diff options
Diffstat (limited to 'sound/pci/hda')
| -rw-r--r-- | sound/pci/hda/hda_codec.c | 140 | ||||
| -rw-r--r-- | sound/pci/hda/hda_codec.h | 4 | ||||
| -rw-r--r-- | sound/pci/hda/hda_generic.c | 128 | ||||
| -rw-r--r-- | sound/pci/hda/hda_intel.c | 122 | ||||
| -rw-r--r-- | sound/pci/hda/hda_local.h | 9 | ||||
| -rw-r--r-- | sound/pci/hda/patch_analog.c | 599 | ||||
| -rw-r--r-- | sound/pci/hda/patch_realtek.c | 1118 | ||||
| -rw-r--r-- | sound/pci/hda/patch_sigmatel.c | 251 |
8 files changed, 2018 insertions, 353 deletions
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 4a6dd97..b42dff7 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -25,6 +25,7 @@ #include <linux/slab.h> #include <linux/pci.h> #include <linux/moduleparam.h> +#include <linux/mutex.h> #include <sound/core.h> #include "hda_codec.h" #include <sound/asoundef.h> @@ -76,12 +77,12 @@ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, int dire unsigned int verb, unsigned int parm) { unsigned int res; - down(&codec->bus->cmd_mutex); + mutex_lock(&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); + mutex_unlock(&codec->bus->cmd_mutex); return res; } @@ -101,9 +102,9 @@ 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); + mutex_lock(&codec->bus->cmd_mutex); err = codec->bus->ops.command(codec, nid, direct, verb, parm); - up(&codec->bus->cmd_mutex); + mutex_unlock(&codec->bus->cmd_mutex); return err; } @@ -371,7 +372,7 @@ int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp, bus->modelname = temp->modelname; bus->ops = temp->ops; - init_MUTEX(&bus->cmd_mutex); + mutex_init(&bus->cmd_mutex); INIT_LIST_HEAD(&bus->codec_list); if ((err = snd_device_new(card, SNDRV_DEV_BUS, bus, &dev_ops)) < 0) { @@ -523,13 +524,19 @@ int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, codec->bus = bus; codec->addr = codec_addr; - init_MUTEX(&codec->spdif_mutex); + mutex_init(&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); + if (codec->vendor_id == -1) + /* read again, hopefully the access method was corrected + * in the last read... + */ + 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); @@ -722,7 +729,8 @@ static void put_vol_mute(struct hda_codec *codec, struct hda_amp_info *info, /* * read AMP value. The volume is between 0 to 0x7f, 0x80 = mute bit. */ -static int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int index) +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) @@ -733,7 +741,8 @@ static int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch /* * update the AMP value, mask = bit mask to set, val = the value */ -static int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int idx, int mask, int val) +int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, + int direction, int idx, int mask, int val) { struct hda_amp_info *info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, idx)); @@ -881,12 +890,12 @@ int snd_hda_mixer_bind_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_ unsigned long pval; int err; - down(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ pval = kcontrol->private_value; kcontrol->private_value = pval & ~AMP_VAL_IDX_MASK; /* index 0 */ err = snd_hda_mixer_amp_switch_get(kcontrol, ucontrol); kcontrol->private_value = pval; - up(&codec->spdif_mutex); + mutex_unlock(&codec->spdif_mutex); return err; } @@ -896,7 +905,7 @@ int snd_hda_mixer_bind_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ unsigned long pval; int i, indices, err = 0, change = 0; - down(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ pval = kcontrol->private_value; indices = (pval & AMP_VAL_IDX_MASK) >> AMP_VAL_IDX_SHIFT; for (i = 0; i < indices; i++) { @@ -907,7 +916,7 @@ int snd_hda_mixer_bind_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ change |= err; } kcontrol->private_value = pval; - up(&codec->spdif_mutex); + mutex_unlock(&codec->spdif_mutex); return err < 0 ? err : change; } @@ -1011,7 +1020,7 @@ static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol, struct snd_c unsigned short val; int change; - down(&codec->spdif_mutex); + mutex_lock(&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) | @@ -1026,7 +1035,7 @@ static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol, struct snd_c snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_2, val >> 8); } - up(&codec->spdif_mutex); + mutex_unlock(&codec->spdif_mutex); return change; } @@ -1054,7 +1063,7 @@ static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kcontrol, struct sn unsigned short val; int change; - down(&codec->spdif_mutex); + mutex_lock(&codec->spdif_mutex); val = codec->spdif_ctls & ~1; if (ucontrol->value.integer.value[0]) val |= 1; @@ -1066,7 +1075,7 @@ static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kcontrol, struct sn AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_OUTPUT | ((val & 1) ? 0 : 0x80)); } - up(&codec->spdif_mutex); + mutex_unlock(&codec->spdif_mutex); return change; } @@ -1150,13 +1159,13 @@ static int snd_hda_spdif_in_switch_put(struct snd_kcontrol *kcontrol, struct snd unsigned int val = !!ucontrol->value.integer.value[0]; int change; - down(&codec->spdif_mutex); + mutex_lock(&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); + mutex_unlock(&codec->spdif_mutex); return change; } @@ -1824,13 +1833,13 @@ int snd_hda_input_mux_put(struct hda_codec *codec, const struct hda_input_mux *i */ int snd_hda_multi_out_dig_open(struct hda_codec *codec, struct hda_multi_out *mout) { - down(&codec->spdif_mutex); + mutex_lock(&codec->spdif_mutex); if (mout->dig_out_used) { - up(&codec->spdif_mutex); + mutex_unlock(&codec->spdif_mutex); return -EBUSY; /* already being used */ } mout->dig_out_used = HDA_DIG_EXCLUSIVE; - up(&codec->spdif_mutex); + mutex_unlock(&codec->spdif_mutex); return 0; } @@ -1839,9 +1848,9 @@ int snd_hda_multi_out_dig_open(struct hda_codec *codec, struct hda_multi_out *mo */ int snd_hda_multi_out_dig_close(struct hda_codec *codec, struct hda_multi_out *mout) { - down(&codec->spdif_mutex); + mutex_lock(&codec->spdif_mutex); mout->dig_out_used = 0; - up(&codec->spdif_mutex); + mutex_unlock(&codec->spdif_mutex); return 0; } @@ -1869,7 +1878,7 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, struct hda_multi_o int chs = substream->runtime->channels; int i; - down(&codec->spdif_mutex); + mutex_lock(&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) && @@ -1883,13 +1892,20 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, struct hda_multi_o snd_hda_codec_setup_stream(codec, mout->dig_out_nid, 0, 0, 0); } } - up(&codec->spdif_mutex); + mutex_unlock(&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); + /* extra outputs copied from front */ + for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++) + if (mout->extra_out_nid[i]) + snd_hda_codec_setup_stream(codec, + mout->extra_out_nid[i], + stream_tag, 0, format); + /* surrounds */ for (i = 1; i < mout->num_dacs; i++) { if (chs >= (i + 1) * 2) /* independent out */ @@ -1914,12 +1930,17 @@ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, struct hda_multi_o 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); + for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++) + if (mout->extra_out_nid[i]) + snd_hda_codec_setup_stream(codec, + mout->extra_out_nid[i], + 0, 0, 0); + mutex_lock(&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); + mutex_unlock(&codec->spdif_mutex); return 0; } @@ -1935,13 +1956,29 @@ static int is_in_nid_list(hda_nid_t nid, hda_nid_t *list) return 0; } -/* parse all pin widgets and store the useful pin nids to cfg */ +/* + * Parse all pin widgets and store the useful pin nids to cfg + * + * The number of line-outs or any primary output is stored in line_outs, + * and the corresponding output pins are assigned to line_out_pins[], + * in the order of front, rear, CLFE, side, ... + * + * If more extra outputs (speaker and headphone) are found, the pins are + * assisnged to hp_pin and speaker_pins[], respectively. If no line-out jack + * is detected, one of speaker of HP pins is assigned as the primary + * output, i.e. to line_out_pins[0]. So, line_outs is always positive + * if any analog output exists. + * + * The analog input pins are assigned to input_pins array. + * The digital input/output pins are assigned to dig_in_pin and dig_out_pin, + * respectively. + */ int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *cfg, hda_nid_t *ignore_nids) { hda_nid_t nid, nid_start; int i, j, nodes; - short seq, sequences[4], assoc_line_out; + short seq, assoc_line_out, sequences[ARRAY_SIZE(cfg->line_out_pins)]; memset(cfg, 0, sizeof(*cfg)); @@ -1983,7 +2020,10 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *c cfg->line_outs++; break; case AC_JACK_SPEAKER: - cfg->speaker_pin = nid; + if (cfg->speaker_outs >= ARRAY_SIZE(cfg->speaker_pins)) + continue; + cfg->speaker_pins[cfg->speaker_outs] = nid; + cfg->speaker_outs++; break; case AC_JACK_HP_OUT: cfg->hp_pin = nid; @@ -2048,6 +2088,46 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *c break; } + /* + * debug prints of the parsed results + */ + snd_printd("autoconfig: line_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n", + cfg->line_outs, cfg->line_out_pins[0], cfg->line_out_pins[1], + cfg->line_out_pins[2], cfg->line_out_pins[3], + cfg->line_out_pins[4]); + snd_printd(" speaker_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n", + cfg->speaker_outs, cfg->speaker_pins[0], + cfg->speaker_pins[1], cfg->speaker_pins[2], + cfg->speaker_pins[3], cfg->speaker_pins[4]); + snd_printd(" hp=0x%x, dig_out=0x%x, din_in=0x%x\n", + cfg->hp_pin, cfg->dig_out_pin, cfg->dig_in_pin); + snd_printd(" inputs: mic=0x%x, fmic=0x%x, line=0x%x, fline=0x%x," + " cd=0x%x, aux=0x%x\n", + cfg->input_pins[AUTO_PIN_MIC], + cfg->input_pins[AUTO_PIN_FRONT_MIC], + cfg->input_pins[AUTO_PIN_LINE], + cfg->input_pins[AUTO_PIN_FRONT_LINE], + cfg->input_pins[AUTO_PIN_CD], + cfg->input_pins[AUTO_PIN_AUX]); + + /* + * FIX-UP: if no line-outs are detected, try to use speaker or HP pin + * as a primary output + */ + if (! cfg->line_outs) { + if (cfg->speaker_outs) { + cfg->line_outs = cfg->speaker_outs; + memcpy(cfg->line_out_pins, cfg->speaker_pins, + sizeof(cfg->speaker_pins)); + cfg->speaker_outs = 0; + memset(cfg->speaker_pins, 0, sizeof(cfg->speaker_pins)); + } else if (cfg->hp_pin) { + cfg->line_outs = 1; + cfg->line_out_pins[0] = cfg->hp_pin; + cfg->hp_pin = 0; + } + } + return 0; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 63e26c7..40520e9 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -438,7 +438,7 @@ struct hda_bus { struct list_head codec_list; struct hda_codec *caddr_tbl[HDA_MAX_CODEC_ADDRESS + 1]; /* caddr -> codec */ - struct semaphore cmd_mutex; + struct mutex cmd_mutex; /* unsolicited event queue */ struct hda_bus_unsolicited *unsol; @@ -559,7 +559,7 @@ struct hda_codec { int amp_info_size; struct hda_amp_info *amp_info; - struct semaphore spdif_mutex; + struct mutex spdif_mutex; unsigned int spdif_status; /* IEC958 status bits */ unsigned short spdif_ctls; /* SPDIF control bits */ unsigned int spdif_in_enable; /* SPDIF input enable? */ diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 39edfcf..85ad164a 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -47,10 +47,10 @@ struct hda_gnode { /* patch-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 *dac_node[2]; /* DAC node */ + struct hda_gnode *out_pin_node[2]; /* Output pin (Line-Out) node */ + struct hda_gnode *pcm_vol_node[2]; /* Node for PCM volume */ + unsigned int pcm_vol_index[2]; /* connection of PCM volume */ struct hda_gnode *adc_node; /* ADC node */ struct hda_gnode *cap_vol_node; /* Node for capture volume */ @@ -69,8 +69,12 @@ struct hda_gspec { /* * retrieve the default device type from the default config value */ -#define defcfg_type(node) (((node)->def_cfg & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT) -#define defcfg_location(node) (((node)->def_cfg & AC_DEFCFG_LOCATION) >> AC_DEFCFG_LOCATION_SHIFT) +#define defcfg_type(node) (((node)->def_cfg & AC_DEFCFG_DEVICE) >> \ + AC_DEFCFG_DEVICE_SHIFT) +#define defcfg_location(node) (((node)->def_cfg & AC_DEFCFG_LOCATION) >> \ + AC_DEFCFG_LOCATION_SHIFT) +#define defcfg_port_conn(node) (((node)->def_cfg & AC_DEFCFG_PORT_CONN) >> \ + AC_DEFCFG_PORT_CONN_SHIFT) /* * destructor @@ -261,7 +265,7 @@ static void clear_check_flags(struct hda_gspec *spec) * 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) + struct hda_gnode *node, int dac_idx) { int i, err; struct hda_gnode *child; @@ -276,14 +280,14 @@ static int parse_output_path(struct hda_codec *codec, struct hda_gspec *spec, return 0; } snd_printdd("AUD_OUT found %x\n", node->nid); - if (spec->dac_node) { + if (spec->dac_node[dac_idx]) { /* already DAC node is assigned, just unmute & connect */ - return node == spec->dac_node; + return node == spec->dac_node[dac_idx]; } - spec->dac_node = node; + spec->dac_node[dac_idx] = node; if (node->wid_caps & AC_WCAP_OUT_AMP) { - spec->pcm_vol_node = node; - spec->pcm_vol_index = 0; + spec->pcm_vol_node[dac_idx] = node; + spec->pcm_vol_index[dac_idx] = 0; } return 1; /* found */ } @@ -292,7 +296,7 @@ static int parse_output_path(struct hda_codec *codec, struct hda_gspec *spec, child = hda_get_node(spec, node->conn_list[i]); if (! child) continue; - err = parse_output_path(codec, spec, child); + err = parse_output_path(codec, spec, child, dac_idx); if (err < 0) return err; else if (err > 0) { @@ -303,13 +307,13 @@ static int parse_output_path(struct hda_codec *codec, struct hda_gspec *spec, select_input_connection(codec, node, i); unmute_input(codec, node, i); unmute_output(codec, node); - if (! spec->pcm_vol_node) { + if (! spec->pcm_vol_node[dac_idx]) { if (node->wid_caps & AC_WCAP_IN_AMP) { - spec->pcm_vol_node = node; - spec->pcm_vol_index = i; + spec->pcm_vol_node[dac_idx] = node; + spec->pcm_vol_index[dac_idx] = i; } else if (node->wid_caps & AC_WCAP_OUT_AMP) { - spec->pcm_vol_node = node; - spec->pcm_vol_index = 0; + spec->pcm_vol_node[dac_idx] = node; + spec->pcm_vol_index[dac_idx] = 0; } } return 1; @@ -339,6 +343,8 @@ static struct hda_gnode *parse_output_jack(struct hda_codec *codec, /* output capable? */ if (! (node->pin_caps & AC_PINCAP_OUT)) continue; + if (defcfg_port_conn(node) == AC_JACK_PORT_NONE) + continue; /* unconnected */ if (jack_type >= 0) { if (jack_type != defcfg_type(node)) continue; @@ -350,10 +356,15 @@ static struct hda_gnode *parse_output_jack(struct hda_codec *codec, continue; } clear_check_flags(spec); - err = parse_output_path(codec, spec, node); + err = parse_output_path(codec, spec, node, 0); if (err < 0) return NULL; - else if (err > 0) { + if (! err && spec->out_pin_node[0]) { + err = parse_output_path(codec, spec, node, 1); + if (err < 0) + return NULL; + } + if (err > 0) { /* unmute the PIN output */ unmute_output(codec, node); /* set PIN-Out enable */ @@ -381,20 +392,28 @@ static int parse_output(struct hda_codec *codec) /* 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; + spec->out_pin_node[0] = node; + else { + /* if no line-out is found, try speaker out */ + node = parse_output_jack(codec, spec, AC_JACK_SPEAKER); + if (node) + spec->out_pin_node[0] = 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[0]) + spec->out_pin_node[0] = node; + else + spec->out_pin_node[1] = node; } - if (! spec->out_pin_node) { + if (! spec->out_pin_node[0]) { /* 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) + spec->out_pin_node[0] = parse_output_jack(codec, spec, -1); + if (! spec->out_pin_node[0]) snd_printd("hda_generic: no proper output path found\n"); } @@ -505,6 +524,9 @@ static int parse_adc_sub_nodes(struct hda_codec *codec, struct hda_gspec *spec, if (! (node->pin_caps & AC_PINCAP_IN)) return 0; + if (defcfg_port_conn(node) == AC_JACK_PORT_NONE) + return 0; /* unconnected */ + if (node->wid_caps & AC_WCAP_DIGITAL) return 0; /* skip SPDIF */ @@ -703,12 +725,16 @@ static int check_existing_control(struct hda_codec *codec, const char *type, con static int build_output_controls(struct hda_codec *codec) { struct hda_gspec *spec = codec->spec; - int err; + static const char *types[2] = { "Master", "Headphone" }; + int i, err; - err = create_mixer(codec, spec->pcm_vol_node, spec->pcm_vol_index, - "PCM", "Playback"); - if (err < 0) - return err; + for (i = 0; i < 2 && spec->pcm_vol_node[i]; i++) { + err = create_mixer(codec, spec->pcm_vol_node[i], + spec->pcm_vol_index[i], + types[i], "Playback"); + if (err < 0) + return err; + } return 0; } @@ -805,7 +831,7 @@ static int build_loopback_controls(struct hda_codec *codec) int err; const char *type; - if (! spec->out_pin_node) + if (! spec->out_pin_node[0]) return 0; list_for_each(p, &spec->nid_list) { @@ -820,7 +846,8 @@ static int build_loopback_controls(struct hda_codec *codec) if (check_existing_control(codec, type, "Playback")) continue; clear_check_flags(spec); - err = parse_loopback_path(codec, spec, spec->out_pin_node, + err = parse_loopback_path(codec, spec, + spec->out_pin_node[0], node, type); if (err < 0) return err; @@ -855,12 +882,37 @@ static struct hda_pcm_stream generic_pcm_playback = { .channels_max = 2, }; +static int generic_pcm2_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct hda_gspec *spec = codec->spec; + + snd_hda_codec_setup_stream(codec, hinfo->nid, stream_tag, 0, format); + snd_hda_codec_setup_stream(codec, spec->dac_node[1]->nid, + stream_tag, 0, format); + return 0; +} + +static int generic_pcm2_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct hda_gspec *spec = codec->spec; + + snd_hda_codec_setup_stream(codec, hinfo->nid, 0, 0, 0); + snd_hda_codec_setup_stream(codec, spec->dac_node[1]->nid, 0, 0, 0); + return 0; +} + 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) { + if (! spec->dac_node[0] && ! spec->adc_node) { snd_printd("hda_generic: no PCM found\n"); return 0; } @@ -869,9 +921,13 @@ static int build_generic_pcms(struct hda_codec *codec) codec->pcm_info = info; info->name = "HDA Generic"; - if (spec->dac_node) { + if (spec->dac_node[0]) { info->stream[0] = generic_pcm_playback; - info->stream[0].nid = spec->dac_node->nid; + info->stream[0].nid = spec->dac_node[0]->nid; + if (spec->dac_node[1]) { + info->stream[0].ops.prepare = generic_pcm2_prepare; + info->stream[0].ops.cleanup = generic_pcm2_cleanup; + } } if (spec->adc_node) { info->stream[1] = generic_pcm_playback; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fd12b69..c096606 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -43,6 +43,7 @@ #include <linux/init.h> #include <linux/slab.h> #include <linux/pci.h> +#include <linux/mutex.h> #include <sound/core.h> #include <sound/initval.h> #include "hda_codec.h" @@ -53,6 +54,7 @@ static char *id = SNDRV_DEFAULT_STR1; static char *model; static int position_fix; static int probe_mask = -1; +static int single_cmd; module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); @@ -64,6 +66,8 @@ module_param(position_fix, int, 0444); MODULE_PARM_DESC(position_fix, "Fix DMA pointer (0 = auto, 1 = none, 2 = POSBUF, 3 = FIFO size)."); module_param(probe_mask, int, 0444); MODULE_PARM_DESC(probe_mask, "Bitmask to probe codecs (default = -1)."); +module_param(single_cmd, bool, 0444); +MODULE_PARM_DESC(single_cmd, "Use single command to communicate with codecs (for debugging only)."); /* just for backward compatibility */ @@ -235,12 +239,6 @@ enum { #define NVIDIA_HDA_ENABLE_COHBITS 0x0f /* - * Use CORB/RIRB for communication from/to codecs. - * This is the way recommended by Intel (see below). - */ -#define USE_CORB_RIRB - -/* */ struct azx_dev { @@ -252,7 +250,6 @@ struct azx_dev { 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 */ - unsigned int last_pos; /* last updated period position */ void __iomem *sd_addr; /* stream descriptor pointer */ @@ -263,10 +260,11 @@ struct azx_dev { 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 */ + /* for sanity check of position buffer */ + unsigned int period_intr; unsigned int opened: 1; unsigned int running: 1; - unsigned int period_updating: 1; }; /* CORB/RIRB */ @@ -300,7 +298,7 @@ struct azx { /* locks */ spinlock_t reg_lock; - struct semaphore open_mutex; + struct mutex open_mutex; /* streams (x num_streams) */ struct azx_dev *azx_dev; @@ -325,6 +323,7 @@ struct azx { /* flags */ int position_fix; unsigned int initialized: 1; + unsigned int single_cmd: 1; }; /* driver types */ @@ -388,7 +387,6 @@ static char *driver_short_names[] __devinitdata = { * Interface for HD codec */ -#ifdef USE_CORB_RIRB /* * CORB / RIRB interface */ @@ -436,11 +434,7 @@ static void azx_init_cmd_io(struct azx *chip) /* 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; } @@ -452,8 +446,8 @@ static void azx_free_cmd_io(struct azx *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) +static int azx_corb_send_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, + unsigned int verb, unsigned int para) { struct azx *chip = codec->bus->private_data; unsigned int wp; @@ -509,18 +503,21 @@ static void azx_update_rirb(struct azx *chip) } /* receive a response */ -static unsigned int azx_get_response(struct hda_codec *codec) +static unsigned int azx_rirb_get_response(struct hda_codec *codec) { struct azx *chip = codec->bus->private_data; int timeout = 50; while (chip->rirb.cmds) { if (! --timeout) { - if (printk_ratelimit()) - snd_printk(KERN_ERR - "azx_get_response timeout\n"); + snd_printk(KERN_ERR + "hda_intel: azx_get_response timeout, " + "switching to single_cmd mode...\n"); chip->rirb.rp = azx_readb(chip, RIRBWP); chip->rirb.cmds = 0; + /* switch to single_cmd mode */ + chip->single_cmd = 1; + azx_free_cmd_io(chip); return -1; } msleep(1); @@ -528,7 +525,6 @@ static unsigned int azx_get_response(struct hda_codec *codec) return chip->rirb.res; /* the last value */ } -#else /* * Use the single immediate command instead of CORB/RIRB for simplicity * @@ -539,13 +535,10 @@ static unsigned int azx_get_response(struct hda_codec *codec) * 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) +static int azx_single_send_cmd(struct hda_codec *codec, hda_nid_t nid, + int direct, unsigned int verb, + unsigned int para) { struct azx *chip = codec->bus->private_data; u32 val; @@ -573,7 +566,7 @@ static int azx_send_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, } /* receive a response */ -static unsigned int azx_get_response(struct hda_codec *codec) +static unsigned int azx_single_get_response(struct hda_codec *codec) { struct azx *chip = codec->bus->private_data; int timeout = 50; @@ -588,9 +581,35 @@ static unsigned int azx_get_response(struct hda_codec *codec) return (unsigned int)-1; } -#define azx_update_rirb(chip) +/* + * The below are the main callbacks from hda_codec. + * + * They are just the skeleton to call sub-callbacks according to the + * current setting of chip->single_cmd. + */ + +/* send a command */ +static int azx_send_cmd(struct hda_codec *codec, hda_nid_t nid, + int direct, unsigned int verb, + unsigned int para) +{ + struct azx *chip = codec->bus->private_data; + if (chip->single_cmd) + return azx_single_send_cmd(codec, nid, direct, verb, para); + else + return azx_corb_send_cmd(codec, nid, direct, verb, para); +} + +/* get a response */ +static unsigned int azx_get_response(struct hda_codec *codec) +{ + struct azx *chip = codec->bus->private_data; + if (chip->single_cmd) + return azx_single_get_response(codec); + else + return azx_rirb_get_response(codec); +} -#endif /* USE_CORB_RIRB */ /* reset codec link */ static int azx_reset(struct azx *chip) @@ -737,7 +756,8 @@ static void azx_init_chip(struct azx *chip) azx_int_enable(chip); /* initialize the codec command I/O */ - azx_init_cmd_io(chip); + if (! chip->single_cmd) + azx_init_cmd_io(chip); /* program the position buffer */ azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr); @@ -784,11 +804,10 @@ static irqreturn_t azx_interrupt(int irq, void* dev_id, struct pt_regs *regs) 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) { - azx_dev->period_updating = 1; + azx_dev->period_intr++; spin_unlock(&chip->reg_lock); snd_pcm_period_elapsed(azx_dev->substream); spin_lock(&chip->reg_lock); - azx_dev->period_updating = 0; } } } @@ -796,7 +815,7 @@ static irqreturn_t azx_interrupt(int irq, void* dev_id, struct pt_regs *regs) /* clear rirb int */ status = azx_readb(chip, RIRBSTS); if (status & RIRB_INT_MASK) { - if (status & RIRB_INT_RESPONSE) + if (! chip->single_cmd && (status & RIRB_INT_RESPONSE)) azx_update_rirb(chip); azx_writeb(chip, RIRBSTS, RIRB_INT_MASK); } @@ -1002,10 +1021,10 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) unsigned long flags; int err; - down(&chip->open_mutex); + mutex_lock(&chip->open_mutex); azx_dev = azx_assign_device(chip, substream->stream); if (azx_dev == NULL) { - up(&chip->open_mutex); + mutex_unlock(&chip->open_mutex); return -EBUSY; } runtime->hw = azx_pcm_hw; @@ -1017,7 +1036,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) 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); + mutex_unlock(&chip->open_mutex); return err; } spin_lock_irqsave(&chip->reg_lock, flags); @@ -1026,7 +1045,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) spin_unlock_irqrestore(&chip->reg_lock, flags); runtime->private_data = azx_dev; - up(&chip->open_mutex); + mutex_unlock(&chip->open_mutex); return 0; } @@ -1038,14 +1057,14 @@ static int azx_pcm_close(struct snd_pcm_substream *substream) struct azx_dev *azx_dev = get_azx_dev(substream); unsigned long flags; - down(&chip->open_mutex); + mutex_lock(&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); + mutex_unlock(&chip->open_mutex); return 0; } @@ -1099,7 +1118,6 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) azx_dev->fifo_size = azx_sd_readw(azx_dev, SD_FIFOSIZE) + 1; else azx_dev->fifo_size = 0; - azx_dev->last_pos = 0; return hinfo->ops.prepare(hinfo, apcm->codec, azx_dev->stream_tag, azx_dev->format_val, substream); @@ -1147,10 +1165,20 @@ static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream) struct azx_dev *azx_dev = get_azx_dev(substream); unsigned int pos; - if (chip->position_fix == POS_FIX_POSBUF) { + if (chip->position_fix == POS_FIX_POSBUF || + chip->position_fix == POS_FIX_AUTO) { /* use the position buffer */ pos = *azx_dev->posbuf; + if (chip->position_fix == POS_FIX_AUTO && + azx_dev->period_intr == 1 && ! pos) { + printk(KERN_WARNING + "hda-intel: Invalid position buffer, " + "using LPIB read method instead.