/regmap.h> +#include <linux/clk.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -42,6 +43,7 @@ struct max98088_priv { struct regmap *regmap; enum max98088_type devtype; struct max98088_pdata *pdata; + struct clk *mclk; unsigned int sysclk; struct max98088_cdata dai[2]; int eq_textcnt; @@ -1103,6 +1105,11 @@ static int max98088_dai_set_sysclk(struct snd_soc_dai *dai, if (freq == max98088->sysclk) return 0; + if (!IS_ERR(max98088->mclk)) { + freq = clk_round_rate(max98088->mclk, freq); + clk_set_rate(max98088->mclk, freq); + } + /* Setup clocks for slave mode, and using the PLL * PSCLK = 0x01 (when master clk is 10MHz to 20MHz) * 0x02 (when master clk is 20MHz to 30MHz).. @@ -1310,6 +1317,20 @@ static int max98088_set_bias_level(struct snd_soc_component *component, break; case SND_SOC_BIAS_PREPARE: + /* + * SND_SOC_BIAS_PREPARE is called while preparing for a + * transition to ON or away from ON. If current bias_level + * is SND_SOC_BIAS_ON, then it is preparing for a transition + * away from ON. Disable the clock in that case, otherwise + * enable it. + */ + if (!IS_ERR(max98088->mclk)) { + if (snd_soc_component_get_bias_level(component) == + SND_SOC_BIAS_ON) + clk_disable_unprepare(max98088->mclk); + else + clk_prepare_enable(max98088->mclk); + } break; case SND_SOC_BIAS_STANDBY: @@ -1725,6 +1746,11 @@ static int max98088_i2c_probe(struct i2c_client *i2c, if (IS_ERR(max98088->regmap)) return PTR_ERR(max98088->regmap); + max98088->mclk = devm_clk_get(&i2c->dev, "mclk"); + if (IS_ERR(max98088->mclk)) + if (PTR_ERR(max98088->mclk) == -EPROBE_DEFER) + return PTR_ERR(max98088->mclk); + max98088->devtype = id->driver_data; i2c_set_clientdata(i2c, max98088); @@ -1742,9 +1768,19 @@ static const struct i2c_device_id max98088_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, max98088_i2c_id); +#if defined(CONFIG_OF) +static const struct of_device_id max98088_of_match[] = { + { .compatible = "maxim,max98088" }, + { .compatible = "maxim,max98089" }, + { } +}; +MODULE_DEVICE_TABLE(of, max98088_of_match); +#endif + static struct i2c_driver max98088_i2c_driver = { .driver = { .name = "max98088", + .of_match_table = of_match_ptr(max98088_of_match), }, .probe = max98088_i2c_probe, .id_table = max98088_i2c_id, diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 1093f766d0d2..9c8616a7b61c 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -2,6 +2,7 @@ // Copyright (c) 2017, Maxim Integrated #include <linux/acpi.h> +#include <linux/delay.h> #include <linux/i2c.h> #include <linux/module.h> #include <linux/regmap.h> @@ -454,7 +455,7 @@ SND_SOC_DAPM_SIGGEN("IMON"), SND_SOC_DAPM_SIGGEN("FBMON"), }; -static DECLARE_TLV_DB_SCALE(max98373_digital_tlv, 0, -50, 0); +static DECLARE_TLV_DB_SCALE(max98373_digital_tlv, -6350, 50, 1); static const DECLARE_TLV_DB_RANGE(max98373_spk_tlv, 0, 8, TLV_DB_SCALE_ITEM(0, 50, 0), 9, 10, TLV_DB_SCALE_ITEM(500, 100, 0), @@ -470,19 +471,19 @@ static const DECLARE_TLV_DB_RANGE(max98373_dht_spkgain_min_tlv, 0, 9, TLV_DB_SCALE_ITEM(800, 100, 0), ); static const DECLARE_TLV_DB_RANGE(max98373_dht_rotation_point_tlv, - 0, 1, TLV_DB_SCALE_ITEM(-50, -50, 0), - 2, 7, TLV_DB_SCALE_ITEM(-200, -100, 0), - 8, 9, TLV_DB_SCALE_ITEM(-1000, -200, 0), - 10, 11, TLV_DB_SCALE_ITEM(-1500, -300, 0), - 12, 13, TLV_DB_SCALE_ITEM(-2000, -200, 0), - 14, 15, TLV_DB_SCALE_ITEM(-2500, -500, 0), + 0, 1, TLV_DB_SCALE_ITEM(-3000, 500, 0), + 2, 4, TLV_DB_SCALE_ITEM(-2200, 200, 0), + 5, 6, TLV_DB_SCALE_ITEM(-1500, 300, 0), + 7, 9, TLV_DB_SCALE_ITEM(-1000, 200, 0), + 10, 13, TLV_DB_SCALE_ITEM(-500, 100, 0), + 14, 15, TLV_DB_SCALE_ITEM(-100, 50, 0), ); static const DECLARE_TLV_DB_RANGE(max98373_limiter_thresh_tlv, - 0, 15, TLV_DB_SCALE_ITEM(0, -100, 0), + 0, 15, TLV_DB_SCALE_ITEM(-1500, 100, 0), ); static const DECLARE_TLV_DB_RANGE(max98373_bde_gain_tlv, - 0, 60, TLV_DB_SCALE_ITEM(0, -25, 0), + 0, 60, TLV_DB_SCALE_ITEM(-1500, 25, 0), ); static bool max98373_readable_register(struct device *dev, unsigned int reg) @@ -604,7 +605,7 @@ SOC_SINGLE("Dither Switch", MAX98373_R203F_AMP_DSP_CFG, SOC_SINGLE("DC Blocker Switch", MAX98373_R203F_AMP_DSP_CFG, MAX98373_AMP_DSP_CFG_DCBLK_SHIFT, 1, 0), SOC_SINGLE_TLV("Digital Volume", MAX98373_R203D_AMP_DIG_VOL_CTRL, - 0, 0x7F, 0, max98373_digital_tlv), + 0, 0x7F, 1, max98373_digital_tlv), SOC_SINGLE_TLV("Speaker Volume", MAX98373_R203E_AMP_PATH_GAIN, MAX98373_SPK_DIGI_GAIN_SHIFT, 10, 0, max98373_spk_tlv), SOC_SINGLE_TLV("FS Max Volume", MAX98373_R203E_AMP_PATH_GAIN, @@ -616,7 +617,7 @@ SOC_SINGLE("DHT Switch", MAX98373_R20D4_DHT_EN, SOC_SINGLE_TLV("DHT Min Volume", MAX98373_R20D1_DHT_CFG, MAX98373_DHT_SPK_GAIN_MIN_SHIFT, 9, 0, max98373_dht_spkgain_min_tlv), SOC_SINGLE_TLV("DHT Rot Pnt Volume", MAX98373_R20D1_DHT_CFG, - MAX98373_DHT_ROT_PNT_SHIFT, 15, 0, max98373_dht_rotation_point_tlv), + MAX98373_DHT_ROT_PNT_SHIFT, 15, 1, max98373_dht_rotation_point_tlv), SOC_SINGLE_TLV("DHT Attack Step Volume", MAX98373_R20D2_DHT_ATTACK_CFG, MAX98373_DHT_ATTACK_STEP_SHIFT, 4, 0, max98373_dht_step_size_tlv), SOC_SINGLE_TLV("DHT Release Step Volume", MAX98373_R20D3_DHT_RELEASE_CFG, @@ -653,29 +654,29 @@ SOC_SINGLE("BDE Hold Time", MAX98373_R2090_BDE_LVL_HOLD, 0, 0xFF, 0), SOC_SINGLE("BDE Attack Rate", MAX98373_R2091_BDE_GAIN_ATK_REL_RATE, 4, 0xF, 0), SOC_SINGLE("BDE Release Rate", MAX98373_R2091_BDE_GAIN_ATK_REL_RATE, 0, 0xF, 0), SOC_SINGLE_TLV("BDE LVL1 Clip Thresh Volume", MAX98373_R20A9_BDE_L1_CFG_2, - 0, 0x3C, 0, max98373_bde_gain_tlv), + 0, 0x3C, 1, max98373_bde_gain_tlv), SOC_SINGLE_TLV("BDE LVL2 Clip Thresh Volume", MAX98373_R20AC_BDE_L2_CFG_2, - 0, 0x3C, 0, max98373_bde_gain_tlv), + 0, 0x3C, 1, max98373_bde_gain_tlv), SOC_SINGLE_TLV("BDE LVL3 Clip Thresh Volume", MAX98373_R20AF_BDE_L3_CFG_2, - 0, 0x3C, 0, max98373_bde_gain_tlv), + 0, 0x3C, 1, max98373_bde_gain_tlv), SOC_SINGLE_TLV("BDE LVL4 Clip Thresh Volume", MAX98373_R20B2_BDE_L4_CFG_2, - 0, 0x3C, 0, max98373_bde_gain_tlv), + 0, 0x3C, 1, max98373_bde_gain_tlv), SOC_SINGLE_TLV("BDE LVL1 Clip Reduction Volume", MAX98373_R20AA_BDE_L1_CFG_3, - 0, 0x3C, 0, max98373_bde_gain_tlv), + 0, 0x3C, 1, max98373_bde_gain_tlv), SOC_SINGLE_TLV("BDE LVL2 Clip Reduction Volume", MAX98373_R20AD_BDE_L2_CFG_3, - 0, 0x3C, 0, max98373_bde_gain_tlv), + 0, 0x3C, 1, max98373_bde_gain_tlv), SOC_SINGLE_TLV("BDE LVL3 Clip Reduction Volume", MAX98373_R20B0_BDE_L3_CFG_3, - 0, 0x3C, 0, max98373_bde_gain_tlv), + 0, 0x3C, 1, max98373_bde_gain_tlv), SOC_SINGLE_TLV("BDE LVL4 Clip Reduction Volume", MAX98373_R20B3_BDE_L4_CFG_3, - 0, 0x3C, 0, max98373_bde_gain_tlv), + 0, 0x3C, 1, max98373_bde_gain_tlv), SOC_SINGLE_TLV("BDE LVL1 Limiter Thresh Volume", MAX98373_R20A8_BDE_L1_CFG_1, - 0, 0xF, 0, max98373_limiter_thresh_tlv), + 0, 0xF, 1, max98373_limiter_thresh_tlv), SOC_SINGLE_TLV("BDE LVL2 Limiter Thresh Volume", MAX98373_R20AB_BDE_L2_CFG_1, - 0, 0xF, 0, max98373_limiter_thresh_tlv), + 0, 0xF, 1, max98373_limiter_thresh_tlv), SOC_SINGLE_TLV("BDE LVL3 Limiter Thresh Volume", MAX98373_R20AE_BDE_L3_CFG_1, - 0, 0xF, 0, max98373_limiter_thresh_tlv), + 0, 0xF, 1, max98373_limiter_thresh_tlv), SOC_SINGLE_TLV("BDE LVL4 Limiter Thresh Volume", MAX98373_R20B1_BDE_L4_CFG_1, - 0, 0xF, 0, max98373_limiter_thresh_tlv), + 0, 0xF, 1, max98373_limiter_thresh_tlv), /* Limiter */ SOC_SINGLE("Limiter Switch", MAX98373_R20E2_LIMITER_EN, MAX98373_LIMITER_EN_SHIFT, 1, 0), @@ -723,14 +724,39 @@ static struct snd_soc_dai_driver max98373_dai[] = { } }; +static void max98373_reset(struct max98373_priv *max98373, struct device *dev) +{ + int ret, reg, count; + + /* Software Reset */ + ret = regmap_update_bits(max98373->regmap, + MAX98373_R2000_SW_RESET, + MAX98373_SOFT_RESET, + MAX98373_SOFT_RESET); + if (ret) + dev_err(dev, "Reset command failed. (ret:%d)\n", ret); + + count = 0; + while (count < 3) { + usleep_range(10000, 11000); + /* Software Reset Verification */ + ret = regmap_read(max98373->regmap, + MAX98373_R21FF_REV_ID, ®); + if (!ret) { + dev_info(dev, "Reset completed (retry:%d)\n", count); + return; + } + count++; + } + dev_err(dev, "Reset failed. (ret:%d)\n", ret); +} + static int max98373_probe(struct snd_soc_component *component) { struct max98373_priv *max98373 = snd_soc_component_get_drvdata(component); /* Software Reset */ - regmap_write(max98373->regmap, - MAX98373_R2000_SW_RESET, MAX98373_SOFT_RESET); - usleep_range(10000, 11000); + max98373_reset(max98373, component->dev); /* IV default slot configuration */ regmap_write(max98373->regmap, @@ -817,9 +843,7 @@ static int max98373_resume(struct device *dev) { struct max98373_priv *max98373 = dev_get_drvdata(dev); - regmap_write(max98373->regmap, - MAX98373_R2000_SW_RESET, MAX98373_SOFT_RESET); - usleep_range(10000, 11000); + max98373_reset(max98373, dev); regcache_cache_only(max98373->regmap, false); regcache_sync(max98373->regmap); return 0; diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 4ea3287162ad..8600c5439e1e 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -1,12 +1,10 @@ -/* - * max9867.c -- max9867 ALSA SoC Audio driver - * - * Copyright 2013-15 Maxim Integrated Products - * - * 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. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// MAX9867 ALSA SoC codec driver +// +// Copyright 2013-2015 Maxim Integrated Products +// Copyright 2018 Ladislav Michl <ladis@linux-mips.org> +// #include <linux/delay.h> #include <linux/i2c.h> @@ -23,254 +21,237 @@ static const char *const max9867_spmode[] = { "Stereo Single", "Mono Single", "Stereo Single Fast", "Mono Single Fast" }; -static const char *const max9867_sidetone_text[] = { - "None", "Left", "Right", "LeftRight", "LeftRightDiv2", -}; static const char *const max9867_filter_text[] = {"IIR", "FIR"}; static SOC_ENUM_SINGLE_DECL(max9867_filter, MAX9867_CODECFLTR, 7, max9867_filter_text); static SOC_ENUM_SINGLE_DECL(max9867_spkmode, MAX9867_MODECONFIG, 0, max9867_spmode); -static SOC_ENUM_SINGLE_DECL(max9867_sidetone, MAX9867_DACGAIN, 6, - max9867_sidetone_text); -static DECLARE_TLV_DB_SCALE(max9860_capture_tlv, -600, 200, 0); -static DECLARE_TLV_DB_SCALE(max9860_mic_tlv, 2000, 100, 1); -static DECLARE_TLV_DB_SCALE(max9860_adc_left_tlv, -1200, 100, 1); -static DECLARE_TLV_DB_SCALE(max9860_adc_right_tlv, -1200, 100, 1); -static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(max98088_micboost_tlv, - 0, 1, TLV_DB_SCALE_ITEM(0, 2000, 0), - 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(max9867_master_tlv, + 0, 2, TLV_DB_SCALE_ITEM(-8600, 200, 1), + 3, 17, TLV_DB_SCALE_ITEM(-7800, 400, 0), + 18, 25, TLV_DB_SCALE_ITEM(-2000, 200, 0), + 26, 34, TLV_DB_SCALE_ITEM( -500, 100, 0), + 35, 40, TLV_DB_SCALE_ITEM( 350, 50, 0), +); +static DECLARE_TLV_DB_SCALE(max9867_mic_tlv, 0, 100, 0); +static DECLARE_TLV_DB_SCALE(max9867_line_tlv, -600, 200, 0); +static DECLARE_TLV_DB_SCALE(max9867_adc_tlv, -1200, 100, 0); +static DECLARE_TLV_DB_SCALE(max9867_dac_tlv, -1500, 100, 0); +static DECLARE_TLV_DB_SCALE(max9867_dacboost_tlv, 0, 600, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(max9867_micboost_tlv, + 0, 2, TLV_DB_SCALE_ITEM(-2000, 2000, 1), + 3, 3, TLV_DB_SCALE_ITEM(3000, 0, 0), ); static const struct snd_kcontrol_new max9867_snd_controls[] = { - SOC_DOUBLE_R("Master Playback Volume", MAX9867_LEFTVOL, - MAX9867_RIGHTVOL, 0, 63, 1), - SOC_DOUBLE_R_TLV("Capture Volume", MAX9867_LEFTMICGAIN, - MAX9867_RIGHTMICGAIN, - 0, 15, 1, max9860_capture_tlv), - SOC_DOUBLE_R_TLV("Mic Volume", MAX9867_LEFTMICGAIN, - MAX9867_RIGHTMICGAIN, 0, 31, 1, max9860_mic_tlv), - SOC_DOUBLE_R_TLV("Mic Boost Volume", MAX9867_LEFTMICGAIN, - MAX9867_RIGHTMICGAIN, 5, 3, 0, max98088_micboost_tlv), - SOC_ENUM("Digital Sidetone Src", max9867_sidetone), - SOC_SINGLE("Sidetone Volume", MAX9867_DACGAIN, 0, 31, 1), - SOC_SINGLE("DAC Volume", MAX9867_DACLEVEL, 4, 3, 0), - SOC_SINGLE("DAC Attenuation", MAX9867_DACLEVEL, 0, 15, 1), - SOC_SINGLE_TLV("ADC Left Volume", MAX9867_ADCLEVEL, - 4, 15, 1, max9860_adc_left_tlv), - SOC_SINGLE_TLV("ADC Right Volume", MAX9867_ADCLEVEL, - 0, 15, 1, max9860_adc_right_tlv), + SOC_DOUBLE_R_TLV("Master Playback Volume", MAX9867_LEFTVOL, + MAX9867_RIGHTVOL, 0, 41, 1, max9867_master_tlv), + SOC_DOUBLE_R_TLV("Line Capture Volume", MAX9867_LEFTLINELVL, + MAX9867_RIGHTLINELVL, 0, 15, 1, max9867_line_tlv), + SOC_DOUBLE_R_TLV("Mic Capture Volume", MAX9867_LEFTMICGAIN, + MAX9867_RIGHTMICGAIN, 0, 20, 1, max9867_mic_tlv), + SOC_DOUBLE_R_TLV("Mic Boost Capture Volume", MAX9867_LEFTMICGAIN, + MAX9867_RIGHTMICGAIN, 5, 4, 0, max9867_micboost_tlv), + SOC_SINGLE("Digital Sidetone Volume", MAX9867_SIDETONE, 0, 31, 1), + SOC_SINGLE_TLV("Digital Playback Volume", MAX9867_DACLEVEL, 0, 15, 1, + max9867_dac_tlv), + SOC_SINGLE_TLV("Digital Boost Playback Volume", MAX9867_DACLEVEL, 4, 3, 0, + max9867_dacboost_tlv), + SOC_DOUBLE_TLV("Digital Capture Volume", MAX9867_ADCLEVEL, 0, 4, 15, 1, + max9867_adc_tlv), SOC_ENUM("Speaker Mode", max9867_spkmode), SOC_SINGLE("Volume Smoothing Switch", MAX9867_MODECONFIG, 6, 1, 0), - SOC_SINGLE("ZCD Switch", MAX9867_MODECONFIG, 5, 1, 0), + SOC_SINGLE("Line ZC Switch", MAX9867_MODECONFIG, 5, 1, 0), SOC_ENUM("DSP Filter", max9867_filter), }; -static const char *const max9867_mux[] = {"None", "Mic", "Line", "Mic_Line"}; +/* Input mixer */ +static const struct snd_kcontrol_new max9867_input_mixer_controls[] = { + SOC_DAPM_DOUBLE("Line Capture Switch", MAX9867_INPUTCONFIG, 7, 5, 1, 0), + SOC_DAPM_DOUBLE("Mic Capture Switch", MAX9867_INPUTCONFIG, 6, 4, 1, 0), +}; + +/* Output mixer */ +static const struct snd_kcontrol_new max9867_output_mixer_controls[] = { + SOC_DAPM_DOUBLE_R("Line Bypass Switch", + MAX9867_LEFTLINELVL, MAX9867_RIGHTLINELVL, 6, 1, 1), +}; -static SOC_ENUM_SINGLE_DECL(max9867_mux_enum, - MAX9867_INPUTCONFIG, MAX9867_INPUT_SHIFT, - max9867_mux); +/* Sidetone mixer */ +static const struct snd_kcontrol_new max9867_sidetone_mixer_controls[] = { + SOC_DAPM_DOUBLE("Sidetone Switch", MAX9867_SIDETONE, 6, 7, 1, 0), +}; -static const struct snd_kcontrol_new max9867_dapm_mux_controls = - SOC_DAPM_ENUM("Route", max9867_mux_enum); +/* Line out switch */ +static const struct snd_kcontrol_new max9867_line_out_control = + SOC_DAPM_DOUBLE_R("Switch", + MAX9867_LEFTVOL, MAX9867_RIGHTVOL, 6, 1, 1); -static const struct snd_kcontrol_new max9867_left_dapm_control = - SOC_DAPM_SINGLE("Switch", MAX9867_PWRMAN, 6, 1, 0); -static const struct snd_kcontrol_new max9867_right_dapm_control = - SOC_DAPM_SINGLE("Switch", MAX9867_PWRMAN, 5, 1, 0); -static const struct snd_kcontrol_new max9867_line_dapm_control = - SOC_DAPM_SINGLE("Switch", MAX9867_LEFTLINELVL, 6, 1, 1); static const struct snd_soc_dapm_widget max9867_dapm_widgets[] = { - SND_SOC_DAPM_AIF_IN("DAI_OUT", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("Left DAC", NULL, MAX9867_PWRMAN, 3, 0), - SND_SOC_DAPM_DAC("Right DAC", NULL, MAX9867_PWRMAN, 2, 0), - SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_OUTPUT("HPOUT"), - - SND_SOC_DAPM_AIF_IN("DAI_IN", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_ADC("Left ADC", "HiFi Capture", MAX9867_PWRMAN, 1, 0), - SND_SOC_DAPM_ADC("Right ADC", "HiFi Capture", MAX9867_PWRMAN, 0, 0), - SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, - &max9867_dapm_mux_controls), - - SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_SWITCH("Left Line", MAX9867_LEFTLINELVL, 6, 1, - &max9867_left_dapm_control), - SND_SOC_DAPM_SWITCH("Right Line", MAX9867_RIGTHLINELVL, 6, 1, - &max9867_right_dapm_control), - SND_SOC_DAPM_SWITCH("Line Mixer", SND_SOC_NOPM, 0, 0, - &max9867_line_dapm_control), - SND_SOC_DAPM_INPUT("LINE_IN"), + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + SND_SOC_DAPM_INPUT("LINL"), + SND_SOC_DAPM_INPUT("LINR"), + + SND_SOC_DAPM_PGA("Left Line Input", MAX9867_PWRMAN, 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Line Input", MAX9867_PWRMAN, 5, 0, NULL, 0), + SND_SOC_DAPM_MIXER_NAMED_CTL("Input Mixer", SND_SOC_NOPM, 0, 0, + max9867_input_mixer_controls, + ARRAY_SIZE(max9867_input_mixer_controls)), + SND_SOC_DAPM_ADC("ADCL", "HiFi Capture", MAX9867_PWRMAN, 1, 0), + SND_SOC_DAPM_ADC("ADCR", "HiFi Capture", MAX9867_PWRMAN, 0, 0), + + SND_SOC_DAPM_MIXER("Digital", SND_SOC_NOPM, 0, 0, + max9867_sidetone_mixer_controls, + ARRAY_SIZE(max9867_sidetone_mixer_controls)), + SND_SOC_DAPM_MIXER_NAMED_CTL("Output Mixer", SND_SOC_NOPM, 0, 0, + max9867_output_mixer_controls, + ARRAY_SIZE(max9867_output_mixer_controls)), + SND_SOC_DAPM_DAC("DACL", "HiFi Playback", MAX9867_PWRMAN, 3, 0), + SND_SOC_DAPM_DAC("DACR", "HiFi Playback", MAX9867_PWRMAN, 2, 0), + SND_SOC_DAPM_SWITCH("Master Playback", SND_SOC_NOPM, 0, 0, + &max9867_line_out_control), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("ROUT"), }; static const struct snd_soc_dapm_route max9867_audio_map[] = { - {"Left DAC", NULL, "DAI_OUT"}, - {"Right DAC", NULL, "DAI_OUT"}, - {"Output Mixer", NULL, "Left DAC"}, - {"Output Mixer", NULL, "Right DAC"}, - {"HPOUT", NULL, "Output Mixer"}, - - {"Left ADC", NULL, "DAI_IN"}, - {"Right ADC", NULL, "DAI_IN"}, - {"Input Mixer", NULL, "Left ADC"}, - {"Input Mixer", NULL, "Right ADC"}, - {"Input Mux", "Line", "Input Mixer"}, - {"Input Mux", "Mic", "Input Mixer"}, - {"Input Mux", "Mic_Line", "Input Mixer"}, - {"Right Line", "Switch", "Input Mux"}, - {"Left Line", "Switch", "Input Mux"}, - {"LINE_IN", NULL, "Left Line"}, - {"LINE_IN", NULL, "Right Line"}, + {"Left Line Input", NULL, "LINL"}, + {"Right Line Input", NULL, "LINR"}, + {"Input Mixer", "Mic Capture Switch", "MICL"}, + {"Input Mixer", "Mic Capture Switch", "MICR"}, + {"Input Mixer", "Line Capture Switch", "Left Line Input"}, + {"Input Mixer", "Line Capture Switch", "Right Line Input"}, + {"ADCL", NULL, "Input Mixer"}, + {"ADCR", NULL, "Input Mixer"}, + + {"Digital", "Sidetone Switch", "ADCL"}, + {"Digital", "Sidetone Switch", "ADCR"}, + {"DACL", NULL, "Digital"}, + {"DACR", NULL, "Digital"}, + + {"Output Mixer", "Line Bypass Switch", "Left Line Input"}, + {"Output Mixer", "Line Bypass Switch", "Right Line Input"}, + {"Output Mixer", NULL, "DACL"}, + {"Output Mixer", NULL, "DACR"}, + {"Master Playback", "Switch", "Output Mixer"}, + {"LOUT", NULL, "Master Playback"}, + {"ROUT", NULL, "Master Playback"}, +}; + +static const unsigned int max9867_rates_44k1[] = { + 11025, 22050, 44100, +}; + +static const struct snd_pcm_hw_constraint_list max9867_constraints_44k1 = { + .list = max9867_rates_44k1, + .count = ARRAY_SIZE(max9867_rates_44k1), }; -enum rates { - pcm_rate_8, pcm_rate_16, pcm_rate_24, - pcm_rate_32, pcm_rate_44, - pcm_rate_48, max_pcm_rate, +static const unsigned int max9867_rates_48k[] = { + 8000, 16000, 32000, 48000, }; -static const struct ni_div_rates { - u32 mclk; - u16 ni[max_pcm_rate]; -} ni_div[] = { - {11289600, {0x116A, 0x22D4, 0x343F, 0x45A9, 0x6000, 0x687D} }, - {12000000, {0x1062, 0x20C5, 0x3127, 0x4189, 0x5A51, 0x624E} }, - {12288000, {0x1000, 0x2000, 0x3000, 0x4000, 0x5833, 0x6000} }, - {13000000, {0x0F20, 0x1E3F, 0x2D5F, 0x3C7F, 0x535F, 0x5ABE} }, - {19200000, {0x0A3D, 0x147B, 0x1EB8, 0x28F6, 0x3873, 0x3D71} }, - {24000000, {0x1062, 0x20C5, 0x1893, 0x4189, 0x5A51, 0x624E} }, - {26000000, {0x0F20, 0x1E3F, 0x16AF, 0x3C7F, 0x535F, 0x5ABE} }, - {27000000, {0x0E90, 0x1D21, 0x15D8, 0x3A41, 0x5048, 0x5762} }, +static const struct snd_pcm_hw_constraint_list max9867_constraints_48k = { + .list = max9867_rates_48k, + .count = ARRAY_SIZE(max9867_rates_48k), }; -static inline int get_ni_value(int mclk, int rate) +struct max9867_priv { + struct regmap *regmap; + const struct snd_pcm_hw_constraint_list *constraints; + unsigned int sysclk, pclk; + bool master, dsp_a; +}; + +static int max9867_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { - int i, ret = 0; + struct max9867_priv *max9867 = + snd_soc_component_get_drvdata(dai->component); - /* find the closest rate index*/ - for (i = 0; i < ARRAY_SIZE(ni_div); i++) { - if (ni_div[i].mclk >= mclk) - break; - } - if (i == ARRAY_SIZE(ni_div)) - return -EINVAL; + if (max9867->constraints) + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, max9867->constraints); - switch (rate) { - case 8000: - return ni_div[i].ni[pcm_rate_8]; - case 16000: - return ni_div[i].ni[pcm_rate_16]; - case 32000: - return ni_div[i].ni[pcm_rate_32]; - case 44100: - return ni_div[i].ni[pcm_rate_44]; - case 48000: - return ni_div[i].ni[pcm_rate_48]; - default: - pr_err("%s wrong rate %d\n", __func__, rate); - ret = -EINVAL; - } - return ret; + return 0; } static int max9867_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + int value; + unsigned long int rate, ratio; struct snd_soc_component *component = dai->component; struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); - unsigned int ni_h, ni_l; - int value; + unsigned int ni = DIV_ROUND_CLOSEST_ULL(96ULL * 0x10000 * params_rate(params), + max9867->pclk); - value = get_ni_value(max9867->sysclk, params_rate(params)); - if (value < 0) - return value; - - ni_h = (0xFF00 & value) >> 8; - ni_l = 0x00FF & value; /* set up the ni value */ regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKHIGH, - MAX9867_NI_HIGH_MASK, ni_h); + MAX9867_NI_HIGH_MASK, (0xFF00 & ni) >> 8); regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKLOW, - MAX9867_NI_LOW_MASK, ni_l); - if (!max9867->master) { - /* - * digital pll locks on to any externally supplied LRCLK signal - * and also enable rapid lock mode. - */ - regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKLOW, - MAX9867_RAPID_LOCK, MAX9867_RAPID_LOCK); - regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKHIGH, - MAX9867_PLL, MAX9867_PLL); - } else { - unsigned long int bclk_rate, pclk_bclk_ratio; - int bclk_value; - - bclk_rate = params_rate(params) * 2 * params_width(params); - pclk_bclk_ratio = max9867->pclk/bclk_rate; - switch (params_width(params)) { - case 8: - case 16: - switch (pclk_bclk_ratio) { - case 2: - bclk_value = MAX9867_IFC1B_PCLK_2; - break; - case 4: - bclk_value = MAX9867_IFC1B_PCLK_4; - break; + MAX9867_NI_LOW_MASK, 0x00FF & ni); + if (max9867->master) { + if (max9867->dsp_a) { + value = MAX9867_IFC1B_48X; + } else { + rate = params_rate(params) * 2 * params_width(params); + ratio = max9867->pclk / rate; + switch (params_width(params)) { case 8: - bclk_value = MAX9867_IFC1B_PCLK_8; - break; case 16: - bclk_value = MAX9867_IFC1B_PCLK_16; + switch (ratio) { + case 2: + value = MAX9867_IFC1B_PCLK_2; + break; + case 4: + value = MAX9867_IFC1B_PCLK_4; + break; + case 8: + value = MAX9867_IFC1B_PCLK_8; + break; + case 16: + value = MAX9867_IFC1B_PCLK_16; + break; + default: + return -EINVAL; + } + break; + case 24: + value = MAX9867_IFC1B_48X; + break; + case 32: + value = MAX9867_IFC1B_64X; break; default: - dev_err(component->dev, - "unsupported sampling rate\n"); return -EINVAL; } - break; - case 24: - bclk_value = MAX9867_IFC1B_24BIT; - break; - case 32: - bclk_value = MAX9867_IFC1B_32BIT; - break; - default: - dev_err(component->dev, "unsupported sampling rate\n"); - return -EINVAL; } regmap_update_bits(max9867->regmap, MAX9867_IFC1B, - MAX9867_IFC1B_BCLK_MASK, bclk_value); + MAX9867_IFC1B_BCLK_MASK, value); + } else { + /* + * digital pll locks on to any externally supplied LRCLK signal + * and also enable rapid lock mode. + */ + regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKLOW, + MAX9867_RAPID_LOCK, MAX9867_RAPID_LOCK); + regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKHIGH, + MAX9867_PLL, MAX9867_PLL); } return 0; } -static int max9867_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_component *component = dai->component; - struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); - - regmap_update_bits(max9867->regmap, MAX9867_PWRMAN, - MAX9867_SHTDOWN_MASK, MAX9867_SHTDOWN_MASK); - return 0; -} - static int max9867_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_component *component = dai->component; struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); - if (mute) - regmap_update_bits(max9867->regmap, MAX9867_DACLEVEL, - MAX9867_DAC_MUTE_MASK, MAX9867_DAC_MUTE_MASK); - else - regmap_update_bits(max9867->regmap, MAX9867_DACLEVEL, - MAX9867_DAC_MUTE_MASK, 0); - return 0; + return regmap_update_bits(max9867->regmap, MAX9867_DACLEVEL, + 1 << 6, !!mute << 6); } static int max9867_set_dai_sysclk(struct snd_soc_dai *codec_dai, @@ -283,21 +264,29 @@ static int max9867_set_dai_sysclk(struct snd_soc_dai *codec_dai, /* Set the prescaler based on the master clock frequency*/ if (freq >= 10000000 && freq <= 20000000) { value |= MAX9867_PSCLK_10_20; - max9867->pclk = freq; + max9867->pclk = freq; } else if (freq >= 20000000 && freq <= 40000000) { value |= MAX9867_PSCLK_20_40; - max9867->pclk = freq/2; + max9867->pclk = freq / 2; } else if (freq >= 40000000 && freq <= 60000000) { value |= MAX9867_PSCLK_40_60; - max9867->pclk = freq/4; + max9867->pclk = freq / 4; } else { dev_err(component->dev, "Invalid clock frequency %uHz (required 10-60MHz)\n", freq); return -EINVAL; } - value = value << MAX9867_PSCLK_SHIFT; + if (freq % 48000 == 0) + max9867->constraints = &max9867_constraints_48k; + else if (freq % 44100 == 0) + max9867->constraints = &max9867_constraints_44k1; + else + dev_warn(component->dev, + "Unable to set exact rate with %uHz clock frequency\n", + freq); max9867->sysclk = freq; + value = value << MAX9867_PSCLK_SHIFT; /* exact integer mode is not supported */ value &= ~MAX9867_FREQ_MASK; regmap_update_bits(max9867->regmap, MAX9867_SYSCLK, @@ -310,16 +299,17 @@ static int max9867_dai_set_fmt(struct snd_soc_dai *codec_dai, { struct snd_soc_component *component = codec_dai->component; struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); - u8 iface1A = 0, iface1B = 0; + u8 iface1A, iface1B; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - max9867->master = 1; - iface1A |= MAX9867_MASTER; + max9867->master = true; + iface1A = MAX9867_MASTER; + iface1B = MAX9867_IFC1B_48X; break; case SND_SOC_DAIFMT_CBS_CFS: - max9867->master = 0; - iface1A &= ~MAX9867_MASTER; + max9867->master = false; + iface1A = iface1B = 0; break; default: return -EINVAL; @@ -327,9 +317,11 @@ static int max9867_dai_set_fmt(struct snd_soc_dai *codec_dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: + max9867->dsp_a = false; iface1A |= MAX9867_I2S_DLY; break; case SND_SOC_DAIFMT_DSP_A: + max9867->dsp_a = true; iface1A |= MAX9867_TDM_MODE | MAX9867_SDOUT_HIZ; break; default: @@ -355,21 +347,18 @@ static int max9867_dai_set_fmt(struct snd_soc_dai *codec_dai, regmap_write(max9867->regmap, MAX9867_IFC1A, iface1A); regmap_write(max9867->regmap, MAX9867_IFC1B, iface1B); + return 0; } static const struct snd_soc_dai_ops max9867_dai_ops = { - .set_fmt = max9867_dai_set_fmt, .set_sysclk = max9867_set_dai_sysclk, - .prepare = max9867_prepare, + .set_fmt = max9867_dai_set_fmt, .digital_mute = max9867_mute, - .hw_params = max9867_dai_hw_params, + .startup = max9867_startup, + .hw_params = max9867_dai_hw_params, }; -#define MAX9867_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ - SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) -#define MAX9867_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) - static struct snd_soc_dai_driver max9867_dai[] = { { .name = "max9867-aif1", @@ -377,42 +366,74 @@ static struct snd_soc_dai_driver max9867_dai[] = { .stream_name = "HiFi Playback", .channels_min = 2, .channels_max = 2, - .rates = MAX9867_RATES, - .formats = MAX9867_FORMATS, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { .stream_name = "HiFi Capture", .channels_min = 2, .channels_max = 2, - .rates = MAX9867_RATES, - .formats = MAX9867_FORMATS, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .ops = &max9867_dai_ops, .symmetric_rates = 1, } }; -#ifdef CONFIG_PM_SLEEP -static int max9867_suspend(struct device *dev) +#ifdef CONFIG_PM +static int max9867_suspend(struct snd_soc_component *component) { - struct max9867_priv *max9867 = dev_get_drvdata(dev); + snd_soc_component_force_bias_level(component, SND_SOC_BIAS_OFF); - /* Drop down to power saving mode when system is suspended */ - regmap_update_bits(max9867->regmap, MAX9867_PWRMAN, - MAX9867_SHTDOWN_MASK, ~MAX9867_SHTDOWN_MASK); return 0; } -static int max9867_resume(struct device *dev) +static int max9867_resume(struct snd_soc_component *component) { - struct max9867_priv *max9867 = dev_get_drvdata(dev); + snd_soc_component_force_bias_level(component, SND_SOC_BIAS_STANDBY); - regmap_update_bits(max9867->regmap, MAX9867_PWRMAN, - MAX9867_SHTDOWN_MASK, MAX9867_SHTDOWN_MASK); return 0; } +#else +#define max9867_suspend NULL +#define max9867_resume NULL #endif +static int max9867_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + int err; + struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); + + switch (level) { + case SND_SOC_BIAS_STANDBY: + if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) { + err = regcache_sync(max9867->regmap); + if (err) + return err; + + err = regmap_update_bits(max9867->regmap, MAX9867_PWRMAN, + MAX9867_SHTDOWN, MAX9867_SHTDOWN); + if (err) + return err; + } + break; + case SND_SOC_BIAS_OFF: + err = regmap_update_bits(max9867->regmap, MAX9867_PWRMAN, + MAX9867_SHTDOWN, 0); + if (err) + return err; + + regcache_mark_dirty(max9867->regmap); + break; + default: + break; + } + + return 0; +} + static const struct snd_soc_component_driver max9867_component = { .controls = max9867_snd_controls, .num_controls = ARRAY_SIZE(max9867_snd_controls), @@ -420,6 +441,9 @@ static const struct snd_soc_component_driver max9867_component = { .num_dapm_routes = ARRAY_SIZE(max9867_audio_map), .dapm_widgets = max9867_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(max9867_dapm_widgets), + .suspend = max9867_suspend, + .resume = max9867_resume, + .set_bias_level = max9867_set_bias_level, .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, @@ -450,8 +474,8 @@ static const struct reg_default max9867_reg[] = { { 0x0B, 0x00 }, { 0x0C, 0x00 }, { 0x0D, 0x00 }, - { 0x0E, 0x00 }, - { 0x0F, 0x00 }, + { 0x0E, 0x40 }, + { 0x0F, 0x40 }, { 0x10, 0x00 }, { 0x11, 0x00 }, { 0x12, 0x00 }, @@ -476,10 +500,9 @@ static int max9867_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct max9867_priv *max9867; - int ret = 0, reg; + int ret, reg; - max9867 = devm_kzalloc(&i2c->dev, - sizeof(*max9867), GFP_KERNEL); + max9867 = devm_kzalloc(&i2c->dev, sizeof(*max9867), GFP_KERNEL); if (!max9867) return -ENOMEM; @@ -490,8 +513,7 @@ static int max9867_i2c_probe(struct i2c_client *i2c, dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); return ret; } - ret = regmap_read(max9867->regmap, - MAX9867_REVISION, ®); + ret = regmap_read(max9867->regmap, MAX9867_REVISION, ®); if (ret < 0) { dev_err(&i2c->dev, "Failed to read: %d\n", ret); return ret; @@ -499,10 +521,8 @@ static int max9867_i2c_probe(struct i2c_client *i2c, dev_info(&i2c->dev, "device revision: %x\n", reg); ret = devm_snd_soc_register_component(&i2c->dev, &max9867_component, max9867_dai, ARRAY_SIZE(max9867_dai)); - if (ret < 0) { + if (ret < 0) dev_err(&i2c->dev, "Failed to register component: %d\n", ret); - return ret; - } return ret; } @@ -518,15 +538,10 @@ static const struct of_device_id max9867_of_match[] = { }; MODULE_DEVICE_TABLE(of, max9867_of_match); -static const struct dev_pm_ops max9867_pm_ops = { - SET_SYSTEM_SLEEP_PM_OPS(max9867_suspend, max9867_resume) -}; - static struct i2c_driver max9867_i2c_driver = { .driver = { .name = "max9867", .of_match_table = of_match_ptr(max9867_of_match), - .pm = &max9867_pm_ops, }, .probe = max9867_i2c_probe, .id_table = max9867_i2c_id, @@ -534,6 +549,6 @@ static struct i2c_driver max9867_i2c_driver = { module_i2c_driver(max9867_i2c_driver); -MODULE_AUTHOR("anish kumar <yesanishhere@gmail.com>"); -MODULE_DESCRIPTION("ALSA SoC MAX9867 driver"); +MODULE_AUTHOR("Ladislav Michl <ladis@linux-mips.org>"); +MODULE_DESCRIPTION("ASoC MAX9867 driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max9867.h b/sound/soc/codecs/max9867.h index 55cd9976ff47..2277798291a1 100644 --- a/sound/soc/codecs/max9867.h +++ b/sound/soc/codecs/max9867.h @@ -26,13 +26,11 @@ #define MAX9867_PSCLK_10_20 0x1 #define MAX9867_PSCLK_20_40 0x2 #define MAX9867_PSCLK_40_60 0x3 -#define MAX9867_AUDIOCLKHIGH 0x06 -#define MAX9867_NI_HIGH_WIDTH 0x7 -#define MAX9867_NI_HIGH_MASK 0x7F -#define MAX9867_NI_LOW_MASK 0x7F -#define MAX9867_NI_LOW_SHIFT 0x1 -#define MAX9867_PLL (1<<7) -#define MAX9867_AUDIOCLKLOW 0x07 +#define MAX9867_AUDIOCLKHIGH 0x06 +#define MAX9867_NI_HIGH_MASK 0x7F +#define MAX9867_NI_LOW_MASK 0xFE +#define MAX9867_PLL (1<<7) +#define MAX9867_AUDIOCLKLOW 0x07 #define MAX9867_RAPID_LOCK 0x01 #define MAX9867_IFC1A 0x08 #define MAX9867_MASTER (1<<7) @@ -43,40 +41,29 @@ #define MAX9867_BCI_MODE (1<<5) #define MAX9867_IFC1B 0x09 #define MAX9867_IFC1B_BCLK_MASK 7 -#define MAX9867_IFC1B_32BIT 0x01 -#define MAX9867_IFC1B_24BIT 0x02 -#define MAX9867_IFC1B_PCLK_2 4 -#define MAX9867_IFC1B_PCLK_4 5 -#define MAX9867_IFC1B_PCLK_8 6 -#define MAX9867_IFC1B_PCLK_16 7 +#define MAX9867_IFC1B_64X 0x01 +#define MAX9867_IFC1B_48X 0x02 +#define MAX9867_IFC1B_PCLK_2 0x04 +#define MAX9867_IFC1B_PCLK_4 0x05 +#define MAX9867_IFC1B_PCLK_8 0x06 +#define MAX9867_IFC1B_PCLK_16 0x07 #define MAX9867_CODECFLTR 0x0a -#define MAX9867_DACGAIN 0x0b +#define MAX9867_SIDETONE 0x0b #define MAX9867_DACLEVEL 0x0c -#define MAX9867_DAC_MUTE_SHIFT 0x6 -#define MAX9867_DAC_MUTE_WIDTH 0x1 -#define MAX9867_DAC_MUTE_MASK (0x1<<MAX9867_DAC_MUTE_SHIFT) #define MAX9867_ADCLEVEL 0x0d #define MAX9867_LEFTLINELVL 0x0e -#define MAX9867_RIGTHLINELVL 0x0f +#define MAX9867_RIGHTLINELVL 0x0f #define MAX9867_LEFTVOL 0x10 #define MAX9867_RIGHTVOL 0x11 #define MAX9867_LEFTMICGAIN 0x12 #define MAX9867_RIGHTMICGAIN 0x13 #define MAX9867_INPUTCONFIG 0x14 -#define MAX9867_INPUT_SHIFT 0x6 #define MAX9867_MICCONFIG 0x15 #define MAX9867_MODECONFIG 0x16 #define MAX9867_PWRMAN 0x17 -#define MAX9867_SHTDOWN_MASK (1<<7) +#define MAX9867_SHTDOWN 0x80 #define MAX9867_REVISION 0xff #define MAX9867_CACHEREGNUM 10 -/* codec private data */ -struct max9867_priv { - struct regmap *regmap; - unsigned int sysclk; - unsigned int pclk; - unsigned int master; -}; #endif diff --git a/sound/soc/codecs/nau8540.c b/sound/soc/codecs/nau8540.c index e3c8cd17daf2..4dd1a609756b 100644 --- a/sound/soc/codecs/nau8540.c +++ b/sound/soc/codecs/nau8540.c @@ -585,7 +585,7 @@ static int nau8540_calc_fll_param(unsigned int fll_in, fvco_max = 0; fvco_sel = ARRAY_SIZE(mclk_src_scaling); for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { - fvco = 256 * fs * 2 * mclk_src_scaling[i].param; + fvco = 256ULL * fs * 2 * mclk_src_scaling[i].param; if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && fvco_max < fvco) { fvco_max = fvco; diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c new file mode 100644 index 000000000000..c6152a044416 --- /dev/null +++ b/sound/soc/codecs/nau8822.c @@ -0,0 +1,1132 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// nau8822.c -- NAU8822 ALSA Soc Audio driver +// +// Copyright 2017 Nuvoton Technology Crop. +// +// Author: David Lin <ctlin0@nuvoton.com> +// Co-author: John Hsu <kchsu0@nuvoton.com> +// Co-author: Seven Li <wtli@nuvoton.com> +// +// Based on WM8974.c + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.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/initval.h> +#include <sound/tlv.h> +#include <asm/div64.h> +#include "nau8822.h" + +#define NAU_PLL_FREQ_MAX 100000000 +#define NAU_PLL_FREQ_MIN 90000000 +#define NAU_PLL_REF_MAX 33000000 +#define NAU_PLL_REF_MIN 8000000 +#define NAU_PLL_OPTOP_MIN 6 + +static const int nau8822_mclk_scaler[] = { 10, 15, 20, 30, 40, 60, 80, 120 }; + +static const struct reg_default nau8822_reg_defaults[] = { + { NAU8822_REG_POWER_MANAGEMENT_1, 0x0000 }, + { NAU8822_REG_POWER_MANAGEMENT_2, 0x0000 }, + { NAU8822_REG_POWER_MANAGEMENT_3, 0x0000 }, + { NAU8822_REG_AUDIO_INTERFACE, 0x0050 }, + { NAU8822_REG_COMPANDING_CONTROL, 0x0000 }, + { NAU8822_REG_CLOCKING, 0x0140 }, + { NAU8822_REG_ADDITIONAL_CONTROL, 0x0000 }, + { NAU8822_REG_GPIO_CONTROL, 0x0000 }, + { NAU8822_REG_JACK_DETECT_CONTROL_1, 0x0000 }, + { NAU8822_REG_DAC_CONTROL, 0x0000 }, + { NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME, 0x00ff }, + { NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME, 0x00ff }, + { NAU8822_REG_JACK_DETECT_CONTROL_2, 0x0000 }, + { NAU8822_REG_ADC_CONTROL, 0x0100 }, + { NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME, 0x00ff }, + { NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME, 0x00ff }, + { NAU8822_REG_EQ1, 0x012c }, + { NAU8822_REG_EQ2, 0x002c }, + { NAU8822_REG_EQ3, 0x002c }, + { NAU8822_REG_EQ4, 0x002c }, + { NAU8822_REG_EQ5, 0x002c }, + { NAU8822_REG_DAC_LIMITER_1, 0x0032 }, + { NAU8822_REG_DAC_LIMITER_2, 0x0000 }, + { NAU8822_REG_NOTCH_FILTER_1, 0x0000 }, + { NAU8822_REG_NOTCH_FILTER_2, 0x0000 }, + { NAU8822_REG_NOTCH_FILTER_3, 0x0000 }, + { NAU8822_REG_NOTCH_FILTER_4, 0x0000 }, + { NAU8822_REG_ALC_CONTROL_1, 0x0038 }, + { NAU8822_REG_ALC_CONTROL_2, 0x000b }, + { NAU8822_REG_ALC_CONTROL_3, 0x0032 }, + { NAU8822_REG_NOISE_GATE, 0x0010 }, + { NAU8822_REG_PLL_N, 0x0008 }, + { NAU8822_REG_PLL_K1, 0x000c }, + { NAU8822_REG_PLL_K2, 0x0093 }, + { NAU8822_REG_PLL_K3, 0x00e9 }, + { NAU8822_REG_3D_CONTROL, 0x0000 }, + { NAU8822_REG_RIGHT_SPEAKER_CONTROL, 0x0000 }, + { NAU8822_REG_INPUT_CONTROL, 0x0033 }, + { NAU8822_REG_LEFT_INP_PGA_CONTROL, 0x0010 }, + { NAU8822_REG_RIGHT_INP_PGA_CONTROL, 0x0010 }, + { NAU8822_REG_LEFT_ADC_BOOST_CONTROL, 0x0100 }, + { NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 0x0100 }, + { NAU8822_REG_OUTPUT_CONTROL, 0x0002 }, + { NAU8822_REG_LEFT_MIXER_CONTROL, 0x0001 }, + { NAU8822_REG_RIGHT_MIXER_CONTROL, 0x0001 }, + { NAU8822_REG_LHP_VOLUME, 0x0039 }, + { NAU8822_REG_RHP_VOLUME, 0x0039 }, + { NAU8822_REG_LSPKOUT_VOLUME, 0x0039 }, + { NAU8822_REG_RSPKOUT_VOLUME, 0x0039 }, + { NAU8822_REG_AUX2_MIXER, 0x0001 }, + { NAU8822_REG_AUX1_MIXER, 0x0001 }, + { NAU8822_REG_POWER_MANAGEMENT_4, 0x0000 }, + { NAU8822_REG_LEFT_TIME_SLOT, 0x0000 }, + { NAU8822_REG_MISC, 0x0020 }, + { NAU8822_REG_RIGHT_TIME_SLOT, 0x0000 }, + { NAU8822_REG_DEVICE_REVISION, 0x007f }, + { NAU8822_REG_DEVICE_ID, 0x001a }, + { NAU8822_REG_DAC_DITHER, 0x0114 }, + { NAU8822_REG_ALC_ENHANCE_1, 0x0000 }, + { NAU8822_REG_ALC_ENHANCE_2, 0x0000 }, + { NAU8822_REG_192KHZ_SAMPLING, 0x0008 }, + { NAU8822_REG_MISC_CONTROL, 0x0000 }, + { NAU8822_REG_INPUT_TIEOFF, 0x0000 }, + { NAU8822_REG_POWER_REDUCTION, 0x0000 }, + { NAU8822_REG_AGC_PEAK2PEAK, 0x0000 }, + { NAU8822_REG_AGC_PEAK_DETECT, 0x0000 }, + { NAU8822_REG_AUTOMUTE_CONTROL, 0x0000 }, + { NAU8822_REG_OUTPUT_TIEOFF, 0x0000 }, +}; + +static bool nau8822_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8822_REG_RESET ... NAU8822_REG_JACK_DETECT_CONTROL_1: + case NAU8822_REG_DAC_CONTROL ... NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME: + case NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME: + case NAU8822_REG_EQ1 ... NAU8822_REG_EQ5: + case NAU8822_REG_DAC_LIMITER_1 ... NAU8822_REG_DAC_LIMITER_2: + case NAU8822_REG_NOTCH_FILTER_1 ... NAU8822_REG_NOTCH_FILTER_4: + case NAU8822_REG_ALC_CONTROL_1 ...NAU8822_REG_PLL_K3: + case NAU8822_REG_3D_CONTROL: + case NAU8822_REG_RIGHT_SPEAKER_CONTROL: + case NAU8822_REG_INPUT_CONTROL ... NAU8822_REG_LEFT_ADC_BOOST_CONTROL: + case NAU8822_REG_RIGHT_ADC_BOOST_CONTROL ... NAU8822_REG_AUX1_MIXER: + case NAU8822_REG_POWER_MANAGEMENT_4 ... NAU8822_REG_DEVICE_ID: + case NAU8822_REG_DAC_DITHER: + case NAU8822_REG_ALC_ENHANCE_1 ... NAU8822_REG_MISC_CONTROL: + case NAU8822_REG_INPUT_TIEOFF ... NAU8822_REG_OUTPUT_TIEOFF: + return true; + default: + return false; + } +} + +static bool nau8822_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8822_REG_RESET ... NAU8822_REG_JACK_DETECT_CONTROL_1: + case NAU8822_REG_DAC_CONTROL ... NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME: + case NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME: + case NAU8822_REG_EQ1 ... NAU8822_REG_EQ5: + case NAU8822_REG_DAC_LIMITER_1 ... NAU8822_REG_DAC_LIMITER_2: + case NAU8822_REG_NOTCH_FILTER_1 ... NAU8822_REG_NOTCH_FILTER_4: + case NAU8822_REG_ALC_CONTROL_1 ...NAU8822_REG_PLL_K3: + case NAU8822_REG_3D_CONTROL: + case NAU8822_REG_RIGHT_SPEAKER_CONTROL: + case NAU8822_REG_INPUT_CONTROL ... NAU8822_REG_LEFT_ADC_BOOST_CONTROL: + case NAU8822_REG_RIGHT_ADC_BOOST_CONTROL ... NAU8822_REG_AUX1_MIXER: + case NAU8822_REG_POWER_MANAGEMENT_4 ... NAU8822_REG_DEVICE_ID: + case NAU8822_REG_DAC_DITHER: + case NAU8822_REG_ALC_ENHANCE_1 ... NAU8822_REG_MISC_CONTROL: + case NAU8822_REG_INPUT_TIEOFF ... NAU8822_REG_OUTPUT_TIEOFF: + return true; + default: + return false; + } +} + +static bool nau8822_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8822_REG_RESET: + case NAU8822_REG_DEVICE_REVISION: + case NAU8822_REG_DEVICE_ID: + case NAU8822_REG_AGC_PEAK2PEAK: + case NAU8822_REG_AGC_PEAK_DETECT: + case NAU8822_REG_AUTOMUTE_CONTROL: + return true; + default: + return false; + } +} + +/* The EQ parameters get function is to get the 5 band equalizer control. + * The regmap raw read can't work here because regmap doesn't provide + * value format for value width of 9 bits. Therefore, the driver reads data + * from cache and makes value format according to the endianness of + * bytes type control element. + */ +static int nau8822_eq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + int i, reg; + u16 reg_val, *val; + + val = (u16 *)ucontrol->value.bytes.data; + reg = NAU8822_REG_EQ1; + for (i = 0; i < params->max / sizeof(u16); i++) { + reg_val = snd_soc_component_read32(component, reg + i); + /* conversion of 16-bit integers between native CPU format + * and big endian format + */ + reg_val = cpu_to_be16(reg_val); + memcpy(val + i, ®_val, sizeof(reg_val)); + } + + return 0; +} + +/* The EQ parameters put function is to make configuration of 5 band equalizer + * control. These configuration includes central frequency, equalizer gain, + * cut-off frequency, bandwidth control, and equalizer path. + * The regmap raw write can't work here because regmap doesn't provide + * register and value format for register with address 7 bits and value 9 bits. + * Therefore, the driver makes value format according to the endianness of + * bytes type control element and writes data to codec. + */ +static int nau8822_eq_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + void *data; + u16 *val, value; + int i, reg, ret; + + data = kmemdup(ucontrol->value.bytes.data, + params->max, GFP_KERNEL | GFP_DMA); + if (!data) + return -ENOMEM; + + val = (u16 *)data; + reg = NAU8822_REG_EQ1; + for (i = 0; i < params->max / sizeof(u16); i++) { + /* conversion of 16-bit integers between native CPU format + * and big endian format + */ + value = be16_to_cpu(*(val + i)); + ret = snd_soc_component_write(component, reg + i, value); + if (ret) { + dev_err(component->dev, + "EQ configuration fail, register: %x ret: %d\n", + reg + i, ret); + kfree(data); + return ret; + } + } + kfree(data); + + return 0; +} + +static const char * const nau8822_companding[] = { + "Off", "NC", "u-law", "A-law"}; + +static const struct soc_enum nau8822_companding_adc_enum = + SOC_ENUM_SINGLE(NAU8822_REG_COMPANDING_CONTROL, NAU8822_ADCCM_SFT, + ARRAY_SIZE(nau8822_companding), nau8822_companding); + +static const struct soc_enum nau8822_companding_dac_enum = + SOC_ENUM_SINGLE(NAU8822_REG_COMPANDING_CONTROL, NAU8822_DACCM_SFT, + ARRAY_SIZE(nau8822_companding), nau8822_companding); + +static const char * const nau8822_eqmode[] = {"Capture", "Playback"}; + +static const struct soc_enum nau8822_eqmode_enum = + SOC_ENUM_SINGLE(NAU8822_REG_EQ1, NAU8822_EQM_SFT, + ARRAY_SIZE(nau8822_eqmode), nau8822_eqmode); + +static const char * const nau8822_alc1[] = {"Off", "Right", "Left", "Both"}; +static const char * const nau8822_alc3[] = {"Normal", "Limiter"}; + +static const struct soc_enum nau8822_alc_enable_enum = + SOC_ENUM_SINGLE(NAU8822_REG_ALC_CONTROL_1, NAU8822_ALCEN_SFT, + ARRAY_SIZE(nau8822_alc1), nau8822_alc1); + +static const struct soc_enum nau8822_alc_mode_enum = + SOC_ENUM_SINGLE(NAU8822_REG_ALC_CONTROL_3, NAU8822_ALCM_SFT, + ARRAY_SIZE(nau8822_alc3), nau8822_alc3); + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0); +static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0); +static const DECLARE_TLV_DB_SCALE(pga_boost_tlv, 0, 2000, 0); +static const DECLARE_TLV_DB_SCALE(boost_tlv, -1500, 300, 1); +static const DECLARE_TLV_DB_SCALE(limiter_tlv, 0, 100, 0); + +static const struct snd_kcontrol_new nau8822_snd_controls[] = { + SOC_ENUM("ADC Companding", nau8822_companding_adc_enum), + SOC_ENUM("DAC Companding", nau8822_companding_dac_enum), + + SOC_ENUM("EQ Function", nau8822_eqmode_enum), + SND_SOC_BYTES_EXT("EQ Parameters", 10, + nau8822_eq_get, nau8822_eq_put), + + SOC_DOUBLE("DAC Inversion Switch", + NAU8822_REG_DAC_CONTROL, 0, 1, 1, 0), + SOC_DOUBLE_R_TLV("PCM Volume", + NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME, + NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME, 0, 255, 0, digital_tlv), + + SOC_SINGLE("High Pass Filter Switch", + NAU8822_REG_ADC_CONTROL, 8, 1, 0), + SOC_SINGLE("High Pass Cut Off", + NAU8822_REG_ADC_CONTROL, 4, 7, 0), + + SOC_DOUBLE("ADC Inversion Switch", + NAU8822_REG_ADC_CONTROL, 0, 1, 1, 0), + SOC_DOUBLE_R_TLV("ADC Volume", + NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME, + NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME, 0, 255, 0, digital_tlv), + + SOC_SINGLE("DAC Limiter Switch", + NAU8822_REG_DAC_LIMITER_1, 8, 1, 0), + SOC_SINGLE("DAC Limiter Decay", + NAU8822_REG_DAC_LIMITER_1, 4, 15, 0), + SOC_SINGLE("DAC Limiter Attack", + NAU8822_REG_DAC_LIMITER_1, 0, 15, 0), + SOC_SINGLE("DAC Limiter Threshold", + NAU8822_REG_DAC_LIMITER_2, 4, 7, 0), + SOC_SINGLE_TLV("DAC Limiter Volume", + NAU8822_REG_DAC_LIMITER_2, 0, 12, 0, limiter_tlv), + + SOC_ENUM("ALC Mode", nau8822_alc_mode_enum), + SOC_ENUM("ALC Enable Switch", nau8822_alc_enable_enum), + SOC_SINGLE("ALC Min Gain", + NAU8822_REG_ALC_CONTROL_1, 0, 7, 0), + SOC_SINGLE("ALC Max Gain", + NAU8822_REG_ALC_CONTROL_1, 3, 7, 0), + SOC_SINGLE("ALC Hold", + NAU8822_REG_ALC_CONTROL_2, 4, 10, 0), + SOC_SINGLE("ALC Target", + NAU8822_REG_ALC_CONTROL_2, 0, 15, 0), + SOC_SINGLE("ALC Decay", + NAU8822_REG_ALC_CONTROL_3, 4, 10, 0), + SOC_SINGLE("ALC Attack", + NAU8822_REG_ALC_CONTROL_3, 0, 10, 0), + SOC_SINGLE("ALC Noise Gate Switch", + NAU8822_REG_NOISE_GATE, 3, 1, 0), + SOC_SINGLE("ALC Noise Gate Threshold", + NAU8822_REG_NOISE_GATE, 0, 7, 0), + + SOC_DOUBLE_R("PGA ZC Switch", + NAU8822_REG_LEFT_INP_PGA_CONTROL, + NAU8822_REG_RIGHT_INP_PGA_CONTROL, + 7, 1, 0), + SOC_DOUBLE_R_TLV("PGA Volume", + NAU8822_REG_LEFT_INP_PGA_CONTROL, + NAU8822_REG_RIGHT_INP_PGA_CONTROL, 0, 63, 0, inpga_tlv), + + SOC_DOUBLE_R("Headphone ZC Switch", + NAU8822_REG_LHP_VOLUME, + NAU8822_REG_RHP_VOLUME, 7, 1, 0), + SOC_DOUBLE_R("Headphone Playback Switch", + NAU8822_REG_LHP_VOLUME, + NAU8822_REG_RHP_VOLUME, 6, 1, 1), + SOC_DOUBLE_R_TLV("Headphone Volume", + NAU8822_REG_LHP_VOLUME, + NAU8822_REG_RHP_VOLUME, 0, 63, 0, spk_tlv), + + SOC_DOUBLE_R("Speaker ZC Switch", + NAU8822_REG_LSPKOUT_VOLUME, + NAU8822_REG_RSPKOUT_VOLUME, 7, 1, 0), + SOC_DOUBLE_R("Speaker Playback Switch", + NAU8822_REG_LSPKOUT_VOLUME, + NAU8822_REG_RSPKOUT_VOLUME, 6, 1, 1), + SOC_DOUBLE_R_TLV("Speaker Volume", + NAU8822_REG_LSPKOUT_VOLUME, + NAU8822_REG_RSPKOUT_VOLUME, 0, 63, 0, spk_tlv), + + SOC_DOUBLE_R("AUXOUT Playback Switch", + NAU8822_REG_AUX2_MIXER, + NAU8822_REG_AUX1_MIXER, 6, 1, 1), + + SOC_DOUBLE_R_TLV("PGA Boost Volume", + NAU8822_REG_LEFT_ADC_BOOST_CONTROL, + NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 8, 1, 0, pga_boost_tlv), + SOC_DOUBLE_R_TLV("L2/R2 Boost Volume", + NAU8822_REG_LEFT_ADC_BOOST_CONTROL, + NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 4, 7, 0, boost_tlv), + SOC_DOUBLE_R_TLV("Aux Boost Volume", + NAU8822_REG_LEFT_ADC_BOOST_CONTROL, + NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 0, 7, 0, boost_tlv), + + SOC_SINGLE("DAC 128x Oversampling Switch", + NAU8822_REG_DAC_CONTROL, 5, 1, 0), + SOC_SINGLE("ADC 128x Oversampling Switch", + NAU8822_REG_ADC_CONTROL, 5, 1, 0), +}; + +/* LMAIN and RMAIN Mixer */ +static const struct snd_kcontrol_new nau8822_left_out_mixer[] = { + SOC_DAPM_SINGLE("LINMIX Switch", + NAU8822_REG_LEFT_MIXER_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("LAUX Switch", + NAU8822_REG_LEFT_MIXER_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("LDAC Switch", + NAU8822_REG_LEFT_MIXER_CONTROL, 0, 1, 0), + SOC_DAPM_SINGLE("RDAC Switch", + NAU8822_REG_OUTPUT_CONTROL, 5, 1, 0), +}; + +static const struct snd_kcontrol_new nau8822_right_out_mixer[] = { + SOC_DAPM_SINGLE("RINMIX Switch", + NAU8822_REG_RIGHT_MIXER_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("RAUX Switch", + NAU8822_REG_RIGHT_MIXER_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("RDAC Switch", + NAU8822_REG_RIGHT_MIXER_CONTROL, 0, 1, 0), + SOC_DAPM_SINGLE("LDAC Switch", + NAU8822_REG_OUTPUT_CONTROL, 6, 1, 0), +}; + +/* AUX1 and AUX2 Mixer */ +static const struct snd_kcontrol_new nau8822_auxout1_mixer[] = { + SOC_DAPM_SINGLE("RDAC Switch", NAU8822_REG_AUX1_MIXER, 0, 1, 0), + SOC_DAPM_SINGLE("RMIX Switch", NAU8822_REG_AUX1_MIXER, 1, 1, 0), + SOC_DAPM_SINGLE("RINMIX Switch", NAU8822_REG_AUX1_MIXER, 2, 1, 0), + SOC_DAPM_SINGLE("LDAC Switch", NAU8822_REG_AUX1_MIXER, 3, 1, 0), + SOC_DAPM_SINGLE("LMIX Switch", NAU8822_REG_AUX1_MIXER, 4, 1, 0), +}; + +static const struct snd_kcontrol_new nau8822_auxout2_mixer[] = { + SOC_DAPM_SINGLE("LDAC Switch", NAU8822_REG_AUX2_MIXER, 0, 1, 0), + SOC_DAPM_SINGLE("LMIX Switch", NAU8822_REG_AUX2_MIXER, 1, 1, 0), + SOC_DAPM_SINGLE("LINMIX Switch", NAU8822_REG_AUX2_MIXER, 2, 1, 0), + SOC_DAPM_SINGLE("AUX1MIX Output Switch", + NAU8822_REG_AUX2_MIXER, 3, 1, 0), +}; + +/* Input PGA */ +static const struct snd_kcontrol_new nau8822_left_input_mixer[] = { + SOC_DAPM_SINGLE("L2 Switch", NAU8822_REG_INPUT_CONTROL, 2, 1, 0), + SOC_DAPM_SINGLE("MicN Switch", NAU8822_REG_INPUT_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("MicP Switch", NAU8822_REG_INPUT_CONTROL, 0, 1, 0), +}; +static const struct snd_kcontrol_new nau8822_right_input_mixer[] = { + SOC_DAPM_SINGLE("R2 Switch", NAU8822_REG_INPUT_CONTROL, 6, 1, 0), + SOC_DAPM_SINGLE("MicN Switch", NAU8822_REG_INPUT_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("MicP Switch", NAU8822_REG_INPUT_CONTROL, 4, 1, 0), +}; + +/* Loopback Switch */ +static const struct snd_kcontrol_new nau8822_loopback = + SOC_DAPM_SINGLE("Switch", NAU8822_REG_COMPANDING_CONTROL, + NAU8822_ADDAP_SFT, 1, 0); + +static int check_mclk_select_pll(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(source->dapm); + unsigned int value; + + value = snd_soc_component_read32(component, NAU8822_REG_CLOCKING); + + return (value & NAU8822_CLKM_MASK); +} + +static const struct snd_soc_dapm_widget nau8822_dapm_widgets[] = { + SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", + NAU8822_REG_POWER_MANAGEMENT_3, 0, 0), + SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", + NAU8822_REG_POWER_MANAGEMENT_3, 1, 0), + SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", + NAU8822_REG_POWER_MANAGEMENT_2, 0, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", + NAU8822_REG_POWER_MANAGEMENT_2, 1, 0), + + SOC_MIXER_ARRAY("Left Output Mixer", + NAU8822_REG_POWER_MANAGEMENT_3, 2, 0, nau8822_left_out_mixer), + SOC_MIXER_ARRAY("Right Output Mixer", + NAU8822_REG_POWER_MANAGEMENT_3, 3, 0, nau8822_right_out_mixer), + SOC_MIXER_ARRAY("AUX1 Output Mixer", + NAU8822_REG_POWER_MANAGEMENT_1, 7, 0, nau8822_auxout1_mixer), + SOC_MIXER_ARRAY("AUX2 Output Mixer", + NAU8822_REG_POWER_MANAGEMENT_1, 6, 0, nau8822_auxout2_mixer), + + SOC_MIXER_ARRAY("Left Input Mixer", + NAU8822_REG_POWER_MANAGEMENT_2, + 2, 0, nau8822_left_input_mixer), + SOC_MIXER_ARRAY("Right Input Mixer", + NAU8822_REG_POWER_MANAGEMENT_2, + 3, 0, nau8822_right_input_mixer), + + SND_SOC_DAPM_PGA("Left Boost Mixer", + NAU8822_REG_POWER_MANAGEMENT_2, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Boost Mixer", + NAU8822_REG_POWER_MANAGEMENT_2, 5, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Left Capture PGA", + NAU8822_REG_LEFT_INP_PGA_CONTROL, 6, 1, NULL, 0), + SND_SOC_DAPM_PGA("Right Capture PGA", + NAU8822_REG_RIGHT_INP_PGA_CONTROL, 6, 1, NULL, 0), + + SND_SOC_DAPM_PGA("Left Headphone Out", + NAU8822_REG_POWER_MANAGEMENT_2, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Headphone Out", + NAU8822_REG_POWER_MANAGEMENT_2, 8, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Left Speaker Out", + NAU8822_REG_POWER_MANAGEMENT_3, 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Speaker Out", + NAU8822_REG_POWER_MANAGEMENT_3, 5, 0, NULL, 0), + + SND_SOC_DAPM_PGA("AUX1 Out", + NAU8822_REG_POWER_MANAGEMENT_3, 8, 0, NULL, 0), + SND_SOC_DAPM_PGA("AUX2 Out", + NAU8822_REG_POWER_MANAGEMENT_3, 7, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("Mic Bias", + NAU8822_REG_POWER_MANAGEMENT_1, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL", + NAU8822_REG_POWER_MANAGEMENT_1, 5, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("Digital Loopback", SND_SOC_NOPM, 0, 0, + &nau8822_loopback), + + SND_SOC_DAPM_INPUT("LMICN"), + SND_SOC_DAPM_INPUT("LMICP"), + SND_SOC_DAPM_INPUT("RMICN"), + SND_SOC_DAPM_INPUT("RMICP"), + SND_SOC_DAPM_INPUT("LAUX"), + SND_SOC_DAPM_INPUT("RAUX"), + SND_SOC_DAPM_INPUT("L2"), + SND_SOC_DAPM_INPUT("R2"), + SND_SOC_DAPM_OUTPUT("LHP"), + SND_SOC_DAPM_OUTPUT("RHP"), + SND_SOC_DAPM_OUTPUT("LSPK"), + SND_SOC_DAPM_OUTPUT("RSPK"), + SND_SOC_DAPM_OUTPUT("AUXOUT1"), + SND_SOC_DAPM_OUTPUT("AUXOUT2"), +}; + +static const struct snd_soc_dapm_route nau8822_dapm_routes[] = { + {"Right DAC", NULL, "PLL", check_mclk_select_pll}, + {"Left DAC", NULL, "PLL", check_mclk_select_pll}, + + /* LMAIN and RMAIN Mixer */ + {"Right Output Mixer", "LDAC Switch", "Left DAC"}, + {"Right Output Mixer", "RDAC Switch", "Right DAC"}, + {"Right Output Mixer", "RAUX Switch", "RAUX"}, + {"Right Output Mixer", "RINMIX Switch", "Right Boost Mixer"}, + + {"Left Output Mixer", "LDAC Switch", "Left DAC"}, + {"Left Output Mixer", "RDAC Switch", "Right DAC"}, + {"Left Output Mixer", "LAUX Switch", "LAUX"}, + {"Left Output Mixer", "LINMIX Switch", "Left Boost Mixer"}, + + /* AUX1 and AUX2 Mixer */ + {"AUX1 Output Mixer", "RDAC Switch", "Right DAC"}, + {"AUX1 Output Mixer", "RMIX Switch", "Right Output Mixer"}, + {"AUX1 Output Mixer", "RINMIX Switch", "Right Boost Mixer"}, + {"AUX1 Output Mixer", "LDAC Switch", "Left DAC"}, + {"AUX1 Output Mixer", "LMIX Switch", "Left Output Mixer"}, + + {"AUX2 Output Mixer", "LDAC Switch", "Left DAC"}, + {"AUX2 Output Mixer", "LMIX Switch", "Left Output Mixer"}, + {"AUX2 Output Mixer", "LINMIX Switch", "Left Boost Mixer"}, + {"AUX2 Output Mixer", "AUX1MIX Output Switch", "AUX1 Output Mixer"}, + + /* Outputs */ + {"Right Headphone Out", NULL, "Right Output Mixer"}, + {"RHP", NULL, "Right Headphone Out"}, + + {"Left Headphone Out", NULL, "Left Output Mixer"}, + {"LHP", NULL, "Left Headphone Out"}, + + {"Right Speaker Out", NULL, "Right Output Mixer"}, + {"RSPK", NULL, "Right Speaker Out"}, + + {"Left Speaker Out", NULL, "Left Output Mixer"}, + {"LSPK", NULL, "Left Speaker Out"}, + + {"AUX1 Out", NULL, "AUX1 Output Mixer"}, + {"AUX2 Out", NULL, "AUX2 Output Mixer"}, + {"AUXOUT1", NULL, "AUX1 Out"}, + {"AUXOUT2", NULL, "AUX2 Out"}, + + /* Boost Mixer */ + {"Right ADC", NULL, "PLL", check_mclk_select_pll}, + {"Left ADC", NULL, "PLL", check_mclk_select_pll}, + + {"Right ADC", NULL, "Right Boost Mixer"}, + + {"Right Boost Mixer", NULL, "RAUX"}, + {"Right Boost Mixer", NULL, "Right Capture PGA"}, + {"Right Boost Mixer", NULL, "R2"}, + + {"Left ADC", NULL, "Left Boost Mixer"}, + + {"Left Boost Mixer", NULL, "LAUX"}, + {"Left Boost Mixer", NULL, "Left Capture PGA"}, + {"Left Boost Mixer", NULL, "L2"}, + + /* Input PGA */ + {"Right Capture PGA", NULL, "Right Input Mixer"}, + {"Left Capture PGA", NULL, "Left Input Mixer"}, + + /* Enable Microphone Power */ + {"Right Capture PGA", NULL, "Mic Bias"}, + {"Left Capture PGA", NULL, "Mic Bias"}, + + {"Right Input Mixer", "R2 Switch", "R2"}, + {"Right Input Mixer", "MicN Switch", "RMICN"}, + {"Right Input Mixer", "MicP Switch", "RMICP"}, + + {"Left Input Mixer", "L2 Switch", "L2"}, + {"Left Input Mixer", "MicN Switch", "LMICN"}, + {"Left Input Mixer", "MicP Switch", "LMICP"}, + + /* Digital Loopback */ + {"Digital Loopback", "Switch", "Left ADC"}, + {"Digital Loopback", "Switch", "Right ADC"}, + {"Left DAC", NULL, "Digital Loopback"}, + {"Right DAC", NULL, "Digital Loopback"}, +}; + +static int nau8822_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_component *component = dai->component; + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + + nau8822->div_id = clk_id; + nau8822->sysclk = freq; + dev_dbg(component->dev, "master sysclk %dHz, source %s\n", freq, + clk_id == NAU8822_CLK_PLL ? "PLL" : "MCLK"); + + return 0; +} + +static int nau8822_calc_pll(unsigned int pll_in, unsigned int fs, + struct nau8822_pll *pll_param) +{ + u64 f2, f2_max, pll_ratio; + int i, scal_sel; + + if (pll_in > NAU_PLL_REF_MAX || pll_in < NAU_PLL_REF_MIN) + return -EINVAL; + f2_max = 0; + scal_sel = ARRAY_SIZE(nau8822_mclk_scaler); + + for (i = 0; i < scal_sel; i++) { + f2 = 256 * fs * 4 * nau8822_mclk_scaler[i] / 10; + if (f2 > NAU_PLL_FREQ_MIN && f2 < NAU_PLL_FREQ_MAX && + f2_max < f2) { + f2_max = f2; + scal_sel = i; + } + } + + if (ARRAY_SIZE(nau8822_mclk_scaler) == scal_sel) + return -EINVAL; + pll_param->mclk_scaler = scal_sel; + f2 = f2_max; + + /* Calculate the PLL 4-bit integer input and the PLL 24-bit fractional + * input; round up the 24+4bit. + */ + pll_ratio = div_u64(f2 << 28, pll_in); + pll_param->pre_factor = 0; + if (((pll_ratio >> 28) & 0xF) < NAU_PLL_OPTOP_MIN) { + pll_ratio <<= 1; + pll_param->pre_factor = 1; + } + pll_param->pll_int = (pll_ratio >> 28) & 0xF; + pll_param->pll_frac = ((pll_ratio & 0xFFFFFFF) >> 4); + + return 0; +} + +static int nau8822_config_clkdiv(struct snd_soc_dai *dai, int div, int rate) +{ + struct snd_soc_component *component = dai->component; + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + struct nau8822_pll *pll = &nau8822->pll; + int i, sclk, imclk; + + switch (nau8822->div_id) { + case NAU8822_CLK_MCLK: + /* Configure the master clock prescaler div to make system + * clock to approximate the internal master clock (IMCLK); + * and large or equal to IMCLK. + */ + div = 0; + imclk = rate * 256; + for (i = 1; i < ARRAY_SIZE(nau8822_mclk_scaler); i++) { + sclk = (nau8822->sysclk * 10) / nau8822_mclk_scaler[i]; + if (sclk < imclk) + break; + div = i; + } + dev_dbg(component->dev, "master clock prescaler %x for fs %d\n", + div, rate); + + /* master clock from MCLK and disable PLL */ + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_MCLKSEL_MASK, + (div << NAU8822_MCLKSEL_SFT)); + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, + NAU8822_CLKM_MCLK); + break; + + case NAU8822_CLK_PLL: + /* master clock from PLL and enable PLL */ + if (pll->mclk_scaler != div) { + dev_err(component->dev, + "master clock prescaler not meet PLL parameters\n"); + return -EINVAL; + } + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_MCLKSEL_MASK, + (div << NAU8822_MCLKSEL_SFT)); + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, + NAU8822_CLKM_PLL); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_component *component = dai->component; + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + struct nau8822_pll *pll_param = &nau8822->pll; + int ret, fs; + + fs = freq_out / 256; + + ret = nau8822_calc_pll(freq_in, fs, pll_param); + if (ret < 0) { + dev_err(component->dev, "Unsupported input clock %d\n", + freq_in); + return ret; + } + + dev_info(component->dev, + "pll_int=%x pll_frac=%x mclk_scaler=%x pre_factor=%x\n", + pll_param->pll_int, pll_param->pll_frac, + pll_param->mclk_scaler, pll_param->pre_factor); + + snd_soc_component_update_bits(component, + NAU8822_REG_PLL_N, NAU8822_PLLMCLK_DIV2 | NAU8822_PLLN_MASK, + (pll_param->pre_factor ? NAU8822_PLLMCLK_DIV2 : 0) | + pll_param->pll_int); + snd_soc_component_write(component, + NAU8822_REG_PLL_K1, (pll_param->pll_frac >> NAU8822_PLLK1_SFT) & + NAU8822_PLLK1_MASK); + snd_soc_component_write(component, + NAU8822_REG_PLL_K2, (pll_param->pll_frac >> NAU8822_PLLK2_SFT) & + NAU8822_PLLK2_MASK); + snd_soc_component_write(component, + NAU8822_REG_PLL_K3, pll_param->pll_frac & NAU8822_PLLK3_MASK); + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_MCLKSEL_MASK, + pll_param->mclk_scaler << NAU8822_MCLKSEL_SFT); + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, NAU8822_CLKM_PLL); + + return 0; +} + +static int nau8822_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + u16 ctrl1_val = 0, ctrl2_val = 0; + + dev_dbg(component->dev, "%s\n", __func__); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl2_val |= 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ctrl2_val &= ~1; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1_val |= 0x10; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1_val |= 0x8; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1_val |= 0x18; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + ctrl1_val |= 0x180; + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1_val |= 0x100; + break; + case SND_SOC_DAIFMT_NB_IF: + ctrl1_val |= 0x80; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, + NAU8822_REG_AUDIO_INTERFACE, + NAU8822_AIFMT_MASK | NAU8822_LRP_MASK | NAU8822_BCLKP_MASK, + ctrl1_val); + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_CLKIOEN_MASK, ctrl2_val); + + return 0; +} + +static int nau8822_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + int val_len = 0, val_rate = 0; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val_len |= NAU8822_WLEN_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val_len |= NAU8822_WLEN_24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + val_len |= NAU8822_WLEN_32; + break; + default: + return -EINVAL; + } + + switch (params_rate(params)) { + case 8000: + val_rate |= NAU8822_SMPLR_8K; + break; + case 11025: + val_rate |= NAU8822_SMPLR_12K; + break; + case 16000: + val_rate |= NAU8822_SMPLR_16K; + break; + case 22050: + val_rate |= NAU8822_SMPLR_24K; + break; + case 32000: + val_rate |= NAU8822_SMPLR_32K; + break; + case 44100: + case 48000: + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, + NAU8822_REG_AUDIO_INTERFACE, NAU8822_WLEN_MASK, val_len); + snd_soc_component_update_bits(component, + NAU8822_REG_ADDITIONAL_CONTROL, NAU8822_SMPLR_MASK, val_rate); + + /* If the master clock is from MCLK, provide the runtime FS for driver + * to get the master clock prescaler configuration. + */ + if (nau8822->div_id == NAU8822_CLK_MCLK) + nau8822_config_clkdiv(dai, 0, params_rate(params)); + + return 0; +} + +static int nau8822_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_component *component = dai->component; + + dev_dbg(component->dev, "%s: %d\n", __func__, mute); + + if (mute) + snd_soc_component_update_bits(component, + NAU8822_REG_DAC_CONTROL, 0x40, 0x40); + else + snd_soc_component_update_bits(component, + NAU8822_REG_DAC_CONTROL, 0x40, 0); + + return 0; +} + +static int nau8822_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, + NAU8822_REFIMP_MASK, NAU8822_REFIMP_80K); + break; + + case SND_SOC_BIAS_STANDBY: + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, + NAU8822_IOBUF_EN | NAU8822_ABIAS_EN, + NAU8822_IOBUF_EN | NAU8822_ABIAS_EN); + + if (snd_soc_component_get_bias_level(component) == + SND_SOC_BIAS_OFF) { + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, + NAU8822_REFIMP_MASK, NAU8822_REFIMP_3K); + mdelay(100); + } + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, + NAU8822_REFIMP_MASK, NAU8822_REFIMP_300K); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_component_write(component, + NAU8822_REG_POWER_MANAGEMENT_1, 0); + snd_soc_component_write(component, + NAU8822_REG_POWER_MANAGEMENT_2, 0); + snd_soc_component_write(component, + NAU8822_REG_POWER_MANAGEMENT_3, 0); + break; + } + + dev_dbg(component->dev, "%s: %d\n", __func__, level); + + return 0; +} + +#define NAU8822_RATES (SNDRV_PCM_RATE_8000_48000) + +#define NAU8822_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static const struct snd_soc_dai_ops nau8822_dai_ops = { + .hw_params = nau8822_hw_params, + .digital_mute = nau8822_mute, + .set_fmt = nau8822_set_dai_fmt, + .set_sysclk = nau8822_set_dai_sysclk, + .set_pll = nau8822_set_pll, +}; + +static struct snd_soc_dai_driver nau8822_dai = { + .name = "nau8822-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8822_RATES, + .formats = NAU8822_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8822_RATES, + .formats = NAU8822_FORMATS, + }, + .ops = &nau8822_dai_ops, + .symmetric_rates = 1, +}; + +static int nau8822_suspend(struct snd_soc_component *component) +{ + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + + snd_soc_component_force_bias_level(component, SND_SOC_BIAS_OFF); + + regcache_mark_dirty(nau8822->regmap); + + return 0; +} + +static int nau8822_resume(struct snd_soc_component *component) +{ + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + + regcache_sync(nau8822->regmap); + + snd_soc_component_force_bias_level(component, SND_SOC_BIAS_STANDBY); + + return 0; +} + +/* + * These registers contain an "update" bit - bit 8. This means, for example, + * that one can write new DAC digital volume for both channels, but only when + * the update bit is set, will also the volume be updated - simultaneously for + * both channels. + */ +static const int update_reg[] = { + NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME, + NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME, + NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME, + NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME, + NAU8822_REG_LEFT_INP_PGA_CONTROL, + NAU8822_REG_RIGHT_INP_PGA_CONTROL, + NAU8822_REG_LHP_VOLUME, + NAU8822_REG_RHP_VOLUME, + NAU8822_REG_LSPKOUT_VOLUME, + NAU8822_REG_RSPKOUT_VOLUME, +}; + +static int nau8822_probe(struct snd_soc_component *component) +{ + int i; + + /* + * Set the update bit in all registers, that have one. This way all + * writes to those registers will also cause the update bit to be + * written. + */ + for (i = 0; i < ARRAY_SIZE(update_reg); i++) + snd_soc_component_update_bits(component, + update_reg[i], 0x100, 0x100); + + return 0; +} + +static const struct snd_soc_component_driver soc_component_dev_nau8822 = { + .probe = nau8822_probe, + .suspend = nau8822_suspend, + .resume = nau8822_resume, + .set_bias_level = nau8822_set_bias_level, + .controls = nau8822_snd_controls, + .num_controls = ARRAY_SIZE(nau8822_snd_controls), + .dapm_widgets = nau8822_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8822_dapm_widgets), + .dapm_routes = nau8822_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8822_dapm_routes), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct regmap_config nau8822_regmap_config = { + .reg_bits = 7, + .val_bits = 9, + + .max_register = NAU8822_REG_MAX_REGISTER, + .volatile_reg = nau8822_volatile, + + .readable_reg = nau8822_readable_reg, + .writeable_reg = nau8822_writeable_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8822_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(nau8822_reg_defaults), +}; + +static int nau8822_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct nau8822 *nau8822 = dev_get_platdata(dev); + int ret; + + if (!nau8822) { + nau8822 = devm_kzalloc(dev, sizeof(*nau8822), GFP_KERNEL); + if (nau8822 == NULL) + return -ENOMEM; + } + i2c_set_clientdata(i2c, nau8822); + + nau8822->regmap = devm_regmap_init_i2c(i2c, &nau8822_regmap_config); + if (IS_ERR(nau8822->regmap)) { + ret = PTR_ERR(nau8822->regmap); + dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); + return ret; + } + nau8822->dev = dev; + + /* Reset the codec */ + ret = regmap_write(nau8822->regmap, NAU8822_REG_RESET, 0x00); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret); + return ret; + } + + ret = devm_snd_soc_register_component(dev, &soc_component_dev_nau8822, + &nau8822_dai, 1); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); + return ret; + } + + return 0; +} + +static const struct i2c_device_id nau8822_i2c_id[] = { + { "nau8822", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, nau8822_i2c_id); + +#ifdef CONFIG_OF +static const struct of_device_id nau8822_of_match[] = { + { .compatible = "nuvoton,nau8822", }, + { } +}; +MODULE_DEVICE_TABLE(of, nau8822_of_match); +#endif + +static struct i2c_driver nau8822_i2c_driver = { + .driver = { + .name = "nau8822", + .of_match_table = of_match_ptr(nau8822_of_match), + }, + .probe = nau8822_i2c_probe, + .id_table = nau8822_i2c_id, +}; +module_i2c_driver(nau8822_i2c_driver); + +MODULE_DESCRIPTION("ASoC NAU8822 codec driver"); +MODULE_AUTHOR("David Lin <ctlin0@nuvoton.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h new file mode 100644 index 000000000000..9c552983a293 --- /dev/null +++ b/sound/soc/codecs/nau8822.h @@ -0,0 +1,203 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * nau8822.h -- NAU8822 ALSA SoC Audio driver + * + * Copyright 2017 Nuvoton Technology Crop. + * + * Author: David Lin <ctlin0@nuvoton.com> + * Co-author: John Hsu <kchsu0@nuvoton.com> + * Co-author: Seven Li <wtli@nuvoton.com> + */ + +#ifndef __NAU8822_H__ +#define __NAU8822_H__ + +#define NAU8822_REG_RESET 0x00 +#define NAU8822_REG_POWER_MANAGEMENT_1 0x01 +#define NAU8822_REG_POWER_MANAGEMENT_2 0x02 +#define NAU8822_REG_POWER_MANAGEMENT_3 0x03 +#define NAU8822_REG_AUDIO_INTERFACE 0x04 +#define NAU8822_REG_COMPANDING_CONTROL 0x05 +#define NAU8822_REG_CLOCKING 0x06 +#define NAU8822_REG_ADDITIONAL_CONTROL 0x07 +#define NAU8822_REG_GPIO_CONTROL 0x08 +#define NAU8822_REG_JACK_DETECT_CONTROL_1 0x09 +#define NAU8822_REG_DAC_CONTROL 0x0A +#define NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME 0x0B +#define NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME 0x0C +#define NAU8822_REG_JACK_DETECT_CONTROL_2 0x0D +#define NAU8822_REG_ADC_CONTROL 0x0E +#define NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME 0x0F +#define NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME 0x10 +#define NAU8822_REG_EQ1 0x12 +#define NAU8822_REG_EQ2 0x13 +#define NAU8822_REG_EQ3 0x14 +#define NAU8822_REG_EQ4 0x15 +#define NAU8822_REG_EQ5 0x16 +#define NAU8822_REG_DAC_LIMITER_1 0x18 +#define NAU8822_REG_DAC_LIMITER_2 0x19 +#define NAU8822_REG_NOTCH_FILTER_1 0x1B +#define NAU8822_REG_NOTCH_FILTER_2 0x1C +#define NAU8822_REG_NOTCH_FILTER_3 0x1D +#define NAU8822_REG_NOTCH_FILTER_4 0x1E +#define NAU8822_REG_ALC_CONTROL_1 0x20 +#define NAU8822_REG_ALC_CONTROL_2 0x21 +#define NAU8822_REG_ALC_CONTROL_3 0x22 +#define NAU8822_REG_NOISE_GATE 0x23 +#define NAU8822_REG_PLL_N 0x24 +#define NAU8822_REG_PLL_K1 0x25 +#define NAU8822_REG_PLL_K2 0x26 +#define NAU8822_REG_PLL_K3 0x27 +#define NAU8822_REG_3D_CONTROL 0x29 +#define NAU8822_REG_RIGHT_SPEAKER_CONTROL 0x2B +#define NAU8822_REG_INPUT_CONTROL 0x2C +#define NAU8822_REG_LEFT_INP_PGA_CONTROL 0x2D +#define NAU8822_REG_RIGHT_INP_PGA_CONTROL 0x2E +#define NAU8822_REG_LEFT_ADC_BOOST_CONTROL 0x2F +#define NAU8822_REG_RIGHT_ADC_BOOST_CONTROL 0x30 +#define NAU8822_REG_OUTPUT_CONTROL 0x31 +#define NAU8822_REG_LEFT_MIXER_CONTROL 0x32 +#define NAU8822_REG_RIGHT_MIXER_CONTROL 0x33 +#define NAU8822_REG_LHP_VOLUME 0x34 +#define NAU8822_REG_RHP_VOLUME 0x35 +#define NAU8822_REG_LSPKOUT_VOLUME 0x36 +#define NAU8822_REG_RSPKOUT_VOLUME 0x37 +#define NAU8822_REG_AUX2_MIXER 0x38 +#define NAU8822_REG_AUX1_MIXER 0x39 +#define NAU8822_REG_POWER_MANAGEMENT_4 0x3A +#define NAU8822_REG_LEFT_TIME_SLOT 0x3B +#define NAU8822_REG_MISC 0x3C +#define NAU8822_REG_RIGHT_TIME_SLOT 0x3D +#define NAU8822_REG_DEVICE_REVISION 0x3E +#define NAU8822_REG_DEVICE_ID 0x3F +#define NAU8822_REG_DAC_DITHER 0x41 +#define NAU8822_REG_ALC_ENHANCE_1 0x46 +#define NAU8822_REG_ALC_ENHANCE_2 0x47 +#define NAU8822_REG_192KHZ_SAMPLING 0x48 +#define NAU8822_REG_MISC_CONTROL 0x49 +#define NAU8822_REG_INPUT_TIEOFF 0x4A +#define NAU8822_REG_POWER_REDUCTION 0x4B +#define NAU8822_REG_AGC_PEAK2PEAK 0x4C +#define NAU8822_REG_AGC_PEAK_DETECT 0x4D +#define NAU8822_REG_AUTOMUTE_CONTROL 0x4E +#define NAU8822_REG_OUTPUT_TIEOFF 0x4F +#define NAU8822_REG_MAX_REGISTER NAU8822_REG_OUTPUT_TIEOFF + +/* NAU8822_REG_POWER_MANAGEMENT_1 (0x1) */ +#define NAU8822_REFIMP_MASK 0x3 +#define NAU8822_REFIMP_80K 0x1 +#define NAU8822_REFIMP_300K 0x2 +#define NAU8822_REFIMP_3K 0x3 +#define NAU8822_IOBUF_EN (0x1 << 2) +#define NAU8822_ABIAS_EN (0x1 << 3) + +/* NAU8822_REG_AUDIO_INTERFACE (0x4) */ +#define NAU8822_AIFMT_MASK (0x3 << 3) +#define NAU8822_WLEN_MASK (0x3 << 5) +#define NAU8822_WLEN_20 (0x1 << 5) +#define NAU8822_WLEN_24 (0x2 << 5) +#define NAU8822_WLEN_32 (0x3 << 5) +#define NAU8822_LRP_MASK (0x1 << 7) +#define NAU8822_BCLKP_MASK (0x1 << 8) + +/* NAU8822_REG_COMPANDING_CONTROL (0x5) */ +#define NAU8822_ADDAP_SFT 0 +#define NAU8822_ADCCM_SFT 1 +#define NAU8822_DACCM_SFT 3 + +/* NAU8822_REG_CLOCKING (0x6) */ +#define NAU8822_CLKIOEN_MASK 0x1 +#define NAU8822_MCLKSEL_SFT 5 +#define NAU8822_MCLKSEL_MASK (0x7 << 5) +#define NAU8822_BCLKSEL_SFT 2 +#define NAU8822_BCLKSEL_MASK (0x7 << 2) +#define NAU8822_CLKM_MASK (0x1 << 8) +#define NAU8822_CLKM_MCLK (0x0 << 8) +#define NAU8822_CLKM_PLL (0x1 << 8) + +/* NAU8822_REG_ADDITIONAL_CONTROL (0x08) */ +#define NAU8822_SMPLR_SFT 1 +#define NAU8822_SMPLR_MASK (0x7 << 1) +#define NAU8822_SMPLR_48K (0x0 << 1) +#define NAU8822_SMPLR_32K (0x1 << 1) +#define NAU8822_SMPLR_24K (0x2 << 1) +#define NAU8822_SMPLR_16K (0x3 << 1) +#define NAU8822_SMPLR_12K (0x4 << 1) +#define NAU8822_SMPLR_8K (0x5 << 1) + +/* NAU8822_REG_EQ1 (0x12) */ +#define NAU8822_EQ1GC_SFT 0 +#define NAU8822_EQ1CF_SFT 5 +#define NAU8822_EQM_SFT 8 + +/* NAU8822_REG_EQ2 (0x13) */ +#define NAU8822_EQ2GC_SFT 0 +#define NAU8822_EQ2CF_SFT 5 +#define NAU8822_EQ2BW_SFT 8 + +/* NAU8822_REG_EQ3 (0x14) */ +#define NAU8822_EQ3GC_SFT 0 +#define NAU8822_EQ3CF_SFT 5 +#define NAU8822_EQ3BW_SFT 8 + +/* NAU8822_REG_EQ4 (0x15) */ +#define NAU8822_EQ4GC_SFT 0 +#define NAU8822_EQ4CF_SFT 5 +#define NAU8822_EQ4BW_SFT 8 + +/* NAU8822_REG_EQ5 (0x16) */ +#define NAU8822_EQ5GC_SFT 0 +#define NAU8822_EQ5CF_SFT 5 + +/* NAU8822_REG_ALC_CONTROL_1 (0x20) */ +#define NAU8822_ALCMINGAIN_SFT 0 +#define NAU8822_ALCMXGAIN_SFT 3 +#define NAU8822_ALCEN_SFT 7 + +/* NAU8822_REG_ALC_CONTROL_2 (0x21) */ +#define NAU8822_ALCSL_SFT 0 +#define NAU8822_ALCHT_SFT 4 + +/* NAU8822_REG_ALC_CONTROL_3 (0x22) */ +#define NAU8822_ALCATK_SFT 0 +#define NAU8822_ALCDCY_SFT 4 +#define NAU8822_ALCM_SFT 8 + +/* NAU8822_REG_PLL_N (0x24) */ +#define NAU8822_PLLMCLK_DIV2 (0x1 << 4) +#define NAU8822_PLLN_MASK 0xF + +#define NAU8822_PLLK1_SFT 18 +#define NAU8822_PLLK1_MASK 0x3F + +/* NAU8822_REG_PLL_K2 (0x26) */ +#define NAU8822_PLLK2_SFT 9 +#define NAU8822_PLLK2_MASK 0x1FF + +/* NAU8822_REG_PLL_K3 (0x27) */ +#define NAU8822_PLLK3_MASK 0x1FF + +/* System Clock Source */ +enum { + NAU8822_CLK_MCLK, + NAU8822_CLK_PLL, +}; + +struct nau8822_pll { + int pre_factor; + int mclk_scaler; + int pll_frac; + int pll_int; +}; + +/* Codec Private Data */ +struct nau8822 { + struct device *dev; + struct regmap *regmap; + int mclk_idx; + struct nau8822_pll pll; + int sysclk; + int div_id; +}; + +#endif /* __NAU8822_H__ */ diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index b9fed99d8b5e..7bbcbf5f05c8 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -424,10 +424,8 @@ static u32 nau8825_xtalk_sidetone(u32 sig_org, u32 sig_cros) { u32 gain, sidetone; - if (unlikely(sig_org == 0) || unlikely(sig_cros == 0)) { - WARN_ON(1); + if (WARN_ON(sig_org == 0 || sig_cros == 0)) return 0; - } sig_org = nau8825_intlog10_dec3(sig_org); sig_cros = nau8825_intlog10_dec3(sig_cros); diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c index 690c26e7389e..809b7e9f03ca 100644 --- a/sound/soc/codecs/pcm186x.c +++ b/sound/soc/codecs/pcm186x.c @@ -401,7 +401,8 @@ static int pcm186x_set_fmt(struct snd_soc_dai *dai, unsigned int format) break; case SND_SOC_DAIFMT_DSP_A: priv->tdm_offset += 1; - /* Fall through... DSP_A uses the same basic config as DSP_B + /* fall through */ + /* DSP_A uses the same basic config as DSP_B * except we need to shift the TDM output by one BCK cycle */ case SND_SOC_DAIFMT_DSP_B: diff --git a/sound/soc/codecs/pcm186x.h b/sound/soc/codecs/pcm186x.h index 2c6ba55bf394..bb3f0c42a1cd 100644 --- a/sound/soc/codecs/pcm186x.h +++ b/sound/soc/codecs/pcm186x.h @@ -139,7 +139,7 @@ enum pcm186x_type { #define PCM186X_MAX_REGISTER PCM186X_CURR_TRIM_CTRL /* PCM186X_PAGE */ -#define PCM186X_RESET 0xff +#define PCM186X_RESET 0xfe /* PCM186X_ADCX_INPUT_SEL_X */ #define PCM186X_ADC_INPUT_SEL_POL BIT(7) diff --git a/sound/soc/codecs/pcm3060-i2c.c b/sound/soc/codecs/pcm3060-i2c.c new file mode 100644 index 000000000000..cdc8314882bc --- /dev/null +++ b/sound/soc/codecs/pcm3060-i2c.c @@ -0,0 +1,60 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// PCM3060 I2C driver +// +// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech> + +#include <linux/i2c.h> +#include <linux/module.h> +#include <sound/soc.h> + +#include "pcm3060.h" + +static int pcm3060_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct pcm3060_priv *priv; + + priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + i2c_set_clientdata(i2c, priv); + + priv->regmap = devm_regmap_init_i2c(i2c, &pcm3060_regmap); + if (IS_ERR(priv->regmap)) + return PTR_ERR(priv->regmap); + + return pcm3060_probe(&i2c->dev); +} + +static const struct i2c_device_id pcm3060_i2c_id[] = { + { .name = "pcm3060" }, + { }, +}; +MODULE_DEVICE_TABLE(i2c, pcm3060_i2c_id); + +#ifdef CONFIG_OF +static const struct of_device_id pcm3060_of_match[] = { + { .compatible = "ti,pcm3060" }, + { }, +}; +MODULE_DEVICE_TABLE(of, pcm3060_of_match); +#endif /* CONFIG_OF */ + +static struct i2c_driver pcm3060_i2c_driver = { + .driver = { + .name = "pcm3060", +#ifdef CONFIG_OF + .of_match_table = pcm3060_of_match, +#endif /* CONFIG_OF */ + }, + .id_table = pcm3060_i2c_id, + .probe = pcm3060_i2c_probe, +}; + +module_i2c_driver(pcm3060_i2c_driver); + +MODULE_DESCRIPTION("PCM3060 I2C driver"); +MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.tech>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/pcm3060-spi.c b/sound/soc/codecs/pcm3060-spi.c new file mode 100644 index 000000000000..f6f19fa80932 --- /dev/null +++ b/sound/soc/codecs/pcm3060-spi.c @@ -0,0 +1,59 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// PCM3060 SPI driver +// +// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech> + +#include <linux/module.h> +#include <linux/spi/spi.h> +#include <sound/soc.h> + +#include "pcm3060.h" + +static int pcm3060_spi_probe(struct spi_device *spi) +{ + struct pcm3060_priv *priv; + + priv = devm_kzalloc(&spi->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + spi_set_drvdata(spi, priv); + + priv->regmap = devm_regmap_init_spi(spi, &pcm3060_regmap); + if (IS_ERR(priv->regmap)) + return PTR_ERR(priv->regmap); + + return pcm3060_probe(&spi->dev); +} + +static const struct spi_device_id pcm3060_spi_id[] = { + { .name = "pcm3060" }, + { }, +}; +MODULE_DEVICE_TABLE(spi, pcm3060_spi_id); + +#ifdef CONFIG_OF +static const struct of_device_id pcm3060_of_match[] = { + { .compatible = "ti,pcm3060" }, + { }, +}; +MODULE_DEVICE_TABLE(of, pcm3060_of_match); +#endif /* CONFIG_OF */ + +static struct spi_driver pcm3060_spi_driver = { + .driver = { + .name = "pcm3060", +#ifdef CONFIG_OF + .of_match_table = pcm3060_of_match, +#endif /* CONFIG_OF */ + }, + .id_table = pcm3060_spi_id, + .probe = pcm3060_spi_probe, +}; + +module_spi_driver(pcm3060_spi_driver); + +MODULE_DESCRIPTION("PCM3060 SPI driver"); +MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.tech>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c new file mode 100644 index 000000000000..6714aa8d9026 --- /dev/null +++ b/sound/soc/codecs/pcm3060.c @@ -0,0 +1,311 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// PCM3060 codec driver +// +// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech> + +#include <linux/module.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "pcm3060.h" + +/* dai */ + +static int pcm3060_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_component *comp = dai->component; + struct pcm3060_priv *priv = snd_soc_component_get_drvdata(comp); + + if (dir != SND_SOC_CLOCK_IN) { + dev_err(comp->dev, "unsupported sysclock dir: %d\n", dir); + return -EINVAL; + } + + priv->dai[dai->id].sclk_freq = freq; + + return 0; +} + +static int pcm3060_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *comp = dai->component; + struct pcm3060_priv *priv = snd_soc_component_get_drvdata(comp); + unsigned int reg; + unsigned int val; + + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) { + dev_err(comp->dev, "unsupported DAI polarity: 0x%x\n", fmt); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + priv->dai[dai->id].is_master = true; + break; + case SND_SOC_DAIFMT_CBS_CFS: + priv->dai[dai->id].is_master = false; + break; + default: + dev_err(comp->dev, "unsupported DAI master mode: 0x%x\n", fmt); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + val = PCM3060_REG_FMT_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val = PCM3060_REG_FMT_RJ; + break; + case SND_SOC_DAIFMT_LEFT_J: + val = PCM3060_REG_FMT_LJ; + break; + default: + dev_err(comp->dev, "unsupported DAI format: 0x%x\n", fmt); + return -EINVAL; + } + + if (dai->id == PCM3060_DAI_ID_DAC) + reg = PCM3060_REG67; + else + reg = PCM3060_REG72; + + regmap_update_bits(priv->regmap, reg, PCM3060_REG_MASK_FMT, val); + + return 0; +} + +static int pcm3060_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *comp = dai->component; + struct pcm3060_priv *priv = snd_soc_component_get_drvdata(comp); + unsigned int rate; + unsigned int ratio; + unsigned int reg; + unsigned int val; + + if (!priv->dai[dai->id].is_master) { + val = PCM3060_REG_MS_S; + goto val_ready; + } + + rate = params_rate(params); + if (!rate) { + dev_err(comp->dev, "rate is not configured\n"); + return -EINVAL; + } + + ratio = priv->dai[dai->id].sclk_freq / rate; + + switch (ratio) { + case 768: + val = PCM3060_REG_MS_M768; + break; + case 512: + val = PCM3060_REG_MS_M512; + break; + case 384: + val = PCM3060_REG_MS_M384; + break; + case 256: + val = PCM3060_REG_MS_M256; + break; + case 192: + val = PCM3060_REG_MS_M192; + break; + case 128: + val = PCM3060_REG_MS_M128; + break; + default: + dev_err(comp->dev, "unsupported ratio: %d\n", ratio); + return -EINVAL; + } + +val_ready: + if (dai->id == PCM3060_DAI_ID_DAC) + reg = PCM3060_REG67; + else + reg = PCM3060_REG72; + + regmap_update_bits(priv->regmap, reg, PCM3060_REG_MASK_MS, val); + + return 0; +} + +static const struct snd_soc_dai_ops pcm3060_dai_ops = { + .set_sysclk = pcm3060_set_sysclk, + .set_fmt = pcm3060_set_fmt, + .hw_params = pcm3060_hw_params, +}; + +#define PCM3060_DAI_RATES_ADC (SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + +#define PCM3060_DAI_RATES_DAC (PCM3060_DAI_RATES_ADC | \ + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000) + +static struct snd_soc_dai_driver pcm3060_dai[] = { + { + .name = "pcm3060-dac", + .id = PCM3060_DAI_ID_DAC, + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = PCM3060_DAI_RATES_DAC, + .formats = SNDRV_PCM_FMTBIT_S24_LE, + }, + .ops = &pcm3060_dai_ops, + }, + { + .name = "pcm3060-adc", + .id = PCM3060_DAI_ID_ADC, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = PCM3060_DAI_RATES_ADC, + .formats = SNDRV_PCM_FMTBIT_S24_LE, + }, + .ops = &pcm3060_dai_ops, + }, +}; + +/* dapm */ + +static DECLARE_TLV_DB_SCALE(pcm3060_dapm_tlv, -10050, 50, 1); + +static const struct snd_kcontrol_new pcm3060_dapm_controls[] = { + SOC_DOUBLE_R_RANGE_TLV("Master Playback Volume", + PCM3060_REG65, PCM3060_REG66, 0, + PCM3060_REG_AT2_MIN, PCM3060_REG_AT2_MAX, + 0, pcm3060_dapm_tlv), + SOC_DOUBLE("Master Playback Switch", PCM3060_REG68, + PCM3060_REG_SHIFT_MUT21, PCM3060_REG_SHIFT_MUT22, 1, 1), + + SOC_DOUBLE_R_RANGE_TLV("Master Capture Volume", + PCM3060_REG70, PCM3060_REG71, 0, + PCM3060_REG_AT1_MIN, PCM3060_REG_AT1_MAX, + 0, pcm3060_dapm_tlv), + SOC_DOUBLE("Master Capture Switch", PCM3060_REG73, + PCM3060_REG_SHIFT_MUT11, PCM3060_REG_SHIFT_MUT12, 1, 1), +}; + +static const struct snd_soc_dapm_widget pcm3060_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", PCM3060_REG64, + PCM3060_REG_SHIFT_DAPSV, 1), + + SND_SOC_DAPM_OUTPUT("OUTL"), + SND_SOC_DAPM_OUTPUT("OUTR"), + + SND_SOC_DAPM_INPUT("INL"), + SND_SOC_DAPM_INPUT("INR"), + + SND_SOC_DAPM_ADC("ADC", "Capture", PCM3060_REG64, + PCM3060_REG_SHIFT_ADPSV, 1), +}; + +static const struct snd_soc_dapm_route pcm3060_dapm_map[] = { + { "OUTL", NULL, "DAC" }, + { "OUTR", NULL, "DAC" }, + + { "ADC", NULL, "INL" }, + { "ADC", NULL, "INR" }, +}; + +/* soc component */ + +static const struct snd_soc_component_driver pcm3060_soc_comp_driver = { + .controls = pcm3060_dapm_controls, + .num_controls = ARRAY_SIZE(pcm3060_dapm_controls), + .dapm_widgets = pcm3060_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(pcm3060_dapm_widgets), + .dapm_routes = pcm3060_dapm_map, + .num_dapm_routes = ARRAY_SIZE(pcm3060_dapm_map), +}; + +/* regmap */ + +static bool pcm3060_reg_writeable(struct device *dev, unsigned int reg) +{ + return (reg >= PCM3060_REG64); +} + +static bool pcm3060_reg_readable(struct device *dev, unsigned int reg) +{ + return (reg >= PCM3060_REG64); +} + +static bool pcm3060_reg_volatile(struct device *dev, unsigned int reg) +{ + /* PCM3060_REG64 is volatile */ + return (reg == PCM3060_REG64); +} + +static const struct reg_default pcm3060_reg_defaults[] = { + { PCM3060_REG64, 0xF0 }, + { PCM3060_REG65, 0xFF }, + { PCM3060_REG66, 0xFF }, + { PCM3060_REG67, 0x00 }, + { PCM3060_REG68, 0x00 }, + { PCM3060_REG69, 0x00 }, + { PCM3060_REG70, 0xD7 }, + { PCM3060_REG71, 0xD7 }, + { PCM3060_REG72, 0x00 }, + { PCM3060_REG73, 0x00 }, +}; + +const struct regmap_config pcm3060_regmap = { + .reg_bits = 8, + .val_bits = 8, + .writeable_reg = pcm3060_reg_writeable, + .readable_reg = pcm3060_reg_readable, + .volatile_reg = pcm3060_reg_volatile, + .max_register = PCM3060_REG73, + .reg_defaults = pcm3060_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(pcm3060_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; +EXPORT_SYMBOL(pcm3060_regmap); + +/* device */ + +static void pcm3060_parse_dt(const struct device_node *np, + struct pcm3060_priv *priv) +{ + priv->out_se = of_property_read_bool(np, "ti,out-single-ended"); +} + +int pcm3060_probe(struct device *dev) +{ + int rc; + struct pcm3060_priv *priv = dev_get_drvdata(dev); + + if (dev->of_node) + pcm3060_parse_dt(dev->of_node, priv); + + if (priv->out_se) + regmap_update_bits(priv->regmap, PCM3060_REG64, + PCM3060_REG_SE, PCM3060_REG_SE); + + rc = devm_snd_soc_register_component(dev, &pcm3060_soc_comp_driver, + pcm3060_dai, + ARRAY_SIZE(pcm3060_dai)); + if (rc) { + dev_err(dev, "failed to register component, rc=%d\n", rc); + return rc; + } + + return 0; +} +EXPORT_SYMBOL(pcm3060_probe); + +MODULE_DESCRIPTION("PCM3060 codec driver"); +MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.tech>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/pcm3060.h b/sound/soc/codecs/pcm3060.h new file mode 100644 index 000000000000..6a027b4a845d --- /dev/null +++ b/sound/soc/codecs/pcm3060.h @@ -0,0 +1,91 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * PCM3060 codec driver + * + * Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech> + */ + +#ifndef _SND_SOC_PCM3060_H +#define _SND_SOC_PCM3060_H + +#include <linux/device.h> +#include <linux/regmap.h> + +extern const struct regmap_config pcm3060_regmap; + +#define PCM3060_DAI_ID_DAC 0 +#define PCM3060_DAI_ID_ADC 1 +#define PCM3060_DAI_IDS_NUM 2 + +struct pcm3060_priv_dai { + bool is_master; + unsigned int sclk_freq; +}; + +struct pcm3060_priv { + struct regmap *regmap; + struct pcm3060_priv_dai dai[PCM3060_DAI_IDS_NUM]; + u8 out_se: 1; +}; + +int pcm3060_probe(struct device *dev); +int pcm3060_remove(struct device *dev); + +/* registers */ + +#define PCM3060_REG64 0x40 +#define PCM3060_REG_MRST 0x80 +#define PCM3060_REG_SRST 0x40 +#define PCM3060_REG_ADPSV 0x20 +#define PCM3060_REG_SHIFT_ADPSV 0x05 +#define PCM3060_REG_DAPSV 0x10 +#define PCM3060_REG_SHIFT_DAPSV 0x04 +#define PCM3060_REG_SE 0x01 + +#define PCM3060_REG65 0x41 +#define PCM3060_REG66 0x42 +#define PCM3060_REG_AT2_MIN 0x36 +#define PCM3060_REG_AT2_MAX 0xFF + +#define PCM3060_REG67 0x43 +#define PCM3060_REG72 0x48 +#define PCM3060_REG_CSEL 0x80 +#define PCM3060_REG_MASK_MS 0x70 +#define PCM3060_REG_MS_S 0x00 +#define PCM3060_REG_MS_M768 (0x01 << 4) +#define PCM3060_REG_MS_M512 (0x02 << 4) +#define PCM3060_REG_MS_M384 (0x03 << 4) +#define PCM3060_REG_MS_M256 (0x04 << 4) +#define PCM3060_REG_MS_M192 (0x05 << 4) +#define PCM3060_REG_MS_M128 (0x06 << 4) +#define PCM3060_REG_MASK_FMT 0x03 +#define PCM3060_REG_FMT_I2S 0x00 +#define PCM3060_REG_FMT_LJ 0x01 +#define PCM3060_REG_FMT_RJ 0x02 + +#define PCM3060_REG68 0x44 +#define PCM3060_REG_OVER 0x40 +#define PCM3060_REG_DREV2 0x04 +#define PCM3060_REG_SHIFT_MUT21 0x00 +#define PCM3060_REG_SHIFT_MUT22 0x01 + +#define PCM3060_REG69 0x45 +#define PCM3060_REG_FLT 0x80 +#define PCM3060_REG_MASK_DMF 0x60 +#define PCM3060_REG_DMC 0x10 +#define PCM3060_REG_ZREV 0x02 +#define PCM3060_REG_AZRO 0x01 + +#define PCM3060_REG70 0x46 +#define PCM3060_REG71 0x47 +#define PCM3060_REG_AT1_MIN 0x0E +#define PCM3060_REG_AT1_MAX 0xFF + +#define PCM3060_REG73 0x49 +#define PCM3060_REG_ZCDD 0x10 +#define PCM3060_REG_BYP 0x08 +#define PCM3060_REG_DREV1 0x04 +#define PCM3060_REG_SHIFT_MUT11 0x00 +#define PCM3060_REG_SHIFT_MUT12 0x01 + +#endif /* _SND_SOC_PCM3060_H */ diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 3356c91f55b0..08d3fe192e65 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -33,6 +33,8 @@ #define PCM3168A_FMT_RIGHT_J_16 0x3 #define PCM3168A_FMT_DSP_A 0x4 #define PCM3168A_FMT_DSP_B 0x5 +#define PCM3168A_FMT_I2S_TDM 0x6 +#define PCM3168A_FMT_LEFT_J_TDM 0x7 #define PCM3168A_FMT_DSP_MASK 0x4 #define PCM3168A_NUM_SUPPLIES 6 @@ -131,10 +133,6 @@ static const struct snd_kcontrol_new pcm3168a_snd_controls[] = { SOC_DOUBLE("DAC2 Invert Switch", PCM3168A_DAC_INV, 2, 3, 1, 0), SOC_DOUBLE("DAC3 Invert Switch", PCM3168A_DAC_INV, 4, 5, 1, 0), SOC_DOUBLE("DAC4 Invert Switch", PCM3168A_DAC_INV, 6, 7, 1, 0), - SOC_DOUBLE_STS("DAC1 Zero Flag", PCM3168A_DAC_ZERO, 0, 1, 1, 0), - SOC_DOUBLE_STS("DAC2 Zero Flag", PCM3168A_DAC_ZERO, 2, 3, 1, 0), - SOC_DOUBLE_STS("DAC3 Zero Flag", PCM3168A_DAC_ZERO, 4, 5, 1, 0), - SOC_DOUBLE_STS("DAC4 Zero Flag", PCM3168A_DAC_ZERO, 6, 7, 1, 0), SOC_ENUM("DAC Volume Control Type", pcm3168a_dac_volume_type), SOC_ENUM("DAC Volume Rate Multiplier", pcm3168a_dac_att_mult), SOC_ENUM("DAC De-Emphasis", pcm3168a_dac_demp), @@ -174,9 +172,6 @@ static const struct snd_kcontrol_new pcm3168a_snd_controls[] = { SOC_DOUBLE("ADC1 Mute Switch", PCM3168A_ADC_MUTE, 0, 1, 1, 0), SOC_DOUBLE("ADC2 Mute Switch", PCM3168A_ADC_MUTE, 2, 3, 1, 0), SOC_DOUBLE("ADC3 Mute Switch", PCM3168A_ADC_MUTE, 4, 5, 1, 0), - SOC_DOUBLE_STS("ADC1 Overflow Flag", PCM3168A_ADC_OV, 0, 1, 1, 0), - SOC_DOUBLE_STS("ADC2 Overflow Flag", PCM3168A_ADC_OV, 2, 3, 1, 0), - SOC_DOUBLE_STS("ADC3 Overflow Flag", PCM3168A_ADC_OV, 4, 5, 1, 0), SOC_ENUM("ADC Volume Control Type", pcm3168a_adc_volume_type), SOC_ENUM("ADC Volume Rate Multiplier", pcm3168a_adc_att_mult), SOC_ENUM("ADC Overflow Flag Polarity", pcm3168a_adc_ov_pol), @@ -401,9 +396,11 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream, bool tx, master_mode; u32 val, mask, shift, reg; unsigned int rate, fmt, ratio, max_ratio; + unsigned int chan; int i, min_frame_size; rate = params_rate(params); + chan = params_channels(params); ratio = pcm3168a->sysclk / rate; @@ -456,6 +453,21 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + /* for TDM */ + if (chan > 2) { + switch (fmt) { + case PCM3168A_FMT_I2S: + fmt = PCM3168A_FMT_I2S_TDM; + break; + case PCM3168A_FMT_LEFT_J: + fmt = PCM3168A_FMT_LEFT_J_TDM; + break; + default: + dev_err(component->dev, "TDM is supported under I2S/Left_J only\n"); + return -EINVAL; + } + } + if (master_mode) val = ((i + 1) << shift); else @@ -476,7 +488,64 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream, return 0; } +static int pcm3168a_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(component); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + unsigned int fmt; + unsigned int sample_min; + unsigned int channel_max; + unsigned int channel_maxs[] = { + 6, /* rx */ + 8 /* tx */ + }; + + if (tx) + fmt = pcm3168a->dac_fmt; + else + fmt = pcm3168a->adc_fmt; + + /* + * Available Data Bits + * + * RIGHT_J : 24 / 16 + * LEFT_J : 24 + * I2S : 24 + * + * TDM available + * + * I2S + * LEFT_J + */ + switch (fmt) { + case PCM3168A_FMT_RIGHT_J: + sample_min = 16; + channel_max = 2; + break; + case PCM3168A_FMT_LEFT_J: + case PCM3168A_FMT_I2S: + sample_min = 24; + channel_max = channel_maxs[tx]; + break; + default: + sample_min = 24; + channel_max = 2; + } + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + sample_min, 32); + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, + 2, channel_max); + + return 0; +} static const struct snd_soc_dai_ops pcm3168a_dac_dai_ops = { + .startup = pcm3168a_startup, .set_fmt = pcm3168a_set_dai_fmt_dac, .set_sysclk = pcm3168a_set_dai_sysclk, .hw_params = pcm3168a_hw_params, @@ -484,6 +553,7 @@ static const struct snd_soc_dai_ops pcm3168a_dac_dai_ops = { }; static const struct snd_soc_dai_ops pcm3168a_adc_dai_ops = { + .startup = pcm3168a_startup, .set_fmt = pcm3168a_set_dai_fmt_adc, .set_sysclk = pcm3168a_set_dai_sysclk, .hw_params = pcm3168a_hw_params @@ -688,15 +758,22 @@ err_clk: } EXPORT_SYMBOL_GPL(pcm3168a_probe); -void pcm3168a_remove(struct device *dev) +static void pcm3168a_disable(struct device *dev) { struct pcm3168a_priv *pcm3168a = dev_get_drvdata(dev); - pm_runtime_disable(dev); regulator_bulk_disable(ARRAY_SIZE(pcm3168a->supplies), - pcm3168a->supplies); + pcm3168a->supplies); clk_disable_unprepare(pcm3168a->scki); } + +void pcm3168a_remove(struct device *dev) +{ + pm_runtime_disable(dev); +#ifndef CONFIG_PM + pcm3168a_disable(dev); +#endif +} EXPORT_SYMBOL_GPL(pcm3168a_remove); #ifdef CONFIG_PM @@ -751,10 +828,7 @@ static int pcm3168a_rt_suspend(struct device *dev) regcache_cache_only(pcm3168a->regmap, true); - regulator_bulk_disable(ARRAY_SIZE(pcm3168a->supplies), - pcm3168a->supplies); - - clk_disable_unprepare(pcm3168a->scki); + pcm3168a_disable(dev); return 0; } diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index f0f2d4fd3769..4cc24a5d5c31 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -53,6 +53,8 @@ struct pcm512x_priv { unsigned long overclock_pll; unsigned long overclock_dac; unsigned long overclock_dsp; + int mute; + struct mutex mutex; }; /* @@ -384,6 +386,61 @@ static const struct soc_enum pcm512x_veds = SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_2, PCM512x_VEDS_SHIFT, 4, pcm512x_ramp_step_text); +static int pcm512x_update_mute(struct pcm512x_priv *pcm512x) +{ + return regmap_update_bits( + pcm512x->regmap, PCM512x_MUTE, PCM512x_RQML | PCM512x_RQMR, + (!!(pcm512x->mute & 0x5) << PCM512x_RQML_SHIFT) + | (!!(pcm512x->mute & 0x3) << PCM512x_RQMR_SHIFT)); +} + +static int pcm512x_digital_playback_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_component_get_drvdata(component); + + mutex_lock(&pcm512x->mutex); + ucontrol->value.integer.value[0] = !(pcm512x->mute & 0x4); + ucontrol->value.integer.value[1] = !