diff options
Diffstat (limited to 'sound/pci/hda')
| -rw-r--r-- | sound/pci/hda/Makefile | 7 | ||||
| -rw-r--r-- | sound/pci/hda/hda_codec.c | 1856 | ||||
| -rw-r--r-- | sound/pci/hda/hda_codec.h | 604 | ||||
| -rw-r--r-- | sound/pci/hda/hda_generic.c | 906 | ||||
| -rw-r--r-- | sound/pci/hda/hda_intel.c | 1449 | ||||
| -rw-r--r-- | sound/pci/hda/hda_local.h | 161 | ||||
| -rw-r--r-- | sound/pci/hda/hda_patch.h | 17 | ||||
| -rw-r--r-- | sound/pci/hda/hda_proc.c | 298 | ||||
| -rw-r--r-- | sound/pci/hda/patch_analog.c | 445 | ||||
| -rw-r--r-- | sound/pci/hda/patch_cmedia.c | 621 | ||||
| -rw-r--r-- | sound/pci/hda/patch_realtek.c | 1503 |
11 files changed, 7867 insertions, 0 deletions
diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile new file mode 100644 index 000000000000..570a59d33b41 --- /dev/null +++ b/sound/pci/hda/Makefile @@ -0,0 +1,7 @@ +snd-hda-intel-objs := hda_intel.o +snd-hda-codec-objs := hda_codec.o hda_generic.o patch_realtek.o patch_cmedia.o patch_analog.o +ifdef CONFIG_PROC_FS +snd-hda-codec-objs += hda_proc.o +endif + +obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-intel.o snd-hda-codec.o diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c new file mode 100644 index 000000000000..9ed117ac0c09 --- /dev/null +++ b/sound/pci/hda/hda_codec.c @@ -0,0 +1,1856 @@ +/* + * Universal Interface for Intel High Definition Audio Codec + * + * Copyright (c) 2004 Takashi Iwai <tiwai@suse.de> + * + * + * This driver 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 driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <sound/driver.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/pci.h> +#include <linux/moduleparam.h> +#include <sound/core.h> +#include "hda_codec.h" +#include <sound/asoundef.h> +#include <sound/initval.h> +#include "hda_local.h" + + +MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>"); +MODULE_DESCRIPTION("Universal interface for High Definition Audio Codec"); +MODULE_LICENSE("GPL"); + + +/* + * vendor / preset table + */ + +struct hda_vendor_id { + unsigned int id; + const char *name; +}; + +/* codec vendor labels */ +static struct hda_vendor_id hda_vendor_ids[] = { + { 0x10ec, "Realtek" }, + { 0x13f6, "C-Media" }, + { 0x434d, "C-Media" }, + {} /* terminator */ +}; + +/* codec presets */ +#include "hda_patch.h" + + +/** + * snd_hda_codec_read - send a command and get the response + * @codec: the HDA codec + * @nid: NID to send the command + * @direct: direct flag + * @verb: the verb to send + * @parm: the parameter for the verb + * + * Send a single command and read the corresponding response. + * + * Returns the obtained response value, or -1 for an error. + */ +unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, int direct, + unsigned int verb, unsigned int parm) +{ + unsigned int res; + down(&codec->bus->cmd_mutex); + if (! codec->bus->ops.command(codec, nid, direct, verb, parm)) + res = codec->bus->ops.get_response(codec); + else + res = (unsigned int)-1; + up(&codec->bus->cmd_mutex); + return res; +} + +/** + * snd_hda_codec_write - send a single command without waiting for response + * @codec: the HDA codec + * @nid: NID to send the command + * @direct: direct flag + * @verb: the verb to send + * @parm: the parameter for the verb + * + * Send a single command without waiting for response. + * + * Returns 0 if successful, or a negative error code. + */ +int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct, + unsigned int verb, unsigned int parm) +{ + int err; + down(&codec->bus->cmd_mutex); + err = codec->bus->ops.command(codec, nid, direct, verb, parm); + up(&codec->bus->cmd_mutex); + return err; +} + +/** + * snd_hda_sequence_write - sequence writes + * @codec: the HDA codec + * @seq: VERB array to send + * + * Send the commands sequentially from the given array. + * The array must be terminated with NID=0. + */ +void snd_hda_sequence_write(struct hda_codec *codec, const struct hda_verb *seq) +{ + for (; seq->nid; seq++) + snd_hda_codec_write(codec, seq->nid, 0, seq->verb, seq->param); +} + +/** + * snd_hda_get_sub_nodes - get the range of sub nodes + * @codec: the HDA codec + * @nid: NID to parse + * @start_id: the pointer to store the start NID + * + * Parse the NID and store the start NID of its sub-nodes. + * Returns the number of sub-nodes. + */ +int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid, hda_nid_t *start_id) +{ + unsigned int parm; + + parm = snd_hda_param_read(codec, nid, AC_PAR_NODE_COUNT); + *start_id = (parm >> 16) & 0x7fff; + return (int)(parm & 0x7fff); +} + +/** + * snd_hda_get_connections - get connection list + * @codec: the HDA codec + * @nid: NID to parse + * @conn_list: connection list array + * @max_conns: max. number of connections to store + * + * Parses the connection list of the given widget and stores the list + * of NIDs. + * + * Returns the number of connections, or a negative error code. + */ +int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, + hda_nid_t *conn_list, int max_conns) +{ + unsigned int parm; + int i, j, conn_len, num_tupples, conns; + unsigned int shift, num_elems, mask; + + snd_assert(conn_list && max_conns > 0, return -EINVAL); + + parm = snd_hda_param_read(codec, nid, AC_PAR_CONNLIST_LEN); + if (parm & AC_CLIST_LONG) { + /* long form */ + shift = 16; + num_elems = 2; + } else { + /* short form */ + shift = 8; + num_elems = 4; + } + conn_len = parm & AC_CLIST_LENGTH; + num_tupples = num_elems / 2; + mask = (1 << (shift-1)) - 1; + + if (! conn_len) + return 0; /* no connection */ + + if (conn_len == 1) { + /* single connection */ + parm = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_LIST, 0); + conn_list[0] = parm & mask; + return 1; + } + + /* multi connection */ + conns = 0; + for (i = 0; i < conn_len; i += num_elems) { + parm = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_LIST, i); + for (j = 0; j < num_tupples; j++) { + int range_val; + hda_nid_t val1, val2, n; + range_val = parm & (1 << (shift-1)); /* ranges */ + val1 = parm & mask; + parm >>= shift; + val2 = parm & mask; + parm >>= shift; + if (range_val) { + /* ranges between val1 and val2 */ + if (val1 > val2) { + snd_printk(KERN_WARNING "hda_codec: invalid dep_range_val %x:%x\n", val1, val2); + continue; + } + for (n = val1; n <= val2; n++) { + if (conns >= max_conns) + return -EINVAL; + conn_list[conns++] = n; + } + } else { + if (! val1) + break; + if (conns >= max_conns) + return -EINVAL; + conn_list[conns++] = val1; + if (! val2) + break; + if (conns >= max_conns) + return -EINVAL; + conn_list[conns++] = val2; + } + } + } + return conns; +} + + +/** + * snd_hda_queue_unsol_event - add an unsolicited event to queue + * @bus: the BUS + * @res: unsolicited event (lower 32bit of RIRB entry) + * @res_ex: codec addr and flags (upper 32bit or RIRB entry) + * + * Adds the given event to the queue. The events are processed in + * the workqueue asynchronously. Call this function in the interrupt + * hanlder when RIRB receives an unsolicited event. + * + * Returns 0 if successful, or a negative error code. + */ +int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex) +{ + struct hda_bus_unsolicited *unsol; + unsigned int wp; + + if ((unsol = bus->unsol) == NULL) + return 0; + + wp = (unsol->wp + 1) % HDA_UNSOL_QUEUE_SIZE; + unsol->wp = wp; + + wp <<= 1; + unsol->queue[wp] = res; + unsol->queue[wp + 1] = res_ex; + + queue_work(unsol->workq, &unsol->work); + + return 0; +} + +/* + * process queueud unsolicited events + */ +static void process_unsol_events(void *data) +{ + struct hda_bus *bus = data; + struct hda_bus_unsolicited *unsol = bus->unsol; + struct hda_codec *codec; + unsigned int rp, caddr, res; + + while (unsol->rp != unsol->wp) { + rp = (unsol->rp + 1) % HDA_UNSOL_QUEUE_SIZE; + unsol->rp = rp; + rp <<= 1; + res = unsol->queue[rp]; + caddr = unsol->queue[rp + 1]; + if (! (caddr & (1 << 4))) /* no unsolicited event? */ + continue; + codec = bus->caddr_tbl[caddr & 0x0f]; + if (codec && codec->patch_ops.unsol_event) + codec->patch_ops.unsol_event(codec, res); + } +} + +/* + * initialize unsolicited queue + */ +static int init_unsol_queue(struct hda_bus *bus) +{ + struct hda_bus_unsolicited *unsol; + + unsol = kcalloc(1, sizeof(*unsol), GFP_KERNEL); + if (! unsol) { + snd_printk(KERN_ERR "hda_codec: can't allocate unsolicited queue\n"); + return -ENOMEM; + } + unsol->workq = create_workqueue("hda_codec"); + if (! unsol->workq) { + snd_printk(KERN_ERR "hda_codec: can't create workqueue\n"); + kfree(unsol); + return -ENOMEM; + } + INIT_WORK(&unsol->work, process_unsol_events, bus); + bus->unsol = unsol; + return 0; +} + +/* + * destructor + */ +static void snd_hda_codec_free(struct hda_codec *codec); + +static int snd_hda_bus_free(struct hda_bus *bus) +{ + struct list_head *p, *n; + + if (! bus) + return 0; + if (bus->unsol) { + destroy_workqueue(bus->unsol->workq); + kfree(bus->unsol); + } + list_for_each_safe(p, n, &bus->codec_list) { + struct hda_codec *codec = list_entry(p, struct hda_codec, list); + snd_hda_codec_free(codec); + } + if (bus->ops.private_free) + bus->ops.private_free(bus); + kfree(bus); + return 0; +} + +static int snd_hda_bus_dev_free(snd_device_t *device) +{ + struct hda_bus *bus = device->device_data; + return snd_hda_bus_free(bus); +} + +/** + * snd_hda_bus_new - create a HDA bus + * @card: the card entry + * @temp: the template for hda_bus information + * @busp: the pointer to store the created bus instance + * + * Returns 0 if successful, or a negative error code. + */ +int snd_hda_bus_new(snd_card_t *card, const struct hda_bus_template *temp, + struct hda_bus **busp) +{ + struct hda_bus *bus; + int err; + static snd_device_ops_t dev_ops = { + .dev_free = snd_hda_bus_dev_free, + }; + + snd_assert(temp, return -EINVAL); + snd_assert(temp->ops.command && temp->ops.get_response, return -EINVAL); + + if (busp) + *busp = NULL; + + bus = kcalloc(1, sizeof(*bus), GFP_KERNEL); + if (bus == NULL) { + snd_printk(KERN_ERR "can't allocate struct hda_bus\n"); + return -ENOMEM; + } + + bus->card = card; + bus->private_data = temp->private_data; + bus->pci = temp->pci; + bus->modelname = temp->modelname; + bus->ops = temp->ops; + + init_MUTEX(&bus->cmd_mutex); + INIT_LIST_HEAD(&bus->codec_list); + + init_unsol_queue(bus); + + if ((err = snd_device_new(card, SNDRV_DEV_BUS, bus, &dev_ops)) < 0) { + snd_hda_bus_free(bus); + return err; + } + if (busp) + *busp = bus; + return 0; +} + + +/* + * find a matching codec preset + */ +static const struct hda_codec_preset *find_codec_preset(struct hda_codec *codec) +{ + const struct hda_codec_preset **tbl, *preset; + + for (tbl = hda_preset_tables; *tbl; tbl++) { + for (preset = *tbl; preset->id; preset++) { + u32 mask = preset->mask; + if (! mask) + mask = ~0; + if (preset->id == (codec->vendor_id & mask)) + return preset; + } + } + return NULL; +} + +/* + * snd_hda_get_codec_name - store the codec name + */ +void snd_hda_get_codec_name(struct hda_codec *codec, + char *name, int namelen) +{ + const struct hda_vendor_id *c; + const char *vendor = NULL; + u16 vendor_id = codec->vendor_id >> 16; + char tmp[16]; + + for (c = hda_vendor_ids; c->id; c++) { + if (c->id == vendor_id) { + vendor = c->name; + break; + } + } + if (! vendor) { + sprintf(tmp, "Generic %04x", vendor_id); + vendor = tmp; + } + if (codec->preset && codec->preset->name) + snprintf(name, namelen, "%s %s", vendor, codec->preset->name); + else + snprintf(name, namelen, "%s ID %x", vendor, codec->vendor_id & 0xffff); +} + +/* + * look for an AFG node + * + * return 0 if not found + */ +static int look_for_afg_node(struct hda_codec *codec) +{ + int i, total_nodes; + hda_nid_t nid; + + total_nodes = snd_hda_get_sub_nodes(codec, AC_NODE_ROOT, &nid); + for (i = 0; i < total_nodes; i++, nid++) { + if ((snd_hda_param_read(codec, nid, AC_PAR_FUNCTION_TYPE) & 0xff) == + AC_GRP_AUDIO_FUNCTION) + return nid; + } + return 0; +} + +/* + * codec destructor + */ +static void snd_hda_codec_free(struct hda_codec *codec) +{ + if (! codec) + return; + list_del(&codec->list); + codec->bus->caddr_tbl[codec->addr] = NULL; + if (codec->patch_ops.free) + codec->patch_ops.free(codec); + kfree(codec); +} + +static void init_amp_hash(struct hda_codec *codec); + +/** + * snd_hda_codec_new - create a HDA codec + * @bus: the bus to assign + * @codec_addr: the codec address + * @codecp: the pointer to store the generated codec + * + * Returns 0 if successful, or a negative error code. + */ +int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, + struct hda_codec **codecp) +{ + struct hda_codec *codec; + char component[13]; + int err; + + snd_assert(bus, return -EINVAL); + snd_assert(codec_addr <= HDA_MAX_CODEC_ADDRESS, return -EINVAL); + + if (bus->caddr_tbl[codec_addr]) { + snd_printk(KERN_ERR "hda_codec: address 0x%x is already occupied\n", codec_addr); + return -EBUSY; + } + + codec = kcalloc(1, sizeof(*codec), GFP_KERNEL); + if (codec == NULL) { + snd_printk(KERN_ERR "can't allocate struct hda_codec\n"); + return -ENOMEM; + } + + codec->bus = bus; + codec->addr = codec_addr; + init_MUTEX(&codec->spdif_mutex); + init_amp_hash(codec); + + list_add_tail(&codec->list, &bus->codec_list); + bus->caddr_tbl[codec_addr] = codec; + + codec->vendor_id = snd_hda_param_read(codec, AC_NODE_ROOT, AC_PAR_VENDOR_ID); + codec->subsystem_id = snd_hda_param_read(codec, AC_NODE_ROOT, AC_PAR_SUBSYSTEM_ID); + codec->revision_id = snd_hda_param_read(codec, AC_NODE_ROOT, AC_PAR_REV_ID); + + /* FIXME: support for multiple AFGs? */ + codec->afg = look_for_afg_node(codec); + if (! codec->afg) { + snd_printk(KERN_ERR "hda_codec: no AFG node found\n"); + snd_hda_codec_free(codec); + return -ENODEV; + } + + codec->preset = find_codec_preset(codec); + if (! *bus->card->mixername) + snd_hda_get_codec_name(codec, bus->card->mixername, + sizeof(bus->card->mixername)); + + if (codec->preset && codec->preset->patch) + err = codec->preset->patch(codec); + else + err = snd_hda_parse_generic_codec(codec); + if (err < 0) { + snd_hda_codec_free(codec); + return err; + } + + snd_hda_codec_proc_new(codec); + + sprintf(component, "HDA:%08x", codec->vendor_id); + snd_component_add(codec->bus->card, component); + + if (codecp) + *codecp = codec; + return 0; +} + +/** + * snd_hda_codec_setup_stream - set up the codec for streaming + * @codec: the CODEC to set up + * @nid: the NID to set up + * @stream_tag: stream tag to pass, it's between 0x1 and 0xf. + * @channel_id: channel id to pass, zero based. + * @format: stream format. + */ +void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, u32 stream_tag, + int channel_id, int format) +{ + snd_printdd("hda_codec_setup_stream: NID=0x%x, stream=0x%x, channel=%d, format=0x%x\n", + nid, stream_tag, channel_id, format); + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, + (stream_tag << 4) | channel_id); + msleep(1); + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, format); +} + + +/* + * amp access functions + */ + +#define HDA_HASH_KEY(nid,dir,idx) (u32)((nid) + (idx) * 32 + (dir) * 64) +#define INFO_AMP_CAPS (1<<0) +#define INFO_AMP_VOL (1<<1) + +/* initialize the hash table */ +static void init_amp_hash(struct hda_codec *codec) +{ + memset(codec->amp_hash, 0xff, sizeof(codec->amp_hash)); + codec->num_amp_entries = 0; +} + +/* query the hash. allocate an entry if not found. */ +static struct hda_amp_info *get_alloc_amp_hash(struct hda_codec *codec, u32 key) +{ + u16 idx = key % (u16)ARRAY_SIZE(codec->amp_hash); + u16 cur = codec->amp_hash[idx]; + struct hda_amp_info *info; + + while (cur != 0xffff) { + info = &codec->amp_info[cur]; + if (info->key == key) + return info; + cur = info->next; + } + + /* add a new hash entry */ + if (codec->num_amp_entries >= ARRAY_SIZE(codec->amp_info)) { + snd_printk(KERN_ERR "hda_codec: Tooooo many amps!\n"); + return NULL; + } + cur = codec->num_amp_entries++; + info = &codec->amp_info[cur]; + info->key = key; + info->status = 0; /* not initialized yet */ + info->next = codec->amp_hash[idx]; + codec->amp_hash[idx] = cur; + + return info; +} + +/* + * query AMP capabilities for the given widget and direction + */ +static u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction) +{ + struct hda_amp_info *info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, 0)); + + if (! info) + return 0; + if (! (info->status & INFO_AMP_CAPS)) { + if (!(snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP) & AC_WCAP_AMP_OVRD)) + nid = codec->afg; + info->amp_caps = snd_hda_param_read(codec, nid, direction == HDA_OUTPUT ? + AC_PAR_AMP_OUT_CAP : AC_PAR_AMP_IN_CAP); + info->status |= INFO_AMP_CAPS; + } + return info->amp_caps; +} + +/* + * read the current volume to info + * if the cache exists, read from the cache. + */ +static void get_vol_mute(struct hda_codec *codec, struct hda_amp_info *info, + hda_nid_t nid, int ch, int direction, int index) +{ + u32 val, parm; + + if (info->status & (INFO_AMP_VOL << ch)) + return; + + parm = ch ? AC_AMP_GET_RIGHT : AC_AMP_GET_LEFT; + parm |= direction == HDA_OUTPUT ? AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT; + parm |= index; + val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_AMP_GAIN_MUTE, parm); + info->vol[ch] = val & 0xff; + info->status |= INFO_AMP_VOL << ch; +} + +/* + * write the current volume in info to the h/w + */ +static void put_vol_mute(struct hda_codec *codec, + hda_nid_t nid, int ch, int direction, int index, int val) +{ + u32 parm; + + parm = ch ? AC_AMP_SET_RIGHT : AC_AMP_SET_LEFT; + parm |= direction == HDA_OUTPUT ? AC_AMP_SET_OUTPUT : AC_AMP_SET_INPUT; + parm |= index << AC_AMP_SET_INDEX_SHIFT; + parm |= val; + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, parm); +} + +/* + * read/write AMP value. The volume is between 0 to 0x7f, 0x80 = mute bit. + */ +int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int index) +{ + struct hda_amp_info *info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, index)); + if (! info) + return 0; + get_vol_mute(codec, info, nid, ch, direction, index); + return info->vol[ch]; +} + +int snd_hda_codec_amp_write(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int idx, int val) +{ + struct hda_amp_info *info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, idx)); + if (! info) + return 0; + get_vol_mute(codec, info, nid, ch, direction, idx); + if (info->vol[ch] == val && ! codec->in_resume) + return 0; + put_vol_mute(codec, nid, ch, direction, idx, val); + info->vol[ch] = val; + return 1; +} + + +/* + * AMP control callbacks + */ +/* retrieve parameters from private_value */ +#define get_amp_nid(kc) ((kc)->private_value & 0xffff) +#define get_amp_channels(kc) (((kc)->private_value >> 16) & 0x3) +#define get_amp_direction(kc) (((kc)->private_value >> 18) & 0x1) +#define get_amp_index(kc) (((kc)->private_value >> 19) & 0xf) + +/* volume */ +int snd_hda_mixer_amp_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + u16 nid = get_amp_nid(kcontrol); + u8 chs = get_amp_channels(kcontrol); + int dir = get_amp_direction(kcontrol); + u32 caps; + + caps = query_amp_caps(codec, nid, dir); + caps = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT; /* num steps */ + if (! caps) { + printk(KERN_WARNING "hda_codec: num_steps = 0 for NID=0x%x\n", nid); + return -EINVAL; + } + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = chs == 3 ? 2 : 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = caps; + return 0; +} + +int snd_hda_mixer_amp_volume_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = get_amp_nid(kcontrol); + int chs = get_amp_channels(kcontrol); + int dir = get_amp_direction(kcontrol); + int idx = get_amp_index(kcontrol); + long *valp = ucontrol->value.integer.value; + + if (chs & 1) + *valp++ = snd_hda_codec_amp_read(codec, nid, 0, dir, idx) & 0x7f; + if (chs & 2) + *valp = snd_hda_codec_amp_read(codec, nid, 1, dir, idx) & 0x7f; + return 0; +} + +int snd_hda_mixer_amp_volume_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = get_amp_nid(kcontrol); + int chs = get_amp_channels(kcontrol); + int dir = get_amp_direction(kcontrol); + int idx = get_amp_index(kcontrol); + int val; + long *valp = ucontrol->value.integer.value; + int change = 0; + + if (chs & 1) { + val = *valp & 0x7f; + val |= snd_hda_codec_amp_read(codec, nid, 0, dir, idx) & 0x80; + change = snd_hda_codec_amp_write(codec, nid, 0, dir, idx, val); + valp++; + } + if (chs & 2) { + val = *valp & 0x7f; + val |= snd_hda_codec_amp_read(codec, nid, 1, dir, idx) & 0x80; + change |= snd_hda_codec_amp_write(codec, nid, 1, dir, idx, val); + } + return change; +} + +/* switch */ +int snd_hda_mixer_amp_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) +{ + int chs = get_amp_channels(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = chs == 3 ? 2 : 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +int snd_hda_mixer_amp_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = get_amp_nid(kcontrol); + int chs = get_amp_channels(kcontrol); + int dir = get_amp_direction(kcontrol); + int idx = get_amp_index(kcontrol); + long *valp = ucontrol->value.integer.value; + + if (chs & 1) + *valp++ = (snd_hda_codec_amp_read(codec, nid, 0, dir, idx) & 0x80) ? 0 : 1; + if (chs & 2) + *valp = (snd_hda_codec_amp_read(codec, nid, 1, dir, idx) & 0x80) ? 0 : 1; + return 0; +} + +int snd_hda_mixer_amp_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = get_amp_nid(kcontrol); + int chs = get_amp_channels(kcontrol); + int dir = get_amp_direction(kcontrol); + int idx = get_amp_index(kcontrol); + int val; + long *valp = ucontrol->value.integer.value; + int change = 0; + + if (chs & 1) { + val = snd_hda_codec_amp_read(codec, nid, 0, dir, idx) & 0x7f; + val |= *valp ? 0 : 0x80; + change = snd_hda_codec_amp_write(codec, nid, 0, dir, idx, val); + valp++; + } + if (chs & 2) { + val = snd_hda_codec_amp_read(codec, nid, 1, dir, idx) & 0x7f; + val |= *valp ? 0 : 0x80; + change = snd_hda_codec_amp_write(codec, nid, 1, dir, idx, val); + } + return change; +} + +/* + * SPDIF out controls + */ + +static int snd_hda_spdif_mask_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + return 0; +} + +static int snd_hda_spdif_cmask_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + ucontrol->value.iec958.status[0] = IEC958_AES0_PROFESSIONAL | + IEC958_AES0_NONAUDIO | + IEC958_AES0_CON_EMPHASIS_5015 | + IEC958_AES0_CON_NOT_COPYRIGHT; + ucontrol->value.iec958.status[1] = IEC958_AES1_CON_CATEGORY | + IEC958_AES1_CON_ORIGINAL; + return 0; +} + +static int snd_hda_spdif_pmask_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + ucontrol->value.iec958.status[0] = IEC958_AES0_PROFESSIONAL | + IEC958_AES0_NONAUDIO | + IEC958_AES0_PRO_EMPHASIS_5015; + return 0; +} + +static int snd_hda_spdif_default_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + + ucontrol->value.iec958.status[0] = codec->spdif_status & 0xff; + ucontrol->value.iec958.status[1] = (codec->spdif_status >> 8) & 0xff; + ucontrol->value.iec958.status[2] = (codec->spdif_status >> 16) & 0xff; + ucontrol->value.iec958.status[3] = (codec->spdif_status >> 24) & 0xff; + + return 0; +} + +/* convert from SPDIF status bits to HDA SPDIF bits + * bit 0 (DigEn) is always set zero (to be filled later) + */ +static unsigned short convert_from_spdif_status(unsigned int sbits) +{ + unsigned short val = 0; + + if (sbits & IEC958_AES0_PROFESSIONAL) + val |= 1 << 6; + if (sbits & IEC958_AES0_NONAUDIO) + val |= 1 << 5; + if (sbits & IEC958_AES0_PROFESSIONAL) { + if ((sbits & IEC958_AES0_PRO_EMPHASIS) == IEC958_AES0_PRO_EMPHASIS_5015) + val |= 1 << 3; + } else { + if ((sbits & IEC958_AES0_CON_EMPHASIS) == IEC958_AES0_CON_EMPHASIS_5015) + val |= 1 << 3; + if (! (sbits & IEC958_AES0_CON_NOT_COPYRIGHT)) + val |= 1 << 4; + if (sbits & (IEC958_AES1_CON_ORIGINAL << 8)) + val |= 1 << 7; + val |= sbits & (IEC958_AES1_CON_CATEGORY << 8); + } + return val; +} + +/* convert to SPDIF status bits from HDA SPDIF bits + */ +static unsigned int convert_to_spdif_status(unsigned short val) +{ + unsigned int sbits = 0; + + if (val & (1 << 5)) + sbits |= IEC958_AES0_NONAUDIO; + if (val & (1 << 6)) + sbits |= IEC958_AES0_PROFESSIONAL; + if (sbits & IEC958_AES0_PROFESSIONAL) { + if (sbits & (1 << 3)) + sbits |= IEC958_AES0_PRO_EMPHASIS_5015; + } else { + if (val & (1 << 3)) + sbits |= IEC958_AES0_CON_EMPHASIS_5015; + if (! (val & (1 << 4))) + sbits |= IEC958_AES0_CON_NOT_COPYRIGHT; + if (val & (1 << 7)) + sbits |= (IEC958_AES1_CON_ORIGINAL << 8); + sbits |= val & (0x7f << 8); + } + return sbits; +} + +static int snd_hda_spdif_default_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value; + unsigned short val; + int change; + + down(&codec->spdif_mutex); + codec->spdif_status = ucontrol->value.iec958.status[0] | + ((unsigned int)ucontrol->value.iec958.status[1] << 8) | + ((unsigned int)ucontrol->value.iec958.status[2] << 16) | + ((unsigned int)ucontrol->value.iec958.status[3] << 24); + val = convert_from_spdif_status(codec->spdif_status); + val |= codec->spdif_ctls & 1; + change = codec->spdif_ctls != val; + codec->spdif_ctls = val; + + if (change || codec->in_resume) { + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, val & 0xff); + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_2, val >> 8); + } + + up(&codec->spdif_mutex); + return change; +} + +static int snd_hda_spdif_out_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int snd_hda_spdif_out_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = codec->spdif_ctls & 1; + return 0; +} + +static int snd_hda_spdif_out_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value; + unsigned short val; + int change; + + down(&codec->spdif_mutex); + val = codec->spdif_ctls & ~1; + if (ucontrol->value.integer.value[0]) + val |= 1; + change = codec->spdif_ctls != val; + if (change || codec->in_resume) { + codec->spdif_ctls = val; + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, val & 0xff); + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, + AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | + AC_AMP_SET_OUTPUT | ((val & 1) ? 