From d031166fecac97fc6b5c35636deace8a3c9ec5f6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 7 Nov 2005 14:38:44 +0100 Subject: [ALSA] hda-codec - Allocate amp hash array dynamically Modules: HDA Codec driver Allocate amp hash array dynamically instead of static array. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 22 +++++++++++++++++++--- sound/pci/hda/hda_codec.h | 3 ++- 2 files changed, 21 insertions(+), 4 deletions(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0dbeeaf6113a..e7fb182f90ed 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -465,6 +465,7 @@ static void snd_hda_codec_free(struct hda_codec *codec) codec->bus->caddr_tbl[codec->addr] = NULL; if (codec->patch_ops.free) codec->patch_ops.free(codec); + kfree(codec->amp_info); kfree(codec); } @@ -586,6 +587,8 @@ static void init_amp_hash(struct hda_codec *codec) { memset(codec->amp_hash, 0xff, sizeof(codec->amp_hash)); codec->num_amp_entries = 0; + codec->amp_info_size = 0; + codec->amp_info = NULL; } /* query the hash. allocate an entry if not found. */ @@ -603,9 +606,22 @@ static struct hda_amp_info *get_alloc_amp_hash(struct hda_codec *codec, u32 key) } /* add a new hash entry */ - if (codec->num_amp_entries >= ARRAY_SIZE(codec->amp_info)) { - snd_printk(KERN_ERR "hda_codec: Tooooo many amps!\n"); - return NULL; + if (codec->num_amp_entries >= codec->amp_info_size) { + /* reallocate the array */ + int new_size = codec->amp_info_size + 64; + struct hda_amp_info *new_info = kcalloc(new_size, sizeof(struct hda_amp_info), + GFP_KERNEL); + if (! new_info) { + snd_printk(KERN_ERR "hda_codec: can't malloc amp_info\n"); + return NULL; + } + if (codec->amp_info) { + memcpy(new_info, codec->amp_info, + codec->amp_info_size * sizeof(struct hda_amp_info)); + kfree(codec->amp_info); + } + codec->amp_info_size = new_size; + codec->amp_info = new_info; } cur = codec->num_amp_entries++; info = &codec->amp_info[cur]; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 1179d6cfa82a..b0123adc569c 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -545,7 +545,8 @@ struct hda_codec { /* hash for amp access */ u16 amp_hash[32]; int num_amp_entries; - struct hda_amp_info amp_info[128]; /* big enough? */ + int amp_info_size; + struct hda_amp_info *amp_info; struct semaphore spdif_mutex; unsigned int spdif_status; /* IEC958 status bits */ -- cgit v1.2.3 From 362775e2125b74cd04f83fd4ef5b72ef1ee6d3a1 Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Mon, 7 Nov 2005 14:43:23 +0100 Subject: [ALSA] sound/hda: rate-limit timeout message Modules: HDA Intel driver Rate-limit the azx_get_response timeout message. A continuous 2 per second is too much. Signed-off-by: Randy Dunlap Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ed525c03c996..429ef3810b16 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -37,6 +37,7 @@ #include #include #include +#include #include #include #include @@ -514,7 +515,9 @@ static unsigned int azx_get_response(struct hda_codec *codec) while (chip->rirb.cmds) { if (! --timeout) { - snd_printk(KERN_ERR "azx_get_response timeout\n"); + if (printk_ratelimit()) + snd_printk(KERN_ERR + "azx_get_response timeout\n"); chip->rirb.rp = azx_readb(chip, RIRBWP); chip->rirb.cmds = 0; return -1; -- cgit v1.2.3 From 26741b5512a99ee35f398ef018d23a38e8dc6e8a Mon Sep 17 00:00:00 2001 From: Daniel Mueller Date: Mon, 14 Nov 2005 17:40:44 +0100 Subject: [ALSA] hda-codec - Fix HDA sound and V.92 modem for notebook Siemens FieldPG-M Modules: HDA Codec driver The patch fixes the problem of mute onboard HDA sound output, buildin V.92 modem idendification and functionality. Signed-off-by: Daniel Mueller Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_si3054.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index 9c7fe0b3200a..12b5de29f42f 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -295,6 +295,7 @@ static int patch_si3054(struct hda_codec *codec) * patch entries */ struct hda_codec_preset snd_hda_preset_si3054[] = { + { .id = 0x163c3055, .name = "Si3054", .patch = patch_si3054 }, { .id = 0x163c3155, .name = "Si3054", .patch = patch_si3054 }, {} }; -- cgit v1.2.3 From d2a6d7dc757da6b57d77bd8b460cf4faa9fd152d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 17 Nov 2005 11:06:29 +0100 Subject: [ALSA] hda-codec - Add channel-mode helper Modules: HDA Codec driver,HDA generic driver Add common channel-mode helper functions for all codec patches. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 48 +++++++++++++++++++++++++++++++ sound/pci/hda/hda_local.h | 17 +++++++++++ sound/pci/hda/patch_analog.c | 2 +- sound/pci/hda/patch_cmedia.c | 62 +++++++++++---------------------------- sound/pci/hda/patch_realtek.c | 67 +++++++++++-------------------------------- 5 files changed, 99 insertions(+), 97 deletions(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index e7fb182f90ed..14a6f5463277 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1662,6 +1662,54 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, snd_kcontrol_new_t *knew) } + /* + * Channel mode helper + */ +int snd_hda_ch_mode_info(struct hda_codec *codec, snd_ctl_elem_info_t *uinfo, + const struct hda_channel_mode *chmode, int num_chmodes) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = num_chmodes; + if (uinfo->value.enumerated.item >= num_chmodes) + uinfo->value.enumerated.item = num_chmodes - 1; + sprintf(uinfo->value.enumerated.name, "%dch", + chmode[uinfo->value.enumerated.item].channels); + return 0; +} + +int snd_hda_ch_mode_get(struct hda_codec *codec, snd_ctl_elem_value_t *ucontrol, + const struct hda_channel_mode *chmode, int num_chmodes, + int max_channels) +{ + int i; + + for (i = 0; i < num_chmodes; i++) { + if (max_channels == chmode[i].channels) { + ucontrol->value.enumerated.item[0] = i; + break; + } + } + return 0; +} + +int snd_hda_ch_mode_put(struct hda_codec *codec, snd_ctl_elem_value_t *ucontrol, + const struct hda_channel_mode *chmode, int num_chmodes, + int *max_channelsp) +{ + unsigned int mode; + + mode = ucontrol->value.enumerated.item[0]; + snd_assert(mode < num_chmodes, return -EINVAL); + if (*max_channelsp && ! codec->in_resume) + return 0; + /* change the current channel setting */ + *max_channelsp = chmode[mode].channels; + if (chmode[mode].sequence) + snd_hda_sequence_write(codec, chmode[mode].sequence); + return 1; +} + /* * input MUX helper */ diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index f51a56f813c8..05a88fb1d652 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -102,6 +102,23 @@ int snd_hda_input_mux_put(struct hda_codec *codec, const struct hda_input_mux *i snd_ctl_elem_value_t *ucontrol, hda_nid_t nid, unsigned int *cur_val); + /* + * Channel mode helper + */ +struct hda_channel_mode { + int channels; + const struct hda_verb *sequence; +}; + +int snd_hda_ch_mode_info(struct hda_codec *codec, snd_ctl_elem_info_t *uinfo, + const struct hda_channel_mode *chmode, int num_chmodes); +int snd_hda_ch_mode_get(struct hda_codec *codec, snd_ctl_elem_value_t *ucontrol, + const struct hda_channel_mode *chmode, int num_chmodes, + int max_channels); +int snd_hda_ch_mode_put(struct hda_codec *codec, snd_ctl_elem_value_t *ucontrol, + const struct hda_channel_mode *chmode, int num_chmodes, + int *max_channelsp); + /* * Multi-channel / digital-out PCM helper */ diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index d7d636decef8..4687736aa0d7 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -52,7 +52,7 @@ struct ad198x_spec { unsigned int cur_mux[3]; /* channel model */ - const struct alc_channel_mode *channel_mode; + const struct hda_channel_mode *channel_mode; int num_channel_mode; /* PCM information */ diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 523c362ec44d..6e0fd92a2be3 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -44,7 +44,6 @@ enum { struct cmi_spec { int board_config; - unsigned int surr_switch: 1; /* switchable line,mic */ unsigned int no_line_in: 1; /* no line-in (5-jack) */ unsigned int front_panel: 1; /* has front-panel 2-jack */ @@ -62,9 +61,8 @@ struct cmi_spec { unsigned int cur_mux[2]; /* channel mode */ - unsigned int num_ch_modes; - unsigned int cur_ch_mode; - const struct cmi_channel_mode *channel_modes; + int num_channel_modes; + const struct hda_channel_mode *channel_modes; struct hda_pcm pcm_rec[2]; /* PCM information */ @@ -158,12 +156,7 @@ static struct hda_verb cmi9880_ch8_init[] = { {} }; -struct cmi_channel_mode { - unsigned int channels; - const struct hda_verb *sequence; -}; - -static struct cmi_channel_mode cmi9880_channel_modes[3] = { +static struct hda_channel_mode cmi9880_channel_modes[3] = { { 2, cmi9880_ch2_init }, { 6, cmi9880_ch6_init }, { 8, cmi9880_ch8_init }, @@ -173,43 +166,24 @@ static int cmi_ch_mode_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct cmi_spec *spec = codec->spec; - - snd_assert(spec->channel_modes, return -EINVAL); - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = spec->num_ch_modes; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - sprintf(uinfo->value.enumerated.name, "%dch", - spec->channel_modes[uinfo->value.enumerated.item].channels); - return 0; + return snd_hda_ch_mode_info(codec, uinfo, spec->channel_modes, + spec->num_channel_modes); } static int cmi_ch_mode_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct cmi_spec *spec = codec->spec; - - ucontrol->value.enumerated.item[0] = spec->cur_ch_mode; - return 0; + return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_modes, + spec->num_channel_modes, spec->multiout.max_channels); } static int cmi_ch_mode_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct cmi_spec *spec = codec->spec; - - snd_assert(spec->channel_modes, return -EINVAL); - if (ucontrol->value.enumerated.item[0] >= spec->num_ch_modes) - ucontrol->value.enumerated.item[0] = spec->num_ch_modes; - if (ucontrol->value.enumerated.item[0] == spec->cur_ch_mode && - ! codec->in_resume) - return 0; - - spec->cur_ch_mode = ucontrol->value.enumerated.item[0]; - snd_hda_sequence_write(codec, spec->channel_modes[spec->cur_ch_mode].sequence); - spec->multiout.max_channels = spec->channel_modes[spec->cur_ch_mode].channels; - return 1; + return snd_hda_ch_mode_put(codec, ucontrol, spec->channel_modes, + spec->num_channel_modes, &spec->multiout.max_channels); } /* @@ -361,7 +335,7 @@ static int cmi9880_build_controls(struct hda_codec *codec) err = snd_hda_add_new_ctls(codec, cmi9880_basic_mixer); if (err < 0) return err; - if (spec->surr_switch) { + if (spec->channel_modes) { err = snd_hda_add_new_ctls(codec, cmi9880_ch_mode_mixer); if (err < 0) return err; @@ -475,7 +449,7 @@ static int cmi9880_resume(struct hda_codec *codec) cmi9880_init(codec); snd_hda_resume_ctls(codec, cmi9880_basic_mixer); - if (spec->surr_switch) + if (spec->channel_modes) snd_hda_resume_ctls(codec, cmi9880_ch_mode_mixer); if (spec->multiout.dig_out_nid) snd_hda_resume_spdif_out(codec); @@ -685,14 +659,13 @@ static int patch_cmi9880(struct hda_codec *codec) switch (spec->board_config) { case CMI_MINIMAL: case CMI_MIN_FP: - spec->surr_switch = 1; + spec->channel_modes = cmi9880_channel_modes; if (spec->board_config == CMI_MINIMAL) - spec->num_ch_modes = 2; + spec->num_channel_modes = 2; else { spec->front_panel = 1; - spec->num_ch_modes = 3; + spec->num_channel_modes = 3; } - spec->channel_modes = cmi9880_channel_modes; spec->multiout.max_channels = cmi9880_channel_modes[0].channels; spec->input_mux = &cmi9880_basic_mux; break; @@ -727,19 +700,18 @@ static int patch_cmi9880(struct hda_codec *codec) get_defcfg_connect(port_f) == AC_JACK_PORT_NONE) { port_g = snd_hda_codec_read(codec, 0x1f, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); port_h = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); - spec->surr_switch = 1; + spec->channel_modes = cmi9880_channel_modes; /* no front panel */ if (get_defcfg_connect(port_g) == AC_JACK_PORT_NONE || get_defcfg_connect(port_h) == AC_JACK_PORT_NONE) { /* no optional rear panel */ spec->board_config = CMI_MINIMAL; spec->front_panel = 0; - spec->num_ch_modes = 2; + spec->num_channel_modes = 2; } else { spec->board_config = CMI_MIN_FP; - spec->num_ch_modes = 3; + spec->num_channel_modes = 3; } - spec->channel_modes = cmi9880_channel_modes; spec->input_mux = &cmi9880_basic_mux; spec->multiout.max_channels = cmi9880_channel_modes[0].channels; } else { diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cffb83fdcff7..a213c19ab06c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -109,7 +109,7 @@ struct alc_spec { unsigned int cur_mux[3]; /* channel model */ - const struct alc_channel_mode *channel_mode; + const struct hda_channel_mode *channel_mode; int num_channel_mode; /* PCM information */ @@ -157,63 +157,28 @@ static int alc_mux_enum_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucon /* * channel mode setting */ -struct alc_channel_mode { - int channels; - const struct hda_verb *sequence; -}; - static int alc880_ch_mode_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; - int items = kcontrol->private_value ? (int)kcontrol->private_value : 2; - - snd_assert(spec->channel_mode, return -ENXIO); - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = items; - if (uinfo->value.enumerated.item >= items) - uinfo->value.enumerated.item = items - 1; - sprintf(uinfo->value.enumerated.name, "%dch", - spec->channel_mode[uinfo->value.enumerated.item].channels); - return 0; + return snd_hda_ch_mode_info(codec, uinfo, spec->channel_mode, + spec->num_channel_mode); } static int alc880_ch_mode_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; - int items = kcontrol->private_value ? (int)kcontrol->private_value : 2; - int i; - - snd_assert(spec->channel_mode, return -ENXIO); - for (i = 0; i < items; i++) { - if (spec->multiout.max_channels == spec->channel_mode[i].channels) { - ucontrol->value.enumerated.item[0] = i; - break; - } - } - return 0; + return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode, + spec->num_channel_mode, spec->multiout.max_channels); } static int alc880_ch_mode_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; - int mode; - - snd_assert(spec->channel_mode, return -ENXIO); - mode = ucontrol->value.enumerated.item[0] ? 1 : 0; - if (spec->multiout.max_channels == spec->channel_mode[mode].channels && - ! codec->in_resume) - return 0; - - /* change the current channel setting */ - spec->multiout.max_channels = spec->channel_mode[mode].channels; - if (spec->channel_mode[mode].sequence) - snd_hda_sequence_write(codec, spec->channel_mode[mode].sequence); - - return 1; + return snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode, + spec->num_channel_mode, &spec->multiout.max_channels); } @@ -328,7 +293,7 @@ static struct hda_verb alc880_threestack_ch6_init[] = { { } /* end */ }; -static struct alc_channel_mode alc880_threestack_modes[2] = { +static struct hda_channel_mode alc880_threestack_modes[2] = { { 2, alc880_threestack_ch2_init }, { 6, alc880_threestack_ch6_init }, }; @@ -443,7 +408,7 @@ static struct hda_verb alc880_fivestack_ch8_init[] = { { } /* end */ }; -static struct alc_channel_mode alc880_fivestack_modes[2] = { +static struct hda_channel_mode alc880_fivestack_modes[2] = { { 6, alc880_fivestack_ch6_init }, { 8, alc880_fivestack_ch8_init }, }; @@ -473,7 +438,7 @@ static struct hda_input_mux alc880_6stack_capture_source = { }; /* fixed 8-channels */ -static struct alc_channel_mode alc880_sixstack_modes[1] = { +static struct hda_channel_mode alc880_sixstack_modes[1] = { { 8, NULL }, }; @@ -540,7 +505,7 @@ static hda_nid_t alc880_w810_dac_nids[3] = { }; /* fixed 6 channels */ -static struct alc_channel_mode alc880_w810_modes[1] = { +static struct hda_channel_mode alc880_w810_modes[1] = { { 6, NULL } }; @@ -572,7 +537,7 @@ static hda_nid_t alc880_z71v_dac_nids[1] = { #define ALC880_Z71V_HP_DAC 0x03 /* fixed 2 channels */ -static struct alc_channel_mode alc880_2_jack_modes[1] = { +static struct hda_channel_mode alc880_2_jack_modes[1] = { { 2, NULL } }; @@ -1227,7 +1192,7 @@ static struct hda_input_mux alc880_test_capture_source = { }, }; -static struct alc_channel_mode alc880_test_modes[4] = { +static struct hda_channel_mode alc880_test_modes[4] = { { 2, NULL }, { 4, NULL }, { 6, NULL }, @@ -1585,7 +1550,7 @@ struct alc_config_preset { unsigned int num_adc_nids; hda_nid_t *adc_nids; unsigned int num_channel_mode; - const struct alc_channel_mode *channel_mode; + const struct hda_channel_mode *channel_mode; const struct hda_input_mux *input_mux; }; @@ -2202,7 +2167,7 @@ static struct hda_input_mux alc260_fujitsu_capture_source = { * element which allows changing the channel mode, so the verb list is * never used. */ -static struct alc_channel_mode alc260_modes[1] = { +static struct hda_channel_mode alc260_modes[1] = { { 2, NULL }, }; @@ -2506,7 +2471,7 @@ static int patch_alc260(struct hda_codec *codec) * driver yet). */ -static struct alc_channel_mode alc882_ch_modes[1] = { +static struct hda_channel_mode alc882_ch_modes[1] = { { 8, NULL } }; -- cgit v1.2.3 From 9f146bb6e68610ab2b62c76e7485900545515613 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 17 Nov 2005 11:07:49 +0100 Subject: [ALSA] hda-codec - Prepare unsol workqueue on demand Modules: HDA Codec driver Prepare unsol workqueue only when a codec really supports. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 14a6f5463277..cfd50b56187b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -288,6 +288,9 @@ static int init_unsol_queue(struct hda_bus *bus) { struct hda_bus_unsolicited *unsol; + if (bus->unsol) /* already initialized */ + return 0; + unsol = kzalloc(sizeof(*unsol), GFP_KERNEL); if (! unsol) { snd_printk(KERN_ERR "hda_codec: can't allocate unsolicited queue\n"); @@ -373,8 +376,6 @@ int snd_hda_bus_new(snd_card_t *card, const struct hda_bus_template *temp, init_MUTEX(&bus->cmd_mutex); INIT_LIST_HEAD(&bus->codec_list); - init_unsol_queue(bus); - if ((err = snd_device_new(card, SNDRV_DEV_BUS, bus, &dev_ops)) < 0) { snd_hda_bus_free(bus); return err; @@ -540,6 +541,9 @@ int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, return err; } + if (codec->patch_ops.unsol_event) + init_unsol_queue(bus); + snd_hda_codec_proc_new(codec); sprintf(component, "HDA:%08x", codec->vendor_id); -- cgit v1.2.3 From a2a20939b1cc82222eb67a4631009338791f1acd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 17 Nov 2005 11:08:23 +0100 Subject: [ALSA] hda-codec - Fix a typo Modules: HDA Codec driver Fix a typo in hda_codec.h. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index b0123adc569c..58b9949aca46 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -98,7 +98,7 @@ enum { #define AC_VERB_SET_UNSOLICITED_ENABLE 0x708 #define AC_VERB_SET_PIN_SENSE 0x709 #define AC_VERB_SET_BEEP_CONTROL 0x70a -#define AC_VERB_SET_EAPD_BTLENALBE 0x70c +#define AC_VERB_SET_EAPD_BTLENABLE 0x70c #define AC_VERB_SET_DIGI_CONVERT_1 0x70d #define AC_VERB_SET_DIGI_CONVERT_2 0x70e #define AC_VERB_SET_VOLUME_KNOB_CONTROL 0x70f -- cgit v1.2.3 From 8d88bc3d361bdd81a214eb9c5d06b291d06c603a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 17 Nov 2005 11:09:23 +0100 Subject: [ALSA] hda-codec - Fix assignment of speaker pin Modules: HDA Codec driver,HDA generic driver Fix the auto-assignment of speaker pin. Handle it independently from line-out pins. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 4 +++- sound/pci/hda/hda_local.h | 15 ++++++++++++++ sound/pci/hda/patch_cmedia.c | 13 ------------ sound/pci/hda/patch_realtek.c | 47 ++++++++++++++++++++----------------------- 4 files changed, 40 insertions(+), 39 deletions(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index cfd50b56187b..2e9f5877386e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1889,7 +1889,6 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *c loc = get_defcfg_location(def_conf); switch (get_defcfg_device(def_conf)) { case AC_JACK_LINE_OUT: - case AC_JACK_SPEAKER: seq = get_defcfg_sequence(def_conf); assoc = get_defcfg_association(def_conf); if (! assoc) @@ -1904,6 +1903,9 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *c sequences[cfg->line_outs] = seq; cfg->line_outs++; break; + case AC_JACK_SPEAKER: + cfg->speaker_pin = nid; + break; case AC_JACK_HP_OUT: cfg->hp_pin = nid; break; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 05a88fb1d652..31d3c7ef5842 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -214,6 +214,7 @@ enum { struct auto_pin_cfg { int line_outs; hda_nid_t line_out_pins[4]; /* sorted in the order of Front/Surr/CLFE/Side */ + hda_nid_t speaker_pin; hda_nid_t hp_pin; hda_nid_t input_pins[AUTO_PIN_LAST]; hda_nid_t dig_out_pin; @@ -228,4 +229,18 @@ struct auto_pin_cfg { int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *cfg); +/* amp values */ +#define AMP_IN_MUTE(idx) (0x7080 | ((idx)<<8)) +#define AMP_IN_UNMUTE(idx) (0x7000 | ((idx)<<8)) +#define AMP_OUT_MUTE 0xb080 +#define AMP_OUT_UNMUTE 0xb000 +#define AMP_OUT_ZERO 0xb000 +/* pinctl values */ +#define PIN_IN 0x20 +#define PIN_VREF80 0x24 +#define PIN_VREF50 0x21 +#define PIN_OUT 0x40 +#define PIN_HP 0xc0 +#define PIN_HP_AMP 0x80 + #endif /* __SOUND_HDA_LOCAL_H */ diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 6e0fd92a2be3..37ee1246b2dd 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -76,19 +76,6 @@ struct cmi_spec { struct hda_verb multi_init[9]; /* 2 verbs for each pin + terminator */ }; -/* amp values */ -#define AMP_IN_MUTE(idx) (0x7080 | ((idx)<<8)) -#define AMP_IN_UNMUTE(idx) (0x7000 | ((idx)<<8)) -#define AMP_OUT_MUTE 0xb080 -#define AMP_OUT_UNMUTE 0xb000 -#define AMP_OUT_ZERO 0xb000 -/* pinctl values */ -#define PIN_IN 0x20 -#define PIN_VREF80 0x24 -#define PIN_VREF50 0x21 -#define PIN_OUT 0x40 -#define PIN_HP 0xc0 - /* * input MUX */ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a213c19ab06c..3ff72c49cd26 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -61,20 +61,6 @@ enum { ALC260_MODEL_LAST /* last tag */ }; -/* amp values */ -#define AMP_IN_MUTE(idx) (0x7080 | ((idx)<<8)) -#define AMP_IN_UNMUTE(idx) (0x7000 | ((idx)<<8)) -#define AMP_OUT_MUTE 0xb080 -#define AMP_OUT_UNMUTE 0xb000 -#define AMP_OUT_ZERO 0xb000 -/* pinctl values */ -#define PIN_IN 0x20 -#define PIN_VREF80 0x24 -#define PIN_VREF50 0x21 -#define PIN_OUT 0x40 -#define PIN_HP 0xc0 -#define PIN_HP_AMP 0x80 - struct alc_spec { /* codec parameterization */ snd_kcontrol_new_t *mixers[3]; /* mixer arrays */ @@ -1833,15 +1819,16 @@ static int alc880_auto_create_multi_out_ctls(struct alc_spec *spec, const struct return err; } } - return 0; } -/* add playback controls for HP output */ -static int alc880_auto_create_hp_ctls(struct alc_spec *spec, hda_nid_t pin) +/* add playback controls for speaker and HP outputs */ +static int alc880_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin, + const char *pfx) { hda_nid_t nid; int err; + char name[32]; if (! pin) return 0; @@ -1854,14 +1841,16 @@ static int alc880_auto_create_hp_ctls(struct alc_spec *spec, hda_nid_t pin) if (! spec->multiout.num_dacs) spec->multiout.num_dacs = 1; } else - /* specify the DAC as the extra HP output */ + /* specify the DAC as the extra output */ spec->multiout.hp_nid = nid; /* control HP volume/switch on the output mixer amp */ nid = alc880_idx_to_mixer(alc880_fixed_pin_idx(pin)); - if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, "Headphone Playback Volume", + sprintf(name, "%s Playback Volume", pfx); + if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, name, HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0) return err; - if ((err = add_control(spec, ALC_CTL_BIND_MUTE, "Headphone Playback Switch", + sprintf(name, "%s Playback Switch", pfx); + if ((err = add_control(spec, ALC_CTL_BIND_MUTE, name, HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT))) < 0) return err; } else if (alc880_is_multi_pin(pin)) { @@ -1873,7 +1862,8 @@ static int alc880_auto_create_hp_ctls(struct alc_spec *spec, hda_nid_t pin) spec->multiout.num_dacs = 1; } /* we have only a switch on HP-out PIN */ - if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, "Headphone Playback Switch", + sprintf(name, "%s Playback Switch", pfx); + if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT))) < 0) return err; } @@ -1947,11 +1937,14 @@ static void alc880_auto_init_multi_out(struct hda_codec *codec) } } -static void alc880_auto_init_hp_out(struct hda_codec *codec) +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; + if (pin) /* connect to front */ + alc880_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); pin = spec->autocfg.hp_pin; if (pin) /* connect to front */ alc880_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); @@ -1985,10 +1978,14 @@ static int alc880_parse_auto_config(struct hda_codec *codec) return err; if ((err = alc880_auto_fill_dac_nids(spec, &spec->autocfg)) < 0) return err; - if (! spec->autocfg.line_outs && ! spec->autocfg.hp_pin) + if (! spec->autocfg.line_outs && ! spec->autocfg.speaker_pin && + ! spec->autocfg.hp_pin) return 0; /* can't find valid BIOS pin config */ if ((err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 || - (err = alc880_auto_create_hp_ctls(spec, spec->autocfg.hp_pin)) < 0 || + (err = alc880_auto_create_extra_out(spec, spec->autocfg.speaker_pin, + "Speaker")) < 0 || + (err = alc880_auto_create_extra_out(spec, spec->autocfg.speaker_pin, + "Headphone")) < 0 || (err = alc880_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0) return err; @@ -2014,7 +2011,7 @@ static int alc880_auto_init(struct hda_codec *codec) { alc_init(codec); alc880_auto_init_multi_out(codec); - alc880_auto_init_hp_out(codec); + alc880_auto_init_extra_out(codec); alc880_auto_init_analog_input(codec); return 0; } -- cgit v1.2.3 From c8b6bf9b5ef1f595a65a3414a5ca2588e8d993b2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 17 Nov 2005 14:57:47 +0100 Subject: [ALSA] Remove xxx_t typedefs: HD-Audio codec Modules: HDA Codec driver,HDA generic driver Remove xxx_t typedefs from the HD-Audio codec support codes. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 90 +++++++++++++++++++++--------------------- sound/pci/hda/hda_codec.h | 14 +++---- sound/pci/hda/hda_generic.c | 20 +++++----- sound/pci/hda/hda_local.h | 36 ++++++++--------- sound/pci/hda/hda_proc.c | 12 +++--- sound/pci/hda/patch_analog.c | 42 ++++++++++---------- sound/pci/hda/patch_cmedia.c | 30 +++++++------- sound/pci/hda/patch_realtek.c | 88 ++++++++++++++++++++--------------------- sound/pci/hda/patch_si3054.c | 20 +++++----- sound/pci/hda/patch_sigmatel.c | 44 ++++++++++----------- 10 files changed, 198 insertions(+), 198 deletions(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 2e9f5877386e..7f4e19951bae 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -332,7 +332,7 @@ static int snd_hda_bus_free(struct hda_bus *bus) return 0; } -static int snd_hda_bus_dev_free(snd_device_t *device) +static int snd_hda_bus_dev_free(struct snd_device *device) { struct hda_bus *bus = device->device_data; return snd_hda_bus_free(bus); @@ -346,12 +346,12 @@ static int snd_hda_bus_dev_free(snd_device_t *device) * * Returns 0 if successful, or a negative error code. */ -int snd_hda_bus_new(snd_card_t *card, const struct hda_bus_template *temp, +int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp, struct hda_bus **busp) { struct hda_bus *bus; int err; - static snd_device_ops_t dev_ops = { + static struct snd_device_ops dev_ops = { .dev_free = snd_hda_bus_dev_free, }; @@ -732,7 +732,7 @@ static int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int #define get_amp_index(kc) (((kc)->private_value >> 19) & 0xf) /* volume */ -int snd_hda_mixer_amp_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) +int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); u16 nid = get_amp_nid(kcontrol); @@ -753,7 +753,7 @@ int snd_hda_mixer_amp_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t return 0; } -int snd_hda_mixer_amp_volume_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = get_amp_nid(kcontrol); @@ -769,7 +769,7 @@ int snd_hda_mixer_amp_volume_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t return 0; } -int snd_hda_mixer_amp_volume_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = get_amp_nid(kcontrol); @@ -791,7 +791,7 @@ int snd_hda_mixer_amp_volume_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t } /* switch */ -int snd_hda_mixer_amp_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) +int snd_hda_mixer_amp_switch_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { int chs = get_amp_channels(kcontrol); @@ -802,7 +802,7 @@ int snd_hda_mixer_amp_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t return 0; } -int snd_hda_mixer_amp_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = get_amp_nid(kcontrol); @@ -818,7 +818,7 @@ int snd_hda_mixer_amp_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t return 0; } -int snd_hda_mixer_amp_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = get_amp_nid(kcontrol); @@ -849,7 +849,7 @@ int snd_hda_mixer_amp_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t #define AMP_VAL_IDX_SHIFT 19 #define AMP_VAL_IDX_MASK (0x0f<<19) -int snd_hda_mixer_bind_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +int snd_hda_mixer_bind_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); unsigned long pval; @@ -864,7 +864,7 @@ int snd_hda_mixer_bind_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t return err; } -int snd_hda_mixer_bind_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +int snd_hda_mixer_bind_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); unsigned long pval; @@ -889,14 +889,14 @@ int snd_hda_mixer_bind_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t * SPDIF out controls */ -static int snd_hda_spdif_mask_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) +static int snd_hda_spdif_mask_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; uinfo->count = 1; return 0; } -static int snd_hda_spdif_cmask_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int snd_hda_spdif_cmask_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { ucontrol->value.iec958.status[0] = IEC958_AES0_PROFESSIONAL | IEC958_AES0_NONAUDIO | @@ -907,7 +907,7 @@ static int snd_hda_spdif_cmask_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_ return 0; } -static int snd_hda_spdif_pmask_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int snd_hda_spdif_pmask_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { ucontrol->value.iec958.status[0] = IEC958_AES0_PROFESSIONAL | IEC958_AES0_NONAUDIO | @@ -915,7 +915,7 @@ static int snd_hda_spdif_pmask_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_ return 0; } -static int snd_hda_spdif_default_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int snd_hda_spdif_default_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); @@ -978,7 +978,7 @@ static unsigned int convert_to_spdif_status(unsigned short val) return sbits; } -static int snd_hda_spdif_default_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int snd_hda_spdif_default_put(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; @@ -1004,7 +1004,7 @@ static int snd_hda_spdif_default_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_valu return change; } -static int snd_hda_spdif_out_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) +static int snd_hda_spdif_out_switch_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; uinfo->count = 1; @@ -1013,7 +1013,7 @@ static int snd_hda_spdif_out_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_ return 0; } -static int snd_hda_spdif_out_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int snd_hda_spdif_out_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); @@ -1021,7 +1021,7 @@ static int snd_hda_spdif_out_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_v return 0; } -static int snd_hda_spdif_out_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int snd_hda_spdif_out_switch_put(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; @@ -1044,7 +1044,7 @@ static int snd_hda_spdif_out_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_v return change; } -static snd_kcontrol_new_t dig_mixes[] = { +static struct snd_kcontrol_new dig_mixes[] = { { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1089,8 +1089,8 @@ static snd_kcontrol_new_t dig_mixes[] = { int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid) { int err; - snd_kcontrol_t *kctl; - snd_kcontrol_new_t *dig_mix; + struct snd_kcontrol *kctl; + struct snd_kcontrol_new *dig_mix; for (dig_mix = dig_mixes; dig_mix->name; dig_mix++) { kctl = snd_ctl_new1(dig_mix, codec); @@ -1109,7 +1109,7 @@ int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid) #define snd_hda_spdif_in_switch_info snd_hda_spdif_out_switch_info -static int snd_hda_spdif_in_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int snd_hda_spdif_in_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); @@ -1117,7 +1117,7 @@ static int snd_hda_spdif_in_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_va return 0; } -static int snd_hda_spdif_in_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int snd_hda_spdif_in_switch_put(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; @@ -1134,7 +1134,7 @@ static int snd_hda_spdif_in_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_va return change; } -static int snd_hda_spdif_in_status_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int snd_hda_spdif_in_status_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; @@ -1150,7 +1150,7 @@ static int snd_hda_spdif_in_status_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_va return 0; } -static snd_kcontrol_new_t dig_in_ctls[] = { +static struct snd_kcontrol_new dig_in_ctls[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), @@ -1181,8 +1181,8 @@ static snd_kcontrol_new_t dig_in_ctls[] = { int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid) { int err; - snd_kcontrol_t *kctl; - snd_kcontrol_new_t *dig_mix; + struct snd_kcontrol *kctl; + struct snd_kcontrol_new *dig_mix; for (dig_mix = dig_in_ctls; dig_mix->name; dig_mix++) { kctl = snd_ctl_new1(dig_mix, codec); @@ -1498,7 +1498,7 @@ int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid, */ static int hda_pcm_default_open_close(struct hda_pcm_stream *hinfo, struct hda_codec *codec, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { return 0; } @@ -1507,7 +1507,7 @@ static int hda_pcm_default_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, unsigned int format, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { snd_hda_codec_setup_stream(codec, hinfo->nid, stream_tag, 0, format); return 0; @@ -1515,7 +1515,7 @@ static int hda_pcm_default_prepare(struct hda_pcm_stream *hinfo, static int hda_pcm_default_cleanup(struct hda_pcm_stream *hinfo, struct hda_codec *codec, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { snd_hda_codec_setup_stream(codec, hinfo->nid, 0, 0, 0); return 0; @@ -1646,14 +1646,14 @@ int snd_hda_check_board_config(struct hda_codec *codec, const struct hda_board_c /** * snd_hda_add_new_ctls - create controls from the array * @codec: the HDA codec - * @knew: the array of snd_kcontrol_new_t + * @knew: the array of struct snd_kcontrol_new * * This helper function creates and add new controls in the given array. * The array must be terminated with an empty entry as terminator. * * Returns 0 if successful, or a negative error code. */ -int snd_hda_add_new_ctls(struct hda_codec *codec, snd_kcontrol_new_t *knew) +int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew) { int err; @@ -1666,10 +1666,10 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, snd_kcontrol_new_t *knew) } - /* +/* * Channel mode helper */ -int