From 74dc8909c1ce38098e6689239ed6ae6b6bf9f92b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 13 Jan 2011 14:14:41 +0100 Subject: ALSA: hda - Remove unused fixup entry for ALC262 ... and a minor cleanup. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 98c9cd8f6471..738d5d8962d0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12453,12 +12453,6 @@ static const struct alc_fixup alc262_fixups[] = { { } } }, - [PINFIX_PB_M5210] = { - .verbs = (const struct hda_verb[]) { - { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50 }, - {} - } - }, }; static struct snd_pci_quirk alc262_fixup_tbl[] = { @@ -14826,7 +14820,7 @@ static void alc269_fixup_hweq(struct hda_codec *codec, enum { ALC269_FIXUP_SONY_VAIO, - ALC275_FIX_SONY_VAIO_GPIO2, + ALC275_FIXUP_SONY_VAIO_GPIO2, ALC269_FIXUP_DELL_M101Z, ALC269_FIXUP_SKU_IGNORE, ALC269_FIXUP_ASUS_G73JW, @@ -14841,7 +14835,7 @@ static const struct alc_fixup alc269_fixups[] = { {} } }, - [ALC275_FIX_SONY_VAIO_GPIO2] = { + [ALC275_FIXUP_SONY_VAIO_GPIO2] = { .verbs = (const struct hda_verb[]) { {0x01, AC_VERB_SET_GPIO_MASK, 0x04}, {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04}, @@ -14886,7 +14880,7 @@ static const struct alc_fixup alc269_fixups[] = { }; static struct snd_pci_quirk alc269_fixup_tbl[] = { - SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIX_SONY_VAIO_GPIO2), + SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2), SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), -- cgit v1.2.3 From 6fc398cb306b0441436c93d6ddead3109b99f884 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 13 Jan 2011 14:36:37 +0100 Subject: ALSA: hda - Apply mario fixup only once The amp-override is necessary only once at initialization time. Also fixed a coding style issue. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 738d5d8962d0..f13920e53847 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -19366,7 +19366,10 @@ static void alc662_auto_init(struct hda_codec *codec) } static void alc272_fixup_mario(struct hda_codec *codec, - const struct alc_fixup *fix, int pre_init) { + const struct alc_fixup *fix, int pre_init) +{ + if (!pre_init) + return; if (snd_hda_override_amp_caps(codec, 0x2, HDA_OUTPUT, (0x3b << AC_AMPCAP_OFFSET_SHIFT) | (0x3b << AC_AMPCAP_NUM_STEPS_SHIFT) | -- cgit v1.2.3 From 9fb1ef25f4d31f07cdaf7c6075b40bbcb00c1f92 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 13 Jan 2011 14:40:43 +0100 Subject: ALSA: hda - Apply Sony VAIO hweq fixup only once This should be applied also only once as a part of the initialization. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f13920e53847..98b4b2e1b930 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14814,6 +14814,8 @@ static void alc269_fixup_hweq(struct hda_codec *codec, { int coef; + if (pre_init) + return; coef = alc_read_coef_idx(codec, 0x1e); alc_write_coef_idx(codec, 0x1e, coef | 0x80); } -- cgit v1.2.3 From b5bfbc670283d1ff21df4cd3f9f036cc47e34ce4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 13 Jan 2011 14:22:32 +0100 Subject: ALSA: hda - Reorganize fixup structure for Realtek Instead of keeping various data types in a single record, put the type field and keep a single value in each entry, but allows chaining multiple fixup entries. This allows more flexible data management (see ALC275_FIXUP_SONY_HWEQ for example). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 317 +++++++++++++++++++++++++----------------- 1 file changed, 193 insertions(+), 124 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 98b4b2e1b930..a06c9437cdeb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -303,6 +303,8 @@ struct alc_customize_define { unsigned int fixup:1; /* Means that this sku is set by driver, not read from hw */ }; +struct alc_fixup; + struct alc_spec { /* codec parameterization */ struct snd_kcontrol_new *mixers[5]; /* mixer arrays */ @@ -404,6 +406,11 @@ struct alc_spec { /* for PLL fix */ hda_nid_t pll_nid; unsigned int pll_coef_idx, pll_coef_bit; + + /* fix-up list */ + int fixup_id; + const struct alc_fixup *fixup_list; + const char *fixup_name; }; /* @@ -1683,88 +1690,130 @@ struct alc_model_fixup { }; struct alc_fixup { - unsigned int sku; - const struct alc_pincfg *pins; - const struct hda_verb *verbs; - void (*func)(struct hda_codec *codec, const struct alc_fixup *fix, - int pre_init); + int type; + union { + unsigned int sku; + const struct alc_pincfg *pins; + const struct hda_verb *verbs; + void (*func)(struct hda_codec *codec, + const struct alc_fixup *fix, + int action); + } v; + bool chained; + int chain_id; }; -static void __alc_pick_fixup(struct hda_codec *codec, - const struct alc_fixup *fix, - const char *modelname, - int pre_init) +enum { + ALC_FIXUP_INVALID, + ALC_FIXUP_SKU, + ALC_FIXUP_PINS, + ALC_FIXUP_VERBS, + ALC_FIXUP_FUNC, +}; + +enum { + ALC_FIXUP_ACT_PRE_PROBE, + ALC_FIXUP_ACT_PROBE, +}; + +static void alc_apply_fixup(struct hda_codec *codec, int action) { - const struct alc_pincfg *cfg; - struct alc_spec *spec; + struct alc_spec *spec = codec->spec; + int id = spec->fixup_id; + const char *modelname = spec->fixup_name; + int depth = 0; - cfg = fix->pins; - if (pre_init && fix->sku) { -#ifdef CONFIG_SND_DEBUG_VERBOSE - snd_printdd(KERN_INFO "hda_codec: %s: Apply sku override for %s\n", - codec->chip_name, modelname); -#endif - spec = codec->spec; - spec->cdefine.sku_cfg = fix->sku; - spec->cdefine.fixup = 1; - } - if (pre_init && cfg) { -#ifdef CONFIG_SND_DEBUG_VERBOSE - snd_printdd(KERN_INFO "hda_codec: %s: Apply pincfg for %s\n", - codec->chip_name, modelname); -#endif - for (; cfg->nid; cfg++) - snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); - } - if (!pre_init && fix->verbs) { -#ifdef CONFIG_SND_DEBUG_VERBOSE - snd_printdd(KERN_INFO "hda_codec: %s: Apply fix-verbs for %s\n", - codec->chip_name, modelname); -#endif - add_verb(codec->spec, fix->verbs); - } - if (fix->func) { -#ifdef CONFIG_SND_DEBUG_VERBOSE - snd_printdd(KERN_INFO "hda_codec: %s: Apply fix-func for %s\n", - codec->chip_name, modelname); -#endif - fix->func(codec, fix, pre_init); + if (!spec->fixup_list) + return; + + while (id >= 0) { + const struct alc_fixup *fix = spec->fixup_list + id; + const struct alc_pincfg *cfg; + + switch (fix->type) { + case ALC_FIXUP_SKU: + if (action != ALC_FIXUP_ACT_PRE_PROBE || !fix->v.sku) + break;; + snd_printdd(KERN_INFO "hda_codec: %s: " + "Apply sku override for %s\n", + codec->chip_name, modelname); + spec->cdefine.sku_cfg = fix->v.sku; + spec->cdefine.fixup = 1; + break; + case ALC_FIXUP_PINS: + cfg = fix->v.pins; + if (action != ALC_FIXUP_ACT_PRE_PROBE || !cfg) + break; + snd_printdd(KERN_INFO "hda_codec: %s: " + "Apply pincfg for %s\n", + codec->chip_name, modelname); + for (; cfg->nid; cfg++) + snd_hda_codec_set_pincfg(codec, cfg->nid, + cfg->val); + break; + case ALC_FIXUP_VERBS: + if (action != ALC_FIXUP_ACT_PROBE || !fix->v.verbs) + break; + snd_printdd(KERN_INFO "hda_codec: %s: " + "Apply fix-verbs for %s\n", + codec->chip_name, modelname); + add_verb(codec->spec, fix->v.verbs); + break; + case ALC_FIXUP_FUNC: + if (!fix->v.func) + break; + snd_printdd(KERN_INFO "hda_codec: %s: " + "Apply fix-func for %s\n", + codec->chip_name, modelname); + fix->v.func(codec, fix, action); + break; + default: + snd_printk(KERN_ERR "hda_codec: %s: " + "Invalid fixup type %d\n", + codec->chip_name, fix->type); + break; + } + if (!fix[id].chained) + break; + if (++depth > 10) + break; + id = fix[id].chain_id; } } static void alc_pick_fixup(struct hda_codec *codec, - const struct snd_pci_quirk *quirk, - const struct alc_fixup *fix, - int pre_init) + const struct alc_model_fixup *models, + const struct snd_pci_quirk *quirk, + const struct alc_fixup *fixlist) { - quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk); - if (quirk) { - fix += quirk->value; -#ifdef CONFIG_SND_DEBUG_VERBOSE - __alc_pick_fixup(codec, fix, quirk->name, pre_init); -#else - __alc_pick_fixup(codec, fix, NULL, pre_init); -#endif - } -} + struct alc_spec *spec = codec->spec; + int id = -1; + const char *name = NULL; -static void alc_pick_fixup_model(struct hda_codec *codec, - const struct alc_model_fixup *models, - const struct snd_pci_quirk *quirk, - const struct alc_fixup *fix, - int pre_init) -{ if (codec->modelname && models) { while (models->name) { if (!strcmp(codec->modelname, models->name)) { - fix += models->id; + id = models->id; + name = models->name; break; } models++; } - __alc_pick_fixup(codec, fix, codec->modelname, pre_init); - } else { - alc_pick_fixup(codec, quirk, fix, pre_init); + } + if (id < 0) { + quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk); + if (quirk) { + id = quirk->value; +#ifdef CONFIG_SND_DEBUG_VERBOSE + name = quirk->name; +#endif + } + } + + spec->fixup_id = id; + if (id >= 0) { + spec->fixup_list = fixlist; + spec->fixup_name = name; } } @@ -7090,7 +7139,8 @@ enum { static const struct alc_fixup alc260_fixups[] = { [PINFIX_HP_DC5750] = { - .pins = (const struct alc_pincfg[]) { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { { 0x11, 0x90130110 }, /* speaker */ { } } @@ -7301,8 +7351,10 @@ static int patch_alc260(struct hda_codec *codec) board_config = ALC260_AUTO; } - if (board_config == ALC260_AUTO) - alc_pick_fixup(codec, alc260_fixup_tbl, alc260_fixups, 1); + if (board_config == ALC260_AUTO) { + alc_pick_fixup(codec, NULL, alc260_fixup_tbl, alc260_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + } if (board_config == ALC260_AUTO) { /* automatic parse from the BIOS config */ @@ -7350,8 +7402,7 @@ static int patch_alc260(struct hda_codec *codec) set_capture_mixer(codec); set_beep_amp(spec, 0x07, 0x05, HDA_INPUT); - if (board_config == ALC260_AUTO) - alc_pick_fixup(codec, alc260_fixup_tbl, alc260_fixups, 0); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); spec->vmaster_nid = 0x08; @@ -10678,7 +10729,8 @@ enum { static const struct alc_fixup alc882_fixups[] = { [PINFIX_ABIT_AW9D_MAX] = { - .pins = (const struct alc_pincfg[]) { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { { 0x15, 0x01080104 }, /* side */ { 0x16, 0x01011012 }, /* rear */ { 0x17, 0x01016011 }, /* clfe */ @@ -10686,13 +10738,15 @@ static const struct alc_fixup alc882_fixups[] = { } }, [PINFIX_PB_M5210] = { - .verbs = (const struct hda_verb[]) { + .type = ALC_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50 }, {} } }, [PINFIX_ACER_ASPIRE_7736] = { - .sku = ALC_FIXUP_SKU_IGNORE, + .type = ALC_FIXUP_SKU, + .v.sku = ALC_FIXUP_SKU_IGNORE, }, }; @@ -10978,8 +11032,10 @@ static int patch_alc882(struct hda_codec *codec) board_config = ALC882_AUTO; } - if (board_config == ALC882_AUTO) - alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups, 1); + if (board_config == ALC882_AUTO) { + alc_pick_fixup(codec, NULL, alc882_fixup_tbl, alc882_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + } alc_auto_parse_customize_define(codec); @@ -11055,8 +11111,7 @@ static int patch_alc882(struct hda_codec *codec) if (has_cdefine_beep(codec)) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); - if (board_config == ALC882_AUTO) - alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups, 0); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); spec->vmaster_nid = 0x0c; @@ -12446,7 +12501,8 @@ enum { static const struct alc_fixup alc262_fixups[] = { [PINFIX_FSC_H270] = { - .pins = (const struct alc_pincfg[]) { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { { 0x14, 0x99130110 }, /* speaker */ { 0x15, 0x0221142f }, /* front HP */ { 0x1b, 0x0121141f }, /* rear HP */ @@ -12883,8 +12939,10 @@ static int patch_alc262(struct hda_codec *codec) board_config = ALC262_AUTO; } - if (board_config == ALC262_AUTO) - alc_pick_fixup(codec, alc262_fixup_tbl, alc262_fixups, 1); + if (board_config == ALC262_AUTO) { + alc_pick_fixup(codec, NULL, alc262_fixup_tbl, alc262_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + } if (board_config == ALC262_AUTO) { /* automatic parse from the BIOS config */ @@ -12954,8 +13012,7 @@ static int patch_alc262(struct hda_codec *codec) if (!spec->no_analog && has_cdefine_beep(codec)) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); - if (board_config == ALC262_AUTO) - alc_pick_fixup(codec, alc262_fixup_tbl, alc262_fixups, 0); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); spec->vmaster_nid = 0x0c; @@ -14810,11 +14867,11 @@ static int alc269_resume(struct hda_codec *codec) #endif /* SND_HDA_NEEDS_RESUME */ static void alc269_fixup_hweq(struct hda_codec *codec, - const struct alc_fixup *fix, int pre_init) + const struct alc_fixup *fix, int action) { int coef; - if (pre_init) + if (action != ALC_FIXUP_ACT_PROBE) return; coef = alc_read_coef_idx(codec, 0x1e); alc_write_coef_idx(codec, 0x1e, coef | 0x80); @@ -14832,22 +14889,26 @@ enum { static const struct alc_fixup alc269_fixups[] = { [ALC269_FIXUP_SONY_VAIO] = { - .verbs = (const struct hda_verb[]) { + .type = ALC_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREFGRD}, {} } }, [ALC275_FIXUP_SONY_VAIO_GPIO2] = { - .verbs = (const struct hda_verb[]) { + .type = ALC_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { {0x01, AC_VERB_SET_GPIO_MASK, 0x04}, {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04}, {0x01, AC_VERB_SET_GPIO_DATA, 0x00}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREFGRD}, { } - } + }, + .chained = true, + .chain_id = ALC269_FIXUP_SONY_VAIO }, [ALC269_FIXUP_DELL_M101Z] = { - .verbs = (const struct hda_verb[]) { + .type = ALC_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { /* Enables internal speaker */ {0x20, AC_VERB_SET_COEF_INDEX, 13}, {0x20, AC_VERB_SET_PROC_COEF, 0x4040}, @@ -14855,29 +14916,28 @@ static const struct alc_fixup alc269_fixups[] = { } }, [ALC269_FIXUP_SKU_IGNORE] = { - .sku = ALC_FIXUP_SKU_IGNORE, + .type = ALC_FIXUP_SKU, + .v.sku = ALC_FIXUP_SKU_IGNORE, }, [ALC269_FIXUP_ASUS_G73JW] = { - .pins = (const struct alc_pincfg[]) { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { { 0x17, 0x99130111 }, /* subwoofer */ { } } }, [ALC269_FIXUP_LENOVO_EAPD] = { - .verbs = (const struct hda_verb[]) { + .type = ALC_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { {0x14, AC_VERB_SET_EAPD_BTLENABLE, 0}, {} } }, [ALC275_FIXUP_SONY_HWEQ] = { - .func = alc269_fixup_hweq, - .verbs = (const struct hda_verb[]) { - {0x01, AC_VERB_SET_GPIO_MASK, 0x04}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x00}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREFGRD}, - { } - } + .type = ALC_FIXUP_FUNC, + .v.func = alc269_fixup_hweq, + .chained = true, + .chain_id = ALC275_FIXUP_SONY_VAIO_GPIO2 } }; @@ -15174,8 +15234,10 @@ static int patch_alc269(struct hda_codec *codec) board_config = ALC269_AUTO; } - if (board_config == ALC269_AUTO) - alc_pick_fixup(codec, alc269_fixup_tbl, alc269_fixups, 1); + if (board_config == ALC269_AUTO) { + alc_pick_fixup(codec, NULL, alc269_fixup_tbl, alc269_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + } if (board_config == ALC269_AUTO) { /* automatic parse from the BIOS config */ @@ -15236,8 +15298,7 @@ static int patch_alc269(struct hda_codec *codec) if (has_cdefine_beep(codec)) set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); - if (board_config == ALC269_AUTO) - alc_pick_fixup(codec, alc269_fixup_tbl, alc269_fixups, 0); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); spec->vmaster_nid = 0x02; @@ -16296,7 +16357,8 @@ enum { static const struct alc_fixup alc861_fixups[] = { [PINFIX_FSC_AMILO_PI1505] = { - .pins = (const struct alc_pincfg[]) { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { { 0x0b, 0x0221101f }, /* HP */ { 0x0f, 0x90170310 }, /* speaker */ { } @@ -16331,8 +16393,10 @@ static int patch_alc861(struct hda_codec *codec) board_config = ALC861_AUTO; } - if (board_config == ALC861_AUTO) - alc_pick_fixup(codec, alc861_fixup_tbl, alc861_fixups, 1); + if (board_config == ALC861_AUTO) { + alc_pick_fixup(codec, NULL, alc861_fixup_tbl, alc861_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + } if (board_config == ALC861_AUTO) { /* automatic parse from the BIOS config */ @@ -16369,8 +16433,7 @@ static int patch_alc861(struct hda_codec *codec) spec->vmaster_nid = 0x03; - if (board_config == ALC861_AUTO) - alc_pick_fixup(codec, alc861_fixup_tbl, alc861_fixups, 0); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); codec->patch_ops = alc_patch_ops; if (board_config == ALC861_AUTO) { @@ -17252,7 +17315,8 @@ enum { /* reset GPIO1 */ static const struct alc_fixup alc861vd_fixups[] = { [ALC660VD_FIX_ASUS_GPIO1] = { - .verbs = (const struct hda_verb[]) { + .type = ALC_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { {0x01, AC_VERB_SET_GPIO_MASK, 0x03}, {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01}, {0x01, AC_VERB_SET_GPIO_DATA, 0x01}, @@ -17287,8 +17351,10 @@ static int patch_alc861vd(struct hda_codec *codec) board_config = ALC861VD_AUTO; } - if (board_config == ALC861VD_AUTO) - alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups, 1); + if (board_config == ALC861VD_AUTO) { + alc_pick_fixup(codec, NULL, alc861vd_fixup_tbl, alc861vd_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + } if (board_config == ALC861VD_AUTO) { /* automatic parse from the BIOS config */ @@ -17336,8 +17402,7 @@ static int patch_alc861vd(struct hda_codec *codec) spec->vmaster_nid = 0x02; - if (board_config == ALC861VD_AUTO) - alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups, 0); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); codec->patch_ops = alc_patch_ops; @@ -19368,9 +19433,9 @@ static void alc662_auto_init(struct hda_codec *codec) } static void alc272_fixup_mario(struct hda_codec *codec, - const struct alc_fixup *fix, int pre_init) + const struct alc_fixup *fix, int action) { - if (!pre_init) + if (action != ALC_FIXUP_ACT_PROBE) return; if (snd_hda_override_amp_caps(codec, 0x2, HDA_OUTPUT, (0x3b << AC_AMPCAP_OFFSET_SHIFT) | @@ -19389,19 +19454,22 @@ enum { static const struct alc_fixup alc662_fixups[] = { [ALC662_FIXUP_ASPIRE] = { - .pins = (const struct alc_pincfg[]) { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { { 0x15, 0x99130112 }, /* subwoofer */ { } } }, [ALC662_FIXUP_IDEAPAD] = { - .pins = (const struct alc_pincfg[]) { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { { 0x17, 0x99130112 }, /* subwoofer */ { } } }, [ALC272_FIXUP_MARIO] = { - .func = alc272_fixup_mario, + .type = ALC_FIXUP_FUNC, + .v.func = alc272_fixup_mario, } }; @@ -19455,7 +19523,9 @@ static int patch_alc662(struct hda_codec *codec) } if (board_config == ALC662_AUTO) { - alc_pick_fixup(codec, alc662_fixup_tbl, alc662_fixups, 1); + alc_pick_fixup(codec, alc662_fixup_models, + alc662_fixup_tbl, alc662_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); /* automatic parse from the BIOS config */ err = alc662_parse_auto_config(codec); if (err < 0) { @@ -19513,12 +19583,11 @@ static int patch_alc662(struct hda_codec *codec) } spec->vmaster_nid = 0x02; + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); + codec->patch_ops = alc_patch_ops; - if (board_config == ALC662_AUTO) { + if (board_config == ALC662_AUTO) spec->init_hook = alc662_auto_init; - alc_pick_fixup_model(codec, alc662_fixup_models, - alc662_fixup_tbl, alc662_fixups, 0); - } alc_init_jacks(codec); -- cgit v1.2.3 From 5870112021fb38e73b25dad3baec4ca0819c594a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 13 Jan 2011 15:41:45 +0100 Subject: ALSA: hda - Add fixup-call in init callback In some cases, the fix-up is required in the init callback to be called both at the first initialization and at the resume. The new action type ALC_FIXUP_ACT_INIT is used for this case. So far, only ALC275_FIXUP_SONY_HWEQ uses this. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a06c9437cdeb..b445ae989421 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1714,6 +1714,7 @@ enum { enum { ALC_FIXUP_ACT_PRE_PROBE, ALC_FIXUP_ACT_PROBE, + ALC_FIXUP_ACT_INIT, }; static void alc_apply_fixup(struct hda_codec *codec, int action) @@ -3910,6 +3911,8 @@ static int alc_init(struct hda_codec *codec) if (spec->init_hook) spec->init_hook(codec); + alc_apply_fixup(codec, ALC_FIXUP_ACT_INIT); + hda_call_check_power_status(codec, 0x01); return 0; } @@ -14871,7 +14874,7 @@ static void alc269_fixup_hweq(struct hda_codec *codec, { int coef; - if (action != ALC_FIXUP_ACT_PROBE) + if (action != ALC_FIXUP_ACT_INIT) return; coef = alc_read_coef_idx(codec, 0x1e); alc_write_coef_idx(codec, 0x1e, coef | 0x80); -- cgit v1.2.3 From ad09fc9d2156f3d37537b34418a6b79309013d33 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 14 Jan 2011 09:42:27 +0100 Subject: ALSA: hda - Suppress the odd number of channels for HDMI It looks like that HDMI codecs don't support the odd number of channels although HD-audio spec doesn't have the restriction. Add the hw_constraint to limit to only the even number of channels. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index f29b97b5de8f..2d288793ceb3 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1238,6 +1238,9 @@ static int simple_playback_pcm_open(struct hda_pcm_stream *hinfo, snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_constraints_channels); + } else { + snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, 2); } return snd_hda_multi_out_dig_open(codec, &spec->multiout); -- cgit v1.2.3 From f8fe80e4383bf5f542beb80bf2abe9fc1505c366 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 14 Jan 2011 08:07:50 +0100 Subject: ALSA: oxygen: Xonar DG: fix CS4245 register writes Accidentally exchanging register addresses and register values leads to many strange errors ... Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/xonar_dg.