\n"); + chip->position_fix = POS_FIX_NONE; + goto read_lpib; + } } else { + read_lpib: /* read LPIB */ pos = azx_sd_readl(azx_dev, SD_LPIB); if (chip->position_fix == POS_FIX_FIFO) @@ -1415,13 +1443,14 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, } spin_lock_init(&chip->reg_lock); - init_MUTEX(&chip->open_mutex); + mutex_init(&chip->open_mutex); chip->card = card; chip->pci = pci; chip->irq = -1; chip->driver_type = driver_type; - chip->position_fix = position_fix ? position_fix : POS_FIX_POSBUF; + chip->position_fix = position_fix; + chip->single_cmd = single_cmd; #if BITS_PER_LONG != 64 /* Fix up base address on ULI M5461 */ @@ -1492,8 +1521,9 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, goto errout; } /* allocate CORB/RIRB */ - if ((err = azx_alloc_cmd_io(chip)) < 0) - goto errout; + if (! chip->single_cmd) + if ((err = azx_alloc_cmd_io(chip)) < 0) + goto errout; /* initialize streams */ azx_init_stream(chip); diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index c82d2a7..14e8aa2 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -66,6 +66,11 @@ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e int snd_hda_mixer_amp_switch_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +/* lowlevel accessor with caching; use carefully */ +int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch, + int direction, int index); +int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, + int direction, int idx, int mask, int val); /* mono switch binding multiple inputs */ #define HDA_BIND_MUTE_MONO(xname, nid, channel, indices, direction) \ @@ -130,6 +135,7 @@ 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 extra_out_nid[3]; /* optional DACs, 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) */ @@ -216,7 +222,8 @@ extern const char *auto_pin_cfg_labels[AUTO_PIN_LAST]; struct auto_pin_cfg { int line_outs; hda_nid_t line_out_pins[5]; /* sorted in the order of Front/Surr/CLFE/Side */ - hda_nid_t speaker_pin; + int speaker_outs; + hda_nid_t speaker_pins[5]; hda_nid_t hp_pin; hda_nid_t input_pins[AUTO_PIN_LAST]; hda_nid_t dig_out_pin; diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 1ada1b0..32401bd 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -23,6 +23,8 @@ #include <linux/delay.h> #include <linux/slab.h> #include <linux/pci.h> +#include <linux/mutex.h> + #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" @@ -60,7 +62,7 @@ struct ad198x_spec { /* PCM information */ struct hda_pcm pcm_rec[2]; /* used in alc_build_pcms() */ - struct semaphore amp_mutex; /* PCM volume/mute control mutex */ + struct mutex amp_mutex; /* PCM volume/mute control mutex */ unsigned int spdif_route; /* dynamic controls, init_verbs and input_mux */ @@ -308,7 +310,7 @@ static int ad198x_resume(struct hda_codec *codec) struct ad198x_spec *spec = codec->spec; int i; - ad198x_init(codec); + codec->patch_ops.init(codec); for (i = 0; i < spec->num_mixers; i++) snd_hda_resume_ctls(codec, spec->mixers[i]); if (spec->multiout.dig_out_nid) @@ -331,6 +333,61 @@ static struct hda_codec_ops ad198x_patch_ops = { /* + * EAPD control + * the private value = nid | (invert << 8) + */ +static int ad198x_eapd_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int ad198x_eapd_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ad198x_spec *spec = codec->spec; + int invert = (kcontrol->private_value >> 8) & 1; + if (invert) + ucontrol->value.integer.value[0] = ! spec->cur_eapd; + else + ucontrol->value.integer.value[0] = spec->cur_eapd; + return 0; +} + +static int ad198x_eapd_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ad198x_spec *spec = codec->spec; + int invert = (kcontrol->private_value >> 8) & 1; + hda_nid_t nid = kcontrol->private_value & 0xff; + unsigned int eapd; + eapd = ucontrol->value.integer.value[0]; + if (invert) + eapd = !eapd; + if (eapd == spec->cur_eapd && ! codec->in_resume) + return 0; + spec->cur_eapd = eapd; + snd_hda_codec_write(codec, nid, + 0, AC_VERB_SET_EAPD_BTLENABLE, + eapd ? 0x02 : 0x00); + return 1; +} + +static int ad198x_ch_mode_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +static int ad198x_ch_mode_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + + +/* * AD1986A specific */ @@ -344,6 +401,7 @@ static hda_nid_t ad1986a_dac_nids[3] = { AD1986A_FRONT_DAC, AD1986A_SURR_DAC, AD1986A_CLFE_DAC }; static hda_nid_t ad1986a_adc_nids[1] = { AD1986A_ADC }; +static hda_nid_t ad1986a_capsrc_nids[1] = { 0x12 }; static struct hda_input_mux ad1986a_capture_source = { .num_items = 7, @@ -371,9 +429,9 @@ static int ad1986a_pcm_amp_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct ad198x_spec *ad = codec->spec; - down(&ad->amp_mutex); + mutex_lock(&ad->amp_mutex); snd_hda_mixer_amp_volume_get(kcontrol, ucontrol); - up(&ad->amp_mutex); + mutex_unlock(&ad->amp_mutex); return 0; } @@ -383,13 +441,13 @@ static int ad1986a_pcm_amp_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl struct ad198x_spec *ad = codec->spec; int i, change = 0; - down(&ad->amp_mutex); + mutex_lock(&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); + mutex_unlock(&ad->amp_mutex); return change; } @@ -400,9 +458,9 @@ static int ad1986a_pcm_amp_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_ struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct ad198x_spec *ad = codec->spec; - down(&ad->amp_mutex); + mutex_lock(&ad->amp_mutex); snd_hda_mixer_amp_switch_get(kcontrol, ucontrol); - up(&ad->amp_mutex); + mutex_unlock(&ad->amp_mutex); return 0; } @@ -412,13 +470,13 @@ static int ad1986a_pcm_amp_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ struct ad198x_spec *ad = codec->spec; int i, change = 0; - down(&ad->amp_mutex); + mutex_lock(&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); + mutex_unlock(&ad->amp_mutex); return change; } @@ -477,6 +535,143 @@ static struct snd_kcontrol_new ad1986a_mixers[] = { { } /* end */ }; +/* additional mixers for 3stack mode */ +static struct snd_kcontrol_new ad1986a_3st_mixers[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Channel Mode", + .info = ad198x_ch_mode_info, + .get = ad198x_ch_mode_get, + .put = ad198x_ch_mode_put, + }, + { } /* end */ +}; + +/* laptop model - 2ch only */ +static hda_nid_t ad1986a_laptop_dac_nids[1] = { AD1986A_FRONT_DAC }; + +static struct snd_kcontrol_new ad1986a_laptop_mixers[] = { + HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Master Playback Volume", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Master Playback Switch", 0x1b, 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 = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + { } /* end */ +}; + +/* laptop-eapd model - 2ch only */ + +/* master controls both pins 0x1a and 0x1b */ +static int ad1986a_laptop_master_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + long *valp = ucontrol->value.integer.value; + int change; + + change = snd_hda_codec_amp_update(codec, 0x1a, 0, HDA_OUTPUT, 0, + 0x7f, valp[0] & 0x7f); + change |= snd_hda_codec_amp_update(codec, 0x1a, 1, HDA_OUTPUT, 0, + 0x7f, valp[1] & 0x7f); + snd_hda_codec_amp_update(codec, 0x1b, 0, HDA_OUTPUT, 0, + 0x7f, valp[0] & 0x7f); + snd_hda_codec_amp_update(codec, 0x1b, 1, HDA_OUTPUT, 0, + 0x7f, valp[1] & 0x7f); + return change; +} + +static int ad1986a_laptop_master_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + long *valp = ucontrol->value.integer.value; + int change; + + change = snd_hda_codec_amp_update(codec, 0x1a, 0, HDA_OUTPUT, 0, + 0x80, valp[0] ? 0 : 0x80); + change |= snd_hda_codec_amp_update(codec, 0x1a, 1, HDA_OUTPUT, 0, + 0x80, valp[1] ? 0 : 0x80); + snd_hda_codec_amp_update(codec, 0x1b, 0, HDA_OUTPUT, 0, + 0x80, valp[0] ? 0 : 0x80); + snd_hda_codec_amp_update(codec, 0x1b, 1, HDA_OUTPUT, 0, + 0x80, valp[1] ? 0 : 0x80); + return change; +} + +static struct hda_input_mux ad1986a_laptop_eapd_capture_source = { + .num_items = 3, + .items = { + { "Mic", 0x0 }, + { "Internal Mic", 0x4 }, + { "Mix", 0x5 }, + }, +}; + +static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Volume", + .info = snd_hda_mixer_amp_volume_info, + .get = snd_hda_mixer_amp_volume_get, + .put = ad1986a_laptop_master_vol_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT), + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = ad1986a_laptop_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT), + }, + HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 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("Capture Volume", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "External Amplifier", + .info = ad198x_eapd_info, + .get = ad198x_eapd_get, + .put = ad198x_eapd_put, + .private_value = 0x1b | (1 << 8), /* port-D, inversed */ + }, + { } /* end */ +}; + /* * initialization verbs */ @@ -535,16 +730,89 @@ static struct hda_verb ad1986a_init_verbs[] = { { } /* end */ }; +/* additional verbs for 3-stack model */ +static struct hda_verb ad1986a_3st_init_verbs[] = { + /* Mic and line-in selectors */ + {0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, + {0x10, AC_VERB_SET_CONNECT_SEL, 0x1}, + { } /* end */ +}; + +static struct hda_verb ad1986a_ch2_init[] = { + /* Surround out -> Line In */ + { 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, + { 0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + /* CLFE -> Mic in */ + { 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, + { 0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + { } /* end */ +}; + +static struct hda_verb ad1986a_ch4_init[] = { + /* Surround out -> Surround */ + { 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, + { 0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + /* CLFE -> Mic in */ + { 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, + { 0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + { } /* end */ +}; + +static struct hda_verb ad1986a_ch6_init[] = { + /* Surround out -> Surround out */ + { 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, + { 0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + /* CLFE -> CLFE */ + { 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, + { 0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + { } /* end */ +}; + +static struct hda_channel_mode ad1986a_modes[3] = { + { 2, ad1986a_ch2_init }, + { 4, ad1986a_ch4_init }, + { 6, ad1986a_ch6_init }, +}; + +/* eapd initialization */ +static struct hda_verb ad1986a_eapd_init_verbs[] = { + {0x1b, AC_VERB_SET_EAPD_BTLENABLE, 0x00}, + {} +}; + +/* models */ +enum { AD1986A_6STACK, AD1986A_3STACK, AD1986A_LAPTOP, AD1986A_LAPTOP_EAPD }; + +static struct hda_board_config ad1986a_cfg_tbl[] = { + { .modelname = "6stack", .config = AD1986A_6STACK }, + { .modelname = "3stack", .config = AD1986A_3STACK }, + { .pci_subvendor = 0x10de, .pci_subdevice = 0xcb84, + .config = AD1986A_3STACK }, /* ASUS A8N-VM CSM */ + { .modelname = "laptop", .config = AD1986A_LAPTOP }, + { .pci_subvendor = 0x144d, .pci_subdevice = 0xc01e, + .config = AD1986A_LAPTOP }, /* FSC V2060 */ + { .pci_subvendor = 0x17c0, .pci_subdevice = 0x2017, + .config = AD1986A_LAPTOP }, /* Samsung M50 */ + { .pci_subvendor = 0x1043, .pci_subdevice = 0x818f, + .config = AD1986A_LAPTOP }, /* ASUS P5GV-MX */ + { .modelname = "laptop-eapd", .config = AD1986A_LAPTOP_EAPD }, + { .pci_subvendor = 0x144d, .pci_subdevice = 0xc024, + .config = AD1986A_LAPTOP_EAPD }, /* Samsung R65-T2300 Charis */ + { .pci_subvendor = 0x1043, .pci_subdevice = 0x1213, + .config = AD1986A_LAPTOP_EAPD }, /* ASUS A6J */ + {} +}; static int patch_ad1986a(struct hda_codec *codec) { struct ad198x_spec *spec; + int board_config; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) return -ENOMEM; - init_MUTEX(&spec->amp_mutex); + mutex_init(&spec->amp_mutex); codec->spec = spec; spec->multiout.max_channels = 6; @@ -553,7 +821,7 @@ static int patch_ad1986a(struct hda_codec *codec) spec->multiout.dig_out_nid = AD1986A_SPDIF_OUT; spec->num_adc_nids = 1; spec->adc_nids = ad1986a_adc_nids; - spec->capsrc_nids = ad1986a_adc_nids; + spec->capsrc_nids = ad1986a_capsrc_nids; spec->input_mux = &ad1986a_capture_source; spec->num_mixers = 1; spec->mixers[0] = ad1986a_mixers; @@ -562,6 +830,35 @@ static int patch_ad1986a(struct hda_codec *codec) codec->patch_ops = ad198x_patch_ops; + /* override some parameters */ + board_config = snd_hda_check_board_config(codec, ad1986a_cfg_tbl); + switch (board_config) { + case AD1986A_3STACK: + spec->num_mixers = 2; + spec->mixers[1] = ad1986a_3st_mixers; + spec->num_init_verbs = 2; + spec->init_verbs[1] = ad1986a_3st_init_verbs; + spec->channel_mode = ad1986a_modes; + spec->num_channel_mode = ARRAY_SIZE(ad1986a_modes); + break; + case AD1986A_LAPTOP: + spec->mixers[0] = ad1986a_laptop_mixers; + spec->multiout.max_channels = 2; + spec->multiout.num_dacs = 1; + spec->multiout.dac_nids = ad1986a_laptop_dac_nids; + break; + case AD1986A_LAPTOP_EAPD: + spec->mixers[0] = ad1986a_laptop_eapd_mixers; + spec->num_init_verbs = 2; + spec->init_verbs[1] = ad1986a_eapd_init_verbs; + spec->multiout.max_channels = 2; + spec->multiout.num_dacs = 1; + spec->multiout.dac_nids = ad1986a_laptop_dac_nids; + spec->multiout.dig_out_nid = 0; + spec->input_mux = &ad1986a_laptop_eapd_capture_source; + break; + } + return 0; } @@ -575,6 +872,7 @@ static int patch_ad1986a(struct hda_codec *codec) static hda_nid_t ad1983_dac_nids[1] = { AD1983_DAC }; static hda_nid_t ad1983_adc_nids[1] = { AD1983_ADC }; +static hda_nid_t ad1983_capsrc_nids[1] = { 0x15 }; static struct hda_input_mux ad1983_capture_source = { .num_items = 4, @@ -708,7 +1006,7 @@ static int patch_ad1983(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; - init_MUTEX(&spec->amp_mutex); + mutex_init(&spec->amp_mutex); codec->spec = spec; spec->multiout.max_channels = 2; @@ -717,7 +1015,7 @@ static int patch_ad1983(struct hda_codec *codec) spec->multiout.dig_out_nid = AD1983_SPDIF_OUT; spec->num_adc_nids = 1; spec->adc_nids = ad1983_adc_nids; - spec->capsrc_nids = ad1983_adc_nids; + spec->capsrc_nids = ad1983_capsrc_nids; spec->input_mux = &ad1983_capture_source; spec->num_mixers = 1; spec->mixers[0] = ad1983_mixers; @@ -741,6 +1039,7 @@ static int patch_ad1983(struct hda_codec *codec) static hda_nid_t ad1981_dac_nids[1] = { AD1981_DAC }; static hda_nid_t ad1981_adc_nids[1] = { AD1981_ADC }; +static hda_nid_t ad1981_capsrc_nids[1] = { 0x15 }; /* 0x0c, 0x09, 0x0e, 0x0f, 0x19, 0x05, 0x18, 0x17 */ static struct hda_input_mux ad1981_capture_source = { @@ -846,15 +1145,200 @@ static struct hda_verb ad1981_init_verbs[] = { { } /* end */ }; +/* + * Patch for HP nx6320 + * + * nx6320 uses EAPD in the reserve way - EAPD-on means the internal + * speaker output enabled _and_ mute-LED off. + */ + +#define AD1981_HP_EVENT 0x37 +#define AD1981_MIC_EVENT 0x38 + +static struct hda_verb ad1981_hp_init_verbs[] = { + {0x05, AC_VERB_SET_EAPD_BTLENABLE, 0x00 }, /* default off */ + /* pin sensing on HP and Mic jacks */ + {0x06, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_HP_EVENT}, + {0x08, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_MIC_EVENT}, + {} +}; + +/* turn on/off EAPD (+ mute HP) as a master switch */ +static int ad1981_hp_master_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ad198x_spec *spec = codec->spec; + + if (! ad198x_eapd_put(kcontrol, ucontrol)) + return 0; + + /* toggle HP mute appropriately */ + snd_hda_codec_amp_update(codec, 0x06, 0, HDA_OUTPUT, 0, + 0x80, spec->cur_eapd ? 