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Code Issues Releases Wiki Activity

Merge branch 'topic/hda' into for-linus

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This commit is contained in:
Takashi Iwai 2011-07-22 08:43:27 +02:00
commit 76531d4166
30 changed files with 19575 additions and 22099 deletions
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100
Documentation/sound/alsa/HD-Audio-Controls.txt Normal file
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@ -0,0 +1,100 @@
This file explains the codec-specific mixer controls.
Realtek codecs
--------------
* Channel Mode
This is an enum control to change the surround-channel setup,
appears only when the surround channels are available.
It gives the number of channels to be used, "2ch", "4ch", "6ch",
and "8ch". According to the configuration, this also controls the
jack-retasking of multi-I/O jacks.
* Auto-Mute Mode
This is an enum control to change the auto-mute behavior of the
headphone and line-out jacks. If built-in speakers and headphone
and/or line-out jacks are available on a machine, this controls
appears.
When there are only either headphones or line-out jacks, it gives
"Disabled" and "Enabled" state. When enabled, the speaker is muted
automatically when a jack is plugged.
When both headphone and line-out jacks are present, it gives
"Disabled", "Speaker Only" and "Line-Out+Speaker". When
speaker-only is chosen, plugging into a headphone or a line-out jack
mutes the speakers, but not line-outs. When line-out+speaker is
selected, plugging to a headphone jack mutes both speakers and
line-outs.
IDT/Sigmatel codecs
-------------------
* Analog Loopback
This control enables/disables the analog-loopback circuit. This
appears only when "loopback" is set to true in a codec hint
(see HD-Audio.txt). Note that on some codecs the analog-loopback
and the normal PCM playback are exclusive, i.e. when this is on, you
won't hear any PCM stream.
* Swap Center/LFE
Swaps the center and LFE channel order. Normally, the left
corresponds to the center and the right to the LFE. When this is
ON, the left to the LFE and the right to the center.
* Headphone as Line Out
When this control is ON, treat the headphone jacks as line-out
jacks. That is, the headphone won't auto-mute the other line-outs,
and no HP-amp is set to the pins.
* Mic Jack Mode, Line Jack Mode, etc
These enum controls the direction and the bias of the input jack
pins. Depending on the jack type, it can set as "Mic In" and "Line
In", for determining the input bias, or it can be set to "Line Out"
when the pin is a multi-I/O jack for surround channels.
VIA codecs
----------
* Smart 5.1
An enum control to re-task the multi-I/O jacks for surround outputs.
When it's ON, the corresponding input jacks (usually a line-in and a
mic-in) are switched as the surround and the CLFE output jacks.
* Independent HP
When this enum control is enabled, the headphone output is routed
from an individual stream (the third PCM such as hw:0,2) instead of
the primary stream. In the case the headphone DAC is shared with a
side or a CLFE-channel DAC, the DAC is switched to the headphone
automatically.
* Loopback Mixing
An enum control to determine whether the analog-loopback route is
enabled or not. When it's enabled, the analog-loopback is mixed to
the front-channel. Also, the same route is used for the headphone
and speaker outputs. As a side-effect, when this mode is set, the
individual volume controls will be no longer available for
headphones and speakers because there is only one DAC connected to a
mixer widget.
* Dynamic Power-Control
This control determines whether the dynamic power-control per jack
detection is enabled or not. When enabled, the widgets power state
(D0/D3) are changed dynamically depending on the jack plugging
state for saving power consumptions. However, if your system
doesn't provide a proper jack-detection, this won't work; in such a
case, turn this control OFF.
* Jack Detect
This control is provided only for VT1708 codec which gives no proper
unsolicited event per jack plug. When this is on, the driver polls
the jack detection so that the headphone auto-mute can work, while
turning this off would reduce the power consumption.
Conexant codecs
---------------
* Auto-Mute Mode
See Reatek codecs.

39
sound/pci/hda/Kconfig
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@ -14,6 +14,19 @@ menuconfig SND_HDA_INTEL
if SND_HDA_INTEL
config SND_HDA_PREALLOC_SIZE
int "Pre-allocated buffer size for HD-audio driver"
range 0 32768
default 64
help
Specifies the default pre-allocated buffer-size in kB for the
HD-audio driver. A larger buffer (e.g. 2048) is preferred
for systems using PulseAudio. The default 64 is chosen just
for compatibility reasons.
Note that the pre-allocation size can be changed dynamically
via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too.
config SND_HDA_HWDEP
bool "Build hwdep interface for HD-audio driver"
select SND_HWDEP
@ -83,6 +96,19 @@ config SND_HDA_CODEC_REALTEK
snd-hda-codec-realtek.
This module is automatically loaded at probing.
config SND_HDA_ENABLE_REALTEK_QUIRKS
bool "Build static quirks for Realtek codecs"
depends on SND_HDA_CODEC_REALTEK
default y
help
Say Y here to build the static quirks codes for Realtek codecs.
If you need the "model" preset that the default BIOS auto-parser
can't handle, turn this option on.
If your device works with model=auto option, basically you don't
need the quirk code. By turning this off, you can reduce the
module size quite a lot.
config SND_HDA_CODEC_ANALOG
bool "Build Analog Device HD-audio codec support"
default y
@ -171,6 +197,19 @@ config SND_HDA_CODEC_CA0110
snd-hda-codec-ca0110.
This module is automatically loaded at probing.
config SND_HDA_CODEC_CA0132
bool "Build Creative CA0132 codec support"
depends on SND_HDA_INTEL
default y
help
Say Y here to include Creative CA0132 codec support in
snd-hda-intel driver.
When the HD-audio driver is built as a module, the codec
support code is also built as another module,
snd-hda-codec-ca0132.
This module is automatically loaded at probing.
config SND_HDA_CODEC_CMEDIA
bool "Build C-Media HD-audio codec support"
default y