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c index e4de0b8d087a..e1fa602eba79 100644 --- a/sound/pci/oxygen/xonar_dg.c +++ b/sound/pci/oxygen/xonar_dg.c @@ -75,7 +75,7 @@ static void cs4245_write(struct oxygen *chip, unsigned int reg, u8 value) OXYGEN_SPI_CEN_LATCH_CLOCK_HI, CS4245_SPI_ADDRESS | CS4245_SPI_WRITE | - (value << 8) | reg); + (reg << 8) | value); data->cs4245_regs[reg] = value; } -- cgit v1.2.3 From 361fe6e90888af83d5bfdfc152d737018cbede43 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 14 Jan 2011 09:55:32 +0100 Subject: ALSA: hda - Rearrange fixup struct in patch_realtek.c Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b445ae989421..69554061c16e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1691,6 +1691,8 @@ struct alc_model_fixup { struct alc_fixup { int type; + bool chained; + int chain_id; union { unsigned int sku; const struct alc_pincfg *pins; @@ -1699,8 +1701,6 @@ struct alc_fixup { const struct alc_fixup *fix, int action); } v; - bool chained; - int chain_id; }; enum { -- cgit v1.2.3 From 639cef0eb6df05d5516520aa89b0c9fe62ee2d3b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 14 Jan 2011 10:30:46 +0100 Subject: ALSA: hda - Store PCM parameters properly in HDMI open callback In hdmi_pcm_open(), the evaluated PCM hw parameters are stored in hinfo, but these aren't properly set back to the current runtime record since these have been set beforehand in azx_pcm_open(). This patch fixes the behavior. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 2d288793ceb3..5980552f5970 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -817,6 +817,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, struct hdmi_spec *spec = codec->spec; struct hdmi_eld *eld; struct hda_pcm_stream *codec_pars; + struct snd_pcm_runtime *runtime = substream->runtime; unsigned int idx; for (idx = 0; idx < spec->num_cvts; idx++) @@ -844,6 +845,11 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, hinfo->formats = codec_pars->formats; hinfo->maxbps = codec_pars->maxbps; } + /* store the updated parameters */ + runtime->hw.channels_min = hinfo->channels_min; + runtime->hw.channels_max = hinfo->channels_max; + runtime->hw.formats = hinfo->formats; + runtime->hw.rates = hinfo->rates; return 0; } -- cgit v1.2.3 From 4fe2ca14678174d9df254ae3ba2b79bacc19e2ee Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 14 Jan 2011 10:33:26 +0100 Subject: ALSA: hda - More coverage for odd-number channels elimination for HDMI The commit ad09fc9d2156f3d37537b34418a6b79309013d33 didn't cover the case for Intel and Nvidia HDMIs, where hdmi_pcm_open() is called. Put the hw_constraint there, too. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 5980552f5970..2d5b83fa8d24 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -850,6 +850,9 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, runtime->hw.channels_max = hinfo->channels_max; runtime->hw.formats = hinfo->formats; runtime->hw.rates = hinfo->rates; + + snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, 2); return 0; } -- cgit v1.2.3 From 228dd545147a07c5e81e4732ad0e829d19ce5daa Mon Sep 17 00:00:00 2001 From: Matti J. Aaltonen Date: Thu, 13 Jan 2011 15:22:45 +0200 Subject: ASoC: WL1273 FM radio: Fix breakage with MFD API changes These changes are needed to keep up with the changes in the MFD core and V4L2 parts of the wl1273 FM radio driver. Use function pointers instead of exported functions for I2C IO. Also move all preprocessor constants from the wl1273.h to include/linux/mfd/wl1273-core.h. Signed-off-by: Matti J. Aaltonen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/wl1273.c | 29 ++++++++----------- sound/soc/codecs/wl1273.h | 71 ----------------------------------------------- 3 files changed, 13 insertions(+), 89 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 883a312bb293..c48b23c1d4fc 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -44,7 +44,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TWL6040 if TWL4030_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C - select SND_SOC_WL1273 if WL1273_CORE + select SND_SOC_WL1273 if RADIO_WL1273 select SND_SOC_WM2000 if I2C select SND_SOC_WM8350 if MFD_WM8350 select SND_SOC_WM8400 if MFD_WM8400 diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index d3ffa2f0122a..861b28f543d2 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -42,7 +42,7 @@ struct wl1273_priv { static int snd_wl1273_fm_set_i2s_mode(struct wl1273_core *core, int rate, int width) { - struct device *dev = &core->i2c_dev->dev; + struct device *dev = &core->client->dev; int r = 0; u16 mode; @@ -123,13 +123,13 @@ static int snd_wl1273_fm_set_i2s_mode(struct wl1273_core *core, dev_dbg(dev, "mode: 0x%04x\n", mode); if (core->i2s_mode != mode) { - r = wl1273_fm_write_cmd(core, WL1273_I2S_MODE_CONFIG_SET, mode); + r = core->write(core, WL1273_I2S_MODE_CONFIG_SET, mode); if (r) goto out; core->i2s_mode = mode; - r = wl1273_fm_write_cmd(core, WL1273_AUDIO_ENABLE, - WL1273_AUDIO_ENABLE_I2S); + r = core->write(core, WL1273_AUDIO_ENABLE, + WL1273_AUDIO_ENABLE_I2S); if (r) goto out; } @@ -142,8 +142,7 @@ out: static int snd_wl1273_fm_set_channel_number(struct wl1273_core *core, int channel_number) { - struct i2c_client *client = core->i2c_dev; - struct device *dev = &client->dev; + struct device *dev = &core->client->dev; int r = 0; dev_dbg(dev, "%s\n", __func__); @@ -154,17 +153,13 @@ static int snd_wl1273_fm_set_channel_number(struct wl1273_core *core, goto out; if (channel_number == 1 && core->mode == WL1273_MODE_RX) - r = wl1273_fm_write_cmd(core, WL1273_MOST_MODE_SET, - WL1273_RX_MONO); + r = core->write(core, WL1273_MOST_MODE_SET, WL1273_RX_MONO); else if (channel_number == 1 && core->mode == WL1273_MODE_TX) - r = wl1273_fm_write_cmd(core, WL1273_MONO_SET, - WL1273_TX_MONO); + r = core->write(core, WL1273_MONO_SET, WL1273_TX_MONO); else if (channel_number == 2 && core->mode == WL1273_MODE_RX) - r = wl1273_fm_write_cmd(core, WL1273_MOST_MODE_SET, - WL1273_RX_STEREO); + r = core->write(core, WL1273_MOST_MODE_SET, WL1273_RX_STEREO); else if (channel_number == 2 && core->mode == WL1273_MODE_TX) - r = wl1273_fm_write_cmd(core, WL1273_MONO_SET, - WL1273_TX_STEREO); + r = core->write(core, WL1273_MONO_SET, WL1273_TX_STEREO); else r = -EINVAL; out: @@ -237,7 +232,7 @@ static int snd_wl1273_fm_audio_put(struct snd_kcontrol *kcontrol, if (wl1273->core->audio_mode == val) return 0; - r = wl1273_fm_set_audio(wl1273->core, val); + r = wl1273->core->set_audio(wl1273->core, val); if (r < 0) return r; @@ -272,8 +267,8 @@ static int snd_wl1273_fm_volume_put(struct snd_kcontrol *kcontrol, dev_dbg(codec->dev, "%s: enter.\n", __func__); - r = wl1273_fm_set_volume(wl1273->core, - ucontrol->value.integer.value[0]); + r = wl1273->core->set_volume(wl1273->core, + ucontrol->value.integer.value[0]); if (r) return r; diff --git a/sound/soc/codecs/wl1273.h b/sound/soc/codecs/wl1273.h index 14ed027fdcfc..43ec7e668c51 100644 --- a/sound/soc/codecs/wl1273.h +++ b/sound/soc/codecs/wl1273.h @@ -25,77 +25,6 @@ #ifndef __WL1273_CODEC_H__ #define __WL1273_CODEC_H__ -/* I2S protocol, left channel first, data width 16 bits */ -#define WL1273_PCM_DEF_MODE 0x00 - -/* Rx */ -#define WL1273_AUDIO_ENABLE_I2S (1 << 0) -#define WL1273_AUDIO_ENABLE_ANALOG (1 << 1) - -/* Tx */ -#define WL1273_AUDIO_IO_SET_ANALOG 0 -#define WL1273_AUDIO_IO_SET_I2S 1 - -#define WL1273_POWER_SET_OFF 0 -#define WL1273_POWER_SET_FM (1 << 0) -#define WL1273_POWER_SET_RDS (1 << 1) -#define WL1273_POWER_SET_RETENTION (1 << 4) - -#define WL1273_PUPD_SET_OFF 0x00 -#define WL1273_PUPD_SET_ON 0x01 -#define WL1273_PUPD_SET_RETENTION 0x10 - -/* I2S mode */ -#define WL1273_IS2_WIDTH_32 0x0 -#define WL1273_IS2_WIDTH_40 0x1 -#define WL1273_IS2_WIDTH_22_23 0x2 -#define WL1273_IS2_WIDTH_23_22 0x3 -#define WL1273_IS2_WIDTH_48 0x4 -#define WL1273_IS2_WIDTH_50 0x5 -#define WL1273_IS2_WIDTH_60 0x6 -#define WL1273_IS2_WIDTH_64 0x7 -#define WL1273_IS2_WIDTH_80 0x8 -#define WL1273_IS2_WIDTH_96 0x9 -#define WL1273_IS2_WIDTH_128 0xa -#define WL1273_IS2_WIDTH 0xf - -#define WL1273_IS2_FORMAT_STD (0x0 << 4) -#define WL1273_IS2_FORMAT_LEFT (0x1 << 4) -#define WL1273_IS2_FORMAT_RIGHT (0x2 << 4) -#define WL1273_IS2_FORMAT_USER (0x3 << 4) - -#define WL1273_IS2_MASTER (0x0 << 6) -#define WL1273_IS2_SLAVEW (0x1 << 6) - -#define WL1273_IS2_TRI_AFTER_SENDING (0x0 << 7) -#define WL1273_IS2_TRI_ALWAYS_ACTIVE (0x1 << 7) - -#define WL1273_IS2_SDOWS_RR (0x0 << 8) -#define WL1273_IS2_SDOWS_RF (0x1 << 8) -#define WL1273_IS2_SDOWS_FR (0x2 << 8) -#define WL1273_IS2_SDOWS_FF (0x3 << 8) - -#define WL1273_IS2_TRI_OPT (0x0 << 10) -#define WL1273_IS2_TRI_ALWAYS (0x1 << 10) - -#define WL1273_IS2_RATE_48K (0x0 << 12) -#define WL1273_IS2_RATE_44_1K (0x1 << 12) -#define WL1273_IS2_RATE_32K (0x2 << 12) -#define WL1273_IS2_RATE_22_05K (0x4 << 12) -#define WL1273_IS2_RATE_16K (0x5 << 12) -#define WL1273_IS2_RATE_12K (0x8 << 12) -#define WL1273_IS2_RATE_11_025 (0x9 << 12) -#define WL1273_IS2_RATE_8K (0xa << 12) -#define WL1273_IS2_RATE (0xf << 12) - -#define WL1273_I2S_DEF_MODE (WL1273_IS2_WIDTH_32 | \ - WL1273_IS2_FORMAT_STD | \ - WL1273_IS2_MASTER | \ - WL1273_IS2_TRI_AFTER_SENDING | \ - WL1273_IS2_SDOWS_RR | \ - WL1273_IS2_TRI_OPT | \ - WL1273_IS2_RATE_48K) - int wl1273_get_format(struct snd_soc_codec *codec, unsigned int *fmt); #endif /* End of __WL1273_CODEC_H__ */ -- cgit v1.2.3 From 3e8b3b90fecedcf20d895c4e6ad01a379fe252bf Mon Sep 17 00:00:00 2001 From: Hanno Boeck Date: Fri, 14 Jan 2011 19:14:47 +0100 Subject: ALSA: constify functions in ac97 Signed-off-by: Hanno Boeck Signed-off-by: Takashi Iwai --- include/sound/ac97_codec.h | 2 +- sound/pci/ac97/ac97_codec.c | 2 +- sound/pci/ac97/ac97_patch.c | 62 ++++++++++++++++++++++----------------------- 3 files changed, 33 insertions(+), 33 deletions(-) (limited to 'sound') diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index 49400459b477..b602f475cdbb 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -477,7 +477,7 @@ struct snd_ac97_template { struct snd_ac97 { /* -- lowlevel (hardware) driver specific -- */ - struct snd_ac97_build_ops * build_ops; + const struct snd_ac97_build_ops *build_ops; void *private_data; void (*private_free) (struct snd_ac97 *ac97); /* --- */ diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 0fc614ce16c1..cb62d178b3e0 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -1961,7 +1961,7 @@ static int snd_ac97_dev_disconnect(struct snd_device *device) } /* build_ops to do nothing */ -static struct snd_ac97_build_ops null_build_ops; +static const struct snd_ac97_build_ops null_build_ops; #ifdef CONFIG_SND_AC97_POWER_SAVE static void do_update_power(struct work_struct *work) diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index e68c98ef4041..bf47574ca1f0 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -371,7 +371,7 @@ static int patch_yamaha_ymf743_build_spdif(struct