0 : 0x80)); + } + up(&codec->spdif_mutex); + return change; +} + +static snd_kcontrol_new_t dig_mixes[] = { + { + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK), + .info = snd_hda_spdif_mask_info, + .get = snd_hda_spdif_cmask_get, + }, + { + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PRO_MASK), + .info = snd_hda_spdif_mask_info, + .get = snd_hda_spdif_pmask_get, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), + .info = snd_hda_spdif_mask_info, + .get = snd_hda_spdif_default_get, + .put = snd_hda_spdif_default_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH), + .info = snd_hda_spdif_out_switch_info, + .get = snd_hda_spdif_out_switch_get, + .put = snd_hda_spdif_out_switch_put, + }, + { } /* end */ +}; + +/** + * snd_hda_create_spdif_out_ctls - create Output SPDIF-related controls + * @codec: the HDA codec + * @nid: audio out widget NID + * + * Creates controls related with the SPDIF output. + * Called from each patch supporting the SPDIF out. + * + * Returns 0 if successful, or a negative error code. + */ +int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid) +{ + int err; + snd_kcontrol_t *kctl; + snd_kcontrol_new_t *dig_mix; + + for (dig_mix = dig_mixes; dig_mix->name; dig_mix++) { + kctl = snd_ctl_new1(dig_mix, codec); + kctl->private_value = nid; + if ((err = snd_ctl_add(codec->bus->card, kctl)) < 0) + return err; + } + codec->spdif_ctls = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT, 0); + codec->spdif_status = convert_to_spdif_status(codec->spdif_ctls); + return 0; +} + +/* + * SPDIF input + */ + +#define snd_hda_spdif_in_switch_info snd_hda_spdif_out_switch_info + +static int snd_hda_spdif_in_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = codec->spdif_in_enable; + return 0; +} + +static int snd_hda_spdif_in_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value; + unsigned int val = !!ucontrol->value.integer.value[0]; + int change; + + down(&codec->spdif_mutex); + change = codec->spdif_in_enable != val; + if (change || codec->in_resume) { + codec->spdif_in_enable = val; + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, val); + } + up(&codec->spdif_mutex); + return change; +} + +static int snd_hda_spdif_in_status_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value; + unsigned short val; + unsigned int sbits; + + val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT, 0); + sbits = convert_to_spdif_status(val); + ucontrol->value.iec958.status[0] = sbits; + ucontrol->value.iec958.status[1] = sbits >> 8; + ucontrol->value.iec958.status[2] = sbits >> 16; + ucontrol->value.iec958.status[3] = sbits >> 24; + return 0; +} + +static snd_kcontrol_new_t dig_in_ctls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), + .info = snd_hda_spdif_in_switch_info, + .get = snd_hda_spdif_in_switch_get, + .put = snd_hda_spdif_in_switch_put, + }, + { + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,DEFAULT), + .info = snd_hda_spdif_mask_info, + .get = snd_hda_spdif_in_status_get, + }, + { } /* end */ +}; + +/** + * snd_hda_create_spdif_in_ctls - create Input SPDIF-related controls + * @codec: the HDA codec + * @nid: audio in widget NID + * + * Creates controls related with the SPDIF input. + * Called from each patch supporting the SPDIF in. + * + * Returns 0 if successful, or a negative error code. + */ +int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid) +{ + int err; + snd_kcontrol_t *kctl; + snd_kcontrol_new_t *dig_mix; + + for (dig_mix = dig_in_ctls; dig_mix->name; dig_mix++) { + kctl = snd_ctl_new1(dig_mix, codec); + kctl->private_value = nid; + if ((err = snd_ctl_add(codec->bus->card, kctl)) < 0) + return err; + } + codec->spdif_in_enable = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT, 0) & 1; + return 0; +} + + +/** + * snd_hda_build_controls - build mixer controls + * @bus: the BUS + * + * Creates mixer controls for each codec included in the bus. + * + * Returns 0 if successful, otherwise a negative error code. + */ +int snd_hda_build_controls(struct hda_bus *bus) +{ + struct list_head *p; + + /* build controls */ + list_for_each(p, &bus->codec_list) { + struct hda_codec *codec = list_entry(p, struct hda_codec, list); + int err; + if (! codec->patch_ops.build_controls) + continue; + err = codec->patch_ops.build_controls(codec); + if (err < 0) + return err; + } + + /* initialize */ + list_for_each(p, &bus->codec_list) { + struct hda_codec *codec = list_entry(p, struct hda_codec, list); + int err; + if (! codec->patch_ops.init) + continue; + err = codec->patch_ops.init(codec); + if (err < 0) + return err; + } + return 0; +} + + +/* + * stream formats + */ +static unsigned int rate_bits[][3] = { + /* rate in Hz, ALSA rate bitmask, HDA format value */ + { 8000, SNDRV_PCM_RATE_8000, 0x0500 }, /* 1/6 x 48 */ + { 11025, SNDRV_PCM_RATE_11025, 0x4300 }, /* 1/4 x 44 */ + { 16000, SNDRV_PCM_RATE_16000, 0x0200 }, /* 1/3 x 48 */ + { 22050, SNDRV_PCM_RATE_22050, 0x4100 }, /* 1/2 x 44 */ + { 32000, SNDRV_PCM_RATE_32000, 0x0a00 }, /* 2/3 x 48 */ + { 44100, SNDRV_PCM_RATE_44100, 0x4000 }, /* 44 */ + { 48000, SNDRV_PCM_RATE_48000, 0x0000 }, /* 48 */ + { 88200, SNDRV_PCM_RATE_88200, 0x4800 }, /* 2 x 44 */ + { 96000, SNDRV_PCM_RATE_96000, 0x0800 }, /* 2 x 48 */ + { 176400, SNDRV_PCM_RATE_176400, 0x5800 },/* 4 x 44 */ + { 192000, SNDRV_PCM_RATE_192000, 0x1800 }, /* 4 x 48 */ + { 0 } +}; + +/** + * snd_hda_calc_stream_format - calculate format bitset + * @rate: the sample rate + * @channels: the number of channels + * @format: the PCM format (SNDRV_PCM_FORMAT_XXX) + * @maxbps: the max. bps + * + * Calculate the format bitset from the given rate, channels and th PCM format. + * + * Return zero if invalid. + */ +unsigned int snd_hda_calc_stream_format(unsigned int rate, + unsigned int channels, + unsigned int format, + unsigned int maxbps) +{ + int i; + unsigned int val = 0; + + for (i = 0; rate_bits[i][0]; i++) + if (rate_bits[i][0] == rate) { + val = rate_bits[i][2]; + break; + } + if (! rate_bits[i][0]) { + snd_printdd("invalid rate %d\n", rate); + return 0; + } + + if (channels == 0 || channels > 8) { + snd_printdd("invalid channels %d\n", channels); + return 0; + } + val |= channels - 1; + + switch (snd_pcm_format_width(format)) { + case 8: val |= 0x00; break; + case 16: val |= 0x10; break; + case 20: + case 24: + case 32: + if (maxbps >= 32) + val |= 0x40; + else if (maxbps >= 24) + val |= 0x30; + else + val |= 0x20; + break; + default: + snd_printdd("invalid format width %d\n", snd_pcm_format_width(format)); + return 0; + } + + return val; +} + +/** + * snd_hda_query_supported_pcm - query the supported PCM rates and formats + * @codec: the HDA codec + * @nid: NID to query + * @ratesp: the pointer to store the detected rate bitflags + * @formatsp: the pointer to store the detected formats + * @bpsp: the pointer to store the detected format widths + * + * Queries the supported PCM rates and formats. The NULL @ratesp, @formatsp + * or @bsps argument is ignored. + * + * Returns 0 if successful, otherwise a negative error code. + */ +int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, + u32 *ratesp, u64 *formatsp, unsigned int *bpsp) +{ + int i; + unsigned int val, streams; + + val = 0; + if (nid != codec->afg && + snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP) & AC_WCAP_FORMAT_OVRD) { + val = snd_hda_param_read(codec, nid, AC_PAR_PCM); + if (val == -1) + return -EIO; + } + if (! val) + val = snd_hda_param_read(codec, codec->afg, AC_PAR_PCM); + + if (ratesp) { + u32 rates = 0; + for (i = 0; rate_bits[i][0]; i++) { + if (val & (1 << i)) + rates |= rate_bits[i][1]; + } + *ratesp = rates; + } + + if (formatsp || bpsp) { + u64 formats = 0; + unsigned int bps; + unsigned int wcaps; + + wcaps = snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP); + streams = snd_hda_param_read(codec, nid, AC_PAR_STREAM); + if (streams == -1) + return -EIO; + if (! streams) { + streams = snd_hda_param_read(codec, codec->afg, AC_PAR_STREAM); + if (streams == -1) + return -EIO; + } + + bps = 0; + if (streams & AC_SUPFMT_PCM) { + if (val & AC_SUPPCM_BITS_8) { + formats |= SNDRV_PCM_FMTBIT_U8; + bps = 8; + } + if (val & AC_SUPPCM_BITS_16) { + formats |= SNDRV_PCM_FMTBIT_S16_LE; + bps = 16; + } + if (wcaps & AC_WCAP_DIGITAL) { + if (val & AC_SUPPCM_BITS_32) + formats |= SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE; + if (val & (AC_SUPPCM_BITS_20|AC_SUPPCM_BITS_24)) + formats |= SNDRV_PCM_FMTBIT_S32_LE; + if (val & AC_SUPPCM_BITS_24) + bps = 24; + else if (val & AC_SUPPCM_BITS_20) + bps = 20; + } else if (val & (AC_SUPPCM_BITS_20|AC_SUPPCM_BITS_24|AC_SUPPCM_BITS_32)) { + formats |= SNDRV_PCM_FMTBIT_S32_LE; + if (val & AC_SUPPCM_BITS_32) + bps = 32; + else if (val & AC_SUPPCM_BITS_20) + bps = 20; + else if (val & AC_SUPPCM_BITS_24) + bps = 24; + } + } + else if (streams == AC_SUPFMT_FLOAT32) { /* should be exclusive */ + formats |= SNDRV_PCM_FMTBIT_FLOAT_LE; + bps = 32; + } else if (streams == AC_SUPFMT_AC3) { /* should be exclusive */ + /* temporary hack: we have still no proper support + * for the direct AC3 stream... + */ + formats |= SNDRV_PCM_FMTBIT_U8; + bps = 8; + } + if (formatsp) + *formatsp = formats; + if (bpsp) + *bpsp = bps; + } + + return 0; +} + +/** + * snd_hda_is_supported_format - check whether the given node supports the format val + * + * Returns 1 if supported, 0 if not. + */ +int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid, + unsigned int format) +{ + int i; + unsigned int val = 0, rate, stream; + + if (nid != codec->afg && + snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP) & AC_WCAP_FORMAT_OVRD) { + val = snd_hda_param_read(codec, nid, AC_PAR_PCM); + if (val == -1) + return 0; + } + if (! val) { + val = snd_hda_param_read(codec, codec->afg, AC_PAR_PCM); + if (val == -1) + return 0; + } + + rate = format & 0xff00; + for (i = 0; rate_bits[i][0]; i++) + if (rate_bits[i][2] == rate) { + if (val & (1 << i)) + break; + return 0; + } + if (! rate_bits[i][0]) + return 0; + + stream = snd_hda_param_read(codec, nid, AC_PAR_STREAM); + if (stream == -1) + return 0; + if (! stream && nid != codec->afg) + stream = snd_hda_param_read(codec, codec->afg, AC_PAR_STREAM); + if (! stream || stream == -1) + return 0; + + if (stream & AC_SUPFMT_PCM) { + switch (format & 0xf0) { + case 0x00: + if (! (val & AC_SUPPCM_BITS_8)) + return 0; + break; + case 0x10: + if (! (val & AC_SUPPCM_BITS_16)) + return 0; + break; + case 0x20: + if (! (val & AC_SUPPCM_BITS_20)) + return 0; + break; + case 0x30: + if (! (val & AC_SUPPCM_BITS_24)) + return 0; + break; + case 0x40: + if (! (val & AC_SUPPCM_BITS_32)) + return 0; + break; + default: + return 0; + } + } else { + /* FIXME: check for float32 and AC3? */ + } + + return 1; +} + +/* + * PCM stuff + */ +static int hda_pcm_default_open_close(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + snd_pcm_substream_t *substream) +{ + return 0; +} + +static int hda_pcm_default_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + snd_pcm_substream_t *substream) +{ + snd_hda_codec_setup_stream(codec, hinfo->nid, stream_tag, 0, format); + return 0; +} + +static int hda_pcm_default_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + snd_pcm_substream_t *substream) +{ + snd_hda_codec_setup_stream(codec, hinfo->nid, 0, 0, 0); + return 0; +} + +static int set_pcm_default_values(struct hda_codec *codec, struct hda_pcm_stream *info) +{ + if (info->nid) { + /* query support PCM information from the given NID */ + if (! info->rates || ! info->formats) + snd_hda_query_supported_pcm(codec, info->nid, + info->rates ? NULL : &info->rates, + info->formats ? NULL : &info->formats, + info->maxbps ? NULL : &info->maxbps); + } + if (info->ops.open == NULL) + info->ops.open = hda_pcm_default_open_close; + if (info->ops.close == NULL) + info->ops.close = hda_pcm_default_open_close; + if (info->ops.prepare == NULL) { + snd_assert(info->nid, return -EINVAL); + info->ops.prepare = hda_pcm_default_prepare; + } + if (info->ops.prepare == NULL) { + snd_assert(info->nid, return -EINVAL); + info->ops.prepare = hda_pcm_default_prepare; + } + if (info->ops.cleanup == NULL) { + snd_assert(info->nid, return -EINVAL); + info->ops.cleanup = hda_pcm_default_cleanup; + } + return 0; +} + +/** + * snd_hda_build_pcms - build PCM information + * @bus: the BUS + * + * Create PCM information for each codec included in the bus. + * + * The build_pcms codec patch is requested to set up codec->num_pcms and + * codec->pcm_info properly. The array is referred by the top-level driver + * to create its PCM instances. + * The allocated codec->pcm_info should be released in codec->patch_ops.free + * callback. + * + * At least, substreams, channels_min and channels_max must be filled for + * each stream. substreams = 0 indicates that the stream doesn't exist. + * When rates and/or formats are zero, the supported values are queried + * from the given nid. The nid is used also by the default ops.prepare + * and ops.cleanup callbacks. + * + * The driver needs to call ops.open in its open callback. Similarly, + * ops.close is supposed to be called in the close callback. + * ops.prepare should be called in the prepare or hw_params callback + * with the proper parameters for set up. + * ops.cleanup should be called in hw_free for clean up of streams. + * + * This function returns 0 if successfull, or a negative error code. + */ +int snd_hda_build_pcms(struct hda_bus *bus) +{ + struct list_head *p; + + list_for_each(p, &bus->codec_list) { + struct hda_codec *codec = list_entry(p, struct hda_codec, list); + unsigned int pcm, s; + int err; + if (! codec->patch_ops.build_pcms) + continue; + err = codec->patch_ops.build_pcms(codec); + if (err < 0) + return err; + for (pcm = 0; pcm < codec->num_pcms; pcm++) { + for (s = 0; s < 2; s++) { + struct hda_pcm_stream *info; + info = &codec->pcm_info[pcm].stream[s]; + if (! info->substreams) + continue; + err = set_pcm_default_values(codec, info); + if (err < 0) + return err; + } + } + } + return 0; +} + + +/** + * snd_hda_check_board_config - compare the current codec with the config table + * @codec: the HDA codec + * @tbl: configuration table, terminated by null entries + * + * Compares the modelname or PCI subsystem id of the current codec with the + * given configuration table. If a matching entry is found, returns its + * config value (supposed to be 0 or positive). + * + * If no entries are matching, the function returns a negative value. + */ +int snd_hda_check_board_config(struct hda_codec *codec, struct hda_board_config *tbl) +{ + struct hda_board_config *c; + + if (codec->bus->modelname) { + for (c = tbl; c->modelname || c->pci_vendor; c++) { + if (c->modelname && + ! strcmp(codec->bus->modelname, c->modelname)) { + snd_printd(KERN_INFO "hda_codec: model '%s' is selected\n", c->modelname); + return c->config; + } + } + } + + if (codec->bus->pci) { + u16 subsystem_vendor, subsystem_device; + pci_read_config_word(codec->bus->pci, PCI_SUBSYSTEM_VENDOR_ID, &subsystem_vendor); + pci_read_config_word(codec->bus->pci, PCI_SUBSYSTEM_ID, &subsystem_device); + for (c = tbl; c->modelname || c->pci_vendor; c++) { + if (c->pci_vendor == subsystem_vendor && + c->pci_device == subsystem_device) + return c->config; + } + } + return -1; +} + +/** + * snd_hda_add_new_ctls - create controls from the array + * @codec: the HDA codec + * @knew: the array of snd_kcontrol_new_t + * + * This helper function creates and add new controls in the given array. + * The array must be terminated with an empty entry as terminator. + * + * Returns 0 if successful, or a negative error code. + */ +int snd_hda_add_new_ctls(struct hda_codec *codec, snd_kcontrol_new_t *knew) +{ + int err; + + for (; knew->name; knew++) { + err = snd_ctl_add(codec->bus->card, snd_ctl_new1(knew, codec)); + if (err < 0) + return err; + } + return 0; +} + + +/* + * input MUX helper + */ +int snd_hda_input_mux_info(const struct hda_input_mux *imux, snd_ctl_elem_info_t *uinfo) +{ + unsigned int index; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = imux->num_items; + index = uinfo->value.enumerated.item; + if (index >= imux->num_items) + index = imux->num_items - 1; + strcpy(uinfo->value.enumerated.name, imux->items[index].label); + return 0; +} + +int snd_hda_input_mux_put(struct hda_codec *codec, const struct hda_input_mux *imux, + snd_ctl_elem_value_t *ucontrol, hda_nid_t nid, + unsigned int *cur_val) +{ + unsigned int idx; + + idx = ucontrol->value.enumerated.item[0]; + if (idx >= imux->num_items) + idx = imux->num_items - 1; + if (*cur_val == idx && ! codec->in_resume) + return 0; + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, + imux->items[idx].index); + *cur_val = idx; + return 1; +} + + +/* + * Multi-channel / digital-out PCM helper functions + */ + +/* + * open the digital out in the exclusive mode + */ +int snd_hda_multi_out_dig_open(struct hda_codec *codec, struct hda_multi_out *mout) +{ + down(&codec->spdif_mutex); + if (mout->dig_out_used) { + up(&codec->spdif_mutex); + return -EBUSY; /* already being used */ + } + mout->dig_out_used = HDA_DIG_EXCLUSIVE; + up(&codec->spdif_mutex); + return 0; +} + +/* + * release the digital out + */ +int snd_hda_multi_out_dig_close(struct hda_codec *codec, struct hda_multi_out *mout) +{ + down(&codec->spdif_mutex); + mout->dig_out_used = 0; + up(&codec->spdif_mutex); + return 0; +} + +/* + * set up more restrictions for analog out + */ +int snd_hda_multi_out_analog_open(struct hda_codec *codec, struct hda_multi_out *mout, + snd_pcm_substream_t *substream) +{ + substream->runtime->hw.channels_max = mout->max_channels; + return snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, 2); +} + +/* + * set up the i/o for analog out + * when the digital out is available, copy the front out to digital out, too. + */ +int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, struct hda_multi_out *mout, + unsigned int stream_tag, + unsigned int format, + snd_pcm_substream_t *substream) +{ + hda_nid_t *nids = mout->dac_nids; + int chs = substream->runtime->channels; + int i; + + down(&codec->spdif_mutex); + if (mout->dig_out_nid && mout->dig_out_used != HDA_DIG_EXCLUSIVE) { + if (chs == 2 && + snd_hda_is_supported_format(codec, mout->dig_out_nid, format) && + ! (codec->spdif_status & IEC958_AES0_NONAUDIO)) { + mout->dig_out_used = HDA_DIG_ANALOG_DUP; + /* setup digital receiver */ + snd_hda_codec_setup_stream(codec, mout->dig_out_nid, + stream_tag, 0, format); + } else { + mout->dig_out_used = 0; + snd_hda_codec_setup_stream(codec, mout->dig_out_nid, 0, 0, 0); + } + } + up(&codec->spdif_mutex); + + /* front */ + snd_hda_codec_setup_stream(codec, nids[HDA_FRONT], stream_tag, 0, format); + if (mout->hp_nid) + /* headphone out will just decode front left/right (stereo) */ + snd_hda_codec_setup_stream(codec, mout->hp_nid, stream_tag, 0, format); + /* surrounds */ + for (i = 1; i < mout->num_dacs; i++) { + if (i == HDA_REAR && chs == 2) /* copy front to rear */ + snd_hda_codec_setup_stream(codec, nids[i], stream_tag, 0, format); + else if (chs >= (i + 1) * 2) /* independent out */ + snd_hda_codec_setup_stream(codec, nids[i], stream_tag, i * 2, + format); + } + return 0; +} + +/* + * clean up the setting for analog out + */ +int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, struct hda_multi_out *mout) +{ + hda_nid_t *nids = mout->dac_nids; + int i; + + for (i = 0; i < mout->num_dacs; i++) + snd_hda_codec_setup_stream(codec, nids[i], 0, 0, 0); + if (mout->hp_nid) + snd_hda_codec_setup_stream(codec, mout->hp_nid, 0, 0, 0); + down(&codec->spdif_mutex); + if (mout->dig_out_nid && mout->dig_out_used == HDA_DIG_ANALOG_DUP) { + snd_hda_codec_setup_stream(codec, mout->dig_out_nid, 0, 0, 0); + mout->dig_out_used = 0; + } + up(&codec->spdif_mutex); + return 0; +} + +#ifdef CONFIG_PM +/* + * power management + */ + +/** + * snd_hda_suspend - suspend the codecs + * @bus: the HDA bus + * @state: suspsend state + * + * Returns 0 if successful. + */ +int snd_hda_suspend(struct hda_bus *bus, pm_message_t state) +{ + struct list_head *p; + + /* FIXME: should handle power widget capabilities */ + list_for_each(p, &bus->codec_list) { + struct hda_codec *codec = list_entry(p, struct hda_codec, list); + if (codec->patch_ops.suspend) + codec->patch_ops.suspend(codec, state); + } + return 0; +} + +/** + * snd_hda_resume - resume the codecs + * @bus: the HDA bus + * @state: resume state + * + * Returns 0 if successful. + */ +int snd_hda_resume(struct hda_bus *bus) +{ + struct list_head *p; + + list_for_each(p, &bus->codec_list) { + struct hda_codec *codec = list_entry(p, struct hda_codec, list); + if (codec->patch_ops.resume) + codec->patch_ops.resume(codec); + } + return 0; +} + +/** + * snd_hda_resume_ctls - resume controls in the new control list + * @codec: the HDA codec + * @knew: the array of snd_kcontrol_new_t + * + * This function resumes the mixer controls in the snd_kcontrol_new_t array, + * originally for snd_hda_add_new_ctls(). + * The array must be terminated with an empty entry as terminator. + */ +int snd_hda_resume_ctls(struct hda_codec *codec, snd_kcontrol_new_t *knew) +{ + snd_ctl_elem_value_t *val; + + val = kmalloc(sizeof(*val), GFP_KERNEL); + if (! val) + return -ENOMEM; + codec->in_resume = 1; + for (; knew->name; knew++) { + int i, count; + count = knew->count ? knew->count : 1; + for (i = 0; i < count; i++) { + memset(val, 0, sizeof(*val)); + val->id.iface = knew->iface; + val->id.device = knew->device; + val->id.subdevice = knew->subdevice; + strcpy(val->id.name, knew->name); + val->id.index = knew->index ? knew->index : i; + /* Assume that get callback reads only from cache, + * not accessing to the real hardware + */ + if (snd_ctl_elem_read(codec->bus->card, val) < 0) + continue; + snd_ctl_elem_write(codec->bus->card, NULL, val); + } + } + codec->in_resume = 0; + kfree(val); + return 0; +} + +/** + * snd_hda_resume_spdif_out - resume the digital out + * @codec: the HDA codec + */ +int snd_hda_resume_spdif_out(struct hda_codec *codec) +{ + return snd_hda_resume_ctls(codec, dig_mixes); +} + +/** + * snd_hda_resume_spdif_in - resume the digital in + * @codec: the HDA codec + */ +int snd_hda_resume_spdif_in(struct hda_codec *codec) +{ + return snd_hda_resume_ctls(codec, dig_in_ctls); +} +#endif + +/* + * symbols exported for controller modules + */ +EXPORT_SYMBOL(snd_hda_codec_read); +EXPORT_SYMBOL(snd_hda_codec_write); +EXPORT_SYMBOL(snd_hda_sequence_write); +EXPORT_SYMBOL(snd_hda_get_sub_nodes); +EXPORT_SYMBOL(snd_hda_queue_unsol_event); +EXPORT_SYMBOL(snd_hda_bus_new); +EXPORT_SYMBOL(snd_hda_codec_new); +EXPORT_SYMBOL(snd_hda_codec_setup_stream); +EXPORT_SYMBOL(snd_hda_calc_stream_format); +EXPORT_SYMBOL(snd_hda_build_pcms); +EXPORT_SYMBOL(snd_hda_build_controls); +#ifdef CONFIG_PM +EXPORT_SYMBOL(snd_hda_suspend); +EXPORT_SYMBOL(snd_hda_resume); +#endif + +/* + * INIT part + */ + +static int __init alsa_hda_init(void) +{ + return 0; +} + +static void __exit alsa_hda_exit(void) +{ +} + +module_init(alsa_hda_init) +module_exit(alsa_hda_exit) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h new file mode 100644 index 000000000000..c9e9dc9c7c98 --- /dev/null +++ b/sound/pci/hda/hda_codec.h @@ -0,0 +1,604 @@ +/* + * 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 <sound/info.h> +#include <sound/control.h> +#include <sound/pcm.h> + +/* + * nodes + */ +#define AC_NODE_ROOT 0x00 + +/* + * function group types + */ +enum { + AC_GRP_AUDIO_FUNCTION = 0x01, + AC_GRP_MODEM_FUNCTION = 0x02, +}; + +/* + * widget types + */ +enum { + AC_WID_AUD_OUT, /* Audio Out */ + AC_WID_AUD_IN, /* Audio In */ + AC_WID_AUD_MIX, /* Audio Mixer */ + AC_WID_AUD_SEL, /* Audio Selector */ + AC_WID_PIN, /* Pin Complex */ + AC_WID_POWER, /* Power */ + AC_WID_VOL_KNB, /* Volume Knob */ + AC_WID_BEEP, /* Beep Generator */ + AC_WID_VENDOR = 0x0f /* Vendor specific */ +}; + +/* + * GET verbs + */ +#define AC_VERB_GET_STREAM_FORMAT 0x0a00 +#define AC_VERB_GET_AMP_GAIN_MUTE 0x0b00 +#define AC_VERB_GET_PROC_COEF 0x0c00 +#define AC_VERB_GET_COEF_INDEX 0x0d00 +#define AC_VERB_PARAMETERS 0x0f00 +#define AC_VERB_GET_CONNECT_SEL 0x0f01 +#define AC_VERB_GET_CONNECT_LIST 0x0f02 +#define AC_VERB_GET_PROC_STATE 0x0f03 +#define AC_VERB_GET_SDI_SELECT 0x0f04 +#define AC_VERB_GET_POWER_STATE 0x0f05 +#define AC_VERB_GET_CONV 0x0f06 +#define AC_VERB_GET_PIN_WIDGET_CONTROL 0x0f07 +#define AC_VERB_GET_UNSOLICITED_RESPONSE 0x0f08 +#define AC_VERB_GET_PIN_SENSE 0x0f09 +#define AC_VERB_GET_BEEP_CONTROL 0x0f0a +#define AC_VERB_GET_EAPD_BTLENABLE 0x0f0c +#define AC_VERB_GET_DIGI_CONVERT 0x0f0d +#define AC_VERB_GET_VOLUME_KNOB_CONTROL 0x0f0f +/* f10-f1a: GPIO */ +#define AC_VERB_GET_CONFIG_DEFAULT 0x0f1c + +/* + * SET verbs + */ +#define AC_VERB_SET_STREAM_FORMAT 0x200 +#define AC_VERB_SET_AMP_GAIN_MUTE 0x300 +#define AC_VERB_SET_PROC_COEF 0x400 +#define AC_VERB_SET_COEF_INDEX 0x500 +#define AC_VERB_SET_CONNECT_SEL 0x701 +#define AC_VERB_SET_PROC_STATE 0x703 +#define AC_VERB_SET_SDI_SELECT 0x704 +#define AC_VERB_SET_POWER_STATE 0x705 +#define AC_VERB_SET_CHANNEL_STREAMID 0x706 +#define AC_VERB_SET_PIN_WIDGET_CONTROL 0x707 +#define AC_VERB_SET_UNSOLICITED_ENABLE 0x708 +#define AC_VERB_SET_PIN_SENSE 0x709 +#define AC_VERB_SET_BEEP_CONTROL 0x70a +#define AC_VERB_SET_EAPD_BTLENALBE 0x70c +#define AC_VERB_SET_DIGI_CONVERT_1 0x70d +#define AC_VERB_SET_DIGI_CONVERT_2 0x70e +#define AC_VERB_SET_VOLUME_KNOB_CONTROL 0x70f +#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 0x71c +#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_1 0x71d +#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_2 0x71e +#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_3 0x71f +#define AC_VERB_SET_CODEC_RESET 0x7ff + +/* + * Parameter IDs + */ +#define AC_PAR_VENDOR_ID 0x00 +#define AC_PAR_SUBSYSTEM_ID 0x01 +#define AC_PAR_REV_ID 0x02 +#define AC_PAR_NODE_COUNT 0x04 +#define AC_PAR_FUNCTION_TYPE 0x05 +#define AC_PAR_AUDIO_FG_CAP 0x08 +#define AC_PAR_AUDIO_WIDGET_CAP 0x09 +#define AC_PAR_PCM 0x0a +#define AC_PAR_STREAM 0x0b +#define AC_PAR_PIN_CAP 0x0c +#define AC_PAR_AMP_IN_CAP 0x0d +#define AC_PAR_CONNLIST_LEN 0x0e +#define AC_PAR_POWER_STATE 0x0f +#define AC_PAR_PROC_CAP 0x10 +#define AC_PAR_GPIO_CAP 0x11 +#define AC_PAR_AMP_OUT_CAP 0x12 + +/* + * AC_VERB_PARAMETERS results (32bit) + */ + +/* Function Group Type */ +#define AC_FGT_TYPE (0xff<<0) +#define AC_FGT_TYPE_SHIFT 0 +#define AC_FGT_UNSOL_CAP (1<<8) + +/* Audio Function Group Capabilities */ +#define AC_AFG_OUT_DELAY (0xf<<0) +#define AC_AFG_IN_DELAY (0xf<<8) +#define AC_AFG_BEEP_GEN (1<<16) + +/* Audio Widget Capabilities */ +#define AC_WCAP_STEREO (1<<0) /* stereo I/O */ +#define AC_WCAP_IN_AMP (1<<1) /* AMP-in present */ +#define AC_WCAP_OUT_AMP (1<<2) /* AMP-out present */ +#define AC_WCAP_AMP_OVRD (1<<3) /* AMP-parameter override */ +#define AC_WCAP_FORMAT_OVRD (1<<4) /* format override */ +#define AC_WCAP_STRIPE (1<<5) /* stripe */ +#define AC_WCAP_PROC_WID (1<<6) /* Proc Widget */ +#define AC_WCAP_UNSOL_CAP (1<<7) /* Unsol capable */ +#define AC_WCAP_CONN_LIST (1<<8) /* connection list */ +#define AC_WCAP_DIGITAL (1<<9) /* digital I/O */ +#define AC_WCAP_POWER (1<<10) /* power control */ +#define AC_WCAP_LR_SWAP (1<<11) /* L/R swap */ +#define AC_WCAP_DELAY (0xf<<16) +#define AC_WCAP_DELAY_SHIFT 16 +#define AC_WCAP_TYPE (0xf<<20) +#define AC_WCAP_TYPE_SHIFT 20 + +/* supported PCM rates and bits */ +#define AC_SUPPCM_RATES (0xfff << 0) +#define AC_SUPPCM_BITS_8 (1<<16) +#define AC_SUPPCM_BITS_16 (1<<17) +#define AC_SUPPCM_BITS_20 (1<<18) +#define AC_SUPPCM_BITS_24 (1<<19) +#define AC_SUPPCM_BITS_32 (1<<20) + +/* supported PCM stream format */ +#define AC_SUPFMT_PCM (1<<0) +#define AC_SUPFMT_FLOAT32 (1<<1) +#define AC_SUPFMT_AC3 (1<<2) + +/* Pin widget capabilies */ +#define AC_PINCAP_IMP_SENSE (1<<0) /* impedance sense capable */ +#define AC_PINCAP_TRIG_REQ (1<<1) /* trigger required */ +#define AC_PINCAP_PRES_DETECT (1<<2) /* presence detect capable */ +#define AC_PINCAP_HP_DRV (1<<3) /* headphone drive capable */ +#define AC_PINCAP_OUT (1<<4) /* output capable */ +#define AC_PINCAP_IN (1<<5) /* input capable */ +#define AC_PINCAP_BALANCE (1<<6) /* balanced I/O capable */ +#define AC_PINCAP_VREF (7<<8) +#define AC_PINCAP_VREF_SHIFT 8 +#define AC_PINCAP_EAPD (1<<16) /* EAPD capable */ +/* Vref status (used in pin cap and pin ctl) */ +#define AC_PIN_VREF_HIZ (1<<0) /* Hi-Z */ +#define AC_PIN_VREF_50 (1<<1) /* 50% */ +#define AC_PIN_VREF_GRD (1<<2) /* ground */ +#define AC_PIN_VREF_80 (1<<4) /* 80% */ +#define AC_PIN_VREF_100 (1<<5) /* 100% */ + + +/* Amplifier capabilities */ +#define AC_AMPCAP_OFFSET (0x7f<<0) /* 0dB offset */ +#define AC_AMPCAP_OFFSET_SHIFT 0 +#define AC_AMPCAP_NUM_STEPS (0x7f<<8) /* number of steps */ +#define AC_AMPCAP_NUM_STEPS_SHIFT 8 +#define AC_AMPCAP_STEP_SIZE (0x7f<<16) /* step size 0-32dB in 0.25dB */ +#define AC_AMPCAP_STEP_SIZE_SHIFT 16 +#define AC_AMPCAP_MUTE (1<<31) /* mute capable */ +#define AC_AMPCAP_MUTE_SHIFT 31 + +/* Connection list */ +#define AC_CLIST_LENGTH (0x7f<<0) +#define AC_CLIST_LONG (1<<7) + +/* Supported power status */ +#define AC_PWRST_D0SUP (1<<0) +#define AC_PWRST_D1SUP (1<<1) +#define AC_PWRST_D2SUP (1<<2) +#define AC_PWRST_D3SUP (1<<3) + +/* Processing capabilies */ +#define AC_PCAP_BENIGN (1<<0) +#define AC_PCAP_NUM_COEF (0xff<<8) + +/* Volume knobs capabilities */ +#define AC_KNBCAP_NUM_STEPS (0x7f<<0) +#define AC_KNBCAP_DELTA (1<<8) + +/* + * Control Parameters + */ + +/* Amp gain/mute */ +#define AC_AMP_MUTE (1<<8) +#define AC_AMP_GAIN (0x7f) +#define AC_AMP_GET_INDEX (0xf<<0) + +#define AC_AMP_GET_LEFT (1<<13) +#define AC_AMP_GET_RIGHT (0<<13) +#define AC_AMP_GET_OUTPUT (1<<15) +#define AC_AMP_GET_INPUT (0<<15) + +#define AC_AMP_SET_INDEX (0xf<<8) +#define AC_AMP_SET_INDEX_SHIFT 8 +#define AC_AMP_SET_RIGHT (1<<12) +#define AC_AMP_SET_LEFT (1<<13) +#define AC_AMP_SET_INPUT (1<<14) +#define AC_AMP_SET_OUTPUT (1<<15) + +/* DIGITAL1 bits */ +#define AC_DIG1_ENABLE (1<<0) +#define AC_DIG1_V (1<<1) +#define AC_DIG1_VCFG (1<<2) +#define AC_DIG1_EMPHASIS (1<<3) +#define AC_DIG1_COPYRIGHT (1<<4) +#define AC_DIG1_NONAUDIO (1<<5) +#define AC_DIG1_PROFESSIONAL (1<<6) +#define AC_DIG1_LEVEL (1<<7) + +/* Pin widget control - 8bit */ +#define AC_PINCTL_VREFEN (0x7<<0) +#define AC_PINCTL_IN_EN (1<<5) +#define AC_PINCTL_OUT_EN (1<<6) +#define AC_PINCTL_HP_EN (1<<7) + +/* configuration default - 32bit */ +#define AC_DEFCFG_SEQUENCE (0xf<<0) +#define AC_DEFCFG_DEF_ASSOC (0xf<<4) +#define AC_DEFCFG_MISC (0xf<<8) +#define AC_DEFCFG_COLOR (0xf<<12) +#define AC_DEFCFG_COLOR_SHIFT 12 +#define AC_DEFCFG_CONN_TYPE (0xf<<16) +#define AC_DEFCFG_CONN_TYPE_SHIFT 16 +#define AC_DEFCFG_DEVICE (0xf<<20) +#define AC_DEFCFG_DEVICE_SHIFT 20 +#define AC_DEFCFG_LOCATION (0x3f<<24) +#define AC_DEFCFG_LOCATION_SHIFT 24 +#define AC_DEFCFG_PORT_CONN (0x3<<30) +#define AC_DEFCFG_PORT_CONN_SHIFT 30 + +/* device device types (0x0-0xf) */ +enum { + AC_JACK_LINE_OUT, + AC_JACK_SPEAKER, + AC_JACK_HP_OUT, + AC_JACK_CD, + AC_JACK_SPDIF_OUT, + AC_JACK_DIG_OTHER_OUT, + AC_JACK_MODEM_LINE_SIDE, + AC_JACK_MODEM_HAND_SIDE, + AC_JACK_LINE_IN, + AC_JACK_AUX, + AC_JACK_MIC_IN, + AC_JACK_TELEPHONY, + AC_JACK_SPDIF_IN, + AC_JACK_DIG_OTHER_IN, + AC_JACK_OTHER = 0xf, +}; + +/* jack connection types (0x0-0xf) */ +enum { + AC_JACK_CONN_UNKNOWN, + AC_JACK_CONN_1_8, + AC_JACK_CONN_1_4, + AC_JACK_CONN_ATAPI, + AC_JACK_CONN_RCA, + AC_JACK_CONN_OPTICAL, + AC_JACK_CONN_OTHER_DIGITAL, + AC_JACK_CONN_OTHER_ANALOG, + AC_JACK_CONN_DIN, + AC_JACK_CONN_XLR, + AC_JACK_CONN_RJ11, + AC_JACK_CONN_COMB, + AC_JACK_CONN_OTHER = 0xf, +}; + +/* jack colors (0x0-0xf) */ +enum { + AC_JACK_COLOR_UNKNOWN, + AC_JACK_COLOR_BLACK, + AC_JACK_COLOR_GREY, + AC_JACK_COLOR_BLUE, + AC_JACK_COLOR_GREEN, + AC_JACK_COLOR_RED, + AC_JACK_COLOR_ORANGE, + AC_JACK_COLOR_YELLOW, + AC_JACK_COLOR_PURPLE, + AC_JACK_COLOR_PINK, + AC_JACK_COLOR_WHITE = 0xe, + AC_JACK_COLOR_OTHER, +}; + +/* Jack location (0x0-0x3f) */ +/* common case */ +enum { + AC_JACK_LOC_NONE, + AC_JACK_LOC_REAR, + AC_JACK_LOC_FRONT, + AC_JACK_LOC_LEFT, + AC_JACK_LOC_RIGHT, + AC_JACK_LOC_TOP, + AC_JACK_LOC_BOTTOM, +}; +/* bits 4-5 */ +enum { + AC_JACK_LOC_EXTERNAL = 0x00, + AC_JACK_LOC_INTERNAL = 0x10, + AC_JACK_LOC_SEPARATE = 0x20, + AC_JACK_LOC_OTHER = 0x30, +}; +enum { + /* external on primary chasis */ + AC_JACK_LOC_REAR_PANEL = 0x07, + AC_JACK_LOC_DRIVE_BAY, + /* internal */ + AC_JACK_LOC_RISER = 0x17, + AC_JACK_LOC_HDMI, + AC_JACK_LOC_ATAPI, + /* others */ + AC_JACK_LOC_MOBILE_IN = 0x37, + AC_JACK_LOC_MOBILE_OUT, +}; + +/* Port connectivity (0-3) */ +enum { + AC_JACK_PORT_COMPLEX, + AC_JACK_PORT_NONE, + AC_JACK_PORT_FIXED, + AC_JACK_PORT_BOTH, +}; + +/* max. connections to a widget */ +#define HDA_MAX_CONNECTIONS 16 + +/* max. codec address */ +#define HDA_MAX_CODEC_ADDRESS 0x0f + +/* + * Structures + */ + +struct hda_bus; +struct hda_codec; +struct hda_pcm; +struct hda_pcm_stream; +struct hda_bus_unsolicited; + +/* NID type */ +typedef u16 hda_nid_t; + +/* bus operators */ +struct hda_bus_ops { + /* send a single command */ + int (*command)(struct hda_codec *codec, hda_nid_t nid, int direct, + unsigned int verb, unsigned int parm); + /* get a response from the last command */ + unsigned int (*get_response)(struct hda_codec *codec); + /* free the private data */ + void (*private_free)(struct hda_bus *); +}; + +/* template to pass to the bus constructor */ +struct hda_bus_template { + void *private_data; + struct pci_dev *pci; + const char *modelname; + struct hda_bus_ops ops; +}; + +/* + * 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 { + snd_card_t *card; + + /* copied from template */ + void *private_data; + struct pci_dev *pci; + const char *modelname; + struct hda_bus_ops ops; + + /* codec linked list */ + struct list_head codec_list; + struct hda_codec *caddr_tbl[HDA_MAX_CODEC_ADDRESS]; /* caddr -> codec */ + + struct semaphore cmd_mutex; + + /* unsolicited event queue */ + struct hda_bus_unsolicited *unsol; + + snd_info_entry_t *proc; +}; + +/* + * codec preset + * + * Known codecs have the patch to build and set up the controls/PCMs + * better than the generic parser. + */ +struct hda_codec_preset { + unsigned int id; + unsigned int mask; + unsigned int subs; + unsigned int subs_mask; + unsigned int rev; + const char *name; + int (*patch)(struct hda_codec *codec); +}; + +/* 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); +#ifdef CONFIG_PM + int (*suspend)(struct hda_codec *codec, pm_message_t state); + int (*resume)(struct hda_codec *codec); +#endif +}; + +/* record for amp information cache */ +struct hda_amp_info { + u32 key; /* hash key */ + u32 amp_caps; /* amp capabilities */ + u16 vol[2]; /* current volume & mute*/ + u16 status; /* update flag */ + u16 next; /* next link */ +}; + +/* PCM callbacks */ +struct hda_pcm_ops { + int (*open)(struct hda_pcm_stream *info, struct hda_codec *codec, + snd_pcm_substream_t *substream); + int (*close)(struct hda_pcm_stream *info, struct hda_codec *codec, + snd_pcm_substream_t *substream); + int (*prepare)(struct hda_pcm_stream *info, struct hda_codec *codec, + unsigned int stream_tag, unsigned int format, + snd_pcm_substream_t *substream); + int (*cleanup)(struct hda_pcm_stream *info, struct hda_codec *codec, + snd_pcm_substream_t *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 */ + struct hda_pcm_ops ops; +}; + +/* for PCM creation */ +struct hda_pcm { + char *name; + struct hda_pcm_stream stream[2]; +}; + +/* codec information */ +struct hda_codec { + struct hda_bus *bus; + unsigned int addr; /* codec addr*/ + struct list_head list; /* list point */ + + hda_nid_t afg; /* AFG node id */ + + /* ids */ + u32 vendor_id; + u32 subsystem_id; + u32 revision_id; + + /* detected preset */ + const struct hda_codec_preset *preset; + + /* set by patch */ + struct hda_codec_ops patch_ops; + + /* resume phase - all controls should update even if + * the values are not changed + */ + unsigned int in_resume; + + /* PCM to create, set by patch_ops.build_pcms callback */ + unsigned int num_pcms; + struct hda_pcm *pcm_info; + + /* codec specific info */ + void *spec; + + /* hash for amp access */ + u16 amp_hash[32]; + int num_amp_entries; + struct hda_amp_info amp_info[128]; /* big enough? */ + + struct semaphore spdif_mutex; + unsigned int spdif_status; /* IEC958 status bits */ + unsigned short spdif_ctls; /* SPDIF control bits */ + unsigned int spdif_in_enable; /* SPDIF input enable? */ +}; + +/* direction */ +enum { + HDA_INPUT, HDA_OUTPUT +}; + + +/* + * constructors + */ +int snd_hda_bus_new(snd_card_t *card, const struct hda_bus_template *temp, + struct hda_bus **busp); +int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, + struct hda_codec **codecp); + +/* + * low level functions + */ +unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, int direct, + unsigned int verb, unsigned int parm); +int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct, + unsigned int verb, unsigned int parm); +#define snd_hda_param_read(codec, nid, param) snd_hda_codec_read(codec, nid, 0, AC_VERB_PARAMETERS, param) +int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid, hda_nid_t *start_id); +int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, hda_nid_t *conn_list, int max_conns); + +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 */ +int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex); + +/* + * Mixer + */ +int snd_hda_build_controls(struct hda_bus *bus); + +/* + * PCM + */ +int snd_hda_build_pcms(struct hda_bus *bus); +void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, u32 stream_tag, + int channel_id, int format); +unsigned int snd_hda_calc_stream_format(unsigned int rate, unsigned int channels, + unsigned int format, unsigned int maxbps); +int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, + u32 *ratesp, u64 *formatsp, unsigned int *bpsp); +int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid, + unsigned int format); + +/* + * Misc + */ +void snd_hda_get_codec_name(struct hda_codec *codec, char *name, int namelen); + +/* + * power management + */ +#ifdef CONFIG_PM +int snd_hda_suspend(struct hda_bus *bus, pm_message_t state); +int snd_hda_resume(struct hda_bus *bus); +#endif + +#endif /* __SOUND_HDA_CODEC_H */ diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c new file mode 100644 index 000000000000..69f7b6c4cf83 --- /dev/null +++ b/sound/pci/hda/hda_generic.c @@ -0,0 +1,906 @@ +/* + * Universal Interface for Intel High Definition Audio Codec + * + * Generic widget tree parser + * + * Copyright (c) 2004 Takashi Iwai <tiwai@suse.de> + * + * This driver 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 driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <sound/driver.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/pci.h> +#include <sound/core.h> +#include "hda_codec.h" +#include "hda_local.h" + +/* widget node for parsing */ +struct hda_gnode { + hda_nid_t nid; /* NID of this widget */ + unsigned short nconns; /* number of input connections */ + hda_nid_t conn_list[HDA_MAX_CONNECTIONS]; /* input connections */ + unsigned int wid_caps; /* widget capabilities */ + unsigned char type; /* widget type */ + unsigned char pin_ctl; /* pin controls */ + unsigned char checked; /* the flag indicates that the node is already parsed */ + unsigned int pin_caps; /* pin widget capabilities */ + unsigned int def_cfg; /* default configuration */ + unsigned int amp_out_caps; /* AMP out capabilities */ + unsigned int amp_in_caps; /* AMP in capabilities */ + struct list_head list; +}; + +/* pathc-specific record */ +struct hda_gspec { + struct hda_gnode *dac_node; /* DAC node */ + struct hda_gnode *out_pin_node; /* Output pin (Line-Out) node */ + struct hda_gnode *pcm_vol_node; /* Node for PCM volume */ + unsigned int pcm_vol_index; /* connection of PCM volume */ + + struct hda_gnode *adc_node; /* ADC node */ + struct hda_gnode *cap_vol_node; /* Node for capture volume */ + unsigned int cur_cap_src; /* current capture source */ + struct hda_input_mux input_mux; + char cap_labels[HDA_MAX_NUM_INPUTS][16]; + + unsigned int def_amp_in_caps; + unsigned int def_amp_out_caps; + + struct hda_pcm pcm_rec; /* PCM information */ + + struct list_head nid_list; /* list of widgets */ +}; + +/* + * retrieve the default device type from the default config value + */ +#define get_defcfg_type(node) (((node)->def_cfg & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT) +#define get_defcfg_location(node) (((node)->def_cfg & AC_DEFCFG_LOCATION) >> AC_DEFCFG_LOCATION_SHIFT) + +/* + * destructor + */ +static void snd_hda_generic_free(struct hda_codec *codec) +{ + struct hda_gspec *spec = codec->spec; + struct list_head *p, *n; + + if (! spec) + return; + /* free all widgets */ + list_for_each_safe(p, n, &spec->nid_list) { + struct hda_gnode *node = list_entry(p, struct hda_gnode, list); + kfree(node); + } + kfree(spec); +} + + +/* + * add a new widget node and read its attributes + */ +static int add_new_node(struct hda_codec *codec, struct hda_gspec *spec, hda_nid_t nid) +{ + struct hda_gnode *node; + int nconns; + + node = kcalloc(1, sizeof(*node), GFP_KERNEL); + if (node == NULL) + return -ENOMEM; + node->nid = nid; + nconns = snd_hda_get_connections(codec, nid, node->conn_list, HDA_MAX_CONNECTIONS); + if (nconns < 0) { + kfree(node); + return nconns; + } + node->nconns = nconns; + node->wid_caps = snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP); + node->type = (node->wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; + + if (node->type == AC_WID_PIN) { + node->pin_caps = snd_hda_param_read(codec, node->nid, AC_PAR_PIN_CAP); + node->pin_ctl = snd_hda_codec_read(codec, node->nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + node->def_cfg = snd_hda_codec_read(codec, node->nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + } + + if (node->wid_caps & AC_WCAP_OUT_AMP) { + if (node->wid_caps & AC_WCAP_AMP_OVRD) + node->amp_out_caps = snd_hda_param_read(codec, node->nid, AC_PAR_AMP_OUT_CAP); + if (! node->amp_out_caps) + node->amp_out_caps = spec->def_amp_out_caps; + } + if (node->wid_caps & AC_WCAP_IN_AMP) { + if (node->wid_caps & AC_WCAP_AMP_OVRD) + node->amp_in_caps = snd_hda_param_read(codec, node->nid, AC_PAR_AMP_IN_CAP); + if (! node->amp_in_caps) + node->amp_in_caps = spec->def_amp_in_caps; + } + list_add_tail(&node->list, &spec->nid_list); + return 0; +} + +/* + * build the AFG subtree + */ +static int build_afg_tree(struct hda_codec *codec) +{ + struct hda_gspec *spec = codec->spec; + int i, nodes, err; + hda_nid_t nid; + + snd_assert(spec, return -EINVAL); + + spec->def_amp_out_caps = snd_hda_param_read(codec, codec->afg, AC_PAR_AMP_OUT_CAP); + spec->def_amp_in_caps = snd_hda_param_read(codec, codec->afg, AC_PAR_AMP_IN_CAP); + + nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid); + if (! nid || nodes < 0) { + printk(KERN_ERR "Invalid AFG subtree\n"); + return -EINVAL; + } + + /* parse all nodes belonging to the AFG */ + for (i = 0; i < nodes; i++, nid++) { + if ((err = add_new_node(codec, spec, nid)) < 0) + return err; + } + + return 0; +} + + +/* + * look for the node record for the given NID + */ +/* FIXME: should avoid the braindead linear search */ +static struct hda_gnode *hda_get_node(struct hda_gspec *spec, hda_nid_t nid) +{ + struct list_head *p; + struct hda_gnode *node; + + list_for_each(p, &spec->nid_list) { + node = list_entry(p, struct hda_gnode, list); + if (node->nid == nid) + return node; + } + return NULL; +} + +/* + * unmute (and set max vol) the output amplifier + */ +static int unmute_output(struct hda_codec *codec, struct hda_gnode *node) +{ + unsigned int val, ofs; + snd_printdd("UNMUTE OUT: NID=0x%x\n", node->nid); + val = (node->amp_out_caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT; + ofs = (node->amp_out_caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT; + if (val >= ofs) + val -= ofs; + val |= AC_AMP_SET_LEFT | AC_AMP_SET_RIGHT; + val |= AC_AMP_SET_OUTPUT; + return snd_hda_codec_write(codec, node->nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, val); +} + +/* + * unmute (and set max vol) the input amplifier + */ +static int unmute_input(struct hda_codec *codec, struct hda_gnode *node, unsigned int index) +{ + unsigned int val, ofs; + snd_printdd("UNMUTE IN: NID=0x%x IDX=0x%x\n", node->nid, index); + val = (node->amp_in_caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT; + ofs = (node->amp_in_caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT; + if (val >= ofs) + val -= ofs; + val |= AC_AMP_SET_LEFT | AC_AMP_SET_RIGHT; + val |= AC_AMP_SET_INPUT; + // awk added - fixed to allow unmuting of indexed amps + val |= index << AC_AMP_SET_INDEX_SHIFT; + return snd_hda_codec_write(codec, node->nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, val); +} + +/* + * select the input connection of the given node. + */ +static int select_input_connection(struct hda_codec *codec, struct hda_gnode *node, + unsigned int index) +{ + snd_printdd("CONNECT: NID=0x%x IDX=0x%x\n", node->nid, index); + return snd_hda_codec_write(codec, node->nid, 0, AC_VERB_SET_CONNECT_SEL, index); +} + +/* + * clear checked flag of each node in the node list + */ +static void clear_check_flags(struct hda_gspec *spec) +{ + struct list_head *p; + struct hda_gnode *node; + + list_for_each(p, &spec->nid_list) { + node = list_entry(p, struct hda_gnode, list); + node->checked = 0; + } +} + +/* + * parse the output path recursively until reach to an audio output widget + * + * returns 0 if not found, 1 if found, or a negative error code. + */ +static int parse_output_path(struct hda_codec *codec, struct hda_gspec *spec, + struct hda_gnode *node) +{ + int i, err; + struct hda_gnode *child; + + if (node->checked) + return 0; + + node->checked = 1; + if (node->type == AC_WID_AUD_OUT) { + if (node->wid_caps & AC_WCAP_DIGITAL) { + snd_printdd("Skip Digital OUT node %x\n", node->nid); + return 0; + } + snd_printdd("AUD_OUT found %x\n", node->nid); + if (spec->dac_node) { + /* already DAC node is assigned, just unmute & connect */ + return node == spec->dac_node; + } + spec->dac_node = node; + if (node->wid_caps & AC_WCAP_OUT_AMP) { + spec->pcm_vol_node = node; + spec->pcm_vol_index = 0; + } + return 1; /* found */ + } + + for (i = 0; i < node->nconns; i++) { + child = hda_get_node(spec, node->conn_list[i]); + if (! child) + continue; + err = parse_output_path(codec, spec, child); + if (err < 0) + return err; + else if (err > 0) { + /* found one, + * select the path, unmute both input and output + */ + if (node->nconns > 1) + select_input_connection(codec, node, i); + unmute_input(codec, node, i); + unmute_output(codec, node); + if (! spec->pcm_vol_node) { + if (node->wid_caps & AC_WCAP_IN_AMP) { + spec->pcm_vol_node = node; + spec->pcm_vol_index = i; + } else if (node->wid_caps & AC_WCAP_OUT_AMP) { + spec->pcm_vol_node = node; + spec->pcm_vol_index = 0; + } + } + return 1; + } + } + return 0; +} + +/* + * Look for the output PIN widget with the given jack type + * and parse the output path to that PIN. + * + * Returns the PIN node when the path to DAC is established. + */ +static struct hda_gnode *parse_output_jack(struct hda_codec *codec, + struct hda_gspec *spec, + int jack_type) +{ + struct list_head *p; + struct hda_gnode *node; + int err; + + list_for_each(p, &spec->nid_list) { + node = list_entry(p, struct hda_gnode, list); + if (node->type != AC_WID_PIN) + continue; + /* output capable? */ + if (! (node->pin_caps & AC_PINCAP_OUT)) + continue; + if (jack_type >= 0) { + if (jack_type != get_defcfg_type(node)) + continue; + if (node->wid_caps & AC_WCAP_DIGITAL) + continue; /* skip SPDIF */ + } else { + /* output as default? */ + if (! (node->pin_ctl & AC_PINCTL_OUT_EN)) + continue; + } + clear_check_flags(spec); + err = parse_output_path(codec, spec, node); + if (err < 0) + return NULL; + else if (err > 0) { + /* unmute the PIN output */ + unmute_output(codec, node); + /* set PIN-Out enable */ + snd_hda_codec_write(codec, node->nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN); + return node; + } + } + return NULL; +} + + +/* + * parse outputs + */ +static int parse_output(struct hda_codec *codec) +{ + struct hda_gspec *spec = codec->spec; + struct hda_gnode *node; + + /* + * Look for the output PIN widget + */ + /* first, look for the line-out pin */ + node = parse_output_jack(codec, spec, AC_JACK_LINE_OUT); + if (node) /* found, remember the PIN node */ + spec->out_pin_node = node; + /* look for the HP-out pin */ + node = parse_output_jack(codec, spec, AC_JACK_HP_OUT); + if (node) { + if (! spec->out_pin_node) + spec->out_pin_node = node; + } + + if (! spec->out_pin_node) { + /* no line-out or HP pins found, + * then choose for the first output pin + */ + spec->out_pin_node = parse_output_jack(codec, spec, -1); + if (! spec->out_pin_node) + snd_printd("hda_generic: no proper output path found\n"); + } + + return 0; +} + +/* + * input MUX + */ + +/* control callbacks */ +static int capture_source_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hda_gspec *spec = codec->spec; + return snd_hda_input_mux_info(&spec->input_mux, uinfo); +} + +static int capture_source_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hda_gspec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->cur_cap_src; + return 0; +} + +static int capture_source_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hda_gspec *spec = codec->spec; + return snd_hda_input_mux_put(codec, &spec->input_mux, ucontrol, + spec->adc_node->nid, &spec->cur_cap_src); +} + +/* + * return the string name of the given input PIN widget + */ +static const char *get_input_type(struct hda_gnode *node, unsigned int *pinctl) +{ + unsigned int location = get_defcfg_location(node); + switch (get_defcfg_type(node)) { + case AC_JACK_LINE_IN: + if ((location & 0x0f) == AC_JACK_LOC_FRONT) + return "Front Line"; + return "Line"; + case AC_JACK_CD: + if (pinctl) + *pinctl |= AC_PIN_VREF_GRD; + return "CD"; + case AC_JACK_AUX: + if ((location & 0x0f) == AC_JACK_LOC_FRONT) + return "Front Aux"; + return "Aux"; + case AC_JACK_MIC_IN: + if ((location & 0x0f) == AC_JACK_LOC_FRONT) + return "Front Mic"; + return "Mic"; + case AC_JACK_SPDIF_IN: + return "SPDIF"; + case AC_JACK_DIG_OTHER_IN: + return "Digital"; + } + return NULL; +} + +/* + * parse the nodes recursively until reach to the input PIN + * + * returns 0 if not found, 1 if found, or a negative error code. + */ +static int parse_adc_sub_nodes(struct hda_codec *codec, struct hda_gspec *spec, + struct hda_gnode *node) +{ + int i, err; + unsigned int pinctl; + char *label; + const char *type; + + if (node->checked) + return 0; + + node->checked = 1; + if (node->type != AC_WID_PIN) { + for (i = 0; i < node->nconns; i++) { + struct hda_gnode *child; + child = hda_get_node(spec, node->conn_list[i]); + if (! child) + continue; + err = parse_adc_sub_nodes(codec, spec, child); + if (err < 0) + return err; + if (err > 0) { + /* found one, + * select the path, unmute both input and output + */ + if (node->nconns > 1) + select_input_connection(codec, node, i); + unmute_input(codec, node, i); + unmute_output(codec, node); + return err; + } + } + return 0; + } + + /* input capable? */ + if (! (node->pin_caps & AC_PINCAP_IN)) + return 0; + + if (node->wid_caps & AC_WCAP_DIGITAL) + return 0; /* skip SPDIF */ + + if (spec->input_mux.num_items >= HDA_MAX_NUM_INPUTS) { + snd_printk(KERN_ERR "hda_generic: Too many items for capture\n"); + return -EINVAL; + } + + pinctl = AC_PINCTL_IN_EN; + /* create a proper capture source label */ + type = get_input_type(node, &pinctl); + if (! type) { + /* input as default? */ + if (! (node->pin_ctl & AC_PINCTL_IN_EN)) + return 0; + type = "Input"; + } + label = spec->cap_labels[spec->input_mux.num_items]; + strcpy(label, type); + spec->input_mux.items[spec->input_mux.num_items].label = label; + + /* unmute the PIN external input */ + unmute_input(codec, node, 0); /* index = 0? */ + /* set PIN-In enable */ + snd_hda_codec_write(codec, node->nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl); + + return 1; /* found */ +} + +/* + * parse input + */ +static int parse_input_path(struct hda_codec *codec, struct hda_gnode *adc_node) +{ + struct hda_gspec *spec = codec->spec; + struct hda_gnode *node; + int i, err; + + snd_printdd("AUD_IN = %x\n", adc_node->nid); + clear_check_flags(spec); + + // awk added - fixed no recording due to muted widget + unmute_input(codec, adc_node, 0); + + /* + * check each connection of the ADC + * if it reaches to a proper input PIN, add the path as the + * input path. + */ + for (i = 0; i < adc_node->nconns; i++) { + node = hda_get_node(spec, adc_node->conn_list[i]); + if (! node) + continue; + err = parse_adc_sub_nodes(codec, spec, node); + if (err < 0) + return err; + else if (err > 0) { + struct hda_input_mux_item *csrc = &spec->input_mux.items[spec->input_mux.num_items]; + char *buf = spec->cap_labels[spec->input_mux.num_items]; + int ocap; + for (ocap = 0; ocap < spec->input_mux.num_items; ocap++) { + if (! strcmp(buf, spec->cap_labels[ocap])) { + /* same label already exists, + * put the index number to be unique + */ + sprintf(buf, "%s %d", spec->cap_labels[ocap], + spec->input_mux.num_items); + } + } + csrc->index = i; + spec->input_mux.num_items++; + } + } + + if (! spec->input_mux.num_items) + return 0; /* no input path found... */ + + snd_printdd("[Capture Source] NID=0x%x, #SRC=%d\n", adc_node->nid, spec->input_mux.num_items); + for (i = 0; i < spec->input_mux.num_items; i++) + snd_printdd(" [%s] IDX=0x%x\n", spec->input_mux.items[i].label, + spec->input_mux.items[i].index); + + spec->adc_node = adc_node; + return 1; +} + +/* + * parse input + */ +static int parse_input(struct hda_codec *codec) +{ + struct hda_gspec *spec = codec->spec; + struct list_head *p; + struct hda_gnode *node; + int err; + + /* + * At first we look for an audio input widget. + * If it reaches to certain input PINs, we take it as the + * input path. + */ + list_for_each(p, &spec->nid_list) { + node = list_entry(p, struct hda_gnode, list); + if (node->wid_caps & AC_WCAP_DIGITAL) + continue; /* skip SPDIF */ + if (node->type == AC_WID_AUD_IN) { + err = parse_input_path(codec, node); + if (err < 0) + return err; + else if (err > 0) + return 0; + } + } + snd_printd("hda_generic: no proper input path found\n"); + return 0; +} + +/* + * create mixer controls if possible + */ +#define DIR_OUT 0x1 +#define DIR_IN 0x2 + +static int create_mixer(struct hda_codec *codec, struct hda_gnode *node, + unsigned int index, const char *type, const char *dir_sfx) +{ + char name[32]; + int err; + int created = 0; + snd_kcontrol_new_t knew; + + if (type) + sprintf(name, "%s %s Switch", type, dir_sfx); + else + sprintf(name, "%s Switch", dir_sfx); + if ((node->wid_caps & AC_WCAP_IN_AMP) && + (node->amp_in_caps & AC_AMPCAP_MUTE)) { + knew = (snd_kcontrol_new_t)HDA_CODEC_MUTE(name, node->nid, index, HDA_INPUT); + snd_printdd("[%s] NID=0x%x, DIR=IN, IDX=0x%x\n", name, node->nid, index); + if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0) + return err; + created = 1; + } else if ((node->wid_caps & AC_WCAP_OUT_AMP) && + (node->amp_out_caps & AC_AMPCAP_MUTE)) { + knew = (snd_kcontrol_new_t)HDA_CODEC_MUTE(name, node->nid, 0, HDA_OUTPUT); + snd_printdd("[%s] NID=0x%x, DIR=OUT\n", name, node->nid); + if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0) + return err; + created = 1; + } + + if (type) + sprintf(name, "%s %s Volume", type, dir_sfx); + else + sprintf(name, "%s Volume", dir_sfx); + if ((node->wid_caps & AC_WCAP_IN_AMP) && + (node->amp_in_caps & AC_AMPCAP_NUM_STEPS)) { + knew = (snd_kcontrol_new_t)HDA_CODEC_VOLUME(name, node->nid, index, HDA_INPUT); + snd_printdd("[%s] NID=0x%x, DIR=IN, IDX=0x%x\n", name, node->nid, index); + if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0) + return err; + created = 1; + } else if ((node->wid_caps & AC_WCAP_OUT_AMP) && + (node->amp_out_caps & AC_AMPCAP_NUM_STEPS)) { + knew = (snd_kcontrol_new_t)HDA_CODEC_VOLUME(name, node->nid, 0, HDA_OUTPUT); + snd_printdd("[%s] NID=0x%x, DIR=OUT\n", name, node->nid); + if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0) + return err; + created = 1; + } + + return created; +} + +/* + * check whether the controls with the given name and direction suffix already exist + */ +static int check_existing_control(struct hda_codec *codec, const char *type, const char *dir) +{ + snd_ctl_elem_id_t id; + memset(&id, 0, sizeof(id)); + sprintf(id.name, "%s %s Volume", type, dir); + id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + if (snd_ctl_find_id(codec->bus->card, &id)) + return 1; + sprintf(id.name, "%s %s Switch", type, dir); + id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + if (snd_ctl_find_id(codec->bus->card, &id)) + return 1; + return 0; +} + +/* + * build output mixer controls + */ +static int build_output_controls(struct hda_codec *codec) +{ + struct hda_gspec *spec = codec->spec; + int err; + + err = create_mixer(codec, spec->pcm_vol_node, spec->pcm_vol_index, + "PCM", "Playback"); + if (err < 0) + return err; + return 0; +} + +/* create capture volume/switch */ +static int build_input_controls(struct hda_codec *codec) +{ + struct hda_gspec *spec = codec->spec; + struct hda_gnode *adc_node = spec->adc_node; + int err; + + if (! adc_node) + return 0; /* not found */ + + /* create capture volume and switch controls if the ADC has an amp */ + err = create_mixer(codec, adc_node, 0, NULL, "Capture"); + + /* create input MUX if multiple sources are available */ + if (spec->input_mux.num_items > 1) { + static snd_kcontrol_new_t cap_sel = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = capture_source_info, + .get = capture_source_get, + .put = capture_source_put, + }; + if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&cap_sel, codec))) < 0) + return err; + spec->cur_cap_src = 0; + select_input_connection(codec, adc_node, spec->input_mux.items[0].index); + } + return 0; +} + + +/* + * parse the nodes recursively until reach to the output PIN. + * + * returns 0 - if not found, + * 1 - if found, but no mixer is created + * 2 - if found and mixer was already created, (just skip) + * a negative error code + */ +static int parse_loopback_path(struct hda_codec *codec, struct hda_gspec *spec, + struct hda_gnode *node, struct hda_gnode *dest_node, + const char *type) +{ + int i, err; + + if (node->checked) + return 0; + + node->checked = 1; + if (node == dest_node) { + /* loopback connection found */ + return 1; + } + + for (i = 0; i < node->nconns; i++) { + struct hda_gnode *child = hda_get_node(spec, node->conn_list[i]); + if (! child) + continue; + err = parse_loopback_path(codec, spec, child, dest_node, type); + if (err < 0) + return err; + else if (err >= 1) { + if (err == 1) { + err = create_mixer(codec, node, i, type, "Playback"); + if (err < 0) + return err; + if (err > 0) + return 2; /* ok, created */ + /* not created, maybe in the lower path */ + err = 1; + } + /* connect and unmute */ + if (node->nconns > 1) + select_input_connection(codec, node, i); + unmute_input(codec, node, i); + unmute_output(codec, node); + return err; + } + } + return 0; +} + +/* + * parse the tree and build the loopback controls + */ +static int build_loopback_controls(struct hda_codec *codec) +{ + struct hda_gspec *spec = codec->spec; + struct list_head *p; + struct hda_gnode *node; + int err; + const char *type; + + if (! spec->out_pin_node) + return 0; + + list_for_each(p, &spec->nid_list) { + node = list_entry(p, struct hda_gnode, list); + if (node->type != AC_WID_PIN) + continue; + /* input capable? */ + if (! (node->pin_caps & AC_PINCAP_IN)) + return 0; + type = get_input_type(node, NULL); + if (type) { + if (check_existing_control(codec, type, "Playback")) + continue; + clear_check_flags(spec); + err = parse_loopback_path(codec, spec, spec->out_pin_node, + node, type); + if (err < 0) + return err; + if (! err) + continue; + } + } + return 0; +} + +/* + * build mixer controls + */ +static int build_generic_controls(struct hda_codec *codec) +{ + int err; + + if ((err = build_input_controls(codec)) < 0 || + (err = build_output_controls(codec)) < 0 || + (err = build_loopback_controls(codec)) < 0) + return err; + + return 0; +} + +/* + * PCM + */ +static struct hda_pcm_stream generic_pcm_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, +}; + +static int build_generic_pcms(struct hda_codec *codec) +{ + struct hda_gspec *spec = codec->spec; + struct hda_pcm *info = &spec->pcm_rec; + + if (! spec->dac_node && ! spec->adc_node) { + snd_printd("hda_generic: no PCM found\n"); + return 0; + } + + codec->num_pcms = 1; + codec->pcm_info = info; + + info->name = "HDA Generic"; + if (spec->dac_node) { + info->stream[0] = generic_pcm_playback; + info->stream[0].nid = spec->dac_node->nid; + } + if (spec->adc_node) { + info->stream[1] = generic_pcm_playback; + info->stream[1].nid = spec->adc_node->nid; + } + + return 0; +} + + +/* + */ +static struct hda_codec_ops generic_patch_ops = { + .build_controls = build_generic_controls, + .build_pcms = build_generic_pcms, + .free = snd_hda_generic_free, +}; + +/* + * the generic parser + */ +int snd_hda_parse_generic_codec(struct hda_codec *codec) +{ + struct hda_gspec *spec; + int err; + + spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); + if (spec == NULL) { + printk(KERN_ERR "hda_generic: can't allocate spec\n"); + return -ENOMEM; + } + codec->spec = spec; + INIT_LIST_HEAD(&spec->nid_list); + + if ((err = build_afg_tree(codec)) < 0) + goto error; + + if ((err = parse_input(codec)) < 0 || + (err = parse_output(codec)) < 0) + goto error; + + codec->patch_ops = generic_patch_ops; + + return 0; + + error: + snd_hda_generic_free(codec); + return err; +} diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c new file mode 100644 index 000000000000..d89647a3d449 --- /dev/null +++ b/sound/pci/hda/hda_intel.c @@ -0,0 +1,1449 @@ +/* + * + * hda_intel.c - Implementation of primary alsa driver code base for Intel HD Audio. + * + * Copyright(c) 2004 Intel Corporation. All rights reserved. + * + * Copyright (c) 2004 Takashi Iwai <tiwai@suse.de> + * PeiSen Hou <pshou@realtek.com.tw> + * + * 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. + * + * CONTACTS: + * + * Matt Jared matt.jared@intel.com + * Andy Kopp andy.kopp@intel.com + * Dan Kogan dan.d.kogan@intel.com + * + * CHANGES: + * + * 2004.12.01 Major rewrite by tiwai, merged the work of pshou + * + */ + +#include <sound/driver.h> +#include <asm/io.h> +#include <linux/delay.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/pci.h> +#include <sound/core.h> +#include <sound/initval.h> +#include "hda_codec.h" + + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static char *model[SNDRV_CARDS]; + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for Intel HD audio interface."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable Intel HD audio interface."); +module_param_array(model, charp, NULL, 0444); +MODULE_PARM_DESC(model, "Use the given board model."); + +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," + "{Intel, ICH6M}," + "{Intel, ICH7}}"); +MODULE_DESCRIPTION("Intel HDA driver"); + +#define SFX "hda-intel: " + +/* + * registers + */ +#define ICH6_REG_GCAP 0x00 +#define ICH6_REG_VMIN 0x02 +#define ICH6_REG_VMAJ 0x03 +#define ICH6_REG_OUTPAY 0x04 +#define ICH6_REG_INPAY 0x06 +#define ICH6_REG_GCTL 0x08 +#define ICH6_REG_WAKEEN 0x0c +#define ICH6_REG_STATESTS 0x0e +#define ICH6_REG_GSTS 0x10 +#define ICH6_REG_INTCTL 0x20 +#define ICH6_REG_INTSTS 0x24 +#define ICH6_REG_WALCLK 0x30 +#define ICH6_REG_SYNC 0x34 +#define ICH6_REG_CORBLBASE 0x40 +#define ICH6_REG_CORBUBASE 0x44 +#define ICH6_REG_CORBWP 0x48 +#define ICH6_REG_CORBRP 0x4A +#define ICH6_REG_CORBCTL 0x4c +#define ICH6_REG_CORBSTS 0x4d +#define ICH6_REG_CORBSIZE 0x4e + +#define ICH6_REG_RIRBLBASE 0x50 +#define ICH6_REG_RIRBUBASE 0x54 +#define ICH6_REG_RIRBWP 0x58 +#define ICH6_REG_RINTCNT 0x5a +#define ICH6_REG_RIRBCTL 0x5c +#define ICH6_REG_RIRBSTS 0x5d +#define ICH6_REG_RIRBSIZE 0x5e + +#define ICH6_REG_IC 0x60 +#define ICH6_REG_IR 0x64 +#define ICH6_REG_IRS 0x68 +#define ICH6_IRS_VALID (1<<1) +#define ICH6_IRS_BUSY (1<<0) + +#define ICH6_REG_DPLBASE 0x70 +#define ICH6_REG_DPUBASE 0x74 +#define ICH6_DPLBASE_ENABLE 0x1 /* Enable position buffer */ + +/* SD offset: SDI0=0x80, SDI1=0xa0, ... SDO3=0x160 */ +enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; + +/* stream register offsets from stream base */ +#define ICH6_REG_SD_CTL 0x00 +#define ICH6_REG_SD_STS 0x03 +#define ICH6_REG_SD_LPIB 0x04 +#define ICH6_REG_SD_CBL 0x08 +#define ICH6_REG_SD_LVI 0x0c +#define ICH6_REG_SD_FIFOW 0x0e +#define ICH6_REG_SD_FIFOSIZE 0x10 +#define ICH6_REG_SD_FORMAT 0x12 +#define ICH6_REG_SD_BDLPL 0x18 +#define ICH6_REG_SD_BDLPU 0x1c + +/* PCI space */ +#define ICH6_PCIREG_TCSEL 0x44 + +/* + * other constants + */ + +/* max number of SDs */ +#define MAX_ICH6_DEV 8 +/* max number of fragments - we may use more if allocating more pages for BDL */ +#define AZX_MAX_FRAG (PAGE_SIZE / (MAX_ICH6_DEV * 16)) +/* max buffer size - no h/w limit, you can increase as you like */ +#define AZX_MAX_BUF_SIZE (1024*1024*1024) +/* max number of PCM devics per card */ +#define AZX_MAX_PCMS 8 + +/* RIRB int mask: overrun[2], response[0] */ +#define RIRB_INT_RESPONSE 0x01 +#define RIRB_INT_OVERRUN 0x04 +#define RIRB_INT_MASK 0x05 + +/* STATESTS int mask: SD2,SD1,SD0 */ +#define STATESTS_INT_MASK 0x07 +#define AZX_MAX_CODECS 3 + +/* SD_CTL bits */ +#define SD_CTL_STREAM_RESET 0x01 /* stream reset bit */ +#define SD_CTL_DMA_START 0x02 /* stream DMA start bit */ +#define SD_CTL_STREAM_TAG_MASK (0xf << 20) +#define SD_CTL_STREAM_TAG_SHIFT 20 + +/* SD_CTL and SD_STS */ +#define SD_INT_DESC_ERR 0x10 /* descriptor error interrupt */ +#define SD_INT_FIFO_ERR 0x08 /* FIFO error interrupt */ +#define SD_INT_COMPLETE 0x04 /* completion interrupt */ +#define SD_INT_MASK (SD_INT_DESC_ERR|SD_INT_FIFO_ERR|SD_INT_COMPLETE) + +/* SD_STS */ +#define SD_STS_FIFO_READY 0x20 /* FIFO ready */ + +/* INTCTL and INTSTS */ +#define ICH6_INT_ALL_STREAM 0xff /* all stream interrupts */ +#define ICH6_INT_CTRL_EN 0x40000000 /* controller interrupt enable bit */ +#define ICH6_INT_GLOBAL_EN 0x80000000 /* global interrupt enable bit */ + +/* GCTL reset bit */ +#define ICH6_GCTL_RESET (1<<0) + +/* CORB/RIRB control, read/write pointer */ +#define ICH6_RBCTL_DMA_EN 0x02 /* enable DMA */ +#define ICH6_RBCTL_IRQ_EN 0x01 /* enable IRQ */ +#define ICH6_RBRWP_CLR 0x8000 /* read/write pointer clear */ +/* below are so far hardcoded - should read registers in future */ +#define ICH6_MAX_CORB_ENTRIES 256 +#define ICH6_MAX_RIRB_ENTRIES 256 + + +/* + * Use CORB/RIRB for communication from/to codecs. + * This is the way recommended by Intel (see below). + */ +#define USE_CORB_RIRB + +/* + * Define this if use the position buffer instead of reading SD_LPIB + * It's not used as default since SD_LPIB seems to give more accurate position + */ +/* #define USE_POSBUF */ + +/* + */ + +typedef struct snd_azx azx_t; +typedef struct snd_azx_rb azx_rb_t; +typedef struct snd_azx_dev azx_dev_t; + +struct snd_azx_dev { + u32 *bdl; /* virtual address of the BDL */ + dma_addr_t bdl_addr; /* physical address of the BDL */ + volatile u32 *posbuf; /* position buffer pointer */ + + unsigned int bufsize; /* size of the play buffer in bytes */ + unsigned int fragsize; /* size of each period in bytes */ + unsigned int frags; /* number for period in the play buffer */ + unsigned int fifo_size; /* FIFO size */ + + void __iomem *sd_addr; /* stream descriptor pointer */ + + u32 sd_int_sta_mask; /* stream int status mask */ + + /* pcm support */ + snd_pcm_substream_t *substream; /* assigned substream, set in PCM open */ + unsigned int format_val; /* format value to be set in the controller and the codec */ + unsigned char stream_tag; /* assigned stream */ + unsigned char index; /* stream index */ + + unsigned int opened: 1; + unsigned int running: 1; +}; + +/* CORB/RIRB */ +struct snd_azx_rb { + u32 *buf; /* CORB/RIRB buffer + * Each CORB entry is 4byte, RIRB is 8byte + */ + dma_addr_t addr; /* physical address of CORB/RIRB buffer */ + /* for RIRB */ + unsigned short rp, wp; /* read/write pointers */ + int cmds; /* number of pending requests */ + u32 res; /* last read value */ +}; + +struct snd_azx { + snd_card_t *card; + struct pci_dev *pci; + + /* pci resources */ + unsigned long addr; + void __iomem *remap_addr; + int irq; + + /* locks */ + spinlock_t reg_lock; + struct semaphore open_mutex; + + /* streams */ + azx_dev_t azx_dev[MAX_ICH6_DEV]; + + /* PCM */ + unsigned int pcm_devs; + snd_pcm_t *pcm[AZX_MAX_PCMS]; + + /* HD codec */ + unsigned short codec_mask; + struct hda_bus *bus; + + /* CORB/RIRB */ + azx_rb_t corb; + azx_rb_t rirb; + + /* BDL, CORB/RIRB and position buffers */ + struct snd_dma_buffer bdl; + struct snd_dma_buffer rb; + struct snd_dma_buffer posbuf; +}; + +/* + * macros for easy use + */ +#define azx_writel(chip,reg,value) \ + writel(value, (chip)->remap_addr + ICH6_REG_##reg) +#define azx_readl(chip,reg) \ + readl((chip)->remap_addr + ICH6_REG_##reg) +#define azx_writew(chip,reg,value) \ + writew(value, (chip)->remap_addr + ICH6_REG_##reg) +#define azx_readw(chip,reg) \ + readw((chip)->remap_addr + ICH6_REG_##reg) +#define azx_writeb(chip,reg,value) \ + writeb(value, (chip)->remap_addr + ICH6_REG_##reg) +#define azx_readb(chip,reg) \ + readb((chip)->remap_addr + ICH6_REG_##reg) + +#define azx_sd_writel(dev,reg,value) \ + writel(value, (dev)->sd_addr + ICH6_REG_##reg) +#define azx_sd_readl(dev,reg) \ + readl((dev)->sd_addr + ICH6_REG_##reg) +#define azx_sd_writew(dev,reg,value) \ + writew(value, (dev)->sd_addr + ICH6_REG_##reg) +#define azx_sd_readw(dev,reg) \ + readw((dev)->sd_addr + ICH6_REG_##reg) +#define azx_sd_writeb(dev,reg,value) \ + writeb(value, (dev)->sd_addr + ICH6_REG_##reg) +#define azx_sd_readb(dev,reg) \ + readb((dev)->sd_addr + ICH6_REG_##reg) + +/* for pcm support */ +#define get_azx_dev(substream) (azx_dev_t*)(substream->runtime->private_data) + +/* Get the upper 32bit of the given dma_addr_t + * Compiler should optimize and eliminate the code if dma_addr_t is 32bit + */ +#define upper_32bit(addr) (sizeof(addr) > 4 ? (u32)((addr) >> 32) : (u32)0) + + +/* + * Interface for HD codec + */ + +#ifdef USE_CORB_RIRB +/* + * CORB / RIRB interface + */ +static int azx_alloc_cmd_io(azx_t *chip) +{ + int err; + + /* single page (at least 4096 bytes) must suffice for both ringbuffes */ + err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), + PAGE_SIZE, &chip->rb); + if (err < 0) { + snd_printk(KERN_ERR SFX "cannot allocate CORB/RIRB\n"); + return err; + } + return 0; +} + +static void azx_init_cmd_io(azx_t *chip) +{ + /* CORB set up */ + chip->corb.addr = chip->rb.addr; + chip->corb.buf = (u32 *)chip->rb.area; + azx_writel(chip, CORBLBASE, (u32)chip->corb.addr); + azx_writel(chip, CORBUBASE, upper_32bit(chip->corb.addr)); + + /* set the corb write pointer to 0 */ + azx_writew(chip, CORBWP, 0); + /* reset the corb hw read pointer */ + azx_writew(chip, CORBRP, ICH6_RBRWP_CLR); + /* enable corb dma */ + azx_writeb(chip, CORBCTL, ICH6_RBCTL_DMA_EN); + + /* RIRB set up */ + chip->rirb.addr = chip->rb.addr + 2048; + chip->rirb.buf = (u32 *)(chip->rb.area + 2048); + azx_writel(chip, RIRBLBASE, (u32)chip->rirb.addr); + azx_writel(chip, RIRBUBASE, upper_32bit(chip->rirb.addr)); + + /* reset the rirb hw write pointer */ + azx_writew(chip, RIRBWP, ICH6_RBRWP_CLR); + /* set N=1, get RIRB response interrupt for new entry */ + azx_writew(chip, RINTCNT, 1); + /* enable rirb dma and response irq */ +#ifdef USE_CORB_RIRB + azx_writeb(chip, RIRBCTL, ICH6_RBCTL_DMA_EN | ICH6_RBCTL_IRQ_EN); +#else + azx_writeb(chip, RIRBCTL, ICH6_RBCTL_DMA_EN); +#endif + chip->rirb.rp = chip->rirb.cmds = 0; +} + +static void azx_free_cmd_io(azx_t *chip) +{ + /* disable ringbuffer DMAs */ + azx_writeb(chip, RIRBCTL, 0); + azx_writeb(chip, CORBCTL, 0); +} + +/* send a command */ +static int azx_send_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, + unsigned int verb, unsigned int para) +{ + azx_t *chip = codec->bus->private_data; + unsigned int wp; + u32 val; + + val = (u32)(codec->addr & 0x0f) << 28; + val |= (u32)direct << 27; + val |= (u32)nid << 20; + val |= verb << 8; + val |= para; + + /* add command to corb */ + wp = azx_readb(chip, CORBWP); + wp++; + wp %= ICH6_MAX_CORB_ENTRIES; + + spin_lock_irq(&chip->reg_lock); + chip->rirb.cmds++; + chip->corb.buf[wp] = cpu_to_le32(val); + azx_writel(chip, CORBWP, wp); + spin_unlock_irq(&chip->reg_lock); + + return 0; +} + +#define ICH6_RIRB_EX_UNSOL_EV (1<<4) + +/* retrieve RIRB entry - called from interrupt handler */ +static void azx_update_rirb(azx_t *chip) +{ + unsigned int rp, wp; + u32 res, res_ex; + + wp = azx_readb(chip, RIRBWP); + if (wp == chip->rirb.wp) + return; + chip->rirb.wp = wp; + + while (chip->rirb.rp != wp) { + chip->rirb.rp++; + chip->rirb.rp %= ICH6_MAX_RIRB_ENTRIES; + + rp = chip->rirb.rp << 1; /* an RIRB entry is 8-bytes */ + res_ex = le32_to_cpu(chip->rirb.buf[rp + 1]); + res = le32_to_cpu(chip->rirb.buf[rp]); + if (res_ex & ICH6_RIRB_EX_UNSOL_EV) + snd_hda_queue_unsol_event(chip->bus, res, res_ex); + else if (chip->rirb.cmds) { + chip->rirb.cmds--; + chip->rirb.res = res; + } + } +} + +/* receive a response */ +static unsigned int azx_get_response(struct hda_codec *codec) +{ + azx_t *chip = codec->bus->private_data; + int timeout = 50; + + while (chip->rirb.cmds) { + if (! --timeout) { + snd_printk(KERN_ERR "azx_get_response timeout\n"); + chip->rirb.rp = azx_readb(chip, RIRBWP); + chip->rirb.cmds = 0; + return -1; + } + msleep(1); + } + return chip->rirb.res; /* the last value */ +} + +#else +/* + * Use the single immediate command instead of CORB/RIRB for simplicity + * + * Note: according to Intel, this is not preferred use. The command was + * intended for the BIOS only, and may get confused with unsolicited + * responses. So, we shouldn't use it for normal operation from the + * driver. + * I left the codes, however, for debugging/testing purposes. + */ + +#define azx_alloc_cmd_io(chip) 0 +#define azx_init_cmd_io(chip) +#define azx_free_cmd_io(chip) + +/* send a command */ +static int azx_send_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, + unsigned int verb, unsigned int para) +{ + azx_t *chip = codec->bus->private_data; + u32 val; + int timeout = 50; + + val = (u32)(codec->addr & 0x0f) << 28; + val |= (u32)direct << 27; + val |= (u32)nid << 20; + val |= verb << 8; + val |= para; + + while (timeout--) { + /* check ICB busy bit */ + if (! (azx_readw(chip, IRS) & ICH6_IRS_BUSY)) { + /* Clear IRV valid bit */ + azx_writew(chip, IRS, azx_readw(chip, IRS) | ICH6_IRS_VALID); + azx_writel(chip, IC, val); + azx_writew(chip, IRS, azx_readw(chip, IRS) | ICH6_IRS_BUSY); + return 0; + } + udelay(1); + } + snd_printd(SFX "send_cmd timeout: IRS=0x%x, val=0x%x\n", azx_readw(chip, IRS), val); + return -EIO; +} + +/* receive a response */ +static unsigned int azx_get_response(struct hda_codec *codec) +{ + azx_t *chip = codec->bus->private_data; + int timeout = 50; + + while (timeout--) { + /* check IRV busy bit */ + if (azx_readw(chip, IRS) & ICH6_IRS_VALID) + return azx_readl(chip, IR); + udelay(1); + } + snd_printd(SFX "get_response timeout: IRS=0x%x\n", azx_readw(chip, IRS)); + return (unsigned int)-1; +} + +#define azx_update_rirb(chip) + +#endif /* USE_CORB_RIRB */ + +/* reset codec link */ +static int azx_reset(azx_t *chip) +{ + int count; + + /* reset controller */ + azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~ICH6_GCTL_RESET); + + count = 50; + while (azx_readb(chip, GCTL) && --count) + msleep(1); + + /* delay for >= 100us for codec PLL to settle per spec + * Rev 0.9 section 5.5.1 + */ + msleep(1); + + /* Bring controller out of reset */ + azx_writeb(chip, GCTL, azx_readb(chip, GCTL) | ICH6_GCTL_RESET); + + count = 50; + while (! azx_readb(chip, GCTL) && --count) + msleep(1); + + /* Brent Chartrand said to wait >= 540us for codecs to intialize */ + msleep(1); + + /* check to see if controller is ready */ + if (! azx_readb(chip, GCTL)) { + snd_printd("azx_reset: controller not ready!\n"); + return -EBUSY; + } + + /* detect codecs */ + if (! chip->codec_mask) { + chip->codec_mask = azx_readw(chip, STATESTS); + snd_printdd("codec_mask = 0x%x\n", chip->codec_mask); + } + + return 0; +} + + +/* + * Lowlevel interface + */ + +/* enable interrupts */ +static void azx_int_enable(azx_t *chip) +{ + /* enable controller CIE and GIE */ + azx_writel(chip, INTCTL, azx_readl(chip, INTCTL) | + ICH6_INT_CTRL_EN | ICH6_INT_GLOBAL_EN); +} + +/* disable interrupts */ +static void azx_int_disable(azx_t *chip) +{ + int i; + + /* disable interrupts in stream descriptor */ + for (i = 0; i < MAX_ICH6_DEV; i++) { + azx_dev_t *azx_dev = &chip->azx_dev[i]; + azx_sd_writeb(azx_dev, SD_CTL, + azx_sd_readb(azx_dev, SD_CTL) & ~SD_INT_MASK); + } + + /* disable SIE for all streams */ + azx_writeb(chip, INTCTL, 0); + + /* disable controller CIE and GIE */ + azx_writel(chip, INTCTL, azx_readl(chip, INTCTL) & + ~(ICH6_INT_CTRL_EN | ICH6_INT_GLOBAL_EN)); +} + +/* clear interrupts */ +static void azx_int_clear(azx_t *chip) +{ + int i; + + /* clear stream status */ + for (i = 0; i < MAX_ICH6_DEV; i++) { + azx_dev_t *azx_dev = &chip->azx_dev[i]; + azx_sd_writeb(azx_dev, SD_STS, SD_INT_MASK); + } + + /* clear STATESTS */ + azx_writeb(chip, STATESTS, STATESTS_INT_MASK); + + /* clear rirb status */ + azx_writeb(chip, RIRBSTS, RIRB_INT_MASK); + + /* clear int status */ + azx_writel(chip, INTSTS, ICH6_INT_CTRL_EN | ICH6_INT_ALL_STREAM); +} + +/* start a stream */ +static void azx_stream_start(azx_t *chip, azx_dev_t *azx_dev) +{ + /* enable SIE */ + azx_writeb(chip, INTCTL, + azx_readb(chip, INTCTL) | (1 << azx_dev->index)); + /* set DMA start and interrupt mask */ + azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) | + SD_CTL_DMA_START | SD_INT_MASK); +} + +/* stop a stream */ +static void azx_stream_stop(azx_t *chip, azx_dev_t *azx_dev) +{ + /* stop DMA */ + azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) & + ~(SD_CTL_DMA_START | SD_INT_MASK)); + azx_sd_writeb(azx_dev, SD_STS, SD_INT_MASK); /* to be sure */ + /* disable SIE */ + azx_writeb(chip, INTCTL, + azx_readb(chip, INTCTL) & ~(1 << azx_dev->index)); +} + + +/* + * initialize the chip + */ +static void azx_init_chip(azx_t *chip) +{ + unsigned char tcsel_reg; + + /* Clear bits 0-2 of PCI register TCSEL (at offset 0x44) + * TCSEL == Traffic Class Select Register, which sets PCI express QOS + * Ensuring these bits are 0 clears playback static on some HD Audio codecs + */ + pci_read_config_byte (chip->pci, ICH6_PCIREG_TCSEL, &tcsel_reg); + pci_write_config_byte(chip->pci, ICH6_PCIREG_TCSEL, tcsel_reg & 0xf8); + + /* reset controller */ + azx_reset(chip); + + /* initialize interrupts */ + azx_int_clear(chip); + azx_int_enable(chip); + + /* initialize the codec command I/O */ + azx_init_cmd_io(chip); + +#ifdef USE_POSBUF + /* program the position buffer */ + azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr); + azx_writel(chip, DPUBASE, upper_32bit(chip->posbuf.addr)); +#endif +} + + +/* + * interrupt handler + */ +static irqreturn_t azx_interrupt(int irq, void* dev_id, struct pt_regs *regs) +{ + azx_t *chip = dev_id; + azx_dev_t *azx_dev; + u32 status; + int i; + + spin_lock(&chip->reg_lock); + + status = azx_readl(chip, INTSTS); + if (status == 0) { + spin_unlock(&chip->reg_lock); + return IRQ_NONE; + } + + for (i = 0; i < MAX_ICH6_DEV; i++) { + azx_dev = &chip->azx_dev[i]; + if (status & azx_dev->sd_int_sta_mask) { + azx_sd_writeb(azx_dev, SD_STS, SD_INT_MASK); + if (azx_dev->substream && azx_dev->running) { + spin_unlock(&chip->reg_lock); + snd_pcm_period_elapsed(azx_dev->substream); + spin_lock(&chip->reg_lock); + } + } + } + + /* clear rirb int */ + status = azx_readb(chip, RIRBSTS); + if (status & RIRB_INT_MASK) { + if (status & RIRB_INT_RESPONSE) + azx_update_rirb(chip); + azx_writeb(chip, RIRBSTS, RIRB_INT_MASK); + } + +#if 0 + /* clear state status int */ + if (azx_readb(chip, STATESTS) & 0x04) + azx_writeb(chip, STATESTS, 0x04); +#endif + spin_unlock(&chip->reg_lock); + + return IRQ_HANDLED; +} + + +/* + * set up BDL entries + */ +static void azx_setup_periods(azx_dev_t *azx_dev) +{ + u32 *bdl = azx_dev->bdl; + dma_addr_t dma_addr = azx_dev->substream->runtime->dma_addr; + int idx; + + /* reset BDL address */ + azx_sd_writel(azx_dev, SD_BDLPL, 0); + azx_sd_writel(azx_dev, SD_BDLPU, 0); + + /* program the initial BDL entries */ + for (idx = 0; idx < azx_dev->frags; idx++) { + unsigned int off = idx << 2; /* 4 dword step */ + dma_addr_t addr = dma_addr + idx * azx_dev->fragsize; + /* program the address field of the BDL entry */ + bdl[off] = cpu_to_le32((u32)addr); + bdl[off+1] = cpu_to_le32(upper_32bit(addr)); + + /* program the size field of the BDL entry */ + bdl[off+2] = cpu_to_le32(azx_dev->fragsize); + + /* program the IOC to enable interrupt when buffer completes */ + bdl[off+3] = cpu_to_le32(0x01); + } +} + +/* + * set up the SD for streaming + */ +static int azx_setup_controller(azx_t *chip, azx_dev_t *azx_dev) +{ + unsigned char val; + int timeout; + + /* make sure the run bit is zero for SD */ + azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) & ~SD_CTL_DMA_START); + /* reset stream */ + azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) | SD_CTL_STREAM_RESET); + udelay(3); + timeout = 300; + while (!