(pcm512x->mute & 0x2); + mutex_unlock(&pcm512x->mutex); + + return 0; +} + +static int pcm512x_digital_playback_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_component_get_drvdata(component); + int ret, changed = 0; + + mutex_lock(&pcm512x->mutex); + + if ((pcm512x->mute & 0x4) == (ucontrol->value.integer.value[0] << 2)) { + pcm512x->mute ^= 0x4; + changed = 1; + } + if ((pcm512x->mute & 0x2) == (ucontrol->value.integer.value[1] << 1)) { + pcm512x->mute ^= 0x2; + changed = 1; + } + + if (changed) { + ret = pcm512x_update_mute(pcm512x); + if (ret != 0) { + dev_err(component->dev, + "Failed to update digital mute: %d\n", ret); + mutex_unlock(&pcm512x->mutex); + return ret; + } + } + + mutex_unlock(&pcm512x->mutex); + + return changed; +} + static const struct snd_kcontrol_new pcm512x_controls[] = { SOC_DOUBLE_R_TLV("Digital Playback Volume", PCM512x_DIGITAL_VOLUME_2, PCM512x_DIGITAL_VOLUME_3, 0, 255, 1, digital_tlv), @@ -391,8 +448,15 @@ SOC_DOUBLE_TLV("Analogue Playback Volume", PCM512x_ANALOG_GAIN_CTRL, PCM512x_LAGN_SHIFT, PCM512x_RAGN_SHIFT, 1, 1, analog_tlv), SOC_DOUBLE_TLV("Analogue Playback Boost Volume", PCM512x_ANALOG_GAIN_BOOST, PCM512x_AGBL_SHIFT, PCM512x_AGBR_SHIFT, 1, 0, boost_tlv), -SOC_DOUBLE("Digital Playback Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT, - PCM512x_RQMR_SHIFT, 1, 1), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Digital Playback Switch", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_stereo_info, + .get = pcm512x_digital_playback_switch_get, + .put = pcm512x_digital_playback_switch_put +}, SOC_SINGLE("Deemphasis Switch", PCM512x_DSP, PCM512x_DEMP_SHIFT, 1, 1), SOC_ENUM("DSP Program", pcm512x_dsp_program), @@ -1319,10 +1383,58 @@ static int pcm512x_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } +static int pcm512x_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_component *component = dai->component; + struct pcm512x_priv *pcm512x = snd_soc_component_get_drvdata(component); + int ret; + unsigned int mute_det; + + mutex_lock(&pcm512x->mutex); + + if (mute) { + pcm512x->mute |= 0x1; + ret = regmap_update_bits(pcm512x->regmap, PCM512x_MUTE, + PCM512x_RQML | PCM512x_RQMR, + PCM512x_RQML | PCM512x_RQMR); + if (ret != 0) { + dev_err(component->dev, + "Failed to set digital mute: %d\n", ret); + goto unlock; + } + + regmap_read_poll_timeout(pcm512x->regmap, + PCM512x_ANALOG_MUTE_DET, + mute_det, (mute_det & 0x3) == 0, + 200, 10000); + } else { + pcm512x->mute &= ~0x1; + ret = pcm512x_update_mute(pcm512x); + if (ret != 0) { + dev_err(component->dev, + "Failed to update digital mute: %d\n", ret); + goto unlock; + } + + regmap_read_poll_timeout(pcm512x->regmap, + PCM512x_ANALOG_MUTE_DET, + mute_det, + (mute_det & 0x3) + == ((~pcm512x->mute >> 1) & 0x3), + 200, 10000); + } + +unlock: + mutex_unlock(&pcm512x->mutex); + + return ret; +} + static const struct snd_soc_dai_ops pcm512x_dai_ops = { .startup = pcm512x_dai_startup, .hw_params = pcm512x_hw_params, .set_fmt = pcm512x_set_fmt, + .digital_mute = pcm512x_digital_mute, }; static struct snd_soc_dai_driver pcm512x_dai = { @@ -1388,6 +1500,8 @@ int pcm512x_probe(struct device *dev, struct regmap *regmap) if (!pcm512x) return -ENOMEM; + mutex_init(&pcm512x->mutex); + dev_set_drvdata(dev, pcm512x); pcm512x->regmap = regmap; diff --git a/sound/soc/codecs/pcm512x.h b/sound/soc/codecs/pcm512x.h index d70d9c0c2088..9dda8693498e 100644 --- a/sound/soc/codecs/pcm512x.h +++ b/sound/soc/codecs/pcm512x.h @@ -112,7 +112,9 @@ #define PCM512x_RQST_SHIFT 4 /* Page 0, Register 3 - mute */ +#define PCM512x_RQMR (1 << 0) #define PCM512x_RQMR_SHIFT 0 +#define PCM512x_RQML (1 << 4) #define PCM512x_RQML_SHIFT 4 /* Page 0, Register 4 - PLL */ diff --git a/sound/soc/codecs/rt1305.c b/sound/soc/codecs/rt1305.c index c4452efc7970..c2c8a68cec97 100644 --- a/sound/soc/codecs/rt1305.c +++ b/sound/soc/codecs/rt1305.c @@ -963,7 +963,8 @@ static const struct regmap_config rt1305_regmap = { .num_reg_defaults = ARRAY_SIZE(rt1305_reg), .ranges = rt1305_ranges, .num_ranges = ARRAY_SIZE(rt1305_ranges), - .use_single_rw = true, + .use_single_read = true, + .use_single_write = true, }; #if defined(CONFIG_OF) diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c index d88e67341083..e2855ab9a2c6 100644 --- a/sound/soc/codecs/rt274.c +++ b/sound/soc/codecs/rt274.c @@ -755,6 +755,7 @@ static int rt274_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, break; default: dev_warn(component->dev, "invalid pll source, use BCLK\n"); + /* fall through */ case RT274_PLL2_S_BCLK: snd_soc_component_update_bits(component, RT274_PLL2_CTRL, RT274_PLL2_SRC_MASK, RT274_PLL2_SRC_BCLK); @@ -782,6 +783,7 @@ static int rt274_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, break; default: dev_warn(component->dev, "invalid freq_in, assume 4.8M\n"); + /* fall through */ case 100: snd_soc_component_write(component, 0x7a, 0xaab6); snd_soc_component_write(component, 0x7b, 0x0301); @@ -1126,8 +1128,11 @@ static int rt274_i2c_probe(struct i2c_client *i2c, return ret; } - regmap_read(rt274->regmap, + ret = regmap_read(rt274->regmap, RT274_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val); + if (ret) + return ret; + if (val != RT274_VENDOR_ID) { dev_err(&i2c->dev, "Device with ID register %#x is not rt274\n", val); diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 6478d10c4f4a..bec2eefa8b0f 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -91,6 +91,14 @@ static void rt5514_spi_copy_work(struct work_struct *work) runtime = rt5514_dsp->substream->runtime; period_bytes = snd_pcm_lib_period_bytes(rt5514_dsp->substream); + if (!period_bytes) { + schedule_delayed_work(&rt5514_dsp->copy_work, 5); + goto done; + } + + if (rt5514_dsp->buf_size % period_bytes) + rt5514_dsp->buf_size = (rt5514_dsp->buf_size / period_bytes) * + period_bytes; if (rt5514_dsp->get_size >= rt5514_dsp->buf_size) { rt5514_spi_burst_read(RT5514_BUFFER_VOICE_WP, (u8 *)&buf, @@ -149,13 +157,11 @@ done: static void rt5514_schedule_copy(struct rt5514_dsp *rt5514_dsp) { - size_t period_bytes; u8 buf[8]; if (!rt5514_dsp->substream) return; - period_bytes = snd_pcm_lib_period_bytes(rt5514_dsp->substream); rt5514_dsp->get_size = 0; /** @@ -183,10 +189,6 @@ static void rt5514_schedule_copy(struct rt5514_dsp *rt5514_dsp) rt5514_dsp->buf_size = rt5514_dsp->buf_limit - rt5514_dsp->buf_base; - if (rt5514_dsp->buf_size % period_bytes) - rt5514_dsp->buf_size = (rt5514_dsp->buf_size / period_bytes) * - period_bytes; - if (rt5514_dsp->buf_base && rt5514_dsp->buf_limit && rt5514_dsp->buf_rp && rt5514_dsp->buf_size) schedule_delayed_work(&rt5514_dsp->copy_work, 0); @@ -278,6 +280,8 @@ static int rt5514_spi_pcm_probe(struct snd_soc_component *component) rt5514_dsp = devm_kzalloc(component->dev, sizeof(*rt5514_dsp), GFP_KERNEL); + if (!rt5514_dsp) + return -ENOMEM; rt5514_dsp->dev = &rt5514_spi->dev; mutex_init(&rt5514_dsp->dma_lock); diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 32fe76c3134a..a67de68b6da6 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -1201,7 +1201,8 @@ static const struct regmap_config rt5514_regmap = { .cache_type = REGCACHE_RBTREE, .reg_defaults = rt5514_reg, .num_reg_defaults = ARRAY_SIZE(rt5514_reg), - .use_single_rw = true, + .use_single_read = true, + .use_single_write = true, }; static const struct i2c_device_id rt5514_i2c_id[] = { diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index 3dc795f444ce..36a9f1c56c8d 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -1313,7 +1313,8 @@ static const struct snd_soc_component_driver soc_component_dev_rt5616 = { static const struct regmap_config rt5616_regmap = { .reg_bits = 8, .val_bits = 16, - .use_single_rw = true, + .use_single_read = true, + .use_single_write = true, .max_register = RT5616_DEVICE_ID + 1 + (ARRAY_SIZE(rt5616_ranges) * RT5616_PR_SPACING), .volatile_reg = rt5616_volatile_register, diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 27770143ae8f..fc530481a6e4 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2704,7 +2704,8 @@ static const struct snd_soc_component_driver soc_component_dev_rt5640 = { static const struct regmap_config rt5640_regmap = { .reg_bits = 8, .val_bits = 16, - .use_single_rw = true, + .use_single_read = true, + .use_single_write = true, .max_register = RT5640_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5640_ranges) * RT5640_PR_SPACING), diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 1dc70f452c1b..be674688dc40 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3559,7 +3559,8 @@ static const struct snd_soc_component_driver soc_component_dev_rt5645 = { static const struct regmap_config rt5645_regmap = { .reg_bits = 8, .val_bits = 16, - .use_single_rw = true, + .use_single_read = true, + .use_single_write = true, .max_register = RT5645_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5645_ranges) * RT5645_PR_SPACING), .volatile_reg = rt5645_volatile_register, @@ -3575,7 +3576,8 @@ static const struct regmap_config rt5645_regmap = { static const struct regmap_config rt5650_regmap = { .reg_bits = 8, .val_bits = 16, - .use_single_rw = true, + .use_single_read = true, + .use_single_write = true, .max_register = RT5645_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5645_ranges) * RT5645_PR_SPACING), .volatile_reg = rt5645_volatile_register, @@ -3592,7 +3594,8 @@ static const struct regmap_config temp_regmap = { .name="nocache", .reg_bits = 8, .val_bits = 16, - .use_single_rw = true, + .use_single_read = true, + .use_single_write = true, .max_register = RT5645_VENDOR_ID2 + 1, .cache_type = REGCACHE_NONE, }; diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 985852fd9723..b7ba64350a07 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -10,7 +10,6 @@ */ #include <linux/module.h> -#include <linux/moduleparam.h> #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> @@ -2124,7 +2123,8 @@ static const struct regmap_config rt5651_regmap = { .num_reg_defaults = ARRAY_SIZE(rt5651_reg), .ranges = rt5651_ranges, .num_ranges = ARRAY_SIZE(rt5651_ranges), - .use_single_rw = true, + .use_single_read = true, + .use_single_write = true, }; #if defined(CONFIG_OF) diff --git a/sound/soc/codecs/rt5660.c b/sound/soc/codecs/rt5660.c index 20a755137e63..e74b2e8cd423 100644 --- a/sound/soc/codecs/rt5660.c +++ b/sound/soc/codecs/rt5660.c @@ -1217,7 +1217,8 @@ static const struct snd_soc_component_driver soc_component_dev_rt5660 = { static const struct regmap_config rt5660_regmap = { .reg_bits = 8, .val_bits = 16, - .use_single_rw = true, + .use_single_read = true, + .use_single_write = true, .max_register = RT5660_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5660_ranges) * RT5660_PR_SPACING), @@ -1245,6 +1246,7 @@ MODULE_DEVICE_TABLE(of, rt5660_of_match); static const struct acpi_device_id rt5660_acpi_match[] = { { "10EC5660", 0 }, + { "10EC3277", 0 }, { }, }; MODULE_DEVICE_TABLE(acpi, rt5660_acpi_match); diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 9bd24ad42240..da6647015708 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -17,6 +17,7 @@ #include <linux/platform_device.h> #include <linux/spi/spi.h> #include <linux/acpi.h> +#include <linux/regulator/consumer.h> #include <linux/workqueue.h> #include <sound/core.h> #include <sound/pcm.h> @@ -33,6 +34,9 @@ #define RT5663_DEVICE_ID_2 0x6451 #define RT5663_DEVICE_ID_1 0x6406 +#define RT5663_POWER_ON_DELAY_MS 300 +#define RT5663_SUPPLY_CURRENT_UA 500000 + enum { CODEC_VER_1, CODEC_VER_0, @@ -48,6 +52,11 @@ struct impedance_mapping_table { unsigned int dc_offset_r_manual_mic; }; +static const char *const rt5663_supply_names[] = { + "avdd", + "cpvdd", +}; + struct rt5663_priv { struct snd_soc_component *component; struct rt5663_platform_data pdata; @@ -56,6 +65,7 @@ struct rt5663_priv { struct snd_soc_jack *hs_jack; struct timer_list btn_check_timer; struct impedance_mapping_table *imp_table; + struct regulator_bulk_data supplies[ARRAY_SIZE(rt5663_supply_names)]; int codec_ver; int sysclk; @@ -72,6 +82,7 @@ struct rt5663_priv { static const struct reg_sequence rt5663_patch_list[] = { { 0x002a, 0x8020 }, { 0x0086, 0x0028 }, + { 0x0100, 0xa020 }, { 0x0117, 0x0f28 }, { 0x02fb, 0x8089 }, }; @@ -580,7 +591,7 @@ static const struct reg_default rt5663_reg[] = { { 0x00fd, 0x0001 }, { 0x00fe, 0x10ec }, { 0x00ff, 0x6406 }, - { 0x0100, 0xa0a0 }, + { 0x0100, 0xa020 }, { 0x0108, 0x4444 }, { 0x0109, 0x4444 }, { 0x010a, 0xaaaa }, @@ -2337,6 +2348,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w, 0x8000); snd_soc_component_update_bits(component, RT5663_DEPOP_1, 0x3000, 0x3000); + snd_soc_component_update_bits(component, + RT5663_DIG_VOL_ZCD, 0x00c0, 0x0080); } break; @@ -2351,6 +2364,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w, RT5663_OVCD_HP_MASK, RT5663_OVCD_HP_EN); snd_soc_component_update_bits(component, RT5663_DACREF_LDO, 0x3e0e, 0); + snd_soc_component_update_bits(component, + RT5663_DIG_VOL_ZCD, 0x00c0, 0); } break; @@ -3252,7 +3267,8 @@ static const struct snd_soc_component_driver soc_component_dev_rt5663 = { static const struct regmap_config rt5663_v2_regmap = { .reg_bits = 16, .val_bits = 16, - .use_single_rw = true, + .use_single_read = true, + .use_single_write = true, .max_register = 0x07fa, .volatile_reg = rt5663_v2_volatile_register, .readable_reg = rt5663_v2_readable_register, @@ -3264,7 +3280,8 @@ static const struct regmap_config rt5663_v2_regmap = { static const struct regmap_config rt5663_regmap = { .reg_bits = 16, .val_bits = 16, - .use_single_rw = true, + .use_single_read = true, + .use_single_write = true, .max_register = 0x03f3, .volatile_reg = rt5663_volatile_register, .readable_reg = rt5663_readable_register, @@ -3277,7 +3294,8 @@ static const struct regmap_config temp_regmap = { .name = "nocache", .reg_bits = 16, .val_bits = 16, - .use_single_rw = true, + .use_single_read = true, + .use_single_write = true, .max_register = 0x03f3, .cache_type = REGCACHE_NONE, }; @@ -3475,7 +3493,7 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, { struct rt5663_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt5663_priv *rt5663; - int ret; + int ret, i; unsigned int val; struct regmap *regmap; @@ -3492,12 +3510,44 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, else rt5663_parse_dp(rt5663, &i2c->dev); + for (i = 0; i < ARRAY_SIZE(rt5663->supplies); i++) + rt5663->supplies[i].supply = rt5663_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, + ARRAY_SIZE(rt5663->supplies), + rt5663->supplies); + if (ret) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + /* Set load for regulator. */ + for (i = 0; i < ARRAY_SIZE(rt5663->supplies); i++) { + ret = regulator_set_load(rt5663->supplies[i].consumer, + RT5663_SUPPLY_CURRENT_UA); + if (ret < 0) { + dev_err(&i2c->dev, + "Failed to set regulator load on %s, ret: %d\n", + rt5663->supplies[i].supply, ret); + return ret; + } + } + + ret = regulator_bulk_enable(ARRAY_SIZE(rt5663->supplies), + rt5663->supplies); + + if (ret) { + dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + msleep(RT5663_POWER_ON_DELAY_MS); + regmap = devm_regmap_init_i2c(i2c, &temp_regmap); if (IS_ERR(regmap)) { ret = PTR_ERR(regmap); dev_err(&i2c->dev, "Failed to allocate temp register map: %d\n", ret); - return ret; + goto err_enable; } ret = regmap_read(regmap, RT5663_VENDOR_ID_2, &val); @@ -3522,14 +3572,15 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, dev_err(&i2c->dev, "Device with ID register %#x is not rt5663\n", val); - return -ENODEV; + ret = -ENODEV; + goto err_enable; } if (IS_ERR(rt5663->regmap)) { ret = PTR_ERR(rt5663->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", ret); - return ret; + goto err_enable; } /* reset and calibrate */ @@ -3627,20 +3678,32 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, ret = request_irq(i2c->irq, rt5663_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5663", rt5663); - if (ret) + if (ret) { dev_err(&i2c->dev, "%s Failed to reguest IRQ: %d\n", __func__, ret); + goto err_enable; + } } ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5663, rt5663_dai, ARRAY_SIZE(rt5663_dai)); - if (ret) { - if (i2c->irq) - free_irq(i2c->irq, rt5663); - } + if (ret) + goto err_enable; + return 0; + + + /* + * Error after enabling regulators should goto err_enable + * to disable regulators. + */ +err_enable: + if (i2c->irq) + free_irq(i2c->irq, rt5663); + + regulator_bulk_disable(ARRAY_SIZE(rt5663->supplies), rt5663->supplies); return ret; } @@ -3651,6 +3714,8 @@ static int rt5663_i2c_remove(struct i2c_client *i2c) if (i2c->irq) free_irq(i2c->irq, rt5663); + regulator_bulk_disable(ARRAY_SIZE(rt5663->supplies), rt5663->supplies); + return 0; } diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 6ba99f5ed3f4..f2ad3a4c3b7f 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4633,7 +4633,8 @@ static const struct regmap_config rt5665_regmap = { .cache_type = REGCACHE_RBTREE, .reg_defaults = rt5665_reg, .num_reg_defaults = ARRAY_SIZE(rt5665_reg), - .use_single_rw = true, + .use_single_read = true, + .use_single_write = true, }; static const struct i2c_device_id rt5665_i2c_id[] = { diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c index 3c19d03f2446..230a21c93b6b 100644 --- a/sound/soc/codecs/rt5668.c +++ b/sound/soc/codecs/rt5668.c @@ -2375,7 +2375,8 @@ static const struct regmap_config rt5668_regmap = { .cache_type = REGCACHE_RBTREE, .reg_defaults = rt5668_reg, .num_reg_defaults = ARRAY_SIZE(rt5668_reg), - .use_single_rw = true, + .use_single_read = true, + .use_single_write = true, }; static const struct i2c_device_id rt5668_i2c_id[] = { @@ -2587,17 +2588,10 @@ static int rt5668_i2c_probe(struct i2c_client *i2c, } - return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5668, + return devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5668, rt5668_dai, ARRAY_SIZE(rt5668_dai)); } -static int rt5668_i2c_remove(struct i2c_client *i2c) -{ - snd_soc_unregister_component(&i2c->dev); - - return 0; -} - static void rt5668_i2c_shutdown(struct i2c_client *client) { struct rt5668_priv *rt5668 = i2c_get_clientdata(client); @@ -2628,7 +2622,6 @@ static struct i2c_driver rt5668_i2c_driver = { .acpi_match_table = ACPI_PTR(rt5668_acpi_match), }, .probe = rt5668_i2c_probe, - .remove = rt5668_i2c_remove, .shutdown = rt5668_i2c_shutdown, .id_table = rt5668_i2c_id, }; diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 732ef928b25d..453328c988c0 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2814,7 +2814,8 @@ static const struct snd_soc_component_driver soc_component_dev_rt5670 = { static const struct regmap_config rt5670_regmap = { .reg_bits = 8, .val_bits = 16, - .use_single_rw = true, + .use_single_read = true, + .use_single_write = true, .max_register = RT5670_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5670_ranges) * RT5670_PR_SPACING), .volatile_reg = rt5670_volatile_register, @@ -2877,6 +2878,18 @@ static const struct dmi_system_id dmi_platform_intel_quirks[] = { }, { .callback = rt5670_quirk_cb, + .ident = "Lenovo Thinkpad Tablet 8", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_VERSION, "ThinkPad 8"), + }, + .driver_data = (unsigned long *)(RT5670_DMIC_EN | + RT5670_DMIC2_INR | + RT5670_DEV_GPIO | + RT5670_JD_MODE1), + }, + { + .callback = rt5670_quirk_cb, .ident = "Lenovo Thinkpad Tablet 10", .matches = { DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"), diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index bd51f3655ee3..84501c2020c7 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -18,7 +18,6 @@ #include <linux/interrupt.h> #include <linux/irq.h> #include <linux/slab.h> -#include <linux/gpio.h> #include <linux/sched.h> #include <linux/uaccess.h> #include <linux/regulator/consumer.h> diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index afe7d5b19313..a9b91bcfcc09 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -67,7 +67,8 @@ struct rt5682_priv { }; static const struct reg_sequence patch_list[] = { - {0x01c1, 0x1000}, + {RT5682_HP_IMP_SENS_CTRL_19, 0x1000}, + {RT5682_DAC_ADC_DIG_VOL1, 0xa020}, }; static const struct reg_default rt5682_reg[] = { @@ -749,7 +750,6 @@ static bool rt5682_readable_register(struct device *dev, unsigned int reg) } } -static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -2250, 150, 0); static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6525, 75, 0); static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); @@ -1108,10 +1108,6 @@ static void rt5682_jack_detect_handler(struct work_struct *work) } static const struct snd_kcontrol_new rt5682_snd_controls[] = { - /* Headphone Output Volume */ - SOC_DOUBLE_R_TLV("Headphone Playback Volume", RT5682_HPL_GAIN, - RT5682_HPR_GAIN, RT5682_G_HP_SFT, 15, 1, hp_vol_tlv), - /* DAC Digital Volume */ SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5682_DAC1_DIG_VOL, RT5682_L_VOL_SFT + 1, RT5682_R_VOL_SFT + 1, 86, 0, dac_vol_tlv), @@ -1437,6 +1433,28 @@ static const struct snd_kcontrol_new hpor_switch = SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682_HP_CTRL_1, RT5682_R_MUTE_SFT, 1, 1); +static int rt5682_charge_pump_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_component_update_bits(component, + RT5682_HP_CHARGE_PUMP_1, RT5682_PM_HP_MASK, RT5682_PM_HP_HV); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_component_update_bits(component, + RT5682_HP_CHARGE_PUMP_1, RT5682_PM_HP_MASK, RT5682_PM_HP_LV); + break; + default: + return 0; + } + + return 0; +} + static int rt5682_hp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1449,10 +1467,10 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w, RT5682_HP_LOGIC_CTRL_2, 0x0012); snd_soc_component_write(component, RT5682_HP_CTRL_2, 0x6000); - snd_soc_component_update_bits(component, RT5682_STO_NG2_CTRL_1, - RT5682_NG2_EN_MASK, RT5682_NG2_EN); snd_soc_component_update_bits(component, RT5682_DEPOP_1, 0x60, 0x60); + snd_soc_component_update_bits(component, + RT5682_DAC_ADC_DIG_VOL1, 0x00c0, 0x0080); break; case SND_SOC_DAPM_POST_PMD: @@ -1460,6 +1478,8 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w, RT5682_DEPOP_1, 0x60, 0x0); snd_soc_component_write(component, RT5682_HP_CTRL_2, 0x0000); + snd_soc_component_update_bits(component, + RT5682_DAC_ADC_DIG_VOL1, 0x00c0, 0x0000); break; default: @@ -1723,7 +1743,8 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("HP Amp R", RT5682_PWR_ANLG_1, RT5682_PWR_HA_R_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("Charge Pump", 1, RT5682_DEPOP_1, - RT5682_PUMP_EN_SFT, 0, NULL, 0), + RT5682_PUMP_EN_SFT, 0, rt5682_charge_pump_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY_S("Capless", 2, RT5682_DEPOP_1, RT5682_CAPLESS_EN_SFT, 0, NULL, 0), @@ -1757,7 +1778,9 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc}, {"DAC Stereo1 Filter", NULL, "DAC STO1 ASRC", is_using_asrc}, {"ADC STO1 ASRC", NULL, "AD ASRC"}, + {"ADC STO1 ASRC", NULL, "DA ASRC"}, {"ADC STO1 ASRC", NULL, "CLKDET"}, + {"DAC STO1 ASRC", NULL, "AD ASRC"}, {"DAC STO1 ASRC", NULL, "DA ASRC"}, {"DAC STO1 ASRC", NULL, "CLKDET"}, @@ -1884,6 +1907,7 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"HP Amp", NULL, "Charge Pump"}, {"HP Amp", NULL, "CLKDET SYS"}, {"HP Amp", NULL, "CBJ Power"}, + {"HP Amp", NULL, "Vref1"}, {"HP Amp", NULL, "Vref2"}, {"HPOL Playback", "Switch", "HP Amp"}, {"HPOR Playback", "Switch", "HP Amp"}, @@ -2419,7 +2443,8 @@ static const struct regmap_config rt5682_regmap = { .cache_type = REGCACHE_RBTREE, .reg_defaults = rt5682_reg, .num_reg_defaults = ARRAY_SIZE(rt5682_reg), - .use_single_rw = true, + .use_single_read = true, + .use_single_write = true, }; static const struct i2c_device_id rt5682_i2c_id[] = { @@ -2451,30 +2476,23 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682) mutex_lock(&rt5682->calibrate_mutex); rt5682_reset(rt5682->regmap); - regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2bf); + regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2af); usleep_range(15000, 20000); - regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2bf); - regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0380); - regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x8001); - regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000); - regmap_write(rt5682->regmap, RT5682_STO1_DAC_MIXER, 0x2080); - regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x4040); - regmap_write(rt5682->regmap, RT5682_DEPOP_1, 0x0069); + regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2af); + regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0300); + regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x8000); + regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0100); + regmap_write(rt5682->regmap, RT5682_HP_IMP_SENS_CTRL_19, 0x3800); regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x3000); - regmap_write(rt5682->regmap, RT5682_HP_CTRL_2, 0x6000); - regmap_write(rt5682->regmap, RT5682_HP_CHARGE_PUMP_1, 0x0f26); - regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7f05); + regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7005); regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x686c); regmap_write(rt5682->regmap, RT5682_CAL_REC, 0x0d0d); - regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_9, 0x000f); - regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x8d01); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_2, 0x0321); regmap_write(rt5682->regmap, RT5682_HP_LOGIC_CTRL_2, 0x0004); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0x7c00); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_3, 0x06a1); regmap_write(rt5682->regmap, RT5682_A_DAC1_MUX, 0x0311); - regmap_write(rt5682->regmap, RT5682_RESET_HPF_CTRL, 0x0000); - regmap_write(rt5682->regmap, RT5682_ADC_STO1_HP_CTRL_1, 0x3320); + regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0x7c00); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0xfc00); @@ -2490,8 +2508,13 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682) pr_err("HP Calibration Failure\n"); /* restore settings */ - regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4); + regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0x02af); + regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0080); + regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x0000); regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000); + regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x2000); + regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x2005); + regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4); mutex_unlock(&rt5682->calibrate_mutex); @@ -2565,7 +2588,7 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, rt5682_calibrate(rt5682); - ret = regmap_register_patch(rt5682->regmap, patch_list, + ret = regmap_multi_reg_write(rt5682->regmap, patch_list, ARRAY_SIZE(patch_list)); if (ret != 0) dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); @@ -2619,6 +2642,10 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, RT5682_GP4_PIN_MASK | RT5682_GP5_PIN_MASK, RT5682_GP4_PIN_ADCDAT1 | RT5682_GP5_PIN_DACDAT1); regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000); + regmap_update_bits(rt5682->regmap, RT5682_BIAS_CUR_CTRL_8, + RT5682_HPA_CP_BIAS_CTRL_MASK, RT5682_HPA_CP_BIAS_3UA); + regmap_update_bits(rt5682->regmap, RT5682_CHARGE_PUMP_1, + RT5682_CP_CLK_HP_MASK, RT5682_CP_CLK_HP_300KHZ); INIT_DELAYED_WORK(&rt5682->jack_detect_work, rt5682_jack_detect_handler); @@ -2636,11 +2663,17 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, } - return devm_snd_soc_register_component(&i2c->dev, - &soc_component_dev_rt5682, + return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5682, rt5682_dai, ARRAY_SIZE(rt5682_dai)); } +static int rt5682_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_component(&i2c->dev); + + return 0; +} + static void rt5682_i2c_shutdown(struct i2c_client *client) { struct rt5682_priv *rt5682 = i2c_get_clientdata(client); @@ -2671,6 +2704,7 @@ static struct i2c_driver rt5682_i2c_driver = { .acpi_match_table = ACPI_PTR(rt5682_acpi_match), }, .probe = rt5682_i2c_probe, + .remove = rt5682_i2c_remove, .shutdown = rt5682_i2c_shutdown, .id_table = rt5682_i2c_id, }; diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index 8068140ebe3f..96944cff0ed7 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -849,18 +849,18 @@ #define RT5682_SCLK_SRC_PLL2 (0x2 << 13) #define RT5682_SCLK_SRC_SDW (0x3 << 13) #define RT5682_SCLK_SRC_RCCLK (0x4 << 13) -#define RT5682_PLL1_SRC_MASK (0x3 << 10) -#define RT5682_PLL1_SRC_SFT 10 -#define RT5682_PLL1_SRC_MCLK (0x0 << 10) -#define RT5682_PLL1_SRC_BCLK1 (0x1 << 10) -#define RT5682_PLL1_SRC_SDW (0x2 << 10) -#define RT5682_PLL1_SRC_RC (0x3 << 10) -#define RT5682_PLL2_SRC_MASK (0x3 << 8) -#define RT5682_PLL2_SRC_SFT 8 -#define RT5682_PLL2_SRC_MCLK (0x0 << 8) -#define RT5682_PLL2_SRC_BCLK1 (0x1 << 8) -#define RT5682_PLL2_SRC_SDW (0x2 << 8) -#define RT5682_PLL2_SRC_RC (0x3 << 8) +#define RT5682_PLL2_SRC_MASK (0x3 << 10) +#define RT5682_PLL2_SRC_SFT 10 +#define RT5682_PLL2_SRC_MCLK (0x0 << 10) +#define RT5682_PLL2_SRC_BCLK1 (0x1 << 10) +#define RT5682_PLL2_SRC_SDW (0x2 << 10) +#define RT5682_PLL2_SRC_RC (0x3 << 10) +#define RT5682_PLL1_SRC_MASK (0x3 << 8) +#define RT5682_PLL1_SRC_SFT 8 +#define RT5682_PLL1_SRC_MCLK (0x0 << 8) +#define RT5682_PLL1_SRC_BCLK1 (0x1 << 8) +#define RT5682_PLL1_SRC_SDW (0x2 << 8) +#define RT5682_PLL1_SRC_RC (0x3 << 8) @@ -1214,6 +1214,20 @@ #define RT5682_JDH_NO_PLUG (0x1 << 4) #define RT5682_JDH_PLUG (0x0 << 4) +/* Bias current control 8 (0x0111) */ +#define RT5682_HPA_CP_BIAS_CTRL_MASK (0x3 << 2) +#define RT5682_HPA_CP_BIAS_2UA (0x0 << 2) +#define RT5682_HPA_CP_BIAS_3UA (0x1 << 2) +#define RT5682_HPA_CP_BIAS_4UA (0x2 << 2) +#define RT5682_HPA_CP_BIAS_6UA (0x3 << 2) + +/* Charge Pump Internal Register1 (0x0125) */ +#define RT5682_CP_CLK_HP_MASK (0x3 << 4) +#define RT5682_CP_CLK_HP_100KHZ (0x0 << 4) +#define RT5682_CP_CLK_HP_200KHZ (0x1 << 4) +#define RT5682_CP_CLK_HP_300KHZ (0x2 << 4) +#define RT5682_CP_CLK_HP_600KHZ (0x3 << 4) + /* Chopper and Clock control for DAC (0x013a)*/ #define RT5682_CKXEN_DAC1_MASK (0x1 << 13) #define RT5682_CKXEN_DAC1_SFT 13 diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 60764f6201b1..add18d6d77da 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1218,7 +1218,7 @@ static int sgtl5000_set_power_regs(struct snd_soc_component *component) * Searching for a suitable index solving this formula: * idx = 40 * log10(vag_val / lo_cagcntrl) + 15 */ - vol_quot = (vag * 100) / lo_vag; + vol_quot = lo_vag ? (vag * 100) / lo_vag : 0; lo_vol = 0; for (i = 0; i < ARRAY_SIZE(vol_quot_table); i++) { if (vol_quot >= vol_quot_table[i]) diff --git a/sound/soc/codecs/simple-amplifier.c b/sound/soc/codecs/simple-amplifier.c index 85524acf3e9c..c07e8a80b4b7 100644 --- a/sound/soc/codecs/simple-amplifier.c +++ b/sound/soc/codecs/simple-amplifier.c @@ -19,6 +19,7 @@ #include <linux/gpio/consumer.h> #include <linux/module.h> +#include <linux/regulator/consumer.h> #include <sound/soc.h> #define DRV_NAME "simple-amplifier" @@ -58,11 +59,14 @@ static const struct snd_soc_dapm_widget simple_amp_dapm_widgets[] = { (SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD)), SND_SOC_DAPM_OUTPUT("OUTL"), SND_SOC_DAPM_OUTPUT("OUTR"), + SND_SOC_DAPM_REGULATOR_SUPPLY("VCC", 20, 0), }; static const struct snd_soc_dapm_route simple_amp_dapm_routes[] = { { "DRV", NULL, "INL" }, { "DRV", NULL, "INR" }, + { "OUTL", NULL, "VCC" }, + { "OUTR", NULL, "VCC" }, { "OUTL", NULL, "DRV" }, { "OUTR", NULL, "DRV" }, }; diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index ce508b4cc85c..f753d2db0a5a 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -21,6 +21,7 @@ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> +#include <linux/clk.h> #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> @@ -142,6 +143,7 @@ static const char *sta32x_supply_names[] = { /* codec private data */ struct sta32x_priv { struct regmap *regmap; + struct clk *xti_clk; struct regulator_bulk_data supplies[ARRAY_SIZE(sta32x_supply_names)]; struct snd_soc_component *component; struct sta32x_platform_data *pdata; @@ -882,6 +884,15 @@ static int sta32x_probe(struct snd_soc_component *component) sta32x->component = component; + if (sta32x->xti_clk) { + ret = clk_prepare_enable(sta32x->xti_clk); + if (ret != 0) { + dev_err(component->dev, + "Failed to enable clock: %d\n", ret); + return ret; + } + } + ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); if (ret != 0) { @@ -984,6 +995,9 @@ static void sta32x_remove(struct snd_soc_component *component) sta32x_watchdog_stop(sta32x); regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + + if (sta32x->xti_clk) + clk_disable_unprepare(sta32x->xti_clk); } static const struct snd_soc_component_driver sta32x_component = { @@ -1041,6 +1055,8 @@ static int sta32x_probe_dt(struct device *dev, struct sta32x_priv *sta32x) of_property_read_u8(np, "st,ch3-output-mapping", &pdata->ch3_output_mapping); + if (of_get_property(np, "st,fault-detect-recovery", NULL)) + pdata->fault_detect_recovery = 1; if (of_get_property(np, "st,thermal-warning-recovery", NULL)) pdata->thermal_warning_recovery = 1; if (of_get_property(np, "st,thermal-warning-adjustment", NULL)) @@ -1098,6 +1114,17 @@ static int sta32x_i2c_probe(struct i2c_client *i2c, } #endif + /* Clock */ + sta32x->xti_clk = devm_clk_get(dev, "xti"); + if (IS_ERR(sta32x->xti_clk)) { + ret = PTR_ERR(sta32x->xti_clk); + + if (ret == -EPROBE_DEFER) + return ret; + + sta32x->xti_clk = NULL; + } + /* GPIOs */ sta32x->gpiod_nreset = devm_gpiod_get_optional(dev, "reset", GPIOD_OUT_LOW); diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c index ae3d032ac35a..6bd0e5d5347f 100644 --- a/sound/soc/codecs/tas5720.c +++ b/sound/soc/codecs/tas5720.c @@ -152,6 +152,7 @@ static int tas5720_set_dai_tdm_slot(struct snd_soc_dai *dai, int slots, int slot_width) { struct snd_soc_component *component = dai->component; + struct tas5720_data *tas5720 = snd_soc_component_get_drvdata(component); unsigned int first_slot; int ret; @@ -185,6 +186,20 @@ static int tas5720_set_dai_tdm_slot(struct snd_soc_dai *dai, if (ret < 0) goto error_snd_soc_component_update_bits; + /* Configure TDM slot width. This is only applicable to TAS5722. */ + switch (tas5720->devtype) { + case TAS5722: + ret = snd_soc_component_update_bits(component, TAS5722_DIGITAL_CTRL2_REG, + TAS5722_TDM_SLOT_16B, + slot_width == 16 ? + TAS5722_TDM_SLOT_16B : 0); + if (ret < 0) + goto error_snd_soc_component_update_bits; + break; + default: + break; + } + return 0; error_snd_soc_component_update_bits: @@ -485,15 +500,56 @@ static const DECLARE_TLV_DB_RANGE(dac_analog_tlv, ); /* - * DAC digital volumes. From -103.5 to 24 dB in 0.5 dB steps. Note that - * setting the gain below -100 dB (register value <0x7) is effectively a MUTE - * as per device datasheet. + * DAC digital volumes. From -103.5 to 24 dB in 0.5 dB or 0.25 dB steps + * depending on the device. Note that setting the gain below -100 dB + * (register value <0x7) is effectively a MUTE as per device datasheet. + * + * Note that for the TAS5722 the digital volume controls are actually split + * over two registers, so we need custom getters/setters for access. */ -static DECLARE_TLV_DB_SCALE(dac_tlv, -10350, 50, 0); +static DECLARE_TLV_DB_SCALE(tas5720_dac_tlv, -10350, 50, 0); +static DECLARE_TLV_DB_SCALE(tas5722_dac_tlv, -10350, 25, 0); + +static int tas5722_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + unsigned int val; + + snd_soc_component_read(component, TAS5720_VOLUME_CTRL_REG, &val); + ucontrol->value.integer.value[0] = val << 1; + + snd_soc_component_read(component, TAS5722_DIGITAL_CTRL2_REG, &val); + ucontrol->value.integer.value[0] |= val & TAS5722_VOL_CONTROL_LSB; + + return 0; +} + +static int tas5722_volume_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + unsigned int sel = ucontrol->value.integer.value[0]; + + snd_soc_component_write(component, TAS5720_VOLUME_CTRL_REG, sel >> 1); + snd_soc_component_update_bits(component, TAS5722_DIGITAL_CTRL2_REG, + TAS5722_VOL_CONTROL_LSB, sel); + + return 0; +} static const struct snd_kcontrol_new tas5720_snd_controls[] = { SOC_SINGLE_TLV("Speaker Driver Playback Volume", - TAS5720_VOLUME_CTRL_REG, 0, 0xff, 0, dac_tlv), + TAS5720_VOLUME_CTRL_REG, 0, 0xff, 0, tas5720_dac_tlv), + SOC_SINGLE_TLV("Speaker Driver Analog Gain", TAS5720_ANALOG_CTRL_REG, + TAS5720_ANALOG_GAIN_SHIFT, 3, 0, dac_analog_tlv), +}; + +static const struct snd_kcontrol_new tas5722_snd_controls[] = { + SOC_SINGLE_EXT_TLV("Speaker Driver Playback Volume", + 0, 0, 511, 0, + tas5722_volume_get, tas5722_volume_set, + tas5722_dac_tlv), SOC_SINGLE_TLV("Speaker Driver Analog Gain", TAS5720_ANALOG_CTRL_REG, TAS5720_ANALOG_GAIN_SHIFT, 3, 0, dac_analog_tlv), }; @@ -527,6 +583,23 @@ static const struct snd_soc_component_driver soc_component_dev_tas5720 = { .non_legacy_dai_naming = 1, }; +static const struct snd_soc_component_driver soc_component_dev_tas5722 = { + .probe = tas5720_codec_probe, + .remove = tas5720_codec_remove, + .suspend = tas5720_suspend, + .resume = tas5720_resume, + .controls = tas5722_snd_controls, + .num_controls = ARRAY_SIZE(tas5722_snd_controls), + .dapm_widgets = tas5720_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tas5720_dapm_widgets), + .dapm_routes = tas5720_audio_map, + .num_dapm_routes = ARRAY_SIZE(tas5720_audio_map), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + /* PCM rates supported by the TAS5720 driver */ #define TAS5720_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) @@ -613,9 +686,23 @@ static int tas5720_probe(struct i2c_client *client, dev_set_drvdata(dev, data); - ret = devm_snd_soc_register_component(&client->dev, - &soc_component_dev_tas5720, - tas5720_dai, ARRAY_SIZE(tas5720_dai)); + switch (id->driver_data) { + case TAS5720: + ret = devm_snd_soc_register_component(&client->dev, + &soc_component_dev_tas5720, + tas5720_dai, + ARRAY_SIZE(tas5720_dai)); + break; + case TAS5722: + ret = devm_snd_soc_register_component(&client->dev, + &soc_component_dev_tas5722, + tas5720_dai, + ARRAY_SIZE(tas5720_dai)); + break; + default: + dev_err(dev, "unexpected private driver data\n"); + return -EINVAL; + } if (ret < 0) { dev_err(dev, "failed to register component: %d\n", ret); return ret; diff --git a/sound/soc/codecs/tas6424.c b/sound/soc/codecs/tas6424.c index 0d6145549a98..aaba39295079 100644 --- a/sound/soc/codecs/tas6424.c +++ b/sound/soc/codecs/tas6424.c @@ -41,6 +41,7 @@ struct tas6424_data { struct regmap *regmap; struct regulator_bulk_data supplies[TAS6424_NUM_SUPPLIES]; struct delayed_work fault_check_work; + unsigned int last_cfault; unsigned int last_fault1; unsigned int last_fault2; unsigned int last_warn; @@ -377,7 +378,7 @@ static struct snd_soc_component_driver soc_codec_dev_tas6424 = { .non_legacy_dai_naming = 1, }; -static struct snd_soc_dai_ops tas6424_speaker_dai_ops = { +static const struct snd_soc_dai_ops tas6424_speaker_dai_ops = { .hw_params = tas6424_hw_params, .set_fmt = tas6424_set_dai_fmt, .set_tdm_slot = tas6424_set_dai_tdm_slot, @@ -406,9 +407,54 @@ static void tas6424_fault_check_work(struct work_struct *work) unsigned int reg; int ret; + ret = regmap_read(tas6424->regmap, TAS6424_CHANNEL_FAULT, ®); + if (ret < 0) { + dev_err(dev, "failed to read CHANNEL_FAULT register: %d\n", ret); + goto out; + } + + if (!reg) { + tas6424->last_cfault = reg; + goto check_global_fault1_reg; + } + + /* + * Only flag errors once for a given occurrence. This is needed as + * the TAS6424 will take time clearing the fault condition internally + * during which we don't want to bombard the system with the same + * error message over and over. + */ + if ((reg & TAS6424_FAULT_OC_CH1) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH1)) + dev_crit(dev, "experienced a channel 1 overcurrent fault\n"); + + if ((reg & TAS6424_FAULT_OC_CH2) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH2)) + dev_crit(dev, "experienced a channel 2 overcurrent fault\n"); + + if ((reg & TAS6424_FAULT_OC_CH3) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH3)) + dev_crit(dev, "experienced a channel 3 overcurrent fault\n"); + + if ((reg & TAS6424_FAULT_OC_CH4) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH4)) + dev_crit(dev, "experienced a channel 4 overcurrent fault\n"); + + if ((reg & TAS6424_FAULT_DC_CH1) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH1)) + dev_crit(dev, "experienced a channel 1 DC fault\n"); + + if ((reg & TAS6424_FAULT_DC_CH2) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH2)) + dev_crit(dev, "experienced a channel 2 DC fault\n"); + + if ((reg & TAS6424_FAULT_DC_CH3) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH3)) + dev_crit(dev, "experienced a channel 3 DC fault\n"); + + if ((reg & TAS6424_FAULT_DC_CH4) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH4)) + dev_crit(dev, "experienced a channel 4 DC fault\n"); + + /* Store current fault1 value so we can detect any changes next time */ + tas6424->last_cfault = reg; + +check_global_fault1_reg: ret = regmap_read(tas6424->regmap, TAS6424_GLOB_FAULT1, ®); if (ret < 0) { - dev_err(dev, "failed to read FAULT1 register: %d\n", ret); + dev_err(dev, "failed to read GLOB_FAULT1 register: %d\n", ret); goto out; } @@ -429,12 +475,6 @@ static void tas6424_fault_check_work(struct work_struct *work) goto check_global_fault2_reg; } - /* - * Only flag errors once for a given occurrence. This is needed as - * the TAS6424 will take time clearing the fault condition internally - * during which we don't want to bombard the system with the same - * error message over and over. - */ if ((reg & TAS6424_FAULT_PVDD_OV) && !(tas6424->last_fault1 & TAS6424_FAULT_PVDD_OV)) dev_crit(dev, "experienced a PVDD overvoltage fault\n"); @@ -453,7 +493,7 @@ static void tas6424_fault_check_work(struct work_struct *work) check_global_fault2_reg: ret = regmap_read(tas6424->regmap, TAS6424_GLOB_FAULT2, ®); if (ret < 0) { - dev_err(dev, "failed to read FAULT2 register: %d\n", ret); + dev_err(dev, "failed to read GLOB_FAULT2 register: %d\n", ret); goto out; } @@ -530,7 +570,7 @@ check_warn_reg: /* Store current warn value so we can detect any changes next time */ tas6424->last_warn = reg; - /* Clear any faults by toggling the CLEAR_FAULT control bit */ + /* Clear any warnings by toggling the CLEAR_FAULT control bit */ ret = regmap_write_bits(tas6424->regmap, TAS6424_MISC_CTRL3, TAS6424_CLEAR_FAULT, TAS6424_CLEAR_FAULT); if (ret < 0) diff --git a/sound/soc/codecs/tas6424.h b/sound/soc/codecs/tas6424.h index b5958c45ed0e..c67a7835ca66 100644 --- a/sound/soc/codecs/tas6424.h +++ b/sound/soc/codecs/tas6424.h @@ -116,6 +116,16 @@ #define TAS6424_LDGBYPASS_MASK BIT(TAS6424_LDGBYPASS_SHIFT) /* TAS6424_GLOB_FAULT1_REG */ +#define TAS6424_FAULT_OC_CH1 BIT(7) +#define TAS6424_FAULT_OC_CH2 BIT(6) +#define TAS6424_FAULT_OC_CH3 BIT(5) +#define TAS6424_FAULT_OC_CH4 BIT(4) +#define TAS6424_FAULT_DC_CH1 BIT(3) +#define TAS6424_FAULT_DC_CH2 BIT(2) +#define TAS6424_FAULT_DC_CH3 BIT(1) +#define TAS6424_FAULT_DC_CH4 BIT(0) + +/* TAS6424_GLOB_FAULT1_REG */ #define TAS6424_FAULT_CLOCK BIT(4) #define TAS6424_FAULT_PVDD_OV BIT(3) #define TAS6424_FAULT_VBAT_OV BIT(2) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index bf92d36b8f8a..c6048d95c6d3 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -167,6 +167,7 @@ struct aic31xx_priv { u8 p_div; int rate_div_line; bool master_dapm_route_applied; + int irq; }; struct aic31xx_rate_divs { @@ -1094,7 +1095,7 @@ static int aic31xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, if (freq/i > 20000000) { dev_err(aic31xx->dev, "%s: Too high mclk frequency %u\n", __func__, freq); - return -EINVAL; + return -EINVAL; } aic31xx->p_div = i; @@ -1391,6 +1392,69 @@ static const struct acpi_device_id aic31xx_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, aic31xx_acpi_match); #endif +static irqreturn_t aic31xx_irq(int irq, void *data) +{ + struct aic31xx_priv *aic31xx = data; + struct device *dev = aic31xx->dev; + unsigned int value; + bool handled = false; + int ret; + + ret = regmap_read(aic31xx->regmap, AIC31XX_INTRDACFLAG, &value); + if (ret) { + dev_err(dev, "Failed to read interrupt mask: %d\n", ret); + goto exit; + } + + if (value) + handled = true; + else + goto read_overflow; + + if (value & AIC31XX_HPLSCDETECT) + dev_err(dev, "Short circuit on Left output is detected\n"); + if (value & AIC31XX_HPRSCDETECT) + dev_err(dev, "Short circuit on Right output is detected\n"); + if (value & ~(AIC31XX_HPLSCDETECT | + AIC31XX_HPRSCDETECT)) + dev_err(dev, "Unknown DAC interrupt flags: 0x%08x\n", value); + +read_overflow: + ret = regmap_read(aic31xx->regmap, AIC31XX_OFFLAG, &value); + if (ret) { + dev_err(dev, "Failed to read overflow flag: %d\n", ret); + goto exit; + } + + if (value) + handled = true; + else + goto exit; + + if (value & AIC31XX_DAC_OF_LEFT) + dev_warn(dev, "Left-channel DAC overflow has occurred\n"); + if (value & AIC31XX_DAC_OF_RIGHT) + dev_warn(dev, "Right-channel DAC overflow has occurred\n"); + if (value & AIC31XX_DAC_OF_SHIFTER) + dev_warn(dev, "DAC barrel shifter overflow has occurred\n"); + if (value & AIC31XX_ADC_OF) + dev_warn(dev, "ADC overflow has occurred\n"); + if (value & AIC31XX_ADC_OF_SHIFTER) + dev_warn(dev, "ADC barrel shifter overflow has occurred\n"); + if (value & ~(AIC31XX_DAC_OF_LEFT | + AIC31XX_DAC_OF_RIGHT | + AIC31XX_DAC_OF_SHIFTER | + AIC31XX_ADC_OF | + AIC31XX_ADC_OF_SHIFTER)) + dev_warn(dev, "Unknown overflow interrupt flags: 0x%08x\n", value); + +exit: + if (handled) + return IRQ_HANDLED; + else + return IRQ_NONE; +} + static int aic31xx_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1413,6 +1477,7 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, return ret; } aic31xx->dev = &i2c->dev; + aic31xx->irq = i2c->irq; aic31xx->codec_type = id->driver_data; @@ -1456,6 +1521,26 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, return ret; } + if (aic31xx->irq > 0) { + regmap_update_bits(aic31xx->regmap, AIC31XX_GPIO1, + AIC31XX_GPIO1_FUNC_MASK, + AIC31XX_GPIO1_INT1 << + AIC31XX_GPIO1_FUNC_SHIFT); + + regmap_write(aic31xx->regmap, AIC31XX_INT1CTRL, + AIC31XX_SC | + AIC31XX_ENGINE); + + ret = devm_request_threaded_irq(aic31xx->dev, aic31xx->irq, + NULL, aic31xx_irq, + IRQF_ONESHOT, "aic31xx-irq", + aic31xx); + if (ret) { + dev_err(aic31xx->dev, "Unable to request IRQ\n"); + return ret; + } + } + if (aic31xx->codec_type & DAC31XX_BIT) return devm_snd_soc_register_component(&i2c->dev, &soc_codec_driver_aic31xx, diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 0b587585b38b..2636f2c6bc79 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -173,6 +173,13 @@ struct aic31xx_pdata { #define AIC31XX_HPRDRVPWRSTATUS_MASK BIT(1) #define AIC31XX_SPRDRVPWRSTATUS_MASK BIT(0) +/* AIC31XX_OFFLAG */ +#define AIC31XX_DAC_OF_LEFT BIT(7) +#define AIC31XX_DAC_OF_RIGHT BIT(6) +#define AIC31XX_DAC_OF_SHIFTER BIT(5) +#define AIC31XX_ADC_OF BIT(3) +#define AIC31XX_ADC_OF_SHIFTER BIT(1) + /* AIC31XX_INTRDACFLAG */ #define AIC31XX_HPLSCDETECT BIT(7) #define AIC31XX_HPRSCDETECT BIT(6) @@ -191,6 +198,22 @@ struct aic31xx_pdata { #define AIC31XX_SC BIT(3) #define AIC31XX_ENGINE BIT(2) +/* AIC31XX_GPIO1 */ +#define AIC31XX_GPIO1_FUNC_MASK GENMASK(5, 2) +#define AIC31XX_GPIO1_FUNC_SHIFT 2 +#define AIC31XX_GPIO1_DISABLED 0x00 +#define AIC31XX_GPIO1_INPUT 0x01 +#define AIC31XX_GPIO1_GPI 0x02 +#define AIC31XX_GPIO1_GPO 0x03 +#define AIC31XX_GPIO1_CLKOUT 0x04 +#define AIC31XX_GPIO1_INT1 0x05 +#define AIC31XX_GPIO1_INT2 0x06 +#define AIC31XX_GPIO1_ADC_WCLK 0x07 +#define AIC31XX_GPIO1_SBCLK 0x08 +#define AIC31XX_GPIO1_SWCLK 0x09 +#define AIC31XX_GPIO1_ADC_MOD_CLK 0x10 +#define AIC31XX_GPIO1_SDOUT 0x11 + /* AIC31XX_DACSETUP */ #define AIC31XX_SOFTSTEP_MASK GENMASK(1, 0) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index e2b5a11b16d1..f03195d2ab2e 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -822,6 +822,10 @@ static int aic32x4_set_bias_level(struct snd_soc_component *component, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: + /* Initial cold start */ + if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) + break; + /* Switch off BCLK_N Divider */ snd_soc_component_update_bits(component, AIC32X4_BCLKN, AIC32X4_BCLKEN, 0); diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6a271e6e6b8f..6aa0edf8c5ef 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1260,6 +1260,16 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, aic3x->master = 0; iface_areg &= ~(BIT_CLK_MASTER | WORD_CLK_MASTER); break; + case SND_SOC_DAIFMT_CBM_CFS: + aic3x->master = 1; + iface_areg |= BIT_CLK_MASTER; + iface_areg &= ~WORD_CLK_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFM: + aic3x->master = 1; + iface_areg |= WORD_CLK_MASTER; + iface_areg &= ~BIT_CLK_MASTER; + break; default: return -EINVAL; } diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index a957eaeb7bc1..32907b1e20cf 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -394,7 +394,7 @@ static int dac33_hard_power(struct snd_soc_component *component, int power) if (ret != 0) { dev_err(component->dev, "Failed to enable supplies: %d\n", ret); - goto exit; + goto exit; } if (dac33->power_gpio >= 0) diff --git a/sound/soc/codecs/tscs454.c b/sound/soc/codecs/tscs454.c index ff85a0bf6170..93d84e5ae2d5 100644 --- a/sound/soc/codecs/tscs454.c +++ b/sound/soc/codecs/tscs454.c @@ -3459,7 +3459,7 @@ static int tscs454_i2c_probe(struct i2c_client *i2c, /* Sync pg sel reg with cache */ regmap_write(tscs454->regmap, R_PAGESEL, 0x00); - ret = snd_soc_register_component(&i2c->dev, &soc_component_dev_tscs454, + ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_tscs454, tscs454_dais, ARRAY_SIZE(tscs454_dais)); if (ret) { dev_err(&i2c->dev, "Failed to register component (%d)\n", ret); diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index c5ae07234a00..bba330e30162 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -88,19 +88,6 @@ static int wm2000_write(struct i2c_client *i2c, unsigned int reg, return regmap_write(wm2000->regmap, reg, value); } -static unsigned int wm2000_read(struct i2c_client *i2c, unsigned int r) -{ - struct wm2000_priv *wm2000 = i2c_get_clientdata(i2c); - unsigned int val; - int ret; - - ret = regmap_read(wm2000->regmap, r, &val); - if (ret < 0) - return -1; - - return val; -} - static void wm2000_reset(struct wm2000_priv *wm2000) { struct i2c_client *i2c = wm2000->i2c; @@ -115,14 +102,15 @@ static void wm2000_reset(struct wm2000_priv *wm2000) static int wm2000_poll_bit(struct i2c_client *i2c, unsigned int reg, u8 mask) { + struct wm2000_priv *wm2000 = i2c_get_clientdata(i2c); int timeout = 4000; - int val; + unsigned int val; - val = wm2000_read(i2c, reg); + regmap_read(wm2000->regmap, reg, &val); while (!(val & mask) && --timeout) { msleep(1); - val = wm2000_read(i2c, reg); + regmap_read(wm2000->regmap, reg, &val); } if (timeout == 0) @@ -135,6 +123,7 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue) { struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); unsigned long rate; + unsigned int val; int ret; if (WARN_ON(wm2000->anc_mode != ANC_OFF)) @@ -213,12 +202,17 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue) WM2000_MODE_THERMAL_ENABLE); } - ret = wm2000_read(i2c, WM2000_REG_SPEECH_CLARITY); + ret = regmap_read(wm2000->regmap, WM2000_REG_SPEECH_CLARITY, &val); + if (ret != 0) { + dev_err(&i2c->dev, "Unable to read Speech Clarity: %d\n", ret); + regulator_bulk_disable(WM2000_NUM_SUPPLIES, wm2000->supplies); + return ret; + } if (wm2000->speech_clarity) - ret |= WM2000_SPEECH_CLARITY; + val |= WM2000_SPEECH_CLARITY; else - ret &= ~WM2000_SPEECH_CLARITY; - wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, ret); + val &= ~WM2000_SPEECH_CLARITY; + wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, val); wm2000_write(i2c, WM2000_REG_SYS_START0, 0x33); wm2000_write(i2c, WM2000_REG_SYS_START1, 0x02); @@ -824,7 +818,7 @@ static int wm2000_i2c_probe(struct i2c_client *i2c, const char *filename; const struct firmware *fw = NULL; int ret, i; - int reg; + unsigned int reg; u16 id; wm2000 = devm_kzalloc(&i2c->dev, sizeof(*wm2000), GFP_KERNEL); @@ -860,9 +854,17 @@ static int wm2000_i2c_probe(struct i2c_client *i2c, } /* Verify that this is a WM2000 */ - reg = wm2000_read(i2c, WM2000_REG_ID1); + ret = regmap_read(wm2000->regmap, WM2000_REG_ID1, ®); + if (ret != 0) { + dev_err(&i2c->dev, "Unable to read ID1: %d\n", ret); + return ret; + } id = reg << 8; - reg = wm2000_read(i2c, WM2000_REG_ID2); + ret = regmap_read(wm2000->regmap, WM2000_REG_ID2, ®); + if (ret != 0) { + dev_err(&i2c->dev, "Unable to read ID2: %d\n", ret); + return ret; + } id |= reg & 0xff; if (id != 0x2000) { @@ -871,7 +873,11 @@ static int wm2000_i2c_probe(struct i2c_client *i2c, goto err_supplies; } - reg = wm2000_read(i2c, WM2000_REG_REVISON); + ret = regmap_read(wm2000->regmap, WM2000_REG_REVISON, ®); + if (ret != 0) { + dev_err(&i2c->dev, "Unable to read Revision: %d\n", ret); + return ret; + } dev_info(&i2c->dev, "revision %c\n", reg + 'A'); wm2000->mclk = devm_clk_get(&i2c->dev, "MCLK"); diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c index 317db9a149a7..cf2cdbece122 100644 --- a/sound/soc/codecs/wm8782.c +++ b/sound/soc/codecs/wm8782.c @@ -20,6 +20,7 @@ #include <linux/module.h> #include <linux/kernel.h> #include <linux/device.h> +#include <linux/regulator/consumer.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/ac97_codec.h> @@ -50,7 +51,51 @@ static struct snd_soc_dai_driver wm8782_dai = { }, }; +/* regulator power supply names */ +static const char *supply_names[] = { + "Vdda", /* analog supply, 2.7V - 3.6V */ + "Vdd", /* digital supply, 2.7V - 5.5V */ +}; + +struct wm8782_priv { + struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; +}; + +static int wm8782_soc_probe(struct snd_soc_component *component) +{ + struct wm8782_priv *priv = snd_soc_component_get_drvdata(component); + return regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies); +} + +static void wm8782_soc_remove(struct snd_soc_component *component) +{ + struct wm8782_priv *priv = snd_soc_component_get_drvdata(component); + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); +} + +#ifdef CONFIG_PM +static int wm8782_soc_suspend(struct snd_soc_component *component) +{ + struct wm8782_priv *priv = snd_soc_component_get_drvdata(component); + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + return 0; +} + +static int wm8782_soc_resume(struct snd_soc_component *component) +{ + struct wm8782_priv *priv = snd_soc_component_get_drvdata(component); + return regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies); +} +#else +#define wm8782_soc_suspend NULL +#define wm8782_soc_resume NULL +#endif /* CONFIG_PM */ + static const struct snd_soc_component_driver soc_component_dev_wm8782 = { + .probe = wm8782_soc_probe, + .remove = wm8782_soc_remove, + .suspend = wm8782_soc_suspend, + .resume = wm8782_soc_resume, .dapm_widgets = wm8782_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm8782_dapm_widgets), .dapm_routes = wm8782_dapm_routes, @@ -63,6 +108,24 @@ static const struct snd_soc_component_driver soc_component_dev_wm8782 = { static int wm8782_probe(struct platform_device *pdev) { + struct device *dev = &pdev->dev; + struct wm8782_priv *priv; + int ret, i; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + dev_set_drvdata(dev, priv); + + for (i = 0; i < ARRAY_SIZE(supply_names); i++) + priv->supplies[i].supply = supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(priv->supplies), + priv->supplies); + if (ret < 0) + return ret; + return devm_snd_soc_register_component(&pdev->dev, &soc_component_dev_wm8782, &wm8782_dai, 1); } diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 1965635ec07c..2a3e5fbd04e4 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -13,7 +13,6 @@ #include <linux/clk.h> #include <linux/module.h> -#include <linux/moduleparam.h> #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 43edaf8cd276..593a11960888 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -11,7 +11,6 @@ */ #include <linux/module.h> -#include <linux/moduleparam.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index 61294c787f27..409bed30a4e4 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -60,7 +60,7 @@ static int wm8998_asrc_ev(struct snd_soc_dapm_widget *w, dev_warn(component->dev, "Unsupported ASRC rate1 (%s)\n", arizona_sample_rate_val_to_name(val)); - return -EINVAL; + return -EINVAL; } break; default: diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index ccdf088461b7..54c306707c02 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -325,8 +325,7 @@ static int wm9705_soc_probe(struct snd_soc_component *component) if (wm9705->mfd_pdata) { wm9705->ac97 = wm9705->mfd_pdata->ac97; regmap = wm9705->mfd_pdata->regmap; - } else { -#ifdef CONFIG_SND_SOC_AC97_BUS + } else if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS)) { wm9705->ac97 = snd_soc_new_ac97_component(component, WM9705_VENDOR_ID, WM9705_VENDOR_ID_MASK); if (IS_ERR(wm9705->ac97)) { @@ -339,7 +338,8 @@ static int wm9705_soc_probe(struct snd_soc_component *component) snd_soc_free_ac97_component(wm9705->ac97); return PTR_ERR(regmap); } -#endif + } else { + return -ENXIO; } snd_soc_component_set_drvdata(component, wm9705->ac97); @@ -350,14 +350,12 @@ static int wm9705_soc_probe(struct snd_soc_component *component) static void wm9705_soc_remove(struct snd_soc_component *component) { -#ifdef CONFIG_SND_SOC_AC97_BUS struct wm9705_priv *wm9705 = snd_soc_component_get_drvdata(component); - if (!wm9705->mfd_pdata) { + if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS) && !wm9705->mfd_pdata) { snd_soc_component_exit_regmap(component); snd_soc_free_ac97_component(wm9705->ac97); } -#endif } static const struct snd_soc_component_driver soc_component_dev_wm9705 = { diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index ade34c26ad2f..01949eaba4fd 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -638,13 +638,13 @@ static int wm9712_soc_probe(struct snd_soc_component *component) { struct wm9712_priv *wm9712 = snd_soc_component_get_drvdata(component); struct regmap *regmap; - int ret; if (wm9712->mfd_pdata) { wm9712->ac97 = wm9712->mfd_pdata->ac97; regmap = wm9712->mfd_pdata->regmap; - } else { -#ifdef CONFIG_SND_SOC_AC97_BUS + } else if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS)) { + int ret; + wm9712->ac97 = snd_soc_new_ac97_component(component, WM9712_VENDOR_ID, WM9712_VENDOR_ID_MASK); if (IS_ERR(wm9712->ac97)) { @@ -659,7 +659,8 @@ static int wm9712_soc_probe(struct snd_soc_component *component) snd_soc_free_ac97_component(wm9712->ac97); return PTR_ERR(regmap); } -#endif + } else { + return -ENXIO; } snd_soc_component_init_regmap(component, regmap); @@ -672,14 +673,12 @@ static int wm9712_soc_probe(struct snd_soc_component *component) static void wm9712_soc_remove(struct snd_soc_component *component) { -#ifdef CONFIG_SND_SOC_AC97_BUS struct wm9712_priv *wm9712 = snd_soc_component_get_drvdata(component); - if (!wm9712->mfd_pdata) { + if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS) && !wm9712->mfd_pdata) { snd_soc_component_exit_regmap(component); snd_soc_free_ac97_component(wm9712->ac97); } -#endif } static const struct snd_soc_component_driver soc_component_dev_wm9712 = { diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 643863bb32e0..5a2fdf4f69bf 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1214,8 +1214,7 @@ static int wm9713_soc_probe(struct snd_soc_component *component) if (wm9713->mfd_pdata) { wm9713->ac97 = wm9713->mfd_pdata->ac97; regmap = wm9713->mfd_pdata->regmap; - } else { -#ifdef CONFIG_SND_SOC_AC97_BUS + } else if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS)) { wm9713->ac97 = snd_soc_new_ac97_component(component, WM9713_VENDOR_ID, WM9713_VENDOR_ID_MASK); if (IS_ERR(wm9713->ac97)) @@ -1225,7 +1224,8 @@ static int wm9713_soc_probe(struct snd_soc_component *component) snd_soc_free_ac97_component(wm9713->ac97); return PTR_ERR(regmap); } -#endif + } else { + return -ENXIO; } snd_soc_component_init_regmap(component, regmap); @@ -1238,14 +1238,12 @@ static int wm9713_soc_probe(struct snd_soc_component *component) static void wm9713_soc_remove(struct snd_soc_component *component) { -#ifdef CONFIG_SND_SOC_AC97_BUS struct wm9713_priv *wm9713 = snd_soc_component_get_drvdata(component); - if (!wm9713->mfd_pdata) { + if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS) && !wm9713->mfd_pdata) { snd_soc_component_exit_regmap(component); snd_soc_free_ac97_component(wm9713->ac97); } -#endif } static const struct snd_soc_component_driver soc_component_dev_wm9713 = { diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f61656070225..1dd291cebe67 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -311,12 +311,12 @@ struct wm_adsp_alg_xm_struct { }; struct wm_adsp_buffer { - __be32 X_buf_base; /* XM base addr of first X area */ - __be32 X_buf_size; /* Size of 1st X area in words */ - __be32 X_buf_base2; /* XM base addr of 2nd X area */ - __be32 X_buf_brk; /* Total X size in words */ - __be32 Y_buf_base; /* YM base addr of Y area */ - __be32 wrap; /* Total size X and Y in words */ + __be32 buf1_base; /* Base addr of first buffer area */ + __be32 buf1_size; /* Size of buf1 area in DSP words */ + __be32 buf2_base; /* Base addr of 2nd buffer area */ + __be32 buf1_buf2_size; /* Size of buf1+buf2 in DSP words */ + __be32 buf3_base; /* Base addr of buf3 area */ + __be32 buf_total_size; /* Size of buf1+buf2+buf3 in DSP words */ __be32 high_water_mark; /* Point at which IRQ is asserted */ __be32 irq_count; /* bits 1-31 count IRQ assertions */ __be32 irq_ack; /* acked IRQ count, bit 0 enables IRQ */ @@ -393,18 +393,18 @@ struct wm_adsp_buffer_region_def { static const struct wm_adsp_buffer_region_def default_regions[] = { { .mem_type = WMFW_ADSP2_XM, - .base_offset = HOST_BUFFER_FIELD(X_buf_base), - .size_offset = HOST_BUFFER_FIELD(X_buf_size), + .base_offset = HOST_BUFFER_FIELD(buf1_base), + .size_offset = HOST_BUFFER_FIELD(buf1_size), }, { .mem_type = WMFW_ADSP2_XM, - .base_offset = HOST_BUFFER_FIELD(X_buf_base2), - .size_offset = HOST_BUFFER_FIELD(X_buf_brk), + .base_offset = HOST_BUFFER_FIELD(buf2_base), + .size_offset = HOST_BUFFER_FIELD(buf1_buf2_size), }, { .mem_type = WMFW_ADSP2_YM, - .base_offset = HOST_BUFFER_FIELD(Y_buf_base), - .size_offset = HOST_BUFFER_FIELD(wrap), + .base_offset = HOST_BUFFER_FIELD(buf3_base), + .size_offset = HOST_BUFFER_FIELD(buf_total_size), }, }; @@ -765,38 +765,41 @@ static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *mem, static void wm_adsp2_show_fw_status(struct wm_adsp *dsp) { - u16 scratch[4]; + unsigned int scratch[4]; + unsigned int addr = dsp->base + ADSP2_SCRATCH0; + unsigned int i; int ret; - ret = regmap_raw_read(dsp->regmap, dsp->base + ADSP2_SCRATCH0, - scratch, sizeof(scratch)); - if (ret) { - adsp_err(dsp, "Failed to read SCRATCH regs: %d\n", ret); - return; + for (i = 0; i < ARRAY_SIZE(scratch); ++i) { + ret = regmap_read(dsp->regmap, addr + i, &scratch[i]); + if (ret) { + adsp_err(dsp, "Failed to read SCRATCH%u: %d\n", i, ret); + return; + } } adsp_dbg(dsp, "FW SCRATCH 0:0x%x 1:0x%x 2:0x%x 3:0x%x\n", - be16_to_cpu(scratch[0]), - be16_to_cpu(scratch[1]), - be16_to_cpu(scratch[2]), - be16_to_cpu(scratch[3])); + scratch[0], scratch[1], scratch[2], scratch[3]); } static void wm_adsp2v2_show_fw_status(struct wm_adsp *dsp) { - u32 scratch[2]; + unsigned int scratch[2]; int ret; - ret = regmap_raw_read(dsp->regmap, dsp->base + ADSP2V2_SCRATCH0_1, - scratch, sizeof(scratch)); - + ret = regmap_read(dsp->regmap, dsp->base + ADSP2V2_SCRATCH0_1, + &scratch[0]); if (ret) { - adsp_err(dsp, "Failed to read SCRATCH regs: %d\n", ret); + adsp_err(dsp, "Failed to read SCRATCH0_1: %d\n", ret); return; } - scratch[0] = be32_to_cpu(scratch[0]); - scratch[1] = be32_to_cpu(scratch[1]); + ret = regmap_read(dsp->regmap, dsp->base + ADSP2V2_SCRATCH2_3, + &scratch[1]); + if (ret) { + adsp_err(dsp, "Failed to read SCRATCH2_3: %d\n", ret); + return; + } adsp_dbg(dsp, "FW SCRATCH 0:0x%x 1:0x%x 2:0x%x 3:0x%x\n", scratch[0] & 0xFFFF, @@ -2416,7 +2419,7 @@ static int wm_adsp_create_name(struct wm_adsp *dsp) return 0; } -int wm_adsp1_init(struct wm_adsp *dsp) +static int wm_adsp_common_init(struct wm_adsp *dsp) { int ret; @@ -2425,11 +2428,17 @@ int wm_adsp1_init(struct wm_adsp *dsp) return ret; INIT_LIST_HEAD(&dsp->alg_regions); + INIT_LIST_HEAD(&dsp->ctl_list); mutex_init(&dsp->pwr_lock); return 0; } + +int wm_adsp1_init(struct wm_adsp *dsp) +{ + return wm_adsp_common_init(dsp); +} EXPORT_SYMBOL_GPL(wm_adsp1_init); int wm_adsp1_event(struct snd_soc_dapm_widget *w, @@ -2914,7 +2923,7 @@ int wm_adsp2_init(struct wm_adsp *dsp) { int ret; - ret = wm_adsp_create_name(dsp); + ret = wm_adsp_common_init(dsp); if (ret) return ret; @@ -2936,12 +2945,8 @@ int wm_adsp2_init(struct wm_adsp *dsp) break; } - INIT_LIST_HEAD(&dsp->alg_regions); - INIT_LIST_HEAD(&dsp->ctl_list); INIT_WORK(&dsp->boot_work, wm_adsp2_boot_work); - mutex_init(&dsp->pwr_lock); - return 0; } EXPORT_SYMBOL_GPL(wm_adsp2_init); @@ -3345,7 +3350,7 @@ static int wm_adsp_buffer_populate(struct wm_adsp_compr_buf *buf) region->cumulative_size = offset; adsp_dbg(buf->dsp, - "region=%d type=%d base=%04x off=%04x size=%04x\n", + "region=%d type=%d base=%08x off=%08x size=%08x\n", i, region->mem_type, region->base_addr, region->offset, region->cumulative_size); } diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig deleted file mode 100644 index 778faff28e0e..000000000000 --- a/sound/soc/davinci/Kconfig +++ /dev/null @@ -1,106 +0,0 @@ -config SND_DAVINCI_SOC - tristate - depends on ARCH_DAVINCI - select SND_EDMA_SOC - -config SND_EDMA_SOC - tristate "SoC Audio for Texas Instruments chips using eDMA" - depends on TI_EDMA - select SND_SOC_GENERIC_DMAENGINE_PCM - help - Say Y or M here if you want audio support for TI SoC which uses eDMA. - The following line of SoCs are supported by this platform driver: - - daVinci devices - - AM335x - - AM437x/AM438x - - DRA7xx family - -config SND_DAVINCI_SOC_I2S - tristate "DaVinci Multichannel Buffered Serial Port (McBSP) support" - depends on SND_EDMA_SOC - help - Say Y or M here if you want to have support for McBSP IP found in - Texas Instruments DaVinci DA850 SoCs. - -config SND_DAVINCI_SOC_MCASP - tristate "Multichannel Audio Serial Port (McASP) support" - depends on SND_SDMA_SOC || SND_EDMA_SOC - help - Say Y or M here if you want to have support for McASP IP found in - various Texas Instruments SoCs like: - - daVinci devices - - Sitara line of SoCs (AM335x, AM438x, etc) - - DRA7x devices - -config SND_DAVINCI_SOC_VCIF - tristate - -config SND_DAVINCI_SOC_GENERIC_EVM - tristate - select SND_SOC_TLV320AIC3X - select SND_DAVINCI_SOC_MCASP - -config SND_AM33XX_SOC_EVM - tristate "SoC Audio for the AM33XX chip based boards" - depends on SND_EDMA_SOC && SOC_AM33XX && I2C - select SND_DAVINCI_SOC_GENERIC_EVM - help - Say Y or M if you want to add support for SoC audio on AM33XX - boards using McASP and TLV320AIC3X codec. For example AM335X-EVM, - AM335X-EVMSK, and BeagelBone with AudioCape boards have this - setup. - -config SND_DAVINCI_SOC_EVM - tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" - depends on SND_EDMA_SOC && I2C - depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM - select SND_DAVINCI_SOC_GENERIC_EVM - help - Say Y if you want to add support for SoC audio on TI - DaVinci DM6446, DM355 or DM365 EVM platforms. - -choice - prompt "DM365 codec select" - depends on SND_DAVINCI_SOC_EVM - depends on MACH_DAVINCI_DM365_EVM - -config SND_DM365_AIC3X_CODEC - tristate "Audio Codec - AIC3101" - help - Say Y if you want to add support for AIC3101 audio codec - -config SND_DM365_VOICE_CODEC - tristate "Voice Codec - CQ93VC" - select MFD_DAVINCI_VOICECODEC - select SND_DAVINCI_SOC_VCIF - select SND_SOC_CQ0093VC - help - Say Y if you want to add support for SoC On-chip voice codec -endchoice - -config SND_DM6467_SOC_EVM - tristate "SoC Audio support for DaVinci DM6467 EVM" - depends on SND_EDMA_SOC && MACH_DAVINCI_DM6467_EVM && I2C - select SND_DAVINCI_SOC_GENERIC_EVM - select SND_SOC_SPDIF - - help - Say Y if you want to add support for SoC audio on TI - -config SND_DA830_SOC_EVM - tristate "SoC Audio support for DA830/OMAP-L137 EVM" - depends on SND_EDMA_SOC && MACH_DAVINCI_DA830_EVM && I2C - select SND_DAVINCI_SOC_GENERIC_EVM - - help - Say Y if you want to add support for SoC audio on TI - DA830/OMAP-L137 EVM - -config SND_DA850_SOC_EVM - tristate "SoC Audio support for DA850/OMAP-L138 EVM" - depends on SND_EDMA_SOC && MACH_DAVINCI_DA850_EVM && I2C - select SND_DAVINCI_SOC_GENERIC_EVM - help - Say Y if you want to add support for SoC audio on TI - DA850/OMAP-L138 EVM - diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile deleted file mode 100644 index 23c6592eb31a..000000000000 --- a/sound/soc/davinci/Makefile +++ /dev/null @@ -1,16 +0,0 @@ -# SPDX-License-Identifier: GPL-2.0 -# DAVINCI Platform Support -snd-soc-edma-objs := edma-pcm.o -snd-soc-davinci-i2s-objs := davinci-i2s.o -snd-soc-davinci-mcasp-objs:= davinci-mcasp.o -snd-soc-davinci-vcif-objs:= davinci-vcif.o - -obj-$(CONFIG_SND_EDMA_SOC) += snd-soc-edma.o -obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o -obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o -obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o - -# Generic DAVINCI/AM33xx Machine Support -snd-soc-evm-objs := davinci-evm.o - -obj-$(CONFIG_SND_DAVINCI_SOC_GENERIC_EVM) += snd-soc-evm.o diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 6ec19fb4a934..2e75b5bc5f1d 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -221,7 +221,7 @@ config SND_SOC_PHYCORE_AC97 config SND_SOC_EUKREA_TLV320 tristate "Eukrea TLV320" - depends on ARCH_MXC && I2C + depends on ARCH_MXC && !ARM64 && I2C select SND_SOC_TLV320AIC23_I2C select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 44433b20435c..81f2fe2c6d23 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -571,17 +571,17 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) } /* Common settings for corresponding Freescale CPU DAI driver */ - if (strstr(cpu_np->name, "ssi")) { + if (of_node_name_eq(cpu_np, "ssi")) { /* Only SSI needs to configure AUDMUX */ ret = fsl_asoc_card_audmux_init(np, priv); if (ret) { dev_err(&pdev->dev, "failed to init audmux\n"); goto asrc_fail; } - } else if (strstr(cpu_np->name, "esai")) { + } else if (of_node_name_eq(cpu_np, "esai")) { priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL; priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL; - } else if (strstr(cpu_np->name, "sai")) { + } else if (of_node_name_eq(cpu_np, "sai")) { priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1; priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1; } diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 1033ac6631b0..01052a0808b0 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -151,7 +151,7 @@ static int fsl_asrc_dma_hw_params(struct snd_pcm_substream *substream, int ret; /* Fetch the Back-End dma_data from DPCM */ - list_for_each_entry(dpcm, &rtd->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(rtd, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *substream_be; struct snd_soc_dai *dai = be->cpu_dai; diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index c1d1d06783e5..57b484768a58 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -807,7 +807,7 @@ static int fsl_esai_probe(struct platform_device *pdev) return -ENOMEM; esai_priv->pdev = pdev; - strncpy(esai_priv->name, np->name, sizeof(esai_priv->name) - 1); + snprintf(esai_priv->name, sizeof(esai_priv->name), "%pOFn", np); /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); diff --git a/sound/soc/fsl/fsl_ssi_dbg.c b/sound/soc/fsl/fsl_ssi_dbg.c index 1255dfe19eef..6f6294149476 100644 --- a/sound/soc/fsl/fsl_ssi_dbg.c +++ b/sound/soc/fsl/fsl_ssi_dbg.c @@ -124,17 +124,7 @@ static int fsl_ssi_stats_show(struct seq_file *s, void *unused) return 0; } -static int fsl_ssi_stats_open(struct inode *inode, struct file *file) -{ - return single_open(file, fsl_ssi_stats_show, inode->i_private); -} - -static const struct file_operations fsl_ssi_stats_ops = { - .open = fsl_ssi_stats_open, - .read = seq_read, - .llseek = seq_lseek, - .release = single_release, -}; +DEFINE_SHOW_ATTRIBUTE(fsl_ssi_stats); int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, struct device *dev) { @@ -144,7 +134,7 @@ int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, struct device *dev) ssi_dbg->dbg_stats = debugfs_create_file("stats", 0444, ssi_dbg->dbg_dir, ssi_dbg, - &fsl_ssi_stats_ops); + &fsl_ssi_stats_fops); if (!ssi_dbg->dbg_stats) { debugfs_remove(ssi_dbg->dbg_dir); return -ENOMEM; diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c index 7f0fa4b52223..9981668ab590 100644 --- a/sound/soc/fsl/fsl_utils.c +++ b/sound/soc/fsl/fsl_utils.c @@ -57,8 +57,8 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np, of_node_put(dma_channel_np); return ret; } - snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s", - (unsigned long long) res.start, dma_channel_np->name); + snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%pOFn", + (unsigned long long) res.start, dma_channel_np); iprop = of_get_property(dma_channel_np, "cell-index", NULL); if (!iprop) { diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 392d5eef356d..99e07b01a2ce 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -86,49 +86,49 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, if (!buf) return -ENOMEM; - ret = snprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n", + ret = scnprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n", pdcr, ptcr); if (ptcr & IMX_AUDMUX_V2_PTCR_TFSDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "TxFS output from %s, ", audmux_port_string((ptcr >> 27) & 0x7)); else - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "TxFS input, "); if (ptcr & IMX_AUDMUX_V2_PTCR_TCLKDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "TxClk output from %s", audmux_port_string((ptcr >> 22) & 0x7)); else - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "TxClk input"); - ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n"); + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\n"); if (ptcr & IMX_AUDMUX_V2_PTCR_SYN) { - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "Port is symmetric"); } else { if (ptcr & IMX_AUDMUX_V2_PTCR_RFSDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "RxFS output from %s, ", audmux_port_string((ptcr >> 17) & 0x7)); else - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "RxFS input, "); if (ptcr & IMX_AUDMUX_V2_PTCR_RCLKDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "RxClk output from %s", audmux_port_string((ptcr >> 12) & 0x7)); else - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "RxClk input"); } - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\nData received from %s\n", audmux_port_string((pdcr >> 13) & 0x7)); diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index ec731223cab3..e339f36cea95 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -57,6 +57,7 @@ static int pcm030_fabric_probe(struct platform_device *op) struct device_node *platform_np; struct snd_soc_card *card = &pcm030_card; struct pcm030_audio_data *pdata; + struct snd_soc_dai_link *dai_link; int ret; int i; @@ -78,8 +79,8 @@ static int pcm030_fabric_probe(struct platform_device *op) return -ENODEV; } - for (i = 0; i < card->num_links; i++) - card->dai_link[i].platform_of_node = platform_np; + for_each_card_prelinks(card, i, dai_link) + dai_link->platform_of_node = platform_np; ret = request_module("snd-soc-wm9712"); if (ret) diff --git a/sound/soc/generic/Kconfig b/sound/soc/generic/Kconfig index c954be0a0f96..92c2cf06f40a 100644 --- a/sound/soc/generic/Kconfig +++ b/sound/soc/generic/Kconfig @@ -6,6 +6,7 @@ config SND_SIMPLE_CARD select SND_SIMPLE_CARD_UTILS help This option enables generic simple sound card support + It also support DPCM of multi CPU single Codec ststem. config SND_SIMPLE_SCU_CARD tristate "ASoC Simple SCU sound card support" @@ -20,8 +21,9 @@ config SND_AUDIO_GRAPH_CARD depends on OF select SND_SIMPLE_CARD_UTILS help - This option enables generic simple simple sound card support + This option enables generic simple sound card support with OF-graph DT bindings. + It also support DPCM of multi CPU single Codec ststem. config SND_AUDIO_GRAPH_SCU_CARD tristate "ASoC Audio Graph SCU sound card support" diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 2094d2c8919f..0d6144560a1e 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -23,17 +23,29 @@ struct graph_card_data { struct snd_soc_card snd_card; struct graph_dai_props { - struct asoc_simple_dai cpu_dai; - struct asoc_simple_dai codec_dai; + struct asoc_simple_dai *cpu_dai; + struct asoc_simple_dai *codec_dai; + struct snd_soc_dai_link_component codecs; /* single codec */ + struct snd_soc_dai_link_component platform; + struct asoc_simple_card_data adata; + struct snd_soc_codec_conf *codec_conf; unsigned int mclk_fs; } *dai_props; - unsigned int mclk_fs; struct asoc_simple_jack hp_jack; struct asoc_simple_jack mic_jack; struct snd_soc_dai_link *dai_link; + struct asoc_simple_dai *dais; + struct snd_soc_codec_conf *codec_conf; struct gpio_desc *pa_gpio; }; +#define graph_priv_to_card(priv) (&(priv)->snd_card) +#define graph_priv_to_props(priv, i) ((priv)->dai_props + (i)) +#define graph_priv_to_dev(priv) (graph_priv_to_card(priv)->dev) +#define graph_priv_to_link(priv, i) (graph_priv_to_card(priv)->dai_link + (i)) + +#define PREFIX "audio-graph-card," + static int asoc_graph_card_outdrv_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) @@ -61,11 +73,6 @@ static const struct snd_soc_dapm_widget asoc_graph_card_dapm_widgets[] = { SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), }; -#define graph_priv_to_card(priv) (&(priv)->snd_card) -#define graph_priv_to_props(priv, i) ((priv)->dai_props + (i)) -#define graph_priv_to_dev(priv) (graph_priv_to_card(priv)->dev) -#define graph_priv_to_link(priv, i) (graph_priv_to_card(priv)->dai_link + (i)) - static int asoc_graph_card_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -73,13 +80,13 @@ static int asoc_graph_card_startup(struct snd_pcm_substream *substream) struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); int ret; - ret = asoc_simple_card_clk_enable(&dai_props->cpu_dai); + ret = asoc_simple_card_clk_enable(dai_props->cpu_dai); if (ret) return ret; - ret = asoc_simple_card_clk_enable(&dai_props->codec_dai); + ret = asoc_simple_card_clk_enable(dai_props->codec_dai); if (ret) - asoc_simple_card_clk_disable(&dai_props->cpu_dai); + asoc_simple_card_clk_disable(dai_props->cpu_dai); return ret; } @@ -90,9 +97,9 @@ static void asoc_graph_card_shutdown(struct snd_pcm_substream *substream) struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); - asoc_simple_card_clk_disable(&dai_props->cpu_dai); + asoc_simple_card_clk_disable(dai_props->cpu_dai); - asoc_simple_card_clk_disable(&dai_props->codec_dai); + asoc_simple_card_clk_disable(dai_props->codec_dai); } static int asoc_graph_card_hw_params(struct snd_pcm_substream *substream, @@ -106,9 +113,7 @@ static int asoc_graph_card_hw_params(struct snd_pcm_substream *substream, unsigned int mclk, mclk_fs = 0; int ret = 0; - if (priv->mclk_fs) - mclk_fs = priv->mclk_fs; - else if (dai_props->mclk_fs) + if (dai_props->mclk_fs) mclk_fs = dai_props->mclk_fs; if (mclk_fs) { @@ -137,85 +142,238 @@ static const struct snd_soc_ops asoc_graph_card_ops = { static int asoc_graph_card_dai_init(struct snd_soc_pcm_runtime *rtd) { struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *codec = rtd->codec_dai; - struct snd_soc_dai *cpu = rtd->cpu_dai; - struct graph_dai_props *dai_props = - graph_priv_to_props(priv, rtd->num); - int ret; + struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); + int ret = 0; - ret = asoc_simple_card_init_dai(codec, &dai_props->codec_dai); + ret = asoc_simple_card_init_dai(rtd->codec_dai, + dai_props->codec_dai); if (ret < 0) return ret; - ret = asoc_simple_card_init_dai(cpu, &dai_props->cpu_dai); + ret = asoc_simple_card_init_dai(rtd->cpu_dai, + dai_props->cpu_dai); if (ret < 0) return ret; return 0; } -static int asoc_graph_card_dai_link_of(struct device_node *cpu_port, - struct graph_card_data *priv, - int idx) +static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); + struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); + + asoc_simple_card_convert_fixup(&dai_props->adata, params); + + return 0; +} + +static int asoc_graph_card_dai_link_of_dpcm(struct device_node *top, + struct device_node *cpu_ep, + struct device_node *codec_ep, + struct graph_card_data *priv, + int *dai_idx, int link_idx, + int *conf_idx, int is_cpu) { struct device *dev = graph_priv_to_dev(priv); - struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, idx); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, idx); - struct asoc_simple_dai *cpu_dai = &dai_props->cpu_dai; - struct asoc_simple_dai *codec_dai = &dai_props->codec_dai; - struct device_node *cpu_ep = of_get_next_child(cpu_port, NULL); - struct device_node *codec_ep = of_graph_get_remote_endpoint(cpu_ep); - struct device_node *rcpu_ep = of_graph_get_remote_endpoint(codec_ep); + struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, link_idx); + struct graph_dai_props *dai_props = graph_priv_to_props(priv, link_idx); + struct device_node *ep = is_cpu ? cpu_ep : codec_ep; + struct device_node *port = of_get_parent(ep); + struct device_node *ports = of_get_parent(port); + struct device_node *node = of_graph_get_port_parent(ep); + struct asoc_simple_dai *dai; + struct snd_soc_dai_link_component *codecs = dai_link->codecs; int ret; - if (rcpu_ep != cpu_ep) { - dev_err(dev, "remote-endpoint mismatch (%s/%s/%s)\n", - cpu_ep->name, codec_ep->name, rcpu_ep->name); - ret = -EINVAL; - goto dai_link_of_err; + dev_dbg(dev, "link_of DPCM (for %s)\n", is_cpu ? "CPU" : "Codec"); + + of_property_read_u32(top, "mclk-fs", &dai_props->mclk_fs); + of_property_read_u32(ports, "mclk-fs", &dai_props->mclk_fs); + of_property_read_u32(port, "mclk-fs", &dai_props->mclk_fs); + of_property_read_u32(ep, "mclk-fs", &dai_props->mclk_fs); + + asoc_simple_card_parse_convert(dev, top, NULL, &dai_props->adata); + asoc_simple_card_parse_convert(dev, node, PREFIX, &dai_props->adata); + asoc_simple_card_parse_convert(dev, ports, NULL, &dai_props->adata); + asoc_simple_card_parse_convert(dev, port, NULL, &dai_props->adata); + asoc_simple_card_parse_convert(dev, ep, NULL, &dai_props->adata); + + of_node_put(ports); + of_node_put(port); + + if (is_cpu) { + + /* BE is dummy */ + codecs->of_node = NULL; + codecs->dai_name = "snd-soc-dummy-dai"; + codecs->name = "snd-soc-dummy"; + + /* FE settings */ + dai_link->dynamic = 1; + dai_link->dpcm_merged_format = 1; + + dai = + dai_props->cpu_dai = &priv->dais[(*dai_idx)++]; + + ret = asoc_simple_card_parse_graph_cpu(ep, dai_link); + if (ret) + return ret; + + ret = asoc_simple_card_parse_clk_cpu(dev, ep, dai_link, dai); + if (ret < 0) + return ret; + + ret = asoc_simple_card_set_dailink_name(dev, dai_link, + "fe.%s", + dai_link->cpu_dai_name); + if (ret < 0) + return ret; + + /* card->num_links includes Codec */ + asoc_simple_card_canonicalize_cpu(dai_link, + of_graph_get_endpoint_count(dai_link->cpu_of_node) == 1); + } else { + struct snd_soc_codec_conf *cconf; + + /* FE is dummy */ + dai_link->cpu_of_node = NULL; + dai_link->cpu_dai_name = "snd-soc-dummy-dai"; + dai_link->cpu_name = "snd-soc-dummy"; + + /* BE settings */ + dai_link->no_pcm = 1; + dai_link->be_hw_params_fixup = asoc_graph_card_be_hw_params_fixup; + + dai = + dai_props->codec_dai = &priv->dais[(*dai_idx)++]; + + cconf = + dai_props->codec_conf = &priv->codec_conf[(*conf_idx)++]; + + ret = asoc_simple_card_parse_graph_codec(ep, dai_link); + if (ret < 0) + return ret; + + ret = asoc_simple_card_parse_clk_codec(dev, ep, dai_link, dai); + if (ret < 0) + return ret; + + ret = asoc_simple_card_set_dailink_name(dev, dai_link, + "be.%s", + codecs->dai_name); + if (ret < 0) + return ret; + + /* check "prefix" from top node */ + snd_soc_of_parse_node_prefix(top, cconf, codecs->of_node, + "prefix"); + snd_soc_of_parse_node_prefix(node, cconf, codecs->of_node, + PREFIX "prefix"); + snd_soc_of_parse_node_prefix(ports, cconf, codecs->of_node, + "prefix"); + snd_soc_of_parse_node_prefix(port, cconf, codecs->of_node, + "prefix"); } + ret = asoc_simple_card_of_parse_tdm(ep, dai); + if (ret) + return ret; + + ret = asoc_simple_card_canonicalize_dailink(dai_link); + if (ret < 0) + return ret; + ret = asoc_simple_card_parse_daifmt(dev, cpu_ep, codec_ep, NULL, &dai_link->dai_fmt); if (ret < 0) - goto dai_link_of_err; + return ret; + + dai_link->dpcm_playback = 1; + dai_link->dpcm_capture = 1; + dai_link->ops = &asoc_graph_card_ops; + dai_link->init = asoc_graph_card_dai_init; + + return 0; +} + +static int asoc_graph_card_dai_link_of(struct device_node *top, + struct device_node *cpu_ep, + struct device_node *codec_ep, + struct graph_card_data *priv, + int *dai_idx, int link_idx) +{ + struct device *dev = graph_priv_to_dev(priv); + struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, link_idx); + struct graph_dai_props *dai_props = graph_priv_to_props(priv, link_idx); + struct device_node *cpu_port = of_get_parent(cpu_ep); + struct device_node *codec_port = of_get_parent(codec_ep); + struct device_node *cpu_ports = of_get_parent(cpu_port); + struct device_node *codec_ports = of_get_parent(codec_port); + struct asoc_simple_dai *cpu_dai; + struct asoc_simple_dai *codec_dai; + int ret; + + dev_dbg(dev, "link_of\n"); + + cpu_dai = + dai_props->cpu_dai = &priv->dais[(*dai_idx)++]; + codec_dai = + dai_props->codec_dai = &priv->dais[(*dai_idx)++]; - of_property_read_u32(rcpu_ep, "mclk-fs", &dai_props->mclk_fs); + /* Factor to mclk, used in hw_params() */ + of_property_read_u32(top, "mclk-fs", &dai_props->mclk_fs); + of_property_read_u32(cpu_ports, "mclk-fs", &dai_props->mclk_fs); + of_property_read_u32(codec_ports, "mclk-fs", &dai_props->mclk_fs); + of_property_read_u32(cpu_port, "mclk-fs", &dai_props->mclk_fs); + of_property_read_u32(codec_port, "mclk-fs", &dai_props->mclk_fs); + of_property_read_u32(cpu_ep, "mclk-fs", &dai_props->mclk_fs); + of_property_read_u32(codec_ep, "mclk-fs", &dai_props->mclk_fs); + of_node_put(cpu_port); + of_node_put(cpu_ports); + of_node_put(codec_port); + of_node_put(codec_ports); + + ret = asoc_simple_card_parse_daifmt(dev, cpu_ep, codec_ep, + NULL, &dai_link->dai_fmt); + if (ret < 0) + return ret; ret = asoc_simple_card_parse_graph_cpu(cpu_ep, dai_link); if (ret < 0) - goto dai_link_of_err; + return ret; ret = asoc_simple_card_parse_graph_codec(codec_ep, dai_link); if (ret < 0) - goto dai_link_of_err; + return ret; ret = asoc_simple_card_of_parse_tdm(cpu_ep, cpu_dai); if (ret < 0) - goto dai_link_of_err; + return ret; ret = asoc_simple_card_of_parse_tdm(codec_ep, codec_dai); if (ret < 0) - goto dai_link_of_err; + return ret; ret = asoc_simple_card_parse_clk_cpu(dev, cpu_ep, dai_link, cpu_dai); if (ret < 0) - goto dai_link_of_err; + return ret; ret = asoc_simple_card_parse_clk_codec(dev, codec_ep, dai_link, codec_dai); if (ret < 0) - goto dai_link_of_err; + return ret; ret = asoc_simple_card_canonicalize_dailink(dai_link); if (ret < 0) - goto dai_link_of_err; + return ret; ret = asoc_simple_card_set_dailink_name(dev, dai_link, "%s-%s", dai_link->cpu_dai_name, - dai_link->codec_dai_name); + dai_link->codecs->dai_name); if (ret < 0) - goto dai_link_of_err; + return ret; dai_link->ops = &asoc_graph_card_ops; dai_link->init = asoc_graph_card_dai_init; @@ -223,12 +381,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_port, asoc_simple_card_canonicalize_cpu(dai_link, of_graph_get_endpoint_count(dai_link->cpu_of_node) == 1); -dai_link_of_err: - of_node_put(cpu_ep); - of_node_put(rcpu_ep); - of_node_put(codec_ep); - - return ret; + return 0; } static int asoc_graph_card_parse_of(struct graph_card_data *priv) @@ -236,44 +389,173 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) struct of_phandle_iterator it; struct device *dev = graph_priv_to_dev(priv); struct snd_soc_card *card = graph_priv_to_card(priv); - struct device_node *node = dev->of_node; - int rc, idx = 0; - int ret; + struct device_node *top = dev->of_node; + struct device_node *node = top; + struct device_node *cpu_port; + struct device_node *cpu_ep = NULL; + struct device_node *codec_ep = NULL; + struct device_node *codec_port = NULL; + struct device_node *codec_port_old = NULL; + int rc, ret; + int link_idx, dai_idx, conf_idx; + int cpu; ret = asoc_simple_card_of_parse_widgets(card, NULL); if (ret < 0) return ret; - ret = asoc_simple_card_of_parse_routing(card, NULL, 1); + ret = asoc_simple_card_of_parse_routing(card, NULL); if (ret < 0) return ret; - /* Factor to mclk, used in hw_params() */ - of_property_read_u32(node, "mclk-fs", &priv->mclk_fs); - - of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { - ret = asoc_graph_card_dai_link_of(it.node, priv, idx++); - if (ret < 0) { - of_node_put(it.node); - - return ret; + link_idx = 0; + dai_idx = 0; + conf_idx = 0; + codec_port_old = NULL; + for (cpu = 1; cpu >= 0; cpu--) { + /* + * Detect all CPU first, and Detect all Codec 2nd. + * + * In Normal sound case, all DAIs are detected + * as "CPU-Codec". + * + * In DPCM sound case, + * all CPUs are detected as "CPU-dummy", and + * all Codecs are detected as "dummy-Codec". + * To avoid random sub-device numbering, + * detect "dummy-Codec" in last; + */ + of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { + cpu_port = it.node; + cpu_ep = NULL; + while (1) { + cpu_ep = of_get_next_child(cpu_port, cpu_ep); + if (!cpu_ep) + break; + + codec_ep = of_graph_get_remote_endpoint(cpu_ep); + codec_port = of_get_parent(codec_ep); + + of_node_put(codec_ep); + of_node_put(codec_port); + + dev_dbg(dev, "%pOFf <-> %pOFf\n", cpu_ep, codec_ep); + + if (of_get_child_count(codec_port) > 1) { + /* + * for DPCM sound + */ + if (!cpu) { + if (codec_port_old == codec_port) + continue; + codec_port_old = codec_port; + } + ret = asoc_graph_card_dai_link_of_dpcm( + top, cpu_ep, codec_ep, priv, + &dai_idx, link_idx++, + &conf_idx, cpu); + } else if (cpu) { + /* + * for Normal sound + */ + ret = asoc_graph_card_dai_link_of( + top, cpu_ep, codec_ep, priv, + &dai_idx, link_idx++); + } + if (ret < 0) + return ret; + } } } return asoc_simple_card_parse_card_name(card, NULL); } -static int asoc_graph_get_dais_count(struct device *dev) +static void asoc_graph_get_dais_count(struct device *dev, + int *link_num, + int *dais_num, + int *ccnf_num) { struct of_phandle_iterator it; struct device_node *node = dev->of_node; - int count = 0; + struct device_node *cpu_port; + struct device_node *cpu_ep; + struct device_node *codec_ep; + struct device_node *codec_port; + struct device_node *codec_port_old; + struct device_node *codec_port_old2; int rc; - of_for_each_phandle(&it, rc, node, "dais", NULL, 0) - count++; - - return count; + /* + * link_num : number of links. + * CPU-Codec / CPU-dummy / dummy-Codec + * dais_num : number of DAIs + * ccnf_num : number of codec_conf + * same number for "dummy-Codec" + * + * ex1) + * CPU0 --- Codec0 link : 5 + * CPU1 --- Codec1 dais : 7 + * CPU2 -/ ccnf : 1 + * CPU3 --- Codec2 + * + * => 5 links = 2xCPU-Codec + 2xCPU-dummy + 1xdummy-Codec + * => 7 DAIs = 4xCPU + 3xCodec + * => 1 ccnf = 1xdummy-Codec + * + * ex2) + * CPU0 --- Codec0 link : 5 + * CPU1 --- Codec1 dais : 6 + * CPU2 -/ ccnf : 1 + * CPU3 -/ + * + * => 5 links = 1xCPU-Codec + 3xCPU-dummy + 1xdummy-Codec + * => 6 DAIs = 4xCPU + 2xCodec + * => 1 ccnf = 1xdummy-Codec + * + * ex3) + * CPU0 --- Codec0 link : 6 + * CPU1 -/ dais : 6 + * CPU2 --- Codec1 ccnf : 2 + * CPU3 -/ + * + * => 6 links = 0xCPU-Codec + 4xCPU-dummy + 2xdummy-Codec + * => 6 DAIs = 4xCPU + 2xCodec + * => 2 ccnf = 2xdummy-Codec + */ + codec_port_old = NULL; + codec_port_old2 = NULL; + of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { + cpu_port = it.node; + cpu_ep = NULL; + while (1) { + cpu_ep = of_get_next_child(cpu_port, cpu_ep); + if (!cpu_ep) + break; + + codec_ep = of_graph_get_remote_endpoint(cpu_ep); + codec_port = of_get_parent(codec_ep); + + of_node_put(codec_ep); + of_node_put(codec_port); + + (*link_num)++; + (*dais_num)++; + + if (codec_port_old == codec_port) { + if (codec_port_old2 != codec_port_old) { + (*link_num)++; + (*ccnf_num)++; + } + + codec_port_old2 = codec_port_old; + continue; + } + + (*dais_num)++; + codec_port_old = codec_port; + } + } } static int asoc_graph_soc_card_probe(struct snd_soc_card *card) @@ -297,24 +579,41 @@ static int asoc_graph_card_probe(struct platform_device *pdev) struct graph_card_data *priv; struct snd_soc_dai_link *dai_link; struct graph_dai_props *dai_props; + struct asoc_simple_dai *dais; struct device *dev = &pdev->dev; struct snd_soc_card *card; - int num, ret; + struct snd_soc_codec_conf *cconf; + int lnum = 0, dnum = 0, cnum = 0; + int ret, i; /* Allocate the private data and the DAI link array */ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) return -ENOMEM; - num = asoc_graph_get_dais_count(dev); - if (num == 0) + asoc_graph_get_dais_count(dev, &lnum, &dnum, &cnum); + if (!lnum || !dnum) return -EINVAL; - dai_props = devm_kcalloc(dev, num, sizeof(*dai_props), GFP_KERNEL); - dai_link = devm_kcalloc(dev, num, sizeof(*dai_link), GFP_KERNEL); - if (!dai_props || !dai_link) + dai_props = devm_kcalloc(dev, lnum, sizeof(*dai_props), GFP_KERNEL); + dai_link = devm_kcalloc(dev, lnum, sizeof(*dai_link), GFP_KERNEL); + dais = devm_kcalloc(dev, dnum, sizeof(*dais), GFP_KERNEL); + cconf = devm_kcalloc(dev, cnum, sizeof(*cconf), GFP_KERNEL); + if (!dai_props || !dai_link || !dais) return -ENOMEM; + /* + * Use snd_soc_dai_link_component instead of legacy style + * It is codec only. but cpu/platform will be supported in the future. + * see + * soc-core.c :: snd_soc_init_multicodec() + */ + for (i = 0; i < lnum; i++) { + dai_link[i].codecs = &dai_props[i].codecs; + dai_link[i].num_codecs = 1; + dai_link[i].platform = &dai_props[i].platform; + } + priv->pa_gpio = devm_gpiod_get_optional(dev, "pa", GPIOD_OUT_LOW); if (IS_ERR(priv->pa_gpio)) { ret = PTR_ERR(priv->pa_gpio); @@ -324,16 +623,20 @@ static int asoc_graph_card_probe(struct platform_device *pdev) priv->dai_props = dai_props; priv->dai_link = dai_link; + priv->dais = dais; + priv->codec_conf = cconf; /* Init snd_soc_card */ card = graph_priv_to_card(priv); - card->owner = THIS_MODULE; - card->dev = dev; - card->dai_link = dai_link; - card->num_links = num; - card->dapm_widgets = asoc_graph_card_dapm_widgets; - card->num_dapm_widgets = ARRAY_SIZE(asoc_graph_card_dapm_widgets); - card->probe = asoc_graph_soc_card_probe; + card->owner = THIS_MODULE; + card->dev = dev; + card->dai_link = dai_link; + card->num_links = lnum; + card->dapm_widgets = asoc_graph_card_dapm_widgets; + card->num_dapm_widgets = ARRAY_SIZE(asoc_graph_card_dapm_widgets); + card->probe = asoc_graph_soc_card_probe; + card->codec_conf = cconf; + card->num_configs = cnum; ret = asoc_graph_card_parse_of(priv); if (ret < 0) { @@ -364,6 +667,7 @@ static int asoc_graph_card_remove(struct platform_device *pdev) static const struct of_device_id asoc_graph_of_match[] = { { .compatible = "audio-graph-card", }, + { .compatible = "audio-graph-scu-card", }, {}, }; MODULE_DEVICE_TABLE(of, asoc_graph_of_match); diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index 92882e392d6c..e1b192ea147b 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -24,10 +24,18 @@ struct graph_card_data { struct snd_soc_card snd_card; - struct snd_soc_codec_conf codec_conf; - struct asoc_simple_dai *dai_props; + struct graph_dai_props { + struct asoc_simple_dai *cpu_dai; + struct asoc_simple_dai *codec_dai; + struct snd_soc_dai_link_component codecs; + struct snd_soc_dai_link_component platform; + struct asoc_simple_card_data adata; + struct snd_soc_codec_conf *codec_conf; + } *dai_props; struct snd_soc_dai_link *dai_link; + struct asoc_simple_dai *dais; struct asoc_simple_card_data adata; + struct snd_soc_codec_conf *codec_conf; }; #define graph_priv_to_card(priv) (&(priv)->snd_card) @@ -35,22 +43,35 @@ struct graph_card_data { #define graph_priv_to_dev(priv) (graph_priv_to_card(priv)->dev) #define graph_priv_to_link(priv, i) (graph_priv_to_card(priv)->dai_link + (i)) +#define PREFIX "audio-graph-card," + static int asoc_graph_card_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct asoc_simple_dai *dai_props = graph_priv_to_props(priv, rtd->num); + struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); + int ret = 0; + + ret = asoc_simple_card_clk_enable(dai_props->cpu_dai); + if (ret) + return ret; - return asoc_simple_card_clk_enable(dai_props); + ret = asoc_simple_card_clk_enable(dai_props->codec_dai); + if (ret) + asoc_simple_card_clk_disable(dai_props->cpu_dai); + + return ret; } static void asoc_graph_card_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct asoc_simple_dai *dai_props = graph_priv_to_props(priv, rtd->num); + struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); + + asoc_simple_card_clk_disable(dai_props->cpu_dai); - asoc_simple_card_clk_disable(dai_props); + asoc_simple_card_clk_disable(dai_props->codec_dai); } static const struct snd_soc_ops asoc_graph_card_ops = { @@ -61,56 +82,72 @@ static const struct snd_soc_ops asoc_graph_card_ops = { static int asoc_graph_card_dai_init(struct snd_soc_pcm_runtime *rtd) { struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai; - struct snd_soc_dai_link *dai_link; - struct asoc_simple_dai *dai_props; - int num = rtd->num; + struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); + int ret = 0; - dai_link = graph_priv_to_link(priv, num); - dai_props = graph_priv_to_props(priv, num); - dai = dai_link->dynamic ? - rtd->cpu_dai : - rtd->codec_dai; + ret = asoc_simple_card_init_dai(rtd->codec_dai, + dai_props->codec_dai); + if (ret < 0) + return ret; - return asoc_simple_card_init_dai(dai, dai_props); + ret = asoc_simple_card_init_dai(rtd->cpu_dai, + dai_props->cpu_dai); + if (ret < 0) + return ret; + + return 0; } static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); + struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); + asoc_simple_card_convert_fixup(&dai_props->adata, params); + + /* overwrite by top level adata if exist */ asoc_simple_card_convert_fixup(&priv->adata, params); return 0; } -static int asoc_graph_card_dai_link_of(struct device_node *ep, +static int asoc_graph_card_dai_link_of(struct device_node *cpu_ep, + struct device_node *codec_ep, struct graph_card_data *priv, - unsigned int daifmt, - int idx, int is_fe) + int *dai_idx, int link_idx, + int *conf_idx, int is_fe) { struct device *dev = graph_priv_to_dev(priv); - struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, idx); - struct asoc_simple_dai *dai_props = graph_priv_to_props(priv, idx); + struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, link_idx); + struct graph_dai_props *dai_props = graph_priv_to_props(priv, link_idx); struct snd_soc_card *card = graph_priv_to_card(priv); + struct device_node *ep = is_fe ? cpu_ep : codec_ep; + struct device_node *node = of_graph_get_port_parent(ep); + struct asoc_simple_dai *dai; int ret; if (is_fe) { + struct snd_soc_dai_link_component *codecs; + /* BE is dummy */ - dai_link->codec_of_node = NULL; - dai_link->codec_dai_name = "snd-soc-dummy-dai"; - dai_link->codec_name = "snd-soc-dummy"; + codecs = dai_link->codecs; + codecs->of_node = NULL; + codecs->dai_name = "snd-soc-dummy-dai"; + codecs->name = "snd-soc-dummy"; /* FE settings */ dai_link->dynamic = 1; dai_link->dpcm_merged_format = 1; + dai = + dai_props->cpu_dai = &priv->dais[(*dai_idx)++]; + ret = asoc_simple_card_parse_graph_cpu(ep, dai_link); if (ret) return ret; - ret = asoc_simple_card_parse_clk_cpu(dev, ep, dai_link, dai_props); + ret = asoc_simple_card_parse_clk_cpu(dev, ep, dai_link, dai); if (ret < 0) return ret; @@ -124,6 +161,8 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep, asoc_simple_card_canonicalize_cpu(dai_link, of_graph_get_endpoint_count(dai_link->cpu_of_node) == 1); } else { + struct snd_soc_codec_conf *cconf; + /* FE is dummy */ dai_link->cpu_of_node = NULL; dai_link->cpu_dai_name = "snd-soc-dummy-dai"; @@ -133,27 +172,40 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep, dai_link->no_pcm = 1; dai_link->be_hw_params_fixup = asoc_graph_card_be_hw_params_fixup; + dai = + dai_props->codec_dai = &priv->dais[(*dai_idx)++]; + + cconf = + dai_props->codec_conf = &priv->codec_conf[(*conf_idx)++]; + ret = asoc_simple_card_parse_graph_codec(ep, dai_link); if (ret < 0) return ret; - ret = asoc_simple_card_parse_clk_codec(dev, ep, dai_link, dai_props); + ret = asoc_simple_card_parse_clk_codec(dev, ep, dai_link, dai); if (ret < 0) return ret; ret = asoc_simple_card_set_dailink_name(dev, dai_link, "be.%s", - dai_link->codec_dai_name); + dai_link->codecs->dai_name); if (ret < 0) return ret; - snd_soc_of_parse_audio_prefix(card, - &priv->codec_conf, - dai_link->codec_of_node, + /* check "prefix" from top node */ + snd_soc_of_parse_audio_prefix(card, cconf, + dai_link->codecs->of_node, "prefix"); + /* check "prefix" from each node if top doesn't have */ + if (!cconf->of_node) + snd_soc_of_parse_node_prefix(node, cconf, + dai_link->codecs->of_node, + PREFIX "prefix"); } - ret = asoc_simple_card_of_parse_tdm(ep, dai_props); + asoc_simple_card_parse_convert(dev, node, PREFIX, &dai_props->adata); + + ret = asoc_simple_card_of_parse_tdm(ep, dai); if (ret) return ret; @@ -161,7 +213,11 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep, if (ret < 0) return ret; - dai_link->dai_fmt = daifmt; + ret = asoc_simple_card_parse_daifmt(dev, cpu_ep, codec_ep, + NULL, &dai_link->dai_fmt); + if (ret < 0) + return ret; + dai_link->dpcm_playback = 1; dai_link->dpcm_capture = 1; dai_link->ops = &asoc_graph_card_ops; @@ -179,11 +235,9 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) struct device_node *cpu_port; struct device_node *cpu_ep; struct device_node *codec_ep; - struct device_node *rcpu_ep; struct device_node *codec_port; struct device_node *codec_port_old; - unsigned int daifmt = 0; - int dai_idx, ret; + int dai_idx, link_idx, conf_idx, ret; int rc, codec; if (!node) @@ -194,47 +248,20 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) * see simple-card */ - ret = asoc_simple_card_of_parse_routing(card, NULL, 0); + ret = asoc_simple_card_of_parse_routing(card, NULL); if (ret < 0) return ret; - asoc_simple_card_parse_convert(dev, NULL, &priv->adata); + asoc_simple_card_parse_convert(dev, node, NULL, &priv->adata); /* * it supports multi CPU, single CODEC only here * see asoc_graph_get_dais_count */ - /* find 1st codec */ - of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { - cpu_port = it.node; - cpu_ep = of_get_next_child(cpu_port, NULL); - codec_ep = of_graph_get_remote_endpoint(cpu_ep); - rcpu_ep = of_graph_get_remote_endpoint(codec_ep); - - of_node_put(cpu_ep); - of_node_put(codec_ep); - of_node_put(rcpu_ep); - - if (!codec_ep) - continue; - - if (rcpu_ep != cpu_ep) { - dev_err(dev, "remote-endpoint missmatch (%s/%s/%s)\n", - cpu_ep->name, codec_ep->name, rcpu_ep->name); - ret = -EINVAL; - goto parse_of_err; - } - - ret = asoc_simple_card_parse_daifmt(dev, cpu_ep, codec_ep, - NULL, &daifmt); - if (ret < 0) { - of_node_put(cpu_port); - goto parse_of_err; - } - } - + link_idx = 0; dai_idx = 0; + conf_idx = 0; codec_port_old = NULL; for (codec = 0; codec < 2; codec++) { /* @@ -250,31 +277,23 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) of_node_put(cpu_ep); of_node_put(codec_ep); + of_node_put(cpu_port); of_node_put(codec_port); + it.node = NULL; if (codec) { - if (!codec_port) - continue; - if (codec_port_old == codec_port) continue; codec_port_old = codec_port; - - /* Back-End (= Codec) */ - ret = asoc_graph_card_dai_link_of(codec_ep, priv, daifmt, dai_idx++, 0); - if (ret < 0) { - of_node_put(cpu_port); - goto parse_of_err; - } - } else { - /* Front-End (= CPU) */ - ret = asoc_graph_card_dai_link_of(cpu_ep, priv, daifmt, dai_idx++, 1); - if (ret < 0) { - of_node_put(cpu_port); - goto parse_of_err; - } } + + ret = asoc_graph_card_dai_link_of(cpu_ep, codec_ep, + priv, &dai_idx, + link_idx++, &conf_idx, + !codec); + if (ret < 0) + goto parse_of_err; } } @@ -282,13 +301,24 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) if (ret) goto parse_of_err; + if ((card->num_links != link_idx) || + (card->num_configs != conf_idx)) { + dev_err(dev, "dai_link or codec_config wrong (%d/%d, %d/%d)\n", + card->num_links, link_idx, card->num_configs, conf_idx); + ret = -EINVAL; + goto parse_of_err; + } + ret = 0; parse_of_err: return ret; } -static int asoc_graph_get_dais_count(struct device *dev) +static void asoc_graph_get_dais_count(struct device *dev, + int *link_num, + int *dais_num, + int *ccnf_num) { struct of_phandle_iterator it; struct device_node *node = dev->of_node; @@ -297,10 +327,48 @@ static int asoc_graph_get_dais_count(struct device *dev) struct device_node *codec_ep; struct device_node *codec_port; struct device_node *codec_port_old; - int count = 0; + struct device_node *codec_port_old2; int rc; + /* + * link_num : number of links. + * CPU-Codec / CPU-dummy / dummy-Codec + * dais_num : number of DAIs + * ccnf_num : number of codec_conf + * same number for dummy-Codec + * + * ex1) + * CPU0 --- Codec0 link : 5 + * CPU1 --- Codec1 dais : 7 + * CPU2 -/ ccnf : 1 + * CPU3 --- Codec2 + * + * => 5 links = 2xCPU-Codec + 2xCPU-dummy + 1xdummy-Codec + * => 7 DAIs = 4xCPU + 3xCodec + * => 1 ccnf = 1xdummy-Codec + * + * ex2) + * CPU0 --- Codec0 link : 5 + * CPU1 --- Codec1 dais : 6 + * CPU2 -/ ccnf : 1 + * CPU3 -/ + * + * => 5 links = 1xCPU-Codec + 3xCPU-dummy + 1xdummy-Codec + * => 6 DAIs = 4xCPU + 2xCodec + * => 1 ccnf = 1xdummy-Codec + * + * ex3) + * CPU0 --- Codec0 link : 6 + * CPU1 -/ dais : 6 + * CPU2 --- Codec1 ccnf : 2 + * CPU3 -/ + * + * => 6 links = 0xCPU-Codec + 4xCPU-dummy + 2xdummy-Codec + * => 6 DAIs = 4xCPU + 2xCodec + * => 2 ccnf = 2xdummy-Codec + */ codec_port_old = NULL; + codec_port_old2 = NULL; of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { cpu_port = it.node; cpu_ep = of_get_next_child(cpu_port, NULL); @@ -311,56 +379,77 @@ static int asoc_graph_get_dais_count(struct device *dev) of_node_put(codec_ep); of_node_put(codec_port); - if (cpu_ep) - count++; + (*link_num)++; + (*dais_num)++; - if (!codec_port) - continue; + if (codec_port_old == codec_port) { + if (codec_port_old2 != codec_port_old) { + (*link_num)++; + (*ccnf_num)++; + } - if (codec_port_old == codec_port) + codec_port_old2 = codec_port_old; continue; + } - count++; + (*dais_num)++; codec_port_old = codec_port; } - - return count; } static int asoc_graph_card_probe(struct platform_device *pdev) { struct graph_card_data *priv; struct snd_soc_dai_link *dai_link; - struct asoc_simple_dai *dai_props; + struct graph_dai_props *dai_props; + struct asoc_simple_dai *dais; struct device *dev = &pdev->dev; struct snd_soc_card *card; - int num, ret; + struct snd_soc_codec_conf *cconf; + int lnum = 0, dnum = 0, cnum = 0; + int ret, i; /* Allocate the private data and the DAI link array */ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) return -ENOMEM; - num = asoc_graph_get_dais_count(dev); - if (num == 0) + asoc_graph_get_dais_count(dev, &lnum, &dnum, &cnum); + if (!lnum || !dnum) return -EINVAL; - dai_props = devm_kcalloc(dev, num, sizeof(*dai_props), GFP_KERNEL); - dai_link = devm_kcalloc(dev, num, sizeof(*dai_link), GFP_KERNEL); - if (!dai_props || !dai_link) + dai_props = devm_kcalloc(dev, lnum, sizeof(*dai_props), GFP_KERNEL); + dai_link = devm_kcalloc(dev, lnum, sizeof(*dai_link), GFP_KERNEL); + dais = devm_kcalloc(dev, dnum, sizeof(*dais), GFP_KERNEL); + cconf = devm_kcalloc(dev, cnum, sizeof(*cconf), GFP_KERNEL); + if (!dai_props || !dai_link || !dais) return -ENOMEM; + /* + * Use snd_soc_dai_link_component instead of legacy style + * It is codec only. but cpu/platform will be supported in the future. + * see + * soc-core.c :: snd_soc_init_multicodec() + */ + for (i = 0; i < lnum; i++) { + dai_link[i].codecs = &dai_props[i].codecs; + dai_link[i].num_codecs = 1; + dai_link[i].platform = &dai_props[i].platform; + } + priv->dai_props = dai_props; priv->dai_link = dai_link; + priv->dais = dais; + priv->codec_conf = cconf; /* Init snd_soc_card */ card = graph_priv_to_card(priv); card->owner = THIS_MODULE; card->dev = dev; card->dai_link = priv->dai_link; - card->num_links = num; - card->codec_conf = &priv->codec_conf; - card->num_configs = 1; + card->num_links = lnum; + card->codec_conf = cconf; + card->num_configs = cnum; ret = asoc_graph_card_parse_of(priv); if (ret < 0) { diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index d3f3f0fec74c..b807a47515eb 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -32,10 +32,11 @@ void asoc_simple_card_convert_fixup(struct asoc_simple_card_data *data, } EXPORT_SYMBOL_GPL(asoc_simple_card_convert_fixup); -void asoc_simple_card_parse_convert(struct device *dev, char *prefix, +void asoc_simple_card_parse_convert(struct device *dev, + struct device_node *np, + char *prefix, struct asoc_simple_card_data *data) { - struct device_node *np = dev->of_node; char prop[128]; if (!prefix) @@ -151,21 +152,19 @@ int asoc_simple_card_parse_card_name(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(asoc_simple_card_parse_card_name); -static void asoc_simple_card_clk_register(struct asoc_simple_dai *dai, - struct clk *clk) -{ - dai->clk = clk; -} - int asoc_simple_card_clk_enable(struct asoc_simple_dai *dai) { - return clk_prepare_enable(dai->clk); + if (dai) + return clk_prepare_enable(dai->clk); + + return 0; } EXPORT_SYMBOL_GPL(asoc_simple_card_clk_enable); void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai) { - clk_disable_unprepare(dai->clk); + if (dai) + clk_disable_unprepare(dai->clk); } EXPORT_SYMBOL_GPL(asoc_simple_card_clk_disable); @@ -173,12 +172,24 @@ int asoc_simple_card_parse_clk(struct device *dev, struct device_node *node, struct device_node *dai_of_node, struct asoc_simple_dai *simple_dai, - const char *name) + const char *dai_name, + struct snd_soc_dai_link_component *dlc) { struct clk *clk; u32 val; /* + * Use snd_soc_dai_link_component instead of legacy style. + * It is only for codec, but cpu will be supported in the future. + * see + * soc-core.c :: snd_soc_init_multicodec() + */ + if (dlc) { + dai_of_node = dlc->of_node; + dai_name = dlc->dai_name; + } + + /* * Parse dai->sysclk come from "clocks = <&xxx>" * (if system has common clock) * or "system-clock-frequency = <xxx>" @@ -188,7 +199,7 @@ int asoc_simple_card_parse_clk(struct device *dev, if (!IS_ERR(clk)) { simple_dai->sysclk = clk_get_rate(clk); - asoc_simple_card_clk_register(simple_dai, clk); + simple_dai->clk = clk; } else if (!of_property_read_u32(node, "system-clock-frequency", &val)) { simple_dai->sysclk = val; } else { @@ -200,7 +211,7 @@ int asoc_simple_card_parse_clk(struct device *dev, if (of_property_read_bool(node, "system-clock-direction-out")) simple_dai->clk_direction = SND_SOC_CLOCK_OUT; - dev_dbg(dev, "%s : sysclk = %d, direction %d\n", name, + dev_dbg(dev, "%s : sysclk = %d, direction %d\n", dai_name, simple_dai->sysclk, simple_dai->clk_direction); return 0; @@ -208,6 +219,7 @@ int asoc_simple_card_parse_clk(struct device *dev, EXPORT_SYMBOL_GPL(asoc_simple_card_parse_clk); int asoc_simple_card_parse_dai(struct device_node *node, + struct snd_soc_dai_link_component *dlc, struct device_node **dai_of_node, const char **dai_name, const char *list_name, @@ -221,6 +233,17 @@ int asoc_simple_card_parse_dai(struct device_node *node, return 0; /* + * Use snd_soc_dai_link_component instead of legacy style. + * It is only for codec, but cpu will be supported in the future. + * see + * soc-core.c :: snd_soc_init_multicodec() + */ + if (dlc) { + dai_name = &dlc->dai_name; + dai_of_node = &dlc->of_node; + } + + /* * Get node via "sound-dai = <&phandle port>" * it will be used as xxx_of_node on soc_bind_dai_link() */ @@ -248,13 +271,24 @@ static int asoc_simple_card_get_dai_id(struct device_node *ep) { struct device_node *node; struct device_node *endpoint; + struct of_endpoint info; int i, id; int ret; + /* use driver specified DAI ID if exist */ ret = snd_soc_get_dai_id(ep); if (ret != -ENOTSUPP) return ret; + /* use endpoint/port reg if exist */ + ret = of_graph_parse_endpoint(ep, &info); + if (ret == 0) { + if (info.id) + return info.id; + if (info.port) + return info.port; + } + node = of_graph_get_port_parent(ep); /* @@ -278,6 +312,7 @@ static int asoc_simple_card_get_dai_id(struct device_node *ep) } int asoc_simple_card_parse_graph_dai(struct device_node *ep, + struct snd_soc_dai_link_component *dlc, struct device_node **dai_of_node, const char **dai_name) { @@ -285,6 +320,17 @@ int asoc_simple_card_parse_graph_dai(struct device_node *ep, struct of_phandle_args args; int ret; + /* + * Use snd_soc_dai_link_component instead of legacy style. + * It is only for codec, but cpu will be supported in the future. + * see + * soc-core.c :: snd_soc_init_multicodec() + */ + if (dlc) { + dai_name = &dlc->dai_name; + dai_of_node = &dlc->of_node; + } + if (!ep) return 0; if (!dai_name) @@ -312,6 +358,9 @@ int asoc_simple_card_init_dai(struct snd_soc_dai *dai, { int ret; + if (!simple_dai) + return 0; + if (simple_dai->sysclk) { ret = snd_soc_dai_set_sysclk(dai, 0, simple_dai->sysclk, simple_dai->clk_direction); @@ -340,10 +389,11 @@ EXPORT_SYMBOL_GPL(asoc_simple_card_init_dai); int asoc_simple_card_canonicalize_dailink(struct snd_soc_dai_link *dai_link) { /* Assumes platform == cpu */ - if (!dai_link->platform_of_node) - dai_link->platform_of_node = dai_link->cpu_of_node; + if (!dai_link->platform->of_node) + dai_link->platform->of_node = dai_link->cpu_of_node; return 0; + } EXPORT_SYMBOL_GPL(asoc_simple_card_canonicalize_dailink); @@ -367,21 +417,18 @@ EXPORT_SYMBOL_GPL(asoc_simple_card_canonicalize_cpu); int asoc_simple_card_clean_reference(struct snd_soc_card *card) { struct snd_soc_dai_link *dai_link; - int num_links; + int i; - for (num_links = 0, dai_link = card->dai_link; - num_links < card->num_links; - num_links++, dai_link++) { + for_each_card_prelinks(card, i, dai_link) { of_node_put(dai_link->cpu_of_node); - of_node_put(dai_link->codec_of_node); + of_node_put(dai_link->codecs->of_node); } return 0; } EXPORT_SYMBOL_GPL(asoc_simple_card_clean_reference); int asoc_simple_card_of_parse_routing(struct snd_soc_card *card, - char *prefix, - int optional) + char *prefix) { struct device_node *node = card->dev->of_node; char prop[128]; @@ -391,11 +438,8 @@ int asoc_simple_card_of_parse_routing(struct snd_soc_card *card, snprintf(prop, sizeof(prop), "%s%s", prefix, "routing"); - if (!of_property_read_bool(node, prop)) { - if (optional) - return 0; - return -EINVAL; - } + if (!of_property_read_bool(node, prop)) + return 0; return snd_soc_of_parse_audio_routing(card, prop); } diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 64bf3560c1d1..3fe34417ec89 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -18,14 +18,19 @@ struct simple_card_data { struct snd_soc_card snd_card; struct simple_dai_props { - struct asoc_simple_dai cpu_dai; - struct asoc_simple_dai codec_dai; + struct asoc_simple_dai *cpu_dai; + struct asoc_simple_dai *codec_dai; + struct snd_soc_dai_link_component codecs; /* single codec */ + struct snd_soc_dai_link_component platform; + struct asoc_simple_card_data adata; + struct snd_soc_codec_conf *codec_conf; unsigned int mclk_fs; } *dai_props; - unsigned int mclk_fs; struct asoc_simple_jack hp_jack; struct asoc_simple_jack mic_jack; struct snd_soc_dai_link *dai_link; + struct asoc_simple_dai *dais; + struct snd_soc_codec_conf *codec_conf; }; #define simple_priv_to_card(priv) (&(priv)->snd_card) @@ -45,13 +50,13 @@ static int asoc_simple_card_startup(struct snd_pcm_substream *substream) simple_priv_to_props(priv, rtd->num); int ret; - ret = asoc_simple_card_clk_enable(&dai_props->cpu_dai); + ret = asoc_simple_card_clk_enable(dai_props->cpu_dai); if (ret) return ret; - ret = asoc_simple_card_clk_enable(&dai_props->codec_dai); + ret = asoc_simple_card_clk_enable(dai_props->codec_dai); if (ret) - asoc_simple_card_clk_disable(&dai_props->cpu_dai); + asoc_simple_card_clk_disable(dai_props->cpu_dai); return ret; } @@ -63,14 +68,17 @@ static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream) struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); - asoc_simple_card_clk_disable(&dai_props->cpu_dai); + asoc_simple_card_clk_disable(dai_props->cpu_dai); - asoc_simple_card_clk_disable(&dai_props->codec_dai); + asoc_simple_card_clk_disable(dai_props->codec_dai); } static int asoc_simple_set_clk_rate(struct asoc_simple_dai *simple_dai, unsigned long rate) { + if (!simple_dai) + return 0; + if (!simple_dai->clk) return 0; @@ -92,19 +100,17 @@ static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream, unsigned int mclk, mclk_fs = 0; int ret = 0; - if (priv->mclk_fs) - mclk_fs = priv->mclk_fs; - else if (dai_props->mclk_fs) + if (dai_props->mclk_fs) mclk_fs = dai_props->mclk_fs; if (mclk_fs) { mclk = params_rate(params) * mclk_fs; - ret = asoc_simple_set_clk_rate(&dai_props->codec_dai, mclk); + ret = asoc_simple_set_clk_rate(dai_props->codec_dai, mclk); if (ret < 0) return ret; - ret = asoc_simple_set_clk_rate(&dai_props->cpu_dai, mclk); + ret = asoc_simple_set_clk_rate(dai_props->cpu_dai, mclk); if (ret < 0) return ret; @@ -132,33 +138,169 @@ static const struct snd_soc_ops asoc_simple_card_ops = { static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) { struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *codec = rtd->codec_dai; - struct snd_soc_dai *cpu = rtd->cpu_dai; - struct simple_dai_props *dai_props = - simple_priv_to_props(priv, rtd->num); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); + int ret; + + ret = asoc_simple_card_init_dai(rtd->codec_dai, + dai_props->codec_dai); + if (ret < 0) + return ret; + + ret = asoc_simple_card_init_dai(rtd->cpu_dai, + dai_props->cpu_dai); + if (ret < 0) + return ret; + + return 0; +} + +static int asoc_simple_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); + + asoc_simple_card_convert_fixup(&dai_props->adata, params); + + return 0; +} + +static int asoc_simple_card_dai_link_of_dpcm(struct device_node *top, + struct device_node *node, + struct device_node *np, + struct device_node *codec, + struct simple_card_data *priv, + int *dai_idx, int link_idx, + int *conf_idx, int is_fe, + bool is_top_level_node) +{ + struct device *dev = simple_priv_to_dev(priv); + struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, link_idx); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, link_idx); + struct asoc_simple_dai *dai; + struct snd_soc_dai_link_component *codecs = dai_link->codecs; + + char prop[128]; + char *prefix = ""; int ret; - ret = asoc_simple_card_init_dai(codec, &dai_props->codec_dai); + /* For single DAI link & old style of DT node */ + if (is_top_level_node) + prefix = PREFIX; + + if (is_fe) { + int is_single_links = 0; + + /* BE is dummy */ + codecs->of_node = NULL; + codecs->dai_name = "snd-soc-dummy-dai"; + codecs->name = "snd-soc-dummy"; + + /* FE settings */ + dai_link->dynamic = 1; + dai_link->dpcm_merged_format = 1; + + dai = + dai_props->cpu_dai = &priv->dais[(*dai_idx)++]; + + ret = asoc_simple_card_parse_cpu(np, dai_link, DAI, CELL, + &is_single_links); + if (ret) + return ret; + + ret = asoc_simple_card_parse_clk_cpu(dev, np, dai_link, dai); + if (ret < 0) + return ret; + + ret = asoc_simple_card_set_dailink_name(dev, dai_link, + "fe.%s", + dai_link->cpu_dai_name); + if (ret < 0) + return ret; + + asoc_simple_card_canonicalize_cpu(dai_link, is_single_links); + } else { + struct snd_soc_codec_conf *cconf; + + /* FE is dummy */ + dai_link->cpu_of_node = NULL; + dai_link->cpu_dai_name = "snd-soc-dummy-dai"; + dai_link->cpu_name = "snd-soc-dummy"; + + /* BE settings */ + dai_link->no_pcm = 1; + dai_link->be_hw_params_fixup = asoc_simple_card_be_hw_params_fixup; + + dai = + dai_props->codec_dai = &priv->dais[(*dai_idx)++]; + + cconf = + dai_props->codec_conf = &priv->codec_conf[(*conf_idx)++]; + + ret = asoc_simple_card_parse_codec(np, dai_link, DAI, CELL); + if (ret < 0) + return ret; + + ret = asoc_simple_card_parse_clk_codec(dev, np, dai_link, dai); + if (ret < 0) + return ret; + + ret = asoc_simple_card_set_dailink_name(dev, dai_link, + "be.