4
sound/pci/hda/Makefile
View file

@ -13,6 +13,7 @@ snd-hda-codec-idt-objs := patch_sigmatel.o
snd-hda-codec-si3054-objs := patch_si3054.o
snd-hda-codec-cirrus-objs := patch_cirrus.o
snd-hda-codec-ca0110-objs := patch_ca0110.o
snd-hda-codec-ca0132-objs := patch_ca0132.o
snd-hda-codec-conexant-objs := patch_conexant.o
snd-hda-codec-via-objs := patch_via.o
snd-hda-codec-hdmi-objs := patch_hdmi.o hda_eld.o
@ -42,6 +43,9 @@ endif
ifdef CONFIG_SND_HDA_CODEC_CA0110
obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-ca0110.o
endif
ifdef CONFIG_SND_HDA_CODEC_CA0132
obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-ca0132.o
endif
ifdef CONFIG_SND_HDA_CODEC_CONEXANT
obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-conexant.o
endif

1272
sound/pci/hda/alc260_quirks.c Normal file
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1353
sound/pci/hda/alc262_quirks.c Normal file
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File diff suppressed because it is too large Load diff

636
sound/pci/hda/alc268_quirks.c Normal file
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@ -0,0 +1,636 @@
/*
* ALC267/ALC268 quirk models
* included by patch_realtek.c
*/
/* ALC268 models */
enum {
ALC268_AUTO,
ALC267_QUANTA_IL1,
ALC268_3ST,
ALC268_TOSHIBA,
ALC268_ACER,
ALC268_ACER_DMIC,
ALC268_ACER_ASPIRE_ONE,
ALC268_DELL,
ALC268_ZEPTO,
#ifdef CONFIG_SND_DEBUG
ALC268_TEST,
#endif
ALC268_MODEL_LAST /* last tag */
};
/*
* ALC268 channel source setting (2 channel)
*/
#define ALC268_DIGOUT_NID ALC880_DIGOUT_NID
#define alc268_modes alc260_modes
static const hda_nid_t alc268_dac_nids[2] = {
/* front, hp */
0x02, 0x03
};
static const hda_nid_t alc268_adc_nids[2] = {
/* ADC0-1 */
0x08, 0x07
};
static const hda_nid_t alc268_adc_nids_alt[1] = {
/* ADC0 */
0x08
};
static const hda_nid_t alc268_capsrc_nids[2] = { 0x23, 0x24 };
static const struct snd_kcontrol_new alc268_base_mixer[] = {
/* output mixer control */
HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT),
{ }
};
static const struct snd_kcontrol_new alc268_toshiba_mixer[] = {
/* output mixer control */
HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT),
ALC262_HIPPO_MASTER_SWITCH,
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT),
{ }
};
static const struct hda_verb alc268_eapd_verbs[] = {
{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
{ }
};
/* Toshiba specific */
static const struct hda_verb alc268_toshiba_verbs[] = {
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
{ } /* end */
};
/* Acer specific */
/* bind volumes of both NID 0x02 and 0x03 */
static const struct hda_bind_ctls alc268_acer_bind_master_vol = {
.ops = &snd_hda_bind_vol,
.values = {
HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
HDA_COMPOSE_AMP_VAL(0x03, 3, 0, HDA_OUTPUT),
0
},
};
static void alc268_acer_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x15;
spec->automute = 1;
spec->automute_mode = ALC_AUTOMUTE_AMP;
}
#define alc268_acer_master_sw_get alc262_hp_master_sw_get
#define alc268_acer_master_sw_put alc262_hp_master_sw_put
static const struct snd_kcontrol_new alc268_acer_aspire_one_mixer[] = {
/* output mixer control */
HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Master Playback Switch",
.subdevice = HDA_SUBDEV_NID_FLAG | 0x15,
.info = snd_ctl_boolean_mono_info,
.get = alc268_acer_master_sw_get,
.put = alc268_acer_master_sw_put,
},
HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x18, 0, HDA_INPUT),
{ }
};
static const struct snd_kcontrol_new alc268_acer_mixer[] = {
/* output mixer control */
HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Master Playback Switch",
.subdevice = HDA_SUBDEV_NID_FLAG | 0x14,
.info = snd_ctl_boolean_mono_info,
.get = alc268_acer_master_sw_get,
.put = alc268_acer_master_sw_put,
},
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT),
{ }
};
static const struct snd_kcontrol_new alc268_acer_dmic_mixer[] = {
/* output mixer control */
HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Master Playback Switch",
.subdevice = HDA_SUBDEV_NID_FLAG | 0x14,
.info = snd_ctl_boolean_mono_info,
.get = alc268_acer_master_sw_get,
.put = alc268_acer_master_sw_put,
},
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT),
{ }
};
static const struct hda_verb alc268_acer_aspire_one_verbs[] = {
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
{0x23, AC_VERB_SET_CONNECT_SEL, 0x06},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, 0xa017},
