X-Git-Url: https://err.no/cgi-bin/gitweb.cgi?a=blobdiff_plain;f=sound%2Fpci%2Fhda%2Fpatch_realtek.c;h=cf6c100940dcebc2c5b8db6df1b898769ff238ef;hb=5e1b1518a53fc62d9f39a13819c849336c6d8dd4;hp=b76755264730d62733b95dcd47242493997c2e86;hpb=934a3595b30c986bab52bc9c08d12c8962c88c8a;p=linux-2.6 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b767552647..cf6c100940 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6,6 +6,7 @@ * Copyright (c) 2004 Kailang Yang * PeiSen Hou * Takashi Iwai + * Jonathan Woithe * * This driver is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by @@ -50,6 +51,8 @@ enum { ALC880_UNIWILL_DIG, ALC880_CLEVO, ALC880_TCL_S700, + ALC880_LG, + ALC880_LG_LW, #ifdef CONFIG_SND_DEBUG ALC880_TEST, #endif @@ -63,6 +66,10 @@ enum { ALC260_HP, ALC260_HP_3013, ALC260_FUJITSU_S702X, + ALC260_ACER, +#ifdef CONFIG_SND_DEBUG + ALC260_TEST, +#endif ALC260_AUTO, ALC260_MODEL_LAST /* last tag */ }; @@ -70,6 +77,7 @@ enum { /* ALC262 models */ enum { ALC262_BASIC, + ALC262_FUJITSU, ALC262_AUTO, ALC262_MODEL_LAST /* last tag */ }; @@ -124,6 +132,7 @@ struct alc_spec { hda_nid_t dig_in_nid; /* digital-in NID; optional */ /* capture source */ + unsigned int num_mux_defs; const struct hda_input_mux *input_mux; unsigned int cur_mux[3]; @@ -132,7 +141,7 @@ struct alc_spec { int num_channel_mode; /* PCM information */ - struct hda_pcm pcm_rec[2]; /* used in alc_build_pcms() */ + struct hda_pcm pcm_rec[3]; /* used in alc_build_pcms() */ /* dynamic controls, init_verbs and input_mux */ struct auto_pin_cfg autocfg; @@ -140,6 +149,14 @@ struct alc_spec { struct snd_kcontrol_new *kctl_alloc; struct hda_input_mux private_imux; hda_nid_t private_dac_nids[5]; + + /* hooks */ + void (*init_hook)(struct hda_codec *codec); + void (*unsol_event)(struct hda_codec *codec, unsigned int res); + + /* for pin sensing */ + unsigned int sense_updated: 1; + unsigned int jack_present: 1; }; /* @@ -157,7 +174,10 @@ struct alc_config_preset { hda_nid_t dig_in_nid; unsigned int num_channel_mode; const struct hda_channel_mode *channel_mode; + unsigned int num_mux_defs; const struct hda_input_mux *input_mux; + void (*unsol_event)(struct hda_codec *, unsigned int); + void (*init_hook)(struct hda_codec *); }; @@ -168,7 +188,10 @@ static int alc_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; - return snd_hda_input_mux_info(spec->input_mux, uinfo); + unsigned int mux_idx = snd_ctl_get_ioffidx(kcontrol, &uinfo->id); + if (mux_idx >= spec->num_mux_defs) + mux_idx = 0; + return snd_hda_input_mux_info(&spec->input_mux[mux_idx], uinfo); } static int alc_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -186,7 +209,8 @@ static int alc_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, + unsigned int mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx; + return snd_hda_input_mux_put(codec, &spec->input_mux[mux_idx], ucontrol, spec->adc_nids[adc_idx], &spec->cur_mux[adc_idx]); } @@ -218,56 +242,241 @@ static int alc_ch_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_va spec->num_channel_mode, &spec->multiout.max_channels); } - /* - * Control of pin widget settings via the mixer. Only boolean settings are - * supported, so VrefEn can't be controlled using these functions as they - * stand. + * Control the mode of pin widget settings via the mixer. "pc" is used + * instead of "%" to avoid consequences of accidently treating the % as + * being part of a format specifier. Maximum allowed length of a value is + * 63 characters plus NULL terminator. + * + * Note: some retasking pin complexes seem to ignore requests for input + * states other than HiZ (eg: PIN_VREFxx) and revert to HiZ if any of these + * are requested. Therefore order this list so that this behaviour will not + * cause problems when mixer clients move through the enum sequentially. + * NIDs 0x0f and 0x10 have been observed to have this behaviour as of + * March 2006. + */ +static char *alc_pin_mode_names[] = { + "Mic 50pc bias", "Mic 80pc bias", + "Line in", "Line out", "Headphone out", +}; +static unsigned char alc_pin_mode_values[] = { + PIN_VREF50, PIN_VREF80, PIN_IN, PIN_OUT, PIN_HP, +}; +/* The control can present all 5 options, or it can limit the options based + * in the pin being assumed to be exclusively an input or an output pin. In + * addition, "input" pins may or may not process the mic bias option + * depending on actual widget capability (NIDs 0x0f and 0x10 don't seem to + * accept requests for bias as of chip versions up to March 2006) and/or + * wiring in the computer. */ -static int alc_pinctl_switch_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +#define ALC_PIN_DIR_IN 0x00 +#define ALC_PIN_DIR_OUT 0x01 +#define ALC_PIN_DIR_INOUT 0x02 +#define ALC_PIN_DIR_IN_NOMICBIAS 0x03 +#define ALC_PIN_DIR_INOUT_NOMICBIAS 0x04 + +/* Info about the pin modes supported by the different pin direction modes. + * For each direction the minimum and maximum values are given. + */ +static signed char alc_pin_mode_dir_info[5][2] = { + { 0, 2 }, /* ALC_PIN_DIR_IN */ + { 3, 4 }, /* ALC_PIN_DIR_OUT */ + { 0, 4 }, /* ALC_PIN_DIR_INOUT */ + { 2, 2 }, /* ALC_PIN_DIR_IN_NOMICBIAS */ + { 2, 4 }, /* ALC_PIN_DIR_INOUT_NOMICBIAS */ +}; +#define alc_pin_mode_min(_dir) (alc_pin_mode_dir_info[_dir][0]) +#define alc_pin_mode_max(_dir) (alc_pin_mode_dir_info[_dir][1]) +#define alc_pin_mode_n_items(_dir) \ + (alc_pin_mode_max(_dir)-alc_pin_mode_min(_dir)+1) + +static int alc_pin_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + unsigned int item_num = uinfo->value.enumerated.item; + unsigned char dir = (kcontrol->private_value >> 16) & 0xff; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; + uinfo->value.enumerated.items = alc_pin_mode_n_items(dir); + + if (item_numalc_pin_mode_max(dir)) + item_num = alc_pin_mode_min(dir); + strcpy(uinfo->value.enumerated.name, alc_pin_mode_names[item_num]); return 0; } -static int alc_pinctl_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int alc_pin_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + unsigned int i; struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = kcontrol->private_value & 0xffff; - long mask = (kcontrol->private_value >> 16) & 0xff; + unsigned char dir = (kcontrol->private_value >> 16) & 0xff; long *valp = ucontrol->value.integer.value; + unsigned int pinctl = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_PIN_WIDGET_CONTROL,0x00); - *valp = 0; - if (snd_hda_codec_read(codec,nid,0,AC_VERB_GET_PIN_WIDGET_CONTROL,0x00) & mask) - *valp = 1; + /* Find enumerated value for current pinctl setting */ + i = alc_pin_mode_min(dir); + while (alc_pin_mode_values[i]!=pinctl && i<=alc_pin_mode_max(dir)) + i++; + *valp = i<=alc_pin_mode_max(dir)?i:alc_pin_mode_min(dir); return 0; } -static int alc_pinctl_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int alc_pin_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + signed int change; struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = kcontrol->private_value & 0xffff; - long mask = (kcontrol->private_value >> 16) & 0xff; - long *valp = ucontrol->value.integer.value; + unsigned char dir = (kcontrol->private_value >> 16) & 0xff; + long val = *ucontrol->value.integer.value; unsigned int pinctl = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_PIN_WIDGET_CONTROL,0x00); - int change = ((pinctl & mask)!