snd_hda_ch_mode_info(struct hda_codec *codec, snd_ctl_elem_info_t *uinfo, +int snd_hda_ch_mode_info(struct hda_codec *codec, struct snd_ctl_elem_info *uinfo, const struct hda_channel_mode *chmode, int num_chmodes) { uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; @@ -1682,7 +1682,7 @@ int snd_hda_ch_mode_info(struct hda_codec *codec, snd_ctl_elem_info_t *uinfo, return 0; } -int snd_hda_ch_mode_get(struct hda_codec *codec, snd_ctl_elem_value_t *ucontrol, +int snd_hda_ch_mode_get(struct hda_codec *codec, struct snd_ctl_elem_value *ucontrol, const struct hda_channel_mode *chmode, int num_chmodes, int max_channels) { @@ -1697,7 +1697,7 @@ int snd_hda_ch_mode_get(struct hda_codec *codec, snd_ctl_elem_value_t *ucontrol, return 0; } -int snd_hda_ch_mode_put(struct hda_codec *codec, snd_ctl_elem_value_t *ucontrol, +int snd_hda_ch_mode_put(struct hda_codec *codec, struct snd_ctl_elem_value *ucontrol, const struct hda_channel_mode *chmode, int num_chmodes, int *max_channelsp) { @@ -1717,7 +1717,7 @@ int snd_hda_ch_mode_put(struct hda_codec *codec, snd_ctl_elem_value_t *ucontrol, /* * input MUX helper */ -int snd_hda_input_mux_info(const struct hda_input_mux *imux, snd_ctl_elem_info_t *uinfo) +int snd_hda_input_mux_info(const struct hda_input_mux *imux, struct snd_ctl_elem_info *uinfo) { unsigned int index; @@ -1732,7 +1732,7 @@ int snd_hda_input_mux_info(const struct hda_input_mux *imux, snd_ctl_elem_info_t } int snd_hda_input_mux_put(struct hda_codec *codec, const struct hda_input_mux *imux, - snd_ctl_elem_value_t *ucontrol, hda_nid_t nid, + struct snd_ctl_elem_value *ucontrol, hda_nid_t nid, unsigned int *cur_val) { unsigned int idx; @@ -1783,7 +1783,7 @@ int snd_hda_multi_out_dig_close(struct hda_codec *codec, struct hda_multi_out *m * set up more restrictions for analog out */ int snd_hda_multi_out_analog_open(struct hda_codec *codec, struct hda_multi_out *mout, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { substream->runtime->hw.channels_max = mout->max_channels; return snd_pcm_hw_constraint_step(substream->runtime, 0, @@ -1797,7 +1797,7 @@ int snd_hda_multi_out_analog_open(struct hda_codec *codec, struct hda_multi_out int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, struct hda_multi_out *mout, unsigned int stream_tag, unsigned int format, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { hda_nid_t *nids = mout->dac_nids; int chs = substream->runtime->channels; @@ -2019,15 +2019,15 @@ int snd_hda_resume(struct hda_bus *bus) /** * snd_hda_resume_ctls - resume controls in the new control list * @codec: the HDA codec - * @knew: the array of snd_kcontrol_new_t + * @knew: the array of struct snd_kcontrol_new * - * This function resumes the mixer controls in the snd_kcontrol_new_t array, + * This function resumes the mixer controls in the struct snd_kcontrol_new array, * originally for snd_hda_add_new_ctls(). * The array must be terminated with an empty entry as terminator. */ -int snd_hda_resume_ctls(struct hda_codec *codec, snd_kcontrol_new_t *knew) +int snd_hda_resume_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew) { - snd_ctl_elem_value_t *val; + struct snd_ctl_elem_value *val; val = kmalloc(sizeof(*val), GFP_KERNEL); if (! val) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 58b9949aca46..0b5c36788898 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -420,7 +420,7 @@ struct hda_bus_template { * A hda_bus contains several codecs in the list codec_list. */ struct hda_bus { - snd_card_t *card; + struct snd_card *card; /* copied from template */ void *private_data; @@ -437,7 +437,7 @@ struct hda_bus { /* unsolicited event queue */ struct hda_bus_unsolicited *unsol; - snd_info_entry_t *proc; + struct snd_info_entry *proc; }; /* @@ -481,14 +481,14 @@ struct hda_amp_info { /* PCM callbacks */ struct hda_pcm_ops { int (*open)(struct hda_pcm_stream *info, struct hda_codec *codec, - snd_pcm_substream_t *substream); + struct snd_pcm_substream *substream); int (*close)(struct hda_pcm_stream *info, struct hda_codec *codec, - snd_pcm_substream_t *substream); + struct snd_pcm_substream *substream); int (*prepare)(struct hda_pcm_stream *info, struct hda_codec *codec, unsigned int stream_tag, unsigned int format, - snd_pcm_substream_t *substream); + struct snd_pcm_substream *substream); int (*cleanup)(struct hda_pcm_stream *info, struct hda_codec *codec, - snd_pcm_substream_t *substream); + struct snd_pcm_substream *substream); }; /* PCM information for each substream */ @@ -563,7 +563,7 @@ enum { /* * constructors */ -int snd_hda_bus_new(snd_card_t *card, const struct hda_bus_template *temp, +int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp, struct hda_bus **busp); int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, struct hda_codec **codecp); diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index d0eb9f2250aa..863e8c6d29a8 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -389,14 +389,14 @@ static int parse_output(struct hda_codec *codec) */ /* control callbacks */ -static int capture_source_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) +static int capture_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_gspec *spec = codec->spec; return snd_hda_input_mux_info(&spec->input_mux, uinfo); } -static int capture_source_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int capture_source_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_gspec *spec = codec->spec; @@ -405,7 +405,7 @@ static int capture_source_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *uc return 0; } -static int capture_source_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int capture_source_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_gspec *spec = codec->spec; @@ -617,7 +617,7 @@ static int create_mixer(struct hda_codec *codec, struct hda_gnode *node, char name[32]; int err; int created = 0; - snd_kcontrol_new_t knew; + struct snd_kcontrol_new knew; if (type) sprintf(name, "%s %s Switch", type, dir_sfx); @@ -625,14 +625,14 @@ static int create_mixer(struct hda_codec *codec, struct hda_gnode *node, sprintf(name, "%s Switch", dir_sfx); if ((node->wid_caps & AC_WCAP_IN_AMP) && (node->amp_in_caps & AC_AMPCAP_MUTE)) { - knew = (snd_kcontrol_new_t)HDA_CODEC_MUTE(name, node->nid, index, HDA_INPUT); + knew = (struct snd_kcontrol_new)HDA_CODEC_MUTE(name, node->nid, index, HDA_INPUT); snd_printdd("[%s] NID=0x%x, DIR=IN, IDX=0x%x\n", name, node->nid, index); if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0) return err; created = 1; } else if ((node->wid_caps & AC_WCAP_OUT_AMP) && (node->amp_out_caps & AC_AMPCAP_MUTE)) { - knew = (snd_kcontrol_new_t)HDA_CODEC_MUTE(name, node->nid, 0, HDA_OUTPUT); + knew = (struct snd_kcontrol_new)HDA_CODEC_MUTE(name, node->nid, 0, HDA_OUTPUT); snd_printdd("[%s] NID=0x%x, DIR=OUT\n", name, node->nid); if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0) return err; @@ -645,14 +645,14 @@ static int create_mixer(struct hda_codec *codec, struct hda_gnode *node, sprintf(name, "%s Volume", dir_sfx); if ((node->wid_caps & AC_WCAP_IN_AMP) && (node->amp_in_caps & AC_AMPCAP_NUM_STEPS)) { - knew = (snd_kcontrol_new_t)HDA_CODEC_VOLUME(name, node->nid, index, HDA_INPUT); + knew = (struct snd_kcontrol_new)HDA_CODEC_VOLUME(name, node->nid, index, HDA_INPUT); snd_printdd("[%s] NID=0x%x, DIR=IN, IDX=0x%x\n", name, node->nid, index); if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0) return err; created = 1; } else if ((node->wid_caps & AC_WCAP_OUT_AMP) && (node->amp_out_caps & AC_AMPCAP_NUM_STEPS)) { - knew = (snd_kcontrol_new_t)HDA_CODEC_VOLUME(name, node->nid, 0, HDA_OUTPUT); + knew = (struct snd_kcontrol_new)HDA_CODEC_VOLUME(name, node->nid, 0, HDA_OUTPUT); snd_printdd("[%s] NID=0x%x, DIR=OUT\n", name, node->nid); if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0) return err; @@ -667,7 +667,7 @@ static int create_mixer(struct hda_codec *codec, struct hda_gnode *node, */ static int check_existing_control(struct hda_codec *codec, const char *type, const char *dir) { - snd_ctl_elem_id_t id; + struct snd_ctl_elem_id id; memset(&id, 0, sizeof(id)); sprintf(id.name, "%s %s Volume", type, dir); id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; @@ -710,7 +710,7 @@ static int build_input_controls(struct hda_codec *codec) /* create input MUX if multiple sources are available */ if (spec->input_mux.num_items > 1) { - static snd_kcontrol_new_t cap_sel = { + static struct snd_kcontrol_new cap_sel = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Source", .info = capture_source_info, diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 31d3c7ef5842..502290424c67 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -60,12 +60,12 @@ #define HDA_CODEC_MUTE(xname, nid, xindex, direction) \ HDA_CODEC_MUTE_MONO(xname, nid, 3, xindex, direction) -int snd_hda_mixer_amp_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo); -int snd_hda_mixer_amp_volume_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol); -int snd_hda_mixer_amp_volume_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol); -int snd_hda_mixer_amp_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo); -int snd_hda_mixer_amp_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol); -int snd_hda_mixer_amp_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol); +int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); +int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +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); /* mono switch binding multiple inputs */ #define HDA_BIND_MUTE_MONO(xname, nid, channel, indices, direction) \ @@ -78,8 +78,8 @@ int snd_hda_mixer_amp_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t /* stereo switch binding multiple inputs */ #define HDA_BIND_MUTE(xname,nid,indices,dir) HDA_BIND_MUTE_MONO(xname,nid,3,indices,dir) -int snd_hda_mixer_bind_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol); -int snd_hda_mixer_bind_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol); +int snd_hda_mixer_bind_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_hda_mixer_bind_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid); int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid); @@ -97,12 +97,12 @@ struct hda_input_mux { struct hda_input_mux_item items[HDA_MAX_NUM_INPUTS]; }; -int snd_hda_input_mux_info(const struct hda_input_mux *imux, snd_ctl_elem_info_t *uinfo); +int snd_hda_input_mux_info(const struct hda_input_mux *imux, struct snd_ctl_elem_info *uinfo); int snd_hda_input_mux_put(struct hda_codec *codec, const struct hda_input_mux *imux, - snd_ctl_elem_value_t *ucontrol, hda_nid_t nid, + struct snd_ctl_elem_value *ucontrol, hda_nid_t nid, unsigned int *cur_val); - /* +/* * Channel mode helper */ struct hda_channel_mode { @@ -110,12 +110,12 @@ struct hda_channel_mode { const struct hda_verb *sequence; }; -int snd_hda_ch_mode_info(struct hda_codec *codec, snd_ctl_elem_info_t *uinfo, +int snd_hda_ch_mode_info(struct hda_codec *codec, struct snd_ctl_elem_info *uinfo, const struct hda_channel_mode *chmode, int num_chmodes); -int snd_hda_ch_mode_get(struct hda_codec *codec, snd_ctl_elem_value_t *ucontrol, +int snd_hda_ch_mode_get(struct hda_codec *codec, struct snd_ctl_elem_value *ucontrol, const struct hda_channel_mode *chmode, int num_chmodes, int max_channels); -int snd_hda_ch_mode_put(struct hda_codec *codec, snd_ctl_elem_value_t *ucontrol, +int snd_hda_ch_mode_put(struct hda_codec *codec, struct snd_ctl_elem_value *ucontrol, const struct hda_channel_mode *chmode, int num_chmodes, int *max_channelsp); @@ -138,11 +138,11 @@ struct hda_multi_out { int snd_hda_multi_out_dig_open(struct hda_codec *codec, struct hda_multi_out *mout); int snd_hda_multi_out_dig_close(struct hda_codec *codec, struct hda_multi_out *mout); int snd_hda_multi_out_analog_open(struct hda_codec *codec, struct hda_multi_out *mout, - snd_pcm_substream_t *substream); + struct snd_pcm_substream *substream); int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, struct hda_multi_out *mout, unsigned int stream_tag, unsigned int format, - snd_pcm_substream_t *substream); + struct snd_pcm_substream *substream); int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, struct hda_multi_out *mout); /* @@ -170,13 +170,13 @@ struct hda_board_config { }; int snd_hda_check_board_config(struct hda_codec *codec, const struct hda_board_config *tbl); -int snd_hda_add_new_ctls(struct hda_codec *codec, snd_kcontrol_new_t *knew); +int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew); /* * power management */ #ifdef CONFIG_PM -int snd_hda_resume_ctls(struct hda_codec *codec, snd_kcontrol_new_t *knew); +int snd_hda_resume_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew); int snd_hda_resume_spdif_out(struct hda_codec *codec); int snd_hda_resume_spdif_in(struct hda_codec *codec); #endif diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 39ddf1cd9019..8cc5773958f6 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -47,7 +47,7 @@ static const char *get_wid_type_name(unsigned int wid_value) return "UNKOWN Widget"; } -static void print_amp_caps(snd_info_buffer_t *buffer, +static void print_amp_caps(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid, int dir) { unsigned int caps; @@ -66,7 +66,7 @@ static void print_amp_caps(snd_info_buffer_t *buffer, (caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT); } -static void print_amp_vals(snd_info_buffer_t *buffer, +static void print_amp_vals(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid, int dir, int stereo, int indices) { @@ -91,7 +91,7 @@ static void print_amp_vals(snd_info_buffer_t *buffer, snd_iprintf(buffer, "\n"); } -static void print_pcm_caps(snd_info_buffer_t *buffer, +static void print_pcm_caps(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid) { unsigned int pcm = snd_hda_param_read(codec, nid, AC_PAR_PCM); @@ -160,7 +160,7 @@ static const char *get_jack_color(u32 cfg) return "UNKNOWN"; } -static void print_pin_caps(snd_info_buffer_t *buffer, +static void print_pin_caps(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid) { static char *jack_conns[4] = { "Jack", "N/A", "Fixed", "Both" }; @@ -194,7 +194,7 @@ static void print_pin_caps(snd_info_buffer_t *buffer, } -static void print_codec_info(snd_info_entry_t *entry, snd_info_buffer_t *buffer) +static void print_codec_info(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct hda_codec *codec = entry->private_data; char buf[32]; @@ -309,7 +309,7 @@ static void print_codec_info(snd_info_entry_t *entry, snd_info_buffer_t *buffer) int snd_hda_codec_proc_new(struct hda_codec *codec) { char name[32]; - snd_info_entry_t *entry; + struct snd_info_entry *entry; int err; snprintf(name, sizeof(name), "codec#%d", codec->addr); diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 4687736aa0d7..1f371fe6b92f 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -28,7 +28,7 @@ #include "hda_local.h" struct ad198x_spec { - snd_kcontrol_new_t *mixers[5]; + struct snd_kcontrol_new *mixers[5]; int num_mixers; const struct hda_verb *init_verbs[3]; /* initialization verbs @@ -65,7 +65,7 @@ struct ad198x_spec { /* * input MUX handling (common part) */ -static int ad198x_mux_enum_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) +static int ad198x_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct ad198x_spec *spec = codec->spec; @@ -73,7 +73,7 @@ static int ad198x_mux_enum_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *u return snd_hda_input_mux_info(spec->input_mux, uinfo); } -static int ad198x_mux_enum_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int ad198x_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct ad198x_spec *spec = codec->spec; @@ -83,7 +83,7 @@ static int ad198x_mux_enum_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *u return 0; } -static int ad198x_mux_enum_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int ad198x_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct ad198x_spec *spec = codec->spec; @@ -135,7 +135,7 @@ static int ad198x_build_controls(struct hda_codec *codec) */ static int ad198x_playback_pcm_open(struct hda_pcm_stream *hinfo, struct hda_codec *codec, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { struct ad198x_spec *spec = codec->spec; return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream); @@ -145,7 +145,7 @@ static int ad198x_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, unsigned int format, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { struct ad198x_spec *spec = codec->spec; return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag, @@ -154,7 +154,7 @@ static int ad198x_playback_pcm_prepare(struct hda_pcm_stream *hinfo, static int ad198x_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, struct hda_codec *codec, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { struct ad198x_spec *spec = codec->spec; return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); @@ -165,7 +165,7 @@ static int ad198x_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, */ static int ad198x_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, struct hda_codec *codec, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { struct ad198x_spec *spec = codec->spec; return snd_hda_multi_out_dig_open(codec, &spec->multiout); @@ -173,7 +173,7 @@ static int ad198x_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, static int ad198x_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, struct hda_codec *codec, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { struct ad198x_spec *spec = codec->spec; return snd_hda_multi_out_dig_close(codec, &spec->multiout); @@ -186,7 +186,7 @@ static int ad198x_capture_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, unsigned int format, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { struct ad198x_spec *spec = codec->spec; snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], @@ -196,7 +196,7 @@ static int ad198x_capture_pcm_prepare(struct hda_pcm_stream *hinfo, static int ad198x_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, struct hda_codec *codec, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { struct ad198x_spec *spec = codec->spec; snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], @@ -348,7 +348,7 @@ static struct hda_input_mux ad1986a_capture_source = { #define ad1986a_pcm_amp_vol_info snd_hda_mixer_amp_volume_info -static int ad1986a_pcm_amp_vol_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int ad1986a_pcm_amp_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct ad198x_spec *ad = codec->spec; @@ -359,7 +359,7 @@ static int ad1986a_pcm_amp_vol_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_ return 0; } -static int ad1986a_pcm_amp_vol_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int ad1986a_pcm_amp_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct ad198x_spec *ad = codec->spec; @@ -377,7 +377,7 @@ static int ad1986a_pcm_amp_vol_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_ #define ad1986a_pcm_amp_sw_info snd_hda_mixer_amp_switch_info -static int ad1986a_pcm_amp_sw_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int ad1986a_pcm_amp_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct ad198x_spec *ad = codec->spec; @@ -388,7 +388,7 @@ static int ad1986a_pcm_amp_sw_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t return 0; } -static int ad1986a_pcm_amp_sw_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int ad1986a_pcm_amp_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct ad198x_spec *ad = codec->spec; @@ -407,7 +407,7 @@ static int ad1986a_pcm_amp_sw_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t /* * mixers */ -static snd_kcontrol_new_t ad1986a_mixers[] = { +static struct snd_kcontrol_new ad1986a_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "PCM Playback Volume", @@ -570,7 +570,7 @@ static struct hda_input_mux ad1983_capture_source = { /* * SPDIF playback route */ -static int ad1983_spdif_route_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) +static int ad1983_spdif_route_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = { "PCM", "ADC" }; @@ -583,7 +583,7 @@ static int ad1983_spdif_route_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t return 0; } -static int ad1983_spdif_route_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int ad1983_spdif_route_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct ad198x_spec *spec = codec->spec; @@ -592,7 +592,7 @@ static int ad1983_spdif_route_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t return 0; } -static int ad1983_spdif_route_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int ad1983_spdif_route_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct ad198x_spec *spec = codec->spec; @@ -606,7 +606,7 @@ static int ad1983_spdif_route_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t return 0; } -static snd_kcontrol_new_t ad1983_mixers[] = { +static struct snd_kcontrol_new ad1983_mixers[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x05, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x05, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x06, 0x0, HDA_OUTPUT), @@ -736,7 +736,7 @@ static struct hda_input_mux ad1981_capture_source = { }, }; -static snd_kcontrol_new_t ad1981_mixers[] = { +static struct snd_kcontrol_new ad1981_mixers[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x05, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x05, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x06, 0x0, HDA_OUTPUT), diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 37ee1246b2dd..9a6981162982 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -79,14 +79,14 @@ struct cmi_spec { /* * input MUX */ -static int cmi_mux_enum_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) +static int cmi_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct cmi_spec *spec = codec->spec; return snd_hda_input_mux_info(spec->input_mux, uinfo); } -static int cmi_mux_enum_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int cmi_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct cmi_spec *spec = codec->spec; @@ -96,7 +96,7 @@ static int cmi_mux_enum_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucon return 0; } -static int cmi_mux_enum_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int cmi_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct cmi_spec *spec = codec->spec; @@ -149,7 +149,7 @@ static struct hda_channel_mode cmi9880_channel_modes[3] = { { 8, cmi9880_ch8_init }, }; -static int cmi_ch_mode_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) +static int cmi_ch_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct cmi_spec *spec = codec->spec; @@ -157,7 +157,7 @@ static int cmi_ch_mode_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo spec->num_channel_modes); } -static int cmi_ch_mode_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int cmi_ch_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct cmi_spec *spec = codec->spec; @@ -165,7 +165,7 @@ static int cmi_ch_mode_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucont spec->num_channel_modes, spec->multiout.max_channels); } -static int cmi_ch_mode_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int cmi_ch_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct cmi_spec *spec = codec->spec; @@ -175,7 +175,7 @@ static int cmi_ch_mode_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucont /* */ -static snd_kcontrol_new_t cmi9880_basic_mixer[] = { +static struct snd_kcontrol_new cmi9880_basic_mixer[] = { /* CMI9880 has no playback volumes! */ HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), /* front */ HDA_CODEC_MUTE("Surround Playback Switch", 0x04, 0x0, HDA_OUTPUT), @@ -207,7 +207,7 @@ static snd_kcontrol_new_t cmi9880_basic_mixer[] = { /* * shared I/O pins */ -static snd_kcontrol_new_t cmi9880_ch_mode_mixer[] = { +static struct snd_kcontrol_new cmi9880_ch_mode_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Channel Mode", @@ -452,7 +452,7 @@ static int cmi9880_resume(struct hda_codec *codec) */ static int cmi9880_playback_pcm_open(struct hda_pcm_stream *hinfo, struct hda_codec *codec, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { struct cmi_spec *spec = codec->spec; return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream); @@ -462,7 +462,7 @@ static int cmi9880_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, unsigned int format, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { struct cmi_spec *spec = codec->spec; return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag, @@ -471,7 +471,7 @@ static int cmi9880_playback_pcm_prepare(struct hda_pcm_stream *hinfo, static int cmi9880_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, struct hda_codec *codec, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { struct cmi_spec *spec = codec->spec; return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); @@ -482,7 +482,7 @@ static int cmi9880_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, */ static int cmi9880_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, struct hda_codec *codec, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { struct cmi_spec *spec = codec->spec; return snd_hda_multi_out_dig_open(codec, &spec->multiout); @@ -490,7 +490,7 @@ static int cmi9880_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, static int cmi9880_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, struct hda_codec *codec, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { struct cmi_spec *spec = codec->spec; return snd_hda_multi_out_dig_close(codec, &spec->multiout); @@ -503,7 +503,7 @@ static int cmi9880_capture_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, unsigned int format, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { struct cmi_spec *spec = codec->spec; @@ -514,7 +514,7 @@ static int cmi9880_capture_pcm_prepare(struct hda_pcm_stream *hinfo, static int cmi9880_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, struct hda_codec *codec, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { struct cmi_spec *spec = codec->spec; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3ff72c49cd26..62e6993056e6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -63,7 +63,7 @@ enum { struct alc_spec { /* codec parameterization */ - snd_kcontrol_new_t *mixers[3]; /* mixer arrays */ + struct snd_kcontrol_new *mixers[3]; /* mixer arrays */ unsigned int num_mixers; const struct hda_verb *init_verbs[3]; /* initialization verbs @@ -104,7 +104,7 @@ struct alc_spec { /* dynamic controls, init_verbs and input_mux */ struct auto_pin_cfg autocfg; unsigned int num_kctl_alloc, num_kctl_used; - snd_kcontrol_new_t *kctl_alloc; + struct snd_kcontrol_new *kctl_alloc; struct hda_input_mux private_imux; hda_nid_t private_dac_nids[4]; }; @@ -113,14 +113,14 @@ struct alc_spec { /* * input MUX handling */ -static int alc_mux_enum_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) +static int alc_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; return snd_hda_input_mux_info(spec->input_mux, uinfo); } -static int alc_mux_enum_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int alc_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; @@ -130,7 +130,7 @@ static int alc_mux_enum_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucon return 0; } -static int alc_mux_enum_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int alc_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; @@ -143,7 +143,7 @@ static int alc_mux_enum_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucon /* * channel mode setting */ -static int alc880_ch_mode_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) +static int alc880_ch_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; @@ -151,7 +151,7 @@ static int alc880_ch_mode_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *ui spec->num_channel_mode); } -static int alc880_ch_mode_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int alc880_ch_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; @@ -159,7 +159,7 @@ static int alc880_ch_mode_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *uc spec->num_channel_mode, spec->multiout.max_channels); } -static int alc880_ch_mode_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int alc880_ch_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; @@ -173,7 +173,7 @@ static int alc880_ch_mode_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *uc * supported, so VrefEn can't be controlled using these functions as they * stand. */ -static int alc_pinctl_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) +static int alc_pinctl_switch_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; uinfo->count = 1; @@ -182,7 +182,7 @@ static int alc_pinctl_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t return 0; } -static int alc_pinctl_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int alc_pinctl_switch_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; @@ -195,7 +195,7 @@ static int alc_pinctl_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t return 0; } -static int alc_pinctl_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int alc_pinctl_switch_put(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; @@ -284,7 +284,7 @@ static struct hda_channel_mode alc880_threestack_modes[2] = { { 6, alc880_threestack_ch6_init }, }; -static snd_kcontrol_new_t alc880_three_stack_mixer[] = { +static struct snd_kcontrol_new alc880_three_stack_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT), @@ -315,7 +315,7 @@ static snd_kcontrol_new_t alc880_three_stack_mixer[] = { }; /* capture mixer elements */ -static snd_kcontrol_new_t alc880_capture_mixer[] = { +static struct snd_kcontrol_new alc880_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT), @@ -339,7 +339,7 @@ static snd_kcontrol_new_t alc880_capture_mixer[] = { }; /* capture mixer elements (in case NID 0x07 not available) */ -static snd_kcontrol_new_t alc880_capture_alt_mixer[] = { +static struct snd_kcontrol_new alc880_capture_alt_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), @@ -371,7 +371,7 @@ static snd_kcontrol_new_t alc880_capture_alt_mixer[] = { */ /* additional mixers to alc880_three_stack_mixer */ -static snd_kcontrol_new_t alc880_five_stack_mixer[] = { +static struct snd_kcontrol_new alc880_five_stack_mixer[] = { HDA_CODEC_VOLUME("Side Playback Volume", 0x0d, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Side Playback Switch", 0x0d, 2, HDA_INPUT), { } /* end */ @@ -428,7 +428,7 @@ static struct hda_channel_mode alc880_sixstack_modes[1] = { { 8, NULL }, }; -static snd_kcontrol_new_t alc880_six_stack_mixer[] = { +static struct snd_kcontrol_new alc880_six_stack_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), @@ -496,7 +496,7 @@ static struct hda_channel_mode alc880_w810_modes[1] = { }; /* Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, HP = 0x1b */ -static snd_kcontrol_new_t alc880_w810_base_mixer[] = { +static struct snd_kcontrol_new alc880_w810_base_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), @@ -527,7 +527,7 @@ static struct hda_channel_mode alc880_2_jack_modes[1] = { { 2, NULL } }; -static snd_kcontrol_new_t alc880_z71v_mixer[] = { +static struct snd_kcontrol_new alc880_z71v_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT), @@ -553,7 +553,7 @@ static hda_nid_t alc880_f1734_dac_nids[1] = { }; #define ALC880_F1734_HP_DAC 0x02 -static snd_kcontrol_new_t alc880_f1734_mixer[] = { +static struct snd_kcontrol_new alc880_f1734_mixer[] = { HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Internal Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), @@ -578,7 +578,7 @@ static snd_kcontrol_new_t alc880_f1734_mixer[] = { #define alc880_asus_dac_nids alc880_w810_dac_nids /* identical with w810 */ #define alc880_asus_modes alc880_threestack_modes /* 2/6 channel mode */ -static snd_kcontrol_new_t alc880_asus_mixer[] = { +static struct snd_kcontrol_new alc880_asus_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), @@ -613,14 +613,14 @@ static snd_kcontrol_new_t alc880_asus_mixer[] = { */ /* additional mixers to alc880_asus_mixer */ -static snd_kcontrol_new_t alc880_asus_w1v_mixer[] = { +static struct snd_kcontrol_new alc880_asus_w1v_mixer[] = { HDA_CODEC_VOLUME("Line2 Playback Volume", 0x0b, 0x03, HDA_INPUT), HDA_CODEC_MUTE("Line2 Playback Switch", 0x0b, 0x03, HDA_INPUT), { } /* end */ }; /* additional mixers to alc880_asus_mixer */ -static snd_kcontrol_new_t alc880_pcbeep_mixer[] = { +static struct snd_kcontrol_new alc880_pcbeep_mixer[] = { HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ @@ -974,7 +974,7 @@ static int alc_resume(struct hda_codec *codec) */ static int alc880_playback_pcm_open(struct hda_pcm_stream *hinfo, struct hda_codec *codec, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { struct alc_spec *spec = codec->spec; return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream); @@ -984,7 +984,7 @@ static int alc880_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, unsigned int format, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { struct alc_spec *spec = codec->spec; return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag, @@ -993,7 +993,7 @@ static int alc880_playback_pcm_prepare(struct hda_pcm_stream *hinfo, static int alc880_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, struct hda_codec *codec, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { struct alc_spec *spec = codec->spec; return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); @@ -1004,7 +1004,7 @@ static int alc880_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, */ static int alc880_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, struct hda_codec *codec, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { struct alc_spec *spec = codec->spec; return snd_hda_multi_out_dig_open(codec, &spec->multiout); @@ -1012,7 +1012,7 @@ static int alc880_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, static int alc880_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, struct hda_codec *codec, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { struct alc_spec *spec = codec->spec; return snd_hda_multi_out_dig_close(codec, &spec->multiout); @@ -1025,7 +1025,7 @@ static int alc880_capture_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, unsigned int format, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { struct alc_spec *spec = codec->spec; @@ -1036,7 +1036,7 @@ static int alc880_capture_pcm_prepare(struct hda_pcm_stream *hinfo, static int alc880_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, struct hda_codec *codec, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { struct alc_spec *spec = codec->spec; @@ -1185,7 +1185,7 @@ static struct hda_channel_mode alc880_test_modes[4] = { { 8, NULL }, }; -static int alc_test_pin_ctl_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) +static int alc_test_pin_ctl_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = { "N/A", "Line Out", "HP Out", @@ -1200,7 +1200,7 @@ static int alc_test_pin_ctl_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * return 0; } -static int alc_test_pin_ctl_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int alc_test_pin_ctl_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = (hda_nid_t)kcontrol->private_value; @@ -1226,7 +1226,7 @@ static int alc_test_pin_ctl_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t * return 0; } -static int alc_test_pin_ctl_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int alc_test_pin_ctl_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = (hda_nid_t)kcontrol->private_value; @@ -1252,7 +1252,7 @@ static int alc_test_pin_ctl_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t * return 0; } -static int alc_test_pin_src_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) +static int alc_test_pin_src_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = { "Front", "Surround", "CLFE", "Side" @@ -1266,7 +1266,7 @@ static int alc_test_pin_src_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * return 0; } -static int alc_test_pin_src_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int alc_test_pin_src_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = (hda_nid_t)kcontrol->private_value; @@ -1277,7 +1277,7 @@ static int alc_test_pin_src_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t * return 0; } -static int alc_test_pin_src_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int alc_test_pin_src_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = (hda_nid_t)kcontrol->private_value; @@ -1310,7 +1310,7 @@ static int alc_test_pin_src_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t * .private_value = nid \ } -static snd_kcontrol_new_t alc880_test_mixer[] = { +static struct snd_kcontrol_new alc880_test_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CLFE Playback Volume", 0x0e, 0x0, HDA_OUTPUT), @@ -1527,7 +1527,7 @@ static struct hda_board_config alc880_cfg_tbl[] = { * configuration template - to be copied to the spec instance */ struct alc_config_preset { - snd_kcontrol_new_t *mixers[4]; + struct snd_kcontrol_new *mixers[4]; const struct hda_verb *init_verbs[4]; unsigned int num_dacs; hda_nid_t *dac_nids; @@ -1698,7 +1698,7 @@ enum { ALC_CTL_WIDGET_MUTE, ALC_CTL_BIND_MUTE, }; -static snd_kcontrol_new_t alc880_control_templates[] = { +static struct snd_kcontrol_new alc880_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), HDA_BIND_MUTE(NULL, 0, 0, 0), @@ -1707,7 +1707,7 @@ static snd_kcontrol_new_t alc880_control_templates[] = { /* add dynamic controls */ static int add_control(struct alc_spec *spec, int type, const char *name, unsigned long val) { - snd_kcontrol_new_t *knew; + struct snd_kcontrol_new *knew; if (spec->num_kctl_used >= spec->num_kctl_alloc) { int num = spec->num_kctl_alloc + NUM_CONTROL_ALLOC; @@ -2168,7 +2168,7 @@ static struct hda_channel_mode alc260_modes[1] = { { 2, NULL }, }; -static snd_kcontrol_new_t alc260_base_mixer[] = { +static struct snd_kcontrol_new alc260_base_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), @@ -2197,7 +2197,7 @@ static snd_kcontrol_new_t alc260_base_mixer[] = { { } /* end */ }; -static snd_kcontrol_new_t alc260_hp_mixer[] = { +static struct snd_kcontrol_new alc260_hp_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), @@ -2224,7 +2224,7 @@ static snd_kcontrol_new_t alc260_hp_mixer[] = { { } /* end */ }; -static snd_kcontrol_new_t alc260_fujitsu_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), @@ -2498,7 +2498,7 @@ static struct hda_input_mux alc882_capture_source = { #define alc882_mux_enum_info alc_mux_enum_info #define alc882_mux_enum_get alc_mux_enum_get -static int alc882_mux_enum_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int alc882_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; @@ -2526,7 +2526,7 @@ static int alc882_mux_enum_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *u /* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17 * Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b */ -static snd_kcontrol_new_t alc882_base_mixer[] = { +static struct snd_kcontrol_new alc882_base_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index 12b5de29f42f..8f8840e6002b 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -95,8 +95,8 @@ struct si3054_spec { #define PRIVATE_REG(val) ((val>>16)&0xffff) #define PRIVATE_MASK(val) (val&0xffff) -static int si3054_switch_info(snd_kcontrol_t *kcontrol, - snd_ctl_elem_info_t *uinfo) +static int si3054_switch_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; uinfo->count = 1; @@ -105,8 +105,8 @@ static int si3054_switch_info(snd_kcontrol_t *kcontrol, return 0; } -static int si3054_switch_get(snd_kcontrol_t *kcontrol, - snd_ctl_elem_value_t *uvalue) +static int si3054_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uvalue) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); u16 reg = PRIVATE_REG(kcontrol->private_value); @@ -115,8 +115,8 @@ static int si3054_switch_get(snd_kcontrol_t *kcontrol, return 0; } -static int si3054_switch_put(snd_kcontrol_t *kcontrol, - snd_ctl_elem_value_t *uvalue) +static int si3054_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uvalue) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); u16 reg = PRIVATE_REG(kcontrol->private_value); @@ -138,7 +138,7 @@ static int si3054_switch_put(snd_kcontrol_t *kcontrol, } -static snd_kcontrol_new_t si3054_modem_mixer[] = { +static struct snd_kcontrol_new si3054_modem_mixer[] = { SI3054_KCONTROL("Off-hook Switch", SI3054_GPIO_CONTROL, SI3054_GPIO_OH), SI3054_KCONTROL("Caller ID Switch", SI3054_GPIO_CONTROL, SI3054_GPIO_CID), {} @@ -158,7 +158,7 @@ static int si3054_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, unsigned int format, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { u16 val; @@ -175,10 +175,10 @@ static int si3054_pcm_prepare(struct hda_pcm_stream *hinfo, static int si3054_pcm_open(struct hda_pcm_stream *hinfo, struct hda_codec *codec, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { static unsigned int rates[] = { 8000, 9600, 16000 }; - static snd_pcm_hw_constraint_list_t hw_constraints_rates = { + static struct snd_pcm_hw_constraint_list hw_constraints_rates = { .count = ARRAY_SIZE(rates), .list = rates, .mask = 0, diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 33a8adaea768..d8d68f5b6131 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -41,7 +41,7 @@ #define STAC_UNSOL_ENABLE (AC_USRSP_EN | STAC_HP_EVENT) struct sigmatel_spec { - snd_kcontrol_new_t *mixers[4]; + struct snd_kcontrol_new *mixers[4]; unsigned int num_mixers; unsigned int surr_switch: 1; @@ -66,7 +66,7 @@ struct sigmatel_spec { /* codec specific stuff */ struct hda_verb *init; - snd_kcontrol_new_t *mixer; + struct snd_kcontrol_new *mixer; /* capture source */ struct hda_input_mux *input_mux; @@ -81,7 +81,7 @@ struct sigmatel_spec { /* dynamic controls and input_mux */ struct auto_pin_cfg autocfg; unsigned int num_kctl_alloc, num_kctl_used; - snd_kcontrol_new_t *kctl_alloc; + struct snd_kcontrol_new *kctl_alloc; struct hda_input_mux private_imux; }; @@ -116,14 +116,14 @@ static hda_nid_t stac922x_pin_nids[10] = { }; #endif -static int stac92xx_mux_enum_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) +static int stac92xx_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; return snd_hda_input_mux_info(spec->input_mux, uinfo); } -static int stac92xx_mux_enum_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int stac92xx_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; @@ -133,7 +133,7 @@ static int stac92xx_mux_enum_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t return 0; } -static int stac92xx_mux_enum_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int stac92xx_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; @@ -157,7 +157,7 @@ static struct hda_verb stac922x_core_init[] = { static int stac922x_channel_modes[3] = {2, 6, 8}; -static int stac922x_ch_mode_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) +static int stac922x_ch_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; @@ -172,7 +172,7 @@ static int stac922x_ch_mode_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * return 0; } -static int stac922x_ch_mode_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int stac922x_ch_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; @@ -181,7 +181,7 @@ static int stac922x_ch_mode_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t * return 0; } -static int stac922x_ch_mode_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +static int stac922x_ch_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; @@ -198,7 +198,7 @@ static int stac922x_ch_mode_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t * return 1; } -static snd_kcontrol_new_t stac9200_mixer[] = { +static struct snd_kcontrol_new stac9200_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0xb, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT), { @@ -216,7 +216,7 @@ static snd_kcontrol_new_t stac9200_mixer[] = { }; /* This needs to be generated dynamically based on sequence */ -static snd_kcontrol_new_t stac922x_mixer[] = { +static struct snd_kcontrol_new stac922x_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Input Source", @@ -231,7 +231,7 @@ static snd_kcontrol_new_t stac922x_mixer[] = { { } /* end */ }; -static snd_kcontrol_new_t stac922x_ch_mode_mixer[] = { +static struct snd_kcontrol_new stac922x_ch_mode_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Channel Mode", @@ -320,7 +320,7 @@ static void stac92xx_set_config_regs(struct hda_codec *codec) */ static int stac92xx_playback_pcm_open(struct hda_pcm_stream *hinfo, struct hda_codec *codec, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { struct sigmatel_spec *spec = codec->spec; return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream); @@ -333,7 +333,7 @@ static int stac92xx_playback_pcm_open(struct hda_pcm_stream *hinfo, static int stac92xx_multi_out_analog_prepare(struct hda_codec *codec, struct hda_multi_out *mout, unsigned int stream_tag, unsigned int format, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { hda_nid_t *nids = mout->dac_nids; int chs = substream->runtime->channels; @@ -380,7 +380,7 @@ static int stac92xx_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, unsigned int format, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { struct sigmatel_spec *spec = codec->spec; return stac92xx_multi_out_analog_prepare(codec, &spec->multiout, stream_tag, @@ -389,7 +389,7 @@ static int stac92xx_playback_pcm_prepare(struct hda_pcm_stream *hinfo, static int stac92xx_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, struct hda_codec *codec, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { struct sigmatel_spec *spec = codec->spec; return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); @@ -400,7 +400,7 @@ static int stac92xx_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, */ static int stac92xx_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, struct hda_codec *codec, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { struct sigmatel_spec *spec = codec->spec; return snd_hda_multi_out_dig_open(codec, &spec->multiout); @@ -408,7 +408,7 @@ static int stac92xx_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, static int stac92xx_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, struct hda_codec *codec, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { struct sigmatel_spec *spec = codec->spec; return snd_hda_multi_out_dig_close(codec, &spec->multiout); @@ -422,7 +422,7 @@ static int stac92xx_capture_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, unsigned int format, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { struct sigmatel_spec *spec = codec->spec; @@ -433,7 +433,7 @@ static int stac92xx_capture_pcm_prepare(struct hda_pcm_stream *hinfo, static int stac92xx_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, struct hda_codec *codec, - snd_pcm_substream_t *substream) + struct snd_pcm_substream *substream) { struct sigmatel_spec *spec = codec->spec; @@ -516,7 +516,7 @@ enum { STAC_CTL_WIDGET_MUTE, }; -static snd_kcontrol_new_t stac92xx_control_templates[] = { +static struct snd_kcontrol_new stac92xx_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), }; @@ -524,7 +524,7 @@ static snd_kcontrol_new_t stac92xx_control_templates[] = { /* add dynamic controls */ static int stac92xx_add_control(struct sigmatel_spec *spec, int type, const char *name, unsigned long val) { - snd_kcontrol_new_t *knew; + struct snd_kcontrol_new *knew; if (spec->num_kctl_used >= spec->num_kctl_alloc) { int num = spec->num_kctl_alloc + NUM_CONTROL_ALLOC; -- cgit v1.2.3 From a98f90fd826913519c3f704ea24fb9bea1e0e494 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 17 Nov 2005 14:59:02 +0100 Subject: [ALSA] Remove xxx_t typedefs: HDA-Intel Modules: HDA Intel driver Remove xxx_t typedefs from the HDA-Intel driver. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 156 ++++++++++++++++++++++------------------------ 1 file changed, 76 insertions(+), 80 deletions(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 429ef3810b16..abdbd96d4c06 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -238,11 +238,7 @@ enum { /* */ -typedef struct snd_azx azx_t; -typedef struct snd_azx_rb azx_rb_t; -typedef struct snd_azx_dev azx_dev_t; - -struct snd_azx_dev { +struct azx_dev { u32 *bdl; /* virtual address of the BDL */ dma_addr_t bdl_addr; /* physical address of the BDL */ volatile u32 *posbuf; /* position buffer pointer */ @@ -258,7 +254,7 @@ struct snd_azx_dev { u32 sd_int_sta_mask; /* stream int status mask */ /* pcm support */ - snd_pcm_substream_t *substream; /* assigned substream, set in PCM open */ + struct snd_pcm_substream *substream; /* assigned substream, set in PCM open */ unsigned int format_val; /* format value to be set in the controller and the codec */ unsigned char stream_tag; /* assigned stream */ unsigned char index; /* stream index */ @@ -269,7 +265,7 @@ struct snd_azx_dev { }; /* CORB/RIRB */ -struct snd_azx_rb { +struct azx_rb { u32 *buf; /* CORB/RIRB buffer * Each CORB entry is 4byte, RIRB is 8byte */ @@ -280,8 +276,8 @@ struct snd_azx_rb { u32 res; /* last read value */ }; -struct snd_azx { - snd_card_t *card; +struct azx { + struct snd_card *card; struct pci_dev *pci; /* chip type specific */ @@ -302,19 +298,19 @@ struct snd_azx { struct semaphore open_mutex; /* streams (x num_streams) */ - azx_dev_t *azx_dev; + struct azx_dev *azx_dev; /* PCM */ unsigned int pcm_devs; - snd_pcm_t *pcm[AZX_MAX_PCMS]; + struct snd_pcm *pcm[AZX_MAX_PCMS]; /* HD codec */ unsigned short codec_mask; struct hda_bus *bus; /* CORB/RIRB */ - azx_rb_t corb; - azx_rb_t rirb; + struct azx_rb corb; + struct azx_rb rirb; /* BDL, CORB/RIRB and position buffers */ struct snd_dma_buffer bdl; @@ -375,7 +371,7 @@ static char *driver_short_names[] __devinitdata = { readb((dev)->sd_addr + ICH6_REG_##reg) /* for pcm support */ -#define get_azx_dev(substream) (azx_dev_t*)(substream->runtime->private_data) +#define get_azx_dev(substream) (substream->runtime->private_data) /* Get the upper 32bit of the given dma_addr_t * Compiler should optimize and eliminate the code if dma_addr_t is 32bit @@ -391,7 +387,7 @@ static char *driver_short_names[] __devinitdata = { /* * CORB / RIRB interface */ -static int azx_alloc_cmd_io(azx_t *chip) +static int azx_alloc_cmd_io(struct azx *chip) { int err; @@ -405,7 +401,7 @@ static int azx_alloc_cmd_io(azx_t *chip) return 0; } -static void azx_init_cmd_io(azx_t *chip) +static void azx_init_cmd_io(struct azx *chip) { /* CORB set up */ chip->corb.addr = chip->rb.addr; @@ -443,7 +439,7 @@ static void azx_init_cmd_io(azx_t *chip) chip->rirb.rp = chip->rirb.cmds = 0; } -static void azx_free_cmd_io(azx_t *chip) +static void azx_free_cmd_io(struct azx *chip) { /* disable ringbuffer DMAs */ azx_writeb(chip, RIRBCTL, 0); @@ -454,7 +450,7 @@ static void azx_free_cmd_io(azx_t *chip) static int azx_send_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, unsigned int verb, unsigned int para) { - azx_t *chip = codec->bus->private_data; + struct azx *chip = codec->bus->private_data; unsigned int wp; u32 val; @@ -481,7 +477,7 @@ static int azx_send_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, #define ICH6_RIRB_EX_UNSOL_EV (1<<4) /* retrieve RIRB entry - called from interrupt handler */ -static void azx_update_rirb(azx_t *chip) +static void azx_update_rirb(struct azx *chip) { unsigned int rp, wp; u32 res, res_ex; @@ -510,7 +506,7 @@ static void azx_update_rirb(azx_t *chip) /* receive a response */ static unsigned int azx_get_response(struct hda_codec *codec) { - azx_t *chip = codec->bus->private_data; + struct azx *chip = codec->bus->private_data; int timeout = 50; while (chip->rirb.cmds) { @@ -546,7 +542,7 @@ static unsigned int azx_get_response(struct hda_codec *codec) static int azx_send_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, unsigned int verb, unsigned int para) { - azx_t *chip = codec->bus->private_data; + struct azx *chip = codec->bus->private_data; u32 val; int timeout = 50; @@ -574,7 +570,7 @@ 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) { - azx_t *chip = codec->bus->private_data; + struct azx *chip = codec->bus->private_data; int timeout = 50; while (timeout--) { @@ -592,7 +588,7 @@ static unsigned int azx_get_response(struct hda_codec *codec) #endif /* USE_CORB_RIRB */ /* reset codec link */ -static int azx_reset(azx_t *chip) +static int azx_reset(struct azx *chip) { int count; @@ -642,7 +638,7 @@ static int azx_reset(azx_t *chip) */ /* enable interrupts */ -static void azx_int_enable(azx_t *chip) +static void azx_int_enable(struct azx *chip) { /* enable controller CIE and GIE */ azx_writel(chip, INTCTL, azx_readl(chip, INTCTL) | @@ -650,13 +646,13 @@ static void azx_int_enable(azx_t *chip) } /* disable interrupts */ -static void azx_int_disable(azx_t *chip) +static void azx_int_disable(struct azx *chip) { int i; /* disable interrupts in stream descriptor */ for (i = 0; i < chip->num_streams; i++) { - azx_dev_t *azx_dev = &chip->azx_dev[i]; + struct azx_dev *azx_dev = &chip->azx_dev[i]; azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) & ~SD_INT_MASK); } @@ -670,13 +666,13 @@ static void azx_int_disable(azx_t *chip) } /* clear interrupts */ -static void azx_int_clear(azx_t *chip) +static void azx_int_clear(struct azx *chip) { int i; /* clear stream status */ for (i = 0; i < chip->num_streams; i++) { - azx_dev_t *azx_dev = &chip->azx_dev[i]; + struct azx_dev *azx_dev = &chip->azx_dev[i]; azx_sd_writeb(azx_dev, SD_STS, SD_INT_MASK); } @@ -691,7 +687,7 @@ static void azx_int_clear(azx_t *chip) } /* start a stream */ -static void azx_stream_start(azx_t *chip, azx_dev_t *azx_dev) +static void azx_stream_start(struct azx *chip, struct azx_dev *azx_dev) { /* enable SIE */ azx_writeb(chip, INTCTL, @@ -702,7 +698,7 @@ static void azx_stream_start(azx_t *chip, azx_dev_t *azx_dev) } /* stop a stream */ -static void azx_stream_stop(azx_t *chip, azx_dev_t *azx_dev) +static void azx_stream_stop(struct azx *chip, struct azx_dev *azx_dev) { /* stop DMA */ azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) & @@ -717,7 +713,7 @@ static void azx_stream_stop(azx_t *chip, azx_dev_t *azx_dev) /* * initialize the chip */ -static void azx_init_chip(azx_t *chip) +static void azx_init_chip(struct azx *chip) { unsigned char reg; @@ -765,8 +761,8 @@ static void azx_init_chip(azx_t *chip) */ static irqreturn_t azx_interrupt(int irq, void* dev_id, struct pt_regs *regs) { - azx_t *chip = dev_id; - azx_dev_t *azx_dev; + struct azx *chip = dev_id; + struct azx_dev *azx_dev; u32 status; int i; @@ -814,7 +810,7 @@ static irqreturn_t azx_interrupt(int irq, void* dev_id, struct pt_regs *regs) /* * set up BDL entries */ -static void azx_setup_periods(azx_dev_t *azx_dev) +static void azx_setup_periods(struct azx_dev *azx_dev) { u32 *bdl = azx_dev->bdl; dma_addr_t dma_addr = azx_dev->substream->runtime->dma_addr; @@ -843,7 +839,7 @@ static void azx_setup_periods(azx_dev_t *azx_dev) /* * set up the SD for streaming */ -static int azx_setup_controller(azx_t *chip, azx_dev_t *azx_dev) +static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev) { unsigned char val; int timeout; @@ -903,7 +899,7 @@ static int azx_setup_controller(azx_t *chip, azx_dev_t *azx_dev) * Codec initialization */ -static int __devinit azx_codec_create(azx_t *chip, const char *model) +static int __devinit azx_codec_create(struct azx *chip, const char *model) { struct hda_bus_template bus_temp; int c, codecs, err; @@ -941,7 +937,7 @@ static int __devinit azx_codec_create(azx_t *chip, const char *model) */ /* assign a stream for the PCM */ -static inline azx_dev_t *azx_assign_device(azx_t *chip, int stream) +static inline struct azx_dev *azx_assign_device(struct azx *chip, int stream) { int dev, i, nums; if (stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -960,12 +956,12 @@ static inline azx_dev_t *azx_assign_device(azx_t *chip, int stream) } /* release the assigned stream */ -static inline void azx_release_device(azx_dev_t *azx_dev) +static inline void azx_release_device(struct azx_dev *azx_dev) { azx_dev->opened = 0; } -static snd_pcm_hardware_t azx_pcm_hw = { +static struct snd_pcm_hardware azx_pcm_hw = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | @@ -986,18 +982,18 @@ static snd_pcm_hardware_t azx_pcm_hw = { }; struct azx_pcm { - azx_t *chip; + struct azx *chip; struct hda_codec *codec; struct hda_pcm_stream *hinfo[2]; }; -static int azx_pcm_open(snd_pcm_substream_t *substream) +static int azx_pcm_open(struct snd_pcm_substream *substream) { struct azx_pcm *apcm = snd_pcm_substream_chip(substream); struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; - azx_t *chip = apcm->chip; - azx_dev_t *azx_dev; - snd_pcm_runtime_t *runtime = substream->runtime; + struct azx *chip = apcm->chip; + struct azx_dev *azx_dev; + struct snd_pcm_runtime *runtime = substream->runtime; unsigned long flags; int err; @@ -1029,12 +1025,12 @@ static int azx_pcm_open(snd_pcm_substream_t *substream) return 0; } -static int azx_pcm_close(snd_pcm_substream_t *substream) +static int azx_pcm_close(struct snd_pcm_substream *substream) { struct azx_pcm *apcm = snd_pcm_substream_chip(substream); struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; - azx_t *chip = apcm->chip; - azx_dev_t *azx_dev = get_azx_dev(substream); + struct azx *chip = apcm->chip; + struct azx_dev *azx_dev = get_azx_dev(substream); unsigned long flags; down(&chip->open_mutex); @@ -1048,15 +1044,15 @@ static int azx_pcm_close(snd_pcm_substream_t *substream) return 0; } -static int azx_pcm_hw_params(snd_pcm_substream_t *substream, snd_pcm_hw_params_t *hw_params) +static int azx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); } -static int azx_pcm_hw_free(snd_pcm_substream_t *substream) +static int azx_pcm_hw_free(struct snd_pcm_substream *substream) { struct azx_pcm *apcm = snd_pcm_substream_chip(substream); - azx_dev_t *azx_dev = get_azx_dev(substream); + struct azx_dev *azx_dev = get_azx_dev(substream); struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; /* reset BDL address */ @@ -1069,13 +1065,13 @@ static int azx_pcm_hw_free(snd_pcm_substream_t *substream) return snd_pcm_lib_free_pages(substream); } -static int azx_pcm_prepare(snd_pcm_substream_t *substream) +static int azx_pcm_prepare(struct snd_pcm_substream *substream) { struct azx_pcm *apcm = snd_pcm_substream_chip(substream); - azx_t *chip = apcm->chip; - azx_dev_t *azx_dev = get_azx_dev(substream); + struct azx *chip = apcm->chip; + struct azx_dev *azx_dev = get_azx_dev(substream); struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; - snd_pcm_runtime_t *runtime = substream->runtime; + struct snd_pcm_runtime *runtime = substream->runtime; azx_dev->bufsize = snd_pcm_lib_buffer_bytes(substream); azx_dev->fragsize = snd_pcm_lib_period_bytes(substream); @@ -1104,11 +1100,11 @@ static int azx_pcm_prepare(snd_pcm_substream_t *substream) azx_dev->format_val, substream); } -static int azx_pcm_trigger(snd_pcm_substream_t *substream, int cmd) +static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct azx_pcm *apcm = snd_pcm_substream_chip(substream); - azx_dev_t *azx_dev = get_azx_dev(substream); - azx_t *chip = apcm->chip; + struct azx_dev *azx_dev = get_azx_dev(substream); + struct azx *chip = apcm->chip; int err = 0; spin_lock(&chip->reg_lock); @@ -1139,11 +1135,11 @@ static int azx_pcm_trigger(snd_pcm_substream_t *substream, int cmd) return err; } -static snd_pcm_uframes_t azx_pcm_pointer(snd_pcm_substream_t *substream) +static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream) { struct azx_pcm *apcm = snd_pcm_substream_chip(substream); - azx_t *chip = apcm->chip; - azx_dev_t *azx_dev = get_azx_dev(substream); + struct azx *chip = apcm->chip; + struct azx_dev *azx_dev = get_azx_dev(substream); unsigned int pos; if (chip->position_fix == POS_FIX_POSBUF) { @@ -1185,7 +1181,7 @@ static snd_pcm_uframes_t azx_pcm_pointer(snd_pcm_substream_t *substream) return bytes_to_frames(substream->runtime, pos); } -static snd_pcm_ops_t azx_pcm_ops = { +static struct snd_pcm_ops azx_pcm_ops = { .open = azx_pcm_open, .close = azx_pcm_close, .ioctl = snd_pcm_lib_ioctl, @@ -1196,16 +1192,16 @@ static snd_pcm_ops_t azx_pcm_ops = { .pointer = azx_pcm_pointer, }; -static void azx_pcm_free(snd_pcm_t *pcm) +static void azx_pcm_free(struct snd_pcm *pcm) { kfree(pcm->private_data); } -static int __devinit create_codec_pcm(azx_t *chip, struct hda_codec *codec, +static int __devinit create_codec_pcm(struct azx *chip, struct hda_codec *codec, struct hda_pcm *cpcm, int pcm_dev) { int err; - snd_pcm_t *pcm; + struct snd_pcm *pcm; struct azx_pcm *apcm; snd_assert(cpcm->stream[0].substreams || cpcm->stream[1].substreams, return -EINVAL); @@ -1239,7 +1235,7 @@ static int __devinit create_codec_pcm(azx_t *chip, struct hda_codec *codec, return 0; } -static int __devinit azx_pcm_create(azx_t *chip) +static int __devinit azx_pcm_create(struct azx *chip) { struct list_head *p; struct hda_codec *codec; @@ -1291,7 +1287,7 @@ static int __devinit azx_pcm_create(azx_t *chip) /* * mixer creation - all stuff is implemented in hda module */ -static int __devinit azx_mixer_create(azx_t *chip) +static int __devinit azx_mixer_create(struct azx *chip) { return snd_hda_build_controls(chip->bus); } @@ -1300,7 +1296,7 @@ static int __devinit azx_mixer_create(azx_t *chip) /* * initialize SD streams */ -static int __devinit azx_init_stream(azx_t *chip) +static int __devinit azx_init_stream(struct azx *chip) { int i; @@ -1309,7 +1305,7 @@ static int __devinit azx_init_stream(azx_t *chip) */ for (i = 0; i < chip->num_streams; i++) { unsigned int off = sizeof(u32) * (i * AZX_MAX_FRAG * 4); - azx_dev_t *azx_dev = &chip->azx_dev[i]; + struct azx_dev *azx_dev = &chip->azx_dev[i]; azx_dev->bdl = (u32 *)(chip->bdl.area + off); azx_dev->bdl_addr = chip->bdl.addr + off; azx_dev->posbuf = (volatile u32 *)(chip->posbuf.area + i * 8); @@ -1330,9 +1326,9 @@ static int __devinit azx_init_stream(azx_t *chip) /* * power management */ -static int azx_suspend(snd_card_t *card, pm_message_t state) +static int azx_suspend(struct snd_card *card, pm_message_t state) { - azx_t *chip = card->pm_private_data; + struct azx *chip = card->pm_private_data; int i; for (i = 0; i < chip->pcm_devs; i++) @@ -1344,9 +1340,9 @@ static int azx_suspend(snd_card_t *card, pm_message_t state) return 0; } -static int azx_resume(snd_card_t *card) +static int azx_resume(struct snd_card *card) { - azx_t *chip = card->pm_private_data; + struct azx *chip = card->pm_private_data; pci_enable_device(chip->pci); pci_set_master(chip->pci); @@ -1360,7 +1356,7 @@ static int azx_resume(snd_card_t *card) /* * destructor */ -static int azx_free(azx_t *chip) +static int azx_free(struct azx *chip) { if (chip->initialized) { int i; @@ -1402,7 +1398,7 @@ static int azx_free(azx_t *chip) return 0; } -static int azx_dev_free(snd_device_t *device) +static int azx_dev_free(struct snd_device *device) { return azx_free(device->device_data); } @@ -1410,13 +1406,13 @@ static int azx_dev_free(snd_device_t *device) /* * constructor */ -static int __devinit azx_create(snd_card_t *card, struct pci_dev *pci, +static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, int posfix, int driver_type, - azx_t **rchip) + struct azx **rchip) { - azx_t *chip; + struct azx *chip; int err = 0; - static snd_device_ops_t ops = { + static struct snd_device_ops ops = { .dev_free = azx_dev_free, }; @@ -1548,8 +1544,8 @@ static int __devinit azx_create(snd_card_t *card, struct pci_dev *pci, static int __devinit azx_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { - snd_card_t *card; - azx_t *chip; + struct snd_card *card; + struct azx *chip; int err = 0; card = snd_card_new(index, id, THIS_MODULE, 0); -- cgit v1.2.3 From fd66e0d0591dd12eb0bea1e9f3aa194bb93cebbd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 17 Nov 2005 15:31:34 +0100 Subject: [ALSA] hda-codec - Add AD1988 support Modules: HDA Codec driver Add AD1988 codec support to hda-codec driver. Still experimental, and no BIOS configuration parser is implemented yet. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 785 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 785 insertions(+) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 1f371fe6b92f..25116a883ca6 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -41,6 +41,7 @@ struct ad198x_spec { * max_channels, dacs must be set * dig_out_nid and hp_nid are optional */ + unsigned int cur_eapd; /* capture */ unsigned int num_adc_nids; @@ -856,6 +857,789 @@ static int patch_ad1981(struct hda_codec *codec) } +/* + * AD1988 + * + * Output pins and routes + * + * Pin Mix Sel DAC + * port-A 0x11 (mute/hp) <- 0x22 <- 0x37 <- 03/04/06 + * port-B 0x14 (mute/hp) <- 0x2b <- 0x30 <- 03/04/06 + * port-C 0x15 (mute) <- 0x2c <- 0x31 <- 05/0a + * port-D 0x12 (mute/hp) <- 0x29 <- 04 + * port-E 0x17 (mute/hp) <- 0x26 <- 0x32 <- 05/0a + * port-F 0x16 (mute) <- 0x2a <- 06 + * port-G 0x24 (mute) <- 0x27 <- 05 + * port-H 0x25 (mute) <- 0x28 <- 0a + * mono 0x13 (mute/amp)<- 0x1e <- 0x36 <- 03/04/06 + * + * + * Input pins and routes + * + * pin boost mix input # / adc input # + * port-A 0x11 -> 0x38 -> mix 2, ADC 0 + * port-B 0x14 -> 0x39 -> mix 0, ADC 1 + * port-C 0x15 -> 0x3a -> 33:0 - mix 1, ADC 2 + * port-D 0x12 -> 0x3d -> mix 3, ADC 8 + * port-E 0x17 -> 0x3c -> 34:0 - mix 4, ADC 4 + * port-F 0x16 -> 0x3b -> mix 5, ADC 3 + * port-G 0x24 -> N/A -> 33:1 - mix 1, 34:1 - mix 4, ADC 6 + * port-H 0x25 -> N/A -> 33:2 - mix 1, 34:2 - mix 4, ADC 7 + * + * + * DAC assignment + * front DAC - 04 + * surr DAC - 06 + * CLFE DAC - 05 + * side DAC - 0a + * opt DAC - 03 + * + * Inputs of Analog Mix (0x20) + * 0:Port-B (front mic) + * 1:Port-C/G/H (line-in) + * 2:Port-A + * 3:Port-D (line-in/2) + * 4:Port-E/G/H (mic-in) + * 5:Port-F (mic2-in) + * 6:CD + * 7:Beep + * + * ADC selection + * 0:Port-A + * 1:Port-B (front mic-in) + * 2:Port-C (line-in) + * 3:Port-F (mic2-in) + * 4:Port-E (mic-in) + * 5:CD + * 6:Port-G + * 7:Port-H + * 8:Port-D (line-in/2) + * 9:Mix + * + * Proposed pin assignments by the datasheet + * + * 6-stack + * Port-A front headphone + * B front mic-in + * C rear line-in + * D rear front-out + * E rear mic-in + * F rear surround + * G rear CLFE + * H rear side + * + * 3-stack + * Port-A front headphone + * B front mic + * C rear line-in/surround + * D rear front-out + * E rear mic-in/CLFE + * + * laptop + * Port-A headphone + * B mic-in + * C docking station + * D internal speaker (with EAPD) + * E/F quad mic array + */ + + +/* models */ +enum { + AD1988_6STACK, + AD1988_6STACK_DIG, + AD1988_3STACK, + AD1988_3STACK_DIG, + AD1988_LAPTOP, + AD1988_LAPTOP_DIG, + AD1988_MODEL_LAST, +}; + + +/* + * mixers + */ + +static hda_nid_t ad1988_dac_nids[4] = { + 0x04, 0x06, 0x05, 0x0a +}; + +static hda_nid_t ad1988_adc_nids[3] = { + 0x08, 0x09, 0x0f +}; + +#define AD1988_SPDIF_OUT 0x02 +#define AD1988_SPDIF_IN 0x07 + +static struct hda_input_mux ad1988_6stack_capture_source = { + .num_items = 5, + .items = { + { "Front Mic", 0x0 }, + { "Line", 0x1 }, + { "Mic", 0x4 }, + { "CD", 0x5 }, + { "Mix", 0x9 }, + }, +}; + +static struct hda_input_mux ad1988_laptop_capture_source = { + .num_items = 3, + .items = { + { "Mic/Line", 0x0 }, + { "CD", 0x5 }, + { "Mix", 0x9 }, + }, +}; + +/* + */ +static int ad198x_ch_mode_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ad198x_spec *spec = codec->spec; + return snd_hda_ch_mode_info(codec, uinfo, spec->channel_mode, + spec->num_channel_mode); +} + +static int ad198x_ch_mode_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; + return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode, + spec->num_channel_mode, spec->multiout.max_channels); +} + +static int ad198x_ch_mode_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; + return snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode, + 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_mixers[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x06, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Side Playback Volume", 0x0a, 0x0, HDA_OUTPUT), + + HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x2a, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x27, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x27, 2, 2, HDA_INPUT), + HDA_BIND_MUTE("Side Playback Switch", 0x28, 2, HDA_INPUT), + HDA_BIND_MUTE("Headphone Playback Switch", 0x22, 2, HDA_INPUT), + HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT), + + HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT), + + HDA_CODEC_VOLUME("Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Front Mic Boost", 0x39, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x3c, 0x0, HDA_OUTPUT), + + { } /* end */ +}; + +/* 3-stack mode */ +static struct snd_kcontrol_new ad1988_3stack_mixers[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x06, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT), + + HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x2a, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x27, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x27, 2, 2, HDA_INPUT), + HDA_BIND_MUTE("Headphone Playback Switch", 0x22, 2, HDA_INPUT), + HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT), + + HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT), + + HDA_CODEC_VOLUME("Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Front Mic Boost", 0x39, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x3c, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Channel Mode", + .info = ad198x_ch_mode_info, + .get = ad198x_ch_mode_get, + .put = ad198x_ch_mode_put, + }, + + { } /* end */ +}; + +/* laptop mode */ +static struct snd_kcontrol_new ad1988_laptop_mixers[] = { + HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x29, 0x0, HDA_INPUT), + HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT), + + HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT), + + HDA_CODEC_VOLUME("Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Mic Boost", 0x39, 0x0, HDA_OUTPUT), + + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "External Amplifier", + .info = ad1988_eapd_info, + .get = ad1988_eapd_get, + .put = ad1988_eapd_put, + }, + + { } /* end */ +}; + +/* capture */ +static struct snd_kcontrol_new ad1988_capture_mixers[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x0e, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x0e, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + * FIXME: the controls appear in the "playback" view! + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 3, + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + { } /* end */ +}; + +static int ad1988_spdif_playback_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[] = { + "PCM", "ADC1", "ADC2", "ADC3" + }; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 4; + if (uinfo->value.enumerated.item >= 4) + uinfo->value.enumerated.item = 3; + strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); + return 0; +} + +static int ad1988_spdif_playback_source_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int sel; + + sel = snd_hda_codec_read(codec, 