%s", + codecs->dai_name); + if (ret < 0) + return ret; + + /* check "prefix" from top node */ + snd_soc_of_parse_node_prefix(top, cconf, codecs->of_node, + PREFIX "prefix"); + snd_soc_of_parse_node_prefix(node, cconf, codecs->of_node, + "prefix"); + snd_soc_of_parse_node_prefix(np, cconf, codecs->of_node, + "prefix"); + } + + asoc_simple_card_parse_convert(dev, top, PREFIX, &dai_props->adata); + asoc_simple_card_parse_convert(dev, node, prefix, &dai_props->adata); + asoc_simple_card_parse_convert(dev, np, NULL, &dai_props->adata); + + ret = asoc_simple_card_of_parse_tdm(np, dai); + if (ret) + return ret; + + ret = asoc_simple_card_canonicalize_dailink(dai_link); if (ret < 0) return ret; - ret = asoc_simple_card_init_dai(cpu, &dai_props->cpu_dai); + snprintf(prop, sizeof(prop), "%smclk-fs", prefix); + of_property_read_u32(top, PREFIX "mclk-fs", &dai_props->mclk_fs); + of_property_read_u32(node, prop, &dai_props->mclk_fs); + of_property_read_u32(np, prop, &dai_props->mclk_fs); + + ret = asoc_simple_card_parse_daifmt(dev, node, codec, + prefix, &dai_link->dai_fmt); if (ret < 0) return ret; + dai_link->dpcm_playback = 1; + dai_link->dpcm_capture = 1; + dai_link->ops = &asoc_simple_card_ops; + dai_link->init = asoc_simple_card_dai_init; + return 0; } -static int asoc_simple_card_dai_link_of(struct device_node *node, +static int asoc_simple_card_dai_link_of(struct device_node *top, + struct device_node *node, struct simple_card_data *priv, - int idx, + int *dai_idx, int link_idx, bool is_top_level_node) { struct device *dev = simple_priv_to_dev(priv); - struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, idx); - struct simple_dai_props *dai_props = simple_priv_to_props(priv, idx); - struct asoc_simple_dai *cpu_dai = &dai_props->cpu_dai; - struct asoc_simple_dai *codec_dai = &dai_props->codec_dai; + struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, link_idx); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, link_idx); + struct asoc_simple_dai *cpu_dai; + struct asoc_simple_dai *codec_dai; struct device_node *cpu = NULL; struct device_node *plat = NULL; struct device_node *codec = NULL; @@ -191,12 +333,21 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, goto dai_link_of_err; } + cpu_dai = + dai_props->cpu_dai = &priv->dais[(*dai_idx)++]; + codec_dai = + dai_props->codec_dai = &priv->dais[(*dai_idx)++]; + ret = asoc_simple_card_parse_daifmt(dev, node, codec, prefix, &dai_link->dai_fmt); if (ret < 0) goto dai_link_of_err; - of_property_read_u32(node, "mclk-fs", &dai_props->mclk_fs); + snprintf(prop, sizeof(prop), "%smclk-fs", prefix); + of_property_read_u32(top, PREFIX "mclk-fs", &dai_props->mclk_fs); + of_property_read_u32(node, prop, &dai_props->mclk_fs); + of_property_read_u32(cpu, prop, &dai_props->mclk_fs); + of_property_read_u32(codec, prop, &dai_props->mclk_fs); ret = asoc_simple_card_parse_cpu(cpu, dai_link, DAI, CELL, &single_cpu); @@ -234,7 +385,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, ret = asoc_simple_card_set_dailink_name(dev, dai_link, "%s-%s", dai_link->cpu_dai_name, - dai_link->codec_dai_name); + dai_link->codecs->dai_name); if (ret < 0) goto dai_link_of_err; @@ -284,61 +435,148 @@ static int asoc_simple_card_parse_aux_devs(struct device_node *node, static int asoc_simple_card_parse_of(struct simple_card_data *priv) { struct device *dev = simple_priv_to_dev(priv); + struct device_node *top = dev->of_node; struct snd_soc_card *card = simple_priv_to_card(priv); - struct device_node *dai_link; - struct device_node *node = dev->of_node; - int ret; - - if (!node) + struct device_node *node; + struct device_node *np; + struct device_node *codec; + bool is_fe; + int ret, loop; + int dai_idx, link_idx, conf_idx; + + if (!top) return -EINVAL; - dai_link = of_get_child_by_name(node, PREFIX "dai-link"); - ret = asoc_simple_card_of_parse_widgets(card, PREFIX); if (ret < 0) - goto card_parse_end; + return ret; - ret = asoc_simple_card_of_parse_routing(card, PREFIX, 1); + ret = asoc_simple_card_of_parse_routing(card, PREFIX); if (ret < 0) - goto card_parse_end; - - /* Factor to mclk, used in hw_params() */ - of_property_read_u32(node, PREFIX "mclk-fs", &priv->mclk_fs); + return ret; /* Single/Muti DAI link(s) & New style of DT node */ - if (dai_link) { - struct device_node *np = NULL; - int i = 0; - - for_each_child_of_node(node, np) { - dev_dbg(dev, "\tlink %d:\n", i); - ret = asoc_simple_card_dai_link_of(np, priv, - i, false); - if (ret < 0) { - of_node_put(np); - goto card_parse_end; + loop = 1; + link_idx = 0; + dai_idx = 0; + conf_idx = 0; + node = of_get_child_by_name(top, PREFIX "dai-link"); + if (!node) { + node = of_node_get(top); + loop = 0; + } + + do { + /* DPCM */ + if (of_get_child_count(node) > 2) { + for_each_child_of_node(node, np) { + codec = of_get_child_by_name(node, + loop ? "codec" : + PREFIX "codec"); + if (!codec) + return -ENODEV; + + is_fe = (np != codec); + + ret = asoc_simple_card_dai_link_of_dpcm( + top, node, np, codec, priv, + &dai_idx, link_idx++, &conf_idx, + is_fe, !loop); } - i++; + } else { + ret = asoc_simple_card_dai_link_of( + top, node, priv, + &dai_idx, link_idx++, !loop); } - } else { - /* For single DAI link & old style of DT node */ - ret = asoc_simple_card_dai_link_of(node, priv, 0, true); if (ret < 0) - goto card_parse_end; - } + return ret; + + node = of_get_next_child(top, node); + } while (loop && node); ret = asoc_simple_card_parse_card_name(card, PREFIX); if (ret < 0) - goto card_parse_end; - - ret = asoc_simple_card_parse_aux_devs(node, priv); + return ret; -card_parse_end: - of_node_put(dai_link); + ret = asoc_simple_card_parse_aux_devs(top, priv); return ret; } +static void asoc_simple_card_get_dais_count(struct device *dev, + int *link_num, + int *dais_num, + int *ccnf_num) +{ + struct device_node *top = dev->of_node; + struct device_node *node; + int loop; + int num; + + /* + * link_num : number of links. + * CPU-Codec / CPU-dummy / dummy-Codec + * dais_num : number of DAIs + * ccnf_num : number of codec_conf + * same number for "dummy-Codec" + * + * ex1) + * CPU0 --- Codec0 link : 5 + * CPU1 --- Codec1 dais : 7 + * CPU2 -/ ccnf : 1 + * CPU3 --- Codec2 + * + * => 5 links = 2xCPU-Codec + 2xCPU-dummy + 1xdummy-Codec + * => 7 DAIs = 4xCPU + 3xCodec + * => 1 ccnf = 1xdummy-Codec + * + * ex2) + * CPU0 --- Codec0 link : 5 + * CPU1 --- Codec1 dais : 6 + * CPU2 -/ ccnf : 1 + * CPU3 -/ + * + * => 5 links = 1xCPU-Codec + 3xCPU-dummy + 1xdummy-Codec + * => 6 DAIs = 4xCPU + 2xCodec + * => 1 ccnf = 1xdummy-Codec + * + * ex3) + * CPU0 --- Codec0 link : 6 + * CPU1 -/ dais : 6 + * CPU2 --- Codec1 ccnf : 2 + * CPU3 -/ + * + * => 6 links = 0xCPU-Codec + 4xCPU-dummy + 2xdummy-Codec + * => 6 DAIs = 4xCPU + 2xCodec + * => 2 ccnf = 2xdummy-Codec + */ + if (!top) { + (*link_num) = 1; + (*dais_num) = 2; + (*ccnf_num) = 0; + return; + } + + loop = 1; + node = of_get_child_by_name(top, PREFIX "dai-link"); + if (!node) { + node = top; + loop = 0; + } + + do { + num = of_get_child_count(node); + (*dais_num) += num; + if (num > 2) { + (*link_num) += num; + (*ccnf_num)++; + } else { + (*link_num)++; + } + node = of_get_next_child(top, node); + } while (loop && node); +} + static int asoc_simple_soc_card_probe(struct snd_soc_card *card) { struct simple_card_data *priv = snd_soc_card_get_drvdata(card); @@ -360,36 +598,55 @@ static int asoc_simple_card_probe(struct platform_device *pdev) struct simple_card_data *priv; struct snd_soc_dai_link *dai_link; struct simple_dai_props *dai_props; + struct asoc_simple_dai *dais; struct device *dev = &pdev->dev; struct device_node *np = dev->of_node; struct snd_soc_card *card; - int num, ret; - - /* Get the number of DAI links */ - if (np && of_get_child_by_name(np, PREFIX "dai-link")) - num = of_get_child_count(np); - else - num = 1; + struct snd_soc_codec_conf *cconf; + int lnum = 0, dnum = 0, cnum = 0; + int ret, i; /* Allocate the private data and the DAI link array */ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) return -ENOMEM; - dai_props = devm_kcalloc(dev, num, sizeof(*dai_props), GFP_KERNEL); - dai_link = devm_kcalloc(dev, num, sizeof(*dai_link), GFP_KERNEL); - if (!dai_props || !dai_link) + asoc_simple_card_get_dais_count(dev, &lnum, &dnum, &cnum); + if (!lnum || !dnum) + return -EINVAL; + + dai_props = devm_kcalloc(dev, lnum, sizeof(*dai_props), GFP_KERNEL); + dai_link = devm_kcalloc(dev, lnum, sizeof(*dai_link), GFP_KERNEL); + dais = devm_kcalloc(dev, dnum, sizeof(*dais), GFP_KERNEL); + cconf = devm_kcalloc(dev, cnum, sizeof(*cconf), GFP_KERNEL); + if (!dai_props || !dai_link || !dais) return -ENOMEM; + /* + * Use snd_soc_dai_link_component instead of legacy style + * It is codec only. but cpu/platform will be supported in the future. + * see + * soc-core.c :: snd_soc_init_multicodec() + */ + for (i = 0; i < lnum; i++) { + dai_link[i].codecs = &dai_props[i].codecs; + dai_link[i].num_codecs = 1; + dai_link[i].platform = &dai_props[i].platform; + } + priv->dai_props = dai_props; priv->dai_link = dai_link; + priv->dais = dais; + priv->codec_conf = cconf; /* Init snd_soc_card */ card = simple_priv_to_card(priv); card->owner = THIS_MODULE; card->dev = dev; card->dai_link = priv->dai_link; - card->num_links = num; + card->num_links = lnum; + card->codec_conf = cconf; + card->num_configs = cnum; card->probe = asoc_simple_soc_card_probe; if (np && of_device_is_available(np)) { @@ -403,6 +660,9 @@ static int asoc_simple_card_probe(struct platform_device *pdev) } else { struct asoc_simple_card_info *cinfo; + struct snd_soc_dai_link_component *codecs; + struct snd_soc_dai_link_component *platform; + int dai_idx = 0; cinfo = dev->platform_data; if (!cinfo) { @@ -419,19 +679,26 @@ static int asoc_simple_card_probe(struct platform_device *pdev) return -EINVAL; } + dai_props->cpu_dai = &priv->dais[dai_idx++]; + dai_props->codec_dai = &priv->dais[dai_idx++]; + + codecs = dai_link->codecs; + codecs->name = cinfo->codec; + codecs->dai_name = cinfo->codec_dai.name; + + platform = dai_link->platform; + platform->name = cinfo->platform; + card->name = (cinfo->card) ? cinfo->card : cinfo->name; dai_link->name = cinfo->name; dai_link->stream_name = cinfo->name; - dai_link->platform_name = cinfo->platform; - dai_link->codec_name = cinfo->codec; dai_link->cpu_dai_name = cinfo->cpu_dai.name; - dai_link->codec_dai_name = cinfo->codec_dai.name; dai_link->dai_fmt = cinfo->daifmt; dai_link->init = asoc_simple_card_dai_init; - memcpy(&priv->dai_props->cpu_dai, &cinfo->cpu_dai, - sizeof(priv->dai_props->cpu_dai)); - memcpy(&priv->dai_props->codec_dai, &cinfo->codec_dai, - sizeof(priv->dai_props->codec_dai)); + memcpy(priv->dai_props->cpu_dai, &cinfo->cpu_dai, + sizeof(*priv->dai_props->cpu_dai)); + memcpy(priv->dai_props->codec_dai, &cinfo->codec_dai, + sizeof(*priv->dai_props->codec_dai)); } snd_soc_card_set_drvdata(card, priv); @@ -456,6 +723,7 @@ static int asoc_simple_card_remove(struct platform_device *pdev) static const struct of_device_id asoc_simple_of_match[] = { { .compatible = "simple-audio-card", }, + { .compatible = "simple-scu-audio-card", }, {}, }; MODULE_DEVICE_TABLE(of, asoc_simple_of_match); diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 16a83bc51e0e..9d7299d536a8 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -21,10 +21,18 @@ struct simple_card_data { struct snd_soc_card snd_card; - struct snd_soc_codec_conf codec_conf; - struct asoc_simple_dai *dai_props; + struct simple_dai_props { + struct asoc_simple_dai *cpu_dai; + struct asoc_simple_dai *codec_dai; + struct snd_soc_dai_link_component codecs; + struct snd_soc_dai_link_component platform; + struct asoc_simple_card_data adata; + struct snd_soc_codec_conf *codec_conf; + } *dai_props; struct snd_soc_dai_link *dai_link; + struct asoc_simple_dai *dais; struct asoc_simple_card_data adata; + struct snd_soc_codec_conf *codec_conf; }; #define simple_priv_to_card(priv) (&(priv)->snd_card) @@ -40,20 +48,31 @@ static int asoc_simple_card_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct asoc_simple_dai *dai_props = + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); + int ret; + + ret = asoc_simple_card_clk_enable(dai_props->cpu_dai); + if (ret) + return ret; + + ret = asoc_simple_card_clk_enable(dai_props->codec_dai); + if (ret) + asoc_simple_card_clk_disable(dai_props->cpu_dai); - return asoc_simple_card_clk_enable(dai_props); + return ret; } static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct asoc_simple_dai *dai_props = + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); - asoc_simple_card_clk_disable(dai_props); + asoc_simple_card_clk_disable(dai_props->cpu_dai); + + asoc_simple_card_clk_disable(dai_props->codec_dai); } static const struct snd_soc_ops asoc_simple_card_ops = { @@ -63,60 +82,80 @@ static const struct snd_soc_ops asoc_simple_card_ops = { static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai; - struct snd_soc_dai_link *dai_link; - struct asoc_simple_dai *dai_props; - int num = rtd->num; + struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); + int ret; - dai_link = simple_priv_to_link(priv, num); - dai_props = simple_priv_to_props(priv, num); - dai = dai_link->dynamic ? - rtd->cpu_dai : - rtd->codec_dai; + ret = asoc_simple_card_init_dai(rtd->codec_dai, + dai_props->codec_dai); + if (ret < 0) + return ret; + + ret = asoc_simple_card_init_dai(rtd->cpu_dai, + dai_props->cpu_dai); + if (ret < 0) + return ret; - return asoc_simple_card_init_dai(dai, dai_props); + return 0; } static int asoc_simple_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); + asoc_simple_card_convert_fixup(&dai_props->adata, params); + + /* overwrite by top level adata if exist */ asoc_simple_card_convert_fixup(&priv->adata, params); return 0; } -static int asoc_simple_card_dai_link_of(struct device_node *np, +static int asoc_simple_card_dai_link_of(struct device_node *link, + struct device_node *np, + struct device_node *codec, struct simple_card_data *priv, - unsigned int daifmt, - int idx, bool is_fe) + int *dai_idx, int link_idx, + int *conf_idx, int is_fe, + bool is_top_level_node) { struct device *dev = simple_priv_to_dev(priv); - struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, idx); - struct asoc_simple_dai *dai_props = simple_priv_to_props(priv, idx); + struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, link_idx); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, link_idx); struct snd_soc_card *card = simple_priv_to_card(priv); + struct asoc_simple_dai *dai; + char *prefix = ""; int ret; + /* For single DAI link & old style of DT node */ + if (is_top_level_node) + prefix = PREFIX; + if (is_fe) { int is_single_links = 0; + struct snd_soc_dai_link_component *codecs; /* BE is dummy */ - dai_link->codec_of_node = NULL; - dai_link->codec_dai_name = "snd-soc-dummy-dai"; - dai_link->codec_name = "snd-soc-dummy"; + codecs = dai_link->codecs; + codecs->of_node = NULL; + codecs->dai_name = "snd-soc-dummy-dai"; + codecs->name = "snd-soc-dummy"; /* FE settings */ dai_link->dynamic = 1; dai_link->dpcm_merged_format = 1; + dai = + dai_props->cpu_dai = &priv->dais[(*dai_idx)++]; + ret = asoc_simple_card_parse_cpu(np, dai_link, DAI, CELL, &is_single_links); if (ret) return ret; - ret = asoc_simple_card_parse_clk_cpu(dev, np, dai_link, dai_props); + ret = asoc_simple_card_parse_clk_cpu(dev, np, dai_link, dai); if (ret < 0) return ret; @@ -128,6 +167,8 @@ static int asoc_simple_card_dai_link_of(struct device_node *np, asoc_simple_card_canonicalize_cpu(dai_link, is_single_links); } else { + struct snd_soc_codec_conf *cconf; + /* FE is dummy */ dai_link->cpu_of_node = NULL; dai_link->cpu_dai_name = "snd-soc-dummy-dai"; @@ -137,27 +178,40 @@ static int asoc_simple_card_dai_link_of(struct device_node *np, dai_link->no_pcm = 1; dai_link->be_hw_params_fixup = asoc_simple_card_be_hw_params_fixup; + dai = + dai_props->codec_dai = &priv->dais[(*dai_idx)++]; + + cconf = + dai_props->codec_conf = &priv->codec_conf[(*conf_idx)++]; + ret = asoc_simple_card_parse_codec(np, dai_link, DAI, CELL); if (ret < 0) return ret; - ret = asoc_simple_card_parse_clk_codec(dev, np, dai_link, dai_props); + ret = asoc_simple_card_parse_clk_codec(dev, np, dai_link, dai); if (ret < 0) return ret; ret = asoc_simple_card_set_dailink_name(dev, dai_link, "be.%s", - dai_link->codec_dai_name); + dai_link->codecs->dai_name); if (ret < 0) return ret; - snd_soc_of_parse_audio_prefix(card, - &priv->codec_conf, - dai_link->codec_of_node, + /* check "prefix" from top node */ + snd_soc_of_parse_audio_prefix(card, cconf, + dai_link->codecs->of_node, PREFIX "prefix"); + /* check "prefix" from each node if top doesn't have */ + if (!cconf->of_node) + snd_soc_of_parse_node_prefix(np, cconf, + dai_link->codecs->of_node, + "prefix"); } - ret = asoc_simple_card_of_parse_tdm(np, dai_props); + asoc_simple_card_parse_convert(dev, link, prefix, &dai_props->adata); + + ret = asoc_simple_card_of_parse_tdm(np, dai); if (ret) return ret; @@ -165,7 +219,11 @@ static int asoc_simple_card_dai_link_of(struct device_node *np, if (ret < 0) return ret; - dai_link->dai_fmt = daifmt; + ret = asoc_simple_card_parse_daifmt(dev, link, codec, + prefix, &dai_link->dai_fmt); + if (ret < 0) + return ret; + dai_link->dpcm_playback = 1; dai_link->dpcm_capture = 1; dai_link->ops = &asoc_simple_card_ops; @@ -178,87 +236,191 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) { struct device *dev = simple_priv_to_dev(priv); + struct device_node *top = dev->of_node; + struct device_node *node; struct device_node *np; + struct device_node *codec; struct snd_soc_card *card = simple_priv_to_card(priv); - struct device_node *node = dev->of_node; - unsigned int daifmt = 0; bool is_fe; - int ret, i; + int ret, loop; + int dai_idx, link_idx, conf_idx; - if (!node) + if (!top) return -EINVAL; ret = asoc_simple_card_of_parse_widgets(card, PREFIX); if (ret < 0) return ret; - ret = asoc_simple_card_of_parse_routing(card, PREFIX, 0); + ret = asoc_simple_card_of_parse_routing(card, PREFIX); if (ret < 0) return ret; - asoc_simple_card_parse_convert(dev, PREFIX, &priv->adata); + asoc_simple_card_parse_convert(dev, top, PREFIX, &priv->adata); - /* find 1st codec */ - np = of_get_child_by_name(node, PREFIX "codec"); - if (!np) - return -ENODEV; + loop = 1; + link_idx = 0; + dai_idx = 0; + conf_idx = 0; + node = of_get_child_by_name(top, PREFIX "dai-link"); + if (!node) { + node = dev->of_node; + loop = 0; + } + + do { + codec = of_get_child_by_name(node, + loop ? "codec" : PREFIX "codec"); + if (!codec) + return -ENODEV; + + for_each_child_of_node(node, np) { + is_fe = (np != codec); + + ret = asoc_simple_card_dai_link_of(node, np, codec, priv, + &dai_idx, link_idx++, + &conf_idx, + is_fe, !loop); + if (ret < 0) + return ret; + } + node = of_get_next_child(top, node); + } while (loop && node); - ret = asoc_simple_card_parse_daifmt(dev, node, np, PREFIX, &daifmt); + ret = asoc_simple_card_parse_card_name(card, PREFIX); if (ret < 0) return ret; - i = 0; - for_each_child_of_node(node, np) { - is_fe = false; - if (strcmp(np->name, PREFIX "cpu") == 0) - is_fe = true; + return 0; +} - ret = asoc_simple_card_dai_link_of(np, priv, daifmt, i, is_fe); - if (ret < 0) - return ret; - i++; +static void asoc_simple_card_get_dais_count(struct device *dev, + int *link_num, + int *dais_num, + int *ccnf_num) +{ + struct device_node *top = dev->of_node; + struct device_node *node; + int loop; + int num; + + /* + * link_num : number of links. + * CPU-Codec / CPU-dummy / dummy-Codec + * dais_num : number of DAIs + * ccnf_num : number of codec_conf + * same number for "dummy-Codec" + * + * ex1) + * CPU0 --- Codec0 link : 5 + * CPU1 --- Codec1 dais : 7 + * CPU2 -/ ccnf : 1 + * CPU3 --- Codec2 + * + * => 5 links = 2xCPU-Codec + 2xCPU-dummy + 1xdummy-Codec + * => 7 DAIs = 4xCPU + 3xCodec + * => 1 ccnf = 1xdummy-Codec + * + * ex2) + * CPU0 --- Codec0 link : 5 + * CPU1 --- Codec1 dais : 6 + * CPU2 -/ ccnf : 1 + * CPU3 -/ + * + * => 5 links = 1xCPU-Codec + 3xCPU-dummy + 1xdummy-Codec + * => 6 DAIs = 4xCPU + 2xCodec + * => 1 ccnf = 1xdummy-Codec + * + * ex3) + * CPU0 --- Codec0 link : 6 + * CPU1 -/ dais : 6 + * CPU2 --- Codec1 ccnf : 2 + * CPU3 -/ + * + * => 6 links = 0xCPU-Codec + 4xCPU-dummy + 2xdummy-Codec + * => 6 DAIs = 4xCPU + 2xCodec + * => 2 ccnf = 2xdummy-Codec + */ + if (!top) { + (*link_num) = 1; + (*dais_num) = 2; + (*ccnf_num) = 0; + return; } - ret = asoc_simple_card_parse_card_name(card, PREFIX); - if (ret < 0) - return ret; + loop = 1; + node = of_get_child_by_name(top, PREFIX "dai-link"); + if (!node) { + node = top; + loop = 0; + } - return 0; + do { + num = of_get_child_count(node); + (*dais_num) += num; + if (num > 2) { + (*link_num) += num; + (*ccnf_num)++; + } else { + (*link_num)++; + } + node = of_get_next_child(top, node); + } while (loop && node); } static int asoc_simple_card_probe(struct platform_device *pdev) { struct simple_card_data *priv; struct snd_soc_dai_link *dai_link; - struct asoc_simple_dai *dai_props; + struct simple_dai_props *dai_props; + struct asoc_simple_dai *dais; struct snd_soc_card *card; + struct snd_soc_codec_conf *cconf; struct device *dev = &pdev->dev; - struct device_node *np = dev->of_node; - int num, ret; + int ret, i; + int lnum = 0, dnum = 0, cnum = 0; /* Allocate the private data */ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) return -ENOMEM; - num = of_get_child_count(np); + asoc_simple_card_get_dais_count(dev, &lnum, &dnum, &cnum); + if (!lnum || !dnum) + return -EINVAL; - dai_props = devm_kcalloc(dev, num, sizeof(*dai_props), GFP_KERNEL); - dai_link = devm_kcalloc(dev, num, sizeof(*dai_link), GFP_KERNEL); - if (!dai_props || !dai_link) + dai_props = devm_kcalloc(dev, lnum, sizeof(*dai_props), GFP_KERNEL); + dai_link = devm_kcalloc(dev, lnum, sizeof(*dai_link), GFP_KERNEL); + dais = devm_kcalloc(dev, dnum, sizeof(*dais), GFP_KERNEL); + cconf = devm_kcalloc(dev, cnum, sizeof(*cconf), GFP_KERNEL); + if (!dai_props || !dai_link || !dais) return -ENOMEM; + /* + * Use snd_soc_dai_link_component instead of legacy style + * It is codec only. but cpu/platform will be supported in the future. + * see + * soc-core.c :: snd_soc_init_multicodec() + */ + for (i = 0; i < lnum; i++) { + dai_link[i].codecs = &dai_props[i].codecs; + dai_link[i].num_codecs = 1; + dai_link[i].platform = &dai_props[i].platform; + } + priv->dai_props = dai_props; priv->dai_link = dai_link; + priv->dais = dais; + priv->codec_conf = cconf; /* Init snd_soc_card */ card = simple_priv_to_card(priv); card->owner = THIS_MODULE; card->dev = dev; card->dai_link = priv->dai_link; - card->num_links = num; - card->codec_conf = &priv->codec_conf; - card->num_configs = 1; + card->num_links = lnum; + card->codec_conf = cconf; + card->num_configs = cnum; ret = asoc_simple_card_parse_of(priv); if (ret < 0) { diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c index 53344a3b7a60..a69e5b11b3da 100644 --- a/sound/soc/hisilicon/hi6210-i2s.c +++ b/sound/soc/hisilicon/hi6210-i2s.c @@ -269,13 +269,13 @@ static int hi6210_i2s_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_U16_LE: signed_data = HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT; - /* fallthru */ + /* fall through */ case SNDRV_PCM_FORMAT_S16_LE: bits = HII2S_BITS_16; break; case SNDRV_PCM_FORMAT_U24_LE: signed_data = HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT; - /* fallthru */ + /* fall through */ case SNDRV_PCM_FORMAT_S24_LE: bits = HII2S_BITS_24; break; diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 0caa1f4eb94d..bd9fd2035c55 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -91,7 +91,7 @@ config SND_SST_ATOM_HIFI2_PLATFORM_PCI config SND_SST_ATOM_HIFI2_PLATFORM_ACPI tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms" default ACPI - depends on X86 && ACPI + depends on X86 && ACPI && PCI select SND_SST_IPC_ACPI select SND_SST_ATOM_HIFI2_PLATFORM select SND_SOC_ACPI_INTEL_MATCH @@ -101,22 +101,101 @@ config SND_SST_ATOM_HIFI2_PLATFORM_ACPI codec, then enable this option by saying Y or m. This is a recommended option +config SND_SOC_INTEL_SKYLAKE + tristate "All Skylake/SST Platforms" + depends on PCI && ACPI + select SND_SOC_INTEL_SKL + select SND_SOC_INTEL_APL + select SND_SOC_INTEL_KBL + select SND_SOC_INTEL_GLK + select SND_SOC_INTEL_CNL + select SND_SOC_INTEL_CFL + help + This is a backwards-compatible option to select all devices + supported by the Intel SST/Skylake driver. This option is no + longer recommended and will be deprecated when the SOF + driver is introduced. Distributions should explicitly + select which platform uses this driver. + +config SND_SOC_INTEL_SKL + tristate "Skylake Platforms" + depends on PCI && ACPI + select SND_SOC_INTEL_SKYLAKE_FAMILY + help + If you have a Intel Skylake platform with the DSP enabled + in the BIOS then enable this option by saying Y or m. + +config SND_SOC_INTEL_APL + tristate "Broxton/ApolloLake Platforms" + depends on PCI && ACPI + select SND_SOC_INTEL_SKYLAKE_FAMILY + help + If you have a Intel Broxton/ApolloLake platform with the DSP + enabled in the BIOS then enable this option by saying Y or m. + +config SND_SOC_INTEL_KBL + tristate "Kabylake Platforms" + depends on PCI && ACPI + select SND_SOC_INTEL_SKYLAKE_FAMILY + help + If you have a Intel Kabylake platform with the DSP + enabled in the BIOS then enable this option by saying Y or m. + +config SND_SOC_INTEL_GLK + tristate "GeminiLake Platforms" + depends on PCI && ACPI + select SND_SOC_INTEL_SKYLAKE_FAMILY + help + If you have a Intel GeminiLake platform with the DSP + enabled in the BIOS then enable this option by saying Y or m. + +config SND_SOC_INTEL_CNL + tristate "CannonLake/WhiskyLake Platforms" + depends on PCI && ACPI + select SND_SOC_INTEL_SKYLAKE_FAMILY + help + If you have a Intel CNL/WHL platform with the DSP + enabled in the BIOS then enable this option by saying Y or m. + +config SND_SOC_INTEL_CFL + tristate "CoffeeLake Platforms" + depends on PCI && ACPI + select SND_SOC_INTEL_SKYLAKE_FAMILY + help + If you have a Intel CoffeeLake platform with the DSP + enabled in the BIOS then enable this option by saying Y or m. + +config SND_SOC_INTEL_SKYLAKE_FAMILY + tristate + select SND_SOC_INTEL_SKYLAKE_COMMON + +if SND_SOC_INTEL_SKYLAKE_FAMILY + config SND_SOC_INTEL_SKYLAKE_SSP_CLK tristate -config SND_SOC_INTEL_SKYLAKE - tristate "SKL/BXT/KBL/GLK/CNL... Platforms" - depends on PCI && ACPI +config SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC + bool "HDAudio codec support" + help + If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ + GeminiLake or CannonLake platform with an HDaudio codec + then enable this option by saying Y + +config SND_SOC_INTEL_SKYLAKE_COMMON + tristate select SND_HDA_EXT_CORE select SND_HDA_DSP_LOADER select SND_SOC_TOPOLOGY select SND_SOC_INTEL_SST + select SND_SOC_HDAC_HDA if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC select SND_SOC_ACPI_INTEL_MATCH help If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ GeminiLake or CannonLake platform with the DSP enabled in the BIOS then enable this option by saying Y or m. +endif ## SND_SOC_INTEL_SKYLAKE_FAMILY + config SND_SOC_ACPI_INTEL_MATCH tristate select SND_SOC_ACPI if ACPI diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 6c36da560877..91a2436ce952 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -399,7 +399,13 @@ static int sst_media_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + int ret; + + ret = + snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(params)); + if (ret) + return ret; memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); return 0; } @@ -765,7 +771,7 @@ static int sst_soc_prepare(struct device *dev) snd_soc_poweroff(drv->soc_card->dev); /* set the SSPs to idle */ - list_for_each_entry(rtd, &drv->soc_card->rtd_list, list) { + for_each_card_rtds(drv->soc_card, rtd) { struct snd_soc_dai *dai = rtd->cpu_dai; if (dai->active) { @@ -786,7 +792,7 @@ static void sst_soc_complete(struct device *dev) return; /* restart SSPs */ - list_for_each_entry(rtd, &drv->soc_card->rtd_list, list) { + for_each_card_rtds(drv->soc_card, rtd) { struct snd_soc_dai *dai = rtd->cpu_dai; if (dai->active) { diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index c90b04cc071d..ac542535b9d5 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -341,6 +341,10 @@ static int sst_acpi_probe(struct platform_device *pdev) byt_rvp_platform_data.res_info = &bytcr_res_info; } + /* update machine parameters */ + mach->mach_params.acpi_ipc_irq_index = + pdata->res_info->acpi_ipc_irq_index; + plat_dev = platform_device_register_data(dev, pdata->platform, -1, NULL, 0); if (IS_ERR(plat_dev)) { diff --git a/sound/soc/intel/atom/sst/sst_loader.c b/sound/soc/intel/atom/sst/sst_loader.c index 27413ebae956..b8c456753f01 100644 --- a/sound/soc/intel/atom/sst/sst_loader.c +++ b/sound/soc/intel/atom/sst/sst_loader.c @@ -354,14 +354,14 @@ static int sst_request_fw(struct intel_sst_drv *sst) const struct firmware *fw; retval = request_firmware(&fw, sst->firmware_name, sst->dev); - if (fw == NULL) { - dev_err(sst->dev, "fw is returning as null\n"); - return -EINVAL; - } if (retval) { dev_err(sst->dev, "request fw failed %d\n", retval); return retval; } + if (fw == NULL) { + dev_err(sst->dev, "fw is returning as null\n"); + return -EINVAL; + } mutex_lock(&sst->sst_lock); retval = sst_cache_and_parse_fw(sst, fw); mutex_unlock(&sst->sst_lock); diff --git a/sound/soc/intel/atom/sst/sst_pvt.c b/sound/soc/intel/atom/sst/sst_pvt.c index af93244b4868..00a37a09dc9b 100644 --- a/sound/soc/intel/atom/sst/sst_pvt.c +++ b/sound/soc/intel/atom/sst/sst_pvt.c @@ -166,11 +166,11 @@ int sst_create_ipc_msg(struct ipc_post **arg, bool large) { struct ipc_post *msg; - msg = kzalloc(sizeof(struct ipc_post), GFP_ATOMIC); + msg = kzalloc(sizeof(*msg), GFP_KERNEL); if (!msg) return -ENOMEM; if (large) { - msg->mailbox_data = kzalloc(SST_MAILBOX_SIZE, GFP_ATOMIC); + msg->mailbox_data = kzalloc(SST_MAILBOX_SIZE, GFP_KERNEL); if (!msg->mailbox_data) { kfree(msg); return -ENOMEM; diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index cccda87f4b34..0a7e40d06395 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -172,7 +172,7 @@ config SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH endif ## SND_SST_ATOM_HIFI2_PLATFORM -if SND_SOC_INTEL_SKYLAKE +if SND_SOC_INTEL_SKL config SND_SOC_INTEL_SKL_RT286_MACH tristate "SKL with RT286 I2S mode" @@ -212,6 +212,10 @@ config SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH Say Y or m if you have such a device. This is a recommended option. If unsure select "N". +endif ## SND_SOC_INTEL_SKL + +if SND_SOC_INTEL_APL + config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH tristate "Broxton with DA7219 and MAX98357A in I2S Mode" depends on MFD_INTEL_LPSS && I2C && ACPI @@ -239,6 +243,10 @@ config SND_SOC_INTEL_BXT_RT298_MACH Say Y or m if you have such a device. This is a recommended option. If unsure select "N". +endif ## SND_SOC_INTEL_APL + +if SND_SOC_INTEL_KBL + config SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH tristate "KBL with RT5663 and MAX98927 in I2S Mode" depends on MFD_INTEL_LPSS && I2C && ACPI @@ -279,8 +287,34 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH This adds support for ASoC Onboard Codec I2S machine driver. This will create an alsa sound card for DA7219 + MAX98357A I2S audio codec. Say Y if you have such a device. + +config SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH + tristate "KBL with DA7219 and MAX98927 in I2S Mode" + depends on MFD_INTEL_LPSS && I2C && ACPI + select SND_SOC_DA7219 + select SND_SOC_MAX98927 + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + help + This adds support for ASoC Onboard Codec I2S machine driver. This will + create an alsa sound card for DA7219 + MAX98927 I2S audio codec. + Say Y if you have such a device. If unsure select "N". +config SND_SOC_INTEL_KBL_RT5660_MACH + tristate "KBL with RT5660 in I2S Mode" + depends on MFD_INTEL_LPSS && I2C && ACPI + select SND_SOC_RT5660 + select SND_SOC_HDAC_HDMI + help + This adds support for ASoC Onboard Codec I2S machine driver. This will + create an alsa sound card for RT5660 I2S audio codec. + Say Y if you have such a device. + +endif ## SND_SOC_INTEL_KBL + +if SND_SOC_INTEL_GLK + config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH tristate "GLK with RT5682 and MAX98357A in I2S Mode" depends on MFD_INTEL_LPSS && I2C && ACPI @@ -295,6 +329,20 @@ config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH Say Y if you have such a device. If unsure select "N". -endif ## SND_SOC_INTEL_SKYLAKE +endif ## SND_SOC_INTEL_GLK + +if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC + +config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH + tristate "SKL/KBL/BXT/APL with HDA Codecs" + select SND_SOC_HDAC_HDMI + # SND_SOC_HDAC_HDA is already selected + help + This adds support for ASoC machine driver for Intel platforms + SKL/KBL/BXT/APL with iDisp, HDA audio codecs. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + +endif ## SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC endif ## SND_SOC_INTEL_MACH diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 87ef8b4058e5..bf072ea299b7 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -17,9 +17,12 @@ snd-soc-sst-byt-cht-da7213-objs := bytcht_da7213.o snd-soc-sst-byt-cht-es8316-objs := bytcht_es8316.o snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o +snd-soc-kbl_da7219_max98927-objs := kbl_da7219_max98927.o snd-soc-kbl_rt5663_max98927-objs := kbl_rt5663_max98927.o snd-soc-kbl_rt5663_rt5514_max98927-objs := kbl_rt5663_rt5514_max98927.o +snd-soc-kbl_rt5660-objs := kbl_rt5660.o snd-soc-skl_rt286-objs := skl_rt286.o +snd-soc-skl_hda_dsp-objs := skl_hda_dsp_generic.o skl_hda_dsp_common.o snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o @@ -41,8 +44,11 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_DA7213_MACH) += snd-soc-sst-byt-cht-da7213.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_ES8316_MACH) += snd-soc-sst-byt-cht-es8316.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) += snd-soc-sst-byt-cht-nocodec.o obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH) += snd-soc-kbl_da7219_max98357a.o +obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH) += snd-soc-kbl_da7219_max98927.o obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH) += snd-soc-kbl_rt5663_max98927.o obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH) += snd-soc-kbl_rt5663_rt5514_max98927.o +obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5660_MACH) += snd-soc-kbl_rt5660.o obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o +obj-$(CONFIG_SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH) += snd-soc-skl_hda_dsp.o diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index 7b0ee67b4fc8..99f2a0156ae8 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -192,7 +192,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .stream_name = "Loopback", .cpu_dai_name = "Loopback Pin", .platform_name = "haswell-pcm-audio", - .dynamic = 0, + .dynamic = 1, .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, @@ -223,7 +223,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { static int broadwell_suspend(struct snd_soc_card *card){ struct snd_soc_component *component; - list_for_each_entry(component, &card->component_dev_list, card_list) { + for_each_card_components(card, component) { if (!strcmp(component->name, "i2c-INT343A:00")) { dev_dbg(component->dev, "disabling jack detect before going to suspend.\n"); @@ -237,7 +237,7 @@ static int broadwell_suspend(struct snd_soc_card *card){ static int broadwell_resume(struct snd_soc_card *card){ struct snd_soc_component *component; - list_for_each_entry(component, &card->component_dev_list, card_list) { + for_each_card_components(card, component) { if (!strcmp(component->name, "i2c-INT343A:00")) { dev_dbg(component->dev, "enabling jack detect for resume.\n"); diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index b6dc524830b2..a22366ce33c4 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -29,7 +29,6 @@ #include <linux/input.h> #include <linux/slab.h> #include <asm/cpu_device_id.h> -#include <asm/platform_sst_audio.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -674,6 +673,33 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, + { /* Point of View Mobii TAB-P1005W-232 (V2.0) */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "POV"), + DMI_EXACT_MATCH(DMI_BOARD_NAME, "I102A"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, + { + /* Prowise PT301 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Prowise"), + DMI_MATCH(DMI_PRODUCT_NAME, "PT301"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "TECLAST"), @@ -1048,7 +1074,7 @@ static int byt_rt5640_suspend(struct snd_soc_card *card) if (!BYT_RT5640_JDSRC(byt_rt5640_quirk)) return 0; - list_for_each_entry(component, &card->component_dev_list, card_list) { + for_each_card_components(card, component) { if (!strcmp(component->name, byt_rt5640_codec_name)) { dev_dbg(component->dev, "disabling jack detect before suspend\n"); snd_soc_component_set_jack(component, NULL, NULL); @@ -1067,7 +1093,7 @@ static int byt_rt5640_resume(struct snd_soc_card *card) if (!BYT_RT5640_JDSRC(byt_rt5640_quirk)) return 0; - list_for_each_entry(component, &card->component_dev_list, card_list) { + for_each_card_components(card, component) { if (!strcmp(component->name, byt_rt5640_codec_name)) { dev_dbg(component->dev, "re-enabling jack detect after resume\n"); snd_soc_component_set_jack(component, &priv->jack, NULL); @@ -1152,10 +1178,7 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) * (will be overridden if DMI quirk is detected) */ if (is_valleyview()) { - struct sst_platform_info *p_info = mach->pdata; - const struct sst_res_info *res_info = p_info->res_info; - - if (res_info->acpi_ipc_irq_index == 0) + if (mach->mach_params.acpi_ipc_irq_index == 0) is_bytcr = true; } diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index f8a68bdb3885..e528995668b7 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -32,7 +32,6 @@ #include <linux/slab.h> #include <asm/cpu_device_id.h> #include <asm/intel-family.h> -#include <asm/platform_sst_audio.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -742,7 +741,7 @@ static int byt_rt5651_suspend(struct snd_soc_card *card) if (!BYT_RT5651_JDSRC(byt_rt5651_quirk)) return 0; - list_for_each_entry(component, &card->component_dev_list, card_list) { + for_each_card_components(card, component) { if (!strcmp(component->name, byt_rt5651_codec_name)) { dev_dbg(component->dev, "disabling jack detect before suspend\n"); snd_soc_component_set_jack(component, NULL, NULL); @@ -761,7 +760,7 @@ static int byt_rt5651_resume(struct snd_soc_card *card) if (!BYT_RT5651_JDSRC(byt_rt5651_quirk)) return 0; - list_for_each_entry(component, &card->component_dev_list, card_list) { + for_each_card_components(card, component) { if (!strcmp(component->name, byt_rt5651_codec_name)) { dev_dbg(component->dev, "re-enabling jack detect after resume\n"); snd_soc_component_set_jack(component, &priv->jack, NULL); @@ -787,7 +786,7 @@ static struct snd_soc_card byt_rt5651_card = { }; static const struct x86_cpu_id baytrail_cpu_ids[] = { - { X86_VENDOR_INTEL, 6, INTEL_FAM6_ATOM_SILVERMONT1 }, /* Valleyview */ + { X86_VENDOR_INTEL, 6, INTEL_FAM6_ATOM_SILVERMONT }, /* Valleyview */ {} }; @@ -920,10 +919,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) * (will be overridden if DMI quirk is detected) */ if (x86_match_cpu(baytrail_cpu_ids)) { - struct sst_platform_info *p_info = mach->pdata; - const struct sst_res_info *res_info = p_info->res_info; - - if (res_info->acpi_ipc_irq_index == 0) + if (mach->mach_params.acpi_ipc_irq_index == 0) is_bytcr = true; } diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index db6976f4ddaa..08a5152e635a 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -19,6 +19,7 @@ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ */ +#include <linux/dmi.h> #include <linux/module.h> #include <linux/platform_device.h> #include <linux/slab.h> @@ -35,6 +36,8 @@ #define CHT_PLAT_CLK_3_HZ 19200000 #define CHT_CODEC_DAI "HiFi" +#define QUIRK_PMC_PLT_CLK_0 0x01 + struct cht_mc_private { struct clk *mclk; struct snd_soc_jack jack; @@ -385,11 +388,43 @@ static struct snd_soc_card snd_soc_card_cht = { .num_controls = ARRAY_SIZE(cht_mc_controls), }; +static const struct dmi_system_id cht_max98090_quirk_table[] = { + { + /* Clapper model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Clapper"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Gnawty model Chromebook (Acer Chromebook CB3-111) */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Gnawty"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Swanky model Chromebook (Toshiba Chromebook 2) */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Swanky"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + {} +}; + static int snd_cht_mc_probe(struct platform_device *pdev) { + const struct dmi_system_id *dmi_id; struct device *dev = &pdev->dev; int ret_val = 0; struct cht_mc_private *drv; + const char *mclk_name; + int quirks = 0; + + dmi_id = dmi_first_match(cht_max98090_quirk_table); + if (dmi_id) + quirks = (unsigned long)dmi_id->driver_data; drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); if (!drv) @@ -411,11 +446,16 @@ static int snd_cht_mc_probe(struct platform_device *pdev) snd_soc_card_cht.dev = &pdev->dev; snd_soc_card_set_drvdata(&snd_soc_card_cht, drv); - drv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); + if (quirks & QUIRK_PMC_PLT_CLK_0) + mclk_name = "pmc_plt_clk_0"; + else + mclk_name = "pmc_plt_clk_3"; + + drv->mclk = devm_clk_get(&pdev->dev, mclk_name); if (IS_ERR(drv->mclk)) { dev_err(&pdev->dev, - "Failed to get MCLK from pmc_plt_clk_3: %ld\n", - PTR_ERR(drv->mclk)); + "Failed to get MCLK from %s: %ld\n", + mclk_name, PTR_ERR(drv->mclk)); return PTR_ERR(drv->mclk); } diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index f5a5ea6a093c..250a356a0cbf 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -27,7 +27,6 @@ #include <linux/dmi.h> #include <linux/slab.h> #include <asm/cpu_device_id.h> -#include <asm/platform_sst_audio.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -585,10 +584,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) * (will be overridden if DMI quirk is detected) */ if (is_valleyview()) { - struct sst_platform_info *p_info = mach->pdata; - const struct sst_res_info *res_info = p_info->res_info; - - if (res_info->acpi_ipc_irq_index == 0) + if (mach->mach_params.acpi_ipc_irq_index == 0) is_bytcr = true; } diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index e5aa13058dd7..9de64f447e7b 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -16,6 +16,7 @@ * General Public License for more details. */ +#include <linux/input.h> #include <linux/module.h> #include <linux/platform_device.h> #include <linux/slab.h> @@ -212,6 +213,10 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) if (ret) return ret; + snd_jack_set_key(ctx->headset.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(ctx->headset.jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(ctx->headset.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + rt5670_set_jack_detect(component, &ctx->headset); if (ctx->mclk) { /* @@ -342,7 +347,7 @@ static int cht_suspend_pre(struct snd_soc_card *card) struct snd_soc_component *component; struct cht_mc_private *ctx = snd_soc_card_get_drvdata(card); - list_for_each_entry(component, &card->component_dev_list, card_list) { + for_each_card_components(card, component) { if (!strncmp(component->name, ctx->codec_name, sizeof(ctx->codec_name))) { @@ -359,7 +364,7 @@ static int cht_resume_post(struct snd_soc_card *card) struct snd_soc_component *component; struct cht_mc_private *ctx = snd_soc_card_get_drvdata(card); - list_for_each_entry(component, &card->component_dev_list, card_list) { + for_each_card_components(card, component) { if (!strncmp(component->name, ctx->codec_name, sizeof(ctx->codec_name))) { @@ -398,7 +403,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) const char *i2c_name; int i; - drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC); + drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); if (!drv) return -ENOMEM; diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index c4b94e2617c5..8f83b182c4f9 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -55,39 +55,6 @@ enum { GLK_DPCM_AUDIO_HDMI3_PB, }; -static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - struct snd_soc_dapm_context *dapm = w->dapm; - struct snd_soc_card *card = dapm->card; - struct snd_soc_dai *codec_dai; - int ret = 0; - - codec_dai = snd_soc_card_get_codec_dai(card, GLK_REALTEK_CODEC_DAI); - if (!codec_dai) { - dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n"); - return -EIO; - } - - if (SND_SOC_DAPM_EVENT_OFF(event)) { - ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0); - if (ret) - dev_err(card->dev, "failed to stop sysclk: %d\n", ret); - } else if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK, - GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ); - if (ret < 0) { - dev_err(card->dev, "can't set codec pll: %d\n", ret); - return ret; - } - } - - if (ret) - dev_err(card->dev, "failed to start internal clk: %d\n", ret); - - return ret; -} - static const struct snd_kcontrol_new geminilake_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), @@ -102,14 +69,10 @@ static const struct snd_soc_dapm_widget geminilake_widgets[] = { SND_SOC_DAPM_SPK("HDMI1", NULL), SND_SOC_DAPM_SPK("HDMI2", NULL), SND_SOC_DAPM_SPK("HDMI3", NULL), - SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, - platform_clock_control, SND_SOC_DAPM_PRE_PMU | - SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route geminilake_map[] = { /* HP jack connectors - unknown if we have jack detection */ - { "Headphone Jack", NULL, "Platform Clock" }, { "Headphone Jack", NULL, "HPOL" }, { "Headphone Jack", NULL, "HPOR" }, @@ -117,7 +80,6 @@ static const struct snd_soc_dapm_route geminilake_map[] = { { "Spk", NULL, "Speaker" }, /* other jacks */ - { "Headset Mic", NULL, "Platform Clock" }, { "IN1P", NULL, "Headset Mic" }, /* digital mics */ @@ -177,6 +139,13 @@ static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_jack *jack; int ret; + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK, + GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } + /* Configure sysclk for codec */ ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1, RT5682_PLL_FREQ, SND_SOC_CLOCK_IN); @@ -603,7 +572,7 @@ static int geminilake_audio_probe(struct platform_device *pdev) { struct glk_card_private *ctx; - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) return -ENOMEM; diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index eab1f439dd3f..a4022983a7ce 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -146,7 +146,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { .stream_name = "Loopback", .cpu_dai_name = "Loopback Pin", .platform_name = "haswell-pcm-audio", - .dynamic = 0, + .dynamic = 1, .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c new file mode 100644 index 000000000000..723a4935ed76 --- /dev/null +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -0,0 +1,983 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2018 Intel Corporation. + +/* + * Intel Kabylake I2S Machine Driver with MAX98927 & DA7219 Codecs + * + * Modified from: + * Intel Kabylake I2S Machine driver supporting MAX98927 and + * RT5663 codecs + */ + +#include <linux/input.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "../../codecs/da7219.h" +#include "../../codecs/hdac_hdmi.h" +#include "../skylake/skl.h" +#include "../../codecs/da7219-aad.h" + +#define KBL_DIALOG_CODEC_DAI "da7219-hifi" +#define MAX98927_CODEC_DAI "max98927-aif1" +#define MAXIM_DEV0_NAME "i2c-MX98927:00" +#define MAXIM_DEV1_NAME "i2c-MX98927:01" +#define DUAL_CHANNEL 2 +#define QUAD_CHANNEL 4 +#define NAME_SIZE 32 + +static struct snd_soc_card *kabylake_audio_card; +static struct snd_soc_jack kabylake_hdmi[3]; + +struct kbl_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct kbl_codec_private { + struct snd_soc_jack kabylake_headset; + struct list_head hdmi_pcm_list; +}; + +enum { + KBL_DPCM_AUDIO_PB = 0, + KBL_DPCM_AUDIO_CP, + KBL_DPCM_AUDIO_ECHO_REF_CP, + KBL_DPCM_AUDIO_REF_CP, + KBL_DPCM_AUDIO_DMIC_CP, + KBL_DPCM_AUDIO_HDMI1_PB, + KBL_DPCM_AUDIO_HDMI2_PB, + KBL_DPCM_AUDIO_HDMI3_PB, + KBL_DPCM_AUDIO_HS_PB, +}; + +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + int ret = 0; + + codec_dai = snd_soc_card_get_codec_dai(card, KBL_DIALOG_CODEC_DAI); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n"); + return -EIO; + } + + /* Configure sysclk for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, 24576000, + SND_SOC_CLOCK_IN); + if (ret) { + dev_err(card->dev, "can't set codec sysclk configuration\n"); + return ret; + } + + if (SND_SOC_DAPM_EVENT_OFF(event)) { + ret = snd_soc_dai_set_pll(codec_dai, 0, + DA7219_SYSCLK_MCLK, 0, 0); + if (ret) + dev_err(card->dev, "failed to stop PLL: %d\n", ret); + } else if (SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_PLL_SRM, + 0, DA7219_PLL_FREQ_OUT_98304); + if (ret) + dev_err(card->dev, "failed to start PLL: %d\n", ret); + } + + return ret; +} + +static const struct snd_kcontrol_new kabylake_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + +static const struct snd_soc_dapm_widget kabylake_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), + SND_SOC_DAPM_SPK("DP", NULL), + SND_SOC_DAPM_SPK("HDMI", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route kabylake_map[] = { + /* speaker */ + { "Left Spk", NULL, "Left BE_OUT" }, + { "Right Spk", NULL, "Right BE_OUT" }, + + /* other jacks */ + { "DMic", NULL, "SoC DMIC" }, + + { "HDMI", NULL, "hif5 Output" }, + { "DP", NULL, "hif6 Output" }, + + /* CODEC BE connections */ + { "Left HiFi Playback", NULL, "ssp0 Tx" }, + { "Right HiFi Playback", NULL, "ssp0 Tx" }, + { "ssp0 Tx", NULL, "spk_out" }, + + /* IV feedback path */ + { "codec0_fb_in", NULL, "ssp0 Rx"}, + { "ssp0 Rx", NULL, "Left HiFi Capture" }, + { "ssp0 Rx", NULL, "Right HiFi Capture" }, + + /* AEC capture path */ + { "echo_ref_out", NULL, "ssp0 Rx" }, + + /* DMIC */ + { "dmic01_hifi", NULL, "DMIC01 Rx" }, + { "DMIC01 Rx", NULL, "DMIC AIF" }, + + { "hifi1", NULL, "iDisp1 Tx" }, + { "iDisp1 Tx", NULL, "iDisp1_out" }, + { "hifi2", NULL, "iDisp2 Tx" }, + { "iDisp2 Tx", NULL, "iDisp2_out" }, + { "hifi3", NULL, "iDisp3 Tx"}, + { "iDisp3 Tx", NULL, "iDisp3_out"}, +}; + +static const struct snd_soc_dapm_route kabylake_ssp1_map[] = { + { "Headphone Jack", NULL, "HPL" }, + { "Headphone Jack", NULL, "HPR" }, + + /* other jacks */ + { "MIC", NULL, "Headset Mic" }, + + /* CODEC BE connections */ + { "Playback", NULL, "ssp1 Tx" }, + { "ssp1 Tx", NULL, "codec1_out" }, + + { "hs_in", NULL, "ssp1 Rx" }, + { "ssp1 Rx", NULL, "Capture" }, + + { "Headphone Jack", NULL, "Platform Clock" }, + { "Headset Mic", NULL, "Platform Clock" }, +}; + +static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *runtime = substream->private_data; + int ret = 0, j; + + for (j = 0; j < runtime->num_codecs; j++) { + struct snd_soc_dai *codec_dai = runtime->codec_dais[j]; + + if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16); + if (ret < 0) { + dev_err(runtime->dev, "DEV0 TDM slot err:%d\n", ret); + return ret; + } + } + if (!strcmp(codec_dai->component->name, MAXIM_DEV1_NAME)) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xC0, 3, 8, 16); + if (ret < 0) { + dev_err(runtime->dev, "DEV1 TDM slot err:%d\n", ret); + return ret; + } + } + } + + return 0; +} + +static struct snd_soc_ops kabylake_ssp0_ops = { + .hw_params = kabylake_ssp0_hw_params, +}; + +static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + struct snd_soc_dpcm *dpcm = container_of( + params, struct snd_soc_dpcm, hw_params); + struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; + struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; + + /* + * The ADSP will convert the FE rate to 48k, stereo, 24 bit + */ + if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || + !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || + !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + snd_mask_none(fmt); + snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + } + + /* + * The speaker on the SSP0 supports S16_LE and not S24_LE. + * thus changing the mask here + */ + if (!strcmp(be_dai_link->name, "SSP0-Codec")) + snd_mask_set(fmt, SNDRV_PCM_FORMAT_S16_LE); + + return 0; +} + +static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_jack *jack; + struct snd_soc_card *card = rtd->card; + int ret; + + + ret = snd_soc_dapm_add_routes(&card->dapm, + kabylake_ssp1_map, + ARRAY_SIZE(kabylake_ssp1_map)); + + /* + * Headset buttons map to the google Reference headset. + * These can be configured by userspace. + */ + ret = snd_soc_card_jack_new(kabylake_audio_card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, + &ctx->kabylake_headset, NULL, 0); + if (ret) { + dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); + return ret; + } + + jack = &ctx->kabylake_headset; + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + + da7219_aad_jack_det(component, &ctx->kabylake_headset); + + ret = snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "SoC DMIC"); + if (ret) + dev_err(rtd->dev, "SoC DMIC - Ignore suspend failed %d\n", ret); + + return ret; +} + +static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device) +{ + struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *dai = rtd->codec_dai; + struct kbl_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + pcm->device = device; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; +} + +static int kabylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd) +{ + return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI1_PB); +} + +static int kabylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd) +{ + return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI2_PB); +} + +static int kabylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) +{ + return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI3_PB); +} + +static int kabylake_da7219_fe_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dapm_context *dapm; + struct snd_soc_component *component = rtd->cpu_dai->component; + + dapm = snd_soc_component_get_dapm(component); + snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); + + return 0; +} + +static const unsigned int rates[] = { + 48000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static const unsigned int channels[] = { + DUAL_CHANNEL, +}; + +static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, +}; + +static unsigned int channels_quad[] = { + QUAD_CHANNEL, +}; + +static struct snd_pcm_hw_constraint_list constraints_channels_quad = { + .count = ARRAY_SIZE(channels_quad), + .list = channels_quad, + .mask = 0, +}; + +static int kbl_fe_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + /* + * On this platform for PCM device we support, + * 48Khz + * stereo + * 16 bit audio + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); + + return 0; +} + +static const struct snd_soc_ops kabylake_da7219_fe_ops = { + .startup = kbl_fe_startup, +}; + +static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* + * set BE channel constraint as user FE channels + */ + + if (params_channels(params) == 2) + channels->min = channels->max = 2; + else + channels->min = channels->max = 4; + + return 0; +} + +static int kabylake_dmic_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels_quad); + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); +} + +static struct snd_soc_ops kabylake_dmic_ops = { + .startup = kabylake_dmic_startup, +}; + +static const unsigned int rates_16000[] = { + 16000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_16000 = { + .count = ARRAY_SIZE(rates_16000), + .list = rates_16000, +}; + +static const unsigned int ch_mono[] = { + 1, +}; +static const struct snd_pcm_hw_constraint_list constraints_refcap = { + .count = ARRAY_SIZE(ch_mono), + .list = ch_mono, +}; + +static int kabylake_refcap_startup(struct snd_pcm_substream *substream) +{ + substream->runtime->hw.channels_max = 1; + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_refcap); + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_16000); +} + + +static struct snd_soc_ops skylake_refcap_ops = { + .startup = kabylake_refcap_startup, +}; + +static struct snd_soc_codec_conf max98927_codec_conf[] = { + + { + .dev_name = MAXIM_DEV0_NAME, + .name_prefix = "Right", + }, + + { + .dev_name = MAXIM_DEV1_NAME, + .name_prefix = "Left", + }, +}; + +static struct snd_soc_dai_link_component ssp0_codec_components[] = { + { /* Left */ + .name = MAXIM_DEV0_NAME, + .dai_name = MAX98927_CODEC_DAI, + }, + + { /* For Right */ + .name = MAXIM_DEV1_NAME, + .dai_name = MAX98927_CODEC_DAI, + }, + +}; + +/* kabylake digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link kabylake_dais[] = { + /* Front End DAI links */ + [KBL_DPCM_AUDIO_PB] = { + .name = "Kbl Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .init = kabylake_da7219_fe_init, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .ops = &kabylake_da7219_fe_ops, + }, + [KBL_DPCM_AUDIO_CP] = { + .name = "Kbl Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + .ops = &kabylake_da7219_fe_ops, + }, + [KBL_DPCM_AUDIO_ECHO_REF_CP] = { + .name = "Kbl Audio Echo Reference cap", + .stream_name = "Echoreference Capture", + .cpu_dai_name = "Echoref Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .capture_only = 1, + .nonatomic = 1, + }, + [KBL_DPCM_AUDIO_REF_CP] = { + .name = "Kbl Audio Reference cap", + .stream_name = "Wake on Voice", + .cpu_dai_name = "Reference Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &skylake_refcap_ops, + }, + [KBL_DPCM_AUDIO_DMIC_CP] = { + .name = "Kbl Audio DMIC cap", + .stream_name = "dmiccap", + .cpu_dai_name = "DMIC Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &kabylake_dmic_ops, + }, + [KBL_DPCM_AUDIO_HDMI1_PB] = { + .name = "Kbl HDMI Port1", + .stream_name = "Hdmi1", + .cpu_dai_name = "HDMI1 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [KBL_DPCM_AUDIO_HDMI2_PB] = { + .name = "Kbl HDMI Port2", + .stream_name = "Hdmi2", + .cpu_dai_name = "HDMI2 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [KBL_DPCM_AUDIO_HDMI3_PB] = { + .name = "Kbl HDMI Port3", + .stream_name = "Hdmi3", + .cpu_dai_name = "HDMI3 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .init = NULL, + .nonatomic = 1, + .dynamic = 1, + }, + [KBL_DPCM_AUDIO_HS_PB] = { + .name = "Kbl Audio Headset Playback", + .stream_name = "Headset Audio", + .cpu_dai_name = "System Pin2", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .nonatomic = 1, + .dynamic = 1, + .init = kabylake_da7219_fe_init, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .ops = &kabylake_da7219_fe_ops, + + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "SSP0-Codec", + .id = 0, + .cpu_dai_name = "SSP0 Pin", + .platform_name = "0000:00:1f.3", + .no_pcm = 1, + .codecs = ssp0_codec_components, + .num_codecs = ARRAY_SIZE(ssp0_codec_components), + .dai_fmt = SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = kabylake_ssp_fixup, + .ops = &kabylake_ssp0_ops, + }, + { + /* SSP1 - Codec */ + .name = "SSP1-Codec", + .id = 1, + .cpu_dai_name = "SSP1 Pin", + .platform_name = "0000:00:1f.3", + .no_pcm = 1, + .codec_name = "i2c-DLGS7219:00", + .codec_dai_name = KBL_DIALOG_CODEC_DAI, + .init = kabylake_da7219_codec_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = kabylake_ssp_fixup, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + { + .name = "dmic01", + .id = 2, + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:1f.3", + .be_hw_params_fixup = kabylake_dmic_fixup, + .ignore_suspend = 1, + .dpcm_capture = 1, + .no_pcm = 1, + }, + { + .name = "iDisp1", + .id = 3, + .cpu_dai_name = "iDisp1 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi1", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = kabylake_hdmi1_init, + .no_pcm = 1, + }, + { + .name = "iDisp2", + .id = 4, + .cpu_dai_name = "iDisp2 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi2", + .platform_name = "0000:00:1f.3", + .init = kabylake_hdmi2_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp3", + .id = 5, + .cpu_dai_name = "iDisp3 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi3", + .platform_name = "0000:00:1f.3", + .init = kabylake_hdmi3_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, +}; + +/* kabylake digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link kabylake_max98927_dais[] = { + /* Front End DAI links */ + [KBL_DPCM_AUDIO_PB] = { + .name = "Kbl Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .init = kabylake_da7219_fe_init, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .ops = &kabylake_da7219_fe_ops, + }, + [KBL_DPCM_AUDIO_CP] = { + .name = "Kbl Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + .ops = &kabylake_da7219_fe_ops, + }, + [KBL_DPCM_AUDIO_ECHO_REF_CP] = { + .name = "Kbl Audio Echo Reference cap", + .stream_name = "Echoreference Capture", + .cpu_dai_name = "Echoref Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .capture_only = 1, + .nonatomic = 1, + }, + [KBL_DPCM_AUDIO_REF_CP] = { + .name = "Kbl Audio Reference cap", + .stream_name = "Wake on Voice", + .cpu_dai_name = "Reference Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &skylake_refcap_ops, + }, + [KBL_DPCM_AUDIO_DMIC_CP] = { + .name = "Kbl Audio DMIC cap", + .stream_name = "dmiccap", + .cpu_dai_name = "DMIC Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &kabylake_dmic_ops, + }, + [KBL_DPCM_AUDIO_HDMI1_PB] = { + .name = "Kbl HDMI Port1", + .stream_name = "Hdmi1", + .cpu_dai_name = "HDMI1 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [KBL_DPCM_AUDIO_HDMI2_PB] = { + .name = "Kbl HDMI Port2", + .stream_name = "Hdmi2", + .cpu_dai_name = "HDMI2 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [KBL_DPCM_AUDIO_HDMI3_PB] = { + .name = "Kbl HDMI Port3", + .stream_name = "Hdmi3", + .cpu_dai_name = "HDMI3 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .init = NULL, + .nonatomic = 1, + .dynamic = 1, + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "SSP0-Codec", + .id = 0, + .cpu_dai_name = "SSP0 Pin", + .platform_name = "0000:00:1f.3", + .no_pcm = 1, + .codecs = ssp0_codec_components, + .num_codecs = ARRAY_SIZE(ssp0_codec_components), + .dai_fmt = SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = kabylake_ssp_fixup, + .ops = &kabylake_ssp0_ops, + }, + { + .name = "dmic01", + .id = 1, + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:1f.3", + .be_hw_params_fixup = kabylake_dmic_fixup, + .ignore_suspend = 1, + .dpcm_capture = 1, + .no_pcm = 1, + }, + { + .name = "iDisp1", + .id = 2, + .cpu_dai_name = "iDisp1 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi1", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = kabylake_hdmi1_init, + .no_pcm = 1, + }, + { + .name = "iDisp2", + .id = 3, + .cpu_dai_name = "iDisp2 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi2", + .platform_name = "0000:00:1f.3", + .init = kabylake_hdmi2_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp3", + .id = 4, + .cpu_dai_name = "iDisp3 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi3", + .platform_name = "0000:00:1f.3", + .init = kabylake_hdmi3_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, +}; + +static int kabylake_card_late_probe(struct snd_soc_card *card) +{ + struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(card); + struct kbl_hdmi_pcm *pcm; + struct snd_soc_component *component = NULL; + int err, i = 0; + char jack_name[NAME_SIZE]; + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + component = pcm->codec_dai->component; + snprintf(jack_name, sizeof(jack_name), + "HDMI/DP, pcm=%d Jack", pcm->device); + err = snd_soc_card_jack_new(card, jack_name, + SND_JACK_AVOUT, &kabylake_hdmi[i], + NULL, 0); + + if (err) + return err; + + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, + &kabylake_hdmi[i]); + if (err < 0) + return err; + + i++; + } + + if (!component) + return -EINVAL; + + return hdac_hdmi_jack_port_init(component, &card->dapm); + + return 0; +} + +/* kabylake audio machine driver for SPT + DA7219 */ +static struct snd_soc_card kbl_audio_card_da7219_m98927 = { + .name = "kblda7219m98927", + .owner = THIS_MODULE, + .dai_link = kabylake_dais, + .num_links = ARRAY_SIZE(kabylake_dais), + .controls = kabylake_controls, + .num_controls = ARRAY_SIZE(kabylake_controls), + .dapm_widgets = kabylake_widgets, + .num_dapm_widgets = ARRAY_SIZE(kabylake_widgets), + .dapm_routes = kabylake_map, + .num_dapm_routes = ARRAY_SIZE(kabylake_map), + .codec_conf = max98927_codec_conf, + .num_configs = ARRAY_SIZE(max98927_codec_conf), + .fully_routed = true, + .late_probe = kabylake_card_late_probe, +}; + +/* kabylake audio machine driver for Maxim98927 */ +static struct snd_soc_card kbl_audio_card_max98927 = { + .name = "kblmax98927", + .owner = THIS_MODULE, + .dai_link = kabylake_max98927_dais, + .num_links = ARRAY_SIZE(kabylake_max98927_dais), + .controls = kabylake_controls, + .num_controls = ARRAY_SIZE(kabylake_controls), + .dapm_widgets = kabylake_widgets, + .num_dapm_widgets = ARRAY_SIZE(kabylake_widgets), + .dapm_routes = kabylake_map, + .num_dapm_routes = ARRAY_SIZE(kabylake_map), + .codec_conf = max98927_codec_conf, + .num_configs = ARRAY_SIZE(max98927_codec_conf), + .fully_routed = true, + .late_probe = kabylake_card_late_probe, +}; + +static int kabylake_audio_probe(struct platform_device *pdev) +{ + struct kbl_codec_private *ctx; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); + if (!ctx) + return -ENOMEM; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + + kabylake_audio_card = + (struct snd_soc_card *)pdev->id_entry->driver_data; + + kabylake_audio_card->dev = &pdev->dev; + snd_soc_card_set_drvdata(kabylake_audio_card, ctx); + + return devm_snd_soc_register_card(&pdev->dev, kabylake_audio_card); +} + +static const struct platform_device_id kbl_board_ids[] = { + { + .name = "kbl_da7219_max98927", + .driver_data = + (kernel_ulong_t)&kbl_audio_card_da7219_m98927, + }, + { + .name = "kbl_max98927", + .driver_data = + (kernel_ulong_t)&kbl_audio_card_max98927, + }, + { } +}; + +static struct platform_driver kabylake_audio = { + .probe = kabylake_audio_probe, + .driver = { + .name = "kbl_da7219_max98927", + .pm = &snd_soc_pm_ops, + }, + .id_table = kbl_board_ids, +}; + +module_platform_driver(kabylake_audio) + +/* Module information */ +MODULE_DESCRIPTION("Audio KabyLake Machine driver for MAX98927 & DA7219"); +MODULE_AUTHOR("Mac Chiang <mac.chiang@intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:kbl_da7219_max98927"); +MODULE_ALIAS("platform:kbl_max98927"); diff --git a/sound/soc/intel/boards/kbl_rt5660.c b/sound/soc/intel/boards/kbl_rt5660.c new file mode 100644 index 000000000000..3255e0029276 --- /dev/null +++ b/sound/soc/intel/boards/kbl_rt5660.c @@ -0,0 +1,543 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2018-19 Canonical Corporation. + +/* + * Intel Kabylake I2S Machine Driver with RT5660 Codec + * + * Modified from: + * Intel Kabylake I2S Machine driver supporting MAXIM98357a and + * DA7219 codecs + * Also referred to: + * Intel Broadwell I2S Machine driver supporting RT5677 codec + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/gpio/consumer.h> +#include <linux/acpi.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "../../codecs/hdac_hdmi.h" +#include "../../codecs/rt5660.h" + +#define KBL_RT5660_CODEC_DAI "rt5660-aif1" +#define DUAL_CHANNEL 2 + +static struct snd_soc_card *kabylake_audio_card; +static struct snd_soc_jack skylake_hdmi[3]; +static struct snd_soc_jack lineout_jack; +static struct snd_soc_jack mic_jack; + +struct kbl_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct kbl_codec_private { + struct gpio_desc *gpio_lo_mute; + struct list_head hdmi_pcm_list; +}; + +enum { + KBL_DPCM_AUDIO_PB = 0, + KBL_DPCM_AUDIO_CP, + KBL_DPCM_AUDIO_HDMI1_PB, + KBL_DPCM_AUDIO_HDMI2_PB, + KBL_DPCM_AUDIO_HDMI3_PB, +}; + +#define GPIO_LINEOUT_MUTE_INDEX 0 +#define GPIO_LINEOUT_DET_INDEX 3 +#define GPIO_LINEIN_DET_INDEX 4 + +static const struct acpi_gpio_params lineout_mute_gpio = { GPIO_LINEOUT_MUTE_INDEX, 0, true }; +static const struct acpi_gpio_params lineout_det_gpio = { GPIO_LINEOUT_DET_INDEX, 0, false }; +static const struct acpi_gpio_params mic_det_gpio = { GPIO_LINEIN_DET_INDEX, 0, false }; + + +static const struct acpi_gpio_mapping acpi_rt5660_gpios[] = { + { "lineout-mute-gpios", &lineout_mute_gpio, 1 }, + { "lineout-det-gpios", &lineout_det_gpio, 1 }, + { "mic-det-gpios", &mic_det_gpio, 1 }, + { NULL }, +}; + +static struct snd_soc_jack_pin lineout_jack_pin = { + .pin = "Line Out", + .mask = SND_JACK_LINEOUT, +}; + +static struct snd_soc_jack_pin mic_jack_pin = { + .pin = "Line In", + .mask = SND_JACK_MICROPHONE, +}; + +static struct snd_soc_jack_gpio lineout_jack_gpio = { + .name = "lineout-det", + .report = SND_JACK_LINEOUT, + .debounce_time = 200, +}; + +static struct snd_soc_jack_gpio mic_jack_gpio = { + .name = "mic-det", + .report = SND_JACK_MICROPHONE, + .debounce_time = 200, +}; + +static int kabylake_5660_event_lineout(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct kbl_codec_private *priv = snd_soc_card_get_drvdata(dapm->card); + + gpiod_set_value_cansleep(priv->gpio_lo_mute, + !(SND_SOC_DAPM_EVENT_ON(event))); + + return 0; +} + +static const struct snd_kcontrol_new kabylake_rt5660_controls[] = { + SOC_DAPM_PIN_SWITCH("Line In"), + SOC_DAPM_PIN_SWITCH("Line Out"), +}; + +static const struct snd_soc_dapm_widget kabylake_rt5660_widgets[] = { + SND_SOC_DAPM_MIC("Line In", NULL), + SND_SOC_DAPM_LINE("Line Out", kabylake_5660_event_lineout), +}; + +static const struct snd_soc_dapm_route kabylake_rt5660_map[] = { + /* other jacks */ + {"IN1P", NULL, "Line In"}, + {"IN2P", NULL, "Line In"}, + {"Line Out", NULL, "LOUTR"}, + {"Line Out", NULL, "LOUTL"}, + + /* CODEC BE connections */ + { "AIF1 Playback", NULL, "ssp0 Tx"}, + { "ssp0 Tx", NULL, "codec0_out"}, + + { "codec0_in", NULL, "ssp0 Rx" }, + { "ssp0 Rx", NULL, "AIF1 Capture" }, + + { "hifi1", NULL, "iDisp1 Tx"}, + { "iDisp1 Tx", NULL, "iDisp1_out"}, + { "hifi2", NULL, "iDisp2 Tx"}, + { "iDisp2 Tx", NULL, "iDisp2_out"}, + { "hifi3", NULL, "iDisp3 Tx"}, + { "iDisp3 Tx", NULL, "iDisp3_out"}, +}; + +static int kabylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = DUAL_CHANNEL; + + /* set SSP0 to 24 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static int kabylake_rt5660_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret; + struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + ret = devm_acpi_dev_add_driver_gpios(component->dev, acpi_rt5660_gpios); + if (ret) + dev_warn(component->dev, "Failed to add driver gpios\n"); + + /* Request rt5660 GPIO for lineout mute control, return if fails */ + ctx->gpio_lo_mute = devm_gpiod_get(component->dev, "lineout-mute", + GPIOD_OUT_HIGH); + if (IS_ERR(ctx->gpio_lo_mute)) { + dev_err(component->dev, "Can't find GPIO_MUTE# gpio\n"); + return PTR_ERR(ctx->gpio_lo_mute); + } + + /* Create and initialize headphone jack, this jack is not mandatory, don't return if fails */ + ret = snd_soc_card_jack_new(rtd->card, "Lineout Jack", + SND_JACK_LINEOUT, &lineout_jack, + &lineout_jack_pin, 1); + if (ret) + dev_warn(component->dev, "Can't create Lineout jack\n"); + else { + lineout_jack_gpio.gpiod_dev = component->dev; + ret = snd_soc_jack_add_gpios(&lineout_jack, 1, + &lineout_jack_gpio); + if (ret) + dev_warn(component->dev, "Can't add Lineout jack gpio\n"); + } + + /* Create and initialize mic jack, this jack is not mandatory, don't return if fails */ + ret = snd_soc_card_jack_new(rtd->card, "Mic Jack", + SND_JACK_MICROPHONE, &mic_jack, + &mic_jack_pin, 1); + if (ret) + dev_warn(component->dev, "Can't create mic jack\n"); + else { + mic_jack_gpio.gpiod_dev = component->dev; + ret = snd_soc_jack_add_gpios(&mic_jack, 1, &mic_jack_gpio); + if (ret) + dev_warn(component->dev, "Can't add mic jack gpio\n"); + } + + /* Here we enable some dapms in advance to reduce the pop noise for recording via line-in */ + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); + snd_soc_dapm_force_enable_pin(dapm, "BST1"); + snd_soc_dapm_force_enable_pin(dapm, "BST2"); + + return 0; +} + +static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device) +{ + struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *dai = rtd->codec_dai; + struct kbl_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + pcm->device = device; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; +} + +static int kabylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd) +{ + return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI1_PB); +} + +static int kabylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd) +{ + return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI2_PB); +} + +static int kabylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) +{ + return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI3_PB); +} + +static int kabylake_rt5660_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_sysclk(codec_dai, + RT5660_SCLK_S_PLL1, params_rate(params) * 512, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, + RT5660_PLL1_S_BCLK, + params_rate(params) * 50, + params_rate(params) * 512); + if (ret < 0) + dev_err(codec_dai->dev, "can't set codec pll: %d\n", ret); + + return ret; +} + +static struct snd_soc_ops kabylake_rt5660_ops = { + .hw_params = kabylake_rt5660_hw_params, +}; + +static const unsigned int rates[] = { + 48000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static const unsigned int channels[] = { + DUAL_CHANNEL, +}; + +static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, +}; + +static int kbl_fe_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + /* + * On this platform for PCM device we support, + * 48Khz + * stereo + * 16 bit audio + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); + + return 0; +} + +static const struct snd_soc_ops kabylake_rt5660_fe_ops = { + .startup = kbl_fe_startup, +}; + +/* kabylake digital audio interface glue - connects rt5660 codec <--> CPU */ +static struct snd_soc_dai_link kabylake_rt5660_dais[] = { + /* Front End DAI links */ + [KBL_DPCM_AUDIO_PB] = { + .name = "Kbl Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .ops = &kabylake_rt5660_fe_ops, + }, + [KBL_DPCM_AUDIO_CP] = { + .name = "Kbl Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + .ops = &kabylake_rt5660_fe_ops, + }, + [KBL_DPCM_AUDIO_HDMI1_PB] = { + .name = "Kbl HDMI Port1", + .stream_name = "Hdmi1", + .cpu_dai_name = "HDMI1 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [KBL_DPCM_AUDIO_HDMI2_PB] = { + .name = "Kbl HDMI Port2", + .stream_name = "Hdmi2", + .cpu_dai_name = "HDMI2 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [KBL_DPCM_AUDIO_HDMI3_PB] = { + .name = "Kbl HDMI Port3", + .stream_name = "Hdmi3", + .cpu_dai_name = "HDMI3 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .init = NULL, + .nonatomic = 1, + .dynamic = 1, + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "SSP0-Codec", + .id = 0, + .cpu_dai_name = "SSP0 Pin", + .platform_name = "0000:00:1f.3", + .no_pcm = 1, + .codec_name = "i2c-10EC3277:00", + .codec_dai_name = KBL_RT5660_CODEC_DAI, + .init = kabylake_rt5660_codec_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = kabylake_ssp0_fixup, + .ops = &kabylake_rt5660_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + { + .name = "iDisp1", + .id = 1, + .cpu_dai_name = "iDisp1 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi1", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = kabylake_hdmi1_init, + .no_pcm = 1, + }, + { + .name = "iDisp2", + .id = 2, + .cpu_dai_name = "iDisp2 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi2", + .platform_name = "0000:00:1f.3", + .init = kabylake_hdmi2_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp3", + .id = 3, + .cpu_dai_name = "iDisp3 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi3", + .platform_name = "0000:00:1f.3", + .init = kabylake_hdmi3_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, +}; + + +#define NAME_SIZE 32 +static int kabylake_card_late_probe(struct snd_soc_card *card) +{ + struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(card); + struct kbl_hdmi_pcm *pcm; + struct snd_soc_component *component = NULL; + int err, i = 0; + char jack_name[NAME_SIZE]; + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + component = pcm->codec_dai->component; + snprintf(jack_name, sizeof(jack_name), + "HDMI/DP, pcm=%d Jack", pcm->device); + err = snd_soc_card_jack_new(card, jack_name, + SND_JACK_AVOUT, &skylake_hdmi[i], + NULL, 0); + + if (err) + return err; + + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, + &skylake_hdmi[i]); + if (err < 0) + return err; + + i++; + + } + + if (!component) + return -EINVAL; + + return hdac_hdmi_jack_port_init(component, &card->dapm); +} + +/* kabylake audio machine driver for rt5660 */ +static struct snd_soc_card kabylake_audio_card_rt5660 = { + .name = "kblrt5660", + .owner = THIS_MODULE, + .dai_link = kabylake_rt5660_dais, + .num_links = ARRAY_SIZE(kabylake_rt5660_dais), + .controls = kabylake_rt5660_controls, + .num_controls = ARRAY_SIZE(kabylake_rt5660_controls), + .dapm_widgets = kabylake_rt5660_widgets, + .num_dapm_widgets = ARRAY_SIZE(kabylake_rt5660_widgets), + .dapm_routes = kabylake_rt5660_map, + .num_dapm_routes = ARRAY_SIZE(kabylake_rt5660_map), + .fully_routed = true, + .late_probe = kabylake_card_late_probe, +}; + +static int kabylake_audio_probe(struct platform_device *pdev) +{ + struct kbl_codec_private *ctx; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); + if (!ctx) + return -ENOMEM; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + + kabylake_audio_card = + (struct snd_soc_card *)pdev->id_entry->driver_data; + + kabylake_audio_card->dev = &pdev->dev; + snd_soc_card_set_drvdata(kabylake_audio_card, ctx); + return devm_snd_soc_register_card(&pdev->dev, kabylake_audio_card); +} + +static const struct platform_device_id kbl_board_ids[] = { + { + .name = "kbl_rt5660", + .driver_data = + (kernel_ulong_t)&kabylake_audio_card_rt5660, + }, + { } +}; + +static struct platform_driver kabylake_audio = { + .probe = kabylake_audio_probe, + .driver = { + .name = "kbl_rt5660", + .pm = &snd_soc_pm_ops, + }, + .id_table = kbl_board_ids, +}; + +module_platform_driver(kabylake_audio) + +/* Module information */ +MODULE_DESCRIPTION("Audio Machine driver-RT5660 in I2S mode"); +MODULE_AUTHOR("Hui Wang <hui.wang@canonical.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:kbl_rt5660"); diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 21a6490746a6..d71475200b08 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -25,9 +25,9 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/soc-acpi.h> #include "../../codecs/rt5663.h" #include "../../codecs/hdac_hdmi.h" -#include "../skylake/skl.h" #include <linux/clk.h> #include <linux/clk-provider.h> #include <linux/clkdev.h> @@ -488,11 +488,10 @@ static int kabylake_ssp0_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; int ret = 0, j; - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; - + for_each_rtd_codec_dai(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) { /* * Use channel 4 and 5 for the first amp @@ -587,7 +586,7 @@ static int kabylake_refcap_startup(struct snd_pcm_substream *substream) &constraints_16000); } -static struct snd_soc_ops skylaye_refcap_ops = { +static struct snd_soc_ops skylake_refcap_ops = { .startup = kabylake_refcap_startup, }; @@ -656,7 +655,7 @@ static struct snd_soc_dai_link kabylake_dais[] = { .dpcm_capture = 1, .nonatomic = 1, .dynamic = 1, - .ops = &skylaye_refcap_ops, + .ops = &skylake_refcap_ops, }, [KBL_DPCM_AUDIO_DMIC_CP] = { .name = "Kbl Audio DMIC cap", @@ -970,7 +969,7 @@ static struct snd_soc_card kabylake_audio_card_rt5663 = { static int kabylake_audio_probe(struct platform_device *pdev) { struct kbl_rt5663_private *ctx; - struct skl_machine_pdata *pdata; + struct snd_soc_acpi_mach *mach; int ret; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); @@ -985,9 +984,9 @@ static int kabylake_audio_probe(struct platform_device *pdev) kabylake_audio_card->dev = &pdev->dev; snd_soc_card_set_drvdata(kabylake_audio_card, ctx); - pdata = dev_get_drvdata(&pdev->dev); - if (pdata) - dmic_constraints = pdata->dmic_num == 2 ? + mach = (&pdev->dev)->platform_data; + if (mach) + dmic_constraints = mach->mach_params.dmic_num == 2 ? &constraints_dmic_2ch : &constraints_dmic_channels; ctx->mclk = devm_clk_get(&pdev->dev, "ssp1_mclk"); diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index a892b37eab7c..7044d8c2b187 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -26,10 +26,10 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/soc-acpi.h> #include "../../codecs/rt5514.h" #include "../../codecs/rt5663.h" #include "../../codecs/hdac_hdmi.h" -#include "../skylake/skl.h" #define KBL_REALTEK_CODEC_DAI "rt5663-aif" #define KBL_REALTEK_DMIC_CODEC_DAI "rt5514-aif1" @@ -353,11 +353,10 @@ static int kabylake_ssp0_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; int ret = 0, j; - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; - + for_each_rtd_codec_dai(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name, RT5514_DEV_NAME)) { ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0, 8, 16); if (ret < 0) { @@ -649,7 +648,7 @@ static struct snd_soc_card kabylake_audio_card = { static int kabylake_audio_probe(struct platform_device *pdev) { struct kbl_codec_private *ctx; - struct skl_machine_pdata *pdata; + struct snd_soc_acpi_mach *mach; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -660,9 +659,9 @@ static int kabylake_audio_probe(struct platform_device *pdev) kabylake_audio_card.dev = &pdev->dev; snd_soc_card_set_drvdata(&kabylake_audio_card, ctx); - pdata = dev_get_drvdata(&pdev->dev); - if (pdata) - dmic_constraints = pdata->dmic_num == 2 ? + mach = (&pdev->dev)->platform_data; + if (mach) + dmic_constraints = mach->mach_params.dmic_num == 2 ? &constraints_dmic_2ch : &constraints_dmic_channels; return devm_snd_soc_register_card(&pdev->dev, &kabylake_audio_card); diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.c b/sound/soc/intel/boards/skl_hda_dsp_common.c new file mode 100644 index 000000000000..3fdbf239da74 --- /dev/null +++ b/sound/soc/intel/boards/skl_hda_dsp_common.c @@ -0,0 +1,127 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2015-18 Intel Corporation. + +/* + * Common functions used in different Intel machine drivers + */ +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "../../codecs/hdac_hdmi.h" +#include "../skylake/skl.h" +#include "skl_hda_dsp_common.h" + +#define NAME_SIZE 32 + +int skl_hda_hdmi_add_pcm(struct snd_soc_card *card, int device) +{ + struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card); + struct skl_hda_hdmi_pcm *pcm; + char dai_name[NAME_SIZE]; + + pcm = devm_kzalloc(card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + snprintf(dai_name, sizeof(dai_name), "intel-hdmi-hifi%d", + ctx->dai_index); + pcm->codec_dai = snd_soc_card_get_codec_dai(card, dai_name); + if (!pcm->codec_dai) + return -EINVAL; + + pcm->device = device; + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; +} + +/* skl_hda_digital audio interface glue - connects codec <--> CPU */ +struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS] = { + /* Back End DAI links */ + { + .name = "iDisp1", + .id = 1, + .cpu_dai_name = "iDisp1 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi1", + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp2", + .id = 2, + .cpu_dai_name = "iDisp2 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi2", + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp3", + .id = 3, + .cpu_dai_name = "iDisp3 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi3", + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "Analog Playback and Capture", + .id = 4, + .cpu_dai_name = "Analog CPU DAI", + .codec_name = "ehdaudio0D0", + .codec_dai_name = "Analog Codec DAI", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .dpcm_capture = 1, + .init = NULL, + .no_pcm = 1, + }, + { + .name = "Digital Playback and Capture", + .id = 5, + .cpu_dai_name = "Digital CPU DAI", + .codec_name = "ehdaudio0D0", + .codec_dai_name = "Digital Codec DAI", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .dpcm_capture = 1, + .init = NULL, + .no_pcm = 1, + }, +}; + +int skl_hda_hdmi_jack_init(struct snd_soc_card *card) +{ + struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_component *component = NULL; + struct skl_hda_hdmi_pcm *pcm; + char jack_name[NAME_SIZE]; + int err; + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + component = pcm->codec_dai->component; + snprintf(jack_name, sizeof(jack_name), + "HDMI/DP, pcm=%d Jack", pcm->device); + err = snd_soc_card_jack_new(card, jack_name, + SND_JACK_AVOUT, &pcm->hdmi_jack, + NULL, 0); + + if (err) + return err; + + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, + &pcm->hdmi_jack); + if (err < 0) + return err; + } + + if (!component) + return -EINVAL; + + return hdac_hdmi_jack_port_init(component, &card->dapm); +} diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.h b/sound/soc/intel/boards/skl_hda_dsp_common.h new file mode 100644 index 000000000000..87c50aff56cd --- /dev/null +++ b/sound/soc/intel/boards/skl_hda_dsp_common.h @@ -0,0 +1,38 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright(c) 2015-18 Intel Corporation. + */ + +/* + * This file defines data structures used in Machine Driver for Intel + * platforms with HDA Codecs. + */ + +#ifndef __SOUND_SOC_HDA_DSP_COMMON_H +#define __SOUND_SOC_HDA_DSP_COMMON_H +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/jack.h> + +#define HDA_DSP_MAX_BE_DAI_LINKS 5 + +struct skl_hda_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + struct snd_soc_jack hdmi_jack; + int device; +}; + +struct skl_hda_private { + struct list_head hdmi_pcm_list; + int pcm_count; + int dai_index; + const char *platform_name; +}; + +extern struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS]; +int skl_hda_hdmi_jack_init(struct snd_soc_card *card); +int skl_hda_hdmi_add_pcm(struct snd_soc_card *card, int device); + +#endif /* __SOUND_SOC_HDA_DSP_COMMON_H */ diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c new file mode 100644 index 000000000000..b9a21e64ead2 --- /dev/null +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -0,0 +1,183 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2015-18 Intel Corporation. + +/* + * Machine Driver for SKL+ platforms with DSP and iDisp, HDA Codecs + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include "../../codecs/hdac_hdmi.h" +#include "skl_hda_dsp_common.h" + +static const struct snd_soc_dapm_widget skl_hda_widgets[] = { + SND_SOC_DAPM_HP("Analog Out", NULL), + SND_SOC_DAPM_MIC("Analog In", NULL), + SND_SOC_DAPM_HP("Alt Analog Out", NULL), + SND_SOC_DAPM_MIC("Alt Analog In", NULL), + SND_SOC_DAPM_SPK("Digital Out", NULL), + SND_SOC_DAPM_MIC("Digital In", NULL), +}; + +static const struct snd_soc_dapm_route skl_hda_map[] = { + { "hifi3", NULL, "iDisp3 Tx"}, + { "iDisp3 Tx", NULL, "iDisp3_out"}, + { "hifi2", NULL, "iDisp2 Tx"}, + { "iDisp2 Tx", NULL, "iDisp2_out"}, + { "hifi1", NULL, "iDisp1 Tx"}, + { "iDisp1 Tx", NULL, "iDisp1_out"}, + + { "Analog Out", NULL, "Codec Output Pin1" }, + { "Digital Out", NULL, "Codec Output Pin2" }, + { "Alt Analog Out", NULL, "Codec Output Pin3" }, + + { "Codec Input Pin1", NULL, "Analog In" }, + { "Codec Input Pin2", NULL, "Digital In" }, + { "Codec Input Pin3", NULL, "Alt Analog In" }, + + /* CODEC BE connections */ + { "Analog Codec Playback", NULL, "Analog CPU Playback" }, + { "Analog CPU Playback", NULL, "codec0_out" }, + { "Digital Codec Playback", NULL, "Digital CPU Playback" }, + { "Digital CPU Playback", NULL, "codec1_out" }, + { "Alt Analog Codec Playback", NULL, "Alt Analog CPU Playback" }, + { "Alt Analog CPU Playback", NULL, "codec2_out" }, + + { "codec0_in", NULL, "Analog CPU Capture" }, + { "Analog CPU Capture", NULL, "Analog Codec Capture" }, + { "codec1_in", NULL, "Digital CPU Capture" }, + { "Digital CPU Capture", NULL, "Digital Codec Capture" }, + { "codec2_in", NULL, "Alt Analog CPU Capture" }, + { "Alt Analog CPU Capture", NULL, "Alt Analog Codec Capture" }, +}; + +static int skl_hda_card_late_probe(struct snd_soc_card *card) +{ + return skl_hda_hdmi_jack_init(card); +} + +static int +skl_hda_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link) +{ + struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card); + int ret = 0; + + dev_dbg(card->dev, "%s: dai link name - %s\n", __func__, link->name); + link->platform_name = ctx->platform_name; + link->nonatomic = 1; + + if (strstr(link->name, "HDMI")) { + ret = skl_hda_hdmi_add_pcm(card, ctx->pcm_count); + + if (ret < 0) + return ret; + + ctx->dai_index++; + } + + ctx->pcm_count++; + return ret; +} + +static struct snd_soc_card hda_soc_card = { + .name = "skl_hda_card", + .owner = THIS_MODULE, + .dai_link = skl_hda_be_dai_links, + .dapm_widgets = skl_hda_widgets, + .dapm_routes = skl_hda_map, + .add_dai_link = skl_hda_add_dai_link, + .fully_routed = true, + .late_probe = skl_hda_card_late_probe, +}; + +#define IDISP_DAI_COUNT 3 +/* there are two routes per iDisp output */ +#define IDISP_ROUTE_COUNT (IDISP_DAI_COUNT * 2) +#define IDISP_CODEC_MASK 0x4 + +static int skl_hda_fill_card_info(struct snd_soc_acpi_mach_params *mach_params) +{ + struct snd_soc_card *card = &hda_soc_card; + struct snd_soc_dai_link *dai_link; + u32 codec_count, codec_mask; + int i, num_links, num_route; + + codec_mask = mach_params->codec_mask; + codec_count = hweight_long(codec_mask); + + if (codec_count == 1 && codec_mask & IDISP_CODEC_MASK) { + num_links = IDISP_DAI_COUNT; + num_route = IDISP_ROUTE_COUNT; + } else if (codec_count == 2 && codec_mask & IDISP_CODEC_MASK) { + num_links = ARRAY_SIZE(skl_hda_be_dai_links); + num_route = ARRAY_SIZE(skl_hda_map), + card->dapm_widgets = skl_hda_widgets; + card->num_dapm_widgets = ARRAY_SIZE(skl_hda_widgets); + } else { + return -EINVAL; + } + + card->num_links = num_links; + card->num_dapm_routes = num_route; + + for_each_card_prelinks(card, i, dai_link) + dai_link->platform_name = mach_params->platform; + + return 0; +} + +static int skl_hda_audio_probe(struct platform_device *pdev) +{ + struct snd_soc_acpi_mach *mach; + struct skl_hda_private *ctx; + int ret; + + dev_dbg(&pdev->dev, "%s: entry\n", __func__); + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); + if (!ctx) + return -ENOMEM; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + + mach = (&pdev->dev)->platform_data; + if (!mach) + return -EINVAL; + + ret = skl_hda_fill_card_info(&mach->mach_params); + if (ret < 0) { + dev_err(&pdev->dev, "Unsupported HDAudio/iDisp configuration found\n"); + return ret; + } + + ctx->pcm_count = hda_soc_card.num_links; + ctx->dai_index = 1; /* hdmi codec dai name starts from index 1 */ + ctx->platform_name = mach->mach_params.platform; + + hda_soc_card.dev = &pdev->dev; + snd_soc_card_set_drvdata(&hda_soc_card, ctx); + + return devm_snd_soc_register_card(&pdev->dev, &hda_soc_card); +} + +static struct platform_driver skl_hda_audio = { + .probe = skl_hda_audio_probe, + .driver = { + .name = "skl_hda_dsp_generic", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(skl_hda_audio) + +/* Module information */ +MODULE_DESCRIPTION("SKL/KBL/BXT/APL HDA Generic Machine driver"); +MODULE_AUTHOR("Rakesh Ughreja <rakesh.a.ughreja@intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:skl_hda_dsp_generic"); diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index d31482b8c9bb..0922106bd323 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -21,9 +21,9 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/soc-acpi.h> #include "../../codecs/nau8825.h" #include "../../codecs/hdac_hdmi.h" -#include "../skylake/skl.h" #define SKL_NUVOTON_CODEC_DAI "nau8825-hifi" #define SKL_MAXIM_CODEC_DAI "HiFi" @@ -400,7 +400,7 @@ static int skylake_refcap_startup(struct snd_pcm_substream *substream) &constraints_16000); } -static const struct snd_soc_ops skylaye_refcap_ops = { +static const struct snd_soc_ops skylake_refcap_ops = { .startup = skylake_refcap_startup, }; @@ -447,7 +447,7 @@ static struct snd_soc_dai_link skylake_dais[] = { .dpcm_capture = 1, .nonatomic = 1, .dynamic = 1, - .ops = &skylaye_refcap_ops, + .ops = &skylake_refcap_ops, }, [SKL_DPCM_AUDIO_DMIC_CP] = { .name = "Skl Audio DMIC cap", @@ -641,7 +641,7 @@ static struct snd_soc_card skylake_audio_card = { static int skylake_audio_probe(struct platform_device *pdev) { struct skl_nau8825_private *ctx; - struct skl_machine_pdata *pdata; + struct snd_soc_acpi_mach *mach; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -652,9 +652,9 @@ static int skylake_audio_probe(struct platform_device *pdev) skylake_audio_card.dev = &pdev->dev; snd_soc_card_set_drvdata(&skylake_audio_card, ctx); - pdata = dev_get_drvdata(&pdev->dev); - if (pdata) - dmic_constraints = pdata->dmic_num == 2 ? + mach = (&pdev->dev)->platform_data; + if (mach) + dmic_constraints = mach->mach_params.dmic_num == 2 ? &constraints_dmic_2ch : &constraints_dmic_channels; return devm_snd_soc_register_card(&pdev->dev, &skylake_audio_card); diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index e877bb60beb1..8433c521d39f 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -23,11 +23,11 @@ #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> +#include <sound/soc-acpi.h> #include <sound/jack.h> #include <sound/pcm_params.h> #include "../../codecs/nau8825.h" #include "../../codecs/hdac_hdmi.h" -#include "../skylake/skl.h" #define SKL_NUVOTON_CODEC_DAI "nau8825-hifi" #define SKL_SSM_CODEC_DAI "ssm4567-hifi" @@ -449,7 +449,7 @@ static int skylake_refcap_startup(struct snd_pcm_substream *substream) &constraints_16000); } -static const struct snd_soc_ops skylaye_refcap_ops = { +static const struct snd_soc_ops skylake_refcap_ops = { .startup = skylake_refcap_startup, }; @@ -496,7 +496,7 @@ static struct snd_soc_dai_link skylake_dais[] = { .dpcm_capture = 1, .nonatomic = 1, .dynamic = 1, - .ops = &skylaye_refcap_ops, + .ops = &skylake_refcap_ops, }, [SKL_DPCM_AUDIO_DMIC_CP] = { .name = "Skl Audio DMIC cap", @@ -694,7 +694,7 @@ static struct snd_soc_card skylake_audio_card = { static int skylake_audio_probe(struct platform_device *pdev) { struct skl_nau88125_private *ctx; - struct skl_machine_pdata *pdata; + struct snd_soc_acpi_mach *mach; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -705,9 +705,9 @@ static int skylake_audio_probe(struct platform_device *pdev) skylake_audio_card.dev = &pdev->dev; snd_soc_card_set_drvdata(&skylake_audio_card, ctx); - pdata = dev_get_drvdata(&pdev->dev); - if (pdata) - dmic_constraints = pdata->dmic_num == 2 ? + mach = (&pdev->dev)->platform_data; + if (mach) + dmic_constraints = mach->mach_params.dmic_num == 2 ? &constraints_dmic_2ch : &constraints_dmic_channels; return devm_snd_soc_register_card(&pdev->dev, &skylake_audio_card); diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile index 915a34cdc8ac..56c81e20b5bf 100644 --- a/sound/soc/intel/common/Makefile +++ b/sound/soc/intel/common/Makefile @@ -7,7 +7,8 @@ snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-m soc-acpi-intel-hsw-bdw-match.o \ soc-acpi-intel-skl-match.o soc-acpi-intel-kbl-match.o \ soc-acpi-intel-bxt-match.o soc-acpi-intel-glk-match.o \ - soc-acpi-intel-cnl-match.o + soc-acpi-intel-cnl-match.o soc-acpi-intel-icl-match.o \ + soc-acpi-intel-hda-match.o obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o diff --git a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c index f39386e540d3..61dedc103b19 100644 --- a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c @@ -6,9 +6,41 @@ * */ +#include <linux/dmi.h> #include <sound/soc-acpi.h> #include <sound/soc-acpi-intel-match.h> +enum { + APL_RVP, +}; + +static const struct dmi_system_id apl_table[] = { + { + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corp."), + DMI_MATCH(DMI_BOARD_NAME, "Apollolake RVP1A"), + }, + .driver_data = (void *)(APL_RVP), + }, + {} +}; + +static struct snd_soc_acpi_mach *apl_quirk(void *arg) +{ + struct snd_soc_acpi_mach *mach = arg; + const struct dmi_system_id *dmi_id; + unsigned long apl_machine_id; + + dmi_id = dmi_first_match(apl_table); + if (dmi_id) { + apl_machine_id = (unsigned long)dmi_id->driver_data; + if (apl_machine_id == APL_RVP) + return NULL; + } + + return mach; +} + static struct snd_soc_acpi_codecs bxt_codecs = { .num_codecs = 1, .codecs = {"MX98357A"} @@ -19,6 +51,9 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_bxt_machines[] = { .id = "INT343A", .drv_name = "bxt_alc298s_i2s", .fw_filename = "intel/dsp_fw_bxtn.bin", + .sof_fw_filename = "intel/sof-apl.ri", + .sof_tplg_filename = "intel/sof-apl-rt298.tplg", + .asoc_plat_name = "0000:00:0e.0", }, { .id = "DLGS7219", @@ -47,6 +82,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_bxt_machines[] = { { .id = "INT34C3", .drv_name = "bxt_tdf8532", + .machine_quirk = apl_quirk, .sof_fw_filename = "intel/sof-apl.ri", .sof_tplg_filename = "intel/sof-apl-tdf8532.tplg", .asoc_plat_name = "0000:00:0e.0", diff --git a/sound/soc/intel/common/soc-acpi-intel-byt-match.c b/sound/soc/intel/common/soc-acpi-intel-byt-match.c index 4daa8a4f0c0c..097dc06377ba 100644 --- a/sound/soc/intel/common/soc-acpi-intel-byt-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-byt-match.c @@ -34,6 +34,13 @@ static const struct dmi_system_id byt_table[] = { .callback = byt_thinkpad10_quirk_cb, .matches = { DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_VERSION, "ThinkPad 8"), + }, + }, + { + .callback = byt_thinkpad10_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_VERSION, "ThinkPad 10"), }, }, diff --git a/sound/soc/intel/common/soc-acpi-intel-hda-match.c b/sound/soc/intel/common/soc-acpi-intel-hda-match.c new file mode 100644 index 000000000000..533c1064f84b --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-hda-match.c @@ -0,0 +1,40 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2018, Intel Corporation. + +/* + * soc-apci-intel-hda-match.c - tables and support for HDA+ACPI enumeration. + * + */ + +#include <sound/soc-acpi.h> +#include <sound/soc-acpi-intel-match.h> +#include "../skylake/skl.h" + +static struct skl_machine_pdata hda_pdata = { + .use_tplg_pcm = true, +}; + +struct snd_soc_acpi_mach snd_soc_acpi_intel_hda_machines[] = { + { + /* .id is not used in this file */ + .drv_name = "skl_hda_dsp_generic", + + /* .fw_filename is dynamically set in skylake driver */ + + /* .sof_fw_filename is dynamically set in sof/intel driver */ + + .sof_tplg_filename = "intel/sof-hda-generic.tplg", + + /* + * .machine_quirk and .quirk_data are not used here but + * can be used if we need a more complicated machine driver + * combining HDA+other device (e.g. DMIC). + */ + .pdata = &hda_pdata, + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_hda_machines); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-icl-match.c b/sound/soc/intel/common/soc-acpi-intel-icl-match.c new file mode 100644 index 000000000000..33b441dca4d3 --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-icl-match.c @@ -0,0 +1,32 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * soc-apci-intel-icl-match.c - tables and support for ICL ACPI enumeration. + * + * Copyright (c) 2018, Intel Corporation. + * + */ + +#include <sound/soc-acpi.h> +#include <sound/soc-acpi-intel-match.h> +#include "../skylake/skl.h" + +static struct skl_machine_pdata icl_pdata = { + .use_tplg_pcm = true, +}; + +struct snd_soc_acpi_mach snd_soc_acpi_intel_icl_machines[] = { + { + .id = "INT34C2", + .drv_name = "icl_rt274", + .fw_filename = "intel/dsp_fw_icl.bin", + .pdata = &icl_pdata, + .sof_fw_filename = "intel/sof-icl.ri", + .sof_tplg_filename = "intel/sof-icl-rt274.tplg", + .asoc_plat_name = "0000:00:1f.3", + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_icl_machines); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c index 0ee173ca437d..e6fa6f470526 100644 --- a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c @@ -32,6 +32,11 @@ static struct snd_soc_acpi_codecs kbl_7219_98357_codecs = { .codecs = {"MX98357A"} }; +static struct snd_soc_acpi_codecs kbl_7219_98927_codecs = { + .num_codecs = 1, + .codecs = {"MX98927"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { { .id = "INT343A", @@ -83,6 +88,24 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { .quirk_data = &kbl_7219_98357_codecs, .pdata = &skl_dmic_data, }, + { + .id = "DLGS7219", + .drv_name = "kbl_da7219_max98927", + .fw_filename = "intel/dsp_fw_kbl.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &kbl_7219_98927_codecs, + .pdata = &skl_dmic_data + }, + { + .id = "10EC5660", + .drv_name = "kbl_rt5660", + .fw_filename = "intel/dsp_fw_kbl.bin", + }, + { + .id = "10EC3277", + .drv_name = "kbl_rt5660", + .fw_filename = "intel/dsp_fw_kbl.bin", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_kbl_machines); diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c index 11041aedea31..1e067504b604 100644 --- a/sound/soc/intel/common/sst-firmware.c +++ b/sound/soc/intel/common/sst-firmware.c @@ -355,7 +355,7 @@ struct sst_fw *sst_fw_new(struct sst_dsp *dsp, /* allocate DMA buffer to store FW data */ sst_fw->dma_buf = dma_alloc_coherent(dsp->dma_dev, sst_fw->size, - &sst_fw->dmable_fw_paddr, GFP_DMA | GFP_KERNEL); + &sst_fw->dmable_fw_paddr, GFP_KERNEL); if (!sst_fw->dma_buf) { dev_err(dsp->dev, "error: DMA alloc failed\n"); kfree(sst_fw); diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index 8bfb8b0fa3d5..b0e6fb93eaf8 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -247,6 +247,14 @@ static const struct skl_dsp_ops dsp_ops[] = { .init_fw = cnl_sst_init_fw, .cleanup = cnl_sst_dsp_cleanup }, + { + .id = 0xa348, + .num_cores = 4, + .loader_ops = bxt_get_loader_ops, + .init = cnl_sst_dsp_init, + .init_fw = cnl_sst_init_fw, + .cleanup = cnl_sst_dsp_cleanup + }, }; const struct skl_dsp_ops *skl_get_dsp_ops(int pci_id) diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index 01a050cf8775..5d125a3df527 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -180,6 +180,9 @@ int skl_get_dmic_geo(struct skl *skl) unsigned int dmic_geo = 0; u8 j; + if (!nhlt) + return 0; + epnt = (struct nhlt_endpoint *)nhlt->desc; for (j = 0; j < nhlt->endpoint_count; j++) { diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 823e39103edd..557f80c0bfe5 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -32,6 +32,7 @@ #define HDA_MONO 1 #define HDA_STEREO 2 #define HDA_QUAD 4 +#define HDA_MAX 8 static const struct snd_pcm_hardware azx_pcm_hw = { .info = (SNDRV_PCM_INFO_MMAP | @@ -494,6 +495,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, stream->lpib); snd_hdac_ext_stream_set_lpib(stream, stream->lpib); } + /* fall through */ case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: @@ -569,7 +571,10 @@ static int skl_link_hw_params(struct snd_pcm_substream *substream, stream_tag = hdac_stream(link_dev)->stream_tag; /* set the stream tag in the codec dai dma params */ - snd_soc_dai_set_tdm_slot(codec_dai, stream_tag, 0, 0, 0); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_dai_set_tdm_slot(codec_dai, stream_tag, 0, 0, 0); + else + snd_soc_dai_set_tdm_slot(codec_dai, 0, stream_tag, 0, 0); p_params.s_fmt = snd_pcm_format_width(params_format(params)); p_params.ch = params_channels(params); @@ -995,21 +1000,63 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { }, }, { - .name = "HD-Codec Pin", + .name = "Analog CPU DAI", .ops = &skl_link_dai_ops, .playback = { - .stream_name = "HD-Codec Tx", - .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .stream_name = "Analog CPU Playback", + .channels_min = HDA_MONO, + .channels_max = HDA_MAX, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, }, .capture = { - .stream_name = "HD-Codec Rx", - .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .stream_name = "Analog CPU Capture", + .channels_min = HDA_MONO, + .channels_max = HDA_MAX, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, +}, +{ + .name = "Alt Analog CPU DAI", + .ops = &skl_link_dai_ops, + .playback = { + .stream_name = "Alt Analog CPU Playback", + .channels_min = HDA_MONO, + .channels_max = HDA_MAX, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .stream_name = "Alt Analog CPU Capture", + .channels_min = HDA_MONO, + .channels_max = HDA_MAX, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, +}, +{ + .name = "Digital CPU DAI", + .ops = &skl_link_dai_ops, + .playback = { + .stream_name = "Digital CPU Playback", + .channels_min = HDA_MONO, + .channels_max = HDA_MAX, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .stream_name = "Digital CPU Capture", + .channels_min = HDA_MONO, + .channels_max = HDA_MAX, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, }, }, }; diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c index 5234fafb758a..9f3ce73593ae 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.c +++ b/sound/soc/intel/skylake/skl-sst-ipc.c @@ -249,6 +249,8 @@ enum skl_ipc_glb_reply { IPC_GLB_REPLY_INVALID_CONFIG_DATA_LEN = 121, IPC_GLB_REPLY_GATEWAY_NOT_INITIALIZED = 140, IPC_GLB_REPLY_GATEWAY_NOT_EXIST = 141, + IPC_GLB_REPLY_SCLK_ALREADY_RUNNING = 150, + IPC_GLB_REPLY_MCLK_ALREADY_RUNNING = 151, IPC_GLB_REPLY_PPL_NOT_INITIALIZED = 160, IPC_GLB_REPLY_PPL_NOT_EXIST = 161, @@ -392,18 +394,47 @@ int skl_ipc_process_notification(struct sst_generic_ipc *ipc, return 0; } -static int skl_ipc_set_reply_error_code(u32 reply) +struct skl_ipc_err_map { + const char *msg; + enum skl_ipc_glb_reply reply; + int err; +}; + +static struct skl_ipc_err_map skl_err_map[] = { + {"DSP out of memory", IPC_GLB_REPLY_OUT_OF_MEMORY, -ENOMEM}, + {"DSP busy", IPC_GLB_REPLY_BUSY, -EBUSY}, + {"SCLK already running", IPC_GLB_REPLY_SCLK_ALREADY_RUNNING, + IPC_GLB_REPLY_SCLK_ALREADY_RUNNING}, + {"MCLK already running", IPC_GLB_REPLY_MCLK_ALREADY_RUNNING, + IPC_GLB_REPLY_MCLK_ALREADY_RUNNING}, +}; + +static int skl_ipc_set_reply_error_code(struct sst_generic_ipc *ipc, u32 reply) { - switch (reply) { - case IPC_GLB_REPLY_OUT_OF_MEMORY: - return -ENOMEM; + int i; - case IPC_GLB_REPLY_BUSY: - return -EBUSY; + for (i = 0; i < ARRAY_SIZE(skl_err_map); i++) { + if (skl_err_map[i].reply == reply) + break; + } - default: + if (i == ARRAY_SIZE(skl_err_map)) { + dev_err(ipc->dev, "ipc FW reply: %d FW Error Code: %u\n", + reply, + ipc->dsp->fw_ops.get_fw_errcode(ipc->dsp)); return -EINVAL; } + + if (skl_err_map[i].err < 0) + dev_err(ipc->dev, "ipc FW reply: %s FW Error Code: %u\n", + skl_err_map[i].msg, + ipc->dsp->fw_ops.get_fw_errcode(ipc->dsp)); + else + dev_info(ipc->dev, "ipc FW reply: %s FW Error Code: %u\n", + skl_err_map[i].msg, + ipc->dsp->fw_ops.get_fw_errcode(ipc->dsp)); + + return skl_err_map[i].err; } void skl_ipc_process_reply(struct sst_generic_ipc *ipc, @@ -441,10 +472,7 @@ void skl_ipc_process_reply(struct sst_generic_ipc *ipc, } } else { - msg->errno = skl_ipc_set_reply_error_code(reply); - dev_err(ipc->dev, "ipc FW reply: reply=%d\n", reply); - dev_err(ipc->dev, "FW Error Code: %u\n", - ipc->dsp->fw_ops.get_fw_errcode(ipc->dsp)); + msg->errno = skl_ipc_set_reply_error_code(ipc, reply); switch (IPC_GLB_NOTIFY_MSG_TYPE(header.primary)) { case IPC_GLB_LOAD_MULTIPLE_MODS: case IPC_GLB_LOAD_LIBRARY: diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index f99c600f86e4..cf8848b779dc 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -898,11 +898,10 @@ static int skl_tplg_set_module_bind_params(struct snd_soc_dapm_widget *w, bc = (struct skl_algo_data *)sb->dobj.private; if (bc->set_params == SKL_PARAM_BIND) { - params = kzalloc(bc->max, GFP_KERNEL); + params = kmemdup(bc->params, bc->max, GFP_KERNEL); if (!params) return -ENOMEM; - memcpy(params, bc->params, bc->max); skl_fill_sink_instance_id(ctx, params, bc->max, mconfig); diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 1d17be0f78a0..4ed5b7e17d44 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -33,9 +33,16 @@ #include <sound/hda_register.h> #include <sound/hdaudio.h> #include <sound/hda_i915.h> +#include <sound/hda_codec.h> #include "skl.h" #include "skl-sst-dsp.h" #include "skl-sst-ipc.h" +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) +#include "../../../soc/codecs/hdac_hda.h" +#endif +static int skl_pci_binding; +module_param_named(pci_binding, skl_pci_binding, int, 0444); +MODULE_PARM_DESC(pci_binding, "PCI binding (0=auto, 1=only legacy, 2=only asoc"); /* * initialize the PCI registers @@ -307,7 +314,7 @@ static int skl_suspend(struct device *dev) struct pci_dev *pci = to_pci_dev(dev); struct hdac_bus *bus = pci_get_drvdata(pci); struct skl *skl = bus_to_skl(bus); - int ret = 0; + int ret; /* * Do not suspend if streams which are marked ignore suspend are @@ -329,14 +336,7 @@ static int skl_suspend(struct device *dev) skl->skl_sst->fw_loaded = false; } - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { - ret = snd_hdac_display_power(bus, false); - if (ret < 0) - dev_err(bus->dev, - "Cannot turn OFF display power on i915\n"); - } - - return ret; + return 0; } static int skl_resume(struct device *dev) @@ -347,16 +347,6 @@ static int skl_resume(struct device *dev) struct hdac_ext_link *hlink = NULL; int ret; - /* Turned OFF in HDMI codec driver after codec reconfiguration */ - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { - ret = snd_hdac_display_power(bus, true); - if (ret < 0) { - dev_err(bus->dev, - "Cannot turn on display power on i915\n"); - return ret; - } - } - /* * resume only when we are not in suspend active, otherwise need to * restore the device @@ -449,8 +439,10 @@ static int skl_free(struct hdac_bus *bus) snd_hdac_ext_bus_exit(bus); cancel_work_sync(&skl->probe_work); - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { + snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, false); snd_hdac_i915_exit(bus); + } return 0; } @@ -472,6 +464,25 @@ static struct skl_ssp_clk skl_ssp_clks[] = { {.name = "ssp5_sclkfs"}, }; +static struct snd_soc_acpi_mach *skl_find_hda_machine(struct skl *skl, + struct snd_soc_acpi_mach *machines) +{ + struct hdac_bus *bus = skl_to_bus(skl); + struct snd_soc_acpi_mach *mach; + + /* check if we have any codecs detected on bus */ + if (bus->codec_mask == 0) + return NULL; + + /* point to common table */ + mach = snd_soc_acpi_intel_hda_machines; + + /* all entries in the machine table use the same firmware */ + mach->fw_filename = machines->fw_filename; + + return mach; +} + static int skl_find_machine(struct skl *skl, void *driver_data) { struct hdac_bus *bus = skl_to_bus(skl); @@ -479,9 +490,13 @@ static int skl_find_machine(struct skl *skl, void *driver_data) struct skl_machine_pdata *pdata; mach = snd_soc_acpi_find_machine(mach); - if (mach == NULL) { - dev_err(bus->dev, "No matching machine driver found\n"); - return -ENODEV; + if (!mach) { + dev_dbg(bus->dev, "No matching I2S machine driver found\n"); + mach = skl_find_hda_machine(skl, driver_data); + if (!mach) { + dev_err(bus->dev, "No matching machine driver found\n"); + return -ENODEV; + } } skl->mach = mach; @@ -490,7 +505,7 @@ static int skl_find_machine(struct skl *skl, void *driver_data) if (pdata) { skl->use_tplg_pcm = pdata->use_tplg_pcm; - pdata->dmic_num = skl_get_dmic_geo(skl); + mach->mach_params.dmic_num = skl_get_dmic_geo(skl); } return 0; @@ -498,8 +513,8 @@ static int skl_find_machine(struct skl *skl, void *driver_data) static int skl_machine_device_register(struct skl *skl) { - struct hdac_bus *bus = skl_to_bus(skl); struct snd_soc_acpi_mach *mach = skl->mach; + struct hdac_bus *bus = skl_to_bus(skl); struct platform_device *pdev; int ret; @@ -509,6 +524,16 @@ static int skl_machine_device_register(struct skl *skl) return -EIO; } + mach->mach_params.platform = dev_name(bus->dev); + mach->mach_params.codec_mask = bus->codec_mask; + + ret = platform_device_add_data(pdev, (const void *)mach, sizeof(*mach)); + if (ret) { + dev_err(bus->dev, "failed to add machine device platform data\n"); + platform_device_put(pdev); + return ret; + } + ret = platform_device_add(pdev); if (ret) { dev_err(bus->dev, "failed to add machine device\n"); @@ -516,8 +541,6 @@ static int skl_machine_device_register(struct skl *skl) return -EIO; } - if (mach->pdata) - dev_set_drvdata(&pdev->dev, mach->pdata); skl->i2s_dev = pdev; @@ -628,6 +651,28 @@ static void skl_clock_device_unregister(struct skl *skl) platform_device_unregister(skl->clk_dev); } +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) + +#define IDISP_INTEL_VENDOR_ID 0x80860000 + +/* + * load the legacy codec driver + */ +static void load_codec_module(struct hda_codec *codec) +{ +#ifdef MODULE + char modalias[MODULE_NAME_LEN]; + const char *mod = NULL; + + snd_hdac_codec_modalias(&codec->core, modalias, sizeof(modalias)); + mod = modalias; + dev_dbg(&codec->core.dev, "loading %s codec module\n", mod); + request_module(mod); +#endif +} + +#endif /* CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC */ + /* * Probe the given codec address */ @@ -637,6 +682,10 @@ static int probe_codec(struct hdac_bus *bus, int addr) (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID; unsigned int res = -1; struct skl *skl = bus_to_skl(bus); +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) + struct hdac_hda_priv *hda_codec; + int err; +#endif struct hdac_device *hdev; mutex_lock(&bus->cmd_mutex); @@ -645,13 +694,34 @@ static int probe_codec(struct hdac_bus *bus, int addr) mutex_unlock(&bus->cmd_mutex); if (res == -1) return -EIO; - dev_dbg(bus->dev, "codec #%d probed OK\n", addr); + dev_dbg(bus->dev, "codec #%d probed OK: %x\n", addr, res); +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) + hda_codec = devm_kzalloc(&skl->pci->dev, sizeof(*hda_codec), + GFP_KERNEL); + if (!hda_codec) + return -ENOMEM; + + hda_codec->codec.bus = skl_to_hbus(skl); + hdev = &hda_codec->codec.core; + + err = snd_hdac_ext_bus_device_init(bus, addr, hdev); + if (err < 0) + return err; + + /* use legacy bus only for HDA codecs, idisp uses ext bus */ + if ((res & 0xFFFF0000) != IDISP_INTEL_VENDOR_ID) { + hdev->type = HDA_DEV_LEGACY; + load_codec_module(&hda_codec->codec); + } + return 0; +#else hdev = devm_kzalloc(&skl->pci->dev, sizeof(*hdev), GFP_KERNEL); if (!hdev) return -ENOMEM; return snd_hdac_ext_bus_device_init(bus, addr, hdev); +#endif /* CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC */ } /* Codec initialization */ @@ -704,11 +774,9 @@ static int skl_i915_init(struct hdac_bus *bus) if (err < 0) return err; - err = snd_hdac_display_power(bus, true); - if (err < 0) - dev_err(bus->dev, "Cannot turn on display power on i915\n"); + snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, true); - return err; + return 0; } static void skl_probe_work(struct work_struct *work) @@ -741,24 +809,13 @@ static void skl_probe_work(struct work_struct *work) err = skl_platform_register(bus->dev); if (err < 0) { dev_err(bus->dev, "platform register failed: %d\n", err); - return; - } - - if (bus->ppcap) { - err = skl_machine_device_register(skl); - if (err < 0) { - dev_err(bus->dev, "machine register failed: %d\n", err); - goto out_err; - } + goto out_err; } - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { - err = snd_hdac_display_power(bus, false); - if (err < 0) { - dev_err(bus->dev, "Cannot turn off display power on i915\n"); - skl_machine_device_unregister(skl); - return; - } + err = skl_machine_device_register(skl); + if (err < 0) { + dev_err(bus->dev, "machine register failed: %d\n", err); + goto out_err; } /* @@ -767,6 +824,9 @@ static void skl_probe_work(struct work_struct *work) list_for_each_entry(hlink, &bus->hlink_list, list) snd_hdac_ext_bus_link_put(bus, hlink); + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) + snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, false); + /* configure PM */ pm_runtime_put_noidle(bus->dev); pm_runtime_allow(bus->dev); @@ -776,7 +836,7 @@ static void skl_probe_work(struct work_struct *work) out_err: if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) - err = snd_hdac_display_power(bus, false); + snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, false); } /* @@ -786,9 +846,10 @@ static int skl_create(struct pci_dev *pci, const struct hdac_io_ops *io_ops, struct skl **rskl) { + struct hdac_ext_bus_ops *ext_ops = NULL; struct skl *skl; struct hdac_bus *bus; - + struct hda_bus *hbus; int err; *rskl = NULL; @@ -803,13 +864,23 @@ static int skl_create(struct pci_dev *pci, return -ENOMEM; } + hbus = skl_to_hbus(skl); bus = skl_to_bus(skl); - snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, io_ops, NULL); + +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) + ext_ops = snd_soc_hdac_hda_get_ops(); +#endif + snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, io_ops, ext_ops); bus->use_posbuf = 1; skl->pci = pci; INIT_WORK(&skl->probe_work, skl_probe_work); bus->bdl_pos_adj = 0; + mutex_init(&hbus->prepare_mutex); + hbus->pci = pci; + hbus->mixer_assigned = -1; + hbus->modelname = "sklbus"; + *rskl = skl; return 0; @@ -838,6 +909,12 @@ static int skl_first_init(struct hdac_bus *bus) snd_hdac_bus_parse_capabilities(bus); + /* check if PPCAP exists */ + if (!bus->ppcap) { + dev_err(bus->dev, "bus ppcap not set, HDaudio or DSP not present?