=0) != *valp; - if (change) + if (valalc_pin_mode_max(dir)) + val = alc_pin_mode_min(dir); + + change = pinctl != alc_pin_mode_values[val]; + if (change) { + /* Set pin mode to that requested */ snd_hda_codec_write(codec,nid,0,AC_VERB_SET_PIN_WIDGET_CONTROL, - *valp?(pinctl|mask):(pinctl&~mask)); + alc_pin_mode_values[val]); + + /* Also enable the retasking pin's input/output as required + * for the requested pin mode. Enum values of 2 or less are + * input modes. + * + * Dynamically switching the input/output buffers probably + * reduces noise slightly (particularly on input) so we'll + * do it. However, having both input and output buffers + * enabled simultaneously doesn't seem to be problematic if + * this turns out to be necessary in the future. + */ + if (val <= 2) { + snd_hda_codec_write(codec,nid,0,AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_MUTE); + snd_hda_codec_write(codec,nid,0,AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_UNMUTE(0)); + } else { + snd_hda_codec_write(codec,nid,0,AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_MUTE(0)); + snd_hda_codec_write(codec,nid,0,AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_UNMUTE); + } + } return change; } -#define ALC_PINCTL_SWITCH(xname, nid, mask) \ +#define ALC_PIN_MODE(xname, nid, dir) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ - .info = alc_pinctl_switch_info, \ - .get = alc_pinctl_switch_get, \ - .put = alc_pinctl_switch_put, \ - .private_value = (nid) | (mask<<16) } + .info = alc_pin_mode_info, \ + .get = alc_pin_mode_get, \ + .put = alc_pin_mode_put, \ + .private_value = nid | (dir<<16) } + +/* A switch control for ALC260 GPIO pins. Multiple GPIOs can be ganged + * together using a mask with more than one bit set. This control is + * currently used only by the ALC260 test model. At this stage they are not + * needed for any "production" models. + */ +#ifdef CONFIG_SND_DEBUG +static int alc_gpio_data_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} +static int alc_gpio_data_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value & 0xffff; + unsigned char mask = (kcontrol->private_value >> 16) & 0xff; + long *valp = ucontrol->value.integer.value; + unsigned int val = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_GPIO_DATA,0x00); + + *valp = (val & mask) != 0; + return 0; +} +static int alc_gpio_data_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + signed int change; + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value & 0xffff; + unsigned char mask = (kcontrol->private_value >> 16) & 0xff; + long val = *ucontrol->value.integer.value; + unsigned int gpio_data = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_GPIO_DATA,0x00); + + /* Set/unset the masked GPIO bit(s) as needed */ + change = (val==0?0:mask) != (gpio_data & mask); + if (val==0) + gpio_data &= ~mask; + else + gpio_data |= mask; + snd_hda_codec_write(codec,nid,0,AC_VERB_SET_GPIO_DATA,gpio_data); + + return change; +} +#define ALC_GPIO_DATA_SWITCH(xname, nid, mask) \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ + .info = alc_gpio_data_info, \ + .get = alc_gpio_data_get, \ + .put = alc_gpio_data_put, \ + .private_value = nid | (mask<<16) } +#endif /* CONFIG_SND_DEBUG */ + +/* A switch control to allow the enabling of the digital IO pins on the + * ALC260. This is incredibly simplistic; the intention of this control is + * to provide something in the test model allowing digital outputs to be + * identified if present. If models are found which can utilise these + * outputs a more complete mixer control can be devised for those models if + * necessary. + */ +#ifdef CONFIG_SND_DEBUG +static int alc_spdif_ctrl_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} +static int alc_spdif_ctrl_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value & 0xffff; + unsigned char mask = (kcontrol->private_value >> 16) & 0xff; + long *valp = ucontrol->value.integer.value; + unsigned int val = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_DIGI_CONVERT,0x00); + + *valp = (val & mask) != 0; + return 0; +} +static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + signed int change; + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value & 0xffff; + unsigned char mask = (kcontrol->private_value >> 16) & 0xff; + long val = *ucontrol->value.integer.value; + unsigned int ctrl_data = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_DIGI_CONVERT,0x00); + + /* Set/unset the masked control bit(s) as needed */ + change = (val==0?0:mask) != (ctrl_data & mask); + if (val==0) + ctrl_data &= ~mask; + else + ctrl_data |= mask; + snd_hda_codec_write(codec,nid,0,AC_VERB_SET_DIGI_CONVERT_1,ctrl_data); + return change; +} +#define ALC_SPDIF_CTRL_SWITCH(xname, nid, mask) \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ + .info = alc_spdif_ctrl_info, \ + .get = alc_spdif_ctrl_get, \ + .put = alc_spdif_ctrl_put, \ + .private_value = nid | (mask<<16) } +#endif /* CONFIG_SND_DEBUG */ /* * set up from the preset table @@ -291,11 +500,17 @@ static void setup_preset(struct alc_spec *spec, const struct alc_config_preset * spec->multiout.dig_out_nid = preset->dig_out_nid; spec->multiout.hp_nid = preset->hp_nid; + spec->num_mux_defs = preset->num_mux_defs; + if (! spec->num_mux_defs) + spec->num_mux_defs = 1; spec->input_mux = preset->input_mux; spec->num_adc_nids = preset->num_adc_nids; spec->adc_nids = preset->adc_nids; spec->dig_in_nid = preset->dig_in_nid; + + spec->unsol_event = preset->unsol_event; + spec->init_hook = preset->init_hook; } /* @@ -1098,6 +1313,217 @@ static struct hda_verb alc880_pin_tcl_S700_init_verbs[] = { }; /* + * LG m1 express dual + * + * Pin assignment: + * Rear Line-In/Out (blue): 0x14 + * Build-in Mic-In: 0x15 + * Speaker-out: 0x17 + * HP-Out (green): 0x1b + * Mic-In/Out (red): 0x19 + * SPDIF-Out: 0x1e + */ + +/* To make 5.1 output working (green=Front, blue=Surr, red=CLFE) */ +static hda_nid_t alc880_lg_dac_nids[3] = { + 0x05, 0x02, 0x03 +}; + +/* seems analog CD is not working */ +static struct hda_input_mux alc880_lg_capture_source = { + .num_items = 3, + .items = { + { "Mic", 0x1 }, + { "Line", 0x5 }, + { "Internal Mic", 0x6 }, + }, +}; + +/* 2,4,6 channel modes */ +static struct hda_verb alc880_lg_ch2_init[] = { + /* set line-in and mic-in to input */ + { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { } +}; + +static struct hda_verb alc880_lg_ch4_init[] = { + /* set line-in to out and mic-in to input */ + { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { } +}; + +static struct hda_verb alc880_lg_ch6_init[] = { + /* set line-in and mic-in to output */ + { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, + { } +}; + +static struct hda_channel_mode alc880_lg_ch_modes[3] = { + { 2, alc880_lg_ch2_init }, + { 4, alc880_lg_ch4_init }, + { 6, alc880_lg_ch6_init }, +}; + +static struct snd_kcontrol_new alc880_lg_mixer[] = { + /* FIXME: it's not really "master" but front channels */ + HDA_CODEC_VOLUME("Master Playback Volume", 0x0f, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Master Playback Switch", 0x0f, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x06, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x06, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x07, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x07, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Channel