0 : 0x80); + snd_hda_codec_amp_update(codec, 0x06, 1, HDA_OUTPUT, 0, + 0x80, spec->cur_eapd ? 0 : 0x80); + return 1; +} + +/* bind volumes of both NID 0x05 and 0x06 */ +static int ad1981_hp_master_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + long *valp = ucontrol->value.integer.value; + int change; + + change = snd_hda_codec_amp_update(codec, 0x05, 0, HDA_OUTPUT, 0, + 0x7f, valp[0] & 0x7f); + change |= snd_hda_codec_amp_update(codec, 0x05, 1, HDA_OUTPUT, 0, + 0x7f, valp[1] & 0x7f); + snd_hda_codec_amp_update(codec, 0x06, 0, HDA_OUTPUT, 0, + 0x7f, valp[0] & 0x7f); + snd_hda_codec_amp_update(codec, 0x06, 1, HDA_OUTPUT, 0, + 0x7f, valp[1] & 0x7f); + return change; +} + +/* mute internal speaker if HP is plugged */ +static void ad1981_hp_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x06, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_update(codec, 0x05, 0, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); + snd_hda_codec_amp_update(codec, 0x05, 1, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); +} + +/* toggle input of built-in and mic jack appropriately */ +static void ad1981_hp_automic(struct hda_codec *codec) +{ + static struct hda_verb mic_jack_on[] = { + {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + {} + }; + static struct hda_verb mic_jack_off[] = { + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + {} + }; + unsigned int present; + + present = snd_hda_codec_read(codec, 0x08, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + if (present) + snd_hda_sequence_write(codec, mic_jack_on); + else + snd_hda_sequence_write(codec, mic_jack_off); +} + +/* unsolicited event for HP jack sensing */ +static void ad1981_hp_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + res >>= 26; + switch (res) { + case AD1981_HP_EVENT: + ad1981_hp_automute(codec); + break; + case AD1981_MIC_EVENT: + ad1981_hp_automic(codec); + break; + } +} + +static struct hda_input_mux ad1981_hp_capture_source = { + .num_items = 3, + .items = { + { "Mic", 0x0 }, + { "Docking-Station", 0x1 }, + { "Mix", 0x2 }, + }, +}; + +static struct snd_kcontrol_new ad1981_hp_mixers[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Volume", + .info = snd_hda_mixer_amp_volume_info, + .get = snd_hda_mixer_amp_volume_get, + .put = ad1981_hp_master_vol_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x05, 3, 0, HDA_OUTPUT), + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = ad198x_eapd_info, + .get = ad198x_eapd_get, + .put = ad1981_hp_master_sw_put, + .private_value = 0x05, + }, + HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT), +#if 0 + /* FIXME: analog mic/line loopback doesn't work with my tests... + * (although recording is OK) + */ + HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Docking-Station Playback Volume", 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Docking-Station Playback Switch", 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x1c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x1c, 0x0, HDA_OUTPUT), + /* FIXME: does this laptop have analog CD connection? */ + HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT), +#endif + HDA_CODEC_VOLUME("Mic Boost", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost", 0x18, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + { } /* end */ +}; + +/* initialize jack-sensing, too */ +static int ad1981_hp_init(struct hda_codec *codec) +{ + ad198x_init(codec); + ad1981_hp_automute(codec); + ad1981_hp_automic(codec); + return 0; +} + +/* models */ +enum { AD1981_BASIC, AD1981_HP }; + +static struct hda_board_config ad1981_cfg_tbl[] = { + { .modelname = "hp", .config = AD1981_HP }, + { .pci_subvendor = 0x103c, .pci_subdevice = 0x30aa, + .config = AD1981_HP }, /* HP nx6320 */ + { .pci_subvendor = 0x103c, .pci_subdevice = 0x309f, + .config = AD1981_HP }, /* HP nx9420 AngelFire */ + { .modelname = "basic", .config = AD1981_BASIC }, + {} +}; + static int patch_ad1981(struct hda_codec *codec) { struct ad198x_spec *spec; + int board_config; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) return -ENOMEM; - init_MUTEX(&spec->amp_mutex); + mutex_init(&spec->amp_mutex); codec->spec = spec; spec->multiout.max_channels = 2; @@ -863,7 +1347,7 @@ static int patch_ad1981(struct hda_codec *codec) spec->multiout.dig_out_nid = AD1981_SPDIF_OUT; spec->num_adc_nids = 1; spec->adc_nids = ad1981_adc_nids; - spec->capsrc_nids = ad1981_adc_nids; + spec->capsrc_nids = ad1981_capsrc_nids; spec->input_mux = &ad1981_capture_source; spec->num_mixers = 1; spec->mixers[0] = ad1981_mixers; @@ -873,6 +1357,21 @@ static int patch_ad1981(struct hda_codec *codec) codec->patch_ops = ad198x_patch_ops; + /* override some parameters */ + board_config = snd_hda_check_board_config(codec, ad1981_cfg_tbl); + switch (board_config) { + case AD1981_HP: + spec->mixers[0] = ad1981_hp_mixers; + spec->num_init_verbs = 2; + spec->init_verbs[1] = ad1981_hp_init_verbs; + spec->multiout.dig_out_nid = 0; + spec->input_mux = &ad1981_hp_capture_source; + + codec->patch_ops.init = ad1981_hp_init; + codec->patch_ops.unsol_event = ad1981_hp_unsol_event; + break; + } + return 0; } @@ -1060,44 +1559,6 @@ static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol, spec->num_channel_mode, &spec->multiout.max_channels); } -/* - * EAPD control - */ -static int ad1988_eapd_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} - -static int ad1988_eapd_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - ucontrol->value.enumerated.item[0] = ! spec->cur_eapd; - return 0; -} - -static int ad1988_eapd_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - unsigned int eapd; - eapd = ! ucontrol->value.enumerated.item[0]; - if (eapd == spec->cur_eapd && ! codec->in_resume) - return 0; - spec->cur_eapd = eapd; - snd_hda_codec_write(codec, 0x12 /* port-D */, - 0, AC_VERB_SET_EAPD_BTLENABLE, - eapd ? 0x02 : 0x00); - return 0; -} - /* 6-stack mode */ static struct snd_kcontrol_new ad1988_6stack_mixers1[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), @@ -1220,9 +1681,10 @@ static struct snd_kcontrol_new ad1988_laptop_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "External Amplifier", - .info = ad1988_eapd_info, - .get = ad1988_eapd_get, - .put = ad1988_eapd_put, + .info = ad198x_eapd_info, + .get = ad198x_eapd_get, + .put = ad198x_eapd_put, + .private_value = 0x12 | (1 << 8), /* port-D, inversed */ }, { } /* end */ @@ -1795,14 +2257,11 @@ static int ad1988_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin, idx = ad1988_pin_idx(pin); nid = ad1988_idx_to_dac(codec, idx); - if (! spec->multiout.dac_nids[0]) { - /* use this as the primary output */ - spec->multiout.dac_nids[0] = nid; - if (! spec->multiout.num_dacs) - spec->multiout.num_dacs = 1; - } else - /* specify the DAC as the extra output */ + /* specify the DAC as the extra output */ + if (! spec->multiout.hp_nid) spec->multiout.hp_nid = nid; + else + spec->multiout.extra_out_nid[0] = nid; /* control HP volume/switch on the output mixer amp */ sprintf(name, "%s Playback Volume", pfx); if ((err = add_control(spec, AD_CTL_WIDGET_VOL, name, @@ -1921,7 +2380,7 @@ static void ad1988_auto_init_extra_out(struct hda_codec *codec) struct ad198x_spec *spec = codec->spec; hda_nid_t pin; - pin = spec->autocfg.speaker_pin; + pin = spec->autocfg.speaker_pins[0]; if (pin) /* connect to front */ ad1988_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); pin = spec->autocfg.hp_pin; @@ -1970,13 +2429,13 @@ static int ad1988_parse_auto_config(struct hda_codec *codec) return err; if ((err = ad1988_auto_fill_dac_nids(codec, &spec->autocfg)) < 0) return err; - if (! spec->autocfg.line_outs && ! spec->autocfg.speaker_pin && - ! spec->autocfg.hp_pin) + if (! spec->autocfg.line_outs) return 0; /* can't find valid BIOS pin config */ if ((err = ad1988_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 || - (err = ad1988_auto_create_extra_out(codec, spec->autocfg.speaker_pin, + (err = ad1988_auto_create_extra_out(codec, + spec->autocfg.speaker_pins[0], "Speaker")) < 0 || - (err = ad1988_auto_create_extra_out(codec, spec->autocfg.speaker_pin, + (err = ad1988_auto_create_extra_out(codec, spec->autocfg.hp_pin, "Headphone")) < 0 || (err = ad1988_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0) return err; @@ -2032,7 +2491,7 @@ static int patch_ad1988(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; - init_MUTEX(&spec->amp_mutex); + mutex_init(&spec->amp_mutex); codec->spec = spec; if (codec->revision_id == AD1988A_REV2) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b767552..4c6c9ec 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6,6 +6,7 @@ * Copyright (c) 2004 Kailang Yang <kailang@realtek.com.tw> * PeiSen Hou <pshou@realtek.com.tw> * Takashi Iwai <tiwai@suse.de> + * Jonathan Woithe <jwoithe@physics.adelaide.edu.au> * * 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 @@ -50,6 +51,7 @@ enum { ALC880_UNIWILL_DIG, ALC880_CLEVO, ALC880_TCL_S700, + ALC880_LG, #ifdef CONFIG_SND_DEBUG ALC880_TEST, #endif @@ -63,6 +65,10 @@ enum { ALC260_HP, ALC260_HP_3013, ALC260_FUJITSU_S702X, + ALC260_ACER, +#ifdef CONFIG_SND_DEBUG + ALC260_TEST, +#endif ALC260_AUTO, ALC260_MODEL_LAST /* last tag */ }; @@ -70,6 +76,7 @@ enum { /* ALC262 models */ enum { ALC262_BASIC, + ALC262_FUJITSU, ALC262_AUTO, ALC262_MODEL_LAST /* last tag */ }; @@ -132,7 +139,7 @@ struct alc_spec { int num_channel_mode; /* PCM information */ - struct hda_pcm pcm_rec[2]; /* used in alc_build_pcms() */ + struct hda_pcm pcm_rec[3]; /* used in alc_build_pcms() */ /* dynamic controls, init_verbs and input_mux */ struct auto_pin_cfg autocfg; @@ -140,6 +147,14 @@ struct alc_spec { struct snd_kcontrol_new *kctl_alloc; struct hda_input_mux private_imux; hda_nid_t private_dac_nids[5]; + + /* hooks */ + void (*init_hook)(struct hda_codec *codec); + void (*unsol_event)(struct hda_codec *codec, unsigned int res); + + /* for pin sensing */ + unsigned int sense_updated: 1; + unsigned int jack_present: 1; }; /* @@ -158,6 +173,8 @@ struct alc_config_preset { unsigned int num_channel_mode; const struct hda_channel_mode *channel_mode; const struct hda_input_mux *input_mux; + void (*unsol_event)(struct hda_codec *, unsigned int); + void (*init_hook)(struct hda_codec *); }; @@ -218,56 +235,231 @@ static int alc_ch_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_va spec->num_channel_mode, &spec->multiout.max_channels); } - /* - * Control of pin widget settings via the mixer. Only boolean settings are - * supported, so VrefEn can't be controlled using these functions as they - * stand. + * Control the mode of pin widget settings via the mixer. "pc" is used + * instead of "%" to avoid consequences of accidently treating the % as + * being part of a format specifier. Maximum allowed length of a value is + * 63 characters plus NULL terminator. + * + * Note: some retasking pin complexes seem to ignore requests for input + * states other than HiZ (eg: PIN_VREFxx) and revert to HiZ if any of these + * are requested. Therefore order this list so that this behaviour will not + * cause problems when mixer clients move through the enum sequentially. + * NIDs 0x0f and 0x10 have been observed to have this behaviour. */ -static int alc_pinctl_switch_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +static char *alc_pin_mode_names[] = { + "Mic 50pc bias", "Mic 80pc bias", + "Line in", "Line out", "Headphone out", +}; +static unsigned char alc_pin_mode_values[] = { + PIN_VREF50, PIN_VREF80, PIN_IN, PIN_OUT, PIN_HP, +}; +/* The control can present all 5 options, or it can limit the options based + * in the pin being assumed to be exclusively an input or an output pin. + */ +#define ALC_PIN_DIR_IN 0x00 +#define ALC_PIN_DIR_OUT 0x01 +#define ALC_PIN_DIR_INOUT 0x02 + +/* Info about the pin modes supported by the three different pin directions. + * For each direction the minimum and maximum values are given. + */ +static signed char alc_pin_mode_dir_info[3][2] = { + { 0, 2 }, /* ALC_PIN_DIR_IN */ + { 3, 4 }, /* ALC_PIN_DIR_OUT */ + { 0, 4 }, /* ALC_PIN_DIR_INOUT */ +}; +#define alc_pin_mode_min(_dir) (alc_pin_mode_dir_info[_dir][0]) +#define alc_pin_mode_max(_dir) (alc_pin_mode_dir_info[_dir][1]) +#define alc_pin_mode_n_items(_dir) \ + (alc_pin_mode_max(_dir)-alc_pin_mode_min(_dir)+1) + +static int alc_pin_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + unsigned int item_num = uinfo->value.enumerated.item; + unsigned char dir = (kcontrol->private_value >> 16) & 0xff; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; + uinfo->value.enumerated.items = alc_pin_mode_n_items(dir); + + if (item_num<alc_pin_mode_min(dir) || item_num>alc_pin_mode_max(dir)) + item_num = alc_pin_mode_min(dir); + strcpy(uinfo->value.enumerated.name, alc_pin_mode_names[item_num]); return 0; } -static int alc_pinctl_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int alc_pin_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + unsigned int i; struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = kcontrol->private_value & 0xffff; - long mask = (kcontrol->private_value >> 16) & 0xff; + unsigned char dir = (kcontrol->private_value >> 16) & 0xff; long *valp = ucontrol->value.integer.value; + unsigned int pinctl = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_PIN_WIDGET_CONTROL,0x00); - *valp = 0; - if (snd_hda_codec_read(codec,nid,0,AC_VERB_GET_PIN_WIDGET_CONTROL,0x00) & mask) - *valp = 1; + /* Find enumerated value for current pinctl setting */ + i = alc_pin_mode_min(dir); + while (alc_pin_mode_values[i]!