((val = azx_sd_readb(azx_dev, SD_CTL)) & SD_CTL_STREAM_RESET) && + --timeout) + ; + val &= ~SD_CTL_STREAM_RESET; + azx_sd_writeb(azx_dev, SD_CTL, val); + udelay(3); + + timeout = 300; + /* waiting for hardware to report that the stream is out of reset */ + while (((val = azx_sd_readb(azx_dev, SD_CTL)) & SD_CTL_STREAM_RESET) && + --timeout) + ; + + /* program the stream_tag */ + azx_sd_writel(azx_dev, SD_CTL, + (azx_sd_readl(azx_dev, SD_CTL) & ~SD_CTL_STREAM_TAG_MASK) | + (azx_dev->stream_tag << SD_CTL_STREAM_TAG_SHIFT)); + + /* program the length of samples in cyclic buffer */ + azx_sd_writel(azx_dev, SD_CBL, azx_dev->bufsize); + + /* program the stream format */ + /* this value needs to be the same as the one programmed */ + azx_sd_writew(azx_dev, SD_FORMAT, azx_dev->format_val); + + /* program the stream LVI (last valid index) of the BDL */ + azx_sd_writew(azx_dev, SD_LVI, azx_dev->frags - 1); + + /* program the BDL address */ + /* lower BDL address */ + azx_sd_writel(azx_dev, SD_BDLPL, (u32)azx_dev->bdl_addr); + /* upper BDL address */ + azx_sd_writel(azx_dev, SD_BDLPU, upper_32bit(azx_dev->bdl_addr)); + +#ifdef USE_POSBUF + /* enable the position buffer */ + if (! (azx_readl(chip, DPLBASE) & ICH6_DPLBASE_ENABLE)) + azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr | ICH6_DPLBASE_ENABLE); +#endif + /* set the interrupt enable bits in the descriptor control register */ + azx_sd_writel(azx_dev, SD_CTL, azx_sd_readl(azx_dev, SD_CTL) | SD_INT_MASK); + + return 0; +} + + +/* + * Codec initialization + */ + +static int __devinit azx_codec_create(azx_t *chip, const char *model) +{ + struct hda_bus_template bus_temp; + int c, codecs, err; + + memset(&bus_temp, 0, sizeof(bus_temp)); + bus_temp.private_data = chip; + bus_temp.modelname = model; + bus_temp.pci = chip->pci; + bus_temp.ops.command = azx_send_cmd; + bus_temp.ops.get_response = azx_get_response; + + if ((err = snd_hda_bus_new(chip->card, &bus_temp, &chip->bus)) < 0) + return err; + + codecs = 0; + for (c = 0; c < AZX_MAX_CODECS; c++) { + if (chip->codec_mask & (1 << c)) { + err = snd_hda_codec_new(chip->bus, c, NULL); + if (err < 0) + continue; + codecs++; + } + } + if (! codecs) { + snd_printk(KERN_ERR SFX "no codecs initialized\n"); + return -ENXIO; + } + + return 0; +} + + +/* + * PCM support + */ + +/* assign a stream for the PCM */ +static inline azx_dev_t *azx_assign_device(azx_t *chip, int stream) +{ + int dev, i; + dev = stream == SNDRV_PCM_STREAM_PLAYBACK ? 4 : 0; + for (i = 0; i < 4; i++, dev++) + if (! chip->azx_dev[dev].opened) { + chip->azx_dev[dev].opened = 1; + return &chip->azx_dev[dev]; + } + return NULL; +} + +/* release the assigned stream */ +static inline void azx_release_device(azx_dev_t *azx_dev) +{ + azx_dev->opened = 0; +} + +static snd_pcm_hardware_t azx_pcm_hw = { + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = AZX_MAX_BUF_SIZE, + .period_bytes_min = 128, + .period_bytes_max = AZX_MAX_BUF_SIZE / 2, + .periods_min = 2, + .periods_max = AZX_MAX_FRAG, + .fifo_size = 0, +}; + +struct azx_pcm { + azx_t *chip; + struct hda_codec *codec; + struct hda_pcm_stream *hinfo[2]; +}; + +static int azx_pcm_open(snd_pcm_substream_t *substream) +{ + struct azx_pcm *apcm = snd_pcm_substream_chip(substream); + struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; + azx_t *chip = apcm->chip; + azx_dev_t *azx_dev; + snd_pcm_runtime_t *runtime = substream->runtime; + unsigned long flags; + int err; + + down(&chip->open_mutex); + azx_dev = azx_assign_device(chip, substream->stream); + if (azx_dev == NULL) { + up(&chip->open_mutex); + return -EBUSY; + } + runtime->hw = azx_pcm_hw; + runtime->hw.channels_min = hinfo->channels_min; + runtime->hw.channels_max = hinfo->channels_max; + runtime->hw.formats = hinfo->formats; + runtime->hw.rates = hinfo->rates; + snd_pcm_limit_hw_rates(runtime); + snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); + if ((err = hinfo->ops.open(hinfo, apcm->codec, substream)) < 0) { + azx_release_device(azx_dev); + up(&chip->open_mutex); + return err; + } + spin_lock_irqsave(&chip->reg_lock, flags); + azx_dev->substream = substream; + azx_dev->running = 0; + spin_unlock_irqrestore(&chip->reg_lock, flags); + + runtime->private_data = azx_dev; + up(&chip->open_mutex); + return 0; +} + +static int azx_pcm_close(snd_pcm_substream_t *substream) +{ + struct azx_pcm *apcm = snd_pcm_substream_chip(substream); + struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; + azx_t *chip = apcm->chip; + azx_dev_t *azx_dev = get_azx_dev(substream); + unsigned long flags; + + down(&chip->open_mutex); + spin_lock_irqsave(&chip->reg_lock, flags); + azx_dev->substream = NULL; + azx_dev->running = 0; + spin_unlock_irqrestore(&chip->reg_lock, flags); + azx_release_device(azx_dev); + hinfo->ops.close(hinfo, apcm->codec, substream); + up(&chip->open_mutex); + return 0; +} + +static int azx_pcm_hw_params(snd_pcm_substream_t *substream, snd_pcm_hw_params_t *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); +} + +static int azx_pcm_hw_free(snd_pcm_substream_t *substream) +{ + struct azx_pcm *apcm = snd_pcm_substream_chip(substream); + azx_dev_t *azx_dev = get_azx_dev(substream); + struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; + + /* reset BDL address */ + azx_sd_writel(azx_dev, SD_BDLPL, 0); + azx_sd_writel(azx_dev, SD_BDLPU, 0); + azx_sd_writel(azx_dev, SD_CTL, 0); + + hinfo->ops.cleanup(hinfo, apcm->codec, substream); + + return snd_pcm_lib_free_pages(substream); +} + +static int azx_pcm_prepare(snd_pcm_substream_t *substream) +{ + struct azx_pcm *apcm = snd_pcm_substream_chip(substream); + azx_t *chip = apcm->chip; + azx_dev_t *azx_dev = get_azx_dev(substream); + struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; + snd_pcm_runtime_t *runtime = substream->runtime; + + azx_dev->bufsize = snd_pcm_lib_buffer_bytes(substream); + azx_dev->fragsize = snd_pcm_lib_period_bytes(substream); + azx_dev->frags = azx_dev->bufsize / azx_dev->fragsize; + azx_dev->format_val = snd_hda_calc_stream_format(runtime->rate, + runtime->channels, + runtime->format, + hinfo->maxbps); + if (! azx_dev->format_val) { + snd_printk(KERN_ERR SFX "invalid format_val, rate=%d, ch=%d, format=%d\n", + runtime->rate, runtime->channels, runtime->format); + return -EINVAL; + } + + snd_printdd("azx_pcm_prepare: bufsize=0x%x, fragsize=0x%x, format=0x%x\n", + azx_dev->bufsize, azx_dev->fragsize, azx_dev->format_val); + azx_setup_periods(azx_dev); + azx_setup_controller(chip, azx_dev); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + azx_dev->fifo_size = azx_sd_readw(azx_dev, SD_FIFOSIZE) + 1; + else + azx_dev->fifo_size = 0; + + return hinfo->ops.prepare(hinfo, apcm->codec, azx_dev->stream_tag, + azx_dev->format_val, substream); +} + +static int azx_pcm_trigger(snd_pcm_substream_t *substream, int cmd) +{ + struct azx_pcm *apcm = snd_pcm_substream_chip(substream); + azx_dev_t *azx_dev = get_azx_dev(substream); + azx_t *chip = apcm->chip; + int err = 0; + + spin_lock(&chip->reg_lock); + switch (cmd) { + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_START: + azx_stream_start(chip, azx_dev); + azx_dev->running = 1; + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_STOP: + azx_stream_stop(chip, azx_dev); + azx_dev->running = 0; + break; + default: + err = -EINVAL; + } + spin_unlock(&chip->reg_lock); + if (cmd == SNDRV_PCM_TRIGGER_PAUSE_PUSH || + cmd == SNDRV_PCM_TRIGGER_STOP) { + int timeout = 5000; + while (azx_sd_readb(azx_dev, SD_CTL) & SD_CTL_DMA_START && --timeout) + ; + } + return err; +} + +static snd_pcm_uframes_t azx_pcm_pointer(snd_pcm_substream_t *substream) +{ + azx_dev_t *azx_dev = get_azx_dev(substream); + unsigned int pos; + +#ifdef USE_POSBUF + /* use the position buffer */ + pos = *azx_dev->posbuf; +#else + /* read LPIB */ + pos = azx_sd_readl(azx_dev, SD_LPIB) + azx_dev->fifo_size; +#endif + if (pos >= azx_dev->bufsize) + pos = 0; + return bytes_to_frames(substream->runtime, pos); +} + +static snd_pcm_ops_t azx_pcm_ops = { + .open = azx_pcm_open, + .close = azx_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = azx_pcm_hw_params, + .hw_free = azx_pcm_hw_free, + .prepare = azx_pcm_prepare, + .trigger = azx_pcm_trigger, + .pointer = azx_pcm_pointer, +}; + +static void azx_pcm_free(snd_pcm_t *pcm) +{ + kfree(pcm->private_data); +} + +static int __devinit create_codec_pcm(azx_t *chip, struct hda_codec *codec, + struct hda_pcm *cpcm, int pcm_dev) +{ + int err; + snd_pcm_t *pcm; + struct azx_pcm *apcm; + + snd_assert(cpcm->stream[0].substreams || cpcm->stream[1].substreams, return -EINVAL); + snd_assert(cpcm->name, return -EINVAL); + + err = snd_pcm_new(chip->card, cpcm->name, pcm_dev, + cpcm->stream[0].substreams, cpcm->stream[1].substreams, + &pcm); + if (err < 0) + return err; + strcpy(pcm->name, cpcm->name); + apcm = kmalloc(sizeof(*apcm), GFP_KERNEL); + if (apcm == NULL) + return -ENOMEM; + apcm->chip = chip; + apcm->codec = codec; + apcm->hinfo[0] = &cpcm->stream[0]; + apcm->hinfo[1] = &cpcm->stream[1]; + pcm->private_data = apcm; + pcm->private_free = azx_pcm_free; + if (cpcm->stream[0].substreams) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &azx_pcm_ops); + if (cpcm->stream[1].substreams) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &azx_pcm_ops); + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(chip->pci), + 1024 * 64, 1024 * 128); + chip->pcm[pcm_dev] = pcm; + + return 0; +} + +static int __devinit azx_pcm_create(azx_t *chip) +{ + struct list_head *p; + struct hda_codec *codec; + int c, err; + int pcm_dev; + + if ((err = snd_hda_build_pcms(chip->bus)) < 0) + return err; + + pcm_dev = 0; + list_for_each(p, &chip->bus->codec_list) { + codec = list_entry(p, struct hda_codec, list); + for (c = 0; c < codec->num_pcms; c++) { + if (pcm_dev >= AZX_MAX_PCMS) { + snd_printk(KERN_ERR SFX "Too many PCMs\n"); + return -EINVAL; + } + err = create_codec_pcm(chip, codec, &codec->pcm_info[c], pcm_dev); + if (err < 0) + return err; + pcm_dev++; + } + } + return 0; +} + +/* + * mixer creation - all stuff is implemented in hda module + */ +static int __devinit azx_mixer_create(azx_t *chip) +{ + return snd_hda_build_controls(chip->bus); +} + + +/* + * initialize SD streams + */ +static int __devinit azx_init_stream(azx_t *chip) +{ + int i; + + /* initialize each stream (aka device) + * assign the starting bdl address to each stream (device) and initialize + */ + for (i = 0; i < MAX_ICH6_DEV; i++) { + unsigned int off = sizeof(u32) * (i * AZX_MAX_FRAG * 4); + azx_dev_t *azx_dev = &chip->azx_dev[i]; + azx_dev->bdl = (u32 *)(chip->bdl.area + off); + azx_dev->bdl_addr = chip->bdl.addr + off; +#ifdef USE_POSBUF + azx_dev->posbuf = (volatile u32 *)(chip->posbuf.area + i * 8); +#endif + /* offset: SDI0=0x80, SDI1=0xa0, ... SDO3=0x160 */ + azx_dev->sd_addr = chip->remap_addr + (0x20 * i + 0x80); + /* int mask: SDI0=0x01, SDI1=0x02, ... SDO3=0x80 */ + azx_dev->sd_int_sta_mask = 1 << i; + /* stream tag: must be non-zero and unique */ + azx_dev->index = i; + azx_dev->stream_tag = i + 1; + } + + return 0; +} + + +#ifdef CONFIG_PM +/* + * power management + */ +static int azx_suspend(snd_card_t *card, pm_message_t state) +{ + azx_t *chip = card->pm_private_data; + int i; + + for (i = 0; i < chip->pcm_devs; i++) + if (chip->pcm[i]) + snd_pcm_suspend_all(chip->pcm[i]); + snd_hda_suspend(chip->bus, state); + azx_free_cmd_io(chip); + pci_disable_device(chip->pci); + return 0; +} + +static int azx_resume(snd_card_t *card) +{ + azx_t *chip = card->pm_private_data; + + pci_enable_device(chip->pci); + pci_set_master(chip->pci); + azx_init_chip(chip); + snd_hda_resume(chip->bus); + return 0; +} +#endif /* CONFIG_PM */ + + +/* + * destructor + */ +static int azx_free(azx_t *chip) +{ + if (chip->remap_addr) { + int i; + + for (i = 0; i < MAX_ICH6_DEV; i++) + azx_stream_stop(chip, &chip->azx_dev[i]); + + /* disable interrupts */ + azx_int_disable(chip); + azx_int_clear(chip); + + /* disable CORB/RIRB */ + azx_free_cmd_io(chip); + + /* disable position buffer */ + azx_writel(chip, DPLBASE, 0); + azx_writel(chip, DPUBASE, 0); + + /* wait a little for interrupts to finish */ + msleep(1); + + iounmap(chip->remap_addr); + } + + if (chip->irq >= 0) + free_irq(chip->irq, (void*)chip); + + if (chip->bdl.area) + snd_dma_free_pages(&chip->bdl); + if (chip->rb.area) + snd_dma_free_pages(&chip->rb); +#ifdef USE_POSBUF + if (chip->posbuf.area) + snd_dma_free_pages(&chip->posbuf); +#endif + pci_release_regions(chip->pci); + pci_disable_device(chip->pci); + kfree(chip); + + return 0; +} + +static int azx_dev_free(snd_device_t *device) +{ + return azx_free(device->device_data); +} + +/* + * constructor + */ +static int __devinit azx_create(snd_card_t *card, struct pci_dev *pci, azx_t **rchip) +{ + azx_t *chip; + int err = 0; + static snd_device_ops_t ops = { + .dev_free = azx_dev_free, + }; + + *rchip = NULL; + + if ((err = pci_enable_device(pci)) < 0) + return err; + + chip = kcalloc(1, sizeof(*chip), GFP_KERNEL); + + if (NULL == chip) { + snd_printk(KERN_ERR SFX "cannot allocate chip\n"); + pci_disable_device(pci); + return -ENOMEM; + } + + spin_lock_init(&chip->reg_lock); + init_MUTEX(&chip->open_mutex); + chip->card = card; + chip->pci = pci; + chip->irq = -1; + + if ((err = pci_request_regions(pci, "ICH HD audio")) < 0) { + kfree(chip); + pci_disable_device(pci); + return err; + } + + chip->addr = pci_resource_start(pci,0); + chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci,0)); + if (chip->remap_addr == NULL) { + snd_printk(KERN_ERR SFX "ioremap error\n"); + err = -ENXIO; + goto errout; + } + + if (request_irq(pci->irq, azx_interrupt, SA_INTERRUPT|SA_SHIRQ, + "HDA Intel", (void*)chip)) { + snd_printk(KERN_ERR SFX "unable to grab IRQ %d\n", pci->irq); + err = -EBUSY; + goto errout; + } + chip->irq = pci->irq; + + pci_set_master(pci); + synchronize_irq(chip->irq); + + /* allocate memory for the BDL for each stream */ + if ((err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), + PAGE_SIZE, &chip->bdl)) < 0) { + snd_printk(KERN_ERR SFX "cannot allocate BDL\n"); + goto errout; + } +#ifdef USE_POSBUF + /* allocate memory for the position buffer */ + if ((err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), + MAX_ICH6_DEV * 8, &chip->posbuf)) < 0) { + snd_printk(KERN_ERR SFX "cannot allocate posbuf\n"); + goto errout; + } +#endif + /* allocate CORB/RIRB */ + if ((err = azx_alloc_cmd_io(chip)) < 0) + goto errout; + + /* initialize streams */ + azx_init_stream(chip); + + /* initialize chip */ + azx_init_chip(chip); + + /* codec detection */ + if (! chip->codec_mask) { + snd_printk(KERN_ERR SFX "no codecs found!\n"); + err = -ENODEV; + goto errout; + } + + if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) <0) { + snd_printk(KERN_ERR SFX "Error creating device [card]!\n"); + goto errout; + } + + *rchip = chip; + return 0; + + errout: + azx_free(chip); + return err; +} + +static int __devinit azx_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) +{ + static int dev; + snd_card_t *card; + azx_t *chip; + int err = 0; + + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (! enable[dev]) { + dev++; + return -ENOENT; + } + + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + if (NULL == card) { + snd_printk(KERN_ERR SFX "Error creating card!\n"); + return -ENOMEM; + } + + if ((err = azx_create(card, pci, &chip)) < 0) { + snd_card_free(card); + return err; + } + + strcpy(card->driver, "HDA-Intel"); + strcpy(card->shortname, "HDA Intel"); + sprintf(card->longname, "%s at 0x%lx irq %i", card->shortname, chip->addr, chip->irq); + + /* create codec instances */ + if ((err = azx_codec_create(chip, model[dev])) < 0) { + snd_card_free(card); + return err; + } + + /* create PCM streams */ + if ((err = azx_pcm_create(chip)) < 0) { + snd_card_free(card); + return err; + } + + /* create mixer controls */ + if ((err = azx_mixer_create(chip)) < 0) { + snd_card_free(card); + return err; + } + + snd_card_set_pm_callback(card, azx_suspend, azx_resume, chip); + snd_card_set_dev(card, &pci->dev); + + if ((err = snd_card_register(card)) < 0) { + snd_card_free(card); + return err; + } + + pci_set_drvdata(pci, card); + dev++; + + return err; +} + +static void __devexit azx_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +/* PCI IDs */ +static struct pci_device_id azx_ids[] = { + { 0x8086, 0x2668, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* ICH6 */ + { 0x8086, 0x27d8, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* ICH7 */ + { 0, } +}; +MODULE_DEVICE_TABLE(pci, azx_ids); + +/* pci_driver definition */ +static struct pci_driver driver = { + .name = "HDA Intel", + .id_table = azx_ids, + .probe = azx_probe, + .remove = __devexit_p(azx_remove), + SND_PCI_PM_CALLBACKS +}; + +static int __init alsa_card_azx_init(void) +{ + return pci_module_init(&driver); +} + +static void __exit alsa_card_azx_exit(void) +{ + pci_unregister_driver(&driver); +} + +module_init(alsa_card_azx_init) +module_exit(alsa_card_azx_exit) diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h new file mode 100644 index 000000000000..7c7b849875a0 --- /dev/null +++ b/sound/pci/hda/hda_local.h @@ -0,0 +1,161 @@ +/* + * Universal Interface for Intel High Definition Audio Codec + * + * Local helper functions + * + * 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_LOCAL_H +#define __SOUND_HDA_LOCAL_H + +/* + * for mixer controls + */ +#define HDA_COMPOSE_AMP_VAL(nid,chs,idx,dir) ((nid) | ((chs)<<16) | ((dir)<<18) | ((idx)<<19)) +#define HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ + .info = snd_hda_mixer_amp_volume_info, \ + .get = snd_hda_mixer_amp_volume_get, \ + .put = snd_hda_mixer_amp_volume_put, \ + .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, xindex, direction) } +#define HDA_CODEC_VOLUME_IDX(xname, xcidx, nid, xindex, direction) \ + HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, 3, xindex, direction) +#define HDA_CODEC_VOLUME_MONO(xname, nid, channel, xindex, direction) \ + HDA_CODEC_VOLUME_MONO_IDX(xname, 0, nid, channel, xindex, direction) +#define HDA_CODEC_VOLUME(xname, nid, xindex, direction) \ + HDA_CODEC_VOLUME_MONO(xname, nid, 3, xindex, direction) +#define HDA_CODEC_MUTE_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ + .info = snd_hda_mixer_amp_switch_info, \ + .get = snd_hda_mixer_amp_switch_get, \ + .put = snd_hda_mixer_amp_switch_put, \ + .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, xindex, direction) } +#define HDA_CODEC_MUTE_IDX(xname, xcidx, nid, xindex, direction) \ + HDA_CODEC_MUTE_MONO_IDX(xname, xcidx, nid, 3, xindex, direction) +#define HDA_CODEC_MUTE_MONO(xname, nid, channel, xindex, direction) \ + HDA_CODEC_MUTE_MONO_IDX(xname, 0, nid, channel, xindex, direction) +#define HDA_CODEC_MUTE(xname, nid, xindex, direction) \ + HDA_CODEC_MUTE_MONO(xname, nid, 3, xindex, direction) + +int snd_hda_mixer_amp_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo); +int snd_hda_mixer_amp_volume_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol); +int snd_hda_mixer_amp_volume_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol); +int snd_hda_mixer_amp_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo); +int snd_hda_mixer_amp_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol); +int snd_hda_mixer_amp_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol); + +int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid); +int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid); + +/* + * input MUX helper + */ +#define HDA_MAX_NUM_INPUTS 8 +struct hda_input_mux_item { + const char *label; + unsigned int index; +}; +struct hda_input_mux { + unsigned int num_items; + struct hda_input_mux_item items[HDA_MAX_NUM_INPUTS]; +}; + +int snd_hda_input_mux_info(const struct hda_input_mux *imux, snd_ctl_elem_info_t *uinfo); +int snd_hda_input_mux_put(struct hda_codec *codec, const struct hda_input_mux *imux, + snd_ctl_elem_value_t *ucontrol, hda_nid_t nid, + unsigned int *cur_val); + +/* + * Multi-channel / digital-out PCM helper + */ + +enum { HDA_FRONT, HDA_REAR, HDA_CLFE, HDA_SIDE }; /* index for dac_nidx */ +enum { HDA_DIG_NONE, HDA_DIG_EXCLUSIVE, HDA_DIG_ANALOG_DUP }; /* dig_out_used */ + +struct hda_multi_out { + int num_dacs; /* # of DACs, must be more than 1 */ + hda_nid_t *dac_nids; /* DAC list */ + hda_nid_t hp_nid; /* optional DAC for HP, 0 when not exists */ + hda_nid_t dig_out_nid; /* digital out audio widget */ + int max_channels; /* currently supported analog channels */ + int dig_out_used; /* current usage of digital out (HDA_DIG_XXX) */ +}; + +int snd_hda_multi_out_dig_open(struct hda_codec *codec, struct hda_multi_out *mout); +int snd_hda_multi_out_dig_close(struct hda_codec *codec, struct hda_multi_out *mout); +int snd_hda_multi_out_analog_open(struct hda_codec *codec, struct hda_multi_out *mout, + snd_pcm_substream_t *substream); +int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, struct hda_multi_out *mout, + unsigned int stream_tag, + unsigned int format, + snd_pcm_substream_t *substream); +int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, struct hda_multi_out *mout); + +/* + * generic codec parser + */ +int snd_hda_parse_generic_codec(struct hda_codec *codec); + +/* + * generic proc interface + */ +#ifdef CONFIG_PROC_FS +int snd_hda_codec_proc_new(struct hda_codec *codec); +#else +static inline int snd_hda_codec_proc_new(struct hda_codec *codec) { return 0; } +#endif + +/* + * Misc + */ +struct hda_board_config { + const char *modelname; + int config; + unsigned short pci_vendor; + unsigned short pci_device; +}; + +int snd_hda_check_board_config(struct hda_codec *codec, struct hda_board_config *tbl); +int snd_hda_add_new_ctls(struct hda_codec *codec, snd_kcontrol_new_t *knew); + +/* + * power management + */ +#ifdef CONFIG_PM +int snd_hda_resume_ctls(struct hda_codec *codec, snd_kcontrol_new_t *knew); +int snd_hda_resume_spdif_out(struct hda_codec *codec); +int snd_hda_resume_spdif_in(struct hda_codec *codec); +#endif + +/* + * unsolicited event handler + */ + +#define HDA_UNSOL_QUEUE_SIZE 64 + +struct hda_bus_unsolicited { + /* ring buffer */ + u32 queue[HDA_UNSOL_QUEUE_SIZE * 2]; + unsigned int rp, wp; + + /* workqueue */ + struct workqueue_struct *workq; + struct work_struct work; +}; + +#endif /* __SOUND_HDA_LOCAL_H */ diff --git a/sound/pci/hda/hda_patch.h b/sound/pci/hda/hda_patch.h new file mode 100644 index 000000000000..cf6abce42bc9 --- /dev/null +++ b/sound/pci/hda/hda_patch.h @@ -0,0 +1,17 @@ +/* + * HDA Patches - included by hda_codec.c + */ + +/* Realtek codecs */ +extern struct hda_codec_preset snd_hda_preset_realtek[]; +/* C-Media codecs */ +extern struct hda_codec_preset snd_hda_preset_cmedia[]; +/* Analog Devices codecs */ +extern struct hda_codec_preset snd_hda_preset_analog[]; + +static const struct hda_codec_preset *hda_preset_tables[] = { + snd_hda_preset_realtek, + snd_hda_preset_cmedia, + snd_hda_preset_analog, + NULL +}; diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c new file mode 100644 index 000000000000..4d5db7faad8d --- /dev/null +++ b/sound/pci/hda/hda_proc.c @@ -0,0 +1,298 @@ +/* + * Universal Interface for Intel High Definition Audio Codec + * + * Generic proc interface + * + * Copyright (c) 2004 Takashi Iwai <tiwai@suse.de> + * + * + * This driver 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 driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <sound/driver.h> +#include <linux/init.h> +#include <linux/pci.h> +#include <sound/core.h> +#include "hda_codec.h" + +static const char *get_wid_type_name(unsigned int wid_value) +{ + static char *names[16] = { + [AC_WID_AUD_OUT] = "Audio