0x02, 0, AC_VERB_GET_CONNECT_SEL, 0); + if (sel > 0) { + sel = snd_hda_codec_read(codec, 0x0b, 0, AC_VERB_GET_CONNECT_SEL, 0); + if (sel <= 3) + sel++; + else + sel = 0; + } + ucontrol->value.enumerated.item[0] = sel; + return 0; +} + +static int ad1988_spdif_playback_source_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int sel; + int change; + + sel = snd_hda_codec_read(codec, 0x02, 0, AC_VERB_GET_CONNECT_SEL, 0); + if (! ucontrol->value.enumerated.item[0]) { + change = sel != 0; + if (change) + snd_hda_codec_write(codec, 0x02, 0, AC_VERB_SET_CONNECT_SEL, 0); + } else { + change = sel == 0; + if (change) + snd_hda_codec_write(codec, 0x02, 0, AC_VERB_SET_CONNECT_SEL, 1); + sel = snd_hda_codec_read(codec, 0x0b, 0, AC_VERB_GET_CONNECT_SEL, 0) + 1; + change |= sel == ucontrol->value.enumerated.item[0]; + if (change) + snd_hda_codec_write(codec, 0x02, 0, AC_VERB_SET_CONNECT_SEL, + ucontrol->value.enumerated.item[0] - 1); + } + return change; +} + +static struct snd_kcontrol_new ad1988_spdif_out_mixers[] = { + HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "IEC958 Playback Source", + .info = ad1988_spdif_playback_source_info, + .get = ad1988_spdif_playback_source_get, + .put = ad1988_spdif_playback_source_put, + }, + { } /* end */ +}; + +static struct snd_kcontrol_new ad1988_spdif_in_mixers[] = { + HDA_CODEC_VOLUME("IEC958 Capture Volume", 0x1c, 0x0, HDA_INPUT), + { } /* end */ +}; + + +/* + * initialization verbs + */ + +/* + * for 6-stack (+dig) + */ +static struct hda_verb ad1988_6stack_init_verbs[] = { + /* Front, Surround, CLFE, side DAC; mute as default */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Port-A front headphon path */ + {0x37, AC_VERB_SET_CONNECT_SEL, 0x01}, /* DAC1:04h */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Port-D line-out path */ + {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* Port-F surround path */ + {0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* Port-G CLFE path */ + {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* Port-H side path */ + {0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* Mono out path */ + {0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */ + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */ + /* Port-B front mic-in path */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + /* Port-C line-in path */ + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x33, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* Port-E mic-in path */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x34, AC_VERB_SET_CONNECT_SEL, 0x0}, + + { } +}; + +static struct hda_verb ad1988_capture_init_verbs[] = { + /* mute analog mix */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, + /* select ADCs - front-mic */ + {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1}, + {0x0d, AC_VERB_SET_CONNECT_SEL, 0x1}, + {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* ADCs; muted */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + { } +}; + +static struct hda_verb ad1988_spdif_init_verbs[] = { + /* SPDIF out sel */ + {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */ + {0x0b, AC_VERB_SET_CONNECT_SEL, 0x0}, /* ADC1 */ + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* SPDIF out pin */ + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x17}, /* 0dB */ + + { } +}; + +/* + * verbs for 3stack (+dig) + */ +static struct hda_verb ad1988_3stack_ch2_init[] = { + /* set port-C to line-in */ + { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + /* set port-E to mic-in */ + { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { } /* end */ +}; + +static struct hda_verb ad1988_3stack_ch6_init[] = { + /* set port-C to surround out */ + { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + /* set port-E to CLFE out */ + { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { } /* end */ +}; + +static struct hda_channel_mode ad1988_3stack_modes[2] = { + { 2, ad1988_3stack_ch2_init }, + { 6, ad1988_3stack_ch6_init }, +}; + +static struct hda_verb ad1988_3stack_init_verbs[] = { + /* Front, Surround, CLFE, side DAC; mute as default */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Port-A front headphon path */ + {0x37, AC_VERB_SET_CONNECT_SEL, 0x01}, /* DAC1:04h */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Port-D line-out path */ + {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* Mono out path */ + {0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */ + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */ + /* Port-B front mic-in path */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + /* Port-C line-in/surround path */ + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x33, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* Port-E mic-in/CLFE path */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x34, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* mute analog mix */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, + /* select ADCs - front-mic */ + {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1}, + {0x0d, AC_VERB_SET_CONNECT_SEL, 0x1}, + {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* ADCs; muted */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + { } +}; + +/* + * verbs for laptop mode (+dig) + */ +static struct hda_verb ad1988_laptop_hp_on[] = { + /* unmute port-A and mute port-D */ + { 0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { } /* end */ +}; +static struct hda_verb ad1988_laptop_hp_off[] = { + /* mute port-A and unmute port-D */ + { 0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { } /* end */ +}; + +#define AD1988_HP_EVENT 0x01 + +static struct hda_verb ad1988_laptop_init_verbs[] = { + /* Front, Surround, CLFE, side DAC; mute as default */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Port-A front headphon path */ + {0x37, AC_VERB_SET_CONNECT_SEL, 0x01}, /* DAC1:04h */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* unsolicited event for pin-sense */ + {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1988_HP_EVENT }, + /* Port-D line-out path + EAPD */ + {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x00}, /* EAPD-off */ + /* Mono out path */ + {0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */ + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */ + /* Port-B mic-in path */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + /* Port-C docking station - try to output */ + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x33, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* mute analog mix */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, + /* select ADCs - mic */ + {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1}, + {0x0d, AC_VERB_SET_CONNECT_SEL, 0x1}, + {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* ADCs; muted */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + { } +}; + +static void ad1988_laptop_unsol_event(struct hda_codec *codec, unsigned int res) +{ + if ((res >> 26) != AD1988_HP_EVENT) + return; + if (snd_hda_codec_read(codec, 0x11, 0, AC_VERB_GET_PIN_SENSE, 0) & (1 << 31)) + snd_hda_sequence_write(codec, ad1988_laptop_hp_on); + else + snd_hda_sequence_write(codec, ad1988_laptop_hp_off); +} + + +/* + */ + +static struct hda_board_config ad1988_cfg_tbl[] = { + { .modelname = "6stack", .config = AD1988_6STACK }, + { .modelname = "6stack-dig", .config = AD1988_6STACK_DIG }, + { .modelname = "3stack", .config = AD1988_3STACK }, + { .modelname = "3stack-dig", .config = AD1988_3STACK_DIG }, + { .modelname = "laptop", .config = AD1988_LAPTOP }, + { .modelname = "laptop-dig", .config = AD1988_LAPTOP_DIG }, + {} +}; + +static int patch_ad1988(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); + codec->spec = spec; + + board_config = snd_hda_check_board_config(codec, ad1988_cfg_tbl); + if (board_config < 0 || board_config >= AD1988_MODEL_LAST) { + printk(KERN_INFO "hda_codec: Unknown model for ALC880, trying auto-probe from BIOS...\n"); + board_config = AD1988_6STACK; + } + + switch (board_config) { + case AD1988_6STACK: + case AD1988_6STACK_DIG: + spec->multiout.max_channels = 8; + spec->multiout.num_dacs = 4; + spec->multiout.dac_nids = ad1988_dac_nids; + spec->num_adc_nids = ARRAY_SIZE(ad1988_adc_nids); + spec->adc_nids = ad1988_adc_nids; + spec->input_mux = &ad1988_6stack_capture_source; + spec->num_mixers = 1; + spec->mixers[0] = ad1988_6stack_mixers; + spec->num_init_verbs = 1; + spec->init_verbs[0] = ad1988_6stack_init_verbs; + if (board_config == AD1988_6STACK_DIG) { + spec->multiout.dig_out_nid = AD1988_SPDIF_OUT; + spec->dig_in_nid = AD1988_SPDIF_IN; + } + break; + case AD1988_3STACK: + case AD1988_3STACK_DIG: + spec->multiout.max_channels = 6; + spec->multiout.num_dacs = 3; + spec->multiout.dac_nids = ad1988_dac_nids; + spec->num_adc_nids = ARRAY_SIZE(ad1988_adc_nids); + spec->adc_nids = ad1988_adc_nids; + spec->input_mux = &ad1988_6stack_capture_source; + spec->channel_mode = ad1988_3stack_modes; + spec->num_channel_mode = ARRAY_SIZE(ad1988_3stack_modes); + spec->num_mixers = 1; + spec->mixers[0] = ad1988_3stack_mixers; + spec->num_init_verbs = 1; + spec->init_verbs[0] = ad1988_3stack_init_verbs; + if (board_config == AD1988_3STACK_DIG) + spec->multiout.dig_out_nid = AD1988_SPDIF_OUT; + break; + case AD1988_LAPTOP: + case AD1988_LAPTOP_DIG: + spec->multiout.max_channels = 2; + spec->multiout.num_dacs = 1; + spec->multiout.dac_nids = ad1988_dac_nids; + spec->num_adc_nids = ARRAY_SIZE(ad1988_adc_nids); + spec->adc_nids = ad1988_adc_nids; + spec->input_mux = &ad1988_laptop_capture_source; + spec->num_mixers = 1; + spec->mixers[0] = ad1988_laptop_mixers; + spec->num_init_verbs = 1; + spec->init_verbs[0] = ad1988_laptop_init_verbs; + if (board_config == AD1988_LAPTOP_DIG) + spec->multiout.dig_out_nid = AD1988_SPDIF_OUT; + break; + } + + spec->mixers[spec->num_mixers++] = ad1988_capture_mixers; + spec->init_verbs[spec->num_init_verbs++] = ad1988_capture_init_verbs; + if (spec->multiout.dig_out_nid) { + spec->mixers[spec->num_mixers++] = ad1988_spdif_out_mixers; + spec->init_verbs[spec->num_init_verbs++] = ad1988_spdif_init_verbs; + } + if (spec->dig_in_nid) + spec->mixers[spec->num_mixers++] = ad1988_spdif_in_mixers; + + codec->patch_ops = ad198x_patch_ops; + switch (board_config) { + case AD1988_LAPTOP: + case AD1988_LAPTOP_DIG: + codec->patch_ops.unsol_event = ad1988_laptop_unsol_event; + break; + } + + return 0; +} + + /* * patch entries */ @@ -863,5 +1647,6 @@ struct hda_codec_preset snd_hda_preset_analog[] = { { .id = 0x11d41981, .name = "AD1981", .patch = patch_ad1981 }, { .id = 0x11d41983, .name = "AD1983", .patch = patch_ad1983 }, { .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a }, + { .id = 0x11d41988, .name = "AD1988", .patch = patch_ad1988 }, {} /* terminator */ }; -- cgit v1.2.3 From 421a12520d4ed4a0c96640fb672ef24ad8019beb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 17 Nov 2005 16:11:09 +0100 Subject: [ALSA] hda-intel - Fix PM support Modules: HDA Intel driver Fix PM support on HDA-Intel driver. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 30 +++++++++++++++++++----------- 1 file changed, 19 insertions(+), 11 deletions(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index abdbd96d4c06..3945c44440cf 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1326,28 +1326,33 @@ static int __devinit azx_init_stream(struct azx *chip) /* * power management */ -static int azx_suspend(struct snd_card *card, pm_message_t state) +static int azx_suspend(struct pci_dev *pci, pm_message_t state) { - struct azx *chip = card->pm_private_data; + struct snd_card *card = pci_get_drvdata(pci); + struct azx *chip = card->private_data; int i; + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); for (i = 0; i < chip->pcm_devs; i++) - if (chip->pcm[i]) - snd_pcm_suspend_all(chip->pcm[i]); + snd_pcm_suspend_all(chip->pcm[i]); snd_hda_suspend(chip->bus, state); azx_free_cmd_io(chip); - pci_disable_device(chip->pci); + pci_disable_device(pci); + pci_save_state(pci); return 0; } -static int azx_resume(struct snd_card *card) +static int azx_resume(struct pci_dev *pci) { - struct azx *chip = card->pm_private_data; + struct snd_card *card = pci_get_drvdata(pci); + struct azx *chip = card->private_data; - pci_enable_device(chip->pci); - pci_set_master(chip->pci); + pci_restore_state(pci); + pci_enable_device(pci); + pci_set_master(pci); azx_init_chip(chip); snd_hda_resume(chip->bus); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } #endif /* CONFIG_PM */ @@ -1559,6 +1564,7 @@ static int __devinit azx_probe(struct pci_dev *pci, const struct pci_device_id * snd_card_free(card); return err; } + card->private_data = chip; /* create codec instances */ if ((err = azx_codec_create(chip, model)) < 0) { @@ -1578,7 +1584,6 @@ static int __devinit azx_probe(struct pci_dev *pci, const struct pci_device_id * return err; } - snd_card_set_pm_callback(card, azx_suspend, azx_resume, chip); snd_card_set_dev(card, &pci->dev); if ((err = snd_card_register(card)) < 0) { @@ -1618,7 +1623,10 @@ static struct pci_driver driver = { .id_table = azx_ids, .probe = azx_probe, .remove = __devexit_p(azx_remove), - SND_PCI_PM_CALLBACKS +#ifdef CONFIG_PM + .suspend = azx_suspend, + .resume = azx_resume, +#endif }; static int __init alsa_card_azx_init(void) -- cgit v1.2.3 From 54d174031576a2855c49611d83d4946bde81b504 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 21 Nov 2005 16:33:22 +0100 Subject: [ALSA] hda-codec - Fix connection list parsing Modules: HDA Codec driver,HDA generic driver - Fix connection list parsing (with ranged flag). - Increase the max number of connections - Introduce widget capabilities cache - Power up/down widgets at init, suspend and resume Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 155 ++++++++++++++++++++++++++++++----------- sound/pci/hda/hda_codec.h | 13 +++- sound/pci/hda/hda_local.h | 14 +++- sound/pci/hda/patch_realtek.c | 3 +- sound/pci/hda/patch_sigmatel.c | 25 +------ 5 files changed, 140 insertions(+), 70 deletions(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 7f4e19951bae..402ce00c6a13 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -155,8 +155,9 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, hda_nid_t *conn_list, int max_conns) { unsigned int parm; - int i, j, conn_len, num_tupples, conns; + int i, conn_len, conns; unsigned int shift, num_elems, mask; + hda_nid_t prev_nid; snd_assert(conn_list && max_conns > 0, return -EINVAL); @@ -171,7 +172,6 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, num_elems = 4; } conn_len = parm & AC_CLIST_LENGTH; - num_tupples = num_elems / 2; mask = (1 << (shift-1)) - 1; if (! conn_len) @@ -186,40 +186,38 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, /* multi connection */ conns = 0; - for (i = 0; i < conn_len; i += num_elems) { - parm = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_LIST, i); - for (j = 0; j < num_tupples; j++) { - int range_val; - hda_nid_t val1, val2, n; - range_val = parm & (1 << (shift-1)); /* ranges */ - val1 = parm & mask; - parm >>= shift; - val2 = parm & mask; - parm >>= shift; - if (range_val) { - /* ranges between val1 and val2 */ - if (val1 > val2) { - snd_printk(KERN_WARNING "hda_codec: invalid dep_range_val %x:%x\n", val1, val2); - continue; - } - for (n = val1; n <= val2; n++) { - if (conns >= max_conns) - return -EINVAL; - conn_list[conns++] = n; - } - } else { - if (! val1) - break; - if (conns >= max_conns) - return -EINVAL; - conn_list[conns++] = val1; - if (! val2) - break; - if (conns >= max_conns) + prev_nid = 0; + for (i = 0; i < conn_len; i++) { + int range_val; + hda_nid_t val, n; + + if (i % num_elems == 0) + parm = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONNECT_LIST, i); + range_val = !! (parm & (1 << (shift-1))); /* ranges */ + val = parm & mask; + parm >>= shift; + if (range_val) { + /* ranges between the previous and this one */ + if (! prev_nid || prev_nid >= val) { + snd_printk(KERN_WARNING "hda_codec: invalid dep_range_val %x:%x\n", prev_nid, val); + continue; + } + for (n = prev_nid + 1; n <= val; n++) { + if (conns >= max_conns) { + snd_printk(KERN_ERR "Too many connections\n"); return -EINVAL; - conn_list[conns++] = val2; + } + conn_list[conns++] = n; } + } else { + if (conns >= max_conns) { + snd_printk(KERN_ERR "Too many connections\n"); + return -EINVAL; + } + conn_list[conns++] = val; } + prev_nid = val; } return conns; } @@ -455,6 +453,27 @@ static void setup_fg_nodes(struct hda_codec *codec) } } +/* + * read widget caps for each widget and store in cache + */ +static int read_widget_caps(struct hda_codec *codec, hda_nid_t fg_node) +{ + int i; + hda_nid_t nid; + + codec->num_nodes = snd_hda_get_sub_nodes(codec, fg_node, + &codec->start_nid); + codec->wcaps = kmalloc(codec->num_nodes * 4, GFP_KERNEL); + if (! codec->wcaps) + return -ENOMEM; + nid = codec->start_nid; + for (i = 0; i < codec->num_nodes; i++, nid++) + codec->wcaps[i] = snd_hda_param_read(codec, nid, + AC_PAR_AUDIO_WIDGET_CAP); + return 0; +} + + /* * codec destructor */ @@ -467,6 +486,7 @@ static void snd_hda_codec_free(struct hda_codec *codec) if (codec->patch_ops.free) codec->patch_ops.free(codec); kfree(codec->amp_info); + kfree(codec->wcaps); kfree(codec); } @@ -520,6 +540,12 @@ int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, return -ENODEV; } + if (read_widget_caps(codec, codec->afg ? codec->afg : codec->mfg) < 0) { + snd_printk(KERN_ERR "hda_codec: cannot malloc\n"); + snd_hda_codec_free(codec); + return -ENOMEM; + } + if (! codec->subsystem_id) { hda_nid_t nid = codec->afg ? codec->afg : codec->mfg; codec->subsystem_id = snd_hda_codec_read(codec, nid, 0, @@ -647,7 +673,7 @@ static u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction) if (! info) return 0; if (! (info->status & INFO_AMP_CAPS)) { - if (!(snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP) & AC_WCAP_AMP_OVRD)) + if (! (get_wcaps(codec, nid) & AC_WCAP_AMP_OVRD)) nid = codec->afg; info->amp_caps = snd_hda_param_read(codec, nid, direction == HDA_OUTPUT ? AC_PAR_AMP_OUT_CAP : AC_PAR_AMP_IN_CAP); @@ -1195,6 +1221,31 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid) } +/* + * set power state of the codec + */ +static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state) +{ + hda_nid_t nid, nid_start; + int nodes; + + snd_hda_codec_write(codec, fg, 0, AC_VERB_SET_POWER_STATE, + power_state); + + nodes = snd_hda_get_sub_nodes(codec, fg, &nid_start); + for (nid = nid_start; nid < nodes + nid_start; nid++) { + if (get_wcaps(codec, nid) & AC_WCAP_POWER) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_POWER_STATE, + power_state); + } + + if (power_state == AC_PWRST_D0) + msleep(10); +} + + /** * snd_hda_build_controls - build mixer controls * @bus: the BUS @@ -1222,6 +1273,9 @@ int snd_hda_build_controls(struct hda_bus *bus) list_for_each(p, &bus->codec_list) { struct hda_codec *codec = list_entry(p, struct hda_codec, list); int err; + hda_set_power_state(codec, + codec->afg ? codec->afg : codec->mfg, + AC_PWRST_D0); if (! codec->patch_ops.init) continue; err = codec->patch_ops.init(codec); @@ -1340,7 +1394,7 @@ int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, val = 0; if (nid != codec->afg && - snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP) & AC_WCAP_FORMAT_OVRD) { + (get_wcaps(codec, nid) & AC_WCAP_FORMAT_OVRD)) { val = snd_hda_param_read(codec, nid, AC_PAR_PCM); if (val == -1) return -EIO; @@ -1362,7 +1416,7 @@ int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, unsigned int bps; unsigned int wcaps; - wcaps = snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP); + wcaps = get_wcaps(codec, nid); streams = snd_hda_param_read(codec, nid, AC_PAR_STREAM); if (streams == -1) return -EIO; @@ -1432,7 +1486,7 @@ int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid, unsigned int val = 0, rate, stream; if (nid != codec->afg && - snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP) & AC_WCAP_FORMAT_OVRD) { + (get_wcaps(codec, nid) & AC_WCAP_FORMAT_OVRD)) { val = snd_hda_param_read(codec, nid, AC_PAR_PCM); if (val == -1) return 0; @@ -1658,9 +1712,21 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew) int err; for (; knew->name; knew++) { - err = snd_ctl_add(codec->bus->card, snd_ctl_new1(knew, codec)); - if (err < 0) - return err; + struct snd_kcontrol *kctl; + kctl = snd_ctl_new1(knew, codec); + if (! kctl) + return -ENOMEM; + err = snd_ctl_add(codec->bus->card, kctl); + if (err < 0) { + if (! codec->addr) + return err; + kctl = snd_ctl_new1(knew, codec); + if (! kctl) + return -ENOMEM; + kctl->id.device = codec->addr; + if ((err = snd_ctl_add(codec->bus->card, kctl)) < 0) + return err; + } } return 0; } @@ -1874,8 +1940,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *c nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid_start); for (nid = nid_start; nid < nodes + nid_start; nid++) { - unsigned int wid_caps = snd_hda_param_read(codec, nid, - AC_PAR_AUDIO_WIDGET_CAP); + unsigned int wid_caps = get_wcaps(codec, nid); unsigned int wid_type = (wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; unsigned int def_conf; short assoc, loc; @@ -1993,6 +2058,9 @@ int snd_hda_suspend(struct hda_bus *bus, pm_message_t state) struct hda_codec *codec = list_entry(p, struct hda_codec, list); if (codec->patch_ops.suspend) codec->patch_ops.suspend(codec, state); + hda_set_power_state(codec, + codec->afg ? codec->afg : codec->mfg, + AC_PWRST_D3); } return 0; } @@ -2010,6 +2078,9 @@ int snd_hda_resume(struct hda_bus *bus) list_for_each(p, &bus->codec_list) { struct hda_codec *codec = list_entry(p, struct hda_codec, list); + hda_set_power_state(codec, + codec->afg ? codec->afg : codec->mfg, + AC_PWRST_D0); if (codec->patch_ops.resume) codec->patch_ops.resume(codec); } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 0b5c36788898..63e26c7a2b7a 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -214,6 +214,12 @@ enum { #define AC_PWRST_D2SUP (1<<2) #define AC_PWRST_D3SUP (1<<3) +/* Power state values */ +#define AC_PWRST_D0 0x00 +#define AC_PWRST_D1 0x01 +#define AC_PWRST_D2 0x02 +#define AC_PWRST_D3 0x03 + /* Processing capabilies */ #define AC_PCAP_BENIGN (1<<0) #define AC_PCAP_NUM_COEF (0xff<<8) @@ -376,7 +382,7 @@ enum { }; /* max. connections to a widget */ -#define HDA_MAX_CONNECTIONS 16 +#define HDA_MAX_CONNECTIONS 32 /* max. codec address */ #define HDA_MAX_CODEC_ADDRESS 0x0f @@ -542,6 +548,11 @@ struct hda_codec { /* codec specific info */ void *spec; + /* widget capabilities cache */ + unsigned int num_nodes; + hda_nid_t start_nid; + u32 *wcaps; + /* hash for amp access */ u16 amp_hash[32]; int num_amp_entries; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 502290424c67..ded32359b654 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -87,7 +87,7 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid); /* * input MUX helper */ -#define HDA_MAX_NUM_INPUTS 8 +#define HDA_MAX_NUM_INPUTS 16 struct hda_input_mux_item { const char *label; unsigned int index; @@ -243,4 +243,16 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *c #define PIN_HP 0xc0 #define PIN_HP_AMP 0x80 +/* + * get widget capabilities + */ +static inline u32 get_wcaps(struct hda_codec *codec, hda_nid_t nid) +{ + if (nid < codec->start_nid || + nid >= codec->start_nid + codec->num_nodes) + return snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP); + return codec->wcaps[nid - codec->start_nid]; +} + + #endif /* __SOUND_HDA_LOCAL_H */ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 62e6993056e6..77c5f95ea55b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2091,8 +2091,7 @@ static int patch_alc880(struct hda_codec *codec) if (! spec->adc_nids && spec->input_mux) { /* check whether NID 0x07 is valid */ - unsigned int wcap = snd_hda_param_read(codec, alc880_adc_nids[0], - AC_PAR_AUDIO_WIDGET_CAP); + unsigned int wcap = get_wcaps(codec, alc880_adc_nids[0]); wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; /* get type */ if (wcap != AC_WID_AUD_IN) { spec->adc_nids = alc880_adc_nids_alt; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d8d68f5b6131..c8c539cb4a8f 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -624,7 +624,7 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, struct auto_pin if (! pin) return 0; - wid_caps = snd_hda_param_read(codec, pin, AC_PAR_AUDIO_WIDGET_CAP); + 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, @@ -786,33 +786,10 @@ static int stac9200_parse_auto_config(struct hda_codec *codec) return 1; } -static int stac92xx_init_pstate(struct hda_codec *codec) -{ - hda_nid_t nid, nid_start; - int nodes; - - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_POWER_STATE, 0x00); - - nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid_start); - for (nid = nid_start; nid < nodes + nid_start; nid++) { - unsigned int wid_caps = snd_hda_param_read(codec, nid, - AC_PAR_AUDIO_WIDGET_CAP); - if (wid_caps & AC_WCAP_POWER) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_POWER_STATE, 0x00); - } - - mdelay(100); - - return 0; -} - static int stac92xx_init(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - stac92xx_init_pstate(codec); - snd_hda_sequence_write(codec, spec->init); stac92xx_auto_init_multi_out(codec); -- cgit v1.2.3 From d25695056ff2e1e048cfc8d7dbafaf80c3c46d5d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 21 Nov 2005 16:33:51 +0100 Subject: [ALSA] hda-codec - Allocate connection lists dynamically in generic parser Modules: HDA generic driver Allocate connection lists dynamically in generic parser. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 23 ++++++++++++++++++++--- 1 file changed, 20 insertions(+), 3 deletions(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 863e8c6d29a8..39edfcfd3abd 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -32,7 +32,8 @@ struct hda_gnode { hda_nid_t nid; /* NID of this widget */ unsigned short nconns; /* number of input connections */ - hda_nid_t conn_list[HDA_MAX_CONNECTIONS]; /* input connections */ + hda_nid_t *conn_list; + hda_nid_t slist[2]; /* temporay list */ unsigned int wid_caps; /* widget capabilities */ unsigned char type; /* widget type */ unsigned char pin_ctl; /* pin controls */ @@ -84,6 +85,8 @@ static void snd_hda_generic_free(struct hda_codec *codec) /* free all widgets */ list_for_each_safe(p, n, &spec->nid_list) { struct hda_gnode *node = list_entry(p, struct hda_gnode, list); + if (node->conn_list != node->slist) + kfree(node->conn_list); kfree(node); } kfree(spec); @@ -97,18 +100,32 @@ static int add_new_node(struct hda_codec *codec, struct hda_gspec *spec, hda_nid { struct hda_gnode *node; int nconns; + hda_nid_t conn_list[HDA_MAX_CONNECTIONS]; node = kzalloc(sizeof(*node), GFP_KERNEL); if (node == NULL) return -ENOMEM; node->nid = nid; - nconns = snd_hda_get_connections(codec, nid, node->conn_list, HDA_MAX_CONNECTIONS); + nconns = snd_hda_get_connections(codec, nid, conn_list, + HDA_MAX_CONNECTIONS); if (nconns < 0) { kfree(node); return nconns; } + if (nconns <= ARRAY_SIZE(node->slist)) + node->conn_list = node->slist; + else { + node->conn_list = kmalloc(sizeof(hda_nid_t) * nconns, + GFP_KERNEL); + if (! node->conn_list) { + snd_printk(KERN_ERR "hda-generic: cannot malloc\n"); + kfree(node); + return -ENOMEM; + } + } + memcpy(node->conn_list, conn_list, nconns); node->nconns = nconns; - node->wid_caps = snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP); + node->wid_caps = get_wcaps(codec, nid); node->type = (node->wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; if (node->type == AC_WID_PIN) { -- cgit v1.2.3 From 2e5b9567f7444673a93cbacdcbeb3feacdb4914f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 21 Nov 2005 16:36:15 +0100 Subject: [ALSA] hda-codec - Fix AD1988 support Modules: HDA Codec driver Fix AD1988 support. As default, 6stack model is used. Still no auto-BIOS setup is implemented. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 52 +++++++++++++++++++++++++++----------------- 1 file changed, 32 insertions(+), 20 deletions(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 25116a883ca6..3799d8a1afae 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -50,6 +50,7 @@ struct ad198x_spec { /* capture source */ const struct hda_input_mux *input_mux; + hda_nid_t *capsrc_nids; unsigned int cur_mux[3]; /* channel model */ @@ -91,7 +92,8 @@ static int ad198x_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ele unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, - spec->adc_nids[adc_idx], &spec->cur_mux[adc_idx]); + spec->capsrc_nids[adc_idx], + &spec->cur_mux[adc_idx]); } /* @@ -536,6 +538,7 @@ 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->input_mux = &ad1986a_capture_source; spec->num_mixers = 1; spec->mixers[0] = ad1986a_mixers; @@ -699,6 +702,7 @@ 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->input_mux = &ad1983_capture_source; spec->num_mixers = 1; spec->mixers[0] = ad1983_mixers; @@ -844,6 +848,7 @@ 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->input_mux = &ad1981_capture_source; spec->num_mixers = 1; spec->mixers[0] = ad1981_mixers; @@ -968,6 +973,10 @@ static hda_nid_t ad1988_adc_nids[3] = { 0x08, 0x09, 0x0f }; +static hda_nid_t ad1988_capsrc_nids[3] = { + 0x0c, 0x0d, 0x0e +}; + #define AD1988_SPDIF_OUT 0x02 #define AD1988_SPDIF_IN 0x07 @@ -1086,7 +1095,7 @@ static struct snd_kcontrol_new ad1988_6stack_mixers[] = { HDA_CODEC_VOLUME("Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x39, 0x0, HDA_OUTPUT), @@ -1121,7 +1130,7 @@ static struct snd_kcontrol_new ad1988_3stack_mixers[] = { HDA_CODEC_VOLUME("Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x39, 0x0, HDA_OUTPUT), @@ -1153,7 +1162,7 @@ static struct snd_kcontrol_new ad1988_laptop_mixers[] = { HDA_CODEC_VOLUME("Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Boost", 0x39, 0x0, HDA_OUTPUT), @@ -1277,11 +1286,11 @@ static struct snd_kcontrol_new ad1988_spdif_in_mixers[] = { * for 6-stack (+dig) */ static struct hda_verb ad1988_6stack_init_verbs[] = { - /* Front, Surround, CLFE, side DAC; mute as default */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Front, Surround, CLFE, side DAC; unmute as default */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Port-A front headphon path */ {0x37, AC_VERB_SET_CONNECT_SEL, 0x01}, /* DAC1:04h */ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, @@ -1396,11 +1405,11 @@ static struct hda_channel_mode ad1988_3stack_modes[2] = { }; static struct hda_verb ad1988_3stack_init_verbs[] = { - /* Front, Surround, CLFE, side DAC; mute as default */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Front, Surround, CLFE, side DAC; unmute as default */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Port-A front headphon path */ {0x37, AC_VERB_SET_CONNECT_SEL, 0x01}, /* DAC1:04h */ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, @@ -1471,11 +1480,11 @@ static struct hda_verb ad1988_laptop_hp_off[] = { #define AD1988_HP_EVENT 0x01 static struct hda_verb ad1988_laptop_init_verbs[] = { - /* Front, Surround, CLFE, side DAC; mute as default */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Front, Surround, CLFE, side DAC; unmute as default */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Port-A front headphon path */ {0x37, AC_VERB_SET_CONNECT_SEL, 0x01}, /* DAC1:04h */ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, @@ -1563,7 +1572,7 @@ static int patch_ad1988(struct hda_codec *codec) board_config = snd_hda_check_board_config(codec, ad1988_cfg_tbl); if (board_config < 0 || board_config >= AD1988_MODEL_LAST) { - printk(KERN_INFO "hda_codec: Unknown model for ALC880, trying auto-probe from BIOS...\n"); + printk(KERN_INFO "hda_codec: Unknown model for AD1988, using 6stack model...\n"); board_config = AD1988_6STACK; } @@ -1575,6 +1584,7 @@ static int patch_ad1988(struct hda_codec *codec) spec->multiout.dac_nids = ad1988_dac_nids; spec->num_adc_nids = ARRAY_SIZE(ad1988_adc_nids); spec->adc_nids = ad1988_adc_nids; + spec->capsrc_nids = ad1988_capsrc_nids; spec->input_mux = &ad1988_6stack_capture_source; spec->num_mixers = 1; spec->mixers[0] = ad1988_6stack_mixers; @@ -1592,6 +1602,7 @@ static int patch_ad1988(struct hda_codec *codec) spec->multiout.dac_nids = ad1988_dac_nids; spec->num_adc_nids = ARRAY_SIZE(ad1988_adc_nids); spec->adc_nids = ad1988_adc_nids; + spec->capsrc_nids = ad1988_capsrc_nids; spec->input_mux = &ad1988_6stack_capture_source; spec->channel_mode = ad1988_3stack_modes; spec->num_channel_mode = ARRAY_SIZE(ad1988_3stack_modes); @@ -1609,6 +1620,7 @@ static int patch_ad1988(struct hda_codec *codec) spec->multiout.dac_nids = ad1988_dac_nids; spec->num_adc_nids = ARRAY_SIZE(ad1988_adc_nids); spec->adc_nids = ad1988_adc_nids; + spec->capsrc_nids = ad1988_capsrc_nids; spec->input_mux = &ad1988_laptop_capture_source; spec->num_mixers = 1; spec->mixers[0] = ad1988_laptop_mixers; -- cgit v1.2.3 From 18612048b3e951f7e0ae9be65efe9e8cfde868a6 Mon Sep 17 00:00:00 2001 From: Adrian Bunk Date: Wed, 23 Nov 2005 13:14:50 +0100 Subject: [ALSA] sound/: possible cleanups Modules: RawMidi Midlevel,HDA generic driver This patch contains the following possible cleanups: - pci/hda/hda_proc.c should #include 'hda_local.h' for including the prototype of it's global function snd_hda_codec_proc_new() - core/rawmidi.c: make the needlessly global and EXPORT_SYMBOL'ed function snd_rawmidi_info() static Signed-off-by: Adrian Bunk Signed-off-by: Takashi Iwai --- sound/core/rawmidi.c | 5 ++--- sound/pci/hda/hda_proc.c | 1 + 2 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/pci/hda') diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 587ea1eb3197..d4d124e21924 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -534,8 +534,8 @@ static int snd_rawmidi_release(struct inode *inode, struct file *file) return err; } -int snd_rawmidi_info(struct snd_rawmidi_substream *substream, - struct snd_rawmidi_info *info) +static int snd_rawmidi_info(struct snd_rawmidi_substream *substream, + struct snd_rawmidi_info *info) { struct snd_rawmidi *rmidi; @@ -1694,7 +1694,6 @@ EXPORT_SYMBOL(snd_rawmidi_transmit_ack); EXPORT_SYMBOL(snd_rawmidi_transmit); EXPORT_SYMBOL(snd_rawmidi_new); EXPORT_SYMBOL(snd_rawmidi_set_ops); -EXPORT_SYMBOL(snd_rawmidi_info); EXPORT_SYMBOL(snd_rawmidi_info_select); EXPORT_SYMBOL(snd_rawmidi_kernel_open); EXPORT_SYMBOL(snd_rawmidi_kernel_release); diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 8cc5773958f6..ca514a6a5875 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -26,6 +26,7 @@ #include #include #include "hda_codec.h" +#include "hda_local.h" static const char *get_wid_type_name(unsigned int wid_value) { -- cgit v1.2.3 From 5014f193166d14e47525a34d65a1c7d77b0f6f38 Mon Sep 17 00:00:00 2001 From: Libin Yang Date: Wed, 23 Nov 2005 15:48:36 +0100 Subject: [ALSA] hda-codec - Fix auto-probe of ALC880 Modules: HDA Codec driver This patch is to fix the problem of calculating the nid incorrectly when auto-probe for ALC880. The problem to be fixed often behaves with such words when using dmesg, 'num_steps = 0 for NID=0x8' when auto-probe for ALC880. The patch contains: - alsa-kernel/pci/hda/patch_realtek.c: replace 'alc880_dac_to_idx' with 'alc880_idx_to_dac' in function 'alc880_auto_fill_dac_nids()' Signed-off-by: Libin Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 77c5f95ea55b..c5fb141f6222 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1760,7 +1760,7 @@ static int alc880_auto_fill_dac_nids(struct alc_spec *spec, const struct auto_pi nid = cfg->line_out_pins[i]; if (alc880_is_fixed_pin(nid)) { int idx = alc880_fixed_pin_idx(nid); - spec->multiout.dac_nids[i] = alc880_dac_to_idx(idx); + spec->multiout.dac_nids[i] = alc880_idx_to_dac(idx); assigned[idx] = 1; } } -- cgit v1.2.3 From 606ad75fb5372c0edb5ee6276c8e29fcb525f3e1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 24 Nov 2005 16:03:40 +0100 Subject: [ALSA] hda-intel - Use position buffer as default Modules: HDA Intel driver - Use the position buffer for obtaining the current DMA position as default. This seems more stable than others. - Add probe_mask module option (mainly for test boards with multiple codecs). Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 37 ++++++++----------------------------- 1 file changed, 8 insertions(+), 29 deletions(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 3945c44440cf..8a0a0a7d5d49 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -52,6 +52,7 @@ static int index = SNDRV_DEFAULT_IDX1; static char *id = SNDRV_DEFAULT_STR1; static char *model; static int position_fix; +static int probe_mask; module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); @@ -61,6 +62,9 @@ module_param(model, charp, 0444); MODULE_PARM_DESC(model, "Use the given board model."); 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)."); + /* just for backward compatibility */ static int enable; @@ -916,7 +920,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model) codecs = 0; for (c = 0; c < AZX_MAX_CODECS; c++) { - if (chip->codec_mask & (1 << c)) { + if ((chip->codec_mask & (1 << c)) & probe_mask) { err = snd_hda_codec_new(chip->bus, c, NULL); if (err < 0) continue; @@ -1150,31 +1154,6 @@ static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream) pos = azx_sd_readl(azx_dev, SD_LPIB); if (chip->position_fix == POS_FIX_FIFO) pos += azx_dev->fifo_size; -#if 0 /* disabled temprarily, auto-correction doesn't work well... */ - else if (chip->position_fix == POS_FIX_AUTO && azx_dev->period_updating) { - /* check the validity of DMA position */ - unsigned int diff = 0; - azx_dev->last_pos += azx_dev->fragsize; - if (azx_dev->last_pos > pos) - diff = azx_dev->last_pos - pos; - if (azx_dev->last_pos >= azx_dev->bufsize) { - if (pos < azx_dev->fragsize) - diff = 0; - azx_dev->last_pos = 0; - } - if (diff > 0 && diff <= azx_dev->fifo_size) - pos += azx_dev->fifo_size; - else { - snd_printdd(KERN_INFO "hda_intel: DMA position fix %d, switching to posbuf\n", diff); - chip->position_fix = POS_FIX_POSBUF; - pos = *azx_dev->posbuf; - } - azx_dev->period_updating = 0; - } -#else - else if (chip->position_fix == POS_FIX_AUTO) - pos += azx_dev->fifo_size; -#endif } if (pos >= azx_dev->bufsize) pos = 0; @@ -1412,7 +1391,7 @@ static int azx_dev_free(struct snd_device *device) * constructor */ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, - int posfix, int driver_type, + int driver_type, struct azx **rchip) { struct azx *chip; @@ -1441,7 +1420,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip->irq = -1; chip->driver_type = driver_type; - chip->position_fix = posfix; + chip->position_fix = position_fix ? position_fix : POS_FIX_POSBUF; #if BITS_PER_LONG != 64 /* Fix up base address on ULI M5461 */ @@ -1559,7 +1538,7 @@ static int __devinit azx_probe(struct pci_dev *pci, const struct pci_device_id * return -ENOMEM; } - if ((err = azx_create(card, pci, position_fix, pci_id->driver_data, + if ((err = azx_create(card, pci, pci_id->driver_data, &chip)) < 0) { snd_card_free(card); return err; -- cgit v1.2.3 From b2ec642362eef10f660e2b857dda12e2d61e0198 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 24 Nov 2005 16:05:04 +0100 Subject: [ALSA] hda-codec - Fix channel mode helper Modules: HDA Codec driver Fix the channel mode helper (for put callback). Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 402ce00c6a13..5ead2a3d05ad 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1771,7 +1771,7 @@ int snd_hda_ch_mode_put(struct hda_codec *codec, struct snd_ctl_elem_value *ucon mode = ucontrol->value.enumerated.item[0]; snd_assert(mode < num_chmodes, return -EINVAL); - if (*max_channelsp && ! codec->in_resume) + if (*max_channelsp == chmode[mode].channels && ! codec->in_resume) return 0; /* change the current channel setting */ *max_channelsp = chmode[mode].channels; -- cgit v1.2.3 From d32410b1095cf93e8e31f8919de46f496d7b3ce0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 24 Nov 2005 16:06:23 +0100 Subject: [ALSA] hda-codec - Fix/enhance AD1988 support Modules: HDA Codec driver Fix/enhance AD1988 support code. - Fix for h/w bug of AD1988A rev 2 - The BIOS auto-configuration is added and used as fallback Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 572 +++++++++++++++++++++++++++++++++++++++---- 1 file changed, 531 insertions(+), 41 deletions(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 3799d8a1afae..fabcbcf77a10 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1,5 +1,5 @@ /* - * HD audio interface patch for AD1981HD, AD1983, AD1986A + * HD audio interface patch for AD1981HD, AD1983, AD1986A, AD1988 * * Copyright (c) 2005 Takashi Iwai * @@ -31,7 +31,7 @@ struct ad198x_spec { struct snd_kcontrol_new *mixers[5]; int num_mixers; - const struct hda_verb *init_verbs[3]; /* initialization verbs + const struct hda_verb *init_verbs[5]; /* initialization verbs * don't forget NULL termination! */ unsigned int num_init_verbs; @@ -62,6 +62,13 @@ struct ad198x_spec { struct semaphore amp_mutex; /* PCM volume/mute control mutex */ unsigned int spdif_route; + + /* dynamic controls, init_verbs and input_mux */ + struct auto_pin_cfg autocfg; + unsigned int num_kctl_alloc, num_kctl_used; + struct snd_kcontrol_new *kctl_alloc; + struct hda_input_mux private_imux; + hda_nid_t private_dac_nids[4]; }; /* @@ -284,6 +291,14 @@ static int ad198x_build_pcms(struct hda_codec *codec) static void ad198x_free(struct hda_codec *codec) { + struct ad198x_spec *spec = codec->spec; + unsigned int i; + + if (spec->kctl_alloc) { + for (i = 0; i < spec->num_kctl_used; i++) + kfree(spec->kctl_alloc[i].name); + kfree(spec->kctl_alloc); + } kfree(codec->spec); } @@ -867,7 +882,7 @@ static int patch_ad1981(struct hda_codec *codec) * * Output pins and routes * - * Pin Mix Sel DAC + * Pin Mix Sel DAC (*) * port-A 0x11 (mute/hp) <- 0x22 <- 0x37 <- 03/04/06 * port-B 0x14 (mute/hp) <- 0x2b <- 0x30 <- 03/04/06 * port-C 0x15 (mute) <- 0x2c <- 0x31 <- 05/0a @@ -878,6 +893,8 @@ static int patch_ad1981(struct hda_codec *codec) * port-H 0x25 (mute) <- 0x28 <- 0a * mono 0x13 (mute/amp)<- 0x1e <- 0x36 <- 03/04/06 * + * DAC0 = 03h, DAC1 = 04h, DAC2 = 05h, DAC3 = 06h, DAC4 = 0ah + * (*) DAC2/3/4 are swapped to DAC3/4/2 on AD198A rev.2 due to a h/w bug. * * Input pins and routes * @@ -893,11 +910,8 @@ static int patch_ad1981(struct hda_codec *codec) * * * DAC assignment - * front DAC - 04 - * surr DAC - 06 - * CLFE DAC - 05 - * side DAC - 0a - * opt DAC - 03 + * 6stack - front/surr/CLFE/side/opt DACs - 04/06/05/0a/03 + * 3stack - front/surr/CLFE/opt DACs - 04/0a/05/03 * * Inputs of Analog Mix (0x20) * 0:Port-B (front mic) @@ -957,18 +971,35 @@ enum { AD1988_3STACK_DIG, AD1988_LAPTOP, AD1988_LAPTOP_DIG, + AD1988_AUTO, AD1988_MODEL_LAST, }; +/* reivision id to check workarounds */ +#define AD1988A_REV2 0x100200 + /* * mixers */ -static hda_nid_t ad1988_dac_nids[4] = { +static hda_nid_t ad1988_6stack_dac_nids[4] = { 0x04, 0x06, 0x05, 0x0a }; +static hda_nid_t ad1988_3stack_dac_nids[3] = { + 0x04, 0x0a, 0x05 +}; + +/* for AD1988A revision-2, DAC2-4 are swapped */ +static hda_nid_t ad1988_6stack_dac_nids_rev2[4] = { + 0x04, 0x05, 0x0a, 0x06 +}; + +static hda_nid_t ad1988_3stack_dac_nids_rev2[3] = { + 0x04, 0x06, 0x0a +}; + static hda_nid_t ad1988_adc_nids[3] = { 0x08, 0x09, 0x0f }; @@ -1068,13 +1099,23 @@ static int ad1988_eapd_put(struct snd_kcontrol *kcontrol, } /* 6-stack mode */ -static struct snd_kcontrol_new ad1988_6stack_mixers[] = { +static struct snd_kcontrol_new ad1988_6stack_mixers1[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x06, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Side Playback Volume", 0x0a, 0x0, HDA_OUTPUT), +}; + +static struct snd_kcontrol_new ad1988_6stack_mixers1_rev2[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x05, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0a, 2, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Side Playback Volume", 0x06, 0x0, HDA_OUTPUT), +}; +static struct snd_kcontrol_new ad1988_6stack_mixers2[] = { HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT), HDA_BIND_MUTE("Surround Playback Switch", 0x2a, 2, HDA_INPUT), HDA_BIND_MUTE_MONO("Center Playback Switch", 0x27, 1, 2, HDA_INPUT), @@ -1105,16 +1146,25 @@ static struct snd_kcontrol_new ad1988_6stack_mixers[] = { }; /* 3-stack mode */ -static struct snd_kcontrol_new ad1988_3stack_mixers[] = { +static struct snd_kcontrol_new ad1988_3stack_mixers1[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x06, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0a, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT), +}; + +static struct snd_kcontrol_new ad1988_3stack_mixers1_rev2[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x06, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0a, 2, 0x0, HDA_OUTPUT), +}; +static struct snd_kcontrol_new ad1988_3stack_mixers2[] = { HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x2a, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x27, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x27, 2, 2, HDA_INPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x2c, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x26, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x26, 2, 2, HDA_INPUT), HDA_BIND_MUTE("Headphone Playback Switch", 0x22, 2, HDA_INPUT), HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT), @@ -1391,11 +1441,11 @@ static struct hda_verb ad1988_3stack_ch2_init[] = { static struct hda_verb ad1988_3stack_ch6_init[] = { /* set port-C to surround out */ - { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, /* set port-E to CLFE out */ - { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, { } /* end */ }; @@ -1431,15 +1481,17 @@ static struct hda_verb ad1988_3stack_init_verbs[] = { {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* Port-C line-in/surround path */ - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + /* Port-C line-in/surround path - 6ch mode as default */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x31, AC_VERB_SET_CONNECT_SEL, 0x0}, /* output sel: DAC 0x05 */ {0x33, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Port-E mic-in/CLFE path */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + /* Port-E mic-in/CLFE path - 6ch mode as default */ + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x32, AC_VERB_SET_CONNECT_SEL, 0x1}, /* output sel: DAC 0x06 */ {0x34, AC_VERB_SET_CONNECT_SEL, 0x0}, /* mute analog mix */ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, @@ -1545,6 +1597,420 @@ static void ad1988_laptop_unsol_event(struct hda_codec *codec, unsigned int res) } +/* + * Automatic parse of I/O pins from the BIOS configuration + */ + +#define NUM_CONTROL_ALLOC 32 +#define NUM_VERB_ALLOC 32 + +enum { + AD_CTL_WIDGET_VOL, + AD_CTL_WIDGET_MUTE, + AD_CTL_BIND_MUTE, +}; +static struct snd_kcontrol_new ad1988_control_templates[] = { + HDA_CODEC_VOLUME(NULL, 0, 0, 0), + HDA_CODEC_MUTE(NULL, 0, 0, 0), + HDA_BIND_MUTE(NULL, 0, 0, 0), +}; + +/* add dynamic controls */ +static int add_control(struct ad198x_spec *spec, int type, const char *name, + unsigned long val) +{ + struct snd_kcontrol_new *knew; + + if (spec->num_kctl_used >= spec->num_kctl_alloc) { + int num = spec->num_kctl_alloc + NUM_CONTROL_ALLOC; + + knew = kcalloc(num + 1, sizeof(*knew), GFP_KERNEL); /* array + terminator */ + if (! knew) + return -ENOMEM; + if (spec->kctl_alloc) { + memcpy(knew, spec->kctl_alloc, sizeof(*knew) * spec->num_kctl_alloc); + kfree(spec->kctl_alloc); + } + spec->kctl_alloc = knew; + spec->num_kctl_alloc = num; + } + + knew = &spec->kctl_alloc[spec->num_kctl_used]; + *knew = ad1988_control_templates[type]; + knew->name = kstrdup(name, GFP_KERNEL); + if (! knew->name) + return -ENOMEM; + knew->private_value = val; + spec->num_kctl_used++; + return 0; +} + +#define AD1988_PIN_CD_NID 0x18 +#define AD1988_PIN_BEEP_NID 0x10 + +static hda_nid_t ad1988_mixer_nids[8] = { + /* A B C D E F G H */ + 0x22, 0x2b, 0x2c, 0x29, 0x26, 0x2a, 0x27, 0x28 +}; + +static inline hda_nid_t ad1988_idx_to_dac(struct hda_codec *codec, int idx) +{ + static hda_nid_t idx_to_dac[8] = { + /* A B C D E F G H */ + 0x04, 0x06, 0x0a, 0x04, 0x05, 0x06, 0x05, 0x0a + }; + static hda_nid_t idx_to_dac_rev2[8] = { + /* A B C D E F G H */ + 0x04, 0x05, 0x06, 0x04, 0x0a, 0x05, 0x0a, 0x06 + }; + if (codec->revision_id == AD1988A_REV2) + return idx_to_dac_rev2[idx]; + else + return idx_to_dac[idx]; +} + +static hda_nid_t ad1988_boost_nids[8] = { + 0x38, 0x39, 0x3a, 0x3d, 0x3c, 0x3b, 0, 0 +}; + +static int ad1988_pin_idx(hda_nid_t nid) +{ + static hda_nid_t ad1988_io_pins[8] = { + 0x11, 0x14, 0x15, 0x12, 0x17, 0x16, 0x24, 0x25 + }; + int i; + for (i = 0; i < ARRAY_SIZE(ad1988_io_pins); i++) + if (ad1988_io_pins[i] == nid) + return i; + return 0; /* should be -1 */ +} + +static int ad1988_pin_to_loopback_idx(hda_nid_t nid) +{ + static int loopback_idx[8] = { + 2, 0, 1, 3, 4, 5, 1, 4 + }; + switch (nid) { + case AD1988_PIN_CD_NID: + return 6; + default: + return loopback_idx[ad1988_pin_idx(nid)]; + } +} + +static int ad1988_pin_to_adc_idx(hda_nid_t nid) +{ + static int adc_idx[8] = { + 0, 1, 2, 8, 4, 3, 6, 7 + }; + switch (nid) { + case AD1988_PIN_CD_NID: + return 5; + default: + return adc_idx[ad1988_pin_idx(nid)]; + } +} + +/* fill in the dac_nids table from the parsed pin configuration */ +static int ad1988_auto_fill_dac_nids(struct hda_codec *codec, + const struct auto_pin_cfg *cfg) +{ + struct ad198x_spec *spec = codec->spec; + int i, idx; + + spec->multiout.dac_nids = spec->private_dac_nids; + + /* check the pins hardwired to audio widget */ + for (i = 0; i < cfg->line_outs; i++) { + idx = ad1988_pin_idx(cfg->line_out_pins[i]); + spec->multiout.dac_nids[i] = ad1988_idx_to_dac(codec, idx); + } + spec->multiout.num_dacs = cfg->line_outs; + return 0; +} + +/* add playback controls from the parsed DAC table */ +static int ad1988_auto_create_multi_out_ctls(struct ad198x_spec *spec, + const struct auto_pin_cfg *cfg) +{ + char name[32]; + static const char *chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; + hda_nid_t nid; + int i, err; + + for (i = 0; i < cfg->line_outs; i++) { + hda_nid_t dac = spec->multiout.dac_nids[i]; + if (! dac) + continue; + nid = ad1988_mixer_nids[ad1988_pin_idx(cfg->line_out_pins[i])]; + if (i == 2) { + /* Center/LFE */ + err = add_control(spec, AD_CTL_WIDGET_VOL, + "Center Playback Volume", + HDA_COMPOSE_AMP_VAL(dac, 1, 0, HDA_OUTPUT)); + if (err < 0) + return err; + err = add_control(spec, AD_CTL_WIDGET_VOL, + "LFE Playback Volume", + HDA_COMPOSE_AMP_VAL(dac, 2, 0, HDA_OUTPUT)); + if (err < 0) + return err; + err = add_control(spec, AD_CTL_BIND_MUTE, + "Center Playback Switch", + HDA_COMPOSE_AMP_VAL(nid, 1, 2, HDA_INPUT)); + if (err < 0) + return err; + err = add_control(spec, AD_CTL_BIND_MUTE, + "LFE Playback Switch", + HDA_COMPOSE_AMP_VAL(nid, 2, 2, HDA_INPUT)); + if (err < 0) + return err; + } else { + sprintf(name, "%s Playback Volume", chname[i]); + err = add_control(spec, AD_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + sprintf(name, "%s Playback Switch", chname[i]); + err = add_control(spec, AD_CTL_BIND_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT)); + if (err < 0) + return err; + } + } + return 0; +} + +/* add playback controls for speaker and HP outputs */ +static int ad1988_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin, + const char *pfx) +{ + struct ad198x_spec *spec = codec->spec; + hda_nid_t nid; + int idx, err; + char name[32]; + + if (! pin) + return 0; + + 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 */ + spec->multiout.hp_nid = 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, + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0) + return err; + nid = ad1988_mixer_nids[idx]; + sprintf(name, "%s Playback Switch", pfx); + if ((err = add_control(spec, AD_CTL_BIND_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT))) < 0) + return err; + return 0; +} + +/* create input playback/capture controls for the given pin */ +static int new_analog_input(struct ad198x_spec *spec, hda_nid_t pin, + const char *ctlname, int boost) +{ + char name[32]; + int err, idx; + + sprintf(name, "%s Playback Volume", ctlname); + idx = ad1988_pin_to_loopback_idx(pin); + if ((err = add_control(spec, AD_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(0x20, 3, idx, HDA_INPUT))) < 0) + return err; + sprintf(name, "%s Playback Switch", ctlname); + if ((err = add_control(spec, AD_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(0x20, 3, idx, HDA_INPUT))) < 0) + return err; + if (boost) { + hda_nid_t bnid; + idx = ad1988_pin_idx(pin); + bnid = ad1988_boost_nids[idx]; + if (bnid) { + sprintf(name, "%s Boost", ctlname); + return add_control(spec, AD_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(bnid, 3, idx, HDA_OUTPUT)); + + } + } + return 0; +} + +/* create playback/capture controls for input pins */ +static int ad1988_auto_create_analog_input_ctls(struct ad198x_spec *spec, + const struct auto_pin_cfg *cfg) +{ + static char *labels[AUTO_PIN_LAST] = { + "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux" + }; + struct hda_input_mux *imux = &spec->private_imux; + int i, err; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + err = new_analog_input(spec, cfg->input_pins[i], labels[i], + i <= AUTO_PIN_FRONT_MIC); + if (err < 0) + return err; + imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].index = ad1988_pin_to_adc_idx(cfg->input_pins[i]); + imux->num_items++; + } + imux->items[imux->num_items].label = "Mix"; + imux->items[imux->num_items].index = 9; + imux->num_items++; + + if ((err = add_control(spec, AD_CTL_WIDGET_VOL, + "Analog Mix Playback Volume", + HDA_COMPOSE_AMP_VAL(0x21, 3, 0x0, HDA_OUTPUT))) < 0) + return err; + if ((err = add_control(spec, AD_CTL_WIDGET_MUTE, + "Analog Mix Playback Switch", + HDA_COMPOSE_AMP_VAL(0x21, 3, 0x0, HDA_OUTPUT))) < 0) + return err; + + return 0; +} + +static void ad1988_auto_set_output_and_unmute(struct hda_codec *codec, + hda_nid_t nid, int pin_type, + int dac_idx) +{ + /* set as output */ + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type); + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); + switch (nid) { + case 0x11: /* port-A - DAC 04 */ + snd_hda_codec_write(codec, 0x37, 0, AC_VERB_SET_CONNECT_SEL, 0x01); + break; + case 0x14: /* port-B - DAC 06 */ + snd_hda_codec_write(codec, 0x30, 0, AC_VERB_SET_CONNECT_SEL, 0x02); + break; + case 0x15: /* port-C - DAC 05 */ + snd_hda_codec_write(codec, 0x31, 0, AC_VERB_SET_CONNECT_SEL, 0x00); + break; + case 0x17: /* port-E - DAC 06 */ + snd_hda_codec_write(codec, 0x32, 0, AC_VERB_SET_CONNECT_SEL, 0x01); + break; + case 0x13: /* mono - DAC 04 */ + snd_hda_codec_write(codec, 0x36, 0, AC_VERB_SET_CONNECT_SEL, 0x01); + break; + } +} + +static void ad1988_auto_init_multi_out(struct hda_codec *codec) +{ + struct ad198x_spec *spec = codec->spec; + int i; + + for (i = 0; i < spec->autocfg.line_outs; i++) { + hda_nid_t nid = spec->autocfg.line_out_pins[i]; + ad1988_auto_set_output_and_unmute(codec, nid, PIN_OUT, i); + } +} + +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; + if (pin) /* connect to front */ + ad1988_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); + pin = spec->autocfg.hp_pin; + if (pin) /* connect to front */ + ad1988_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); +} + +static void ad1988_auto_init_analog_input(struct hda_codec *codec) +{ + struct ad198x_spec *spec = codec->spec; + int i, idx; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + hda_nid_t nid = spec->autocfg.input_pins[i]; + if (! nid) + continue; + switch (nid) { + case 0x15: /* port-C */ + snd_hda_codec_write(codec, 0x33, 0, AC_VERB_SET_CONNECT_SEL, 0x0); + break; + case 0x17: /* port-E */ + snd_hda_codec_write(codec, 0x34, 0, AC_VERB_SET_CONNECT_SEL, 0x0); + break; + } + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + i <= AUTO_PIN_FRONT_MIC ? PIN_VREF80 : PIN_IN); + if (nid != AD1988_PIN_CD_NID) + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_MUTE); + idx = ad1988_pin_idx(nid); + if (ad1988_boost_nids[idx]) + snd_hda_codec_write(codec, ad1988_boost_nids[idx], 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_ZERO); + } +} + +/* parse the BIOS configuration and set up the alc_spec */ +/* return 1 if successful, 0 if the proper config is not found, or a negative error code */ +static int ad1988_parse_auto_config(struct hda_codec *codec) +{ + struct ad198x_spec *spec = codec->spec; + int err; + + if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg)) < 0) + 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) + 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, + "Speaker")) < 0 || + (err = ad1988_auto_create_extra_out(codec, spec->autocfg.speaker_pin, + "Headphone")) < 0 || + (err = ad1988_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0) + return err; + + spec->multiout.max_channels = spec->multiout.num_dacs * 2; + + if (spec->autocfg.dig_out_pin) + spec->multiout.dig_out_nid = AD1988_SPDIF_OUT; + if (spec->autocfg.dig_in_pin) + spec->dig_in_nid = AD1988_SPDIF_IN; + + if (spec->kctl_alloc) + spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + + spec->init_verbs[spec->num_init_verbs++] = ad1988_6stack_init_verbs; + + spec->input_mux = &spec->private_imux; + + return 1; +} + +/* init callback for auto-configuration model -- overriding the default init */ +static int ad1988_auto_init(struct hda_codec *codec) +{ + ad198x_init(codec); + ad1988_auto_init_multi_out(codec); + ad1988_auto_init_extra_out(codec); + ad1988_auto_init_analog_input(codec); + return 0; +} + + /* */ @@ -1555,6 +2021,7 @@ static struct hda_board_config ad1988_cfg_tbl[] = { { .modelname = "3stack-dig", .config = AD1988_3STACK_DIG }, { .modelname = "laptop", .config = AD1988_LAPTOP }, { .modelname = "laptop-dig", .config = AD1988_LAPTOP_DIG }, + { .modelname = "auto", .config = AD1988_AUTO }, {} }; @@ -1572,8 +2039,20 @@ static int patch_ad1988(struct hda_codec *codec) board_config = snd_hda_check_board_config(codec, ad1988_cfg_tbl); if (board_config < 0 || board_config >= AD1988_MODEL_LAST) { - printk(KERN_INFO "hda_codec: Unknown model for AD1988, using 6stack model...\n"); - board_config = AD1988_6STACK; + printk(KERN_INFO "hda_codec: Unknown model for AD1988, trying auto-probe from BIOS...\n"); + board_config = AD1988_AUTO; + } + + if (board_config == AD1988_AUTO) { + /* automatic parse from the BIOS config */ + int err = ad1988_parse_auto_config(codec); + if (err < 0) { + ad198x_free(codec); + return err; + } else if (! err) { + printk(KERN_INFO "hda_codec: Cannot set up configuration from BIOS. Using 6-stack mode...\n"); + board_config = AD1988_6STACK; + } } switch (board_config) { @@ -1581,13 +2060,17 @@ static int patch_ad1988(struct hda_codec *codec) case AD1988_6STACK_DIG: spec->multiout.max_channels = 8; spec->multiout.num_dacs = 4; - spec->multiout.dac_nids = ad1988_dac_nids; - spec->num_adc_nids = ARRAY_SIZE(ad1988_adc_nids); - spec->adc_nids = ad1988_adc_nids; - spec->capsrc_nids = ad1988_capsrc_nids; + if (codec->revision_id == AD1988A_REV2) + spec->multiout.dac_nids = ad1988_6stack_dac_nids_rev2; + else + spec->multiout.dac_nids = ad1988_6stack_dac_nids; spec->input_mux = &ad1988_6stack_capture_source; - spec->num_mixers = 1; - spec->mixers[0] = ad1988_6stack_mixers; + spec->num_mixers = 2; + if (codec->revision_id == AD1988A_REV2) + spec->mixers[0] = ad1988_6stack_mixers1_rev2; + else + spec->mixers[0] = ad1988_6stack_mixers1; + spec->mixers[1] = ad1988_6stack_mixers2; spec->num_init_verbs = 1; spec->init_verbs[0] = ad1988_6stack_init_verbs; if (board_config == AD1988_6STACK_DIG) { @@ -1599,15 +2082,19 @@ static int patch_ad1988(struct hda_codec *codec) case AD1988_3STACK_DIG: spec->multiout.max_channels = 6; spec->multiout.num_dacs = 3; - spec->multiout.dac_nids = ad1988_dac_nids; - spec->num_adc_nids = ARRAY_SIZE(ad1988_adc_nids); - spec->adc_nids = ad1988_adc_nids; - spec->capsrc_nids = ad1988_capsrc_nids; + if (codec->revision_id == AD1988A_REV2) + spec->multiout.dac_nids = ad1988_3stack_dac_nids_rev2; + else + spec->multiout.dac_nids = ad1988_3stack_dac_nids; spec->input_mux = &ad1988_6stack_capture_source; spec->channel_mode = ad1988_3stack_modes; spec->num_channel_mode = ARRAY_SIZE(ad1988_3stack_modes); - spec->num_mixers = 1; - spec->mixers[0] = ad1988_3stack_mixers; + spec->num_mixers = 2; + if (codec->revision_id == AD1988A_REV2) + spec->mixers[0] = ad1988_3stack_mixers1_rev2; + else + spec->mixers[0] = ad1988_3stack_mixers1; + spec->mixers[1] = ad1988_3stack_mixers2; spec->num_init_verbs = 1; spec->init_verbs[0] = ad1988_3stack_init_verbs; if (board_config == AD1988_3STACK_DIG) @@ -1617,10 +2104,7 @@ static int patch_ad1988(struct hda_codec *codec) case AD1988_LAPTOP_DIG: spec->multiout.max_channels = 2; spec->multiout.num_dacs = 1; - spec->multiout.dac_nids = ad1988_dac_nids; - spec->num_adc_nids = ARRAY_SIZE(ad1988_adc_nids); - spec->adc_nids = ad1988_adc_nids; - spec->capsrc_nids = ad1988_capsrc_nids; + spec->multiout.dac_nids = ad1988_3stack_dac_nids; spec->input_mux = &ad1988_laptop_capture_source; spec->num_mixers = 1; spec->mixers[0] = ad1988_laptop_mixers; @@ -1631,6 +2115,9 @@ static int patch_ad1988(struct hda_codec *codec) break; } + spec->num_adc_nids = ARRAY_SIZE(ad1988_adc_nids); + spec->adc_nids = ad1988_adc_nids; + spec->capsrc_nids = ad1988_capsrc_nids; spec->mixers[spec->num_mixers++] = ad1988_capture_mixers; spec->init_verbs[spec->num_init_verbs++] = ad1988_capture_init_verbs; if (spec->multiout.dig_out_nid) { @@ -1642,6 +2129,9 @@ static int patch_ad1988(struct hda_codec *codec) codec->patch_ops = ad198x_patch_ops; switch (board_config) { + case AD1988_AUTO: + codec->patch_ops.init = ad1988_auto_init; + break; case AD1988_LAPTOP: case AD1988_LAPTOP_DIG: codec->patch_ops.unsol_event = ad1988_laptop_unsol_event; -- cgit v1.2.3 From f8c7c7b8dd2828b42c1230c6b0235e7d1dcf57e5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 24 Nov 2005 16:17:20 +0100 Subject: [ALSA] hda-codec - Fix surrounds on 3stack mode of AD1988 Modules: HDA Codec driver Fixed the swapped surround/CLFE on 3stack mode of AD1988. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 23 +++++++++++++---------- 1 file changed, 13 insertions(+), 10 deletions(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index fabcbcf77a10..fc144155f7a6 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -911,7 +911,7 @@ static int patch_ad1981(struct hda_codec *codec) * * DAC assignment * 6stack - front/surr/CLFE/side/opt DACs - 04/06/05/0a/03 - * 3stack - front/surr/CLFE/opt DACs - 04/0a/05/03 + * 3stack - front/surr/CLFE/opt DACs - 04/05/0a/03 * * Inputs of Analog Mix (0x20) * 0:Port-B (front mic) @@ -988,7 +988,7 @@ static hda_nid_t ad1988_6stack_dac_nids[4] = { }; static hda_nid_t ad1988_3stack_dac_nids[3] = { - 0x04, 0x0a, 0x05 + 0x04, 0x05, 0x0a }; /* for AD1988A revision-2, DAC2-4 are swapped */ @@ -997,7 +997,7 @@ static hda_nid_t ad1988_6stack_dac_nids_rev2[4] = { }; static hda_nid_t ad1988_3stack_dac_nids_rev2[3] = { - 0x04, 0x06, 0x0a + 0x04, 0x0a, 0x06 }; static hda_nid_t ad1988_adc_nids[3] = { @@ -1155,9 +1155,9 @@ static struct snd_kcontrol_new ad1988_3stack_mixers1[] = { static struct snd_kcontrol_new ad1988_3stack_mixers1_rev2[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x06, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0a, 2, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0a, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x06, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x06, 2, 0x0, HDA_OUTPUT), }; static struct snd_kcontrol_new ad1988_3stack_mixers2[] = { @@ -1491,7 +1491,7 @@ static struct hda_verb ad1988_3stack_init_verbs[] = { {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x32, AC_VERB_SET_CONNECT_SEL, 0x1}, /* output sel: DAC 0x06 */ + {0x32, AC_VERB_SET_CONNECT_SEL, 0x1}, /* output sel: DAC 0x0a */ {0x34, AC_VERB_SET_CONNECT_SEL, 0x0}, /* mute analog mix */ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, @@ -1657,11 +1657,11 @@ static inline hda_nid_t ad1988_idx_to_dac(struct hda_codec *codec, int idx) { static hda_nid_t idx_to_dac[8] = { /* A B C D E F G H */ - 0x04, 0x06, 0x0a, 0x04, 0x05, 0x06, 0x05, 0x0a + 0x04, 0x06, 0x05, 0x04, 0x0a, 0x06, 0x05, 0x0a }; static hda_nid_t idx_to_dac_rev2[8] = { /* A B C D E F G H */ - 0x04, 0x05, 0x06, 0x04, 0x0a, 0x05, 0x0a, 0x06 + 0x04, 0x05, 0x0a, 0x04, 0x06, 0x05, 0x0a, 0x06 }; if (codec->revision_id == AD1988A_REV2) return idx_to_dac_rev2[idx]; @@ -1898,7 +1898,7 @@ static void ad1988_auto_set_output_and_unmute(struct hda_codec *codec, case 0x15: /* port-C - DAC 05 */ snd_hda_codec_write(codec, 0x31, 0, AC_VERB_SET_CONNECT_SEL, 0x00); break; - case 0x17: /* port-E - DAC 06 */ + case 0x17: /* port-E - DAC 0a */ snd_hda_codec_write(codec, 0x32, 0, AC_VERB_SET_CONNECT_SEL, 0x01); break; case 0x13: /* mono - DAC 04 */ @@ -2037,6 +2037,9 @@ static int patch_ad1988(struct hda_codec *codec) init_MUTEX(&spec->amp_mutex); codec->spec = spec; + if (codec->revision_id == AD1988A_REV2) + snd_printk(KERN_INFO "patch_analog: AD1988A rev.2 is detected, enable workarounds\n"); + board_config = snd_hda_check_board_config(codec, ad1988_cfg_tbl); if (board_config < 0 || board_config >= AD1988_MODEL_LAST) { printk(KERN_INFO "hda_codec: Unknown model for AD1988, trying auto-probe from BIOS...\n"); -- cgit v1.2.3 From 954fa19ab7a14c3f54044780a90cd6a95149f90b Mon Sep 17 00:00:00 2001 From: Matt Porter Date: Tue, 29 Nov 2005 14:46:01 +0100 Subject: [ALSA] hda-intel - Fix HDA probe_mask default Modules: HDA Intel driver The probe_mask module parameter comment notes that the intended default is -1. Fix it to be so, otherwise all codecs are skipped and init fails. Signed-off-by: Matt Porter Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 8a0a0a7d5d49..a983deba4025 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -52,7 +52,7 @@ static int index = SNDRV_DEFAULT_IDX1; static char *id = SNDRV_DEFAULT_STR1; static char *model; static int position_fix; -static int probe_mask; +static int probe_mask = -1; module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); -- cgit v1.2.3 From 403d19446bd0cabee70110415d2f3bc466f46448 Mon Sep 17 00:00:00 2001 From: Matt Porter Date: Tue, 29 Nov 2005 15:00:51 +0100 Subject: [ALSA] hda-codec - update sigmatel support and bug fixes Modules: HDA Codec driver - Explictly set pin control as input for all input pins - Fix bug in 922x mixer (no mute on adc0vol) - Remove broken ch_mode control - Add support for jack retasking mixer controls to use rear line and mic as surround outputs - Add board tables to support autodetect and pin config defaults for systems with broken bioses - Add support for several Intel mobos - Add support for DFI mobo with reference boards attached (gets rid of compile time switch to use reference boards) Signed-off-by: Matt Porter Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 356 +++++++++++++++++++++++------------------ 1 file changed, 200 insertions(+), 156 deletions(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index c8c539cb4a8f..78662d3539e2 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4,7 +4,7 @@ * HD audio interface patch for SigmaTel STAC92xx * * Copyright (c) 2005 Embedded Alley Solutions, Inc. - * + * Matt Porter * * Based on patch_cmedia.c and patch_realtek.c * Copyright (c) 2004 Takashi Iwai @@ -34,17 +34,22 @@ #include "hda_codec.h" #include "hda_local.h" -#undef STAC_TEST - #define NUM_CONTROL_ALLOC 32 #define STAC_HP_EVENT 0x37 #define STAC_UNSOL_ENABLE (AC_USRSP_EN | STAC_HP_EVENT) +#define STAC_REF 0 +#define STAC_D945GTP3 1 +#define STAC_D945GTP5 2 + struct sigmatel_spec { struct snd_kcontrol_new *mixers[4]; unsigned int num_mixers; + int board_config; unsigned int surr_switch: 1; + unsigned int line_switch: 1; + unsigned int mic_switch: 1; /* playback */ struct hda_multi_out multiout; @@ -57,12 +62,10 @@ struct sigmatel_spec { unsigned int num_muxes; hda_nid_t dig_in_nid; -#ifdef STAC_TEST /* pin widgets */ hda_nid_t *pin_nids; unsigned int num_pins; unsigned int *pin_configs; -#endif /* codec specific stuff */ struct hda_verb *init; @@ -72,9 +75,8 @@ struct sigmatel_spec { struct hda_input_mux *input_mux; unsigned int cur_mux[2]; - /* channel mode */ - unsigned int num_ch_modes; - unsigned int cur_ch_mode; + /* i/o switches */ + unsigned int io_switch[2]; struct hda_pcm pcm_rec[2]; /* PCM information */ @@ -105,7 +107,6 @@ static hda_nid_t stac922x_mux_nids[2] = { 0x12, 0x13, }; -#ifdef STAC_TEST static hda_nid_t stac9200_pin_nids[8] = { 0x08, 0x09, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x12, }; @@ -114,7 +115,6 @@ static hda_nid_t stac922x_pin_nids[10] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x15, 0x1b, }; -#endif static int stac92xx_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -155,49 +155,6 @@ static struct hda_verb stac922x_core_init[] = { {} }; -static int stac922x_channel_modes[3] = {2, 6, 8}; - -static int stac922x_ch_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = spec->num_ch_modes; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - sprintf(uinfo->value.enumerated.name, "%dch", - stac922x_channel_modes[uinfo->value.enumerated.item]); - return 0; -} - -static int stac922x_ch_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - - ucontrol->value.enumerated.item[0] = spec->cur_ch_mode; - return 0; -} - -static int stac922x_ch_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - - if (ucontrol->value.enumerated.item[0] >= spec->num_ch_modes) - ucontrol->value.enumerated.item[0] = spec->num_ch_modes; - if (ucontrol->value.enumerated.item[0] == spec->cur_ch_mode && - ! codec->in_resume) - return 0; - - spec->cur_ch_mode = ucontrol->value.enumerated.item[0]; - spec->multiout.max_channels = stac922x_channel_modes[spec->cur_ch_mode]; - - return 1; -} - static struct snd_kcontrol_new stac9200_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0xb, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT), @@ -226,22 +183,10 @@ static struct snd_kcontrol_new stac922x_mixer[] = { .put = stac92xx_mux_enum_put, }, HDA_CODEC_VOLUME("Capture Volume", 0x17, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x17, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mux Capture Volume", 0x12, 0x0, HDA_OUTPUT), { } /* end */ }; -static struct snd_kcontrol_new stac922x_ch_mode_mixer[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Channel Mode", - .info = stac922x_ch_mode_info, - .get = stac922x_ch_mode_get, - .put = stac922x_ch_mode_put, - }, - { } /* end */ -}; - static int stac92xx_build_controls(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -258,11 +203,6 @@ static int stac92xx_build_controls(struct hda_codec *codec) return err; } - if (spec->surr_switch) { - err = snd_hda_add_new_ctls(codec, stac922x_ch_mode_mixer); - if (err < 0) - return err; - } if (spec->multiout.dig_out_nid) { err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid); if (err < 0) @@ -276,18 +216,67 @@ static int stac92xx_build_controls(struct hda_codec *codec) return 0; } -#ifdef STAC_TEST -static unsigned int stac9200_pin_configs[8] = { +static unsigned int ref9200_pin_configs[8] = { 0x01c47010, 0x01447010, 0x0221401f, 0x01114010, 0x02a19020, 0x01a19021, 0x90100140, 0x01813122, }; -static unsigned int stac922x_pin_configs[10] = { - 0x01014010, 0x01014011, 0x01014012, 0x0221401f, - 0x01813122, 0x01014014, 0x01441030, 0x01c41030, +static unsigned int *stac9200_brd_tbl[] = { + ref9200_pin_configs, +}; + +static struct hda_board_config stac9200_cfg_tbl[] = { + { .modelname = "ref", + .