snd_ac97 *ac97) return 0; } -static struct snd_ac97_build_ops patch_yamaha_ymf743_ops = { +static const struct snd_ac97_build_ops patch_yamaha_ymf743_ops = { .build_spdif = patch_yamaha_ymf743_build_spdif, .build_3d = patch_yamaha_ymf7x3_3d, }; @@ -455,7 +455,7 @@ static int patch_yamaha_ymf753_post_spdif(struct snd_ac97 * ac97) return 0; } -static struct snd_ac97_build_ops patch_yamaha_ymf753_ops = { +static const struct snd_ac97_build_ops patch_yamaha_ymf753_ops = { .build_3d = patch_yamaha_ymf7x3_3d, .build_post_spdif = patch_yamaha_ymf753_post_spdif }; @@ -502,7 +502,7 @@ static int patch_wolfson_wm9703_specific(struct snd_ac97 * ac97) return 0; } -static struct snd_ac97_build_ops patch_wolfson_wm9703_ops = { +static const struct snd_ac97_build_ops patch_wolfson_wm9703_ops = { .build_specific = patch_wolfson_wm9703_specific, }; @@ -533,7 +533,7 @@ static int patch_wolfson_wm9704_specific(struct snd_ac97 * ac97) return 0; } -static struct snd_ac97_build_ops patch_wolfson_wm9704_ops = { +static const struct snd_ac97_build_ops patch_wolfson_wm9704_ops = { .build_specific = patch_wolfson_wm9704_specific, }; @@ -677,7 +677,7 @@ static int patch_wolfson_wm9711_specific(struct snd_ac97 * ac97) return 0; } -static struct snd_ac97_build_ops patch_wolfson_wm9711_ops = { +static const struct snd_ac97_build_ops patch_wolfson_wm9711_ops = { .build_specific = patch_wolfson_wm9711_specific, }; @@ -871,7 +871,7 @@ static void patch_wolfson_wm9713_resume (struct snd_ac97 * ac97) } #endif -static struct snd_ac97_build_ops patch_wolfson_wm9713_ops = { +static const struct snd_ac97_build_ops patch_wolfson_wm9713_ops = { .build_specific = patch_wolfson_wm9713_specific, .build_3d = patch_wolfson_wm9713_3d, #ifdef CONFIG_PM @@ -976,7 +976,7 @@ static int patch_sigmatel_stac97xx_specific(struct snd_ac97 * ac97) return 0; } -static struct snd_ac97_build_ops patch_sigmatel_stac9700_ops = { +static const struct snd_ac97_build_ops patch_sigmatel_stac9700_ops = { .build_3d = patch_sigmatel_stac9700_3d, .build_specific = patch_sigmatel_stac97xx_specific }; @@ -1023,7 +1023,7 @@ static int patch_sigmatel_stac9708_specific(struct snd_ac97 *ac97) return patch_sigmatel_stac97xx_specific(ac97); } -static struct snd_ac97_build_ops patch_sigmatel_stac9708_ops = { +static const struct snd_ac97_build_ops patch_sigmatel_stac9708_ops = { .build_3d = patch_sigmatel_stac9708_3d, .build_specific = patch_sigmatel_stac9708_specific }; @@ -1252,7 +1252,7 @@ static int patch_sigmatel_stac9758_specific(struct snd_ac97 *ac97) return 0; } -static struct snd_ac97_build_ops patch_sigmatel_stac9758_ops = { +static const struct snd_ac97_build_ops patch_sigmatel_stac9758_ops = { .build_3d = patch_sigmatel_stac9700_3d, .build_specific = patch_sigmatel_stac9758_specific }; @@ -1327,7 +1327,7 @@ static int patch_cirrus_build_spdif(struct snd_ac97 * ac97) return 0; } -static struct snd_ac97_build_ops patch_cirrus_ops = { +static const struct snd_ac97_build_ops patch_cirrus_ops = { .build_spdif = patch_cirrus_build_spdif }; @@ -1384,7 +1384,7 @@ static int patch_conexant_build_spdif(struct snd_ac97 * ac97) return 0; } -static struct snd_ac97_build_ops patch_conexant_ops = { +static const struct snd_ac97_build_ops patch_conexant_ops = { .build_spdif = patch_conexant_build_spdif }; @@ -1560,7 +1560,7 @@ static void patch_ad1881_chained(struct snd_ac97 * ac97, int unchained_idx, int } } -static struct snd_ac97_build_ops patch_ad1881_build_ops = { +static const struct snd_ac97_build_ops patch_ad1881_build_ops = { #ifdef CONFIG_PM .resume = ad18xx_resume #endif @@ -1647,7 +1647,7 @@ static int patch_ad1885_specific(struct snd_ac97 * ac97) return 0; } -static struct snd_ac97_build_ops patch_ad1885_build_ops = { +static const struct snd_ac97_build_ops patch_ad1885_build_ops = { .build_specific = &patch_ad1885_specific, #ifdef CONFIG_PM .resume = ad18xx_resume @@ -1674,7 +1674,7 @@ static int patch_ad1886_specific(struct snd_ac97 * ac97) return 0; } -static struct snd_ac97_build_ops patch_ad1886_build_ops = { +static const struct snd_ac97_build_ops patch_ad1886_build_ops = { .build_specific = &patch_ad1886_specific, #ifdef CONFIG_PM .resume = ad18xx_resume @@ -1881,7 +1881,7 @@ static int patch_ad1981a_specific(struct snd_ac97 * ac97) ARRAY_SIZE(snd_ac97_ad1981x_jack_sense)); } -static struct snd_ac97_build_ops patch_ad1981a_build_ops = { +static const struct snd_ac97_build_ops patch_ad1981a_build_ops = { .build_post_spdif = patch_ad198x_post_spdif, .build_specific = patch_ad1981a_specific, #ifdef CONFIG_PM @@ -1936,7 +1936,7 @@ static int patch_ad1981b_specific(struct snd_ac97 *ac97) ARRAY_SIZE(snd_ac97_ad1981x_jack_sense)); } -static struct snd_ac97_build_ops patch_ad1981b_build_ops = { +static const struct snd_ac97_build_ops patch_ad1981b_build_ops = { .build_post_spdif = patch_ad198x_post_spdif, .build_specific = patch_ad1981b_specific, #ifdef CONFIG_PM @@ -2075,7 +2075,7 @@ static int patch_ad1888_specific(struct snd_ac97 *ac97) return patch_build_controls(ac97, snd_ac97_ad1888_controls, ARRAY_SIZE(snd_ac97_ad1888_controls)); } -static struct snd_ac97_build_ops patch_ad1888_build_ops = { +static const struct snd_ac97_build_ops patch_ad1888_build_ops = { .build_post_spdif = patch_ad198x_post_spdif, .build_specific = patch_ad1888_specific, #ifdef CONFIG_PM @@ -2124,7 +2124,7 @@ static int patch_ad1980_specific(struct snd_ac97 *ac97) return patch_build_controls(ac97, &snd_ac97_ad198x_2cmic, 1); } -static struct snd_ac97_build_ops patch_ad1980_build_ops = { +static const struct snd_ac97_build_ops patch_ad1980_build_ops = { .build_post_spdif = patch_ad198x_post_spdif, .build_specific = patch_ad1980_specific, #ifdef CONFIG_PM @@ -2239,7 +2239,7 @@ static int patch_ad1985_specific(struct snd_ac97 *ac97) ARRAY_SIZE(snd_ac97_ad1985_controls)); } -static struct snd_ac97_build_ops patch_ad1985_build_ops = { +static const struct snd_ac97_build_ops patch_ad1985_build_ops = { .build_post_spdif = patch_ad198x_post_spdif, .build_specific = patch_ad1985_specific, #ifdef CONFIG_PM @@ -2531,7 +2531,7 @@ static int patch_ad1986_specific(struct snd_ac97 *ac97) ARRAY_SIZE(snd_ac97_ad1985_controls)); } -static struct snd_ac97_build_ops patch_ad1986_build_ops = { +static const struct snd_ac97_build_ops patch_ad1986_build_ops = { .build_post_spdif = patch_ad198x_post_spdif, .build_specific = patch_ad1986_specific, #ifdef CONFIG_PM @@ -2636,7 +2636,7 @@ static int patch_alc650_specific(struct snd_ac97 * ac97) return 0; } -static struct snd_ac97_build_ops patch_alc650_ops = { +static const struct snd_ac97_build_ops patch_alc650_ops = { .build_specific = patch_alc650_specific, .update_jacks = alc650_update_jacks }; @@ -2788,7 +2788,7 @@ static int patch_alc655_specific(struct snd_ac97 * ac97) return 0; } -static struct snd_ac97_build_ops patch_alc655_ops = { +static const struct snd_ac97_build_ops patch_alc655_ops = { .build_specific = patch_alc655_specific, .update_jacks = alc655_update_jacks }; @@ -2900,7 +2900,7 @@ static int patch_alc850_specific(struct snd_ac97 *ac97) return 0; } -static struct snd_ac97_build_ops patch_alc850_ops = { +static const struct snd_ac97_build_ops patch_alc850_ops = { .build_specific = patch_alc850_specific, .update_jacks = alc850_update_jacks }; @@ -2962,7 +2962,7 @@ static int patch_cm9738_specific(struct snd_ac97 * ac97) return patch_build_controls(ac97, snd_ac97_cm9738_controls, ARRAY_SIZE(snd_ac97_cm9738_controls)); } -static struct snd_ac97_build_ops patch_cm9738_ops = { +static const struct snd_ac97_build_ops patch_cm9738_ops = { .build_specific = patch_cm9738_specific, .update_jacks = cm9738_update_jacks }; @@ -3053,7 +3053,7 @@ static int patch_cm9739_post_spdif(struct snd_ac97 * ac97) return patch_build_controls(ac97, snd_ac97_cm9739_controls_spdif, ARRAY_SIZE(snd_ac97_cm9739_controls_spdif)); } -static struct snd_ac97_build_ops patch_cm9739_ops = { +static const struct snd_ac97_build_ops patch_cm9739_ops = { .build_specific = patch_cm9739_specific, .build_post_spdif = patch_cm9739_post_spdif, .update_jacks = cm9739_update_jacks @@ -3227,7 +3227,7 @@ static int patch_cm9761_specific(struct snd_ac97 * ac97) return patch_build_controls(ac97, snd_ac97_cm9761_controls, ARRAY_SIZE(snd_ac97_cm9761_controls)); } -static struct snd_ac97_build_ops patch_cm9761_ops = { +static const struct snd_ac97_build_ops patch_cm9761_ops = { .build_specific = patch_cm9761_specific, .build_post_spdif = patch_cm9761_post_spdif, .update_jacks = cm9761_update_jacks @@ -3323,7 +3323,7 @@ static int patch_cm9780_specific(struct snd_ac97 *ac97) return patch_build_controls(ac97, cm9780_controls, ARRAY_SIZE(cm9780_controls)); } -static struct snd_ac97_build_ops patch_cm9780_ops = { +static const struct snd_ac97_build_ops patch_cm9780_ops = { .build_specific = patch_cm9780_specific, .build_post_spdif = patch_cm9761_post_spdif /* identical with CM9761 */ }; @@ -3443,7 +3443,7 @@ static int patch_vt1616_specific(struct snd_ac97 * ac97) return 0; } -static struct snd_ac97_build_ops patch_vt1616_ops = { +static const struct snd_ac97_build_ops patch_vt1616_ops = { .build_specific = patch_vt1616_specific }; @@ -3797,7 +3797,7 @@ static int patch_it2646_specific(struct snd_ac97 * ac97) return 0; } -static struct snd_ac97_build_ops patch_it2646_ops = { +static const struct snd_ac97_build_ops patch_it2646_ops = { .build_specific = patch_it2646_specific, .update_jacks = it2646_update_jacks }; @@ -3831,7 +3831,7 @@ static int patch_si3036_specific(struct snd_ac97 * ac97) return 0; } -static struct snd_ac97_build_ops patch_si3036_ops = { +static const struct snd_ac97_build_ops patch_si3036_ops = { .build_specific = patch_si3036_specific, }; @@ -3898,7 +3898,7 @@ static int patch_ucb1400_specific(struct snd_ac97 * ac97) return 0; } -static struct snd_ac97_build_ops patch_ucb1400_ops = { +static const struct snd_ac97_build_ops patch_ucb1400_ops = { .build_specific = patch_ucb1400_specific, }; -- cgit v1.2.3 From d9ab344336f74c012f6643ed3d1ad8ca0136de3b Mon Sep 17 00:00:00 2001 From: Raymond Yau Date: Sun, 16 Jan 2011 10:55:54 +0800 Subject: ALSA : au88x0 - Limit number of channels to fix Oops via OSS emu Fix playback/capture channels patch to change supported playback channels of au8830 to 1,2,4 and capture channels to 1,2. This prevent oops when oss emulation use SNDCTL_DSP_CHANNELS to set 3 Channels Signed-off-by: Raymond Yau Cc: Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0_pcm.c | 24 ++++++++++++++++++++---- 1 file changed, 20 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index b9d2f202cf9b..5439d662d104 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -42,11 +42,7 @@ static struct snd_pcm_hardware snd_vortex_playback_hw_adb = { .rate_min = 5000, .rate_max = 48000, .channels_min = 1, -#ifdef CHIP_AU8830 - .channels_max = 4, -#else .channels_max = 2, -#endif .buffer_bytes_max = 0x10000, .period_bytes_min = 0x1, .period_bytes_max = 0x1000, @@ -115,6 +111,17 @@ static struct snd_pcm_hardware snd_vortex_playback_hw_wt = { .periods_max = 64, }; #endif +#ifdef CHIP_AU8830 +static unsigned int au8830_channels[3] = { + 1, 2, 4, +}; + +static struct snd_pcm_hw_constraint_list hw_constraints_au8830_channels = { + .count = ARRAY_SIZE(au8830_channels), + .list = au8830_channels, + .mask = 0, +}; +#endif /* open callback */ static int snd_vortex_pcm_open(struct snd_pcm_substream *substream) { @@ -156,6 +163,15 @@ static int snd_vortex_pcm_open(struct snd_pcm_substream *substream) if (VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_ADB || VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_I2S) runtime->hw = snd_vortex_playback_hw_adb; +#ifdef CHIP_AU8830 + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_ADB) { + runtime->hw.channels_max = 4; + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &hw_constraints_au8830_channels); + } +#endif substream->runtime->private_data = NULL; } #ifndef CHIP_AU8810 -- cgit v1.2.3 From 7ebcf5d6021a696680ee77d9162a2edec2d671dd Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Fri, 14 Jan 2011 15:59:13 +0000 Subject: ASoC: WM8990: msleep() takes milliseconds not jiffies Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8990.