\n"); + return -ENODEV; + } + if (skl_acquire_irq(bus, 0) < 0) return -EBUSY; @@ -847,23 +924,25 @@ static int skl_first_init(struct hdac_bus *bus) gcap = snd_hdac_chip_readw(bus, GCAP); dev_dbg(bus->dev, "chipset global capabilities = 0x%x\n", gcap); - /* allow 64bit DMA address if supported by H/W */ - if (!dma_set_mask(bus->dev, DMA_BIT_MASK(64))) { - dma_set_coherent_mask(bus->dev, DMA_BIT_MASK(64)); - } else { - dma_set_mask(bus->dev, DMA_BIT_MASK(32)); - dma_set_coherent_mask(bus->dev, DMA_BIT_MASK(32)); - } - /* read number of streams from GCAP register */ cp_streams = (gcap >> 8) & 0x0f; pb_streams = (gcap >> 12) & 0x0f; - if (!pb_streams && !cp_streams) + if (!pb_streams && !cp_streams) { + dev_err(bus->dev, "no streams found in GCAP definitions?\n"); return -EIO; + } bus->num_streams = cp_streams + pb_streams; + /* allow 64bit DMA address if supported by H/W */ + if (!dma_set_mask(bus->dev, DMA_BIT_MASK(64))) { + dma_set_coherent_mask(bus->dev, DMA_BIT_MASK(64)); + } else { + dma_set_mask(bus->dev, DMA_BIT_MASK(32)); + dma_set_coherent_mask(bus->dev, DMA_BIT_MASK(32)); + } + /* initialize streams */ snd_hdac_ext_stream_init_all (bus, 0, cp_streams, SNDRV_PCM_STREAM_CAPTURE); @@ -888,6 +967,36 @@ static int skl_probe(struct pci_dev *pci, struct hdac_bus *bus = NULL; int err; + switch (skl_pci_binding) { + case SND_SKL_PCI_BIND_AUTO: + /* + * detect DSP by checking class/subclass/prog-id information + * class=04 subclass 03 prog-if 00: no DSP, use legacy driver + * class=04 subclass 01 prog-if 00: DSP is present + * (and may be required e.g. for DMIC or SSP support) + * class=04 subclass 03 prog-if 80: use DSP or legacy mode + */ + if (pci->class == 0x040300) { + dev_info(&pci->dev, "The DSP is not enabled on this platform, aborting probe\n"); + return -ENODEV; + } + if (pci->class != 0x040100 && pci->class != 0x040380) { + dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, aborting probe\n", pci->class); + return -ENODEV; + } + dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); + break; + case SND_SKL_PCI_BIND_LEGACY: + dev_info(&pci->dev, "Module parameter forced binding with HDaudio legacy, aborting probe\n"); + return -ENODEV; + case SND_SKL_PCI_BIND_ASOC: + dev_info(&pci->dev, "Module parameter forced binding with SKL driver, bypassed detection logic\n"); + break; + default: + dev_err(&pci->dev, "invalid value for skl_pci_binding module parameter, ignored\n"); + break; + } + /* we use ext core ops, so provide NULL for ops here */ err = skl_create(pci, NULL, &skl); if (err < 0) @@ -896,8 +1005,10 @@ static int skl_probe(struct pci_dev *pci, bus = skl_to_bus(skl); err = skl_first_init(bus); - if (err < 0) + if (err < 0) { + dev_err(bus->dev, "skl_first_init failed with err: %d\n", err); goto out_free; + } skl->pci_id = pci->device; @@ -906,37 +1017,48 @@ static int skl_probe(struct pci_dev *pci, skl->nhlt = skl_nhlt_init(bus->dev); if (skl->nhlt == NULL) { +#if !IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) + dev_err(bus->dev, "no nhlt info found\n"); err = -ENODEV; goto out_free; - } - - err = skl_nhlt_create_sysfs(skl); - if (err < 0) - goto out_nhlt_free; +#else + dev_warn(bus->dev, "no nhlt info found, continuing to try to enable HDaudio codec\n"); +#endif + } else { - skl_nhlt_update_topology_bin(skl); + err = skl_nhlt_create_sysfs(skl); + if (err < 0) { + dev_err(bus->dev, "skl_nhlt_create_sysfs failed with err: %d\n", err); + goto out_nhlt_free; + } - pci_set_drvdata(skl->pci, bus); + skl_nhlt_update_topology_bin(skl); - /* check if dsp is there */ - if (bus->ppcap) { /* create device for dsp clk */ err = skl_clock_device_register(skl); - if (err < 0) + if (err < 0) { + dev_err(bus->dev, "skl_clock_device_register failed with err: %d\n", err); goto out_clk_free; + } + } - err = skl_find_machine(skl, (void *)pci_id->driver_data); - if (err < 0) - goto out_nhlt_free; + pci_set_drvdata(skl->pci, bus); - err = skl_init_dsp(skl); - if (err < 0) { - dev_dbg(bus->dev, "error failed to register dsp\n"); - goto out_nhlt_free; - } - skl->skl_sst->enable_miscbdcge = skl_enable_miscbdcge; - skl->skl_sst->clock_power_gating = skl_clock_power_gating; + + err = skl_find_machine(skl, (void *)pci_id->driver_data); + if (err < 0) { + dev_err(bus->dev, "skl_find_machine failed with err: %d\n", err); + goto out_nhlt_free; } + + err = skl_init_dsp(skl); + if (err < 0) { + dev_dbg(bus->dev, "error failed to register dsp\n"); + goto out_nhlt_free; + } + skl->skl_sst->enable_miscbdcge = skl_enable_miscbdcge; + skl->skl_sst->clock_power_gating = skl_clock_power_gating; + if (bus->mlcap) snd_hdac_ext_bus_get_ml_capabilities(bus); @@ -944,8 +1066,10 @@ static int skl_probe(struct pci_dev *pci, /* create device for soc dmic */ err = skl_dmic_device_register(skl); - if (err < 0) + if (err < 0) { + dev_err(bus->dev, "skl_dmic_device_register failed with err: %d\n", err); goto out_dsp_free; + } schedule_work(&skl->probe_work); @@ -1013,21 +1137,36 @@ static void skl_remove(struct pci_dev *pci) /* PCI IDs */ static const struct pci_device_id skl_ids[] = { +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKL) /* Sunrise Point-LP */ { PCI_DEVICE(0x8086, 0x9d70), .driver_data = (unsigned long)&snd_soc_acpi_intel_skl_machines}, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_APL) /* BXT-P */ { PCI_DEVICE(0x8086, 0x5a98), .driver_data = (unsigned long)&snd_soc_acpi_intel_bxt_machines}, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_KBL) /* KBL */ { PCI_DEVICE(0x8086, 0x9D71), .driver_data = (unsigned long)&snd_soc_acpi_intel_kbl_machines}, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_GLK) /* GLK */ { PCI_DEVICE(0x8086, 0x3198), .driver_data = (unsigned long)&snd_soc_acpi_intel_glk_machines}, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_CNL) /* CNL */ { PCI_DEVICE(0x8086, 0x9dc8), .driver_data = (unsigned long)&snd_soc_acpi_intel_cnl_machines}, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_CFL) + /* CFL */ + { PCI_DEVICE(0x8086, 0xa348), + .driver_data = (unsigned long)&snd_soc_acpi_intel_cnl_machines}, +#endif { 0, } }; MODULE_DEVICE_TABLE(pci, skl_ids); diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index 78aa8bdcb619..85f8bb6687dc 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -23,6 +23,7 @@ #include <sound/hda_register.h> #include <sound/hdaudio_ext.h> +#include <sound/hda_codec.h> #include <sound/soc.h> #include "skl-nhlt.h" #include "skl-ssp-clk.h" @@ -71,7 +72,7 @@ struct skl_fw_config { }; struct skl { - struct hdac_bus hbus; + struct hda_bus hbus; struct pci_dev *pci; unsigned int init_done:1; /* delayed init status */ @@ -105,8 +106,11 @@ struct skl { struct snd_soc_acpi_mach *mach; }; -#define skl_to_bus(s) (&(s)->hbus) -#define bus_to_skl(bus) container_of(bus, struct skl, hbus) +#define skl_to_bus(s) (&(s)->hbus.core) +#define bus_to_skl(bus) container_of(bus, struct skl, hbus.core) + +#define skl_to_hbus(s) (&(s)->hbus) +#define hbus_to_skl(hbus) container_of((hbus), struct skl, (hbus)) /* to pass dai dma data */ struct skl_dma_params { @@ -115,7 +119,6 @@ struct skl_dma_params { }; struct skl_machine_pdata { - u32 dmic_num; bool use_tplg_pcm; /* use dais and dai links from topology */ }; diff --git a/sound/soc/mediatek/mt2701/mt2701-cs42448.c b/sound/soc/mediatek/mt2701/mt2701-cs42448.c index 666282b865a8..97f9f38ce6b3 100644 --- a/sound/soc/mediatek/mt2701/mt2701-cs42448.c +++ b/sound/soc/mediatek/mt2701/mt2701-cs42448.c @@ -299,6 +299,7 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev) devm_kzalloc(&pdev->dev, sizeof(struct mt2701_cs42448_private), GFP_KERNEL); struct device *dev = &pdev->dev; + struct snd_soc_dai_link *dai_link; if (!priv) return -ENOMEM; @@ -309,10 +310,10 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); return -EINVAL; } - for (i = 0; i < card->num_links; i++) { - if (mt2701_cs42448_dai_links[i].platform_name) + for_each_card_prelinks(card, i, dai_link) { + if (dai_link->platform_name) continue; - mt2701_cs42448_dai_links[i].platform_of_node = platform_node; + dai_link->platform_of_node = platform_node; } card->dev = dev; @@ -324,10 +325,10 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev) "Property 'audio-codec' missing or invalid\n"); return -EINVAL; } - for (i = 0; i < card->num_links; i++) { - if (mt2701_cs42448_dai_links[i].codec_name) + for_each_card_prelinks(card, i, dai_link) { + if (dai_link->codec_name) continue; - mt2701_cs42448_dai_links[i].codec_of_node = codec_node; + dai_link->codec_of_node = codec_node; } codec_node_bt_mrg = of_parse_phandle(pdev->dev.of_node, diff --git a/sound/soc/mediatek/mt2701/mt2701-wm8960.c b/sound/soc/mediatek/mt2701/mt2701-wm8960.c index 89f34efd9747..6bc1d3d58e64 100644 --- a/sound/soc/mediatek/mt2701/mt2701-wm8960.c +++ b/sound/soc/mediatek/mt2701/mt2701-wm8960.c @@ -97,6 +97,7 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt2701_wm8960_card; struct device_node *platform_node, *codec_node; + struct snd_soc_dai_link *dai_link; int ret, i; platform_node = of_parse_phandle(pdev->dev.of_node, @@ -105,10 +106,10 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); return -EINVAL; } - for (i = 0; i < card->num_links; i++) { - if (mt2701_wm8960_dai_links[i].platform_name) + for_each_card_prelinks(card, i, dai_link) { + if (dai_link->platform_name) continue; - mt2701_wm8960_dai_links[i].platform_of_node = platform_node; + dai_link->platform_of_node = platform_node; } card->dev = &pdev->dev; @@ -120,10 +121,10 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev) "Property 'audio-codec' missing or invalid\n"); return -EINVAL; } - for (i = 0; i < card->num_links; i++) { - if (mt2701_wm8960_dai_links[i].codec_name) + for_each_card_prelinks(card, i, dai_link) { + if (dai_link->codec_name) continue; - mt2701_wm8960_dai_links[i].codec_of_node = codec_node; + dai_link->codec_of_node = codec_node; } ret = snd_soc_of_parse_audio_routing(card, "audio-routing"); @@ -150,7 +151,6 @@ static const struct of_device_id mt2701_wm8960_machine_dt_match[] = { static struct platform_driver mt2701_wm8960_machine = { .driver = { .name = "mt2701-wm8960", - .owner = THIS_MODULE, #ifdef CONFIG_OF .of_match_table = mt2701_wm8960_machine_dt_match, #endif diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c index 192f4d7b37b6..bff7d71d0742 100644 --- a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c +++ b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c @@ -828,7 +828,7 @@ static int mt6797_afe_pcm_dev_probe(struct platform_device *pdev) /* request irq */ irq_id = platform_get_irq(pdev, 0); if (!irq_id) { - dev_err(dev, "%s no irq found\n", dev->of_node->name); + dev_err(dev, "%pOFn no irq found\n", dev->of_node); return -ENXIO; } ret = devm_request_irq(dev, irq_id, mt6797_afe_irq_handler, diff --git a/sound/soc/mediatek/mt6797/mt6797-mt6351.c b/sound/soc/mediatek/mt6797/mt6797-mt6351.c index b1558c57b9ca..cc41eb531653 100644 --- a/sound/soc/mediatek/mt6797/mt6797-mt6351.c +++ b/sound/soc/mediatek/mt6797/mt6797-mt6351.c @@ -158,6 +158,7 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt6797_mt6351_card; struct device_node *platform_node, *codec_node; + struct snd_soc_dai_link *dai_link; int ret, i; card->dev = &pdev->dev; @@ -168,10 +169,10 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); return -EINVAL; } - for (i = 0; i < card->num_links; i++) { - if (mt6797_mt6351_dai_links[i].platform_name) + for_each_card_prelinks(card, i, dai_link) { + if (dai_link->platform_name) continue; - mt6797_mt6351_dai_links[i].platform_of_node = platform_node; + dai_link->platform_of_node = platform_node; } codec_node = of_parse_phandle(pdev->dev.of_node, @@ -181,10 +182,10 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev) "Property 'audio-codec' missing or invalid\n"); return -EINVAL; } - for (i = 0; i < card->num_links; i++) { - if (mt6797_mt6351_dai_links[i].codec_name) + for_each_card_prelinks(card, i, dai_link) { + if (dai_link->codec_name) continue; - mt6797_mt6351_dai_links[i].codec_of_node = codec_node; + dai_link->codec_of_node = codec_node; } ret = devm_snd_soc_register_card(&pdev->dev, card); @@ -205,7 +206,6 @@ static const struct of_device_id mt6797_mt6351_dt_match[] = { static struct platform_driver mt6797_mt6351_driver = { .driver = { .name = "mt6797-mt6351", - .owner = THIS_MODULE, #ifdef CONFIG_OF .of_match_table = mt6797_mt6351_dt_match, #endif diff --git a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c index c0b6697503fd..166aed28330d 100644 --- a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c +++ b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c @@ -1092,7 +1092,7 @@ static int mt8173_afe_pcm_dev_probe(struct platform_device *pdev) irq_id = platform_get_irq(pdev, 0); if (irq_id <= 0) { - dev_err(afe->dev, "np %s no irq\n", afe->dev->of_node->name); + dev_err(afe->dev, "np %pOFn no irq\n", afe->dev->of_node); return irq_id < 0 ? irq_id : -ENXIO; } ret = devm_request_irq(afe->dev, irq_id, mt8173_afe_irq_handler, diff --git a/sound/soc/mediatek/mt8173/mt8173-max98090.c b/sound/soc/mediatek/mt8173/mt8173-max98090.c index 902d111016d6..4d6596d5cb07 100644 --- a/sound/soc/mediatek/mt8173/mt8173-max98090.c +++ b/sound/soc/mediatek/mt8173/mt8173-max98090.c @@ -137,6 +137,7 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8173_max98090_card; struct device_node *codec_node, *platform_node; + struct snd_soc_dai_link *dai_link; int ret, i; platform_node = of_parse_phandle(pdev->dev.of_node, @@ -145,10 +146,10 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); return -EINVAL; } - for (i = 0; i < card->num_links; i++) { - if (mt8173_max98090_dais[i].platform_name) + for_each_card_prelinks(card, i, dai_link) { + if (dai_link->platform_name) continue; - mt8173_max98090_dais[i].platform_of_node = platform_node; + dai_link->platform_of_node = platform_node; } codec_node = of_parse_phandle(pdev->dev.of_node, @@ -158,10 +159,10 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev) "Property 'audio-codec' missing or invalid\n"); return -EINVAL; } - for (i = 0; i < card->num_links; i++) { - if (mt8173_max98090_dais[i].codec_name) + for_each_card_prelinks(card, i, dai_link) { + if (dai_link->codec_name) continue; - mt8173_max98090_dais[i].codec_of_node = codec_node; + dai_link->codec_of_node = codec_node; } card->dev = &pdev->dev; diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c index 582174d98c6c..da5b58ce791b 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c @@ -44,11 +44,10 @@ static int mt8173_rt5650_rt5514_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; int i, ret; - for (i = 0; i < rtd->num_codecs; i++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; - + for_each_rtd_codec_dai(rtd, i, codec_dai) { /* pll from mclk 12.288M */ ret = snd_soc_dai_set_pll(codec_dai, 0, 0, MCLK_FOR_CODECS, params_rate(params) * 512); @@ -179,6 +178,7 @@ static int mt8173_rt5650_rt5514_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8173_rt5650_rt5514_card; struct device_node *platform_node; + struct snd_soc_dai_link *dai_link; int i, ret; platform_node = of_parse_phandle(pdev->dev.of_node, @@ -188,10 +188,10 @@ static int mt8173_rt5650_rt5514_dev_probe(struct platform_device *pdev) return -EINVAL; } - for (i = 0; i < card->num_links; i++) { - if (mt8173_rt5650_rt5514_dais[i].platform_name) + for_each_card_prelinks(card, i, dai_link) { + if (dai_link->platform_name) continue; - mt8173_rt5650_rt5514_dais[i].platform_of_node = platform_node; + dai_link->platform_of_node = platform_node; } mt8173_rt5650_rt5514_codecs[0].of_node = diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c index b3670c8a5b8d..d83cd039b413 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c @@ -48,11 +48,10 @@ static int mt8173_rt5650_rt5676_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; int i, ret; - for (i = 0; i < rtd->num_codecs; i++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; - + for_each_rtd_codec_dai(rtd, i, codec_dai) { /* pll from mclk 12.288M */ ret = snd_soc_dai_set_pll(codec_dai, 0, 0, MCLK_FOR_CODECS, params_rate(params) * 512); @@ -225,6 +224,7 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8173_rt5650_rt5676_card; struct device_node *platform_node; + struct snd_soc_dai_link *dai_link; int i, ret; platform_node = of_parse_phandle(pdev->dev.of_node, @@ -234,10 +234,10 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev) return -EINVAL; } - for (i = 0; i < card->num_links; i++) { - if (mt8173_rt5650_rt5676_dais[i].platform_name) + for_each_card_prelinks(card, i, dai_link) { + if (dai_link->platform_name) continue; - mt8173_rt5650_rt5676_dais[i].platform_of_node = platform_node; + dai_link->platform_of_node = platform_node; } mt8173_rt5650_rt5676_codecs[0].of_node = diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index 7a89b4aad182..7edf250c8fb1 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -59,6 +59,7 @@ static int mt8173_rt5650_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; unsigned int mclk_clock; + struct snd_soc_dai *codec_dai; int i, ret; switch (mt8173_rt5650_priv.pll_from) { @@ -76,9 +77,7 @@ static int mt8173_rt5650_hw_params(struct snd_pcm_substream *substream, break; } - for (i = 0; i < rtd->num_codecs; i++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; - + for_each_rtd_codec_dai(rtd, i, codec_dai) { /* pll from mclk */ ret = snd_soc_dai_set_pll(codec_dai, 0, 0, mclk_clock, params_rate(params) * 512); @@ -240,6 +239,7 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) struct device_node *platform_node; struct device_node *np; const char *codec_capture_dai; + struct snd_soc_dai_link *dai_link; int i, ret; platform_node = of_parse_phandle(pdev->dev.of_node, @@ -249,10 +249,10 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) return -EINVAL; } - for (i = 0; i < card->num_links; i++) { - if (mt8173_rt5650_dais[i].platform_name) + for_each_card_prelinks(card, i, dai_link) { + if (dai_link->platform_name) continue; - mt8173_rt5650_dais[i].platform_of_node = platform_node; + dai_link->platform_of_node = platform_node; } mt8173_rt5650_codecs[0].of_node = diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 8af8bc358a90..8779fe23671d 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -4,6 +4,8 @@ menu "ASoC support for Amlogic platforms" config SND_MESON_AXG_FIFO tristate select REGMAP_MMIO + imply COMMON_CLK_AXG_AUDIO + imply RESET_MESON_AUDIO_ARB config SND_MESON_AXG_FRDDR tristate "Amlogic AXG Playback FIFO support" @@ -22,6 +24,7 @@ config SND_MESON_AXG_TODDR config SND_MESON_AXG_TDM_FORMATTER tristate select REGMAP_MMIO + imply COMMON_CLK_AXG_AUDIO config SND_MESON_AXG_TDM_INTERFACE tristate @@ -51,6 +54,8 @@ config SND_MESON_AXG_SOUND_CARD imply SND_MESON_AXG_TDMIN imply SND_MESON_AXG_TDMOUT imply SND_MESON_AXG_SPDIFOUT + imply SND_MESON_AXG_SPDIFIN + imply SND_MESON_AXG_PDM help Select Y or M to add support for the AXG SoC sound card @@ -58,8 +63,23 @@ config SND_MESON_AXG_SPDIFOUT tristate "Amlogic AXG SPDIF Output Support" select SND_PCM_IEC958 imply SND_SOC_SPDIF + imply COMMON_CLK_AXG_AUDIO help Select Y or M to add support for SPDIF output serializer embedded in the Amlogic AXG SoC family +config SND_MESON_AXG_SPDIFIN + tristate "Amlogic AXG SPDIF Input Support" + imply SND_SOC_SPDIF + help + Select Y or M to add support for SPDIF input embedded + in the Amlogic AXG SoC family + +config SND_MESON_AXG_PDM + tristate "Amlogic AXG PDM Input Support" + imply SND_SOC_DMIC + imply COMMON_CLK_AXG_AUDIO + help + Select Y or M to add support for PDM input embedded + in the Amlogic AXG SoC family endmenu diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index c5e003b093db..b45dfb9e2f88 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -8,7 +8,9 @@ snd-soc-meson-axg-tdm-interface-objs := axg-tdm-interface.o snd-soc-meson-axg-tdmin-objs := axg-tdmin.o snd-soc-meson-axg-tdmout-objs := axg-tdmout.o snd-soc-meson-axg-sound-card-objs := axg-card.o +snd-soc-meson-axg-spdifin-objs := axg-spdifin.o snd-soc-meson-axg-spdifout-objs := axg-spdifout.o +snd-soc-meson-axg-pdm-objs := axg-pdm.o obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o @@ -18,4 +20,6 @@ obj-$(CONFIG_SND_MESON_AXG_TDM_INTERFACE) += snd-soc-meson-axg-tdm-interface.o obj-$(CONFIG_SND_MESON_AXG_TDMIN) += snd-soc-meson-axg-tdmin.o obj-$(CONFIG_SND_MESON_AXG_TDMOUT) += snd-soc-meson-axg-tdmout.o obj-$(CONFIG_SND_MESON_AXG_SOUND_CARD) += snd-soc-meson-axg-sound-card.o +obj-$(CONFIG_SND_MESON_AXG_SPDIFIN) += snd-soc-meson-axg-spdifin.o obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o +obj-$(CONFIG_SND_MESON_AXG_PDM) += snd-soc-meson-axg-pdm.o diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 2914ba0d965b..aa54d2c612c9 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -97,14 +97,14 @@ static void axg_card_clean_references(struct axg_card *priv) { struct snd_soc_card *card = &priv->card; struct snd_soc_dai_link *link; + struct snd_soc_dai_link_component *codec; int i, j; if (card->dai_link) { - for (i = 0; i < card->num_links; i++) { - link = &card->dai_link[i]; + for_each_card_prelinks(card, i, link) { of_node_put(link->cpu_of_node); - for (j = 0; j < link->num_codecs; j++) - of_node_put(link->codecs[j].of_node); + for_each_link_codecs(link, j, codec) + of_node_put(codec->of_node); } } @@ -167,8 +167,7 @@ static int axg_card_tdm_be_hw_params(struct snd_pcm_substream *substream, if (be->mclk_fs) { mclk = params_rate(params) * be->mclk_fs; - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, SND_SOC_CLOCK_IN); if (ret && ret != -ENOTSUPP) @@ -196,8 +195,7 @@ static int axg_card_tdm_dai_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dai *codec_dai; int ret, i; - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { ret = snd_soc_dai_set_tdm_slot(codec_dai, be->codec_masks[i].tx, be->codec_masks[i].rx, @@ -478,7 +476,7 @@ static int axg_card_set_be_link(struct snd_soc_card *card, ret = axg_card_set_link_name(card, link, "be"); if (ret) - dev_err(card->dev, "error setting %s link name\n", np->name); + dev_err(card->dev, "error setting %pOFn link name\n", np); return ret; } diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index 30262550e37b..0e4f65e654c4 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -203,6 +203,8 @@ static int axg_fifo_pcm_open(struct snd_pcm_substream *ss) ret = request_irq(fifo->irq, axg_fifo_pcm_irq_block, 0, dev_name(dev), ss); + if (ret) + return ret; /* Enable pclk to access registers and clock the fifo ip */ ret = clk_prepare_enable(fifo->pclk); diff --git a/sound/soc/meson/axg-fifo.h b/sound/soc/meson/axg-fifo.h index cb6c4013ca33..d9f516cfbeda 100644 --- a/sound/soc/meson/axg-fifo.h +++ b/sound/soc/meson/axg-fifo.h @@ -25,7 +25,8 @@ struct snd_soc_pcm_runtime; SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S20_LE | \ SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) + SNDRV_PCM_FMTBIT_S32_LE | \ + SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE) #define AXG_FIFO_BURST 8 #define AXG_FIFO_MIN_CNT 64 diff --git a/sound/soc/meson/axg-pdm.c b/sound/soc/meson/axg-pdm.c new file mode 100644 index 000000000000..9d5684493ffc --- /dev/null +++ b/sound/soc/meson/axg-pdm.c @@ -0,0 +1,654 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/of_irq.h> +#include <linux/of_platform.h> +#include <linux/regmap.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/pcm_params.h> + +#define PDM_CTRL 0x00 +#define PDM_CTRL_EN BIT(31) +#define PDM_CTRL_OUT_MODE BIT(29) +#define PDM_CTRL_BYPASS_MODE BIT(28) +#define PDM_CTRL_RST_FIFO BIT(16) +#define PDM_CTRL_CHAN_RSTN_MASK GENMASK(15, 8) +#define PDM_CTRL_CHAN_RSTN(x) ((x) << 8) +#define PDM_CTRL_CHAN_EN_MASK GENMASK(7, 0) +#define PDM_CTRL_CHAN_EN(x) ((x) << 0) +#define PDM_HCIC_CTRL1 0x04 +#define PDM_FILTER_EN BIT(31) +#define PDM_HCIC_CTRL1_GAIN_SFT_MASK GENMASK(29, 24) +#define PDM_HCIC_CTRL1_GAIN_SFT(x) ((x) << 24) +#define PDM_HCIC_CTRL1_GAIN_MULT_MASK GENMASK(23, 16) +#define PDM_HCIC_CTRL1_GAIN_MULT(x) ((x) << 16) +#define PDM_HCIC_CTRL1_DSR_MASK GENMASK(8, 4) +#define PDM_HCIC_CTRL1_DSR(x) ((x) << 4) +#define PDM_HCIC_CTRL1_STAGE_NUM_MASK GENMASK(3, 0) +#define PDM_HCIC_CTRL1_STAGE_NUM(x) ((x) << 0) +#define PDM_HCIC_CTRL2 0x08 +#define PDM_F1_CTRL 0x0c +#define PDM_LPF_ROUND_MODE_MASK GENMASK(17, 16) +#define PDM_LPF_ROUND_MODE(x) ((x) << 16) +#define PDM_LPF_DSR_MASK GENMASK(15, 12) +#define PDM_LPF_DSR(x) ((x) << 12) +#define PDM_LPF_STAGE_NUM_MASK GENMASK(8, 0) +#define PDM_LPF_STAGE_NUM(x) ((x) << 0) +#define PDM_LPF_MAX_STAGE 336 +#define PDM_LPF_NUM 3 +#define PDM_F2_CTRL 0x10 +#define PDM_F3_CTRL 0x14 +#define PDM_HPF_CTRL 0x18 +#define PDM_HPF_SFT_STEPS_MASK GENMASK(20, 16) +#define PDM_HPF_SFT_STEPS(x) ((x) << 16) +#define PDM_HPF_OUT_FACTOR_MASK GENMASK(15, 0) +#define PDM_HPF_OUT_FACTOR(x) ((x) << 0) +#define PDM_CHAN_CTRL 0x1c +#define PDM_CHAN_CTRL_POINTER_WIDTH 8 +#define PDM_CHAN_CTRL_POINTER_MAX ((1 << PDM_CHAN_CTRL_POINTER_WIDTH) - 1) +#define PDM_CHAN_CTRL_NUM 4 +#define PDM_CHAN_CTRL1 0x20 +#define PDM_COEFF_ADDR 0x24 +#define PDM_COEFF_DATA 0x28 +#define PDM_CLKG_CTRL 0x2c +#define PDM_STS 0x30 + +struct axg_pdm_lpf { + unsigned int ds; + unsigned int round_mode; + const unsigned int *tap; + unsigned int tap_num; +}; + +struct axg_pdm_hcic { + unsigned int shift; + unsigned int mult; + unsigned int steps; + unsigned int ds; +}; + +struct axg_pdm_hpf { + unsigned int out_factor; + unsigned int steps; +}; + +struct axg_pdm_filters { + struct axg_pdm_hcic hcic; + struct axg_pdm_hpf hpf; + struct axg_pdm_lpf lpf[PDM_LPF_NUM]; +}; + +struct axg_pdm_cfg { + const struct axg_pdm_filters *filters; + unsigned int sys_rate; +}; + +struct axg_pdm { + const struct axg_pdm_cfg *cfg; + struct regmap *map; + struct clk *dclk; + struct clk *sysclk; + struct clk *pclk; +}; + +static void axg_pdm_enable(struct regmap *map) +{ + /* Reset AFIFO */ + regmap_update_bits(map, PDM_CTRL, PDM_CTRL_RST_FIFO, PDM_CTRL_RST_FIFO); + regmap_update_bits(map, PDM_CTRL, PDM_CTRL_RST_FIFO, 0); + + /* Enable PDM */ + regmap_update_bits(map, PDM_CTRL, PDM_CTRL_EN, PDM_CTRL_EN); +} + +static void axg_pdm_disable(struct regmap *map) +{ + regmap_update_bits(map, PDM_CTRL, PDM_CTRL_EN, 0); +} + +static void axg_pdm_filters_enable(struct regmap *map, bool enable) +{ + unsigned int val = enable ? PDM_FILTER_EN : 0; + + regmap_update_bits(map, PDM_HCIC_CTRL1, PDM_FILTER_EN, val); + regmap_update_bits(map, PDM_F1_CTRL, PDM_FILTER_EN, val); + regmap_update_bits(map, PDM_F2_CTRL, PDM_FILTER_EN, val); + regmap_update_bits(map, PDM_F3_CTRL, PDM_FILTER_EN, val); + regmap_update_bits(map, PDM_HPF_CTRL, PDM_FILTER_EN, val); +} + +static int axg_pdm_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + axg_pdm_enable(priv->map); + return 0; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + axg_pdm_disable(priv->map); + return 0; + + default: + return -EINVAL; + } +} + +static unsigned int axg_pdm_get_os(struct axg_pdm *priv) +{ + const struct axg_pdm_filters *filters = priv->cfg->filters; + unsigned int os = filters->hcic.ds; + int i; + + /* + * The global oversampling factor is defined by the down sampling + * factor applied by each filter (HCIC and LPFs) + */ + + for (i = 0; i < PDM_LPF_NUM; i++) + os *= filters->lpf[i].ds; + + return os; +} + +static int axg_pdm_set_sysclk(struct axg_pdm *priv, unsigned int os, + unsigned int rate) +{ + unsigned int sys_rate = os * 2 * rate * PDM_CHAN_CTRL_POINTER_MAX; + + /* + * Set the default system clock rate unless it is too fast for + * for the requested sample rate. In this case, the sample pointer + * counter could overflow so set a lower system clock rate + */ + if (sys_rate < priv->cfg->sys_rate) + return clk_set_rate(priv->sysclk, sys_rate); + + return clk_set_rate(priv->sysclk, priv->cfg->sys_rate); +} + +static int axg_pdm_set_sample_pointer(struct axg_pdm *priv) +{ + unsigned int spmax, sp, val; + int i; + + /* Max sample counter value per half period of dclk */ + spmax = DIV_ROUND_UP_ULL((u64)clk_get_rate(priv->sysclk), + clk_get_rate(priv->dclk) * 2); + + /* Check if sysclk is not too fast - should not happen */ + if (WARN_ON(spmax > PDM_CHAN_CTRL_POINTER_MAX)) + return -EINVAL; + + /* Capture the data when we are at 75% of the half period */ + sp = spmax * 3 / 4; + + for (i = 0, val = 0; i < PDM_CHAN_CTRL_NUM; i++) + val |= sp << (PDM_CHAN_CTRL_POINTER_WIDTH * i); + + regmap_write(priv->map, PDM_CHAN_CTRL, val); + regmap_write(priv->map, PDM_CHAN_CTRL1, val); + + return 0; +} + +static void axg_pdm_set_channel_mask(struct axg_pdm *priv, + unsigned int channels) +{ + unsigned int mask = GENMASK(channels - 1, 0); + + /* Put all channel in reset */ + regmap_update_bits(priv->map, PDM_CTRL, + PDM_CTRL_CHAN_RSTN_MASK, 0); + + /* Take the necessary channels out of reset and enable them */ + regmap_update_bits(priv->map, PDM_CTRL, + PDM_CTRL_CHAN_RSTN_MASK | + PDM_CTRL_CHAN_EN_MASK, + PDM_CTRL_CHAN_RSTN(mask) | + PDM_CTRL_CHAN_EN(mask)); +} + +static int axg_pdm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai); + unsigned int os = axg_pdm_get_os(priv); + unsigned int rate = params_rate(params); + unsigned int val; + int ret; + + switch (params_width(params)) { + case 24: + val = PDM_CTRL_OUT_MODE; + break; + case 32: + val = 0; + break; + default: + dev_err(dai->dev, "unsupported sample width\n"); + return -EINVAL; + } + + regmap_update_bits(priv->map, PDM_CTRL, PDM_CTRL_OUT_MODE, val); + + ret = axg_pdm_set_sysclk(priv, os, rate); + if (ret) { + dev_err(dai->dev, "failed to set system clock\n"); + return ret; + } + + ret = clk_set_rate(priv->dclk, rate * os); + if (ret) { + dev_err(dai->dev, "failed to set dclk\n"); + return ret; + } + + ret = axg_pdm_set_sample_pointer(priv); + if (ret) { + dev_err(dai->dev, "invalid clock setting\n"); + return ret; + } + + axg_pdm_set_channel_mask(priv, params_channels(params)); + + return 0; +} + +static int axg_pdm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai); + int ret; + + ret = clk_prepare_enable(priv->dclk); + if (ret) { + dev_err(dai->dev, "enabling dclk failed\n"); + return ret; + } + + /* Enable the filters */ + axg_pdm_filters_enable(priv->map, true); + + return ret; +} + +static void axg_pdm_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai); + + axg_pdm_filters_enable(priv->map, false); + clk_disable_unprepare(priv->dclk); +} + +static const struct snd_soc_dai_ops axg_pdm_dai_ops = { + .trigger = axg_pdm_trigger, + .hw_params = axg_pdm_hw_params, + .startup = axg_pdm_startup, + .shutdown = axg_pdm_shutdown, +}; + +static void axg_pdm_set_hcic_ctrl(struct axg_pdm *priv) +{ + const struct axg_pdm_hcic *hcic = &priv->cfg->filters->hcic; + unsigned int val; + + val = PDM_HCIC_CTRL1_STAGE_NUM(hcic->steps); + val |= PDM_HCIC_CTRL1_DSR(hcic->ds); + val |= PDM_HCIC_CTRL1_GAIN_MULT(hcic->mult); + val |= PDM_HCIC_CTRL1_GAIN_SFT(hcic->shift); + + regmap_update_bits(priv->map, PDM_HCIC_CTRL1, + PDM_HCIC_CTRL1_STAGE_NUM_MASK | + PDM_HCIC_CTRL1_DSR_MASK | + PDM_HCIC_CTRL1_GAIN_MULT_MASK | + PDM_HCIC_CTRL1_GAIN_SFT_MASK, + val); +} + +static void axg_pdm_set_lpf_ctrl(struct axg_pdm *priv, unsigned int index) +{ + const struct axg_pdm_lpf *lpf = &priv->cfg->filters->lpf[index]; + unsigned int offset = index * regmap_get_reg_stride(priv->map) + + PDM_F1_CTRL; + unsigned int val; + + val = PDM_LPF_STAGE_NUM(lpf->tap_num); + val |= PDM_LPF_DSR(lpf->ds); + val |= PDM_LPF_ROUND_MODE(lpf->round_mode); + + regmap_update_bits(priv->map, offset, + PDM_LPF_STAGE_NUM_MASK | + PDM_LPF_DSR_MASK | + PDM_LPF_ROUND_MODE_MASK, + val); +} + +static void axg_pdm_set_hpf_ctrl(struct axg_pdm *priv) +{ + const struct axg_pdm_hpf *hpf = &priv->cfg->filters->hpf; + unsigned int val; + + val = PDM_HPF_OUT_FACTOR(hpf->out_factor); + val |= PDM_HPF_SFT_STEPS(hpf->steps); + + regmap_update_bits(priv->map, PDM_HPF_CTRL, + PDM_HPF_OUT_FACTOR_MASK | + PDM_HPF_SFT_STEPS_MASK, + val); +} + +static int axg_pdm_set_lpf_filters(struct axg_pdm *priv) +{ + const struct axg_pdm_lpf *lpf = priv->cfg->filters->lpf; + unsigned int count = 0; + int i, j; + + for (i = 0; i < PDM_LPF_NUM; i++) + count += lpf[i].tap_num; + + /* Make sure the coeffs fit in the memory */ + if (count >= PDM_LPF_MAX_STAGE) + return -EINVAL; + + /* Set the initial APB bus register address */ + regmap_write(priv->map, PDM_COEFF_ADDR, 0); + + /* Set the tap filter values of all 3 filters */ + for (i = 0; i < PDM_LPF_NUM; i++) { + axg_pdm_set_lpf_ctrl(priv, i); + + for (j = 0; j < lpf[i].tap_num; j++) + regmap_write(priv->map, PDM_COEFF_DATA, lpf[i].tap[j]); + } + + return 0; +} + +static int axg_pdm_dai_probe(struct snd_soc_dai *dai) +{ + struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai); + int ret; + + ret = clk_prepare_enable(priv->pclk); + if (ret) { + dev_err(dai->dev, "enabling pclk failed\n"); + return ret; + } + + /* + * sysclk must be set and enabled as well to access the pdm registers + * Accessing the register w/o it will give a bus error. + */ + ret = clk_set_rate(priv->sysclk, priv->cfg->sys_rate); + if (ret) { + dev_err(dai->dev, "setting sysclk failed\n"); + goto err_pclk; + } + + ret = clk_prepare_enable(priv->sysclk); + if (ret) { + dev_err(dai->dev, "enabling sysclk failed\n"); + goto err_pclk; + } + + /* Make sure the device is initially disabled */ + axg_pdm_disable(priv->map); + + /* Make sure filter bypass is disabled */ + regmap_update_bits(priv->map, PDM_CTRL, PDM_CTRL_BYPASS_MODE, 0); + + /* Load filter settings */ + axg_pdm_set_hcic_ctrl(priv); + axg_pdm_set_hpf_ctrl(priv); + + ret = axg_pdm_set_lpf_filters(priv); + if (ret) { + dev_err(dai->dev, "invalid filter configuration\n"); + goto err_sysclk; + } + + return 0; + +err_sysclk: + clk_disable_unprepare(priv->sysclk); +err_pclk: + clk_disable_unprepare(priv->pclk); + return ret; +} + +static int axg_pdm_dai_remove(struct snd_soc_dai *dai) +{ + struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai); + + clk_disable_unprepare(priv->sysclk); + clk_disable_unprepare(priv->pclk); + + return 0; +} + +static struct snd_soc_dai_driver axg_pdm_dai_drv = { + .name = "PDM", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 48000, + .formats = (SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE), + }, + .ops = &axg_pdm_dai_ops, + .probe = axg_pdm_dai_probe, + .remove = axg_pdm_dai_remove, +}; + +static const struct snd_soc_component_driver axg_pdm_component_drv = {}; + +static const struct regmap_config axg_pdm_regmap_cfg = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = PDM_STS, +}; + +static const unsigned int lpf1_default_tap[] = { + 0x000014, 0xffffb2, 0xfffed9, 0xfffdce, 0xfffd45, + 0xfffe32, 0x000147, 0x000645, 0x000b86, 0x000e21, + 0x000ae3, 0x000000, 0xffeece, 0xffdca8, 0xffd212, + 0xffd7d1, 0xfff2a7, 0x001f4c, 0x0050c2, 0x0072aa, + 0x006ff1, 0x003c32, 0xffdc4e, 0xff6a18, 0xff0fef, + 0xfefbaf, 0xff4c40, 0x000000, 0x00ebc8, 0x01c077, + 0x02209e, 0x01c1a4, 0x008e60, 0xfebe52, 0xfcd690, + 0xfb8fa5, 0xfba498, 0xfd9812, 0x0181ce, 0x06f5f3, + 0x0d112f, 0x12a958, 0x169686, 0x18000e, 0x169686, + 0x12a958, 0x0d112f, 0x06f5f3, 0x0181ce, 0xfd9812, + 0xfba498, 0xfb8fa5, 0xfcd690, 0xfebe52, 0x008e60, + 0x01c1a4, 0x02209e, 0x01c077, 0x00ebc8, 0x000000, + 0xff4c40, 0xfefbaf, 0xff0fef, 0xff6a18, 0xffdc4e, + 0x003c32, 0x006ff1, 0x0072aa, 0x0050c2, 0x001f4c, + 0xfff2a7, 0xffd7d1, 0xffd212, 0xffdca8, 0xffeece, + 0x000000, 0x000ae3, 0x000e21, 0x000b86, 0x000645, + 0x000147, 0xfffe32, 0xfffd45, 0xfffdce, 0xfffed9, + 0xffffb2, 0x000014, +}; + +static const unsigned int lpf2_default_tap[] = { + 0x00050a, 0xfff004, 0x0002c1, 0x003c12, 0xffa818, + 0xffc87d, 0x010aef, 0xff5223, 0xfebd93, 0x028f41, + 0xff5c0e, 0xfc63f8, 0x055f81, 0x000000, 0xf478a0, + 0x11c5e3, 0x2ea74d, 0x11c5e3, 0xf478a0, 0x000000, + 0x055f81, 0xfc63f8, 0xff5c0e, 0x028f41, 0xfebd93, + 0xff5223, 0x010aef, 0xffc87d, 0xffa818, 0x003c12, + 0x0002c1, 0xfff004, 0x00050a, +}; + +static const unsigned int lpf3_default_tap[] = { + 0x000000, 0x000081, 0x000000, 0xfffedb, 0x000000, + 0x00022d, 0x000000, 0xfffc46, 0x000000, 0x0005f7, + 0x000000, 0xfff6eb, 0x000000, 0x000d4e, 0x000000, + 0xffed1e, 0x000000, 0x001a1c, 0x000000, 0xffdcb0, + 0x000000, 0x002ede, 0x000000, 0xffc2d1, 0x000000, + 0x004ebe, 0x000000, 0xff9beb, 0x000000, 0x007dd7, + 0x000000, 0xff633a, 0x000000, 0x00c1d2, 0x000000, + 0xff11d5, 0x000000, 0x012368, 0x000000, 0xfe9c45, + 0x000000, 0x01b252, 0x000000, 0xfdebf6, 0x000000, + 0x0290b8, 0x000000, 0xfcca0d, 0x000000, 0x041d7c, + 0x000000, 0xfa8152, 0x000000, 0x07e9c6, 0x000000, + 0xf28fb5, 0x000000, 0x28b216, 0x3fffde, 0x28b216, + 0x000000, 0xf28fb5, 0x000000, 0x07e9c6, 0x000000, + 0xfa8152, 0x000000, 0x041d7c, 0x000000, 0xfcca0d, + 0x000000, 0x0290b8, 0x000000, 0xfdebf6, 0x000000, + 0x01b252, 0x000000, 0xfe9c45, 0x000000, 0x012368, + 0x000000, 0xff11d5, 0x000000, 0x00c1d2, 0x000000, + 0xff633a, 0x000000, 0x007dd7, 0x000000, 0xff9beb, + 0x000000, 0x004ebe, 0x000000, 0xffc2d1, 0x000000, + 0x002ede, 0x000000, 0xffdcb0, 0x000000, 0x001a1c, + 0x000000, 0xffed1e, 0x000000, 0x000d4e, 0x000000, + 0xfff6eb, 0x000000, 0x0005f7, 0x000000, 0xfffc46, + 0x000000, 0x00022d, 0x000000, 0xfffedb, 0x000000, + 0x000081, 0x000000, +}; + +/* + * These values are sane defaults for the axg platform: + * - OS = 64 + * - Latency = 38700 (?) + * + * TODO: There is a lot of different HCIC, LPFs and HPF configurations possible. + * the configuration may depend on the dmic used by the platform, the + * expected tradeoff between latency and quality, etc ... If/When other + * settings are required, we should add a fw interface to this driver to + * load new filter settings. + */ +static const struct axg_pdm_filters axg_default_filters = { + .hcic = { + .shift = 0x15, + .mult = 0x80, + .steps = 7, + .ds = 8, + }, + .hpf = { + .out_factor = 0x8000, + .steps = 13, + }, + .lpf = { + [0] = { + .ds = 2, + .round_mode = 1, + .tap = lpf1_default_tap, + .tap_num = ARRAY_SIZE(lpf1_default_tap), + }, + [1] = { + .ds = 2, + .round_mode = 0, + .tap = lpf2_default_tap, + .tap_num = ARRAY_SIZE(lpf2_default_tap), + }, + [2] = { + .ds = 2, + .round_mode = 1, + .tap = lpf3_default_tap, + .tap_num = ARRAY_SIZE(lpf3_default_tap) + }, + }, +}; + +static const struct axg_pdm_cfg axg_pdm_config = { + .filters = &axg_default_filters, + .sys_rate = 250000000, +}; + +static const struct of_device_id axg_pdm_of_match[] = { + { + .compatible = "amlogic,axg-pdm", + .data = &axg_pdm_config, + }, {} +}; +MODULE_DEVICE_TABLE(of, axg_pdm_of_match); + +static int axg_pdm_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct axg_pdm *priv; + struct resource *res; + void __iomem *regs; + int ret; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + platform_set_drvdata(pdev, priv); + + priv->cfg = of_device_get_match_data(dev); + if (!priv->cfg) { + dev_err(dev, "failed to match device\n"); + return -ENODEV; + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + priv->map = devm_regmap_init_mmio(dev, regs, &axg_pdm_regmap_cfg); + if (IS_ERR(priv->map)) { + dev_err(dev, "failed to init regmap: %ld\n", + PTR_ERR(priv->map)); + return PTR_ERR(priv->map); + } + + priv->pclk = devm_clk_get(dev, "pclk"); + if (IS_ERR(priv->pclk)) { + ret = PTR_ERR(priv->pclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get pclk: %d\n", ret); + return ret; + } + + priv->dclk = devm_clk_get(dev, "dclk"); + if (IS_ERR(priv->dclk)) { + ret = PTR_ERR(priv->dclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get dclk: %d\n", ret); + return ret; + } + + priv->sysclk = devm_clk_get(dev, "sysclk"); + if (IS_ERR(priv->sysclk)) { + ret = PTR_ERR(priv->sysclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get dclk: %d\n", ret); + return ret; + } + + return devm_snd_soc_register_component(dev, &axg_pdm_component_drv, + &axg_pdm_dai_drv, 1); +} + +static struct platform_driver axg_pdm_pdrv = { + .probe = axg_pdm_probe, + .driver = { + .name = "axg-pdm", + .of_match_table = axg_pdm_of_match, + }, +}; +module_platform_driver(axg_pdm_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG PDM Input driver"); +MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/axg-spdifin.c b/sound/soc/meson/axg-spdifin.c new file mode 100644 index 000000000000..01b2035fa841 --- /dev/null +++ b/sound/soc/meson/axg-spdifin.c @@ -0,0 +1,521 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/bitfield.h> +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/regmap.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/pcm_params.h> + +#define SPDIFIN_CTRL0 0x00 +#define SPDIFIN_CTRL0_EN BIT(31) +#define SPDIFIN_CTRL0_RST_OUT BIT(29) +#define SPDIFIN_CTRL0_RST_IN BIT(28) +#define SPDIFIN_CTRL0_WIDTH_SEL BIT(24) +#define SPDIFIN_CTRL0_STATUS_CH_SHIFT 11 +#define SPDIFIN_CTRL0_STATUS_SEL GENMASK(10, 8) +#define SPDIFIN_CTRL0_SRC_SEL GENMASK(5, 4) +#define SPDIFIN_CTRL0_CHK_VALID BIT(3) +#define SPDIFIN_CTRL1 0x04 +#define SPDIFIN_CTRL1_BASE_TIMER GENMASK(19, 0) +#define SPDIFIN_CTRL1_IRQ_MASK GENMASK(27, 20) +#define SPDIFIN_CTRL2 0x08 +#define SPDIFIN_THRES_PER_REG 3 +#define SPDIFIN_THRES_WIDTH 10 +#define SPDIFIN_CTRL3 0x0c +#define SPDIFIN_CTRL4 0x10 +#define SPDIFIN_TIMER_PER_REG 4 +#define SPDIFIN_TIMER_WIDTH 8 +#define SPDIFIN_CTRL5 0x14 +#define SPDIFIN_CTRL6 0x18 +#define SPDIFIN_STAT0 0x1c +#define SPDIFIN_STAT0_MODE GENMASK(30, 28) +#define SPDIFIN_STAT0_MAXW GENMASK(17, 8) +#define SPDIFIN_STAT0_IRQ GENMASK(7, 0) +#define SPDIFIN_IRQ_MODE_CHANGED BIT(2) +#define SPDIFIN_STAT1 0x20 +#define SPDIFIN_STAT2 0x24 +#define SPDIFIN_MUTE_VAL 0x28 + +#define SPDIFIN_MODE_NUM 7 + +struct axg_spdifin_cfg { + const unsigned int *mode_rates; + unsigned int ref_rate; +}; + +struct axg_spdifin { + const struct axg_spdifin_cfg *conf; + struct regmap *map; + struct clk *refclk; + struct clk *pclk; +}; + +/* + * TODO: + * It would have been nice to check the actual rate against the sample rate + * requested in hw_params(). Unfortunately, I was not able to make the mode + * detection and IRQ work reliably: + * + * 1. IRQs are generated on mode change only, so there is no notification + * on transition between no signal and mode 0 (32kHz). + * 2. Mode detection very often has glitches, and may detects the + * lowest or the highest mode before zeroing in on the actual mode. + * + * This makes calling snd_pcm_stop() difficult to get right. Even notifying + * the kcontrol would be very unreliable at this point. + * Let's keep things simple until the magic spell that makes this work is + * found. + */ + +static unsigned int axg_spdifin_get_rate(struct axg_spdifin *priv) +{ + unsigned int stat, mode, rate = 0; + + regmap_read(priv->map, SPDIFIN_STAT0, &stat); + mode = FIELD_GET(SPDIFIN_STAT0_MODE, stat); + + /* + * If max width is zero, we are not capturing anything. + * Also Sometimes, when the capture is on but there is no data, + * mode is SPDIFIN_MODE_NUM, but not always ... + */ + if (FIELD_GET(SPDIFIN_STAT0_MAXW, stat) && + mode < SPDIFIN_MODE_NUM) + rate = priv->conf->mode_rates[mode]; + + return rate; +} + +static int axg_spdifin_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_spdifin *priv = snd_soc_dai_get_drvdata(dai); + + /* Apply both reset */ + regmap_update_bits(priv->map, SPDIFIN_CTRL0, + SPDIFIN_CTRL0_RST_OUT | + SPDIFIN_CTRL0_RST_IN, + 0); + + /* Clear out reset before in reset */ + regmap_update_bits(priv->map, SPDIFIN_CTRL0, + SPDIFIN_CTRL0_RST_OUT, SPDIFIN_CTRL0_RST_OUT); + regmap_update_bits(priv->map, SPDIFIN_CTRL0, + SPDIFIN_CTRL0_RST_IN, SPDIFIN_CTRL0_RST_IN); + + return 0; +} + +static int axg_spdifin_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_spdifin *priv = snd_soc_dai_get_drvdata(dai); + int ret; + + ret = clk_prepare_enable(priv->refclk); + if (ret) { + dev_err(dai->dev, + "failed to enable spdifin reference clock\n"); + return ret; + } + + regmap_update_bits(priv->map, SPDIFIN_CTRL0, SPDIFIN_CTRL0_EN, + SPDIFIN_CTRL0_EN); + + return 0; +} + +static void axg_spdifin_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_spdifin *priv = snd_soc_dai_get_drvdata(dai); + + regmap_update_bits(priv->map, SPDIFIN_CTRL0, SPDIFIN_CTRL0_EN, 0); + clk_disable_unprepare(priv->refclk); +} + +static void axg_spdifin_write_mode_param(struct regmap *map, int mode, + unsigned int val, + unsigned int num_per_reg, + unsigned int base_reg, + unsigned int width) +{ + uint64_t offset = mode; + unsigned int reg, shift, rem; + + rem = do_div(offset, num_per_reg); + + reg = offset * regmap_get_reg_stride(map) + base_reg; + shift = width * (num_per_reg - 1 - rem); + + regmap_update_bits(map, reg, GENMASK(width - 1, 0) << shift, + val << shift); +} + +static void axg_spdifin_write_timer(struct regmap *map, int mode, + unsigned int val) +{ + axg_spdifin_write_mode_param(map, mode, val, SPDIFIN_TIMER_PER_REG, + SPDIFIN_CTRL4, SPDIFIN_TIMER_WIDTH); +} + +static void axg_spdifin_write_threshold(struct regmap *map, int mode, + unsigned int val) +{ + axg_spdifin_write_mode_param(map, mode, val, SPDIFIN_THRES_PER_REG, + SPDIFIN_CTRL2, SPDIFIN_THRES_WIDTH); +} + +static unsigned int axg_spdifin_mode_timer(struct axg_spdifin *priv, + int mode, + unsigned int rate) +{ + /* + * Number of period of the reference clock during a period of the + * input signal reference clock + */ + return rate / (128 * priv->conf->mode_rates[mode]); +} + +static int axg_spdifin_sample_mode_config(struct snd_soc_dai *dai, + struct axg_spdifin *priv) +{ + unsigned int rate, t_next; + int ret, i = SPDIFIN_MODE_NUM - 1; + + /* Set spdif input reference clock */ + ret = clk_set_rate(priv->refclk, priv->conf->ref_rate); + if (ret) { + dev_err(dai->dev, "reference clock rate set failed\n"); + return ret; + } + + /* + * The rate actually set might be slightly different, get + * the actual rate for the following mode calculation + */ + rate = clk_get_rate(priv->refclk); + + /* HW will update mode every 1ms */ + regmap_update_bits(priv->map, SPDIFIN_CTRL1, + SPDIFIN_CTRL1_BASE_TIMER, + FIELD_PREP(SPDIFIN_CTRL1_BASE_TIMER, rate / 1000)); + + /* Threshold based on the minimum width between two edges */ + regmap_update_bits(priv->map, SPDIFIN_CTRL0, + SPDIFIN_CTRL0_WIDTH_SEL, SPDIFIN_CTRL0_WIDTH_SEL); + + /* Calculate the last timer which has no threshold */ + t_next = axg_spdifin_mode_timer(priv, i, rate); + axg_spdifin_write_timer(priv->map, i, t_next); + + do { + unsigned int t; + + i -= 1; + + /* Calculate the timer */ + t = axg_spdifin_mode_timer(priv, i, rate); + + /* Set the timer value */ + axg_spdifin_write_timer(priv->map, i, t); + + /* Set the threshold value */ + axg_spdifin_write_threshold(priv->map, i, t + t_next); + + /* Save the current timer for the next threshold calculation */ + t_next = t; + + } while (i > 0); + + return 0; +} + +static int axg_spdifin_dai_probe(struct snd_soc_dai *dai) +{ + struct axg_spdifin *priv = snd_soc_dai_get_drvdata(dai); + int ret; + + ret = clk_prepare_enable(priv->pclk); + if (ret) { + dev_err(dai->dev, "failed to enable pclk\n"); + return ret; + } + + ret = axg_spdifin_sample_mode_config(dai, priv); + if (ret) { + dev_err(dai->dev, "mode configuration failed\n"); + clk_disable_unprepare(priv->pclk); + return ret; + } + + return 0; +} + +static int axg_spdifin_dai_remove(struct snd_soc_dai *dai) +{ + struct axg_spdifin *priv = snd_soc_dai_get_drvdata(dai); + + clk_disable_unprepare(priv->pclk); + return 0; +} + +static const struct snd_soc_dai_ops axg_spdifin_ops = { + .prepare = axg_spdifin_prepare, + .startup = axg_spdifin_startup, + .shutdown = axg_spdifin_shutdown, +}; + +static int axg_spdifin_iec958_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 axg_spdifin_get_status_mask(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int i; + + for (i = 0; i < 24; i++) + ucontrol->value.iec958.status[i] = 0xff; + + return 0; +} + +static int axg_spdifin_get_status(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *c = snd_kcontrol_chip(kcontrol); + struct axg_spdifin *priv = snd_soc_component_get_drvdata(c); + int i, j; + + for (i = 0; i < 6; i++) { + unsigned int val; + + regmap_update_bits(priv->map, SPDIFIN_CTRL0, + SPDIFIN_CTRL0_STATUS_SEL, + FIELD_PREP(SPDIFIN_CTRL0_STATUS_SEL, i)); + + regmap_read(priv->map, SPDIFIN_STAT1, &val); + + for (j = 0; j < 4; j++) { + unsigned int offset = i * 4 + j; + + ucontrol->value.iec958.status[offset] = + (val >> (j * 8)) & 0xff; + } + } + + return 0; +} + +#define AXG_SPDIFIN_IEC958_MASK \ + { \ + .access = SNDRV_CTL_ELEM_ACCESS_READ, \ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, \ + .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, MASK), \ + .info = axg_spdifin_iec958_info, \ + .get = axg_spdifin_get_status_mask, \ + } + +#define AXG_SPDIFIN_IEC958_STATUS \ + { \ + .access = (SNDRV_CTL_ELEM_ACCESS_READ | \ + SNDRV_CTL_ELEM_ACCESS_VOLATILE), \ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, \ + .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, NONE), \ + .info = axg_spdifin_iec958_info, \ + .get = axg_spdifin_get_status, \ + } + +static const char * const spdifin_chsts_src_texts[] = { + "A", "B", +}; + +static SOC_ENUM_SINGLE_DECL(axg_spdifin_chsts_src_enum, SPDIFIN_CTRL0, + SPDIFIN_CTRL0_STATUS_CH_SHIFT, + spdifin_chsts_src_texts); + +static int axg_spdifin_rate_lock_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 = 0; + uinfo->value.integer.max = 192000; + + return 0; +} + +static int axg_spdifin_rate_lock_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *c = snd_kcontrol_chip(kcontrol); + struct axg_spdifin *priv = snd_soc_component_get_drvdata(c); + + ucontrol->value.integer.value[0] = axg_spdifin_get_rate(priv); + + return 0; +} + +#define AXG_SPDIFIN_LOCK_RATE(xname) \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, \ + .access = (SNDRV_CTL_ELEM_ACCESS_READ | \ + SNDRV_CTL_ELEM_ACCESS_VOLATILE), \ + .get = axg_spdifin_rate_lock_get, \ + .info = axg_spdifin_rate_lock_info, \ + .name = xname, \ + } + +static const struct snd_kcontrol_new axg_spdifin_controls[] = { + AXG_SPDIFIN_LOCK_RATE("Capture Rate Lock"), + SOC_DOUBLE("Capture Switch", SPDIFIN_CTRL0, 7, 6, 1, 1), + SOC_ENUM(SNDRV_CTL_NAME_IEC958("", CAPTURE, NONE) "Src", + axg_spdifin_chsts_src_enum), + AXG_SPDIFIN_IEC958_MASK, + AXG_SPDIFIN_IEC958_STATUS, +}; + +static const struct snd_soc_component_driver axg_spdifin_component_drv = { + .controls = axg_spdifin_controls, + .num_controls = ARRAY_SIZE(axg_spdifin_controls), +}; + +static const struct regmap_config axg_spdifin_regmap_cfg = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = SPDIFIN_MUTE_VAL, +}; + +static const unsigned int axg_spdifin_mode_rates[SPDIFIN_MODE_NUM] = { + 32000, 44100, 48000, 88200, 96000, 176400, 192000, +}; + +static const struct axg_spdifin_cfg axg_cfg = { + .mode_rates = axg_spdifin_mode_rates, + .ref_rate = 333333333, +}; + +static const struct of_device_id axg_spdifin_of_match[] = { + { + .compatible = "amlogic,axg-spdifin", + .data = &axg_cfg, + }, {} +}; +MODULE_DEVICE_TABLE(of, axg_spdifin_of_match); + +static struct snd_soc_dai_driver * +axg_spdifin_get_dai_drv(struct device *dev, struct axg_spdifin *priv) +{ + struct snd_soc_dai_driver *drv; + int i; + + drv = devm_kzalloc(dev, sizeof(*drv), GFP_KERNEL); + if (!drv) + return ERR_PTR(-ENOMEM); + + drv->name = "SPDIF Input"; + drv->ops = &axg_spdifin_ops; + drv->probe = axg_spdifin_dai_probe; + drv->remove = axg_spdifin_dai_remove; + drv->capture.stream_name = "Capture"; + drv->capture.channels_min = 1; + drv->capture.channels_max = 2; + drv->capture.formats = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE; + + for (i = 0; i < SPDIFIN_MODE_NUM; i++) { + unsigned int rb = + snd_pcm_rate_to_rate_bit(priv->conf->mode_rates[i]); + + if (rb == SNDRV_PCM_RATE_KNOT) + return ERR_PTR(-EINVAL); + + drv->capture.rates |= rb; + } + + return drv; +} + +static int axg_spdifin_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct axg_spdifin *priv; + struct snd_soc_dai_driver *dai_drv; + struct resource *res; + void __iomem *regs; + int ret; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + platform_set_drvdata(pdev, priv); + + priv->conf = of_device_get_match_data(dev); + if (!priv->conf) { + dev_err(dev, "failed to match device\n"); + return -ENODEV; + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + priv->map = devm_regmap_init_mmio(dev, regs, &axg_spdifin_regmap_cfg); + if (IS_ERR(priv->map)) { + dev_err(dev, "failed to init regmap: %ld\n", + PTR_ERR(priv->map)); + return PTR_ERR(priv->map); + } + + priv->pclk = devm_clk_get(dev, "pclk"); + if (IS_ERR(priv->pclk)) { + ret = PTR_ERR(priv->pclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get pclk: %d\n", ret); + return ret; + } + + priv->refclk = devm_clk_get(dev, "refclk"); + if (IS_ERR(priv->refclk)) { + ret = PTR_ERR(priv->refclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get mclk: %d\n", ret); + return ret; + } + + dai_drv = axg_spdifin_get_dai_drv(dev, priv); + if (IS_ERR(dai_drv)) { + dev_err(dev, "failed to get dai driver: %ld\n", + PTR_ERR(dai_drv)); + return PTR_ERR(dai_drv); + } + + return devm_snd_soc_register_component(dev, &axg_spdifin_component_drv, + dai_drv, 1); +} + +static struct platform_driver axg_spdifin_pdrv = { + .probe = axg_spdifin_probe, + .driver = { + .name = "axg-spdifin", + .of_match_table = axg_spdifin_of_match, + }, +}; +module_platform_driver(axg_spdifin_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG SPDIF Input driver"); +MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index 7b8baf46d968..585ce030b79b 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -42,6 +42,7 @@ int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask, struct axg_tdm_stream *rx = (struct axg_tdm_stream *) dai->capture_dma_data; unsigned int tx_slots, rx_slots; + unsigned int fmt = 0; tx_slots = axg_tdm_slots_total(tx_mask); rx_slots = axg_tdm_slots_total(rx_mask); @@ -52,38 +53,45 @@ int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask, return -EINVAL; } - /* - * Amend the dai driver channel number and let dpcm channel merge do - * its job - */ - if (tx) { - tx->mask = tx_mask; - dai->driver->playback.channels_max = tx_slots; - } - - if (rx) { - rx->mask = rx_mask; - dai->driver->capture.channels_max = rx_slots; - } - iface->slots = slots; switch (slot_width) { case 0: - /* defaults width to 32 if not provided */ - iface->slot_width = 32; - break; - case 8: - case 16: - case 24: + slot_width = 32; + /* Fall-through */ case 32: - iface->slot_width = slot_width; + fmt |= SNDRV_PCM_FMTBIT_S32_LE; + /* Fall-through */ + case 24: + fmt |= SNDRV_PCM_FMTBIT_S24_LE; + fmt |= SNDRV_PCM_FMTBIT_S20_LE; + /* Fall-through */ + case 16: + fmt |= SNDRV_PCM_FMTBIT_S16_LE; + /* Fall-through */ + case 8: + fmt |= SNDRV_PCM_FMTBIT_S8; break; default: dev_err(dai->dev, "unsupported slot width: %d\n", slot_width); return -EINVAL; } + iface->slot_width = slot_width; + + /* Amend the dai driver and let dpcm merge do its job */ + if (tx) { + tx->mask = tx_mask; + dai->driver->playback.channels_max = tx_slots; + dai->driver->playback.formats = fmt; + } + + if (rx) { + rx->mask = rx_mask; + dai->driver->capture.channels_max = rx_slots; + dai->driver->capture.formats = fmt; + } + return 0; } EXPORT_SYMBOL_GPL(axg_tdm_set_tdm_slots); diff --git a/sound/soc/meson/axg-toddr.c b/sound/soc/meson/axg-toddr.c index c2c9bb312586..0e9ca3882ae5 100644 --- a/sound/soc/meson/axg-toddr.c +++ b/sound/soc/meson/axg-toddr.c @@ -25,6 +25,8 @@ #define CTRL0_TODDR_LSB_POS_MASK GENMASK(7, 3) #define CTRL0_TODDR_LSB_POS(x) ((x) << 3) +#define TODDR_MSB_POS 31 + static int axg_toddr_pcm_new(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { @@ -36,14 +38,7 @@ static int axg_toddr_dai_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); - unsigned int type, width, msb = 31; - - /* - * NOTE: - * Almost all backend will place the MSB at bit 31, except SPDIF Input - * which will put it at index 28. When adding support for the SPDIF - * Input, we'll need to find which type of backend we are connected to. - */ + unsigned int type, width; switch (params_physical_width(params)) { case 8: @@ -66,8 +61,8 @@ static int axg_toddr_dai_hw_params(struct snd_pcm_substream *substream, CTRL0_TODDR_MSB_POS_MASK | CTRL0_TODDR_LSB_POS_MASK, CTRL0_TODDR_TYPE(type) | - CTRL0_TODDR_MSB_POS(msb) | - CTRL0_TODDR_LSB_POS(msb - (width - 1))); + CTRL0_TODDR_MSB_POS(TODDR_MSB_POS) | + CTRL0_TODDR_LSB_POS(TODDR_MSB_POS - (width - 1))); return 0; } diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c index 81b09d740ed9..6384bb6dacfd 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -356,7 +356,7 @@ static int nuc900_ac97_drvprobe(struct platform_device *pdev) if (ret) goto out; - ret = snd_soc_register_component(&pdev->dev, &nuc900_ac97_component, + ret = devm_snd_soc_register_component(&pdev->dev, &nuc900_ac97_component, &nuc900_ac97_dai, 1); if (ret) goto out; @@ -373,8 +373,6 @@ out: static int nuc900_ac97_drvremove(struct platform_device *pdev) { - snd_soc_unregister_component(&pdev->dev); - nuc900_ac97_data = NULL; snd_soc_set_ac97_ops(NULL); diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig deleted file mode 100644 index 6dccea6fdaeb..000000000000 --- a/sound/soc/omap/Kconfig +++ /dev/null @@ -1,129 +0,0 @@ -config SND_OMAP_SOC - tristate "SoC Audio for Texas Instruments OMAP chips (deprecated)" - depends on (ARCH_OMAP && DMA_OMAP) || (ARM && COMPILE_TEST) - select SND_SDMA_SOC - -config SND_SDMA_SOC - tristate "SoC Audio for Texas Instruments chips using sDMA" - depends on DMA_OMAP || COMPILE_TEST - select SND_SOC_GENERIC_DMAENGINE_PCM - -config SND_OMAP_SOC_DMIC - tristate - -config SND_OMAP_SOC_MCBSP - tristate - -config SND_OMAP_SOC_MCPDM - tristate - -config SND_OMAP_SOC_HDMI_AUDIO - tristate "HDMI audio support for OMAP4+ based SoCs" - depends on SND_SDMA_SOC - help - For HDMI audio to work OMAPDSS HDMI support should be - enabled. - The hdmi audio driver implements cpu-dai component using the - callbacks provided by OMAPDSS and registers the component - under DSS HDMI device. Omap-pcm is registered for platform - component also under DSS HDMI device. Dummy codec is used as - as codec component. The hdmi audio driver implements also - the card and registers it under its own platform device. - The device for the driver is registered by OMAPDSS hdmi - driver. - -config SND_OMAP_SOC_N810 - tristate "SoC Audio support for Nokia N810" - depends on SND_SDMA_SOC && MACH_NOKIA_N810 && I2C - select SND_OMAP_SOC_MCBSP - select SND_SOC_TLV320AIC3X - help - Say Y if you want to add support for SoC audio on Nokia N810. - -config SND_OMAP_SOC_RX51 - tristate "SoC Audio support for Nokia N900 (RX-51)" - depends on SND_SDMA_SOC && ARM && I2C - select SND_OMAP_SOC_MCBSP - select SND_SOC_TLV320AIC3X - select SND_SOC_TPA6130A2 - depends on GPIOLIB - help - Say Y if you want to add support for SoC audio on Nokia N900 - cellphone. - -config SND_OMAP_SOC_AMS_DELTA - tristate "SoC Audio support for Amstrad E3 (Delta) videophone" - depends on SND_SDMA_SOC && MACH_AMS_DELTA && TTY - select SND_OMAP_SOC_MCBSP - select SND_SOC_CX20442 - help - Say Y if you want to add support for SoC audio device connected to - a handset and a speakerphone found on Amstrad E3 (Delta) videophone. - - Note that in order to get those devices fully supported, you have to - build the kernel with standard serial port driver included and - configured for at least 4 ports. Then, from userspace, you must load - a line discipline #19 on the modem (ttyS3) serial line. The simplest - way to achieve this is to install util-linux-ng and use the included - ldattach utility. This can be started automatically from udev, - a simple rule like this one should do the trick (it does for me): - ACTION=="add", KERNEL=="controlC0", \ - RUN+="/usr/sbin/ldattach 19 /dev/ttyS3" - -config SND_OMAP_SOC_OSK5912 - tristate "SoC Audio support for omap osk5912" - depends on SND_SDMA_SOC && MACH_OMAP_OSK && I2C - select SND_OMAP_SOC_MCBSP - select SND_SOC_TLV320AIC23_I2C - help - Say Y if you want to add support for SoC audio on osk5912. - -config SND_OMAP_SOC_AM3517EVM - tristate "SoC Audio support for OMAP3517 / AM3517 EVM" - depends on SND_SDMA_SOC && MACH_OMAP3517EVM && I2C - select SND_OMAP_SOC_MCBSP - select SND_SOC_TLV320AIC23_I2C - help - Say Y if you want to add support for SoC audio on the OMAP3517 / AM3517 - EVM. - -config SND_OMAP_SOC_OMAP_TWL4030 - tristate "SoC Audio support for TI SoC based boards with twl4030 codec" - depends on TWL4030_CORE && SND_SDMA_SOC - select SND_OMAP_SOC_MCBSP - select SND_SOC_TWL4030 - help - Say Y if you want to add support for SoC audio on TI SoC based boards - using twl4030 as c codec. This driver currently supports: - - Beagleboard or Devkit8000 - - Gumstix Overo or CompuLab CM-T35/CM-T3730 - - IGEP v2 - - OMAP3EVM - - SDP3430 - - Zoom2 - -config SND_OMAP_SOC_OMAP_ABE_TWL6040 - tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" - depends on TWL6040_CORE && SND_SDMA_SOC && COMMON_CLK - depends on ARCH_OMAP4 || (SOC_OMAP5 && MFD_PALMAS) || COMPILE_TEST - select SND_OMAP_SOC_DMIC - select SND_OMAP_SOC_MCPDM - select SND_SOC_TWL6040 - select SND_SOC_DMIC - select COMMON_CLK_PALMAS if (SOC_OMAP5 && MFD_PALMAS) - select CLK_TWL6040 - help - Say Y if you want to add support for SoC audio on OMAP boards using - ABE and twl6040 codec. This driver currently supports: - - SDP4430/Blaze boards - - PandaBoard (4430) - - PandaBoardES (4460) - - omap5-uevm (5432) - -config SND_OMAP_SOC_OMAP3_PANDORA - tristate "SoC Audio support for OMAP3 Pandora" - depends on TWL4030_CORE && SND_SDMA_SOC && MACH_OMAP3_PANDORA - select SND_OMAP_SOC_MCBSP - select SND_SOC_TWL4030 - help - Say Y if you want to add support for SoC audio on the OMAP3 Pandora. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile deleted file mode 100644 index 53eba3413485..000000000000 --- a/sound/soc/omap/Makefile +++ /dev/null @@ -1,32 +0,0 @@ -# SPDX-License-Identifier: GPL-2.0 -# OMAP Platform Support -snd-soc-sdma-objs := sdma-pcm.o -snd-soc-omap-dmic-objs := omap-dmic.o -snd-soc-omap-mcbsp-objs := omap-mcbsp.o mcbsp.o -snd-soc-omap-mcpdm-objs := omap-mcpdm.o -snd-soc-omap-hdmi-audio-objs := omap-hdmi-audio.o - -obj-$(CONFIG_SND_SDMA_SOC) += snd-soc-sdma.o -obj-$(CONFIG_SND_OMAP_SOC_DMIC) += snd-soc-omap-dmic.o -obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o -obj-$(CONFIG_SND_OMAP_SOC_MCPDM) += snd-soc-omap-mcpdm.o -obj-$(CONFIG_SND_OMAP_SOC_HDMI_AUDIO) += snd-soc-omap-hdmi-audio.o - -# OMAP Machine Support -snd-soc-n810-objs := n810.o -snd-soc-rx51-objs := rx51.o -snd-soc-ams-delta-objs := ams-delta.o -snd-soc-osk5912-objs := osk5912.o -snd-soc-am3517evm-objs := am3517evm.o -snd-soc-omap-abe-twl6040-objs := omap-abe-twl6040.o -snd-soc-omap-twl4030-objs := omap-twl4030.o -snd-soc-omap3pandora-objs := omap3pandora.o - -obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o -obj-$(CONFIG_SND_OMAP_SOC_RX51) += snd-soc-rx51.o -obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o -obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o -obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o -obj-$(CONFIG_SND_OMAP_SOC_OMAP_ABE_TWL6040) += snd-soc-omap-abe-twl6040.o -obj-$(CONFIG_SND_OMAP_SOC_OMAP_TWL4030) += snd-soc-omap-twl4030.o -obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c deleted file mode 100644 index d5651026ec10..000000000000 --- a/sound/soc/omap/am3517evm.c +++ /dev/null @@ -1,141 +0,0 @@ -/* - * am3517evm.c -- ALSA SoC support for OMAP3517 / AM3517 EVM - * - * Author: Anuj Aggarwal <anuj.aggarwal@ti.com> - * - * Based on sound/soc/omap/beagle.c by Steve Sakoman - * - * Copyright (C) 2009 Texas Instruments Incorporated - * - * 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 version 2. - * - * This program is distributed "as is" WITHOUT ANY WARRANTY of any kind, - * whether express or implied; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - */ - -#include <linux/clk.h> -#include <linux/platform_device.h> -#include <linux/module.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> - -#include <asm/mach-types.h> -#include <linux/platform_data/asoc-ti-mcbsp.h> - -#include "omap-mcbsp.h" - -#include "../codecs/tlv320aic23.h" - -#define CODEC_CLOCK 12000000 - -static int am3517evm_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; - - /* Set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, - CODEC_CLOCK, SND_SOC_CLOCK_IN); - if (ret < 0) - printk(KERN_ERR "can't set codec system clock\n"); - - return ret; -} - -static const struct snd_soc_ops am3517evm_ops = { - .hw_params = am3517evm_hw_params, -}; - -/* am3517evm machine dapm widgets */ -static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { - SND_SOC_DAPM_HP("Line Out", NULL), - SND_SOC_DAPM_LINE("Line In", NULL), - SND_SOC_DAPM_MIC("Mic In", NULL), -}; - -static const struct snd_soc_dapm_route audio_map[] = { - /* Line Out connected to LLOUT, RLOUT */ - {"Line Out", NULL, "LOUT"}, - {"Line Out", NULL, "ROUT"}, - - {"LLINEIN", NULL, "Line In"}, - {"RLINEIN", NULL, "Line In"}, - - {"MICIN", NULL, "Mic In"}, -}; - -/* Digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link am3517evm_dai = { - .name = "TLV320AIC23", - .stream_name = "AIC23", - .cpu_dai_name = "omap-mcbsp.1", - .codec_dai_name = "tlv320aic23-hifi", - .platform_name = "omap-mcbsp.1", - .codec_name = "tlv320aic23-codec.2-001a", - .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, - .ops = &am3517evm_ops, -}; - -/* Audio machine driver */ -static struct snd_soc_card snd_soc_am3517evm = { - .name = "am3517evm", - .owner = THIS_MODULE, - .dai_link = &am3517evm_dai, - .num_links = 1, - - .dapm_widgets = tlv320aic23_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), - .dapm_routes = audio_map, - .num_dapm_routes = ARRAY_SIZE(audio_map), -}; - -static struct platform_device *am3517evm_snd_device; - -static int __init am3517evm_soc_init(void) -{ - int ret; - - if (!machine_is_omap3517evm()) - return -ENODEV; - pr_info("OMAP3517 / AM3517 EVM SoC init\n"); - - am3517evm_snd_device = platform_device_alloc("soc-audio", -1); - if (!am3517evm_snd_device) { - printk(KERN_ERR "Platform device allocation failed\n"); - return -ENOMEM; - } - - platform_set_drvdata(am3517evm_snd_device, &snd_soc_am3517evm); - - ret = platform_device_add(am3517evm_snd_device); - if (ret) - goto err1; - - return 0; - -err1: - printk(KERN_ERR "Unable to add platform device\n"); - platform_device_put(am3517evm_snd_device); - - return ret; -} - -static void __exit am3517evm_soc_exit(void) -{ - platform_device_unregister(am3517evm_snd_device); -} - -module_init(am3517evm_soc_init); -module_exit(am3517evm_soc_exit); - -MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal@ti.com>"); -MODULE_DESCRIPTION("ALSA SoC OMAP3517 / AM3517 EVM"); -MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c deleted file mode 100644 index 79d4dc785e5c..000000000000 --- a/sound/soc/omap/mcbsp.c +++ /dev/null @@ -1,1104 +0,0 @@ -/* - * sound/soc/omap/mcbsp.c - * - * Copyright (C) 2004 Nokia Corporation - * Author: Samuel Ortiz <samuel.ortiz@nokia.com> - * - * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com> - * Peter Ujfalusi <peter.ujfalusi@ti.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. - * - * Multichannel mode not supported. - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/device.h> -#include <linux/platform_device.h> -#include <linux/interrupt.h> -#include <linux/err.h> -#include <linux/clk.h> -#include <linux/delay.h> -#include <linux/io.h> -#include <linux/slab.h> -#include <linux/pm_runtime.h> - -#include <linux/platform_data/asoc-ti-mcbsp.h> - -#include "mcbsp.h" - -static void omap_mcbsp_write(struct omap_mcbsp *mcbsp, u16 reg, u32 val) -{ - void __iomem *addr = mcbsp->io_base + reg * mcbsp->pdata->reg_step; - - if (mcbsp->pdata->reg_size == 2) { - ((u16 *)mcbsp->reg_cache)[reg] = (u16)val; - writew_relaxed((u16)val, addr); - } else { - ((u32 *)mcbsp->reg_cache)[reg] = val; - writel_relaxed(val, addr); - } -} - -static int omap_mcbsp_read(struct omap_mcbsp *mcbsp, u16 reg, bool from_cache) -{ - void __iomem *addr = mcbsp->io_base + reg * mcbsp->pdata->reg_step; - - if (mcbsp->pdata->reg_size == 2) { - return !from_cache ? readw_relaxed(addr) : - ((u16 *)mcbsp->reg_cache)[reg]; - } else { - return !from_cache ? readl_relaxed(addr) : - ((u32 *)mcbsp->reg_cache)[reg]; - } -} - -static void omap_mcbsp_st_write(struct omap_mcbsp *mcbsp, u16 reg, u32 val) -{ - writel_relaxed(val, mcbsp->st_data->io_base_st + reg); -} - -static int omap_mcbsp_st_read(struct omap_mcbsp *mcbsp, u16 reg) -{ - return readl_relaxed(mcbsp->st_data->io_base_st + reg); -} - -#define MCBSP_READ(mcbsp, reg) \ - omap_mcbsp_read(mcbsp, OMAP_MCBSP_REG_##reg, 0) -#define MCBSP_WRITE(mcbsp, reg, val) \ - omap_mcbsp_write(mcbsp, OMAP_MCBSP_REG_##reg, val) -#define MCBSP_READ_CACHE(mcbsp, reg) \ - omap_mcbsp_read(mcbsp, OMAP_MCBSP_REG_##reg, 1) - -#define MCBSP_ST_READ(mcbsp, reg) \ - omap_mcbsp_st_read(mcbsp, OMAP_ST_REG_##reg) -#define MCBSP_ST_WRITE(mcbsp, reg, val) \ - omap_mcbsp_st_write(mcbsp, OMAP_ST_REG_##reg, val) - -static void omap_mcbsp_dump_reg(struct omap_mcbsp *mcbsp) -{ - dev_dbg(mcbsp->dev, "**** McBSP%d regs ****\n", mcbsp->id); - dev_dbg(mcbsp->dev, "DRR2: 0x%04x\n", - MCBSP_READ(mcbsp, DRR2)); - dev_dbg(mcbsp->dev, "DRR1: 0x%04x\n", - MCBSP_READ(mcbsp, DRR1)); - dev_dbg(mcbsp->dev, "DXR2: 0x%04x\n", - MCBSP_READ(mcbsp, DXR2)); - dev_dbg(mcbsp->dev, "DXR1: 0x%04x\n", - MCBSP_READ(mcbsp, DXR1)); - dev_dbg(mcbsp->dev, "SPCR2: 0x%04x\n", - MCBSP_READ(mcbsp, SPCR2)); - dev_dbg(mcbsp->dev, "SPCR1: 0x%04x\n", - MCBSP_READ(mcbsp, SPCR1)); - dev_dbg(mcbsp->dev, "RCR2: 0x%04x\n", - MCBSP_READ(mcbsp, RCR2)); - dev_dbg(mcbsp->dev, "RCR1: 0x%04x\n", - MCBSP_READ(mcbsp, RCR1)); - dev_dbg(mcbsp->dev, "XCR2: 0x%04x\n", - MCBSP_READ(mcbsp, XCR2)); - dev_dbg(mcbsp->dev, "XCR1: 0x%04x\n", - MCBSP_READ(mcbsp, XCR1)); - dev_dbg(mcbsp->dev, "SRGR2: 0x%04x\n", - MCBSP_READ(mcbsp, SRGR2)); - dev_dbg(mcbsp->dev, "SRGR1: 0x%04x\n", - MCBSP_READ(mcbsp, SRGR1)); - dev_dbg(mcbsp->dev, "PCR0: 0x%04x\n", - MCBSP_READ(mcbsp, PCR0)); - dev_dbg(mcbsp->dev, "***********************\n"); -} - -static irqreturn_t omap_mcbsp_irq_handler(int irq, void *dev_id) -{ - struct omap_mcbsp *mcbsp = dev_id; - u16 irqst; - - irqst = MCBSP_READ(mcbsp, IRQST); - dev_dbg(mcbsp->dev, "IRQ callback : 0x%x\n", irqst); - - if (irqst & RSYNCERREN) - dev_err(mcbsp->dev, "RX Frame Sync Error!\n"); - if (irqst & RFSREN) - dev_dbg(mcbsp->dev, "RX Frame Sync\n"); - if (irqst & REOFEN) - dev_dbg(mcbsp->dev, "RX End Of Frame\n"); - if (irqst & RRDYEN) - dev_dbg(mcbsp->dev, "RX Buffer Threshold Reached\n"); - if (irqst & RUNDFLEN) - dev_err(mcbsp->dev, "RX Buffer Underflow!\n"); - if (irqst & ROVFLEN) - dev_err(mcbsp->dev, "RX Buffer Overflow!\n"); - - if (irqst & XSYNCERREN) - dev_err(mcbsp->dev, "TX Frame Sync Error!\n"); - if (irqst & XFSXEN) - dev_dbg(mcbsp->dev, "TX Frame Sync\n"); - if (irqst & XEOFEN) - dev_dbg(mcbsp->dev, "TX End Of Frame\n"); - if (irqst & XRDYEN) - dev_dbg(mcbsp->dev, "TX Buffer threshold Reached\n"); - if (irqst & XUNDFLEN) - dev_err(mcbsp->dev, "TX Buffer Underflow!\n"); - if (irqst & XOVFLEN) - dev_err(mcbsp->dev, "TX Buffer Overflow!\n"); - if (irqst & XEMPTYEOFEN) - dev_dbg(mcbsp->dev, "TX Buffer empty at end of frame\n"); - - MCBSP_WRITE(mcbsp, IRQST, irqst); - - return IRQ_HANDLED; -} - -static irqreturn_t omap_mcbsp_tx_irq_handler(int irq, void *dev_id) -{ - struct omap_mcbsp *mcbsp_tx = dev_id; - u16 irqst_spcr2; - - irqst_spcr2 = MCBSP_READ(mcbsp_tx, SPCR2); - dev_dbg(mcbsp_tx->dev, "TX IRQ callback : 0x%x\n", irqst_spcr2); - - if (irqst_spcr2 & XSYNC_ERR) { - dev_err(mcbsp_tx->dev, "TX Frame Sync Error! : 0x%x\n", - irqst_spcr2); - /* Writing zero to XSYNC_ERR clears the IRQ */ - MCBSP_WRITE(mcbsp_tx, SPCR2, MCBSP_READ_CACHE(mcbsp_tx, SPCR2)); - } - - return IRQ_HANDLED; -} - -static irqreturn_t omap_mcbsp_rx_irq_handler(int irq, void *dev_id) -{ - struct omap_mcbsp *mcbsp_rx = dev_id; - u16 irqst_spcr1; - - irqst_spcr1 = MCBSP_READ(mcbsp_rx, SPCR1); - dev_dbg(mcbsp_rx->dev, "RX IRQ callback : 0x%x\n", irqst_spcr1); - - if (irqst_spcr1 & RSYNC_ERR) { - dev_err(mcbsp_rx->dev, "RX Frame Sync Error! : 0x%x\n", - irqst_spcr1); - /* Writing zero to RSYNC_ERR clears the IRQ */ - MCBSP_WRITE(mcbsp_rx, SPCR1, MCBSP_READ_CACHE(mcbsp_rx, SPCR1)); - } - - return IRQ_HANDLED; -} - -/* - * omap_mcbsp_config simply write a config to the - * appropriate McBSP. - * You either call this function or set the McBSP registers - * by yourself before calling omap_mcbsp_start(). - */ -void omap_mcbsp_config(struct omap_mcbsp *mcbsp, - const struct omap_mcbsp_reg_cfg *config) -{ - dev_dbg(mcbsp->dev, "Configuring McBSP%d phys_base: 0x%08lx\n", - mcbsp->id, mcbsp->phys_base); - - /* We write the given config */ - MCBSP_WRITE(mcbsp, SPCR2, config->spcr2); - MCBSP_WRITE(mcbsp, SPCR1, config->spcr1); - MCBSP_WRITE(mcbsp, RCR2, config->rcr2); - MCBSP_WRITE(mcbsp, RCR1, config->rcr1); - MCBSP_WRITE(mcbsp, XCR2, config->xcr2); - MCBSP_WRITE(mcbsp, XCR1, config->xcr1); - MCBSP_WRITE(mcbsp, SRGR2, config->srgr2); - MCBSP_WRITE(mcbsp, SRGR1, config->srgr1); - MCBSP_WRITE(mcbsp, MCR2, config->mcr2); - MCBSP_WRITE(mcbsp, MCR1, config->mcr1); - MCBSP_WRITE(mcbsp, PCR0, config->pcr0); - if (mcbsp->pdata->has_ccr) { - MCBSP_WRITE(mcbsp, XCCR, config->xccr); - MCBSP_WRITE(mcbsp, RCCR, config->rccr); - } - /* Enable wakeup behavior */ - if (mcbsp->pdata->has_wakeup) - MCBSP_WRITE(mcbsp, WAKEUPEN, XRDYEN | RRDYEN); - - /* Enable TX/RX sync error interrupts by default */ - if (mcbsp->irq) - MCBSP_WRITE(mcbsp, IRQEN, RSYNCERREN | XSYNCERREN | - RUNDFLEN | ROVFLEN | XUNDFLEN | XOVFLEN); -} - -/** - * omap_mcbsp_dma_reg_params - returns the address of mcbsp data register - * @id - mcbsp id - * @stream - indicates the direction of data flow (rx or tx) - * - * Returns the address of mcbsp data transmit register or data receive register - * to be used by DMA for transferring/receiving data based on the value of - * @stream for the requested mcbsp given by @id - */ -static int omap_mcbsp_dma_reg_params(struct omap_mcbsp *mcbsp, - unsigned int stream) -{ - int data_reg; - - if (mcbsp->pdata->reg_size == 2) { - if (stream) - data_reg = OMAP_MCBSP_REG_DRR1; - else - data_reg = OMAP_MCBSP_REG_DXR1; - } else { - if (stream) - data_reg = OMAP_MCBSP_REG_DRR; - else - data_reg = OMAP_MCBSP_REG_DXR; - } - - return mcbsp->phys_dma_base + data_reg * mcbsp->pdata->reg_step; -} - -static void omap_st_on(struct omap_mcbsp *mcbsp) -{ - unsigned int w; - - if (mcbsp->pdata->force_ick_on) - mcbsp->pdata->force_ick_on(mcbsp->st_data->mcbsp_iclk, true); - - /* Disable Sidetone clock auto-gating for normal operation */ - w = MCBSP_ST_READ(mcbsp, SYSCONFIG); - MCBSP_ST_WRITE(mcbsp, SYSCONFIG, w & ~(ST_AUTOIDLE)); - - /* Enable McBSP Sidetone */ - w = MCBSP_READ(mcbsp, SSELCR); - MCBSP_WRITE(mcbsp, SSELCR, w | SIDETONEEN); - - /* Enable Sidetone from Sidetone Core */ - w = MCBSP_ST_READ(mcbsp, SSELCR); - MCBSP_ST_WRITE(mcbsp, SSELCR, w | ST_SIDETONEEN); -} - -static void omap_st_off(struct omap_mcbsp *mcbsp) -{ - unsigned int w; - - w = MCBSP_ST_READ(mcbsp, SSELCR); - MCBSP_ST_WRITE(mcbsp, SSELCR, w & ~(ST_SIDETONEEN)); - - w = MCBSP_READ(mcbsp, SSELCR); - MCBSP_WRITE(mcbsp, SSELCR, w & ~(SIDETONEEN)); - - /* Enable Sidetone clock auto-gating to reduce power consumption */ - w = MCBSP_ST_READ(mcbsp, SYSCONFIG); - MCBSP_ST_WRITE(mcbsp, SYSCONFIG, w | ST_AUTOIDLE); - - if (mcbsp->pdata->force_ick_on) - mcbsp->pdata->force_ick_on(mcbsp->st_data->mcbsp_iclk, false); -} - -static void omap_st_fir_write(struct omap_mcbsp *mcbsp, s16 *fir) -{ - u16 val, i; - - val = MCBSP_ST_READ(mcbsp, SSELCR); - - if (val & ST_COEFFWREN) - MCBSP_ST_WRITE(mcbsp, SSELCR, val & ~(ST_COEFFWREN)); - - MCBSP_ST_WRITE(mcbsp, SSELCR, val | ST_COEFFWREN); - - for (i = 0; i < 128; i++) - MCBSP_ST_WRITE(mcbsp, SFIRCR, fir[i]); - - i = 0; - - val = MCBSP_ST_READ(mcbsp, SSELCR); - while (!(val & ST_COEFFWRDONE) && (++i < 1000)) - val = MCBSP_ST_READ(mcbsp, SSELCR); - - MCBSP_ST_WRITE(mcbsp, SSELCR, val & ~(ST_COEFFWREN)); - - if (i == 1000) - dev_err(mcbsp->dev, "McBSP FIR load error!\n"); -} - -static void omap_st_chgain(struct omap_mcbsp *mcbsp) -{ - u16 w; - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - - w = MCBSP_ST_READ(mcbsp, SSELCR); - - MCBSP_ST_WRITE(mcbsp, SGAINCR, ST_CH0GAIN(st_data->ch0gain) | \ - ST_CH1GAIN(st_data->ch1gain)); -} - -int omap_st_set_chgain(struct omap_mcbsp *mcbsp, int channel, s16 chgain) -{ - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - int ret = 0; - - if (!st_data) - return -ENOENT; - - spin_lock_irq(&mcbsp->lock); - if (channel == 0) - st_data->ch0gain = chgain; - else if (channel == 1) - st_data->ch1gain = chgain; - else - ret = -EINVAL; - - if (st_data->enabled) - omap_st_chgain(mcbsp); - spin_unlock_irq(&mcbsp->lock); - - return ret; -} - -int omap_st_get_chgain(struct omap_mcbsp *mcbsp, int channel, s16 *chgain) -{ - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - int ret = 0; - - if (!st_data) - return -ENOENT; - - spin_lock_irq(&mcbsp->lock); - if (channel == 0) - *chgain = st_data->ch0gain; - else if (channel == 1) - *chgain = st_data->ch1gain; - else - ret = -EINVAL; - spin_unlock_irq(&mcbsp->lock); - - return ret; -} - -static int omap_st_start(struct omap_mcbsp *mcbsp) -{ - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - - if (st_data->enabled && !st_data->running) { - omap_st_fir_write(mcbsp, st_data->taps); - omap_st_chgain(mcbsp); - - if (!mcbsp->free) { - omap_st_on(mcbsp); - st_data->running = 1; - } - } - - return 0; -} - -int omap_st_enable(struct omap_mcbsp *mcbsp) -{ - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - - if (!st_data) - return -ENODEV; - - spin_lock_irq(&mcbsp->lock); - st_data->enabled = 1; - omap_st_start(mcbsp); - spin_unlock_irq(&mcbsp->lock); - - return 0; -} - -static int omap_st_stop(struct omap_mcbsp *mcbsp) -{ - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - - if (st_data->running) { - if (!mcbsp->free) { - omap_st_off(mcbsp); - st_data->running = 0; - } - } - - return 0; -} - -int omap_st_disable(struct omap_mcbsp *mcbsp) -{ - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - int ret = 0; - - if (!st_data) - return -ENODEV; - - spin_lock_irq(&mcbsp->lock); - omap_st_stop(mcbsp); - st_data->enabled = 0; - spin_unlock_irq(&mcbsp->lock); - - return ret; -} - -int omap_st_is_enabled(struct omap_mcbsp *mcbsp) -{ - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - - if (!st_data) - return -ENODEV; - - return st_data->enabled; -} - -/* - * omap_mcbsp_set_rx_threshold configures the transmit threshold in words. - * The threshold parameter is 1 based, and it is converted (threshold - 1) - * for the THRSH2 register. - */ -void omap_mcbsp_set_tx_threshold(struct omap_mcbsp *mcbsp, u16 threshold) -{ - if (mcbsp->pdata->buffer_size == 0) - return; - - if (threshold && threshold <= mcbsp->max_tx_thres) - MCBSP_WRITE(mcbsp, THRSH2, threshold - 1); -} - -/* - * omap_mcbsp_set_rx_threshold configures the receive threshold in words. - * The threshold parameter is 1 based, and it is converted (threshold - 1) - * for the THRSH1 register. - */ -void omap_mcbsp_set_rx_threshold(struct omap_mcbsp *mcbsp, u16 threshold) -{ - if (mcbsp->pdata->buffer_size == 0) - return; - - if (threshold && threshold <= mcbsp->max_rx_thres) - MCBSP_WRITE(mcbsp, THRSH1, threshold - 1); -} - -/* - * omap_mcbsp_get_tx_delay returns the number of used slots in the McBSP FIFO - */ -u16 omap_mcbsp_get_tx_delay(struct omap_mcbsp *mcbsp) -{ - u16 buffstat; - - if (mcbsp->pdata->buffer_size == 0) - return 0; - - /* Returns the number of free locations in the buffer */ - buffstat = MCBSP_READ(mcbsp, XBUFFSTAT); - - /* Number of slots are different in McBSP ports */ - return mcbsp->pdata->buffer_size - buffstat; -} - -/* - * omap_mcbsp_get_rx_delay returns the number of free slots in the McBSP FIFO - * to reach the threshold value (when the DMA will be triggered to read it) - */ -u16 omap_mcbsp_get_rx_delay(struct omap_mcbsp *mcbsp) -{ - u16 buffstat, threshold; - - if (mcbsp->pdata->buffer_size == 0) - return 0; - - /* Returns the number of used locations in the buffer */ - buffstat = MCBSP_READ(mcbsp, RBUFFSTAT); - /* RX threshold */ - threshold = MCBSP_READ(mcbsp, THRSH1); - - /* Return the number of location till we reach the threshold limit */ - if (threshold <= buffstat) - return 0; - else - return threshold - buffstat; -} - -int omap_mcbsp_request(struct omap_mcbsp *mcbsp) -{ - void *reg_cache; - int err; - - reg_cache = kzalloc(mcbsp->reg_cache_size, GFP_KERNEL); - if (!reg_cache) { - return -ENOMEM; - } - - spin_lock(&mcbsp->lock); - if (!mcbsp->free) { - dev_err(mcbsp->dev, "McBSP%d is currently in use\n", - mcbsp->id); - err = -EBUSY; - goto err_kfree; - } - - mcbsp->free = false; - mcbsp->reg_cache = reg_cache; - spin_unlock(&mcbsp->lock); - - if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->request) - mcbsp->pdata->ops->request(mcbsp->id - 1); - - /* - * Make sure that transmitter, receiver and sample-rate generator are - * not running before activating IRQs. - */ - MCBSP_WRITE(mcbsp, SPCR1, 0); - MCBSP_WRITE(mcbsp, SPCR2, 0); - - if (mcbsp->irq) { - err = request_irq(mcbsp->irq, omap_mcbsp_irq_handler, 0, - "McBSP", (void *)mcbsp); - if (err != 0) { - dev_err(mcbsp->dev, "Unable to request IRQ\n"); - goto err_clk_disable; - } - } else { - err = request_irq(mcbsp->tx_irq, omap_mcbsp_tx_irq_handler, 0, - "McBSP TX", (void *)mcbsp); - if (err != 0) { - dev_err(mcbsp->dev, "Unable to request TX IRQ\n"); - goto err_clk_disable; - } - - err = request_irq(mcbsp->rx_irq, omap_mcbsp_rx_irq_handler, 0, - "McBSP RX", (void *)mcbsp); - if (err != 0) { - dev_err(mcbsp->dev, "Unable to request RX IRQ\n"); - goto err_free_irq; - } - } - - return 0; -err_free_irq: - free_irq(mcbsp->tx_irq, (void *)mcbsp); -err_clk_disable: - if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->free) - mcbsp->pdata->ops->free(mcbsp->id - 1); - - /* Disable wakeup behavior */ - if (mcbsp->pdata->has_wakeup) - MCBSP_WRITE(mcbsp, WAKEUPEN, 0); - - spin_lock(&mcbsp->lock); - mcbsp->free = true; - mcbsp->reg_cache = NULL; -err_kfree: - spin_unlock(&mcbsp->lock); - kfree(reg_cache); - - return err; -} - -void omap_mcbsp_free(struct omap_mcbsp *mcbsp) -{ - void *reg_cache; - - if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->free) - mcbsp->pdata->ops->free(mcbsp->id - 1); - - /* Disable wakeup behavior */ - if (mcbsp->pdata->has_wakeup) - MCBSP_WRITE(mcbsp, WAKEUPEN, 0); - - /* Disable interrupt requests */ - if (mcbsp->irq) - MCBSP_WRITE(mcbsp, IRQEN, 0); - - if (mcbsp->irq) { - free_irq(mcbsp->irq, (void *)mcbsp); - } else { - free_irq(mcbsp->rx_irq, (void *)mcbsp); - free_irq(mcbsp->tx_irq, (void *)mcbsp); - } - - reg_cache = mcbsp->reg_cache; - - /* - * Select CLKS source from internal source unconditionally before - * marking the McBSP port as free. - * If the external clock source via MCBSP_CLKS pin has been selected the - * system will refuse to enter idle if the CLKS pin source is not reset - * back to internal source. - */ - if (!mcbsp_omap1()) - omap2_mcbsp_set_clks_src(mcbsp, MCBSP_CLKS_PRCM_SRC); - - spin_lock(&mcbsp->lock); - if (mcbsp->free) - dev_err(mcbsp->dev, "McBSP%d was not reserved\n", mcbsp->id); - else - mcbsp->free = true; - mcbsp->reg_cache = NULL; - spin_unlock(&mcbsp->lock); - - kfree(reg_cache); -} - -/* - * Here we start the McBSP, by enabling transmitter, receiver or both. - * If no transmitter or receiver is active prior calling, then sample-rate - * generator and frame sync are started. - */ -void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int tx, int rx) -{ - int enable_srg = 0; - u16 w; - - if (mcbsp->st_data) - omap_st_start(mcbsp); - - /* Only enable SRG, if McBSP is master */ - w = MCBSP_READ_CACHE(mcbsp, PCR0); - if (w & (FSXM | FSRM | CLKXM | CLKRM)) - enable_srg = !((MCBSP_READ_CACHE(mcbsp, SPCR2) | - MCBSP_READ_CACHE(mcbsp, SPCR1)) & 1); - - if (enable_srg) { - /* Start the sample generator */ - w = MCBSP_READ_CACHE(mcbsp, SPCR2); - MCBSP_WRITE(mcbsp, SPCR2, w | (1 << 6)); - } - - /* Enable transmitter and receiver */ - tx &= 1; - w = MCBSP_READ_CACHE(mcbsp, SPCR2); - MCBSP_WRITE(mcbsp, SPCR2, w | tx); - - rx &= 1; - w = MCBSP_READ_CACHE(mcbsp, SPCR1); - MCBSP_WRITE(mcbsp, SPCR1, w | rx); - - /* - * Worst case: CLKSRG*2 = 8000khz: (1/8000) * 2 * 2 usec - * REVISIT: 100us may give enough time for two CLKSRG, however - * due to some unknown PM related, clock gating etc. reason it - * is now at 500us. - */ - udelay(500); - - if (enable_srg) { - /* Start frame sync */ - w = MCBSP_READ_CACHE(mcbsp, SPCR2); - MCBSP_WRITE(mcbsp, SPCR2, w | (1 << 7)); - } - - if (mcbsp->pdata->has_ccr) { - /* Release the transmitter and receiver */ - w = MCBSP_READ_CACHE(mcbsp, XCCR); - w &= ~(tx ? XDISABLE : 0); - MCBSP_WRITE(mcbsp, XCCR, w); - w = MCBSP_READ_CACHE(mcbsp, RCCR); - w &= ~(rx ? RDISABLE : 0); - MCBSP_WRITE(mcbsp, RCCR, w); - } - - /* Dump McBSP Regs */ - omap_mcbsp_dump_reg(mcbsp); -} - -void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int tx, int rx) -{ - int idle; - u16 w; - - /* Reset transmitter */ - tx &= 1; - if (mcbsp->pdata->has_ccr) { - w = MCBSP_READ_CACHE(mcbsp, XCCR); - w |= (tx ? XDISABLE : 0); - MCBSP_WRITE(mcbsp, XCCR, w); - } - w = MCBSP_READ_CACHE(mcbsp, SPCR2); - MCBSP_WRITE(mcbsp, SPCR2, w & ~tx); - - /* Reset receiver */ - rx &= 1; - if (mcbsp->pdata->has_ccr) { - w = MCBSP_READ_CACHE(mcbsp, RCCR); - w |= (rx ? RDISABLE : 0); - MCBSP_WRITE(mcbsp, RCCR, w); - } - w = MCBSP_READ_CACHE(mcbsp, SPCR1); - MCBSP_WRITE(mcbsp, SPCR1, w & ~rx); - - idle = !((MCBSP_READ_CACHE(mcbsp, SPCR2) | - MCBSP_READ_CACHE(mcbsp, SPCR1)) & 1); - - if (idle) { - /* Reset the sample rate generator */ - w = MCBSP_READ_CACHE(mcbsp, SPCR2); - MCBSP_WRITE(mcbsp, SPCR2, w & ~(1 << 6)); - } - - if (mcbsp->st_data) - omap_st_stop(mcbsp); -} - -int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) -{ - struct clk *fck_src; - const char *src; - int r; - - if (fck_src_id == MCBSP_CLKS_PAD_SRC) - src = "pad_fck"; - else if (fck_src_id == MCBSP_CLKS_PRCM_SRC) - src = "prcm_fck"; - else - return -EINVAL; - - fck_src = clk_get(mcbsp->dev, src); - if (IS_ERR(fck_src)) { - dev_err(mcbsp->dev, "CLKS: could not clk_get() %s\n", src); - return -EINVAL; - } - - pm_runtime_put_sync(mcbsp->dev); - - r = clk_set_parent(mcbsp->fclk, fck_src); - if (r) { - dev_err(mcbsp->dev, "CLKS: could not clk_set_parent() to %s\n", - src); - clk_put(fck_src); - return r; - } - - pm_runtime_get_sync(mcbsp->dev); - - clk_put(fck_src); - - return 0; - -} - -#define max_thres(m) (mcbsp->pdata->buffer_size) -#define valid_threshold(m, val) ((val) <= max_thres(m)) -#define THRESHOLD_PROP_BUILDER(prop) \ -static ssize_t prop##_show(struct device *dev, \ - struct device_attribute *attr, char *buf) \ -{ \ - struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); \ - \ - return sprintf(buf, "%u\n", mcbsp->prop); \ -} \ - \ -static ssize_t prop##_store(struct device *dev, \ - struct device_attribute *attr, \ - const char *buf, size_t size) \ -{ \ - struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); \ - unsigned long val; \ - int status; \ - \ - status = kstrtoul(buf, 0, &val); \ - if (status) \ - return status; \ - \ - if (!valid_threshold(mcbsp, val)) \ - return -EDOM; \ - \ - mcbsp->prop = val; \ - return size; \ -} \ - \ -static DEVICE_ATTR(prop, 0644, prop##_show, prop##_store); - -THRESHOLD_PROP_BUILDER(max_tx_thres); -THRESHOLD_PROP_BUILDER(max_rx_thres); - -static const char *dma_op_modes[] = { - "element", "threshold", -}; - -static ssize_t dma_op_mode_show(struct device *dev, - struct device_attribute *attr, char *buf) -{ - struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); - int dma_op_mode, i = 0; - ssize_t len = 0; - const char * const *s; - - dma_op_mode = mcbsp->dma_op_mode; - - for (s = &dma_op_modes[i]; i < ARRAY_SIZE(dma_op_modes); s++, i++) { - if (dma_op_mode == i) - len += sprintf(buf + len, "[%s] ", *s); - else - len += sprintf(buf + len, "%s ", *s); - } - len += sprintf(buf + len, "\n"); - - return len; -} - -static ssize_t dma_op_mode_store(struct device *dev, - struct device_attribute *attr, - const char *buf, size_t size) -{ - struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); - int i; - - i = sysfs_match_string(dma_op_modes, buf); - if (i < 0) - return i; - - spin_lock_irq(&mcbsp->lock); - if (!mcbsp->free) { - size = -EBUSY; - goto unlock; - } - mcbsp->dma_op_mode = i; - -unlock: - spin_unlock_irq(&mcbsp->lock); - - return size; -} - -static DEVICE_ATTR_RW(dma_op_mode); - -static const struct attribute *additional_attrs[] = { - &dev_attr_max_tx_thres.attr, - &dev_attr_max_rx_thres.attr, - &dev_attr_dma_op_mode.attr, - NULL, -}; - -static const struct attribute_group additional_attr_group = { - .attrs = (struct attribute **)additional_attrs, -}; - -static ssize_t st_taps_show(struct device *dev, - struct device_attribute *attr, char *buf) -{ - struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - ssize_t status = 0; - int i; - - spin_lock_irq(&mcbsp->lock); - for (i = 0; i < st_data->nr_taps; i++) - status += sprintf(&buf[status], (i ? ", %d" : "%d"), - st_data->taps[i]); - if (i) - status += sprintf(&buf[status], "\n"); - spin_unlock_irq(&mcbsp->lock); - - return status; -} - -static ssize_t st_taps_store(struct device *dev, - struct device_attribute *attr, - const char *buf, size_t size) -{ - struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - int val, tmp, status, i = 0; - - spin_lock_irq(&mcbsp->lock); - memset(st_data->taps, 0, sizeof(st_data->taps)); - st_data->nr_taps = 0; - - do { - status = sscanf(buf, "%d%n", &val, &tmp); - if (status < 0 || status == 0) { - size = -EINVAL; - goto out; - } - if (val < -32768 || val > 32767) { - size = -EINVAL; - goto out; - } - st_data->taps[i++] = val; - buf += tmp; - if (*buf != ',') - break; - buf++; - } while (1); - - st_data->nr_taps = i; - -out: - spin_unlock_irq(&mcbsp->lock); - - return size; -} - -static DEVICE_ATTR_RW(st_taps); - -static const struct attribute *sidetone_attrs[] = { - &dev_attr_st_taps.attr, - NULL, -}; - -static const struct attribute_group sidetone_attr_group = { - .attrs = (struct attribute **)sidetone_attrs, -}; - -static int omap_st_add(struct omap_mcbsp *mcbsp, struct resource *res) -{ - struct omap_mcbsp_st_data *st_data; - int err; - - st_data = devm_kzalloc(mcbsp->dev, sizeof(*mcbsp->st_data), GFP_KERNEL); - if (!st_data) - return -ENOMEM; - - st_data->mcbsp_iclk = clk_get(mcbsp->dev, "ick"); - if (IS_ERR(st_data->mcbsp_iclk)) { - dev_warn(mcbsp->dev, - "Failed to get ick, sidetone might be broken\n"); - st_data->mcbsp_iclk = NULL; - } - - st_data->io_base_st = devm_ioremap(mcbsp->dev, res->start, - resource_size(res)); - if (!st_data->io_base_st) - return -ENOMEM; - - err = sysfs_create_group(&mcbsp->dev->kobj, &sidetone_attr_group); - if (err) - return err; - - mcbsp->st_data = st_data; - return 0; -} - -/* - * McBSP1 and McBSP3 are directly mapped on 1610 and 1510. - * 730 has only 2 McBSP, and both of them are MPU peripherals. - */ -int omap_mcbsp_init(struct platform_device *pdev) -{ - struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); - struct resource *res; - int ret = 0; - - spin_lock_init(&mcbsp->lock); - mcbsp->free = true; - - res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); - if (!res) - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - - mcbsp->io_base = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(mcbsp->io_base)) - return PTR_ERR(mcbsp->io_base); - - mcbsp->phys_base = res->start; - mcbsp->reg_cache_size = resource_size(res); - - res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dma"); - if (!res) - mcbsp->phys_dma_base = mcbsp->phys_base; - else - mcbsp->phys_dma_base = res->start; - - /* - * OMAP1, 2 uses two interrupt lines: TX, RX - * OMAP2430, OMAP3 SoC have combined IRQ line as well. - * OMAP4 and newer SoC only have the combined IRQ line. - * Use the combined IRQ if available since it gives better debugging - * possibilities. - */ - mcbsp->irq = platform_get_irq_byname(pdev, "common"); - if (mcbsp->irq == -ENXIO) { - mcbsp->tx_irq = platform_get_irq_byname(pdev, "tx"); - - if (mcbsp->tx_irq == -ENXIO) { - mcbsp->irq = platform_get_irq(pdev, 0); - mcbsp->tx_irq = 0; - } else { - mcbsp->rx_irq = platform_get_irq_byname(pdev, "rx"); - mcbsp->irq = 0; - } - } - - if (!pdev->dev.of_node) { - res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx"); - if (!res) { - dev_err(&pdev->dev, "invalid tx DMA channel\n"); - return -ENODEV; - } - mcbsp->dma_req[0] = res->start; - mcbsp->dma_data[0].filter_data = &mcbsp->dma_req[0]; - - res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx"); - if (!res) { - dev_err(&pdev->dev, "invalid rx DMA channel\n"); - return -ENODEV; - } - mcbsp->dma_req[1] = res->start; - mcbsp->dma_data[1].filter_data = &mcbsp->dma_req[1]; - } else { - mcbsp->dma_data[0].filter_data = "tx"; - mcbsp->dma_data[1].filter_data = "rx"; - } - - mcbsp->dma_data[0].addr = omap_mcbsp_dma_reg_params(mcbsp, 0); - mcbsp->dma_data[0].maxburst = 4; - - mcbsp->dma_data[1].addr = omap_mcbsp_dma_reg_params(mcbsp, 1); - mcbsp->dma_data[1].maxburst = 4; - - mcbsp->fclk = clk_get(&pdev->dev, "fck"); - if (IS_ERR(mcbsp->fclk)) { - ret = PTR_ERR(mcbsp->fclk); - dev_err(mcbsp->dev, "unable to get fck: %d\n", ret); - return ret; - } - - mcbsp->dma_op_mode = MCBSP_DMA_MODE_ELEMENT; - if (mcbsp->pdata->buffer_size) { - /* - * Initially configure the maximum thresholds to a safe value. - * The McBSP FIFO usage with these values should not go under - * 16 locations. - * If the whole FIFO without safety buffer is used, than there - * is a possibility that the DMA will be not able to push the - * new data on time, causing channel shifts in runtime. - */ - mcbsp->max_tx_thres = max_thres(mcbsp) - 0x10; - mcbsp->max_rx_thres = max_thres(mcbsp) - 0x10; - - ret = sysfs_create_group(&mcbsp->dev->kobj, - &additional_attr_group); - if (ret) { - dev_err(mcbsp->dev, - "Unable to create additional controls\n"); - goto err_thres; - } - } else { - mcbsp->max_tx_thres = -EINVAL; - mcbsp->max_rx_thres = -EINVAL; - } - - res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "sidetone"); - if (res) { - ret = omap_st_add(mcbsp, res); - if (ret) { - dev_err(mcbsp->dev, - "Unable to create sidetone controls\n"); - goto err_st; - } - } - - return 0; - -err_st: - if (mcbsp->pdata->buffer_size) - sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); -err_thres: - clk_put(mcbsp->fclk); - return ret; -} - -void omap_mcbsp_cleanup(struct omap_mcbsp *mcbsp) -{ - if (mcbsp->pdata->buffer_size) - sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); - - if (mcbsp->st_data) { - sysfs_remove_group(&mcbsp->dev->kobj, &sidetone_attr_group); - clk_put(mcbsp->st_data->mcbsp_iclk); - } -} diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 776e148b0aa2..67159a6b90a8 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -19,14 +19,13 @@ config SND_MMP_SOC config SND_PXA2XX_AC97 tristate - select SND_AC97_CODEC config SND_PXA2XX_SOC_AC97 tristate - select AC97_BUS + select AC97_BUS_NEW select SND_PXA2XX_LIB select SND_PXA2XX_LIB_AC97 - select SND_SOC_AC97_BUS + select SND_SOC_AC97_BUS_NEW config SND_PXA2XX_SOC_I2S select SND_PXA2XX_LIB @@ -80,6 +79,7 @@ config SND_PXA2XX_SOC_TOSA tristate "SoC AC97 Audio support for Tosa" depends on SND_PXA2XX_SOC && MACH_TOSA depends on MFD_TC6393XB + depends on AC97_BUS=n select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -89,6 +89,7 @@ config SND_PXA2XX_SOC_TOSA config SND_PXA2XX_SOC_E740 tristate "SoC AC97 Audio support for e740" depends on SND_PXA2XX_SOC && MACH_E740 + depends on AC97_BUS=n select SND_SOC_WM9705 select SND_PXA2XX_SOC_AC97 help @@ -98,6 +99,7 @@ config SND_PXA2XX_SOC_E740 config SND_PXA2XX_SOC_E750 tristate "SoC AC97 Audio support for e750" depends on SND_PXA2XX_SOC && MACH_E750 + depends on AC97_BUS=n select SND_SOC_WM9705 select SND_PXA2XX_SOC_AC97 help @@ -107,6 +109,7 @@ config SND_PXA2XX_SOC_E750 config SND_PXA2XX_SOC_E800 tristate "SoC AC97 Audio support for e800" depends on SND_PXA2XX_SOC && MACH_E800 + depends on AC97_BUS=n select SND_SOC_WM9712 select SND_PXA2XX_SOC_AC97 help @@ -117,6 +120,7 @@ config SND_PXA2XX_SOC_EM_X270 tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300" depends on SND_PXA2XX_SOC && (MACH_EM_X270 || MACH_EXEDA || \ MACH_CM_X300) + depends on AC97_BUS=n select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -127,6 +131,7 @@ config SND_PXA2XX_SOC_PALM27X bool "SoC Audio support for Palm T|X, T5, E2 and LifeDrive" depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || \ MACH_PALMT5 || MACH_PALMTE2) + depends on AC97_BUS=n select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -156,6 +161,7 @@ config SND_SOC_TTC_DKB config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" depends on SND_PXA2XX_SOC && MACH_ZYLONITE + depends on AC97_BUS=n select SND_PXA2XX_SOC_AC97 select SND_PXA_SOC_SSP select SND_SOC_WM9713 @@ -163,16 +169,6 @@ config SND_SOC_ZYLONITE Say Y if you want to add support for SoC audio on the Marvell Zylonite reference platform. -config SND_SOC_RAUMFELD - tristate "SoC Audio support Raumfeld audio adapter" - depends on SND_PXA2XX_SOC && (MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR) - depends on I2C && SPI_MASTER - select SND_PXA_SOC_SSP - select SND_SOC_CS4270 - select SND_SOC_AK4104 - help - Say Y if you want to add support for SoC audio on Raumfeld devices - config SND_PXA2XX_SOC_HX4700 tristate "SoC Audio support for HP iPAQ hx4700" depends on SND_PXA2XX_SOC && MACH_H4700 && I2C @@ -195,6 +191,7 @@ config SND_PXA2XX_SOC_MAGICIAN config SND_PXA2XX_SOC_MIOA701 tristate "SoC Audio support for MIO A701" depends on SND_PXA2XX_SOC && MACH_MIOA701 + depends on AC97_BUS=n select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9713 help diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 5b265662f04f..0ab2a9dcb720 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -49,6 +49,5 @@ obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o -obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o obj-$(CONFIG_SND_MMP_SOC_BROWNSTONE) += snd-soc-brownstone.o obj-$(CONFIG_SND_SOC_TTC_DKB) += snd-soc-ttc-dkb.o diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 69033e1a84e6..adcf8ba9d287 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -103,6 +103,9 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, pxa_ssp_disable(ssp); } + if (priv->extclk) + clk_prepare_enable(priv->extclk); + dma = kzalloc(sizeof(struct snd_dmaengine_dai_dma_data), GFP_KERNEL); if (!dma) return -ENOMEM; @@ -125,6 +128,9 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, clk_disable_unprepare(ssp->clk); } + if (priv->extclk) + clk_disable_unprepare(priv->extclk); + kfree(snd_soc_dai_get_dma_data(cpu_dai, substream)); snd_soc_dai_set_dma_data(cpu_dai, substream, NULL); } diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 9f779657bc86..f8a3aa6c6d4e 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -17,6 +17,7 @@ #include <linux/dmaengine.h> #include <linux/dma/pxa-dma.h> +#include <sound/ac97/controller.h> #include <sound/core.h> #include <sound/ac97_codec.h> #include <sound/soc.h> @@ -27,43 +28,35 @@ #include <mach/regs-ac97.h> #include <mach/audio.h> -static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) +static void pxa2xx_ac97_warm_reset(struct ac97_controller *adrv) { pxa2xx_ac97_try_warm_reset(); pxa2xx_ac97_finish_reset(); } -static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97) +static void pxa2xx_ac97_cold_reset(struct ac97_controller *adrv) { pxa2xx_ac97_try_cold_reset(); pxa2xx_ac97_finish_reset(); } -static unsigned short pxa2xx_ac97_legacy_read(struct snd_ac97 *ac97, - unsigned short reg) +static int pxa2xx_ac97_read_actrl(struct ac97_controller *adrv, int slot, + unsigned short reg) { - int ret; - - ret = pxa2xx_ac97_read(ac97->num, reg); - if (ret < 0) - return 0; - else - return (unsigned short)(ret & 0xffff); + return pxa2xx_ac97_read(slot, reg); } -static void pxa2xx_ac97_legacy_write(struct snd_ac97 *ac97, - unsigned short reg, unsigned short val) +static int pxa2xx_ac97_write_actrl(struct ac97_controller *adrv, int slot, + unsigned short reg, unsigned short val) { - int ret; - - ret = pxa2xx_ac97_write(ac97->num, reg, val); + return pxa2xx_ac97_write(slot, reg, val); } -static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { - .read = pxa2xx_ac97_legacy_read, - .write = pxa2xx_ac97_legacy_write, +static struct ac97_controller_ops pxa2xx_ac97_ops = { + .read = pxa2xx_ac97_read_actrl, + .write = pxa2xx_ac97_write_actrl, .warm_reset = pxa2xx_ac97_warm_reset, .reset = pxa2xx_ac97_cold_reset, }; @@ -233,6 +226,9 @@ MODULE_DEVICE_TABLE(of, pxa2xx_ac97_dt_ids); static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) { int ret; + struct ac97_controller *ctrl; + pxa2xx_audio_ops_t *pdata = pdev->dev.platform_data; + void **codecs_pdata; if (pdev->id != -1) { dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n"); @@ -245,10 +241,14 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) return ret; } - ret = snd_soc_set_ac97_ops(&pxa2xx_ac97_ops); - if (ret != 0) - return ret; + codecs_pdata = pdata ? pdata->codec_pdata : NULL; + ctrl = snd_ac97_controller_register(&pxa2xx_ac97_ops, &pdev->dev, + AC97_SLOTS_AVAILABLE_ALL, + codecs_pdata); + if (IS_ERR(ctrl)) + return PTR_ERR(ctrl); + platform_set_drvdata(pdev, ctrl); /* Punt most of the init to the SoC probe; we may need the machine * driver to do interesting things with the clocking to get us up * and running. @@ -259,8 +259,10 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) static int pxa2xx_ac97_dev_remove(struct platform_device *pdev) { + struct ac97_controller *ctrl = platform_get_drvdata(pdev); + snd_soc_unregister_component(&pdev->dev); - snd_soc_set_ac97_ops(NULL); + snd_ac97_controller_unregister(ctrl); pxa2xx_ac97_hw_remove(pdev); return 0; } diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c deleted file mode 100644 index 111a907c4eb9..000000000000 --- a/sound/soc/pxa/raumfeld.c +++ /dev/null @@ -1,318 +0,0 @@ -/* - * raumfeld_audio.c -- SoC audio for Raumfeld audio devices - * - * Copyright (c) 2009 Daniel Mack <daniel@caiaq.de> - * - * based on code from: - * - * Wolfson Microelectronics PLC. - * Openedhand Ltd. - * Liam Girdwood <lrg@slimlogic.co.uk> - * Richard Purdie <richard@openedhand.