Mode", + .info = alc_ch_mode_info, + .get = alc_ch_mode_get, + .put = alc_ch_mode_put, + }, + { } /* end */ +}; + +static struct hda_verb alc880_lg_init_verbs[] = { + /* set capture source to mic-in */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* mute all amp mixer inputs */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(6)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(7)}, + /* line-in to input */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* built-in mic */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* speaker-out */ + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* mic-in to input */ + {0x11, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* HP-out */ + {0x13, AC_VERB_SET_CONNECT_SEL, 0x03}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* jack sense */ + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | 0x1}, + { } +}; + +/* toggle speaker-output according to the hp-jack state */ +static void alc880_lg_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_update(codec, 0x17, 0, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); + snd_hda_codec_amp_update(codec, 0x17, 1, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); +} + +static void alc880_lg_unsol_event(struct hda_codec *codec, unsigned int res) +{ + /* Looks like the unsol event is incompatible with the standard + * definition. 4bit tag is placed at 28 bit! + */ + if ((res >> 28) == 0x01) + alc880_lg_automute(codec); +} + +/* + * LG LW20 + * + * Pin assignment: + * Speaker-out: 0x14 + * Mic-In: 0x18 + * Built-in Mic-In: 0x19 (?) + * HP-Out: 0x1b + * SPDIF-Out: 0x1e + */ + +/* seems analog CD is not working */ +static struct hda_input_mux alc880_lg_lw_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x0 }, + { "Internal Mic", 0x1 }, + }, +}; + +static struct snd_kcontrol_new alc880_lg_lw_mixer[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), + { } /* end */ +}; + +static struct hda_verb alc880_lg_lw_init_verbs[] = { + /* set capture source to mic-in */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(7)}, + /* speaker-out */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* HP-out */ + {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* mic-in to input */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* built-in mic */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* jack sense */ + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | 0x1}, + { } +}; + +/* toggle speaker-output according to the hp-jack state */ +static void alc880_lg_lw_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); + snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); +} + +static void alc880_lg_lw_unsol_event(struct hda_codec *codec, unsigned int res) +{ + /* Looks like the unsol event is incompatible with the standard + * definition. 4bit tag is placed at 28 bit! + */ + if ((res >> 28) == 0x01) + alc880_lg_lw_automute(codec); +} + +/* + * Common callbacks */ static int alc_init(struct hda_codec *codec) @@ -1107,9 +1533,21 @@ static int alc_init(struct hda_codec *codec) for (i = 0; i < spec->num_init_verbs; i++) snd_hda_sequence_write(codec, spec->init_verbs[i]); + + if (spec->init_hook) + spec->init_hook(codec); + return 0; } +static void alc_unsol_event(struct hda_codec *codec, unsigned int res) +{ + struct alc_spec *spec = codec->spec; + + if (spec->unsol_event) + spec->unsol_event(codec, res); +} + #ifdef CONFIG_PM /* * resume @@ -1250,6 +1688,13 @@ static struct hda_pcm_stream alc880_pcm_digital_capture = { /* NID is set in alc_build_pcms */ }; +/* Used by alc_build_pcms to flag that a PCM has no playback stream */ +static struct hda_pcm_stream alc_pcm_null_playback = { + .substreams = 0, + .channels_min = 0, + .channels_max = 0, +}; + static int alc_build_pcms(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -1280,6 +1725,23 @@ static int alc_build_pcms(struct hda_codec *codec) } } + /* If the use of more than one ADC is requested for the current + * model, configure a second analog capture-only PCM. + */ + if (spec->num_adc_nids > 1) { + codec->num_pcms++; + info++; + info->name = spec->stream_name_analog; + /* No playback stream for second PCM */ + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = alc_pcm_null_playback; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = 0; + if (spec->stream_analog_capture) { + snd_assert(spec->adc_nids, return -EINVAL); + info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture); + info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[1]; + } + } + if (spec->multiout.dig_out_nid || spec->dig_in_nid) { codec->num_pcms++; info++; @@ -1322,6 +1784,7 @@ static struct hda_codec_ops alc_patch_ops = { .build_pcms = alc_build_pcms, .init = alc_init, .free = alc_free, + .unsol_event = alc_unsol_event, #ifdef CONFIG_PM .resume = alc_resume, #endif @@ -1340,13 +1803,15 @@ static hda_nid_t alc880_test_dac_nids[4] = { }; static struct hda_input_mux alc880_test_capture_source = { - .num_items = 5, + .num_items = 7, .items = { { "In-1", 0x0 }, { "In-2", 0x1 }, { "In-3", 0x2 }, { "In-4", 0x3 }, { "CD", 0x4 }, + { "Front", 0x5 }, + { "Surround", 0x6 }, }, }; @@ -1653,6 +2118,8 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .pci_subvendor = 0x8086, .pci_subdevice = 0xa100, .config = ALC880_5ST_DIG }, { .pci_subvendor = 0x1565, .pci_subdevice = 0x8202, .config = ALC880_5ST_DIG }, { .pci_subvendor = 0x1019, .pci_subdevice = 0xa880, .config = ALC880_5ST_DIG }, + { .pci_subvendor = 0xa0a0, .pci_subdevice = 0x0560, + .config = ALC880_5ST_DIG }, /* Aopen i915GMm-HFS */ /* { .pci_subvendor = 0x1019, .pci_subdevice = 0xa884, .config = ALC880_5ST_DIG }, */ /* conflict with 6stack */ { .pci_subvendor = 0x1695, .pci_subdevice = 0x400d, .config = ALC880_5ST_DIG }, /* note subvendor = 0 below */ @@ -1680,6 +2147,8 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .pci_subvendor = 0x1025, .pci_subdevice = 0x0078, .config = ALC880_6ST_DIG }, { .pci_subvendor = 0x1025, .pci_subdevice = 0x0087, .config = ALC880_6ST_DIG }, { .pci_subvendor = 0x1297, .pci_subdevice = 0xc790, .config = ALC880_6ST_DIG }, /* Shuttle ST20G5 */ + { .pci_subvendor = 0x1509, .pci_subdevice = 0x925d, .config = ALC880_6ST_DIG }, /* FIC P4M-915GD1 */ + { .pci_subvendor = 0x1695, .pci_subdevice = 0x4012, .config = ALC880_5ST_DIG }, /* Epox EP-5LDA+ GLi */ { .modelname = "asus", .config = ALC880_ASUS }, { .pci_subvendor = 0x1043, .pci_subdevice = 0x1964, .config = ALC880_ASUS_DIG }, @@ -1693,6 +2162,7 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .pci_subvendor = 0x1043, .pci_subdevice = 0x1123, .config = ALC880_ASUS_DIG }, { .pci_subvendor = 0x1043, .pci_subdevice = 0x1143, .config = ALC880_ASUS }, { .pci_subvendor = 0x1043, .pci_subdevice = 0x10b3, .config = ALC880_ASUS_W1V }, + { .pci_subvendor = 0x1043, .pci_subdevice = 0x8181, .config = ALC880_ASUS_DIG }, /* ASUS P4GPL-X */ { .pci_subvendor = 0x1558, .pci_subdevice = 0x5401, .config = ALC880_ASUS_DIG2 }, { .modelname = "uniwill", .config = ALC880_UNIWILL_DIG }, @@ -1702,6 +2172,12 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .pci_subvendor = 0x1734, .pci_subdevice = 0x107c, .config = ALC880_F1734 }, { .pci_subvendor = 0x1584, .pci_subdevice = 0x9054, .config = ALC880_F1734 }, + { .modelname = "lg", .config = ALC880_LG }, + { .pci_subvendor = 0x1854, .pci_subdevice = 0x003b, .config = ALC880_LG }, + + { .modelname = "lg-lw", .config = ALC880_LG_LW }, + { .pci_subvendor = 0x1854, .pci_subdevice = 0x0018, .config = ALC880_LG_LW }, + #ifdef CONFIG_SND_DEBUG { .modelname = "test", .config = ALC880_TEST }, #endif @@ -1879,6 +2355,32 @@ static struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_threestack_modes, .input_mux = &alc880_capture_source, }, + [ALC880_LG] = { + .mixers = { alc880_lg_mixer }, + .init_verbs = { alc880_volume_init_verbs, + alc880_lg_init_verbs }, + .num_dacs = ARRAY_SIZE(alc880_lg_dac_nids), + .dac_nids = alc880_lg_dac_nids, + .dig_out_nid = ALC880_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc880_lg_ch_modes), + .channel_mode = alc880_lg_ch_modes, + .input_mux = &alc880_lg_capture_source, + .unsol_event = alc880_lg_unsol_event, + .init_hook = alc880_lg_automute, + }, + [ALC880_LG_LW] = { + .mixers = { alc880_lg_lw_mixer }, + .init_verbs = { alc880_volume_init_verbs, + alc880_lg_lw_init_verbs }, + .num_dacs = 1, + .dac_nids = alc880_dac_nids, + .dig_out_nid = ALC880_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes), + .channel_mode = alc880_2_jack_modes, + .input_mux = &alc880_lg_lw_capture_source, + .unsol_event = alc880_lg_lw_unsol_event, + .init_hook = alc880_lg_lw_automute, + }, #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = { .mixers = { alc880_test_mixer }, @@ -2043,14 +2545,11 @@ static int alc880_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin, if (alc880_is_fixed_pin(pin)) { nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin)); - if (! spec->multiout.dac_nids[0]) { - /* use this as the primary output */ - spec->multiout.dac_nids[0] = nid; - if (! spec->multiout.num_dacs) - spec->multiout.num_dacs = 1; - } else - /* specify the DAC as the extra output */ + /* specify the DAC as the extra output */ + if (! spec->multiout.hp_nid) spec->multiout.hp_nid = nid; + else + spec->multiout.extra_out_nid[0] = nid; /* control HP volume/switch on the output mixer amp */ nid = alc880_idx_to_mixer(alc880_fixed_pin_idx(pin)); sprintf(name, "%s Playback Volume", pfx); @@ -2063,12 +2562,6 @@ static int alc880_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin, return err; } else if (alc880_is_multi_pin(pin)) { /* set manual connection */ - if (! spec->multiout.dac_nids[0]) { - /* use this as the primary output */ - spec->multiout.dac_nids[0] = alc880_idx_to_dac(alc880_multi_pin_idx(pin)); - if (! spec->multiout.num_dacs) - spec->multiout.num_dacs = 1; - } /* we have only a switch on HP-out PIN */ sprintf(name, "%s Playback Switch", pfx); if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, @@ -2152,7 +2645,7 @@ static void alc880_auto_init_extra_out(struct hda_codec *codec) struct alc_spec *spec = codec->spec; hda_nid_t pin; - pin = spec->autocfg.speaker_pin; + pin = spec->autocfg.speaker_pins[0]; if (pin) /* connect to front */ alc880_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); pin = spec->autocfg.hp_pin; @@ -2188,15 +2681,15 @@ static int alc880_parse_auto_config(struct hda_codec *codec) if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, alc880_ignore)) < 0) return err; - if (! spec->autocfg.line_outs && ! spec->autocfg.speaker_pin && - ! spec->autocfg.hp_pin) + if (! spec->autocfg.line_outs) return 0; /* can't find valid BIOS pin config */ if ((err = alc880_auto_fill_dac_nids(spec, &spec->autocfg)) < 0 || (err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 || - (err = alc880_auto_create_extra_out(spec, spec->autocfg.speaker_pin, + (err = alc880_auto_create_extra_out(spec, + spec->autocfg.speaker_pins[0], "Speaker")) < 0 || - (err = alc880_auto_create_extra_out(spec, spec->autocfg.speaker_pin, + (err = alc880_auto_create_extra_out(spec, spec->autocfg.hp_pin, "Headphone")) < 0 || (err = alc880_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0) return err; @@ -2213,19 +2706,18 @@ static int alc880_parse_auto_config(struct hda_codec *codec) spec->init_verbs[spec->num_init_verbs++] = alc880_volume_init_verbs; + spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux; return 1; } -/* init callback for auto-configuration model -- overriding the default init */ -static int alc880_auto_init(struct hda_codec *codec) +/* additional initialization for auto-configuration model */ +static void alc880_auto_init(struct hda_codec *codec) { - alc_init(codec); alc880_auto_init_multi_out(codec); alc880_auto_init_extra_out(codec); alc880_auto_init_analog_input(codec); - return 0; } /* @@ -2292,7 +2784,7 @@ static int patch_alc880(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC880_AUTO) - codec->patch_ops.init = alc880_auto_init; + spec->init_hook = alc880_auto_init; return 0; } @@ -2322,6 +2814,14 @@ static hda_nid_t alc260_hp_adc_nids[2] = { 0x05, 0x04 }; +/* NIDs used when simultaneous access to both ADCs makes sense. Note that + * alc260_capture_mixer assumes ADC0 (nid 0x04) is the first ADC. + */ +static hda_nid_t alc260_dual_adc_nids[2] = { + /* ADC0, ADC1 */ + 0x04, 0x05 +}; + #define ALC260_DIGOUT_NID 0x03 #define ALC260_DIGIN_NID 0x06 @@ -2335,17 +2835,57 @@ static struct hda_input_mux alc260_capture_source = { }, }; -/* On Fujitsu S702x laptops capture only makes sense from Mic/LineIn jack - * and the internal CD lines. +/* On Fujitsu S702x laptops capture only makes sense from Mic/LineIn jack, + * headphone jack and the internal CD lines since these are the only pins at + * which audio can appear. For flexibility, also allow the option of + * recording the mixer output on the second ADC (ADC0 doesn't have a + * connection to the mixer output). */ -static struct hda_input_mux alc260_fujitsu_capture_source = { - .num_items = 2, - .items = { - { "Mic/Line", 0x0 }, - { "CD", 0x4 }, +static struct hda_input_mux alc260_fujitsu_capture_sources[2] = { + { + .num_items = 3, + .items = { + { "Mic/Line", 0x0 }, + { "CD", 0x4 }, + { "Headphone", 0x2 }, + }, + }, + { + .num_items = 4, + .items = { + { "Mic/Line", 0x0 }, + { "CD", 0x4 }, + { "Headphone", 0x2 }, + { "Mixer", 0x5 }, + }, }, + }; +/* Acer TravelMate(/Extensa/Aspire) notebooks have similar configuration to + * the Fujitsu S702x, but jacks are marked differently. + */ +static struct hda_input_mux alc260_acer_capture_sources[2] = { + { + .num_items = 4, + .items = { + { "Mic", 0x0 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + { "Headphone", 0x5 }, + }, + }, + { + .num_items = 5, + .items = { + { "Mic", 0x0 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + { "Headphone", 0x6 }, + { "Mixer", 0x5 }, + }, + }, +}; /* * This is just place-holder, so there's something for alc_build_pcms to look * at when it calculates the maximum number of channels. ALC260 has no mixer @@ -2363,6 +2903,7 @@ static struct hda_channel_mode alc260_modes[1] = { * HP: base_output + input + capture_alt * HP_3013: hp_3013 + input + capture * fujitsu: fujitsu + capture + * acer: acer + capture */ static struct snd_kcontrol_new alc260_base_output_mixer[] = { @@ -2405,14 +2946,18 @@ static struct snd_kcontrol_new alc260_hp_3013_mixer[] = { { } /* end */ }; +/* Fujitsu S702x series laptops. ALC260 pin usage: Mic/Line jack = 0x12, + * HP