Output", + [AC_WID_AUD_IN] = "Audio Input", + [AC_WID_AUD_MIX] = "Audio Mixer", + [AC_WID_AUD_SEL] = "Audio Selector", + [AC_WID_PIN] = "Pin Complex", + [AC_WID_POWER] = "Power Widget", + [AC_WID_VOL_KNB] = "Volume Knob Widget", + [AC_WID_BEEP] = "Beep Generator Widget", + [AC_WID_VENDOR] = "Vendor Defined Widget", + }; + wid_value &= 0xf; + if (names[wid_value]) + return names[wid_value]; + else + return "UNKOWN Widget"; +} + +static void print_amp_caps(snd_info_buffer_t *buffer, + struct hda_codec *codec, hda_nid_t nid, int dir) +{ + unsigned int caps; + if (dir == HDA_OUTPUT) + caps = snd_hda_param_read(codec, nid, AC_PAR_AMP_OUT_CAP); + else + caps = snd_hda_param_read(codec, nid, AC_PAR_AMP_IN_CAP); + if (caps == -1 || caps == 0) { + snd_iprintf(buffer, "N/A\n"); + return; + } + snd_iprintf(buffer, "ofs=0x%02x, nsteps=0x%02x, stepsize=0x%02x, mute=%x\n", + caps & AC_AMPCAP_OFFSET, + (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT, + (caps & AC_AMPCAP_STEP_SIZE) >> AC_AMPCAP_STEP_SIZE_SHIFT, + (caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT); +} + +static void print_amp_vals(snd_info_buffer_t *buffer, + struct hda_codec *codec, hda_nid_t nid, + int dir, int stereo) +{ + unsigned int val; + if (stereo) { + val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_AMP_GAIN_MUTE, + AC_AMP_GET_LEFT | + (dir == HDA_OUTPUT ? AC_AMP_GET_OUTPUT : + AC_AMP_GET_INPUT)); + snd_iprintf(buffer, "0x%02x ", val); + } + val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_AMP_GAIN_MUTE, + AC_AMP_GET_RIGHT | + (dir == HDA_OUTPUT ? AC_AMP_GET_OUTPUT : + AC_AMP_GET_INPUT)); + snd_iprintf(buffer, "0x%02x\n", val); +} + +static void print_pcm_caps(snd_info_buffer_t *buffer, + struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int pcm = snd_hda_param_read(codec, nid, AC_PAR_PCM); + unsigned int stream = snd_hda_param_read(codec, nid, AC_PAR_STREAM); + if (pcm == -1 || stream == -1) { + snd_iprintf(buffer, "N/A\n"); + return; + } + snd_iprintf(buffer, "rates 0x%03x, bits 0x%02x, types 0x%x\n", + pcm & AC_SUPPCM_RATES, (pcm >> 16) & 0xff, stream & 0xf); +} + +static const char *get_jack_location(u32 cfg) +{ + static char *bases[7] = { + "N/A", "Rear", "Front", "Left", "Right", "Top", "Bottom", + }; + static unsigned char specials_idx[] = { + 0x07, 0x08, + 0x17, 0x18, 0x19, + 0x37, 0x38 + }; + static char *specials[] = { + "Rear Panel", "Drive Bar", + "Riser", "HDMI", "ATAPI", + "Mobile-In", "Mobile-Out" + }; + int i; + cfg = (cfg & AC_DEFCFG_LOCATION) >> AC_DEFCFG_LOCATION_SHIFT; + if ((cfg & 0x0f) < 7) + return bases[cfg & 0x0f]; + for (i = 0; i < ARRAY_SIZE(specials_idx); i++) { + if (cfg == specials_idx[i]) + return specials[i]; + } + return "UNKNOWN"; +} + +static const char *get_jack_connection(u32 cfg) +{ + static char *names[16] = { + "Unknown", "1/8", "1/4", "ATAPI", + "RCA", "Optical","Digital", "Analog", + "DIN", "XLR", "RJ11", "Comb", + NULL, NULL, NULL, "Other" + }; + cfg = (cfg & AC_DEFCFG_CONN_TYPE) >> AC_DEFCFG_CONN_TYPE_SHIFT; + if (names[cfg]) + return names[cfg]; + else + return "UNKNOWN"; +} + +static const char *get_jack_color(u32 cfg) +{ + static char *names[16] = { + "Unknown", "Black", "Grey", "Blue", + "Green", "Red", "Orange", "Yellow", + "Purple", "Pink", NULL, NULL, + NULL, NULL, "White", "Other", + }; + cfg = (cfg & AC_DEFCFG_COLOR) >> AC_DEFCFG_COLOR_SHIFT; + if (names[cfg]) + return names[cfg]; + else + return "UNKNOWN"; +} + +static void print_pin_caps(snd_info_buffer_t *buffer, + struct hda_codec *codec, hda_nid_t nid) +{ + static char *jack_types[16] = { + "Line Out", "Speaker", "HP Out", "CD", + "SPDIF Out", "Digital Out", "Modem Line", "Modem Hand", + "Line In", "Aux", "Mic", "Telephony", + "SPDIF In", "Digitial In", "Reserved", "Other" + }; + static char *jack_locations[4] = { "Ext", "Int", "Sep", "Oth" }; + unsigned int caps; + + caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + snd_iprintf(buffer, " Pincap 0x08%x:", caps); + if (caps & AC_PINCAP_IN) + snd_iprintf(buffer, " IN"); + if (caps & AC_PINCAP_OUT) + snd_iprintf(buffer, " OUT"); + if (caps & AC_PINCAP_HP_DRV) + snd_iprintf(buffer, " HP"); + snd_iprintf(buffer, "\n"); + caps = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + snd_iprintf(buffer, " Pin Default 0x%08x: %s at %s %s\n", caps, + jack_types[(caps & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT], + jack_locations[(caps >> (AC_DEFCFG_LOCATION_SHIFT + 4)) & 3], + get_jack_location(caps)); + snd_iprintf(buffer, " Conn = %s, Color = %s\n", + get_jack_connection(caps), + get_jack_color(caps)); +} + + +static void print_codec_info(snd_info_entry_t *entry, snd_info_buffer_t *buffer) +{ + struct hda_codec *codec = entry->private_data; + char buf[32]; + hda_nid_t nid; + int i, nodes; + + snd_hda_get_codec_name(codec, buf, sizeof(buf)); + snd_iprintf(buffer, "Codec: %s\n", buf); + snd_iprintf(buffer, "Address: %d\n", codec->addr); + snd_iprintf(buffer, "Vendor Id: 0x%x\n", codec->vendor_id); + snd_iprintf(buffer, "Subsystem Id: 0x%x\n", codec->subsystem_id); + snd_iprintf(buffer, "Revision Id: 0x%x\n", codec->revision_id); + snd_iprintf(buffer, "Default PCM: "); + print_pcm_caps(buffer, codec, codec->afg); + snd_iprintf(buffer, "Default Amp-In caps: "); + print_amp_caps(buffer, codec, codec->afg, HDA_INPUT); + snd_iprintf(buffer, "Default Amp-Out caps: "); + print_amp_caps(buffer, codec, codec->afg, HDA_OUTPUT); + + nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid); + if (! nid || nodes < 0) { + snd_iprintf(buffer, "Invalid AFG subtree\n"); + return; + } + for (i = 0; i < nodes; i++, nid++) { + unsigned int wid_caps = snd_hda_param_read(codec, nid, + AC_PAR_AUDIO_WIDGET_CAP); + unsigned int wid_type = (wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; + snd_iprintf(buffer, "Node 0x%02x [%s] wcaps 0x%x:", nid, + get_wid_type_name(wid_type), wid_caps); + if (wid_caps & AC_WCAP_STEREO) + snd_iprintf(buffer, " Stereo"); + else + snd_iprintf(buffer, " Mono"); + if (wid_caps & AC_WCAP_DIGITAL) + snd_iprintf(buffer, " Digital"); + if (wid_caps & AC_WCAP_IN_AMP) + snd_iprintf(buffer, " Amp-In"); + if (wid_caps & AC_WCAP_OUT_AMP) + snd_iprintf(buffer, " Amp-Out"); + snd_iprintf(buffer, "\n"); + + if (wid_caps & AC_WCAP_IN_AMP) { + snd_iprintf(buffer, " Amp-In caps: "); + print_amp_caps(buffer, codec, nid, HDA_INPUT); + snd_iprintf(buffer, " Amp-In vals: "); + print_amp_vals(buffer, codec, nid, HDA_INPUT, + wid_caps & AC_WCAP_STEREO); + } + if (wid_caps & AC_WCAP_OUT_AMP) { + snd_iprintf(buffer, " Amp-Out caps: "); + print_amp_caps(buffer, codec, nid, HDA_OUTPUT); + snd_iprintf(buffer, " Amp-Out vals: "); + print_amp_vals(buffer, codec, nid, HDA_OUTPUT, + wid_caps & AC_WCAP_STEREO); + } + + if (wid_type == AC_WID_PIN) { + unsigned int pinctls; + print_pin_caps(buffer, codec, nid); + pinctls = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_iprintf(buffer, " Pin-ctls: 0x%02x:", pinctls); + if (pinctls & AC_PINCTL_IN_EN) + snd_iprintf(buffer, " IN"); + if (pinctls & AC_PINCTL_OUT_EN) + snd_iprintf(buffer, " OUT"); + if (pinctls & AC_PINCTL_HP_EN) + snd_iprintf(buffer, " HP"); + snd_iprintf(buffer, "\n"); + } + + if ((wid_type == AC_WID_AUD_OUT || wid_type == AC_WID_AUD_IN) && + (wid_caps & AC_WCAP_FORMAT_OVRD)) { + snd_iprintf(buffer, " PCM: "); + print_pcm_caps(buffer, codec, nid); + } + + if (wid_caps & AC_WCAP_CONN_LIST) { + hda_nid_t conn[HDA_MAX_CONNECTIONS]; + int c, conn_len; + conn_len = snd_hda_get_connections(codec, nid, conn, + HDA_MAX_CONNECTIONS); + snd_iprintf(buffer, " Connection: %d\n", conn_len); + snd_iprintf(buffer, " "); + for (c = 0; c < conn_len; c++) + snd_iprintf(buffer, " 0x%02x", conn[c]); + snd_iprintf(buffer, "\n"); + } + } +} + +/* + * create a proc read + */ +int snd_hda_codec_proc_new(struct hda_codec *codec) +{ + char name[32]; + snd_info_entry_t *entry; + int err; + + snprintf(name, sizeof(name), "codec#%d", codec->addr); + err = snd_card_proc_new(codec->bus->card, name, &entry); + if (err < 0) + return err; + + snd_info_set_text_ops(entry, codec, 32 * 1024, print_codec_info); + return 0; +} + diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c new file mode 100644 index 000000000000..75d23849f71a --- /dev/null +++ b/sound/pci/hda/patch_analog.c @@ -0,0 +1,445 @@ +/* + * HD audio interface patch for AD1986A + * + * Copyright (c) 2005 Takashi Iwai <tiwai@suse.de> + * + * This driver 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 driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <sound/driver.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/pci.h> +#include <sound/core.h> +#include "hda_codec.h" +#include "hda_local.h" + +struct ad1986a_spec { + struct semaphore amp_mutex; /* PCM volume/mute control mutex */ + struct hda_multi_out multiout; /* playback */ + unsigned int cur_mux; /* capture source */ + struct hda_pcm pcm_rec[2]; /* PCM information */ +}; + +#define AD1986A_SPDIF_OUT 0x02 +#define AD1986A_FRONT_DAC 0x03 +#define AD1986A_SURR_DAC 0x04 +#define AD1986A_CLFE_DAC 0x05 +#define AD1986A_ADC 0x06 + +static hda_nid_t ad1986a_dac_nids[3] = { + AD1986A_FRONT_DAC, AD1986A_SURR_DAC, AD1986A_CLFE_DAC +}; + +static struct hda_input_mux ad1986a_capture_source = { + .num_items = 7, + .items = { + { "Mic", 0x0 }, + { "CD", 0x1 }, + { "Aux", 0x3 }, + { "Line", 0x4 }, + { "Mix", 0x5 }, + { "Mono", 0x6 }, + { "Phone", 0x7 }, + }, +}; + +/* + * PCM control + * + * bind volumes/mutes of 3 DACs as a single PCM control for simplicity + */ + +#define ad1986a_pcm_amp_vol_info snd_hda_mixer_amp_volume_info + +static int ad1986a_pcm_amp_vol_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ad1986a_spec *ad = codec->spec; + + down(&ad->amp_mutex); + snd_hda_mixer_amp_volume_get(kcontrol, ucontrol); + up(&ad->amp_mutex); + return 0; +} + +static int ad1986a_pcm_amp_vol_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ad1986a_spec *ad = codec->spec; + int i, change = 0; + + down(&ad->amp_mutex); + for (i = 0; i < ARRAY_SIZE(ad1986a_dac_nids); i++) { + kcontrol->private_value = HDA_COMPOSE_AMP_VAL(ad1986a_dac_nids[i], 3, 0, HDA_OUTPUT); + change |= snd_hda_mixer_amp_volume_put(kcontrol, ucontrol); + } + kcontrol->private_value = HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT); + up(&ad->amp_mutex); + return change; +} + +#define ad1986a_pcm_amp_sw_info snd_hda_mixer_amp_volume_info + +static int ad1986a_pcm_amp_sw_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ad1986a_spec *ad = codec->spec; + + down(&ad->amp_mutex); + snd_hda_mixer_amp_switch_get(kcontrol, ucontrol); + up(&ad->amp_mutex); + return 0; +} + +static int ad1986a_pcm_amp_sw_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ad1986a_spec *ad = codec->spec; + int i, change = 0; + + down(&ad->amp_mutex); + for (i = 0; i < ARRAY_SIZE(ad1986a_dac_nids); i++) { + kcontrol->private_value = HDA_COMPOSE_AMP_VAL(ad1986a_dac_nids[i], 3, 0, HDA_OUTPUT); + change |= snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); + } + kcontrol->private_value = HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT); + up(&ad->amp_mutex); + return change; +} + +/* + * input MUX handling + */ +static int ad1986a_mux_enum_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) +{ + return snd_hda_input_mux_info(&ad1986a_capture_source, uinfo); +} + +static int ad1986a_mux_enum_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ad1986a_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->cur_mux; + return 0; +} + +static int ad1986a_mux_enum_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ad1986a_spec *spec = codec->spec; + + return snd_hda_input_mux_put(codec, &ad1986a_capture_source, ucontrol, + AD1986A_ADC, &spec->cur_mux); +} + +/* + * mixers + */ +static snd_kcontrol_new_t ad1986a_mixers[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .info = ad1986a_pcm_amp_vol_info, + .get = ad1986a_pcm_amp_vol_get, + .put = ad1986a_pcm_amp_vol_put, + .private_value = HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT) + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Switch", + .info = ad1986a_pcm_amp_sw_info, + .get = ad1986a_pcm_amp_sw_get, + .put = ad1986a_pcm_amp_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT) + }, + HDA_CODEC_VOLUME("Front Playback Volume", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x1c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Surround Playback Switch", 0x1c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x1d, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x1d, 2, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x1d, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x1d, 2, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x1a, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Aux Playback Volume", 0x16, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Aux Playback Switch", 0x16, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x18, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x18, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = ad1986a_mux_enum_info, + .get = ad1986a_mux_enum_get, + .put = ad1986a_mux_enum_put, + }, + HDA_CODEC_MUTE("Stereo Downmix Switch", 0x09, 0x0, HDA_OUTPUT), + { } /* end */ +}; + +/* + * initialization verbs + */ +static struct hda_verb ad1986a_init_verbs[] = { + /* Front, Surround, CLFE DAC; mute as default */ + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + /* Downmix - off */ + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + /* HP, Line-Out, Surround, CLFE selectors */ + {0x0a, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x0b, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* Mono selector */ + {0x0e, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* Mic selector: Mic 1/2 pin */ + {0x0f, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* Line-in selector: Line-in */ + {0x10, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* Mic 1/2 swap */ + {0x11, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* Record selector: mic */ + {0x12, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* Mic, Phone, CD, Aux, Line-In amp; mute as default */ + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + /* PC beep */ + {0x18, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* HP, Line-Out, Surround, CLFE, Mono pins; mute as default */ + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + { } /* end */ +}; + + +static int ad1986a_init(struct hda_codec *codec) +{ + snd_hda_sequence_write(codec, ad1986a_init_verbs); + return 0; +} + +static int ad1986a_build_controls(struct hda_codec *codec) +{ + int err; + + err = snd_hda_add_new_ctls(codec, ad1986a_mixers); + if (err < 0) + return err; + err = snd_hda_create_spdif_out_ctls(codec, AD1986A_SPDIF_OUT); + if (err < 0) + return err; + return 0; +} + +/* + * Analog playback callbacks + */ +static int ad1986a_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + snd_pcm_substream_t *substream) +{ + struct ad1986a_spec *spec = codec->spec; + return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream); +} + +static int ad1986a_playback_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + snd_pcm_substream_t *substream) +{ + struct ad1986a_spec *spec = codec->spec; + return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag, + format, substream); +} + +static int ad1986a_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + snd_pcm_substream_t *substream) +{ + struct ad1986a_spec *spec = codec->spec; + return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); +} + +/* + * Digital out + */ +static int ad1986a_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + snd_pcm_substream_t *substream) +{ + struct ad1986a_spec *spec = codec->spec; + return snd_hda_multi_out_dig_open(codec, &spec->multiout); +} + +static int ad1986a_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + snd_pcm_substream_t *substream) +{ + struct ad1986a_spec *spec = codec->spec; + return snd_hda_multi_out_dig_close(codec, &spec->multiout); +} + +/* + * Analog capture + */ +static int ad1986a_capture_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + snd_pcm_substream_t *substream) +{ + snd_hda_codec_setup_stream(codec, AD1986A_ADC, stream_tag, 0, format); + return 0; +} + +static int ad1986a_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + snd_pcm_substream_t *substream) +{ + snd_hda_codec_setup_stream(codec, AD1986A_ADC, 0, 0, 0); + return 0; +} + + +/* + */ +static struct hda_pcm_stream ad1986a_pcm_analog_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 6, + .nid = AD1986A_FRONT_DAC, /* NID to query formats and rates */ + .ops = { + .open = ad1986a_playback_pcm_open, + .prepare = ad1986a_playback_pcm_prepare, + .cleanup = ad1986a_playback_pcm_cleanup + }, +}; + +static struct hda_pcm_stream ad1986a_pcm_analog_capture = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = AD1986A_ADC, /* NID to query formats and rates */ + .ops = { + .prepare = ad1986a_capture_pcm_prepare, + .cleanup = ad1986a_capture_pcm_cleanup + }, +}; + +static struct hda_pcm_stream ad1986a_pcm_digital_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .nid = AD1986A_SPDIF_OUT, + .ops = { + .open = ad1986a_dig_playback_pcm_open, + .close = ad1986a_dig_playback_pcm_close + }, +}; + +static int ad1986a_build_pcms(struct hda_codec *codec) +{ + struct ad1986a_spec *spec = codec->spec; + struct hda_pcm *info = spec->pcm_rec; + + codec->num_pcms = 2; + codec->pcm_info = info; + + info->name = "AD1986A Analog"; + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad1986a_pcm_analog_playback; + info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad1986a_pcm_analog_capture; + info++; + + info->name = "AD1986A Digital"; + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad1986a_pcm_digital_playback; + + return 0; +} + +static void ad1986a_free(struct hda_codec *codec) +{ + kfree(codec->spec); +} + +#ifdef CONFIG_PM +static int ad1986a_resume(struct hda_codec *codec) +{ + ad1986a_init(codec); + snd_hda_resume_ctls(codec, ad1986a_mixers); + snd_hda_resume_spdif_out(codec); + return 0; +} +#endif + +static struct hda_codec_ops ad1986a_patch_ops = { + .build_controls = ad1986a_build_controls, + .build_pcms = ad1986a_build_pcms, + .init = ad1986a_init, + .free = ad1986a_free, +#ifdef CONFIG_PM + .resume = ad1986a_resume, +#endif +}; + +static int patch_ad1986a(struct hda_codec *codec) +{ + struct ad1986a_spec *spec; + + spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + init_MUTEX(&spec->amp_mutex); + codec->spec = spec; + + spec->multiout.max_channels = 6; + spec->multiout.num_dacs = ARRAY_SIZE(ad1986a_dac_nids); + spec->multiout.dac_nids = ad1986a_dac_nids; + spec->multiout.dig_out_nid = AD1986A_SPDIF_OUT; + + codec->patch_ops = ad1986a_patch_ops; + + return 0; +} + +/* + * patch entries + */ +struct hda_codec_preset snd_hda_preset_analog[] = { + { .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a }, + {} /* terminator */ +}; diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c new file mode 100644 index 000000000000..b7cc8e4bffb7 --- /dev/null +++ b/sound/pci/hda/patch_cmedia.c @@ -0,0 +1,621 @@ +/* + * Universal Interface for Intel High Definition Audio Codec + * + * HD audio interface patch for C-Media CMI9880 + * + * Copyright (c) 2004 Takashi Iwai <tiwai@suse.de> + * + * + * This driver 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 driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <sound/driver.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/pci.h> +#include <sound/core.h> +#include "hda_codec.h" +#include "hda_local.h" + + +/* board config type */ +enum { + CMI_MINIMAL, /* back 3-jack */ + CMI_MIN_FP, /* back 3-jack + front-panel 2-jack */ + CMI_FULL, /* back 6-jack + front-panel 2-jack */ + CMI_FULL_DIG, /* back 6-jack + front-panel 2-jack + digital I/O */ + CMI_ALLOUT, /* back 5-jack + front-panel 2-jack + digital out */ +}; + +struct cmi_spec { + int board_config; + unsigned int surr_switch: 1; /* switchable line,mic */ + unsigned int no_line_in: 1; /* no line-in (5-jack) */ + unsigned int front_panel: 1; /* has front-panel 2-jack */ + + /* playback */ + struct hda_multi_out multiout; + + /* capture */ + hda_nid_t *adc_nids; + hda_nid_t dig_in_nid; + + /* capture source */ + const struct hda_input_mux *input_mux; + unsigned int cur_mux[2]; + + /* channel mode */ + unsigned int num_ch_modes; + unsigned int cur_ch_mode; + const struct cmi_channel_mode *channel_modes; + + struct hda_pcm pcm_rec[2]; /* PCM information */ +}; + +/* + * input MUX + */ +static int cmi_mux_enum_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cmi_spec *spec = codec->spec; + return snd_hda_input_mux_info(spec->input_mux, uinfo); +} + +static int cmi_mux_enum_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cmi_spec *spec = codec->spec; + unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + + ucontrol->value.enumerated.item[0] = spec->cur_mux[adc_idx]; + return 0; +} + +static int cmi_mux_enum_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cmi_spec *spec = codec->spec; + unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + + return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, + spec->adc_nids[adc_idx], &spec->cur_mux[adc_idx]); +} + +/* + * shared line-in, mic for surrounds + */ + +/* 3-stack / 2 channel */ +static struct hda_verb cmi9880_ch2_init[] = { + /* set line-in PIN for input */ + { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, + /* set mic PIN for input, also enable vref */ + { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, + /* route front PCM (DAC1) to HP */ + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 }, + {} +}; + +/* 3-stack / 6 channel */ +static struct hda_verb cmi9880_ch6_init[] = { + /* set line-in PIN for output */ + { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, + /* set mic PIN for output */ + { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, + /* route front PCM (DAC1) to HP */ + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 }, + {} +}; + +/* 3-stack+front / 8 channel */ +static struct hda_verb cmi9880_ch8_init[] = { + /* set line-in PIN for output */ + { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, + /* set mic PIN for output */ + { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, + /* route rear-surround PCM (DAC4) to HP */ + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x03 }, + {} +}; + +struct cmi_channel_mode { + unsigned int channels; + const struct hda_verb *sequence; +}; + +static struct cmi_channel_mode cmi9880_channel_modes[3] = { + { 2, cmi9880_ch2_init }, + { 6, cmi9880_ch6_init }, + { 8, cmi9880_ch8_init }, +}; + +static int cmi_ch_mode_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cmi_spec *spec = codec->spec; + + snd_assert(spec->channel_modes, return -EINVAL); + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = spec->num_ch_modes; + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) + uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; + sprintf(uinfo->value.enumerated.name, "%dch", + spec->channel_modes[uinfo->value.enumerated.item].channels); + return 0; +} + +static int cmi_ch_mode_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cmi_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->cur_ch_mode; + return 0; +} + +static int cmi_ch_mode_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cmi_spec *spec = codec->spec; + + snd_assert(spec->channel_modes, return -EINVAL); + if (ucontrol->value.enumerated.item[0] >= spec->num_ch_modes) + ucontrol->value.enumerated.item[0] = spec->num_ch_modes; + if (ucontrol->value.enumerated.item[0] == spec->cur_ch_mode && + ! codec->in_resume) + return 0; + + spec->cur_ch_mode = ucontrol->value.enumerated.item[0]; + snd_hda_sequence_write(codec, spec->channel_modes[spec->cur_ch_mode].sequence); + spec->multiout.max_channels = spec->channel_modes[spec->cur_ch_mode].channels; + return 1; +} + +/* + */ +static snd_kcontrol_new_t cmi9880_basic_mixer[] = { + /* CMI9880 has no playback volumes! */ + HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), /* front */ + HDA_CODEC_MUTE("Surround Playback Switch", 0x04, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Side Playback Switch", 0x06, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + * FIXME: the controls appear in the "playback" view! + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = cmi_mux_enum_info, + .get = cmi_mux_enum_get, + .put = cmi_mux_enum_put, + }, + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0, HDA_INPUT), + HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x23, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x23, 0, HDA_OUTPUT), + { } /* end */ +}; + +/* + * shared I/O pins + */ +static snd_kcontrol_new_t cmi9880_ch_mode_mixer[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Channel Mode", + .info = cmi_ch_mode_info, + .get = cmi_ch_mode_get, + .put = cmi_ch_mode_put, + }, + { } /* end */ +}; + +/* AUD-in selections: + * 0x0b 0x0c 0x0d 0x0e 0x0f 0x10 0x11 0x1f 0x20 + */ +static struct hda_input_mux cmi9880_basic_mux = { + .num_items = 4, + .items = { + { "Front Mic", 0x5 }, + { "Rear Mic", 0x2 }, + { "Line", 0x1 }, + { "CD", 0x7 }, + } +}; + +static struct hda_input_mux cmi9880_no_line_mux = { + .num_items = 3, + .items = { + { "Front Mic", 0x5 }, + { "Rear Mic", 0x2 }, + { "CD", 0x7 }, + } +}; + +/* front, rear, clfe, rear_surr */ +static hda_nid_t cmi9880_dac_nids[4] = { + 0x03, 0x04, 0x05, 0x06 +}; +/* ADC0, ADC1 */ +static hda_nid_t cmi9880_adc_nids[2] = { + 0x08, 0x09 +}; + +#define CMI_DIG_OUT_NID 0x07 +#define CMI_DIG_IN_NID 0x0a + +/* + */ +static struct hda_verb cmi9880_basic_init[] = { + /* port-D for line out (rear panel) */ + { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, + /* port-E for HP out (front panel) */ + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, + /* route front PCM to HP */ + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 }, + /* port-A for surround (rear panel) */ + { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, + /* port-G for CLFE (rear panel) */ + { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, + /* port-H for side (rear panel) */ + { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, + /* port-C for line-in (rear