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 5c87a634fc04..100aeee5ba96 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1183,7 +1183,7 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, WM8990_VMIDTOG); /* Delay to allow output caps to discharge */ - msleep(msecs_to_jiffies(300)); + msleep(300); /* Disable VMIDTOG */ snd_soc_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | @@ -1195,17 +1195,17 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, /* Enable outputs */ snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1b00); - msleep(msecs_to_jiffies(50)); + msleep(50); /* Enable VMID at 2x50k */ snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f02); - msleep(msecs_to_jiffies(100)); + msleep(100); /* Enable VREF */ snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03); - msleep(msecs_to_jiffies(600)); + msleep(600); /* Enable BUFIOEN */ snd_soc_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | @@ -1250,7 +1250,7 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, /* Disable VMID */ snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f01); - msleep(msecs_to_jiffies(300)); + msleep(300); /* Enable all output discharge bits */ snd_soc_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE | -- cgit v1.2.3 From 7322ce21cde92777a9b11e17429d61d1cda6d2c2 Mon Sep 17 00:00:00 2001 From: Alexander Sverdlin Date: Sun, 16 Jan 2011 15:48:05 +0300 Subject: ASoC: EP93xx: fixed LRCLK rate and DMA oper. in I2S code Changelog: 1. I2S module of EP93xx should be feed by 32bit DMA transfers. This is hardware limitation and that's the way original Cirrus's driver worked. This will fix distorted sound playback and make capture actually work in present ep93xx drivers. I've found, that author of code, on which modern ep93xx-i2s.c and ep93xx-pcm.c are based, had faced this problem also in 2007: http://blog.gmane.org/gmane.linux.ports.arm.cirrus/month=20070101/page=3 Now SoC code uses his developments, but not overcomes the hardware issues. Some details from EP93xx users guide: Both I2S transmitter and receiver have similar 16x32bit FIFO, where they store 8 samples for both left and right channels. The FIFO is always 32bit wide and should be properly aligned if you use samples of other width. Transmitter and receiver have configuration registers for selection of I2S word length (16, 24, 32). They are I2STXWrdLen and I2SRXWrdLen. Yes, EP93xx DMA can do byte, word and quad-word transfers. The width for transfers to and from peripherals is selected by particular module configuration. Lucky AC97 module has such configuration: AC97RXCRx registers, bit CM (Compact mode enable) switches between 16 and 32 bit samples. AC97TXCRx registers have the same bits for transmitters. ep93xx-ac97.c enables this compact mode and so has all the rights to use S16_LE format. No one has found such a configuration in I2S module until now in any Cirrus manuals. I2S module always feeds it's 32bit wide FIFO with 32bit samples consecutively for left and right channels. You cannot use 32-bit DMA transfers to transfer two 16-bit samples. So we can use two formats for AC97, but should remove all but S32_LE for I2S. Always using 32 bit chunks is not a problem for I2S, the codec I use uses less bits too (24), it's permitted by I2S standard. In proposed patch formats list shortened to just S32_LE, this makes all the DMA transactions right, while ALSA will do all sample format translation for us. 2. Incorrect setting of LRCLK (2 times slower) in original ep93xx-i2s.c masks the first problem. DMA takes two 16 bit samples instead of one, overall sound speed seems to be normal, but you get actually 4000 sampling rate instead of requested 8000 and therefore some noise... This is also the reason why the capture function not worked at all in this driver... If we take a look into I2S specification, we will figure that LRCLK MUST be equal to sample rate, if we are talking about stereo (in mono too, but it's not our case at all). In proposed patch SCLK and LRCLK rates are corrected, assuming we always send 32 bits * 2 channels to codec. Signed-off-by: Alexander Sverdlin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/ep93xx/ep93xx-i2s.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c index 9ac93f6b4f85..fff579a1c134 100644 --- a/sound/soc/ep93xx/ep93xx-i2s.c +++ b/sound/soc/ep93xx/ep93xx-i2s.c @@ -267,14 +267,16 @@ static int ep93xx_i2s_hw_params(struct snd_pcm_substream *substream, ep93xx_i2s_write_reg(info, EP93XX_I2S_RXWRDLEN, word_len); /* - * Calculate the sdiv (bit clock) and lrdiv (left/right clock) values. - * If the lrclk is pulse length is larger than the word size, then the - * bit clock will be gated for the unused bits. + * EP93xx I2S module can be setup so SCLK / LRCLK value can be + * 32, 64, 128. MCLK / SCLK value can be 2 and 4. + * We set LRCLK equal to `rate' and minimum SCLK / LRCLK + * value is 64, because our sample size is 32 bit * 2 channels. + * I2S standard permits us to transmit more bits than + * the codec uses. */ - div = (clk_get_rate(info->mclk) / params_rate(params)) * - params_channels(params); + div = clk_get_rate(info->mclk) / params_rate(params); for (sdiv = 2; sdiv <= 4; sdiv += 2) - for (lrdiv = 32; lrdiv <= 128; lrdiv <<= 1) + for (lrdiv = 64; lrdiv <= 128; lrdiv <<= 1) if (sdiv * lrdiv == div) { found = 1; goto out; @@ -341,9 +343,7 @@ static struct snd_soc_dai_ops ep93xx_i2s_dai_ops = { .set_fmt = ep93xx_i2s_set_dai_fmt, }; -#define EP93XX_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ - SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) +#define EP93XX_I2S_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) static struct snd_soc_dai_driver ep93xx_i2s_dai = { .symmetric_rates= 1, -- cgit v1.2.3 From c66ddf32dda0d5bcf9db7b4cc42ef5da7baadd3e Mon Sep 17 00:00:00 2001 From: Raymond Yau Date: Mon, 17 Jan 2011 11:19:03 +0100 Subject: ALSA: hda - Add add multi-streaming playback for AD1988 Attached a patch which add a new model to support multi-streaming playback for ad1988. playback another stereo stream through the front panel headphone on device 2 while playback through the speakers connected to rear panel on device 0 at the same time. Tested with ad1988a rev2 codec on asus P5B-V motherboard. Signed-off-by: Raymond Yau Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 182 ++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 178 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 46780670162b..34ee1169f2e0 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -46,6 +46,9 @@ struct ad198x_spec { unsigned int cur_eapd; unsigned int need_dac_fix; + hda_nid_t *alt_dac_nid; + struct hda_pcm_stream *stream_analog_alt_playback; + /* capture */ unsigned int num_adc_nids; hda_nid_t *adc_nids; @@ -156,6 +159,25 @@ static const char *ad_slave_sws[] = { NULL }; +static const char *ad1988_6stack_fp_slave_vols[] = { + "Front Playback Volume", + "Surround Playback Volume", + "Center Playback Volume", + "LFE Playback Volume", + "Side Playback Volume", + "IEC958 Playback Volume", + NULL +}; + +static const char *ad1988_6stack_fp_slave_sws[] = { + "Front Playback Switch", + "Surround Playback Switch", + "Center Playback Switch", + "LFE Playback Switch", + "Side Playback Switch", + "IEC958 Playback Switch", + NULL +}; static void ad198x_free_kctls(struct hda_codec *codec); #ifdef CONFIG_SND_HDA_INPUT_BEEP @@ -309,6 +331,38 @@ static int ad198x_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); } +static int ad198x_alt_playback_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct ad198x_spec *spec = codec->spec; + snd_hda_codec_setup_stream(codec, spec->alt_dac_nid[0], stream_tag, + 0, format); + return 0; +} + +static int ad198x_alt_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ad198x_spec *spec = codec->spec; + snd_hda_codec_cleanup_stream(codec, spec->alt_dac_nid[0]); + return 0; +} + +static struct hda_pcm_stream ad198x_pcm_analog_alt_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + /* NID is set in ad198x_build_pcms */ + .ops = { + .prepare = ad198x_alt_playback_pcm_prepare, + .cleanup = ad198x_alt_playback_pcm_cleanup + }, +}; + /* * Digital out */ @@ -446,6 +500,17 @@ static int ad198x_build_pcms(struct hda_codec *codec) } } + if (spec->alt_dac_nid && spec->stream_analog_alt_playback) { + codec->num_pcms++; + info = spec->pcm_rec + 2; + info->name = "AD198x Headphone"; + info->pcm_type = HDA_PCM_TYPE_AUDIO; + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = + *spec->stream_analog_alt_playback; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = + spec->alt_dac_nid[0]; + } + return 0; } @@ -2015,6 +2080,7 @@ static int patch_ad1981(struct hda_codec *codec) enum { AD1988_6STACK, AD1988_6STACK_DIG, + AD1988_6STACK_DIG_FP, AD1988_3STACK, AD1988_3STACK_DIG, AD1988_LAPTOP, @@ -2047,6 +2113,10 @@ static hda_nid_t ad1988_6stack_dac_nids_rev2[4] = { 0x04, 0x05, 0x0a, 0x06 }; +static hda_nid_t ad1988_alt_dac_nid[1] = { + 0x03 +}; + static hda_nid_t ad1988_3stack_dac_nids_rev2[3] = { 0x04, 0x0a, 0x06 }; @@ -2166,6 +2236,35 @@ static struct snd_kcontrol_new ad1988_6stack_mixers2[] = { { } /* end */ }; +static struct snd_kcontrol_new ad1988_6stack_fp_mixers[] = { + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), + + HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x2a, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x27, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x27, 2, 