jack = 0x14, CD audio = 0x16, internal speaker = 0x10. + */ static struct snd_kcontrol_new alc260_fujitsu_mixer[] = { HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Headphone Playback Switch", 0x08, 2, HDA_INPUT), - ALC_PINCTL_SWITCH("Headphone Amp Switch", 0x14, PIN_HP_AMP), + ALC_PIN_MODE("Headphone Jack Mode", 0x14, ALC_PIN_DIR_INOUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Mic/Line Playback Volume", 0x07, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic/Line Playback Switch", 0x07, 0x0, HDA_INPUT), + ALC_PIN_MODE("Mic/Line Jack Mode", 0x12, ALC_PIN_DIR_IN), HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT), HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT), HDA_CODEC_VOLUME("Internal Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT), @@ -2420,6 +2965,41 @@ static struct snd_kcontrol_new alc260_fujitsu_mixer[] = { { } /* end */ }; +/* Mixer for Acer TravelMate(/Extensa/Aspire) notebooks. Note that current + * versions of the ALC260 don't act on requests to enable mic bias from NID + * 0x0f (used to drive the headphone jack in these laptops). The ALC260 + * datasheet doesn't mention this restriction. At this stage it's not clear + * whether this behaviour is intentional or is a hardware bug in chip + * revisions available in early 2006. Therefore for now allow the + * "Headphone Jack Mode" control to span all choices, but if it turns out + * that the lack of mic bias for this NID is intentional we could change the + * mode from ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS. + * + * In addition, Acer TravelMate(/Extensa/Aspire) notebooks in early 2006 + * don't appear to make the mic bias available from the "line" jack, even + * though the NID used for this jack (0x14) can supply it. The theory is + * that perhaps Acer have included blocking capacitors between the ALC260 + * and the output jack. If this turns out to be the case for all such + * models the "Line Jack Mode" mode could be changed from ALC_PIN_DIR_INOUT + * to ALC_PIN_DIR_INOUT_NOMICBIAS. + */ +static struct snd_kcontrol_new alc260_acer_mixer[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT), + ALC_PIN_MODE("Headphone Jack Mode", 0x0f, ALC_PIN_DIR_INOUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT), + ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN), + HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT), + ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT), + HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT), + { } /* end */ +}; + /* capture mixer elements */ static struct snd_kcontrol_new alc260_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x04, 0x0, HDA_INPUT), @@ -2620,7 +3200,8 @@ static struct hda_verb alc260_hp_3013_init_verbs[] = { }; /* Initialisation sequence for ALC260 as configured in Fujitsu S702x - * laptops. + * laptops. ALC260 pin usage: Mic/Line jack = 0x12, HP jack = 0x14, CD + * audio = 0x16, internal speaker = 0x10. */ static struct hda_verb alc260_fujitsu_init_verbs[] = { /* Disable all GPIOs */ @@ -2629,51 +3210,345 @@ static struct hda_verb alc260_fujitsu_init_verbs[] = { {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, /* Headphone/Line-out jack connects to Line1 pin; make it an output */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* Mic/Line-in jack is connected to mic1 pin, so make it an input */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - /* Ensure all other unused pins are disabled and muted. - * Note: trying to set widget 0x15 to anything blocks all audio - * output for some reason, so just leave that at the default. + /* Mic/Line-in jack is connected to mic1 pin, so make it an input */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + /* Ensure all other unused pins are disabled and muted. */ + {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + + /* Disable digital (SPDIF) pins */ + {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, + {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, + + /* Ensure Line1 pin widget takes its input from the OUT1 sum bus + * when acting as an output. + */ + {0x0d, AC_VERB_SET_CONNECT_SEL, 0}, + + /* Start with output sum widgets muted and their output gains at min */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + + /* Unmute HP pin widget amp left and right (no equiv mixer ctrl) */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Unmute Line1 pin widget output buffer since it starts as an output. + * If the pin mode is changed by the user the pin mode control will + * take care of enabling the pin's input/output buffers as needed. + * Therefore there's no need to enable the input buffer at this + * stage. + */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Unmute input buffer of pin widget used for Line-in (no equiv + * mixer ctrl) */ - {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + /* Mute capture amp left and right */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + /* Set ADC connection select to match default mixer setting - line + * in (on mic1 pin) + */ + {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Do the same for the second ADC: mute capture input amp and + * set ADC connection to line in (on mic1 pin) + */ + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x05, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Mute all inputs to mixer widget (even unconnected ones) */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */ + + { } +}; + +/* Initialisation sequence for ALC260 as configured in Acer TravelMate and + * similar laptops (adapted from Fujitsu init verbs). + */ +static struct hda_verb alc260_acer_init_verbs[] = { + /* On TravelMate laptops, GPIO 0 enables the internal speaker and + * the headphone jack. Turn this on and rely on the standard mute + * methods whenever the user wants to turn these outputs off. + */ + {0x01, AC_VERB_SET_GPIO_MASK, 0x01}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x01}, + /* Internal speaker/Headphone jack is connected to Line-out pin */ + {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Internal microphone/Mic jack is connected to Mic1 pin */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50}, + /* Line In jack is connected to Line1 pin */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + /* Ensure all other unused pins are disabled and muted. */ + {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - /* Disable digital (SPDIF) pins */ - {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, - {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, - - /* Start with mixer outputs muted */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - - /* Unmute HP pin widget amp left and right (no equiv mixer ctrl) */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Unmute Line1 pin widget amp left and right (no equiv mixer ctrl) */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Unmute pin widget used for Line-in (no equiv mixer ctrl) */ - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - - /* Mute capture amp left and right */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - /* Set ADC connection select to line in (on mic1 pin) */ - {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* Mute all inputs to mixer widget (even unconnected