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#include <linux/module.h> -#include <linux/i2c.h> -#include <linux/delay.h> -#include <linux/gpio.h> -#include <sound/pcm.h> -#include <sound/soc.h> - -#include <asm/mach-types.h> - -#include "pxa-ssp.h" - -#define GPIO_SPDIF_RESET (38) -#define GPIO_MCLK_RESET (111) -#define GPIO_CODEC_RESET (120) - -static struct i2c_client *max9486_client; -static struct i2c_board_info max9486_hwmon_info = { - I2C_BOARD_INFO("max9485", 0x63), -}; - -#define MAX9485_MCLK_FREQ_112896 0x22 -#define MAX9485_MCLK_FREQ_122880 0x23 -#define MAX9485_MCLK_FREQ_225792 0x32 -#define MAX9485_MCLK_FREQ_245760 0x33 - -static void set_max9485_clk(char clk) -{ - i2c_master_send(max9486_client, &clk, 1); -} - -static void raumfeld_enable_audio(bool en) -{ - if (en) { - gpio_set_value(GPIO_MCLK_RESET, 1); - - /* wait some time to let the clocks become stable */ - msleep(100); - - gpio_set_value(GPIO_SPDIF_RESET, 1); - gpio_set_value(GPIO_CODEC_RESET, 1); - } else { - gpio_set_value(GPIO_MCLK_RESET, 0); - gpio_set_value(GPIO_SPDIF_RESET, 0); - gpio_set_value(GPIO_CODEC_RESET, 0); - } -} - -/* CS4270 */ -static int raumfeld_cs4270_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - - /* set freq to 0 to enable all possible codec sample rates */ - return snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0); -} - -static void raumfeld_cs4270_shutdown(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - - /* set freq to 0 to enable all possible codec sample rates */ - snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0); -} - -static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int clk = 0; - int ret = 0; - - switch (params_rate(params)) { - case 44100: - set_max9485_clk(MAX9485_MCLK_FREQ_112896); - clk = 11289600; - break; - case 48000: - set_max9485_clk(MAX9485_MCLK_FREQ_122880); - clk = 12288000; - break; - case 88200: - set_max9485_clk(MAX9485_MCLK_FREQ_225792); - clk = 22579200; - break; - case 96000: - set_max9485_clk(MAX9485_MCLK_FREQ_245760); - clk = 24576000; - break; - default: - return -EINVAL; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, 0); - if (ret < 0) - return ret; - - /* setup the CPU DAI */ - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); - if (ret < 0) - return ret; - - return 0; -} - -static const struct snd_soc_ops raumfeld_cs4270_ops = { - .startup = raumfeld_cs4270_startup, - .shutdown = raumfeld_cs4270_shutdown, - .hw_params = raumfeld_cs4270_hw_params, -}; - -static int raumfeld_analog_suspend(struct snd_soc_card *card) -{ - raumfeld_enable_audio(false); - return 0; -} - -static int raumfeld_analog_resume(struct snd_soc_card *card) -{ - raumfeld_enable_audio(true); - return 0; -} - -/* AK4104 */ - -static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int ret = 0, clk = 0; - - switch (params_rate(params)) { - case 44100: - set_max9485_clk(MAX9485_MCLK_FREQ_112896); - clk = 11289600; - break; - case 48000: - set_max9485_clk(MAX9485_MCLK_FREQ_122880); - clk = 12288000; - break; - case 88200: - set_max9485_clk(MAX9485_MCLK_FREQ_225792); - clk = 22579200; - break; - case 96000: - set_max9485_clk(MAX9485_MCLK_FREQ_245760); - clk = 24576000; - break; - default: - return -EINVAL; - } - - /* setup the CPU DAI */ - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); - if (ret < 0) - return ret; - - return 0; -} - -static struct snd_soc_ops raumfeld_ak4104_ops = { - .hw_params = raumfeld_ak4104_hw_params, -}; - -#define DAI_LINK_CS4270 \ -{ \ - .name = "CS4270", \ - .stream_name = "CS4270", \ - .cpu_dai_name = "pxa-ssp-dai.0", \ - .platform_name = "pxa-pcm-audio", \ - .codec_dai_name = "cs4270-hifi", \ - .codec_name = "cs4270.0-0048", \ - .dai_fmt = SND_SOC_DAIFMT_I2S | \ - SND_SOC_DAIFMT_NB_NF | \ - SND_SOC_DAIFMT_CBS_CFS, \ - .ops = &raumfeld_cs4270_ops, \ -} - -#define DAI_LINK_AK4104 \ -{ \ - .name = "ak4104", \ - .stream_name = "Playback", \ - .cpu_dai_name = "pxa-ssp-dai.1", \ - .codec_dai_name = "ak4104-hifi", \ - .platform_name = "pxa-pcm-audio", \ - .dai_fmt = SND_SOC_DAIFMT_I2S | \ - SND_SOC_DAIFMT_NB_NF | \ - SND_SOC_DAIFMT_CBS_CFS, \ - .ops = &raumfeld_ak4104_ops, \ - .codec_name = "spi0.0", \ -} - -static struct snd_soc_dai_link snd_soc_raumfeld_connector_dai[] = { - DAI_LINK_CS4270, - DAI_LINK_AK4104, -}; - -static struct snd_soc_dai_link snd_soc_raumfeld_speaker_dai[] = { - DAI_LINK_CS4270, -}; - -static struct snd_soc_card snd_soc_raumfeld_connector = { - .name = "Raumfeld Connector", - .owner = THIS_MODULE, - .dai_link = snd_soc_raumfeld_connector_dai, - .num_links = ARRAY_SIZE(snd_soc_raumfeld_connector_dai), - .suspend_post = raumfeld_analog_suspend, - .resume_pre = raumfeld_analog_resume, -}; - -static struct snd_soc_card snd_soc_raumfeld_speaker = { - .name = "Raumfeld Speaker", - .owner = THIS_MODULE, - .dai_link = snd_soc_raumfeld_speaker_dai, - .num_links = ARRAY_SIZE(snd_soc_raumfeld_speaker_dai), - .suspend_post = raumfeld_analog_suspend, - .resume_pre = raumfeld_analog_resume, -}; - -static struct platform_device *raumfeld_audio_device; - -static int __init raumfeld_audio_init(void) -{ - int ret; - - if (!machine_is_raumfeld_speaker() && - !machine_is_raumfeld_connector()) - return 0; - - max9486_client = i2c_new_device(i2c_get_adapter(0), - &max9486_hwmon_info); - - if (!max9486_client) - return -ENOMEM; - - set_max9485_clk(MAX9485_MCLK_FREQ_122880); - - /* Register analog device */ - raumfeld_audio_device = platform_device_alloc("soc-audio", 0); - if (!raumfeld_audio_device) - return -ENOMEM; - - if (machine_is_raumfeld_speaker()) - platform_set_drvdata(raumfeld_audio_device, - &snd_soc_raumfeld_speaker); - - if (machine_is_raumfeld_connector()) - platform_set_drvdata(raumfeld_audio_device, - &snd_soc_raumfeld_connector); - - ret = platform_device_add(raumfeld_audio_device); - if (ret < 0) { - platform_device_put(raumfeld_audio_device); - return ret; - } - - raumfeld_enable_audio(true); - return 0; -} - -static void __exit raumfeld_audio_exit(void) -{ - raumfeld_enable_audio(false); - - platform_device_unregister(raumfeld_audio_device); - - i2c_unregister_device(max9486_client); - - gpio_free(GPIO_MCLK_RESET); - gpio_free(GPIO_CODEC_RESET); - gpio_free(GPIO_SPDIF_RESET); -} - -module_init(raumfeld_audio_init); -module_exit(raumfeld_audio_exit); - -/* Module information */ -MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>"); -MODULE_DESCRIPTION("Raumfeld audio SoC"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 2a4c912d1e48..804ae0d93058 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -66,6 +66,7 @@ config SND_SOC_QDSP6_ASM tristate config SND_SOC_QDSP6_ASM_DAI + select SND_SOC_COMPRESS tristate config SND_SOC_QDSP6 @@ -100,6 +101,7 @@ config SND_SOC_SDM845 depends on QCOM_APR select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON + select SND_SOC_RT5663 help To add support for audio on Qualcomm Technologies Inc. SDM845 SoC-based systems. diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index 1543e85629f8..fb45f396ab4a 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -25,13 +25,12 @@ static int apq8096_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, static void apq8096_add_be_ops(struct snd_soc_card *card) { - struct snd_soc_dai_link *link = card->dai_link; - int i, num_links = card->num_links; + struct snd_soc_dai_link *link; + int i; - for (i = 0; i < num_links; i++) { + for_each_card_prelinks(card, i, link) { if (link->no_pcm == 1) link->be_hw_params_fixup = apq8096_be_hw_params_fixup; - link++; } } diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c index eb1b9da05dd4..4715527054e5 100644 --- a/sound/soc/qcom/common.c +++ b/sound/soc/qcom/common.c @@ -13,6 +13,7 @@ int qcom_snd_parse_of(struct snd_soc_card *card) struct device_node *cpu = NULL; struct device *dev = card->dev; struct snd_soc_dai_link *link; + struct of_phandle_args args; int ret, num_links; ret = snd_soc_of_parse_card_name(card, "model"); @@ -47,12 +48,14 @@ int qcom_snd_parse_of(struct snd_soc_card *card) goto err; } - link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0); - if (!link->cpu_of_node) { + ret = of_parse_phandle_with_args(cpu, "sound-dai", + "#sound-dai-cells", 0, &args); + if (ret) { dev_err(card->dev, "error getting cpu phandle\n"); - ret = -EINVAL; goto err; } + link->cpu_of_node = args.np; + link->id = args.args[0]; ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name); if (ret) { diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index d07271ea4c45..028bce671cbc 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -91,7 +91,7 @@ static int lpass_platform_pcmops_open(struct snd_pcm_substream *substream) if (ret) { dev_err(soc_runtime->dev, "error writing to rdmactl reg: %d\n", ret); - return ret; + return ret; } data->dma_ch = dma_ch; diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c index 932c3ebfd252..da242515e146 100644 --- a/sound/soc/qcom/qdsp6/q6adm.c +++ b/sound/soc/qcom/qdsp6/q6adm.c @@ -2,25 +2,24 @@ // Copyright (c) 2011-2017, The Linux Foundation. All rights reserved. // Copyright (c) 2018, Linaro Limited -#include <linux/slab.h> -#include <linux/wait.h> -#include <linux/kernel.h> #include <linux/device.h> -#include <linux/module.h> -#include <linux/sched.h> #include <linux/jiffies.h> +#include <linux/kernel.h> +#include <linux/kref.h> +#include <linux/module.h> #include <linux/of.h> #include <linux/of_platform.h> -#include <linux/kref.h> -#include <linux/wait.h> -#include <linux/soc/qcom/apr.h> #include <linux/platform_device.h> +#include <linux/sched.h> +#include <linux/slab.h> +#include <linux/soc/qcom/apr.h> +#include <linux/wait.h> #include <sound/asound.h> #include "q6adm.h" #include "q6afe.h" #include "q6core.h" -#include "q6dsp-errno.h" #include "q6dsp-common.h" +#include "q6dsp-errno.h" #define ADM_CMD_DEVICE_OPEN_V5 0x00010326 #define ADM_CMDRSP_DEVICE_OPEN_V5 0x00010329 diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index 60ff4a2d3577..dc645ba4d8d0 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -341,6 +341,7 @@ static int q6afe_dai_prepare(struct snd_pcm_substream *substream, switch (dai->id) { case HDMI_RX: + case DISPLAY_PORT_RX: q6afe_hdmi_port_prepare(dai_data->port[dai->id], &dai_data->port_config[dai->id].hdmi); break; @@ -445,6 +446,7 @@ static int q6afe_mi2s_set_sysclk(struct snd_soc_dai *dai, static const struct snd_soc_dapm_route q6afe_dapm_routes[] = { {"HDMI Playback", NULL, "HDMI_RX"}, + {"Display Port Playback", NULL, "DISPLAY_PORT_RX"}, {"Slimbus1 Playback", NULL, "SLIMBUS_1_RX"}, {"Slimbus2 Playback", NULL, "SLIMBUS_2_RX"}, {"Slimbus3 Playback", NULL, "SLIMBUS_3_RX"}, @@ -561,13 +563,13 @@ static const struct snd_soc_dapm_route q6afe_dapm_routes[] = { {"QUAT_MI2S_TX", NULL, "Quaternary MI2S Capture"}, }; -static struct snd_soc_dai_ops q6hdmi_ops = { +static const struct snd_soc_dai_ops q6hdmi_ops = { .prepare = q6afe_dai_prepare, .hw_params = q6hdmi_hw_params, .shutdown = q6afe_dai_shutdown, }; -static struct snd_soc_dai_ops q6i2s_ops = { +static const struct snd_soc_dai_ops q6i2s_ops = { .prepare = q6afe_dai_prepare, .hw_params = q6i2s_hw_params, .set_fmt = q6i2s_set_fmt, @@ -575,14 +577,14 @@ static struct snd_soc_dai_ops q6i2s_ops = { .set_sysclk = q6afe_mi2s_set_sysclk, }; -static struct snd_soc_dai_ops q6slim_ops = { +static const struct snd_soc_dai_ops q6slim_ops = { .prepare = q6afe_dai_prepare, .hw_params = q6slim_hw_params, .shutdown = q6afe_dai_shutdown, .set_channel_map = q6slim_set_channel_map, }; -static struct snd_soc_dai_ops q6tdm_ops = { +static const struct snd_soc_dai_ops q6tdm_ops = { .prepare = q6afe_dai_prepare, .shutdown = q6afe_dai_shutdown, .set_sysclk = q6afe_mi2s_set_sysclk, @@ -1090,6 +1092,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = { Q6AFE_TDM_CAP_DAI("Quinary", 5, QUINARY_TDM_TX_5), Q6AFE_TDM_CAP_DAI("Quinary", 6, QUINARY_TDM_TX_6), Q6AFE_TDM_CAP_DAI("Quinary", 7, QUINARY_TDM_TX_7), + { + .playback = { + .stream_name = "Display Port Playback", + .rates = SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 2, + .channels_max = 8, + .rate_max = 192000, + .rate_min = 48000, + }, + .ops = &q6hdmi_ops, + .id = DISPLAY_PORT_RX, + .name = "DISPLAY_PORT", + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + }, }; static int q6afe_of_xlate_dai_name(struct snd_soc_component *component, @@ -1112,205 +1133,206 @@ static int q6afe_of_xlate_dai_name(struct snd_soc_component *component, } static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = { - SND_SOC_DAPM_AIF_OUT("HDMI_RX", "HDMI Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SLIMBUS_0_RX", "Slimbus Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SLIMBUS_1_RX", "Slimbus1 Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SLIMBUS_2_RX", "Slimbus2 Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SLIMBUS_3_RX", "Slimbus3 Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SLIMBUS_4_RX", "Slimbus4 Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SLIMBUS_5_RX", "Slimbus5 Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SLIMBUS_6_RX", "Slimbus6 Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SLIMBUS_0_TX", "Slimbus Capture", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SLIMBUS_1_TX", "Slimbus1 Capture", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SLIMBUS_2_TX", "Slimbus2 Capture", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SLIMBUS_3_TX", "Slimbus3 Capture", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SLIMBUS_4_TX", "Slimbus4 Capture", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SLIMBUS_5_TX", "Slimbus5 Capture", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SLIMBUS_6_TX", "Slimbus6 Capture", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUAT_MI2S_RX", "Quaternary MI2S Playback", + SND_SOC_DAPM_AIF_IN("HDMI_RX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_0_RX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_1_RX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_2_RX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_3_RX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_4_RX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_5_RX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_6_RX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("SLIMBUS_0_TX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("SLIMBUS_1_TX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("SLIMBUS_2_TX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("SLIMBUS_3_TX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("SLIMBUS_4_TX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("SLIMBUS_5_TX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("SLIMBUS_6_TX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("QUAT_MI2S_RX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUAT_MI2S_TX", "Quaternary MI2S Capture", + SND_SOC_DAPM_AIF_OUT("QUAT_MI2S_TX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("TERT_MI2S_RX", "Tertiary MI2S Playback", + SND_SOC_DAPM_AIF_IN("TERT_MI2S_RX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("TERT_MI2S_TX", "Tertiary MI2S Capture", + SND_SOC_DAPM_AIF_OUT("TERT_MI2S_TX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SEC_MI2S_RX", "Secondary MI2S Playback", + SND_SOC_DAPM_AIF_IN("SEC_MI2S_RX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SEC_MI2S_TX", "Secondary MI2S Capture", + SND_SOC_DAPM_AIF_OUT("SEC_MI2S_TX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SEC_MI2S_RX_SD1", + SND_SOC_DAPM_AIF_IN("SEC_MI2S_RX_SD1", "Secondary MI2S Playback SD1", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("PRI_MI2S_RX", "Primary MI2S Playback", + SND_SOC_DAPM_AIF_IN("PRI_MI2S_RX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("PRI_MI2S_TX", "Primary MI2S Capture", + SND_SOC_DAPM_AIF_OUT("PRI_MI2S_TX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_0", "Primary TDM0 Playback", + SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_0", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_1", "Primary TDM1 Playback", + SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_1", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_2", "Primary TDM2 Playback", + SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_2", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_3", "Primary TDM3 Playback", + SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_3", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_4", "Primary TDM4 Playback", + SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_4", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_5", "Primary TDM5 Playback", + SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_5", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_6", "Primary TDM6 Playback", + SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_6", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_7", "Primary TDM7 Playback", + SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_7", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_0", "Primary TDM0 Capture", + SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_0", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_1", "Primary TDM1 Capture", + SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_1", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_2", "Primary TDM2 Capture", + SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_2", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_3", "Primary TDM3 Capture", + SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_3", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_4", "Primary TDM4 Capture", + SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_4", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_5", "Primary TDM5 Capture", + SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_5", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_6", "Primary TDM6 Capture", + SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_6", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_7", "Primary TDM7 Capture", + SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_7", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_0", "Secondary TDM0 Playback", + SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_0", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_1", "Secondary TDM1 Playback", + SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_1", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_2", "Secondary TDM2 Playback", + SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_2", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_3", "Secondary TDM3 Playback", + SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_3", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_4", "Secondary TDM4 Playback", + SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_4", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_5", "Secondary TDM5 Playback", + SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_5", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_6", "Secondary TDM6 Playback", + SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_6", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_7", "Secondary TDM7 Playback", + SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_7", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_0", "Secondary TDM0 Capture", + SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_0", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_1", "Secondary TDM1 Capture", + SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_1", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_2", "Secondary TDM2 Capture", + SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_2", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_3", "Secondary TDM3 Capture", + SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_3", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_4", "Secondary TDM4 Capture", + SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_4", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_5", "Secondary TDM5 Capture", + SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_5", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_6", "Secondary TDM6 Capture", + SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_6", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_7", "Secondary TDM7 Capture", + SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_7", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_0", "Tertiary TDM0 Playback", + SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_0", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_1", "Tertiary TDM1 Playback", + SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_1", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_2", "Tertiary TDM2 Playback", + SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_2", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_3", "Tertiary TDM3 Playback", + SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_3", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_4", "Tertiary TDM4 Playback", + SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_4", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_5", "Tertiary TDM5 Playback", + SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_5", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_6", "Tertiary TDM6 Playback", + SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_6", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_7", "Tertiary TDM7 Playback", + SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_7", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_0", "Tertiary TDM0 Capture", + SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_0", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_1", "Tertiary TDM1 Capture", + SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_1", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_2", "Tertiary TDM2 Capture", + SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_2", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_3", "Tertiary TDM3 Capture", + SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_3", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_4", "Tertiary TDM4 Capture", + SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_4", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_5", "Tertiary TDM5 Capture", + SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_5", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_6", "Tertiary TDM6 Capture", + SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_6", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_7", "Tertiary TDM7 Capture", + SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_7", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_0", "Quaternary TDM0 Playback", + SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_0", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_1", "Quaternary TDM1 Playback", + SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_1", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_2", "Quaternary TDM2 Playback", + SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_2", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_3", "Quaternary TDM3 Playback", + SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_3", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_4", "Quaternary TDM4 Playback", + SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_4", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_5", "Quaternary TDM5 Playback", + SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_5", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_6", "Quaternary TDM6 Playback", + SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_6", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_7", "Quaternary TDM7 Playback", + SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_7", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_0", "Quaternary TDM0 Capture", + SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_0", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_1", "Quaternary TDM1 Capture", + SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_1", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_2", "Quaternary TDM2 Capture", + SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_2", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_3", "Quaternary TDM3 Capture", + SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_3", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_4", "Quaternary TDM4 Capture", + SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_4", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_5", "Quaternary TDM5 Capture", + SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_5", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_6", "Quaternary TDM6 Capture", + SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_6", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_7", "Quaternary TDM7 Capture", + SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_7", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_0", "Quinary TDM0 Playback", + SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_0", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_1", "Quinary TDM1 Playback", + SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_1", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_2", "Quinary TDM2 Playback", + SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_2", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_3", "Quinary TDM3 Playback", + SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_3", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_4", "Quinary TDM4 Playback", + SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_4", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_5", "Quinary TDM5 Playback", + SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_5", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_6", "Quinary TDM6 Playback", + SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_6", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_7", "Quinary TDM7 Playback", + SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_7", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_0", "Quinary TDM0 Capture", + SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_0", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_1", "Quinary TDM1 Capture", + SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_1", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_2", "Quinary TDM2 Capture", + SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_2", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_3", "Quinary TDM3 Capture", + SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_3", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_4", "Quinary TDM4 Capture", + SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_4", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_5", "Quinary TDM5 Capture", + SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_5", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_6", "Quinary TDM6 Capture", + SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_6", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_7", "Quinary TDM7 Capture", + SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_7", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("DISPLAY_PORT_RX", "NULL", 0, 0, 0, 0), }; static const struct snd_soc_component_driver q6afe_dai_component = { diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index 000775b4bba8..e0945f7a58c8 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -49,14 +49,14 @@ #define AFE_PORT_I2S_SD1 0x2 #define AFE_PORT_I2S_SD2 0x3 #define AFE_PORT_I2S_SD3 0x4 -#define AFE_PORT_I2S_SD0_MASK BIT(0x1) -#define AFE_PORT_I2S_SD1_MASK BIT(0x2) -#define AFE_PORT_I2S_SD2_MASK BIT(0x3) -#define AFE_PORT_I2S_SD3_MASK BIT(0x4) -#define AFE_PORT_I2S_SD0_1_MASK GENMASK(2, 1) -#define AFE_PORT_I2S_SD2_3_MASK GENMASK(4, 3) -#define AFE_PORT_I2S_SD0_1_2_MASK GENMASK(3, 1) -#define AFE_PORT_I2S_SD0_1_2_3_MASK GENMASK(4, 1) +#define AFE_PORT_I2S_SD0_MASK BIT(0x0) +#define AFE_PORT_I2S_SD1_MASK BIT(0x1) +#define AFE_PORT_I2S_SD2_MASK BIT(0x2) +#define AFE_PORT_I2S_SD3_MASK BIT(0x3) +#define AFE_PORT_I2S_SD0_1_MASK GENMASK(1, 0) +#define AFE_PORT_I2S_SD2_3_MASK GENMASK(3, 2) +#define AFE_PORT_I2S_SD0_1_2_MASK GENMASK(2, 0) +#define AFE_PORT_I2S_SD0_1_2_3_MASK GENMASK(3, 0) #define AFE_PORT_I2S_QUAD01 0x5 #define AFE_PORT_I2S_QUAD23 0x6 #define AFE_PORT_I2S_6CHS 0x7 @@ -71,6 +71,7 @@ /* Port IDs */ #define AFE_API_VERSION_HDMI_CONFIG 0x1 #define AFE_PORT_ID_MULTICHAN_HDMI_RX 0x100E +#define AFE_PORT_ID_HDMI_OVER_DP_RX 0x6020 #define AFE_API_VERSION_SLIMBUS_CONFIG 0x1 /* Clock set API version */ @@ -704,6 +705,8 @@ static struct afe_port_map port_maps[AFE_PORT_MAX] = { QUINARY_TDM_RX_7, 1, 1}, [QUINARY_TDM_TX_7] = { AFE_PORT_ID_QUINARY_TDM_TX_7, QUINARY_TDM_TX_7, 0, 1}, + [DISPLAY_PORT_RX] = { AFE_PORT_ID_HDMI_OVER_DP_RX, + DISPLAY_PORT_RX, 1, 1}, }; static void q6afe_port_free(struct kref *ref) @@ -1384,6 +1387,7 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id) switch (port_id) { case AFE_PORT_ID_MULTICHAN_HDMI_RX: + case AFE_PORT_ID_HDMI_OVER_DP_RX: cfg_type = AFE_PARAM_ID_HDMI_CONFIG; break; case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX: diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 9db9a2944ef2..548eb4fa2da6 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -8,9 +8,10 @@ #include <linux/platform_device.h> #include <linux/slab.h> #include <sound/soc.h> -#include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/pcm.h> +#include <linux/spinlock.h> +#include <sound/compress_driver.h> #include <asm/dma.h> #include <linux/dma-mapping.h> #include <linux/of_device.h> @@ -31,6 +32,15 @@ #define CAPTURE_MIN_PERIOD_SIZE 320 #define SID_MASK_DEFAULT 0xF +/* Default values used if user space does not set */ +#define COMPR_PLAYBACK_MIN_FRAGMENT_SIZE (8 * 1024) +#define COMPR_PLAYBACK_MAX_FRAGMENT_SIZE (128 * 1024) +#define COMPR_PLAYBACK_MIN_NUM_FRAGMENTS (4) +#define COMPR_PLAYBACK_MAX_NUM_FRAGMENTS (16 * 4) +#define Q6ASM_DAI_TX_RX 0 +#define Q6ASM_DAI_TX 1 +#define Q6ASM_DAI_RX 2 + enum stream_state { Q6ASM_STREAM_IDLE = 0, Q6ASM_STREAM_STOPPED, @@ -39,11 +49,18 @@ enum stream_state { struct q6asm_dai_rtd { struct snd_pcm_substream *substream; + struct snd_compr_stream *cstream; + struct snd_compr_params codec_param; + struct snd_dma_buffer dma_buffer; + spinlock_t lock; phys_addr_t phys; unsigned int pcm_size; unsigned int pcm_count; unsigned int pcm_irq_pos; /* IRQ position */ unsigned int periods; + unsigned int bytes_sent; + unsigned int bytes_received; + unsigned int copied_total; uint16_t bits_per_sample; uint16_t source; /* Encoding source bit mask */ struct audio_client *audio_client; @@ -123,7 +140,6 @@ static struct snd_pcm_hardware q6asm_dai_hardware_playback = { .rate_max = 48000, \ }, \ .name = "MultiMedia"#num, \ - .probe = fe_dai_probe, \ .id = MSM_FRONTEND_DAI_MULTIMEDIA##num, \ } @@ -139,6 +155,21 @@ static struct snd_pcm_hw_constraint_list constraints_sample_rates = { .mask = 0, }; +static const struct snd_compr_codec_caps q6asm_compr_caps = { + .num_descriptors = 1, + .descriptor[0].max_ch = 2, + .descriptor[0].sample_rates = { 8000, 11025, 12000, 16000, 22050, + 24000, 32000, 44100, 48000, 88200, + 96000, 176400, 192000 }, + .descriptor[0].num_sample_rates = 13, + .descriptor[0].bit_rate[0] = 320, + .descriptor[0].bit_rate[1] = 128, + .descriptor[0].num_bitrates = 2, + .descriptor[0].profiles = 0, + .descriptor[0].modes = SND_AUDIOCHANMODE_MP3_STEREO, + .descriptor[0].formats = 0, +}; + static void event_handler(uint32_t opcode, uint32_t token, uint32_t *payload, void *priv) { @@ -319,10 +350,11 @@ static int q6asm_dai_open(struct snd_pcm_substream *substream) prtd->audio_client = q6asm_audio_client_alloc(dev, (q6asm_cb)event_handler, prtd, stream_id, LEGACY_PCM_MODE); - if (!prtd->audio_client) { + if (IS_ERR(prtd->audio_client)) { pr_info("%s: Could not allocate memory\n", __func__); + ret = PTR_ERR(prtd->audio_client); kfree(prtd); - return -ENOMEM; + return ret; } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -461,6 +493,313 @@ static struct snd_pcm_ops q6asm_dai_ops = { .mmap = q6asm_dai_mmap, }; +static void compress_event_handler(uint32_t opcode, uint32_t token, + uint32_t *payload, void *priv) +{ + struct q6asm_dai_rtd *prtd = priv; + struct snd_compr_stream *substream = prtd->cstream; + unsigned long flags; + uint64_t avail; + + switch (opcode) { + case ASM_CLIENT_EVENT_CMD_RUN_DONE: + spin_lock_irqsave(&prtd->lock, flags); + if (!prtd->bytes_sent) { + q6asm_write_async(prtd->audio_client, prtd->pcm_count, + 0, 0, NO_TIMESTAMP); + prtd->bytes_sent += prtd->pcm_count; + } + + spin_unlock_irqrestore(&prtd->lock, flags); + break; + + case ASM_CLIENT_EVENT_CMD_EOS_DONE: + prtd->state = Q6ASM_STREAM_STOPPED; + break; + + case ASM_CLIENT_EVENT_DATA_WRITE_DONE: + spin_lock_irqsave(&prtd->lock, flags); + + prtd->copied_total += prtd->pcm_count; + snd_compr_fragment_elapsed(substream); + + if (prtd->state != Q6ASM_STREAM_RUNNING) { + spin_unlock_irqrestore(&prtd->lock, flags); + break; + } + + avail = prtd->bytes_received - prtd->bytes_sent; + + if (avail >= prtd->pcm_count) { + q6asm_write_async(prtd->audio_client, + prtd->pcm_count, 0, 0, NO_TIMESTAMP); + prtd->bytes_sent += prtd->pcm_count; + } + + spin_unlock_irqrestore(&prtd->lock, flags); + break; + + default: + break; + } +} + +static int q6asm_dai_compr_open(struct snd_compr_stream *stream) +{ + struct snd_soc_pcm_runtime *rtd = stream->private_data; + struct snd_soc_component *c = snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct snd_compr_runtime *runtime = stream->runtime; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct q6asm_dai_data *pdata; + struct device *dev = c->dev; + struct q6asm_dai_rtd *prtd; + int stream_id, size, ret; + + stream_id = cpu_dai->driver->id; + pdata = snd_soc_component_get_drvdata(c); + if (!pdata) { + dev_err(dev, "Drv data not found ..\n"); + return -EINVAL; + } + + prtd = kzalloc(sizeof(*prtd), GFP_KERNEL); + if (!prtd) + return -ENOMEM; + + prtd->cstream = stream; + prtd->audio_client = q6asm_audio_client_alloc(dev, + (q6asm_cb)compress_event_handler, + prtd, stream_id, LEGACY_PCM_MODE); + if (IS_ERR(prtd->audio_client)) { + dev_err(dev, "Could not allocate memory\n"); + ret = PTR_ERR(prtd->audio_client); + goto free_prtd; + } + + size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE * + COMPR_PLAYBACK_MAX_NUM_FRAGMENTS; + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dev, size, + &prtd->dma_buffer); + if (ret) { + dev_err(dev, "Cannot allocate buffer(s)\n"); + goto free_client; + } + + if (pdata->sid < 0) + prtd->phys = prtd->dma_buffer.addr; + else + prtd->phys = prtd->dma_buffer.addr | (pdata->sid << 32); + + snd_compr_set_runtime_buffer(stream, &prtd->dma_buffer); + spin_lock_init(&prtd->lock); + runtime->private_data = prtd; + + return 0; + +free_client: + q6asm_audio_client_free(prtd->audio_client); +free_prtd: + kfree(prtd); + + return ret; +} + +static int q6asm_dai_compr_free(struct snd_compr_stream *stream) +{ + struct snd_compr_runtime *runtime = stream->runtime; + struct q6asm_dai_rtd *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = stream->private_data; + + if (prtd->audio_client) { + if (prtd->state) + q6asm_cmd(prtd->audio_client, CMD_CLOSE); + + snd_dma_free_pages(&prtd->dma_buffer); + q6asm_unmap_memory_regions(stream->direction, + prtd->audio_client); + q6asm_audio_client_free(prtd->audio_client); + prtd->audio_client = NULL; + } + q6routing_stream_close(rtd->dai_link->id, stream->direction); + kfree(prtd); + + return 0; +} + +static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream, + struct snd_compr_params *params) +{ + struct snd_compr_runtime *runtime = stream->runtime; + struct q6asm_dai_rtd *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = stream->private_data; + struct snd_soc_component *c = snd_soc_rtdcom_lookup(rtd, DRV_NAME); + int dir = stream->direction; + struct q6asm_dai_data *pdata; + struct device *dev = c->dev; + int ret; + + memcpy(&prtd->codec_param, params, sizeof(*params)); + + pdata = snd_soc_component_get_drvdata(c); + if (!pdata) + return -EINVAL; + + if (!prtd || !prtd->audio_client) { + dev_err(dev, "private data null or audio client freed\n"); + return -EINVAL; + } + + prtd->periods = runtime->fragments; + prtd->pcm_count = runtime->fragment_size; + prtd->pcm_size = runtime->fragments * runtime->fragment_size; + prtd->bits_per_sample = 16; + if (dir == SND_COMPRESS_PLAYBACK) { + ret = q6asm_open_write(prtd->audio_client, params->codec.id, + prtd->bits_per_sample); + + if (ret < 0) { + dev_err(dev, "q6asm_open_write failed\n"); + q6asm_audio_client_free(prtd->audio_client); + prtd->audio_client = NULL; + return ret; + } + } + + prtd->session_id = q6asm_get_session_id(prtd->audio_client); + ret = q6routing_stream_open(rtd->dai_link->id, LEGACY_PCM_MODE, + prtd->session_id, dir); + if (ret) { + dev_err(dev, "Stream reg failed ret:%d\n", ret); + return ret; + } + + ret = q6asm_map_memory_regions(dir, prtd->audio_client, prtd->phys, + (prtd->pcm_size / prtd->periods), + prtd->periods); + + if (ret < 0) { + dev_err(dev, "Buffer Mapping failed ret:%d\n", ret); + return -ENOMEM; + } + + prtd->state = Q6ASM_STREAM_RUNNING; + + return 0; +} + +static int q6asm_dai_compr_trigger(struct snd_compr_stream *stream, int cmd) +{ + struct snd_compr_runtime *runtime = stream->runtime; + struct q6asm_dai_rtd *prtd = runtime->private_data; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = q6asm_run_nowait(prtd->audio_client, 0, 0, 0); + break; + case SNDRV_PCM_TRIGGER_STOP: + prtd->state = Q6ASM_STREAM_STOPPED; + ret = q6asm_cmd_nowait(prtd->audio_client, CMD_EOS); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE); + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static int q6asm_dai_compr_pointer(struct snd_compr_stream *stream, + struct snd_compr_tstamp *tstamp) +{ + struct snd_compr_runtime *runtime = stream->runtime; + struct q6asm_dai_rtd *prtd = runtime->private_data; + unsigned long flags; + + spin_lock_irqsave(&prtd->lock, flags); + + tstamp->copied_total = prtd->copied_total; + tstamp->byte_offset = prtd->copied_total % prtd->pcm_size; + + spin_unlock_irqrestore(&prtd->lock, flags); + + return 0; +} + +static int q6asm_dai_compr_ack(struct snd_compr_stream *stream, + size_t count) +{ + struct snd_compr_runtime *runtime = stream->runtime; + struct q6asm_dai_rtd *prtd = runtime->private_data; + unsigned long flags; + + spin_lock_irqsave(&prtd->lock, flags); + prtd->bytes_received += count; + spin_unlock_irqrestore(&prtd->lock, flags); + + return count; +} + +static int q6asm_dai_compr_mmap(struct snd_compr_stream *stream, + struct vm_area_struct *vma) +{ + struct snd_compr_runtime *runtime = stream->runtime; + struct q6asm_dai_rtd *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = stream->private_data; + struct snd_soc_component *c = snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct device *dev = c->dev; + + return dma_mmap_coherent(dev, vma, + prtd->dma_buffer.area, prtd->dma_buffer.addr, + prtd->dma_buffer.bytes); +} + +static int q6asm_dai_compr_get_caps(struct snd_compr_stream *stream, + struct snd_compr_caps *caps) +{ + caps->direction = SND_COMPRESS_PLAYBACK; + caps->min_fragment_size = COMPR_PLAYBACK_MIN_FRAGMENT_SIZE; + caps->max_fragment_size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE; + caps->min_fragments = COMPR_PLAYBACK_MIN_NUM_FRAGMENTS; + caps->max_fragments = COMPR_PLAYBACK_MAX_NUM_FRAGMENTS; + caps->num_codecs = 1; + caps->codecs[0] = SND_AUDIOCODEC_MP3; + + return 0; +} + +static int q6asm_dai_compr_get_codec_caps(struct snd_compr_stream *stream, + struct snd_compr_codec_caps *codec) +{ + switch (codec->codec) { + case SND_AUDIOCODEC_MP3: + *codec = q6asm_compr_caps; + break; + default: + break; + } + + return 0; +} + +static struct snd_compr_ops q6asm_dai_compr_ops = { + .open = q6asm_dai_compr_open, + .free = q6asm_dai_compr_free, + .set_params = q6asm_dai_compr_set_params, + .pointer = q6asm_dai_compr_pointer, + .trigger = q6asm_dai_compr_trigger, + .get_caps = q6asm_dai_compr_get_caps, + .get_codec_caps = q6asm_dai_compr_get_codec_caps, + .mmap = q6asm_dai_compr_mmap, + .ack = q6asm_dai_compr_ack, +}; + static int q6asm_dai_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_pcm_substream *psubstream, *csubstream; @@ -493,7 +832,7 @@ static int q6asm_dai_pcm_new(struct snd_soc_pcm_runtime *rtd) } } - return ret; + return 0; } static void q6asm_dai_pcm_free(struct snd_pcm *pcm) @@ -511,44 +850,12 @@ static void q6asm_dai_pcm_free(struct snd_pcm *pcm) } } -static const struct snd_soc_dapm_route afe_pcm_routes[] = { - {"MM_DL1", NULL, "MultiMedia1 Playback" }, - {"MM_DL2", NULL, "MultiMedia2 Playback" }, - {"MM_DL3", NULL, "MultiMedia3 Playback" }, - {"MM_DL4", NULL, "MultiMedia4 Playback" }, - {"MM_DL5", NULL, "MultiMedia5 Playback" }, - {"MM_DL6", NULL, "MultiMedia6 Playback" }, - {"MM_DL7", NULL, "MultiMedia7 Playback" }, - {"MM_DL7", NULL, "MultiMedia8 Playback" }, - {"MultiMedia1 Capture", NULL, "MM_UL1"}, - {"MultiMedia2 Capture", NULL, "MM_UL2"}, - {"MultiMedia3 Capture", NULL, "MM_UL3"}, - {"MultiMedia4 Capture", NULL, "MM_UL4"}, - {"MultiMedia5 Capture", NULL, "MM_UL5"}, - {"MultiMedia6 Capture", NULL, "MM_UL6"}, - {"MultiMedia7 Capture", NULL, "MM_UL7"}, - {"MultiMedia8 Capture", NULL, "MM_UL8"}, - -}; - -static int fe_dai_probe(struct snd_soc_dai *dai) -{ - struct snd_soc_dapm_context *dapm; - - dapm = snd_soc_component_get_dapm(dai->component); - snd_soc_dapm_add_routes(dapm, afe_pcm_routes, - ARRAY_SIZE(afe_pcm_routes)); - - return 0; -} - - static const struct snd_soc_component_driver q6asm_fe_dai_component = { .name = DRV_NAME, .ops = &q6asm_dai_ops, .pcm_new = q6asm_dai_pcm_new, .pcm_free = q6asm_dai_pcm_free, - + .compr_ops = &q6asm_dai_compr_ops, }; static struct snd_soc_dai_driver q6asm_fe_dais[] = { @@ -562,6 +869,41 @@ static struct snd_soc_dai_driver q6asm_fe_dais[] = { Q6ASM_FEDAI_DRIVER(8), }; +static int of_q6asm_parse_dai_data(struct device *dev, + struct q6asm_dai_data *pdata) +{ + static struct snd_soc_dai_driver *dai_drv; + struct snd_soc_pcm_stream empty_stream; + struct device_node *node; + int ret, id, dir; + + memset(&empty_stream, 0, sizeof(empty_stream)); + + for_each_child_of_node(dev->of_node, node) { + ret = of_property_read_u32(node, "reg", &id); + if (ret || id >= MAX_SESSIONS || id < 0) { + dev_err(dev, "valid dai id not found:%d\n", ret); + continue; + } + + dai_drv = &q6asm_fe_dais[id]; + + ret = of_property_read_u32(node, "direction", &dir); + if (ret) + continue; + + if (dir == Q6ASM_DAI_RX) + dai_drv->capture = empty_stream; + else if (dir == Q6ASM_DAI_TX) + dai_drv->playback = empty_stream; + + if (of_property_read_bool(node, "is-compress-dai")) + dai_drv->compress_new = snd_soc_new_compress; + } + + return 0; +} + static int q6asm_dai_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; @@ -582,6 +924,8 @@ static int q6asm_dai_probe(struct platform_device *pdev) dev_set_drvdata(dev, pdata); + of_q6asm_parse_dai_data(dev, pdata); + return devm_snd_soc_register_component(dev, &q6asm_fe_dai_component, q6asm_fe_dais, ARRAY_SIZE(q6asm_fe_dais)); diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 2b2c7233bb5f..4f85cb19a309 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -11,8 +11,8 @@ #include <linux/spinlock.h> #include <linux/kref.h> #include <linux/of.h> -#include <linux/of_platform.h> #include <uapi/sound/asound.h> +#include <uapi/sound/compress_params.h> #include <linux/delay.h> #include <linux/slab.h> #include <linux/mm.h> @@ -37,6 +37,7 @@ #define ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2 0x00010DA3 #define ASM_SESSION_CMD_RUN_V2 0x00010DAA #define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 0x00010DA5 +#define ASM_MEDIA_FMT_MP3 0x00010BE9 #define ASM_DATA_CMD_WRITE_V2 0x00010DAB #define ASM_DATA_CMD_READ_V2 0x00010DAC #define ASM_SESSION_CMD_SUSPEND 0x00010DEC @@ -869,6 +870,9 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format, open->postprocopo_id = ASM_NULL_POPP_TOPOLOGY; switch (format) { + case SND_AUDIOCODEC_MP3: + open->dec_fmt_id = ASM_MEDIA_FMT_MP3; + break; case FORMAT_LINEAR_PCM: open->dec_fmt_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2; break; diff --git a/sound/soc/qcom/qdsp6/q6core.c b/sound/soc/qcom/qdsp6/q6core.c index 06f03a5fe9bd..cdfc8ab6cfc0 100644 --- a/sound/soc/qcom/qdsp6/q6core.c +++ b/sound/soc/qcom/qdsp6/q6core.c @@ -10,7 +10,6 @@ #include <linux/of.h> #include <linux/of_platform.h> #include <linux/jiffies.h> -#include <linux/wait.h> #include <linux/soc/qcom/apr.h> #include "q6core.h" #include "q6dsp-errno.h" @@ -105,12 +104,10 @@ static int q6core_callback(struct apr_device *adev, struct apr_resp_pkt *data) bytes = sizeof(*fwk) + fwk->num_services * sizeof(fwk->svc_api_info[0]); - core->fwk_version = kzalloc(bytes, GFP_ATOMIC); + core->fwk_version = kmemdup(data->payload, bytes, GFP_ATOMIC); if (!core->fwk_version) return -ENOMEM; - memcpy(core->fwk_version, data->payload, bytes); - core->fwk_version_supported = true; core->resp_received = true; @@ -124,12 +121,10 @@ static int q6core_callback(struct apr_device *adev, struct apr_resp_pkt *data) len = sizeof(*v) + v->num_services * sizeof(v->svc_api_info[0]); - core->svc_version = kzalloc(len, GFP_ATOMIC); + core->svc_version = kmemdup(data->payload, len, GFP_ATOMIC); if (!core->svc_version) return -ENOMEM; - memcpy(core->svc_version, data->payload, len); - core->get_version_supported = true; core->resp_received = true; diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index c6b51571be94..ddcd9978cf57 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -453,6 +453,9 @@ static int msm_routing_put_audio_mixer(struct snd_kcontrol *kcontrol, static const struct snd_kcontrol_new hdmi_mixer_controls[] = { Q6ROUTING_RX_MIXERS(HDMI_RX) }; +static const struct snd_kcontrol_new display_port_mixer_controls[] = { + Q6ROUTING_RX_MIXERS(DISPLAY_PORT_RX) }; + static const struct snd_kcontrol_new primary_mi2s_rx_mixer_controls[] = { Q6ROUTING_RX_MIXERS(PRIMARY_MI2S_RX) }; @@ -655,6 +658,10 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { hdmi_mixer_controls, ARRAY_SIZE(hdmi_mixer_controls)), + SND_SOC_DAPM_MIXER("DISPLAY_PORT_RX Audio Mixer", SND_SOC_NOPM, 0, 0, + display_port_mixer_controls, + ARRAY_SIZE(display_port_mixer_controls)), + SND_SOC_DAPM_MIXER("SLIMBUS_0_RX Audio Mixer", SND_SOC_NOPM, 0, 0, slimbus_rx_mixer_controls, ARRAY_SIZE(slimbus_rx_mixer_controls)), @@ -833,6 +840,8 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { static const struct snd_soc_dapm_route intercon[] = { Q6ROUTING_RX_DAPM_ROUTE("HDMI Mixer", "HDMI_RX"), + Q6ROUTING_RX_DAPM_ROUTE("DISPLAY_PORT_RX Audio Mixer", + "DISPLAY_PORT_RX"), Q6ROUTING_RX_DAPM_ROUTE("SLIMBUS_0_RX Audio Mixer", "SLIMBUS_0_RX"), Q6ROUTING_RX_DAPM_ROUTE("SLIMBUS_1_RX Audio Mixer", "SLIMBUS_1_RX"), Q6ROUTING_RX_DAPM_ROUTE("SLIMBUS_2_RX Audio Mixer", "SLIMBUS_2_RX"), @@ -909,6 +918,25 @@ static const struct snd_soc_dapm_route intercon[] = { {"MM_UL6", NULL, "MultiMedia6 Mixer"}, {"MM_UL7", NULL, "MultiMedia7 Mixer"}, {"MM_UL8", NULL, "MultiMedia8 Mixer"}, + + {"MM_DL1", NULL, "MultiMedia1 Playback" }, + {"MM_DL2", NULL, "MultiMedia2 Playback" }, + {"MM_DL3", NULL, "MultiMedia3 Playback" }, + {"MM_DL4", NULL, "MultiMedia4 Playback" }, + {"MM_DL5", NULL, "MultiMedia5 Playback" }, + {"MM_DL6", NULL, "MultiMedia6 Playback" }, + {"MM_DL7", NULL, "MultiMedia7 Playback" }, + {"MM_DL8", NULL, "MultiMedia8 Playback" }, + + {"MultiMedia1 Capture", NULL, "MM_UL1"}, + {"MultiMedia2 Capture", NULL, "MM_UL2"}, + {"MultiMedia3 Capture", NULL, "MM_UL3"}, + {"MultiMedia4 Capture", NULL, "MM_UL4"}, + {"MultiMedia5 Capture", NULL, "MM_UL5"}, + {"MultiMedia6 Capture", NULL, "MM_UL6"}, + {"MultiMedia7 Capture", NULL, "MM_UL7"}, + {"MultiMedia8 Capture", NULL, "MM_UL8"}, + }; static int routing_hw_params(struct snd_pcm_substream *substream, diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 2a781d87ee65..6f66a58e23ca 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -6,18 +6,31 @@ #include <linux/module.h> #include <linux/platform_device.h> #include <linux/of_device.h> +#include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> +#include <sound/jack.h> +#include <sound/soc.h> +#include <uapi/linux/input-event-codes.h> #include "common.h" #include "qdsp6/q6afe.h" +#include "../codecs/rt5663.h" #define DEFAULT_SAMPLE_RATE_48K 48000 #define DEFAULT_MCLK_RATE 24576000 -#define DEFAULT_BCLK_RATE 12288000 +#define TDM_BCLK_RATE 6144000 +#define MI2S_BCLK_RATE 1536000 +#define LEFT_SPK_TDM_TX_MASK 0x30 +#define RIGHT_SPK_TDM_TX_MASK 0xC0 +#define SPK_TDM_RX_MASK 0x03 +#define NUM_TDM_SLOTS 8 struct sdm845_snd_data { + struct snd_soc_jack jack; + bool jack_setup; struct snd_soc_card *card; uint32_t pri_mi2s_clk_count; + uint32_t sec_mi2s_clk_count; uint32_t quat_tdm_clk_count; }; @@ -28,12 +41,12 @@ static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int ret = 0; + int ret = 0, j; int channels, slot_width; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: - slot_width = 32; + slot_width = 16; break; default: dev_err(rtd->dev, "%s: invalid param format 0x%x\n", @@ -75,6 +88,35 @@ static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream, goto end; } } + + for (j = 0; j < rtd->num_codecs; j++) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + + if (!strcmp(codec_dai->component->name_prefix, "Left")) { + ret = snd_soc_dai_set_tdm_slot( + codec_dai, LEFT_SPK_TDM_TX_MASK, + SPK_TDM_RX_MASK, NUM_TDM_SLOTS, + slot_width); + if (ret < 0) { + dev_err(rtd->dev, + "DEV0 TDM slot err:%d\n", ret); + return ret; + } + } + + if (!strcmp(codec_dai->component->name_prefix, "Right")) { + ret = snd_soc_dai_set_tdm_slot( + codec_dai, RIGHT_SPK_TDM_TX_MASK, + SPK_TDM_RX_MASK, NUM_TDM_SLOTS, + slot_width); + if (ret < 0) { + dev_err(rtd->dev, + "DEV1 TDM slot err:%d\n", ret); + return ret; + } + } + } + end: return ret; } @@ -84,9 +126,27 @@ static int sdm845_snd_hw_params(struct snd_pcm_substream *substream, { 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; int ret = 0; switch (cpu_dai->id) { + case PRIMARY_MI2S_RX: + case PRIMARY_MI2S_TX: + /* + * Use ASRC for internal clocks, as PLL rate isn't multiple + * of BCLK. + */ + rt5663_sel_asrc_clk_src( + codec_dai->component, + RT5663_DA_STEREO_FILTER | RT5663_AD_STEREO_FILTER, + RT5663_CLK_SEL_I2S1_ASRC); + ret = snd_soc_dai_set_sysclk( + codec_dai, RT5663_SCLK_S_MCLK, DEFAULT_MCLK_RATE, + SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(rtd->dev, + "snd_soc_dai_set_sysclk err = %d\n", ret); + break; case QUATERNARY_TDM_RX_0: case QUATERNARY_TDM_TX_0: ret = sdm845_tdm_snd_hw_params(substream, params); @@ -98,24 +158,100 @@ static int sdm845_snd_hw_params(struct snd_pcm_substream *substream, return ret; } +static void sdm845_jack_free(struct snd_jack *jack) +{ + struct snd_soc_component *component = jack->private_data; + + snd_soc_component_set_jack(component, NULL, NULL); +} + +static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *component; + struct snd_soc_card *card = rtd->card; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct sdm845_snd_data *pdata = snd_soc_card_get_drvdata(card); + struct snd_jack *jack; + int rval; + + if (!pdata->jack_setup) { + rval = snd_soc_card_jack_new(card, "Headset Jack", + SND_JACK_HEADSET | + SND_JACK_HEADPHONE | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &pdata->jack, NULL, 0); + + if (rval < 0) { + dev_err(card->dev, "Unable to add Headphone Jack\n"); + return rval; + } + + jack = pdata->jack.jack; + + snd_jack_set_key(jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + pdata->jack_setup = true; + } + + switch (cpu_dai->id) { + case PRIMARY_MI2S_RX: + jack = pdata->jack.jack; + component = codec_dai->component; + + jack->private_data = component; + jack->private_free = sdm845_jack_free; + rval = snd_soc_component_set_jack(component, + &pdata->jack, NULL); + if (rval != 0 && rval != -ENOTSUPP) { + dev_warn(card->dev, "Failed to set jack: %d\n", rval); + return rval; + } + break; + default: + break; + } + + return 0; +} + + static int sdm845_snd_startup(struct snd_pcm_substream *substream) { unsigned int fmt = SND_SOC_DAIFMT_CBS_CFS; + unsigned int codec_dai_fmt = SND_SOC_DAIFMT_CBS_CFS; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_card *card = rtd->card; struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card); struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int j; + int ret; switch (cpu_dai->id) { case PRIMARY_MI2S_RX: case PRIMARY_MI2S_TX: + codec_dai_fmt |= SND_SOC_DAIFMT_NB_NF; if (++(data->pri_mi2s_clk_count) == 1) { snd_soc_dai_set_sysclk(cpu_dai, Q6AFE_LPASS_CLK_ID_MCLK_1, DEFAULT_MCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); snd_soc_dai_set_sysclk(cpu_dai, Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT, - DEFAULT_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + MI2S_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + } + snd_soc_dai_set_fmt(cpu_dai, fmt); + snd_soc_dai_set_fmt(codec_dai, codec_dai_fmt); + break; + + case SECONDARY_MI2S_TX: + if (++(data->sec_mi2s_clk_count) == 1) { + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_SEC_MI2S_IBIT, + MI2S_BCLK_RATE, SNDRV_PCM_STREAM_CAPTURE); } snd_soc_dai_set_fmt(cpu_dai, fmt); break; @@ -125,7 +261,35 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream) if (++(data->quat_tdm_clk_count) == 1) { snd_soc_dai_set_sysclk(cpu_dai, Q6AFE_LPASS_CLK_ID_QUAD_TDM_IBIT, - DEFAULT_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + TDM_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + } + + codec_dai_fmt |= SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_DSP_B; + + for (j = 0; j < rtd->num_codecs; j++) { + codec_dai = rtd->codec_dais[j]; + + if (!strcmp(codec_dai->component->name_prefix, + "Left")) { + ret = snd_soc_dai_set_fmt( + codec_dai, codec_dai_fmt); + if (ret < 0) { + dev_err(rtd->dev, + "Left TDM fmt err:%d\n", ret); + return ret; + } + } + + if (!strcmp(codec_dai->component->name_prefix, + "Right")) { + ret = snd_soc_dai_set_fmt( + codec_dai, codec_dai_fmt); + if (ret < 0) { + dev_err(rtd->dev, + "Right TDM slot err:%d\n", ret); + return ret; + } + } } break; @@ -156,6 +320,14 @@ static void sdm845_snd_shutdown(struct snd_pcm_substream *substream) }; break; + case SECONDARY_MI2S_TX: + if (--(data->sec_mi2s_clk_count) == 0) { + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_SEC_MI2S_IBIT, + 0, SNDRV_PCM_STREAM_CAPTURE); + } + break; + case QUATERNARY_TDM_RX_0: case QUATERNARY_TDM_TX_0: if (--(data->quat_tdm_clk_count) == 0) { @@ -171,7 +343,7 @@ static void sdm845_snd_shutdown(struct snd_pcm_substream *substream) } } -static struct snd_soc_ops sdm845_be_ops = { +static const struct snd_soc_ops sdm845_be_ops = { .hw_params = sdm845_snd_hw_params, .startup = sdm845_snd_startup, .shutdown = sdm845_snd_shutdown, @@ -193,17 +365,25 @@ static int sdm845_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static void sdm845_add_be_ops(struct snd_soc_card *card) +static const struct snd_soc_dapm_widget sdm845_snd_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), +}; + +static void sdm845_add_ops(struct snd_soc_card *card) { - struct snd_soc_dai_link *link = card->dai_link; - int i, num_links = card->num_links; + struct snd_soc_dai_link *link; + int i; - for (i = 0; i < num_links; i++) { + for_each_card_prelinks(card, i, link) { if (link->no_pcm == 1) { link->ops = &sdm845_be_ops; link->be_hw_params_fixup = sdm845_be_hw_params_fixup; } - link++; + link->init = sdm845_dai_init; } } @@ -225,6 +405,8 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) goto data_alloc_fail; } + card->dapm_widgets = sdm845_snd_widgets; + card->num_dapm_widgets = ARRAY_SIZE(sdm845_snd_widgets); card->dev = dev; dev_set_drvdata(dev, card); ret = qcom_snd_parse_of(card); @@ -236,7 +418,7 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) data->card = card; snd_soc_card_set_drvdata(card, data); - sdm845_add_be_ops(card); + sdm845_add_ops(card); ret = snd_soc_register_card(card); if (ret) { dev_err(dev, "Sound card registration failed\n"); diff --git a/sound/soc/rockchip/rk3288_hdmi_analog.c b/sound/soc/rockchip/rk3288_hdmi_analog.c index 929b3fe289b0..a472d5eb2950 100644 --- a/sound/soc/rockchip/rk3288_hdmi_analog.c +++ b/sound/soc/rockchip/rk3288_hdmi_analog.c @@ -286,7 +286,6 @@ static struct platform_driver rockchip_sound_driver = { .probe = snd_rk_mc_probe, .driver = { .name = DRV_NAME, - .owner = THIS_MODULE, .pm = &snd_soc_pm_ops, .of_match_table = rockchip_sound_of_match, }, diff --git a/sound/soc/rockchip/rockchip_pcm.c b/sound/soc/rockchip/rockchip_pcm.c index f77538319221..4ac78d7a4b2d 100644 --- a/sound/soc/rockchip/rockchip_pcm.c +++ b/sound/soc/rockchip/rockchip_pcm.c @@ -21,7 +21,8 @@ static const struct snd_pcm_hardware snd_rockchip_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME, + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_INTERLEAVED, .period_bytes_min = 32, .period_bytes_max = 8192, .periods_min = 1, @@ -32,6 +33,7 @@ static const struct snd_pcm_hardware snd_rockchip_hardware = { static const struct snd_dmaengine_pcm_config rk_dmaengine_pcm_config = { .pcm_hardware = &snd_rockchip_hardware, + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, .prealloc_buffer_size = 32 * 1024, }; diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index d6c62aa13041..d4bde4834ce5 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -604,6 +604,7 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct i2s_dai *i2s = to_info(dai); + struct i2s_dai *other = get_other_dai(i2s); int lrp_shift, sdf_shift, sdf_mask, lrp_rlow, mod_slave; u32 mod, tmp = 0; unsigned long flags; @@ -661,7 +662,8 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, * CLK_I2S_RCLK_SRC clock is not exposed so we ensure any * clock configuration assigned in DT is not overwritten. */ - if (i2s->rclk_srcrate == 0 && i2s->clk_data.clks == NULL) + if (i2s->rclk_srcrate == 0 && i2s->clk_data.clks == NULL && + other->clk_data.clks == NULL) i2s_set_sysclk(dai, SAMSUNG_I2S_RCLKSRC_0, 0, SND_SOC_CLOCK_IN); break; @@ -699,7 +701,9 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct i2s_dai *i2s = to_info(dai); + struct i2s_dai *other = get_other_dai(i2s); u32 mod, mask = 0, val = 0; + struct clk *rclksrc; unsigned long flags; WARN_ON(!pm_runtime_active(dai->dev)); @@ -782,6 +786,13 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, i2s->frmclk = params_rate(params); + rclksrc = i2s->clk_table[CLK_I2S_RCLK_SRC]; + if (!rclksrc || IS_ERR(rclksrc)) + rclksrc = other->clk_table[CLK_I2S_RCLK_SRC]; + + if (rclksrc && !IS_ERR(rclksrc)) + i2s->rclk_srcrate = clk_get_rate(rclksrc); + return 0; } @@ -886,11 +897,6 @@ static int config_setup(struct i2s_dai *i2s) return 0; if (!(i2s->quirks & QUIRK_NO_MUXPSR)) { - struct clk *rclksrc = i2s->clk_table[CLK_I2S_RCLK_SRC]; - - if (rclksrc && !IS_ERR(rclksrc)) - i2s->rclk_srcrate = clk_get_rate(rclksrc); - psr = i2s->rclk_srcrate / i2s->frmclk / rfs; writel(((psr - 1) << 8) | PSR_PSREN, i2s->addr + I2SPSR); dev_dbg(&i2s->pdev->dev, diff --git a/sound/soc/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c index 43332c32d7e9..dc93941e01c3 100644 --- a/sound/soc/samsung/tm2_wm5110.c +++ b/sound/soc/samsung/tm2_wm5110.c @@ -491,6 +491,7 @@ static int tm2_probe(struct platform_device *pdev) struct snd_soc_card *card = &tm2_card; struct tm2_machine_priv *priv; struct of_phandle_args args; + struct snd_soc_dai_link *dai_link; int num_codecs, ret, i; priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); @@ -558,18 +559,18 @@ static int tm2_probe(struct platform_device *pdev) } /* Initialize WM5110 - I2S and HDMI - I2S1 DAI links */ - for (i = 0; i < card->num_links; i++) { + for_each_card_prelinks(card, i, dai_link) { unsigned int dai_index = 0; /* WM5110 */ - card->dai_link[i].cpu_name = NULL; - card->dai_link[i].platform_name = NULL; + dai_link->cpu_name = NULL; + dai_link->platform_name = NULL; if (num_codecs > 1 && i == card->num_links - 1) dai_index = 1; /* HDMI */ - card->dai_link[i].codec_of_node = codec_dai_node[dai_index]; - card->dai_link[i].cpu_of_node = cpu_dai_node[dai_index]; - card->dai_link[i].platform_of_node = cpu_dai_node[dai_index]; + dai_link->codec_of_node = codec_dai_node[dai_index]; + dai_link->cpu_of_node = cpu_dai_node[dai_index]; + dai_link->platform_of_node = cpu_dai_node[dai_index]; } if (num_codecs > 1) { diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 922fb6aa3ed1..5aee11c94f2a 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -202,7 +202,7 @@ static int camelot_prepare(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id]; - pr_debug("PCM data: addr 0x%08ulx len %d\n", + pr_debug("PCM data: addr 0x%08lx len %d\n", (u32)runtime->dma_addr, runtime->dma_bytes); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index c2b496398e6b..17622ceb98c0 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -319,13 +319,12 @@ static int hac_soc_platform_probe(struct platform_device *pdev) if (ret != 0) return ret; - return snd_soc_register_component(&pdev->dev, &sh4_hac_component, + return devm_snd_soc_register_component(&pdev->dev, &sh4_hac_component, sh4_hac_dai, ARRAY_SIZE(sh4_hac_dai)); } static int hac_soc_platform_remove(struct platform_device *pdev) { - snd_soc_unregister_component(&pdev->dev); snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 051f96405346..e821ccc70f47 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -249,28 +249,8 @@ int rsnd_adg_set_src_timesel_gen2(struct rsnd_mod *src_mod, out = out << shift; mask = 0x0f1f << shift; - switch (id / 2) { - case 0: - rsnd_mod_bset(adg_mod, SRCIN_TIMSEL0, mask, in); - rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL0, mask, out); - break; - case 1: - rsnd_mod_bset(adg_mod, SRCIN_TIMSEL1, mask, in); - rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL1, mask, out); - break; - case 2: - rsnd_mod_bset(adg_mod, SRCIN_TIMSEL2, mask, in); - rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL2, mask, out); - break; - case 3: - rsnd_mod_bset(adg_mod, SRCIN_TIMSEL3, mask, in); - rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL3, mask, out); - break; - case 4: - rsnd_mod_bset(adg_mod, SRCIN_TIMSEL4, mask, in); - rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL4, mask, out); - break; - } + rsnd_mod_bset(adg_mod, SRCIN_TIMSEL(id / 2), mask, in); + rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL(id / 2), mask, out); if (en) rsnd_mod_bset(adg_mod, DIV_EN, en, en); @@ -299,17 +279,7 @@ static void rsnd_adg_set_ssi_clk(struct rsnd_mod *ssi_mod, u32 val) if (id == 8) return; - switch (id / 4) { - case 0: - rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL0, mask, val); - break; - case 1: - rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL1, mask, val); - break; - case 2: - rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL2, mask, val); - break; - } + rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL(id / 4), mask, val); dev_dbg(dev, "AUDIO_CLK_SEL is 0x%x\n", val); } @@ -582,7 +552,7 @@ static void rsnd_adg_clk_dbg_info(struct rsnd_priv *priv, struct rsnd_adg *adg) int i; for_each_rsnd_clk(clk, adg, i) - dev_dbg(dev, "%s : %p : %ld\n", + dev_dbg(dev, "%s : %pa : %ld\n", clk_name[i], clk, clk_get_rate(clk)); dev_dbg(dev, "BRGCKR = 0x%08x, BRRA/BRRB = 0x%x/0x%x\n", @@ -595,7 +565,7 @@ static void rsnd_adg_clk_dbg_info(struct rsnd_priv *priv, struct rsnd_adg *adg) * by BRGCKR::BRGCKR_31 */ for_each_rsnd_clkout(clk, adg, i) - dev_dbg(dev, "clkout %d : %p : %ld\n", i, + dev_dbg(dev, "clkout %d : %pa : %ld\n", i, clk, clk_get_rate(clk)); } #else @@ -613,7 +583,7 @@ int rsnd_adg_probe(struct rsnd_priv *priv) return -ENOMEM; ret = rsnd_mod_init(priv, &adg->mod, &adg_ops, - NULL, NULL, 0, 0); + NULL, 0, 0); if (ret) return ret; diff --git a/sound/soc/sh/rcar/cmd.c b/sound/soc/sh/rcar/cmd.c index cc191cd5fb82..e6bb6a9a0684 100644 --- a/sound/soc/sh/rcar/cmd.c +++ b/sound/soc/sh/rcar/cmd.c @@ -116,10 +116,11 @@ static int rsnd_cmd_stop(struct rsnd_mod *mod, } static struct rsnd_mod_ops rsnd_cmd_ops = { - .name = CMD_NAME, - .init = rsnd_cmd_init, - .start = rsnd_cmd_start, - .stop = rsnd_cmd_stop, + .name = CMD_NAME, + .init = rsnd_cmd_init, + .start = rsnd_cmd_start, + .stop = rsnd_cmd_stop, + .get_status = rsnd_mod_get_status, }; static struct rsnd_mod *rsnd_cmd_mod_get(struct rsnd_priv *priv, int id) @@ -162,7 +163,7 @@ int rsnd_cmd_probe(struct rsnd_priv *priv) for_each_rsnd_cmd(cmd, priv, i) { ret = rsnd_mod_init(priv, rsnd_mod_get(cmd), &rsnd_cmd_ops, NULL, - rsnd_mod_get_status, RSND_MOD_CMD, i); + RSND_MOD_CMD, i); if (ret) return ret; } diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index d23c2bbff0cf..e819e965e1db 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -102,7 +102,9 @@ #include "rsnd.h" #define RSND_RATES SNDRV_PCM_RATE_8000_192000 -#define RSND_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) +#define RSND_FMTS (SNDRV_PCM_FMTBIT_S8 |\ + SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE) static const struct of_device_id rsnd_of_match[] = { { .compatible = "renesas,rcar_sound-gen1", .data = (void *)RSND_GEN1 }, @@ -121,8 +123,8 @@ void rsnd_mod_make_sure(struct rsnd_mod *mod, enum rsnd_mod_type type) struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct device *dev = rsnd_priv_to_dev(priv); - dev_warn(dev, "%s[%d] is not your expected module\n", - rsnd_mod_name(mod), rsnd_mod_id(mod)); + dev_warn(dev, "%s is not your expected module\n", + rsnd_mod_name(mod)); } } @@ -135,20 +137,69 @@ struct dma_chan *rsnd_mod_dma_req(struct rsnd_dai_stream *io, return mod->ops->dma_req(io, mod); } -u32 *rsnd_mod_get_status(struct rsnd_dai_stream *io, - struct rsnd_mod *mod, +#define MOD_NAME_NUM 5 +#define MOD_NAME_SIZE 16 +char *rsnd_mod_name(struct rsnd_mod *mod) +{ + static char names[MOD_NAME_NUM][MOD_NAME_SIZE]; + static int num; + char *name = names[num]; + + num++; + if (num >= MOD_NAME_NUM) + num = 0; + + /* + * Let's use same char to avoid pointlessness memory + * Thus, rsnd_mod_name() should be used immediately + * Don't keep pointer + */ + if ((mod)->ops->id_sub) { + snprintf(name, MOD_NAME_SIZE, "%s[%d%d]", + mod->ops->name, + rsnd_mod_id(mod), + rsnd_mod_id_sub(mod)); + } else { + snprintf(name, MOD_NAME_SIZE, "%s[%d]", + mod->ops->name, + rsnd_mod_id(mod)); + } + + return name; +} + +u32 *rsnd_mod_get_status(struct rsnd_mod *mod, + struct rsnd_dai_stream *io, enum rsnd_mod_type type) { return &mod->status; } +int rsnd_mod_id_raw(struct rsnd_mod *mod) +{ + return mod->id; +} + +int rsnd_mod_id(struct rsnd_mod *mod) +{ + if ((mod)->ops->id) + return (mod)->ops->id(mod); + + return rsnd_mod_id_raw(mod); +} + +int rsnd_mod_id_sub(struct rsnd_mod *mod) +{ + if ((mod)->ops->id_sub) + return (mod)->ops->id_sub(mod); + + return 0; +} + int rsnd_mod_init(struct rsnd_priv *priv, struct rsnd_mod *mod, struct rsnd_mod_ops *ops, struct clk *clk, - u32* (*get_status)(struct rsnd_dai_stream *io, - struct rsnd_mod *mod, - enum rsnd_mod_type type), enum rsnd_mod_type type, int id) { @@ -162,7 +213,6 @@ int rsnd_mod_init(struct rsnd_priv *priv, mod->type = type; mod->clk = clk; mod->priv = priv; - mod->get_status = get_status; return ret; } @@ -226,7 +276,20 @@ int rsnd_runtime_channel_after_ctu_with_params(struct rsnd_dai_stream *io, struct rsnd_mod *ctu_mod = rsnd_io_to_mod_ctu(io); if (ctu_mod) { - u32 converted_chan = rsnd_ctu_converted_channel(ctu_mod); + u32 converted_chan = rsnd_io_converted_chan(io); + + /* + * !! Note !! + * + * converted_chan will be used for CTU, + * or TDM Split mode. + * User shouldn't use CTU with TDM Split mode. + */ + if (rsnd_runtime_is_tdm_split(io)) { + struct device *dev = rsnd_priv_to_dev(rsnd_io_to_priv(io)); + + dev_err(dev, "CTU and TDM Split should be used\n"); + } if (converted_chan) return converted_chan; @@ -244,7 +307,7 @@ int rsnd_runtime_channel_for_ssi_with_params(struct rsnd_dai_stream *io, rsnd_runtime_channel_original_with_params(io, params); /* Use Multi SSI */ - if (rsnd_runtime_is_ssi_multi(io)) + if (rsnd_runtime_is_multi_ssi(io)) chan /= rsnd_rdai_ssi_lane_get(rdai); /* TDM Extend Mode needs 8ch */ @@ -254,7 +317,7 @@ int rsnd_runtime_channel_for_ssi_with_params(struct rsnd_dai_stream *io, return chan; } -int rsnd_runtime_is_ssi_multi(struct rsnd_dai_stream *io) +int rsnd_runtime_is_multi_ssi(struct rsnd_dai_stream *io) { struct rsnd_dai *rdai = rsnd_io_to_rdai(io); int lane = rsnd_rdai_ssi_lane_get(rdai); @@ -265,11 +328,16 @@ int rsnd_runtime_is_ssi_multi(struct rsnd_dai_stream *io) return (chan > 2) && (lane > 1); } -int rsnd_runtime_is_ssi_tdm(struct rsnd_dai_stream *io) +int rsnd_runtime_is_tdm(struct rsnd_dai_stream *io) { return rsnd_runtime_channel_for_ssi(io) >= 6; } +int rsnd_runtime_is_tdm_split(struct rsnd_dai_stream *io) +{ + return !!rsnd_flags_has(io, RSND_STREAM_TDM_SPLIT); +} + /* * ADINR function */ @@ -280,6 +348,8 @@ u32 rsnd_get_adinr_bit(struct rsnd_mod *mod, struct rsnd_dai_stream *io) struct device *dev = rsnd_priv_to_dev(priv); switch (snd_pcm_format_width(runtime->format)) { + case 8: + return 16 << 16; case 16: return 8 << 16; case 24: @@ -331,7 +401,7 @@ u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io) target = cmd ? cmd : ssiu; } - /* Non target mod or 24bit data needs normal DALIGN */ + /* Non target mod or non 16bit needs normal DALIGN */ if ((snd_pcm_format_width(runtime->format) != 16) || (mod != target)) return 0x76543210; @@ -367,7 +437,7 @@ u32 rsnd_get_busif_shift(struct rsnd_dai_stream *io, struct rsnd_mod *mod) * HW 24bit data is located as 0x******00 * */ - if (snd_pcm_format_width(runtime->format) == 16) + if (snd_pcm_format_width(runtime->format) != 24) return 0; for (i = 0; i < ARRAY_SIZE(playback_mods); i++) { @@ -468,20 +538,19 @@ static int rsnd_status_update(u32 *status, enum rsnd_mod_type *types = rsnd_mod_sequence[is_play]; \ for_each_rsnd_mod_arrays(i, mod, io, types, RSND_MOD_MAX) { \ int tmp = 0; \ - u32 *status = mod->get_status(io, mod, types[i]); \ + u32 *status = mod->ops->get_status(mod, io, types[i]); \ int func_call = rsnd_status_update(status, \ __rsnd_mod_shift_##fn, \ __rsnd_mod_add_##fn, \ __rsnd_mod_call_##fn); \ - rsnd_dbg_dai_call(dev, "%s[%d]\t0x%08x %s\n", \ - rsnd_mod_name(mod), rsnd_mod_id(mod), *status, \ + rsnd_dbg_dai_call(dev, "%s\t0x%08x %s\n", \ + rsnd_mod_name(mod), *status, \ (func_call && (mod)->ops->fn) ? #fn : ""); \ if (func_call && (mod)->ops->fn) \ tmp = (mod)->ops->fn(mod, io, param); \ if (tmp && (tmp != -EPROBE_DEFER)) \ - dev_err(dev, "%s[%d] : %s error %d\n", \ - rsnd_mod_name(mod), rsnd_mod_id(mod), \ - #fn, tmp); \ + dev_err(dev, "%s : %s error %d\n", \ + rsnd_mod_name(mod), #fn, tmp); \ ret |= tmp; \ } \ ret; \ @@ -508,8 +577,8 @@ int rsnd_dai_connect(struct rsnd_mod *mod, io->mod[type] = mod; - dev_dbg(dev, "%s[%d] is connected to io (%s)\n", - rsnd_mod_name(mod), rsnd_mod_id(mod), + dev_dbg(dev, "%s is connected to io (%s)\n", + rsnd_mod_name(mod), rsnd_io_is_play(io) ? "Playback" : "Capture"); return 0; @@ -540,6 +609,14 @@ int rsnd_rdai_ssi_lane_ctrl(struct rsnd_dai *rdai, return rdai->ssi_lane; } +int rsnd_rdai_width_ctrl(struct rsnd_dai *rdai, int width) +{ + if (width > 0) + rdai->chan_width = width; + + return rdai->chan_width; +} + struct rsnd_dai *rsnd_rdai_get(struct rsnd_priv *priv, int id) { if ((id < 0) || (id >= rsnd_rdai_nr(priv))) @@ -681,6 +758,7 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) rdai->frm_clk_inv = 0; break; case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_DSP_B: rdai->sys_delay = 1; rdai->data_alignment = 0; rdai->frm_clk_inv = 1; @@ -690,6 +768,11 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) rdai->data_alignment = 1; rdai->frm_clk_inv = 1; break; + case SND_SOC_DAIFMT_DSP_A: + rdai->sys_delay = 0; + rdai->data_alignment = 0; + rdai->frm_clk_inv = 1; + break; } /* set clock inversion */ @@ -720,13 +803,25 @@ static int rsnd_soc_set_dai_tdm_slot(struct snd_soc_dai *dai, struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); struct device *dev = rsnd_priv_to_dev(priv); + switch (slot_width) { + case 16: + case 24: + case 32: + break; + default: + /* use default */ + slot_width = 32; + } + switch (slots) { case 2: + /* TDM Split Mode */ case 6: case 8: /* TDM Extend Mode */ rsnd_rdai_channels_set(rdai, slots); rsnd_rdai_ssi_lane_set(rdai, 1); + rsnd_rdai_width_set(rdai, slot_width); break; default: dev_err(dev, "unsupported TDM slots (%d)\n", slots); @@ -755,7 +850,7 @@ static unsigned int rsnd_soc_hw_rate_list[] = { 192000, }; -static int rsnd_soc_hw_rule(struct rsnd_priv *priv, +static int rsnd_soc_hw_rule(struct rsnd_dai *rdai, unsigned int *list, int list_num, struct snd_interval *baseline, struct snd_interval *iv) { @@ -772,14 +867,14 @@ static int rsnd_soc_hw_rule(struct rsnd_priv *priv, if (!snd_interval_test(iv, list[i])) continue; - rate = rsnd_ssi_clk_query(priv, + rate = rsnd_ssi_clk_query(rdai, baseline->min, list[i], NULL); if (rate > 0) { p.min = min(p.min, list[i]); p.max = max(p.max, list[i]); } - rate = rsnd_ssi_clk_query(priv, + rate = rsnd_ssi_clk_query(rdai, baseline->max, list[i], NULL); if (rate > 0) { p.min = min(p.min, list[i]); @@ -790,17 +885,14 @@ static int rsnd_soc_hw_rule(struct rsnd_priv *priv, return snd_interval_refine(iv, &p); } -static int __rsnd_soc_hw_rule_rate(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule, - int is_play) +static int rsnd_soc_hw_rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) { struct snd_interval *ic_ = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_interval *ir = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); struct snd_interval ic; - struct snd_soc_dai *dai = rule->private; - struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); - struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); - struct rsnd_dai_stream *io = is_play ? &rdai->playback : &rdai->capture; + struct rsnd_dai_stream *io = rule->private; + struct rsnd_dai *rdai = rsnd_io_to_rdai(io); /* * possible sampling rate limitation is same as @@ -811,34 +903,19 @@ static int __rsnd_soc_hw_rule_rate(struct snd_pcm_hw_params *params, ic.min = ic.max = rsnd_runtime_channel_for_ssi_with_params(io, params); - return rsnd_soc_hw_rule(priv, rsnd_soc_hw_rate_list, + return rsnd_soc_hw_rule(rdai, rsnd_soc_hw_rate_list, ARRAY_SIZE(rsnd_soc_hw_rate_list), &ic, ir); } -static int rsnd_soc_hw_rule_rate_playback(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - return __rsnd_soc_hw_rule_rate(params, rule, 1); -} - -static int rsnd_soc_hw_rule_rate_capture(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - return __rsnd_soc_hw_rule_rate(params, rule, 0); -} - -static int __rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule, - int is_play) +static int rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) { struct snd_interval *ic_ = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_interval *ir = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); struct snd_interval ic; - struct snd_soc_dai *dai = rule->private; - struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); - struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); - struct rsnd_dai_stream *io = is_play ? &rdai->playback : &rdai->capture; + struct rsnd_dai_stream *io = rule->private; + struct rsnd_dai *rdai = rsnd_io_to_rdai(io); /* * possible sampling rate limitation is same as @@ -849,23 +926,11 @@ static int __rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params, ic.min = ic.max = rsnd_runtime_channel_for_ssi_with_params(io, params); - return rsnd_soc_hw_rule(priv, rsnd_soc_hw_channels_list, + return rsnd_soc_hw_rule(rdai, rsnd_soc_hw_channels_list, ARRAY_SIZE(rsnd_soc_hw_channels_list), ir, &ic); } -static int rsnd_soc_hw_rule_channels_playback(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - return __rsnd_soc_hw_rule_channels(params, rule, 1); -} - -static int rsnd_soc_hw_rule_channels_capture(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - return __rsnd_soc_hw_rule_channels(params, rule, 0); -} - static const struct snd_pcm_hardware rsnd_pcm_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | @@ -882,12 +947,10 @@ static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); - struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream); struct snd_pcm_hw_constraint_list *constraint = &rdai->constraint; struct snd_pcm_runtime *runtime = substream->runtime; unsigned int max_channels = rsnd_rdai_channels_get(rdai); - int ret; int i; rsnd_dai_stream_init(io, substream); @@ -922,25 +985,16 @@ static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream, int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - is_play ? rsnd_soc_hw_rule_rate_playback : - rsnd_soc_hw_rule_rate_capture, - dai, + rsnd_soc_hw_rule_rate, + is_play ? &rdai->playback : &rdai->capture, SNDRV_PCM_HW_PARAM_CHANNELS, -1); snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - is_play ? rsnd_soc_hw_rule_channels_playback : - rsnd_soc_hw_rule_channels_capture, - dai, + rsnd_soc_hw_rule_channels, + is_play ? &rdai->playback : &rdai->capture, SNDRV_PCM_HW_PARAM_RATE, -1); } - /* - * call rsnd_dai_call without spinlock - */ - ret = rsnd_dai_call(nolock_start, io, priv); - if (ret < 0) - rsnd_dai_call(nolock_stop, io, priv); - - return ret; + return 0; } static void rsnd_soc_dai_shutdown(struct snd_pcm_substream *substream, @@ -953,7 +1007,7 @@ static void rsnd_soc_dai_shutdown(struct snd_pcm_substream *substream, /* * call rsnd_dai_call without spinlock */ - rsnd_dai_call(nolock_stop, io, priv); + rsnd_dai_call(cleanup, io, priv); rsnd_dai_stream_quit(io); } @@ -977,6 +1031,82 @@ static const struct snd_soc_dai_ops rsnd_soc_dai_ops = { .prepare = rsnd_soc_dai_prepare, }; +static void rsnd_parse_connect_simple(struct rsnd_priv *priv, + struct device_node *dai_np, + int dai_i, int is_play) +{ + struct device *dev = rsnd_priv_to_dev(priv); + struct rsnd_dai *rdai = rsnd_rdai_get(priv, dai_i); + struct rsnd_dai_stream *io = is_play ? + &rdai->playback : + &rdai->capture; + struct device_node *ssiu_np = rsnd_ssiu_of_node(priv); + struct device_node *np; + int i, j; + + if (!ssiu_np) + return; + + if (!rsnd_io_to_mod_ssi(io)) + return; + + /* + * This driver assumes that it is TDM Split mode + * if it includes ssiu node + */ + for (i = 0;; i++) { + struct device_node *node = is_play ? + of_parse_phandle(dai_np, "playback", i) : + of_parse_phandle(dai_np, "capture", i); + + if (!node) + break; + + j = 0; + for_each_child_of_node(ssiu_np, np) { + if (np == node) { + rsnd_flags_set(io, RSND_STREAM_TDM_SPLIT); + dev_dbg(dev, "%s is part of TDM Split\n", io->name); + } + j++; + } + + } +} + +static void rsnd_parse_connect_graph(struct rsnd_priv *priv, + struct rsnd_dai_stream *io, + struct device_node *endpoint) +{ + struct device *dev = rsnd_priv_to_dev(priv); + struct device_node *remote_port = of_graph_get_remote_port(endpoint); + struct device_node *remote_node = of_graph_get_remote_port_parent(endpoint); + + if (!rsnd_io_to_mod_ssi(io)) + return; + + /* HDMI0 */ + if (strstr(remote_node->full_name, "hdmi@fead0000")) { + rsnd_flags_set(io, RSND_STREAM_HDMI0); + dev_dbg(dev, "%s connected to HDMI0\n", io->name); + } + + /* HDMI1 */ + if (strstr(remote_node->full_name, "hdmi@feae0000")) { + rsnd_flags_set(io, RSND_STREAM_HDMI1); + dev_dbg(dev, "%s connected to HDMI1\n", io->name); + } + + /* + * This driver assumes that it is TDM Split mode + * if remote node has multi endpoint + */ + if (of_get_child_count(remote_port) > 1) { + rsnd_flags_set(io, RSND_STREAM_TDM_SPLIT); + dev_dbg(dev, "%s is part of TDM Split\n", io->name); + } +} + void rsnd_parse_connect_common(struct rsnd_dai *rdai, struct rsnd_mod* (*mod_get)(struct rsnd_priv *priv, int id), struct device_node *node, @@ -1063,26 +1193,27 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv, drv->name = rdai->name; drv->ops = &rsnd_soc_dai_ops; - snprintf(rdai->playback.name, RSND_DAI_NAME_SIZE, + snprintf(io_playback->name, RSND_DAI_NAME_SIZE, "DAI%d Playback", dai_i); drv->playback.rates = RSND_RATES; drv->playback.formats = RSND_FMTS; drv->playback.channels_min = 2; drv->playback.channels_max = 8; - drv->playback.stream_name = rdai->playback.name; + drv->playback.stream_name = io_playback->name; - snprintf(rdai->capture.name, RSND_DAI_NAME_SIZE, + snprintf(io_capture->name, RSND_DAI_NAME_SIZE, "DAI%d Capture", dai_i); drv->capture.rates = RSND_RATES; drv->capture.formats = RSND_FMTS; drv->capture.channels_min = 2; drv->capture.channels_max = 8; - drv->capture.stream_name = rdai->capture.name; + drv->capture.stream_name = io_capture->name; - rdai->playback.rdai = rdai; - rdai->capture.rdai = rdai; + io_playback->rdai = rdai; + io_capture->rdai = rdai; rsnd_rdai_channels_set(rdai, 2); /* default 2ch */ rsnd_rdai_ssi_lane_set(rdai, 1); /* default 1lane */ + rsnd_rdai_width_set(rdai, 32); /* default 32bit width */ for (io_i = 0;; io_i++) { playback = of_parse_phandle(dai_np, "playback", io_i); @@ -1092,6 +1223,7 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv, break; rsnd_parse_connect_ssi(rdai, playback, capture); + rsnd_parse_connect_ssiu(rdai, playback, capture); rsnd_parse_connect_src(rdai, playback, capture); rsnd_parse_connect_ctu(rdai, playback, capture); rsnd_parse_connect_mix(rdai, playback, capture); @@ -1148,12 +1280,23 @@ static int rsnd_dai_probe(struct rsnd_priv *priv) if (is_graph) { for_each_endpoint_of_node(dai_node, dai_np) { __rsnd_dai_probe(priv, dai_np, dai_i); - rsnd_ssi_parse_hdmi_connection(priv, dai_np, dai_i); + if (rsnd_is_gen3(priv)) { + struct rsnd_dai *rdai = rsnd_rdai_get(priv, dai_i); + + rsnd_parse_connect_graph(priv, &rdai->playback, dai_np); + rsnd_parse_connect_graph(priv, &rdai->capture, dai_np); + } dai_i++; } } else { - for_each_child_of_node(dai_node, dai_np) - __rsnd_dai_probe(priv, dai_np, dai_i++); + for_each_child_of_node(dai_node, dai_np) { + __rsnd_dai_probe(priv, dai_np, dai_i); + if (rsnd_is_gen3(priv)) { + rsnd_parse_connect_simple(priv, dai_np, dai_i, 1); + rsnd_parse_connect_simple(priv, dai_np, dai_i, 0); + } + dai_i++; + } } return 0; @@ -1168,8 +1311,40 @@ static int rsnd_hw_params(struct snd_pcm_substream *substream, 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); + struct snd_soc_pcm_runtime *fe = substream->private_data; int ret; + /* + * rsnd assumes that it might be used under DPCM if user want to use + * channel / rate convert. Then, rsnd should be FE. + * And then, this function will be called *after* BE settings. + * this means, each BE already has fixuped hw_params. + * see + * dpcm_fe_dai_hw_params() + * dpcm_be_dai_hw_params() + */ + io->converted_rate = 0; + io->converted_chan = 0; + if (fe->dai_link->dynamic) { + struct rsnd_priv *priv = rsnd_io_to_priv(io); + struct device *dev = rsnd_priv_to_dev(priv); + struct snd_soc_dpcm *dpcm; + struct snd_pcm_hw_params *be_params; + int stream = substream->stream; + + for_each_dpcm_be(fe, stream, dpcm) { + be_params = &dpcm->hw_params; + if (params_channels(hw_params) != params_channels(be_params)) + io->converted_chan = params_channels(be_params); + if (params_rate(hw_params) != params_rate(be_params)) + io->converted_rate = params_rate(be_params); + } + if (io->converted_chan) + dev_dbg(dev, "convert channels = %d\n", io->converted_chan); + if (io->converted_rate) + dev_dbg(dev, "convert rate = %d\n", io->converted_rate); + } + ret = rsnd_dai_call(hw_params, io, substream, hw_params); if (ret) return ret; @@ -1274,8 +1449,15 @@ int rsnd_kctrl_accept_anytime(struct rsnd_dai_stream *io) int rsnd_kctrl_accept_runtime(struct rsnd_dai_stream *io) { struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + struct rsnd_priv *priv = rsnd_io_to_priv(io); + struct device *dev = rsnd_priv_to_dev(priv); + + if (!runtime) { + dev_warn(dev, "Can't update kctrl when idle\n"); + return 0; + } - return !!runtime; + return 1; } struct rsnd_kctrl_cfg *rsnd_kctrl_init_m(struct rsnd_kctrl_cfg_m *cfg) @@ -1343,6 +1525,18 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, }; int ret; + /* + * 1) Avoid duplicate register for DVC with MIX case + * 2) Allow duplicate register for MIX + * 3) re-register if card was rebinded + */ + list_for_each_entry(kctrl, &card->controls, list) { + struct rsnd_kctrl_cfg *c = kctrl->private_data; + + if (c == cfg) + return 0; + } + if (size > RSND_MAX_CHANNELS) return -EINVAL; diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index 6a55aa753003..8cb06dab234e 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -72,10 +72,7 @@ struct rsnd_ctu { struct rsnd_mod mod; struct rsnd_kctrl_cfg_m pass; - struct rsnd_kctrl_cfg_m sv0; - struct rsnd_kctrl_cfg_m sv1; - struct rsnd_kctrl_cfg_m sv2; - struct rsnd_kctrl_cfg_m sv3; + struct rsnd_kctrl_cfg_m sv[4]; struct rsnd_kctrl_cfg_s reset; int channels; u32 flags; @@ -107,13 +104,6 @@ static void rsnd_ctu_halt(struct rsnd_mod *mod) rsnd_mod_write(mod, CTU_SWRSR, 0); } -int rsnd_ctu_converted_channel(struct rsnd_mod *mod) -{ - struct rsnd_ctu *ctu = rsnd_mod_to_ctu(mod); - - return ctu->channels; -} - static int rsnd_ctu_probe_(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) @@ -127,7 +117,7 @@ static void rsnd_ctu_value_init(struct rsnd_dai_stream *io, struct rsnd_ctu *ctu = rsnd_mod_to_ctu(mod); u32 cpmdr = 0; u32 scmdr = 0; - int i; + int i, j; for (i = 0; i < RSND_MAX_CHANNELS; i++) { u32 val = rsnd_kctrl_valm(ctu->pass, i); @@ -146,45 +136,13 @@ static void rsnd_ctu_value_init(struct rsnd_dai_stream *io, rsnd_mod_write(mod, CTU_SCMDR, scmdr); - if (scmdr > 0) { - rsnd_mod_write(mod, CTU_SV00R, rsnd_kctrl_valm(ctu->sv0, 0)); - rsnd_mod_write(mod, CTU_SV01R, rsnd_kctrl_valm(ctu->sv0, 1)); - rsnd_mod_write(mod, CTU_SV02R, rsnd_kctrl_valm(ctu->sv0, 2)); - rsnd_mod_write(mod, CTU_SV03R, rsnd_kctrl_valm(ctu->sv0, 3)); - rsnd_mod_write(mod, CTU_SV04R, rsnd_kctrl_valm(ctu->sv0, 4)); - rsnd_mod_write(mod, CTU_SV05R, rsnd_kctrl_valm(ctu->sv0, 5)); - rsnd_mod_write(mod, CTU_SV06R, rsnd_kctrl_valm(ctu->sv0, 6)); - rsnd_mod_write(mod, CTU_SV07R, rsnd_kctrl_valm(ctu->sv0, 7)); - } - if (scmdr > 1) { - rsnd_mod_write(mod, CTU_SV10R, rsnd_kctrl_valm(ctu->sv1, 0)); - rsnd_mod_write(mod, CTU_SV11R, rsnd_kctrl_valm(ctu->sv1, 1)); - rsnd_mod_write(mod, CTU_SV12R, rsnd_kctrl_valm(ctu->sv1, 2)); - rsnd_mod_write(mod, CTU_SV13R, rsnd_kctrl_valm(ctu->sv1, 3)); - rsnd_mod_write(mod, CTU_SV14R, rsnd_kctrl_valm(ctu->sv1, 4)); - rsnd_mod_write(mod, CTU_SV15R, rsnd_kctrl_valm(ctu->sv1, 5)); - rsnd_mod_write(mod, CTU_SV16R, rsnd_kctrl_valm(ctu->sv1, 6)); - rsnd_mod_write(mod, CTU_SV17R, rsnd_kctrl_valm(ctu->sv1, 7)); - } - if (scmdr > 2) { - rsnd_mod_write(mod, CTU_SV20R, rsnd_kctrl_valm(ctu->sv2, 0)); - rsnd_mod_write(mod, CTU_SV21R, rsnd_kctrl_valm(ctu->sv2, 1)); - rsnd_mod_write(mod, CTU_SV22R, rsnd_kctrl_valm(ctu->sv2, 2)); - rsnd_mod_write(mod, CTU_SV23R, rsnd_kctrl_valm(ctu->sv2, 3)); - rsnd_mod_write(mod, CTU_SV24R, rsnd_kctrl_valm(ctu->sv2, 4)); - rsnd_mod_write(mod, CTU_SV25R, rsnd_kctrl_valm(ctu->sv2, 5)); - rsnd_mod_write(mod, CTU_SV26R, rsnd_kctrl_valm(ctu->sv2, 6)); - rsnd_mod_write(mod, CTU_SV27R, rsnd_kctrl_valm(ctu->sv2, 7)); - } - if (scmdr > 3) { - rsnd_mod_write(mod, CTU_SV30R, rsnd_kctrl_valm(ctu->sv3, 0)); - rsnd_mod_write(mod, CTU_SV31R, rsnd_kctrl_valm(ctu->sv3, 1)); - rsnd_mod_write(mod, CTU_SV32R, rsnd_kctrl_valm(ctu->sv3, 2)); - rsnd_mod_write(mod, CTU_SV33R, rsnd_kctrl_valm(ctu->sv3, 3)); - rsnd_mod_write(mod, CTU_SV34R, rsnd_kctrl_valm(ctu->sv3, 4)); - rsnd_mod_write(mod, CTU_SV35R, rsnd_kctrl_valm(ctu->sv3, 5)); - rsnd_mod_write(mod, CTU_SV36R, rsnd_kctrl_valm(ctu->sv3, 6)); - rsnd_mod_write(mod, CTU_SV37R, rsnd_kctrl_valm(ctu->sv3, 7)); + for (i = 0; i < 4; i++) { + + if (i >= scmdr) + break; + + for (j = 0; j < RSND_MAX_CHANNELS; j++) + rsnd_mod_write(mod, CTU_SVxxR(i, j), rsnd_kctrl_valm(ctu->sv[i], j)); } rsnd_mod_write(mod, CTU_CTUIR, 0); @@ -201,10 +159,10 @@ static void rsnd_ctu_value_reset(struct rsnd_dai_stream *io, for (i = 0; i < RSND_MAX_CHANNELS; i++) { rsnd_kctrl_valm(ctu->pass, i) = 0; - rsnd_kctrl_valm(ctu->sv0, i) = 0; - rsnd_kctrl_valm(ctu->sv1, i) = 0; - rsnd_kctrl_valm(ctu->sv2, i) = 0; - rsnd_kctrl_valm(ctu->sv3, i) = 0; + rsnd_kctrl_valm(ctu->sv[0], i) = 0; + rsnd_kctrl_valm(ctu->sv[1], i) = 0; + rsnd_kctrl_valm(ctu->sv[2], i) = 0; + rsnd_kctrl_valm(ctu->sv[3], i) = 0; } rsnd_kctrl_vals(ctu->reset) = 0; } @@ -233,43 +191,6 @@ static int rsnd_ctu_quit(struct rsnd_mod *mod, return 0; } -static int rsnd_ctu_hw_params(struct rsnd_mod *mod, - struct rsnd_dai_stream *io, - struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *fe_params) -{ - struct rsnd_ctu *ctu = rsnd_mod_to_ctu(mod); - struct snd_soc_pcm_runtime *fe = substream->private_data; - - /* - * CTU assumes that it is used under DPCM if user want to use - * channel transfer. Then, CTU should be FE. - * And then, this function will be called *after* BE settings. - * this means, each BE already has fixuped hw_params. - * see - * dpcm_fe_dai_hw_params() - * dpcm_be_dai_hw_params() - */ - ctu->channels = 0; - if (fe->dai_link->dynamic) { - struct rsnd_priv *priv = rsnd_mod_to_priv(mod); - struct device *dev = rsnd_priv_to_dev(priv); - struct snd_soc_dpcm *dpcm; - struct snd_pcm_hw_params *be_params; - int stream = substream->stream; - - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { - be_params = &dpcm->hw_params; - if (params_channels(fe_params) != params_channels(be_params)) - ctu->channels = params_channels(be_params); - } - - dev_dbg(dev, "CTU convert channels %d\n", ctu->channels); - } - - return 0; -} - static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct snd_soc_pcm_runtime *rtd) @@ -291,7 +212,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV0", rsnd_kctrl_accept_anytime, NULL, - &ctu->sv0, RSND_MAX_CHANNELS, + &ctu->sv[0], RSND_MAX_CHANNELS, 0x00FFFFFF); if (ret < 0) return ret; @@ -300,7 +221,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV1", rsnd_kctrl_accept_anytime, NULL, - &ctu->sv1, RSND_MAX_CHANNELS, + &ctu->sv[1], RSND_MAX_CHANNELS, 0x00FFFFFF); if (ret < 0) return ret; @@ -309,7 +230,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV2", rsnd_kctrl_accept_anytime, NULL, - &ctu->sv2, RSND_MAX_CHANNELS, + &ctu->sv[2], RSND_MAX_CHANNELS, 0x00FFFFFF); if (ret < 0) return ret; @@ -318,7 +239,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV3", rsnd_kctrl_accept_anytime, NULL, - &ctu->sv3, RSND_MAX_CHANNELS, + &ctu->sv[3], RSND_MAX_CHANNELS, 0x00FFFFFF); if (ret < 0) return ret; @@ -334,13 +255,34 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, return ret; } +static int rsnd_ctu_id(struct rsnd_mod *mod) +{ + /* + * ctu00: -> 0, ctu01: -> 0, ctu02: -> 0, ctu03: -> 0 + * ctu10: -> 1, ctu11: -> 1, ctu12: -> 1, ctu13: -> 1 + */ + return mod->id / 4; +} + +static int rsnd_ctu_id_sub(struct rsnd_mod *mod) +{ + /* + * ctu00: -> 0, ctu01: -> 1, ctu02: -> 2, ctu03: -> 3 + * ctu10: -> 0, ctu11: -> 1, ctu12: -> 2, ctu13: -> 3 + */ + return mod->id % 4; +} + static struct rsnd_mod_ops rsnd_ctu_ops = { .name = CTU_NAME, .probe = rsnd_ctu_probe_, .init = rsnd_ctu_init, .quit = rsnd_ctu_quit, - .hw_params = rsnd_ctu_hw_params, .pcm_new = rsnd_ctu_pcm_new, + .get_status = rsnd_mod_get_status, + .id = rsnd_ctu_id, + .id_sub = rsnd_ctu_id_sub, + .id_cmd = rsnd_mod_id_raw, }; struct rsnd_mod *rsnd_ctu_mod_get(struct rsnd_priv *priv, int id) @@ -404,7 +346,7 @@ int rsnd_ctu_probe(struct rsnd_priv *priv) } ret = rsnd_mod_init(priv, rsnd_mod_get(ctu), &rsnd_ctu_ops, - clk, rsnd_mod_get_status, RSND_MOD_CTU, i); + clk, RSND_MOD_CTU, i); if (ret) { of_node_put(np); goto rsnd_ctu_probe_done; diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index d65ea7bc4dac..0324a5c39619 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -106,9 +106,9 @@ static int rsnd_dmaen_stop(struct rsnd_mod *mod, return 0; } -static int rsnd_dmaen_nolock_stop(struct rsnd_mod *mod, - struct rsnd_dai_stream *io, - struct rsnd_priv *priv) +static int rsnd_dmaen_cleanup(struct rsnd_mod *mod, + struct rsnd_dai_stream *io, + struct rsnd_priv *priv) { struct rsnd_dma *dma = rsnd_mod_to_dma(mod); struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); @@ -116,7 +116,7 @@ static int rsnd_dmaen_nolock_stop(struct rsnd_mod *mod, /* * DMAEngine release uses mutex lock. * Thus, it shouldn't be called under spinlock. - * Let's call it under nolock_start + * Let's call it under prepare */ if (dmaen->chan) dma_release_channel(dmaen->chan); @@ -126,23 +126,22 @@ static int rsnd_dmaen_nolock_stop(struct rsnd_mod *mod, return 0; } -static int rsnd_dmaen_nolock_start(struct rsnd_mod *mod, - struct rsnd_dai_stream *io, - struct rsnd_priv *priv) +static int rsnd_dmaen_prepare(struct rsnd_mod *mod, + struct rsnd_dai_stream *io, + struct rsnd_priv *priv) { struct rsnd_dma *dma = rsnd_mod_to_dma(mod); struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); struct device *dev = rsnd_priv_to_dev(priv); - if (dmaen->chan) { - dev_err(dev, "it already has dma channel\n"); - return -EIO; - } + /* maybe suspended */ + if (dmaen->chan) + return 0; /* * DMAEngine request uses mutex lock. * Thus, it shouldn't be called under spinlock. - * Let's call it under nolock_start + * Let's call it under prepare */ dmaen->chan = rsnd_dmaen_request_channel(io, dma->mod_from, @@ -175,8 +174,8 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, cfg.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; cfg.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; - dev_dbg(dev, "%s[%d] %pad -> %pad\n", - rsnd_mod_name(mod), rsnd_mod_id(mod), + dev_dbg(dev, "%s %pad -> %pad\n", + rsnd_mod_name(mod), &cfg.src_addr, &cfg.dst_addr); ret = dmaengine_slave_config(dmaen->chan, &cfg); @@ -219,7 +218,7 @@ struct dma_chan *rsnd_dma_request_channel(struct device_node *of_node, int i = 0; for_each_child_of_node(of_node, np) { - if (i == rsnd_mod_id(mod) && (!chan)) + if (i == rsnd_mod_id_raw(mod) && (!chan)) chan = of_dma_request_slave_channel(np, name); i++; } @@ -290,28 +289,39 @@ static int rsnd_dmaen_pointer(struct rsnd_mod *mod, } static struct rsnd_mod_ops rsnd_dmaen_ops = { - .name = "audmac", - .nolock_start = rsnd_dmaen_nolock_start, - .nolock_stop = rsnd_dmaen_nolock_stop, - .start = rsnd_dmaen_start, - .stop = rsnd_dmaen_stop, - .pointer= rsnd_dmaen_pointer, + .name = "audmac", + .prepare = rsnd_dmaen_prepare, + .cleanup = rsnd_dmaen_cleanup, + .start = rsnd_dmaen_start, + .stop = rsnd_dmaen_stop, + .pointer = rsnd_dmaen_pointer, + .get_status = rsnd_mod_get_status, }; /* * Audio DMAC peri peri */ static const u8 gen2_id_table_ssiu[] = { - 0x00, /* SSI00 */ - 0x04, /* SSI10 */ - 0x08, /* SSI20 */ - 0x0c, /* SSI3 */ - 0x0d, /* SSI4 */ - 0x0e, /* SSI5 */ - 0x0f, /* SSI6 */ - 0x10, /* SSI7 */ - 0x11, /* SSI8 */ - 0x12, /* SSI90 */ + /* SSI00 ~ SSI07 */ + 0x00, 0x01, 0x02, 0x03, 0x39, 0x3a, 0x3b, 0x3c, + /* SSI10 ~ SSI17 */ + 0x04, 0x05, 0x06, 0x07, 0x3d, 0x3e, 0x3f, 0x40, + /* SSI20 ~ SSI27 */ + 0x08, 0x09, 0x0a, 0x0b, 0x41, 0x42, 0x43, 0x44, + /* SSI30 ~ SSI37 */ + 0x0c, 0x45, 0x46, 0x47, 0x48, 0x49, 0x4a, 0x4b, + /* SSI40 ~ SSI47 */ + 0x0d, 0x4c, 0x4d, 0x4e, 0x4f, 0x50, 0x51, 0x52, + /* SSI5 */ + 0x0e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + /* SSI6 */ + 0x0f, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + /* SSI7 */ + 0x10, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + /* SSI8 */ + 0x11, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + /* SSI90 ~ SSI97 */ + 0x12, 0x13, 0x14, 0x15, 0x53, 0x54, 0x55, 0x56, }; static const u8 gen2_id_table_scu[] = { 0x2d, /* SCU_SRCI0 */ @@ -334,28 +344,34 @@ static u32 rsnd_dmapp_get_id(struct rsnd_dai_stream *io, struct rsnd_mod *mod) { struct rsnd_mod *ssi = rsnd_io_to_mod_ssi(io); + struct rsnd_mod *ssiu = rsnd_io_to_mod_ssiu(io); struct rsnd_mod *src = rsnd_io_to_mod_src(io); struct rsnd_mod *dvc = rsnd_io_to_mod_dvc(io); const u8 *entry = NULL; - int id = rsnd_mod_id(mod); + int id = 255; int size = 0; - if (mod == ssi) { + if ((mod == ssi) || + (mod == ssiu)) { + int busif = rsnd_mod_id_sub(ssiu); + entry = gen2_id_table_ssiu; size = ARRAY_SIZE(gen2_id_table_ssiu); + id = (rsnd_mod_id(mod) * 8) + busif; } else if (mod == src) { entry = gen2_id_table_scu; size = ARRAY_SIZE(gen2_id_table_scu); + id = rsnd_mod_id(mod); } else if (mod == dvc) { entry = gen2_id_table_cmd; size = ARRAY_SIZE(gen2_id_table_cmd); + id = rsnd_mod_id(mod); } if ((!entry) || (size <= id)) { struct device *dev = rsnd_priv_to_dev(rsnd_io_to_priv(io)); - dev_err(dev, "unknown connection (%s[%d])\n", - rsnd_mod_name(mod), rsnd_mod_id(mod)); + dev_err(dev, "unknown connection (%s)\n", rsnd_mod_name(mod)); /* use non-prohibited SRS number as error */ return 0x00; /* SSI00 */ @@ -382,7 +398,7 @@ static void rsnd_dmapp_write(struct rsnd_dma *dma, u32 data, u32 reg) struct rsnd_dma_ctrl *dmac = rsnd_priv_to_dmac(priv); struct device *dev = rsnd_priv_to_dev(priv); - dev_dbg(dev, "w %p : %08x\n", rsnd_dmapp_addr(dmac, dma, reg), data); + dev_dbg(dev, "w 0x%px : %08x\n", rsnd_dmapp_addr(dmac, dma, reg), data); iowrite32(data, rsnd_dmapp_addr(dmac, dma, reg)); } @@ -463,10 +479,11 @@ static int rsnd_dmapp_attach(struct rsnd_dai_stream *io, } static struct rsnd_mod_ops rsnd_dmapp_ops = { - .name = "audmac-pp", - .start = rsnd_dmapp_start, - .stop = rsnd_dmapp_stop, - .quit = rsnd_dmapp_stop, + .name = "audmac-pp", + .start = rsnd_dmapp_start, + .stop = rsnd_dmapp_stop, + .quit = rsnd_dmapp_stop, + .get_status = rsnd_mod_get_status, }; /* @@ -491,11 +508,11 @@ static struct rsnd_mod_ops rsnd_dmapp_ops = { #define RDMA_SSI_I_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0x8) #define RDMA_SSI_O_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0xc) -#define RDMA_SSIU_I_N(addr, i) (addr ##_reg - 0x00441000 + (0x1000 * i)) -#define RDMA_SSIU_O_N(addr, i) (addr ##_reg - 0x00441000 + (0x1000 * i)) +#define RDMA_SSIU_I_N(addr, i, j) (addr ##_reg - 0x00441000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400)) +#define RDMA_SSIU_O_N(addr, i, j) RDMA_SSIU_I_N(addr, i, j) -#define RDMA_SSIU_I_P(addr, i) (addr ##_reg - 0x00141000 + (0x1000 * i)) -#define RDMA_SSIU_O_P(addr, i) (addr ##_reg - 0x00141000 + (0x1000 * i)) +#define RDMA_SSIU_I_P(addr, i, j) (addr ##_reg - 0x00141000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400)) +#define RDMA_SSIU_O_P(addr, i, j) RDMA_SSIU_I_P(addr, i, j) #define RDMA_SRC_I_N(addr, i) (addr ##_reg - 0x00500000 + (0x400 * i)) #define RDMA_SRC_O_N(addr, i) (addr ##_reg - 0x004fc000 + (0x400 * i)) @@ -515,12 +532,14 @@ rsnd_gen2_dma_addr(struct rsnd_dai_stream *io, struct device *dev = rsnd_priv_to_dev(priv); phys_addr_t ssi_reg = rsnd_gen_get_phy_addr(priv, RSND_GEN2_SSI); phys_addr_t src_reg = rsnd_gen_get_phy_addr(priv, RSND_GEN2_SCU); - int is_ssi = !!(rsnd_io_to_mod_ssi(io) == mod); + int is_ssi = !!(rsnd_io_to_mod_ssi(io) == mod) || + !!