2, HDA_INPUT), + HDA_BIND_MUTE("Side Playback Switch", 0x28, 2, HDA_INPUT), + HDA_BIND_MUTE("Headphone Playback Switch", 0x22, 2, HDA_INPUT), + HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT), + + HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT), + + HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT), + + { } /* end */ +}; + /* 3-stack mode */ static struct snd_kcontrol_new ad1988_3stack_mixers1[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), @@ -2445,6 +2544,68 @@ static struct hda_verb ad1988_6stack_init_verbs[] = { { } }; +static struct hda_verb ad1988_6stack_fp_init_verbs[] = { + /* Front, Surround, CLFE, side DAC; unmute as default */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Headphone; unmute as default */ + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Port-A front headphon path */ + {0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Port-D line-out path */ + {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* Port-F surround path */ + {0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* Port-G CLFE path */ + {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* Port-H side path */ + {0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* Mono out path */ + {0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */ + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */ + /* Port-B front mic-in path */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + /* Port-C line-in path */ + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x33, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* Port-E mic-in path */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x34, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* Analog CD Input */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + /* Analog Mix output amp */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ + + { } +}; + static struct hda_verb ad1988_capture_init_verbs[] = { /* mute analog mix */ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, @@ -3074,13 +3235,13 @@ static int ad1988_auto_init(struct hda_codec *codec) return 0; } - /* */ static const char *ad1988_models[AD1988_MODEL_LAST] = { [AD1988_6STACK] = "6stack", [AD1988_6STACK_DIG] = "6stack-dig", + [AD1988_6STACK_DIG_FP] = "6stack-dig-fp", [AD1988_3STACK] = "3stack", [AD1988_3STACK_DIG] = "3stack-dig", [AD1988_LAPTOP] = "laptop", @@ -3140,6 +3301,7 @@ static int patch_ad1988(struct hda_codec *codec) switch (board_config) { case AD1988_6STACK: case AD1988_6STACK_DIG: + case AD1988_6STACK_DIG_FP: spec->multiout.max_channels = 8; spec->multiout.num_dacs = 4; if (is_rev2(codec)) @@ -3152,10 +3314,22 @@ static int patch_ad1988(struct hda_codec *codec) spec->mixers[0] = ad1988_6stack_mixers1_rev2; else spec->mixers[0] = ad1988_6stack_mixers1; - spec->mixers[1] = ad1988_6stack_mixers2; + if (board_config == AD1988_6STACK_DIG_FP) { + spec->mixers[1] = ad1988_6stack_fp_mixers; + spec->slave_vols = ad1988_6stack_fp_slave_vols; + spec->slave_sws = ad1988_6stack_fp_slave_sws; + spec->alt_dac_nid = ad1988_alt_dac_nid; + spec->stream_analog_alt_playback = + &ad198x_pcm_analog_alt_playback; + } else + 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) { + if (board_config == AD1988_6STACK_DIG_FP) + spec->init_verbs[0] = ad1988_6stack_fp_init_verbs; + else + spec->init_verbs[0] = ad1988_6stack_init_verbs; + if ((board_config == AD1988_6STACK_DIG) || + (board_config == AD1988_6STACK_DIG_FP)) { spec->multiout.dig_out_nid = AD1988_SPDIF_OUT; spec->dig_in_nid = AD1988_SPDIF_IN; } -- cgit v1.2.3 From ea73496324c1d990504e27f551e159388f891a4c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 17 Jan 2011 11:29:34 +0100 Subject: ALSA: hda - consitify string arrays Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 10 +++++----- sound/pci/hda/hda_generic.c | 7 ++++--- sound/pci/hda/hda_local.h | 6 +++--- sound/pci/hda/hda_proc.c | 2 +- sound/pci/hda/patch_analog.c | 30 ++++++++++++++++-------------- sound/pci/hda/patch_cirrus.c | 4 ++-- sound/pci/hda/patch_cmedia.c | 2 +- sound/pci/hda/patch_conexant.c | 14 +++++++------- sound/pci/hda/patch_realtek.c | 34 ++++++++++++++++++---------------- sound/pci/hda/patch_sigmatel.c | 36 ++++++++++++++++++------------------ sound/pci/hda/patch_via.c | 28 ++++++++++++++++++++-------- 11 files changed, 95 insertions(+), 78 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 05e5ec88c2d9..ae5c5d5e4b7c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2134,10 +2134,10 @@ int snd_hda_codec_reset(struct hda_codec *codec) * This function returns zero if successful or a negative error code. */ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, - unsigned int *tlv, const char **slaves) + unsigned int *tlv, const char * const *slaves) { struct snd_kcontrol *kctl; - const char **s; + const char * const *s; int err; for (s = slaves; *s && !snd_hda_find_mixer_ctl(codec, *s); s++) @@ -3689,7 +3689,7 @@ EXPORT_SYMBOL_HDA(snd_hda_build_pcms); * If no entries are matching, the function returns a negative value. */ int snd_hda_check_board_config(struct hda_codec *codec, - int num_configs, const char **models, + int num_configs, const char * const *models, const struct snd_pci_quirk *tbl) { if (codec->modelname && models) { @@ -3753,7 +3753,7 @@ EXPORT_SYMBOL_HDA(snd_hda_check_board_config); * If no entries are matching, the function returns a negative value. */ int snd_hda_check_board_codec_sid_config(struct hda_codec *codec, - int num_configs, const char **models, + int num_configs, const char * const *models, const struct snd_pci_quirk *tbl) { const struct snd_pci_quirk *q; @@ -4690,7 +4690,7 @@ const char *hda_get_input_pin_label(struct hda_codec *codec, hda_nid_t pin, int check_location) { unsigned int def_conf; - static const char *mic_names[] = { + static const char * const mic_names[] = { "Internal Mic", "Dock Mic", "Mic", "Front Mic", "Rear Mic", }; int attr; diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index fb0582f8d725..a63c54d9d767 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -762,7 +762,8 @@ static int check_existing_control(struct hda_codec *codec, const char *type, con /* * build output mixer controls */ -static int create_output_mixers(struct hda_codec *codec, const char **names) +static int create_output_mixers(struct hda_codec *codec, + const char * const *names) { struct hda_gspec *spec = codec->spec; int i, err; @@ -780,8 +781,8 @@ static int create_output_mixers(struct hda_codec *codec, const char **names) static int build_output_controls(struct hda_codec *codec) { struct hda_gspec *spec = codec->spec; - static const char *types_speaker[] = { "Speaker", "Headphone" }; - static const char *types_line[] = { "Front", "Headphone" }; + static const char * const types_speaker[] = { "Speaker", "Headphone" }; + static const char * const types_line[] = { "Front", "Headphone" }; switch (spec->pcm_vol_nodes) { case 1: diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 46bbefe2e4a9..3ab5e7a303db 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -140,7 +140,7 @@ void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir, struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, const char *name); int snd_hda_add_vmaster(struct hda_codec *codec, char *name, - unsigned int *tlv, const char **slaves); + unsigned int *tlv, const char * const *slaves); int snd_hda_codec_reset(struct hda_codec *codec); /* amp value bits */ @@ -341,10 +341,10 @@ void snd_print_pcm_bits(int pcm, char *buf, int buflen); * Misc */ int snd_hda_check_board_config(struct hda_codec *codec, int num_configs, - const char **modelnames, + const char * const *modelnames, const struct snd_pci_quirk *pci_list); int snd_hda_check_board_codec_sid_config(struct hda_codec *codec, - int num_configs, const char **models, + int num_configs, const char * const *models, const struct snd_pci_quirk *tbl); int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew); diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index f025200f2a62..bfe74c2fb079 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -418,7 +418,7 @@ static void print_digital_conv(struct snd_info_buffer *buffer, static const char *get_pwr_state(u32 state) { - static const char *buf[4] = { + static const char * const buf[4] = { "D0", "D1", "D2", "D3" }; if (state < 4) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 34ee1169f2e0..8dabab798689 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -84,8 +84,8 @@ struct ad198x_spec { #endif /* for virtual master */ hda_nid_t vmaster_nid; - const char **slave_vols; - const char **slave_sws; + const char * const *slave_vols; + const char * const *slave_sws; }; /* @@ -133,7 +133,7 @@ static int ad198x_init(struct hda_codec *codec) return 0; } -static const char *ad_slave_vols[] = { +static const char * const ad_slave_vols[] = { "Front Playback Volume", "Surround Playback Volume", "Center Playback Volume", @@ -146,7 +146,7 @@ static const char *ad_slave_vols[] = { NULL }; -static const char *ad_slave_sws[] = { +static const char * const ad_slave_sws[] = { "Front Playback Switch", "Surround Playback Switch", "Center Playback Switch", @@ -159,7 +159,7 @@ static const char *ad_slave_sws[] = { NULL }; -static const char *ad1988_6stack_fp_slave_vols[] = { +static const char * const ad1988_6stack_fp_slave_vols[] = { "Front Playback Volume", "Surround Playback Volume", "Center Playback Volume", @@ -169,7 +169,7 @@ static const char *ad1988_6stack_fp_slave_vols[] = { NULL }; -static const char *ad1988_6stack_fp_slave_sws[] = { +static const char * const ad1988_6stack_fp_slave_sws[] = { "Front Playback Switch", "Surround Playback Switch", "Center Playback Switch", @@ -1134,7 +1134,7 @@ enum { AD1986A_MODELS }; -static const char *ad1986a_models[AD1986A_MODELS] = { +static const char * const ad1986a_models[AD1986A_MODELS] = { [AD1986A_6STACK] = "6stack", [AD1986A_3STACK] = "3stack", [AD1986A_LAPTOP] = "laptop", @@ -1878,7 +1878,7 @@ enum { AD1981_MODELS }; -static const char *ad1981_models[AD1981_MODELS] = { +static const char * const ad1981_models[AD1981_MODELS] = { [AD1981_HP] = "hp", [AD1981_THINKPAD] = "thinkpad", [AD1981_BASIC] = "basic", @@ -2953,7 +2953,9 @@ 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" }; + static const char * const chname[4] = { + "Front", "Surround", NULL /*CLFE*/, "Side" + }; hda_nid_t nid; int i, err; @@ -3238,7 +3240,7 @@ static int ad1988_auto_init(struct hda_codec *codec) /* */ -static const char *ad1988_models[AD1988_MODEL_LAST] = { +static const char * const ad1988_models[AD1988_MODEL_LAST] = { [AD1988_6STACK] = "6stack", [AD1988_6STACK_DIG] = "6stack-dig", [AD1988_6STACK_DIG_FP] = "6stack-dig-fp", @@ -3573,7 +3575,7 @@ static struct hda_amp_list ad1884_loopbacks[] = { }; #endif -static const char *ad1884_slave_vols[] = { +static const char * const