ones) */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */ + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + /* Disable digital (SPDIF) pins */ + {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, + {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, + + /* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum + * bus when acting as outputs. + */ + {0x0b, AC_VERB_SET_CONNECT_SEL, 0}, + {0x0d, AC_VERB_SET_CONNECT_SEL, 0}, + + /* Start with output sum widgets muted and their output gains at min */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + + /* Unmute Line-out pin widget amp left and right (no equiv mixer ctrl) */ + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Unmute Mic1 and Line1 pin widget input buffers since they start as + * inputs. If the pin mode is changed by the user the pin mode control + * will take care of enabling the pin's input/output buffers as needed. + * Therefore there's no need to enable the input buffer at this + * stage. + */ + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + /* Mute capture amp left and right */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + /* Set ADC connection select to match default mixer setting - mic + * (on mic1 pin) + */ + {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Do similar with the second ADC: mute capture input amp and + * set ADC connection to mic to match ALSA's default state. + */ + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x05, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Mute all inputs to mixer widget (even unconnected ones) */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */ + + { } +}; + +/* Test configuration for debugging, modelled after the ALC880 test + * configuration. + */ +#ifdef CONFIG_SND_DEBUG +static hda_nid_t alc260_test_dac_nids[1] = { + 0x02, +}; +static hda_nid_t alc260_test_adc_nids[2] = { + 0x04, 0x05, +}; +/* For testing the ALC260, each input MUX needs its own definition since + * the signal assignments are different. This assumes that the first ADC + * is NID 0x04. + */ +static struct hda_input_mux alc260_test_capture_sources[2] = { + { + .num_items = 7, + .items = { + { "MIC1 pin", 0x0 }, + { "MIC2 pin", 0x1 }, + { "LINE1 pin", 0x2 }, + { "LINE2 pin", 0x3 }, + { "CD pin", 0x4 }, + { "LINE-OUT pin", 0x5 }, + { "HP-OUT pin", 0x6 }, + }, + }, + { + .num_items = 8, + .items = { + { "MIC1 pin", 0x0 }, + { "MIC2 pin", 0x1 }, + { "LINE1 pin", 0x2 }, + { "LINE2 pin", 0x3 }, + { "CD pin", 0x4 }, + { "Mixer", 0x5 }, + { "LINE-OUT pin", 0x6 }, + { "HP-OUT pin", 0x7 }, + }, + }, +}; +static struct snd_kcontrol_new alc260_test_mixer[] = { + /* Output driver widgets */ + HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT), + HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x09, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("LOUT2 Playback Switch", 0x09, 2, HDA_INPUT), + HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x08, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("LOUT1 Playback Switch", 0x08, 2, HDA_INPUT), + + /* Modes for retasking pin widgets + * Note: the ALC260 doesn't seem to act on requests to enable mic + * bias from NIDs 0x0f and 0x10. The ALC260 datasheet doesn't + * mention this restriction. At this stage it's not clear whether + * this behaviour is intentional or is a hardware bug in chip + * revisions available at least up until early 2006. Therefore for + * now allow the "HP-OUT" and "LINE-OUT" Mode controls to span all + * choices, but if it turns out that the lack of mic bias for these + * NIDs is intentional we could change their modes from + * ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS. + */ + ALC_PIN_MODE("HP-OUT pin mode", 0x10, ALC_PIN_DIR_INOUT), + ALC_PIN_MODE("LINE-OUT pin mode", 0x0f, ALC_PIN_DIR_INOUT), + ALC_PIN_MODE("LINE2 pin mode", 0x15, ALC_PIN_DIR_INOUT), + ALC_PIN_MODE("LINE1 pin mode", 0x14, ALC_PIN_DIR_INOUT), + ALC_PIN_MODE("MIC2 pin mode", 0x13, ALC_PIN_DIR_INOUT), + ALC_PIN_MODE("MIC1 pin mode", 0x12, ALC_PIN_DIR_INOUT), + + /* Loopback mixer controls */ + HDA_CODEC_VOLUME("MIC1 Playback Volume", 0x07, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("MIC1 Playback Switch", 0x07, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("MIC2 Playback Volume", 0x07, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("MIC2 Playback Switch", 0x07, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("LINE1 Playback Volume", 0x07, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("LINE1 Playback Switch", 0x07, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("LINE2 Playback Volume", 0x07, 0x03, HDA_INPUT), + HDA_CODEC_MUTE("LINE2 Playback Switch", 0x07, 0x03, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT), + HDA_CODEC_VOLUME("LINE-OUT loopback Playback Volume", 0x07, 0x06, HDA_INPUT), + HDA_CODEC_MUTE("LINE-OUT loopback Playback Switch", 0x07, 0x06, HDA_INPUT), + HDA_CODEC_VOLUME("HP-OUT loopback Playback Volume", 0x07, 0x7, HDA_INPUT), + HDA_CODEC_MUTE("HP-OUT loopback Playback Switch", 0x07, 0x7, HDA_INPUT), + + /* Controls for GPIO pins, assuming they are configured as outputs */ + ALC_GPIO_DATA_SWITCH("GPIO pin 0", 0x01, 0x01), + ALC_GPIO_DATA_SWITCH("GPIO pin 1", 0x01, 0x02), + ALC_GPIO_DATA_SWITCH("GPIO pin 2", 0x01, 0x04), + ALC_GPIO_DATA_SWITCH("GPIO pin 3", 0x01, 0x08), + + /* Switches to allow the digital IO pins to be enabled. The datasheet + * is ambigious as to which NID is which; testing on laptops which + * make this output available should provide clarification. + */ + ALC_SPDIF_CTRL_SWITCH("SPDIF Playback Switch", 0x03, 0x01), + ALC_SPDIF_CTRL_SWITCH("SPDIF Capture Switch", 0x06, 0x01), + + { } /* end */ +}; +static struct hda_verb alc260_test_init_verbs[] = { + /* Enable all GPIOs as outputs with an initial value of 0 */ + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x0f}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x00}, + {0x01, AC_VERB_SET_GPIO_MASK, 0x0f}, + + /* Enable retasking pins as output, initially without power amp */ + {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + + /* Disable digital (SPDIF) pins initially, but users can enable + * them via a mixer switch. In the case of SPDIF-out, this initverb + * payload also sets the generation to 0, output to be in "consumer" + * PCM format, copyright asserted, no pre-emphasis and no validity + * control. + */ + {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, + {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, + + /* Ensure mic1, mic2, line1 and line2 pin widgets take input from the + * OUT1 sum bus when acting as an output. + */ + {0x0b, AC_VERB_SET_CONNECT_SEL, 0}, + {0x0c, AC_VERB_SET_CONNECT_SEL, 0}, + {0x0d, AC_VERB_SET_CONNECT_SEL, 0}, + {0x0e, AC_VERB_SET_CONNECT_SEL, 0}, + + /* Start with output sum widgets muted and their output gains at min */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + + /* Unmute retasking pin widget output buffers since the default + * state appears to be output. As the pin mode is changed by the + * user the pin mode control will take care of enabling the pin's + * input/output buffers as needed. + */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Also unmute the mono-out pin widget */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* Mute capture amp left and