(rsnd_io_to_mod_ssiu(io) == mod); int use_src = !!rsnd_io_to_mod_src(io); int use_cmd = !!rsnd_io_to_mod_dvc(io) || !!rsnd_io_to_mod_mix(io) || !!rsnd_io_to_mod_ctu(io); int id = rsnd_mod_id(mod); + int busif = rsnd_mod_id_sub(rsnd_io_to_mod_ssiu(io)); struct dma_addr { dma_addr_t out_addr; dma_addr_t in_addr; @@ -537,25 +556,35 @@ rsnd_gen2_dma_addr(struct rsnd_dai_stream *io, }, /* SSI */ /* Capture */ - {{{ RDMA_SSI_O_N(ssi, id), 0 }, - { RDMA_SSIU_O_P(ssi, id), 0 }, - { RDMA_SSIU_O_P(ssi, id), 0 } }, + {{{ RDMA_SSI_O_N(ssi, id), 0 }, + { RDMA_SSIU_O_P(ssi, id, busif), 0 }, + { RDMA_SSIU_O_P(ssi, id, busif), 0 } }, /* Playback */ - {{ 0, RDMA_SSI_I_N(ssi, id) }, - { 0, RDMA_SSIU_I_P(ssi, id) }, - { 0, RDMA_SSIU_I_P(ssi, id) } } + {{ 0, RDMA_SSI_I_N(ssi, id) }, + { 0, RDMA_SSIU_I_P(ssi, id, busif) }, + { 0, RDMA_SSIU_I_P(ssi, id, busif) } } }, /* SSIU */ /* Capture */ - {{{ RDMA_SSIU_O_N(ssi, id), 0 }, - { RDMA_SSIU_O_P(ssi, id), 0 }, - { RDMA_SSIU_O_P(ssi, id), 0 } }, + {{{ RDMA_SSIU_O_N(ssi, id, busif), 0 }, + { RDMA_SSIU_O_P(ssi, id, busif), 0 }, + { RDMA_SSIU_O_P(ssi, id, busif), 0 } }, /* Playback */ - {{ 0, RDMA_SSIU_I_N(ssi, id) }, - { 0, RDMA_SSIU_I_P(ssi, id) }, - { 0, RDMA_SSIU_I_P(ssi, id) } } }, + {{ 0, RDMA_SSIU_I_N(ssi, id, busif) }, + { 0, RDMA_SSIU_I_P(ssi, id, busif) }, + { 0, RDMA_SSIU_I_P(ssi, id, busif) } } }, }; + /* + * FIXME + * + * We can't support SSI9-4/5/6/7, because its address is + * out of calculation rule + */ + if ((id == 9) && (busif >= 4)) + dev_err(dev, "This driver doesn't support SSI%d-%d, so far", + id, busif); + /* it shouldn't happen */ if (use_cmd && !use_src) dev_err(dev, "DVC is selected without SRC\n"); @@ -594,7 +623,7 @@ static void rsnd_dma_of_path(struct rsnd_mod *this, struct rsnd_mod **mod_from, struct rsnd_mod **mod_to) { - struct rsnd_mod *ssi = rsnd_io_to_mod_ssi(io); + struct rsnd_mod *ssi; struct rsnd_mod *src = rsnd_io_to_mod_src(io); struct rsnd_mod *ctu = rsnd_io_to_mod_ctu(io); struct rsnd_mod *mix = rsnd_io_to_mod_mix(io); @@ -605,6 +634,28 @@ static void rsnd_dma_of_path(struct rsnd_mod *this, struct device *dev = rsnd_priv_to_dev(priv); int nr, i, idx; + /* + * It should use "rcar_sound,ssiu" on DT. + * But, we need to keep compatibility for old version. + * + * If it has "rcar_sound.ssiu", it will be used. + * If not, "rcar_sound.ssi" will be used. + * see + * rsnd_ssiu_dma_req() + * rsnd_ssi_dma_req() + */ + if (rsnd_ssiu_of_node(priv)) { + struct rsnd_mod *ssiu = rsnd_io_to_mod_ssiu(io); + + /* use SSIU */ + ssi = ssiu; + if (this == rsnd_io_to_mod_ssi(io)) + this = ssiu; + } else { + /* keep compatible, use SSI */ + ssi = rsnd_io_to_mod_ssi(io); + } + if (!ssi) return; @@ -665,12 +716,10 @@ static void rsnd_dma_of_path(struct rsnd_mod *this, *mod_to = mod[1]; } - dev_dbg(dev, "module connection (this is %s[%d])\n", - rsnd_mod_name(this), rsnd_mod_id(this)); + dev_dbg(dev, "module connection (this is %s)\n", rsnd_mod_name(this)); for (i = 0; i <= idx; i++) { - dev_dbg(dev, " %s[%d]%s\n", + dev_dbg(dev, " %s%s\n", rsnd_mod_name(mod[i] ? mod[i] : &mem), - rsnd_mod_id (mod[i] ? mod[i] : &mem), (mod[i] == *mod_from) ? " from" : (mod[i] == *mod_to) ? " to" : ""); } @@ -731,16 +780,14 @@ static int rsnd_dma_alloc(struct rsnd_dai_stream *io, struct rsnd_mod *mod, *dma_mod = rsnd_mod_get(dma); ret = rsnd_mod_init(priv, *dma_mod, ops, NULL, - rsnd_mod_get_status, type, dma_id); + type, dma_id); if (ret < 0) return ret; - dev_dbg(dev, "%s[%d] %s[%d] -> %s[%d]\n", - rsnd_mod_name(*dma_mod), rsnd_mod_id(*dma_mod), + dev_dbg(dev, "%s %s -> %s\n", + rsnd_mod_name(*dma_mod), rsnd_mod_name(mod_from ? mod_from : &mem), - rsnd_mod_id (mod_from ? mod_from : &mem), - rsnd_mod_name(mod_to ? mod_to : &mem), - rsnd_mod_id (mod_to ? mod_to : &mem)); + rsnd_mod_name(mod_to ? mod_to : &mem)); ret = attach(io, dma, mod_from, mod_to); if (ret < 0) @@ -798,5 +845,5 @@ int rsnd_dma_probe(struct rsnd_priv *priv) priv->dma = dmac; /* dummy mem mod for debug */ - return rsnd_mod_init(NULL, &mem, &mem_ops, NULL, NULL, 0, 0); + return rsnd_mod_init(NULL, &mem, &mem_ops, NULL, 0, 0); } diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 2b16e0ce6bc5..8d91c0eb0880 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -40,11 +40,8 @@ struct rsnd_dvc { struct rsnd_kctrl_cfg_s ren; /* Ramp Enable */ struct rsnd_kctrl_cfg_s rup; /* Ramp Rate Up */ struct rsnd_kctrl_cfg_s rdown; /* Ramp Rate Down */ - u32 flags; }; -#define KCTRL_INITIALIZED (1 << 0) - #define rsnd_dvc_get(priv, id) ((struct rsnd_dvc *)(priv->dvc) + id) #define rsnd_dvc_nr(priv) ((priv)->dvc_nr) @@ -89,14 +86,8 @@ static void rsnd_dvc_volume_parameter(struct rsnd_dai_stream *io, val[i] = rsnd_kctrl_valm(dvc->volume, i); /* Enable Digital Volume */ - rsnd_mod_write(mod, DVC_VOL0R, val[0]); - rsnd_mod_write(mod, DVC_VOL1R, val[1]); - rsnd_mod_write(mod, DVC_VOL2R, val[2]); - rsnd_mod_write(mod, DVC_VOL3R, val[3]); - rsnd_mod_write(mod, DVC_VOL4R, val[4]); - rsnd_mod_write(mod, DVC_VOL5R, val[5]); - rsnd_mod_write(mod, DVC_VOL6R, val[6]); - rsnd_mod_write(mod, DVC_VOL7R, val[7]); + for (i = 0; i < RSND_MAX_CHANNELS; i++) + rsnd_mod_write(mod, DVC_VOLxR(i), val[i]); } static void rsnd_dvc_volume_init(struct rsnd_dai_stream *io, @@ -227,9 +218,6 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, int channels = rsnd_rdai_channels_get(rdai); int ret; - if (rsnd_flags_has(dvc, KCTRL_INITIALIZED)) - return 0; - /* Volume */ ret = rsnd_kctrl_new_m(mod, io, rtd, is_play ? @@ -285,8 +273,6 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, if (ret < 0) return ret; - rsnd_flags_set(dvc, KCTRL_INITIALIZED); - return 0; } @@ -306,6 +292,7 @@ static struct rsnd_mod_ops rsnd_dvc_ops = { .init = rsnd_dvc_init, .quit = rsnd_dvc_quit, .pcm_new = rsnd_dvc_pcm_new, + .get_status = rsnd_mod_get_status, }; struct rsnd_mod *rsnd_dvc_mod_get(struct rsnd_priv *priv, int id) @@ -365,7 +352,7 @@ int rsnd_dvc_probe(struct rsnd_priv *priv) } ret = rsnd_mod_init(priv, rsnd_mod_get(dvc), &rsnd_dvc_ops, - clk, rsnd_mod_get_status, RSND_MOD_DVC, i); + clk, RSND_MOD_DVC, i); if (ret) { of_node_put(np); goto rsnd_dvc_probe_done; diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 0230301fe078..7cda60188f41 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -26,8 +26,8 @@ struct rsnd_gen { struct regmap *regmap[RSND_BASE_MAX]; /* RSND_REG_MAX base */ - struct regmap_field *regs[RSND_REG_MAX]; - const char *reg_name[RSND_REG_MAX]; + struct regmap_field *regs[REG_MAX]; + const char *reg_name[REG_MAX]; }; #define rsnd_priv_to_gen(p) ((struct rsnd_gen *)(p)->gen) @@ -49,11 +49,11 @@ struct rsnd_regmap_field_conf { } /* single address mapping */ #define RSND_GEN_S_REG(id, offset) \ - RSND_REG_SET(RSND_REG_##id, offset, 0, #id) + RSND_REG_SET(id, offset, 0, #id) /* multi address mapping */ #define RSND_GEN_M_REG(id, offset, _id_offset) \ - RSND_REG_SET(RSND_REG_##id, offset, _id_offset, #id) + RSND_REG_SET(id, offset, _id_offset, #id) /* * basic function @@ -71,9 +71,17 @@ static int rsnd_is_accessible_reg(struct rsnd_priv *priv, return 1; } -u32 rsnd_read(struct rsnd_priv *priv, - struct rsnd_mod *mod, enum rsnd_reg reg) +static int rsnd_mod_id_cmd(struct rsnd_mod *mod) { + if (mod->ops->id_cmd) + return mod->ops->id_cmd(mod); + + return rsnd_mod_id(mod); +} + +u32 rsnd_mod_read(struct rsnd_mod *mod, enum rsnd_reg reg) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_gen *gen = rsnd_priv_to_gen(priv); u32 val; @@ -81,35 +89,36 @@ u32 rsnd_read(struct rsnd_priv *priv, if (!rsnd_is_accessible_reg(priv, gen, reg)) return 0; - regmap_fields_read(gen->regs[reg], rsnd_mod_id(mod), &val); + regmap_fields_read(gen->regs[reg], rsnd_mod_id_cmd(mod), &val); - dev_dbg(dev, "r %s[%d] - %-18s (%4d) : %08x\n", - rsnd_mod_name(mod), rsnd_mod_id(mod), + dev_dbg(dev, "r %s - %-18s (%4d) : %08x\n", + rsnd_mod_name(mod), rsnd_reg_name(gen, reg), reg, val); return val; } -void rsnd_write(struct rsnd_priv *priv, - struct rsnd_mod *mod, - enum rsnd_reg reg, u32 data) +void rsnd_mod_write(struct rsnd_mod *mod, + enum rsnd_reg reg, u32 data) { + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_gen *gen = rsnd_priv_to_gen(priv); if (!rsnd_is_accessible_reg(priv, gen, reg)) return; - regmap_fields_force_write(gen->regs[reg], rsnd_mod_id(mod), data); + regmap_fields_force_write(gen->regs[reg], rsnd_mod_id_cmd(mod), data); - dev_dbg(dev, "w %s[%d] - %-18s (%4d) : %08x\n", - rsnd_mod_name(mod), rsnd_mod_id(mod), + dev_dbg(dev, "w %s - %-18s (%4d) : %08x\n", + rsnd_mod_name(mod), rsnd_reg_name(gen, reg), reg, data); } -void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, - enum rsnd_reg reg, u32 mask, u32 data) +void rsnd_mod_bset(struct rsnd_mod *mod, + enum rsnd_reg reg, u32 mask, u32 data) { + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_gen *gen = rsnd_priv_to_gen(priv); @@ -117,10 +126,10 @@ void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, return; regmap_fields_force_update_bits(gen->regs[reg], - rsnd_mod_id(mod), mask, data); + rsnd_mod_id_cmd(mod), mask, data); - dev_dbg(dev, "b %s[%d] - %-18s (%4d) : %08x/%08x\n", - rsnd_mod_name(mod), rsnd_mod_id(mod), + dev_dbg(dev, "b %s - %-18s (%4d) : %08x/%08x\n", + rsnd_mod_name(mod), rsnd_reg_name(gen, reg), reg, data, mask); } @@ -219,12 +228,33 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv) RSND_GEN_S_REG(HDMI1_SEL, 0x9e4), /* FIXME: it needs SSI_MODE2/3 in the future */ - RSND_GEN_M_REG(SSI_BUSIF_MODE, 0x0, 0x80), - RSND_GEN_M_REG(SSI_BUSIF_ADINR, 0x4, 0x80), - RSND_GEN_M_REG(SSI_BUSIF_DALIGN,0x8, 0x80), - RSND_GEN_M_REG(SSI_MODE, 0xc, 0x80), - RSND_GEN_M_REG(SSI_CTRL, 0x10, 0x80), - RSND_GEN_M_REG(SSI_INT_ENABLE, 0x18, 0x80), + RSND_GEN_M_REG(SSI_BUSIF0_MODE, 0x0, 0x80), + RSND_GEN_M_REG(SSI_BUSIF0_ADINR, 0x4, 0x80), + RSND_GEN_M_REG(SSI_BUSIF0_DALIGN, 0x8, 0x80), + RSND_GEN_M_REG(SSI_BUSIF1_MODE, 0x20, 0x80), + RSND_GEN_M_REG(SSI_BUSIF1_ADINR, 0x24, 0x80), + RSND_GEN_M_REG(SSI_BUSIF1_DALIGN, 0x28, 0x80), + RSND_GEN_M_REG(SSI_BUSIF2_MODE, 0x40, 0x80), + RSND_GEN_M_REG(SSI_BUSIF2_ADINR, 0x44, 0x80), + RSND_GEN_M_REG(SSI_BUSIF2_DALIGN, 0x48, 0x80), + RSND_GEN_M_REG(SSI_BUSIF3_MODE, 0x60, 0x80), + RSND_GEN_M_REG(SSI_BUSIF3_ADINR, 0x64, 0x80), + RSND_GEN_M_REG(SSI_BUSIF3_DALIGN, 0x68, 0x80), + RSND_GEN_M_REG(SSI_BUSIF4_MODE, 0x500, 0x80), + RSND_GEN_M_REG(SSI_BUSIF4_ADINR, 0x504, 0x80), + RSND_GEN_M_REG(SSI_BUSIF4_DALIGN, 0x508, 0x80), + RSND_GEN_M_REG(SSI_BUSIF5_MODE, 0x520, 0x80), + RSND_GEN_M_REG(SSI_BUSIF5_ADINR, 0x524, 0x80), + RSND_GEN_M_REG(SSI_BUSIF5_DALIGN, 0x528, 0x80), + RSND_GEN_M_REG(SSI_BUSIF6_MODE, 0x540, 0x80), + RSND_GEN_M_REG(SSI_BUSIF6_ADINR, 0x544, 0x80), + RSND_GEN_M_REG(SSI_BUSIF6_DALIGN, 0x548, 0x80), + RSND_GEN_M_REG(SSI_BUSIF7_MODE, 0x560, 0x80), + RSND_GEN_M_REG(SSI_BUSIF7_ADINR, 0x564, 0x80), + RSND_GEN_M_REG(SSI_BUSIF7_DALIGN, 0x568, 0x80), + RSND_GEN_M_REG(SSI_MODE, 0xc, 0x80), + RSND_GEN_M_REG(SSI_CTRL, 0x10, 0x80), + RSND_GEN_M_REG(SSI_INT_ENABLE, 0x18, 0x80), }; static const struct rsnd_regmap_field_conf conf_scu[] = { diff --git a/sound/soc/sh/rcar/mix.c b/sound/soc/sh/rcar/mix.c index 8e3b57eaa708..a3e0370f5704 100644 --- a/sound/soc/sh/rcar/mix.c +++ b/sound/soc/sh/rcar/mix.c @@ -256,6 +256,7 @@ static struct rsnd_mod_ops rsnd_mix_ops = { .init = rsnd_mix_init, .quit = rsnd_mix_quit, .pcm_new = rsnd_mix_pcm_new, + .get_status = rsnd_mod_get_status, }; struct rsnd_mod *rsnd_mix_mod_get(struct rsnd_priv *priv, int id) @@ -315,7 +316,7 @@ int rsnd_mix_probe(struct rsnd_priv *priv) } ret = rsnd_mod_init(priv, rsnd_mod_get(mix), &rsnd_mix_ops, - clk, rsnd_mod_get_status, RSND_MOD_MIX, i); + clk, RSND_MOD_MIX, i); if (ret) { of_node_put(np); goto rsnd_mix_probe_done; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 8f7a0abfa751..605e4b934982 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -42,144 +42,175 @@ */ enum rsnd_reg { /* SCU (MIX/CTU/DVC) */ - RSND_REG_SRC_I_BUSIF_MODE, - RSND_REG_SRC_O_BUSIF_MODE, - RSND_REG_SRC_ROUTE_MODE0, - RSND_REG_SRC_SWRSR, - RSND_REG_SRC_SRCIR, - RSND_REG_SRC_ADINR, - RSND_REG_SRC_IFSCR, - RSND_REG_SRC_IFSVR, - RSND_REG_SRC_SRCCR, - RSND_REG_SRC_CTRL, - RSND_REG_SRC_BSDSR, - RSND_REG_SRC_BSISR, - RSND_REG_SRC_INT_ENABLE0, - RSND_REG_SRC_BUSIF_DALIGN, - RSND_REG_SRCIN_TIMSEL0, - RSND_REG_SRCIN_TIMSEL1, - RSND_REG_SRCIN_TIMSEL2, - RSND_REG_SRCIN_TIMSEL3, - RSND_REG_SRCIN_TIMSEL4, - RSND_REG_SRCOUT_TIMSEL0, - RSND_REG_SRCOUT_TIMSEL1, - RSND_REG_SRCOUT_TIMSEL2, - RSND_REG_SRCOUT_TIMSEL3, - RSND_REG_SRCOUT_TIMSEL4, - RSND_REG_SCU_SYS_STATUS0, - RSND_REG_SCU_SYS_STATUS1, - RSND_REG_SCU_SYS_INT_EN0, - RSND_REG_SCU_SYS_INT_EN1, - RSND_REG_CMD_CTRL, - RSND_REG_CMD_BUSIF_MODE, - RSND_REG_CMD_BUSIF_DALIGN, - RSND_REG_CMD_ROUTE_SLCT, - RSND_REG_CMDOUT_TIMSEL, - RSND_REG_CTU_SWRSR, - RSND_REG_CTU_CTUIR, - RSND_REG_CTU_ADINR, - RSND_REG_CTU_CPMDR, - RSND_REG_CTU_SCMDR, - RSND_REG_CTU_SV00R, - RSND_REG_CTU_SV01R, - RSND_REG_CTU_SV02R, - RSND_REG_CTU_SV03R, - RSND_REG_CTU_SV04R, - RSND_REG_CTU_SV05R, - RSND_REG_CTU_SV06R, - RSND_REG_CTU_SV07R, - RSND_REG_CTU_SV10R, - RSND_REG_CTU_SV11R, - RSND_REG_CTU_SV12R, - RSND_REG_CTU_SV13R, - RSND_REG_CTU_SV14R, - RSND_REG_CTU_SV15R, - RSND_REG_CTU_SV16R, - RSND_REG_CTU_SV17R, - RSND_REG_CTU_SV20R, - RSND_REG_CTU_SV21R, - RSND_REG_CTU_SV22R, - RSND_REG_CTU_SV23R, - RSND_REG_CTU_SV24R, - RSND_REG_CTU_SV25R, - RSND_REG_CTU_SV26R, - RSND_REG_CTU_SV27R, - RSND_REG_CTU_SV30R, - RSND_REG_CTU_SV31R, - RSND_REG_CTU_SV32R, - RSND_REG_CTU_SV33R, - RSND_REG_CTU_SV34R, - RSND_REG_CTU_SV35R, - RSND_REG_CTU_SV36R, - RSND_REG_CTU_SV37R, - RSND_REG_MIX_SWRSR, - RSND_REG_MIX_MIXIR, - RSND_REG_MIX_ADINR, - RSND_REG_MIX_MIXMR, - RSND_REG_MIX_MVPDR, - RSND_REG_MIX_MDBAR, - RSND_REG_MIX_MDBBR, - RSND_REG_MIX_MDBCR, - RSND_REG_MIX_MDBDR, - RSND_REG_MIX_MDBER, - RSND_REG_DVC_SWRSR, - RSND_REG_DVC_DVUIR, - RSND_REG_DVC_ADINR, - RSND_REG_DVC_DVUCR, - RSND_REG_DVC_ZCMCR, - RSND_REG_DVC_VOL0R, - RSND_REG_DVC_VOL1R, - RSND_REG_DVC_VOL2R, - RSND_REG_DVC_VOL3R, - RSND_REG_DVC_VOL4R, - RSND_REG_DVC_VOL5R, - RSND_REG_DVC_VOL6R, - RSND_REG_DVC_VOL7R, - RSND_REG_DVC_DVUER, - RSND_REG_DVC_VRCTR, - RSND_REG_DVC_VRPDR, - RSND_REG_DVC_VRDBR, + SRC_I_BUSIF_MODE, + SRC_O_BUSIF_MODE, + SRC_ROUTE_MODE0, + SRC_SWRSR, + SRC_SRCIR, + SRC_ADINR, + SRC_IFSCR, + SRC_IFSVR, + SRC_SRCCR, + SRC_CTRL, + SRC_BSDSR, + SRC_BSISR, + SRC_INT_ENABLE0, + SRC_BUSIF_DALIGN, + SRCIN_TIMSEL0, + SRCIN_TIMSEL1, + SRCIN_TIMSEL2, + SRCIN_TIMSEL3, + SRCIN_TIMSEL4, + SRCOUT_TIMSEL0, + SRCOUT_TIMSEL1, + SRCOUT_TIMSEL2, + SRCOUT_TIMSEL3, + SRCOUT_TIMSEL4, + SCU_SYS_STATUS0, + SCU_SYS_STATUS1, + SCU_SYS_INT_EN0, + SCU_SYS_INT_EN1, + CMD_CTRL, + CMD_BUSIF_MODE, + CMD_BUSIF_DALIGN, + CMD_ROUTE_SLCT, + CMDOUT_TIMSEL, + CTU_SWRSR, + CTU_CTUIR, + CTU_ADINR, + CTU_CPMDR, + CTU_SCMDR, + CTU_SV00R, + CTU_SV01R, + CTU_SV02R, + CTU_SV03R, + CTU_SV04R, + CTU_SV05R, + CTU_SV06R, + CTU_SV07R, + CTU_SV10R, + CTU_SV11R, + CTU_SV12R, + CTU_SV13R, + CTU_SV14R, + CTU_SV15R, + CTU_SV16R, + CTU_SV17R, + CTU_SV20R, + CTU_SV21R, + CTU_SV22R, + CTU_SV23R, + CTU_SV24R, + CTU_SV25R, + CTU_SV26R, + CTU_SV27R, + CTU_SV30R, + CTU_SV31R, + CTU_SV32R, + CTU_SV33R, + CTU_SV34R, + CTU_SV35R, + CTU_SV36R, + CTU_SV37R, + MIX_SWRSR, + MIX_MIXIR, + MIX_ADINR, + MIX_MIXMR, + MIX_MVPDR, + MIX_MDBAR, + MIX_MDBBR, + MIX_MDBCR, + MIX_MDBDR, + MIX_MDBER, + DVC_SWRSR, + DVC_DVUIR, + DVC_ADINR, + DVC_DVUCR, + DVC_ZCMCR, + DVC_VOL0R, + DVC_VOL1R, + DVC_VOL2R, + DVC_VOL3R, + DVC_VOL4R, + DVC_VOL5R, + DVC_VOL6R, + DVC_VOL7R, + DVC_DVUER, + DVC_VRCTR, + DVC_VRPDR, + DVC_VRDBR, /* ADG */ - RSND_REG_BRRA, - RSND_REG_BRRB, - RSND_REG_BRGCKR, - RSND_REG_DIV_EN, - RSND_REG_AUDIO_CLK_SEL0, - RSND_REG_AUDIO_CLK_SEL1, - RSND_REG_AUDIO_CLK_SEL2, + BRRA, + BRRB, + BRGCKR, + DIV_EN, + AUDIO_CLK_SEL0, + AUDIO_CLK_SEL1, + AUDIO_CLK_SEL2, /* SSIU */ - RSND_REG_SSI_MODE, - RSND_REG_SSI_MODE0, - RSND_REG_SSI_MODE1, - RSND_REG_SSI_MODE2, - RSND_REG_SSI_CONTROL, - RSND_REG_SSI_CTRL, - RSND_REG_SSI_BUSIF_MODE, - RSND_REG_SSI_BUSIF_ADINR, - RSND_REG_SSI_BUSIF_DALIGN, - RSND_REG_SSI_INT_ENABLE, - RSND_REG_SSI_SYS_STATUS0, - RSND_REG_SSI_SYS_STATUS1, - RSND_REG_SSI_SYS_STATUS2, - RSND_REG_SSI_SYS_STATUS3, - RSND_REG_SSI_SYS_STATUS4, - RSND_REG_SSI_SYS_STATUS5, - RSND_REG_SSI_SYS_STATUS6, - RSND_REG_SSI_SYS_STATUS7, - RSND_REG_HDMI0_SEL, - RSND_REG_HDMI1_SEL, + SSI_MODE, + SSI_MODE0, + SSI_MODE1, + SSI_MODE2, + SSI_CONTROL, + SSI_CTRL, + SSI_BUSIF0_MODE, + SSI_BUSIF1_MODE, + SSI_BUSIF2_MODE, + SSI_BUSIF3_MODE, + SSI_BUSIF4_MODE, + SSI_BUSIF5_MODE, + SSI_BUSIF6_MODE, + SSI_BUSIF7_MODE, + SSI_BUSIF0_ADINR, + SSI_BUSIF1_ADINR, + SSI_BUSIF2_ADINR, + SSI_BUSIF3_ADINR, + SSI_BUSIF4_ADINR, + SSI_BUSIF5_ADINR, + SSI_BUSIF6_ADINR, + SSI_BUSIF7_ADINR, + SSI_BUSIF0_DALIGN, + SSI_BUSIF1_DALIGN, + SSI_BUSIF2_DALIGN, + SSI_BUSIF3_DALIGN, + SSI_BUSIF4_DALIGN, + SSI_BUSIF5_DALIGN, + SSI_BUSIF6_DALIGN, + SSI_BUSIF7_DALIGN, + SSI_INT_ENABLE, + SSI_SYS_STATUS0, + SSI_SYS_STATUS1, + SSI_SYS_STATUS2, + SSI_SYS_STATUS3, + SSI_SYS_STATUS4, + SSI_SYS_STATUS5, + SSI_SYS_STATUS6, + SSI_SYS_STATUS7, + HDMI0_SEL, + HDMI1_SEL, /* SSI */ - RSND_REG_SSICR, - RSND_REG_SSISR, - RSND_REG_SSITDR, - RSND_REG_SSIRDR, - RSND_REG_SSIWSR, + SSICR, + SSISR, + SSITDR, + SSIRDR, + SSIWSR, - RSND_REG_MAX, + REG_MAX, }; +#define SRCIN_TIMSEL(i) (SRCIN_TIMSEL0 + (i)) +#define SRCOUT_TIMSEL(i) (SRCOUT_TIMSEL0 + (i)) +#define CTU_SVxxR(i, j) (CTU_SV00R + (i * 8) + (j)) +#define DVC_VOLxR(i) (DVC_VOL0R + (i)) +#define AUDIO_CLK_SEL(i) (AUDIO_CLK_SEL0 + (i)) +#define SSI_BUSIF_MODE(i) (SSI_BUSIF0_MODE + (i)) +#define SSI_BUSIF_ADINR(i) (SSI_BUSIF0_ADINR + (i)) +#define SSI_BUSIF_DALIGN(i) (SSI_BUSIF0_DALIGN + (i)) +#define SSI_SYS_STATUS(i) (SSI_SYS_STATUS0 + (i)) + struct rsnd_priv; struct rsnd_mod; @@ -189,20 +220,9 @@ 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) - -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_force_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); +u32 rsnd_mod_read(struct rsnd_mod *mod, enum rsnd_reg reg); +void rsnd_mod_write(struct rsnd_mod *mod, enum rsnd_reg reg, u32 data); +void rsnd_mod_bset(struct rsnd_mod *mod, enum rsnd_reg reg, u32 mask, u32 data); u32 rsnd_get_adinr_bit(struct rsnd_mod *mod, struct rsnd_dai_stream *io); u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io); u32 rsnd_get_busif_shift(struct rsnd_dai_stream *io, struct rsnd_mod *mod); @@ -274,15 +294,18 @@ struct rsnd_mod_ops { int (*fallback)(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv); - int (*nolock_start)(struct rsnd_mod *mod, - struct rsnd_dai_stream *io, - struct rsnd_priv *priv); - int (*nolock_stop)(struct rsnd_mod *mod, - struct rsnd_dai_stream *io, - struct rsnd_priv *priv); int (*prepare)(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv); + int (*cleanup)(struct rsnd_mod *mod, + struct rsnd_dai_stream *io, + struct rsnd_priv *priv); + u32 *(*get_status)(struct rsnd_mod *mod, + struct rsnd_dai_stream *io, + enum rsnd_mod_type type); + int (*id)(struct rsnd_mod *mod); + int (*id_sub)(struct rsnd_mod *mod); + int (*id_cmd)(struct rsnd_mod *mod); }; struct rsnd_dai_stream; @@ -292,17 +315,13 @@ struct rsnd_mod { struct rsnd_mod_ops *ops; struct rsnd_priv *priv; struct clk *clk; - u32 *(*get_status)(struct rsnd_dai_stream *io, - struct rsnd_mod *mod, - enum rsnd_mod_type type); u32 status; }; /* * status * - * 0xH0000CBA + * 0xH0000CB0 * - * A 0: nolock_start 1: nolock_stop * B 0: init 1: quit * C 0: start 1: stop * @@ -313,9 +332,8 @@ struct rsnd_mod { * H 0: hw_params * H 0: pointer * H 0: prepare + * H 0: cleanup */ -#define __rsnd_mod_shift_nolock_start 0 -#define __rsnd_mod_shift_nolock_stop 0 #define __rsnd_mod_shift_init 4 #define __rsnd_mod_shift_quit 4 #define __rsnd_mod_shift_start 8 @@ -328,11 +346,12 @@ struct rsnd_mod { #define __rsnd_mod_shift_hw_params 28 /* always called */ #define __rsnd_mod_shift_pointer 28 /* always called */ #define __rsnd_mod_shift_prepare 28 /* always called */ +#define __rsnd_mod_shift_cleanup 28 /* always called */ #define __rsnd_mod_add_probe 0 #define __rsnd_mod_add_remove 0 -#define __rsnd_mod_add_nolock_start 1 -#define __rsnd_mod_add_nolock_stop -1 +#define __rsnd_mod_add_prepare 0 +#define __rsnd_mod_add_cleanup 0 #define __rsnd_mod_add_init 1 #define __rsnd_mod_add_quit -1 #define __rsnd_mod_add_start 1 @@ -342,10 +361,11 @@ struct rsnd_mod { #define __rsnd_mod_add_fallback 0 #define __rsnd_mod_add_hw_params 0 #define __rsnd_mod_add_pointer 0 -#define __rsnd_mod_add_prepare 0 #define __rsnd_mod_call_probe 0 #define __rsnd_mod_call_remove 0 +#define __rsnd_mod_call_prepare 0 +#define __rsnd_mod_call_cleanup 0 #define __rsnd_mod_call_init 0 #define __rsnd_mod_call_quit 1 #define __rsnd_mod_call_start 0 @@ -355,13 +375,8 @@ struct rsnd_mod { #define __rsnd_mod_call_fallback 0 #define __rsnd_mod_call_hw_params 0 #define __rsnd_mod_call_pointer 0 -#define __rsnd_mod_call_nolock_start 0 -#define __rsnd_mod_call_nolock_stop 1 -#define __rsnd_mod_call_prepare 0 #define rsnd_mod_to_priv(mod) ((mod)->priv) -#define rsnd_mod_name(mod) ((mod)->ops->name) -#define rsnd_mod_id(mod) ((mod)->id) #define rsnd_mod_power_on(mod) clk_enable((mod)->clk) #define rsnd_mod_power_off(mod) clk_disable((mod)->clk) #define rsnd_mod_get(ip) (&(ip)->mod) @@ -370,9 +385,6 @@ int rsnd_mod_init(struct rsnd_priv *priv, struct rsnd_mod *mod, struct rsnd_mod_ops *ops, struct clk *clk, - u32* (*get_status)(struct rsnd_dai_stream *io, - struct rsnd_mod *mod, - enum rsnd_mod_type type), enum rsnd_mod_type type, int id); void rsnd_mod_quit(struct rsnd_mod *mod); @@ -381,9 +393,13 @@ struct dma_chan *rsnd_mod_dma_req(struct rsnd_dai_stream *io, void rsnd_mod_interrupt(struct rsnd_mod *mod, void (*callback)(struct rsnd_mod *mod, struct rsnd_dai_stream *io)); -u32 *rsnd_mod_get_status(struct rsnd_dai_stream *io, - struct rsnd_mod *mod, +u32 *rsnd_mod_get_status(struct rsnd_mod *mod, + struct rsnd_dai_stream *io, enum rsnd_mod_type type); +int rsnd_mod_id(struct rsnd_mod *mod); +int rsnd_mod_id_raw(struct rsnd_mod *mod); +int rsnd_mod_id_sub(struct rsnd_mod *mod); +char *rsnd_mod_name(struct rsnd_mod *mod); struct rsnd_mod *rsnd_mod_next(int *iterator, struct rsnd_dai_stream *io, enum rsnd_mod_type *array, @@ -415,8 +431,9 @@ int rsnd_runtime_channel_after_ctu_with_params(struct rsnd_dai_stream *io, rsnd_runtime_channel_for_ssi_with_params(io, NULL) int rsnd_runtime_channel_for_ssi_with_params(struct rsnd_dai_stream *io, struct snd_pcm_hw_params *params); -int rsnd_runtime_is_ssi_multi(struct rsnd_dai_stream *io); -int rsnd_runtime_is_ssi_tdm(struct rsnd_dai_stream *io); +int rsnd_runtime_is_multi_ssi(struct rsnd_dai_stream *io); +int rsnd_runtime_is_tdm(struct rsnd_dai_stream *io); +int rsnd_runtime_is_tdm_split(struct rsnd_dai_stream *io); /* * DT @@ -425,6 +442,7 @@ int rsnd_runtime_is_ssi_tdm(struct rsnd_dai_stream *io); of_get_child_by_name(rsnd_priv_to_dev(priv)->of_node, node) #define RSND_NODE_DAI "rcar_sound,dai" #define RSND_NODE_SSI "rcar_sound,ssi" +#define RSND_NODE_SSIU "rcar_sound,ssiu" #define RSND_NODE_SRC "rcar_sound,src" #define RSND_NODE_CTU "rcar_sound,ctu" #define RSND_NODE_MIX "rcar_sound,mix" @@ -438,10 +456,20 @@ struct rsnd_dai_stream { char name[RSND_DAI_NAME_SIZE]; struct snd_pcm_substream *substream; struct rsnd_mod *mod[RSND_MOD_MAX]; + struct rsnd_mod *dma; struct rsnd_dai *rdai; struct device *dmac_dev; /* for IPMMU */ + u32 converted_rate; /* converted sampling rate */ + int converted_chan; /* converted channels */ u32 parent_ssi_status; + u32 flags; }; + +/* flags */ +#define RSND_STREAM_HDMI0 (1 << 0) /* for HDMI0 */ +#define RSND_STREAM_HDMI1 (1 << 1) /* for HDMI1 */ +#define RSND_STREAM_TDM_SPLIT (1 << 2) /* for TDM split mode */ + #define rsnd_io_to_mod(io, i) ((i) < RSND_MOD_MAX ? (io)->mod[(i)] : NULL) #define rsnd_io_to_mod_ssi(io) rsnd_io_to_mod((io), RSND_MOD_SSI) #define rsnd_io_to_mod_ssiu(io) rsnd_io_to_mod((io), RSND_MOD_SSIU) @@ -456,6 +484,8 @@ struct rsnd_dai_stream { #define rsnd_io_is_play(io) (&rsnd_io_to_rdai(io)->playback == io) #define rsnd_io_to_runtime(io) ((io)->substream ? \ (io)->substream->runtime : NULL) +#define rsnd_io_converted_rate(io) ((io)->converted_rate) +#define rsnd_io_converted_chan(io) ((io)->converted_chan) int rsnd_io_is_working(struct rsnd_dai_stream *io); struct rsnd_dai { @@ -467,6 +497,7 @@ struct rsnd_dai { int max_channels; /* 2ch - 16ch */ int ssi_lane; /* 1lane - 4lane */ + int chan_width; /* 16/24/32 bit width */ unsigned int clk_master:1; unsigned int bit_clk_inv:1; @@ -500,6 +531,11 @@ int rsnd_rdai_channels_ctrl(struct rsnd_dai *rdai, int rsnd_rdai_ssi_lane_ctrl(struct rsnd_dai *rdai, int ssi_lane); +#define rsnd_rdai_width_set(rdai, width) \ + rsnd_rdai_width_ctrl(rdai, width) +#define rsnd_rdai_width_get(rdai) \ + rsnd_rdai_width_ctrl(rdai, 0) +int rsnd_rdai_width_ctrl(struct rsnd_dai *rdai, int width); void rsnd_dai_period_elapsed(struct rsnd_dai_stream *io); int rsnd_dai_connect(struct rsnd_mod *mod, struct rsnd_dai_stream *io, @@ -690,17 +726,9 @@ extern const char * const volume_ramp_rate[]; int rsnd_ssi_probe(struct rsnd_priv *priv); void rsnd_ssi_remove(struct rsnd_priv *priv); struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id); -int rsnd_ssi_is_dma_mode(struct rsnd_mod *mod); int rsnd_ssi_use_busif(struct rsnd_dai_stream *io); u32 rsnd_ssi_multi_slaves_runtime(struct rsnd_dai_stream *io); -#define RSND_SSI_HDMI_PORT0 0xf0 -#define RSND_SSI_HDMI_PORT1 0xf1 -int rsnd_ssi_hdmi_port(struct rsnd_dai_stream *io); -void rsnd_ssi_parse_hdmi_connection(struct rsnd_priv *priv, - struct device_node *endpoint, - int dai_i); - #define rsnd_ssi_is_pin_sharing(io) \ __rsnd_ssi_is_pin_sharing(rsnd_io_to_mod_ssi(io)) int __rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod); @@ -709,7 +737,7 @@ int __rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod); void rsnd_parse_connect_ssi(struct rsnd_dai *rdai, struct device_node *playback, struct device_node *capture); -unsigned int rsnd_ssi_clk_query(struct rsnd_priv *priv, +unsigned int rsnd_ssi_clk_query(struct rsnd_dai *rdai, int param1, int param2, int *idx); /* @@ -719,6 +747,10 @@ int rsnd_ssiu_attach(struct rsnd_dai_stream *io, struct rsnd_mod *mod); int rsnd_ssiu_probe(struct rsnd_priv *priv); void rsnd_ssiu_remove(struct rsnd_priv *priv); +void rsnd_parse_connect_ssiu(struct rsnd_dai *rdai, + struct device_node *playback, + struct device_node *capture); +#define rsnd_ssiu_of_node(priv) rsnd_parse_of_node(priv, RSND_NODE_SSIU) /* * R-Car SRC @@ -744,7 +776,6 @@ unsigned int rsnd_src_get_rate(struct rsnd_priv *priv, */ int rsnd_ctu_probe(struct rsnd_priv *priv); void rsnd_ctu_remove(struct rsnd_priv *priv); -int rsnd_ctu_converted_channel(struct rsnd_mod *mod); struct rsnd_mod *rsnd_ctu_mod_get(struct rsnd_priv *priv, int id); #define rsnd_ctu_of_node(priv) rsnd_parse_of_node(priv, RSND_NODE_CTU) #define rsnd_parse_connect_ctu(rdai, playback, capture) \ diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index beccfbac7581..50348a2c9203 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -25,7 +25,6 @@ struct rsnd_src { struct rsnd_mod *dma; struct rsnd_kctrl_cfg_s sen; /* sync convert enable */ struct rsnd_kctrl_cfg_s sync; /* sync convert */ - u32 convert_rate; /* sampling rate convert */ int irq; }; @@ -89,12 +88,12 @@ static u32 rsnd_src_convert_rate(struct rsnd_dai_stream *io, return 0; if (!rsnd_src_sync_is_enabled(mod)) - return src->convert_rate; + return rsnd_io_converted_rate(io); convert_rate = src->sync.val; if (!convert_rate) - convert_rate = src->convert_rate; + convert_rate = rsnd_io_converted_rate(io); if (!convert_rate) convert_rate = runtime->rate; @@ -135,40 +134,6 @@ unsigned int rsnd_src_get_rate(struct rsnd_priv *priv, return rate; } -static int rsnd_src_hw_params(struct rsnd_mod *mod, - struct rsnd_dai_stream *io, - struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *fe_params) -{ - struct rsnd_src *src = rsnd_mod_to_src(mod); - struct snd_soc_pcm_runtime *fe = substream->private_data; - - /* - * SRC assumes that it is used under DPCM if user want to use - * sampling rate convert. Then, SRC should be FE. - * And then, this function will be called *after* BE settings. - * this means, each BE already has fixuped hw_params. - * see - * dpcm_fe_dai_hw_params() - * dpcm_be_dai_hw_params() - */ - src->convert_rate = 0; - if (fe->dai_link->dynamic) { - int stream = substream->stream; - struct snd_soc_dpcm *dpcm; - struct snd_pcm_hw_params *be_params; - - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { - be_params = &dpcm->hw_params; - - if (params_rate(fe_params) != params_rate(be_params)) - src->convert_rate = params_rate(be_params); - } - } - - return 0; -} - static void rsnd_src_set_convert_rate(struct rsnd_dai_stream *io, struct rsnd_mod *mod) { @@ -349,9 +314,8 @@ static bool rsnd_src_error_occurred(struct rsnd_mod *mod) status0 = rsnd_mod_read(mod, SCU_SYS_STATUS0); status1 = rsnd_mod_read(mod, SCU_SYS_STATUS1); if ((status0 & val0) || (status1 & val1)) { - rsnd_dbg_irq_status(dev, "%s[%d] err status : 0x%08x, 0x%08x\n", - rsnd_mod_name(mod), rsnd_mod_id(mod), - status0, status1); + rsnd_dbg_irq_status(dev, "%s err status : 0x%08x, 0x%08x\n", + rsnd_mod_name(mod), status0, status1); ret = true; } @@ -527,16 +491,16 @@ static int rsnd_src_pcm_new(struct rsnd_mod *mod, } static struct rsnd_mod_ops rsnd_src_ops = { - .name = SRC_NAME, - .dma_req = rsnd_src_dma_req, - .probe = rsnd_src_probe_, - .init = rsnd_src_init, - .quit = rsnd_src_quit, - .start = rsnd_src_start, - .stop = rsnd_src_stop, - .irq = rsnd_src_irq, - .hw_params = rsnd_src_hw_params, - .pcm_new = rsnd_src_pcm_new, + .name = SRC_NAME, + .dma_req = rsnd_src_dma_req, + .probe = rsnd_src_probe_, + .init = rsnd_src_init, + .quit = rsnd_src_quit, + .start = rsnd_src_start, + .stop = rsnd_src_stop, + .irq = rsnd_src_irq, + .pcm_new = rsnd_src_pcm_new, + .get_status = rsnd_mod_get_status, }; struct rsnd_mod *rsnd_src_mod_get(struct rsnd_priv *priv, int id) @@ -605,8 +569,7 @@ int rsnd_src_probe(struct rsnd_priv *priv) } ret = rsnd_mod_init(priv, rsnd_mod_get(src), - &rsnd_src_ops, clk, rsnd_mod_get_status, - RSND_MOD_SRC, i); + &rsnd_src_ops, clk, RSND_MOD_SRC, i); if (ret) { of_node_put(np); goto rsnd_src_probe_done; diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 3f880ec66459..f5afab631abb 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -42,7 +42,13 @@ #define DWL_24 (5 << 19) /* Data Word Length */ #define DWL_32 (6 << 19) /* Data Word Length */ +/* + * System word length + */ +#define SWL_16 (1 << 16) /* R/W System Word Length */ +#define SWL_24 (2 << 16) /* R/W System 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 */ @@ -72,7 +78,6 @@ struct rsnd_ssi { struct rsnd_mod mod; - struct rsnd_mod *dma; u32 flags; u32 cr_own; @@ -94,9 +99,7 @@ struct rsnd_ssi { /* flags */ #define RSND_SSI_CLK_PIN_SHARE (1 << 0) #define RSND_SSI_NO_BUSIF (1 << 1) /* SSI+DMA without BUSIF */ -#define RSND_SSI_HDMI0 (1 << 2) /* for HDMI0 */ -#define RSND_SSI_HDMI1 (1 << 3) /* for HDMI1 */ -#define RSND_SSI_PROBED (1 << 4) +#define RSND_SSI_PROBED (1 << 2) #define for_each_rsnd_ssi(pos, priv, i) \ for (i = 0; \ @@ -114,19 +117,7 @@ struct rsnd_ssi { (rsnd_ssi_run_mods(io) & (1 << rsnd_mod_id(mod))) #define rsnd_ssi_can_output_clk(mod) (!__rsnd_ssi_is_pin_sharing(mod)) -int rsnd_ssi_hdmi_port(struct rsnd_dai_stream *io) -{ - struct rsnd_mod *mod = rsnd_io_to_mod_ssi(io); - struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); - - if (rsnd_flags_has(ssi, RSND_SSI_HDMI0)) - return RSND_SSI_HDMI_PORT0; - - if (rsnd_flags_has(ssi, RSND_SSI_HDMI1)) - return RSND_SSI_HDMI_PORT1; - - return 0; -} +static int rsnd_ssi_is_dma_mode(struct rsnd_mod *mod); int rsnd_ssi_use_busif(struct rsnd_dai_stream *io) { @@ -171,8 +162,7 @@ static void rsnd_ssi_status_check(struct rsnd_mod *mod, udelay(5); } - dev_warn(dev, "%s[%d] status check failed\n", - rsnd_mod_name(mod), rsnd_mod_id(mod)); + dev_warn(dev, "%s status check failed\n", rsnd_mod_name(mod)); } static u32 rsnd_ssi_multi_slaves(struct rsnd_dai_stream *io) @@ -214,20 +204,38 @@ static u32 rsnd_ssi_run_mods(struct rsnd_dai_stream *io) u32 rsnd_ssi_multi_slaves_runtime(struct rsnd_dai_stream *io) { - if (rsnd_runtime_is_ssi_multi(io)) + if (rsnd_runtime_is_multi_ssi(io)) return rsnd_ssi_multi_slaves(io); return 0; } -unsigned int rsnd_ssi_clk_query(struct rsnd_priv *priv, +static u32 rsnd_rdai_width_to_swl(struct rsnd_dai *rdai) +{ + struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device *dev = rsnd_priv_to_dev(priv); + int width = rsnd_rdai_width_get(rdai); + + switch (width) { + case 32: return SWL_32; + case 24: return SWL_24; + case 16: return SWL_16; + } + + dev_err(dev, "unsupported slot width value: %d\n", width); + return 0; +} + +unsigned int rsnd_ssi_clk_query(struct rsnd_dai *rdai, int param1, int param2, int *idx) { + struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); int ssi_clk_mul_table[] = { 1, 2, 4, 8, 16, 6, 12, }; int j, ret; unsigned int main_rate; + int width = rsnd_rdai_width_get(rdai); for (j = 0; j < ARRAY_SIZE(ssi_clk_mul_table); j++) { @@ -240,12 +248,7 @@ unsigned int rsnd_ssi_clk_query(struct rsnd_priv *priv, if (j == 0) continue; - /* - * this driver is assuming that - * system word is 32bit x chan - * see rsnd_ssi_init() - */ - main_rate = 32 * param1 * param2 * ssi_clk_mul_table[j]; + main_rate = width * param1 * param2 * ssi_clk_mul_table[j]; ret = rsnd_adg_clk_query(priv, main_rate); if (ret < 0) @@ -283,16 +286,24 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, if (rsnd_ssi_is_multi_slave(mod, io)) return 0; - if (ssi->rate) { + if (ssi->usrcnt > 0) { if (ssi->rate != rate) { dev_err(dev, "SSI parent/child should use same rate\n"); return -EINVAL; } + if (ssi->chan != chan) { + dev_err(dev, "SSI parent/child should use same chan\n"); + return -EINVAL; + } + return 0; } - main_rate = rsnd_ssi_clk_query(priv, rate, chan, &idx); + if (rsnd_runtime_is_tdm_split(io)) + chan = rsnd_io_converted_chan(io); + + main_rate = rsnd_ssi_clk_query(rdai, rate, chan, &idx); if (!main_rate) { dev_err(dev, "unsupported clock rate\n"); return -EIO; @@ -312,13 +323,14 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, * SSICR : FORCE, SCKD, SWSD * SSIWSR : CONT */ - ssi->cr_clk = FORCE | SWL_32 | SCKD | SWSD | CKDV(idx); + ssi->cr_clk = FORCE | rsnd_rdai_width_to_swl(rdai) | + SCKD | SWSD | CKDV(idx); ssi->wsr = CONT; ssi->rate = rate; + ssi->chan = chan; - dev_dbg(dev, "%s[%d] outputs %u Hz\n", - rsnd_mod_name(mod), - rsnd_mod_id(mod), rate); + dev_dbg(dev, "%s outputs %d chan %u Hz\n", + rsnd_mod_name(mod), chan, rate); return 0; } @@ -340,6 +352,7 @@ static void rsnd_ssi_master_clk_stop(struct rsnd_mod *mod, ssi->cr_clk = 0; ssi->rate = 0; + ssi->chan = 0; rsnd_adg_ssi_clk_stop(mod); } @@ -348,24 +361,29 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, struct rsnd_dai_stream *io) { struct rsnd_dai *rdai = rsnd_io_to_rdai(io); + struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device *dev = rsnd_priv_to_dev(priv); struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); u32 cr_own = ssi->cr_own; u32 cr_mode = ssi->cr_mode; u32 wsr = ssi->wsr; - int is_tdm; + int width; + int is_tdm, is_tdm_split; - is_tdm = rsnd_runtime_is_ssi_tdm(io); + is_tdm = rsnd_runtime_is_tdm(io); + is_tdm_split = rsnd_runtime_is_tdm_split(io); - /* - * always use 32bit system word. - * see also rsnd_ssi_master_clk_enable() - */ - cr_own |= FORCE | SWL_32; + if (is_tdm) + dev_dbg(dev, "TDM mode\n"); + if (is_tdm_split) + dev_dbg(dev, "TDM Split mode\n"); + + cr_own |= FORCE | rsnd_rdai_width_to_swl(rdai); if (rdai->bit_clk_inv) cr_own |= SCKP; - if (rdai->frm_clk_inv ^ is_tdm) + if (rdai->frm_clk_inv && !is_tdm) cr_own |= SWSP; if (rdai->data_alignment) cr_own |= SDTA; @@ -373,6 +391,17 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, cr_own |= DEL; /* + * TDM Mode + * see + * rsnd_ssiu_init_gen2() + */ + wsr = ssi->wsr; + if (is_tdm || is_tdm_split) { + wsr |= WS_MODE; + cr_own |= CHNL_8; + } + + /* * We shouldn't exchange SWSP after running. * This means, parent needs to care it. */ @@ -383,13 +412,30 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, cr_own |= TRMD; cr_own &= ~DWL_MASK; - switch (snd_pcm_format_width(runtime->format)) { + width = snd_pcm_format_width(runtime->format); + if (is_tdm_split) { + /* + * The SWL and DWL bits in SSICR should be fixed at 32-bit + * setting when TDM split mode. + * see datasheet + * Operation :: TDM Format Split Function (TDM Split Mode) + */ + width = 32; + } + + switch (width) { + case 8: + cr_own |= DWL_8; + break; case 16: cr_own |= DWL_16; break; case 24: cr_own |= DWL_24; break; + case 32: + cr_own |= DWL_32; + break; } if (rsnd_ssi_is_dma_mode(mod)) { @@ -399,16 +445,6 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, cr_mode = DIEN; /* PIO : enable Data interrupt */ } - /* - * TDM Extend Mode - * see - * rsnd_ssiu_init_gen2() - */ - wsr = ssi->wsr; - if (is_tdm) { - wsr |= WS_MODE; - cr_own |= CHNL_8; - } init_end: ssi->cr_own = cr_own; ssi->cr_mode = cr_mode; @@ -463,8 +499,7 @@ static int rsnd_ssi_quit(struct rsnd_mod *mod, return 0; if (!ssi->usrcnt) { - dev_err(dev, "%s[%d] usrcnt error\n", - rsnd_mod_name(mod), rsnd_mod_id(mod)); + dev_err(dev, "%s usrcnt error\n", rsnd_mod_name(mod)); return -EIO; } @@ -488,26 +523,16 @@ static int rsnd_ssi_hw_params(struct rsnd_mod *mod, struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); - int chan = params_channels(params); + struct rsnd_dai *rdai = rsnd_io_to_rdai(io); + unsigned int fmt_width = snd_pcm_format_width(params_format(params)); - /* - * snd_pcm_ops::hw_params will be called *before* - * snd_soc_dai_ops::trigger. Thus, ssi->usrcnt is 0 - * in 1st call. - */ - if (ssi->usrcnt) { - /* - * Already working. - * It will happen if SSI has parent/child connection. - * it is error if child <-> parent SSI uses - * different channels. - */ - if (ssi->chan != chan) - return -EIO; - } + if (fmt_width > rdai->chan_width) { + struct rsnd_priv *priv = rsnd_io_to_priv(io); + struct device *dev = rsnd_priv_to_dev(priv); - ssi->chan = chan; + dev_err(dev, "invalid combination of slot-width and format-data-width\n"); + return -EINVAL; + } return 0; } @@ -633,8 +658,8 @@ static void __rsnd_ssi_interrupt(struct rsnd_mod *mod, /* DMA only */ if (is_dma && (status & (UIRQ | OIRQ))) { - rsnd_dbg_irq_status(dev, "%s[%d] err status : 0x%08x\n", - rsnd_mod_name(mod), rsnd_mod_id(mod), status); + rsnd_dbg_irq_status(dev, "%s err status : 0x%08x\n", + rsnd_mod_name(mod), status); stop = true; } @@ -660,6 +685,41 @@ static irqreturn_t rsnd_ssi_interrupt(int irq, void *data) return IRQ_HANDLED; } +static u32 *rsnd_ssi_get_status(struct rsnd_mod *mod, + struct rsnd_dai_stream *io, + enum rsnd_mod_type type) +{ + /* + * SSIP (= SSI parent) needs to be special, otherwise, + * 2nd SSI might doesn't start. see also rsnd_mod_call() + * + * We can't include parent SSI status on SSI, because we don't know + * how many SSI requests parent SSI. Thus, it is localed on "io" now. + * ex) trouble case + * Playback: SSI0 + * Capture : SSI1 (needs SSI0) + * + * 1) start Capture -> SSI0/SSI1 are started. + * 2) start Playback -> SSI0 doesn't work, because it is already + * marked as "started" on 1) + * + * OTOH, using each mod's status is good for MUX case. + * It doesn't need to start in 2nd start + * ex) + * IO-0: SRC0 -> CTU1 -+-> MUX -> DVC -> SSIU -> SSI0 + * | + * IO-1: SRC1 -> CTU2 -+ + * + * 1) start IO-0 -> start SSI0 + * 2) start IO-1 -> SSI0 doesn't need to start, because it is + * already started on 1) + */ + if (type == RSND_MOD_SSIP) + return &io->parent_ssi_status; + + return rsnd_mod_get_status(mod, io, type); +} + /* * SSI PIO */ @@ -709,7 +769,7 @@ static int rsnd_ssi_common_probe(struct rsnd_mod *mod, { struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); - int ret; + int ret = 0; /* * SSIP/SSIU/IRQ are not needed on @@ -723,10 +783,6 @@ static int rsnd_ssi_common_probe(struct rsnd_mod *mod, * see rsnd_ssi_pcm_new() */ - ret = rsnd_ssiu_attach(io, mod); - if (ret < 0) - return ret; - /* * SSI might be called again as PIO fallback * It is easy to manual handling for IRQ request/free @@ -855,25 +911,25 @@ static int rsnd_ssi_prepare(struct rsnd_mod *mod, } static struct rsnd_mod_ops rsnd_ssi_pio_ops = { - .name = SSI_NAME, - .probe = rsnd_ssi_common_probe, - .remove = rsnd_ssi_common_remove, - .init = rsnd_ssi_pio_init, - .quit = rsnd_ssi_quit, - .start = rsnd_ssi_start, - .stop = rsnd_ssi_stop, - .irq = rsnd_ssi_irq, - .pointer = rsnd_ssi_pio_pointer, - .pcm_new = rsnd_ssi_pcm_new, - .hw_params = rsnd_ssi_hw_params, - .prepare = rsnd_ssi_prepare, + .name = SSI_NAME, + .probe = rsnd_ssi_common_probe, + .remove = rsnd_ssi_common_remove, + .init = rsnd_ssi_pio_init, + .quit = rsnd_ssi_quit, + .start = rsnd_ssi_start, + .stop = rsnd_ssi_stop, + .irq = rsnd_ssi_irq, + .pointer = rsnd_ssi_pio_pointer, + .pcm_new = rsnd_ssi_pcm_new, + .hw_params = rsnd_ssi_hw_params, + .prepare = rsnd_ssi_prepare, + .get_status = rsnd_ssi_get_status, }; static int rsnd_ssi_dma_probe(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); int ret; /* @@ -888,7 +944,7 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod, return ret; /* SSI probe might be called many times in MUX multi path */ - ret = rsnd_dma_attach(io, mod, &ssi->dma); + ret = rsnd_dma_attach(io, mod, &io->dma); return ret; } @@ -908,8 +964,7 @@ static int rsnd_ssi_fallback(struct rsnd_mod *mod, */ mod->ops = &rsnd_ssi_pio_ops; - dev_info(dev, "%s[%d] fallback to PIO mode\n", - rsnd_mod_name(mod), rsnd_mod_id(mod)); + dev_info(dev, "%s fallback to PIO mode\n", rsnd_mod_name(mod)); return 0; } @@ -921,6 +976,17 @@ static struct dma_chan *rsnd_ssi_dma_req(struct rsnd_dai_stream *io, int is_play = rsnd_io_is_play(io); char *name; + /* + * It should use "rcar_sound,ssiu" on DT. + * But, we need to keep compatibility for old version. + * + * If it has "rcar_sound.ssiu", it will be used. + * If not, "rcar_sound.ssi" will be used. + * see + * rsnd_ssiu_dma_req() + * rsnd_dma_of_path() + */ + if (rsnd_ssi_use_busif(io)) name = is_play ? "rxu" : "txu"; else @@ -931,27 +997,27 @@ static struct dma_chan *rsnd_ssi_dma_req(struct rsnd_dai_stream *io, } static struct rsnd_mod_ops rsnd_ssi_dma_ops = { - .name = SSI_NAME, - .dma_req = rsnd_ssi_dma_req, - .probe = rsnd_ssi_dma_probe, - .remove = rsnd_ssi_common_remove, - .init = rsnd_ssi_init, - .quit = rsnd_ssi_quit, - .start = rsnd_ssi_start, - .stop = rsnd_ssi_stop, - .irq = rsnd_ssi_irq, - .pcm_new = rsnd_ssi_pcm_new, - .fallback = rsnd_ssi_fallback, - .hw_params = rsnd_ssi_hw_params, - .prepare = rsnd_ssi_prepare, + .name = SSI_NAME, + .dma_req = rsnd_ssi_dma_req, + .probe = rsnd_ssi_dma_probe, + .remove = rsnd_ssi_common_remove, + .init = rsnd_ssi_init, + .quit = rsnd_ssi_quit, + .start = rsnd_ssi_start, + .stop = rsnd_ssi_stop, + .irq = rsnd_ssi_irq, + .pcm_new = rsnd_ssi_pcm_new, + .fallback = rsnd_ssi_fallback, + .hw_params = rsnd_ssi_hw_params, + .prepare = rsnd_ssi_prepare, + .get_status = rsnd_ssi_get_status, }; -int rsnd_ssi_is_dma_mode(struct rsnd_mod *mod) +static int rsnd_ssi_is_dma_mode(struct rsnd_mod *mod) { return mod->ops == &rsnd_ssi_dma_ops; } - /* * ssi mod function */ @@ -1007,54 +1073,6 @@ void rsnd_parse_connect_ssi(struct rsnd_dai *rdai, of_node_put(node); } -static void __rsnd_ssi_parse_hdmi_connection(struct rsnd_priv *priv, - struct rsnd_dai_stream *io, - struct device_node *remote_ep) -{ - struct device *dev = rsnd_priv_to_dev(priv); - struct rsnd_mod *mod = rsnd_io_to_mod_ssi(io); - struct rsnd_ssi *ssi; - struct device_node *remote_node = of_graph_get_port_parent(remote_ep); - - /* support Gen3 only */ - if (!rsnd_is_gen3(priv)) - return; - - if (!mod) - return; - - ssi = rsnd_mod_to_ssi(mod); - - /* HDMI0 */ - if (strstr(remote_node->full_name, "hdmi@fead0000")) { - rsnd_flags_set(ssi, RSND_SSI_HDMI0); - dev_dbg(dev, "%s[%d] connected to HDMI0\n", - rsnd_mod_name(mod), rsnd_mod_id(mod)); - } - - /* HDMI1 */ - if (strstr(remote_node->full_name, "hdmi@feae0000")) { - rsnd_flags_set(ssi, RSND_SSI_HDMI1); - dev_dbg(dev, "%s[%d] connected to HDMI1\n", - rsnd_mod_name(mod), rsnd_mod_id(mod)); - } -} - -void rsnd_ssi_parse_hdmi_connection(struct rsnd_priv *priv, - struct device_node *endpoint, - int dai_i) -{ - struct rsnd_dai *rdai = rsnd_rdai_get(priv, dai_i); - struct device_node *remote_ep; - - remote_ep = of_graph_get_remote_endpoint(endpoint); - if (!remote_ep) - return; - - __rsnd_ssi_parse_hdmi_connection(priv, &rdai->playback, remote_ep); - __rsnd_ssi_parse_hdmi_connection(priv, &rdai->capture, remote_ep); -} - struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id) { if (WARN_ON(id < 0 || id >= rsnd_ssi_nr(priv))) @@ -1071,41 +1089,6 @@ int __rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod) return !!(rsnd_flags_has(rsnd_mod_to_ssi(mod), RSND_SSI_CLK_PIN_SHARE)); } -static u32 *rsnd_ssi_get_status(struct rsnd_dai_stream *io, - struct rsnd_mod *mod, - enum rsnd_mod_type type) -{ - /* - * SSIP (= SSI parent) needs to be special, otherwise, - * 2nd SSI might doesn't start. see also rsnd_mod_call() - * - * We can't include parent SSI status on SSI, because we don't know - * how many SSI requests parent SSI. Thus, it is localed on "io" now. - * ex) trouble case - * Playback: SSI0 - * Capture : SSI1 (needs SSI0) - * - * 1) start Capture -> SSI0/SSI1 are started. - * 2) start Playback -> SSI0 doesn't work, because it is already - * marked as "started" on 1) - * - * OTOH, using each mod's status is good for MUX case. - * It doesn't need to start in 2nd start - * ex) - * IO-0: SRC0 -> CTU1 -+-> MUX -> DVC -> SSIU -> SSI0 - * | - * IO-1: SRC1 -> CTU2 -+ - * - * 1) start IO-0 -> start SSI0 - * 2) start IO-1 -> SSI0 doesn't need to start, because it is - * already started on 1) - */ - if (type == RSND_MOD_SSIP) - return &io->parent_ssi_status; - - return rsnd_mod_get_status(io, mod, type); -} - int rsnd_ssi_probe(struct rsnd_priv *priv) { struct device_node *node; @@ -1172,7 +1155,7 @@ int rsnd_ssi_probe(struct rsnd_priv *priv) ops = &rsnd_ssi_dma_ops; ret = rsnd_mod_init(priv, rsnd_mod_get(ssi), ops, clk, - rsnd_ssi_get_status, RSND_MOD_SSI, i); + RSND_MOD_SSI, i); if (ret) { of_node_put(np); goto rsnd_ssi_probe_done; diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 016fbf5ac242..c74991dd18ab 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -10,15 +10,51 @@ struct rsnd_ssiu { struct rsnd_mod mod; + u32 busif_status[8]; /* for BUSIF0 - BUSIF7 */ + unsigned int usrcnt; + int id; + int id_sub; }; +/* SSI_MODE */ +#define TDM_EXT (1 << 0) +#define TDM_SPLIT (1 << 8) + #define rsnd_ssiu_nr(priv) ((priv)->ssiu_nr) +#define rsnd_mod_to_ssiu(_mod) container_of((_mod), struct rsnd_ssiu, mod) #define for_each_rsnd_ssiu(pos, priv, i) \ for (i = 0; \ (i < rsnd_ssiu_nr(priv)) && \ ((pos) = ((struct rsnd_ssiu *)(priv)->ssiu + i)); \ i++) +/* + * SSI Gen2 Gen3 + * 0 BUSIF0-3 BUSIF0-7 + * 1 BUSIF0-3 BUSIF0-7 + * 2 BUSIF0-3 BUSIF0-7 + * 3 BUSIF0 BUSIF0-7 + * 4 BUSIF0 BUSIF0-7 + * 5 BUSIF0 BUSIF0 + * 6 BUSIF0 BUSIF0 + * 7 BUSIF0 BUSIF0 + * 8 BUSIF0 BUSIF0 + * 9 BUSIF0-3 BUSIF0-7 + * total 22 52 + */ +static const int gen2_id[] = { 0, 4, 8, 12, 13, 14, 15, 16, 17, 18 }; +static const int gen3_id[] = { 0, 8, 16, 24, 32, 40, 41, 42, 43, 44 }; + +static u32 *rsnd_ssiu_get_status(struct rsnd_mod *mod, + struct rsnd_dai_stream *io, + enum rsnd_mod_type type) +{ + struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod); + int busif = rsnd_mod_id_sub(mod); + + return &ssiu->busif_status[busif]; +} + static int rsnd_ssiu_init(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) @@ -29,6 +65,7 @@ static int rsnd_ssiu_init(struct rsnd_mod *mod, int id = rsnd_mod_id(mod); u32 mask1, val1; u32 mask2, val2; + int i; /* clear status */ switch (id) { @@ -37,16 +74,12 @@ static int rsnd_ssiu_init(struct rsnd_mod *mod, case 2: case 3: case 4: - rsnd_mod_write(mod, SSI_SYS_STATUS0, 0xf << (id * 4)); - rsnd_mod_write(mod, SSI_SYS_STATUS2, 0xf << (id * 4)); - rsnd_mod_write(mod, SSI_SYS_STATUS4, 0xf << (id * 4)); - rsnd_mod_write(mod, SSI_SYS_STATUS6, 0xf << (id * 4)); + for (i = 0; i < 4; i++) + rsnd_mod_write(mod, SSI_SYS_STATUS(i * 2), 0xf << (id * 4)); break; case 9: - rsnd_mod_write(mod, SSI_SYS_STATUS1, 0xf << 4); - rsnd_mod_write(mod, SSI_SYS_STATUS3, 0xf << 4); - rsnd_mod_write(mod, SSI_SYS_STATUS5, 0xf << 4); - rsnd_mod_write(mod, SSI_SYS_STATUS7, 0xf << 4); + for (i = 0; i < 4; i++) + rsnd_mod_write(mod, SSI_SYS_STATUS((i * 2) + 1), 0xf << 4); break; } @@ -112,15 +145,18 @@ static int rsnd_ssiu_init(struct rsnd_mod *mod, } static struct rsnd_mod_ops rsnd_ssiu_ops_gen1 = { - .name = SSIU_NAME, - .init = rsnd_ssiu_init, + .name = SSIU_NAME, + .init = rsnd_ssiu_init, + .get_status = rsnd_ssiu_get_status, }; static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - int hdmi = rsnd_ssi_hdmi_port(io); + struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod); + u32 has_hdmi0 = rsnd_flags_has(io, RSND_STREAM_HDMI0); + u32 has_hdmi1 = rsnd_flags_has(io, RSND_STREAM_HDMI1); int ret; u32 mode = 0; @@ -128,30 +164,49 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, if (ret < 0) return ret; - if (rsnd_runtime_is_ssi_tdm(io)) { - /* - * TDM Extend Mode - * see - * rsnd_ssi_config_init() - */ - mode = 0x1; - } + ssiu->usrcnt++; + + /* + * TDM Extend/Split Mode + * see + * rsnd_ssi_config_init() + */ + if (rsnd_runtime_is_tdm(io)) + mode = TDM_EXT; + else if (rsnd_runtime_is_tdm_split(io)) + mode = TDM_SPLIT; rsnd_mod_write(mod, SSI_MODE, mode); if (rsnd_ssi_use_busif(io)) { - rsnd_mod_write(mod, SSI_BUSIF_ADINR, + int id = rsnd_mod_id(mod); + int busif = rsnd_mod_id_sub(mod); + + /* + * FIXME + * + * We can't support SSI9-4/5/6/7, because its address is + * out of calculation rule + */ + if ((id == 9) && (busif >= 4)) { + struct device *dev = rsnd_priv_to_dev(priv); + + dev_err(dev, "This driver doesn't support SSI%d-%d, so far", + id, busif); + } + + rsnd_mod_write(mod, SSI_BUSIF_ADINR(busif), rsnd_get_adinr_bit(mod, io) | (rsnd_io_is_play(io) ? rsnd_runtime_channel_after_ctu(io) : rsnd_runtime_channel_original(io))); - rsnd_mod_write(mod, SSI_BUSIF_MODE, + rsnd_mod_write(mod, SSI_BUSIF_MODE(busif), rsnd_get_busif_shift(io, mod) | 1); - rsnd_mod_write(mod, SSI_BUSIF_DALIGN, + rsnd_mod_write(mod, SSI_BUSIF_DALIGN(busif), rsnd_get_dalign(mod, io)); } - if (hdmi) { + if (has_hdmi0 || has_hdmi1) { enum rsnd_mod_type rsnd_ssi_array[] = { RSND_MOD_SSIM1, RSND_MOD_SSIM2, @@ -177,14 +232,10 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, rsnd_mod_id(pos) << shift; } - switch (hdmi) { - case RSND_SSI_HDMI_PORT0: + if (has_hdmi0) rsnd_mod_write(mod, HDMI0_SEL, val); - break; - case RSND_SSI_HDMI_PORT1: + if (has_hdmi1) rsnd_mod_write(mod, HDMI1_SEL, val); - break; - } } return 0; @@ -194,10 +245,12 @@ static int rsnd_ssiu_start_gen2(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { + int busif = rsnd_mod_id_sub(mod); + if (!rsnd_ssi_use_busif(io)) return 0; - rsnd_mod_write(mod, SSI_CTRL, 0x1); + rsnd_mod_bset(mod, SSI_CTRL, 1 << (busif * 4), 1 << (busif * 4)); if (rsnd_ssi_multi_slaves_runtime(io)) rsnd_mod_write(mod, SSI_CONTROL, 0x1); @@ -209,10 +262,16 @@ static int rsnd_ssiu_stop_gen2(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { + struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod); + int busif = rsnd_mod_id_sub(mod); + if (!rsnd_ssi_use_busif(io)) return 0; - rsnd_mod_write(mod, SSI_CTRL, 0); + rsnd_mod_bset(mod, SSI_CTRL, 1 << (busif * 4), 0); + + if (--ssiu->usrcnt) + return 0; if (rsnd_ssi_multi_slaves_runtime(io)) rsnd_mod_write(mod, SSI_CONTROL, 0); @@ -220,11 +279,53 @@ static int rsnd_ssiu_stop_gen2(struct rsnd_mod *mod, return 0; } +static int rsnd_ssiu_id(struct rsnd_mod *mod) +{ + struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod); + + /* see rsnd_ssiu_probe() */ + return ssiu->id; +} + +static int rsnd_ssiu_id_sub(struct rsnd_mod *mod) +{ + struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod); + + /* see rsnd_ssiu_probe() */ + return ssiu->id_sub; +} + +static struct dma_chan *rsnd_ssiu_dma_req(struct rsnd_dai_stream *io, + struct rsnd_mod *mod) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + int is_play = rsnd_io_is_play(io); + char *name; + + /* + * It should use "rcar_sound,ssiu" on DT. + * But, we need to keep compatibility for old version. + * + * If it has "rcar_sound.ssiu", it will be used. + * If not, "rcar_sound.ssi" will be used. + * see + * rsnd_ssi_dma_req() + * rsnd_dma_of_path() + */ + + name = is_play ? "rx" : "tx"; + + return rsnd_dma_request_channel(rsnd_ssiu_of_node(priv), + mod, name); +} + static struct rsnd_mod_ops rsnd_ssiu_ops_gen2 = { - .name = SSIU_NAME, - .init = rsnd_ssiu_init_gen2, - .start = rsnd_ssiu_start_gen2, - .stop = rsnd_ssiu_stop_gen2, + .name = SSIU_NAME, + .dma_req = rsnd_ssiu_dma_req, + .init = rsnd_ssiu_init_gen2, + .start = rsnd_ssiu_start_gen2, + .stop = rsnd_ssiu_stop_gen2, + .get_status = rsnd_ssiu_get_status, }; static struct rsnd_mod *rsnd_ssiu_mod_get(struct rsnd_priv *priv, int id) @@ -235,26 +336,85 @@ static struct rsnd_mod *rsnd_ssiu_mod_get(struct rsnd_priv *priv, int id) return rsnd_mod_get((struct rsnd_ssiu *)(priv->ssiu) + id); } -int rsnd_ssiu_attach(struct rsnd_dai_stream *io, - struct rsnd_mod *ssi_mod) +static void rsnd_parse_connect_ssiu_compatible(struct rsnd_priv *priv, + struct rsnd_dai_stream *io) +{ + struct rsnd_mod *ssi_mod = rsnd_io_to_mod_ssi(io); + struct rsnd_mod *mod; + struct rsnd_ssiu *ssiu; + int i; + + if (!ssi_mod) + return; + + /* select BUSIF0 */ + for_each_rsnd_ssiu(ssiu, priv, i) { + mod = rsnd_mod_get(ssiu); + + if ((rsnd_mod_id(ssi_mod) == rsnd_mod_id(mod)) && + (rsnd_mod_id_sub(mod) == 0)) { + rsnd_dai_connect(mod, io, mod->type); + return; + } + } +} + +void rsnd_parse_connect_ssiu(struct rsnd_dai *rdai, + struct device_node *playback, + struct device_node *capture) { - struct rsnd_priv *priv = rsnd_io_to_priv(io); - struct rsnd_mod *mod = rsnd_ssiu_mod_get(priv, rsnd_mod_id(ssi_mod)); + struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device_node *node = rsnd_ssiu_of_node(priv); + struct device_node *np; + struct rsnd_mod *mod; + struct rsnd_dai_stream *io_p = &rdai->playback; + struct rsnd_dai_stream *io_c = &rdai->capture; + int i; - rsnd_mod_confirm_ssi(ssi_mod); + /* use rcar_sound,ssiu if exist */ + if (node) { + i = 0; + for_each_child_of_node(node, np) { + mod = rsnd_ssiu_mod_get(priv, i); + if (np == playback) + rsnd_dai_connect(mod, io_p, mod->type); + if (np == capture) + rsnd_dai_connect(mod, io_c, mod->type); + i++; + } + + of_node_put(node); + } - return rsnd_dai_connect(mod, io, mod->type); + /* Keep DT compatibility */ + if (!rsnd_io_to_mod_ssiu(io_p)) + rsnd_parse_connect_ssiu_compatible(priv, io_p); + if (!rsnd_io_to_mod_ssiu(io_c)) + rsnd_parse_connect_ssiu_compatible(priv, io_c); } int rsnd_ssiu_probe(struct rsnd_priv *priv) { struct device *dev = rsnd_priv_to_dev(priv); + struct device_node *node; struct rsnd_ssiu *ssiu; struct rsnd_mod_ops *ops; + const int *list = NULL; int i, nr, ret; - /* same number to SSI */ - nr = priv->ssi_nr; + /* + * Keep DT compatibility. + * if it has "rcar_sound,ssiu", use it. + * if not, use "rcar_sound,ssi" + * see + * rsnd_ssiu_bufsif_to_id() + */ + node = rsnd_ssiu_of_node(priv); + if (node) + nr = of_get_child_count(node); + else + nr = priv->ssi_nr; + ssiu = devm_kcalloc(dev, nr, sizeof(*ssiu), GFP_KERNEL); if (!ssiu) return -ENOMEM; @@ -267,10 +427,46 @@ int rsnd_ssiu_probe(struct rsnd_priv *priv) else ops = &rsnd_ssiu_ops_gen2; + /* Keep compatibility */ + nr = 0; + if ((node) && + (ops == &rsnd_ssiu_ops_gen2)) { + ops->id = rsnd_ssiu_id; + ops->id_sub = rsnd_ssiu_id_sub; + + if (rsnd_is_gen2(priv)) { + list = gen2_id; + nr = ARRAY_SIZE(gen2_id); + } else if (rsnd_is_gen3(priv)) { + list = gen3_id; + nr = ARRAY_SIZE(gen3_id); + } else { + dev_err(dev, "unknown SSIU\n"); + return -ENODEV; + } + } + for_each_rsnd_ssiu(ssiu, priv, i) { + if (node) { + int j; + + /* + * see + * rsnd_ssiu_get_id() + * rsnd_ssiu_get_id_sub() + */ + for (j = 0; j < nr; j++) { + if (list[j] > i) + break; + ssiu->id = j; + ssiu->id_sub = i - list[ssiu->id]; + } + } else { + ssiu->id = i; + } + ret = rsnd_mod_init(priv, rsnd_mod_get(ssiu), - ops, NULL, rsnd_mod_get_status, - RSND_MOD_SSIU, i); + ops, NULL, RSND_MOD_SSIU, i); if (ret) return ret; } diff --git a/sound/soc/soc-acpi.c b/sound/soc/soc-acpi.c index b8e72b52db30..4fb29f0e561e 100644 --- a/sound/soc/soc-acpi.c +++ b/sound/soc/soc-acpi.c @@ -10,11 +10,17 @@ struct snd_soc_acpi_mach * snd_soc_acpi_find_machine(struct snd_soc_acpi_mach *machines) { struct snd_soc_acpi_mach *mach; + struct snd_soc_acpi_mach *mach_alt; for (mach = machines; mach->id[0]; mach++) { if (acpi_dev_present(mach->id, NULL, -1)) { - if (mach->machine_quirk) - mach = mach->machine_quirk(mach); + if (mach->machine_quirk) { + mach_alt = mach->machine_quirk(mach); + if (!mach_alt) + continue; /* not full match, ignore */ + mach = mach_alt; + } + return mach; } } diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 409d082e80d1..699397a09167 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -157,7 +157,7 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) ret = dpcm_be_dai_startup(fe, stream); if (ret < 0) { /* clean up all links */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) + for_each_dpcm_be(fe, stream, dpcm) dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; dpcm_be_disconnect(fe, stream); @@ -321,7 +321,7 @@ static int soc_compr_free_fe(struct snd_compr_stream *cstream) ret = dpcm_be_dai_shutdown(fe, stream); /* mark FE's links ready to prune */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) + for_each_dpcm_be(fe, stream, dpcm) dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 473eefe8658e..50617db05c46 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -52,6 +52,10 @@ EXPORT_SYMBOL_GPL(snd_soc_debugfs_root); static DEFINE_MUTEX(client_mutex); static LIST_HEAD(component_list); +static LIST_HEAD(unbind_card_list); + +#define for_each_component(component) \ + list_for_each_entry(component, &component_list, list) /* * This is a timeout to do a DAPM powerdown after a stream is closed(). @@ -62,8 +66,9 @@ static int pmdown_time = 5000; module_param(pmdown_time, int, 0); MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)"); -/* If a DMI filed contain strings in this blacklist (e.g. - * "Type2 - Board Manufacturer" or "Type1 - TBD by OEM"), it will be taken +/* + * If a DMI filed contain strings in this blacklist (e.g. + * "Type2 - Board Manufacturer" or "Type1 - TBD by OEM"), it will be taken * as invalid and dropped when setting the card long name from DMI info. */ static const char * const dmi_blacklist[] = { @@ -175,8 +180,8 @@ static int dai_list_show(struct seq_file *m, void *v) mutex_lock(&client_mutex); - list_for_each_entry(component, &component_list, list) - list_for_each_entry(dai, &component->dai_list, list) + for_each_component(component) + for_each_component_dais(component, dai) seq_printf(m, "%s\n", dai->name); mutex_unlock(&client_mutex); @@ -191,7 +196,7 @@ static int component_list_show(struct seq_file *m, void *v) mutex_lock(&client_mutex); - list_for_each_entry(component, &component_list, list) + for_each_component(component) seq_printf(m, "%s\n", component->name); mutex_unlock(&client_mutex); @@ -218,7 +223,7 @@ static void soc_init_card_debugfs(struct snd_soc_card *card) &card->pop_time); if (!card->debugfs_pop_time) dev_warn(card->dev, - "ASoC: Failed to create pop time debugfs file\n"); + "ASoC: Failed to create pop time debugfs file\n"); } static void soc_cleanup_card_debugfs(struct snd_soc_card *card) @@ -341,7 +346,7 @@ struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card, { struct snd_soc_pcm_runtime *rtd; - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { if (rtd->dai_link->no_pcm && !strcmp(rtd->dai_link->name, dai_link)) return rtd->pcm->streams[stream].substream; @@ -398,7 +403,7 @@ static void soc_remove_pcm_runtimes(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd, *_rtd; - list_for_each_entry_safe(rtd, _rtd, &card->rtd_list, list) { + for_each_card_rtds_safe(card, rtd, _rtd) { list_del(&rtd->list); soc_free_pcm_runtime(rtd); } @@ -411,7 +416,7 @@ struct snd_soc_pcm_runtime *snd_soc_get_pcm_runtime(struct snd_soc_card *card, { struct snd_soc_pcm_runtime *rtd; - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { if (!strcmp(rtd->dai_link->name, dai_link)) return rtd; } @@ -422,7 +427,8 @@ EXPORT_SYMBOL_GPL(snd_soc_get_pcm_runtime); static void codec2codec_close_delayed_work(struct work_struct *work) { - /* Currently nothing to do for c2c links + /* + * 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. @@ -442,8 +448,9 @@ int snd_soc_suspend(struct device *dev) if (!card->instantiated) return 0; - /* Due to the resume being scheduled into a workqueue we could - * suspend before that's finished - wait for it to complete. + /* + * Due to the resume being scheduled into a workqueue we could + * suspend before that's finished - wait for it to complete. */ snd_power_wait(card->snd_card, SNDRV_CTL_POWER_D0); @@ -451,13 +458,13 @@ int snd_soc_suspend(struct device *dev) snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D3hot); /* mute any active DACs */ - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { + struct snd_soc_dai *dai; if (rtd->dai_link->ignore_suspend) continue; - for (i = 0; i < rtd->num_codecs; i++) { - struct snd_soc_dai *dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, dai) { struct snd_soc_dai_driver *drv = dai->driver; if (drv->ops->digital_mute && dai->playback_active) @@ -466,7 +473,7 @@ int snd_soc_suspend(struct device *dev) } /* suspend all pcms */ - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { if (rtd->dai_link->ignore_suspend) continue; @@ -476,7 +483,7 @@ int snd_soc_suspend(struct device *dev) if (card->suspend_pre) card->suspend_pre(card); - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; if (rtd->dai_link->ignore_suspend) @@ -487,10 +494,10 @@ int snd_soc_suspend(struct device *dev) } /* close any waiting streams */ - list_for_each_entry(rtd, &card->rtd_list, list) + for_each_card_rtds(card, rtd) flush_delayed_work(&rtd->delayed_work); - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { if (rtd->dai_link->ignore_suspend) continue; @@ -509,11 +516,14 @@ int snd_soc_suspend(struct device *dev) snd_soc_dapm_sync(&card->dapm); /* suspend all COMPONENTs */ - list_for_each_entry(component, &card->component_dev_list, card_list) { - struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + for_each_card_components(card, component) { + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); - /* If there are paths active then the COMPONENT will be held with - * bias _ON and should not be suspended. */ + /* + * If there are paths active then the COMPONENT will be held + * with bias _ON and should not be suspended. + */ if (!component->suspended) { switch (snd_soc_dapm_get_bias_level(dapm)) { case SND_SOC_BIAS_STANDBY: @@ -547,7 +557,7 @@ int snd_soc_suspend(struct device *dev) } } - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; if (rtd->dai_link->ignore_suspend) @@ -567,18 +577,21 @@ int snd_soc_suspend(struct device *dev) } EXPORT_SYMBOL_GPL(snd_soc_suspend); -/* deferred resume work, so resume can complete before we finished +/* + * deferred resume work, so resume can complete before we finished * setting our codec back up, which can be very slow on I2C */ static void soc_resume_deferred(struct work_struct *work) { struct snd_soc_card *card = - container_of(work, struct snd_soc_card, deferred_resume_work); + container_of(work, struct snd_soc_card, + deferred_resume_work); struct snd_soc_pcm_runtime *rtd; struct snd_soc_component *component; int i; - /* our power state is still SNDRV_CTL_POWER_D3hot from suspend time, + /* + * our power state is still SNDRV_CTL_POWER_D3hot from suspend time, * so userspace apps are blocked from touching us */ @@ -591,7 +604,7 @@ static void soc_resume_deferred(struct work_struct *work) card->resume_pre(card); /* resume control bus DAIs */ - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; if (rtd->dai_link->ignore_suspend) @@ -601,7 +614,7 @@ static void soc_resume_deferred(struct work_struct *work) cpu_dai->driver->resume(cpu_dai); } - list_for_each_entry(component, &card->component_dev_list, card_list) { + for_each_card_components(card, component) { if (component->suspended) { if (component->driver->resume) component->driver->resume(component); @@ -609,7 +622,7 @@ static void soc_resume_deferred(struct work_struct *work) } } - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { if (rtd->dai_link->ignore_suspend) continue; @@ -624,13 +637,13 @@ static void soc_resume_deferred(struct work_struct *work) } /* unmute any active DACs */ - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { + struct snd_soc_dai *dai; if (rtd->dai_link->ignore_suspend) continue; - for (i = 0; i < rtd->num_codecs; i++) { - struct snd_soc_dai *dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, dai) { struct snd_soc_dai_driver *drv = dai->driver; if (drv->ops->digital_mute && dai->playback_active) @@ -638,7 +651,7 @@ static void soc_resume_deferred(struct work_struct *work) } } - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; if (rtd->dai_link->ignore_suspend) @@ -673,16 +686,15 @@ int snd_soc_resume(struct device *dev) return 0; /* activate pins from sleep state */ - list_for_each_entry(rtd, &card->rtd_list, list) { - struct snd_soc_dai **codec_dais = rtd->codec_dais; + for_each_card_rtds(card, rtd) { + struct snd_soc_dai *codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int j; if (cpu_dai->active) pinctrl_pm_select_default_state(cpu_dai->dev); - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = codec_dais[j]; + for_each_rtd_codec_dai(rtd, j, codec_dai) { if (codec_dai->active) pinctrl_pm_select_default_state(codec_dai->dev); } @@ -694,8 +706,9 @@ int snd_soc_resume(struct device *dev) * have that problem and may take a substantial amount of time to resume * due to I/O costs and anti-pop so handle them out of line. */ - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + bus_control |= cpu_dai->driver->bus_control; } if (bus_control) { @@ -722,14 +735,19 @@ static struct snd_soc_component *soc_find_component( const struct device_node *of_node, const char *name) { struct snd_soc_component *component; + struct device_node *component_of_node; lockdep_assert_held(&client_mutex); - list_for_each_entry(component, &component_list, list) { + for_each_component(component) { if (of_node) { - if (component->dev->of_node == of_node) + component_of_node = component->dev->of_node; + if (!component_of_node && component->dev->parent) + component_of_node = component->dev->parent->of_node; + + if (component_of_node == of_node) return component; - } else if (strcmp(component->name, name) == 0) { + } else if (name && strcmp(component->name, name) == 0) { return component; } } @@ -737,6 +755,24 @@ static struct snd_soc_component *soc_find_component( return NULL; } +static int snd_soc_is_matching_component( + const struct snd_soc_dai_link_component *dlc, + struct snd_soc_component *component) +{ + struct device_node *component_of_node; + + component_of_node = component->dev->of_node; + if (!component_of_node && component->dev->parent) + component_of_node = component->dev->parent->of_node; + + if (dlc->of_node && component_of_node != dlc->of_node) + return 0; + if (dlc->name && strcmp(component->name, dlc->name)) + return 0; + + return 1; +} + /** * snd_soc_find_dai - Find a registered DAI * @@ -753,21 +789,14 @@ struct snd_soc_dai *snd_soc_find_dai( { struct snd_soc_component *component; struct snd_soc_dai *dai; - struct device_node *component_of_node; lockdep_assert_held(&client_mutex); - /* Find CPU DAI from registered DAIs*/ - list_for_each_entry(component, &component_list, list) { - component_of_node = component->dev->of_node; - if (!component_of_node && component->dev->parent) - component_of_node = component->dev->parent->of_node; - - if (dlc->of_node && component_of_node != dlc->of_node) - continue; - if (dlc->name && strcmp(component->name, dlc->name)) + /* Find CPU DAI from registered DAIs */ + for_each_component(component) { + if (!snd_soc_is_matching_component(dlc, component)) continue; - list_for_each_entry(dai, &component->dai_list, list) { + for_each_component_dais(component, dai) { if (dlc->dai_name && strcmp(dai->name, dlc->dai_name) && (!dai->driver->name || strcmp(dai->driver->name, dlc->dai_name))) @@ -781,7 +810,6 @@ struct snd_soc_dai *snd_soc_find_dai( } EXPORT_SYMBOL_GPL(snd_soc_find_dai); - /** * snd_soc_find_dai_link - Find a DAI link * @@ -805,7 +833,7 @@ struct snd_soc_dai_link *snd_soc_find_dai_link(struct snd_soc_card *card, lockdep_assert_held(&client_mutex); - list_for_each_entry_safe(link, _link, &card->dai_link_list, list) { + for_each_card_links_safe(card, link, _link) { if (link->id != id) continue; @@ -828,7 +856,7 @@ static bool soc_is_dai_link_bound(struct snd_soc_card *card, { struct snd_soc_pcm_runtime *rtd; - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { if (rtd->dai_link == dai_link) return true; } @@ -844,8 +872,6 @@ static int soc_bind_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link_component cpu_dai_component; struct snd_soc_component *component; struct snd_soc_dai **codec_dais; - struct device_node *platform_of_node; - const char *platform_name; int i; if (dai_link->ignore) @@ -877,6 +903,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, rtd->num_codecs = dai_link->num_codecs; /* Find CODEC from registered CODECs */ + /* we can use for_each_rtd_codec_dai() after this */ codec_dais = rtd->codec_dais; for (i = 0; i < rtd->num_codecs; i++) { codec_dais[i] = snd_soc_find_dai(&codecs[i]); @@ -891,24 +918,11 @@ static int soc_bind_dai_link(struct snd_soc_card *card, /* Single codec links expect codec and codec_dai in runtime data */ rtd->codec_dai = codec_dais[0]; - /* if there's no platform we match on the empty platform */ - platform_name = dai_link->platform_name; - if (!platform_name && !dai_link->platform_of_node) - platform_name = "snd-soc-dummy"; - /* find one from the set of registered platforms */ - list_for_each_entry(component, &component_list, list) { - platform_of_node = component->dev->of_node; - if (!platform_of_node && component->dev->parent->of_node) - platform_of_node = component->dev->parent->of_node; - - if (dai_link->platform_of_node) { - if (platform_of_node != dai_link->platform_of_node) - continue; - } else { - if (strcmp(component->name, platform_name)) - continue; - } + for_each_component(component) { + if (!snd_soc_is_matching_component(dai_link->platform, + component)) + continue; snd_soc_rtdcom_add(rtd, component); } @@ -918,7 +932,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, _err_defer: soc_free_pcm_runtime(rtd); - return -EPROBE_DEFER; + return -EPROBE_DEFER; } static void soc_remove_component(struct snd_soc_component *component) @@ -942,23 +956,25 @@ static void soc_remove_dai(struct snd_soc_dai *dai, int order) { int err; - if (dai && dai->probed && - dai->driver->remove_order == order) { - if (dai->driver->remove) { - err = dai->driver->remove(dai); - if (err < 0) - dev_err(dai->dev, - "ASoC: failed to remove %s: %d\n", - dai->name, err); - } - dai->probed = 0; + if (!dai || !dai->probed || !dai->driver || + dai->driver->remove_order != order) + return; + + if (dai->driver->remove) { + err = dai->driver->remove(dai); + if (err < 0) + dev_err(dai->dev, + "ASoC: failed to remove %s: %d\n", + dai->name, err); } + dai->probed = 0; } static void soc_remove_link_dais(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd, int order) { int i; + struct snd_soc_dai *codec_dai; /* unregister the rtd device */ if (rtd->dev_registered) { @@ -967,8 +983,8 @@ static void soc_remove_link_dais(struct snd_soc_card *card, } /* remove the CODEC DAI */ - for (i = 0; i < rtd->num_codecs; i++) - soc_remove_dai(rtd->codec_dais[i], order); + for_each_rtd_codec_dai(rtd, i, codec_dai) + soc_remove_dai(codec_dai, order); soc_remove_dai(rtd->cpu_dai, order); } @@ -993,28 +1009,58 @@ static void soc_remove_dai_links(struct snd_soc_card *card) struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai_link *link, *_link; - for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; - order++) { - list_for_each_entry(rtd, &card->rtd_list, list) + for_each_comp_order(order) { + for_each_card_rtds(card, rtd) soc_remove_link_dais(card, rtd, order); } - for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; - order++) { - list_for_each_entry(rtd, &card->rtd_list, list) + for_each_comp_order(order) { + for_each_card_rtds(card, rtd) soc_remove_link_components(card, rtd, order); } - list_for_each_entry_safe(link, _link, &card->dai_link_list, list) { + for_each_card_links_safe(card, link, _link) { if (link->dobj.type == SND_SOC_DOBJ_DAI_LINK) dev_warn(card->dev, "Topology forgot to remove link %s?\n", link->name); list_del(&link->list); - card->num_dai_links--; } } +static int snd_soc_init_platform(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_link) +{ + struct snd_soc_dai_link_component *platform = dai_link->platform; + + /* + * FIXME + * + * this function should be removed in the future + */ + /* convert Legacy platform link */ + if (!platform || dai_link->legacy_platform) { + platform = devm_kzalloc(card->dev, + sizeof(struct snd_soc_dai_link_component), + GFP_KERNEL); + if (!platform) + return -ENOMEM; + + dai_link->platform = platform; + dai_link->legacy_platform = 1; + platform->name = dai_link->platform_name; + platform->of_node = dai_link->platform_of_node; + platform->dai_name = NULL; + } + + /* if there's no platform we match on the empty platform */ + if (!platform->name && + !platform->of_node) + platform->name = "snd-soc-dummy"; + + return 0; +} + static int snd_soc_init_multicodec(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { @@ -1043,9 +1089,16 @@ static int snd_soc_init_multicodec(struct snd_soc_card *card, } static int soc_init_dai_link(struct snd_soc_card *card, - struct snd_soc_dai_link *link) + struct snd_soc_dai_link *link) { int i, ret; + struct snd_soc_dai_link_component *codec; + + ret = snd_soc_init_platform(card, link); + if (ret) { + dev_err(card->dev, "ASoC: failed to init multiplatform\n"); + return ret; + } ret = snd_soc_init_multicodec(card, link); if (ret) { @@ -1053,19 +1106,19 @@ static int soc_init_dai_link(struct snd_soc_card *card, return ret; } - for (i = 0; i < link->num_codecs; i++) { + for_each_link_codecs(link, i, codec) { /* * Codec must be specified by 1 of name or OF node, * not both or neither. */ - if (!!link->codecs[i].name == - !!link->codecs[i].of_node) { + if (!!codec->name == + !!codec->of_node) { dev_err(card->dev, "ASoC: Neither/both codec name/of_node are set for %s\n", link->name); return -EINVAL; } /* Codec DAI name must be specified */ - if (!link->codecs[i].dai_name) { + if (!codec->dai_name) { dev_err(card->dev, "ASoC: codec_dai_name not set for %s\n", link->name); return -EINVAL; @@ -1076,7 +1129,7 @@ static int soc_init_dai_link(struct snd_soc_card *card, * Platform may be specified by either name or OF node, but * can be left unspecified, and a dummy platform will be used. */ - if (link->platform_name && link->platform_of_node) { + if (link->platform->name && link->platform->of_node) { dev_err(card->dev, "ASoC: Both platform name/of_node are set for %s\n", link->name); @@ -1084,6 +1137,14 @@ static int soc_init_dai_link(struct snd_soc_card *card, } /* + * Defer card registartion if platform dai component is not added to + * component list. + */ + if ((link->platform->of_node || link->platform->name) && + !soc_find_component(link->platform->of_node, link->platform->name)) + return -EPROBE_DEFER; + + /* * CPU device may be specified by either name or OF node, but * can be left unspecified, and will be matched based on DAI * name alone.. @@ -1094,6 +1155,15 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if cpu dai component is not added to + * component list. + */ + if ((link->cpu_of_node || link->cpu_name) && + !soc_find_component(link->cpu_of_node, link->cpu_name)) + return -EPROBE_DEFER; + /* * At least one of CPU DAI name or CPU device name/node must be * specified @@ -1111,7 +1181,8 @@ static int soc_init_dai_link(struct snd_soc_card *card, void snd_soc_disconnect_sync(struct device *dev) { - struct snd_soc_component *component = snd_soc_lookup_component(dev, NULL); + struct snd_soc_component *component = + snd_soc_lookup_component(dev, NULL); if (!component || !component->card) return; @@ -1142,14 +1213,14 @@ int snd_soc_add_dai_link(struct snd_soc_card *card, } lockdep_assert_held(&client_mutex); - /* Notify the machine driver for extra initialization + /* + * Notify the machine driver for extra initialization * on the link created by topology. */ if (dai_link->dobj.type && card->add_dai_link) card->add_dai_link(card, dai_link); list_add_tail(&dai_link->list, &card->dai_link_list); - card->num_dai_links++; return 0; } @@ -1178,16 +1249,16 @@ void snd_soc_remove_dai_link(struct snd_soc_card *card, } lockdep_assert_held(&client_mutex); - /* Notify the machine driver for extra destruction + /* + * Notify the machine driver for extra destruction * on the link created by topology. */ if (dai_link->dobj.type && card->remove_dai_link) card->remove_dai_link(card, dai_link); - list_for_each_entry_safe(link, _link, &card->dai_link_list, list) { + for_each_card_links_safe(card, link, _link) { if (link == dai_link) { list_del(&link->list); - card->num_dai_links--; return; } } @@ -1239,7 +1310,8 @@ static void soc_set_name_prefix(struct snd_soc_card *card, static int soc_probe_component(struct snd_soc_card *card, struct snd_soc_component *component) { - struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); struct snd_soc_dai *dai; int ret; @@ -1277,7 +1349,7 @@ static int soc_probe_component(struct snd_soc_card *card, } } - list_for_each_entry(dai, &component->dai_list, list) { + for_each_component_dais(component, dai) { ret = snd_soc_dapm_new_dai_widgets(dapm, dai); if (ret != 0) { dev_err(component->dev, @@ -1320,6 +1392,7 @@ static int soc_probe_component(struct snd_soc_card *card, component->driver->num_dapm_routes); list_add(&dapm->list, &card->dapm_list); + /* see for_each_card_components */ list_add(&component->card_list, &card->component_dev_list); return 0; @@ -1370,8 +1443,7 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, } static int soc_probe_link_components(struct snd_soc_card *card, - struct snd_soc_pcm_runtime *rtd, - int order) + struct snd_soc_pcm_runtime *rtd, int order) { struct snd_soc_component *component; struct snd_soc_rtdcom_list *rtdcom; @@ -1398,6 +1470,7 @@ static int soc_probe_dai(struct snd_soc_dai *dai, int order) if (dai->driver->probe) { int ret = dai->driver->probe(dai); + if (ret < 0) { dev_err(dai->dev, "ASoC: failed to probe DAI %s: %d\n", dai->name, ret); @@ -1418,7 +1491,7 @@ static int soc_link_dai_pcm_new(struct snd_soc_dai **dais, int num_dais, for (i = 0; i < num_dais; ++i) { struct snd_soc_dai_driver *drv = dais[i]->driver; - if (!rtd->dai_link->no_pcm && drv->pcm_new) + if (drv->pcm_new) ret = drv->pcm_new(rtd, dais[i]); if (ret < 0) { dev_err(dais[i]->dev, @@ -1431,48 +1504,6 @@ static int soc_link_dai_pcm_new(struct snd_soc_dai **dais, int num_dais, return 0; } -static int soc_link_dai_widgets(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_link, - struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dapm_widget *sink, *source; - int ret; - - if (rtd->num_codecs > 1) - dev_warn(card->dev, "ASoC: Multiple codecs not supported yet\n"); - - /* link the DAI widgets */ - sink = codec_dai->playback_widget; - source = cpu_dai->capture_widget; - if (sink && source) { - ret = snd_soc_dapm_new_pcm(card, rtd, dai_link->params, - dai_link->num_params, - source, sink); - if (ret != 0) { - dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", - sink->name, source->name, ret); - return ret; - } - } - - sink = cpu_dai->playback_widget; - source = codec_dai->capture_widget; - if (sink && source) { - ret = snd_soc_dapm_new_pcm(card, rtd, dai_link->params, - dai_link->num_params, - source, sink); - if (ret != 0) { - dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", - sink->name, source->name, ret); - return ret; - } - } - - return 0; -} - static int soc_probe_link_dais(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd, int order) { @@ -1480,6 +1511,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_component *component; + struct snd_soc_dai *codec_dai; int i, ret, num; dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n", @@ -1493,8 +1525,8 @@ static int soc_probe_link_dais(struct snd_soc_card *card, return ret; /* probe the CODEC DAI */ - for (i = 0; i < rtd->num_codecs; i++) { - ret = soc_probe_dai(rtd->codec_dais[i], order); + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = soc_probe_dai(codec_dai, order); if (ret) return ret; } @@ -1546,7 +1578,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, } if (cpu_dai->driver->compress_new) { - /*create compress_device"*/ + /* create compress_device" */ ret = cpu_dai->driver->compress_new(rtd, num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create compress %s\n", @@ -1560,7 +1592,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, ret = soc_new_pcm(rtd, num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create pcm %s :%d\n", - dai_link->stream_name, ret); + dai_link->stream_name, ret); return ret; } ret = soc_link_dai_pcm_new(&cpu_dai, 1, rtd); @@ -1573,11 +1605,6 @@ static int soc_probe_link_dais(struct snd_soc_card *card, } else { INIT_DELAYED_WORK(&rtd->delayed_work, codec2codec_close_delayed_work); - - /* link the DAI widgets */ - ret = soc_link_dai_widgets(card, dai_link, rtd); - if (ret) - return ret; } } @@ -1628,8 +1655,7 @@ static int soc_probe_aux_devices(struct snd_soc_card *card) int order; int ret; - for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; - order++) { + for_each_comp_order(order) { list_for_each_entry(comp, &card->aux_comp_list, card_aux_list) { if (comp->driver->probe_order == order) { ret = soc_probe_component(card, comp); @@ -1651,8 +1677,7 @@ static void soc_remove_aux_devices(struct snd_soc_card *card) struct snd_soc_component *comp, *_comp; int order; - for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; - order++) { + for_each_comp_order(order) { list_for_each_entry_safe(comp, _comp, &card->aux_comp_list, card_aux_list) { @@ -1681,14 +1706,12 @@ static void soc_remove_aux_devices(struct snd_soc_card *card) int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, unsigned int dai_fmt) { - struct snd_soc_dai **codec_dais = rtd->codec_dais; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; unsigned int i; int ret; - for (i = 0; i < rtd->num_codecs; i++) { - struct snd_soc_dai *codec_dai = codec_dais[i]; - + for_each_rtd_codec_dai(rtd, i, codec_dai) { ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt); if (ret != 0 && ret != -ENOTSUPP) { dev_warn(codec_dai->dev, @@ -1697,8 +1720,10 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, } } - /* Flip the polarity for the "CPU" end of a CODEC<->CODEC link */ - /* the component which has non_legacy_dai_naming is Codec */ + /* + * Flip the polarity for the "CPU" end of a CODEC<->CODEC link + * the component which has non_legacy_dai_naming is Codec + */ if (cpu_dai->component->driver->non_legacy_dai_naming) { unsigned int inv_dai_fmt; @@ -1732,9 +1757,9 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, } EXPORT_SYMBOL_GPL(snd_soc_runtime_set_dai_fmt); - #ifdef CONFIG_DMI -/* Trim special characters, and replace '-' with '_' since '-' is used to +/* + * Trim special characters, and replace '-' with '_' since '-' is used to * separate different DMI fields in the card long name. Only number and * alphabet characters and a few separator characters are kept. */ @@ -1753,7 +1778,8 @@ static void cleanup_dmi_name(char *name) name[j] = '\0'; } -/* Check if a DMI field is valid, i.e. not containing any string +/* + * Check if a DMI field is valid, i.e. not containing any string * in the black list. */ static int is_dmi_valid(const char *field) @@ -1816,7 +1842,6 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) return 0; } - snprintf(card->dmi_longname, sizeof(card->snd_card->longname), "%s", vendor); cleanup_dmi_name(card->dmi_longname); @@ -1832,7 +1857,8 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) if (len < longname_buf_size) cleanup_dmi_name(card->dmi_longname + len); - /* some vendors like Lenovo may only put a self-explanatory + /* + * some vendors like Lenovo may only put a self-explanatory * name in the product version field */ product_version = dmi_get_system_info(DMI_PRODUCT_VERSION); @@ -1891,7 +1917,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) struct snd_soc_dai_link *dai_link; int i; - list_for_each_entry(component, &component_list, list) { + for_each_component(component) { /* does this component override FEs ? */ if (!component->driver->ignore_machine) @@ -1903,9 +1929,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) continue; /* machine matches, so override the rtd data */ - for (i = 0; i < card->num_links; i++) { - - dai_link = &card->dai_link[i]; + for_each_card_prelinks(card, i, dai_link) { /* ignore this FE */ if (dai_link->dynamic) { @@ -1917,7 +1941,11 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) card->dai_link[i].name); /* override platform component */ - dai_link->platform_name = component->name; + if (snd_soc_init_platform(card, dai_link) < 0) { + dev_err(card->dev, "init platform error"); + continue; + } + dai_link->platform->name = component->name; /* convert non BE into BE */ dai_link->no_pcm = 1; @@ -1926,7 +1954,8 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) dai_link->be_hw_params_fixup = component->driver->be_hw_params_fixup; - /* most BE links don't set stream name, so set it to + /* + * most BE links don't set stream name, so set it to * dai link name if it's NULL to help bind widgets. */ if (!dai_link->stream_name) @@ -1936,7 +1965,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) /* Inform userspace we are using alternate topology */ if (component->driver->topology_name_prefix) { - /* topology shortname created ? */ + /* topology shortname created? */ if (!card->topology_shortname_created) { comp_drv = component->driver; @@ -1965,8 +1994,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) soc_check_tplg_fes(card); /* bind DAIs */ - for (i = 0; i < card->num_links; i++) { - ret = soc_bind_dai_link(card, &card->dai_link[i]); + for_each_card_prelinks(card, i, dai_link) { + ret = soc_bind_dai_link(card, dai_link); if (ret != 0) goto base_error; } @@ -1979,8 +2008,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) } /* add predefined DAI links to the list */ - for (i = 0; i < card->num_links; i++) - snd_soc_add_dai_link(card, card->dai_link+i); + for_each_card_prelinks(card, i, dai_link) + snd_soc_add_dai_link(card, dai_link); /* card bind complete so register a sound card */ ret = snd_card_new(card->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, @@ -2024,9 +2053,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) } /* probe all components used by DAI links on this card */ - for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; - order++) { - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_comp_order(order) { + for_each_card_rtds(card, rtd) { ret = soc_probe_link_components(card, rtd, order); if (ret < 0) { dev_err(card->dev, @@ -2042,10 +2070,11 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) if (ret < 0) goto probe_dai_err; - /* Find new DAI links added during probing components and bind them. + /* + * Find new DAI links added during probing components and bind them. * Components with topology may bring new DAIs and DAI links. */ - list_for_each_entry(dai_link, &card->dai_link_list, list) { + for_each_card_links(card, dai_link) { if (soc_is_dai_link_bound(card, dai_link)) continue; @@ -2058,9 +2087,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) } /* probe all DAI links on this card */ - for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; - order++) { - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_comp_order(order) { + for_each_card_rtds(card, rtd) { ret = soc_probe_link_dais(card, rtd, order); if (ret < 0) { dev_err(card->dev, @@ -2075,7 +2103,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_connect_dai_link_widgets(card); if (card->controls) - snd_soc_add_card_controls(card, card->controls, card->num_controls); + snd_soc_add_card_controls(card, card->controls, + card->num_controls); if (card->dapm_routes) snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes, @@ -2126,6 +2155,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) } card->instantiated = 1; + dapm_mark_endpoints_dirty(card); snd_soc_dapm_sync(&card->dapm); mutex_unlock(&card->mutex); mutex_unlock(&client_mutex); @@ -2181,7 +2211,7 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card) struct snd_soc_pcm_runtime *rtd; /* make sure any delayed work runs */ - list_for_each_entry(rtd, &card->rtd_list, list) + for_each_card_rtds(card, rtd) flush_delayed_work(&rtd->delayed_work); /* free the ALSA card at first; this syncs with pending operations */ @@ -2221,21 +2251,23 @@ int snd_soc_poweroff(struct device *dev) if (!card->instantiated) return 0; - /* Flush out pmdown_time work - we actually do want to run it - * now, we're shutting down so no imminent restart. */ - list_for_each_entry(rtd, &card->rtd_list, list) + /* + * Flush out pmdown_time work - we actually do want to run it + * now, we're shutting down so no imminent restart. + */ + for_each_card_rtds(card, rtd) flush_delayed_work(&rtd->delayed_work); snd_soc_dapm_shutdown(card); /* deactivate pins to sleep state */ - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; int i; pinctrl_pm_select_sleep_state(cpu_dai->dev); - for (i = 0; i < rtd->num_codecs; i++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { pinctrl_pm_select_sleep_state(codec_dai->dev); } } @@ -2315,6 +2347,7 @@ static int snd_soc_add_controls(struct snd_card *card, struct device *dev, for (i = 0; i < num_controls; i++) { const struct snd_kcontrol_new *control = &controls[i]; + err = snd_ctl_add(card, snd_soc_cnew(control, data, control->name, prefix)); if (err < 0) { @@ -2432,8 +2465,9 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); * * Configures the CODEC master (MCLK) or system (SYSCLK) clocking. */ -int snd_soc_component_set_sysclk(struct snd_soc_component *component, int clk_id, - int source, unsigned int freq, int dir) +int snd_soc_component_set_sysclk(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, + int dir) { if (component->driver->set_sysclk) return component->driver->set_sysclk(component, clk_id, source, @@ -2501,7 +2535,7 @@ int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id, { if (component->driver->set_pll) return component->driver->set_pll(component, pll_id, source, - freq_in, freq_out); + freq_in, freq_out); return -EINVAL; } @@ -2532,8 +2566,6 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_bclk_ratio); */ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { - if (dai->driver == NULL) - return -EINVAL; if (dai->driver->ops->set_fmt == NULL) return -ENOTSUPP; return dai->driver->ops->set_fmt(dai, fmt); @@ -2549,8 +2581,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); * Generates the TDM tx and rx slot default masks for DAI. */ static int snd_soc_xlate_tdm_slot_mask(unsigned int slots, - unsigned int *tx_mask, - unsigned int *rx_mask) + unsigned int *tx_mask, + unsigned int *rx_mask) { if (*tx_mask || *rx_mask) return 0; @@ -2680,9 +2712,6 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate); int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, int direction) { - if (!dai->driver) - return -ENOTSUPP; - if (dai->driver->ops->mute_stream) return dai->driver->ops->mute_stream(dai, mute, direction); else if (direction == SNDRV_PCM_STREAM_PLAYBACK && @@ -2693,6 +2722,33 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, } EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); +static int snd_soc_bind_card(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd; + int ret; + + ret = snd_soc_instantiate_card(card); + if (ret != 0) + return ret; + + /* deactivate pins to sleep state */ + for_each_card_rtds(card, rtd) { + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; + int j; + + for_each_rtd_codec_dai(rtd, j, codec_dai) { + if (!codec_dai->active) + pinctrl_pm_select_sleep_state(codec_dai->dev); + } + + if (!cpu_dai->active) + pinctrl_pm_select_sleep_state(cpu_dai->dev); + } + + return ret; +} + /** * snd_soc_register_card - Register a card with the ASoC core * @@ -2702,28 +2758,29 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); int snd_soc_register_card(struct snd_soc_card *card) { int i, ret; - struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai_link *link; if (!card->name || !card->dev) return -EINVAL; - for (i = 0; i < card->num_links; i++) { - struct snd_soc_dai_link *link = &card->dai_link[i]; + mutex_lock(&client_mutex); + for_each_card_prelinks(card, i, link) { ret = soc_init_dai_link(card, link); if (ret) { dev_err(card->dev, "ASoC: failed to init link %s\n", link->name); + mutex_unlock(&client_mutex); return ret; } } + mutex_unlock(&client_mutex); dev_set_drvdata(card->dev, card); snd_soc_initialize_card_lists(card); INIT_LIST_HEAD(&card->dai_link_list); - card->num_dai_links = 0; INIT_LIST_HEAD(&card->rtd_list); card->num_rtd = 0; @@ -2734,28 +2791,23 @@ int snd_soc_register_card(struct snd_soc_card *card) mutex_init(&card->mutex); mutex_init(&card->dapm_mutex); - ret = snd_soc_instantiate_card(card); - if (ret != 0) - return ret; - - /* deactivate pins to sleep state */ - list_for_each_entry(rtd, &card->rtd_list, list) { - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int j; - - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; - if (!codec_dai->active) - pinctrl_pm_select_sleep_state(codec_dai->dev); - } + return snd_soc_bind_card(card); +} +EXPORT_SYMBOL_GPL(snd_soc_register_card); - if (!cpu_dai->active) - pinctrl_pm_select_sleep_state(cpu_dai->dev); +static void snd_soc_unbind_card(struct snd_soc_card *card, bool unregister) +{ + if (card->instantiated) { + card->instantiated = false; + snd_soc_dapm_shutdown(card); + soc_cleanup_card_resources(card); + if (!unregister) + list_add(&card->list, &unbind_card_list); + } else { + if (unregister) + list_del(&card->list); } - - return ret; } -EXPORT_SYMBOL_GPL(snd_soc_register_card); /** * snd_soc_unregister_card - Unregister a card with the ASoC core @@ -2765,12 +2817,8 @@ EXPORT_SYMBOL_GPL(snd_soc_register_card); */ int snd_soc_unregister_card(struct snd_soc_card *card) { - if (card->instantiated) { - card->instantiated = false; - snd_soc_dapm_shutdown(card); - soc_cleanup_card_resources(card); - dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); - } + snd_soc_unbind_card(card, true); + dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); return 0; } @@ -2802,7 +2850,7 @@ static char *fmt_single_name(struct device *dev, int *id) } } else { - /* I2C component devices are named "bus-addr" */ + /* I2C component devices are named "bus-addr" */ if (sscanf(name, "%x-%x", &id1, &id2) == 2) { char tmp[NAME_SIZE]; @@ -2810,7 +2858,8 @@ static char *fmt_single_name(struct device *dev, int *id) *id = ((id1 & 0xffff) << 16) + id2; /* sanitize component name for DAI link creation */ - snprintf(tmp, NAME_SIZE, "%s.%s", dev->driver->name, name); + snprintf(tmp, NAME_SIZE, "%s.%s", dev->driver->name, + name); strlcpy(name, tmp, NAME_SIZE); } else *id = 0; @@ -2845,7 +2894,7 @@ static void snd_soc_unregister_dais(struct snd_soc_component *component) { struct snd_soc_dai *dai, *_dai; - list_for_each_entry_safe(dai, _dai, &component->dai_list, list) { + for_each_component_dais_safe(component, dai, _dai) { dev_dbg(component->dev, "ASoC: Unregistered DAI '%s'\n", dai->name); list_del(&dai->list); @@ -2877,7 +2926,7 @@ static struct snd_soc_dai *soc_add_dai(struct snd_soc_component *component, * component-less anymore. */ if (legacy_dai_naming && - (dai_drv->id == 0 || dai_drv->name == NULL)) { + (dai_drv->id == 0 || dai_drv->name == NULL)) { dai->name = fmt_single_name(dev, &dai->id); } else { dai->name = fmt_multiple_name(dev, dai_drv); @@ -2897,6 +2946,7 @@ static struct snd_soc_dai *soc_add_dai(struct snd_soc_component *component, if (!dai->driver->ops) dai->driver->ops = &null_dai_ops; + /* see for_each_component_dais */ list_add_tail(&dai->list, &component->dai_list); component->num_dai++; @@ -2910,11 +2960,10 @@ static struct snd_soc_dai *soc_add_dai(struct snd_soc_component *component, * @component: The component the DAIs are registered for * @dai_drv: DAI driver to use for the DAIs * @count: Number of DAIs - * @legacy_dai_naming: Use the legacy naming scheme and let the DAI inherit the - * parent's name. */ static int snd_soc_register_dais(struct snd_soc_component *component, - struct snd_soc_dai_driver *dai_drv, size_t count) + struct snd_soc_dai_driver *dai_drv, + size_t count) { struct device *dev = component->dev; struct snd_soc_dai *dai; @@ -2925,8 +2974,8 @@ static int snd_soc_register_dais(struct snd_soc_component *component, for (i = 0; i < count; i++) { - dai = soc_add_dai(component, dai_drv + i, - count == 1 && !component->driver->non_legacy_dai_naming); + dai = soc_add_dai(component, dai_drv + i, count == 1 && + !component->driver->non_legacy_dai_naming); if (dai == NULL) { ret = -ENOMEM; goto err; @@ -2970,7 +3019,8 @@ int snd_soc_register_dai(struct snd_soc_component *component, if (!dai) return -ENOMEM; - /* Create the DAI widgets here. After adding DAIs, topology may + /* + * Create the DAI widgets here. After adding DAIs, topology may * also add routes that need these widgets as source or sink. */ ret = snd_soc_dapm_new_dai_widgets(dapm, dai); @@ -3052,7 +3102,8 @@ static void snd_soc_component_setup_regmap(struct snd_soc_component *component) #ifdef CONFIG_REGMAP /** - * snd_soc_component_init_regmap() - Initialize regmap instance for the component + * snd_soc_component_init_regmap() - Initialize regmap instance for the + * component * @component: The component for which to initialize the regmap instance * @regmap: The regmap instance that should be used by the component * @@ -3070,7 +3121,8 @@ void snd_soc_component_init_regmap(struct snd_soc_component *component, EXPORT_SYMBOL_GPL(snd_soc_component_init_regmap); /** - * snd_soc_component_exit_regmap() - De-initialize regmap instance for the component + * snd_soc_component_exit_regmap() - De-initialize regmap instance for the + * component * @component: The component for which to de-initialize the regmap instance * * Calls regmap_exit() on the regmap instance associated to the component and @@ -3094,11 +3146,13 @@ static void snd_soc_component_add(struct snd_soc_component *component) if (!component->driver->write && !component->driver->read) { if (!component->regmap) - component->regmap = dev_get_regmap(component->dev, NULL); + component->regmap = dev_get_regmap(component->dev, + NULL); if (component->regmap) snd_soc_component_setup_regmap(component); } + /* see for_each_component */ list_add(&component->list, &component_list); INIT_LIST_HEAD(&component->dobj_list); @@ -3116,7 +3170,7 @@ static void snd_soc_component_del_unlocked(struct snd_soc_component *component) struct snd_soc_card *card = component->card; if (card) - snd_soc_unregister_card(card); + snd_soc_unbind_card(card, false); list_del(&component->list); } @@ -3156,6 +3210,18 @@ static void convert_endianness_formats(struct snd_soc_pcm_stream *stream) stream->formats |= endianness_format_map[i]; } +static void snd_soc_try_rebind_card(void) +{ + struct snd_soc_card *card, *c; + + if (!list_empty(&unbind_card_list)) { + list_for_each_entry_safe(card, c, &unbind_card_list, list) { + if (!snd_soc_bind_card(card)) + list_del(&card->list); + } + } +} + int snd_soc_add_component(struct device *dev, struct snd_soc_component *component, const struct snd_soc_component_driver *component_driver, @@ -3183,6 +3249,7 @@ int snd_soc_add_component(struct device *dev, } snd_soc_component_add(component); + snd_soc_try_rebind_card(); return 0; @@ -3221,27 +3288,28 @@ static int __snd_soc_unregister_component(struct device *dev) int found = 0; mutex_lock(&client_mutex); - list_for_each_entry(component, &component_list, list) { + for_each_component(component) { if (dev != component->dev) continue; - snd_soc_tplg_component_remove(component, SND_SOC_TPLG_INDEX_ALL); + snd_soc_tplg_component_remove(component, + SND_SOC_TPLG_INDEX_ALL); snd_soc_component_del_unlocked(component); found = 1; break; } mutex_unlock(&client_mutex); - if (found) { + if (found) snd_soc_component_cleanup(component); - } return found; } void snd_soc_unregister_component(struct device *dev) { - while (__snd_soc_unregister_component(dev)); + while (__snd_soc_unregister_component(dev)) + ; } EXPORT_SYMBOL_GPL(snd_soc_unregister_component); @@ -3253,7 +3321,7 @@ struct snd_soc_component *snd_soc_lookup_component(struct device *dev, ret = NULL; mutex_lock(&client_mutex); - list_for_each_entry(component, &component_list, list) { + for_each_component(component) { if (dev != component->dev) continue; @@ -3444,12 +3512,11 @@ int snd_soc_of_parse_tdm_slot(struct device_node *np, } EXPORT_SYMBOL_GPL(snd_soc_of_parse_tdm_slot); -void snd_soc_of_parse_audio_prefix(struct snd_soc_card *card, - struct snd_soc_codec_conf *codec_conf, - struct device_node *of_node, - const char *propname) +void snd_soc_of_parse_node_prefix(struct device_node *np, + struct snd_soc_codec_conf *codec_conf, + struct device_node *of_node, + const char *propname) { - struct device_node *np = card->dev->of_node; const char *str; int ret; @@ -3462,7 +3529,7 @@ void snd_soc_of_parse_audio_prefix(struct snd_soc_card *card, codec_conf->of_node = of_node; codec_conf->name_prefix = str; } -EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_prefix); +EXPORT_SYMBOL_GPL(snd_soc_of_parse_node_prefix); int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname) @@ -3653,7 +3720,7 @@ int snd_soc_get_dai_id(struct device_node *ep) */ ret = -ENOTSUPP; mutex_lock(&client_mutex); - list_for_each_entry(pos, &component_list, list) { + for_each_component(pos) { struct device_node *component_of_node = pos->dev->of_node; if (!component_of_node && pos->dev->parent) @@ -3683,7 +3750,7 @@ int snd_soc_get_dai_name(struct of_phandle_args *args, int ret = -EPROBE_DEFER; mutex_lock(&client_mutex); - list_for_each_entry(pos, &component_list, list) { + for_each_component(pos) { component_of_node = pos->dev->of_node; if (!component_of_node && pos->dev->parent) component_of_node = pos->dev->parent->of_node; @@ -3719,7 +3786,7 @@ int snd_soc_get_dai_name(struct of_phandle_args *args, ret = 0; /* find target DAI */ - list_for_each_entry(dai, &pos->dai_list, list) { + for_each_component_dais(pos, dai) { if (id == 0) break; id--; @@ -3764,10 +3831,10 @@ EXPORT_SYMBOL_GPL(snd_soc_of_get_dai_name); */ void snd_soc_of_put_dai_link_codecs(struct snd_soc_dai_link *dai_link) { - struct snd_soc_dai_link_component *component = dai_link->codecs; + struct snd_soc_dai_link_component *component; int index; - for (index = 0; index < dai_link->num_codecs; index++, component++) { + for_each_link_codecs(dai_link, index, component) { if (!component->of_node) break; of_node_put(component->of_node); @@ -3819,12 +3886,10 @@ int snd_soc_of_get_dai_link_codecs(struct device *dev, dai_link->num_codecs = num_codecs; /* Parse the list */ - for (index = 0, component = dai_link->codecs; - index < dai_link->num_codecs; - index++, component++) { + for_each_link_codecs(dai_link, index, component) { ret = of_parse_phandle_with_args(of_node, name, "#sound-dai-cells", - index, &args); + index, &args); if (ret) goto err; component->of_node = args.np; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 461d951917c0..20bad755888b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -18,7 +18,6 @@ // device reopen. #include <linux/module.h> -#include <linux/moduleparam.h> #include <linux/init.h> #include <linux/async.h> #include <linux/delay.h> @@ -71,12 +70,16 @@ static int dapm_up_seq[] = { [snd_soc_dapm_clock_supply] = 1, [snd_soc_dapm_supply] = 2, [snd_soc_dapm_micbias] = 3, + [snd_soc_dapm_vmid] = 3, [snd_soc_dapm_dai_link] = 2, [snd_soc_dapm_dai_in] = 4, [snd_soc_dapm_dai_out] = 4, [snd_soc_dapm_aif_in] = 4, [snd_soc_dapm_aif_out] = 4, [snd_soc_dapm_mic] = 5, + [snd_soc_dapm_siggen] = 5, + [snd_soc_dapm_input] = 5, + [snd_soc_dapm_output] = 5, [snd_soc_dapm_mux] = 6, [snd_soc_dapm_demux] = 6, [snd_soc_dapm_dac] = 7, @@ -84,11 +87,19 @@ static int dapm_up_seq[] = { [snd_soc_dapm_mixer] = 8, [snd_soc_dapm_mixer_named_ctl] = 8, [snd_soc_dapm_pga] = 9, + [snd_soc_dapm_buffer] = 9, + [snd_soc_dapm_scheduler] = 9, + [snd_soc_dapm_effect] = 9, + [snd_soc_dapm_src] = 9, + [snd_soc_dapm_asrc] = 9, + [snd_soc_dapm_encoder] = 9, + [snd_soc_dapm_decoder] = 9, [snd_soc_dapm_adc] = 10, [snd_soc_dapm_out_drv] = 11, [snd_soc_dapm_hp] = 11, [snd_soc_dapm_spk] = 11, [snd_soc_dapm_line] = 11, + [snd_soc_dapm_sink] = 11, [snd_soc_dapm_kcontrol] = 12, [snd_soc_dapm_post] = 13, }; @@ -101,13 +112,25 @@ static int dapm_down_seq[] = { [snd_soc_dapm_spk] = 3, [snd_soc_dapm_line] = 3, [snd_soc_dapm_out_drv] = 3, + [snd_soc_dapm_sink] = 3, [snd_soc_dapm_pga] = 4, + [snd_soc_dapm_buffer] = 4, + [snd_soc_dapm_scheduler] = 4, + [snd_soc_dapm_effect] = 4, + [snd_soc_dapm_src] = 4, + [snd_soc_dapm_asrc] = 4, + [snd_soc_dapm_encoder] = 4, + [snd_soc_dapm_decoder] = 4, [snd_soc_dapm_switch] = 5, [snd_soc_dapm_mixer_named_ctl] = 5, [snd_soc_dapm_mixer] = 5, [snd_soc_dapm_dac] = 6, [snd_soc_dapm_mic] = 7, + [snd_soc_dapm_siggen] = 7, + [snd_soc_dapm_input] = 7, + [snd_soc_dapm_output] = 7, [snd_soc_dapm_micbias] = 8, + [snd_soc_dapm_vmid] = 8, [snd_soc_dapm_mux] = 9, [snd_soc_dapm_demux] = 9, [snd_soc_dapm_aif_in] = 10, @@ -364,10 +387,6 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, ret = PTR_ERR(data->widget); goto err_data; } - if (!data->widget) { - ret = -ENOMEM; - goto err_data; - } } break; case snd_soc_dapm_demux: @@ -402,10 +421,6 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, ret = PTR_ERR(data->widget); goto err_data; } - if (!data->widget) { - ret = -ENOMEM; - goto err_data; - } snd_soc_dapm_add_path(widget->dapm, data->widget, widget, NULL, NULL); @@ -1026,9 +1041,10 @@ static int dapm_new_dai_link(struct snd_soc_dapm_widget *w) struct snd_kcontrol *kcontrol; struct snd_soc_dapm_context *dapm = w->dapm; struct snd_card *card = dapm->card->snd_card; + struct snd_soc_pcm_runtime *rtd = w->priv; /* create control for links with > 1 config */ - if (w->num_params <= 1) + if (rtd->dai_link->num_params <= 1) return 0; /* add kcontrol */ @@ -1320,14 +1336,13 @@ int dapm_clock_event(struct snd_soc_dapm_widget *w, soc_dapm_async_complete(w->dapm); -#ifdef CONFIG_HAVE_CLK if (SND_SOC_DAPM_EVENT_ON(event)) { return clk_prepare_enable(w->clk); } else { clk_disable_unprepare(w->clk); return 0; } -#endif + return 0; } EXPORT_SYMBOL_GPL(dapm_clock_event); @@ -1953,7 +1968,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) dapm_pre_sequence_async(&card->dapm, 0); /* Run other bias changes in parallel */ list_for_each_entry(d, &card->dapm_list, list) { - if (d != &card->dapm) + if (d != &card->dapm && d->bias_level != d->target_bias_level) async_schedule_domain(dapm_pre_sequence_async, d, &async_domain); } @@ -1977,7 +1992,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) /* Run all the bias changes in parallel */ list_for_each_entry(d, &card->dapm_list, list) { - if (d != &card->dapm) + if (d != &card->dapm && d->bias_level != d->target_bias_level) async_schedule_domain(dapm_post_sequence_async, d, &async_domain); } @@ -2028,19 +2043,19 @@ static ssize_t dapm_widget_power_read_file(struct file *file, out = is_connected_output_ep(w, NULL, NULL); } - ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", + ret = scnprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", w->name, w->power ? "On" : "Off", w->force ? " (forced)" : "", in, out); if (w->reg >= 0) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, " - R%d(0x%x) mask 0x%x", w->reg, w->reg, w->mask << w->shift); - ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n"); + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\n"); if (w->sname) - ret += snprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n", + ret += scnprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n", w->sname, w->active ? "active" : "inactive"); @@ -2053,7 +2068,7 @@ static ssize_t dapm_widget_power_read_file(struct file *file, if (!p->connect) continue; - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, " %s \"%s\" \"%s\"\n", (rdir == SND_SOC_DAPM_DIR_IN) ? "in" : "out", p->name ? p->name : "static", @@ -2371,12 +2386,13 @@ static ssize_t dapm_widget_show(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); + struct snd_soc_dai *codec_dai; int i, count = 0; mutex_lock(&rtd->card->dapm_mutex); - for (i = 0; i < rtd->num_codecs; i++) { - struct snd_soc_component *cmpnt = rtd->codec_dais[i]->component; + for_each_rtd_codec_dai(rtd, i, codec_dai) { + struct snd_soc_component *cmpnt = codec_dai->component; count += dapm_widget_show_component(cmpnt, buf + count); } @@ -3426,35 +3442,6 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch); struct snd_soc_dapm_widget * -snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, - const struct snd_soc_dapm_widget *widget) -{ - struct snd_soc_dapm_widget *w; - - mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - w = snd_soc_dapm_new_control_unlocked(dapm, widget); - /* Do not nag about probe deferrals */ - if (IS_ERR(w)) { - int ret = PTR_ERR(w); - - if (ret != -EPROBE_DEFER) - dev_err(dapm->dev, - "ASoC: Failed to create DAPM control %s (%d)\n", - widget->name, ret); - goto out_unlock; - } - if (!w) - dev_err(dapm->dev, - "ASoC: Failed to create DAPM control %s\n", - widget->name); - -out_unlock: - mutex_unlock(&dapm->card->dapm_mutex); - return w; -} -EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control); - -struct snd_soc_dapm_widget * snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_widget *widget) { @@ -3464,53 +3451,37 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, int ret; if ((w = dapm_cnew_widget(widget)) == NULL) - return NULL; + return ERR_PTR(-ENOMEM); switch (w->id) { case snd_soc_dapm_regulator_supply: w->regulator = devm_regulator_get(dapm->dev, w->name); if (IS_ERR(w->regulator)) { ret = PTR_ERR(w->regulator); - if (ret == -EPROBE_DEFER) - return ERR_PTR(ret); - dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", - w->name, ret); - return NULL; + goto request_failed; } if (w->on_val & SND_SOC_DAPM_REGULATOR_BYPASS) { ret = regulator_allow_bypass(w->regulator, true); if (ret != 0) - dev_warn(w->dapm->dev, + dev_warn(dapm->dev, "ASoC: Failed to bypass %s: %d\n", w->name, ret); } break; case snd_soc_dapm_pinctrl: w->pinctrl = devm_pinctrl_get(dapm->dev); - if (IS_ERR_OR_NULL(w->pinctrl)) { + if (IS_ERR(w->pinctrl)) { ret = PTR_ERR(w->pinctrl); - if (ret == -EPROBE_DEFER) - return ERR_PTR(ret); - dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", - w->name, ret); - return NULL; + goto request_failed; } break; case snd_soc_dapm_clock_supply: -#ifdef CONFIG_CLKDEV_LOOKUP w->clk = devm_clk_get(dapm->dev, w->name); if (IS_ERR(w->clk)) { ret = PTR_ERR(w->clk); - if (ret == -EPROBE_DEFER) - return ERR_PTR(ret); - dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", - w->name, ret); - return NULL; + goto request_failed; } -#else - return NULL; -#endif break; default: break; @@ -3523,7 +3494,7 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, w->name = kstrdup_const(widget->name, GFP_KERNEL); if (w->name == NULL) { kfree(w); - return NULL; + return ERR_PTR(-ENOMEM); } switch (w->id) { @@ -3600,7 +3571,37 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, /* machine layer sets up unconnected pins and insertions */ w->connected = 1; return w; + +request_failed: + if (ret != -EPROBE_DEFER) + dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", + w->name, ret); + + return ERR_PTR(ret); +} + +/** + * snd_soc_dapm_new_control - create new dapm control + * @dapm: DAPM context + * @widget: widget template + * + * Creates new DAPM control based upon a template. + * + * Returns a widget pointer on success or an error pointer on failure + */ +struct snd_soc_dapm_widget * +snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, + const struct snd_soc_dapm_widget *widget) +{ + struct snd_soc_dapm_widget *w; + + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + w = snd_soc_dapm_new_control_unlocked(dapm, widget); + mutex_unlock(&dapm->card->dapm_mutex); + + return w; } +EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control); /** * snd_soc_dapm_new_controls - create new dapm controls @@ -3625,19 +3626,6 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm, w = snd_soc_dapm_new_control_unlocked(dapm, widget); if (IS_ERR(w)) { ret = PTR_ERR(w); - /* Do not nag about probe deferrals */ - if (ret == -EPROBE_DEFER) - break; - dev_err(dapm->dev, - "ASoC: Failed to create DAPM control %s (%d)\n", - widget->name, ret); - break; - } - if (!w) { - dev_err(dapm->dev, - "ASoC: Failed to create DAPM control %s\n", - widget->name); - ret = -ENOMEM; break; } widget++; @@ -3650,32 +3638,23 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls); static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_dapm_path *source_p, *sink_p; + struct snd_soc_dapm_path *path; struct snd_soc_dai *source, *sink; struct snd_soc_pcm_runtime *rtd = w->priv; - const struct snd_soc_pcm_stream *config = w->params + w->params_select; + const struct snd_soc_pcm_stream *config; struct snd_pcm_substream substream; struct snd_pcm_hw_params *params = NULL; struct snd_pcm_runtime *runtime = NULL; unsigned int fmt; - int ret; + int ret = 0; + + config = rtd->dai_link->params + rtd->params_select; if (WARN_ON(!config) || WARN_ON(list_empty(&w->edges[SND_SOC_DAPM_DIR_OUT]) || list_empty(&w->edges[SND_SOC_DAPM_DIR_IN]))) return -EINVAL; - /* We only support a single source and sink, pick the first */ - source_p = list_first_entry(&w->edges[SND_SOC_DAPM_DIR_OUT], - struct snd_soc_dapm_path, - list_node[SND_SOC_DAPM_DIR_OUT]); - sink_p = list_first_entry(&w->edges[SND_SOC_DAPM_DIR_IN], - struct snd_soc_dapm_path, - list_node[SND_SOC_DAPM_DIR_IN]); - - source = source_p->source->priv; - sink = sink_p->sink->priv; - /* Be a little careful as we don't want to overflow the mask array */ if (config->formats) { fmt = ffs(config->formats) - 1; @@ -3717,59 +3696,95 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: substream.stream = SNDRV_PCM_STREAM_CAPTURE; - if (source->driver->ops->startup) { - ret = source->driver->ops->startup(&substream, source); - if (ret < 0) { - dev_err(source->dev, - "ASoC: startup() failed: %d\n", ret); - goto out; + snd_soc_dapm_widget_for_each_source_path(w, path) { + source = path->source->priv; + + if (source->driver->ops->startup) { + ret = source->driver->ops->startup(&substream, + source); + if (ret < 0) { + dev_err(source->dev, + "ASoC: startup() failed: %d\n", + ret); + goto out; + } + source->active++; } - source->active++; + ret = soc_dai_hw_params(&substream, params, source); + if (ret < 0) + goto out; } - ret = soc_dai_hw_params(&substream, params, source); - if (ret < 0) - goto out; substream.stream = SNDRV_PCM_STREAM_PLAYBACK; - if (sink->driver->ops->startup) { - ret = sink->driver->ops->startup(&substream, sink); - if (ret < 0) { - dev_err(sink->dev, - "ASoC: startup() failed: %d\n", ret); - goto out; + snd_soc_dapm_widget_for_each_sink_path(w, path) { + sink = path->sink->priv; + + if (sink->driver->ops->startup) { + ret = sink->driver->ops->startup(&substream, + sink); + if (ret < 0) { + dev_err(sink->dev, + "ASoC: startup() failed: %d\n", + ret); + goto out; + } + sink->active++; } - sink->active++; + ret = soc_dai_hw_params(&substream, params, sink); + if (ret < 0) + goto out; } - ret = soc_dai_hw_params(&substream, params, sink); - if (ret < 0) - goto out; break; case SND_SOC_DAPM_POST_PMU: - ret = snd_soc_dai_digital_mute(sink, 0, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret != 0 && ret != -ENOTSUPP) - dev_warn(sink->dev, "ASoC: Failed to unmute: %d\n", ret); - ret = 0; + snd_soc_dapm_widget_for_each_sink_path(w, path) { + sink = path->sink->priv; + + ret = snd_soc_dai_digital_mute(sink, 0, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret != 0 && ret != -ENOTSUPP) + dev_warn(sink->dev, + "ASoC: Failed to unmute: %d\n", ret); + ret = 0; + } break; case SND_SOC_DAPM_PRE_PMD: - ret = snd_soc_dai_digital_mute(sink, 1, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret != 0 && ret != -ENOTSUPP) - dev_warn(sink->dev, "ASoC: Failed to mute: %d\n", ret); - ret = 0; + snd_soc_dapm_widget_for_each_sink_path(w, path) { + sink = path->sink->priv; + + ret = snd_soc_dai_digital_mute(sink, 1, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret != 0 && ret != -ENOTSUPP) + dev_warn(sink->dev, + "ASoC: Failed to mute: %d\n", ret); + ret = 0; + } - source->active--; - if (source->driver->ops->shutdown) { - substream.stream = SNDRV_PCM_STREAM_CAPTURE; - source->driver->ops->shutdown(&substream, source); + substream.stream = SNDRV_PCM_STREAM_CAPTURE; + snd_soc_dapm_widget_for_each_source_path(w, path) { + source = path->source->priv; + + if (source->driver->ops->hw_free) + source->driver->ops->hw_free(&substream, + source); + + source->active--; + if (source->driver->ops->shutdown) + source->driver->ops->shutdown(&substream, + source); } - sink->active--; - if (sink->driver->ops->shutdown) { - substream.stream = SNDRV_PCM_STREAM_PLAYBACK; - sink->driver->ops->shutdown(&substream, sink); + substream.stream = SNDRV_PCM_STREAM_PLAYBACK; + snd_soc_dapm_widget_for_each_sink_path(w, path) { + sink = path->sink->priv; + + if (sink->driver->ops->hw_free) + sink->driver->ops->hw_free(&substream, sink); + + sink->active--; + if (sink->driver->ops->shutdown) + sink->driver->ops->shutdown(&substream, sink); } break; @@ -3788,8 +3803,9 @@ static int snd_soc_dapm_dai_link_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + struct snd_soc_pcm_runtime *rtd = w->priv; - ucontrol->value.enumerated.item[0] = w->params_select; + ucontrol->value.enumerated.item[0] = rtd->params_select; return 0; } @@ -3798,18 +3814,19 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + struct snd_soc_pcm_runtime *rtd = w->priv; /* Can't change the config when widget is already powered */ if (w->power) return -EBUSY; - if (ucontrol->value.enumerated.item[0] == w->params_select) + if (ucontrol->value.enumerated.item[0] == rtd->params_select) return 0; - if (ucontrol->value.enumerated.item[0] >= w->num_params) + if (ucontrol->value.enumerated.item[0] >= rtd->dai_link->num_params) return -EINVAL; - w->params_select = ucontrol->value.enumerated.item[0]; + rtd->params_select = ucontrol->value.enumerated.item[0]; return 0; } @@ -3896,12 +3913,10 @@ outfree_w_param: return NULL; } -int snd_soc_dapm_new_pcm(struct snd_soc_card *card, - struct snd_soc_pcm_runtime *rtd, - const struct snd_soc_pcm_stream *params, - unsigned int num_params, - struct snd_soc_dapm_widget *source, - struct snd_soc_dapm_widget *sink) +static struct snd_soc_dapm_widget * +snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) { struct snd_soc_dapm_widget template; struct snd_soc_dapm_widget *w; @@ -3913,7 +3928,7 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", source->name, sink->name); if (!link_name) - return -ENOMEM; + return ERR_PTR(-ENOMEM); memset(&template, 0, sizeof(template)); template.reg = SND_SOC_NOPM; @@ -3925,9 +3940,10 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, template.kcontrol_news = NULL; /* allocate memory for control, only in case of multiple configs */ - if (num_params > 1) { - w_param_text = devm_kcalloc(card->dev, num_params, - sizeof(char *), GFP_KERNEL); + if (rtd->dai_link->num_params > 1) { + w_param_text = devm_kcalloc(card->dev, + rtd->dai_link->num_params, + sizeof(char *), GFP_KERNEL); if (!w_param_text) { ret = -ENOMEM; goto param_fail; @@ -3936,7 +3952,9 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, template.num_kcontrols = 1; template.kcontrol_news = snd_soc_dapm_alloc_kcontrol(card, - link_name, params, num_params, + link_name, + rtd->dai_link->params, + rtd->dai_link->num_params, w_param_text, &private_value); if (!template.kcontrol_news) { ret = -ENOMEM; @@ -3950,37 +3968,20 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); if (IS_ERR(w)) { ret = PTR_ERR(w); - /* Do not nag about probe deferrals */ - if (ret != -EPROBE_DEFER) - dev_err(card->dev, - "ASoC: Failed to create %s widget (%d)\n", - link_name, ret); - goto outfree_kcontrol_news; - } - if (!w) { - dev_err(card->dev, "ASoC: Failed to create %s widget\n", - link_name); - ret = -ENOMEM; goto outfree_kcontrol_news; } - w->params = params; - w->num_params = num_params; w->priv = rtd; - ret = snd_soc_dapm_add_path(&card->dapm, source, w, NULL, NULL); - if (ret) - goto outfree_w; - return snd_soc_dapm_add_path(&card->dapm, w, sink, NULL, NULL); + return w; -outfree_w: - devm_kfree(card->dev, w); outfree_kcontrol_news: devm_kfree(card->dev, (void *)template.kcontrol_news); - snd_soc_dapm_free_kcontrol(card, &private_value, num_params, w_param_text); + snd_soc_dapm_free_kcontrol(card, &private_value, + rtd->dai_link->num_params, w_param_text); param_fail: devm_kfree(card->dev, link_name); - return ret; + return ERR_PTR(ret); } int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, @@ -4003,21 +4004,8 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, template.name); w = snd_soc_dapm_new_control_unlocked(dapm, &template); - if (IS_ERR(w)) { - int ret = PTR_ERR(w); - - /* Do not nag about probe deferrals */ - if (ret != -EPROBE_DEFER) - dev_err(dapm->dev, - "ASoC: Failed to create %s widget (%d)\n", - dai->driver->playback.stream_name, ret); - return ret; - } - if (!w) { - dev_err(dapm->dev, "ASoC: Failed to create %s widget\n", - dai->driver->playback.stream_name); - return -ENOMEM; - } + if (IS_ERR(w)) + return PTR_ERR(w); w->priv = dai; dai->playback_widget = w; @@ -4032,21 +4020,8 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, template.name); w = snd_soc_dapm_new_control_unlocked(dapm, &template); - if (IS_ERR(w)) { - int ret = PTR_ERR(w); - - /* Do not nag about probe deferrals */ - if (ret != -EPROBE_DEFER) - dev_err(dapm->dev, - "ASoC: Failed to create %s widget (%d)\n", - dai->driver->playback.stream_name, ret); - return ret; - } - if (!w) { - dev_err(dapm->dev, "ASoC: Failed to create %s widget\n", - dai->driver->capture.stream_name); - return -ENOMEM; - } + if (IS_ERR(w)) + return PTR_ERR(w); w->priv = dai; dai->capture_widget = w; @@ -4115,34 +4090,79 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dapm_widget *sink, *source; + struct snd_soc_dai *codec_dai; + struct snd_soc_dapm_widget *playback = NULL, *capture = NULL; + struct snd_soc_dapm_widget *codec, *playback_cpu, *capture_cpu; int i; - for (i = 0; i < rtd->num_codecs; i++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; + if (rtd->dai_link->params) { + playback_cpu = cpu_dai->capture_widget; + capture_cpu = cpu_dai->playback_widget; + } else { + playback = cpu_dai->playback_widget; + capture = cpu_dai->capture_widget; + playback_cpu = playback; + capture_cpu = capture; + } + + for_each_rtd_codec_dai(rtd, i, codec_dai) { /* connect BE DAI playback if widgets are valid */ - if (codec_dai->playback_widget && cpu_dai->playback_widget) { - source = cpu_dai->playback_widget; - sink = codec_dai->playback_widget; + codec = codec_dai->playback_widget; + + if (playback_cpu && codec) { + if (!playback) { + playback = snd_soc_dapm_new_dai(card, rtd, + playback_cpu, + codec); + if (IS_ERR(playback)) { + dev_err(rtd->dev, + "ASoC: Failed to create DAI %s: %ld\n", + codec_dai->name, + PTR_ERR(playback)); + continue; + } + + snd_soc_dapm_add_path(&card->dapm, playback_cpu, + playback, NULL, NULL); + } + dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", - cpu_dai->component->name, source->name, - codec_dai->component->name, sink->name); + cpu_dai->component->name, playback_cpu->name, + codec_dai->component->name, codec->name); - snd_soc_dapm_add_path(&card->dapm, source, sink, - NULL, NULL); + snd_soc_dapm_add_path(&card->dapm, playback, codec, + NULL, NULL); } + } + for_each_rtd_codec_dai(rtd, i, codec_dai) { /* connect BE DAI capture if widgets are valid */ - if (codec_dai->capture_widget && cpu_dai->capture_widget) { - source = codec_dai->capture_widget; - sink = cpu_dai->capture_widget; + codec = codec_dai->capture_widget; + + if (codec && capture_cpu) { + if (!capture) { + capture = snd_soc_dapm_new_dai(card, rtd, + codec, + capture_cpu); + if (IS_ERR(capture)) { + dev_err(rtd->dev, + "ASoC: Failed to create DAI %s: %ld\n", + codec_dai->name, + PTR_ERR(capture)); + continue; + } + + snd_soc_dapm_add_path(&card->dapm, capture, + capture_cpu, NULL, NULL); + } + dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", - codec_dai->component->name, source->name, - cpu_dai->component->name, sink->name); + codec_dai->component->name, codec->name, + cpu_dai->component->name, capture_cpu->name); - snd_soc_dapm_add_path(&card->dapm, source, sink, - NULL, NULL); + snd_soc_dapm_add_path(&card->dapm, codec, capture, + NULL, NULL); } } } @@ -4192,12 +4212,12 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) struct snd_soc_pcm_runtime *rtd; /* for each BE DAI link... */ - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { /* * dynamic FE links have no fixed DAI mapping. * CODEC<->CODEC links have no direct connection. */ - if (rtd->dai_link->dynamic || rtd->dai_link->params) + if (rtd->dai_link->dynamic) continue; dapm_connect_dai_link_widgets(card, rtd); @@ -4207,11 +4227,12 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, int event) { + struct snd_soc_dai *codec_dai; int i; soc_dapm_dai_stream_event(rtd->cpu_dai, stream, event); - for (i = 0; i < rtd->num_codecs; i++) - soc_dapm_dai_stream_event(rtd->codec_dais[i], stream, event); + for_each_rtd_codec_dai(rtd, i, codec_dai) + soc_dapm_dai_stream_event(codec_dai, stream, event); dapm_power_widgets(rtd->card, event); } diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 592efb370c44..f4dc3d445aae 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -373,7 +373,7 @@ int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol, unsigned int rshift = mc->rshift; int max = mc->max; int min = mc->min; - unsigned int mask = (1 << (fls(min + max) - 1)) - 1; + unsigned int mask = (1U << (fls(min + max) - 1)) - 1; unsigned int val; int ret; @@ -418,7 +418,7 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, unsigned int rshift = mc->rshift; int max = mc->max; int min = mc->min; - unsigned int mask = (1 << (fls(min + max) - 1)) - 1; + unsigned int mask = (1U << (fls(min + max) - 1)) - 1; int err = 0; unsigned int val, val_mask, val2 = 0; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e8b98bfd4cf1..03f36e534050 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -59,25 +59,26 @@ static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream) void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; int i; lockdep_assert_held(&rtd->pcm_mutex); if (stream == SNDRV_PCM_STREAM_PLAYBACK) { cpu_dai->playback_active++; - for (i = 0; i < rtd->num_codecs; i++) - rtd->codec_dais[i]->playback_active++; + for_each_rtd_codec_dai(rtd, i, codec_dai) + codec_dai->playback_active++; } else { cpu_dai->capture_active++; - for (i = 0; i < rtd->num_codecs; i++) - rtd->codec_dais[i]->capture_active++; + for_each_rtd_codec_dai(rtd, i, codec_dai) + codec_dai->capture_active++; } cpu_dai->active++; cpu_dai->component->active++; - for (i = 0; i < rtd->num_codecs; i++) { - rtd->codec_dais[i]->active++; - rtd->codec_dais[i]->component->active++; + for_each_rtd_codec_dai(rtd, i, codec_dai) { + codec_dai->active++; + codec_dai->component->active++; } } @@ -94,25 +95,26 @@ void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream) void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; int i; lockdep_assert_held(&rtd->pcm_mutex); if (stream == SNDRV_PCM_STREAM_PLAYBACK) { cpu_dai->playback_active--; - for (i = 0; i < rtd->num_codecs; i++) - rtd->codec_dais[i]->playback_active--; + for_each_rtd_codec_dai(rtd, i, codec_dai) + codec_dai->playback_active--; } else { cpu_dai->capture_active--; - for (i = 0; i < rtd->num_codecs; i++) - rtd->codec_dais[i]->capture_active--; + for_each_rtd_codec_dai(rtd, i, codec_dai) + codec_dai->capture_active--; } cpu_dai->active--; cpu_dai->component->active--; - for (i = 0; i < rtd->num_codecs; i++) { - rtd->codec_dais[i]->component->active--; - rtd->codec_dais[i]->active--; + for_each_rtd_codec_dai(rtd, i, codec_dai) { + codec_dai->component->active--; + codec_dai->active--; } } @@ -172,7 +174,7 @@ int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, { struct snd_soc_dpcm *dpcm; - list_for_each_entry(dpcm, &fe->dpcm[dir].be_clients, list_be) { + for_each_dpcm_be(fe, dir, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; @@ -253,6 +255,7 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; unsigned int rate, channels, sample_bits, symmetry, i; rate = params_rate(params); @@ -263,8 +266,8 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, symmetry = cpu_dai->driver->symmetric_rates || rtd->dai_link->symmetric_rates; - for (i = 0; i < rtd->num_codecs; i++) - symmetry |= rtd->codec_dais[i]->driver->symmetric_rates; + for_each_rtd_codec_dai(rtd, i, codec_dai) + symmetry |= codec_dai->driver->symmetric_rates; if (symmetry && cpu_dai->rate && cpu_dai->rate != rate) { dev_err(rtd->dev, "ASoC: unmatched rate symmetry: %d - %d\n", @@ -275,8 +278,8 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, symmetry = cpu_dai->driver->symmetric_channels || rtd->dai_link->symmetric_channels; - for (i = 0; i < rtd->num_codecs; i++) - symmetry |= rtd->codec_dais[i]->driver->symmetric_channels; + for_each_rtd_codec_dai(rtd, i, codec_dai) + symmetry |= codec_dai->driver->symmetric_channels; if (symmetry && cpu_dai->channels && cpu_dai->channels != channels) { dev_err(rtd->dev, "ASoC: unmatched channel symmetry: %d - %d\n", @@ -287,8 +290,8 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, symmetry = cpu_dai->driver->symmetric_samplebits || rtd->dai_link->symmetric_samplebits; - for (i = 0; i < rtd->num_codecs; i++) - symmetry |= rtd->codec_dais[i]->driver->symmetric_samplebits; + for_each_rtd_codec_dai(rtd, i, codec_dai) + symmetry |= codec_dai->driver->symmetric_samplebits; if (symmetry && cpu_dai->sample_bits && cpu_dai->sample_bits != sample_bits) { dev_err(rtd->dev, "ASoC: unmatched sample bits symmetry: %d - %d\n", @@ -304,17 +307,18 @@ static bool soc_pcm_has_symmetry(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai_driver *cpu_driver = rtd->cpu_dai->driver; struct snd_soc_dai_link *link = rtd->dai_link; + struct snd_soc_dai *codec_dai; unsigned int symmetry, i; symmetry = cpu_driver->symmetric_rates || link->symmetric_rates || cpu_driver->symmetric_channels || link->symmetric_channels || cpu_driver->symmetric_samplebits || link->symmetric_samplebits; - for (i = 0; i < rtd->num_codecs; i++) + for_each_rtd_codec_dai(rtd, i, codec_dai) symmetry = symmetry || - rtd->codec_dais[i]->driver->symmetric_rates || - rtd->codec_dais[i]->driver->symmetric_channels || - rtd->codec_dais[i]->driver->symmetric_samplebits; + codec_dai->driver->symmetric_rates || + codec_dai->driver->symmetric_channels || + codec_dai->driver->symmetric_samplebits; return symmetry; } @@ -342,8 +346,7 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream) unsigned int bits = 0, cpu_bits; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { if (codec_dai->driver->playback.sig_bits == 0) { bits = 0; break; @@ -352,8 +355,7 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream) } cpu_bits = cpu_dai->driver->playback.sig_bits; } else { - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { if (codec_dai->driver->capture.sig_bits == 0) { bits = 0; break; @@ -372,6 +374,7 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct snd_pcm_hardware *hw = &runtime->hw; struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai; struct snd_soc_dai_driver *cpu_dai_drv = rtd->cpu_dai->driver; struct snd_soc_dai_driver *codec_dai_drv; struct snd_soc_pcm_stream *codec_stream; @@ -388,7 +391,7 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) cpu_stream = &cpu_dai_drv->capture; /* first calculate min/max only for CODECs in the DAI link */ - for (i = 0; i < rtd->num_codecs; i++) { + for_each_rtd_codec_dai(rtd, i, codec_dai) { /* * Skip CODECs which don't support the current stream type. @@ -399,11 +402,11 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) * bailed out on a higher level, since there would be no * CODEC to support the transfer direction in that case. */ - if (!snd_soc_dai_stream_valid(rtd->codec_dais[i], + if (!snd_soc_dai_stream_valid(codec_dai, substream->stream)) continue; - codec_dai_drv = rtd->codec_dais[i]->driver; + codec_dai_drv = codec_dai->driver; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) codec_stream = &codec_dai_drv->playback; else @@ -482,8 +485,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) int i, ret = 0; pinctrl_pm_select_default_state(cpu_dai->dev); - for (i = 0; i < rtd->num_codecs; i++) - pinctrl_pm_select_default_state(rtd->codec_dais[i]->dev); + for_each_rtd_codec_dai(rtd, i, codec_dai) + pinctrl_pm_select_default_state(codec_dai->dev); for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -520,8 +523,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } component = NULL; - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { if (codec_dai->driver->ops->startup) { ret = codec_dai->driver->ops->startup(substream, codec_dai); @@ -588,10 +590,9 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) goto config_err; } - for (i = 0; i < rtd->num_codecs; i++) { - if (rtd->codec_dais[i]->active) { - ret = soc_pcm_apply_symmetry(substream, - rtd->codec_dais[i]); + for_each_rtd_codec_dai(rtd, i, codec_dai) { + if (codec_dai->active) { + ret = soc_pcm_apply_symmetry(substream, codec_dai); if (ret != 0) goto config_err; } @@ -620,8 +621,7 @@ machine_err: i = rtd->num_codecs; codec_dai_err: - while (--i >= 0) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai_rollback(rtd, i, codec_dai) { if (codec_dai->driver->ops->shutdown) codec_dai->driver->ops->shutdown(substream, codec_dai); } @@ -641,9 +641,9 @@ out: pm_runtime_put_autosuspend(component->dev); } - for (i = 0; i < rtd->num_codecs; i++) { - if (!rtd->codec_dais[i]->active) - pinctrl_pm_select_sleep_state(rtd->codec_dais[i]->dev); + for_each_rtd_codec_dai(rtd, i, codec_dai) { + if (!codec_dai->active) + pinctrl_pm_select_sleep_state(codec_dai->dev); } if (!cpu_dai->active) pinctrl_pm_select_sleep_state(cpu_dai->dev); @@ -701,8 +701,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) if (!cpu_dai->active) cpu_dai->rate = 0; - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { if (!codec_dai->active) codec_dai->rate = 0; } @@ -712,8 +711,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) if (cpu_dai->driver->ops->shutdown) cpu_dai->driver->ops->shutdown(substream, cpu_dai); - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { if (codec_dai->driver->ops->shutdown) codec_dai->driver->ops->shutdown(substream, codec_dai); } @@ -751,9 +749,9 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) pm_runtime_put_autosuspend(component->dev); } - for (i = 0; i < rtd->num_codecs; i++) { - if (!rtd->codec_dais[i]->active) - pinctrl_pm_select_sleep_state(rtd->codec_dais[i]->dev); + for_each_rtd_codec_dai(rtd, i, codec_dai) { + if (!codec_dai->active) + pinctrl_pm_select_sleep_state(codec_dai->dev); } if (!cpu_dai->active) pinctrl_pm_select_sleep_state(cpu_dai->dev); @@ -801,8 +799,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } } - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { if (codec_dai->driver->ops->prepare) { ret = codec_dai->driver->ops->prepare(substream, codec_dai); @@ -834,8 +831,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) snd_soc_dapm_stream_event(rtd, substream->stream, SND_SOC_DAPM_STREAM_START); - for (i = 0; i < rtd->num_codecs; i++) - snd_soc_dai_digital_mute(rtd->codec_dais[i], 0, + for_each_rtd_codec_dai(rtd, i, codec_dai) + snd_soc_dai_digital_mute(codec_dai, 0, substream->stream); snd_soc_dai_digital_mute(cpu_dai, 0, substream->stream); @@ -920,6 +917,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component; struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; int i, ret = 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -932,8 +930,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } - for (i = 0; i < rtd->num_codecs; i++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { struct snd_pcm_hw_params codec_params; /* @@ -1018,8 +1015,7 @@ interface_err: i = rtd->num_codecs; codec_err: - while (--i >= 0) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai_rollback(rtd, i, codec_dai) { if (codec_dai->driver->ops->hw_free) codec_dai->driver->ops->hw_free(substream, codec_dai); codec_dai->rate = 0; @@ -1052,8 +1048,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) cpu_dai->sample_bits = 0; } - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { if (codec_dai->active == 1) { codec_dai->rate = 0; codec_dai->channels = 0; @@ -1062,10 +1057,10 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) } /* apply codec digital mute */ - for (i = 0; i < rtd->num_codecs; i++) { - if ((playback && rtd->codec_dais[i]->playback_active == 1) || - (!playback && rtd->codec_dais[i]->capture_active == 1)) - snd_soc_dai_digital_mute(rtd->codec_dais[i], 1, + for_each_rtd_codec_dai(rtd, i, codec_dai) { + if ((playback && codec_dai->playback_active == 1) || + (!playback && codec_dai->capture_active == 1)) + snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); } @@ -1077,8 +1072,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) soc_pcm_components_hw_free(substream, NULL); /* now free hw params for the DAIs */ - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { if (codec_dai->driver->ops->hw_free) codec_dai->driver->ops->hw_free(substream, codec_dai); } @@ -1099,8 +1093,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_dai *codec_dai; int i, ret; - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { if (codec_dai->driver->ops->trigger) { ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai); @@ -1144,8 +1137,7 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai; int i, ret; - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { if (codec_dai->driver->ops->bespoke_trigger) { ret = codec_dai->driver->ops->bespoke_trigger(substream, cmd, codec_dai); @@ -1199,8 +1191,7 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) if (cpu_dai->driver->ops->delay) delay += cpu_dai->driver->ops->delay(substream, cpu_dai); - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { if (codec_dai->driver->ops->delay) codec_delay = max(codec_delay, codec_dai->driver->ops->delay(substream, @@ -1220,7 +1211,7 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, struct snd_soc_dpcm *dpcm; /* only add new dpcms */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { if (dpcm->be == be && dpcm->fe == fe) return 0; } @@ -1261,7 +1252,7 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, be_substream = snd_soc_dpcm_get_substream(be, stream); - list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) { + for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) continue; @@ -1281,7 +1272,7 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm, *d; - list_for_each_entry_safe(dpcm, d, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be_safe(fe, stream, dpcm, d) { dev_dbg(fe->dev, "ASoC: BE %s disconnect check for %s\n", stream ? "capture" : "playback", dpcm->be->dai_link->name); @@ -1310,12 +1301,13 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, struct snd_soc_dapm_widget *widget, int stream) { struct snd_soc_pcm_runtime *be; + struct snd_soc_dai *dai; int i; dev_dbg(card->dev, "ASoC: find BE for widget %s\n", widget->name); if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - list_for_each_entry(be, &card->rtd_list, list) { + for_each_card_rtds(card, be) { if (!be->dai_link->no_pcm) continue; @@ -1327,15 +1319,14 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, if (be->cpu_dai->playback_widget == widget) return be; - for (i = 0; i < be->num_codecs; i++) { - struct snd_soc_dai *dai = be->codec_dais[i]; + for_each_rtd_codec_dai(be, i, dai) { if (dai->playback_widget == widget) return be; } } } else { - list_for_each_entry(be, &card->rtd_list, list) { + for_each_card_rtds(card, be) { if (!be->dai_link->no_pcm) continue; @@ -1347,8 +1338,7 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, if (be->cpu_dai->capture_widget == widget) return be; - for (i = 0; i < be->num_codecs; i++) { - struct snd_soc_dai *dai = be->codec_dais[i]; + for_each_rtd_codec_dai(be, i, dai) { if (dai->capture_widget == widget) return be; } @@ -1388,32 +1378,31 @@ static bool dpcm_end_walk_at_be(struct snd_soc_dapm_widget *widget, { struct snd_soc_card *card = widget->dapm->card; struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *dai; int i; if (dir == SND_SOC_DAPM_DIR_OUT) { - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { if (!rtd->dai_link->no_pcm) continue; if (rtd->cpu_dai->playback_widget == widget) return true; - for (i = 0; i < rtd->num_codecs; ++i) { - struct snd_soc_dai *dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, dai) { if (dai->playback_widget == widget) return true; } } } else { /* SND_SOC_DAPM_DIR_IN */ - list_for_each_entry(rtd, &card->rtd_list, list) { + for_each_card_rtds(card, rtd) { if (!rtd->dai_link->no_pcm) continue; if (rtd->cpu_dai->capture_widget == widget) return true; - for (i = 0; i < rtd->num_codecs; ++i) { - struct snd_soc_dai *dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, dai) { if (dai->capture_widget == widget) return true; } @@ -1445,10 +1434,11 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dpcm *dpcm; struct snd_soc_dapm_widget_list *list = *list_; struct snd_soc_dapm_widget *widget; + struct snd_soc_dai *dai; int prune = 0; /* Destroy any old FE <--> BE connections */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { unsigned int i; /* is there a valid CPU DAI widget for this BE */ @@ -1459,8 +1449,7 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, continue; /* is there a valid CODEC DAI widget for this BE */ - for (i = 0; i < dpcm->be->num_codecs; i++) { - struct snd_soc_dai *dai = dpcm->be->codec_dais[i]; + for_each_rtd_codec_dai(dpcm->be, i, dai) { widget = dai_get_widget(dai, stream); /* prune the BE if it's no longer in our active list */ @@ -1555,7 +1544,7 @@ void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) + for_each_dpcm_be(fe, stream, dpcm) dpcm->be->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; } @@ -1566,7 +1555,7 @@ static void dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe, struct snd_soc_dpcm *dpcm; /* disable any enabled and non active backends */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *be_substream = @@ -1595,7 +1584,7 @@ int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) int err, count = 0; /* only startup BE DAIs that are either sinks or sources to this FE DAI */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *be_substream = @@ -1649,7 +1638,7 @@ int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) unwind: /* disable any enabled and non active backends */ - list_for_each_entry_continue_reverse(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be_rollback(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *be_substream = snd_soc_dpcm_get_substream(be, stream); @@ -1680,7 +1669,7 @@ static void dpcm_init_runtime_hw(struct snd_pcm_runtime *runtime, struct snd_soc_pcm_stream *stream) { runtime->hw.rate_min = stream->rate_min; - runtime->hw.rate_max = stream->rate_max; + runtime->hw.rate_max = min_not_zero(stream->rate_max, UINT_MAX); runtime->hw.channels_min = stream->channels_min; runtime->hw.channels_max = stream->channels_max; if (runtime->hw.formats) @@ -1695,6 +1684,7 @@ static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *fe = substream->private_data; struct snd_soc_dpcm *dpcm; + struct snd_soc_dai *dai; int stream = substream->stream; if (!fe->dai_link->dpcm_merged_format) @@ -1705,22 +1695,21 @@ static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream, * if FE want to use it (= dpcm_merged_format) */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_soc_dai_driver *codec_dai_drv; struct snd_soc_pcm_stream *codec_stream; int i; - for (i = 0; i < be->num_codecs; i++) { + for_each_rtd_codec_dai(be, i, dai) { /* * Skip CODECs which don't support the current stream * type. See soc_pcm_init_runtime_hw() for more details */ - if (!snd_soc_dai_stream_valid(be->codec_dais[i], - stream)) + if (!snd_soc_dai_stream_valid(dai, stream)) continue; - codec_dai_drv = be->codec_dais[i]->driver; + codec_dai_drv = dai->driver; if (stream == SNDRV_PCM_STREAM_PLAYBACK) codec_stream = &codec_dai_drv->playback; else @@ -1747,7 +1736,7 @@ static void dpcm_runtime_merge_chan(struct snd_pcm_substream *substream, * if FE want to use it (= dpcm_merged_chan) */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_soc_dai_driver *cpu_dai_drv = be->cpu_dai->driver; struct snd_soc_dai_driver *codec_dai_drv; @@ -1799,12 +1788,13 @@ static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream, * if FE want to use it (= dpcm_merged_chan) */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_soc_dai_driver *cpu_dai_drv = be->cpu_dai->driver; struct snd_soc_dai_driver *codec_dai_drv; struct snd_soc_pcm_stream *codec_stream; struct snd_soc_pcm_stream *cpu_stream; + struct snd_soc_dai *dai; int i; if (stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -1816,16 +1806,15 @@ static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream, *rate_max = min_not_zero(*rate_max, cpu_stream->rate_max); *rates = snd_pcm_rate_mask_intersect(*rates, cpu_stream->rates); - for (i = 0; i < be->num_codecs; i++) { + for_each_rtd_codec_dai(be, i, dai) { /* * Skip CODECs which don't support the current stream * type. See soc_pcm_init_runtime_hw() for more details */ - if (!snd_soc_dai_stream_valid(be->codec_dais[i], - stream)) + if (!snd_soc_dai_stream_valid(dai, stream)) continue; - codec_dai_drv = be->codec_dais[i]->driver; + codec_dai_drv = dai->driver; if (stream == SNDRV_PCM_STREAM_PLAYBACK) codec_stream = &codec_dai_drv->playback; else @@ -1902,11 +1891,12 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream, } /* apply symmetry for BE */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *be_substream = snd_soc_dpcm_get_substream(be, stream); struct snd_soc_pcm_runtime *rtd = be_substream->private_data; + struct snd_soc_dai *codec_dai; int i; if (rtd->dai_link->be_hw_params_fixup) @@ -1923,10 +1913,10 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream, return err; } - for (i = 0; i < rtd->num_codecs; i++) { - if (rtd->codec_dais[i]->active) { + for_each_rtd_codec_dai(rtd, i, codec_dai) { + if (codec_dai->active) { err = soc_pcm_apply_symmetry(fe_substream, - rtd->codec_dais[i]); + codec_dai); if (err < 0) return err; } @@ -1986,7 +1976,7 @@ int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream) struct snd_soc_dpcm *dpcm; /* only shutdown BEs that are either sinks or sources to this FE DAI */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *be_substream = @@ -2050,7 +2040,7 @@ int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) /* only hw_params backends that are either sinks or sources * to this frontend DAI */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *be_substream = @@ -2119,7 +2109,7 @@ int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) struct snd_soc_dpcm *dpcm; int ret; - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *be_substream = @@ -2170,7 +2160,7 @@ int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) unwind: /* disable any enabled and non active backends */ - list_for_each_entry_continue_reverse(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be_rollback(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *be_substream = snd_soc_dpcm_get_substream(be, stream); @@ -2250,7 +2240,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dpcm *dpcm; int ret = 0; - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *be_substream = @@ -2436,7 +2426,7 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) struct snd_soc_dpcm *dpcm; int ret = 0; - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *be_substream = @@ -2646,7 +2636,7 @@ close: dpcm_be_dai_shutdown(fe, stream); disconnect: /* disconnect any non started BEs */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; @@ -2771,14 +2761,14 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card) mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME); /* shutdown all old paths first */ - list_for_each_entry(fe, &card->rtd_list, list) { + for_each_card_rtds(card, fe) { ret = soc_dpcm_fe_runtime_update(fe, 0); if (ret) goto out; } /* bring new paths up */ - list_for_each_entry(fe, &card->rtd_list, list) { + for_each_card_rtds(card, fe) { ret = soc_dpcm_fe_runtime_update(fe, 1); if (ret) goto out; @@ -2791,10 +2781,9 @@ out: int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute) { struct snd_soc_dpcm *dpcm; - struct list_head *clients = - &fe->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients; + struct snd_soc_dai *dai; - list_for_each_entry(dpcm, clients, list_be) { + for_each_dpcm_be(fe, SNDRV_PCM_STREAM_PLAYBACK, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; int i; @@ -2802,8 +2791,7 @@ int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute) if (be->dai_link->ignore_suspend) continue; - for (i = 0; i < be->num_codecs; i++) { - struct snd_soc_dai *dai = be->codec_dais[i]; + for_each_rtd_codec_dai(be, i, dai) { struct snd_soc_dai_driver *drv = dai->driver; dev_dbg(be->dev, "ASoC: BE digital mute %s\n", @@ -2844,7 +2832,7 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) ret = dpcm_fe_dai_startup(fe_substream); if (ret < 0) { /* clean up all links */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) + for_each_dpcm_be(fe, stream, dpcm) dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; dpcm_be_disconnect(fe, stream); @@ -2867,7 +2855,7 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) ret = dpcm_fe_dai_shutdown(fe_substream); /* mark FE's links ready to prune */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) + for_each_dpcm_be(fe, stream, dpcm) dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; dpcm_be_disconnect(fe, stream); @@ -3041,8 +3029,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) playback = rtd->dai_link->dpcm_playback; capture = rtd->dai_link->dpcm_capture; } else { - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { if (codec_dai->driver->playback.channels_min) playback = 1; if (codec_dai->driver->capture.channels_min) @@ -3230,7 +3217,7 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, struct snd_soc_dpcm *dpcm; int state; - list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) { + for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) continue; @@ -3257,7 +3244,7 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, struct snd_soc_dpcm *dpcm; int state; - list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) { + for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) continue; @@ -3337,7 +3324,7 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, goto out; } - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; params = &dpcm->hw_params; diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 66e77e020745..731b963b6995 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -502,6 +502,7 @@ static void remove_dai(struct snd_soc_component *comp, { struct snd_soc_dai_driver *dai_drv = container_of(dobj, struct snd_soc_dai_driver, dobj); + struct snd_soc_dai *dai; if (pass != SOC_TPLG_PASS_PCM_DAI) return; @@ -509,6 +510,10 @@ static void remove_dai(struct snd_soc_component *comp, if (dobj->ops && dobj->ops->dai_unload) dobj->ops->dai_unload(comp, dobj); + list_for_each_entry(dai, &comp->dai_list, list) + if (dai->driver == dai_drv) + dai->driver = NULL; + kfree(dai_drv->name); list_del(&dobj->list); kfree(dai_drv); @@ -993,7 +998,7 @@ static int soc_tplg_denum_create(struct soc_tplg *tplg, unsigned int count, kfree(se); continue; } - /* fall through and create texts */ + /* fall through */ case SND_SOC_TPLG_CTL_ENUM: case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: @@ -1310,7 +1315,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( ec->hdr.name); goto err_se; } - /* fall through to create texts */ + /* fall through */ case SND_SOC_TPLG_CTL_ENUM: case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: @@ -1565,17 +1570,6 @@ widget: widget = snd_soc_dapm_new_control_unlocked(dapm, &template); if (IS_ERR(widget)) { ret = PTR_ERR(widget); - /* Do not nag about probe deferrals */ - if (ret != -EPROBE_DEFER) - dev_err(tplg->dev, - "ASoC: failed to create widget %s controls (%d)\n", - w->name, ret); - goto hdr_err; - } - if (widget == NULL) { - dev_err(tplg->dev, "ASoC: failed to create widget %s controls\n", - w->name); - ret = -ENOMEM; goto hdr_err; } @@ -2493,6 +2487,7 @@ int snd_soc_tplg_component_load(struct snd_soc_component *comp, struct snd_soc_tplg_ops *ops, const struct firmware *fw, u32 id) { struct soc_tplg tplg; + int ret; /* setup parsing context */ memset(&tplg, 0, sizeof(tplg)); @@ -2506,7 +2501,12 @@ int snd_soc_tplg_component_load(struct snd_soc_component *comp, tplg.bytes_ext_ops = ops->bytes_ext_ops; tplg.bytes_ext_ops_count = ops->bytes_ext_ops_count; - return soc_tplg_load(&tplg); + ret = soc_tplg_load(&tplg); + /* free the created components if fail to load topology */ + if (ret) + snd_soc_tplg_component_remove(comp, SND_SOC_TPLG_INDEX_ALL); + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_tplg_component_load); diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index e0c93496c0cd..e3b9dd634c6d 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -273,13 +273,13 @@ static int dummy_dma_open(struct snd_pcm_substream *substream) return 0; } -static const struct snd_pcm_ops dummy_dma_ops = { +static const struct snd_pcm_ops snd_dummy_dma_ops = { .open = dummy_dma_open, .ioctl = snd_pcm_lib_ioctl, }; static const struct snd_soc_component_driver dummy_platform = { - .ops = &dummy_dma_ops, + .ops = &snd_dummy_dma_ops, }; static const struct snd_soc_component_driver dummy_codec = { diff --git a/sound/soc/stm/Kconfig b/sound/soc/stm/Kconfig index 9b2681397dba..c66ffa72057e 100644 --- a/sound/soc/stm/Kconfig +++ b/sound/soc/stm/Kconfig @@ -3,6 +3,7 @@ menu "STMicroelectronics STM32 SOC audio support" config SND_SOC_STM32_SAI tristate "STM32 SAI interface (Serial Audio Interface) support" depends on (ARCH_STM32 && OF) || COMPILE_TEST + depends on COMMON_CLK depends on SND_SOC select SND_SOC_GENERIC_DMAENGINE_PCM select REGMAP_MMIO diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index f22654253c43..bcb35cae2a2c 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -74,14 +74,14 @@ static int stm32_sai_sync_conf_provider(struct stm32_sai_data *sai, int synco) return ret; } - dev_dbg(&sai->pdev->dev, "Set %s%s as synchro provider\n", - sai->pdev->dev.of_node->name, + dev_dbg(&sai->pdev->dev, "Set %pOFn%s as synchro provider\n", + sai->pdev->dev.of_node, synco == STM_SAI_SYNC_OUT_A ? "A" : "B"); prev_synco = FIELD_GET(SAI_GCR_SYNCOUT_MASK, readl_relaxed(sai->base)); if (prev_synco != STM_SAI_SYNC_OUT_NONE && synco != prev_synco) { - dev_err(&sai->pdev->dev, "%s%s already set as sync provider\n", - sai->pdev->dev.of_node->name, + dev_err(&sai->pdev->dev, "%pOFn%s already set as sync provider\n", + sai->pdev->dev.of_node, prev_synco == STM_SAI_SYNC_OUT_A ? "A" : "B"); clk_disable_unprepare(sai->pclk); return -EINVAL; @@ -104,7 +104,7 @@ static int stm32_sai_set_sync(struct stm32_sai_data *sai_client, if (!pdev) { dev_err(&sai_client->pdev->dev, - "Device not found for node %s\n", np_provider->name); + "Device not found for node %pOFn\n", np_provider); return -ENODEV; } diff --git a/sound/soc/stm/stm32_sai.h b/sound/soc/stm/stm32_sai.h index f25422174909..08de899c766b 100644 --- a/sound/soc/stm/stm32_sai.h +++ b/sound/soc/stm/stm32_sai.h @@ -91,6 +91,9 @@ #define SAI_XCR1_OSR_SHIFT 26 #define SAI_XCR1_OSR BIT(SAI_XCR1_OSR_SHIFT) +#define SAI_XCR1_MCKEN_SHIFT 27 +#define SAI_XCR1_MCKEN BIT(SAI_XCR1_MCKEN_SHIFT) + /******************* Bit definition for SAI_XCR2 register *******************/ #define SAI_XCR2_FTH_SHIFT 0 #define SAI_XCR2_FTH_MASK GENMASK(2, SAI_XCR2_FTH_SHIFT) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 06fba9650ac4..d4825700b63f 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -17,6 +17,7 @@ */ #include <linux/clk.h> +#include <linux/clk-provider.h> #include <linux/kernel.h> #include <linux/module.h> #include <linux/of_irq.h> @@ -68,6 +69,8 @@ #define SAI_IEC60958_BLOCK_FRAMES 192 #define SAI_IEC60958_STATUS_BYTES 24 +#define SAI_MCLK_NAME_LEN 32 + /** * struct stm32_sai_sub_data - private data of SAI sub block (block A or B) * @pdev: device data pointer @@ -80,6 +83,7 @@ * @pdata: SAI block parent data pointer * @np_sync_provider: synchronization provider node * @sai_ck: kernel clock feeding the SAI clock generator + * @sai_mclk: master clock from SAI mclk provider * @phys_addr: SAI registers physical base address * @mclk_rate: SAI block master clock frequency (Hz). set at init * @id: SAI sub block id corresponding to sub-block A or B @@ -110,6 +114,7 @@ struct stm32_sai_sub_data { struct stm32_sai_data *pdata; struct device_node *np_sync_provider; struct clk *sai_ck; + struct clk *sai_mclk; dma_addr_t phys_addr; unsigned int mclk_rate; unsigned int id; @@ -251,6 +256,175 @@ static const struct snd_kcontrol_new iec958_ctls = { .put = snd_pcm_iec958_put, }; +struct stm32_sai_mclk_data { + struct clk_hw hw; + unsigned long freq; + struct stm32_sai_sub_data *sai_data; +}; + +#define to_mclk_data(_hw) container_of(_hw, struct stm32_sai_mclk_data, hw) +#define STM32_SAI_MAX_CLKS 1 + +static int stm32_sai_get_clk_div(struct stm32_sai_sub_data *sai, + unsigned long input_rate, + unsigned long output_rate) +{ + int version = sai->pdata->conf->version; + int div; + + div = DIV_ROUND_CLOSEST(input_rate, output_rate); + if (div > SAI_XCR1_MCKDIV_MAX(version)) { + dev_err(&sai->pdev->dev, "Divider %d out of range\n", div); + return -EINVAL; + } + dev_dbg(&sai->pdev->dev, "SAI divider %d\n", div); + + if (input_rate % div) + dev_dbg(&sai->pdev->dev, + "Rate not accurate. requested (%ld), actual (%ld)\n", + output_rate, input_rate / div); + + return div; +} + +static int stm32_sai_set_clk_div(struct stm32_sai_sub_data *sai, + unsigned int div) +{ + int version = sai->pdata->conf->version; + int ret, cr1, mask; + + if (div > SAI_XCR1_MCKDIV_MAX(version)) { + dev_err(&sai->pdev->dev, "Divider %d out of range\n", div); + return -EINVAL; + } + + mask = SAI_XCR1_MCKDIV_MASK(SAI_XCR1_MCKDIV_WIDTH(version)); + cr1 = SAI_XCR1_MCKDIV_SET(div); + ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, mask, cr1); + if (ret < 0) + dev_err(&sai->pdev->dev, "Failed to update CR1 register\n"); + + return ret; +} + +static long stm32_sai_mclk_round_rate(struct clk_hw *hw, unsigned long rate, + unsigned long *prate) +{ + struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); + struct stm32_sai_sub_data *sai = mclk->sai_data; + int div; + + div = stm32_sai_get_clk_div(sai, *prate, rate); + if (div < 0) + return div; + + mclk->freq = *prate / div; + + return mclk->freq; +} + +static unsigned long stm32_sai_mclk_recalc_rate(struct clk_hw *hw, + unsigned long parent_rate) +{ + struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); + + return mclk->freq; +} + +static int stm32_sai_mclk_set_rate(struct clk_hw *hw, unsigned long rate, + unsigned long parent_rate) +{ + struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); + struct stm32_sai_sub_data *sai = mclk->sai_data; + int div, ret; + + div = stm32_sai_get_clk_div(sai, parent_rate, rate); + if (div < 0) + return div; + + ret = stm32_sai_set_clk_div(sai, div); + if (ret) + return ret; + + mclk->freq = rate; + + return 0; +} + +static int stm32_sai_mclk_enable(struct clk_hw *hw) +{ + struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); + struct stm32_sai_sub_data *sai = mclk->sai_data; + + dev_dbg(&sai->pdev->dev, "Enable master clock\n"); + + return regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, + SAI_XCR1_MCKEN, SAI_XCR1_MCKEN); +} + +static void stm32_sai_mclk_disable(struct clk_hw *hw) +{ + struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); + struct stm32_sai_sub_data *sai = mclk->sai_data; + + dev_dbg(&sai->pdev->dev, "Disable master clock\n"); + + regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_MCKEN, 0); +} + +static const struct clk_ops mclk_ops = { + .enable = stm32_sai_mclk_enable, + .disable = stm32_sai_mclk_disable, + .recalc_rate = stm32_sai_mclk_recalc_rate, + .round_rate = stm32_sai_mclk_round_rate, + .set_rate = stm32_sai_mclk_set_rate, +}; + +static int stm32_sai_add_mclk_provider(struct stm32_sai_sub_data *sai) +{ + struct clk_hw *hw; + struct stm32_sai_mclk_data *mclk; + struct device *dev = &sai->pdev->dev; + const char *pname = __clk_get_name(sai->sai_ck); + char *mclk_name, *p, *s = (char *)pname; + int ret, i = 0; + + mclk = devm_kzalloc(dev, sizeof(*mclk), GFP_KERNEL); + if (!mclk) + return -ENOMEM; + + mclk_name = devm_kcalloc(dev, sizeof(char), + SAI_MCLK_NAME_LEN, GFP_KERNEL); + if (!mclk_name) + return -ENOMEM; + + /* + * Forge mclk clock name from parent clock name and suffix. + * String after "_" char is stripped in parent name. + */ + p = mclk_name; + while (*s && *s != '_' && (i < (SAI_MCLK_NAME_LEN - 7))) { + *p++ = *s++; + i++; + } + STM_SAI_IS_SUB_A(sai) ? strcat(p, "a_mclk") : strcat(p, "b_mclk"); + + mclk->hw.init = CLK_HW_INIT(mclk_name, pname, &mclk_ops, 0); + mclk->sai_data = sai; + hw = &mclk->hw; + + dev_dbg(dev, "Register master clock %s\n", mclk_name); + ret = devm_clk_hw_register(&sai->pdev->dev, hw); + if (ret) { + dev_err(dev, "mclk register returned %d\n", ret); + return ret; + } + sai->sai_mclk = hw->clk; + + /* register mclk provider */ + return devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, hw); +} + static irqreturn_t stm32_sai_isr(int irq, void *devid) { struct stm32_sai_sub_data *sai = (struct stm32_sai_sub_data *)devid; @@ -312,15 +486,25 @@ static int stm32_sai_set_sysclk(struct snd_soc_dai *cpu_dai, struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai); int ret; - if ((dir == SND_SOC_CLOCK_OUT) && sai->master) { + if (dir == SND_SOC_CLOCK_OUT) { ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_NODIV, (unsigned int)~SAI_XCR1_NODIV); if (ret < 0) return ret; - sai->mclk_rate = freq; dev_dbg(cpu_dai->dev, "SAI MCLK frequency is %uHz\n", freq); + sai->mclk_rate = freq; + + if (sai->sai_mclk) { + ret = clk_set_rate_exclusive(sai->sai_mclk, + sai->mclk_rate); + if (ret) { + dev_err(cpu_dai->dev, + "Could not set mclk rate\n"); + return ret; + } + } } return 0; @@ -715,15 +899,9 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai, { struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai); int cr1, mask, div = 0; - int sai_clk_rate, mclk_ratio, den, ret; - int version = sai->pdata->conf->version; + int sai_clk_rate, mclk_ratio, den; unsigned int rate = params_rate(params); - if (!sai->mclk_rate) { - dev_err(cpu_dai->dev, "Mclk rate is null\n"); - return -EINVAL; - } - if (!(rate % 11025)) clk_set_parent(sai->sai_ck, sai->pdata->clk_x11k); else @@ -731,14 +909,22 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai, sai_clk_rate = clk_get_rate(sai->sai_ck); if (STM_SAI_IS_F4(sai->pdata)) { - /* - * mclk_rate = 256 * fs - * MCKDIV = 0 if sai_ck < 3/2 * mclk_rate - * MCKDIV = sai_ck / (2 * mclk_rate) otherwise + /* mclk on (NODIV=0) + * mclk_rate = 256 * fs + * MCKDIV = 0 if sai_ck < 3/2 * mclk_rate + * MCKDIV = sai_ck / (2 * mclk_rate) otherwise + * mclk off (NODIV=1) + * MCKDIV ignored. sck = sai_ck */ - if (2 * sai_clk_rate >= 3 * sai->mclk_rate) - div = DIV_ROUND_CLOSEST(sai_clk_rate, - 2 * sai->mclk_rate); + if (!sai->mclk_rate) + return 0; + + if (2 * sai_clk_rate >= 3 * sai->mclk_rate) { + div = stm32_sai_get_clk_div(sai, sai_clk_rate, + 2 * sai->mclk_rate); + if (div < 0) + return div; + } } else { /* * TDM mode : @@ -750,8 +936,10 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai, * Note: NOMCK/NODIV correspond to same bit. */ if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) { - div = DIV_ROUND_CLOSEST(sai_clk_rate, - (params_rate(params) * 128)); + div = stm32_sai_get_clk_div(sai, sai_clk_rate, + rate * 128); + if (div < 0) + return div; } else { if (sai->mclk_rate) { mclk_ratio = sai->mclk_rate / rate; @@ -764,31 +952,22 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai, mclk_ratio); return -EINVAL; } - div = DIV_ROUND_CLOSEST(sai_clk_rate, - sai->mclk_rate); + div = stm32_sai_get_clk_div(sai, sai_clk_rate, + sai->mclk_rate); + if (div < 0) + return div; } else { /* mclk-fs not set, master clock not active */ den = sai->fs_length * params_rate(params); - div = DIV_ROUND_CLOSEST(sai_clk_rate, den); + div = stm32_sai_get_clk_div(sai, sai_clk_rate, + den); + if (div < 0) + return div; } } } - if (div > SAI_XCR1_MCKDIV_MAX(version)) { - dev_err(cpu_dai->dev, "Divider %d out of range\n", div); - return -EINVAL; - } - dev_dbg(cpu_dai->dev, "SAI clock %d, divider %d\n", sai_clk_rate, div); - - mask = SAI_XCR1_MCKDIV_MASK(SAI_XCR1_MCKDIV_WIDTH(version)); - cr1 = SAI_XCR1_MCKDIV_SET(div); - ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, mask, cr1); - if (ret < 0) { - dev_err(cpu_dai->dev, "Failed to update CR1 register\n"); - return ret; - } - - return 0; + return stm32_sai_set_clk_div(sai, div); } static int stm32_sai_hw_params(struct snd_pcm_substream *substream, @@ -881,6 +1060,9 @@ static void stm32_sai_shutdown(struct snd_pcm_substream *substream, SAI_XCR1_NODIV); clk_disable_unprepare(sai->sai_ck); + + clk_rate_exclusive_put(sai->sai_mclk); + sai->substream = NULL; } @@ -903,6 +1085,8 @@ static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai) struct stm32_sai_sub_data *sai = dev_get_drvdata(cpu_dai->dev); int cr1 = 0, cr1_mask; + sai->cpu_dai = cpu_dai; + sai->dma_params.addr = (dma_addr_t)(sai->phys_addr + STM_SAI_DR_REGX); /* * DMA supports 4, 8 or 16 burst sizes. Burst size 4 is the best choice, @@ -1124,16 +1308,15 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev, sai->sync = SAI_SYNC_NONE; if (args.np) { if (args.np == np) { - dev_err(&pdev->dev, "%s sync own reference\n", - np->name); + dev_err(&pdev->dev, "%pOFn sync own reference\n", np); of_node_put(args.np); return -EINVAL; } sai->np_sync_provider = of_get_parent(args.np); if (!sai->np_sync_provider) { - dev_err(&pdev->dev, "%s parent node not found\n", - np->name); + dev_err(&pdev->dev, "%pOFn parent node not found\n", + np); of_node_put(args.np); return -ENODEV; } @@ -1182,6 +1365,23 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev, return PTR_ERR(sai->sai_ck); } + if (STM_SAI_IS_F4(sai->pdata)) + return 0; + + /* Register mclk provider if requested */ + if (of_find_property(np, "#clock-cells", NULL)) { + ret = stm32_sai_add_mclk_provider(sai); + if (ret < 0) + return ret; + } else { + sai->sai_mclk = devm_clk_get(&pdev->dev, "MCLK"); + if (IS_ERR(sai->sai_mclk)) { + if (PTR_ERR(sai->sai_mclk) != -ENOENT) + return PTR_ERR(sai->sai_mclk); + sai->sai_mclk = NULL; + } + } + return 0; } diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index 22408bc2d6ec..8134c3c94229 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -12,7 +12,7 @@ config SND_SUN4I_CODEC config SND_SUN8I_CODEC tristate "Allwinner SUN8I audio codec" depends on OF - depends on MACH_SUN8I || COMPILE_TEST + depends on MACH_SUN8I || (ARM64 && ARCH_SUNXI) || COMPILE_TEST select REGMAP_MMIO help This option enables the digital part of the internal audio codec for @@ -23,11 +23,19 @@ config SND_SUN8I_CODEC config SND_SUN8I_CODEC_ANALOG tristate "Allwinner sun8i Codec Analog Controls Support" depends on MACH_SUN8I || (ARM64 && ARCH_SUNXI) || COMPILE_TEST - select REGMAP + select SND_SUN8I_ADDA_PR_REGMAP help Say Y or M if you want to add support for the analog controls for the codec embedded in newer Allwinner SoCs. +config SND_SUN50I_CODEC_ANALOG + tristate "Allwinner sun50i Codec Analog Controls Support" + depends on (ARM64 && ARCH_SUNXI) || COMPILE_TEST + select SND_SUN8I_ADDA_PR_REGMAP + help + Say Y or M if you want to add support for the analog controls for + the codec embedded in Allwinner A64 SoC. + config SND_SUN4I_I2S tristate "Allwinner A10 I2S Support" select SND_SOC_GENERIC_DMAENGINE_PCM @@ -45,4 +53,9 @@ config SND_SUN4I_SPDIF help Say Y or M to add support for the S/PDIF audio block in the Allwinner A10 and affiliated SoCs. + +config SND_SUN8I_ADDA_PR_REGMAP + tristate + select REGMAP + endmenu diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index 4a9ef67386ca..a86be340a076 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -3,4 +3,6 @@ obj-$(CONFIG_SND_SUN4I_CODEC) += sun4i-codec.o obj-$(CONFIG_SND_SUN4I_I2S) += sun4i-i2s.o obj-$(CONFIG_SND_SUN4I_SPDIF) += sun4i-spdif.o obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o +obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o +obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index a4aa931ebfae..d5ec1a20499d 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -644,40 +644,6 @@ static int sun4i_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int sun4i_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai); - - /* Enable the whole hardware block */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_GL_EN, SUN4I_I2S_CTRL_GL_EN); - - /* Enable the first output line */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_SDO_EN_MASK, - SUN4I_I2S_CTRL_SDO_EN(0)); - - - return clk_prepare_enable(i2s->mod_clk); -} - -static void sun4i_i2s_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai); - - clk_disable_unprepare(i2s->mod_clk); - - /* Disable our output lines */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_SDO_EN_MASK, 0); - - /* Disable the whole hardware block */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_GL_EN, 0); -} - static int sun4i_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { @@ -695,8 +661,6 @@ static const struct snd_soc_dai_ops sun4i_i2s_dai_ops = { .hw_params = sun4i_i2s_hw_params, .set_fmt = sun4i_i2s_set_fmt, .set_sysclk = sun4i_i2s_set_sysclk, - .shutdown = sun4i_i2s_shutdown, - .startup = sun4i_i2s_startup, .trigger = sun4i_i2s_trigger, }; @@ -869,6 +833,21 @@ static int sun4i_i2s_runtime_resume(struct device *dev) goto err_disable_clk; } + /* Enable the whole hardware block */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_GL_EN, SUN4I_I2S_CTRL_GL_EN); + + /* Enable the first output line */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_SDO_EN_MASK, + SUN4I_I2S_CTRL_SDO_EN(0)); + + ret = clk_prepare_enable(i2s->mod_clk); + if (ret) { + dev_err(dev, "Failed to enable module clock\n"); + goto err_disable_clk; + } + return 0; err_disable_clk: @@ -880,6 +859,16 @@ static int sun4i_i2s_runtime_suspend(struct device *dev) { struct sun4i_i2s *i2s = dev_get_drvdata(dev); + clk_disable_unprepare(i2s->mod_clk); + + /* Disable our output lines */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_SDO_EN_MASK, 0); + + /* Disable the whole hardware block */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_GL_EN, 0); + regcache_cache_only(i2s->regmap, true); clk_disable_unprepare(i2s->bus_clk); @@ -961,6 +950,23 @@ static const struct sun4i_i2s_quirks sun8i_h3_i2s_quirks = { .field_rxchansel = REG_FIELD(SUN8I_I2S_RX_CHAN_SEL_REG, 0, 2), }; +static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = { + .has_reset = true, + .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG, + .sun4i_i2s_regmap = &sun4i_i2s_regmap_config, + .has_slave_select_bit = true, + .field_clkdiv_mclk_en = REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 7, 7), + .field_fmt_wss = REG_FIELD(SUN4I_I2S_FMT0_REG, 2, 3), + .field_fmt_sr = REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5), + .field_fmt_bclk = REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6), + .field_fmt_lrclk = REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7), + .field_fmt_mode = REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 1), + .field_txchanmap = REG_FIELD(SUN4I_I2S_TX_CHAN_MAP_REG, 0, 31), + .field_rxchanmap = REG_FIELD(SUN4I_I2S_RX_CHAN_MAP_REG, 0, 31), + .field_txchansel = REG_FIELD(SUN4I_I2S_TX_CHAN_SEL_REG, 0, 2), + .field_rxchansel = REG_FIELD(SUN4I_I2S_RX_CHAN_SEL_REG, 0, 2), +}; + static int sun4i_i2s_init_regmap_fields(struct device *dev, struct sun4i_i2s *i2s) { @@ -1169,6 +1175,10 @@ static const struct of_device_id sun4i_i2s_match[] = { .compatible = "allwinner,sun8i-h3-i2s", .data = &sun8i_h3_i2s_quirks, }, + { + .compatible = "allwinner,sun50i-a64-codec-i2s", + .data = &sun50i_a64_codec_i2s_quirks, + }, {} }; MODULE_DEVICE_TABLE(of, sun4i_i2s_match); diff --git a/sound/soc/sunxi/sun50i-codec-analog.c b/sound/soc/sunxi/sun50i-codec-analog.c new file mode 100644 index 000000000000..df1fed0aa001 --- /dev/null +++ b/sound/soc/sunxi/sun50i-codec-analog.c @@ -0,0 +1,446 @@ +// SPDX-License-Identifier: GPL-2.0+ +/* + * This driver supports the analog controls for the internal codec + * found in Allwinner's A64 SoC. + * + * Copyright (C) 2016 Chen-Yu Tsai <wens@csie.org> + * Copyright (C) 2017 Marcus Cooper <codekipper@gmail.com> + * Copyright (C) 2018 Vasily Khoruzhick <anarsoul@gmail.com> + * + * Based on sun8i-codec-analog.c + * + */ + +#include <linux/io.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_device.h> +#include <linux/platform_device.h> +#include <linux/regmap.h> + +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> + +#include "sun8i-adda-pr-regmap.h" + +/* Codec analog control register offsets and bit fields */ +#define SUN50I_ADDA_HP_CTRL 0x00 +#define SUN50I_ADDA_HP_CTRL_PA_CLK_GATE 7 +#define SUN50I_ADDA_HP_CTRL_HPPA_EN 6 +#define SUN50I_ADDA_HP_CTRL_HPVOL 0 + +#define SUN50I_ADDA_OL_MIX_CTRL 0x01 +#define SUN50I_ADDA_OL_MIX_CTRL_MIC1 6 +#define SUN50I_ADDA_OL_MIX_CTRL_MIC2 5 +#define SUN50I_ADDA_OL_MIX_CTRL_PHONE 4 +#define SUN50I_ADDA_OL_MIX_CTRL_PHONEN 3 +#define SUN50I_ADDA_OL_MIX_CTRL_LINEINL 2 +#define SUN50I_ADDA_OL_MIX_CTRL_DACL 1 +#define SUN50I_ADDA_OL_MIX_CTRL_DACR 0 + +#define SUN50I_ADDA_OR_MIX_CTRL 0x02 +#define SUN50I_ADDA_OR_MIX_CTRL_MIC1 6 +#define SUN50I_ADDA_OR_MIX_CTRL_MIC2 5 +#define SUN50I_ADDA_OR_MIX_CTRL_PHONE 4 +#define SUN50I_ADDA_OR_MIX_CTRL_PHONEP 3 +#define SUN50I_ADDA_OR_MIX_CTRL_LINEINR 2 +#define SUN50I_ADDA_OR_MIX_CTRL_DACR 1 +#define SUN50I_ADDA_OR_MIX_CTRL_DACL 0 + +#define SUN50I_ADDA_LINEOUT_CTRL0 0x05 +#define SUN50I_ADDA_LINEOUT_CTRL0_LEN 7 +#define SUN50I_ADDA_LINEOUT_CTRL0_REN 6 +#define SUN50I_ADDA_LINEOUT_CTRL0_LSRC_SEL 5 +#define SUN50I_ADDA_LINEOUT_CTRL0_RSRC_SEL 4 + +#define SUN50I_ADDA_LINEOUT_CTRL1 0x06 +#define SUN50I_ADDA_LINEOUT_CTRL1_VOL 0 + +#define SUN50I_ADDA_MIC1_CTRL 0x07 +#define SUN50I_ADDA_MIC1_CTRL_MIC1G 4 +#define SUN50I_ADDA_MIC1_CTRL_MIC1AMPEN 3 +#define SUN50I_ADDA_MIC1_CTRL_MIC1BOOST 0 + +#define SUN50I_ADDA_MIC2_CTRL 0x08 +#define SUN50I_ADDA_MIC2_CTRL_MIC2G 4 +#define SUN50I_ADDA_MIC2_CTRL_MIC2AMPEN 3 +#define SUN50I_ADDA_MIC2_CTRL_MIC2BOOST 0 + +#define SUN50I_ADDA_LINEIN_CTRL 0x09 +#define SUN50I_ADDA_LINEIN_CTRL_LINEING 0 + +#define SUN50I_ADDA_MIX_DAC_CTRL 0x0a +#define SUN50I_ADDA_MIX_DAC_CTRL_DACAREN 7 +#define SUN50I_ADDA_MIX_DAC_CTRL_DACALEN 6 +#define SUN50I_ADDA_MIX_DAC_CTRL_RMIXEN 5 +#define SUN50I_ADDA_MIX_DAC_CTRL_LMIXEN 4 +#define SUN50I_ADDA_MIX_DAC_CTRL_RHPPAMUTE 3 +#define SUN50I_ADDA_MIX_DAC_CTRL_LHPPAMUTE 2 +#define SUN50I_ADDA_MIX_DAC_CTRL_RHPIS 1 +#define SUN50I_ADDA_MIX_DAC_CTRL_LHPIS 0 + +#define SUN50I_ADDA_L_ADCMIX_SRC 0x0b +#define SUN50I_ADDA_L_ADCMIX_SRC_MIC1 6 +#define SUN50I_ADDA_L_ADCMIX_SRC_MIC2 5 +#define SUN50I_ADDA_L_ADCMIX_SRC_PHONE 4 +#define SUN50I_ADDA_L_ADCMIX_SRC_PHONEN 3 +#define SUN50I_ADDA_L_ADCMIX_SRC_LINEINL 2 +#define SUN50I_ADDA_L_ADCMIX_SRC_OMIXRL 1 +#define SUN50I_ADDA_L_ADCMIX_SRC_OMIXRR 0 + +#define SUN50I_ADDA_R_ADCMIX_SRC 0x0c +#define SUN50I_ADDA_R_ADCMIX_SRC_MIC1 6 +#define SUN50I_ADDA_R_ADCMIX_SRC_MIC2 5 +#define SUN50I_ADDA_R_ADCMIX_SRC_PHONE 4 +#define SUN50I_ADDA_R_ADCMIX_SRC_PHONEP 3 +#define SUN50I_ADDA_R_ADCMIX_SRC_LINEINR 2 +#define SUN50I_ADDA_R_ADCMIX_SRC_OMIXR 1 +#define SUN50I_ADDA_R_ADCMIX_SRC_OMIXL 0 + +#define SUN50I_ADDA_ADC_CTRL 0x0d +#define SUN50I_ADDA_ADC_CTRL_ADCREN 7 +#define SUN50I_ADDA_ADC_CTRL_ADCLEN 6 +#define SUN50I_ADDA_ADC_CTRL_ADCG 0 + +#define SUN50I_ADDA_HS_MBIAS_CTRL 0x0e +#define SUN50I_ADDA_HS_MBIAS_CTRL_MMICBIASEN 7 + +#define SUN50I_ADDA_JACK_MIC_CTRL 0x1d +#define SUN50I_ADDA_JACK_MIC_CTRL_HMICBIASEN 5 + +/* mixer controls */ +static const struct snd_kcontrol_new sun50i_a64_codec_mixer_controls[] = { + SOC_DAPM_DOUBLE_R("DAC Playback Switch", + SUN50I_ADDA_OL_MIX_CTRL, + SUN50I_ADDA_OR_MIX_CTRL, + SUN50I_ADDA_OL_MIX_CTRL_DACL, 1, 0), + SOC_DAPM_DOUBLE_R("DAC Reversed Playback Switch", + SUN50I_ADDA_OL_MIX_CTRL, + SUN50I_ADDA_OR_MIX_CTRL, + SUN50I_ADDA_OL_MIX_CTRL_DACR, 1, 0), + SOC_DAPM_DOUBLE_R("Line In Playback Switch", + SUN50I_ADDA_OL_MIX_CTRL, + SUN50I_ADDA_OR_MIX_CTRL, + SUN50I_ADDA_OL_MIX_CTRL_LINEINL, 1, 0), + SOC_DAPM_DOUBLE_R("Mic1 Playback Switch", + SUN50I_ADDA_OL_MIX_CTRL, + SUN50I_ADDA_OR_MIX_CTRL, + SUN50I_ADDA_OL_MIX_CTRL_MIC1, 1, 0), + SOC_DAPM_DOUBLE_R("Mic2 Playback Switch", + SUN50I_ADDA_OL_MIX_CTRL, + SUN50I_ADDA_OR_MIX_CTRL, + SUN50I_ADDA_OL_MIX_CTRL_MIC2, 1, 0), +}; + +/* ADC mixer controls */ +static const struct snd_kcontrol_new sun50i_codec_adc_mixer_controls[] = { + SOC_DAPM_DOUBLE_R("Mixer Capture Switch", + SUN50I_ADDA_L_ADCMIX_SRC, + SUN50I_ADDA_R_ADCMIX_SRC, + SUN50I_ADDA_L_ADCMIX_SRC_OMIXRL, 1, 0), + SOC_DAPM_DOUBLE_R("Mixer Reversed Capture Switch", + SUN50I_ADDA_L_ADCMIX_SRC, + SUN50I_ADDA_R_ADCMIX_SRC, + SUN50I_ADDA_L_ADCMIX_SRC_OMIXRR, 1, 0), + SOC_DAPM_DOUBLE_R("Line In Capture Switch", + SUN50I_ADDA_L_ADCMIX_SRC, + SUN50I_ADDA_R_ADCMIX_SRC, + SUN50I_ADDA_L_ADCMIX_SRC_LINEINL, 1, 0), + SOC_DAPM_DOUBLE_R("Mic1 Capture Switch", + SUN50I_ADDA_L_ADCMIX_SRC, + SUN50I_ADDA_R_ADCMIX_SRC, + SUN50I_ADDA_L_ADCMIX_SRC_MIC1, 1, 0), + SOC_DAPM_DOUBLE_R("Mic2 Capture Switch", + SUN50I_ADDA_L_ADCMIX_SRC, + SUN50I_ADDA_R_ADCMIX_SRC, + SUN50I_ADDA_L_ADCMIX_SRC_MIC2, 1, 0), +}; + +static const DECLARE_TLV_DB_SCALE(sun50i_codec_out_mixer_pregain_scale, + -450, 150, 0); +static const DECLARE_TLV_DB_RANGE(sun50i_codec_mic_gain_scale, + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 7, TLV_DB_SCALE_ITEM(2400, 300, 0), +); + +static const DECLARE_TLV_DB_SCALE(sun50i_codec_hp_vol_scale, -6300, 100, 1); + +static const DECLARE_TLV_DB_RANGE(sun50i_codec_lineout_vol_scale, + 0, 1, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + 2, 31, TLV_DB_SCALE_ITEM(-4350, 150, 0), +); + + +/* volume / mute controls */ +static const struct snd_kcontrol_new sun50i_a64_codec_controls[] = { + SOC_SINGLE_TLV("Headphone Playback Volume", + SUN50I_ADDA_HP_CTRL, + SUN50I_ADDA_HP_CTRL_HPVOL, 0x3f, 0, + sun50i_codec_hp_vol_scale), + + SOC_DOUBLE("Headphone Playback Switch", + SUN50I_ADDA_MIX_DAC_CTRL, + SUN50I_ADDA_MIX_DAC_CTRL_LHPPAMUTE, + SUN50I_ADDA_MIX_DAC_CTRL_RHPPAMUTE, 1, 0), + + /* Mixer pre-gain */ + SOC_SINGLE_TLV("Mic1 Playback Volume", SUN50I_ADDA_MIC1_CTRL, + SUN50I_ADDA_MIC1_CTRL_MIC1G, + 0x7, 0, sun50i_codec_out_mixer_pregain_scale), + + /* Microphone Amp boost gain */ + SOC_SINGLE_TLV("Mic1 Boost Volume", SUN50I_ADDA_MIC1_CTRL, + SUN50I_ADDA_MIC1_CTRL_MIC1BOOST, 0x7, 0, + sun50i_codec_mic_gain_scale), + + /* Mixer pre-gain */ + SOC_SINGLE_TLV("Mic2 Playback Volume", + SUN50I_ADDA_MIC2_CTRL, SUN50I_ADDA_MIC2_CTRL_MIC2G, + 0x7, 0, sun50i_codec_out_mixer_pregain_scale), + + /* Microphone Amp boost gain */ + SOC_SINGLE_TLV("Mic2 Boost Volume", SUN50I_ADDA_MIC2_CTRL, + SUN50I_ADDA_MIC2_CTRL_MIC2BOOST, 0x7, 0, + sun50i_codec_mic_gain_scale), + + /* ADC */ + SOC_SINGLE_TLV("ADC Gain Capture Volume", SUN50I_ADDA_ADC_CTRL, + SUN50I_ADDA_ADC_CTRL_ADCG, 0x7, 0, + sun50i_codec_out_mixer_pregain_scale), + + /* Mixer pre-gain */ + SOC_SINGLE_TLV("Line In Playback Volume", SUN50I_ADDA_LINEIN_CTRL, + SUN50I_ADDA_LINEIN_CTRL_LINEING, + 0x7, 0, sun50i_codec_out_mixer_pregain_scale), + + SOC_SINGLE_TLV("Line Out Playback Volume", + SUN50I_ADDA_LINEOUT_CTRL1, + SUN50I_ADDA_LINEOUT_CTRL1_VOL, 0x1f, 0, + sun50i_codec_lineout_vol_scale), + + SOC_DOUBLE("Line Out Playback Switch", + SUN50I_ADDA_LINEOUT_CTRL0, + SUN50I_ADDA_LINEOUT_CTRL0_LEN, + SUN50I_ADDA_LINEOUT_CTRL0_REN, 1, 0), + +}; + +static const char * const sun50i_codec_hp_src_enum_text[] = { + "DAC", "Mixer", +}; + +static SOC_ENUM_DOUBLE_DECL(sun50i_codec_hp_src_enum, + SUN50I_ADDA_MIX_DAC_CTRL, + SUN50I_ADDA_MIX_DAC_CTRL_LHPIS, + SUN50I_ADDA_MIX_DAC_CTRL_RHPIS, + sun50i_codec_hp_src_enum_text); + +static const struct snd_kcontrol_new sun50i_codec_hp_src[] = { + SOC_DAPM_ENUM("Headphone Source Playback Route", + sun50i_codec_hp_src_enum), +}; + +static const char * const sun50i_codec_lineout_src_enum_text[] = { + "Stereo", "Mono Differential", +}; + +static SOC_ENUM_DOUBLE_DECL(sun50i_codec_lineout_src_enum, + SUN50I_ADDA_LINEOUT_CTRL0, + SUN50I_ADDA_LINEOUT_CTRL0_LSRC_SEL, + SUN50I_ADDA_LINEOUT_CTRL0_RSRC_SEL, + sun50i_codec_lineout_src_enum_text); + +static const struct snd_kcontrol_new sun50i_codec_lineout_src[] = { + SOC_DAPM_ENUM("Line Out Source Playback Route", + sun50i_codec_lineout_src_enum), +}; + +static const struct snd_soc_dapm_widget sun50i_a64_codec_widgets[] = { + /* DAC */ + SND_SOC_DAPM_DAC("Left DAC", NULL, SUN50I_ADDA_MIX_DAC_CTRL, + SUN50I_ADDA_MIX_DAC_CTRL_DACALEN, 0), + SND_SOC_DAPM_DAC("Right DAC", NULL, SUN50I_ADDA_MIX_DAC_CTRL, + SUN50I_ADDA_MIX_DAC_CTRL_DACAREN, 0), + /* ADC */ + SND_SOC_DAPM_ADC("Left ADC", NULL, SUN50I_ADDA_ADC_CTRL, + SUN50I_ADDA_ADC_CTRL_ADCLEN, 0), + SND_SOC_DAPM_ADC("Right ADC", NULL, SUN50I_ADDA_ADC_CTRL, + SUN50I_ADDA_ADC_CTRL_ADCREN, 0), + /* + * Due to this component and the codec belonging to separate DAPM + * contexts, we need to manually link the above widgets to their + * stream widgets at the card level. + */ + + SND_SOC_DAPM_REGULATOR_SUPPLY("hpvcc", 0, 0), + SND_SOC_DAPM_MUX("Headphone Source Playback Route", + SND_SOC_NOPM, 0, 0, sun50i_codec_hp_src), + SND_SOC_DAPM_OUT_DRV("Headphone Amp", SUN50I_ADDA_HP_CTRL, + SUN50I_ADDA_HP_CTRL_HPPA_EN, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("HP"), + + SND_SOC_DAPM_MUX("Line Out Source Playback Route", + SND_SOC_NOPM, 0, 0, sun50i_codec_lineout_src), + SND_SOC_DAPM_OUTPUT("LINEOUT"), + + /* Microphone inputs */ + SND_SOC_DAPM_INPUT("MIC1"), + + /* Microphone Bias */ + SND_SOC_DAPM_SUPPLY("MBIAS", SUN50I_ADDA_HS_MBIAS_CTRL, + SUN50I_ADDA_HS_MBIAS_CTRL_MMICBIASEN, + 0, NULL, 0), + + /* Mic input path */ + SND_SOC_DAPM_PGA("Mic1 Amplifier", SUN50I_ADDA_MIC1_CTRL, + SUN50I_ADDA_MIC1_CTRL_MIC1AMPEN, 0, NULL, 0), + + /* Microphone input */ + SND_SOC_DAPM_INPUT("MIC2"), + + /* Microphone Bias */ + SND_SOC_DAPM_SUPPLY("HBIAS", SUN50I_ADDA_JACK_MIC_CTRL, + SUN50I_ADDA_JACK_MIC_CTRL_HMICBIASEN, + 0, NULL, 0), + + /* Mic input path */ + SND_SOC_DAPM_PGA("Mic2 Amplifier", SUN50I_ADDA_MIC2_CTRL, + SUN50I_ADDA_MIC2_CTRL_MIC2AMPEN, 0, NULL, 0), + + /* Line input */ + SND_SOC_DAPM_INPUT("LINEIN"), + + /* Mixers */ + SND_SOC_DAPM_MIXER("Left Mixer", SUN50I_ADDA_MIX_DAC_CTRL, + SUN50I_ADDA_MIX_DAC_CTRL_LMIXEN, 0, + sun50i_a64_codec_mixer_controls, + ARRAY_SIZE(sun50i_a64_codec_mixer_controls)), + SND_SOC_DAPM_MIXER("Right Mixer", SUN50I_ADDA_MIX_DAC_CTRL, + SUN50I_ADDA_MIX_DAC_CTRL_RMIXEN, 0, + sun50i_a64_codec_mixer_controls, + ARRAY_SIZE(sun50i_a64_codec_mixer_controls)), + SND_SOC_DAPM_MIXER("Left ADC Mixer", SUN50I_ADDA_ADC_CTRL, + SUN50I_ADDA_ADC_CTRL_ADCLEN, 0, + sun50i_codec_adc_mixer_controls, + ARRAY_SIZE(sun50i_codec_adc_mixer_controls)), + SND_SOC_DAPM_MIXER("Right ADC Mixer", SUN50I_ADDA_ADC_CTRL, + SUN50I_ADDA_ADC_CTRL_ADCREN, 0, + sun50i_codec_adc_mixer_controls, + ARRAY_SIZE(sun50i_codec_adc_mixer_controls)), +}; + +static const struct snd_soc_dapm_route sun50i_a64_codec_routes[] = { + /* Left Mixer Routes */ + { "Left Mixer", "DAC Playback Switch", "Left DAC" }, + { "Left Mixer", "DAC Reversed Playback Switch", "Right DAC" }, + { "Left Mixer", "Mic1 Playback Switch", "Mic1 Amplifier" }, + + /* Right Mixer Routes */ + { "Right Mixer", "DAC Playback Switch", "Right DAC" }, + { "Right Mixer", "DAC Reversed Playback Switch", "Left DAC" }, + { "Right Mixer", "Mic1 Playback Switch", "Mic1 Amplifier" }, + + /* Left ADC Mixer Routes */ + { "Left ADC Mixer", "Mixer Capture Switch", "Left Mixer" }, + { "Left ADC Mixer", "Mixer Reversed Capture Switch", "Right Mixer" }, + { "Left ADC Mixer", "Mic1 Capture Switch", "Mic1 Amplifier" }, + + /* Right ADC Mixer Routes */ + { "Right ADC Mixer", "Mixer Capture Switch", "Right Mixer" }, + { "Right ADC Mixer", "Mixer Reversed Capture Switch", "Left Mixer" }, + { "Right ADC Mixer", "Mic1 Capture Switch", "Mic1 Amplifier" }, + + /* ADC Routes */ + { "Left ADC", NULL, "Left ADC Mixer" }, + { "Right ADC", NULL, "Right ADC Mixer" }, + + /* Headphone Routes */ + { "Headphone Source Playback Route", "DAC", "Left DAC" }, + { "Headphone Source Playback Route", "DAC", "Right DAC" }, + { "Headphone Source Playback Route", "Mixer", "Left Mixer" }, + { "Headphone Source Playback Route", "Mixer", "Right Mixer" }, + { "Headphone Amp", NULL, "Headphone Source Playback Route" }, + { "Headphone Amp", NULL, "hpvcc" }, + { "HP", NULL, "Headphone Amp" }, + + /* Microphone Routes */ + { "Mic1 Amplifier", NULL, "MIC1"}, + + /* Microphone Routes */ + { "Mic2 Amplifier", NULL, "MIC2"}, + { "Left Mixer", "Mic2 Playback Switch", "Mic2 Amplifier" }, + { "Right Mixer", "Mic2 Playback Switch", "Mic2 Amplifier" }, + { "Left ADC Mixer", "Mic2 Capture Switch", "Mic2 Amplifier" }, + { "Right ADC Mixer", "Mic2 Capture Switch", "Mic2 Amplifier" }, + + /* Line-in Routes */ + { "Left Mixer", "Line In Playback Switch", "LINEIN" }, + { "Right Mixer", "Line In Playback Switch", "LINEIN" }, + { "Left ADC Mixer", "Line In Capture Switch", "LINEIN" }, + { "Right ADC Mixer", "Line In Capture Switch", "LINEIN" }, + + /* Line-out Routes */ + { "Line Out Source Playback Route", "Stereo", "Left Mixer" }, + { "Line Out Source Playback Route", "Stereo", "Right Mixer" }, + { "Line Out Source Playback Route", "Mono Differential", "Left Mixer" }, + { "Line Out Source Playback Route", "Mono Differential", + "Right Mixer" }, + { "LINEOUT", NULL, "Line Out Source Playback Route" }, +}; + +static const struct snd_soc_component_driver sun50i_codec_analog_cmpnt_drv = { + .controls = sun50i_a64_codec_controls, + .num_controls = ARRAY_SIZE(sun50i_a64_codec_controls), + .dapm_widgets = sun50i_a64_codec_widgets, + .num_dapm_widgets = ARRAY_SIZE(sun50i_a64_codec_widgets), + .dapm_routes = sun50i_a64_codec_routes, + .num_dapm_routes = ARRAY_SIZE(sun50i_a64_codec_routes), +}; + +static const struct of_device_id sun50i_codec_analog_of_match[] = { + { + .compatible = "allwinner,sun50i-a64-codec-analog", + }, + {} +}; +MODULE_DEVICE_TABLE(of, sun50i_codec_analog_of_match); + +static int sun50i_codec_analog_probe(struct platform_device *pdev) +{ + struct resource *res; + struct regmap *regmap; + void __iomem *base; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) { + dev_err(&pdev->dev, "Failed to map the registers\n"); + return PTR_ERR(base); + } + + regmap = sun8i_adda_pr_regmap_init(&pdev->dev, base); + if (IS_ERR(regmap)) { + dev_err(&pdev->dev, "Failed to create regmap\n"); + return PTR_ERR(regmap); + } + + return devm_snd_soc_register_component(&pdev->dev, + &sun50i_codec_analog_cmpnt_drv, + NULL, 0); +} + +static struct platform_driver sun50i_codec_analog_driver = { + .driver = { + .name = "sun50i-codec-analog", + .of_match_table = sun50i_codec_analog_of_match, + }, + .probe = sun50i_codec_analog_probe, +}; +module_platform_driver(sun50i_codec_analog_driver); + +MODULE_DESCRIPTION("Allwinner internal codec analog controls driver for A64"); +MODULE_AUTHOR("Vasily Khoruzhick <anarsoul@gmail.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sun50i-codec-analog"); diff --git a/sound/soc/sunxi/sun8i-adda-pr-regmap.c b/sound/soc/sunxi/sun8i-adda-pr-regmap.c new file mode 100644 index 000000000000..e68ce9d2884d --- /dev/null +++ b/sound/soc/sunxi/sun8i-adda-pr-regmap.c @@ -0,0 +1,102 @@ +// SPDX-License-Identifier: GPL-2.0+ +/* + * This driver provides regmap to access to analog part of audio codec + * found on Allwinner A23, A31s, A33, H3 and A64 Socs + * + * Copyright 2016 Chen-Yu Tsai <wens@csie.org> + * Copyright (C) 2018 Vasily Khoruzhick <anarsoul@gmail.com> + */ + +#include <linux/io.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/regmap.h> + +#include "sun8i-adda-pr-regmap.h" + +/* Analog control register access bits */ +#define ADDA_PR 0x0 /* PRCM base + 0x1c0 */ +#define ADDA_PR_RESET BIT(28) +#define ADDA_PR_WRITE BIT(24) +#define ADDA_PR_ADDR_SHIFT 16 +#define ADDA_PR_ADDR_MASK GENMASK(4, 0) +#define ADDA_PR_DATA_IN_SHIFT 8 +#define ADDA_PR_DATA_IN_MASK GENMASK(7, 0) +#define ADDA_PR_DATA_OUT_SHIFT 0 +#define ADDA_PR_DATA_OUT_MASK GENMASK(7, 0) + +/* regmap access bits */ +static int adda_reg_read(void *context, unsigned int reg, unsigned int *val) +{ + void __iomem *base = (void __iomem *)context; + u32 tmp; + + /* De-assert reset */ + writel(readl(base) | ADDA_PR_RESET, base); + + /* Clear write bit */ + writel(readl(base) & ~ADDA_PR_WRITE, base); + + /* Set register address */ + tmp = readl(base); + tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT); + tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT; + writel(tmp, base); + + /* Read back value */ + *val = readl(base) & ADDA_PR_DATA_OUT_MASK; + + return 0; +} + +static int adda_reg_write(void *context, unsigned int reg, unsigned int val) +{ + void __iomem *base = (void __iomem *)context; + u32 tmp; + + /* De-assert reset */ + writel(readl(base) | ADDA_PR_RESET, base); + + /* Set register address */ + tmp = readl(base); + tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT); + tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT; + writel(tmp, base); + + /* Set data to write */ + tmp = readl(base); + tmp &= ~(ADDA_PR_DATA_IN_MASK << ADDA_PR_DATA_IN_SHIFT); + tmp |= (val & ADDA_PR_DATA_IN_MASK) << ADDA_PR_DATA_IN_SHIFT; + writel(tmp, base); + + /* Set write bit to signal a write */ + writel(readl(base) | ADDA_PR_WRITE, base); + + /* Clear write bit */ + writel(readl(base) & ~ADDA_PR_WRITE, base); + + return 0; +} + +static const struct regmap_config adda_pr_regmap_cfg = { + .name = "adda-pr", + .reg_bits = 5, + .reg_stride = 1, + .val_bits = 8, + .reg_read = adda_reg_read, + .reg_write = adda_reg_write, + .fast_io = true, + .max_register = 31, +}; + +struct regmap *sun8i_adda_pr_regmap_init(struct device *dev, + void __iomem *base) +{ + return devm_regmap_init(dev, NULL, base, &adda_pr_regmap_cfg); +} +EXPORT_SYMBOL_GPL(sun8i_adda_pr_regmap_init); + +MODULE_DESCRIPTION("Allwinner analog audio codec regmap driver"); +MODULE_AUTHOR("Vasily Khoruzhick <anarsoul@gmail.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sunxi-adda-pr"); diff --git a/sound/soc/sunxi/sun8i-adda-pr-regmap.h b/sound/soc/sunxi/sun8i-adda-pr-regmap.h new file mode 100644 index 000000000000..a5ae95dfebc1 --- /dev/null +++ b/sound/soc/sunxi/sun8i-adda-pr-regmap.h @@ -0,0 +1,7 @@ +/* SPDX-License-Identifier: GPL-2.0+ */ +/* + * Copyright (C) 2018 Vasily Khoruzhick <anarsoul@gmail.com> + */ + +struct regmap *sun8i_adda_pr_regmap_init(struct device *dev, + void __iomem *base); diff --git a/sound/soc/sunxi/sun8i-codec-analog.c b/sound/soc/sunxi/sun8i-codec-analog.c index 485e79f292c4..916a46bbc1c8 100644 --- a/sound/soc/sunxi/sun8i-codec-analog.c +++ b/sound/soc/sunxi/sun8i-codec-analog.c @@ -27,6 +27,8 @@ #include <sound/soc-dapm.h> #include <sound/tlv.h> +#include "sun8i-adda-pr-regmap.h" + /* Codec analog control register offsets and bit fields */ #define SUN8I_ADDA_HP_VOLC 0x00 #define SUN8I_ADDA_HP_VOLC_PA_CLK_GATE 7 @@ -120,81 +122,6 @@ #define SUN8I_ADDA_ADC_AP_EN_ADCLEN 6 #define SUN8I_ADDA_ADC_AP_EN_ADCG 0 -/* Analog control register access bits */ -#define ADDA_PR 0x0 /* PRCM base + 0x1c0 */ -#define ADDA_PR_RESET BIT(28) -#define ADDA_PR_WRITE BIT(24) -#define ADDA_PR_ADDR_SHIFT 16 -#define ADDA_PR_ADDR_MASK GENMASK(4, 0) -#define ADDA_PR_DATA_IN_SHIFT 8 -#define ADDA_PR_DATA_IN_MASK GENMASK(7, 0) -#define ADDA_PR_DATA_OUT_SHIFT 0 -#define ADDA_PR_DATA_OUT_MASK GENMASK(7, 0) - -/* regmap access bits */ -static int adda_reg_read(void *context, unsigned int reg, unsigned int *val) -{ - void __iomem *base = (void __iomem *)context; - u32 tmp; - - /* De-assert reset */ - writel(readl(base) | ADDA_PR_RESET, base); - - /* Clear write bit */ - writel(readl(base) & ~ADDA_PR_WRITE, base); - - /* Set register address */ - tmp = readl(base); - tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT); - tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT; - writel(tmp, base); - - /* Read back value */ - *val = readl(base) & ADDA_PR_DATA_OUT_MASK; - - return 0; -} - -static int adda_reg_write(void *context, unsigned int reg, unsigned int val) -{ - void __iomem *base = (void __iomem *)context; - u32 tmp; - - /* De-assert reset */ - writel(readl(base) | ADDA_PR_RESET, base); - - /* Set register address */ - tmp = readl(base); - tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT); - tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT; - writel(tmp, base); - - /* Set data to write */ - tmp = readl(base); - tmp &= ~(ADDA_PR_DATA_IN_MASK << ADDA_PR_DATA_IN_SHIFT); - tmp |= (val & ADDA_PR_DATA_IN_MASK) << ADDA_PR_DATA_IN_SHIFT; - writel(tmp, base); - - /* Set write bit to signal a write */ - writel(readl(base) | ADDA_PR_WRITE, base); - - /* Clear write bit */ - writel(readl(base) & ~ADDA_PR_WRITE, base); - - return 0; -} - -static const struct regmap_config adda_pr_regmap_cfg = { - .name = "adda-pr", - .reg_bits = 5, - .reg_stride = 1, - .val_bits = 8, - .reg_read = adda_reg_read, - .reg_write = adda_reg_write, - .fast_io = true, - .max_register = 24, -}; - /* mixer controls */ static const struct snd_kcontrol_new sun8i_codec_mixer_controls[] = { SOC_DAPM_DOUBLE_R("DAC Playback Switch", @@ -912,7 +839,7 @@ static int sun8i_codec_analog_probe(struct platform_device *pdev) return PTR_ERR(base); } - regmap = devm_regmap_init(&pdev->dev, NULL, base, &adda_pr_regmap_cfg); + regmap = sun8i_adda_pr_regmap_init(&pdev->dev, base); if (IS_ERR(regmap)) { dev_err(&pdev->dev, "Failed to create regmap\n"); return PTR_ERR(regmap); diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index fb37dd927e33..92c5de026c43 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -24,6 +24,7 @@ #include <linux/io.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> +#include <linux/log2.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -52,7 +53,6 @@ #define SUN8I_AIF1CLK_CTRL_AIF1_LRCK_INV 13 #define SUN8I_AIF1CLK_CTRL_AIF1_BCLK_DIV 9 #define SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV 6 -#define SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV_16 (1 << 6) #define SUN8I_AIF1CLK_CTRL_AIF1_WORD_SIZ 4 #define SUN8I_AIF1CLK_CTRL_AIF1_WORD_SIZ_16 (1 << 4) #define SUN8I_AIF1CLK_CTRL_AIF1_DATA_FMT 2 @@ -300,12 +300,23 @@ static u8 sun8i_codec_get_bclk_div(struct sun8i_codec *scodec, return best_val; } +static int sun8i_codec_get_lrck_div(unsigned int channels, + unsigned int word_size) +{ + unsigned int div = word_size * channels; + + if (div < 16 || div > 256) + return -EINVAL; + + return ilog2(div) - 4; +} + static int sun8i_codec_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct sun8i_codec *scodec = snd_soc_component_get_drvdata(dai->component); - int sample_rate; + int sample_rate, lrck_div; u8 bclk_div; /* @@ -321,9 +332,14 @@ static int sun8i_codec_hw_params(struct snd_pcm_substream *substream, SUN8I_AIF1CLK_CTRL_AIF1_BCLK_DIV_MASK, bclk_div << SUN8I_AIF1CLK_CTRL_AIF1_BCLK_DIV); + lrck_div = sun8i_codec_get_lrck_div(params_channels(params), + params_physical_width(params)); + if (lrck_div < 0) + return lrck_div; + regmap_update_bits(scodec->regmap, SUN8I_AIF1CLK_CTRL, SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV_MASK, - SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV_16); + lrck_div << SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV); sample_rate = sun8i_codec_get_hw_rate(params); if (sample_rate < 0) @@ -465,7 +481,11 @@ static const struct snd_soc_dapm_route sun8i_codec_dapm_routes[] = { { "Right Digital DAC Mixer", "AIF1 Slot 0 Digital DAC Playback Switch", "AIF1 Slot 0 Right"}, - /* ADC routes */ + /* ADC Routes */ + { "AIF1 Slot 0 Right ADC", NULL, "ADC" }, + { "AIF1 Slot 0 Left ADC", NULL, "ADC" }, + + /* ADC Mixer Routes */ { "Left Digital ADC Mixer", "AIF1 Data Digital ADC Capture Switch", "AIF1 Slot 0 Left ADC" }, { "Right Digital ADC Mixer", "AIF1 Data Digital ADC Capture Switch", @@ -589,16 +609,10 @@ err_pm_disable: static int sun8i_codec_remove(struct platform_device *pdev) { - struct snd_soc_card *card = platform_get_drvdata(pdev); - struct sun8i_codec *scodec = snd_soc_card_get_drvdata(card); - pm_runtime_disable(&pdev->dev); if (!pm_runtime_status_suspended(&pdev->dev)) sun8i_codec_runtime_suspend(&pdev->dev); - clk_disable_unprepare(scodec->clk_module); - clk_disable_unprepare(scodec->clk_bus); - return 0; } diff --git a/sound/soc/tegra/tegra_sgtl5000.c b/sound/soc/tegra/tegra_sgtl5000.c index 45a4aa9d2a47..901457da25ec 100644 --- a/sound/soc/tegra/tegra_sgtl5000.c +++ b/sound/soc/tegra/tegra_sgtl5000.c @@ -149,14 +149,14 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing/invalid\n"); ret = -EINVAL; - goto err; + goto err_put_codec_of_node; } tegra_sgtl5000_dai.platform_of_node = tegra_sgtl5000_dai.cpu_of_node; ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); if (ret) - goto err; + goto err_put_cpu_of_node; ret = snd_soc_register_card(card); if (ret) { @@ -169,6 +169,13 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev) err_fini_utils: tegra_asoc_utils_fini(&machine->util_data); +err_put_cpu_of_node: + of_node_put(tegra_sgtl5000_dai.cpu_of_node); + tegra_sgtl5000_dai.cpu_of_node = NULL; + tegra_sgtl5000_dai.platform_of_node = NULL; +err_put_codec_of_node: + of_node_put(tegra_sgtl5000_dai.codec_of_node); + tegra_sgtl5000_dai.codec_of_node = NULL; err: return ret; } @@ -183,6 +190,12 @@ static int tegra_sgtl5000_driver_remove(struct platform_device *pdev) tegra_asoc_utils_fini(&machine->util_data); + of_node_put(tegra_sgtl5000_dai.cpu_of_node); + tegra_sgtl5000_dai.cpu_of_node = NULL; + tegra_sgtl5000_dai.platform_of_node = NULL; + of_node_put(tegra_sgtl5000_dai.codec_of_node); + tegra_sgtl5000_dai.codec_of_node = NULL; + return ret; } diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig new file mode 100644 index 000000000000..4bf3c15d4e51 --- /dev/null +++ b/sound/soc/ti/Kconfig @@ -0,0 +1,209 @@ +menu "Audio support for Texas Instruments SoCs" +depends on DMA_OMAP || TI_EDMA || COMPILE_TEST + +config SND_SOC_TI_EDMA_PCM + tristate + select SND_SOC_GENERIC_DMAENGINE_PCM + +config SND_SOC_TI_SDMA_PCM + tristate + select SND_SOC_GENERIC_DMAENGINE_PCM + +comment "Texas Instruments DAI support for:" +config SND_SOC_DAVINCI_ASP + tristate "daVinci Audio Serial Port (ASP) or McBSP suport" + depends on ARCH_DAVINCI || COMPILE_TEST + select SND_SOC_TI_EDMA_PCM + help + Say Y or M here if you want audio support via daVinci ASP or McBSP. + The driver only implements the ASP support which is a subset of + daVinci McBSP (w/o the multichannel support). + +config SND_SOC_DAVINCI_MCASP + tristate "Multichannel Audio Serial Port (McASP) support" + select SND_SOC_TI_EDMA_PCM if TI_EDMA + select SND_SOC_TI_SDMA_PCM if DMA_OMAP + help + Say Y or M here if you want to have support for McASP IP found in + various Texas Instruments SoCs like: + - daVinci devices + - Sitara line of SoCs (AM335x, AM438x, etc) + - DRA7x devices + - Keystone devices + +config SND_SOC_DAVINCI_VCIF + tristate "daVinci Voice Interface (VCIF) suport" + depends on ARCH_DAVINCI || COMPILE_TEST + select SND_SOC_TI_EDMA_PCM + help + Say Y or M here if you want audio support via daVinci VCIF. + +config SND_SOC_OMAP_DMIC + tristate "Digital Microphone Module (DMIC) support" + depends on ARCH_OMAP4 || SOC_OMAP5 || COMPILE_TEST + select SND_SOC_TI_SDMA_PCM + help + Say Y or M here if you want to have support for DMIC IP found in + OMAP4 and OMAP5. + +config SND_SOC_OMAP_MCBSP + tristate "Multichannel Buffered Serial Port (McBSP) support" + depends on ARCH_OMAP || ARCH_OMAP1 || COMPILE_TEST + select SND_SOC_TI_SDMA_PCM + help + Say Y or M here if you want to have support for McBSP IP found in + Texas Instruments OMAP1/2/3/4/5 SoCs. + +config SND_SOC_OMAP_MCPDM + tristate "Multichannel PDM Controller (McPDM) support" + depends on ARCH_OMAP4 || SOC_OMAP5 || COMPILE_TEST + select SND_SOC_TI_SDMA_PCM + help + Say Y or M here if you want to have support for McPDM IP found in + OMAP4 and OMAP5. + +comment "Audio support for boards with Texas Instruments SoCs" +config SND_SOC_NOKIA_N810 + tristate "SoC Audio support for Nokia N810" + depends on MACH_NOKIA_N810 && I2C + select SND_SOC_OMAP_MCBSP + select SND_SOC_TLV320AIC3X + help + Say Y or M if you want to add support for SoC audio on Nokia N810. + +config SND_SOC_NOKIA_RX51 + tristate "SoC Audio support for Nokia RX-51" + depends on ARCH_OMAP3 && I2C && GPIOLIB + select SND_SOC_OMAP_MCBSP + select SND_SOC_TLV320AIC3X + select SND_SOC_TPA6130A2 + help + Say Y or M if you want to add support for SoC audio on Nokia RX-51 + hardware. This is also known as Nokia N900 product. + +config SND_SOC_OMAP3_PANDORA + tristate "SoC Audio support for OMAP3 Pandora" + depends on ARCH_OMAP3 + depends on TWL4030_CORE + select SND_SOC_OMAP_MCBSP + select SND_SOC_TWL4030 + help + Say Y or M if you want to add support for SoC audio on the OMAP3 Pandora. + +config SND_SOC_OMAP3_TWL4030 + tristate "SoC Audio support for OMAP3 based boards with twl4030 codec" + depends on ARCH_OMAP3 || COMPILE_TEST + depends on TWL4030_CORE + select SND_SOC_OMAP_MCBSP + select SND_SOC_TWL4030 + help + Say Y or M if you want to add support for SoC audio on OMAP3 based + boards using twl4030 as codec. This driver currently supports: + - Beagleboard or Devkit8000 + - Gumstix Overo or CompuLab CM-T35/CM-T3730 + - IGEP v2 + - OMAP3EVM + - SDP3430 + - Zoom2 + +config SND_SOC_OMAP_ABE_TWL6040 + tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" + depends on TWL6040_CORE && COMMON_CLK + depends on ARCH_OMAP4 || (SOC_OMAP5 && MFD_PALMAS) || COMPILE_TEST + select SND_SOC_OMAP_DMIC + select SND_SOC_OMAP_MCPDM + select SND_SOC_TWL6040 + help + Say Y or M if you want to add support for SoC audio on OMAP boards + using ABE and twl6040 codec. This driver currently supports: + - SDP4430/Blaze boards + - PandaBoard (4430) + - PandaBoardES (4460) + - OMAP5 uEVM + +config SND_SOC_OMAP_AMS_DELTA + tristate "SoC Audio support for Amstrad E3 (Delta) videophone" + depends on MACH_AMS_DELTA && TTY + select SND_SOC_OMAP_MCBSP + select SND_SOC_CX20442 + help + Say Y or M if you want to add support for SoC audio device + connected to a handset and a speakerphone found on Amstrad E3 (Delta) + videophone. + + Note that in order to get those devices fully supported, you have to + build the kernel with standard serial port driver included and + configured for at least 4 ports. Then, from userspace, you must load + a line discipline #19 on the modem (ttyS3) serial line. The simplest + way to achieve this is to install util-linux-ng and use the included + ldattach utility. This can be started automatically from udev, + a simple rule like this one should do the trick (it does for me): + ACTION=="add", KERNEL=="controlC0", \ + RUN+="/usr/sbin/ldattach 19 /dev/ttyS3" + +config SND_SOC_OMAP_HDMI + tristate "OMAP4/5 HDMI audio support" + depends on OMAP4_DSS_HDMI || OMAP5_DSS_HDMI || COMPILE_TEST + select SND_SOC_TI_SDMA_PCM + help + For HDMI audio to work OMAPDSS HDMI support should be + enabled. + The hdmi audio driver implements cpu-dai component using the + callbacks provided by OMAPDSS and registers the component + under DSS HDMI device. Omap-pcm is registered for platform + component also under DSS HDMI device. Dummy codec is used as + as codec component. The hdmi audio driver implements also + the card and registers it under its own platform device. + The device for the driver is registered by OMAPDSS hdmi + driver. + +config SND_SOC_OMAP_OSK5912 + tristate "SoC Audio support for omap osk5912" + depends on MACH_OMAP_OSK && I2C + select SND_SOC_OMAP_MCBSP + select SND_SOC_TLV320AIC23_I2C + help + Say Y or M if you want to add support for SoC audio on osk5912. + +config SND_SOC_DAVINCI_EVM + tristate "SoC Audio support for DaVinci EVMs" + depends on ARCH_DAVINCI && I2C + select SND_SOC_DAVINCI_ASP if MACH_DAVINCI_DM355_EVM + select SND_SOC_DAVINCI_ASP if SND_SOC_DM365_AIC3X_CODEC + select SND_SOC_DAVINCI_VCIF if SND_SOC_DM365_VOICE_CODEC + select SND_SOC_DAVINCI_ASP if MACH_DAVINCI_EVM # DM6446 + select SND_SOC_DAVINCI_MCASP if MACH_DAVINCI_DM6467_EVM + select SND_SOC_SPDIF if MACH_DAVINCI_DM6467_EVM + select SND_SOC_DAVINCI_MCASP if MACH_DAVINCI_DA830_EVM + select SND_SOC_DAVINCI_MCASP if MACH_DAVINCI_DA850_EVM + select SND_SOC_TLV320AIC3X + help + Say Y if you want to add support for SoC audio on the following TI + DaVinci EVM platforms: + - DM355 + - DM365 + - DM6446 + - DM6447 + - DM830 + - DM850 + +choice + prompt "DM365 codec select" + depends on SND_SOC_DAVINCI_EVM + depends on MACH_DAVINCI_DM365_EVM + +config SND_SOC_DM365_AIC3X_CODEC + bool "Audio Codec - AIC3101" + help + Say Y if you want to add support for AIC3101 audio codec + +config SND_SOC_DM365_VOICE_CODEC + bool "Voice Codec - CQ93VC" + select MFD_DAVINCI_VOICECODEC + select SND_SOC_CQ0093VC + help + Say Y if you want to add support for SoC On-chip voice codec +endchoice + +endmenu + diff --git a/sound/soc/ti/Makefile b/sound/soc/ti/Makefile new file mode 100644 index 000000000000..08c44d56ef3e --- /dev/null +++ b/sound/soc/ti/Makefile @@ -0,0 +1,44 @@ +# SPDX-License-Identifier: GPL-2.0 + +# Platform drivers +snd-soc-ti-edma-objs := edma-pcm.o +snd-soc-ti-sdma-objs := sdma-pcm.o + +obj-$(CONFIG_SND_SOC_TI_EDMA_PCM) += snd-soc-ti-edma.o +obj-$(CONFIG_SND_SOC_TI_SDMA_PCM) += snd-soc-ti-sdma.o + +# CPU DAI drivers +snd-soc-davinci-asp-objs := davinci-i2s.o +snd-soc-davinci-mcasp-objs := davinci-mcasp.o +snd-soc-davinci-vcif-objs := davinci-vcif.o +snd-soc-omap-dmic-objs := omap-dmic.o +snd-soc-omap-mcbsp-objs := omap-mcbsp.o omap-mcbsp-st.o +snd-soc-omap-mcpdm-objs := omap-mcpdm.o + +obj-$(CONFIG_SND_SOC_DAVINCI_ASP) += snd-soc-davinci-asp.o +obj-$(CONFIG_SND_SOC_DAVINCI_MCASP) += snd-soc-davinci-mcasp.o +obj-$(CONFIG_SND_SOC_DAVINCI_VCIF) += snd-soc-davinci-vcif.o +obj-$(CONFIG_SND_SOC_OMAP_DMIC) += snd-soc-omap-dmic.o +obj-$(CONFIG_SND_SOC_OMAP_MCBSP) += snd-soc-omap-mcbsp.o +obj-$(CONFIG_SND_SOC_OMAP_MCPDM) += snd-soc-omap-mcpdm.o + +# Machine drivers +snd-soc-davinci-evm-objs := davinci-evm.o +snd-soc-n810-objs := n810.o +snd-soc-rx51-objs := rx51.o +snd-soc-omap3pandora-objs := omap3pandora.o +snd-soc-omap-twl4030-objs := omap-twl4030.o +snd-soc-omap-abe-twl6040-objs := omap-abe-twl6040.o +snd-soc-ams-delta-objs := ams-delta.o +snd-soc-omap-hdmi-objs := omap-hdmi.o +snd-soc-osk5912-objs := osk5912.o + +obj-$(CONFIG_SND_SOC_DAVINCI_EVM) += snd-soc-davinci-evm.o +obj-$(CONFIG_SND_SOC_NOKIA_N810) += snd-soc-n810.o +obj-$(CONFIG_SND_SOC_NOKIA_RX51) += snd-soc-rx51.o +obj-$(CONFIG_SND_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o +obj-$(CONFIG_SND_SOC_OMAP3_TWL4030) += snd-soc-omap-twl4030.o +obj-$(CONFIG_SND_SOC_OMAP_ABE_TWL6040) += snd-soc-omap-abe-twl6040.o +obj-$(CONFIG_SND_SOC_OMAP_AMS_DELTA) += snd-soc-ams-delta.o +obj-$(CONFIG_SND_SOC_OMAP_HDMI) += snd-soc-omap-hdmi.o +obj-$(CONFIG_SND_SOC_OMAP_OSK5912) += snd-soc-osk5912.o diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/ti/ams-delta.c index 4dce494dfbd3..4dce494dfbd3 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/ti/ams-delta.c diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/ti/davinci-evm.c index 7a369e0f2093..4869d6311510 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/ti/davinci-evm.c @@ -170,7 +170,7 @@ static struct snd_soc_dai_link dm355_evm_dai = { }; static struct snd_soc_dai_link dm365_evm_dai = { -#ifdef CONFIG_SND_DM365_AIC3X_CODEC +#ifdef CONFIG_SND_SOC_DM365_AIC3X_CODEC .name = "TLV320AIC3X", .stream_name = "AIC3X", .cpu_dai_name = "davinci-mcbsp", @@ -181,7 +181,7 @@ static struct snd_soc_dai_link dm365_evm_dai = { .ops = &evm_ops, .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF, -#elif defined(CONFIG_SND_DM365_VOICE_CODEC) +#elif defined(CONFIG_SND_SOC_DM365_VOICE_CODEC) .name = "Voice Codec - CQ93VC", .stream_name = "CQ93", .cpu_dai_name = "davinci-vcif", diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c index a3206e65e5e5..a3206e65e5e5 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/ti/davinci-i2s.c diff --git a/sound/soc/davinci/davinci-i2s.h b/sound/soc/ti/davinci-i2s.h index 48dac3e2521a..48dac3e2521a 100644 --- a/sound/soc/davinci/davinci-i2s.h +++ b/sound/soc/ti/davinci-i2s.h diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index f70db8412c7c..a10fcb5963c6 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -28,6 +28,7 @@ #include <linux/of_device.h> #include <linux/platform_data/davinci_asp.h> #include <linux/math64.h> +#include <linux/bitmap.h> #include <sound/asoundef.h> #include <sound/core.h> @@ -38,7 +39,7 @@ #include <sound/dmaengine_pcm.h> #include "edma-pcm.h" -#include "../omap/sdma-pcm.h" +#include "sdma-pcm.h" #include "davinci-mcasp.h" #define MCASP_MAX_AFIFO_DEPTH 64 @@ -84,6 +85,7 @@ struct davinci_mcasp { u32 tdm_mask[2]; int slot_width; u8 op_mode; + u8 dismod; u8 num_serializer; u8 *serial_dir; u8 version; @@ -95,6 +97,8 @@ struct davinci_mcasp { int sysclk_freq; bool bclk_master; + unsigned long pdir; /* Pin direction bitfield */ + /* McASP FIFO related */ u8 txnumevt; u8 rxnumevt; @@ -104,7 +108,7 @@ struct davinci_mcasp { /* Used for comstraint setting on the second stream */ u32 channels; -#ifdef CONFIG_PM_SLEEP +#ifdef CONFIG_PM struct davinci_mcasp_context context; #endif @@ -169,6 +173,30 @@ static bool mcasp_is_synchronous(struct davinci_mcasp *mcasp) return !(aclkxctl & TX_ASYNC) && rxfmctl & AFSRE; } +static inline void mcasp_set_clk_pdir(struct davinci_mcasp *mcasp, bool enable) +{ + u32 bit = PIN_BIT_AMUTE; + + for_each_set_bit_from(bit, &mcasp->pdir, PIN_BIT_AFSR + 1) { + if (enable) + mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, BIT(bit)); + else + mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, BIT(bit)); + } +} + +static inline void mcasp_set_axr_pdir(struct davinci_mcasp *mcasp, bool enable) +{ + u32 bit; + + for_each_set_bit(bit, &mcasp->pdir, PIN_BIT_AFSR) { + if (enable) + mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, BIT(bit)); + else + mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, BIT(bit)); + } +} + static void mcasp_start_rx(struct davinci_mcasp *mcasp) { if (mcasp->rxnumevt) { /* enable FIFO */ @@ -192,6 +220,7 @@ static void mcasp_start_rx(struct davinci_mcasp *mcasp) } /* Activate serializer(s) */ + mcasp_set_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR); /* Release RX state machine */ mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); @@ -219,7 +248,10 @@ static void mcasp_start_tx(struct davinci_mcasp *mcasp) /* Start clocks */ mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST); mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); + mcasp_set_clk_pdir(mcasp, true); + /* Activate serializer(s) */ + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR); /* wait for XDATA to be cleared */ @@ -228,6 +260,8 @@ static void mcasp_start_tx(struct davinci_mcasp *mcasp) (cnt < 100000)) cnt++; + mcasp_set_axr_pdir(mcasp, true); + /* Release TX state machine */ mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); /* Release Frame Sync generator */ @@ -258,8 +292,10 @@ static void mcasp_stop_rx(struct davinci_mcasp *mcasp) * In synchronous mode stop the TX clocks if no other stream is * running */ - if (mcasp_is_synchronous(mcasp) && !mcasp->streams) + if (mcasp_is_synchronous(mcasp) && !mcasp->streams) { + mcasp_set_clk_pdir(mcasp, false); mcasp_set_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, 0); + } mcasp_set_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, 0); mcasp_set_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); @@ -285,6 +321,9 @@ static void mcasp_stop_tx(struct davinci_mcasp *mcasp) */ if (mcasp_is_synchronous(mcasp) && mcasp->streams) val = TXHCLKRST | TXCLKRST | TXFSRST; + else + mcasp_set_clk_pdir(mcasp, false); + mcasp_set_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, val); mcasp_set_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); @@ -294,6 +333,8 @@ static void mcasp_stop_tx(struct davinci_mcasp *mcasp) mcasp_clr_bits(mcasp, reg, FIFO_ENABLE); } + + mcasp_set_axr_pdir(mcasp, false); } static void davinci_mcasp_stop(struct davinci_mcasp *mcasp, int stream) @@ -444,8 +485,13 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE); - mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, ACLKX | ACLKR); - mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR); + /* BCLK */ + set_bit(PIN_BIT_ACLKX, &mcasp->pdir); + set_bit(PIN_BIT_ACLKR, &mcasp->pdir); + /* Frame Sync */ + set_bit(PIN_BIT_AFSX, &mcasp->pdir); + set_bit(PIN_BIT_AFSR, &mcasp->pdir); + mcasp->bclk_master = 1; break; case SND_SOC_DAIFMT_CBS_CFM: @@ -456,8 +502,13 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE); - mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, ACLKX | ACLKR); - mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR); + /* BCLK */ + set_bit(PIN_BIT_ACLKX, &mcasp->pdir); + set_bit(PIN_BIT_ACLKR, &mcasp->pdir); + /* Frame Sync */ + clear_bit(PIN_BIT_AFSX, &mcasp->pdir); + clear_bit(PIN_BIT_AFSR, &mcasp->pdir); + mcasp->bclk_master = 1; break; case SND_SOC_DAIFMT_CBM_CFS: @@ -468,8 +519,13 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE); - mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, ACLKX | ACLKR); - mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR); + /* BCLK */ + clear_bit(PIN_BIT_ACLKX, &mcasp->pdir); + clear_bit(PIN_BIT_ACLKR, &mcasp->pdir); + /* Frame Sync */ + set_bit(PIN_BIT_AFSX, &mcasp->pdir); + set_bit(PIN_BIT_AFSR, &mcasp->pdir); + mcasp->bclk_master = 0; break; case SND_SOC_DAIFMT_CBM_CFM: @@ -480,8 +536,13 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE); - mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, - ACLKX | AFSX | ACLKR | AHCLKR | AFSR); + /* BCLK */ + clear_bit(PIN_BIT_ACLKX, &mcasp->pdir); + clear_bit(PIN_BIT_ACLKR, &mcasp->pdir); + /* Frame Sync */ + clear_bit(PIN_BIT_AFSX, &mcasp->pdir); + clear_bit(PIN_BIT_AFSR, &mcasp->pdir); + mcasp->bclk_master = 0; break; default: @@ -596,11 +657,11 @@ static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, if (dir == SND_SOC_CLOCK_OUT) { mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); - mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AHCLKX); + set_bit(PIN_BIT_AHCLKX, &mcasp->pdir); } else { mcasp_clr_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); mcasp_clr_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); - mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AHCLKX); + clear_bit(PIN_BIT_AHCLKX, &mcasp->pdir); } mcasp->sysclk_freq = freq; @@ -773,17 +834,23 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, mcasp->serial_dir[i]); if (mcasp->serial_dir[i] == TX_MODE && tx_ser < max_active_serializers) { - mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AXR(i)); mcasp_mod_bits(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i), - DISMOD_LOW, DISMOD_MASK); + mcasp->dismod, DISMOD_MASK); + set_bit(PIN_BIT_AXR(i), &mcasp->pdir); tx_ser++; } else if (mcasp->serial_dir[i] == RX_MODE && rx_ser < max_active_serializers) { - mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AXR(i)); + clear_bit(PIN_BIT_AXR(i), &mcasp->pdir); rx_ser++; } else if (mcasp->serial_dir[i] == INACTIVE_MODE) { mcasp_mod_bits(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i), SRMOD_INACTIVE, SRMOD_MASK); + clear_bit(PIN_BIT_AXR(i), &mcasp->pdir); + } else if (mcasp->serial_dir[i] == TX_MODE) { + /* Unused TX pins, clear PDIR */ + mcasp_mod_bits(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i), + mcasp->dismod, DISMOD_MASK); + clear_bit(PIN_BIT_AXR(i), &mcasp->pdir); } } @@ -1041,6 +1108,42 @@ static int davinci_mcasp_calc_clk_div(struct davinci_mcasp *mcasp, return error_ppm; } +static inline u32 davinci_mcasp_tx_delay(struct davinci_mcasp *mcasp) +{ + if (!mcasp->txnumevt) + return 0; + + return mcasp_get_reg(mcasp, mcasp->fifo_base + MCASP_WFIFOSTS_OFFSET); +} + +static inline u32 davinci_mcasp_rx_delay(struct davinci_mcasp *mcasp) +{ + if (!mcasp->rxnumevt) + return 0; + + return mcasp_get_reg(mcasp, mcasp->fifo_base + MCASP_RFIFOSTS_OFFSET); +} + +static snd_pcm_sframes_t davinci_mcasp_delay( + struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + u32 fifo_use; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + fifo_use = davinci_mcasp_tx_delay(mcasp); + else + fifo_use = davinci_mcasp_rx_delay(mcasp); + + /* + * Divide the used locations with the channel count to get the + * FIFO usage in samples (don't care about partial samples in the + * buffer). + */ + return fifo_use / substream->runtime->channels; +} + static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) @@ -1365,6 +1468,7 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = { .startup = davinci_mcasp_startup, .shutdown = davinci_mcasp_shutdown, .trigger = davinci_mcasp_trigger, + .delay = davinci_mcasp_delay, .hw_params = davinci_mcasp_hw_params, .set_fmt = davinci_mcasp_set_dai_fmt, .set_clkdiv = davinci_mcasp_set_clkdiv, @@ -1382,74 +1486,6 @@ static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai) return 0; } -#ifdef CONFIG_PM_SLEEP -static int davinci_mcasp_suspend(struct snd_soc_dai *dai) -{ - struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); - struct davinci_mcasp_context *context = &mcasp->context; - u32 reg; - int i; - - context->pm_state = pm_runtime_active(mcasp->dev); - if (!context->pm_state) - pm_runtime_get_sync(mcasp->dev); - - for (i = 0; i < ARRAY_SIZE(context_regs); i++) - context->config_regs[i] = mcasp_get_reg(mcasp, context_regs[i]); - - if (mcasp->txnumevt) { - reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; - context->afifo_regs[0] = mcasp_get_reg(mcasp, reg); - } - if (mcasp->rxnumevt) { - reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; - context->afifo_regs[1] = mcasp_get_reg(mcasp, reg); - } - - for (i = 0; i < mcasp->num_serializer; i++) - context->xrsr_regs[i] = mcasp_get_reg(mcasp, - DAVINCI_MCASP_XRSRCTL_REG(i)); - - pm_runtime_put_sync(mcasp->dev); - - return 0; -} - -static int davinci_mcasp_resume(struct snd_soc_dai *dai) -{ - struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); - struct davinci_mcasp_context *context = &mcasp->context; - u32 reg; - int i; - - pm_runtime_get_sync(mcasp->dev); - - for (i = 0; i < ARRAY_SIZE(context_regs); i++) - mcasp_set_reg(mcasp, context_regs[i], context->config_regs[i]); - - if (mcasp->txnumevt) { - reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; - mcasp_set_reg(mcasp, reg, context->afifo_regs[0]); - } - if (mcasp->rxnumevt) { - reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; - mcasp_set_reg(mcasp, reg, context->afifo_regs[1]); - } - - for (i = 0; i < mcasp->num_serializer; i++) - mcasp_set_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i), - context->xrsr_regs[i]); - - if (!context->pm_state) - pm_runtime_put_sync(mcasp->dev); - - return 0; -} -#else -#define davinci_mcasp_suspend NULL -#define davinci_mcasp_resume NULL -#endif - #define DAVINCI_MCASP_RATES SNDRV_PCM_RATE_8000_192000 #define DAVINCI_MCASP_PCM_FMTS (SNDRV_PCM_FMTBIT_S8 | \ @@ -1467,8 +1503,6 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { { .name = "davinci-mcasp.0", .probe = davinci_mcasp_dai_probe, - .suspend = davinci_mcasp_suspend, - .resume = davinci_mcasp_resume, .playback = { .channels_min = 1, .channels_max = 32 * 16, @@ -1608,6 +1642,7 @@ static struct davinci_mcasp_pdata *davinci_mcasp_set_pdata_from_of( if (pdev->dev.platform_data) { pdata = pdev->dev.platform_data; + pdata->dismod = DISMOD_LOW; return pdata; } else if (match) { pdata = devm_kmemdup(&pdev->dev, match->data, sizeof(*pdata), @@ -1697,6 +1732,18 @@ static struct davinci_mcasp_pdata *davinci_mcasp_set_pdata_from_of( if (ret >= 0) pdata->sram_size_capture = val; + ret = of_property_read_u32(np, "dismod", &val); + if (ret >= 0) { + if (val == 0 || val == 2 || val == 3) { + pdata->dismod = DISMOD_VAL(val); + } else { + dev_warn(&pdev->dev, "Invalid dismod value: %u\n", val); + pdata->dismod = DISMOD_LOW; + } + } else { + pdata->dismod = DISMOD_LOW; + } + return pdata; nodata: @@ -1859,7 +1906,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) } mcasp->num_serializer = pdata->num_serializer; -#ifdef CONFIG_PM_SLEEP +#ifdef CONFIG_PM mcasp->context.xrsr_regs = devm_kcalloc(&pdev->dev, mcasp->num_serializer, sizeof(u32), GFP_KERNEL); @@ -1872,6 +1919,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp->version = pdata->version; mcasp->txnumevt = pdata->txnumevt; mcasp->rxnumevt = pdata->rxnumevt; + mcasp->dismod = pdata->dismod; mcasp->dev = &pdev->dev; @@ -2031,9 +2079,9 @@ static int davinci_mcasp_probe(struct platform_device *pdev) ret = davinci_mcasp_get_dma_type(mcasp); switch (ret) { case PCM_EDMA: -#if IS_BUILTIN(CONFIG_SND_EDMA_SOC) || \ - (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ - IS_MODULE(CONFIG_SND_EDMA_SOC)) +#if IS_BUILTIN(CONFIG_SND_SOC_TI_EDMA_PCM) || \ + (IS_MODULE(CONFIG_SND_SOC_DAVINCI_MCASP) && \ + IS_MODULE(CONFIG_SND_SOC_TI_EDMA_PCM)) ret = edma_pcm_platform_register(&pdev->dev); #else dev_err(&pdev->dev, "Missing SND_EDMA_SOC\n"); @@ -2042,9 +2090,9 @@ static int davinci_mcasp_probe(struct platform_device *pdev) #endif break; case PCM_SDMA: -#if IS_BUILTIN(CONFIG_SND_SDMA_SOC) || \ - (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ - IS_MODULE(CONFIG_SND_SDMA_SOC)) +#if IS_BUILTIN(CONFIG_SND_SOC_TI_SDMA_PCM) || \ + (IS_MODULE(CONFIG_SND_SOC_DAVINCI_MCASP) && \ + IS_MODULE(CONFIG_SND_SOC_TI_SDMA_PCM)) ret = sdma_pcm_platform_register(&pdev->dev, NULL, NULL); #else dev_err(&pdev->dev, "Missing SND_SDMA_SOC\n"); @@ -2078,11 +2126,73 @@ static int davinci_mcasp_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM +static int davinci_mcasp_runtime_suspend(struct device *dev) +{ + struct davinci_mcasp *mcasp = dev_get_drvdata(dev); + struct davinci_mcasp_context *context = &mcasp->context; + u32 reg; + int i; + + for (i = 0; i < ARRAY_SIZE(context_regs); i++) + context->config_regs[i] = mcasp_get_reg(mcasp, context_regs[i]); + + if (mcasp->txnumevt) { + reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; + context->afifo_regs[0] = mcasp_get_reg(mcasp, reg); + } + if (mcasp->rxnumevt) { + reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; + context->afifo_regs[1] = mcasp_get_reg(mcasp, reg); + } + + for (i = 0; i < mcasp->num_serializer; i++) + context->xrsr_regs[i] = mcasp_get_reg(mcasp, + DAVINCI_MCASP_XRSRCTL_REG(i)); + + return 0; +} + +static int davinci_mcasp_runtime_resume(struct device *dev) +{ + struct davinci_mcasp *mcasp = dev_get_drvdata(dev); + struct davinci_mcasp_context *context = &mcasp->context; + u32 reg; + int i; + + for (i = 0; i < ARRAY_SIZE(context_regs); i++) + mcasp_set_reg(mcasp, context_regs[i], context->config_regs[i]); + + if (mcasp->txnumevt) { + reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; + mcasp_set_reg(mcasp, reg, context->afifo_regs[0]); + } + if (mcasp->rxnumevt) { + reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; + mcasp_set_reg(mcasp, reg, context->afifo_regs[1]); + } + + for (i = 0; i < mcasp->num_serializer; i++) + mcasp_set_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i), + context->xrsr_regs[i]); + + return 0; +} + +#endif + +static const struct dev_pm_ops davinci_mcasp_pm_ops = { + SET_RUNTIME_PM_OPS(davinci_mcasp_runtime_suspend, + davinci_mcasp_runtime_resume, + NULL) +}; + static struct platform_driver davinci_mcasp_driver = { .probe = davinci_mcasp_probe, .remove = davinci_mcasp_remove, .driver = { .name = "davinci-mcasp", + .pm = &davinci_mcasp_pm_ops, .of_match_table = mcasp_dt_ids, }, }; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/ti/davinci-mcasp.h index afddc8010c54..5e4060d8fe56 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/ti/davinci-mcasp.h @@ -108,27 +108,18 @@ /* * DAVINCI_MCASP_PFUNC_REG - Pin Function / GPIO Enable Register Bits - */ -#define AXR(n) (1<<n) -#define PFUNC_AMUTE BIT(25) -#define ACLKX BIT(26) -#define AHCLKX BIT(27) -#define AFSX BIT(28) -#define ACLKR BIT(29) -#define AHCLKR BIT(30) -#define AFSR BIT(31) - -/* * DAVINCI_MCASP_PDIR_REG - Pin Direction Register Bits + * DAVINCI_MCASP_PDOUT_REG - Pin output in GPIO mode + * DAVINCI_MCASP_PDSET_REG - Pin input in GPIO mode */ -#define AXR(n) (1<<n) -#define PDIR_AMUTE BIT(25) -#define ACLKX BIT(26) -#define AHCLKX BIT(27) -#define AFSX BIT(28) -#define ACLKR BIT(29) -#define AHCLKR BIT(30) -#define AFSR BIT(31) +#define PIN_BIT_AXR(n) (n) +#define PIN_BIT_AMUTE 25 +#define PIN_BIT_ACLKX 26 +#define PIN_BIT_AHCLKX 27 +#define PIN_BIT_AFSX 28 +#define PIN_BIT_ACLKR 29 +#define PIN_BIT_AHCLKR 30 +#define PIN_BIT_AFSR 31 /* * DAVINCI_MCASP_TXDITCTL_REG - Transmit DIT Control Register Bits @@ -218,6 +209,7 @@ #define DISMOD_3STATE (0x0) #define DISMOD_LOW (0x2 << 2) #define DISMOD_HIGH (0x3 << 2) +#define DISMOD_VAL(x) ((x) << 2) #define DISMOD_MASK DISMOD_HIGH #define TXSTATE BIT(4) #define RXSTATE BIT(5) diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/ti/davinci-vcif.c index 5415b72393fa..5415b72393fa 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/ti/davinci-vcif.c diff --git a/sound/soc/davinci/edma-pcm.c b/sound/soc/ti/edma-pcm.c index 59e588abe54b..59e588abe54b 100644 --- a/sound/soc/davinci/edma-pcm.c +++ b/sound/soc/ti/edma-pcm.c diff --git a/sound/soc/davinci/edma-pcm.h b/sound/soc/ti/edma-pcm.h index b0957744851c..8058bdb0f032 100644 --- a/sound/soc/davinci/edma-pcm.h +++ b/sound/soc/ti/edma-pcm.h @@ -20,13 +20,13 @@ #ifndef __EDMA_PCM_H__ #define __EDMA_PCM_H__ -#if IS_ENABLED(CONFIG_SND_EDMA_SOC) +#if IS_ENABLED(CONFIG_SND_SOC_TI_EDMA_PCM) int edma_pcm_platform_register(struct device *dev); #else static inline int edma_pcm_platform_register(struct device *dev) { return 0; } -#endif /* CONFIG_SND_EDMA_SOC */ +#endif /* CONFIG_SND_SOC_TI_EDMA_PCM */ #endif /* __EDMA_PCM_H__ */ diff --git a/sound/soc/omap/n810.c b/sound/soc/ti/n810.c index 9cfefe44a75f..9cfefe44a75f 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/ti/n810.c diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/ti/omap-abe-twl6040.c index d5ae9eb8c756..fed45b41f9d3 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/ti/omap-abe-twl6040.c @@ -36,6 +36,8 @@ #include "../codecs/twl6040.h" struct abe_twl6040 { + struct snd_soc_card card; + struct snd_soc_dai_link dai_links[2]; int jack_detection; /* board can detect jack events */ int mclk_freq; /* MCLK frequency speed for twl6040 */ }; @@ -208,40 +210,10 @@ static int omap_abe_dmic_init(struct snd_soc_pcm_runtime *rtd) ARRAY_SIZE(dmic_audio_map)); } -/* Digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link abe_twl6040_dai_links[] = { - { - .name = "TWL6040", - .stream_name = "TWL6040", - .codec_dai_name = "twl6040-legacy", - .codec_name = "twl6040-codec", - .init = omap_abe_twl6040_init, - .ops = &omap_abe_ops, - }, - { - .name = "DMIC", - .stream_name = "DMIC Capture", - .codec_dai_name = "dmic-hifi", - .codec_name = "dmic-codec", - .init = omap_abe_dmic_init, - .ops = &omap_abe_dmic_ops, - }, -}; - -/* Audio machine driver */ -static struct snd_soc_card omap_abe_card = { - .owner = THIS_MODULE, - - .dapm_widgets = twl6040_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets), - .dapm_routes = audio_map, - .num_dapm_routes = ARRAY_SIZE(audio_map), -}; - static int omap_abe_probe(struct platform_device *pdev) { struct device_node *node = pdev->dev.of_node; - struct snd_soc_card *card = &omap_abe_card; + struct snd_soc_card *card; struct device_node *dai_node; struct abe_twl6040 *priv; int num_links = 0; @@ -252,12 +224,18 @@ static int omap_abe_probe(struct platform_device *pdev) return -ENODEV; } - card->dev = &pdev->dev; - priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL); if (priv == NULL) return -ENOMEM; + card = &priv->card; + card->dev = &pdev->dev; + card->owner = THIS_MODULE; + card->dapm_widgets = twl6040_dapm_widgets; + card->num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets); + card->dapm_routes = audio_map; + card->num_dapm_routes = ARRAY_SIZE(audio_map); + if (snd_soc_of_parse_card_name(card, "ti,model")) { dev_err(&pdev->dev, "Card name is not provided\n"); return -ENODEV; @@ -274,14 +252,27 @@ static int omap_abe_probe(struct platform_device *pdev) dev_err(&pdev->dev, "McPDM node is not provided\n"); return -EINVAL; } - abe_twl6040_dai_links[0].cpu_of_node = dai_node; - abe_twl6040_dai_links[0].platform_of_node = dai_node; + + priv->dai_links[0].name = "DMIC"; + priv->dai_links[0].stream_name = "TWL6040"; + priv->dai_links[0].cpu_of_node = dai_node; + priv->dai_links[0].platform_of_node = dai_node; + priv->dai_links[0].codec_dai_name = "twl6040-legacy"; + priv->dai_links[0].codec_name = "twl6040-codec"; + priv->dai_links[0].init = omap_abe_twl6040_init; + priv->dai_links[0].ops = &omap_abe_ops; dai_node = of_parse_phandle(node, "ti,dmic", 0); if (dai_node) { num_links = 2; - abe_twl6040_dai_links[1].cpu_of_node = dai_node; - abe_twl6040_dai_links[1].platform_of_node = dai_node; + priv->dai_links[1].name = "TWL6040"; + priv->dai_links[1].stream_name = "DMIC Capture"; + priv->dai_links[1].cpu_of_node = dai_node; + priv->dai_links[1].platform_of_node = dai_node; + priv->dai_links[1].codec_dai_name = "dmic-hifi"; + priv->dai_links[1].codec_name = "dmic-codec"; + priv->dai_links[1].init = omap_abe_dmic_init; + priv->dai_links[1].ops = &omap_abe_dmic_ops; } else { num_links = 1; } @@ -300,7 +291,7 @@ static int omap_abe_probe(struct platform_device *pdev) return -ENODEV; } - card->dai_link = abe_twl6040_dai_links; + card->dai_link = priv->dai_links; card->num_links = num_links; snd_soc_card_set_drvdata(card, priv); diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/ti/omap-dmic.c index fe966272bd0c..cba9645b6487 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/ti/omap-dmic.c @@ -48,6 +48,8 @@ struct omap_dmic { struct device *dev; void __iomem *io_base; struct clk *fclk; + struct pm_qos_request pm_qos_req; + int latency; int fclk_freq; int out_freq; int clk_div; @@ -124,6 +126,8 @@ static void omap_dmic_dai_shutdown(struct snd_pcm_substream *substream, mutex_lock(&dmic->mutex); + pm_qos_remove_request(&dmic->pm_qos_req); + if (!dai->active) dmic->active = 0; @@ -228,6 +232,8 @@ static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream, /* packet size is threshold * channels */ dma_data = snd_soc_dai_get_dma_data(dai, substream); dma_data->maxburst = dmic->threshold * channels; + dmic->latency = (OMAP_DMIC_THRES_MAX - dmic->threshold) * USEC_PER_SEC / + params_rate(params); return 0; } @@ -238,6 +244,9 @@ static int omap_dmic_dai_prepare(struct snd_pcm_substream *substream, struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai); u32 ctrl; + if (pm_qos_request_active(&dmic->pm_qos_req)) + pm_qos_update_request(&dmic->pm_qos_req, dmic->latency); + /* Configure uplink threshold */ omap_dmic_write(dmic, OMAP_DMIC_FIFO_CTRL_REG, dmic->threshold); diff --git a/sound/soc/omap/omap-dmic.h b/sound/soc/ti/omap-dmic.h index 231e728bff0e..231e728bff0e 100644 --- a/sound/soc/omap/omap-dmic.h +++ b/sound/soc/ti/omap-dmic.h diff --git a/sound/soc/omap/omap-hdmi-audio.c b/sound/soc/ti/omap-hdmi.c index 8a99a8837dc9..673a9eb153b2 100644 --- a/sound/soc/omap/omap-hdmi-audio.c +++ b/sound/soc/ti/omap-hdmi.c @@ -348,7 +348,7 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) default: return -EINVAL; } - ret = snd_soc_register_component(ad->dssdev, &omap_hdmi_component, + ret = devm_snd_soc_register_component(ad->dssdev, &omap_hdmi_component, dai_drv, 1); if (ret) return ret; @@ -383,7 +383,6 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) ret = snd_soc_register_card(card); if (ret) { dev_err(dev, "snd_soc_register_card failed (%d)\n", ret); - snd_soc_unregister_component(ad->dssdev); return ret; } @@ -400,7 +399,6 @@ static int omap_hdmi_audio_remove(struct platform_device *pdev) struct hdmi_audio_data *ad = platform_get_drvdata(pdev); snd_soc_unregister_card(ad->card); - snd_soc_unregister_component(ad->dssdev); return 0; } diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/ti/omap-mcbsp-priv.h index 46ae1269a698..7865cda4bf0a 100644 --- a/sound/soc/omap/mcbsp.h +++ b/sound/soc/ti/omap-mcbsp-priv.h @@ -1,28 +1,15 @@ +/* SPDX-License-Identifier: GPL-2.0 */ /* - * sound/soc/omap/mcbsp.h - * * OMAP Multi-Channel Buffered Serial Port * * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com> * Peter Ujfalusi <peter.ujfalusi@ti.