panel) */ + { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, + /* port-B for mic-in (rear panel) with vref */ + { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, + /* port-F for mic-in (front panel) with vref */ + { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, + /* CD-in */ + { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, + /* route front mic to ADC1/2 */ + { 0x08, AC_VERB_SET_CONNECT_SEL, 0x05 }, + { 0x09, AC_VERB_SET_CONNECT_SEL, 0x05 }, + {} /* terminator */ +}; + +static struct hda_verb cmi9880_allout_init[] = { + /* port-D for line out (rear panel) */ + { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, + /* port-E for HP out (front panel) */ + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, + /* route front PCM to HP */ + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 }, + /* port-A for side (rear panel) */ + { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, + /* port-G for CLFE (rear panel) */ + { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, + /* port-C for surround (rear panel) */ + { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, + /* port-B for mic-in (rear panel) with vref */ + { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, + /* port-F for mic-in (front panel) with vref */ + { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, + /* CD-in */ + { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, + /* route front mic to ADC1/2 */ + { 0x08, AC_VERB_SET_CONNECT_SEL, 0x05 }, + { 0x09, AC_VERB_SET_CONNECT_SEL, 0x05 }, + {} /* terminator */ +}; + +/* + */ +static int cmi9880_build_controls(struct hda_codec *codec) +{ + struct cmi_spec *spec = codec->spec; + int err; + + err = snd_hda_add_new_ctls(codec, cmi9880_basic_mixer); + if (err < 0) + return err; + if (spec->surr_switch) { + err = snd_hda_add_new_ctls(codec, cmi9880_ch_mode_mixer); + if (err < 0) + return err; + } + if (spec->multiout.dig_out_nid) { + err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid); + if (err < 0) + return err; + } + if (spec->dig_in_nid) { + err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); + if (err < 0) + return err; + } + return 0; +} + +static int cmi9880_init(struct hda_codec *codec) +{ + struct cmi_spec *spec = codec->spec; + if (spec->board_config == CMI_ALLOUT) + snd_hda_sequence_write(codec, cmi9880_allout_init); + else + snd_hda_sequence_write(codec, cmi9880_basic_init); + return 0; +} + +#ifdef CONFIG_PM +/* + * resume + */ +static int cmi9880_resume(struct hda_codec *codec) +{ + struct cmi_spec *spec = codec->spec; + + cmi9880_init(codec); + snd_hda_resume_ctls(codec, cmi9880_basic_mixer); + if (spec->surr_switch) + snd_hda_resume_ctls(codec, cmi9880_ch_mode_mixer); + if (spec->multiout.dig_out_nid) + snd_hda_resume_spdif_out(codec); + if (spec->dig_in_nid) + snd_hda_resume_spdif_in(codec); + + return 0; +} +#endif + +/* + * Analog playback callbacks + */ +static int cmi9880_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + snd_pcm_substream_t *substream) +{ + struct cmi_spec *spec = codec->spec; + return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream); +} + +static int cmi9880_playback_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + snd_pcm_substream_t *substream) +{ + struct cmi_spec *spec = codec->spec; + return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag, + format, substream); +} + +static int cmi9880_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + snd_pcm_substream_t *substream) +{ + struct cmi_spec *spec = codec->spec; + return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); +} + +/* + * Digital out + */ +static int cmi9880_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + snd_pcm_substream_t *substream) +{ + struct cmi_spec *spec = codec->spec; + return snd_hda_multi_out_dig_open(codec, &spec->multiout); +} + +static int cmi9880_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + snd_pcm_substream_t *substream) +{ + struct cmi_spec *spec = codec->spec; + return snd_hda_multi_out_dig_close(codec, &spec->multiout); +} + +/* + * Analog capture + */ +static int cmi9880_capture_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + snd_pcm_substream_t *substream) +{ + struct cmi_spec *spec = codec->spec; + + snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], + stream_tag, 0, format); + return 0; +} + +static int cmi9880_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + snd_pcm_substream_t *substream) +{ + struct cmi_spec *spec = codec->spec; + + snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], 0, 0, 0); + return 0; +} + + +/* + */ +static struct hda_pcm_stream cmi9880_pcm_analog_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 8, + .nid = 0x03, /* NID to query formats and rates */ + .ops = { + .open = cmi9880_playback_pcm_open, + .prepare = cmi9880_playback_pcm_prepare, + .cleanup = cmi9880_playback_pcm_cleanup + }, +}; + +static struct hda_pcm_stream cmi9880_pcm_analog_capture = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x08, /* NID to query formats and rates */ + .ops = { + .prepare = cmi9880_capture_pcm_prepare, + .cleanup = cmi9880_capture_pcm_cleanup + }, +}; + +static struct hda_pcm_stream cmi9880_pcm_digital_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + /* NID is set in cmi9880_build_pcms */ + .ops = { + .open = cmi9880_dig_playback_pcm_open, + .close = cmi9880_dig_playback_pcm_close + }, +}; + +static struct hda_pcm_stream cmi9880_pcm_digital_capture = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + /* NID is set in cmi9880_build_pcms */ +}; + +static int cmi9880_build_pcms(struct hda_codec *codec) +{ + struct cmi_spec *spec = codec->spec; + struct hda_pcm *info = spec->pcm_rec; + + codec->num_pcms = 1; + codec->pcm_info = info; + + info->name = "CMI9880"; + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = cmi9880_pcm_analog_playback; + info->stream[SNDRV_PCM_STREAM_CAPTURE] = cmi9880_pcm_analog_capture; + + if (spec->multiout.dig_out_nid || spec->dig_in_nid) { + codec->num_pcms++; + info++; + info->name = "CMI9880 Digital"; + if (spec->multiout.dig_out_nid) { + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = cmi9880_pcm_digital_playback; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid; + } + if (spec->dig_in_nid) { + info->stream[SNDRV_PCM_STREAM_CAPTURE] = cmi9880_pcm_digital_capture; + info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid; + } + } + + return 0; +} + +static void cmi9880_free(struct hda_codec *codec) +{ + kfree(codec->spec); +} + +/* + */ + +static struct hda_board_config cmi9880_cfg_tbl[] = { + { .modelname = "minimal", .config = CMI_MINIMAL }, + { .modelname = "min_fp", .config = CMI_MIN_FP }, + { .modelname = "full", .config = CMI_FULL }, + { .modelname = "full_dig", .config = CMI_FULL_DIG }, + { .modelname = "allout", .config = CMI_ALLOUT }, + {} /* terminator */ +}; + +static struct hda_codec_ops cmi9880_patch_ops = { + .build_controls = cmi9880_build_controls, + .build_pcms = cmi9880_build_pcms, + .init = cmi9880_init, + .free = cmi9880_free, +#ifdef CONFIG_PM + .resume = cmi9880_resume, +#endif +}; + +static int patch_cmi9880(struct hda_codec *codec) +{ + struct cmi_spec *spec; + + spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + spec->board_config = snd_hda_check_board_config(codec, cmi9880_cfg_tbl); + if (spec->board_config < 0) { + snd_printd(KERN_INFO "hda_codec: Unknown model for CMI9880\n"); + spec->board_config = CMI_FULL_DIG; /* try everything */ + } + + switch (spec->board_config) { + case CMI_MINIMAL: + case CMI_MIN_FP: + spec->surr_switch = 1; + if (spec->board_config == CMI_MINIMAL) + spec->num_ch_modes = 2; + else { + spec->front_panel = 1; + spec->num_ch_modes = 3; + } + spec->channel_modes = cmi9880_channel_modes; + spec->multiout.max_channels = cmi9880_channel_modes[0].channels; + spec->input_mux = &cmi9880_basic_mux; + break; + case CMI_FULL: + case CMI_FULL_DIG: + spec->front_panel = 1; + spec->multiout.max_channels = 8; + spec->input_mux = &cmi9880_basic_mux; + if (spec->board_config == CMI_FULL_DIG) { + spec->multiout.dig_out_nid = CMI_DIG_OUT_NID; + spec->dig_in_nid = CMI_DIG_IN_NID; + } + break; + case CMI_ALLOUT: + spec->front_panel = 1; + spec->multiout.max_channels = 8; + spec->no_line_in = 1; + spec->input_mux = &cmi9880_no_line_mux; + spec->multiout.dig_out_nid = CMI_DIG_OUT_NID; + break; + } + + spec->multiout.num_dacs = 4; + spec->multiout.dac_nids = cmi9880_dac_nids; + + spec->adc_nids = cmi9880_adc_nids; + + codec->patch_ops = cmi9880_patch_ops; + + return 0; +} + +/* + * patch entries + */ +struct hda_codec_preset snd_hda_preset_cmedia[] = { + { .id = 0x13f69880, .name = "CMI9880", .patch = patch_cmi9880 }, + { .id = 0x434d4980, .name = "CMI9880", .patch = patch_cmi9880 }, + {} /* terminator */ +}; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c new file mode 100644 index 000000000000..17c5062423ae --- /dev/null +++ b/sound/pci/hda/patch_realtek.c @@ -0,0 +1,1503 @@ +/* + * Universal Interface for Intel High Definition Audio Codec + * + * HD audio interface patch for ALC 260/880/882 codecs + * + * Copyright (c) 2004 PeiSen Hou <pshou@realtek.com.tw> + * Takashi Iwai <tiwai@suse.de> + * + * This driver 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 driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <sound/driver.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/pci.h> +#include <sound/core.h> +#include "hda_codec.h" +#include "hda_local.h" + + +/* ALC880 board config type */ +enum { + ALC880_MINIMAL, + ALC880_3ST, + ALC880_3ST_DIG, + ALC880_5ST, + ALC880_5ST_DIG, + ALC880_W810, +}; + +struct alc_spec { + /* codec parameterization */ + unsigned int front_panel: 1; + + snd_kcontrol_new_t* mixers[2]; + unsigned int num_mixers; + + struct hda_verb *init_verbs; + + char* stream_name_analog; + struct hda_pcm_stream *stream_analog_playback; + struct hda_pcm_stream *stream_analog_capture; + + char* stream_name_digital; + struct hda_pcm_stream *stream_digital_playback; + struct hda_pcm_stream *stream_digital_capture; + + /* playback */ + struct hda_multi_out multiout; + + /* capture */ + unsigned int num_adc_nids; + hda_nid_t *adc_nids; + hda_nid_t dig_in_nid; + + /* capture source */ + const struct hda_input_mux *input_mux; + unsigned int cur_mux[3]; + + /* channel model */ + const struct alc_channel_mode *channel_mode; + int num_channel_mode; + + /* PCM information */ + struct hda_pcm pcm_rec[2]; +}; + +/* DAC/ADC assignment */ + +static hda_nid_t alc880_dac_nids[4] = { + /* front, rear, clfe, rear_surr */ + 0x02, 0x05, 0x04, 0x03 +}; + +static hda_nid_t alc880_w810_dac_nids[3] = { + /* front, rear/surround, clfe */ + 0x02, 0x03, 0x04 +}; + +static hda_nid_t alc880_adc_nids[3] = { + /* ADC0-2 */ + 0x07, 0x08, 0x09, +}; + +#define ALC880_DIGOUT_NID 0x06 +#define ALC880_DIGIN_NID 0x0a + +static hda_nid_t alc260_dac_nids[1] = { + /* front */ + 0x02, +}; + +static hda_nid_t alc260_adc_nids[2] = { + /* ADC0-1 */ + 0x04, 0x05, +}; + +#define ALC260_DIGOUT_NID 0x03 +#define ALC260_DIGIN_NID 0x06 + +static struct hda_input_mux alc880_capture_source = { + .num_items = 4, + .items = { + { "Mic", 0x0 }, + { "Front Mic", 0x3 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + }, +}; + +static struct hda_input_mux alc260_capture_source = { + .num_items = 4, + .items = { + { "Mic", 0x0 }, + { "Front Mic", 0x1 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + }, +}; + +/* + * input MUX handling + */ +static int alc_mux_enum_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + return snd_hda_input_mux_info(spec->input_mux, uinfo); +} + +static int alc_mux_enum_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + + ucontrol->value.enumerated.item[0] = spec->cur_mux[adc_idx]; + return 0; +} + +static int alc_mux_enum_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, + spec->adc_nids[adc_idx], &spec->cur_mux[adc_idx]); +} + +/* + * channel mode setting + */ +struct alc_channel_mode { + int channels; + const struct hda_verb *sequence; +}; + + +/* + * channel source setting (2/6 channel selection for 3-stack) + */ + +/* + * set the path ways for 2 channel output + * need to set the codec line out and mic 1 pin widgets to inputs + */ +static struct hda_verb alc880_threestack_ch2_init[] = { + /* set pin widget 1Ah (line in) for input */ + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, + /* set pin widget 18h (mic1) for input, for mic also enable the vref */ + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, + /* mute the output for Line In PW */ + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 }, + /* mute for Mic1 PW */ + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 }, + { } /* end */ +}; + +/* + * 6ch mode + * need to set the codec line out and mic 1 pin widgets to outputs + */ +static struct hda_verb alc880_threestack_ch6_init[] = { + /* set pin widget 1Ah (line in) for output */ + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, + /* set pin widget 18h (mic1) for output */ + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, + /* unmute the output for Line In PW */ + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000 }, + /* unmute for Mic1 PW */ + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000 }, + /* for rear channel output using Line In 1 + * set select widget connection (nid = 0x12) - to summer node + * for rear NID = 0x0f...offset 3 in connection list + */ + { 0x12, AC_VERB_SET_CONNECT_SEL, 0x3 }, + /* for Mic1 - retask for center/lfe */ + /* set select widget connection (nid = 0x10) - to summer node for + * front CLFE NID = 0x0e...offset 2 in connection list + */ + { 0x10, AC_VERB_SET_CONNECT_SEL, 0x2 }, + { } /* end */ +}; + +static struct alc_channel_mode alc880_threestack_modes[2] = { + { 2, alc880_threestack_ch2_init }, + { 6, alc880_threestack_ch6_init }, +}; + + +/* + * channel source setting (6/8 channel selection for 5-stack) + */ + +/* set the path ways for 6 channel output + * need to set the codec line out and mic 1 pin widgets to inputs + */ +static struct hda_verb alc880_fivestack_ch6_init[] = { + /* set pin widget 1Ah (line in) for input */ + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, + /* mute the output for Line In PW */ + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 }, + { } /* end */ +}; + +/* need to set the codec line out and mic 1 pin widgets to outputs */ +static struct hda_verb alc880_fivestack_ch8_init[] = { + /* set pin widget 1Ah (line in) for output */ + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, + /* unmute the output for Line In PW */ + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000 }, + /* output for surround channel output using Line In 1 */ + /* set select widget connection (nid = 0x12) - to summer node + * for surr_rear NID = 0x0d...offset 1 in connection list + */ + { 0x12, AC_VERB_SET_CONNECT_SEL, 0x1 }, + { } /* end */ +}; + +static struct alc_channel_mode alc880_fivestack_modes[2] = { + { 6, alc880_fivestack_ch6_init }, + { 8, alc880_fivestack_ch8_init }, +}; + +/* + * channel source setting for W810 system + * + * W810 has rear IO for: + * Front (DAC 02) + * Surround (DAC 03) + * Center/LFE (DAC 04) + * Digital out (06) + * + * The system also has a pair of internal speakers, and a headphone jack. + * These are both connected to Line2 on the codec, hence to DAC 02. + * + * There is a variable resistor to control the speaker or headphone + * volume. This is a hardware-only device without a software API. + * + * Plugging headphones in will disable the internal speakers. This is + * implemented in hardware, not via the driver using jack sense. In + * a similar fashion, plugging into the rear socket marked "front" will + * disable both the speakers and headphones. + * + * For input, there's a microphone jack, and an "audio in" jack. + * These may not do anything useful with this driver yet, because I + * haven't setup any initialization verbs for these yet... + */ + +static struct alc_channel_mode alc880_w810_modes[1] = { + { 6, NULL } +}; + +/* + */ +static int alc880_ch_mode_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + + snd_assert(spec->channel_mode, return -ENXIO); + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 2; + if (uinfo->value.enumerated.item >= 2) + uinfo->value.enumerated.item = 1; + sprintf(uinfo->value.enumerated.name, "%dch", + spec->channel_mode[uinfo->value.enumerated.item].channels); + return 0; +} + +static int alc880_ch_mode_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + + snd_assert(spec->channel_mode, return -ENXIO); + ucontrol->value.enumerated.item[0] = + (spec->multiout.max_channels == spec->channel_mode[0].channels) ? 0 : 1; + return 0; +} + +static int alc880_ch_mode_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + int mode; + + snd_assert(spec->channel_mode, return -ENXIO); + mode = ucontrol->value.enumerated.item[0] ? 1 : 0; + if (spec->multiout.max_channels == spec->channel_mode[mode].channels && + ! codec->in_resume) + return 0; + + /* change the current channel setting */ + spec->multiout.max_channels = spec->channel_mode[mode].channels; + if (spec->channel_mode[mode].sequence) + snd_hda_sequence_write(codec, spec->channel_mode[mode].sequence); + + return 1; +} + + +/* + */ + +/* 3-stack mode + * Pin assignment: Front=0x14, Line-In/Rear=0x1a, Mic/CLFE=0x18, F-Mic=0x1b + * HP=0x19 + */ +static snd_kcontrol_new_t alc880_base_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Surround Playback Switch", 0x1a, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x18, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x18, 2, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x3, HDA_INPUT), + HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x19, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + * FIXME: the controls appear in the "playback" view! + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc_mux_enum_info, + .get = alc_mux_enum_get, + .put = alc_mux_enum_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Channel Mode", + .info = alc880_ch_mode_info, + .get = alc880_ch_mode_get, + .put = alc880_ch_mode_put, + }, + { } /* end */ +}; + +/* 5-stack mode + * Pin assignment: Front=0x14, Rear=0x17, CLFE=0x16 + * Line-In/Side=0x1a, Mic=0x18, F-Mic=0x1b, HP=0x19 + */ +static snd_kcontrol_new_t alc880_five_stack_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Surround Playback Switch", 0x17, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x16, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Side Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Side Playback Switch", 0x1a, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x3, HDA_INPUT), + HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x19, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + * FIXME: the controls appear in the "playback" view! + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc_mux_enum_info, + .get = alc_mux_enum_get, + .put = alc_mux_enum_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Channel Mode", + .info = alc880_ch_mode_info, + .get = alc880_ch_mode_get, + .put = alc880_ch_mode_put, + }, + { } /* end */ +}; + +static snd_kcontrol_new_t alc880_w810_base_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Surround Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x16, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + * FIXME: the controls appear in the "playback" view! + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 3, + .info = alc_mux_enum_info, + .get = alc_mux_enum_get, + .put = alc_mux_enum_put, + }, + { } /* end */ +}; + +/* + */ +static int alc_build_controls(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int err; + int i; + + for (i = 0; i < spec->num_mixers; i++) { + err = snd_hda_add_new_ctls(codec, spec->mixers[i]); + if (err < 0) + return err; + } + + if (spec->multiout.dig_out_nid) { + err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid); + if (err < 0) + return err; + } + if (spec->dig_in_nid) { + err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); + if (err < 0) + return err; + } + return 0; +} + +/* + * initialize the codec volumes, etc + */ + +static struct hda_verb alc880_init_verbs_three_stack[] = { + /* Line In pin widget for input */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + /* CD pin widget for input */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + /* Mic1 (rear panel) pin widget for input and vref at 80% */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + /* Mic2 (front panel) pin widget for input and vref at 80% */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + /* unmute amp left and right */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, + /* set connection select to line in (default select for this ADC) */ + {0x07, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* unmute front mixer amp left (volume = 0) */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + /* mute pin widget amp left and right (no gain on this amp) */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, + /* unmute rear mixer amp left and right (volume = 0) */ + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + /* mute pin widget amp left and right (no gain on this amp) */ + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, + /* unmute rear mixer amp left and right (volume = 0) */ + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + /* mute pin widget amp left and right (no gain on this amp) */ + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, + + /* using rear surround as the path for headphone output */ + /* unmute rear surround mixer amp left and right (volume = 0) */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + /* PASD 3 stack boards use the Mic 2 as the headphone output */ + /* need to program the selector associated with the Mic 2 pin widget to + * surround path (index 0x01) for headphone output */ + {0x11, AC_VERB_SET_CONNECT_SEL, 0x01}, + /* mute pin widget amp left and right (no gain on this amp) */ + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, + /* need to retask the Mic 2 pin widget to output */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + + /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) for mixer widget(nid=0x0B) + * to support the input path of analog loopback + * Note: PASD motherboards uses the Line In 2 as the input for front panel + * mic (mic 2) + */ + /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 & Line In 2 = 0x03 */ + /* unmute CD */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))}, + /* unmute Line In */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, + /* unmute Mic 1 */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + /* unmute Line In 2 (for PASD boards Mic 2) */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))}, + + /* Unmute input amps for the line out paths to support the output path of + * analog loopback + * the mixers on the output path has 2 inputs, one from the DAC and one + * from the mixer + */ + /* Amp Indexes: DAC = 0x01 & mixer = 0x00 */ + /* Unmute Front out path */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + /* Unmute Surround (used as HP) out path */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + /* Unmute C/LFE out path */ + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))}, /* mute */ + /* Unmute rear Surround out path */ + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + + { } +}; + +static struct hda_verb alc880_init_verbs_five_stack[] = { + /* Line In pin widget for input */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + /* CD pin widget for input */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + /* Mic1 (rear panel) pin widget for input and vref at 80% */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + /* Mic2 (front panel) pin widget for input and vref at 80% */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + /* unmute amp left and right */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, + /* set connection select to line in (default select for this ADC) */ + {0x07, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* unmute front mixer amp left and right (volume = 0) */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + /* mute pin widget amp left and right (no gain on this amp) */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, + /* five rear and clfe */ + /* unmute rear mixer amp left and right (volume = 0) */ + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + /* mute pin widget amp left and right (no gain on this amp) */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, + /* unmute clfe mixer amp left and right (volume = 0) */ + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + /* mute pin widget amp left and right (no gain on this amp) */ + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, + + /* using rear surround as the path for headphone output */ + /* unmute rear surround mixer amp left and right (volume = 0) */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + /* PASD 3 stack boards use the Mic 2 as the headphone output */ + /* need to program the selector associated with the Mic 2 pin widget to + * surround path (index 0x01) for headphone output + */ + {0x11, AC_VERB_SET_CONNECT_SEL, 0x01}, + /* mute pin widget amp left and right (no gain on this amp) */ + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, + /* need to retask the Mic 2 pin widget to output */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + + /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) for mixer + * widget(nid=0x0B) to support the input path of analog loopback + */ + /* Note: PASD motherboards uses the Line In 2 as the input for front panel mic (mic 2) */ + /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 & Line In 2 = 0x03*/ + /* unmute CD */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))}, + /* unmute Line In */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, + /* unmute Mic 1 */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + /* unmute Line In 2 (for PASD boards Mic 2) */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))}, + + /* Unmute input amps for the line out paths to support the output path of + * analog loopback + * the mixers