right */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + /* Set ADC connection select to match default mixer setting (mic1 + * pin) + */ + {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Do the same for the second ADC: mute capture input amp and + * set ADC connection to mic1 pin + */ + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x05, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Mute all inputs to mixer widget (even unconnected ones) */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */ { } }; +#endif static struct hda_pcm_stream alc260_pcm_analog_playback = { .substreams = 1, @@ -2744,7 +3619,7 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec, return err; } - nid = cfg->speaker_pin; + nid = cfg->speaker_pins[0]; if (nid) { err = alc260_add_playback_controls(spec, nid, "Speaker"); if (err < 0) @@ -2817,7 +3692,7 @@ static void alc260_auto_init_multi_out(struct hda_codec *codec) if (nid) alc260_auto_set_output_and_unmute(codec, nid, PIN_OUT, 0); - nid = spec->autocfg.speaker_pin; + nid = spec->autocfg.speaker_pins[0]; if (nid) alc260_auto_set_output_and_unmute(codec, nid, PIN_OUT, 0); @@ -2913,6 +3788,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec) spec->init_verbs[spec->num_init_verbs++] = alc260_volume_init_verbs; + spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux; /* check whether NID 0x04 is valid */ @@ -2932,13 +3808,11 @@ static int alc260_parse_auto_config(struct hda_codec *codec) return 1; } -/* init callback for auto-configuration model -- overriding the default init */ -static int alc260_auto_init(struct hda_codec *codec) +/* additional initialization for auto-configuration model */ +static void alc260_auto_init(struct hda_codec *codec) { - alc_init(codec); alc260_auto_init_multi_out(codec); alc260_auto_init_analog_input(codec); - return 0; } /* @@ -2948,6 +3822,10 @@ static struct hda_board_config alc260_cfg_tbl[] = { { .modelname = "basic", .config = ALC260_BASIC }, { .pci_subvendor = 0x104d, .pci_subdevice = 0x81bb, .config = ALC260_BASIC }, /* Sony VAIO */ + { .pci_subvendor = 0x104d, .pci_subdevice = 0x81cd, + .config = ALC260_BASIC }, /* Sony VAIO */ + { .pci_subvendor = 0x152d, .pci_subdevice = 0x0729, + .config = ALC260_BASIC }, /* CTL Travel Master U553W */ { .modelname = "hp", .config = ALC260_HP }, { .pci_subvendor = 0x103c, .pci_subdevice = 0x3010, .config = ALC260_HP }, { .pci_subvendor = 0x103c, .pci_subdevice = 0x3011, .config = ALC260_HP }, @@ -2958,6 +3836,11 @@ static struct hda_board_config alc260_cfg_tbl[] = { { .pci_subvendor = 0x103c, .pci_subdevice = 0x3016, .config = ALC260_HP }, { .modelname = "fujitsu", .config = ALC260_FUJITSU_S702X }, { .pci_subvendor = 0x10cf, .pci_subdevice = 0x1326, .config = ALC260_FUJITSU_S702X }, + { .modelname = "acer", .config = ALC260_ACER }, + { .pci_subvendor = 0x1025, .pci_subdevice = 0x008f, .config = ALC260_ACER }, +#ifdef CONFIG_SND_DEBUG + { .modelname = "test", .config = ALC260_TEST }, +#endif { .modelname = "auto", .config = ALC260_AUTO }, {} }; @@ -3009,12 +3892,41 @@ static struct alc_config_preset alc260_presets[] = { .init_verbs = { alc260_fujitsu_init_verbs }, .num_dacs = ARRAY_SIZE(alc260_dac_nids), .dac_nids = alc260_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc260_adc_nids), - .adc_nids = alc260_adc_nids, + .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids), + .adc_nids = alc260_dual_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc260_modes), + .channel_mode = alc260_modes, + .num_mux_defs = ARRAY_SIZE(alc260_fujitsu_capture_sources), + .input_mux = alc260_fujitsu_capture_sources, + }, + [ALC260_ACER] = { + .mixers = { alc260_acer_mixer, + alc260_capture_mixer }, + .init_verbs = { alc260_acer_init_verbs }, + .num_dacs = ARRAY_SIZE(alc260_dac_nids), + .dac_nids = alc260_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids), + .adc_nids = alc260_dual_adc_nids, .num_channel_mode = ARRAY_SIZE(alc260_modes), .channel_mode = alc260_modes, - .input_mux = &alc260_fujitsu_capture_source, + .num_mux_defs = ARRAY_SIZE(alc260_acer_capture_sources), + .input_mux = alc260_acer_capture_sources, }, +#ifdef CONFIG_SND_DEBUG + [ALC260_TEST] = { + .mixers = { alc260_test_mixer, + alc260_capture_mixer }, + .init_verbs = { alc260_test_init_verbs }, + .num_dacs = ARRAY_SIZE(alc260_test_dac_nids), + .dac_nids = alc260_test_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc260_test_adc_nids), + .adc_nids = alc260_test_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc260_modes), + .channel_mode = alc260_modes, + .num_mux_defs = ARRAY_SIZE(alc260_test_capture_sources), + .input_mux = alc260_test_capture_sources, + }, +#endif }; static int patch_alc260(struct hda_codec *codec) @@ -3059,7 +3971,7 @@ static int patch_alc260(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC260_AUTO) - codec->patch_ops.init = alc260_auto_init; + spec->init_hook = alc260_auto_init; return 0; } @@ -3104,7 +4016,6 @@ static struct hda_input_mux alc882_capture_source = { { "CD", 0x4 }, }, }; - #define alc882_mux_enum_info alc_mux_enum_info #define alc882_mux_enum_get alc_mux_enum_get @@ -3534,14 +4445,12 @@ static int alc882_parse_auto_config(struct hda_codec *codec) return err; } -/* init callback for auto-configuration model -- overriding the default init */ -static int alc882_auto_init(struct hda_codec *codec) +/* additional initialization for auto-configuration model */ +static void alc882_auto_init(struct hda_codec *codec) { - alc_init(codec); alc882_auto_init_multi_out(codec); alc882_auto_init_hp_out(codec); alc882_auto_init_analog_input(codec); - return 0; } /* @@ -3608,7 +4517,7 @@ static int patch_alc882(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC882_AUTO) - codec->patch_ops.init = alc882_auto_init; + spec->init_hook = alc882_auto_init; return 0; } @@ -3644,19 +4553,9 @@ static struct snd_kcontrol_new alc262_base_mixer[] = { HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .count = 1, - .info = alc882_mux_enum_info, - .get = alc882_mux_enum_get, - .put = alc882_mux_enum_put, - }, { } /* end */ -}; - +}; + #define alc262_capture_mixer alc882_capture_mixer #define alc262_capture_alt_mixer alc882_capture_alt_mixer @@ -3739,6 +4638,129 @@ static struct hda_verb alc262_init_verbs[] = { { } }; +/* + * fujitsu model + * 0x14 = headphone/spdif-out, 0x15 = internal speaker + */ + +#define ALC_HP_EVENT 0x37 + +static struct hda_verb alc262_fujitsu_unsol_verbs[] = { + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {} +}; + +static struct hda_input_mux alc262_fujitsu_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x0 }, + { "CD", 0x4 }, + }, +}; + +/* mute/unmute internal speaker according to the hp jack and mute state */ +static void alc262_fujitsu_automute(struct hda_codec *codec, int force) +{ + struct alc_spec *spec = codec->spec; + unsigned int mute; + + if (force || ! spec->sense_updated) { + unsigned int present; + /* need to execute and sync at first */ + snd_hda_codec_read(codec, 0x14, 0, AC_VERB_SET_PIN_SENSE, 0); + present = snd_hda_codec_read(codec, 0x14, 0, + AC_VERB_GET_PIN_SENSE, 0); + spec->jack_present = (present & 0x80000000) != 0; + spec->sense_updated = 1; + } + if (spec->jack_present) { + /* mute internal speaker */ + snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0, + 