=pinctl && i<=alc_pin_mode_max(dir)) + i++; + *valp = i<=alc_pin_mode_max(dir)?i:alc_pin_mode_min(dir); return 0; } -static int alc_pinctl_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int alc_pin_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + signed int change; struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = kcontrol->private_value & 0xffff; - long mask = (kcontrol->private_value >> 16) & 0xff; - long *valp = ucontrol->value.integer.value; + unsigned char dir = (kcontrol->private_value >> 16) & 0xff; + long val = *ucontrol->value.integer.value; unsigned int pinctl = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_PIN_WIDGET_CONTROL,0x00); - int change = ((pinctl & mask)!=0) != *valp; - if (change) + if (val<alc_pin_mode_min(dir) || val>alc_pin_mode_max(dir)) + val = alc_pin_mode_min(dir); + + change = pinctl != alc_pin_mode_values[val]; + if (change) { + /* Set pin mode to that requested */ snd_hda_codec_write(codec,nid,0,AC_VERB_SET_PIN_WIDGET_CONTROL, - *valp?(pinctl|mask):(pinctl&~mask)); + alc_pin_mode_values[val]); + + /* Also enable the retasking pin's input/output as required + * for the requested pin mode. Enum values of 2 or less are + * input modes. + * + * Dynamically switching the input/output buffers probably + * reduces noise slightly, particularly on input. However, + * havingboth input and output buffers enabled + * simultaneously doesn't seem to be problematic. + */ + if (val <= 2) { + snd_hda_codec_write(codec,nid,0,AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_MUTE); + snd_hda_codec_write(codec,nid,0,AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_UNMUTE(0)); + } else { + snd_hda_codec_write(codec,nid,0,AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_MUTE(0)); + snd_hda_codec_write(codec,nid,0,AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_UNMUTE); + } + } return change; } -#define ALC_PINCTL_SWITCH(xname, nid, mask) \ +#define ALC_PIN_MODE(xname, nid, dir) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ - .info = alc_pinctl_switch_info, \ - .get = alc_pinctl_switch_get, \ - .put = alc_pinctl_switch_put, \ - .private_value = (nid) | (mask<<16) } + .info = alc_pin_mode_info, \ + .get = alc_pin_mode_get, \ + .put = alc_pin_mode_put, \ + .private_value = nid | (dir<<16) } + +/* A switch control for ALC260 GPIO pins. Multiple GPIOs can be ganged + * together using a mask with more than one bit set. This control is + * currently used only by the ALC260 test model. At this stage they are not + * needed for any "production" models. + */ +#ifdef CONFIG_SND_DEBUG +static int alc_gpio_data_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} +static int alc_gpio_data_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value & 0xffff; + unsigned char mask = (kcontrol->private_value >> 16) & 0xff; + long *valp = ucontrol->value.integer.value; + unsigned int val = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_GPIO_DATA,0x00); + *valp = (val & mask) != 0; + return 0; +} +static int alc_gpio_data_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + signed int change; + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value & 0xffff; + unsigned char mask = (kcontrol->private_value >> 16) & 0xff; + long val = *ucontrol->value.integer.value; + unsigned int gpio_data = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_GPIO_DATA,0x00); + + /* Set/unset the masked GPIO bit(s) as needed */ + change = (val==0?0:mask) != (gpio_data & mask); + if (val==0) + gpio_data &= ~mask; + else + gpio_data |= mask; + snd_hda_codec_write(codec,nid,0,AC_VERB_SET_GPIO_DATA,gpio_data); + + return change; +} +#define ALC_GPIO_DATA_SWITCH(xname, nid, mask) \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ + .info = alc_gpio_data_info, \ + .get = alc_gpio_data_get, \ + .put = alc_gpio_data_put, \ + .private_value = nid | (mask<<16) } +#endif /* CONFIG_SND_DEBUG */ + +/* A switch control to allow the enabling of the digital IO pins on the + * ALC260. This is incredibly simplistic; the intention of this control is + * to provide something in the test model allowing digital outputs to be + * identified if present. If models are found which can utilise these + * outputs a more complete mixer control can be devised for those models if + * necessary. + */ +#ifdef CONFIG_SND_DEBUG +static int alc_spdif_ctrl_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} +static int alc_spdif_ctrl_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value & 0xffff; + unsigned char mask = (kcontrol->private_value >> 16) & 0xff; + long *valp = ucontrol->value.integer.value; + unsigned int val = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_DIGI_CONVERT,0x00); + + *valp = (val & mask) != 0; + return 0; +} +static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + signed int change; + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value & 0xffff; + unsigned char mask = (kcontrol->private_value >> 16) & 0xff; + long val = *ucontrol->value.integer.value; + unsigned int ctrl_data = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_DIGI_CONVERT,0x00); + + /* Set/unset the masked control bit(s) as needed */ + change = (val==0?0:mask) != (ctrl_data & mask); + if (val==0) + ctrl_data &= ~mask; + else + ctrl_data |= mask; + snd_hda_codec_write(codec,nid,0,AC_VERB_SET_DIGI_CONVERT_1,ctrl_data); + + return change; +} +#define ALC_SPDIF_CTRL_SWITCH(xname, nid, mask) \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ + .info = alc_spdif_ctrl_info, \ + .get = alc_spdif_ctrl_get, \ + .put = alc_spdif_ctrl_put, \ + .private_value = nid | (mask<<16) } +#endif /* CONFIG_SND_DEBUG */ /* * set up from the preset table @@ -296,6 +488,9 @@ static void setup_preset(struct alc_spec *spec, const struct alc_config_preset * spec->num_adc_nids = preset->num_adc_nids; spec->adc_nids = preset->adc_nids; spec->dig_in_nid = preset->dig_in_nid; + + spec->unsol_event = preset->unsol_event; + spec->init_hook = preset->init_hook; } /* @@ -1098,6 +1293,141 @@ static struct hda_verb alc880_pin_tcl_S700_init_verbs[] = { }; /* + * LG m1 express dual + * + * Pin assignment: + * Rear Line-In/Out (blue): 0x14 + * Build-in Mic-In: 0x15 + * Speaker-out: 0x17 + * HP-Out (green): 0x1b + * Mic-In/Out (red): 0x19 + * SPDIF-Out: 0x1e + */ + +/* To make 5.1 output working (green=Front, blue=Surr, red=CLFE) */ +static hda_nid_t alc880_lg_dac_nids[3] = { + 0x05, 0x02, 0x03 +}; + +/* seems analog CD is not working */ +static struct hda_input_mux alc880_lg_capture_source = { + .num_items = 3, + .items = { + { "Mic", 0x1 }, + { "Line", 0x5 }, + { "Internal Mic", 0x6 }, + }, +}; + +/* 2,4,6 channel modes */ +static struct hda_verb alc880_lg_ch2_init[] = { + /* set line-in and mic-in to input */ + { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { } +}; + +static struct hda_verb alc880_lg_ch4_init[] = { + /* set line-in to out and mic-in to input */ + { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { } +}; + +static struct hda_verb alc880_lg_ch6_init[] = { + /* set line-in and mic-in to output */ + { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, + { } +}; + +static struct hda_channel_mode alc880_lg_ch_modes[3] = { + { 2, alc880_lg_ch2_init }, + { 4, alc880_lg_ch4_init }, + { 6, alc880_lg_ch6_init }, +}; + +static struct snd_kcontrol_new alc880_lg_mixer[] = { + /* FIXME: it's not really "master" but front channels */ + HDA_CODEC_VOLUME("Master Playback Volume", 0x0f, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Master Playback Switch", 0x0f, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x06, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x06, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x07, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x07, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Channel Mode", + .info = alc_ch_mode_info, + .get = alc_ch_mode_get, + .put = alc_ch_mode_put, + }, + { } /* end */ +}; + +static struct hda_verb alc880_lg_init_verbs[] = { + /* set capture source to mic-in */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* mute all amp mixer inputs */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(6)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(7)}, + /* line-in to input */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* built-in mic */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* speaker-out */ + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* mic-in to input */ + {0x11, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* HP-out */ + {0x13, AC_VERB_SET_CONNECT_SEL, 0x03}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* jack sense */ + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | 0x1}, + { } +}; + +/* toggle speaker-output according to the hp-jack state */ +static void alc880_lg_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_update(codec, 0x17, 0, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); + snd_hda_codec_amp_update(codec, 0x17, 1, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); +} + +static void alc880_lg_unsol_event(struct hda_codec *codec, unsigned int res) +{ + /* Looks like the unsol event is incompatible with the standard + * definition. 4bit tag is placed at 28 bit! + */ + if ((res >> 28) == 0x01) + alc880_lg_automute(codec); +} + +/* + * Common callbacks */ static int alc_init(struct hda_codec *codec) @@ -1107,9 +1437,21 @@ static int alc_init(struct hda_codec *codec) for (i = 0; i < spec->num_init_verbs; i++) snd_hda_sequence_write(codec, spec->init_verbs[i]); + + if (spec->init_hook) + spec->init_hook(codec); + return 0; } +static void alc_unsol_event(struct hda_codec *codec, unsigned int res) +{ + struct alc_spec *spec = codec->spec; + + if (spec->unsol_event) + spec->unsol_event(codec, res); +} + #ifdef CONFIG_PM /* * resume @@ -1250,6 +1592,13 @@ static struct hda_pcm_stream alc880_pcm_digital_capture = { /* NID is set in alc_build_pcms */ }; +/* Used by alc_build_pcms to flag that a PCM has no playback stream */ +static struct hda_pcm_stream alc_pcm_null_playback = { + .substreams = 0, + .channels_min = 0, + .channels_max = 0, +}; + static int alc_build_pcms(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -1280,6 +1629,23 @@ static int alc_build_pcms(struct hda_codec *codec) } } + /* If the use of more than one ADC is requested for the current + * model, configure a second analog capture-only PCM. + */ + if (spec->num_adc_nids > 1) { + codec->num_pcms++; + info++; + info->name = spec->stream_name_analog; + /* No playback stream for second PCM */ + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = alc_pcm_null_playback; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = 0; + if (spec->stream_analog_capture) { + snd_assert(spec->adc_nids, return -EINVAL); + info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture); + info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[1]; + } + } + if (spec->multiout.dig_out_nid || spec->dig_in_nid) { codec->num_pcms++; info++; @@ -1322,6 +1688,7 @@ static struct hda_codec_ops alc_patch_ops = { .build_pcms = alc_build_pcms, .init = alc_init, .free = alc_free, + .unsol_event = alc_unsol_event, #ifdef CONFIG_PM .resume = alc_resume, #endif @@ -1340,13 +1707,15 @@ static hda_nid_t alc880_test_dac_nids[4] = { }; static struct hda_input_mux alc880_test_capture_source = { - .num_items = 5, + .num_items = 7, .items = { { "In-1", 0x0 }, { "In-2", 0x1 }, { "In-3", 0x2 }, { "In-4", 0x3 }, { "CD", 0x4 }, + { "Front", 0x5 }, + { "Surround", 0x6 }, }, }; @@ -1653,6 +2022,8 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .pci_subvendor = 0x8086, .pci_subdevice = 0xa100, .config = ALC880_5ST_DIG }, { .pci_subvendor = 0x1565, .pci_subdevice = 0x8202, .config = ALC880_5ST_DIG }, { .pci_subvendor = 0x1019, .pci_subdevice = 0xa880, .config = ALC880_5ST_DIG }, + { .pci_subvendor = 0xa0a0, .pci_subdevice = 0x0560, + .config = ALC880_5ST_DIG }, /* Aopen i915GMm-HFS */ /* { .pci_subvendor = 0x1019, .pci_subdevice = 0xa884, .config = ALC880_5ST_DIG }, */ /* conflict with 6stack */ { .pci_subvendor = 0x1695, .pci_subdevice = 0x400d, .config = ALC880_5ST_DIG }, /* note subvendor = 0 below */ @@ -1680,6 +2051,7 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .pci_subvendor = 0x1025, .pci_subdevice = 0x0078, .config = ALC880_6ST_DIG }, { .pci_subvendor = 0x1025, .pci_subdevice = 0x0087, .config = ALC880_6ST_DIG }, { .pci_subvendor = 0x1297, .pci_subdevice = 0xc790, .config = ALC880_6ST_DIG }, /* Shuttle ST20G5 */ + { .pci_subvendor = 0x1509, .pci_subdevice = 0x925d, .config = ALC880_6ST_DIG }, /* FIC P4M-915GD1 */ { .modelname = "asus", .config = ALC880_ASUS }, { .pci_subvendor = 0x1043, .pci_subdevice = 0x1964, .config = ALC880_ASUS_DIG }, @@ -1693,6 +2065,7 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .pci_subvendor = 0x1043, .pci_subdevice = 0x1123, .config = ALC880_ASUS_DIG }, { .pci_subvendor = 0x1043, .pci_subdevice = 0x1143, .config = ALC880_ASUS }, { .pci_subvendor = 0x1043, .pci_subdevice = 0x10b3, .config = ALC880_ASUS_W1V }, + { .pci_subvendor = 0x1043, .pci_subdevice = 0x8181, .config = ALC880_ASUS_DIG }, /* ASUS P4GPL-X */ { .pci_subvendor = 0x1558, .pci_subdevice = 0x5401, .config = ALC880_ASUS_DIG2 }, { .modelname = "uniwill", .config = ALC880_UNIWILL_DIG }, @@ -1702,6 +2075,9 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .pci_subvendor = 0x1734, .pci_subdevice = 0x107c, .config = ALC880_F1734 }, { .pci_subvendor = 0x1584, .pci_subdevice = 0x9054, .config = ALC880_F1734 }, + { .modelname = "lg", .config = ALC880_LG }, + { .pci_subvendor = 0x1854, .pci_subdevice = 0x003b, .config = ALC880_LG }, + #ifdef CONFIG_SND_DEBUG { .modelname = "test", .config = ALC880_TEST }, #endif @@ -1879,6 +2255,19 @@ static struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_threestack_modes, .input_mux = &alc880_capture_source, }, + [ALC880_LG] = { + .mixers = { alc880_lg_mixer }, + .init_verbs = { alc880_volume_init_verbs, + alc880_lg_init_verbs }, + .num_dacs = ARRAY_SIZE(alc880_lg_dac_nids), + .dac_nids = alc880_lg_dac_nids, + .dig_out_nid = ALC880_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc880_lg_ch_modes), + .channel_mode = alc880_lg_ch_modes, + .input_mux = &alc880_lg_capture_source, + .unsol_event = alc880_lg_unsol_event, + .init_hook = alc880_lg_automute, + }, #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = { .mixers = { alc880_test_mixer }, @@ -2043,14 +2432,11 @@ static int alc880_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin, if (alc880_is_fixed_pin(pin)) { nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin)); - if (! spec->multiout.dac_nids[0]) { - /* use this as the primary output */ - spec->multiout.dac_nids[0] = nid; - if (! spec->multiout.num_dacs) - spec->multiout.num_dacs = 1; - } else - /* specify the DAC as the extra output */ + /* specify the DAC as the extra output */ + if (! spec->multiout.hp_nid) spec->multiout.hp_nid = nid; + else + spec->multiout.extra_out_nid[0] = nid; /* control HP volume/switch on the output mixer amp */ nid = alc880_idx_to_mixer(alc880_fixed_pin_idx(pin)); sprintf(name, "%s Playback Volume", pfx); @@ -2063,12 +2449,6 @@ static int alc880_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin, return err; } else if (alc880_is_multi_pin(pin)) { /* set manual connection */ - if (! spec->multiout.dac_nids[0]) { - /* use this as the primary output */ - spec->multiout.dac_nids[0] = alc880_idx_to_dac(alc880_multi_pin_idx(pin)); - if (! spec->multiout.num_dacs) - spec->multiout.num_dacs = 1; - } /* we have only a switch on HP-out PIN */ sprintf(name, "%s Playback Switch", pfx); if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, @@ -2152,7 +2532,7 @@ static void alc880_auto_init_extra_out(struct hda_codec *codec) struct alc_spec *spec = codec->spec; hda_nid_t pin; - pin = spec->autocfg.speaker_pin; + pin = spec->autocfg.speaker_pins[0]; if (pin) /* connect to front */ alc880_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); pin = spec->autocfg.hp_pin; @@ -2188,15 +2568,15 @@ static int alc880_parse_auto_config(struct hda_codec *codec) if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, alc880_ignore)) < 0) return err; - if (! spec->autocfg.line_outs && ! spec->autocfg.speaker_pin && - ! spec->autocfg.hp_pin) + if (! spec->autocfg.line_outs) return 0; /* can't find valid BIOS pin config */ if ((err = alc880_auto_fill_dac_nids(spec, &spec->autocfg)) < 0 || (err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 || - (err = alc880_auto_create_extra_out(spec, spec->autocfg.speaker_pin, + (err = alc880_auto_create_extra_out(spec, + spec->autocfg.speaker_pins[0], "Speaker")) < 0 || - (err = alc880_auto_create_extra_out(spec, spec->autocfg.speaker_pin, + (err = alc880_auto_create_extra_out(spec, spec->autocfg.hp_pin, "Headphone")) < 0 || (err = alc880_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0) return err; @@ -2218,14 +2598,12 @@ static int alc880_parse_auto_config(struct hda_codec *codec) return 1; } -/* init callback for auto-configuration model -- overriding the default init */ -static int alc880_auto_init(struct hda_codec *codec) +/* additional initialization for auto-configuration model */ +static void alc880_auto_init(struct hda_codec *codec) { - alc_init(codec); alc880_auto_init_multi_out(codec); alc880_auto_init_extra_out(codec); alc880_auto_init_analog_input(codec); - return 0; } /* @@ -2292,7 +2670,7 @@ static int patch_alc880(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC880_AUTO) - codec->patch_ops.init = alc880_auto_init; + spec->init_hook = alc880_auto_init; return 0; } @@ -2322,6 +2700,14 @@ static hda_nid_t alc260_hp_adc_nids[2] = { 0x05, 0x04 }; +/* NIDs used when simultaneous access to both ADCs makes sense. Note that + * alc260_capture_mixer assumes ADC0 (nid 0x04) is the first ADC. + */ +static hda_nid_t alc260_dual_adc_nids[2] = { + /* ADC0, ADC1 */ + 0x04, 0x05 +}; + #define ALC260_DIGOUT_NID 0x03 #define ALC260_DIGIN_NID 0x06 @@ -2335,14 +2721,28 @@ static struct hda_input_mux alc260_capture_source = { }, }; -/* On Fujitsu S702x laptops capture only makes sense from Mic/LineIn jack - * and the internal CD lines. +/* On Fujitsu S702x laptops capture only makes sense from Mic/LineIn jack, + * headphone jack and the internal CD lines. */ static struct hda_input_mux alc260_fujitsu_capture_source = { - .num_items = 2, + .num_items = 3, .items = { { "Mic/Line", 0x0 }, { "CD", 0x4 }, + { "Headphone", 0x2 }, + }, +}; + +/* Acer TravelMate(/Extensa/Aspire) notebooks have similar configutation to + * the Fujitsu S702x, but jacks are marked differently. We won't allow + * retasking the Headphone jack, so it won't be available here. + */ +static struct hda_input_mux alc260_acer_capture_source = { + .num_items = 3, + .items = { + { "Mic", 0x0 }, + { "Line", 0x2 }, + { "CD", 0x4 }, }, }; @@ -2363,6 +2763,7 @@ static struct hda_channel_mode alc260_modes[1] = { * HP: base_output + input + capture_alt * HP_3013: hp_3013 + input + capture * fujitsu: fujitsu + capture + * acer: acer + capture */ static struct snd_kcontrol_new alc260_base_output_mixer[] = { @@ -2408,11 +2809,12 @@ static struct snd_kcontrol_new alc260_hp_3013_mixer[] = { static struct snd_kcontrol_new alc260_fujitsu_mixer[] = { HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Headphone Playback Switch", 0x08, 2, HDA_INPUT), - ALC_PINCTL_SWITCH("Headphone Amp Switch", 0x14, PIN_HP_AMP), + ALC_PIN_MODE("Headphone Jack Mode", 0x14, ALC_PIN_DIR_INOUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Mic/Line Playback Volume", 0x07, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic/Line Playback Switch", 0x07, 0x0, HDA_INPUT), + ALC_PIN_MODE("Mic/Line Jack Mode", 0x12, ALC_PIN_DIR_IN), HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT), HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT), HDA_CODEC_VOLUME("Internal Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT), @@ -2420,6 +2822,22 @@ static struct snd_kcontrol_new alc260_fujitsu_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc260_acer_mixer[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT), + ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN), + HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT), + ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT), + HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT), + { } /* end */ +}; + /* capture mixer elements */ static struct snd_kcontrol_new alc260_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x04, 0x0, HDA_INPUT), @@ -2629,52 +3047,327 @@ static struct hda_verb alc260_fujitsu_init_verbs[] = { {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, /* Headphone/Line-out jack connects to Line1 pin; make it an output */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* Mic/Line-in jack is connected to mic1 pin, so make it an input */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - /* Ensure all other unused pins are disabled and muted. - * Note: trying to set widget 0x15 to anything blocks all audio - * output for some reason, so just leave that at the default. + /* Mic/Line-in jack is connected to mic1 pin, so make it an input */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + /* Ensure all other unused pins are disabled and muted. */ + {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + + /* Disable digital (SPDIF) pins */ + {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, + {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, + + /* Ensure Line1 pin widget takes its input from the OUT1 sum bus + * when acting as an output. + */ + {0x0d, AC_VERB_SET_CONNECT_SEL, 0}, + + /* Start with output sum widgets muted and their output gains at min */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + + /* Unmute HP pin widget amp left and right (no equiv mixer ctrl) */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Unmute Line1 pin widget output buffer since it starts as an output. + * If the pin mode is changed by the user the pin mode control will + * take care of enabling the pin's input/output buffers as needed. + * Therefore there's no need to enable the input buffer at this + * stage. + */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Unmute input buffer of pin widget used for Line-in (no equiv + * mixer ctrl) + */ + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + /* Mute capture amp left and right */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + /* Set ADC connection select to match default mixer setting - line + * in (on mic1 pin) */ - {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Do the same for the second ADC: mute capture input amp and + * set ADC connection to line in (on mic1 pin) + */ + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x05, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Mute all inputs to mixer widget (even unconnected ones) */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */ + + { } +}; + +/* Initialisation sequence for ALC260 as configured in Acer TravelMate and + * similar laptops (adapted from Fujitsu init verbs). + */ +static struct hda_verb alc260_acer_init_verbs[] = { + /* On TravelMate laptops, GPIO 0 enables the internal speaker and + * the headphone jack. Turn this on and rely on the standard mute + * methods whenever the user wants to turn these outputs off. + */ + {0x01, AC_VERB_SET_GPIO_MASK, 0x01}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x01}, + /* Internal speaker/Headphone jack is connected to Line-out pin */ + {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Internal microphone/Mic jack is connected to Mic1 pin */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50}, + /* Line In jack is connected to Line1 pin */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + /* Ensure all other unused pins are disabled and muted. */ + {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - /* Disable digital (SPDIF) pins */ - {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, - {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, - - /* Start with mixer outputs muted */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - - /* Unmute HP pin widget amp left and right (no equiv mixer ctrl) */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Unmute Line1 pin widget amp left and right (no equiv mixer ctrl) */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Unmute pin widget used for Line-in (no equiv mixer ctrl) */ - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - - /* Mute capture amp left and right */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - /* Set ADC connection select to line in (on mic1 pin) */ - {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* Mute all inputs to mixer widget (even unconnected ones) */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */ + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + /* Disable digital (SPDIF) pins */ + {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, + {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, + + /* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum + * bus when acting as outputs. + */ + {0x0b, AC_VERB_SET_CONNECT_SEL, 0}, + {0x0d, AC_VERB_SET_CONNECT_SEL, 0}, + + /* Start with output sum widgets muted and their output gains at min */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + + /* Unmute Line-out pin widget amp left and right (no equiv mixer ctrl) */ + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Unmute Mic1 and Line1 pin widget input buffers since they start as + * inputs. If the pin mode is changed by the user the pin mode control + * will take care of enabling the pin's input/output buffers as needed. + * Therefore there's no need to enable the input buffer at this + * stage. + */ + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + /* Mute capture amp left and right */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + /* Set ADC connection select to match default mixer setting - mic + * (on mic1 pin) + */ + {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Do similar with the second ADC: mute capture input amp and + * set ADC connection to line (on line1 pin) + */ + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x05, AC_VERB_SET_CONNECT_SEL, 0x02}, + + /* Mute all inputs to mixer widget (even unconnected ones) */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */ { } }; +/* Test configuration for debugging, modelled after the ALC880 test + * configuration. + */ +#ifdef CONFIG_SND_DEBUG +static hda_nid_t alc260_test_dac_nids[1] = { + 0x02, +}; +static hda_nid_t alc260_test_adc_nids[2] = { + 0x04, 0x05, +}; +/* This is a bit messy since the two input muxes in the ALC260 have slight + * variations in their signal assignments. The ideal way to deal with this + * is to extend alc_spec.input_mux to allow a different input MUX for each + * ADC. For the purposes of the test model it's sufficient to just list + * both options for affected signal indices. The separate input mux + * functionality only needs to be considered if a model comes along which + * actually uses signals 0x5, 0x6 and 0x7 for something which makes sense to + * record. + */ +static struct hda_input_mux alc260_test_capture_source = { + .num_items = 8, + .items = { + { "MIC1 pin", 0x0 }, + { "MIC2 pin", 0x1 }, + { "LINE1 pin", 0x2 }, + { "LINE2 pin", 0x3 }, + { "CD pin", 0x4 }, + { "LINE-OUT pin (cap1), Mixer (cap2)", 0x5 }, + { "HP-OUT pin (cap1), LINE-OUT pin (cap2)", 0x6 }, + { "HP-OUT pin (cap2 only)", 0x7 }, + }, +}; +static struct snd_kcontrol_new alc260_test_mixer[] = { + /* Output driver widgets */ + HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT), + HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x09, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("LOUT2 Playback Switch", 0x09, 2, HDA_INPUT), + HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x08, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("LOUT1 Playback Switch", 0x08, 2, HDA_INPUT), + + /* Modes for retasking pin widgets */ + ALC_PIN_MODE("HP-OUT pin mode", 0x10, ALC_PIN_DIR_INOUT), + ALC_PIN_MODE("LINE-OUT pin mode", 0x0f, ALC_PIN_DIR_INOUT), + ALC_PIN_MODE("LINE2 pin mode", 0x15, ALC_PIN_DIR_INOUT), + ALC_PIN_MODE("LINE1 pin mode", 0x14, ALC_PIN_DIR_INOUT), + ALC_PIN_MODE("MIC2 pin mode", 0x13, ALC_PIN_DIR_INOUT), + ALC_PIN_MODE("MIC1 pin mode", 0x12, ALC_PIN_DIR_INOUT), + + /* Loopback mixer controls */ + HDA_CODEC_VOLUME("MIC1 Playback Volume", 0x07, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("MIC1 Playback Switch", 0x07, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("MIC2 Playback Volume", 0x07, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("MIC2 Playback Switch", 0x07, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("LINE1 Playback Volume", 0x07, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("LINE1 Playback Switch", 0x07, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("LINE2 Playback Volume", 0x07, 0x03, HDA_INPUT), + HDA_CODEC_MUTE("LINE2 Playback Switch", 0x07, 0x03, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT), + HDA_CODEC_VOLUME("LINE-OUT loopback Playback Volume", 0x07, 0x06, HDA_INPUT), + HDA_CODEC_MUTE("LINE-OUT loopback Playback Switch", 0x07, 0x06, HDA_INPUT), + HDA_CODEC_VOLUME("HP-OUT loopback Playback Volume", 0x07, 0x7, HDA_INPUT), + HDA_CODEC_MUTE("HP-OUT loopback Playback Switch", 0x07, 0x7, HDA_INPUT), + + /* Controls for GPIO pins, assuming they are configured as outputs */ + ALC_GPIO_DATA_SWITCH("GPIO pin 0", 0x01, 0x01), + ALC_GPIO_DATA_SWITCH("GPIO pin 1", 0x01, 0x02), + ALC_GPIO_DATA_SWITCH("GPIO pin 2", 0x01, 0x04), + ALC_GPIO_DATA_SWITCH("GPIO pin 3", 0x01, 0x08), + + /* Switches to allow the digital IO pins to be enabled. The datasheet + * is ambigious as to which NID is which; testing on laptops which + * make this output available should provide clarification. + */ + ALC_SPDIF_CTRL_SWITCH("SPDIF Playback Switch", 0x03, 0x01), + ALC_SPDIF_CTRL_SWITCH("SPDIF Capture Switch", 0x06, 0x01), + + { } /* end */ +}; +static struct hda_verb alc260_test_init_verbs[] = { + /* Enable all GPIOs as outputs with an initial value of 0 */ + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x0f}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x00}, + {0x01, AC_VERB_SET_GPIO_MASK, 0x0f}, + + /* Enable retasking pins as output, initially without power amp */ + {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + + /* Disable digital (SPDIF) pins initially, but users can enable + * them via a mixer switch. In the case of SPDIF-out, this initverb + * payload also sets the generation to 0, output to be in "consumer" + * PCM format, copyright asserted, no pre-emphasis and no validity + * control. + */ + {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, + {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, + + /* Ensure mic1, mic2, line1 and line2 pin widgets take input from the + * OUT1 sum bus when acting as an output. + */ + {0x0b, AC_VERB_SET_CONNECT_SEL, 0}, + {0x0c, AC_VERB_SET_CONNECT_SEL, 0}, + {0x0d, AC_VERB_SET_CONNECT_SEL, 0}, + {0x0e, AC_VERB_SET_CONNECT_SEL, 0}, + + /* Start with output sum widgets muted and their output gains at min */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + + /* Unmute retasking pin widget output buffers since the default + * state appears to be output. As the pin mode is changed by the + * user the pin mode control will take care of enabling the pin's + * input/output buffers as needed. + */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Also unmute the mono-out pin widget */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* Mute capture amp left and right */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + /* Set ADC connection select to match default mixer setting (mic1 + * pin) + */ + {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Do the same for the second ADC: mute capture input amp and + * set ADC connection to mic1 pin + */ + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x05, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Mute all inputs to mixer widget (even unconnected ones) */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */ + + { } +}; +#endif + static struct hda_pcm_stream alc260_pcm_analog_playback = { .substreams = 1, .channels_min = 2, @@ -2744,7 +3437,7 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec, return err; } - nid = cfg->speaker_pin; + nid = cfg->speaker_pins[0]; if (nid) { err = alc260_add_playback_controls(spec, nid, "Speaker"); if (err < 0) @@ -2817,7 +3510,7 @@ static void alc260_auto_init_multi_out(struct hda_codec *codec) if (nid) alc260_auto_set_output_and_unmute(codec, nid, PIN_OUT, 0); - nid = spec->autocfg.speaker_pin; + nid = spec->autocfg.speaker_pins[0]; if (nid) alc260_auto_set_output_and_unmute(codec, nid, PIN_OUT, 0); @@ -2932,13 +3625,11 @@ static int alc260_parse_auto_config(struct hda_codec *codec) return 1; } -/* init callback for auto-configuration model -- overriding the default init */ -static int alc260_auto_init(struct hda_codec *codec) +/* additional initialization for auto-configuration model */ +static void alc260_auto_init(struct hda_codec *codec) { - alc_init(codec); alc260_auto_init_multi_out(codec); alc260_auto_init_analog_input(codec); - return 0; } /* @@ -2948,6 +3639,8 @@ static struct hda_board_config alc260_cfg_tbl[] = { { .modelname = "basic", .config = ALC260_BASIC }, { .pci_subvendor = 0x104d, .pci_subdevice = 0x81bb, .config = ALC260_BASIC }, /* Sony VAIO */ + { .pci_subvendor = 0x152d, .pci_subdevice = 0x0729, + .config = ALC260_BASIC }, /* CTL Travel Master U553W */ { .modelname = "hp", .config = ALC260_HP }, { .pci_subvendor = 0x103c, .pci_subdevice = 0x3010, .config = ALC260_HP }, { .pci_subvendor = 0x103c, .pci_subdevice = 0x3011, .config = ALC260_HP }, @@ -2958,6 +3651,11 @@ static struct hda_board_config alc260_cfg_tbl[] = { { .pci_subvendor = 0x103c, .pci_subdevice = 0x3016, .config = ALC260_HP }, { .modelname = "fujitsu", .config = ALC260_FUJITSU_S702X }, { .pci_subvendor = 0x10cf, .pci_subdevice = 0x1326, .config = ALC260_FUJITSU_S702X }, + { .modelname = "acer", .config = ALC260_ACER }, + { .pci_subvendor = 0x1025, .pci_subdevice = 0x008f, .config = ALC260_ACER }, +#ifdef CONFIG_SND_DEBUG + { .modelname = "test", .config = ALC260_TEST }, +#endif { .modelname = "auto", .config = ALC260_AUTO }, {} }; @@ -3009,12 +3707,38 @@ static struct alc_config_preset alc260_presets[] = { .init_verbs = { alc260_fujitsu_init_verbs }, .num_dacs = ARRAY_SIZE(alc260_dac_nids), .dac_nids = alc260_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc260_adc_nids), - .adc_nids = alc260_adc_nids, + .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids), + .adc_nids = alc260_dual_adc_nids, .num_channel_mode = ARRAY_SIZE(alc260_modes), .channel_mode = alc260_modes, .input_mux = &alc260_fujitsu_capture_source, }, + [ALC260_ACER] = { + .mixers = { alc260_acer_mixer, + alc260_capture_mixer }, + .init_verbs = { alc260_acer_init_verbs }, + .num_dacs = ARRAY_SIZE(alc260_dac_nids), + .dac_nids = alc260_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids), + .adc_nids = alc260_dual_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc260_modes), + .channel_mode = alc260_modes, + .input_mux = &alc260_acer_capture_source, + }, +#ifdef CONFIG_SND_DEBUG + [ALC260_TEST] = { + .mixers = { alc260_test_mixer, + alc260_capture_mixer }, + .init_verbs = { alc260_test_init_verbs }, + .num_dacs = ARRAY_SIZE(alc260_test_dac_nids), + .dac_nids = alc260_test_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc260_test_adc_nids), + .adc_nids = alc260_test_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc260_modes), + .channel_mode = alc260_modes, + .input_mux = &alc260_test_capture_source, + }, +#endif }; static int patch_alc260(struct hda_codec *codec) @@ -3059,7 +3783,7 @@ static int patch_alc260(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC260_AUTO) - codec->patch_ops.init = alc260_auto_init; + spec->init_hook = alc260_auto_init; return 0; } @@ -3534,14 +4258,12 @@ static int alc882_parse_auto_config(struct hda_codec *codec) return err; } -/* init callback for auto-configuration model -- overriding the default init */ -static int alc882_auto_init(struct hda_codec *codec) +/* additional initialization for auto-configuration model */ +static void alc882_auto_init(struct hda_codec *codec) { - alc_init(codec); alc882_auto_init_multi_out(codec); alc882_auto_init_hp_out(codec); alc882_auto_init_analog_input(codec); - return 0; } /* @@ -3608,7 +4330,7 @@ static int patch_alc882(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC882_AUTO) - codec->patch_ops.init = alc882_auto_init; + spec->init_hook = alc882_auto_init; return 0; } @@ -3644,19 +4366,9 @@ static struct snd_kcontrol_new alc262_base_mixer[] = { HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .count = 1, - .info = alc882_mux_enum_info, - .get = alc882_mux_enum_get, - .put = alc882_mux_enum_put, - }, { } /* end */ -}; - +}; + #define alc262_capture_mixer alc882_capture_mixer #define alc262_capture_alt_mixer alc882_capture_alt_mixer @@ -3739,6 +4451,129 @@ static struct hda_verb alc262_init_verbs[] = { { } }; +/* + * fujitsu model + * 0x14 = headphone/spdif-out, 0x15 = internal speaker + */ + +#define ALC_HP_EVENT 0x37 + +static struct hda_verb alc262_fujitsu_unsol_verbs[] = { + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {} +}; + +static struct hda_input_mux alc262_fujitsu_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x0 }, + { "CD", 0x4 }, + }, +}; + +/* mute/unmute internal speaker according to the hp jack and mute state */ +static void alc262_fujitsu_automute(struct hda_codec *codec, int force) +{ + struct alc_spec *spec = codec->spec; + unsigned int mute; + + if (force || ! spec->sense_updated) { + unsigned int present; + /* need to execute and sync at first */ + snd_hda_codec_read(codec, 0x14, 0, AC_VERB_SET_PIN_SENSE, 0); + present = snd_hda_codec_read(codec, 0x14, 0, + AC_VERB_GET_PIN_SENSE, 0); + spec->jack_present = (present & 0x80000000) != 0; + spec->sense_updated = 1; + } + if (spec->jack_present) { + /* mute internal speaker */ + snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0, + 0x80, 0x80); + snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0, + 0x80, 0x80); + } else { + /* unmute internal speaker if necessary */ + mute = snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0); + snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0, + 0x80, mute & 0x80); + mute = snd_hda_codec_amp_read(codec, 0x14, 1, HDA_OUTPUT, 0); + snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0, + 0x80, mute & 0x80); + } +} + +/* unsolicited event for HP jack sensing */ +static void alc262_fujitsu_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) != ALC_HP_EVENT) + return; + alc262_fujitsu_automute(codec, 1); +} + +/* bind volumes of both NID 0x0c and 0x0d */ +static int alc262_fujitsu_master_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + long *valp = ucontrol->value.integer.value; + int change; + + change = snd_hda_codec_amp_update(codec, 0x0c, 0, HDA_OUTPUT, 0, + 0x7f, valp[0] & 0x7f); + change |= snd_hda_codec_amp_update(codec, 0x0c, 1, HDA_OUTPUT, 0, + 0x7f, valp[1] & 0x7f); + snd_hda_codec_amp_update(codec, 0x0d, 0, HDA_OUTPUT, 0, + 0x7f, valp[0] & 0x7f); + snd_hda_codec_amp_update(codec, 0x0d, 1, HDA_OUTPUT, 0, + 0x7f, valp[1] & 0x7f); + return change; +} + +/* bind hp and internal speaker mute (with plug check) */ +static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + long *valp = ucontrol->value.integer.value; + int change; + + change = snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0, + 0x80, valp[0] ? 0 : 0x80); + change |= snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0, + 0x80, valp[1] ? 