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. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * */ -#ifndef __ASOC_MCBSP_H -#define __ASOC_MCBSP_H + +#ifndef __OMAP_MCBSP_PRIV_H__ +#define __OMAP_MCBSP_PRIV_H__ + +#include <linux/platform_data/asoc-ti-mcbsp.h> #ifdef CONFIG_ARCH_OMAP1 #define mcbsp_omap1() 1 @@ -30,8 +17,6 @@ #define mcbsp_omap1() 0 #endif -#include <sound/dmaengine_pcm.h> - /* McBSP register numbers. Register address offset = num * reg_step */ enum { /* Common registers */ @@ -85,15 +70,6 @@ enum { OMAP_MCBSP_REG_SSELCR, }; -/* OMAP3 sidetone control registers */ -#define OMAP_ST_REG_REV 0x00 -#define OMAP_ST_REG_SYSCONFIG 0x10 -#define OMAP_ST_REG_IRQSTATUS 0x18 -#define OMAP_ST_REG_IRQENABLE 0x1C -#define OMAP_ST_REG_SGAINCR 0x24 -#define OMAP_ST_REG_SFIRCR 0x28 -#define OMAP_ST_REG_SSELCR 0x2C - /************************** McBSP SPCR1 bit definitions ***********************/ #define RRST BIT(0) #define RRDY BIT(1) @@ -202,24 +178,6 @@ enum { #define SIDLEMODE(value) (((value) & 0x3) << 3) #define CLOCKACTIVITY(value) (((value) & 0x3) << 8) -/********************** McBSP SSELCR bit definitions ***********************/ -#define SIDETONEEN BIT(10) - -/********************** McBSP Sidetone SYSCONFIG bit definitions ***********/ -#define ST_AUTOIDLE BIT(0) - -/********************** McBSP Sidetone SGAINCR bit definitions *************/ -#define ST_CH0GAIN(value) ((value) & 0xffff) /* Bits 0:15 */ -#define ST_CH1GAIN(value) (((value) & 0xffff) << 16) /* Bits 16:31 */ - -/********************** McBSP Sidetone SFIRCR bit definitions **************/ -#define ST_FIRCOEFF(value) ((value) & 0xffff) /* Bits 0:15 */ - -/********************** McBSP Sidetone SSELCR bit definitions **************/ -#define ST_SIDETONEEN BIT(0) -#define ST_COEFFWREN BIT(1) -#define ST_COEFFWRDONE BIT(2) - /********************** McBSP DMA operating modes **************************/ #define MCBSP_DMA_MODE_ELEMENT 0 #define MCBSP_DMA_MODE_THRESHOLD 1 @@ -278,16 +236,7 @@ struct omap_mcbsp_reg_cfg { u16 rccr; }; -struct omap_mcbsp_st_data { - void __iomem *io_base_st; - struct clk *mcbsp_iclk; - bool running; - bool enabled; - s16 taps[128]; /* Sidetone filter coefficients */ - int nr_taps; /* Number of filter coefficients in use */ - s16 ch0gain; - s16 ch1gain; -}; +struct omap_mcbsp_st_data; struct omap_mcbsp { struct device *dev; @@ -330,29 +279,46 @@ struct omap_mcbsp { struct pm_qos_request pm_qos_req; }; -void omap_mcbsp_config(struct omap_mcbsp *mcbsp, - const struct omap_mcbsp_reg_cfg *config); -void omap_mcbsp_set_tx_threshold(struct omap_mcbsp *mcbsp, u16 threshold); -void omap_mcbsp_set_rx_threshold(struct omap_mcbsp *mcbsp, u16 threshold); -u16 omap_mcbsp_get_tx_delay(struct omap_mcbsp *mcbsp); -u16 omap_mcbsp_get_rx_delay(struct omap_mcbsp *mcbsp); -int omap_mcbsp_get_dma_op_mode(struct omap_mcbsp *mcbsp); -int omap_mcbsp_request(struct omap_mcbsp *mcbsp); -void omap_mcbsp_free(struct omap_mcbsp *mcbsp); -void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int tx, int rx); -void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int tx, int rx); - -/* McBSP functional clock source changing function */ -int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id); +static inline void omap_mcbsp_write(struct omap_mcbsp *mcbsp, u16 reg, u32 val) +{ + void __iomem *addr = mcbsp->io_base + reg * mcbsp->pdata->reg_step; + + if (mcbsp->pdata->reg_size == 2) { + ((u16 *)mcbsp->reg_cache)[reg] = (u16)val; + writew_relaxed((u16)val, addr); + } else { + ((u32 *)mcbsp->reg_cache)[reg] = val; + writel_relaxed(val, addr); + } +} + +static inline int omap_mcbsp_read(struct omap_mcbsp *mcbsp, u16 reg, + bool from_cache) +{ + void __iomem *addr = mcbsp->io_base + reg * mcbsp->pdata->reg_step; + + if (mcbsp->pdata->reg_size == 2) { + return !from_cache ? readw_relaxed(addr) : + ((u16 *)mcbsp->reg_cache)[reg]; + } else { + return !from_cache ? readl_relaxed(addr) : + ((u32 *)mcbsp->reg_cache)[reg]; + } +} + +#define MCBSP_READ(mcbsp, reg) \ + omap_mcbsp_read(mcbsp, OMAP_MCBSP_REG_##reg, 0) +#define MCBSP_WRITE(mcbsp, reg, val) \ + omap_mcbsp_write(mcbsp, OMAP_MCBSP_REG_##reg, val) +#define MCBSP_READ_CACHE(mcbsp, reg) \ + omap_mcbsp_read(mcbsp, OMAP_MCBSP_REG_##reg, 1) + /* Sidetone specific API */ -int omap_st_set_chgain(struct omap_mcbsp *mcbsp, int channel, s16 chgain); -int omap_st_get_chgain(struct omap_mcbsp *mcbsp, int channel, s16 *chgain); -int omap_st_enable(struct omap_mcbsp *mcbsp); -int omap_st_disable(struct omap_mcbsp *mcbsp); -int omap_st_is_enabled(struct omap_mcbsp *mcbsp); +int omap_mcbsp_st_init(struct platform_device *pdev); +void omap_mcbsp_st_cleanup(struct platform_device *pdev); -int omap_mcbsp_init(struct platform_device *pdev); -void omap_mcbsp_cleanup(struct omap_mcbsp *mcbsp); +int omap_mcbsp_st_start(struct omap_mcbsp *mcbsp); +int omap_mcbsp_st_stop(struct omap_mcbsp *mcbsp); -#endif /* __ASOC_MCBSP_H */ +#endif /* __OMAP_MCBSP_PRIV_H__ */ diff --git a/sound/soc/ti/omap-mcbsp-st.c b/sound/soc/ti/omap-mcbsp-st.c new file mode 100644 index 000000000000..1a3fe854e856 --- /dev/null +++ b/sound/soc/ti/omap-mcbsp-st.c @@ -0,0 +1,516 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * McBSP Sidetone support + * + * Copyright (C) 2004 Nokia Corporation + * Author: Samuel Ortiz <samuel.ortiz@nokia.com> + * + * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com> + * Peter Ujfalusi <peter.ujfalusi@ti.com> + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/device.h> +#include <linux/platform_device.h> +#include <linux/interrupt.h> +#include <linux/err.h> +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/slab.h> +#include <linux/pm_runtime.h> + +#include "omap-mcbsp.h" +#include "omap-mcbsp-priv.h" + +/* OMAP3 sidetone control registers */ +#define OMAP_ST_REG_REV 0x00 +#define OMAP_ST_REG_SYSCONFIG 0x10 +#define OMAP_ST_REG_IRQSTATUS 0x18 +#define OMAP_ST_REG_IRQENABLE 0x1C +#define OMAP_ST_REG_SGAINCR 0x24 +#define OMAP_ST_REG_SFIRCR 0x28 +#define OMAP_ST_REG_SSELCR 0x2C + +/********************** McBSP SSELCR bit definitions ***********************/ +#define SIDETONEEN BIT(10) + +/********************** McBSP Sidetone SYSCONFIG bit definitions ***********/ +#define ST_AUTOIDLE BIT(0) + +/********************** McBSP Sidetone SGAINCR bit definitions *************/ +#define ST_CH0GAIN(value) ((value) & 0xffff) /* Bits 0:15 */ +#define ST_CH1GAIN(value) (((value) & 0xffff) << 16) /* Bits 16:31 */ + +/********************** McBSP Sidetone SFIRCR bit definitions **************/ +#define ST_FIRCOEFF(value) ((value) & 0xffff) /* Bits 0:15 */ + +/********************** McBSP Sidetone SSELCR bit definitions **************/ +#define ST_SIDETONEEN BIT(0) +#define ST_COEFFWREN BIT(1) +#define ST_COEFFWRDONE BIT(2) + +struct omap_mcbsp_st_data { + void __iomem *io_base_st; + struct clk *mcbsp_iclk; + bool running; + bool enabled; + s16 taps[128]; /* Sidetone filter coefficients */ + int nr_taps; /* Number of filter coefficients in use */ + s16 ch0gain; + s16 ch1gain; +}; + +static void omap_mcbsp_st_write(struct omap_mcbsp *mcbsp, u16 reg, u32 val) +{ + writel_relaxed(val, mcbsp->st_data->io_base_st + reg); +} + +static int omap_mcbsp_st_read(struct omap_mcbsp *mcbsp, u16 reg) +{ + return readl_relaxed(mcbsp->st_data->io_base_st + reg); +} + +#define MCBSP_ST_READ(mcbsp, reg) omap_mcbsp_st_read(mcbsp, OMAP_ST_REG_##reg) +#define MCBSP_ST_WRITE(mcbsp, reg, val) \ + omap_mcbsp_st_write(mcbsp, OMAP_ST_REG_##reg, val) + +static void omap_mcbsp_st_on(struct omap_mcbsp *mcbsp) +{ + unsigned int w; + + if (mcbsp->pdata->force_ick_on) + mcbsp->pdata->force_ick_on(mcbsp->st_data->mcbsp_iclk, true); + + /* Disable Sidetone clock auto-gating for normal operation */ + w = MCBSP_ST_READ(mcbsp, SYSCONFIG); + MCBSP_ST_WRITE(mcbsp, SYSCONFIG, w & ~(ST_AUTOIDLE)); + + /* Enable McBSP Sidetone */ + w = MCBSP_READ(mcbsp, SSELCR); + MCBSP_WRITE(mcbsp, SSELCR, w | SIDETONEEN); + + /* Enable Sidetone from Sidetone Core */ + w = MCBSP_ST_READ(mcbsp, SSELCR); + MCBSP_ST_WRITE(mcbsp, SSELCR, w | ST_SIDETONEEN); +} + +static void omap_mcbsp_st_off(struct omap_mcbsp *mcbsp) +{ + unsigned int w; + + w = MCBSP_ST_READ(mcbsp, SSELCR); + MCBSP_ST_WRITE(mcbsp, SSELCR, w & ~(ST_SIDETONEEN)); + + w = MCBSP_READ(mcbsp, SSELCR); + MCBSP_WRITE(mcbsp, SSELCR, w & ~(SIDETONEEN)); + + /* Enable Sidetone clock auto-gating to reduce power consumption */ + w = MCBSP_ST_READ(mcbsp, SYSCONFIG); + MCBSP_ST_WRITE(mcbsp, SYSCONFIG, w | ST_AUTOIDLE); + + if (mcbsp->pdata->force_ick_on) + mcbsp->pdata->force_ick_on(mcbsp->st_data->mcbsp_iclk, false); +} + +static void omap_mcbsp_st_fir_write(struct omap_mcbsp *mcbsp, s16 *fir) +{ + u16 val, i; + + val = MCBSP_ST_READ(mcbsp, SSELCR); + + if (val & ST_COEFFWREN) + MCBSP_ST_WRITE(mcbsp, SSELCR, val & ~(ST_COEFFWREN)); + + MCBSP_ST_WRITE(mcbsp, SSELCR, val | ST_COEFFWREN); + + for (i = 0; i < 128; i++) + MCBSP_ST_WRITE(mcbsp, SFIRCR, fir[i]); + + i = 0; + + val = MCBSP_ST_READ(mcbsp, SSELCR); + while (!(val & ST_COEFFWRDONE) && (++i < 1000)) + val = MCBSP_ST_READ(mcbsp, SSELCR); + + MCBSP_ST_WRITE(mcbsp, SSELCR, val & ~(ST_COEFFWREN)); + + if (i == 1000) + dev_err(mcbsp->dev, "McBSP FIR load error!\n"); +} + +static void omap_mcbsp_st_chgain(struct omap_mcbsp *mcbsp) +{ + u16 w; + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + + w = MCBSP_ST_READ(mcbsp, SSELCR); + + MCBSP_ST_WRITE(mcbsp, SGAINCR, ST_CH0GAIN(st_data->ch0gain) | + ST_CH1GAIN(st_data->ch1gain)); +} + +static int omap_mcbsp_st_set_chgain(struct omap_mcbsp *mcbsp, int channel, + s16 chgain) +{ + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + int ret = 0; + + if (!st_data) + return -ENOENT; + + spin_lock_irq(&mcbsp->lock); + if (channel == 0) + st_data->ch0gain = chgain; + else if (channel == 1) + st_data->ch1gain = chgain; + else + ret = -EINVAL; + + if (st_data->enabled) + omap_mcbsp_st_chgain(mcbsp); + spin_unlock_irq(&mcbsp->lock); + + return ret; +} + +static int omap_mcbsp_st_get_chgain(struct omap_mcbsp *mcbsp, int channel, + s16 *chgain) +{ + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + int ret = 0; + + if (!st_data) + return -ENOENT; + + spin_lock_irq(&mcbsp->lock); + if (channel == 0) + *chgain = st_data->ch0gain; + else if (channel == 1) + *chgain = st_data->ch1gain; + else + ret = -EINVAL; + spin_unlock_irq(&mcbsp->lock); + + return ret; +} + +static int omap_mcbsp_st_enable(struct omap_mcbsp *mcbsp) +{ + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + + if (!st_data) + return -ENODEV; + + spin_lock_irq(&mcbsp->lock); + st_data->enabled = 1; + omap_mcbsp_st_start(mcbsp); + spin_unlock_irq(&mcbsp->lock); + + return 0; +} + +static int omap_mcbsp_st_disable(struct omap_mcbsp *mcbsp) +{ + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + int ret = 0; + + if (!st_data) + return -ENODEV; + + spin_lock_irq(&mcbsp->lock); + omap_mcbsp_st_stop(mcbsp); + st_data->enabled = 0; + spin_unlock_irq(&mcbsp->lock); + + return ret; +} + +static int omap_mcbsp_st_is_enabled(struct omap_mcbsp *mcbsp) +{ + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + + if (!st_data) + return -ENODEV; + + return st_data->enabled; +} + +static ssize_t st_taps_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + ssize_t status = 0; + int i; + + spin_lock_irq(&mcbsp->lock); + for (i = 0; i < st_data->nr_taps; i++) + status += sprintf(&buf[status], (i ? ", %d" : "%d"), + st_data->taps[i]); + if (i) + status += sprintf(&buf[status], "\n"); + spin_unlock_irq(&mcbsp->lock); + + return status; +} + +static ssize_t st_taps_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t size) +{ + struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + int val, tmp, status, i = 0; + + spin_lock_irq(&mcbsp->lock); + memset(st_data->taps, 0, sizeof(st_data->taps)); + st_data->nr_taps = 0; + + do { + status = sscanf(buf, "%d%n", &val, &tmp); + if (status < 0 || status == 0) { + size = -EINVAL; + goto out; + } + if (val < -32768 || val > 32767) { + size = -EINVAL; + goto out; + } + st_data->taps[i++] = val; + buf += tmp; + if (*buf != ',') + break; + buf++; + } while (1); + + st_data->nr_taps = i; + +out: + spin_unlock_irq(&mcbsp->lock); + + return size; +} + +static DEVICE_ATTR_RW(st_taps); + +static const struct attribute *sidetone_attrs[] = { + &dev_attr_st_taps.attr, + NULL, +}; + +static const struct attribute_group sidetone_attr_group = { + .attrs = (struct attribute **)sidetone_attrs, +}; + +int omap_mcbsp_st_start(struct omap_mcbsp *mcbsp) +{ + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + + if (st_data->enabled && !st_data->running) { + omap_mcbsp_st_fir_write(mcbsp, st_data->taps); + omap_mcbsp_st_chgain(mcbsp); + + if (!mcbsp->free) { + omap_mcbsp_st_on(mcbsp); + st_data->running = 1; + } + } + + return 0; +} + +int omap_mcbsp_st_stop(struct omap_mcbsp *mcbsp) +{ + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + + if (st_data->running) { + if (!mcbsp->free) { + omap_mcbsp_st_off(mcbsp); + st_data->running = 0; + } + } + + return 0; +} + +int omap_mcbsp_st_init(struct platform_device *pdev) +{ + struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); + struct omap_mcbsp_st_data *st_data; + struct resource *res; + int ret; + + res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "sidetone"); + if (!res) + return 0; + + st_data = devm_kzalloc(mcbsp->dev, sizeof(*mcbsp->st_data), GFP_KERNEL); + if (!st_data) + return -ENOMEM; + + st_data->mcbsp_iclk = clk_get(mcbsp->dev, "ick"); + if (IS_ERR(st_data->mcbsp_iclk)) { + dev_warn(mcbsp->dev, + "Failed to get ick, sidetone might be broken\n"); + st_data->mcbsp_iclk = NULL; + } + + st_data->io_base_st = devm_ioremap(mcbsp->dev, res->start, + resource_size(res)); + if (!st_data->io_base_st) + return -ENOMEM; + + ret = sysfs_create_group(&mcbsp->dev->kobj, &sidetone_attr_group); + if (ret) + return ret; + + mcbsp->st_data = st_data; + + return 0; +} + +void omap_mcbsp_st_cleanup(struct platform_device *pdev) +{ + struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); + + if (mcbsp->st_data) { + sysfs_remove_group(&mcbsp->dev->kobj, &sidetone_attr_group); + clk_put(mcbsp->st_data->mcbsp_iclk); + } +} + +static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int max = mc->max; + int min = mc->min; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = min; + uinfo->value.integer.max = max; + return 0; +} + +#define OMAP_MCBSP_ST_CHANNEL_VOLUME(channel) \ +static int \ +omap_mcbsp_set_st_ch##channel##_volume(struct snd_kcontrol *kc, \ + struct snd_ctl_elem_value *uc) \ +{ \ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kc); \ + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); \ + struct soc_mixer_control *mc = \ + (struct soc_mixer_control *)kc->private_value; \ + int max = mc->max; \ + int min = mc->min; \ + int val = uc->value.integer.value[0]; \ + \ + if (val < min || val > max) \ + return -EINVAL; \ + \ + /* OMAP McBSP implementation uses index values 0..4 */ \ + return omap_mcbsp_st_set_chgain(mcbsp, channel, val); \ +} \ + \ +static int \ +omap_mcbsp_get_st_ch##channel##_volume(struct snd_kcontrol *kc, \ + struct snd_ctl_elem_value *uc) \ +{ \ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kc); \ + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); \ + s16 chgain; \ + \ + if (omap_mcbsp_st_get_chgain(mcbsp, channel, &chgain)) \ + return -EAGAIN; \ + \ + uc->value.integer.value[0] = chgain; \ + return 0; \ +} + +OMAP_MCBSP_ST_CHANNEL_VOLUME(0) +OMAP_MCBSP_ST_CHANNEL_VOLUME(1) + +static int omap_mcbsp_st_put_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); + u8 value = ucontrol->value.integer.value[0]; + + if (value == omap_mcbsp_st_is_enabled(mcbsp)) + return 0; + + if (value) + omap_mcbsp_st_enable(mcbsp); + else + omap_mcbsp_st_disable(mcbsp); + + return 1; +} + +static int omap_mcbsp_st_get_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); + + ucontrol->value.integer.value[0] = omap_mcbsp_st_is_enabled(mcbsp); + return 0; +} + +#define OMAP_MCBSP_SOC_SINGLE_S16_EXT(xname, xmin, xmax, \ + xhandler_get, xhandler_put) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = omap_mcbsp_st_info_volsw, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.min = xmin, .max = xmax} } + +#define OMAP_MCBSP_ST_CONTROLS(port) \ +static const struct snd_kcontrol_new omap_mcbsp##port##_st_controls[] = { \ +SOC_SINGLE_EXT("McBSP" #port " Sidetone Switch", 1, 0, 1, 0, \ + omap_mcbsp_st_get_mode, omap_mcbsp_st_put_mode), \ +OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP" #port " Sidetone Channel 0 Volume", \ + -32768, 32767, \ + omap_mcbsp_get_st_ch0_volume, \ + omap_mcbsp_set_st_ch0_volume), \ +OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP" #port " Sidetone Channel 1 Volume", \ + -32768, 32767, \ + omap_mcbsp_get_st_ch1_volume, \ + omap_mcbsp_set_st_ch1_volume), \ +} + +OMAP_MCBSP_ST_CONTROLS(2); +OMAP_MCBSP_ST_CONTROLS(3); + +int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd, int port_id) +{ + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); + + if (!mcbsp->st_data) { + dev_warn(mcbsp->dev, "No sidetone data for port\n"); + return 0; + } + + switch (port_id) { + case 2: /* McBSP 2 */ + return snd_soc_add_dai_controls(cpu_dai, + omap_mcbsp2_st_controls, + ARRAY_SIZE(omap_mcbsp2_st_controls)); + case 3: /* McBSP 3 */ + return snd_soc_add_dai_controls(cpu_dai, + omap_mcbsp3_st_controls, + ARRAY_SIZE(omap_mcbsp3_st_controls)); + default: + dev_err(mcbsp->dev, "Port %d not supported\n", port_id); + break; + } + + return -EINVAL; +} +EXPORT_SYMBOL_GPL(omap_mcbsp_st_add_controls); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index d0ebb6b9bfac..a395598f1f20 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -35,21 +35,12 @@ #include <sound/soc.h> #include <sound/dmaengine_pcm.h> -#include <linux/platform_data/asoc-ti-mcbsp.h> -#include "mcbsp.h" +#include "omap-mcbsp-priv.h" #include "omap-mcbsp.h" #include "sdma-pcm.h" #define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_96000) -#define OMAP_MCBSP_SOC_SINGLE_S16_EXT(xname, xmin, xmax, \ - xhandler_get, xhandler_put) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = omap_mcbsp_st_info_volsw, \ - .get = xhandler_get, .put = xhandler_put, \ - .private_value = (unsigned long) &(struct soc_mixer_control) \ - {.min = xmin, .max = xmax} } - enum { OMAP_MCBSP_WORD_8 = 0, OMAP_MCBSP_WORD_12, @@ -59,6 +50,699 @@ enum { OMAP_MCBSP_WORD_32, }; +static void omap_mcbsp_dump_reg(struct omap_mcbsp *mcbsp) +{ + dev_dbg(mcbsp->dev, "**** McBSP%d regs ****\n", mcbsp->id); + dev_dbg(mcbsp->dev, "DRR2: 0x%04x\n", MCBSP_READ(mcbsp, DRR2)); + dev_dbg(mcbsp->dev, "DRR1: 0x%04x\n", MCBSP_READ(mcbsp, DRR1)); + dev_dbg(mcbsp->dev, "DXR2: 0x%04x\n", MCBSP_READ(mcbsp, DXR2)); + dev_dbg(mcbsp->dev, "DXR1: 0x%04x\n", MCBSP_READ(mcbsp, DXR1)); + dev_dbg(mcbsp->dev, "SPCR2: 0x%04x\n", MCBSP_READ(mcbsp, SPCR2)); + dev_dbg(mcbsp->dev, "SPCR1: 0x%04x\n", MCBSP_READ(mcbsp, SPCR1)); + dev_dbg(mcbsp->dev, "RCR2: 0x%04x\n", MCBSP_READ(mcbsp, RCR2)); + dev_dbg(mcbsp->dev, "RCR1: 0x%04x\n", MCBSP_READ(mcbsp, RCR1)); + dev_dbg(mcbsp->dev, "XCR2: 0x%04x\n", MCBSP_READ(mcbsp, XCR2)); + dev_dbg(mcbsp->dev, "XCR1: 0x%04x\n", MCBSP_READ(mcbsp, XCR1)); + dev_dbg(mcbsp->dev, "SRGR2: 0x%04x\n", MCBSP_READ(mcbsp, SRGR2)); + dev_dbg(mcbsp->dev, "SRGR1: 0x%04x\n", MCBSP_READ(mcbsp, SRGR1)); + dev_dbg(mcbsp->dev, "PCR0: 0x%04x\n", MCBSP_READ(mcbsp, PCR0)); + dev_dbg(mcbsp->dev, "***********************\n"); +} + +static int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) +{ + struct clk *fck_src; + const char *src; + int r; + + if (fck_src_id == MCBSP_CLKS_PAD_SRC) + src = "pad_fck"; + else if (fck_src_id == MCBSP_CLKS_PRCM_SRC) + src = "prcm_fck"; + else + return -EINVAL; + + fck_src = clk_get(mcbsp->dev, src); + if (IS_ERR(fck_src)) { + dev_err(mcbsp->dev, "CLKS: could not clk_get() %s\n", src); + return -EINVAL; + } + + pm_runtime_put_sync(mcbsp->dev); + + r = clk_set_parent(mcbsp->fclk, fck_src); + if (r) { + dev_err(mcbsp->dev, "CLKS: could not clk_set_parent() to %s\n", + src); + clk_put(fck_src); + return r; + } + + pm_runtime_get_sync(mcbsp->dev); + + clk_put(fck_src); + + return 0; +} + +static irqreturn_t omap_mcbsp_irq_handler(int irq, void *data) +{ + struct omap_mcbsp *mcbsp = data; + u16 irqst; + + irqst = MCBSP_READ(mcbsp, IRQST); + dev_dbg(mcbsp->dev, "IRQ callback : 0x%x\n", irqst); + + if (irqst & RSYNCERREN) + dev_err(mcbsp->dev, "RX Frame Sync Error!\n"); + if (irqst & RFSREN) + dev_dbg(mcbsp->dev, "RX Frame Sync\n"); + if (irqst & REOFEN) + dev_dbg(mcbsp->dev, "RX End Of Frame\n"); + if (irqst & RRDYEN) + dev_dbg(mcbsp->dev, "RX Buffer Threshold Reached\n"); + if (irqst & RUNDFLEN) + dev_err(mcbsp->dev, "RX Buffer Underflow!\n"); + if (irqst & ROVFLEN) + dev_err(mcbsp->dev, "RX Buffer Overflow!\n"); + + if (irqst & XSYNCERREN) + dev_err(mcbsp->dev, "TX Frame Sync Error!\n"); + if (irqst & XFSXEN) + dev_dbg(mcbsp->dev, "TX Frame Sync\n"); + if (irqst & XEOFEN) + dev_dbg(mcbsp->dev, "TX End Of Frame\n"); + if (irqst & XRDYEN) + dev_dbg(mcbsp->dev, "TX Buffer threshold Reached\n"); + if (irqst & XUNDFLEN) + dev_err(mcbsp->dev, "TX Buffer Underflow!\n"); + if (irqst & XOVFLEN) + dev_err(mcbsp->dev, "TX Buffer Overflow!\n"); + if (irqst & XEMPTYEOFEN) + dev_dbg(mcbsp->dev, "TX Buffer empty at end of frame\n"); + + MCBSP_WRITE(mcbsp, IRQST, irqst); + + return IRQ_HANDLED; +} + +static irqreturn_t omap_mcbsp_tx_irq_handler(int irq, void *data) +{ + struct omap_mcbsp *mcbsp = data; + u16 irqst_spcr2; + + irqst_spcr2 = MCBSP_READ(mcbsp, SPCR2); + dev_dbg(mcbsp->dev, "TX IRQ callback : 0x%x\n", irqst_spcr2); + + if (irqst_spcr2 & XSYNC_ERR) { + dev_err(mcbsp->dev, "TX Frame Sync Error! : 0x%x\n", + irqst_spcr2); + /* Writing zero to XSYNC_ERR clears the IRQ */ + MCBSP_WRITE(mcbsp, SPCR2, MCBSP_READ_CACHE(mcbsp, SPCR2)); + } + + return IRQ_HANDLED; +} + +static irqreturn_t omap_mcbsp_rx_irq_handler(int irq, void *data) +{ + struct omap_mcbsp *mcbsp = data; + u16 irqst_spcr1; + + irqst_spcr1 = MCBSP_READ(mcbsp, SPCR1); + dev_dbg(mcbsp->dev, "RX IRQ callback : 0x%x\n", irqst_spcr1); + + if (irqst_spcr1 & RSYNC_ERR) { + dev_err(mcbsp->dev, "RX Frame Sync Error! : 0x%x\n", + irqst_spcr1); + /* Writing zero to RSYNC_ERR clears the IRQ */ + MCBSP_WRITE(mcbsp, SPCR1, MCBSP_READ_CACHE(mcbsp, SPCR1)); + } + + return IRQ_HANDLED; +} + +/* + * omap_mcbsp_config simply write a config to the + * appropriate McBSP. + * You either call this function or set the McBSP registers + * by yourself before calling omap_mcbsp_start(). + */ +static void omap_mcbsp_config(struct omap_mcbsp *mcbsp, + const struct omap_mcbsp_reg_cfg *config) +{ + dev_dbg(mcbsp->dev, "Configuring McBSP%d phys_base: 0x%08lx\n", + mcbsp->id, mcbsp->phys_base); + + /* We write the given config */ + MCBSP_WRITE(mcbsp, SPCR2, config->spcr2); + MCBSP_WRITE(mcbsp, SPCR1, config->spcr1); + MCBSP_WRITE(mcbsp, RCR2, config->rcr2); + MCBSP_WRITE(mcbsp, RCR1, config->rcr1); + MCBSP_WRITE(mcbsp, XCR2, config->xcr2); + MCBSP_WRITE(mcbsp, XCR1, config->xcr1); + MCBSP_WRITE(mcbsp, SRGR2, config->srgr2); + MCBSP_WRITE(mcbsp, SRGR1, config->srgr1); + MCBSP_WRITE(mcbsp, MCR2, config->mcr2); + MCBSP_WRITE(mcbsp, MCR1, config->mcr1); + MCBSP_WRITE(mcbsp, PCR0, config->pcr0); + if (mcbsp->pdata->has_ccr) { + MCBSP_WRITE(mcbsp, XCCR, config->xccr); + MCBSP_WRITE(mcbsp, RCCR, config->rccr); + } + /* Enable wakeup behavior */ + if (mcbsp->pdata->has_wakeup) + MCBSP_WRITE(mcbsp, WAKEUPEN, XRDYEN | RRDYEN); + + /* Enable TX/RX sync error interrupts by default */ + if (mcbsp->irq) + MCBSP_WRITE(mcbsp, IRQEN, RSYNCERREN | XSYNCERREN | + RUNDFLEN | ROVFLEN | XUNDFLEN | XOVFLEN); +} + +/** + * omap_mcbsp_dma_reg_params - returns the address of mcbsp data register + * @mcbsp: omap_mcbsp struct for the McBSP instance + * @stream: Stream direction (playback/capture) + * + * Returns the address of mcbsp data transmit register or data receive register + * to be used by DMA for transferring/receiving data + */ +static int omap_mcbsp_dma_reg_params(struct omap_mcbsp *mcbsp, + unsigned int stream) +{ + int data_reg; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (mcbsp->pdata->reg_size == 2) + data_reg = OMAP_MCBSP_REG_DXR1; + else + data_reg = OMAP_MCBSP_REG_DXR; + } else { + if (mcbsp->pdata->reg_size == 2) + data_reg = OMAP_MCBSP_REG_DRR1; + else + data_reg = OMAP_MCBSP_REG_DRR; + } + + return mcbsp->phys_dma_base + data_reg * mcbsp->pdata->reg_step; +} + +/* + * omap_mcbsp_set_rx_threshold configures the transmit threshold in words. + * The threshold parameter is 1 based, and it is converted (threshold - 1) + * for the THRSH2 register. + */ +static void omap_mcbsp_set_tx_threshold(struct omap_mcbsp *mcbsp, u16 threshold) +{ + if (threshold && threshold <= mcbsp->max_tx_thres) + MCBSP_WRITE(mcbsp, THRSH2, threshold - 1); +} + +/* + * omap_mcbsp_set_rx_threshold configures the receive threshold in words. + * The threshold parameter is 1 based, and it is converted (threshold - 1) + * for the THRSH1 register. + */ +static void omap_mcbsp_set_rx_threshold(struct omap_mcbsp *mcbsp, u16 threshold) +{ + if (threshold && threshold <= mcbsp->max_rx_thres) + MCBSP_WRITE(mcbsp, THRSH1, threshold - 1); +} + +/* + * omap_mcbsp_get_tx_delay returns the number of used slots in the McBSP FIFO + */ +static u16 omap_mcbsp_get_tx_delay(struct omap_mcbsp *mcbsp) +{ + u16 buffstat; + + /* Returns the number of free locations in the buffer */ + buffstat = MCBSP_READ(mcbsp, XBUFFSTAT); + + /* Number of slots are different in McBSP ports */ + return mcbsp->pdata->buffer_size - buffstat; +} + +/* + * omap_mcbsp_get_rx_delay returns the number of free slots in the McBSP FIFO + * to reach the threshold value (when the DMA will be triggered to read it) + */ +static u16 omap_mcbsp_get_rx_delay(struct omap_mcbsp *mcbsp) +{ + u16 buffstat, threshold; + + /* Returns the number of used locations in the buffer */ + buffstat = MCBSP_READ(mcbsp, RBUFFSTAT); + /* RX threshold */ + threshold = MCBSP_READ(mcbsp, THRSH1); + + /* Return the number of location till we reach the threshold limit */ + if (threshold <= buffstat) + return 0; + else + return threshold - buffstat; +} + +static int omap_mcbsp_request(struct omap_mcbsp *mcbsp) +{ + void *reg_cache; + int err; + + reg_cache = kzalloc(mcbsp->reg_cache_size, GFP_KERNEL); + if (!reg_cache) + return -ENOMEM; + + spin_lock(&mcbsp->lock); + if (!mcbsp->free) { + dev_err(mcbsp->dev, "McBSP%d is currently in use\n", mcbsp->id); + err = -EBUSY; + goto err_kfree; + } + + mcbsp->free = false; + mcbsp->reg_cache = reg_cache; + spin_unlock(&mcbsp->lock); + + if(mcbsp->pdata->ops && mcbsp->pdata->ops->request) + mcbsp->pdata->ops->request(mcbsp->id - 1); + + /* + * Make sure that transmitter, receiver and sample-rate generator are + * not running before activating IRQs. + */ + MCBSP_WRITE(mcbsp, SPCR1, 0); + MCBSP_WRITE(mcbsp, SPCR2, 0); + + if (mcbsp->irq) { + err = request_irq(mcbsp->irq, omap_mcbsp_irq_handler, 0, + "McBSP", (void *)mcbsp); + if (err != 0) { + dev_err(mcbsp->dev, "Unable to request IRQ\n"); + goto err_clk_disable; + } + } else { + err = request_irq(mcbsp->tx_irq, omap_mcbsp_tx_irq_handler, 0, + "McBSP TX", (void *)mcbsp); + if (err != 0) { + dev_err(mcbsp->dev, "Unable to request TX IRQ\n"); + goto err_clk_disable; + } + + err = request_irq(mcbsp->rx_irq, omap_mcbsp_rx_irq_handler, 0, + "McBSP RX", (void *)mcbsp); + if (err != 0) { + dev_err(mcbsp->dev, "Unable to request RX IRQ\n"); + goto err_free_irq; + } + } + + return 0; +err_free_irq: + free_irq(mcbsp->tx_irq, (void *)mcbsp); +err_clk_disable: + if(mcbsp->pdata->ops && mcbsp->pdata->ops->free) + mcbsp->pdata->ops->free(mcbsp->id - 1); + + /* Disable wakeup behavior */ + if (mcbsp->pdata->has_wakeup) + MCBSP_WRITE(mcbsp, WAKEUPEN, 0); + + spin_lock(&mcbsp->lock); + mcbsp->free = true; + mcbsp->reg_cache = NULL; +err_kfree: + spin_unlock(&mcbsp->lock); + kfree(reg_cache); + + return err; +} + +static void omap_mcbsp_free(struct omap_mcbsp *mcbsp) +{ + void *reg_cache; + + if(mcbsp->pdata->ops && mcbsp->pdata->ops->free) + mcbsp->pdata->ops->free(mcbsp->id - 1); + + /* Disable wakeup behavior */ + if (mcbsp->pdata->has_wakeup) + MCBSP_WRITE(mcbsp, WAKEUPEN, 0); + + /* Disable interrupt requests */ + if (mcbsp->irq) + MCBSP_WRITE(mcbsp, IRQEN, 0); + + if (mcbsp->irq) { + free_irq(mcbsp->irq, (void *)mcbsp); + } else { + free_irq(mcbsp->rx_irq, (void *)mcbsp); + free_irq(mcbsp->tx_irq, (void *)mcbsp); + } + + reg_cache = mcbsp->reg_cache; + + /* + * Select CLKS source from internal source unconditionally before + * marking the McBSP port as free. + * If the external clock source via MCBSP_CLKS pin has been selected the + * system will refuse to enter idle if the CLKS pin source is not reset + * back to internal source. + */ + if (!mcbsp_omap1()) + omap2_mcbsp_set_clks_src(mcbsp, MCBSP_CLKS_PRCM_SRC); + + spin_lock(&mcbsp->lock); + if (mcbsp->free) + dev_err(mcbsp->dev, "McBSP%d was not reserved\n", mcbsp->id); + else + mcbsp->free = true; + mcbsp->reg_cache = NULL; + spin_unlock(&mcbsp->lock); + + kfree(reg_cache); +} + +/* + * Here we start the McBSP, by enabling transmitter, receiver or both. + * If no transmitter or receiver is active prior calling, then sample-rate + * generator and frame sync are started. + */ +static void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int stream) +{ + int tx = (stream == SNDRV_PCM_STREAM_PLAYBACK); + int rx = !tx; + int enable_srg = 0; + u16 w; + + if (mcbsp->st_data) + omap_mcbsp_st_start(mcbsp); + + /* Only enable SRG, if McBSP is master */ + w = MCBSP_READ_CACHE(mcbsp, PCR0); + if (w & (FSXM | FSRM | CLKXM | CLKRM)) + enable_srg = !((MCBSP_READ_CACHE(mcbsp, SPCR2) | + MCBSP_READ_CACHE(mcbsp, SPCR1)) & 1); + + if (enable_srg) { + /* Start the sample generator */ + w = MCBSP_READ_CACHE(mcbsp, SPCR2); + MCBSP_WRITE(mcbsp, SPCR2, w | (1 << 6)); + } + + /* Enable transmitter and receiver */ + tx &= 1; + w = MCBSP_READ_CACHE(mcbsp, SPCR2); + MCBSP_WRITE(mcbsp, SPCR2, w | tx); + + rx &= 1; + w = MCBSP_READ_CACHE(mcbsp, SPCR1); + MCBSP_WRITE(mcbsp, SPCR1, w | rx); + + /* + * Worst case: CLKSRG*2 = 8000khz: (1/8000) * 2 * 2 usec + * REVISIT: 100us may give enough time for two CLKSRG, however + * due to some unknown PM related, clock gating etc. reason it + * is now at 500us. + */ + udelay(500); + + if (enable_srg) { + /* Start frame sync */ + w = MCBSP_READ_CACHE(mcbsp, SPCR2); + MCBSP_WRITE(mcbsp, SPCR2, w | (1 << 7)); + } + + if (mcbsp->pdata->has_ccr) { + /* Release the transmitter and receiver */ + w = MCBSP_READ_CACHE(mcbsp, XCCR); + w &= ~(tx ? XDISABLE : 0); + MCBSP_WRITE(mcbsp, XCCR, w); + w = MCBSP_READ_CACHE(mcbsp, RCCR); + w &= ~(rx ? RDISABLE : 0); + MCBSP_WRITE(mcbsp, RCCR, w); + } + + /* Dump McBSP Regs */ + omap_mcbsp_dump_reg(mcbsp); +} + +static void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int stream) +{ + int tx = (stream == SNDRV_PCM_STREAM_PLAYBACK); + int rx = !tx; + int idle; + u16 w; + + /* Reset transmitter */ + tx &= 1; + if (mcbsp->pdata->has_ccr) { + w = MCBSP_READ_CACHE(mcbsp, XCCR); + w |= (tx ? XDISABLE : 0); + MCBSP_WRITE(mcbsp, XCCR, w); + } + w = MCBSP_READ_CACHE(mcbsp, SPCR2); + MCBSP_WRITE(mcbsp, SPCR2, w & ~tx); + + /* Reset receiver */ + rx &= 1; + if (mcbsp->pdata->has_ccr) { + w = MCBSP_READ_CACHE(mcbsp, RCCR); + w |= (rx ? RDISABLE : 0); + MCBSP_WRITE(mcbsp, RCCR, w); + } + w = MCBSP_READ_CACHE(mcbsp, SPCR1); + MCBSP_WRITE(mcbsp, SPCR1, w & ~rx); + + idle = !((MCBSP_READ_CACHE(mcbsp, SPCR2) | + MCBSP_READ_CACHE(mcbsp, SPCR1)) & 1); + + if (idle) { + /* Reset the sample rate generator */ + w = MCBSP_READ_CACHE(mcbsp, SPCR2); + MCBSP_WRITE(mcbsp, SPCR2, w & ~(1 << 6)); + } + + if (mcbsp->st_data) + omap_mcbsp_st_stop(mcbsp); +} + +#define max_thres(m) (mcbsp->pdata->buffer_size) +#define valid_threshold(m, val) ((val) <= max_thres(m)) +#define THRESHOLD_PROP_BUILDER(prop) \ +static ssize_t prop##_show(struct device *dev, \ + struct device_attribute *attr, char *buf) \ +{ \ + struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); \ + \ + return sprintf(buf, "%u\n", mcbsp->prop); \ +} \ + \ +static ssize_t prop##_store(struct device *dev, \ + struct device_attribute *attr, \ + const char *buf, size_t size) \ +{ \ + struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); \ + unsigned long val; \ + int status; \ + \ + status = kstrtoul(buf, 0, &val); \ + if (status) \ + return status; \ + \ + if (!valid_threshold(mcbsp, val)) \ + return -EDOM; \ + \ + mcbsp->prop = val; \ + return size; \ +} \ + \ +static DEVICE_ATTR(prop, 0644, prop##_show, prop##_store) + +THRESHOLD_PROP_BUILDER(max_tx_thres); +THRESHOLD_PROP_BUILDER(max_rx_thres); + +static const char * const dma_op_modes[] = { + "element", "threshold", +}; + +static ssize_t dma_op_mode_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); + int dma_op_mode, i = 0; + ssize_t len = 0; + const char * const *s; + + dma_op_mode = mcbsp->dma_op_mode; + + for (s = &dma_op_modes[i]; i < ARRAY_SIZE(dma_op_modes); s++, i++) { + if (dma_op_mode == i) + len += sprintf(buf + len, "[%s] ", *s); + else + len += sprintf(buf + len, "%s ", *s); + } + len += sprintf(buf + len, "\n"); + + return len; +} + +static ssize_t dma_op_mode_store(struct device *dev, + struct device_attribute *attr, const char *buf, + size_t size) +{ + struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); + int i; + + i = sysfs_match_string(dma_op_modes, buf); + if (i < 0) + return i; + + spin_lock_irq(&mcbsp->lock); + if (!mcbsp->free) { + size = -EBUSY; + goto unlock; + } + mcbsp->dma_op_mode = i; + +unlock: + spin_unlock_irq(&mcbsp->lock); + + return size; +} + +static DEVICE_ATTR_RW(dma_op_mode); + +static const struct attribute *additional_attrs[] = { + &dev_attr_max_tx_thres.attr, + &dev_attr_max_rx_thres.attr, + &dev_attr_dma_op_mode.attr, + NULL, +}; + +static const struct attribute_group additional_attr_group = { + .attrs = (struct attribute **)additional_attrs, +}; + +/* + * McBSP1 and McBSP3 are directly mapped on 1610 and 1510. + * 730 has only 2 McBSP, and both of them are MPU peripherals. + */ +static int omap_mcbsp_init(struct platform_device *pdev) +{ + struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); + struct resource *res; + int ret = 0; + + spin_lock_init(&mcbsp->lock); + mcbsp->free = true; + + res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); + if (!res) + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + + mcbsp->io_base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(mcbsp->io_base)) + return PTR_ERR(mcbsp->io_base); + + mcbsp->phys_base = res->start; + mcbsp->reg_cache_size = resource_size(res); + + res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dma"); + if (!res) + mcbsp->phys_dma_base = mcbsp->phys_base; + else + mcbsp->phys_dma_base = res->start; + + /* + * OMAP1, 2 uses two interrupt lines: TX, RX + * OMAP2430, OMAP3 SoC have combined IRQ line as well. + * OMAP4 and newer SoC only have the combined IRQ line. + * Use the combined IRQ if available since it gives better debugging + * possibilities. + */ + mcbsp->irq = platform_get_irq_byname(pdev, "common"); + if (mcbsp->irq == -ENXIO) { + mcbsp->tx_irq = platform_get_irq_byname(pdev, "tx"); + + if (mcbsp->tx_irq == -ENXIO) { + mcbsp->irq = platform_get_irq(pdev, 0); + mcbsp->tx_irq = 0; + } else { + mcbsp->rx_irq = platform_get_irq_byname(pdev, "rx"); + mcbsp->irq = 0; + } + } + + if (!pdev->dev.of_node) { + res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx"); + if (!res) { + dev_err(&pdev->dev, "invalid tx DMA channel\n"); + return -ENODEV; + } + mcbsp->dma_req[0] = res->start; + mcbsp->dma_data[0].filter_data = &mcbsp->dma_req[0]; + + res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx"); + if (!res) { + dev_err(&pdev->dev, "invalid rx DMA channel\n"); + return -ENODEV; + } + mcbsp->dma_req[1] = res->start; + mcbsp->dma_data[1].filter_data = &mcbsp->dma_req[1]; + } else { + mcbsp->dma_data[0].filter_data = "tx"; + mcbsp->dma_data[1].filter_data = "rx"; + } + + mcbsp->dma_data[0].addr = omap_mcbsp_dma_reg_params(mcbsp, + SNDRV_PCM_STREAM_PLAYBACK); + mcbsp->dma_data[1].addr = omap_mcbsp_dma_reg_params(mcbsp, + SNDRV_PCM_STREAM_CAPTURE); + + mcbsp->fclk = clk_get(&pdev->dev, "fck"); + if (IS_ERR(mcbsp->fclk)) { + ret = PTR_ERR(mcbsp->fclk); + dev_err(mcbsp->dev, "unable to get fck: %d\n", ret); + return ret; + } + + mcbsp->dma_op_mode = MCBSP_DMA_MODE_ELEMENT; + if (mcbsp->pdata->buffer_size) { + /* + * Initially configure the maximum thresholds to a safe value. + * The McBSP FIFO usage with these values should not go under + * 16 locations. + * If the whole FIFO without safety buffer is used, than there + * is a possibility that the DMA will be not able to push the + * new data on time, causing channel shifts in runtime. + */ + mcbsp->max_tx_thres = max_thres(mcbsp) - 0x10; + mcbsp->max_rx_thres = max_thres(mcbsp) - 0x10; + + ret = sysfs_create_group(&mcbsp->dev->kobj, + &additional_attr_group); + if (ret) { + dev_err(mcbsp->dev, + "Unable to create additional controls\n"); + goto err_thres; + } + } + + ret = omap_mcbsp_st_init(pdev); + if (ret) + goto err_st; + + return 0; + +err_st: + if (mcbsp->pdata->buffer_size) + sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); +err_thres: + clk_put(mcbsp->fclk); + return ret; +} + /* * Stream DMA parameters. DMA request line and port address are set runtime * since they are different between OMAP1 and later OMAPs @@ -71,6 +755,10 @@ static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream, struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); int words; + /* No need to proceed further if McBSP does not have FIFO */ + if (mcbsp->pdata->buffer_size == 0) + return; + /* * Configure McBSP threshold based on either: * packet_size, when the sDMA is in packet mode, or based on the @@ -201,27 +889,26 @@ static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); - int err = 0, play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: mcbsp->active++; - omap_mcbsp_start(mcbsp, play, !play); + omap_mcbsp_start(mcbsp, substream->stream); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - omap_mcbsp_stop(mcbsp, play, !play); + omap_mcbsp_stop(mcbsp, substream->stream); mcbsp->active--; break; default: - err = -EINVAL; + return -EINVAL; } - return err; + return 0; } static snd_pcm_sframes_t omap_mcbsp_dai_delay( @@ -234,6 +921,10 @@ static snd_pcm_sframes_t omap_mcbsp_dai_delay( u16 fifo_use; snd_pcm_sframes_t delay; + /* No need to proceed further if McBSP does not have FIFO */ + if (mcbsp->pdata->buffer_size == 0) + return 0; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) fifo_use = omap_mcbsp_get_tx_delay(mcbsp); else @@ -308,9 +999,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, pkt_size = channels; } - latency = ((((buffer_size - pkt_size) / channels) * 1000) - / (params->rate_num / params->rate_den)); - + latency = (buffer_size - pkt_size) / channels; + latency = latency * USEC_PER_SEC / + (params->rate_num / params->rate_den); mcbsp->latency[substream->stream] = latency; omap_mcbsp_set_threshold(substream, pkt_size); @@ -649,132 +1340,6 @@ static const struct snd_soc_component_driver omap_mcbsp_component = { .name = "omap-mcbsp", }; -static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int max = mc->max; - int min = mc->min; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = min; - uinfo->value.integer.max = max; - return 0; -} - -#define OMAP_MCBSP_ST_CHANNEL_VOLUME(channel) \ -static int \ -omap_mcbsp_set_st_ch##channel##_volume(struct snd_kcontrol *kc, \ - struct snd_ctl_elem_value *uc) \ -{ \ - struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kc); \ - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); \ - struct soc_mixer_control *mc = \ - (struct soc_mixer_control *)kc->private_value; \ - int max = mc->max; \ - int min = mc->min; \ - int val = uc->value.integer.value[0]; \ - \ - if (val < min || val > max) \ - return -EINVAL; \ - \ - /* OMAP McBSP implementation uses index values 0..4 */ \ - return omap_st_set_chgain(mcbsp, channel, val); \ -} \ - \ -static int \ -omap_mcbsp_get_st_ch##channel##_volume(struct snd_kcontrol *kc, \ - struct snd_ctl_elem_value *uc) \ -{ \ - struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kc); \ - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); \ - s16 chgain; \ - \ - if (omap_st_get_chgain(mcbsp, channel, &chgain)) \ - return -EAGAIN; \ - \ - uc->value.integer.value[0] = chgain; \ - return 0; \ -} - -OMAP_MCBSP_ST_CHANNEL_VOLUME(0) -OMAP_MCBSP_ST_CHANNEL_VOLUME(1) - -static int omap_mcbsp_st_put_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); - u8 value = ucontrol->value.integer.value[0]; - - if (value == omap_st_is_enabled(mcbsp)) - return 0; - - if (value) - omap_st_enable(mcbsp); - else - omap_st_disable(mcbsp); - - return 1; -} - -static int omap_mcbsp_st_get_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); - - ucontrol->value.integer.value[0] = omap_st_is_enabled(mcbsp); - return 0; -} - -#define OMAP_MCBSP_ST_CONTROLS(port) \ -static const struct snd_kcontrol_new omap_mcbsp##port##_st_controls[] = { \ -SOC_SINGLE_EXT("McBSP" #port " Sidetone Switch", 1, 0, 1, 0, \ - omap_mcbsp_st_get_mode, omap_mcbsp_st_put_mode), \ -OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP" #port " Sidetone Channel 0 Volume", \ - -32768, 32767, \ - omap_mcbsp_get_st_ch0_volume, \ - omap_mcbsp_set_st_ch0_volume), \ -OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP" #port " Sidetone Channel 1 Volume", \ - -32768, 32767, \ - omap_mcbsp_get_st_ch1_volume, \ - omap_mcbsp_set_st_ch1_volume), \ -} - -OMAP_MCBSP_ST_CONTROLS(2); -OMAP_MCBSP_ST_CONTROLS(3); - -int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd, int port_id) -{ - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); - - if (!mcbsp->st_data) { - dev_warn(mcbsp->dev, "No sidetone data for port\n"); - return 0; - } - - switch (port_id) { - case 2: /* McBSP 2 */ - return snd_soc_add_dai_controls(cpu_dai, - omap_mcbsp2_st_controls, - ARRAY_SIZE(omap_mcbsp2_st_controls)); - case 3: /* McBSP 3 */ - return snd_soc_add_dai_controls(cpu_dai, - omap_mcbsp3_st_controls, - ARRAY_SIZE(omap_mcbsp3_st_controls)); - default: - dev_err(mcbsp->dev, "Port %d not supported\n", port_id); - break; - } - - return -EINVAL; -} -EXPORT_SYMBOL_GPL(omap_mcbsp_st_add_controls); - static struct omap_mcbsp_platform_data omap2420_pdata = { .reg_step = 4, .reg_size = 2, @@ -862,6 +1427,11 @@ static int asoc_mcbsp_probe(struct platform_device *pdev) if (ret) return ret; + if (mcbsp->pdata->reg_size == 2) { + omap_mcbsp_dai.playback.formats = SNDRV_PCM_FMTBIT_S16_LE; + omap_mcbsp_dai.capture.formats = SNDRV_PCM_FMTBIT_S16_LE; + } + ret = devm_snd_soc_register_component(&pdev->dev, &omap_mcbsp_component, &omap_mcbsp_dai, 1); @@ -881,7 +1451,10 @@ static int asoc_mcbsp_remove(struct platform_device *pdev) if (pm_qos_request_active(&mcbsp->pm_qos_req)) pm_qos_remove_request(&mcbsp->pm_qos_req); - omap_mcbsp_cleanup(mcbsp); + if (mcbsp->pdata->buffer_size) + sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); + + omap_mcbsp_st_cleanup(pdev); clk_put(mcbsp->fclk); diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/ti/omap-mcbsp.h index 2e3369c27be3..7911d24898c9 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/ti/omap-mcbsp.h @@ -22,8 +22,10 @@ * */ -#ifndef __OMAP_I2S_H__ -#define __OMAP_I2S_H__ +#ifndef __OMAP_MCBSP_H__ +#define __OMAP_MCBSP_H__ + +#include <sound/dmaengine_pcm.h> /* Source clocks for McBSP sample rate generator */ enum omap_mcbsp_clksrg_clk { @@ -41,4 +43,4 @@ enum omap_mcbsp_div { int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd, int port_id); -#endif +#endif /* __OMAP_MCBSP_H__ */ diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/ti/omap-mcpdm.c index 4c1be36c2207..7d5bdc5a2890 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/ti/omap-mcpdm.c @@ -54,6 +54,8 @@ struct omap_mcpdm { unsigned long phys_base; void __iomem *io_base; int irq; + struct pm_qos_request pm_qos_req; + int latency[2]; struct mutex mutex; @@ -277,6 +279,9 @@ static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); + int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE; + int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; mutex_lock(&mcpdm->mutex); @@ -289,6 +294,14 @@ static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream, } } + if (mcpdm->latency[stream2]) + pm_qos_update_request(&mcpdm->pm_qos_req, + mcpdm->latency[stream2]); + else if (mcpdm->latency[stream1]) + pm_qos_remove_request(&mcpdm->pm_qos_req); + + mcpdm->latency[stream1] = 0; + mutex_unlock(&mcpdm->mutex); } @@ -300,7 +313,7 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, int stream = substream->stream; struct snd_dmaengine_dai_dma_data *dma_data; u32 threshold; - int channels; + int channels, latency; int link_mask = 0; channels = params_channels(params); @@ -344,14 +357,25 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, dma_data->maxburst = (MCPDM_DN_THRES_MAX - threshold) * channels; + latency = threshold; } else { /* If playback is not running assume a stereo stream to come */ if (!mcpdm->config[!stream].link_mask) mcpdm->config[!stream].link_mask = (0x3 << 3); dma_data->maxburst = threshold * channels; + latency = (MCPDM_DN_THRES_MAX - threshold); } + /* + * The DMA must act to a DMA request within latency time (usec) to avoid + * under/overflow + */ + mcpdm->latency[stream] = latency * USEC_PER_SEC / params_rate(params); + + if (!mcpdm->latency[stream]) + mcpdm->latency[stream] = 10; + /* Check if we need to restart McPDM with this stream */ if (mcpdm->config[stream].link_mask && mcpdm->config[stream].link_mask != link_mask) @@ -366,6 +390,20 @@ static int omap_mcpdm_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); + struct pm_qos_request *pm_qos_req = &mcpdm->pm_qos_req; + int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE; + int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; + int latency = mcpdm->latency[stream2]; + + /* Prevent omap hardware from hitting off between FIFO fills */ + if (!latency || mcpdm->latency[stream1] < latency) + latency = mcpdm->latency[stream1]; + + if (pm_qos_request_active(pm_qos_req)) + pm_qos_update_request(pm_qos_req, latency); + else if (latency) + pm_qos_add_request(pm_qos_req, PM_QOS_CPU_DMA_LATENCY, latency); if (!omap_mcpdm_active(mcpdm)) { omap_mcpdm_start(mcpdm); @@ -427,6 +465,9 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai) free_irq(mcpdm->irq, (void *)mcpdm); pm_runtime_disable(mcpdm->dev); + if (pm_qos_request_active(&mcpdm->pm_qos_req)) + pm_qos_remove_request(&mcpdm->pm_qos_req); + return 0; } diff --git a/sound/soc/omap/omap-mcpdm.h b/sound/soc/ti/omap-mcpdm.h index de8cf26595b1..de8cf26595b1 100644 --- a/sound/soc/omap/omap-mcpdm.h +++ b/sound/soc/ti/omap-mcpdm.h diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/ti/omap-twl4030.c index cccc316743fa..cccc316743fa 100644 --- a/sound/soc/omap/omap-twl4030.c +++ b/sound/soc/ti/omap-twl4030.c diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/ti/omap3pandora.c index 4e3de712159c..4e3de712159c 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/ti/omap3pandora.c diff --git a/sound/soc/omap/osk5912.c b/sound/soc/ti/osk5912.c index e4096779ca05..e4096779ca05 100644 --- a/sound/soc/omap/osk5912.c +++ b/sound/soc/ti/osk5912.c diff --git a/sound/soc/omap/rx51.c b/sound/soc/ti/rx51.c index 57448bd5ad77..57448bd5ad77 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/ti/rx51.c diff --git a/sound/soc/omap/sdma-pcm.c b/sound/soc/ti/sdma-pcm.c index 21a9c2499d48..21a9c2499d48 100644 --- a/sound/soc/omap/sdma-pcm.c +++ b/sound/soc/ti/sdma-pcm.c diff --git a/sound/soc/omap/sdma-pcm.h b/sound/soc/ti/sdma-pcm.h index 34a7f90b2587..cb0627c8dd34 100644 --- a/sound/soc/omap/sdma-pcm.h +++ b/sound/soc/ti/sdma-pcm.h @@ -7,7 +7,7 @@ #ifndef __SDMA_PCM_H__ #define __SDMA_PCM_H__ -#if IS_ENABLED(CONFIG_SND_SDMA_SOC) +#if IS_ENABLED(CONFIG_SND_SOC_TI_SDMA_PCM) int sdma_pcm_platform_register(struct device *dev, char *txdmachan, char *rxdmachan); #else @@ -16,6 +16,6 @@ static inline int sdma_pcm_platform_register(struct device *dev, { return -ENODEV; } -#endif /* CONFIG_SND_SDMA_SOC */ +#endif /* CONFIG_SND_SOC_TI_SDMA_PCM */ #endif /* __SDMA_PCM_H__ */ diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c index e2ad00e3cae1..1cfca698ae4b 100644 --- a/sound/soc/txx9/txx9aclc-ac97.c +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -208,13 +208,12 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev) if (err < 0) return err; - return snd_soc_register_component(&pdev->dev, &txx9aclc_ac97_component, + return devm_snd_soc_register_component(&pdev->dev, &txx9aclc_ac97_component, &txx9aclc_ac97_dai, 1); } static int txx9aclc_ac97_dev_remove(struct platform_device *pdev) { - snd_soc_unregister_component(&pdev->dev); snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/xilinx/Kconfig b/sound/soc/xilinx/Kconfig new file mode 100644 index 000000000000..723a583a8d57 --- /dev/null +++ b/sound/soc/xilinx/Kconfig @@ -0,0 +1,8 @@ +config SND_SOC_XILINX_I2S + tristate "Audio support for the Xilinx I2S" + help + Select this option to enable Xilinx I2S Audio. This enables + I2S playback and capture using xilinx soft IP. In transmitter + mode, IP receives audio in AES format, extracts PCM and sends + PCM data. In receiver mode, IP receives PCM audio and + encapsulates PCM in AES format and sends AES data. diff --git a/sound/soc/xilinx/Makefile b/sound/soc/xilinx/Makefile new file mode 100644 index 000000000000..6c1209b9ee75 --- /dev/null +++ b/sound/soc/xilinx/Makefile @@ -0,0 +1,2 @@ +snd-soc-xlnx-i2s-objs := xlnx_i2s.o +obj-$(CONFIG_SND_SOC_XILINX_I2S) += snd-soc-xlnx-i2s.o diff --git a/sound/soc/xilinx/xlnx_i2s.c b/sound/soc/xilinx/xlnx_i2s.c new file mode 100644 index 000000000000..8b353166ad44 --- /dev/null +++ b/sound/soc/xilinx/xlnx_i2s.c @@ -0,0 +1,184 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Xilinx ASoC I2S audio support +// +// Copyright (C) 2018 Xilinx, Inc. +// +// Author: Praveen Vuppala <praveenv@xilinx.com> +// Author: Maruthi Srinivas Bayyavarapu <maruthis@xilinx.com> + +#include <linux/io.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_platform.h> +#include <linux/platform_device.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#define DRV_NAME "xlnx_i2s" + +#define I2S_CORE_CTRL_OFFSET 0x08 +#define I2S_I2STIM_OFFSET 0x20 +#define I2S_CH0_OFFSET 0x30 +#define I2S_I2STIM_VALID_MASK GENMASK(7, 0) + +static int xlnx_i2s_set_sclkout_div(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + void __iomem *base = snd_soc_dai_get_drvdata(cpu_dai); + + if (!div || (div & ~I2S_I2STIM_VALID_MASK)) + return -EINVAL; + + writel(div, base + I2S_I2STIM_OFFSET); + + return 0; +} + +static int xlnx_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *i2s_dai) +{ + u32 reg_off, chan_id; + void __iomem *base = snd_soc_dai_get_drvdata(i2s_dai); + + chan_id = params_channels(params) / 2; + + while (chan_id > 0) { + reg_off = I2S_CH0_OFFSET + ((chan_id - 1) * 4); + writel(chan_id, base + reg_off); + chan_id--; + } + + return 0; +} + +static int xlnx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *i2s_dai) +{ + void __iomem *base = snd_soc_dai_get_drvdata(i2s_dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + writel(1, base + I2S_CORE_CTRL_OFFSET); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + writel(0, base + I2S_CORE_CTRL_OFFSET); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dai_ops xlnx_i2s_dai_ops = { + .trigger = xlnx_i2s_trigger, + .set_clkdiv = xlnx_i2s_set_sclkout_div, + .hw_params = xlnx_i2s_hw_params +}; + +static const struct snd_soc_component_driver xlnx_i2s_component = { + .name = DRV_NAME, +}; + +static const struct of_device_id xlnx_i2s_of_match[] = { + { .compatible = "xlnx,i2s-transmitter-1.0", }, + { .compatible = "xlnx,i2s-receiver-1.0", }, + {}, +}; +MODULE_DEVICE_TABLE(of, xlnx_i2s_of_match); + +static int xlnx_i2s_probe(struct platform_device *pdev) +{ + struct resource *res; + void __iomem *base; + struct snd_soc_dai_driver *dai_drv; + int ret; + u32 ch, format, data_width; + struct device *dev = &pdev->dev; + struct device_node *node = dev->of_node; + + dai_drv = devm_kzalloc(&pdev->dev, sizeof(*dai_drv), GFP_KERNEL); + if (!dai_drv) + return -ENOMEM; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return PTR_ERR(base); + + ret = of_property_read_u32(node, "xlnx,num-channels", &ch); + if (ret < 0) { + dev_err(dev, "cannot get supported channels\n"); + return ret; + } + ch = ch * 2; + + ret = of_property_read_u32(node, "xlnx,dwidth", &data_width); + if (ret < 0) { + dev_err(dev, "cannot get data width\n"); + return ret; + } + switch (data_width) { + case 16: + format = SNDRV_PCM_FMTBIT_S16_LE; + break; + case 24: + format = SNDRV_PCM_FMTBIT_S24_LE; + break; + default: + return -EINVAL; + } + + if (of_device_is_compatible(node, "xlnx,i2s-transmitter-1.0")) { + dai_drv->name = "xlnx_i2s_playback"; + dai_drv->playback.stream_name = "Playback"; + dai_drv->playback.formats = format; + dai_drv->playback.channels_min = ch; + dai_drv->playback.channels_max = ch; + dai_drv->playback.rates = SNDRV_PCM_RATE_8000_192000; + dai_drv->ops = &xlnx_i2s_dai_ops; + } else if (of_device_is_compatible(node, "xlnx,i2s-receiver-1.0")) { + dai_drv->name = "xlnx_i2s_capture"; + dai_drv->capture.stream_name = "Capture"; + dai_drv->capture.formats = format; + dai_drv->capture.channels_min = ch; + dai_drv->capture.channels_max = ch; + dai_drv->capture.rates = SNDRV_PCM_RATE_8000_192000; + dai_drv->ops = &xlnx_i2s_dai_ops; + } else { + return -ENODEV; + } + + dev_set_drvdata(&pdev->dev, base); + + ret = devm_snd_soc_register_component(&pdev->dev, &xlnx_i2s_component, + dai_drv, 1); + if (ret) { + dev_err(&pdev->dev, "i2s component registration failed\n"); + return ret; + } + + dev_info(&pdev->dev, "%s DAI registered\n", dai_drv->name); + + return ret; +} + +static struct platform_driver xlnx_i2s_aud_driver = { + .driver = { + .name = DRV_NAME, + .of_match_table = xlnx_i2s_of_match, + }, + .probe = xlnx_i2s_probe, +}; + +module_platform_driver(xlnx_i2s_aud_driver); + +MODULE_LICENSE("GPL v2"); +MODULE_AUTHOR("Praveen Vuppala <praveenv@xilinx.com>"); +MODULE_AUTHOR("Maruthi Srinivas Bayyavarapu <maruthis@xilinx.com>"); diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index e73c962590eb..883678ee971c 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -1146,10 +1146,8 @@ static int snd_cs4231_playback_open(struct snd_pcm_substream *substream) runtime->hw = snd_cs4231_playback; err = snd_cs4231_open(chip, CS4231_MODE_PLAY); - if (err < 0) { - snd_free_pages(runtime->dma_area, runtime->dma_bytes); + if (err < 0) return err; - } chip->playback_substream = substream; chip->p_periods_sent = 0; snd_pcm_set_sync(substream); @@ -1167,10 +1165,8 @@ static int snd_cs4231_capture_open(struct snd_pcm_substream *substream) runtime->hw = snd_cs4231_capture; err = snd_cs4231_open(chip, CS4231_MODE_RECORD); - if (err < 0) { - snd_free_pages(runtime->dma_area, runtime->dma_bytes); + if (err < 0) return err; - } chip->capture_substream = substream; chip->c_periods_sent = 0; snd_pcm_set_sync(substream); @@ -2075,12 +2071,12 @@ static int cs4231_ebus_probe(struct platform_device *op) static int cs4231_probe(struct platform_device *op) { #ifdef EBUS_SUPPORT - if (!strcmp(op->dev.of_node->parent->name, "ebus")) + if (of_node_name_eq(op->dev.of_node->parent, "ebus")) return cs4231_ebus_probe(op); #endif #ifdef SBUS_SUPPORT - if (!strcmp(op->dev.of_node->parent->name, "sbus") || - !strcmp(op->dev.of_node->parent->name, "sbi")) + if (of_node_name_eq(op->dev.of_node->parent, "sbus") || + of_node_name_eq(op->dev.of_node->parent, "sbi")) return cs4231_sbus_probe(op); #endif return -ENODEV; diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 7609eceba1a2..9e71d7cda999 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -2541,8 +2541,8 @@ static int snd_dbri_create(struct snd_card *card, dbri->op = op; dbri->irq = irq; - dbri->dma = dma_zalloc_coherent(&op->dev, sizeof(struct dbri_dma), - &dbri->dma_dvma, GFP_KERNEL); + dbri->dma = dma_alloc_coherent(&op->dev, sizeof(struct dbri_dma), + &dbri->dma_dvma, GFP_KERNEL); if (!dbri->dma) return -ENOMEM; diff --git a/sound/synth/emux/emux_hwdep.c b/sound/synth/emux/emux_hwdep.c index e557946718a9..d9fcae071b47 100644 --- a/sound/synth/emux/emux_hwdep.c +++ b/sound/synth/emux/emux_hwdep.c @@ -22,9 +22,9 @@ #include <sound/core.h> #include <sound/hwdep.h> #include <linux/uaccess.h> +#include <linux/nospec.h> #include "emux_voice.h" - #define TMP_CLIENT_ID 0x1001 /* @@ -66,13 +66,16 @@ snd_emux_hwdep_misc_mode(struct snd_emux *emu, void __user *arg) return -EFAULT; if (info.mode < 0 || info.mode >= EMUX_MD_END) return -EINVAL; + info.mode = array_index_nospec(info.mode, EMUX_MD_END); if (info.port < 0) { for (i = 0; i < emu->num_ports; i++) emu->portptrs[i]->ctrls[info.mode] = info.value; } else { - if (info.port < emu->num_ports) + if (info.port < emu->num_ports) { + info.port = array_index_nospec(info.port, emu->num_ports); emu->portptrs[info.port]->ctrls[info.mode] = info.value; + } } return 0; } diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index d55ca48de3ea..f4a72e39ffa9 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -200,6 +200,7 @@ static void usb_ep1_command_reply_dispatch (struct urb* urb) break; } #ifdef CONFIG_SND_USB_CAIAQ_INPUT + /* fall through */ case EP1_CMD_READ_ERP: case EP1_CMD_READ_ANALOG: snd_usb_caiaq_input_dispatch(cdev, buf, urb->actual_length); diff --git a/sound/usb/card.c b/sound/usb/card.c index 2bfe4e80a6b9..746a72e23cf9 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -246,7 +246,7 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) h1 = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, NULL, UAC_HEADER); - if (!h1) { + if (!h1 || h1->bLength < sizeof(*h1)) { dev_err(&dev->dev, "cannot find UAC_HEADER\n"); return -EINVAL; } @@ -682,9 +682,12 @@ static int usb_audio_probe(struct usb_interface *intf, __error: if (chip) { + /* chip->active is inside the chip->card object, + * decrement before memory is possibly returned. + */ + atomic_dec(&chip->active); if (!chip->num_interfaces) snd_card_free(chip->card); - atomic_dec(&chip->active); } mutex_unlock(®ister_mutex); return err; diff --git a/sound/usb/midi.c b/sound/usb/midi.c index dcfc546d81b9..b737f0ec77d0 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1175,8 +1175,7 @@ static void snd_usbmidi_output_trigger(struct snd_rawmidi_substream *substream, if (port->ep->umidi->disconnected) { /* gobble up remaining bytes to prevent wait in * snd_rawmidi_drain_output */ - while (!snd_rawmidi_transmit_empty(substream)) - snd_rawmidi_transmit_ack(substream, 1); + snd_rawmidi_proceed(substream); return; } tasklet_schedule(&port->ep->tasklet); diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index c63c84b54969..e7d441d0e839 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -753,8 +753,9 @@ static int uac_mixer_unit_get_channels(struct mixer_build *state, struct uac_mixer_unit_descriptor *desc) { int mu_channels; + void *c; - if (desc->bLength < 11) + if (desc->bLength < sizeof(*desc)) return -EINVAL; if (!desc->bNrInPins) return -EINVAL; @@ -763,6 +764,8 @@ static int uac_mixer_unit_get_channels(struct mixer_build *state, case UAC_VERSION_1: case UAC_VERSION_2: default: + if (desc->bLength < sizeof(*desc) + desc->bNrInPins + 1) + return 0; /* no bmControls -> skip */ mu_channels = uac_mixer_unit_bNrChannels(desc); break; case UAC_VERSION_3: @@ -772,7 +775,11 @@ static int uac_mixer_unit_get_channels(struct mixer_build *state, } if (!mu_channels) - return -EINVAL; + return 0; + + c = uac_mixer_unit_bmControls(desc, state->mixer->protocol); + if (c - (void *)desc + (mu_channels - 1) / 8 >= desc->bLength) + return 0; /* no bmControls -> skip */ return mu_channels; } @@ -944,7 +951,7 @@ static int check_input_term(struct mixer_build *state, int id, struct uac_mixer_unit_descriptor *d = p1; err = uac_mixer_unit_get_channels(state, d); - if (err < 0) + if (err <= 0) return err; term->channels = err; @@ -2068,11 +2075,15 @@ static int parse_audio_input_terminal(struct mixer_build *state, int unitid, if (state->mixer->protocol == UAC_VERSION_2) { struct uac2_input_terminal_descriptor *d_v2 = raw_desc; + if (d_v2->bLength < sizeof(*d_v2)) + return -EINVAL; control = UAC2_TE_CONNECTOR; term_id = d_v2->bTerminalID; bmctls = le16_to_cpu(d_v2->bmControls); } else if (state->mixer->protocol == UAC_VERSION_3) { struct uac3_input_terminal_descriptor *d_v3 = raw_desc; + if (d_v3->bLength < sizeof(*d_v3)) + return -EINVAL; control = UAC3_TE_INSERTION; term_id = d_v3->bTerminalID; bmctls = le32_to_cpu(d_v3->bmControls); @@ -2118,7 +2129,7 @@ static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, if (err < 0) continue; /* no bmControls field (e.g. Maya44) -> ignore */ - if (desc->bLength <= 10 + input_pins) + if (!num_outs) continue; err = check_input_term(state, desc->baSourceID[pin], &iterm); if (err < 0) @@ -2314,7 +2325,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, char *name) { struct uac_processing_unit_descriptor *desc = raw_desc; - int num_ins = desc->bNrInPins; + int num_ins; struct usb_mixer_elem_info *cval; struct snd_kcontrol *kctl; int i, err, nameid, type, len; @@ -2329,7 +2340,13 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, 0, NULL, default_value_info }; - if (desc->bLength < 13 || desc->bLength < 13 + num_ins || + if (desc->bLength < 13) { + usb_audio_err(state->chip, "invalid %s descriptor (id %d)\n", name, unitid); + return -EINVAL; + } + + num_ins = desc->bNrInPins; + if (desc->bLength < 13 + num_ins || desc->bLength < num_ins + uac_processing_unit_bControlSize(desc, state->mixer->protocol)) { usb_audio_err(state->chip, "invalid %s descriptor (id %d)\n", name, unitid); return -EINVAL; diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index cbfb48bdea51..85ae0ff2382a 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -29,6 +29,7 @@ #include <linux/hid.h> #include <linux/init.h> +#include <linux/math64.h> #include <linux/slab.h> #include <linux/usb.h> #include <linux/usb/audio.h> @@ -1817,6 +1818,380 @@ static int dell_dock_mixer_init(struct usb_mixer_interface *mixer) return 0; } +/* RME Class Compliant device quirks */ + +#define SND_RME_GET_STATUS1 23 +#define SND_RME_GET_CURRENT_FREQ 17 +#define SND_RME_CLK_SYSTEM_SHIFT 16 +#define SND_RME_CLK_SYSTEM_MASK 0x1f +#define SND_RME_CLK_AES_SHIFT 8 +#define SND_RME_CLK_SPDIF_SHIFT 12 +#define SND_RME_CLK_AES_SPDIF_MASK 0xf +#define SND_RME_CLK_SYNC_SHIFT 6 +#define SND_RME_CLK_SYNC_MASK 0x3 +#define SND_RME_CLK_FREQMUL_SHIFT 18 +#define SND_RME_CLK_FREQMUL_MASK 0x7 +#define SND_RME_CLK_SYSTEM(x) \ + ((x >> SND_RME_CLK_SYSTEM_SHIFT) & SND_RME_CLK_SYSTEM_MASK) +#define SND_RME_CLK_AES(x) \ + ((x >> SND_RME_CLK_AES_SHIFT) & SND_RME_CLK_AES_SPDIF_MASK) +#define SND_RME_CLK_SPDIF(x) \ + ((x >> SND_RME_CLK_SPDIF_SHIFT) & SND_RME_CLK_AES_SPDIF_MASK) +#define SND_RME_CLK_SYNC(x) \ + ((x >> SND_RME_CLK_SYNC_SHIFT) & SND_RME_CLK_SYNC_MASK) +#define SND_RME_CLK_FREQMUL(x) \ + ((x >> SND_RME_CLK_FREQMUL_SHIFT) & SND_RME_CLK_FREQMUL_MASK) +#define SND_RME_CLK_AES_LOCK 0x1 +#define SND_RME_CLK_AES_SYNC 0x4 +#define SND_RME_CLK_SPDIF_LOCK 0x2 +#define SND_RME_CLK_SPDIF_SYNC 0x8 +#define SND_RME_SPDIF_IF_SHIFT 4 +#define SND_RME_SPDIF_FORMAT_SHIFT 5 +#define SND_RME_BINARY_MASK 0x1 +#define SND_RME_SPDIF_IF(x) \ + ((x >> SND_RME_SPDIF_IF_SHIFT) & SND_RME_BINARY_MASK) +#define SND_RME_SPDIF_FORMAT(x) \ + ((x >> SND_RME_SPDIF_FORMAT_SHIFT) & SND_RME_BINARY_MASK) + +static const u32 snd_rme_rate_table[] = { + 32000, 44100, 48000, 50000, + 64000, 88200, 96000, 100000, + 128000, 176400, 192000, 200000, + 256000, 352800, 384000, 400000, + 512000, 705600, 768000, 800000 +}; +/* maximum number of items for AES and S/PDIF rates for above table */ +#define SND_RME_RATE_IDX_AES_SPDIF_NUM 12 + +enum snd_rme_domain { + SND_RME_DOMAIN_SYSTEM, + SND_RME_DOMAIN_AES, + SND_RME_DOMAIN_SPDIF +}; + +enum snd_rme_clock_status { + SND_RME_CLOCK_NOLOCK, + SND_RME_CLOCK_LOCK, + SND_RME_CLOCK_SYNC +}; + +static int snd_rme_read_value(struct snd_usb_audio *chip, + unsigned int item, + u32 *value) +{ + struct usb_device *dev = chip->dev; + int err; + + err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), + item, + USB_DIR_IN | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0, 0, + value, sizeof(*value)); + if (err < 0) + dev_err(&dev->dev, + "unable to issue vendor read request %d (ret = %d)", + item, err); + return err; +} + +static int snd_rme_get_status1(struct snd_kcontrol *kcontrol, + u32 *status1) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct snd_usb_audio *chip = list->mixer->chip; + int err; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + err = snd_rme_read_value(chip, SND_RME_GET_STATUS1, status1); + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_rme_rate_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + u32 rate = 0; + int idx; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + switch (kcontrol->private_value) { + case SND_RME_DOMAIN_SYSTEM: + idx = SND_RME_CLK_SYSTEM(status1); + if (idx < ARRAY_SIZE(snd_rme_rate_table)) + rate = snd_rme_rate_table[idx]; + break; + case SND_RME_DOMAIN_AES: + idx = SND_RME_CLK_AES(status1); + if (idx < SND_RME_RATE_IDX_AES_SPDIF_NUM) + rate = snd_rme_rate_table[idx]; + break; + case SND_RME_DOMAIN_SPDIF: + idx = SND_RME_CLK_SPDIF(status1); + if (idx < SND_RME_RATE_IDX_AES_SPDIF_NUM) + rate = snd_rme_rate_table[idx]; + break; + default: + return -EINVAL; + } + ucontrol->value.integer.value[0] = rate; + return 0; +} + +static int snd_rme_sync_state_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + int idx = SND_RME_CLOCK_NOLOCK; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + switch (kcontrol->private_value) { + case SND_RME_DOMAIN_AES: /* AES */ + if (status1 & SND_RME_CLK_AES_SYNC) + idx = SND_RME_CLOCK_SYNC; + else if (status1 & SND_RME_CLK_AES_LOCK) + idx = SND_RME_CLOCK_LOCK; + break; + case SND_RME_DOMAIN_SPDIF: /* SPDIF */ + if (status1 & SND_RME_CLK_SPDIF_SYNC) + idx = SND_RME_CLOCK_SYNC; + else if (status1 & SND_RME_CLK_SPDIF_LOCK) + idx = SND_RME_CLOCK_LOCK; + break; + default: + return -EINVAL; + } + ucontrol->value.enumerated.item[0] = idx; + return 0; +} + +static int snd_rme_spdif_if_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + ucontrol->value.enumerated.item[0] = SND_RME_SPDIF_IF(status1); + return 0; +} + +static int snd_rme_spdif_format_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + ucontrol->value.enumerated.item[0] = SND_RME_SPDIF_FORMAT(status1); + return 0; +} + +static int snd_rme_sync_source_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + ucontrol->value.enumerated.item[0] = SND_RME_CLK_SYNC(status1); + return 0; +} + +static int snd_rme_current_freq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct snd_usb_audio *chip = list->mixer->chip; + u32 status1; + const u64 num = 104857600000000ULL; + u32 den; + unsigned int freq; + int err; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + err = snd_rme_read_value(chip, SND_RME_GET_STATUS1, &status1); + if (err < 0) + goto end; + err = snd_rme_read_value(chip, SND_RME_GET_CURRENT_FREQ, &den); + if (err < 0) + goto end; + freq = (den == 0) ? 0 : div64_u64(num, den); + freq <<= SND_RME_CLK_FREQMUL(status1); + ucontrol->value.integer.value[0] = freq; + +end: + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_rme_rate_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + switch (kcontrol->private_value) { + case SND_RME_DOMAIN_SYSTEM: + uinfo->value.integer.min = 32000; + uinfo->value.integer.max = 800000; + break; + case SND_RME_DOMAIN_AES: + case SND_RME_DOMAIN_SPDIF: + default: + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 200000; + } + uinfo->value.integer.step = 0; + return 0; +} + +static int snd_rme_sync_state_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const sync_states[] = { + "No Lock", "Lock", "Sync" + }; + + return snd_ctl_enum_info(uinfo, 1, + ARRAY_SIZE(sync_states), sync_states); +} + +static int snd_rme_spdif_if_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const spdif_if[] = { + "Coaxial", "Optical" + }; + + return snd_ctl_enum_info(uinfo, 1, + ARRAY_SIZE(spdif_if), spdif_if); +} + +static int snd_rme_spdif_format_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const optical_type[] = { + "Consumer", "Professional" + }; + + return snd_ctl_enum_info(uinfo, 1, + ARRAY_SIZE(optical_type), optical_type); +} + +static int snd_rme_sync_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const sync_sources[] = { + "Internal", "AES", "SPDIF", "Internal" + }; + + return snd_ctl_enum_info(uinfo, 1, + ARRAY_SIZE(sync_sources), sync_sources); +} + +static struct snd_kcontrol_new snd_rme_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "AES Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_rate_info, + .get = snd_rme_rate_get, + .private_value = SND_RME_DOMAIN_AES + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "AES Sync", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_sync_state_info, + .get = snd_rme_sync_state_get, + .private_value = SND_RME_DOMAIN_AES + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_rate_info, + .get = snd_rme_rate_get, + .private_value = SND_RME_DOMAIN_SPDIF + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Sync", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_sync_state_info, + .get = snd_rme_sync_state_get, + .private_value = SND_RME_DOMAIN_SPDIF + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Interface", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_spdif_if_info, + .get = snd_rme_spdif_if_get, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Format", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_spdif_format_info, + .get = snd_rme_spdif_format_get, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Sync Source", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_sync_source_info, + .get = snd_rme_sync_source_get + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "System Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_rate_info, + .get = snd_rme_rate_get, + .private_value = SND_RME_DOMAIN_SYSTEM + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Current Frequency", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_rate_info, + .get = snd_rme_current_freq_get + } +}; + +static int snd_rme_controls_create(struct usb_mixer_interface *mixer) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(snd_rme_controls); ++i) { + err = add_single_ctl_with_resume(mixer, 0, + NULL, + &snd_rme_controls[i], + NULL); + if (err < 0) + return err; + } + + return 0; +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -1904,6 +2279,12 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x0bda, 0x4014): /* Dell WD15 dock */ err = dell_dock_mixer_init(mixer); break; + + case USB_ID(0x2a39, 0x3fd2): /* RME ADI-2 Pro */ + case USB_ID(0x2a39, 0x3fd3): /* RME ADI-2 DAC */ + case USB_ID(0x2a39, 0x3fd4): /* RME */ + err = snd_rme_controls_create(mixer); + break; } return err; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 382847154227..db114f3977e0 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -314,6 +314,9 @@ static int search_roland_implicit_fb(struct usb_device *dev, int ifnum, return 0; } +/* Setup an implicit feedback endpoint from a quirk. Returns 0 if no quirk + * applies. Returns 1 if a quirk was found. + */ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, struct usb_device *dev, struct usb_interface_descriptor *altsd, @@ -384,7 +387,7 @@ add_sync_ep: subs->data_endpoint->sync_master = subs->sync_endpoint; - return 0; + return 1; } static int set_sync_endpoint(struct snd_usb_substream *subs, @@ -423,6 +426,10 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, if (err < 0) return err; + /* endpoint set by quirk */ + if (err > 0) + return 0; + if (altsd->bNumEndpoints < 2) return 0; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 08aa78007020..b345beb447bd 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3326,6 +3326,9 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, + { + .ifnum = -1 + }, } } }, @@ -3346,19 +3349,14 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .ifnum = 0, .type = QUIRK_AUDIO_STANDARD_MIXER, }, - /* Capture */ - { - .ifnum = 1, - .type = QUIRK_IGNORE_INTERFACE, - }, /* Playback */ { - .ifnum = 2, + .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = &(const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels = 2, - .iface = 2, + .iface = 1, .altsetting = 1, .altset_idx = 1, .attributes = UAC_EP_CS_ATTR_FILL_MAX | @@ -3374,6 +3372,9 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, + { + .ifnum = -1 + }, } } }, @@ -3387,5 +3388,15 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .ifnum = QUIRK_NO_INTERFACE } }, +/* Dell WD19 Dock */ +{ + USB_DEVICE(0x0bda, 0x402e), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Dell", + .product_name = "WD19 Dock", + .profile_name = "Dell-WD15-Dock", + .ifnum = QUIRK_NO_INTERFACE + } +}, #undef USB_DEVICE_VENDOR_SPEC diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 8a945ece9869..7e65fe853ee3 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -19,6 +19,7 @@ #include <linux/usb.h> #include <linux/usb/audio.h> #include <linux/usb/midi.h> +#include <linux/bits.h> #include <sound/control.h> #include <sound/core.h> @@ -668,15 +669,133 @@ static int snd_usb_cm106_boot_quirk(struct usb_device *dev) } /* - * C-Media CM6206 is based on CM106 with two additional - * registers that are not documented in the data sheet. - * Values here are chosen based on sniffing USB traffic - * under Windows. + * CM6206 registers from the CM6206 datasheet rev 2.1 */ +#define CM6206_REG0_DMA_MASTER BIT(15) +#define CM6206_REG0_SPDIFO_RATE_48K (2 << 12) +#define CM6206_REG0_SPDIFO_RATE_96K (7 << 12) +/* Bit 4 thru 11 is the S/PDIF category code */ +#define CM6206_REG0_SPDIFO_CAT_CODE_GENERAL (0 << 4) +#define CM6206_REG0_SPDIFO_EMPHASIS_CD BIT(3) +#define CM6206_REG0_SPDIFO_COPYRIGHT_NA BIT(2) +#define CM6206_REG0_SPDIFO_NON_AUDIO BIT(1) +#define CM6206_REG0_SPDIFO_PRO_FORMAT BIT(0) + +#define CM6206_REG1_TEST_SEL_CLK BIT(14) +#define CM6206_REG1_PLLBIN_EN BIT(13) +#define CM6206_REG1_SOFT_MUTE_EN BIT(12) +#define CM6206_REG1_GPIO4_OUT BIT(11) +#define CM6206_REG1_GPIO4_OE BIT(10) +#define CM6206_REG1_GPIO3_OUT BIT(9) +#define CM6206_REG1_GPIO3_OE BIT(8) +#define CM6206_REG1_GPIO2_OUT BIT(7) +#define CM6206_REG1_GPIO2_OE BIT(6) +#define CM6206_REG1_GPIO1_OUT BIT(5) +#define CM6206_REG1_GPIO1_OE BIT(4) +#define CM6206_REG1_SPDIFO_INVALID BIT(3) +#define CM6206_REG1_SPDIF_LOOP_EN BIT(2) +#define CM6206_REG1_SPDIFO_DIS BIT(1) +#define CM6206_REG1_SPDIFI_MIX BIT(0) + +#define CM6206_REG2_DRIVER_ON BIT(15) +#define CM6206_REG2_HEADP_SEL_SIDE_CHANNELS (0 << 13) +#define CM6206_REG2_HEADP_SEL_SURROUND_CHANNELS (1 << 13) +#define CM6206_REG2_HEADP_SEL_CENTER_SUBW (2 << 13) +#define CM6206_REG2_HEADP_SEL_FRONT_CHANNELS (3 << 13) +#define CM6206_REG2_MUTE_HEADPHONE_RIGHT BIT(12) +#define CM6206_REG2_MUTE_HEADPHONE_LEFT BIT(11) +#define CM6206_REG2_MUTE_REAR_SURROUND_RIGHT BIT(10) +#define CM6206_REG2_MUTE_REAR_SURROUND_LEFT BIT(9) +#define CM6206_REG2_MUTE_SIDE_SURROUND_RIGHT BIT(8) +#define CM6206_REG2_MUTE_SIDE_SURROUND_LEFT BIT(7) +#define CM6206_REG2_MUTE_SUBWOOFER BIT(6) +#define CM6206_REG2_MUTE_CENTER BIT(5) +#define CM6206_REG2_MUTE_RIGHT_FRONT BIT(3) +#define CM6206_REG2_MUTE_LEFT_FRONT BIT(3) +#define CM6206_REG2_EN_BTL BIT(2) +#define CM6206_REG2_MCUCLKSEL_1_5_MHZ (0) +#define CM6206_REG2_MCUCLKSEL_3_MHZ (1) +#define CM6206_REG2_MCUCLKSEL_6_MHZ (2) +#define CM6206_REG2_MCUCLKSEL_12_MHZ (3) + +/* Bit 11..13 sets the sensitivity to FLY tuner volume control VP/VD signal */ +#define CM6206_REG3_FLYSPEED_DEFAULT (2 << 11) +#define CM6206_REG3_VRAP25EN BIT(10) +#define CM6206_REG3_MSEL1 BIT(9) +#define CM6206_REG3_SPDIFI_RATE_44_1K BIT(0 << 7) +#define CM6206_REG3_SPDIFI_RATE_48K BIT(2 << 7) +#define CM6206_REG3_SPDIFI_RATE_32K BIT(3 << 7) +#define CM6206_REG3_PINSEL BIT(6) +#define CM6206_REG3_FOE BIT(5) +#define CM6206_REG3_ROE BIT(4) +#define CM6206_REG3_CBOE BIT(3) +#define CM6206_REG3_LOSE BIT(2) +#define CM6206_REG3_HPOE BIT(1) +#define CM6206_REG3_SPDIFI_CANREC BIT(0) + +#define CM6206_REG5_DA_RSTN BIT(13) +#define CM6206_REG5_AD_RSTN BIT(12) +#define CM6206_REG5_SPDIFO_AD2SPDO BIT(12) +#define CM6206_REG5_SPDIFO_SEL_FRONT (0 << 9) +#define CM6206_REG5_SPDIFO_SEL_SIDE_SUR (1 << 9) +#define CM6206_REG5_SPDIFO_SEL_CEN_LFE (2 << 9) +#define CM6206_REG5_SPDIFO_SEL_REAR_SUR (3 << 9) +#define CM6206_REG5_CODECM BIT(8) +#define CM6206_REG5_EN_HPF BIT(7) +#define CM6206_REG5_T_SEL_DSDA4 BIT(6) +#define CM6206_REG5_T_SEL_DSDA3 BIT(5) +#define CM6206_REG5_T_SEL_DSDA2 BIT(4) +#define CM6206_REG5_T_SEL_DSDA1 BIT(3) +#define CM6206_REG5_T_SEL_DSDAD_NORMAL 0 +#define CM6206_REG5_T_SEL_DSDAD_FRONT 4 +#define CM6206_REG5_T_SEL_DSDAD_S_SURROUND 5 +#define CM6206_REG5_T_SEL_DSDAD_CEN_LFE 6 +#define CM6206_REG5_T_SEL_DSDAD_R_SURROUND 7 + static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) { int err = 0, reg; - int val[] = {0x2004, 0x3000, 0xf800, 0x143f, 0x0000, 0x3000}; + int val[] = { + /* + * Values here are chosen based on sniffing USB traffic + * under Windows. + * + * REG0: DAC is master, sample rate 48kHz, no copyright + */ + CM6206_REG0_SPDIFO_RATE_48K | + CM6206_REG0_SPDIFO_COPYRIGHT_NA, + /* + * REG1: PLL binary search enable, soft mute enable. + */ + CM6206_REG1_PLLBIN_EN | + CM6206_REG1_SOFT_MUTE_EN, + /* + * REG2: enable output drivers, + * select front channels to the headphone output, + * then mute the headphone channels, run the MCU + * at 1.5 MHz. + */ + CM6206_REG2_DRIVER_ON | + CM6206_REG2_HEADP_SEL_FRONT_CHANNELS | + CM6206_REG2_MUTE_HEADPHONE_RIGHT | + CM6206_REG2_MUTE_HEADPHONE_LEFT, + /* + * REG3: default flyspeed, set 2.5V mic bias + * enable all line out ports and enable SPDIF + */ + CM6206_REG3_FLYSPEED_DEFAULT | + CM6206_REG3_VRAP25EN | + CM6206_REG3_FOE | + CM6206_REG3_ROE | + CM6206_REG3_CBOE | + CM6206_REG3_LOSE | + CM6206_REG3_HPOE | + CM6206_REG3_SPDIFI_CANREC, + /* REG4 is just a bunch of GPIO lines */ + 0x0000, + /* REG5: de-assert AD/DA reset signals */ + CM6206_REG5_DA_RSTN | + CM6206_REG5_AD_RSTN }; for (reg = 0; reg < ARRAY_SIZE(val); reg++) { err = snd_usb_cm106_write_int_reg(dev, reg, val[reg]); @@ -1373,6 +1492,8 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; + case USB_ID(0x10cb, 0x0103): /* The Bit Opus #3; with fp->dsd_raw */ + case USB_ID(0x152a, 0x85de): /* SMSL D1 DAC */ case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */ case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */ @@ -1446,6 +1567,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case 0x20b1: /* XMOS based devices */ case 0x152a: /* Thesycon devices */ case 0x25ce: /* Mytek devices */ + case 0x2ab6: /* T+A devices */ if (fp->dsd_raw) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 67cf849aa16b..d9e3de495c16 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -596,12 +596,8 @@ static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip, csep = snd_usb_find_desc(alts->extra, alts->extralen, NULL, USB_DT_CS_ENDPOINT); if (!csep || csep->bLength < 7 || - csep->bDescriptorSubtype != UAC_EP_GENERAL) { - usb_audio_warn(chip, - "%u:%d : no or invalid class specific endpoint descriptor\n", - iface_no, altsd->bAlternateSetting); - return 0; - } + csep->bDescriptorSubtype != UAC_EP_GENERAL) + goto error; if (protocol == UAC_VERSION_1) { attributes = csep->bmAttributes; @@ -609,6 +605,8 @@ static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip, struct uac2_iso_endpoint_descriptor *csep2 = (struct uac2_iso_endpoint_descriptor *) csep; + if (csep2->bLength < sizeof(*csep2)) + goto error; attributes = csep->bmAttributes & UAC_EP_CS_ATTR_FILL_MAX; /* emulate the endpoint attributes of a v1 device */ @@ -618,12 +616,20 @@ static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip, struct uac3_iso_endpoint_descriptor *csep3 = (struct uac3_iso_endpoint_descriptor *) csep; + if (csep3->bLength < sizeof(*csep3)) + goto error; /* emulate the endpoint attributes of a v1 device */ if (le32_to_cpu(csep3->bmControls) & UAC2_CONTROL_PITCH) attributes |= UAC_EP_CS_ATTR_PITCH_CONTROL; } return attributes; + + error: + usb_audio_warn(chip, + "%u:%d : no or invalid class specific endpoint descriptor\n", + iface_no, altsd->bAlternateSetting); + return 0; } /* find an input terminal descriptor (either UAC1 or UAC2) with the given @@ -631,13 +637,17 @@ static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip, */ static void * snd_usb_find_input_terminal_descriptor(struct usb_host_interface *ctrl_iface, - int terminal_id) + int terminal_id, bool uac23) { struct uac2_input_terminal_descriptor *term = NULL; + size_t minlen = uac23 ? sizeof(struct uac2_input_terminal_descriptor) : + sizeof(struct uac_input_terminal_descriptor); while ((term = snd_usb_find_csint_desc(ctrl_iface->extra, ctrl_iface->extralen, term, UAC_INPUT_TERMINAL))) { + if (term->bLength < minlen) + continue; if (term->bTerminalID == terminal_id) return term; } @@ -655,7 +665,8 @@ snd_usb_find_output_terminal_descriptor(struct usb_host_interface *ctrl_iface, while ((term = snd_usb_find_csint_desc(ctrl_iface->extra, ctrl_iface->extralen, term, UAC_OUTPUT_TERMINAL))) { - if (term->bTerminalID == terminal_id) + if (term->bLength >= sizeof(*term) && + term->bTerminalID == terminal_id) return term; } @@ -729,7 +740,8 @@ snd_usb_get_audioformat_uac12(struct snd_usb_audio *chip, format = le16_to_cpu(as->wFormatTag); /* remember the format value */ iterm = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf, - as->bTerminalLink); + as->bTerminalLink, + false); if (iterm) { num_channels = iterm->bNrChannels; chconfig = le16_to_cpu(iterm->wChannelConfig); @@ -764,7 +776,8 @@ snd_usb_get_audioformat_uac12(struct snd_usb_audio *chip, * to extract the clock */ input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf, - as->bTerminalLink); + as->bTerminalLink, + true); if (input_term) { clock = input_term->bCSourceID; if (!chconfig && (num_channels == input_term->bNrChannels)) @@ -998,7 +1011,8 @@ snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip, * to extract the clock */ input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf, - as->bTerminalLink); + as->bTerminalLink, + true); if (input_term) { clock = input_term->bCSourceID; goto found_clock; diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index fa7dca5a68c8..00c92eb854ce 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -30,7 +30,6 @@ #include <linux/pm_runtime.h> #include <linux/dma-mapping.h> #include <linux/delay.h> -#include <asm/set_memory.h> #include <sound/core.h> #include <sound/asoundef.h> #include <sound/pcm.h> @@ -1141,8 +1140,7 @@ static int had_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_intelhad *intelhaddata; - unsigned long addr; - int pages, buf_size, retval; + int buf_size, retval; intelhaddata = snd_pcm_substream_chip(substream); buf_size = params_buffer_bytes(hw_params); @@ -1151,17 +1149,6 @@ static int had_pcm_hw_params(struct snd_pcm_substream *substream, return retval; dev_dbg(intelhaddata->dev, "%s:allocated memory = %d\n", __func__, buf_size); - /* mark the pages as uncached region */ - addr = (unsigned long) substream->runtime->dma_area; - pages = (substream->runtime->dma_bytes + PAGE_SIZE - 1) / PAGE_SIZE; - retval = set_memory_uc(addr, pages); - if (retval) { - dev_err(intelhaddata->dev, "set_memory_uc failed.Error:%d\n", - retval); - return retval; - } - memset(substream->runtime->dma_area, 0, buf_size); - return retval; } @@ -1171,21 +1158,11 @@ static int had_pcm_hw_params(struct snd_pcm_substream *substream, static int had_pcm_hw_free(struct snd_pcm_substream *substream) { struct snd_intelhad *intelhaddata; - unsigned long addr; - u32 pages; intelhaddata = snd_pcm_substream_chip(substream); had_do_reset(intelhaddata); - /* mark back the pages as cached/writeback region before the free */ - if (substream->runtime->dma_area != NULL) { - addr = (unsigned long) substream->runtime->dma_area; - pages = (substream->runtime->dma_bytes + PAGE_SIZE - 1) / - PAGE_SIZE; - set_memory_wb(addr, pages); - return snd_pcm_lib_free_pages(substream); - } - return 0; + return snd_pcm_lib_free_pages(substream); } /* @@ -1671,7 +1648,7 @@ static int had_create_jack(struct snd_intelhad *ctx, * PM callbacks */ -static int hdmi_lpe_audio_runtime_suspend(struct device *dev) +static int __maybe_unused hdmi_lpe_audio_suspend(struct device *dev) { struct snd_intelhad_card *card_ctx = dev_get_drvdata(dev); int port; @@ -1687,23 +1664,8 @@ static int hdmi_lpe_audio_runtime_suspend(struct device *dev) } } - return 0; -} + snd_power_change_state(card_ctx->card, SNDRV_CTL_POWER_D3hot); -static int __maybe_unused hdmi_lpe_audio_suspend(struct device *dev) -{ - struct snd_intelhad_card *card_ctx = dev_get_drvdata(dev); - int err; - - err = hdmi_lpe_audio_runtime_suspend(dev); - if (!err) - snd_power_change_state(card_ctx->card, SNDRV_CTL_POWER_D3hot); - return err; -} - -static int hdmi_lpe_audio_runtime_resume(struct device *dev) -{ - pm_runtime_mark_last_busy(dev); return 0; } @@ -1711,8 +1673,10 @@ static int __maybe_unused hdmi_lpe_audio_resume(struct device *dev) { struct snd_intelhad_card *card_ctx = dev_get_drvdata(dev); - hdmi_lpe_audio_runtime_resume(dev); + pm_runtime_mark_last_busy(dev); + snd_power_change_state(card_ctx->card, SNDRV_CTL_POWER_D0); + return 0; } @@ -1860,7 +1824,7 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev) * try to allocate 600k buffer as default which is large enough */ snd_pcm_lib_preallocate_pages_for_all(pcm, - SNDRV_DMA_TYPE_DEV, NULL, + SNDRV_DMA_TYPE_DEV_UC, NULL, HAD_DEFAULT_BUFFER, HAD_MAX_BUFFER); /* create controls */ @@ -1900,7 +1864,6 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev) pm_runtime_use_autosuspend(&pdev->dev); pm_runtime_mark_last_busy(&pdev->dev); - pm_runtime_set_active(&pdev->dev); dev_dbg(&pdev->dev, "%s: handle pending notification\n", __func__); for_each_port(card_ctx, port) { @@ -1931,8 +1894,6 @@ static int hdmi_lpe_audio_remove(struct platform_device *pdev) static const struct dev_pm_ops hdmi_lpe_audio_pm = { SET_SYSTEM_SLEEP_PM_OPS(hdmi_lpe_audio_suspend, hdmi_lpe_audio_resume) - SET_RUNTIME_PM_OPS(hdmi_lpe_audio_runtime_suspend, - hdmi_lpe_audio_runtime_resume, NULL) }; static struct platform_driver hdmi_lpe_audio_driver = { diff --git a/sound/xen/Kconfig b/sound/xen/Kconfig index 4f1fceea82d2..e4d7beb4df1c 100644 --- a/sound/xen/Kconfig +++ b/sound/xen/Kconfig @@ -5,6 +5,7 @@ config SND_XEN_FRONTEND depends on XEN select SND_PCM select XEN_XENBUS_FRONTEND + select XEN_FRONT_PGDIR_SHBUF help Choose this option if you want to enable a para-virtualized frontend sound driver for Xen guest OSes. diff --git a/sound/xen/Makefile b/sound/xen/Makefile index 1e6470ecc2f2..24031775b715 100644 --- a/sound/xen/Makefile +++ b/sound/xen/Makefile @@ -3,7 +3,6 @@ snd_xen_front-objs := xen_snd_front.o \ xen_snd_front_cfg.o \ xen_snd_front_evtchnl.o \ - xen_snd_front_shbuf.o \ xen_snd_front_alsa.o obj-$(CONFIG_SND_XEN_FRONTEND) += snd_xen_front.o diff --git a/sound/xen/xen_snd_front.c b/sound/xen/xen_snd_front.c index b089b13b5160..a9e5c2cd7698 100644 --- a/sound/xen/xen_snd_front.c +++ b/sound/xen/xen_snd_front.c @@ -16,12 +16,12 @@ #include <xen/xen.h> #include <xen/xenbus.h> +#include <xen/xen-front-pgdir-shbuf.h> #include <xen/interface/io/sndif.h> #include "xen_snd_front.h" #include "xen_snd_front_alsa.h" #include "xen_snd_front_evtchnl.h" -#include "xen_snd_front_shbuf.h" static struct xensnd_req * be_stream_prepare_req(struct xen_snd_front_evtchnl *evtchnl, u8 operation) @@ -82,7 +82,7 @@ int xen_snd_front_stream_query_hw_param(struct xen_snd_front_evtchnl *evtchnl, } int xen_snd_front_stream_prepare(struct xen_snd_front_evtchnl *evtchnl, - struct xen_snd_front_shbuf *sh_buf, + struct xen_front_pgdir_shbuf *shbuf, u8 format, unsigned int channels, unsigned int rate, u32 buffer_sz, u32 period_sz) @@ -99,7 +99,8 @@ int xen_snd_front_stream_prepare(struct xen_snd_front_evtchnl *evtchnl, req->op.open.pcm_rate = rate; req->op.open.buffer_sz = buffer_sz; req->op.open.period_sz = period_sz; - req->op.open.gref_directory = xen_snd_front_shbuf_get_dir_start(sh_buf); + req->op.open.gref_directory = + xen_front_pgdir_shbuf_get_dir_start(shbuf); mutex_unlock(&evtchnl->ring_io_lock); ret = be_stream_do_io(evtchnl); diff --git a/sound/xen/xen_snd_front.h b/sound/xen/xen_snd_front.h index a2ea2463bcc5..05611f113b94 100644 --- a/sound/xen/xen_snd_front.h +++ b/sound/xen/xen_snd_front.h @@ -16,7 +16,7 @@ struct xen_snd_front_card_info; struct xen_snd_front_evtchnl; struct xen_snd_front_evtchnl_pair; -struct xen_snd_front_shbuf; +struct xen_front_pgdir_shbuf; struct xensnd_query_hw_param; struct xen_snd_front_info { @@ -35,7 +35,7 @@ int xen_snd_front_stream_query_hw_param(struct xen_snd_front_evtchnl *evtchnl, struct xensnd_query_hw_param *hw_param_resp); int xen_snd_front_stream_prepare(struct xen_snd_front_evtchnl *evtchnl, - struct xen_snd_front_shbuf *sh_buf, + struct xen_front_pgdir_shbuf *shbuf, u8 format, unsigned int channels, unsigned int rate, u32 buffer_sz, u32 period_sz); diff --git a/sound/xen/xen_snd_front_alsa.c b/sound/xen/xen_snd_front_alsa.c index 129180e17db1..a7f413cb704d 100644 --- a/sound/xen/xen_snd_front_alsa.c +++ b/sound/xen/xen_snd_front_alsa.c @@ -15,17 +15,24 @@ #include <sound/pcm_params.h> #include <xen/xenbus.h> +#include <xen/xen-front-pgdir-shbuf.h> #include "xen_snd_front.h" #include "xen_snd_front_alsa.h" #include "xen_snd_front_cfg.h" #include "xen_snd_front_evtchnl.h" -#include "xen_snd_front_shbuf.h" struct xen_snd_front_pcm_stream_info { struct xen_snd_front_info *front_info; struct xen_snd_front_evtchnl_pair *evt_pair; - struct xen_snd_front_shbuf sh_buf; + + /* This is the shared buffer with its backing storage. */ + struct xen_front_pgdir_shbuf shbuf; + u8 *buffer; + size_t buffer_sz; + int num_pages; + struct page **pages; + int index; bool is_open; @@ -214,12 +221,20 @@ static void stream_clear(struct xen_snd_front_pcm_stream_info *stream) stream->out_frames = 0; atomic_set(&stream->hw_ptr, 0); xen_snd_front_evtchnl_pair_clear(stream->evt_pair); - xen_snd_front_shbuf_clear(&stream->sh_buf); + memset(&stream->shbuf, 0, sizeof(stream->shbuf)); + stream->buffer = NULL; + stream->buffer_sz = 0; + stream->pages = NULL; + stream->num_pages = 0; } static void stream_free(struct xen_snd_front_pcm_stream_info *stream) { - xen_snd_front_shbuf_free(&stream->sh_buf); + xen_front_pgdir_shbuf_unmap(&stream->shbuf); + xen_front_pgdir_shbuf_free(&stream->shbuf); + if (stream->buffer) + free_pages_exact(stream->buffer, stream->buffer_sz); + kfree(stream->pages); stream_clear(stream); } @@ -421,10 +436,34 @@ static int alsa_close(struct snd_pcm_substream *substream) return 0; } +static int shbuf_setup_backstore(struct xen_snd_front_pcm_stream_info *stream, + size_t buffer_sz) +{ + int i; + + stream->buffer = alloc_pages_exact(stream->buffer_sz, GFP_KERNEL); + if (!stream->buffer) + return -ENOMEM; + + stream->buffer_sz = buffer_sz; + stream->num_pages = DIV_ROUND_UP(stream->buffer_sz, PAGE_SIZE); + stream->pages = kcalloc(stream->num_pages, sizeof(struct page *), + GFP_KERNEL); + if (!stream->pages) + return -ENOMEM; + + for (i = 0; i < stream->num_pages; i++) + stream->pages[i] = virt_to_page(stream->buffer + i * PAGE_SIZE); + + return 0; +} + static int alsa_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); + struct xen_snd_front_info *front_info = stream->front_info; + struct xen_front_pgdir_shbuf_cfg buf_cfg; int ret; /* @@ -432,19 +471,32 @@ static int alsa_hw_params(struct snd_pcm_substream *substream, * so free the previously allocated shared buffer if any. */ stream_free(stream); + ret = shbuf_setup_backstore(stream, params_buffer_bytes(params)); + if (ret < 0) + goto fail; - ret = xen_snd_front_shbuf_alloc(stream->front_info->xb_dev, - &stream->sh_buf, - params_buffer_bytes(params)); - if (ret < 0) { - stream_free(stream); - dev_err(&stream->front_info->xb_dev->dev, - "Failed to allocate buffers for stream with index %d\n", - stream->index); - return ret; - } + memset(&buf_cfg, 0, sizeof(buf_cfg)); + buf_cfg.xb_dev = front_info->xb_dev; + buf_cfg.pgdir = &stream->shbuf; + buf_cfg.num_pages = stream->num_pages; + buf_cfg.pages = stream->pages; + + ret = xen_front_pgdir_shbuf_alloc(&buf_cfg); + if (ret < 0) + goto fail; + + ret = xen_front_pgdir_shbuf_map(&stream->shbuf); + if (ret < 0) + goto fail; return 0; + +fail: + stream_free(stream); + dev_err(&front_info->xb_dev->dev, + "Failed to allocate buffers for stream with index %d\n", + stream->index); + return ret; } static int alsa_hw_free(struct snd_pcm_substream *substream) @@ -476,7 +528,7 @@ static int alsa_prepare(struct snd_pcm_substream *substream) sndif_format = ret; ret = xen_snd_front_stream_prepare(&stream->evt_pair->req, - &stream->sh_buf, + &stream->shbuf, sndif_format, runtime->channels, runtime->rate, @@ -556,10 +608,10 @@ static int alsa_pb_copy_user(struct snd_pcm_substream *substream, { struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); - if (unlikely(pos + count > stream->sh_buf.buffer_sz)) + if (unlikely(pos + count > stream->buffer_sz)) return -EINVAL; - if (copy_from_user(stream->sh_buf.buffer + pos, src, count)) + if (copy_from_user(stream->buffer + pos, src, count)) return -EFAULT; return xen_snd_front_stream_write(&stream->evt_pair->req, pos, count); @@ -571,10 +623,10 @@ static int alsa_pb_copy_kernel(struct snd_pcm_substream *substream, { struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); - if (unlikely(pos + count > stream->sh_buf.buffer_sz)) + if (unlikely(pos + count > stream->buffer_sz)) return -EINVAL; - memcpy(stream->sh_buf.buffer + pos, src, count); + memcpy(stream->buffer + pos, src, count); return xen_snd_front_stream_write(&stream->evt_pair->req, pos, count); } @@ -586,14 +638,14 @@ static int alsa_cap_copy_user(struct snd_pcm_substream *substream, struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); int ret; - if (unlikely(pos + count > stream->sh_buf.buffer_sz)) + if (unlikely(pos + count > stream->buffer_sz)) return -EINVAL; ret = xen_snd_front_stream_read(&stream->evt_pair->req, pos, count); if (ret < 0) return ret; - return copy_to_user(dst, stream->sh_buf.buffer + pos, count) ? + return copy_to_user(dst, stream->buffer + pos, count) ? -EFAULT : 0; } @@ -604,14 +656,14 @@ static int alsa_cap_copy_kernel(struct snd_pcm_substream *substream, struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); int ret; - if (unlikely(pos + count > stream->sh_buf.buffer_sz)) + if (unlikely(pos + count > stream->buffer_sz)) return -EINVAL; ret = xen_snd_front_stream_read(&stream->evt_pair->req, pos, count); if (ret < 0) return ret; - memcpy(dst, stream->sh_buf.buffer + pos, count); + memcpy(dst, stream->buffer + pos, count); return 0; } @@ -622,10 +674,10 @@ static int alsa_pb_fill_silence(struct snd_pcm_substream *substream, { struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); - if (unlikely(pos + count > stream->sh_buf.buffer_sz)) + if (unlikely(pos + count > stream->buffer_sz)) return -EINVAL; - memset(stream->sh_buf.buffer + pos, 0, count); + memset(stream->buffer + pos, 0, count); return xen_snd_front_stream_write(&stream->evt_pair->req, pos, count); } @@ -637,31 +689,31 @@ static int alsa_pb_fill_silence(struct snd_pcm_substream *substream, * to know when the buffer can be transferred to the backend. */ -static struct snd_pcm_ops snd_drv_alsa_playback_ops = { - .open = alsa_open, - .close = alsa_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = alsa_hw_params, - .hw_free = alsa_hw_free, - .prepare = alsa_prepare, - .trigger = alsa_trigger, - .pointer = alsa_pointer, - .copy_user = alsa_pb_copy_user, - .copy_kernel = alsa_pb_copy_kernel, - .fill_silence = alsa_pb_fill_silence, +static const struct snd_pcm_ops snd_drv_alsa_playback_ops = { + .open = alsa_open, + .close = alsa_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = alsa_hw_params, + .hw_free = alsa_hw_free, + .prepare = alsa_prepare, + .trigger = alsa_trigger, + .pointer = alsa_pointer, + .copy_user = alsa_pb_copy_user, + .copy_kernel = alsa_pb_copy_kernel, + .fill_silence = alsa_pb_fill_silence, }; -static struct snd_pcm_ops snd_drv_alsa_capture_ops = { - .open = alsa_open, - .close = alsa_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = alsa_hw_params, - .hw_free = alsa_hw_free, - .prepare = alsa_prepare, - .trigger = alsa_trigger, - .pointer = alsa_pointer, - .copy_user = alsa_cap_copy_user, - .copy_kernel = alsa_cap_copy_kernel, +static const struct snd_pcm_ops snd_drv_alsa_capture_ops = { + .open = alsa_open, + .close = alsa_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = alsa_hw_params, + .hw_free = alsa_hw_free, + .prepare = alsa_prepare, + .trigger = alsa_trigger, + .pointer = alsa_pointer, + .copy_user = alsa_cap_copy_user, + .copy_kernel = alsa_cap_copy_kernel, }; static int new_pcm_instance(struct xen_snd_front_card_info *card_info, diff --git a/sound/xen/xen_snd_front_shbuf.c b/sound/xen/xen_snd_front_shbuf.c deleted file mode 100644 index 07ac176a41ba..000000000000 --- a/sound/xen/xen_snd_front_shbuf.c +++ /dev/null @@ -1,194 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0 OR MIT - -/* - * Xen para-virtual sound device - * - * Copyright (C) 2016-2018 EPAM Systems Inc. - * - * Author: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com> - */ - -#include <linux/kernel.h> -#include <xen/xen.h> -#include <xen/xenbus.h> - -#include "xen_snd_front_shbuf.h" - -grant_ref_t xen_snd_front_shbuf_get_dir_start(struct xen_snd_front_shbuf *buf) -{ - if (!buf->grefs) - return GRANT_INVALID_REF; - - return buf->grefs[0]; -} - -void xen_snd_front_shbuf_clear(struct xen_snd_front_shbuf *buf) -{ - memset(buf, 0, sizeof(*buf)); -} - -void xen_snd_front_shbuf_free(struct xen_snd_front_shbuf *buf) -{ - int i; - - if (buf->grefs) { - for (i = 0; i < buf->num_grefs; i++) - if (buf->grefs[i] != GRANT_INVALID_REF) - gnttab_end_foreign_access(buf->grefs[i], - 0, 0UL); - kfree(buf->grefs); - } - kfree(buf->directory); - free_pages_exact(buf->buffer, buf->buffer_sz); - xen_snd_front_shbuf_clear(buf); -} - -/* - * number of grant references a page can hold with respect to the - * xensnd_page_directory header - */ -#define XENSND_NUM_GREFS_PER_PAGE ((XEN_PAGE_SIZE - \ - offsetof(struct xensnd_page_directory, gref)) / \ - sizeof(grant_ref_t)) - -static void fill_page_dir(struct xen_snd_front_shbuf *buf, - int num_pages_dir) -{ - struct xensnd_page_directory *page_dir; - unsigned char *ptr; - int i, cur_gref, grefs_left, to_copy; - - ptr = buf->directory; - grefs_left = buf->num_grefs - num_pages_dir; - /* - * skip grant references at the beginning, they are for pages granted - * for the page directory itself - */ - cur_gref = num_pages_dir; - for (i = 0; i < num_pages_dir; i++) { - page_dir = (struct xensnd_page_directory *)ptr; - if (grefs_left <= XENSND_NUM_GREFS_PER_PAGE) { - to_copy = grefs_left; - page_dir->gref_dir_next_page = GRANT_INVALID_REF; - } else { - to_copy = XENSND_NUM_GREFS_PER_PAGE; - page_dir->gref_dir_next_page = buf->grefs[i + 1]; - } - - memcpy(&page_dir->gref, &buf->grefs[cur_gref], - to_copy * sizeof(grant_ref_t)); - - ptr += XEN_PAGE_SIZE; - grefs_left -= to_copy; - cur_gref += to_copy; - } -} - -static int grant_references(struct xenbus_device *xb_dev, - struct xen_snd_front_shbuf *buf, - int num_pages_dir, int num_pages_buffer, - int num_grefs) -{ - grant_ref_t priv_gref_head; - unsigned long frame; - int ret, i, j, cur_ref; - int otherend_id; - - ret = gnttab_alloc_grant_references(num_grefs, &priv_gref_head); - if (ret) - return ret; - - buf->num_grefs = num_grefs; - otherend_id = xb_dev->otherend_id; - j = 0; - - for (i = 0; i < num_pages_dir; i++) { - cur_ref = gnttab_claim_grant_reference(&priv_gref_head); - if (cur_ref < 0) { - ret = cur_ref; - goto fail; - } - - frame = xen_page_to_gfn(virt_to_page(buf->directory + - XEN_PAGE_SIZE * i)); - gnttab_grant_foreign_access_ref(cur_ref, otherend_id, frame, 0); - buf->grefs[j++] = cur_ref; - } - - for (i = 0; i < num_pages_buffer; i++) { - cur_ref = gnttab_claim_grant_reference(&priv_gref_head); - if (cur_ref < 0) { - ret = cur_ref; - goto fail; - } - - frame = xen_page_to_gfn(virt_to_page(buf->buffer + - XEN_PAGE_SIZE * i)); - gnttab_grant_foreign_access_ref(cur_ref, otherend_id, frame, 0); - buf->grefs[j++] = cur_ref; - } - - gnttab_free_grant_references(priv_gref_head); - fill_page_dir(buf, num_pages_dir); - return 0; - -fail: - gnttab_free_grant_references(priv_gref_head); - return ret; -} - -static int alloc_int_buffers(struct xen_snd_front_shbuf *buf, - int num_pages_dir, int num_pages_buffer, - int num_grefs) -{ - buf->grefs = kcalloc(num_grefs, sizeof(*buf->grefs), GFP_KERNEL); - if (!buf->grefs) - return -ENOMEM; - - buf->directory = kcalloc(num_pages_dir, XEN_PAGE_SIZE, GFP_KERNEL); - if (!buf->directory) - goto fail; - - buf->buffer_sz = num_pages_buffer * XEN_PAGE_SIZE; - buf->buffer = alloc_pages_exact(buf->buffer_sz, GFP_KERNEL); - if (!buf->buffer) - goto fail; - - return 0; - -fail: - kfree(buf->grefs); - buf->grefs = NULL; - kfree(buf->directory); - buf->directory = NULL; - return -ENOMEM; -} - -int xen_snd_front_shbuf_alloc(struct xenbus_device *xb_dev, - struct xen_snd_front_shbuf *buf, - unsigned int buffer_sz) -{ - int num_pages_buffer, num_pages_dir, num_grefs; - int ret; - - xen_snd_front_shbuf_clear(buf); - - num_pages_buffer = DIV_ROUND_UP(buffer_sz, XEN_PAGE_SIZE); - /* number of pages the page directory consumes itself */ - num_pages_dir = DIV_ROUND_UP(num_pages_buffer, - XENSND_NUM_GREFS_PER_PAGE); - num_grefs = num_pages_buffer + num_pages_dir; - - ret = alloc_int_buffers(buf, num_pages_dir, - num_pages_buffer, num_grefs); - if (ret < 0) - return ret; - - ret = grant_references(xb_dev, buf, num_pages_dir, num_pages_buffer, - num_grefs); - if (ret < 0) - return ret; - - fill_page_dir(buf, num_pages_dir); - return 0; -} diff --git a/sound/xen/xen_snd_front_shbuf.h b/sound/xen/xen_snd_front_shbuf.h deleted file mode 100644 index d28e97c47b2c..000000000000 --- a/sound/xen/xen_snd_front_shbuf.h +++ /dev/null @@ -1,36 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0 OR MIT */ - -/* - * Xen para-virtual sound device - * - * Copyright (C) 2016-2018 EPAM Systems Inc. - * - * Author: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com> - */ - -#ifndef __XEN_SND_FRONT_SHBUF_H -#define __XEN_SND_FRONT_SHBUF_H - -#include <xen/grant_table.h> - -#include "xen_snd_front_evtchnl.h" - -struct xen_snd_front_shbuf { - int num_grefs; - grant_ref_t *grefs; - u8 *directory; - u8 *buffer; - size_t buffer_sz; -}; - -grant_ref_t xen_snd_front_shbuf_get_dir_start(struct xen_snd_front_shbuf *buf); - -int xen_snd_front_shbuf_alloc(struct xenbus_device *xb_dev, - struct xen_snd_front_shbuf *buf, - unsigned int buffer_sz); - -void xen_snd_front_shbuf_clear(struct xen_snd_front_shbuf *buf); - -void xen_snd_front_shbuf_free(struct xen_snd_front_shbuf *buf); - -#endif /* __XEN_SND_FRONT_SHBUF_H */ |