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x2668, /* DFI LanParty */ + .config = STAC_REF }, + {} /* terminator */ +}; + +static unsigned int ref922x_pin_configs[10] = { + 0x01014010, 0x01016011, 0x01012012, 0x0221401f, + 0x01813122, 0x01011014, 0x01441030, 0x01c41030, 0x40000100, 0x40000100, }; +static unsigned int d945gtp3_pin_configs[10] = { + 0x0221401f, 0x01a19022, 0x01813021, 0x01114010, + 0x40000100, 0x40000100, 0x40000100, 0x40000100, + 0x02a19120, 0x40000100, +}; + +static unsigned int d945gtp5_pin_configs[10] = { + 0x0221401f, 0x01111012, 0x01813024, 0x01114010, + 0x01a19021, 0x01116011, 0x01452130, 0x40000100, + 0x02a19320, 0x40000100, +}; + +static unsigned int *stac922x_brd_tbl[] = { + ref922x_pin_configs, + d945gtp3_pin_configs, + d945gtp5_pin_configs, +}; + +static struct hda_board_config stac922x_cfg_tbl[] = { + { .modelname = "ref", + .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x2668, /* DFI LanParty */ + .config = STAC_REF }, /* SigmaTel reference board */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x0101, + .config = STAC_D945GTP3 }, /* Intel D945GTP - 3 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x0404, + .config = STAC_D945GTP5 }, /* Intel D945GTP - 5 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x0303, + .config = STAC_D945GTP5 }, /* Intel D945GNT - 5 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x0013, + .config = STAC_D945GTP5 }, /* Intel D955XBK - 5 Stack */ + {} /* terminator */ +}; + static void stac92xx_set_config_regs(struct hda_codec *codec) { int i; @@ -310,10 +299,9 @@ static void stac92xx_set_config_regs(struct hda_codec *codec) pin_cfg = snd_hda_codec_read(codec, spec->pin_nids[i], 0, AC_VERB_GET_CONFIG_DEFAULT, 0x00); - printk("pin nid %2.2x pin config %8.8x\n", spec->pin_nids[i], pin_cfg); + snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x pin config %8.8x\n", spec->pin_nids[i], pin_cfg); } } -#endif /* * Analog playback callbacks @@ -326,56 +314,6 @@ static int stac92xx_playback_pcm_open(struct hda_pcm_stream *hinfo, return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream); } -/* - * set up the i/o for analog out - * when the digital out is available, copy the front out to digital out, too. - */ -static int stac92xx_multi_out_analog_prepare(struct hda_codec *codec, struct hda_multi_out *mout, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - hda_nid_t *nids = mout->dac_nids; - int chs = substream->runtime->channels; - int i; - - down(&codec->spdif_mutex); - if (mout->dig_out_nid && mout->dig_out_used != HDA_DIG_EXCLUSIVE) { - if (chs == 2 && - snd_hda_is_supported_format(codec, mout->dig_out_nid, format) && - ! (codec->spdif_status & IEC958_AES0_NONAUDIO)) { - mout->dig_out_used = HDA_DIG_ANALOG_DUP; - /* setup digital receiver */ - snd_hda_codec_setup_stream(codec, mout->dig_out_nid, - stream_tag, 0, format); - } else { - mout->dig_out_used = 0; - snd_hda_codec_setup_stream(codec, mout->dig_out_nid, 0, 0, 0); - } - } - up(&codec->spdif_mutex); - - /* front */ - snd_hda_codec_setup_stream(codec, nids[HDA_FRONT], stream_tag, 0, format); - if (mout->hp_nid) - /* headphone out will just decode front left/right (stereo) */ - snd_hda_codec_setup_stream(codec, mout->hp_nid, stream_tag, 0, format); - /* surrounds */ - if (mout->max_channels > 2) - for (i = 1; i < mout->num_dacs; i++) { - if ((mout->max_channels == 6) && (i == 3)) - break; - if (chs >= (i + 1) * 2) /* independent out */ - snd_hda_codec_setup_stream(codec, nids[i], stream_tag, i * 2, - format); - else /* copy front */ - snd_hda_codec_setup_stream(codec, nids[i], stream_tag, 0, - format); - } - return 0; -} - - static int stac92xx_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, @@ -383,8 +321,7 @@ static int stac92xx_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct sigmatel_spec *spec = codec->spec; - return stac92xx_multi_out_analog_prepare(codec, &spec->multiout, stream_tag, - format, substream); + return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag, format, substream); } static int stac92xx_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, @@ -511,14 +448,70 @@ static int stac92xx_build_pcms(struct hda_codec *codec) return 0; } +static void stac92xx_auto_set_pinctl(struct hda_codec *codec, hda_nid_t nid, int pin_type) + +{ + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type); +} + +static int stac92xx_io_switch_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 stac92xx_io_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + int io_idx = kcontrol-> private_value & 0xff; + + ucontrol->value.integer.value[0] = spec->io_switch[io_idx]; + return 0; +} + +static int stac92xx_io_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + hda_nid_t nid = kcontrol->private_value >> 8; + int io_idx = kcontrol-> private_value & 0xff; + unsigned short val = ucontrol->value.integer.value[0]; + + spec->io_switch[io_idx] = val; + + if (val) + stac92xx_auto_set_pinctl(codec, nid, AC_PINCTL_OUT_EN); + else + stac92xx_auto_set_pinctl(codec, nid, AC_PINCTL_IN_EN); + + return 1; +} + +#define STAC_CODEC_IO_SWITCH(xname, xpval) \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = 0, \ + .info = stac92xx_io_switch_info, \ + .get = stac92xx_io_switch_get, \ + .put = stac92xx_io_switch_put, \ + .private_value = xpval, \ + } + + enum { STAC_CTL_WIDGET_VOL, STAC_CTL_WIDGET_MUTE, + STAC_CTL_WIDGET_IO_SWITCH, }; static struct snd_kcontrol_new stac92xx_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), + STAC_CODEC_IO_SWITCH(NULL, 0), }; /* add dynamic controls */ @@ -550,6 +543,51 @@ static int stac92xx_add_control(struct sigmatel_spec *spec, int type, const char return 0; } +/* flag inputs as additional dynamic lineouts */ +static int stac92xx_add_dyn_out_pins(struct hda_codec *codec, struct auto_pin_cfg *cfg) +{ + struct sigmatel_spec *spec = codec->spec; + + switch (cfg->line_outs) { + case 3: + /* add line-in as side */ + if (cfg->input_pins[AUTO_PIN_LINE]) { + cfg->line_out_pins[3] = cfg->input_pins[AUTO_PIN_LINE]; + spec->line_switch = 1; + cfg->line_outs++; + } + break; + case 2: + /* add line-in as clfe and mic as side */ + if (cfg->input_pins[AUTO_PIN_LINE]) { + cfg->line_out_pins[2] = cfg->input_pins[AUTO_PIN_LINE]; + spec->line_switch = 1; + cfg->line_outs++; + } + if (cfg->input_pins[AUTO_PIN_MIC]) { + cfg->line_out_pins[3] = cfg->input_pins[AUTO_PIN_MIC]; + spec->mic_switch = 1; + cfg->line_outs++; + } + break; + case 1: + /* add line-in as surr and mic as clfe */ + if (cfg->input_pins[AUTO_PIN_LINE]) { + cfg->line_out_pins[1] = cfg->input_pins[AUTO_PIN_LINE]; + spec->line_switch = 1; + cfg->line_outs++; + } + if (cfg->input_pins[AUTO_PIN_MIC]) { + cfg->line_out_pins[2] = cfg->input_pins[AUTO_PIN_MIC]; + spec->mic_switch = 1; + cfg->line_outs++; + } + break; + } + + return 0; +} + /* fill in the dac_nids table from the parsed pin configuration */ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { @@ -578,7 +616,7 @@ static int stac92xx_auto_create_multi_out_ctls(struct sigmatel_spec *spec, const int i, err; for (i = 0; i < cfg->line_outs; i++) { - if (! spec->multiout.dac_nids[i]) + if (!spec->multiout.dac_nids[i]) continue; nid = spec->multiout.dac_nids[i]; @@ -609,6 +647,14 @@ static int stac92xx_auto_create_multi_out_ctls(struct sigmatel_spec *spec, const } } + if (spec->line_switch) + if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, "Line In as Output Switch", cfg->input_pins[AUTO_PIN_LINE] << 8)) < 0) + return err; + + if (spec->mic_switch) + if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, "Mic as Output Switch", (cfg->input_pins[AUTO_PIN_MIC] << 8) | 1)) < 0) + return err; + return 0; } @@ -666,6 +712,9 @@ 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 = labels[i]; for (j=0; jnum_muxes; j++) { @@ -686,12 +735,6 @@ static int stac92xx_auto_create_analog_input_ctls(struct hda_codec *codec, const return 0; } -static void stac92xx_auto_set_pinctl(struct hda_codec *codec, hda_nid_t nid, int pin_type) - -{ - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type); -} - static void stac92xx_auto_init_multi_out(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -720,6 +763,8 @@ static int stac922x_parse_auto_config(struct hda_codec *codec) if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg)) < 0) return err; + if ((err = stac92xx_add_dyn_out_pins(codec, &spec->autocfg)) < 0) + return err; if ((err = stac92xx_auto_fill_dac_nids(codec, &spec->autocfg)) < 0) return err; if (! spec->autocfg.line_outs && ! spec->autocfg.hp_pin) @@ -731,15 +776,8 @@ static int stac922x_parse_auto_config(struct hda_codec *codec) return err; spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->multiout.max_channels > 2) { + if (spec->multiout.max_channels > 2) spec->surr_switch = 1; - spec->cur_ch_mode = 1; - spec->num_ch_modes = 2; - if (spec->multiout.max_channels == 8) { - spec->cur_ch_mode++; - spec->num_ch_modes++; - } - } if (spec->autocfg.dig_out_pin) { spec->multiout.dig_out_nid = 0x08; @@ -901,13 +939,16 @@ static int patch_stac9200(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + spec->board_config = snd_hda_check_board_config(codec, stac9200_cfg_tbl); + if (spec->board_config < 0) + snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9200, using BIOS defaults\n"); + else { + spec->num_pins = 8; + spec->pin_nids = stac9200_pin_nids; + spec->pin_configs = stac9200_brd_tbl[spec->board_config]; + stac92xx_set_config_regs(codec); + } -#ifdef STAC_TEST - spec->pin_nids = stac9200_pin_nids; - spec->num_pins = 8; - spec->pin_configs = stac9200_pin_configs; - stac92xx_set_config_regs(codec); -#endif spec->multiout.max_channels = 2; spec->multiout.num_dacs = 1; spec->multiout.dac_nids = stac9200_dac_nids; @@ -939,13 +980,16 @@ static int patch_stac922x(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + spec->board_config = snd_hda_check_board_config(codec, stac922x_cfg_tbl); + if (spec->board_config < 0) + snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC922x, using BIOS defaults\n"); + else { + spec->num_pins = 10; + spec->pin_nids = stac922x_pin_nids; + spec->pin_configs = stac922x_brd_tbl[spec->board_config]; + stac92xx_set_config_regs(codec); + } -#ifdef STAC_TEST - spec->num_pins = 10; - spec->pin_nids = stac922x_pin_nids; - spec->pin_configs = stac922x_pin_configs; - stac92xx_set_config_regs(codec); -#endif spec->adc_nids = stac922x_adc_nids; spec->mux_nids = stac922x_mux_nids; spec->num_muxes = 2; -- cgit v1.2.3 From df694daa3c0135202e4702cb2d11e68a43f6c51e Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Mon, 5 Dec 2005 19:42:22 +0100 Subject: [ALSA] hda-codec - Add the support of ALC262,ALC883,ALC885,ALC861 Modules: HDA Codec driver,HDA generic driver This patch adds the support of ALC262,ALC883,ALC885,ALC861 to driver More models and improvements for ALC880, ALC260 and ALC882 codecs, too. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 16 +- sound/pci/hda/hda_local.h | 5 +- sound/pci/hda/patch_analog.c | 2 +- sound/pci/hda/patch_cmedia.c | 2 +- sound/pci/hda/patch_realtek.c | 2399 ++++++++++++++++++++++++++++++++++++---- sound/pci/hda/patch_sigmatel.c | 4 +- 6 files changed, 2215 insertions(+), 213 deletions(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5ead2a3d05ad..bd375f895ec0 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1926,8 +1926,18 @@ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, struct hda_multi_o /* * Helper for automatic ping configuration */ + +static int is_in_nid_list(hda_nid_t nid, hda_nid_t *list) +{ + for (; *list; list++) + if (*list == nid) + return 1; + return 0; +} + /* parse all pin widgets and store the useful pin nids to cfg */ -int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *cfg) +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; @@ -1948,6 +1958,10 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *c /* read all default configuration for pin complex */ if (wid_type != AC_WID_PIN) continue; + /* ignore the given nids (e.g. pc-beep returns error) */ + if (ignore_nids && is_in_nid_list(nid, ignore_nids)) + continue; + def_conf = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) continue; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index ded32359b654..a9863eb20c75 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -213,7 +213,7 @@ enum { struct auto_pin_cfg { int line_outs; - hda_nid_t line_out_pins[4]; /* sorted in the order of Front/Surr/CLFE/Side */ + hda_nid_t line_out_pins[5]; /* sorted in the order of Front/Surr/CLFE/Side */ hda_nid_t speaker_pin; hda_nid_t hp_pin; hda_nid_t input_pins[AUTO_PIN_LAST]; @@ -227,7 +227,8 @@ struct auto_pin_cfg { #define get_defcfg_sequence(cfg) (cfg & AC_DEFCFG_SEQUENCE) #define get_defcfg_device(cfg) ((cfg & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT) -int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *cfg); +int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *cfg, + hda_nid_t *ignore_nids); /* amp values */ #define AMP_IN_MUTE(idx) (0x7080 | ((idx)<<8)) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index fc144155f7a6..d1e1ded27532 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1968,7 +1968,7 @@ static int ad1988_parse_auto_config(struct hda_codec *codec) struct ad198x_spec *spec = codec->spec; int err; - if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg)) < 0) + if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL)) < 0) return err; if ((err = ad1988_auto_fill_dac_nids(codec, &spec->autocfg)) < 0) return err; diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 9a6981162982..d38ce22507ae 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -711,7 +711,7 @@ static int patch_cmi9880(struct hda_codec *codec) spec->dig_in_nid = CMI_DIG_IN_NID; spec->multiout.max_channels = 8; } - snd_hda_parse_pin_def_config(codec, &cfg); + snd_hda_parse_pin_def_config(codec, &cfg, NULL); if (cfg.line_outs) { spec->multiout.max_channels = cfg.line_outs * 2; cmi9880_fill_multi_dac_nids(codec, &cfg); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c5fb141f6222..a98c0e4da0ac 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3,7 +3,8 @@ * * HD audio interface patch for ALC 260/880/882 codecs * - * Copyright (c) 2004 PeiSen Hou + * Copyright (c) 2004 Kailang Yang + * PeiSen Hou * Takashi Iwai * * This driver is free software; you can redistribute it and/or modify @@ -39,17 +40,20 @@ enum { ALC880_5ST_DIG, ALC880_W810, ALC880_Z71V, - ALC880_AUTO, ALC880_6ST, ALC880_6ST_DIG, ALC880_F1734, ALC880_ASUS, ALC880_ASUS_DIG, ALC880_ASUS_W1V, + ALC880_ASUS_DIG2, ALC880_UNIWILL_DIG, + ALC880_CLEVO, + ALC880_TCL_S700, #ifdef CONFIG_SND_DEBUG ALC880_TEST, #endif + ALC880_AUTO, ALC880_MODEL_LAST /* last tag */ }; @@ -57,16 +61,45 @@ enum { enum { ALC260_BASIC, ALC260_HP, - ALC260_FUJITSU_S702x, + ALC260_HP_3013, + ALC260_FUJITSU_S702X, + ALC260_AUTO, ALC260_MODEL_LAST /* last tag */ }; +/* ALC262 models */ +enum { + ALC262_BASIC, + ALC262_AUTO, + ALC262_MODEL_LAST /* last tag */ +}; + +/* ALC861 models */ +enum { + ALC861_3ST, + ALC861_3ST_DIG, + ALC861_6ST_DIG, + ALC861_AUTO, + ALC861_MODEL_LAST, +}; + +/* ALC882 models */ +enum { + ALC882_3ST_DIG, + ALC882_6ST_DIG, + ALC882_AUTO, + ALC882_MODEL_LAST, +}; + +/* for GPIO Poll */ +#define GPIO_MASK 0x03 + struct alc_spec { /* codec parameterization */ - struct snd_kcontrol_new *mixers[3]; /* mixer arrays */ + struct snd_kcontrol_new *mixers[5]; /* mixer arrays */ unsigned int num_mixers; - const struct hda_verb *init_verbs[3]; /* initialization verbs + const struct hda_verb *init_verbs[5]; /* initialization verbs * don't forget NULL termination! */ unsigned int num_init_verbs; @@ -106,7 +139,25 @@ struct alc_spec { unsigned int num_kctl_alloc, num_kctl_used; struct snd_kcontrol_new *kctl_alloc; struct hda_input_mux private_imux; - hda_nid_t private_dac_nids[4]; + hda_nid_t private_dac_nids[5]; +}; + +/* + * configuration template - to be copied to the spec instance + */ +struct alc_config_preset { + struct snd_kcontrol_new *mixers[5]; /* should be identical size with spec */ + const struct hda_verb *init_verbs[5]; + unsigned int num_dacs; + hda_nid_t *dac_nids; + hda_nid_t dig_out_nid; /* optional */ + hda_nid_t hp_nid; /* optional */ + unsigned int num_adc_nids; + hda_nid_t *adc_nids; + hda_nid_t dig_in_nid; + unsigned int num_channel_mode; + const struct hda_channel_mode *channel_mode; + const struct hda_input_mux *input_mux; }; @@ -143,7 +194,7 @@ static int alc_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v /* * channel mode setting */ -static int alc880_ch_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +static int alc_ch_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; @@ -151,7 +202,7 @@ static int alc880_ch_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_ele spec->num_channel_mode); } -static int alc880_ch_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int alc_ch_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; @@ -159,7 +210,7 @@ static int alc880_ch_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem spec->num_channel_mode, spec->multiout.max_channels); } -static int alc880_ch_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int alc_ch_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; @@ -217,6 +268,36 @@ static int alc_pinctl_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e .put = alc_pinctl_switch_put, \ .private_value = (nid) | (mask<<16) } + +/* + * set up from the preset table + */ +static void setup_preset(struct alc_spec *spec, const struct alc_config_preset *preset) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(preset->mixers) && preset->mixers[i]; i++) + spec->mixers[spec->num_mixers++] = preset->mixers[i]; + for (i = 0; i < ARRAY_SIZE(preset->init_verbs) && preset->init_verbs[i]; i++) + spec->init_verbs[spec->num_init_verbs++] = preset->init_verbs[i]; + + spec->channel_mode = preset->channel_mode; + spec->num_channel_mode = preset->num_channel_mode; + + spec->multiout.max_channels = spec->channel_mode[0].channels; + + spec->multiout.num_dacs = preset->num_dacs; + spec->multiout.dac_nids = preset->dac_nids; + spec->multiout.dig_out_nid = preset->dig_out_nid; + spec->multiout.hp_nid = preset->hp_nid; + + spec->input_mux = preset->input_mux; + + spec->num_adc_nids = preset->num_adc_nids; + spec->adc_nids = preset->adc_nids; + spec->dig_in_nid = preset->dig_in_nid; +} + /* * ALC880 3-stack model * @@ -237,6 +318,7 @@ static hda_nid_t alc880_adc_nids[3] = { /* The datasheet says the node 0x07 is connected from inputs, * but it shows zero connection in the real implementation on some devices. + * Note: this is a 915GAV bug, fixed on 915GLV */ static hda_nid_t alc880_adc_nids_alt[2] = { /* ADC1-2 */ @@ -307,9 +389,9 @@ static struct snd_kcontrol_new alc880_three_stack_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Channel Mode", - .info = alc880_ch_mode_info, - .get = alc880_ch_mode_get, - .put = alc880_ch_mode_put, + .info = alc_ch_mode_info, + .get = alc_ch_mode_get, + .put = alc_ch_mode_put, }, { } /* end */ }; @@ -452,9 +534,9 @@ static struct snd_kcontrol_new alc880_six_stack_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Channel Mode", - .info = alc880_ch_mode_info, - .get = alc880_ch_mode_get, - .put = alc880_ch_mode_put, + .info = alc_ch_mode_info, + .get = alc_ch_mode_get, + .put = alc_ch_mode_put, }, { } /* end */ }; @@ -596,9 +678,9 @@ static struct snd_kcontrol_new alc880_asus_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Channel Mode", - .info = alc880_ch_mode_info, - .get = alc880_ch_mode_get, - .put = alc880_ch_mode_put, + .info = alc_ch_mode_info, + .get = alc_ch_mode_get, + .put = alc_ch_mode_put, }, { } /* end */ }; @@ -626,6 +708,33 @@ static struct snd_kcontrol_new alc880_pcbeep_mixer[] = { { } /* end */ }; +/* TCL S700 */ +static struct snd_kcontrol_new alc880_tcl_s700_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0B, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0B, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0B, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0B, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + * FIXME: the controls appear in the "playback" view! + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 1, + .info = alc_mux_enum_info, + .get = alc_mux_enum_get, + .put = alc_mux_enum_put, + }, + { } /* end */ +}; + /* * build control elements */ @@ -925,6 +1034,8 @@ static struct hda_verb alc880_gpio1_init_verbs[] = { {0x01, AC_VERB_SET_GPIO_MASK, 0x01}, {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01}, {0x01, AC_VERB_SET_GPIO_DATA, 0x01}, + + { } }; /* Enable GPIO mask and set output */ @@ -932,8 +1043,59 @@ static struct hda_verb alc880_gpio2_init_verbs[] = { {0x01, AC_VERB_SET_GPIO_MASK, 0x02}, {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02}, {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, + + { } +}; + +/* Clevo m520g init */ +static struct hda_verb alc880_pin_clevo_init_verbs[] = { + /* headphone output */ + {0x11, AC_VERB_SET_CONNECT_SEL, 0x01}, + /* line-out */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Line-in */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* CD */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Mic1 (rear panel) */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Mic2 (front panel) */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* headphone */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* change to EAPD mode */ + {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, + {0x20, AC_VERB_SET_PROC_COEF, 0x3060}, + + { } }; +static struct hda_verb alc880_pin_tcl_S700_init_verbs[] = { + /* Headphone output */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Front output*/ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Line In pin widget for input */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + /* CD pin widget for input */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + /* Mic1 (rear panel) pin widget for input and vref at 80% */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + + /* change to EAPD mode */ + {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, + {0x20, AC_VERB_SET_PROC_COEF, 0x3070}, + + { } +}; /* */ @@ -1344,9 +1506,9 @@ static struct snd_kcontrol_new alc880_test_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Channel Mode", - .info = alc880_ch_mode_info, - .get = alc880_ch_mode_get, - .put = alc880_ch_mode_put, + .info = alc_ch_mode_info, + .get = alc_ch_mode_get, + .put = alc_ch_mode_put, }, { } /* end */ }; @@ -1442,6 +1604,8 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .pci_subvendor = 0x107b, .pci_subdevice = 0x4038, .config = ALC880_3ST }, { .pci_subvendor = 0x107b, .pci_subdevice = 0x4040, .config = ALC880_3ST }, { .pci_subvendor = 0x107b, .pci_subdevice = 0x4041, .config = ALC880_3ST }, + /* TCL S700 */ + { .pci_subvendor = 0x19db, .pci_subdevice = 0x4188, .config = ALC880_TCL_S700 }, /* Back 3 jack, front 2 jack (Internal add Aux-In) */ { .pci_subvendor = 0x1025, .pci_subdevice = 0xe310, .config = ALC880_3ST }, @@ -1452,6 +1616,8 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .modelname = "3stack-digout", .config = ALC880_3ST_DIG }, { .pci_subvendor = 0x8086, .pci_subdevice = 0xe308, .config = ALC880_3ST_DIG }, { .pci_subvendor = 0x1025, .pci_subdevice = 0x0070, .config = ALC880_3ST_DIG }, + /* Clevo m520G NB */ + { .pci_subvendor = 0x1558, .pci_subdevice = 0x0520, .config = ALC880_CLEVO }, /* Back 3 jack plus 1 SPDIF out jack, front 2 jack (Internal add Aux-In)*/ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe305, .config = ALC880_3ST_DIG }, @@ -1489,6 +1655,7 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .pci_subvendor = 0x1043, .pci_subdevice = 0x1964, .config = ALC880_Z71V }, { .modelname = "6stack", .config = ALC880_6ST }, + { .pci_subvendor = 0x1043, .pci_subdevice = 0x81b4, .config = ALC880_6ST }, { .pci_subvendor = 0x1019, .pci_subdevice = 0xa884, .config = ALC880_6ST }, /* Acer APFV */ { .modelname = "6stack-digout", .config = ALC880_6ST_DIG }, @@ -1496,6 +1663,10 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .pci_subvendor = 0x8086, .pci_subdevice = 0x2668, .config = ALC880_6ST_DIG }, { .pci_subvendor = 0x1462, .pci_subdevice = 0x1150, .config = ALC880_6ST_DIG }, { .pci_subvendor = 0xe803, .pci_subdevice = 0x1019, .config = ALC880_6ST_DIG }, + { .pci_subvendor = 0x1039, .pci_subdevice = 0x1234, .config = ALC880_6ST_DIG }, + { .pci_subvendor = 0x1025, .pci_subdevice = 0x0077, .config = ALC880_6ST_DIG }, + { .pci_subvendor = 0x1025, .pci_subdevice = 0x0078, .config = ALC880_6ST_DIG }, + { .pci_subvendor = 0x1025, .pci_subdevice = 0x0087, .config = ALC880_6ST_DIG }, { .modelname = "asus", .config = ALC880_ASUS }, { .pci_subvendor = 0x1043, .pci_subdevice = 0x1964, .config = ALC880_ASUS_DIG }, @@ -1509,37 +1680,26 @@ 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 = 0x1558, .pci_subdevice = 0x5401, .config = ALC880_ASUS_DIG2 }, { .modelname = "uniwill", .config = ALC880_UNIWILL_DIG }, { .pci_subvendor = 0x1584, .pci_subdevice = 0x9050, .config = ALC880_UNIWILL_DIG }, { .modelname = "F1734", .config = ALC880_F1734 }, { .pci_subvendor = 0x1734, .pci_subdevice = 0x107c, .config = ALC880_F1734 }, + { .pci_subvendor = 0x1584, .pci_subdevice = 0x9054, .config = ALC880_F1734 }, #ifdef CONFIG_SND_DEBUG { .modelname = "test", .config = ALC880_TEST }, #endif + { .modelname = "auto", .config = ALC880_AUTO }, {} }; /* - * configuration template - to be copied to the spec instance + * ALC880 codec presets */ -struct alc_config_preset { - struct snd_kcontrol_new *mixers[4]; - const struct hda_verb *init_verbs[4]; - unsigned int num_dacs; - hda_nid_t *dac_nids; - hda_nid_t dig_out_nid; /* optional */ - hda_nid_t hp_nid; /* optional */ - unsigned int num_adc_nids; - hda_nid_t *adc_nids; - unsigned int num_channel_mode; - const struct hda_channel_mode *channel_mode; - const struct hda_input_mux *input_mux; -}; - static struct alc_config_preset alc880_presets[] = { [ALC880_3ST] = { .mixers = { alc880_three_stack_mixer }, @@ -1560,6 +1720,18 @@ static struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_threestack_modes, .input_mux = &alc880_capture_source, }, + [ALC880_TCL_S700] = { + .mixers = { alc880_tcl_s700_mixer }, + .init_verbs = { alc880_volume_init_verbs, + alc880_pin_tcl_S700_init_verbs, + alc880_gpio2_init_verbs }, + .num_dacs = ARRAY_SIZE(alc880_dac_nids), + .dac_nids = alc880_dac_nids, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes), + .channel_mode = alc880_2_jack_modes, + .input_mux = &alc880_capture_source, + }, [ALC880_5ST] = { .mixers = { alc880_three_stack_mixer, alc880_five_stack_mixer}, .init_verbs = { alc880_volume_init_verbs, alc880_pin_5stack_init_verbs }, @@ -1651,6 +1823,17 @@ static struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_asus_modes, .input_mux = &alc880_capture_source, }, + [ALC880_ASUS_DIG2] = { + .mixers = { alc880_asus_mixer }, + .init_verbs = { alc880_volume_init_verbs, alc880_pin_asus_init_verbs, + alc880_gpio2_init_verbs }, /* use GPIO2 */ + .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids), + .dac_nids = alc880_asus_dac_nids, + .dig_out_nid = ALC880_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc880_asus_modes), + .channel_mode = alc880_asus_modes, + .input_mux = &alc880_capture_source, + }, [ALC880_ASUS_W1V] = { .mixers = { alc880_asus_mixer, alc880_asus_w1v_mixer }, .init_verbs = { alc880_volume_init_verbs, alc880_pin_asus_init_verbs, @@ -1672,6 +1855,17 @@ static struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_asus_modes, .input_mux = &alc880_capture_source, }, + [ALC880_CLEVO] = { + .mixers = { alc880_three_stack_mixer }, + .init_verbs = { alc880_volume_init_verbs, + alc880_pin_clevo_init_verbs }, + .num_dacs = ARRAY_SIZE(alc880_dac_nids), + .dac_nids = alc880_dac_nids, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), + .channel_mode = alc880_threestack_modes, + .input_mux = &alc880_capture_source, + }, #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = { .mixers = { alc880_test_mixer }, @@ -1783,7 +1977,8 @@ static int alc880_auto_fill_dac_nids(struct alc_spec *spec, const struct auto_pi } /* add playback controls from the parsed DAC table */ -static int alc880_auto_create_multi_out_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) +static int alc880_auto_create_multi_out_ctls(struct alc_spec *spec, + const struct auto_pin_cfg *cfg) { char name[32]; static const char *chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; @@ -1871,35 +2066,38 @@ static int alc880_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin, } /* create input playback/capture controls for the given pin */ -static int new_analog_input(struct alc_spec *spec, hda_nid_t pin, const char *ctlname) +static int new_analog_input(struct alc_spec *spec, hda_nid_t pin, const char *ctlname, + int idx, hda_nid_t mix_nid) { char name[32]; - int err, idx; + int err; sprintf(name, "%s Playback Volume", ctlname); - idx = alc880_input_pin_idx(pin); if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(0x0b, 3, idx, HDA_INPUT))) < 0) + HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT))) < 0) return err; sprintf(name, "%s Playback Switch", ctlname); if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, - HDA_COMPOSE_AMP_VAL(0x0b, 3, idx, HDA_INPUT))) < 0) + HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT))) < 0) return err; return 0; } /* create playback/capture controls for input pins */ -static int alc880_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) +static int alc880_auto_create_analog_input_ctls(struct alc_spec *spec, + const struct auto_pin_cfg *cfg) { static char *labels[AUTO_PIN_LAST] = { "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux" }; struct hda_input_mux *imux = &spec->private_imux; - int i, err; + int i, err, idx; for (i = 0; i < AUTO_PIN_LAST; i++) { if (alc880_is_input_pin(cfg->input_pins[i])) { - err = new_analog_input(spec, cfg->input_pins[i], labels[i]); + idx = alc880_input_pin_idx(cfg->input_pins[i]); + err = new_analog_input(spec, cfg->input_pins[i], labels[i], + idx, 0x0b); if (err < 0) return err; imux->items[imux->num_items].label = labels[i]; @@ -1910,7 +2108,8 @@ static int alc880_auto_create_analog_input_ctls(struct alc_spec *spec, const str return 0; } -static void alc880_auto_set_output_and_unmute(struct hda_codec *codec, hda_nid_t nid, int pin_type, +static void alc880_auto_set_output_and_unmute(struct hda_codec *codec, + hda_nid_t nid, int pin_type, int dac_idx) { /* set as output */ @@ -1973,15 +2172,17 @@ static int alc880_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; int err; + static hda_nid_t alc880_ignore[] = { 0x1d, 0 }; - if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg)) < 0) - return err; - if ((err = alc880_auto_fill_dac_nids(spec, &spec->autocfg)) < 0) + 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) return 0; /* can't find valid BIOS pin config */ - if ((err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 || + + 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, "Speaker")) < 0 || (err = alc880_auto_create_extra_out(spec, spec->autocfg.speaker_pin, @@ -2024,7 +2225,7 @@ static int patch_alc880(struct hda_codec *codec) { struct alc_spec *spec; int board_config; - int i, err; + int err; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -2050,36 +2251,8 @@ static int patch_alc880(struct hda_codec *codec) } } - if (board_config != ALC880_AUTO) { - /* set up from the preset table */ - const struct alc_config_preset *preset; - - preset = &alc880_presets[board_config]; - - for (i = 0; preset->mixers[i]; i++) { - snd_assert(spec->num_mixers < ARRAY_SIZE(spec->mixers), break); - spec->mixers[spec->num_mixers++] = preset->mixers[i]; - } - for (i = 0; preset->init_verbs[i]; i++) { - snd_assert(spec->num_init_verbs < ARRAY_SIZE(spec->init_verbs), break); - spec->init_verbs[spec->num_init_verbs++] = preset->init_verbs[i]; - } - - spec->channel_mode = preset->channel_mode; - spec->num_channel_mode = preset->num_channel_mode; - - spec->multiout.max_channels = spec->channel_mode[0].channels; - - spec->multiout.num_dacs = preset->num_dacs; - spec->multiout.dac_nids = preset->dac_nids; - spec->multiout.dig_out_nid = preset->dig_out_nid; - spec->multiout.hp_nid = preset->hp_nid; - - spec->input_mux = preset->input_mux; - - spec->num_adc_nids = preset->num_adc_nids; - spec->adc_nids = preset->adc_nids; - } + if (board_config != ALC880_AUTO) + setup_preset(spec, &alc880_presets[board_config]); spec->stream_name_analog = "ALC880 Analog"; spec->stream_analog_playback = &alc880_pcm_analog_playback; @@ -2128,11 +2301,16 @@ static hda_nid_t alc260_adc_nids[1] = { 0x04, }; -static hda_nid_t alc260_hp_adc_nids[1] = { +static hda_nid_t alc260_adc_nids_alt[1] = { /* ADC1 */ 0x05, }; +static hda_nid_t alc260_hp_adc_nids[2] = { + /* ADC1, 0 */ + 0x05, 0x04 +}; + #define ALC260_DIGOUT_NID 0x03 #define ALC260_DIGIN_NID 0x06 @@ -2167,38 +2345,26 @@ static struct hda_channel_mode alc260_modes[1] = { { 2, NULL }, }; -static struct snd_kcontrol_new alc260_base_mixer[] = { + +/* Mixer combinations + * + * basic: base_output + input + pc_beep + capture + * HP: base_output + input + capture_alt + * HP_3013: hp_3013 + input + capture + * fujitsu: fujitsu + capture + */ + +static struct snd_kcontrol_new alc260_base_output_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front 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("Line Playback Volume", 0x07, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x07, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x07, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x07, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x07, 0x05, HDA_INPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT), 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("Capture Volume", 0x04, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x04, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = alc_mux_enum_info, - .get = alc_mux_enum_get, - .put = alc_mux_enum_put, - }, { } /* end */ -}; +}; -static struct snd_kcontrol_new alc260_hp_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT), +static struct snd_kcontrol_new alc260_input_mixer[] = { HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT), @@ -2207,19 +2373,24 @@ static struct snd_kcontrol_new alc260_hp_mixer[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x07, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x07, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT), - 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("Capture Volume", 0x05, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x05, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = alc_mux_enum_info, - .get = alc_mux_enum_get, - .put = alc_mux_enum_put, - }, + { } /* end */ +}; + +static struct snd_kcontrol_new alc260_pc_beep_mixer[] = { + HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x07, 0x05, HDA_INPUT), + HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x07, 0x05, HDA_INPUT), + { } /* end */ +}; + +static struct snd_kcontrol_new alc260_hp_3013_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x09, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x10, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Aux-In Playback Volume", 0x07, 0x06, HDA_INPUT), + HDA_CODEC_MUTE("Aux-In Playback