ad1884_slave_vols[] = { "PCM Playback Volume", "Mic Playback Volume", "Mono Playback Volume", @@ -3811,7 +3813,7 @@ enum { AD1984_MODELS }; -static const char *ad1984_models[AD1984_MODELS] = { +static const char * const ad1984_models[AD1984_MODELS] = { [AD1984_BASIC] = "basic", [AD1984_THINKPAD] = "thinkpad", [AD1984_DELL_DESKTOP] = "dell_desktop", @@ -4482,7 +4484,7 @@ enum { AD1884A_MODELS }; -static const char *ad1884a_models[AD1884A_MODELS] = { +static const char * const ad1884a_models[AD1884A_MODELS] = { [AD1884A_DESKTOP] = "desktop", [AD1884A_LAPTOP] = "laptop", [AD1884A_MOBILE] = "mobile", @@ -4870,7 +4872,7 @@ enum { AD1882_MODELS }; -static const char *ad1882_models[AD1986A_MODELS] = { +static const char * const ad1882_models[AD1986A_MODELS] = { [AD1882_3STACK] = "3stack", [AD1882_6STACK] = "6stack", }; diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 18af38ebf757..a07b031090d8 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -490,7 +490,7 @@ static int parse_digital_input(struct hda_codec *codec) * create mixer controls */ -static const char *dir_sfx[2] = { "Playback", "Capture" }; +static const char * const dir_sfx[2] = { "Playback", "Capture" }; static int add_mute(struct hda_codec *codec, const char *name, int index, unsigned int pval, int dir, struct snd_kcontrol **kctlp) @@ -1156,7 +1156,7 @@ static int cs_parse_auto_config(struct hda_codec *codec) return 0; } -static const char *cs420x_models[CS420X_MODELS] = { +static const char * const cs420x_models[CS420X_MODELS] = { [CS420X_MBP53] = "mbp53", [CS420X_MBP55] = "mbp55", [CS420X_IMAC27] = "imac27", diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index ff60908f4554..1f8bbcd0f802 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -608,7 +608,7 @@ static void cmi9880_free(struct hda_codec *codec) /* */ -static const char *cmi9880_models[CMI_MODELS] = { +static const char * const cmi9880_models[CMI_MODELS] = { [CMI_MINIMAL] = "minimal", [CMI_MIN_FP] = "min_fp", [CMI_FULL] = "full", diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e96581fcdbdb..9bb030a469cd 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -537,13 +537,13 @@ static struct snd_kcontrol_new cxt_beep_mixer[] = { }; #endif -static const char *slave_vols[] = { +static const char * const slave_vols[] = { "Headphone Playback Volume", "Speaker Playback Volume", NULL }; -static const char *slave_sws[] = { +static const char * const slave_sws[] = { "Headphone Playback Switch", "Speaker Playback Switch", NULL @@ -1134,7 +1134,7 @@ enum { CXT5045_MODELS }; -static const char *cxt5045_models[CXT5045_MODELS] = { +static const char * const cxt5045_models[CXT5045_MODELS] = { [CXT5045_LAPTOP_HPSENSE] = "laptop-hpsense", [CXT5045_LAPTOP_MICSENSE] = "laptop-micsense", [CXT5045_LAPTOP_HPMICSENSE] = "laptop-hpmicsense", @@ -1579,7 +1579,7 @@ enum { CXT5047_MODELS }; -static const char *cxt5047_models[CXT5047_MODELS] = { +static const char * const cxt5047_models[CXT5047_MODELS] = { [CXT5047_LAPTOP] = "laptop", [CXT5047_LAPTOP_HP] = "laptop-hp", [CXT5047_LAPTOP_EAPD] = "laptop-eapd", @@ -1995,7 +1995,7 @@ enum { CXT5051_MODELS }; -static const char *cxt5051_models[CXT5051_MODELS] = { +static const char *const cxt5051_models[CXT5051_MODELS] = { [CXT5051_LAPTOP] = "laptop", [CXT5051_HP] = "hp", [CXT5051_HP_DV6736] = "hp-dv6736", @@ -3084,7 +3084,7 @@ enum { CXT5066_MODELS }; -static const char *cxt5066_models[CXT5066_MODELS] = { +static const char * const cxt5066_models[CXT5066_MODELS] = { [CXT5066_LAPTOP] = "laptop", [CXT5066_DELL_LAPTOP] = "dell-laptop", [CXT5066_OLPC_XO_1_5] = "olpc-xo-1_5", @@ -3746,7 +3746,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec) struct conexant_spec *spec = codec->spec; int i, err; int num_line = 0, num_hp = 0, num_spk = 0; - static const char *texts[3] = { "Front", "Surround", "CLFE" }; + static const char * const texts[3] = { "Front", "Surround", "CLFE" }; if (spec->dac_info_filled == 1) return cx_auto_add_pb_volume(codec, spec->dac_info[0].dac, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 69554061c16e..4f006eedd7ef 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2916,7 +2916,7 @@ static struct snd_kcontrol_new alc880_uniwill_p53_mixer[] = { /* * slave controls for virtual master */ -static const char *alc_slave_vols[] = { +static const char * const alc_slave_vols[] = { "Front Playback Volume", "Surround Playback Volume", "Center Playback Volume", @@ -2930,7 +2930,7 @@ static const char *alc_slave_vols[] = { NULL, }; -static const char *alc_slave_sws[] = { +static const char * const alc_slave_sws[] = { "Front Playback Switch", "Surround Playback Switch", "Center Playback Switch", @@ -4611,7 +4611,7 @@ static struct hda_verb alc880_test_init_verbs[] = { /* */ -static const char *alc880_models[ALC880_MODEL_LAST] = { +static const char * const alc880_models[ALC880_MODEL_LAST] = { [ALC880_3ST] = "3stack", [ALC880_TCL_S700] = "tcl", [ALC880_3ST_DIG] = "3stack-digout", @@ -5144,7 +5144,7 @@ static const char *alc_get_line_out_pfx(const struct auto_pin_cfg *cfg, static int alc880_auto_create_multi_out_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - static const char *chname[4] = { + static const char * const chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; const char *pfx = alc_get_line_out_pfx(cfg, false); @@ -7158,7 +7158,7 @@ static struct snd_pci_quirk alc260_fixup_tbl[] = { /* * ALC260 configurations */ -static const char *alc260_models[ALC260_MODEL_LAST] = { +static const char * const alc260_models[ALC260_MODEL_LAST] = { [ALC260_BASIC] = "basic", [ALC260_HP] = "hp", [ALC260_HP_3013] = "hp-3013", @@ -9781,7 +9781,7 @@ static hda_nid_t alc1200_slave_dig_outs[] = { /* * configuration and preset */ -static const char *alc882_models[ALC882_MODEL_LAST] = { +static const char * const alc882_models[ALC882_MODEL_LAST] = { [ALC882_3ST_DIG] = "3stack-dig", [ALC882_6ST_DIG] = "6stack-dig", [ALC882_ARIMA] = "arima", @@ -12601,7 +12601,7 @@ static void alc262_auto_init(struct hda_codec *codec) /* * configuration and preset */ -static const char *alc262_models[ALC262_MODEL_LAST] = { +static const char * const alc262_models[ALC262_MODEL_LAST] = { [ALC262_BASIC] = "basic", [ALC262_HIPPO] = "hippo", [ALC262_HIPPO_1] = "hippo_1", @@ -13789,7 +13789,7 @@ static void alc268_auto_init(struct hda_codec *codec) /* * configuration and preset */ -static const char *alc268_models[ALC268_MODEL_LAST] = { +static const char * const alc268_models[ALC268_MODEL_LAST] = { [ALC267_QUANTA_IL1] = "quanta-il1", [ALC268_3ST] = "3stack", [ALC268_TOSHIBA] = "toshiba", @@ -14961,7 +14961,7 @@ static struct snd_pci_quirk alc269_fixup_tbl[] = { /* * configuration and preset */ -static const char *alc269_models[ALC269_MODEL_LAST] = { +static const char * const alc269_models[ALC269_MODEL_LAST] = { [ALC269_BASIC] = "basic", [ALC269_QUANTA_FL1] = "quanta", [ALC269_AMIC] = "laptop-amic", @@ -16004,7 +16004,7 @@ static int alc861_auto_create_multi_out_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { struct alc_spec *spec = codec->spec; - static const char *chname[4] = { + static const char * const chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; const char *pfx = alc_get_line_out_pfx(cfg, true); @@ -16210,7 +16210,7 @@ static struct hda_amp_list alc861_loopbacks[] = { /* * configuration and preset */ -static const char *alc861_models[ALC861_MODEL_LAST] = { +static const char * const alc861_models[ALC861_MODEL_LAST] = { [ALC861_3ST] = "3stack", [ALC660_3ST] = "3stack-660", [ALC861_3ST_DIG] = "3stack-dig", @@ -16913,7 +16913,7 @@ static void alc861vd_dallas_setup(struct hda_codec *codec) /* * configuration and preset */ -static const char *alc861vd_models[ALC861VD_MODEL_LAST] = { +static const char * const alc861vd_models[ALC861VD_MODEL_LAST] = { [ALC660VD_3ST] = "3stack-660", [ALC660VD_3ST_DIG] = "3stack-660-digout", [ALC660VD_ASUS_V1S] = "asus-v1s", @@ -17133,7 +17133,9 @@ static void alc861vd_auto_init_analog_input(struct hda_codec *codec) static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - static const char *chname[4] = {"Front", "Surround", "CLFE", "Side"}; + static const char * const chname[4] = { + "Front", "Surround", "CLFE", "Side" + }; const char *pfx = alc_get_line_out_pfx(cfg, true); hda_nid_t nid_v, nid_s; int i, err; @@ -18688,7 +18690,7 @@ static struct snd_kcontrol_new alc272_nc10_mixer[] = { /* * configuration and preset */ -static const char *alc662_models[ALC662_MODEL_LAST] = { +static const char * const alc662_models[ALC662_MODEL_LAST] = { [ALC662_3ST_2ch_DIG] = "3stack-dig", [ALC662_3ST_6ch_DIG] = "3stack-6ch-dig", [ALC662_3ST_6ch] = "3stack-6ch", @@ -19203,7 +19205,7 @@ static int alc662_auto_create_multi_out_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { struct alc_spec *spec = codec->spec; - static const char *chname[4] = { + static const char * const chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; const char *pfx = alc_get_line_out_pfx(cfg, true); @@ -19978,7 +19980,7 @@ static void alc680_auto_init(struct hda_codec *codec) /* * configuration and preset */ -static const char *alc680_models[ALC680_MODEL_LAST] = { +static const char * const alc680_models[ALC680_MODEL_LAST] = { [ALC680_BASE] = "base", [ALC680_AUTO] = "auto", }; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 4ab019d0924e..056f52df68cd 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -266,7 +266,7 @@ struct sigmatel_spec { struct sigmatel_mic_route int_mic; struct sigmatel_mic_route dock_mic; - const char **spdif_labels; + const char * const *spdif_labels; hda_nid_t dig_in_nid; hda_nid_t mono_nid; @@ -524,7 +524,7 @@ static unsigned long stac927x_capsws[] = { HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT), }; -static const char *stac927x_spdif_labels[5] = { +static const char * const stac927x_spdif_labels[5] = { "Digital Playback", "ADAT", "Analog Mux 1", "Analog Mux 2", "Analog Mux 3" }; @@ -1062,7 +1062,7 @@ static struct snd_kcontrol_new stac_smux_mixer = { .put = stac92xx_smux_enum_put, }; -static const char *slave_vols[] = { +static const char * const slave_vols[] = { "Front Playback Volume", "Surround Playback Volume", "Center Playback Volume", @@ -1073,7 +1073,7 @@ static const char *slave_vols[] = { NULL }; -static const char *slave_sws[] = { +static const char * const slave_sws[] = { "Front Playback Switch", "Surround Playback Switch", "Center Playback Switch", @@ -1354,7 +1354,7 @@ static unsigned int *stac9200_brd_tbl[STAC_9200_MODELS] = { [STAC_9200_PANASONIC] = ref9200_pin_configs, }; -static const char *stac9200_models[STAC_9200_MODELS] = { +static const char * const stac9200_models[STAC_9200_MODELS] = { [STAC_AUTO] = "auto", [STAC_REF] = "ref", [STAC_9200_OQO] = "oqo", @@ -1500,7 +1500,7 @@ static unsigned int *stac925x_brd_tbl[STAC_925x_MODELS] = { [STAC_M6] = stac925xM6_pin_configs, }; -static