0 : 0x80); + if (change || codec->in_resume) + alc262_fujitsu_automute(codec, codec->in_resume); + return change; +} + +static struct snd_kcontrol_new alc262_fujitsu_mixer[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Volume", + .info = snd_hda_mixer_amp_volume_info, + .get = snd_hda_mixer_amp_volume_get, + .put = alc262_fujitsu_master_vol_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT), + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = alc262_fujitsu_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, 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("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + { } /* end */ +}; + /* add playback controls from the parsed DAC table */ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { @@ -3759,7 +4594,7 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, const struct return err; } - nid = cfg->speaker_pin; + nid = cfg->speaker_pins[0]; if (nid) { if (nid == 0x16) { if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, "Speaker Playback Volume", @@ -3769,10 +4604,6 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, const struct HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT))) < 0) return err; } else { - if (! cfg->line_out_pins[0]) - if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, "Speaker Playback Volume", - HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT))) < 0) - return err; if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, "Speaker Playback Switch", HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0) return err; @@ -3789,10 +4620,6 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, const struct HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT))) < 0) return err; } else { - if (! cfg->line_out_pins[0]) - if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, "Headphone Playback Volume", - HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT))) < 0) - return err; if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, "Headphone Playback Switch", HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0) return err; @@ -3886,8 +4713,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec) if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, alc262_ignore)) < 0) return err; - if (! spec->autocfg.line_outs && ! spec->autocfg.speaker_pin && - ! spec->autocfg.hp_pin) + if (! spec->autocfg.line_outs) return 0; /* can't find valid BIOS pin config */ if ((err = alc262_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 || (err = alc262_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0) @@ -3915,13 +4741,11 @@ static int alc262_parse_auto_config(struct hda_codec *codec) /* init callback for auto-configuration model -- overriding the default init */ -static int alc262_auto_init(struct hda_codec *codec) +static void alc262_auto_init(struct hda_codec *codec) { - alc_init(codec); alc262_auto_init_multi_out(codec); alc262_auto_init_hp_out(codec); alc262_auto_init_analog_input(codec); - return 0; } /* @@ -3929,6 +4753,8 @@ static int alc262_auto_init(struct hda_codec *codec) */ static struct hda_board_config alc262_cfg_tbl[] = { { .modelname = "basic", .config = ALC262_BASIC }, + { .modelname = "fujitsu", .config = ALC262_FUJITSU }, + { .pci_subvendor = 0x10cf, .pci_subdevice = 0x1397, .config = ALC262_FUJITSU }, { .modelname = "auto", .config = ALC262_AUTO }, {} }; @@ -3944,6 +4770,18 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, }, + [ALC262_FUJITSU] = { + .mixers = { alc262_fujitsu_mixer }, + .init_verbs = { alc262_init_verbs, alc262_fujitsu_unsol_verbs }, + .num_dacs = ARRAY_SIZE(alc262_dac_nids), + .dac_nids = alc262_dac_nids, + .hp_nid = 0x03, + .dig_out_nid = ALC262_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc262_modes), + .channel_mode = alc262_modes, + .input_mux = &alc262_fujitsu_capture_source, + .unsol_event = alc262_fujitsu_unsol_event, + }, }; static int patch_alc262(struct hda_codec *codec) @@ -4017,8 +4855,8 @@ static int patch_alc262(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC262_AUTO) - codec->patch_ops.init = alc262_auto_init; - + spec->init_hook = alc262_auto_init; + return 0; } @@ -4549,8 +5387,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec) if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, alc861_ignore)) < 0) return err; - if (! spec->autocfg.line_outs && ! spec->autocfg.speaker_pin && - ! spec->autocfg.hp_pin) + if (! spec->autocfg.line_outs) return 0; /* can't find valid BIOS pin config */ if ((err = alc861_auto_fill_dac_nids(spec, &spec->autocfg)) < 0 || @@ -4579,15 +5416,12 @@ static int alc861_parse_auto_config(struct hda_codec *codec) return 1; } -/* init callback for auto-configuration model -- overriding the default init */ -static int alc861_auto_init(struct hda_codec *codec) +/* additional initialization for auto-configuration model */ +static void alc861_auto_init(struct hda_codec *codec) { - alc_init(codec); alc861_auto_init_multi_out(codec); alc861_auto_init_hp_out(codec); alc861_auto_init_analog_input(codec); - - return 0; } @@ -4685,7 +5519,7 @@ static int patch_alc861(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC861_AUTO) - codec->patch_ops.init = alc861_auto_init; + spec->init_hook = alc861_auto_init; return 0; } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 35c2823..b56ca40 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -51,6 +51,7 @@ struct sigmatel_spec { unsigned int line_switch: 1; unsigned int mic_switch: 1; unsigned int alt_switch: 1; + unsigned int hp_detect: 1; /* playback */ struct hda_multi_out multiout; @@ -303,6 +304,12 @@ static struct hda_board_config stac922x_cfg_tbl[] = { .pci_subdevice = 0x0101, .config = STAC_D945GTP3 }, /* Intel D945GTP - 3 Stack */ { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x0202, + .config = STAC_D945GTP3 }, /* Intel D945GNT - 3 Stack, 9221 A1 */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x0b0b, + .config = STAC_D945GTP3 }, /* Intel D945PSN - 3 Stack, 9221 A1 */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, .pci_subdevice = 0x0404, .config = STAC_D945GTP5 }, /* Intel D945GTP - 5 Stack */ { .pci_subvendor = PCI_VENDOR_ID_INTEL, @@ -691,13 +698,7 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec, const struct aut AC_VERB_GET_CONNECT_LIST, 0) & 0xff; } - if (cfg->line_outs) - spec->multiout.num_dacs = cfg->line_outs; - else if (cfg->hp_pin) { - spec->multiout.dac_nids[0] = snd_hda_codec_read(codec, cfg->hp_pin, 0, - AC_VERB_GET_CONNECT_LIST, 0) & 0xff; - spec->multiout.num_dacs = 1; - } + spec->multiout.num_dacs = cfg->line_outs; return 0; } @@ -766,11 +767,13 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, struct auto_pin return 0; wid_caps = get_wcaps(codec, pin); - if (wid_caps & AC_WCAP_UNSOL_CAP) + if (wid_caps & AC_WCAP_UNSOL_CAP) { /* Enable unsolicited responses on the HP widget */ snd_hda_codec_write(codec, pin, 0, AC_VERB_SET_UNSOLICITED_ENABLE, STAC_UNSOL_ENABLE); + spec->hp_detect = 1; + } nid = snd_hda_codec_read(codec, pin, 0, AC_VERB_GET_CONNECT_LIST, 0) & 0xff; for (i = 0; i < cfg->line_outs; i++) { @@ -804,9 +807,6 @@ static int stac92xx_auto_create_analog_input_ctls(struct hda_codec *codec, const for (i = 0; i < AUTO_PIN_LAST; i++) { int index = -1; if (cfg->input_pins[i]) { - /* Enable active pin widget as an input */ - stac92xx_auto_set_pinctl(codec, cfg->input_pins[i], AC_PINCTL_IN_EN); - imux->items[imux->num_items].label = auto_pin_cfg_labels[i]; for (j=0; j<spec->num_muxes; j++) { @@ -855,10 +855,8 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL)) < 0) return err; - if (! spec->autocfg.line_outs && ! spec->autocfg.hp_pin) + if (! spec->autocfg.line_outs) return 0; /* can't find valid pin config */ - stac92xx_auto_init_multi_out(codec); - stac92xx_auto_init_hp_out(codec); if ((err = stac92xx_add_dyn_out_pins(codec, &spec->autocfg)) < 0) return err; if ((err = stac92xx_auto_fill_dac_nids(codec, &spec->autocfg)) < 0) @@ -873,14 +871,10 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if (spec->multiout.max_channels > 2) spec->surr_switch = 1; - if (spec->autocfg.dig_out_pin) { + if (spec->autocfg.dig_out_pin) spec->multiout.dig_out_nid = dig_out; - stac92xx_auto_set_pinctl(codec, spec->autocfg.dig_out_pin, AC_PINCTL_OUT_EN); - } - if (spec->autocfg.dig_in_pin) { + if (spec->autocfg.dig_in_pin) spec->dig_in_nid = dig_in; - stac92xx_auto_set_pinctl(codec, spec->autocfg.dig_in_pin, AC_PINCTL_IN_EN); - } if (spec->kctl_alloc) spec->mixers[spec->num_mixers++] = spec->kctl_alloc; @@ -890,6 +884,29 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out return 1; } +/* add playback controls for HP output */ +static int stac9200_auto_create_hp_ctls(struct hda_codec *codec, + struct auto_pin_cfg *cfg) +{ + struct sigmatel_spec *spec = codec->spec; + hda_nid_t pin = cfg->hp_pin; + unsigned int wid_caps; + + if (! pin) + return 0; + + wid_caps = get_wcaps(codec, pin); + if (wid_caps & AC_WCAP_UNSOL_CAP) { + /* Enable unsolicited responses on the HP widget */ + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + STAC_UNSOL_ENABLE); + spec->hp_detect = 1; + } + + return 0; +} + static int stac9200_parse_auto_config(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -901,14 +918,13 @@ static int stac9200_parse_auto_config(struct hda_codec *codec) if ((err = stac92xx_auto_create_analog_input_ctls(codec, &spec->autocfg)) < 0) return err; - if (spec->autocfg.dig_out_pin) { + if ((err = stac9200_auto_create_hp_ctls(codec, &spec->autocfg)) < 0) + return err; + + if (spec->autocfg.dig_out_pin) spec->multiout.dig_out_nid = 0x05; - stac92xx_auto_set_pinctl(codec, spec->autocfg.dig_out_pin, AC_PINCTL_OUT_EN); - } - if (spec->autocfg.dig_in_pin) { + if (spec->autocfg.dig_in_pin) spec->dig_in_nid = 0x04; - stac92xx_auto_set_pinctl(codec, spec->autocfg.dig_in_pin, AC_PINCTL_IN_EN); - } if (spec->kctl_alloc) spec->mixers[spec->num_mixers++] = spec->kctl_alloc; @@ -921,9 +937,31 @@ static int stac9200_parse_auto_config(struct hda_codec *codec) static int stac92xx_init(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; snd_hda_sequence_write(codec, spec->init); + /* set up pins */ + if (spec->hp_detect) { + /* fake event to set up pins */ + codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); + } else { + stac92xx_auto_init_multi_out(codec); + stac92xx_auto_init_hp_out(codec); + } + for (i = 0; i < AUTO_PIN_LAST; i++) { + if (cfg->input_pins[i]) + stac92xx_auto_set_pinctl(codec, cfg->input_pins[i], + AC_PINCTL_IN_EN); + } + if (cfg->dig_out_pin) + stac92xx_auto_set_pinctl(codec, cfg->dig_out_pin, + AC_PINCTL_OUT_EN); + if (cfg->dig_in_pin) + stac92xx_auto_set_pinctl(codec, cfg->dig_in_pin, + AC_PINCTL_IN_EN); + return 0; } @@ -1142,6 +1180,166 @@ static int patch_stac927x(struct hda_codec *codec) } /* + * STAC 7661(?) hack + */ + +/* static config for Sony VAIO FE550G */ +static hda_nid_t vaio_dacs[] = { 0x2 }; +#define VAIO_HP_DAC 0x5 +static hda_nid_t vaio_adcs[] = { 0x8 /*,0x6*/ }; +static hda_nid_t vaio_mux_nids[] = { 0x15 }; + +static struct hda_input_mux vaio_mux = { + .num_items = 2, + .items = { + /* { "HP", 0x0 }, + { "Unknown", 0x1 }, */ + { "Mic", 0x2 }, + { "PCM", 0x3 }, + } +}; + +static struct hda_verb vaio_init[] = { + {0x0a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP <- 0x2 */ + {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Speaker <- 0x5 */ + {0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? (<- 0x2) */ + {0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x2}, /* mic-sel: 0a,0d,14,02 */ + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */ + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */ + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* capture sw/vol -> 0x8 */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* CD-in -> 0x6 */ + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Mic-in -> 0x9 */ + {} +}; + +/* bind volumes of both NID 0x02 and 0x05 */ +static int vaio_master_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + long *valp = ucontrol->value.integer.value; + int change; + + change = snd_hda_codec_amp_update(codec, 0x02, 0, HDA_OUTPUT, 0, + 0x7f, valp[0] & 0x7f); + change |= snd_hda_codec_amp_update(codec, 0x02, 1, HDA_OUTPUT, 0, + 0x7f, valp[1] & 0x7f); + snd_hda_codec_amp_update(codec, 0x05, 0, HDA_OUTPUT, 0, + 0x7f, valp[0] & 0x7f); + snd_hda_codec_amp_update(codec, 0x05, 1, HDA_OUTPUT, 0, + 0x7f, valp[1] & 0x7f); + return change; +} + +/* bind volumes of both NID 0x02 and 0x05 */ +static int vaio_master_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + long *valp = ucontrol->value.integer.value; + int change; + + change = snd_hda_codec_amp_update(codec, 0x02, 0, HDA_OUTPUT, 0, + 0x80, valp[0] & 0x80); + change |= snd_hda_codec_amp_update(codec, 0x02, 1, HDA_OUTPUT, 0, + 0x80, valp[1] & 0x80); + snd_hda_codec_amp_update(codec, 0x05, 0, HDA_OUTPUT, 0, + 0x80, valp[0] & 0x80); + snd_hda_codec_amp_update(codec, 0x05, 1, HDA_OUTPUT, 0, + 0x80, valp[1] & 0x80); + return change; +} + +static struct snd_kcontrol_new vaio_mixer[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Volume", + .info = snd_hda_mixer_amp_volume_info, + .get = snd_hda_mixer_amp_volume_get, + .put = vaio_master_vol_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT), + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = vaio_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT), + }, + /* HDA_CODEC_VOLUME("CD Capture Volume", 0x07, 0, HDA_INPUT), */ + HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .count = 1, + .info = stac92xx_mux_enum_info, + .get = stac92xx_mux_enum_get, + .put = stac92xx_mux_enum_put, + }, + {} +}; + +static struct hda_codec_ops stac7661_patch_ops = { + .build_controls = stac92xx_build_controls, + .build_pcms = stac92xx_build_pcms, + .init = stac92xx_init, + .free = stac92xx_free, +#ifdef CONFIG_PM + .resume = stac92xx_resume, +#endif +}; + +enum { STAC7661_VAIO }; + +static struct hda_board_config stac7661_cfg_tbl[] = { + { .modelname = "vaio", .config = STAC7661_VAIO }, + { .pci_subvendor = 0x104d, .pci_subdevice = 0x81e6, + .config = STAC7661_VAIO }, + { .pci_subvendor = 0x104d, .pci_subdevice = 0x81ef, + .config = STAC7661_VAIO }, + {} +}; + +static int patch_stac7661(struct hda_codec *codec) +{ + struct sigmatel_spec *spec; + int board_config; + + board_config = snd_hda_check_board_config(codec, stac7661_cfg_tbl); + if (board_config < 0) + /* unknown config, let generic-parser do its job... */ + return snd_hda_parse_generic_codec(codec); + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + switch (board_config) { + case STAC7661_VAIO: + spec->mixer = vaio_mixer; + spec->init = vaio_init; + spec->multiout.max_channels = 2; + spec->multiout.num_dacs = ARRAY_SIZE(vaio_dacs); + spec->multiout.dac_nids = vaio_dacs; + spec->multiout.hp_nid = VAIO_HP_DAC; + spec->num_adcs = ARRAY_SIZE(vaio_adcs); + spec->adc_nids = vaio_adcs; + spec->input_mux = &vaio_mux; + spec->mux_nids = vaio_mux_nids; + break; + } + + codec->patch_ops = stac7661_patch_ops; + return 0; +} + + +/* * patch entries */ struct hda_codec_preset snd_hda_preset_sigmatel[] = { @@ -1162,5 +1360,6 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x83847627, .name = "STAC9271D", .patch = patch_stac927x }, { .id = 0x83847628, .name = "STAC9274X5NH", .patch = patch_stac927x }, { .id = 0x83847629, .name = "STAC9274D5NH", .patch = patch_stac927x }, + { .id = 0x83847661, .name = "STAC7661", .patch = patch_stac7661 }, {} /* terminator */ }; |
