* Copyright (c) 2004 Kailang Yang <kailang@realtek.com.tw>
* PeiSen Hou <pshou@realtek.com.tw>
* Takashi Iwai <tiwai@suse.de>
+ * Jonathan Woithe <jwoithe@physics.adelaide.edu.au>
*
* This driver is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
ALC880_UNIWILL_DIG,
ALC880_CLEVO,
ALC880_TCL_S700,
+ ALC880_LG,
+ ALC880_LG_LW,
#ifdef CONFIG_SND_DEBUG
ALC880_TEST,
#endif
ALC260_HP,
ALC260_HP_3013,
ALC260_FUJITSU_S702X,
+ ALC260_ACER,
+#ifdef CONFIG_SND_DEBUG
+ ALC260_TEST,
+#endif
ALC260_AUTO,
ALC260_MODEL_LAST /* last tag */
};
/* ALC262 models */
enum {
ALC262_BASIC,
+ ALC262_FUJITSU,
ALC262_AUTO,
ALC262_MODEL_LAST /* last tag */
};
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];
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;
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;
};
/*
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 *);
};
{
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)
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]);
}
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_num<alc_pin_mode_min(dir) || item_num>alc_pin_mode_max(dir))
+ item_num = alc_pin_mode_min(dir);
+ strcpy(uinfo->value.enumerated.name, alc_pin_mode_names[item_num]);
return 0;
}
-static int alc_pinctl_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+static int alc_pin_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
+ unsigned int i;
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
- long mask = (kcontrol->private_value >> 16) & 0xff;
+ unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
long *valp = ucontrol->value.integer.value;
+ unsigned int pinctl = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_PIN_WIDGET_CONTROL,0x00);
- *valp = 0;
- if (snd_hda_codec_read(codec,nid,0,AC_VERB_GET_PIN_WIDGET_CONTROL,0x00) & mask)
- *valp = 1;
+ /* Find enumerated value for current pinctl setting */
+ i = alc_pin_mode_min(dir);
+ while (alc_pin_mode_values[i]!=pinctl && i<=alc_pin_mode_max(dir))
+ i++;
+ *valp = i<=alc_pin_mode_max(dir)?i:alc_pin_mode_min(dir);
return 0;
}
-static int alc_pinctl_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+static int alc_pin_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
+ signed int change;
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
- long mask = (kcontrol->private_value >> 16) & 0xff;
- long *valp = ucontrol->value.integer.value;
+ unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
+ long val = *ucontrol->value.integer.value;
unsigned int pinctl = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_PIN_WIDGET_CONTROL,0x00);
- int change = ((pinctl & mask)!=0) != *valp;
- if (change)
+ if (val<alc_pin_mode_min(dir) || val>alc_pin_mode_max(dir))
+ val = alc_pin_mode_min(dir);
+
+ change = pinctl != alc_pin_mode_values[val];
+ if (change) {
+ /* Set pin mode to that requested */
snd_hda_codec_write(codec,nid,0,AC_VERB_SET_PIN_WIDGET_CONTROL,
- *valp?(pinctl|mask):(pinctl&~mask));
+ alc_pin_mode_values[val]);
+
+ /* Also enable the retasking pin's input/output as required
+ * for the requested pin mode. Enum values of 2 or less are
+ * input modes.
+ *
+ * Dynamically switching the input/output buffers probably
+ * reduces noise slightly (particularly on input) 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
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;
}
/*
};
/*
+ * 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)
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
/* 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;
}
}
+ /* 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++;
.build_pcms = alc_build_pcms,
.init = alc_init,
.free = alc_free,
+ .unsol_event = alc_unsol_event,
#ifdef CONFIG_PM
.resume = alc_resume,
#endif
};
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 },
},
};
{ .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 */
{ .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 },
{ .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 },
{ .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
.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 },
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);
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,
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;
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;
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;
}
/*
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;
}
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
},
};
-/* 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
* 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[] = {
{ } /* 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),
{ } /* 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),
};
/* 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 */
{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,
.channels_min = 2,
return err;
}
- nid = cfg->speaker_pin;
+ nid = cfg->speaker_pins[0];
if (nid) {
err = alc260_add_playback_controls(spec, nid, "Speaker");
if (err < 0)
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);
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 */
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;
}
/*
{ .modelname = "basic", .config = ALC260_BASIC },
{ .pci_subvendor = 0x104d, .pci_subdevice = 0x81bb,
.config = ALC260_BASIC }, /* Sony VAIO */
+ { .pci_subvendor = 0x152d, .pci_subdevice = 0x0729,
+ .config = ALC260_BASIC }, /* CTL Travel Master U553W */
{ .modelname = "hp", .config = ALC260_HP },
{ .pci_subvendor = 0x103c, .pci_subdevice = 0x3010, .config = ALC260_HP },
{ .pci_subvendor = 0x103c, .pci_subdevice = 0x3011, .config = ALC260_HP },
{ .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 },
{}
};
.init_verbs = { alc260_fujitsu_init_verbs },
.num_dacs = ARRAY_SIZE(alc260_dac_nids),
.dac_nids = alc260_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc260_adc_nids),
- .adc_nids = alc260_adc_nids,
+ .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
+ .adc_nids = alc260_dual_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
- .input_mux = &alc260_fujitsu_capture_source,
+ .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,
+ .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)
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;
}
{ "CD", 0x4 },
},
};
-
#define alc882_mux_enum_info alc_mux_enum_info
#define alc882_mux_enum_get alc_mux_enum_get
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;
}
/*
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;
}
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
{ }
};
+/*
+ * 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)
{
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",
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;
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;
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)
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;
/* 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;
}
/*
*/
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 },
{}
};
.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)
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;
}
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 ||
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;
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;
}
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;
}
git.openpandora.org / pandora-kernel.git / blobdiff