on the output path has 2 inputs, one from the DAC and + * one from the mixer + */ + /* Amp Indexes: DAC = 0x01 & mixer = 0x00 */ + /* Unmute Front out path */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + /* Unmute Surround (used as HP) out path */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + /* Unmute C/LFE out path */ + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))}, /* mute */ + /* Unmute rear Surround out path */ + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + + { } +}; + +static struct hda_verb alc880_w810_init_verbs[] = { + /* front channel selector/amp: input 0: DAC: unmuted, (no volume selection) */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, + + /* front channel selector/amp: input 1: capture mix: muted, (no volume selection) */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7180}, + + /* front channel selector/amp: output 0: unmuted, max volume */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + + /* front out pin: muted, (no volume selection) */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + + /* front out pin: NOT headphone enable, out enable, vref disabled */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + + + /* surround channel selector/amp: input 0: DAC: unmuted, (no volume selection) */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, + + /* surround channel selector/amp: input 1: capture mix: muted, (no volume selection) */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7180}, + + /* surround channel selector/amp: output 0: unmuted, max volume */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + + /* surround out pin: muted, (no volume selection) */ + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + + /* surround out pin: NOT headphone enable, out enable, vref disabled */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + + + /* c/lfe channel selector/amp: input 0: DAC: unmuted, (no volume selection) */ + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, + + /* c/lfe channel selector/amp: input 1: capture mix: muted, (no volume selection) */ + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0x7180}, + + /* c/lfe channel selector/amp: output 0: unmuted, max volume */ + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + + /* c/lfe out pin: muted, (no volume selection) */ + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + + /* c/lfe out pin: NOT headphone enable, out enable, vref disabled */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + + + /* hphone/speaker input selector: front DAC */ + {0x13, AC_VERB_SET_CONNECT_SEL, 0x0}, + + /* hphone/speaker out pin: muted, (no volume selection) */ + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + + /* hphone/speaker out pin: NOT headphone enable, out enable, vref disabled */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + + + { } +}; + +static int alc_init(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + snd_hda_sequence_write(codec, spec->init_verbs); + return 0; +} + +#ifdef CONFIG_PM +/* + * resume + */ +static int alc_resume(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int i; + + alc_init(codec); + for (i = 0; i < spec->num_mixers; i++) { + snd_hda_resume_ctls(codec, spec->mixers[i]); + } + if (spec->multiout.dig_out_nid) + snd_hda_resume_spdif_out(codec); + if (spec->dig_in_nid) + snd_hda_resume_spdif_in(codec); + + return 0; +} +#endif + +/* + * Analog playback callbacks + */ +static int alc880_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + snd_pcm_substream_t *substream) +{ + struct alc_spec *spec = codec->spec; + return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream); +} + +static int alc880_playback_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + snd_pcm_substream_t *substream) +{ + struct alc_spec *spec = codec->spec; + return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag, + format, substream); +} + +static int alc880_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + snd_pcm_substream_t *substream) +{ + struct alc_spec *spec = codec->spec; + return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); +} + +/* + * Digital out + */ +static int alc880_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + snd_pcm_substream_t *substream) +{ + struct alc_spec *spec = codec->spec; + return snd_hda_multi_out_dig_open(codec, &spec->multiout); +} + +static int alc880_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + snd_pcm_substream_t *substream) +{ + struct alc_spec *spec = codec->spec; + return snd_hda_multi_out_dig_close(codec, &spec->multiout); +} + +/* + * Analog capture + */ +static int alc880_capture_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + snd_pcm_substream_t *substream) +{ + struct alc_spec *spec = codec->spec; + + snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], + stream_tag, 0, format); + return 0; +} + +static int alc880_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + snd_pcm_substream_t *substream) +{ + struct alc_spec *spec = codec->spec; + + snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], 0, 0, 0); + return 0; +} + + +/* + */ +static struct hda_pcm_stream alc880_pcm_analog_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 8, + .nid = 0x02, /* NID to query formats and rates */ + .ops = { + .open = alc880_playback_pcm_open, + .prepare = alc880_playback_pcm_prepare, + .cleanup = alc880_playback_pcm_cleanup + }, +}; + +static struct hda_pcm_stream alc880_pcm_analog_capture = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x07, /* NID to query formats and rates */ + .ops = { + .prepare = alc880_capture_pcm_prepare, + .cleanup = alc880_capture_pcm_cleanup + }, +}; + +static struct hda_pcm_stream alc880_pcm_digital_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + /* NID is set in alc_build_pcms */ + .ops = { + .open = alc880_dig_playback_pcm_open, + .close = alc880_dig_playback_pcm_close + }, +}; + +static struct hda_pcm_stream alc880_pcm_digital_capture = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + /* NID is set in alc_build_pcms */ +}; + +static int alc_build_pcms(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + struct hda_pcm *info = spec->pcm_rec; + int i; + + codec->num_pcms = 1; + codec->pcm_info = info; + + info->name = spec->stream_name_analog; + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_analog_playback); + info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture); + + info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = 0; + for (i = 0; i < spec->num_channel_mode; i++) { + if (spec->channel_mode[i].channels > info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max) { + info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = spec->channel_mode[i].channels; + } + } + + if (spec->multiout.dig_out_nid || spec->dig_in_nid) { + codec->num_pcms++; + info++; + info->name = spec->stream_name_digital; + if (spec->multiout.dig_out_nid) { + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_digital_playback); + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid; + } + if (spec->dig_in_nid) { + info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_digital_capture); + info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid; + } + } + + return 0; +} + +static void alc_free(struct hda_codec *codec) +{ + kfree(codec->spec); +} + +/* + */ +static struct hda_codec_ops alc_patch_ops = { + .build_controls = alc_build_controls, + .build_pcms = alc_build_pcms, + .init = alc_init, + .free = alc_free, +#ifdef CONFIG_PM + .resume = alc_resume, +#endif +}; + +/* + */ + +static struct hda_board_config alc880_cfg_tbl[] = { + /* Back 3 jack, front 2 jack */ + { .modelname = "3stack", .config = ALC880_3ST }, + { .pci_vendor = 0x8086, .pci_device = 0xe200, .config = ALC880_3ST }, + { .pci_vendor = 0x8086, .pci_device = 0xe201, .config = ALC880_3ST }, + { .pci_vendor = 0x8086, .pci_device = 0xe202, .config = ALC880_3ST }, + { .pci_vendor = 0x8086, .pci_device = 0xe203, .config = ALC880_3ST }, + { .pci_vendor = 0x8086, .pci_device = 0xe204, .config = ALC880_3ST }, + { .pci_vendor = 0x8086, .pci_device = 0xe205, .config = ALC880_3ST }, + { .pci_vendor = 0x8086, .pci_device = 0xe206, .config = ALC880_3ST }, + { .pci_vendor = 0x8086, .pci_device = 0xe207, .config = ALC880_3ST }, + { .pci_vendor = 0x8086, .pci_device = 0xe208, .config = ALC880_3ST }, + { .pci_vendor = 0x8086, .pci_device = 0xe209, .config = ALC880_3ST }, + { .pci_vendor = 0x8086, .pci_device = 0xe20a, .config = ALC880_3ST }, + { .pci_vendor = 0x8086, .pci_device = 0xe20b, .config = ALC880_3ST }, + { .pci_vendor = 0x8086, .pci_device = 0xe20c, .config = ALC880_3ST }, + { .pci_vendor = 0x8086, .pci_device = 0xe20d, .config = ALC880_3ST }, + { .pci_vendor = 0x8086, .pci_device = 0xe20e, .config = ALC880_3ST }, + { .pci_vendor = 0x8086, .pci_device = 0xe20f, .config = ALC880_3ST }, + { .pci_vendor = 0x8086, .pci_device = 0xe210, .config = ALC880_3ST }, + { .pci_vendor = 0x8086, .pci_device = 0xe211, .config = ALC880_3ST }, + { .pci_vendor = 0x8086, .pci_device = 0xe214, .config = ALC880_3ST }, + { .pci_vendor = 0x8086, .pci_device = 0xe302, .config = ALC880_3ST }, + { .pci_vendor = 0x8086, .pci_device = 0xe303, .config = ALC880_3ST }, + { .pci_vendor = 0x8086, .pci_device = 0xe304, .config = ALC880_3ST }, + { .pci_vendor = 0x8086, .pci_device = 0xe306, .config = ALC880_3ST }, + { .pci_vendor = 0x8086, .pci_device = 0xe307, .config = ALC880_3ST }, + { .pci_vendor = 0x8086, .pci_device = 0xe404, .config = ALC880_3ST }, + { .pci_vendor = 0x8086, .pci_device = 0xa101, .config = ALC880_3ST }, + { .pci_vendor = 0x107b, .pci_device = 0x3031, .config = ALC880_3ST }, + { .pci_vendor = 0x107b, .pci_device = 0x4036, .config = ALC880_3ST }, + { .pci_vendor = 0x107b, .pci_device = 0x4037, .config = ALC880_3ST }, + { .pci_vendor = 0x107b, .pci_device = 0x4038, .config = ALC880_3ST }, + { .pci_vendor = 0x107b, .pci_device = 0x4040, .config = ALC880_3ST }, + { .pci_vendor = 0x107b, .pci_device = 0x4041, .config = ALC880_3ST }, + + /* Back 3 jack, front 2 jack (Internal add Aux-In) */ + { .pci_vendor = 0x1025, .pci_device = 0xe310, .config = ALC880_3ST }, + + /* Back 3 jack plus 1 SPDIF out jack, front 2 jack */ + { .modelname = "3stack-digout", .config = ALC880_3ST_DIG }, + { .pci_vendor = 0x8086, .pci_device = 0xe308, .config = ALC880_3ST_DIG }, + + /* Back 3 jack plus 1 SPDIF out jack, front 2 jack (Internal add Aux-In)*/ + { .pci_vendor = 0x8086, .pci_device = 0xe305, .config = ALC880_3ST_DIG }, + { .pci_vendor = 0x8086, .pci_device = 0xd402, .config = ALC880_3ST_DIG }, + { .pci_vendor = 0x1025, .pci_device = 0xe309, .config = ALC880_3ST_DIG }, + + /* Back 5 jack, front 2 jack */ + { .modelname = "5stack", .config = ALC880_5ST }, + { .pci_vendor = 0x107b, .pci_device = 0x3033, .config = ALC880_5ST }, + { .pci_vendor = 0x107b, .pci_device = 0x4039, .config = ALC880_5ST }, + { .pci_vendor = 0x107b, .pci_device = 0x3032, .config = ALC880_5ST }, + { .pci_vendor = 0x103c, .pci_device = 0x2a09, .config = ALC880_5ST }, + + /* Back 5 jack plus 1 SPDIF out jack, front 2 jack */ + { .modelname = "5stack-digout", .config = ALC880_5ST_DIG }, + { .pci_vendor = 0x8086, .pci_device = 0xe224, .config = ALC880_5ST_DIG }, + { .pci_vendor = 0x8086, .pci_device = 0xe400, .config = ALC880_5ST_DIG }, + { .pci_vendor = 0x8086, .pci_device = 0xe401, .config = ALC880_5ST_DIG }, + { .pci_vendor = 0x8086, .pci_device = 0xe402, .config = ALC880_5ST_DIG }, + { .pci_vendor = 0x8086, .pci_device = 0xd400, .config = ALC880_5ST_DIG }, + { .pci_vendor = 0x8086, .pci_device = 0xd401, .config = ALC880_5ST_DIG }, + { .pci_vendor = 0x8086, .pci_device = 0xa100, .config = ALC880_5ST_DIG }, + { .pci_vendor = 0x1565, .pci_device = 0x8202, .config = ALC880_5ST_DIG }, + + { .modelname = "w810", .config = ALC880_W810 }, + { .pci_vendor = 0x161f, .pci_device = 0x203d, .config = ALC880_W810 }, + + {} +}; + +static int patch_alc880(struct hda_codec *codec) +{ + struct alc_spec *spec; + int board_config; + + spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + board_config = snd_hda_check_board_config(codec, alc880_cfg_tbl); + if (board_config < 0) { + snd_printd(KERN_INFO "hda_codec: Unknown model for ALC880\n"); + board_config = ALC880_MINIMAL; + } + + switch (board_config) { + case ALC880_W810: + spec->mixers[spec->num_mixers] = alc880_w810_base_mixer; + spec->num_mixers++; + break; + case ALC880_5ST: + case ALC880_5ST_DIG: + spec->mixers[spec->num_mixers] = alc880_five_stack_mixer; + spec->num_mixers++; + break; + default: + spec->mixers[spec->num_mixers] = alc880_base_mixer; + spec->num_mixers++; + break; + } + + switch (board_config) { + case ALC880_3ST_DIG: + case ALC880_5ST_DIG: + case ALC880_W810: + spec->multiout.dig_out_nid = ALC880_DIGOUT_NID; + break; + default: + break; + } + + switch (board_config) { + case ALC880_3ST: + case ALC880_3ST_DIG: + case ALC880_5ST: + case ALC880_5ST_DIG: + case ALC880_W810: + spec->front_panel = 1; + break; + default: + break; + } + + switch (board_config) { + case ALC880_5ST: + case ALC880_5ST_DIG: + spec->init_verbs = alc880_init_verbs_five_stack; + spec->channel_mode = alc880_fivestack_modes; + spec->num_channel_mode = ARRAY_SIZE(alc880_fivestack_modes); + break; + case ALC880_W810: + spec->init_verbs = alc880_w810_init_verbs; + spec->channel_mode = alc880_w810_modes; + spec->num_channel_mode = ARRAY_SIZE(alc880_w810_modes); + break; + default: + spec->init_verbs = alc880_init_verbs_three_stack; + spec->channel_mode = alc880_threestack_modes; + spec->num_channel_mode = ARRAY_SIZE(alc880_threestack_modes); + break; + } + + spec->stream_name_analog = "ALC880 Analog"; + spec->stream_analog_playback = &alc880_pcm_analog_playback; + spec->stream_analog_capture = &alc880_pcm_analog_capture; + + spec->stream_name_digital = "ALC880 Digital"; + spec->stream_digital_playback = &alc880_pcm_digital_playback; + spec->stream_digital_capture = &alc880_pcm_digital_capture; + + spec->multiout.max_channels = spec->channel_mode[0].channels; + + switch (board_config) { + case ALC880_W810: + spec->multiout.num_dacs = ARRAY_SIZE(alc880_w810_dac_nids); + spec->multiout.dac_nids = alc880_w810_dac_nids; + // No dedicated headphone socket - it's shared with built-in speakers. + break; + default: + spec->multiout.num_dacs = ARRAY_SIZE(alc880_dac_nids); + spec->multiout.dac_nids = alc880_dac_nids; + spec->multiout.hp_nid = 0x03; /* rear-surround NID */ + break; + } + + spec->input_mux = &alc880_capture_source; + spec->num_adc_nids = ARRAY_SIZE(alc880_adc_nids); + spec->adc_nids = alc880_adc_nids; + + codec->patch_ops = alc_patch_ops; + + return 0; +} + +/* + * ALC260 support + */ + +/* + * This is just place-holder, so there's something for alc_build_pcms to look + * at when it calculates the maximum number of channels. ALC260 has no mixer + * element which allows changing the channel mode, so the verb list is + * never used. + */ +static struct alc_channel_mode alc260_modes[1] = { + { 2, NULL }, +}; + +snd_kcontrol_new_t alc260_base_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT), + /* use LINE2 for the output */ + /* HDA_CODEC_MUTE("Front Playback Switch", 0x0f, 0x0, HDA_OUTPUT), */ + HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x07, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x07, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x07, 0x05, HDA_INPUT), + HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x07, 0x05, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x10, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x11, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x04, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x04, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = alc_mux_enum_info, + .get = alc_mux_enum_get, + .put = alc_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_verb alc260_init_verbs[] = { + /* Line In pin widget for input */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + /* CD pin widget for input */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + /* Mic1 (rear panel) pin widget for input and vref at 80% */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + /* Mic2 (front panel) pin widget for input and vref at 80% */ + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + /* LINE-2 is used for line-out in rear */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* select line-out */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* LINE-OUT pin */ + {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* enable HP */ + {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* enable Mono */ + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* unmute amp left and right */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, + /* set connection select to line in (default select for this ADC) */ + {0x04, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* unmute Line-Out mixer amp left and right (volume = 0) */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + /* mute pin widget amp left and right (no gain on this amp) */ + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + /* unmute HP mixer amp left and right (volume = 0) */ + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + /* mute pin widget amp left and right (no gain on this amp) */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + /* unmute Mono mixer amp left and right (volume = 0) */ + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + /* mute pin widget amp left and right (no gain on this amp) */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + /* mute LINE-2 out */ + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 & Line In 2 = 0x03 */ + /* unmute CD */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))}, + /* unmute Line In */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, + /* unmute Mic */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + /* Amp Indexes: DAC = 0x01 & mixer = 0x00 */ + /* Unmute Front out path */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + /* Unmute Headphone out path */ + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + /* Unmute Mono out path */ + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + { } +}; + +static struct hda_pcm_stream alc260_pcm_analog_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .nid = 0x2, +}; + +static struct hda_pcm_stream alc260_pcm_analog_capture = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .nid = 0x4, +}; + +static int patch_alc260(struct hda_codec *codec) +{ + struct alc_spec *spec; + + spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + spec->mixers[spec->num_mixers] = alc260_base_mixer; + spec->num_mixers++; + + spec->init_verbs = alc260_init_verbs; + spec->channel_mode = alc260_modes; + spec->num_channel_mode = ARRAY_SIZE(alc260_modes); + + spec->stream_name_analog = "ALC260 Analog"; + spec->stream_analog_playback = &alc260_pcm_analog_playback; + spec->stream_analog_capture = &alc260_pcm_analog_capture; + + spec->multiout.max_channels = spec->channel_mode[0].channels; + spec->multiout.num_dacs = ARRAY_SIZE(alc260_dac_nids); + spec->multiout.dac_nids = alc260_dac_nids; + + spec->input_mux = &alc260_capture_source; + spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids); + spec->adc_nids = alc260_adc_nids; + + codec->patch_ops = alc_patch_ops; + + return 0; +} + +/* + * ALC882 support + * + * ALC882 is almost identical with ALC880 but has cleaner and more flexible + * configuration. Each pin widget can choose any input DACs and a mixer. + * Each ADC is connected from a mixer of all inputs. This makes possible + * 6-channel independent captures. + * + * In addition, an independent DAC for the multi-playback (not used in this + * driver yet). + */ + +static struct alc_channel_mode alc882_ch_modes[1] = { + { 8, NULL } +}; + +static hda_nid_t alc882_dac_nids[4] = { + /* front, rear, clfe, rear_surr */ + 0x02, 0x03, 0x04, 0x05 +}; + +static hda_nid_t alc882_adc_nids[3] = { + /* ADC0-2 */ + 0x07, 0x08, 0x09, +}; + +/* input MUX */ +/* FIXME: should be a matrix-type input source selection */ + +static struct hda_input_mux alc882_capture_source = { + .num_items = 4, + .items = { + { "Mic", 0x0 }, + { "Front Mic", 0x1 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + }, +}; + +#define alc882_mux_enum_info alc_mux_enum_info +#define alc882_mux_enum_get alc_mux_enum_get + +static int alc882_mux_enum_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + const struct hda_input_mux *imux = spec->input_mux; + unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + static hda_nid_t capture_mixers[3] = { 0x24, 0x23, 0x22 }; + hda_nid_t nid = capture_mixers[adc_idx]; + unsigned int *cur_val = &spec->cur_mux[adc_idx]; + unsigned int i, idx; + + idx = ucontrol->value.enumerated.item[0]; + if (idx >= imux->num_items) + idx = imux->num_items - 1; + if (*cur_val == idx && ! codec->in_resume) + return 0; + for (i = 0; i < imux->num_items; i++) { + unsigned int v = (i == idx) ? 0x7000 : 0x7080; + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, + v | (imux->items[i].index << 8)); + } + *cur_val = idx; + return 1; +} + +/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17 + * Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b + */ +static snd_kcontrol_new_t alc882_base_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Surround Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x16, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Side Playback Switch", 0x17, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 3, + .info = alc882_mux_enum_info, + .get = alc882_mux_enum_get, + .put = alc882_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_verb alc882_init_verbs[] = { + /* Front mixer: unmute input/output amp left and right (volume = 0) */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + /* Rear mixer */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + /* CLFE mixer */ + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + /* Side mixer */ + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + + /* Front Pin: to output mode */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* Front Pin: mute amp left and right (no volume) */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, + /* select Front mixer (0x0c, index 0) */ + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* Rear Pin */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* Rear Pin: mute amp left and right (no volume) */ + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, + /* select Rear mixer (0x0d, index 1) */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, + /* CLFE Pin */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* CLFE Pin: mute amp left and right (no volume) */ + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, + /* select CLFE mixer (0x0e, index 2) */ + {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* Side Pin */ + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* Side Pin: mute amp left and right (no volume) */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, + /* select Side mixer (0x0f, index 3) */ + {0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, + /* Headphone Pin */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* Headphone Pin: mute amp left and right (no volume) */ + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, + /* select Front mixer (0x0c, index 0) */ + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* Mic (rear) pin widget for input and vref at 80% */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + /* Front Mic pin widget for input and vref at 80% */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + /* Line In pin widget for input */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + /* CD pin widget for input */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + + /* FIXME: use matrix-type input source selection */ + /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ + /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, + /* Input mixer2 */ + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, + /* Input mixer3 */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, + /* ADC1: unmute amp left and right */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, + /* ADC2: unmute amp left and right */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, + /* ADC3: unmute amp left and right */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, + + /* Unmute front loopback */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + /* Unmute rear loopback */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + /* Mute CLFE loopback */ + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))}, + /* Unmute side loopback */ + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + + { } +}; + +static int patch_alc882(struct hda_codec *codec) +{ + struct alc_spec *spec; + + spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + spec->mixers[spec->num_mixers] = alc882_base_mixer; + spec->num_mixers++; + + spec->multiout.dig_out_nid = ALC880_DIGOUT_NID; + spec->dig_in_nid = ALC880_DIGIN_NID; + spec->front_panel = 1; + spec->init_verbs = alc882_init_verbs; + spec->channel_mode = alc882_ch_modes; + spec->num_channel_mode = ARRAY_SIZE(alc882_ch_modes); + + spec->stream_name_analog = "ALC882 Analog"; + spec->stream_analog_playback = &alc880_pcm_analog_playback; + spec->stream_analog_capture = &alc880_pcm_analog_capture; + + spec->stream_name_digital = "ALC882 Digital"; + spec->stream_digital_playback = &alc880_pcm_digital_playback; + spec->stream_digital_capture = &alc880_pcm_digital_capture; + + spec->multiout.max_channels = spec->channel_mode[0].channels; + spec->multiout.num_dacs = ARRAY_SIZE(alc882_dac_nids); + spec->multiout.dac_nids = alc882_dac_nids; + + spec->input_mux = &alc882_capture_source; + spec->num_adc_nids = ARRAY_SIZE(alc882_adc_nids); + spec->adc_nids = alc882_adc_nids; + + codec->patch_ops = alc_patch_ops; + + return 0; +} + +/* + * patch entries + */ +struct hda_codec_preset snd_hda_preset_realtek[] = { + { .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 }, + { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 }, + { .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 }, + {} /* terminator */ +}; |