const char *stac925x_models[STAC_925x_MODELS] = { +static const char * const stac925x_models[STAC_925x_MODELS] = { [STAC_925x_AUTO] = "auto", [STAC_REF] = "ref", [STAC_M1] = "m1", @@ -1574,7 +1574,7 @@ static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = { [STAC_92HD73XX_INTEL] = intel_dg45id_pin_configs, }; -static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = { +static const char * const stac92hd73xx_models[STAC_92HD73XX_MODELS] = { [STAC_92HD73XX_AUTO] = "auto", [STAC_92HD73XX_NO_JD] = "no-jd", [STAC_92HD73XX_REF] = "ref", @@ -1660,7 +1660,7 @@ static unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = { [STAC_HP_DV7_4000] = hp_dv7_4000_pin_configs, }; -static const char *stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { +static const char * const stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_AUTO] = "auto", [STAC_92HD83XXX_REF] = "ref", [STAC_92HD83XXX_PWR_REF] = "mic-ref", @@ -1722,7 +1722,7 @@ static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { [STAC_HP_DV4_1222NR] = NULL, }; -static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { +static const char * const stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { [STAC_92HD71BXX_AUTO] = "auto", [STAC_92HD71BXX_REF] = "ref", [STAC_DELL_M4_1] = "dell-m4-1", @@ -1915,7 +1915,7 @@ static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = { [STAC_922X_DELL_M82] = dell_922x_m82_pin_configs, }; -static const char *stac922x_models[STAC_922X_MODELS] = { +static const char * const stac922x_models[STAC_922X_MODELS] = { [STAC_922X_AUTO] = "auto", [STAC_D945_REF] = "ref", [STAC_D945GTP5] = "5stack", @@ -2077,7 +2077,7 @@ static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = { [STAC_927X_VOLKNOB] = NULL, }; -static const char *stac927x_models[STAC_927X_MODELS] = { +static const char * const stac927x_models[STAC_927X_MODELS] = { [STAC_927X_AUTO] = "auto", [STAC_D965_REF_NO_JD] = "ref-no-jd", [STAC_D965_REF] = "ref", @@ -2180,7 +2180,7 @@ static unsigned int *stac9205_brd_tbl[STAC_9205_MODELS] = { [STAC_9205_EAPD] = NULL, }; -static const char *stac9205_models[STAC_9205_MODELS] = { +static const char * const stac9205_models[STAC_9205_MODELS] = { [STAC_9205_AUTO] = "auto", [STAC_9205_REF] = "ref", [STAC_9205_DELL_M42] = "dell-m42", @@ -3123,7 +3123,7 @@ static int create_multi_out_ctls(struct hda_codec *codec, int num_outs, int type) { struct sigmatel_spec *spec = codec->spec; - static const char *chname[4] = { + static const char * const chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; hda_nid_t nid; @@ -3256,7 +3256,7 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, } /* labels for mono mux outputs */ -static const char *stac92xx_mono_labels[4] = { +static const char * const stac92xx_mono_labels[4] = { "DAC0", "DAC1", "Mixer", "DAC2" }; @@ -3380,7 +3380,7 @@ static int stac92xx_auto_create_mux_input_ctls(struct hda_codec *codec) return 0; }; -static const char *stac92xx_spdif_labels[3] = { +static const char * const stac92xx_spdif_labels[3] = { "Digital Playback", "Analog Mux 1", "Analog Mux 2", }; @@ -3388,7 +3388,7 @@ static int stac92xx_auto_create_spdif_mux_ctls(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; struct hda_input_mux *spdif_mux = &spec->private_smux; - const char **labels = spec->spdif_labels; + const char * const *labels = spec->spdif_labels; int i, num_cons; hda_nid_t con_lst[HDA_MAX_NUM_INPUTS]; @@ -3409,7 +3409,7 @@ static int stac92xx_auto_create_spdif_mux_ctls(struct hda_codec *codec) } /* labels for dmic mux inputs */ -static const char *stac92xx_dmic_labels[5] = { +static const char * const stac92xx_dmic_labels[5] = { "Analog Inputs", "Digital Mic 1", "Digital Mic 2", "Digital Mic 3", "Digital Mic 4" }; @@ -6270,7 +6270,7 @@ static unsigned int stac9872_vaio_pin_configs[9] = { 0x90a7013e }; -static const char *stac9872_models[STAC_9872_MODELS] = { +static const char * const stac9872_models[STAC_9872_MODELS] = { [STAC_9872_AUTO] = "auto", [STAC_9872_VAIO] = "vaio", }; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index d1c3f8defc48..71f78456d682 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -2282,7 +2282,9 @@ static int vt1708_auto_create_multi_out_ctls(struct via_spec *spec, const struct auto_pin_cfg *cfg) { char name[32]; - static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" }; + static const char * const chname[4] = { + "Front", "Surround", "C/LFE", "Side" + }; hda_nid_t nid, nid_vol, nid_vols[] = {0x17, 0x19, 0x1a, 0x1b}; int i, err; @@ -2371,7 +2373,7 @@ static void create_hp_imux(struct via_spec *spec) { int i; struct hda_input_mux *imux = &spec->private_imux[1]; - static const char *texts[] = { "OFF", "ON", NULL}; + static const char * const texts[] = { "OFF", "ON", NULL}; /* for hp mode select */ for (i = 0; texts[i]; i++) @@ -2891,7 +2893,9 @@ static int vt1709_auto_create_multi_out_ctls(struct via_spec *spec, const struct auto_pin_cfg *cfg) { char name[32]; - static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" }; + static const char * const chname[4] = { + "Front", "Surround", "C/LFE", "Side" + }; hda_nid_t nid, nid_vol, nid_vols[] = {0x18, 0x1a, 0x1b, 0x29}; int i, err; @@ -3434,7 +3438,9 @@ static int vt1708B_auto_create_multi_out_ctls(struct via_spec *spec, const struct auto_pin_cfg *cfg) { char name[32]; - static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" }; + static const char * const chname[4] = { + "Front", "Surround", "C/LFE", "Side" + }; hda_nid_t nid_vols[] = {0x16, 0x18, 0x26, 0x27}; hda_nid_t nid, nid_vol = 0; int i, err; @@ -3862,7 +3868,9 @@ static int vt1708S_auto_create_multi_out_ctls(struct via_spec *spec, const struct auto_pin_cfg *cfg) { char name[32]; - static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" }; + static const char * const chname[4] = { + "Front", "Surround", "C/LFE", "Side" + }; hda_nid_t nid_vols[] = {0x10, 0x11, 0x24, 0x25}; hda_nid_t nid_mutes[] = {0x1C, 0x18, 0x26, 0x27}; hda_nid_t nid, nid_vol, nid_mute; @@ -4305,7 +4313,7 @@ static int vt1702_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) { int err, i; struct hda_input_mux *imux; - static const char *texts[] = { "ON", "OFF", NULL}; + static const char * const texts[] = { "ON", "OFF", NULL}; if (!pin) return 0; spec->multiout.hp_nid = 0x1D; @@ -4616,7 +4624,9 @@ static int vt1718S_auto_create_multi_out_ctls(struct via_spec *spec, const struct auto_pin_cfg *cfg) { char name[32]; - static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" }; + static const char * const chname[4] = { + "Front", "Surround", "C/LFE", "Side" + }; hda_nid_t nid_vols[] = {0x8, 0x9, 0xa, 0xb}; hda_nid_t nid_mutes[] = {0x24, 0x25, 0x26, 0x27}; hda_nid_t nid, nid_vol, nid_mute = 0; @@ -5065,7 +5075,9 @@ static int vt1716S_auto_create_multi_out_ctls(struct via_spec *spec, const struct auto_pin_cfg *cfg) { char name[32]; - static const char *chname[3] = { "Front", "Surround", "C/LFE" }; + static const char * const chname[3] = { + "Front", "Surround", "C/LFE" + }; hda_nid_t nid_vols[] = {0x10, 0x11, 0x25}; hda_nid_t nid_mutes[] = {0x1C, 0x18, 0x27}; hda_nid_t nid, nid_vol, nid_mute; -- cgit v1.2.3 From 0f0714c5ed0a98fdeaa2287d3b159989bbe6d842 Mon Sep 17 00:00:00 2001 From: Bankim Bhavsar Date: Mon, 17 Jan 2011 15:23:21 +0100 Subject: ALSA: hda - Add support for VMware controller Add the new PCI ID 0x15ad and device ID 0x1977 for VMware HDAudio Controller. [changed to use AZX_DRIVER_GENERIC by tiwai] Signed-off-by: Bankim Bhavsar Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a1c4008af891..07c522fd2b10 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2806,6 +2806,8 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { #endif /* Vortex86MX */ { PCI_DEVICE(0x17f3, 0x3010), .driver_data = AZX_DRIVER_GENERIC }, + /* VMware HDAudio */ + { PCI_DEVICE(0x15ad, 0x1977), .driver_data = AZX_DRIVER_GENERIC }, /* AMD/ATI Generic, PCI class code and Vendor ID for HD Audio */ { PCI_DEVICE(PCI_VENDOR_ID_ATI, PCI_ANY_ID), .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, -- cgit v1.2.3 From cbbf50b22f9693218f9f0d460432266b04fc960d Mon Sep 17 00:00:00 2001 From: Vitaliy Kulikov Date: Fri, 14 Jan 2011 17:21:13 -0600 Subject: ALSA: hda - Fix initialization for HP 2011 notebooks Fixes for HP 2011 notebooks: enable dock ports and disable BTL initialization in the driver. Signed-off-by: Vitaliy Kulikov Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 53 ++++++++---------------------------------- 1 file changed, 10 insertions(+), 43 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 056f52df68cd..9ea48b425d0b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5333,7 +5333,7 @@ again: return 0; } -static int stac92hd83xxx_set_system_btl_amp(struct hda_codec *codec) +static int hp_bnb2011_with_dock(struct hda_codec *codec) { if (codec->vendor_id != 0x111d7605 && codec->vendor_id != 0x111d76d1) @@ -5348,10 +5348,6 @@ static int stac92hd83xxx_set_system_btl_amp(struct hda_codec *codec) case 0x103c161d: case 0x103c161e: case 0x103c161f: - case 0x103c1620: - case 0x103c1621: - case 0x103c1622: - case 0x103c1623: case 0x103c162a: case 0x103c162b: @@ -5360,41 +5356,9 @@ static int stac92hd83xxx_set_system_btl_amp(struct hda_codec *codec) case 0x103c1631: case 0x103c1633: - + case 0x103c1634: case 0x103c1635: - case 0x103c164f: - - case 0x103c1676: - case 0x103c1677: - case 0x103c1678: - case 0x103c1679: - case 0x103c167a: - case 0x103c167b: - case 0x103c167c: - case 0x103c167d: - case 0x103c167e: - case 0x103c167f: - case 0x103c1680: - case 0x103c1681: - case 0x103c1682: - case 0x103c1683: - case 0x103c1684: - case 0x103c1685: - case 0x103c1686: - case 0x103c1687: - case 0x103c1688: - case 0x103c1689: - case 0x103c168a: - case 0x103c168b: - case 0x103c168c: - case 0x103c168d: - case 0x103c168e: - case 0x103c168f: - case 0x103c1690: - case 0x103c1691: - case 0x103c1692: - case 0x103c3587: case 0x103c3588: case 0x103c3589: @@ -5402,9 +5366,9 @@ static int stac92hd83xxx_set_system_btl_amp(struct hda_codec *codec) case 0x103c3667: case 0x103c3668: - /* set BTL amp level to 13.43dB for louder speaker output */ - return snd_hda_codec_write_cache(codec, codec->afg, 0, - 0x7F4, 0x14); + case 0x103c3669: + + return 1; } return 0; } @@ -5420,6 +5384,11 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; + if (hp_bnb2011_with_dock(codec)) { + snd_hda_codec_set_pincfg(codec, 0xa, 0x2101201f); + snd_hda_codec_set_pincfg(codec, 0xf, 0x2181205e); + } + /* reset pin power-down; Windows may leave these bits after reboot */ snd_hda_codec_write_cache(codec, codec->afg, 0, 0x7EC, 0); snd_hda_codec_write_cache(codec, codec->afg, 0, 0x7ED, 0); @@ -5546,8 +5515,6 @@ again: AC_VERB_SET_CONNECT_SEL, num_dacs); } - stac92hd83xxx_set_system_btl_amp(codec); - codec->proc_widget_hook = stac92hd_proc_hook; return 0; -- cgit v1.2.3