Switch", 0x07, 0x06, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("iSpeaker Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("iSpeaker Playback Switch", 0x11, 1, 0x0, HDA_OUTPUT), { } /* end */ }; @@ -2235,11 +2406,43 @@ static struct snd_kcontrol_new alc260_fujitsu_mixer[] = { HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT), HDA_CODEC_VOLUME("Internal Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Internal Speaker Playback Switch", 0x09, 2, HDA_INPUT), + { } /* end */ +}; + +/* capture mixer elements */ +static struct snd_kcontrol_new alc260_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x04, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x04, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x05, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x05, 0x0, HDA_INPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + * FIXME: the controls appear in the "playback" view! + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc_mux_enum_info, + .get = alc_mux_enum_get, + .put = alc_mux_enum_put, + }, + { } /* end */ +}; + +static struct snd_kcontrol_new alc260_capture_alt_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x05, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x05, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + * FIXME: the controls appear in the "playback" view! + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 1, .info = alc_mux_enum_info, .get = alc_mux_enum_get, .put = alc_mux_enum_put, @@ -2247,6 +2450,9 @@ static struct snd_kcontrol_new alc260_fujitsu_mixer[] = { { } /* end */ }; +/* + * initialization verbs + */ static struct hda_verb alc260_init_verbs[] = { /* Line In pin widget for input */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, @@ -2308,6 +2514,100 @@ static struct hda_verb alc260_init_verbs[] = { { } }; +static struct hda_verb alc260_hp_init_verbs[] = { + /* Headphone and output */ + {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0}, + /* mono output */ + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* Mic1 (rear panel) pin widget for input and vref at 80% */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + /* Mic2 (front panel) pin widget for input and vref at 80% */ + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + /* Line In pin widget for input */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + /* Line-2 pin widget for output */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* CD pin widget for input */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + /* unmute amp left and right */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, + /* set connection select to line in (default select for this ADC) */ + {0x04, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* unmute Line-Out mixer amp left and right (volume = 0) */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + /* mute pin widget amp left and right (no gain on this amp) */ + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, + /* unmute HP mixer amp left and right (volume = 0) */ + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + /* mute pin widget amp left and right (no gain on this amp) */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, + /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 & Line In 2 = 0x03 */ + /* unmute CD */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))}, + /* unmute Line In */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, + /* unmute Mic */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + /* Amp Indexes: DAC = 0x01 & mixer = 0x00 */ + /* Unmute Front out path */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + /* Unmute Headphone out path */ + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + /* Unmute Mono out path */ + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + { } +}; + +static struct hda_verb alc260_hp_3013_init_verbs[] = { + /* Line out and output */ + {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* mono output */ + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* Mic1 (rear panel) pin widget for input and vref at 80% */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + /* Mic2 (front panel) pin widget for input and vref at 80% */ + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + /* Line In pin widget for input */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + /* Headphone pin widget for output */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0}, + /* CD pin widget for input */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + /* unmute amp left and right */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, + /* set connection select to line in (default select for this ADC) */ + {0x04, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* unmute Line-Out mixer amp left and right (volume = 0) */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + /* mute pin widget amp left and right (no gain on this amp) */ + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, + /* unmute HP mixer amp left and right (volume = 0) */ + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + /* mute pin widget amp left and right (no gain on this amp) */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, + /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 & Line In 2 = 0x03 */ + /* unmute CD */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))}, + /* unmute Line In */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, + /* unmute Mic */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + /* Amp Indexes: DAC = 0x01 & mixer = 0x00 */ + /* Unmute Front out path */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + /* Unmute Headphone out path */ + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + /* Unmute Mono out path */ + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + { } +}; + /* Initialisation sequence for ALC260 as configured in Fujitsu S702x * laptops. */ @@ -2374,18 +2674,337 @@ static struct hda_pcm_stream alc260_pcm_analog_capture = { .channels_max = 2, }; -static struct hda_board_config alc260_cfg_tbl[] = { - { .modelname = "hp", .config = ALC260_HP }, - { .pci_subvendor = 0x103c, .config = ALC260_HP }, - { .modelname = "fujitsu", .config = ALC260_FUJITSU_S702x }, - { .pci_subvendor = 0x10cf, .pci_subdevice = 0x1326, .config = ALC260_FUJITSU_S702x }, - {} -}; +/* + * for BIOS auto-configuration + */ -static int patch_alc260(struct hda_codec *codec) +static int alc260_add_playback_controls(struct alc_spec *spec, hda_nid_t nid, + const char *pfx) +{ + hda_nid_t nid_vol; + unsigned long vol_val, sw_val; + char name[32]; + int err; + + if (nid >= 0x0f && nid < 0x11) { + nid_vol = nid - 0x7; + vol_val = HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT); + sw_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT); + } else if (nid == 0x11) { + nid_vol = nid - 0x7; + vol_val = HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, HDA_OUTPUT); + sw_val = HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT); + } else if (nid >= 0x12 && nid <= 0x15) { + nid_vol = 0x08; + vol_val = HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT); + sw_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT); + } else + return 0; /* N/A */ + + snprintf(name, sizeof(name), "%s Playback Volume", pfx); + if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, name, vol_val)) < 0) + return err; + snprintf(name, sizeof(name), "%s Playback Switch", pfx); + if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, sw_val)) < 0) + return err; + return 1; +} + +/* add playback controls from the parsed DAC table */ +static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec, + const struct auto_pin_cfg *cfg) +{ + hda_nid_t nid; + int err; + + spec->multiout.num_dacs = 1; + spec->multiout.dac_nids = spec->private_dac_nids; + spec->multiout.dac_nids[0] = 0x02; + + nid = cfg->line_out_pins[0]; + if (nid) { + err = alc260_add_playback_controls(spec, nid, "Front"); + if (err < 0) + return err; + } + + nid = cfg->speaker_pin; + if (nid) { + err = alc260_add_playback_controls(spec, nid, "Speaker"); + if (err < 0) + return err; + } + + nid = cfg->hp_pin; + if (nid) { + err = alc260_add_playback_controls(spec, nid, "Headphone"); + if (err < 0) + return err; + } + return 0; +} + +/* create playback/capture controls for input pins */ +static int alc260_auto_create_analog_input_ctls(struct alc_spec *spec, + const struct auto_pin_cfg *cfg) +{ + static char *labels[AUTO_PIN_LAST] = { + "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux" + }; + struct hda_input_mux *imux = &spec->private_imux; + int i, err, idx; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + if (cfg->input_pins[i] >= 0x12) { + idx = cfg->input_pins[i] - 0x12; + err = new_analog_input(spec, cfg->input_pins[i], labels[i], idx, 0x07); + if (err < 0) + return err; + imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].index = idx; + imux->num_items++; + } + if ((cfg->input_pins[i] >= 0x0f) && (cfg->input_pins[i] <= 0x10)){ + idx = cfg->input_pins[i] - 0x09; + err = new_analog_input(spec, cfg->input_pins[i], labels[i], idx, 0x07); + if (err < 0) + return err; + imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].index = idx; + imux->num_items++; + } + } + return 0; +} + +static void alc260_auto_set_output_and_unmute(struct hda_codec *codec, + hda_nid_t nid, int pin_type, + int sel_idx) +{ + /* set as output */ + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type); + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); + /* need the manual connection? */ + if (nid >= 0x12) { + int idx = nid - 0x12; + snd_hda_codec_write(codec, idx + 0x0b, 0, + AC_VERB_SET_CONNECT_SEL, sel_idx); + + } +} + +static void alc260_auto_init_multi_out(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t nid; + + nid = spec->autocfg.line_out_pins[0]; + if (nid) + alc260_auto_set_output_and_unmute(codec, nid, PIN_OUT, 0); + + nid = spec->autocfg.speaker_pin; + if (nid) + alc260_auto_set_output_and_unmute(codec, nid, PIN_OUT, 0); + + nid = spec->autocfg.hp_pin; + if (nid) + alc260_auto_set_output_and_unmute(codec, nid, PIN_OUT, 0); +} + +#define ALC260_PIN_CD_NID 0x16 +static void alc260_auto_init_analog_input(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int i; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + hda_nid_t nid = spec->autocfg.input_pins[i]; + if (nid >= 0x12) { + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + i <= AUTO_PIN_FRONT_MIC ? PIN_VREF80 : PIN_IN); + if (nid != ALC260_PIN_CD_NID) + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_MUTE); + } + } +} + +/* + * generic initialization of ADC, input mixers and output mixers + */ +static struct hda_verb alc260_volume_init_verbs[] = { + /* + * Unmute ADC0-1 and set the default input to mic-in + */ + {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x05, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + * Note: PASD motherboards uses the Line In 2 as the input for front panel + * mic (mic 2) + */ + /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, + + /* + * Set up output mixers (0x08 - 0x0a) + */ + /* set vol=0 to output mixers */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + /* set up input amps for analog loopback */ + /* Amp Indices: DAC = 0, mixer = 1 */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + { } +}; + +static int alc260_parse_auto_config(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + unsigned int wcap; + int err; + static hda_nid_t alc260_ignore[] = { 0x17, 0 }; + + if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, + alc260_ignore)) < 0) + return err; + if (! spec->autocfg.line_outs && ! spec->autocfg.speaker_pin && + ! spec->autocfg.hp_pin) + return 0; /* can't find valid BIOS pin config */ + if ((err = alc260_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 || + (err = alc260_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0) + return err; + + spec->multiout.max_channels = 2; + + if (spec->autocfg.dig_out_pin) + spec->multiout.dig_out_nid = ALC260_DIGOUT_NID; + if (spec->kctl_alloc) + spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + + spec->init_verbs[spec->num_init_verbs++] = alc260_volume_init_verbs; + + spec->input_mux = &spec->private_imux; + + /* check whether NID 0x04 is valid */ + wcap = snd_hda_param_read(codec, alc260_adc_nids[0], AC_PAR_AUDIO_WIDGET_CAP); + wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; /* get type */ + if (wcap != AC_WID_AUD_IN) { + spec->adc_nids = alc260_adc_nids_alt; + spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt); + spec->mixers[spec->num_mixers] = alc260_capture_alt_mixer; + spec->num_mixers++; + } else { + spec->adc_nids = alc260_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids); + spec->mixers[spec->num_mixers] = alc260_capture_mixer; + spec->num_mixers++; + } + + return 1; +} + +/* init callback for auto-configuration model -- overriding the default init */ +static int alc260_auto_init(struct hda_codec *codec) +{ + alc_init(codec); + alc260_auto_init_multi_out(codec); + alc260_auto_init_analog_input(codec); + return 0; +} + +/* + * ALC260 configurations + */ +static struct hda_board_config alc260_cfg_tbl[] = { + { .modelname = "basic", .config = ALC260_BASIC }, + { .modelname = "hp", .config = ALC260_HP }, + { .pci_subvendor = 0x103c, .pci_subdevice = 0x3010, .config = ALC260_HP }, + { .pci_subvendor = 0x103c, .pci_subdevice = 0x3011, .config = ALC260_HP }, + { .pci_subvendor = 0x103c, .pci_subdevice = 0x3012, .config = ALC260_HP }, + { .pci_subvendor = 0x103c, .pci_subdevice = 0x3013, .config = ALC260_HP_3013 }, + { .pci_subvendor = 0x103c, .pci_subdevice = 0x3014, .config = ALC260_HP }, + { .pci_subvendor = 0x103c, .pci_subdevice = 0x3015, .config = ALC260_HP }, + { .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 = "auto", .config = ALC260_AUTO }, + {} +}; + +static struct alc_config_preset alc260_presets[] = { + [ALC260_BASIC] = { + .mixers = { alc260_base_output_mixer, + alc260_input_mixer, + alc260_pc_beep_mixer, + alc260_capture_mixer }, + .init_verbs = { alc260_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_channel_mode = ARRAY_SIZE(alc260_modes), + .channel_mode = alc260_modes, + .input_mux = &alc260_capture_source, + }, + [ALC260_HP] = { + .mixers = { alc260_base_output_mixer, + alc260_input_mixer, + alc260_capture_alt_mixer }, + .init_verbs = { alc260_hp_init_verbs }, + .num_dacs = ARRAY_SIZE(alc260_dac_nids), + .dac_nids = alc260_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc260_hp_adc_nids), + .adc_nids = alc260_hp_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc260_modes), + .channel_mode = alc260_modes, + .input_mux = &alc260_capture_source, + }, + [ALC260_HP_3013] = { + .mixers = { alc260_hp_3013_mixer, + alc260_input_mixer, + alc260_capture_alt_mixer }, + .init_verbs = { alc260_hp_3013_init_verbs }, + .num_dacs = ARRAY_SIZE(alc260_dac_nids), + .dac_nids = alc260_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc260_hp_adc_nids), + .adc_nids = alc260_hp_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc260_modes), + .channel_mode = alc260_modes, + .input_mux = &alc260_capture_source, + }, + [ALC260_FUJITSU_S702X] = { + .mixers = { alc260_fujitsu_mixer, + alc260_capture_mixer }, + .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_channel_mode = ARRAY_SIZE(alc260_modes), + .channel_mode = alc260_modes, + .input_mux = &alc260_fujitsu_capture_source, + }, +}; + +static int patch_alc260(struct hda_codec *codec) { struct alc_spec *spec; - int board_config; + int err, board_config; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -2396,60 +3015,31 @@ static int patch_alc260(struct hda_codec *codec) board_config = snd_hda_check_board_config(codec, alc260_cfg_tbl); if (board_config < 0 || board_config >= ALC260_MODEL_LAST) { snd_printd(KERN_INFO "hda_codec: Unknown model for ALC260\n"); - board_config = ALC260_BASIC; - } - - switch (board_config) { - case ALC260_HP: - spec->mixers[spec->num_mixers] = alc260_hp_mixer; - spec->num_mixers++; - break; - case ALC260_FUJITSU_S702x: - spec->mixers[spec->num_mixers] = alc260_fujitsu_mixer; - spec->num_mixers++; - break; - default: - spec->mixers[spec->num_mixers] = alc260_base_mixer; - spec->num_mixers++; - break; + board_config = ALC260_AUTO; } - if (board_config != ALC260_FUJITSU_S702x) { - spec->init_verbs[0] = alc260_init_verbs; - spec->num_init_verbs = 1; - } else { - spec->init_verbs[0] = alc260_fujitsu_init_verbs; - spec->num_init_verbs = 1; + if (board_config == ALC260_AUTO) { + /* automatic parse from the BIOS config */ + err = alc260_parse_auto_config(codec); + if (err < 0) { + alc_free(codec); + return err; + } else if (! err) { + printk(KERN_INFO "hda_codec: Cannot set up configuration from BIOS. Using base mode...\n"); + board_config = ALC260_BASIC; + } } - spec->channel_mode = alc260_modes; - spec->num_channel_mode = ARRAY_SIZE(alc260_modes); + if (board_config != ALC260_AUTO) + setup_preset(spec, &alc260_presets[board_config]); spec->stream_name_analog = "ALC260 Analog"; spec->stream_analog_playback = &alc260_pcm_analog_playback; spec->stream_analog_capture = &alc260_pcm_analog_capture; - spec->multiout.max_channels = spec->channel_mode[0].channels; - spec->multiout.num_dacs = ARRAY_SIZE(alc260_dac_nids); - spec->multiout.dac_nids = alc260_dac_nids; - - if (board_config != ALC260_FUJITSU_S702x) { - spec->input_mux = &alc260_capture_source; - } else { - spec->input_mux = &alc260_fujitsu_capture_source; - } - switch (board_config) { - case ALC260_HP: - spec->num_adc_nids = ARRAY_SIZE(alc260_hp_adc_nids); - spec->adc_nids = alc260_hp_adc_nids; - break; - default: - spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids); - spec->adc_nids = alc260_adc_nids; - break; - } - codec->patch_ops = alc_patch_ops; + if (board_config == ALC260_AUTO) + codec->patch_ops.init = alc260_auto_init; return 0; } @@ -2466,6 +3056,8 @@ static int patch_alc260(struct hda_codec *codec) * In addition, an independent DAC for the multi-playback (not used in this * driver yet). */ +#define ALC882_DIGOUT_NID 0x06 +#define ALC882_DIGIN_NID 0x0a static struct hda_channel_mode alc882_ch_modes[1] = { { 8, NULL } @@ -2476,10 +3068,9 @@ static hda_nid_t alc882_dac_nids[4] = { 0x02, 0x03, 0x04, 0x05 }; -static hda_nid_t alc882_adc_nids[3] = { - /* ADC0-2 */ - 0x07, 0x08, 0x09, -}; +/* identical with ALC880 */ +#define alc882_adc_nids alc880_adc_nids +#define alc882_adc_nids_alt alc880_adc_nids_alt /* input MUX */ /* FIXME: should be a matrix-type input source selection */ @@ -2522,6 +3113,33 @@ static int alc882_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ele return 1; } +/* + * 6ch mode + */ +static struct hda_verb alc882_sixstack_ch6_init[] = { + { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 }, + { 0x16, 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 }, + { } /* end */ +}; + +/* + * 8ch mode + */ +static struct hda_verb alc882_sixstack_ch8_init[] = { + { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x16, 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 }, + { } /* end */ +}; + +static struct hda_channel_mode alc882_sixstack_modes[2] = { + { 6, alc882_sixstack_ch6_init }, + { 8, alc882_sixstack_ch8_init }, +}; + /* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17 * Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b */ @@ -2565,6 +3183,17 @@ static struct snd_kcontrol_new alc882_base_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc882_chmode_mixer[] = { + { + .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 alc882_init_verbs[] = { /* Front mixer: unmute input/output amp left and right (volume = 0) */ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, @@ -2645,45 +3274,1399 @@ static struct hda_verb alc882_init_verbs[] = { { } }; -static int patch_alc882(struct hda_codec *codec) -{ - struct alc_spec *spec; - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - codec->spec = spec; - - spec->mixers[spec->num_mixers] = alc882_base_mixer; - spec->num_mixers++; +/* + * generic initialization of ADC, input mixers and output mixers + */ +static struct hda_verb alc882_auto_init_verbs[] = { + /* + * Unmute ADC0-2 and set the default input to mic-in + */ + {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - spec->multiout.dig_out_nid = ALC880_DIGOUT_NID; - spec->dig_in_nid = ALC880_DIGIN_NID; - spec->init_verbs[0] = alc882_init_verbs; - spec->num_init_verbs = 1; + /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + * Note: PASD motherboards uses the Line In 2 as the input for front panel + * mic (mic 2) + */ + /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, - spec->channel_mode = alc882_ch_modes; - spec->num_channel_mode = ARRAY_SIZE(alc882_ch_modes); + /* + * Set up output mixers (0x0c - 0x0f) + */ + /* set vol=0 to output mixers */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + /* set up input amps for analog loopback */ + /* Amp Indices: DAC = 0, mixer = 1 */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + /* FIXME: use matrix-type input source selection */ + /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ + /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, + /* Input mixer2 */ + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, + /* Input mixer3 */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, + + { } +}; + +/* capture mixer elements */ +static struct snd_kcontrol_new alc882_capture_alt_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + * FIXME: the controls appear in the "playback" view! + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc882_mux_enum_info, + .get = alc882_mux_enum_get, + .put = alc882_mux_enum_put, + }, + { } /* end */ +}; + +static struct snd_kcontrol_new alc882_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + * FIXME: the controls appear in the "playback" view! + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 3, + .info = alc882_mux_enum_info, + .get = alc882_mux_enum_get, + .put = alc882_mux_enum_put, + }, + { } /* end */ +}; + +/* pcm configuration: identiacal with ALC880 */ +#define alc882_pcm_analog_playback alc880_pcm_analog_playback +#define alc882_pcm_analog_capture alc880_pcm_analog_capture +#define alc882_pcm_digital_playback alc880_pcm_digital_playback +#define alc882_pcm_digital_capture alc880_pcm_digital_capture + +/* + * configuration and preset + */ +static struct hda_board_config alc882_cfg_tbl[] = { + { .modelname = "3stack-dig", .config = ALC861_3ST_DIG }, + { .modelname = "6stack-dig", .config = ALC861_6ST_DIG }, + { .pci_subvendor = 0x1462, .pci_subdevice = 0x6668, .config = ALC882_6ST_DIG }, /* MSI */ + { .pci_subvendor = 0x105b, .pci_subdevice = 0x6668, .config = ALC882_6ST_DIG }, /* Foxconn */ + { .pci_subvendor = 0x1019, .pci_subdevice = 0x6668, .config = ALC882_6ST_DIG }, /* ECS */ + { .modelname = "auto", .config = ALC861_AUTO }, + {} +}; + +static struct alc_config_preset alc882_presets[] = { + [ALC882_3ST_DIG] = { + .mixers = { alc882_base_mixer }, + .init_verbs = { alc882_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .dig_out_nid = ALC882_DIGOUT_NID, + .num_adc_nids = ARRAY_SIZE(alc882_adc_nids), + .adc_nids = alc882_adc_nids, + .dig_in_nid = ALC882_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc882_ch_modes), + .channel_mode = alc882_ch_modes, + .input_mux = &alc882_capture_source, + }, + [ALC882_6ST_DIG] = { + .mixers = { alc882_base_mixer, alc882_chmode_mixer }, + .init_verbs = { alc882_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .dig_out_nid = ALC882_DIGOUT_NID, + .num_adc_nids = ARRAY_SIZE(alc882_adc_nids), + .adc_nids = alc882_adc_nids, + .dig_in_nid = ALC882_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes), + .channel_mode = alc882_sixstack_modes, + .input_mux = &alc882_capture_source, + }, +}; + + +/* + * BIOS auto configuration + */ +static void alc882_auto_set_output_and_unmute(struct hda_codec *codec, + hda_nid_t nid, int pin_type, + int dac_idx) +{ + /* set as output */ + struct alc_spec *spec = codec->spec; + int idx; + + if (spec->multiout.dac_nids[dac_idx] == 0x25) + idx = 4; + else + idx = spec->multiout.dac_nids[dac_idx] - 2; + + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type); + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx); + +} + +static void alc882_auto_init_multi_out(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int i; + + for (i = 0; i <= HDA_SIDE; i++) { + hda_nid_t nid = spec->autocfg.line_out_pins[i]; + if (nid) + alc882_auto_set_output_and_unmute(codec, nid, PIN_OUT, i); + } +} + +static void alc882_auto_init_hp_out(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t pin; + + pin = spec->autocfg.hp_pin; + if (pin) /* connect to front */ + alc882_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); /* use dac 0 */ +} + +#define alc882_is_input_pin(nid) alc880_is_input_pin(nid) +#define ALC882_PIN_CD_NID ALC880_PIN_CD_NID + +static void alc882_auto_init_analog_input(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int i; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + hda_nid_t nid = spec->autocfg.input_pins[i]; + if (alc882_is_input_pin(nid)) { + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + i <= AUTO_PIN_FRONT_MIC ? PIN_VREF80 : PIN_IN); + if (nid != ALC882_PIN_CD_NID) + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_MUTE); + } + } +} + +/* almost identical with ALC880 parser... */ +static int alc882_parse_auto_config(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int err = alc880_parse_auto_config(codec); + + if (err < 0) + return err; + /* hack - override the init verbs */ + spec->init_verbs[0] = alc882_auto_init_verbs; + return 0; +} + +/* init callback for auto-configuration model -- overriding the default init */ +static int 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; +} + +/* + * ALC882 Headphone poll in 3.5.1a or 3.5.2 + */ + +static int patch_alc882(struct hda_codec *codec) +{ + struct alc_spec *spec; + int err, board_config; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + board_config = snd_hda_check_board_config(codec, alc882_cfg_tbl); + + if (board_config < 0 || board_config >= ALC882_MODEL_LAST) { + printk(KERN_INFO "hda_codec: Unknown model for ALC882, trying auto-probe from BIOS...\n"); + board_config = ALC882_AUTO; + } + + if (board_config == ALC882_AUTO) { + /* automatic parse from the BIOS config */ + err = alc882_parse_auto_config(codec); + if (err < 0) { + alc_free(codec); + return err; + } else if (! err) { + printk(KERN_INFO "hda_codec: Cannot set up configuration from BIOS. Using base mode...\n"); + board_config = ALC882_3ST_DIG; + } + } + + if (board_config != ALC882_AUTO) + setup_preset(spec, &alc882_presets[board_config]); spec->stream_name_analog = "ALC882 Analog"; - spec->stream_analog_playback = &alc880_pcm_analog_playback; - spec->stream_analog_capture = &alc880_pcm_analog_capture; + spec->stream_analog_playback = &alc882_pcm_analog_playback; + spec->stream_analog_capture = &alc882_pcm_analog_capture; spec->stream_name_digital = "ALC882 Digital"; - spec->stream_digital_playback = &alc880_pcm_digital_playback; - spec->stream_digital_capture = &alc880_pcm_digital_capture; + spec->stream_digital_playback = &alc882_pcm_digital_playback; + spec->stream_digital_capture = &alc882_pcm_digital_capture; - spec->multiout.max_channels = spec->channel_mode[0].channels; - spec->multiout.num_dacs = ARRAY_SIZE(alc882_dac_nids); - spec->multiout.dac_nids = alc882_dac_nids; - - spec->input_mux = &alc882_capture_source; - spec->num_adc_nids = ARRAY_SIZE(alc882_adc_nids); - spec->adc_nids = alc882_adc_nids; + if (! spec->adc_nids && spec->input_mux) { + /* check whether NID 0x07 is valid */ + unsigned int wcap = snd_hda_param_read(codec, 0x07, + AC_PAR_AUDIO_WIDGET_CAP); + wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; /* get type */ + if (wcap != AC_WID_AUD_IN) { + spec->adc_nids = alc882_adc_nids_alt; + spec->num_adc_nids = ARRAY_SIZE(alc882_adc_nids_alt); + spec->mixers[spec->num_mixers] = alc882_capture_alt_mixer; + spec->num_mixers++; + } else { + spec->adc_nids = alc882_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(alc882_adc_nids); + spec->mixers[spec->num_mixers] = alc882_capture_mixer; + spec->num_mixers++; + } + } codec->patch_ops = alc_patch_ops; + if (board_config == ALC882_AUTO) + codec->patch_ops.init = alc882_auto_init; + + return 0; +} + +/* + * ALC262 support + */ + +#define ALC262_DIGOUT_NID ALC880_DIGOUT_NID +#define ALC262_DIGIN_NID ALC880_DIGIN_NID + +#define alc262_dac_nids alc260_dac_nids +#define alc262_adc_nids alc882_adc_nids +#define alc262_adc_nids_alt alc882_adc_nids_alt + +#define alc262_modes alc260_modes + +static struct snd_kcontrol_new alc262_base_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), + /* HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_MUTE("PC Beelp Playback Switch", 0x0b, 0x05, HDA_INPUT), */ + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT), + 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 + +/* + * generic initialization of ADC, input mixers and output mixers + */ +static struct hda_verb alc262_init_verbs[] = { + /* + * Unmute ADC0-2 and set the default input to mic-in + */ + {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + * Note: PASD motherboards uses the Line In 2 as the input for front panel + * mic (mic 2) + */ + /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, + + /* + * Set up output mixers (0x0c - 0x0e) + */ + /* set vol=0 to output mixers */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + /* set up input amps for analog loopback */ + /* Amp Indices: DAC = 0, mixer = 1 */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0}, + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, + + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, + + /* FIXME: use matrix-type input source selection */ + /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ + /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, + /* Input mixer2 */ + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, + /* Input mixer3 */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, + + { } +}; +/* 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) +{ + hda_nid_t nid; + int err; + + spec->multiout.num_dacs = 1; /* only use one dac */ + spec->multiout.dac_nids = spec->private_dac_nids; + spec->multiout.dac_nids[0] = 2; + + nid = cfg->line_out_pins[0]; + if (nid) { + if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, "Front Playback Volume", + HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT))) < 0) + return err; + if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, "Front Playback Switch", + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0) + return err; + } + + nid = cfg->speaker_pin; + if (nid) { + if (nid == 0x16) { + if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, "Speaker Playback Volume", + HDA_COMPOSE_AMP_VAL(0x0e, 2, 0, HDA_OUTPUT))) < 0) + return err; + if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, "Speaker Playback Switch", + 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; + } + } + nid = cfg->hp_pin; + if (nid) { + /* spec->multiout.hp_nid = 2; */ + if (nid == 0x16) { + if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, "Headphone Playback Volume", + HDA_COMPOSE_AMP_VAL(0x0e, 2, 0, HDA_OUTPUT))) < 0) + return err; + if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, "Headphone Playback Switch", + 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; + } + } + return 0; +} + +/* identical with ALC880 */ +#define alc262_auto_create_analog_input_ctls alc880_auto_create_analog_input_ctls + +/* + * generic initialization of ADC, input mixers and output mixers + */ +static struct hda_verb alc262_volume_init_verbs[] = { + /* + * Unmute ADC0-2 and set the default input to mic-in + */ + {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + * Note: PASD motherboards uses the Line In 2 as the input for front panel + * mic (mic 2) + */ + /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, + + /* + * Set up output mixers (0x0c - 0x0f) + */ + /* set vol=0 to output mixers */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + + /* set up input amps for analog loopback */ + /* Amp Indices: DAC = 0, mixer = 1 */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + /* FIXME: use matrix-type input source selection */ + /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ + /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, + /* Input mixer2 */ + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, + /* Input mixer3 */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, + + { } +}; + +/* pcm configuration: identiacal with ALC880 */ +#define alc262_pcm_analog_playback alc880_pcm_analog_playback +#define alc262_pcm_analog_capture alc880_pcm_analog_capture +#define alc262_pcm_digital_playback alc880_pcm_digital_playback +#define alc262_pcm_digital_capture alc880_pcm_digital_capture + +/* + * BIOS auto configuration + */ +static int alc262_parse_auto_config(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int err; + static hda_nid_t alc262_ignore[] = { 0x1d, 0 }; + + 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) + 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) + return err; + + spec->multiout.max_channels = spec->multiout.num_dacs * 2; + + if (spec->autocfg.dig_out_pin) + spec->multiout.dig_out_nid = ALC262_DIGOUT_NID; + if (spec->autocfg.dig_in_pin) + spec->dig_in_nid = ALC262_DIGIN_NID; + + if (spec->kctl_alloc) + spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + + spec->init_verbs[spec->num_init_verbs++] = alc262_volume_init_verbs; + spec->input_mux = &spec->private_imux; + + return 1; +} + +#define alc262_auto_init_multi_out alc882_auto_init_multi_out +#define alc262_auto_init_hp_out alc882_auto_init_hp_out +#define alc262_auto_init_analog_input alc882_auto_init_analog_input + + +/* init callback for auto-configuration model -- overriding the default init */ +static int 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; +} + +/* + * configuration and preset + */ +static struct hda_board_config alc262_cfg_tbl[] = { + { .modelname = "basic", .config = ALC262_BASIC }, + { .modelname = "auto", .config = ALC262_AUTO }, + {} +}; + +static struct alc_config_preset alc262_presets[] = { + [ALC262_BASIC] = { + .mixers = { alc262_base_mixer }, + .init_verbs = { alc262_init_verbs }, + .num_dacs = ARRAY_SIZE(alc262_dac_nids), + .dac_nids = alc262_dac_nids, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc262_modes), + .channel_mode = alc262_modes, + }, +}; + +static int patch_alc262(struct hda_codec *codec) +{ + struct alc_spec *spec; + int board_config; + int err; + + spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; +#if 0 + /* pshou 07/11/05 set a zero PCM sample to DAC when FIFO is under-run */ + { + int tmp; + snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_COEF_INDEX, 7); + tmp = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_PROC_COEF, 0); + snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_COEF_INDEX, 7); + snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PROC_COEF, tmp | 0x80); + } +#endif + + board_config = snd_hda_check_board_config(codec, alc262_cfg_tbl); + if (board_config < 0 || board_config >= ALC262_MODEL_LAST) { + printk(KERN_INFO "hda_codec: Unknown model for ALC262, trying auto-probe from BIOS...\n"); + board_config = ALC262_AUTO; + } + + if (board_config == ALC262_AUTO) { + /* automatic parse from the BIOS config */ + err = alc262_parse_auto_config(codec); + if (err < 0) { + alc_free(codec); + return err; + } else if (! err) { + printk(KERN_INFO "hda_codec: Cannot set up configuration from BIOS. Using base mode...