0x80, 0x80); + snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0, + 0x80, 0x80); + } else { + /* unmute internal speaker if necessary */ + mute = snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0); + snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0, + 0x80, mute & 0x80); + mute = snd_hda_codec_amp_read(codec, 0x14, 1, HDA_OUTPUT, 0); + snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0, + 0x80, mute & 0x80); + } +} + +/* unsolicited event for HP jack sensing */ +static void alc262_fujitsu_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) != ALC_HP_EVENT) + return; + alc262_fujitsu_automute(codec, 1); +} + +/* bind volumes of both NID 0x0c and 0x0d */ +static int alc262_fujitsu_master_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + long *valp = ucontrol->value.integer.value; + int change; + + change = snd_hda_codec_amp_update(codec, 0x0c, 0, HDA_OUTPUT, 0, + 0x7f, valp[0] & 0x7f); + change |= snd_hda_codec_amp_update(codec, 0x0c, 1, HDA_OUTPUT, 0, + 0x7f, valp[1] & 0x7f); + snd_hda_codec_amp_update(codec, 0x0d, 0, HDA_OUTPUT, 0, + 0x7f, valp[0] & 0x7f); + snd_hda_codec_amp_update(codec, 0x0d, 1, HDA_OUTPUT, 0, + 0x7f, valp[1] & 0x7f); + return change; +} + +/* bind hp and internal speaker mute (with plug check) */ +static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + long *valp = ucontrol->value.integer.value; + int change; + + change = snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0, + 0x80, valp[0] ? 0 : 0x80); + change |= snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0, + 0x80, valp[1] ? 0 : 0x80); + if (change || codec->in_resume) + alc262_fujitsu_automute(codec, codec->in_resume); + return change; +} + +static struct snd_kcontrol_new alc262_fujitsu_mixer[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Volume", + .info = snd_hda_mixer_amp_volume_info, + .get = snd_hda_mixer_amp_volume_get, + .put = alc262_fujitsu_master_vol_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT), + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = alc262_fujitsu_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), + }, + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + { } /* end */ +}; + /* add playback controls from the parsed DAC table */ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { @@ -3759,7 +4781,7 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, const struct return err; } - nid = cfg->speaker_pin; + nid = cfg->speaker_pins[0]; if (nid) { if (nid == 0x16) { if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, "Speaker Playback Volume", @@ -3769,10 +4791,6 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, const struct HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT))) < 0) return err; } else { - if (! cfg->line_out_pins[0]) - if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, "Speaker Playback Volume", - HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT))) < 0) - return err; if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, "Speaker Playback Switch", HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0) return err; @@ -3789,10 +4807,6 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, const struct HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT))) < 0) return err; } else { - if (! cfg->line_out_pins[0]) - if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, "Headphone Playback Volume", - HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT))) < 0) - return err; if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, "Headphone Playback Switch", HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0) return err; @@ -3886,8 +4900,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec) if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, alc262_ignore)) < 0) return err; - if (! spec->autocfg.line_outs && ! spec->autocfg.speaker_pin && - ! spec->autocfg.hp_pin) + if (! spec->autocfg.line_outs) return 0; /* can't find valid BIOS pin config */ if ((err = alc262_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 || (err = alc262_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0) @@ -3904,6 +4917,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec) spec->mixers[spec->num_mixers++] = spec->kctl_alloc; spec->init_verbs[spec->num_init_verbs++] = alc262_volume_init_verbs; + spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux; return 1; @@ -3915,13 +4929,11 @@ static int alc262_parse_auto_config(struct hda_codec *codec) /* init callback for auto-configuration model -- overriding the default init */ -static int alc262_auto_init(struct hda_codec *codec) +static void alc262_auto_init(struct hda_codec *codec) { - alc_init(codec); alc262_auto_init_multi_out(codec); alc262_auto_init_hp_out(codec); alc262_auto_init_analog_input(codec); - return 0; } /* @@ -3929,6 +4941,8 @@ static int alc262_auto_init(struct hda_codec *codec) */ static struct hda_board_config alc262_cfg_tbl[] = { { .modelname = "basic", .config = ALC262_BASIC }, + { .modelname = "fujitsu", .config = ALC262_FUJITSU }, + { .pci_subvendor = 0x10cf, .pci_subdevice = 0x1397, .config = ALC262_FUJITSU }, { .modelname = "auto", .config = ALC262_AUTO }, {} }; @@ -3944,6 +4958,18 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, }, + [ALC262_FUJITSU] = { + .mixers = { alc262_fujitsu_mixer }, + .init_verbs = { alc262_init_verbs, alc262_fujitsu_unsol_verbs }, + .num_dacs = ARRAY_SIZE(alc262_dac_nids), + .dac_nids = alc262_dac_nids, + .hp_nid = 0x03, + .dig_out_nid = ALC262_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc262_modes), + .channel_mode = alc262_modes, + .input_mux = &alc262_fujitsu_capture_source, + .unsol_event = alc262_fujitsu_unsol_event, + }, }; static int patch_alc262(struct hda_codec *codec) @@ -4017,8 +5043,8 @@ static int patch_alc262(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC262_AUTO) - codec->patch_ops.init = alc262_auto_init; - + spec->init_hook = alc262_auto_init; + return 0; } @@ -4549,8 +5575,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec) if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, alc861_ignore)) < 0) return err; - if (! spec->autocfg.line_outs && ! spec->autocfg.speaker_pin && - ! spec->autocfg.hp_pin) + if (! spec->autocfg.line_outs) return 0; /* can't find valid BIOS pin config */ if ((err = alc861_auto_fill_dac_nids(spec, &spec->autocfg)) < 0 || @@ -4569,6 +5594,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec) spec->init_verbs[spec->num_init_verbs++] = alc861_auto_init_verbs; + spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux; spec->adc_nids = alc861_adc_nids; @@ -4579,15 +5605,12 @@ static int alc861_parse_auto_config(struct hda_codec *codec) return 1; } -/* init callback for auto-configuration model -- overriding the default init */ -static int alc861_auto_init(struct hda_codec *codec) +/* additional initialization for auto-configuration model */ +static void alc861_auto_init(struct hda_codec *codec) { - alc_init(codec); alc861_auto_init_multi_out(codec); alc861_auto_init_hp_out(codec); alc861_auto_init_analog_input(codec); - - return 0; } @@ -4685,7 +5708,7 @@ static int patch_alc861(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC861_AUTO) - codec->patch_ops.init = alc861_auto_init; + spec->init_hook = alc861_auto_init; return 0; }