\n"); + board_config = ALC262_BASIC; + } + } + + if (board_config != ALC262_AUTO) + setup_preset(spec, &alc262_presets[board_config]); + + spec->stream_name_analog = "ALC262 Analog"; + spec->stream_analog_playback = &alc262_pcm_analog_playback; + spec->stream_analog_capture = &alc262_pcm_analog_capture; + + spec->stream_name_digital = "ALC262 Digital"; + spec->stream_digital_playback = &alc262_pcm_digital_playback; + spec->stream_digital_capture = &alc262_pcm_digital_capture; + + if (! spec->adc_nids && spec->input_mux) { + /* check whether NID 0x07 is valid */ + unsigned int wcap = snd_hda_param_read(codec, 0x07, + AC_PAR_AUDIO_WIDGET_CAP); + wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; /* get type */ + if (wcap != AC_WID_AUD_IN) { + spec->adc_nids = alc262_adc_nids_alt; + spec->num_adc_nids = ARRAY_SIZE(alc262_adc_nids_alt); + spec->mixers[spec->num_mixers] = alc262_capture_alt_mixer; + spec->num_mixers++; + } else { + spec->adc_nids = alc262_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(alc262_adc_nids); + spec->mixers[spec->num_mixers] = alc262_capture_mixer; + spec->num_mixers++; + } + } + + codec->patch_ops = alc_patch_ops; + if (board_config == ALC262_AUTO) + codec->patch_ops.init = alc262_auto_init; + + return 0; +} + + +/* + * ALC861 channel source setting (2/6 channel selection for 3-stack) + */ + +/* + * set the path ways for 2 channel output + * need to set the codec line out and mic 1 pin widgets to inputs + */ +static struct hda_verb alc861_threestack_ch2_init[] = { + /* set pin widget 1Ah (line in) for input */ + { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, + /* set pin widget 18h (mic1/2) for input, for mic also enable the vref */ + { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, + + { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c }, + { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, //mic + { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8)) }, //line in + { } /* end */ +}; +/* + * 6ch mode + * need to set the codec line out and mic 1 pin widgets to outputs + */ +static struct hda_verb alc861_threestack_ch6_init[] = { + /* set pin widget 1Ah (line in) for output (Back Surround)*/ + { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, + /* set pin widget 18h (mic1) for output (CLFE)*/ + { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, + + { 0x0c, AC_VERB_SET_CONNECT_SEL, 0x00 }, + { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00 }, + + { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 }, + { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, //mic + { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8)) }, //line in + { } /* end */ +}; + +static struct hda_channel_mode alc861_threestack_modes[2] = { + { 2, alc861_threestack_ch2_init }, + { 6, alc861_threestack_ch6_init }, +}; + +/* patch-ALC861 */ + +static struct snd_kcontrol_new alc861_base_mixer[] = { + /* output mixer control */ + HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT), + + /*Input mixer control */ + /* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */ + HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT), + + /* Capture mixer control */ + 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 = alc_mux_enum_info, + .get = alc_mux_enum_get, + .put = alc_mux_enum_put, + }, + { } /* end */ +}; + +static struct snd_kcontrol_new alc861_3ST_mixer[] = { + /* output mixer control */ + HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT), + /*HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT), */ + + /* Input mixer control */ + /* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */ + HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT), + + /* Capture mixer control */ + 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 = alc_mux_enum_info, + .get = alc_mux_enum_get, + .put = alc_mux_enum_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Channel Mode", + .info = alc_ch_mode_info, + .get = alc_ch_mode_get, + .put = alc_ch_mode_put, + .private_value = ARRAY_SIZE(alc861_threestack_modes), + }, + { } /* end */ +}; + +/* + * generic initialization of ADC, input mixers and output mixers + */ +static struct hda_verb alc861_base_init_verbs[] = { + /* + * Unmute ADC0 and set the default input to mic-in + */ + /* port-A for surround (rear panel) */ + { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, + { 0x0e, AC_VERB_SET_CONNECT_SEL, 0x00 }, + /* port-B for mic-in (rear panel) with vref */ + { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, + /* port-C for line-in (rear panel) */ + { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, + /* port-D for Front */ + { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, + { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 }, + /* port-E for HP out (front panel) */ + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, + /* route front PCM to HP */ + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x01 }, + /* port-F for mic-in (front panel) with vref */ + { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, + /* port-G for CLFE (rear panel) */ + { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, + { 0x1f, AC_VERB_SET_CONNECT_SEL, 0x00 }, + /* port-H for side (rear panel) */ + { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, + { 0x20, AC_VERB_SET_CONNECT_SEL, 0x00 }, + /* CD-in */ + { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, + /* route front mic to ADC1*/ + {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + /* Unmute DAC0~3 & spdif out*/ + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* Unmute Mixer 14 (mic) 1c (Line in)*/ + {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + /* Unmute Stereo Mixer 15 */ + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c }, //Output 0~12 step + + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, // hp used DAC 3 (Front) + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, + + { } +}; + +static struct hda_verb alc861_threestack_init_verbs[] = { + /* + * Unmute ADC0 and set the default input to mic-in + */ + /* port-A for surround (rear panel) */ + { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 }, + /* port-B for mic-in (rear panel) with vref */ + { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, + /* port-C for line-in (rear panel) */ + { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, + /* port-D for Front */ + { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, + { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 }, + /* port-E for HP out (front panel) */ + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, + /* route front PCM to HP */ + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x01 }, + /* port-F for mic-in (front panel) with vref */ + { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, + /* port-G for CLFE (rear panel) */ + { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 }, + /* port-H for side (rear panel) */ + { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 }, + /* CD-in */ + { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, + /* route front mic to ADC1*/ + {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + /* Unmute DAC0~3 & spdif out*/ + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* Unmute Mixer 14 (mic) 1c (Line in)*/ + {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + /* Unmute Stereo Mixer 15 */ + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c }, //Output 0~12 step + + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, // hp used DAC 3 (Front) + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, + { } +}; +/* + * generic initialization of ADC, input mixers and output mixers + */ +static struct hda_verb alc861_auto_init_verbs[] = { + /* + * Unmute ADC0 and set the default input to mic-in + */ +// {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + /* Unmute DAC0~3 & spdif out*/ + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* Unmute Mixer 14 (mic) 1c (Line in)*/ + {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + /* Unmute Stereo Mixer 15 */ + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, + + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, + + {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, // set Mic 1 + + { } +}; + +/* pcm configuration: identiacal with ALC880 */ +#define alc861_pcm_analog_playback alc880_pcm_analog_playback +#define alc861_pcm_analog_capture alc880_pcm_analog_capture +#define alc861_pcm_digital_playback alc880_pcm_digital_playback +#define alc861_pcm_digital_capture alc880_pcm_digital_capture + + +#define ALC861_DIGOUT_NID 0x07 + +static struct hda_channel_mode alc861_8ch_modes[1] = { + { 8, NULL } +}; + +static hda_nid_t alc861_dac_nids[4] = { + /* front, surround, clfe, side */ + 0x03, 0x06, 0x05, 0x04 +}; + +static hda_nid_t alc861_adc_nids[1] = { + /* ADC0-2 */ + 0x08, +}; + +static struct hda_input_mux alc861_capture_source = { + .num_items = 5, + .items = { + { "Mic", 0x0 }, + { "Front Mic", 0x3 }, + { "Line", 0x1 }, + { "CD", 0x4 }, + { "Mixer", 0x5 }, + }, +}; + +/* fill in the dac_nids table from the parsed pin configuration */ +static int alc861_auto_fill_dac_nids(struct alc_spec *spec, const struct auto_pin_cfg *cfg) +{ + int i; + hda_nid_t nid; + + spec->multiout.dac_nids = spec->private_dac_nids; + for (i = 0; i < cfg->line_outs; i++) { + nid = cfg->line_out_pins[i]; + if (nid) { + if (i >= ARRAY_SIZE(alc861_dac_nids)) + continue; + spec->multiout.dac_nids[i] = alc861_dac_nids[i]; + } + } + spec->multiout.num_dacs = cfg->line_outs; + return 0; +} + +/* add playback controls from the parsed DAC table */ +static int alc861_auto_create_multi_out_ctls(struct alc_spec *spec, + const struct auto_pin_cfg *cfg) +{ + char name[32]; + static const char *chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; + hda_nid_t nid; + int i, idx, err; + + for (i = 0; i < cfg->line_outs; i++) { + nid = spec->multiout.dac_nids[i]; + if (! nid) + continue; + if (nid == 0x05) { + /* Center/LFE */ + if ((err = add_control(spec, ALC_CTL_BIND_MUTE, "Center Playback Switch", + HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT))) < 0) + return err; + if ((err = add_control(spec, ALC_CTL_BIND_MUTE, "LFE Playback Switch", + HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT))) < 0) + return err; + } else { + for (idx = 0; idx < ARRAY_SIZE(alc861_dac_nids) - 1; idx++) + if (nid == alc861_dac_nids[idx]) + break; + sprintf(name, "%s Playback Switch", chname[idx]); + if ((err = add_control(spec, ALC_CTL_BIND_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0) + return err; + } + } + return 0; +} + +static int alc861_auto_create_hp_ctls(struct alc_spec *spec, hda_nid_t pin) +{ + int err; + hda_nid_t nid; + + if (! pin) + return 0; + + if ((pin >= 0x0b && pin <= 0x10) || pin == 0x1f || pin == 0x20) { + nid = 0x03; + if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, "Headphone Playback Switch", + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0) + return err; + spec->multiout.hp_nid = nid; + } + return 0; +} + +/* create playback/capture controls for input pins */ +static int alc861_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) +{ + static char *labels[AUTO_PIN_LAST] = { + "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux" + }; + struct hda_input_mux *imux = &spec->private_imux; + int i, err, idx, idx1; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + switch(cfg->input_pins[i]) { + case 0x0c: + idx1 = 1; + idx = 2; // Line In + break; + case 0x0f: + idx1 = 2; + idx = 2; // Line In + break; + case 0x0d: + idx1 = 0; + idx = 1; // Mic In + break; + case 0x10: + idx1 = 3; + idx = 1; // Mic In + break; + case 0x11: + idx1 = 4; + idx = 0; // CD + break; + default: + continue; + } + + err = new_analog_input(spec, cfg->input_pins[i], labels[i], idx, 0x15); + if (err < 0) + return err; + + imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].index = idx1; + imux->num_items++; + } + return 0; +} + +static struct snd_kcontrol_new alc861_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + *FIXME: the controls appear in the "playback" view! + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 1, + .info = alc_mux_enum_info, + .get = alc_mux_enum_get, + .put = alc_mux_enum_put, + }, + { } /* end */ +}; + +static void alc861_auto_set_output_and_unmute(struct hda_codec *codec, hda_nid_t nid, + int pin_type, int dac_idx) +{ + /* set as output */ + + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type); + snd_hda_codec_write(codec, dac_idx, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); + +} + +static void alc861_auto_init_multi_out(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int i; + + for (i = 0; i < spec->autocfg.line_outs; i++) { + hda_nid_t nid = spec->autocfg.line_out_pins[i]; + if (nid) + alc861_auto_set_output_and_unmute(codec, nid, PIN_OUT, spec->multiout.dac_nids[i]); + } +} + +static void alc861_auto_init_hp_out(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t pin; + + pin = spec->autocfg.hp_pin; + if (pin) /* connect to front */ + alc861_auto_set_output_and_unmute(codec, pin, PIN_HP, spec->multiout.dac_nids[0]); +} + +static void alc861_auto_init_analog_input(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int i; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + hda_nid_t nid = spec->autocfg.input_pins[i]; + if ((nid>=0x0c) && (nid <=0x11)) { + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + i <= AUTO_PIN_FRONT_MIC ? PIN_VREF80 : PIN_IN); + } + } +} + +/* parse the BIOS configuration and set up the alc_spec */ +/* return 1 if successful, 0 if the proper config is not found, or a negative error code */ +static int alc861_parse_auto_config(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int err; + static hda_nid_t alc861_ignore[] = { 0x1d, 0 }; + + 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) + return 0; /* can't find valid BIOS pin config */ + + if ((err = alc861_auto_fill_dac_nids(spec, &spec->autocfg)) < 0 || + (err = alc861_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 || + (err = alc861_auto_create_hp_ctls(spec, spec->autocfg.hp_pin)) < 0 || + (err = alc861_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0) + return err; + + spec->multiout.max_channels = spec->multiout.num_dacs * 2; + + if (spec->autocfg.dig_out_pin) + spec->multiout.dig_out_nid = ALC861_DIGOUT_NID; + + if (spec->kctl_alloc) + spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + + spec->init_verbs[spec->num_init_verbs++] = alc861_auto_init_verbs; + + spec->input_mux = &spec->private_imux; + + spec->adc_nids = alc861_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(alc861_adc_nids); + spec->mixers[spec->num_mixers] = alc861_capture_mixer; + spec->num_mixers++; + + return 1; +} + +/* init callback for auto-configuration model -- overriding the default init */ +static int 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; +} + + +/* + * configuration and preset + */ +static struct hda_board_config alc861_cfg_tbl[] = { + { .modelname = "3stack", .config = ALC861_3ST }, + { .pci_subvendor = 0x8086, .pci_subdevice = 0xd600, .config = ALC861_3ST }, + { .modelname = "3stack-dig", .config = ALC861_3ST_DIG }, + { .modelname = "6stack-dig", .config = ALC861_6ST_DIG }, + { .modelname = "auto", .config = ALC861_AUTO }, + {} +}; + +static struct alc_config_preset alc861_presets[] = { + [ALC861_3ST] = { + .mixers = { alc861_3ST_mixer }, + .init_verbs = { alc861_threestack_init_verbs }, + .num_dacs = ARRAY_SIZE(alc861_dac_nids), + .dac_nids = alc861_dac_nids, + .num_channel_mode = ARRAY_SIZE(alc861_threestack_modes), + .channel_mode = alc861_threestack_modes, + .num_adc_nids = ARRAY_SIZE(alc861_adc_nids), + .adc_nids = alc861_adc_nids, + .input_mux = &alc861_capture_source, + }, + [ALC861_3ST_DIG] = { + .mixers = { alc861_base_mixer }, + .init_verbs = { alc861_threestack_init_verbs }, + .num_dacs = ARRAY_SIZE(alc861_dac_nids), + .dac_nids = alc861_dac_nids, + .dig_out_nid = ALC861_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc861_threestack_modes), + .channel_mode = alc861_threestack_modes, + .num_adc_nids = ARRAY_SIZE(alc861_adc_nids), + .adc_nids = alc861_adc_nids, + .input_mux = &alc861_capture_source, + }, + [ALC861_6ST_DIG] = { + .mixers = { alc861_base_mixer }, + .init_verbs = { alc861_base_init_verbs }, + .num_dacs = ARRAY_SIZE(alc861_dac_nids), + .dac_nids = alc861_dac_nids, + .dig_out_nid = ALC861_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc861_8ch_modes), + .channel_mode = alc861_8ch_modes, + .num_adc_nids = ARRAY_SIZE(alc861_adc_nids), + .adc_nids = alc861_adc_nids, + .input_mux = &alc861_capture_source, + }, +}; + + +static int patch_alc861(struct hda_codec *codec) +{ + struct alc_spec *spec; + int board_config; + int err; + + spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + board_config = snd_hda_check_board_config(codec, alc861_cfg_tbl); + if (board_config < 0 || board_config >= ALC861_MODEL_LAST) { + printk(KERN_INFO "hda_codec: Unknown model for ALC861, trying auto-probe from BIOS...\n"); + board_config = ALC861_AUTO; + } + + if (board_config == ALC861_AUTO) { + /* automatic parse from the BIOS config */ + err = alc861_parse_auto_config(codec); + if (err < 0) { + alc_free(codec); + return err; + } else if (! err) { + printk(KERN_INFO "hda_codec: Cannot set up configuration from BIOS. Using base mode...\n"); + board_config = ALC861_3ST_DIG; + } + } + + if (board_config != ALC861_AUTO) + setup_preset(spec, &alc861_presets[board_config]); + + spec->stream_name_analog = "ALC861 Analog"; + spec->stream_analog_playback = &alc861_pcm_analog_playback; + spec->stream_analog_capture = &alc861_pcm_analog_capture; + + spec->stream_name_digital = "ALC861 Digital"; + spec->stream_digital_playback = &alc861_pcm_digital_playback; + spec->stream_digital_capture = &alc861_pcm_digital_capture; + + codec->patch_ops = alc_patch_ops; + if (board_config == ALC861_AUTO) + codec->patch_ops.init = alc861_auto_init; + return 0; } @@ -2692,7 +4675,11 @@ static int patch_alc882(struct hda_codec *codec) */ struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 }, + { .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 }, { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 }, { .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 }, + { .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc882 }, + { .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 }, + { .id = 0x10ec0861, .name = "ALC861", .patch = patch_alc861 }, {} /* terminator */ }; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 78662d3539e2..8311c9fa0527 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -761,7 +761,7 @@ static int stac922x_parse_auto_config(struct hda_codec *codec) struct sigmatel_spec *spec = codec->spec; int err; - if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg)) < 0) + if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL)) < 0) return err; if ((err = stac92xx_add_dyn_out_pins(codec, &spec->autocfg)) < 0) return err; @@ -801,7 +801,7 @@ static int stac9200_parse_auto_config(struct hda_codec *codec) struct sigmatel_spec *spec = codec->spec; int err; - if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg)) < 0) + if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL)) < 0) return err; if ((err = stac92xx_auto_create_analog_input_ctls(codec, &spec->autocfg)) < 0) -- cgit v1.2.3 From c5f2ea08fbd8911e2c975094780d2b16e65f27e0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 6 Dec 2005 18:54:31 +0100 Subject: [ALSA] hda-codec - Fix ALC codec probing Modules: HDA Codec driver - Fix ALC882 BIOS auto-probe - Fix ALC262 basic model - Add assert for debugging Oops Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a98c0e4da0ac..63aeddb731d9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1259,6 +1259,8 @@ static int alc_build_pcms(struct hda_codec *codec) codec->num_pcms = 1; codec->pcm_info = info; + snd_assert(spec->stream_analog_playback, return -EINVAL); + snd_assert(spec->stream_analog_capture, return -EINVAL); info->name = spec->stream_name_analog; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_analog_playback); info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0]; @@ -1277,10 +1279,12 @@ static int alc_build_pcms(struct hda_codec *codec) info++; info->name = spec->stream_name_digital; if (spec->multiout.dig_out_nid) { + snd_assert(spec->stream_digital_playback, return -EINVAL); info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_digital_playback); info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid; } if (spec->dig_in_nid) { + snd_assert(spec->stream_digital_capture, return -EINVAL); info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_digital_capture); info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid; } @@ -3508,9 +3512,10 @@ static int alc882_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - /* hack - override the init verbs */ - spec->init_verbs[0] = alc882_auto_init_verbs; - return 0; + else if (err > 0) + /* hack - override the init verbs */ + spec->init_verbs[0] = alc882_auto_init_verbs; + return err; } /* init callback for auto-configuration model -- overriding the default init */ @@ -3605,6 +3610,7 @@ static int patch_alc882(struct hda_codec *codec) #define alc262_adc_nids_alt alc882_adc_nids_alt #define alc262_modes alc260_modes +#define alc262_capture_source alc882_capture_source static struct snd_kcontrol_new alc262_base_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -3921,6 +3927,7 @@ static struct alc_config_preset alc262_presets[] = { .hp_nid = 0x03, .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, + .input_mux = alc262_capture_source, }, }; -- cgit v1.2.3 From a3bcba384c2f2448ad204ea52baa15f1227d0d40 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 6 Dec 2005 19:05:29 +0100 Subject: [ALSA] hda-codec - Fix Oops with ALC260 auto-probe Modules: HDA Codec driver - Fix Oops with auto-probing of ALC260 with digital I/O - Fix a typo Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 63aeddb731d9..2a6a4804cb92 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2678,6 +2678,9 @@ static struct hda_pcm_stream alc260_pcm_analog_capture = { .channels_max = 2, }; +#define alc260_pcm_digital_playback alc880_pcm_digital_playback +#define alc260_pcm_digital_capture alc880_pcm_digital_capture + /* * for BIOS auto-configuration */ @@ -3041,6 +3044,10 @@ static int patch_alc260(struct hda_codec *codec) spec->stream_analog_playback = &alc260_pcm_analog_playback; spec->stream_analog_capture = &alc260_pcm_analog_capture; + spec->stream_name_digital = "ALC260 Digital"; + spec->stream_digital_playback = &alc260_pcm_digital_playback; + spec->stream_digital_capture = &alc260_pcm_digital_capture; + codec->patch_ops = alc_patch_ops; if (board_config == ALC260_AUTO) codec->patch_ops.init = alc260_auto_init; @@ -3927,7 +3934,7 @@ static struct alc_config_preset alc262_presets[] = { .hp_nid = 0x03, .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, - .input_mux = alc262_capture_source, + .input_mux = &alc262_capture_source, }, }; -- cgit v1.2.3 From 4a471b7ddfe76e39c1633d5a23a687f4b5fc0d8d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Dec 2005 13:56:29 +0100 Subject: [ALSA] hda-codec - Small clean up and fixes Modules: HDA Codec driver,HDA generic driver - Common labels for input pins - Fix and clean up of Realtek codec parsers Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 6 +++ sound/pci/hda/hda_local.h | 2 + sound/pci/hda/patch_analog.c | 8 ++-- sound/pci/hda/patch_realtek.c | 87 +++++++++++++++++++++--------------------- sound/pci/hda/patch_sigmatel.c | 5 +-- 5 files changed, 56 insertions(+), 52 deletions(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index bd375f895ec0..4a6dd97deba6 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2051,6 +2051,12 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *c return 0; } +/* labels for input pins */ +const char *auto_pin_cfg_labels[AUTO_PIN_LAST] = { + "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux" +}; + + #ifdef CONFIG_PM /* * power management diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index a9863eb20c75..c82d2a72d13e 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -211,6 +211,8 @@ enum { AUTO_PIN_LAST }; +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 */ diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index d1e1ded27532..1ada1b075c9a 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1850,18 +1850,16 @@ static int new_analog_input(struct ad198x_spec *spec, hda_nid_t pin, static int ad1988_auto_create_analog_input_ctls(struct ad198x_spec *spec, const struct auto_pin_cfg *cfg) { - static char *labels[AUTO_PIN_LAST] = { - "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux" - }; struct hda_input_mux *imux = &spec->private_imux; int i, err; for (i = 0; i < AUTO_PIN_LAST; i++) { - err = new_analog_input(spec, cfg->input_pins[i], labels[i], + err = new_analog_input(spec, cfg->input_pins[i], + auto_pin_cfg_labels[i], i <= AUTO_PIN_FRONT_MIC); if (err < 0) return err; - imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].label = auto_pin_cfg_labels[i]; imux->items[imux->num_items].index = ad1988_pin_to_adc_idx(cfg->input_pins[i]); imux->num_items++; } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2a6a4804cb92..cac109268f73 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1259,18 +1259,24 @@ static int alc_build_pcms(struct hda_codec *codec) codec->num_pcms = 1; codec->pcm_info = info; - snd_assert(spec->stream_analog_playback, return -EINVAL); - snd_assert(spec->stream_analog_capture, return -EINVAL); info->name = spec->stream_name_analog; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_analog_playback); - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0]; - info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture); - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0]; - - info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = 0; - for (i = 0; i < spec->num_channel_mode; i++) { - if (spec->channel_mode[i].channels > info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max) { - info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = spec->channel_mode[i].channels; + if (spec->stream_analog_playback) { + snd_assert(spec->multiout.dac_nids, return -EINVAL); + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_analog_playback); + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[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[0]; + } + + if (spec->channel_mode) { + info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = 0; + for (i = 0; i < spec->num_channel_mode; i++) { + if (spec->channel_mode[i].channels > info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max) { + info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = spec->channel_mode[i].channels; + } } } @@ -1278,13 +1284,13 @@ static int alc_build_pcms(struct hda_codec *codec) codec->num_pcms++; info++; info->name = spec->stream_name_digital; - if (spec->multiout.dig_out_nid) { - snd_assert(spec->stream_digital_playback, return -EINVAL); + if (spec->multiout.dig_out_nid && + spec->stream_digital_playback) { info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_digital_playback); info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid; } - if (spec->dig_in_nid) { - snd_assert(spec->stream_digital_capture, return -EINVAL); + if (spec->dig_in_nid && + spec->stream_digital_capture) { info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_digital_capture); info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid; } @@ -2091,20 +2097,18 @@ static int new_analog_input(struct alc_spec *spec, hda_nid_t pin, const char *ct static int alc880_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - static char *labels[AUTO_PIN_LAST] = { - "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux" - }; struct hda_input_mux *imux = &spec->private_imux; int i, err, idx; for (i = 0; i < AUTO_PIN_LAST; i++) { if (alc880_is_input_pin(cfg->input_pins[i])) { idx = alc880_input_pin_idx(cfg->input_pins[i]); - err = new_analog_input(spec, cfg->input_pins[i], labels[i], + err = new_analog_input(spec, cfg->input_pins[i], + auto_pin_cfg_labels[i], idx, 0x0b); if (err < 0) return err; - imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].label = auto_pin_cfg_labels[i]; imux->items[imux->num_items].index = alc880_input_pin_idx(cfg->input_pins[i]); imux->num_items++; } @@ -2664,6 +2668,8 @@ static struct hda_verb alc260_fujitsu_init_verbs[] = { {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 */ + + { } }; static struct hda_pcm_stream alc260_pcm_analog_playback = { @@ -2755,28 +2761,27 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec, static int alc260_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - static char *labels[AUTO_PIN_LAST] = { - "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux" - }; struct hda_input_mux *imux = &spec->private_imux; int i, err, idx; for (i = 0; i < AUTO_PIN_LAST; i++) { if (cfg->input_pins[i] >= 0x12) { idx = cfg->input_pins[i] - 0x12; - err = new_analog_input(spec, cfg->input_pins[i], labels[i], idx, 0x07); + err = new_analog_input(spec, cfg->input_pins[i], + auto_pin_cfg_labels[i], idx, 0x07); if (err < 0) return err; - imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].label = auto_pin_cfg_labels[i]; imux->items[imux->num_items].index = idx; imux->num_items++; } if ((cfg->input_pins[i] >= 0x0f) && (cfg->input_pins[i] <= 0x10)){ idx = cfg->input_pins[i] - 0x09; - err = new_analog_input(spec, cfg->input_pins[i], labels[i], idx, 0x07); + err = new_analog_input(spec, cfg->input_pins[i], + auto_pin_cfg_labels[i], idx, 0x07); if (err < 0) return err; - imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].label = auto_pin_cfg_labels[i]; imux->items[imux->num_items].index = idx; imux->num_items++; } @@ -2889,11 +2894,11 @@ static int alc260_parse_auto_config(struct hda_codec *codec) if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, alc260_ignore)) < 0) return err; - if (! spec->autocfg.line_outs && ! spec->autocfg.speaker_pin && - ! spec->autocfg.hp_pin) + if ((err = alc260_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0) + return err; + if (! spec->kctl_alloc) return 0; /* can't find valid BIOS pin config */ - if ((err = alc260_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 || - (err = alc260_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0) + if ((err = alc260_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0) return err; spec->multiout.max_channels = 2; @@ -2908,19 +2913,18 @@ static int alc260_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux; /* check whether NID 0x04 is valid */ - wcap = snd_hda_param_read(codec, alc260_adc_nids[0], AC_PAR_AUDIO_WIDGET_CAP); + wcap = get_wcaps(codec, 0x04); wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; /* get type */ if (wcap != AC_WID_AUD_IN) { spec->adc_nids = alc260_adc_nids_alt; spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt); spec->mixers[spec->num_mixers] = alc260_capture_alt_mixer; - spec->num_mixers++; } else { spec->adc_nids = alc260_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids); spec->mixers[spec->num_mixers] = alc260_capture_mixer; - spec->num_mixers++; } + spec->num_mixers++; return 1; } @@ -3582,8 +3586,7 @@ static int patch_alc882(struct hda_codec *codec) if (! spec->adc_nids && spec->input_mux) { /* check whether NID 0x07 is valid */ - unsigned int wcap = snd_hda_param_read(codec, 0x07, - AC_PAR_AUDIO_WIDGET_CAP); + unsigned int wcap = get_wcaps(codec, 0x07); wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; /* get type */ if (wcap != AC_WID_AUD_IN) { spec->adc_nids = alc882_adc_nids_alt; @@ -3991,8 +3994,8 @@ static int patch_alc262(struct hda_codec *codec) if (! spec->adc_nids && spec->input_mux) { /* check whether NID 0x07 is valid */ - unsigned int wcap = snd_hda_param_read(codec, 0x07, - AC_PAR_AUDIO_WIDGET_CAP); + unsigned int wcap = get_wcaps(codec, 0x07); + wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; /* get type */ if (wcap != AC_WID_AUD_IN) { spec->adc_nids = alc262_adc_nids_alt; @@ -4423,9 +4426,6 @@ static int alc861_auto_create_hp_ctls(struct alc_spec *spec, hda_nid_t pin) /* create playback/capture controls for input pins */ static int alc861_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - static char *labels[AUTO_PIN_LAST] = { - "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux" - }; struct hda_input_mux *imux = &spec->private_imux; int i, err, idx, idx1; @@ -4455,11 +4455,12 @@ static int alc861_auto_create_analog_input_ctls(struct alc_spec *spec, const str continue; } - err = new_analog_input(spec, cfg->input_pins[i], labels[i], idx, 0x15); + err = new_analog_input(spec, cfg->input_pins[i], + auto_pin_cfg_labels[i], idx, 0x15); if (err < 0) return err; - imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].label = auto_pin_cfg_labels[i]; imux->items[imux->num_items].index = idx1; imux->num_items++; } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 8311c9fa0527..61903848cd43 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -702,9 +702,6 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, struct auto_pin static int stac92xx_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { struct sigmatel_spec *spec = codec->spec; - static char *labels[AUTO_PIN_LAST] = { - "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux" - }; struct hda_input_mux *imux = &spec->private_imux; hda_nid_t con_lst[HDA_MAX_NUM_INPUTS]; int i, j, k; @@ -715,7 +712,7 @@ static int stac92xx_auto_create_analog_input_ctls(struct hda_codec *codec, const /* 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 = labels[i]; + imux->items[imux->num_items].label = auto_pin_cfg_labels[i]; for (j=0; jnum_muxes; j++) { int num_cons = snd_hda_get_connections(codec, spec->mux_nids[j], con_lst, HDA_MAX_NUM_INPUTS); -- cgit v1.2.3 From 7632c7b4443057e1294208a0d9a55d8558f2f6ca Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Dec 2005 18:25:47 +0100 Subject: [ALSA] hda-codec - Add the model entry for ASUS P5GD1-HVM Modules: HDA Codec driver Add the model entry (ALC880 6stack) for ASUS P5GD1-HVM. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cac109268f73..55bda26ce126 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1665,6 +1665,7 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .pci_subvendor = 0x1043, .pci_subdevice = 0x1964, .config = ALC880_Z71V }, { .modelname = "6stack", .config = ALC880_6ST }, + { .pci_subvendor = 0x1043, .pci_subdevice = 0x8196, .config = ALC880_6ST }, /* ASUS P5GD1-HVM */ { .pci_subvendor = 0x1043, .pci_subdevice = 0x81b4, .config = ALC880_6ST }, { .pci_subvendor = 0x1019, .pci_subdevice = 0xa884, .config = ALC880_6ST }, /* Acer APFV */ -- cgit v1.2.3 From 041dec01736c59df43b0600c0fd154e50d8ccf6e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 23 Dec 2005 12:27:52 +0100 Subject: [ALSA] hda-codec - Add model entry for Shuttle ST20G5 Modules: HDA Codec driver Added the model entry for Shuttle ST20G5. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 55bda26ce126..ad9e501a9818 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1678,6 +1678,7 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .pci_subvendor = 0x1025, .pci_subdevice = 0x0077, .config = ALC880_6ST_DIG }, { .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 */ { .modelname = "asus", .config = ALC880_ASUS }, { .pci_subvendor = 0x1043, .pci_subdevice = 0x1964, .config = ALC880_ASUS_DIG }, -- cgit v1.2.3