{ }
};
static const struct hda_verb alc268_acer_verbs[] = {
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* internal dmic? */
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
{ }
};
/* unsolicited event for HP jack sensing */
#define alc268_toshiba_setup alc262_hippo_setup
static void alc268_acer_lc_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->automute = 1;
spec->automute_mode = ALC_AUTOMUTE_AMP;
spec->ext_mic_pin = 0x18;
spec->int_mic_pin = 0x12;
spec->auto_mic = 1;
}
static const struct snd_kcontrol_new alc268_dell_mixer[] = {
/* output mixer control */
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
{ }
};
static const struct hda_verb alc268_dell_verbs[] = {
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN},
{ }
};
/* mute/unmute internal speaker according to the hp jack and mute state */
static void alc268_dell_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->ext_mic_pin = 0x18;
spec->int_mic_pin = 0x19;
spec->auto_mic = 1;
spec->automute = 1;
spec->automute_mode = ALC_AUTOMUTE_PIN;
}
static const struct snd_kcontrol_new alc267_quanta_il1_mixer[] = {
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Capture Volume", 0x23, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Mic Capture Switch", 0x23, 2, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
{ }
};
static const struct hda_verb alc267_quanta_il1_verbs[] = {
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN},
{ }
};
static void alc267_quanta_il1_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->ext_mic_pin = 0x18;
spec->int_mic_pin = 0x19;
spec->auto_mic = 1;
spec->automute = 1;
spec->automute_mode = ALC_AUTOMUTE_PIN;
}
/*
* generic initialization of ADC, input mixers and output mixers
*/
static const struct hda_verb alc268_base_init_verbs[] = {
/* Unmute DAC0-1 and set vol = 0 */
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/*
* Set up output mixers (0x0c - 0x0e)
*/
/* set vol=0 to output mixers */
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0e, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* set PCBEEP vol = 0, mute connections */
{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Unmute Selector 23h,24h and set the default input to mic-in */
{0x23, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x24, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{ }
};
/* only for model=test */
#ifdef CONFIG_SND_DEBUG
/*
* generic initialization of ADC, input mixers and output mixers
*/
static const struct hda_verb alc268_volume_init_verbs[] = {
/* set output DAC */
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{ }
};
#endif /* CONFIG_SND_DEBUG */
static const struct snd_kcontrol_new alc268_capture_nosrc_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc268_capture_alt_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
_DEFINE_CAPSRC(1),
{ } /* end */
};
static const struct snd_kcontrol_new alc268_capture_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x24, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x24, 0x0, HDA_OUTPUT),
_DEFINE_CAPSRC(2),
{ } /* end */
};
static const struct hda_input_mux alc268_capture_source = {
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x1 },
{ "Line", 0x2 },
{ "CD", 0x3 },
},
};
static const struct hda_input_mux alc268_acer_capture_source = {
.num_items = 3,
.items = {
{ "Mic", 0x0 },
{ "Internal Mic", 0x1 },
{ "Line", 0x2 },
},
};
static const struct hda_input_mux alc268_acer_dmic_capture_source = {
.num_items = 3,
.items = {
{ "Mic", 0x0 },
{ "Internal Mic", 0x6 },
{ "Line", 0x2 },
},
};
#ifdef CONFIG_SND_DEBUG
static const struct snd_kcontrol_new alc268_test_mixer[] = {
/* Volume widgets */
HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x03, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Mono sum Playback Switch", 0x0e, 1, 2, HDA_INPUT),
HDA_BIND_MUTE("LINE-OUT sum Playback Switch", 0x0f, 2, HDA_INPUT),
HDA_BIND_MUTE("HP-OUT sum Playback Switch", 0x10, 2, HDA_INPUT),
HDA_BIND_MUTE("LINE-OUT Playback Switch", 0x14, 2, HDA_OUTPUT),
HDA_BIND_MUTE("HP-OUT Playback Switch", 0x15, 2, HDA_OUTPUT),
HDA_BIND_MUTE("Mono Playback Switch", 0x16, 2, HDA_OUTPUT),
HDA_CODEC_VOLUME("MIC1 Capture Volume", 0x18, 0x0, HDA_INPUT),
HDA_BIND_MUTE("MIC1 Capture Switch", 0x18, 2, HDA_OUTPUT),
HDA_CODEC_VOLUME("MIC2 Capture Volume", 0x19, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("LINE1 Capture Volume", 0x1a, 0x0, HDA_INPUT),
HDA_BIND_MUTE("LINE1 Capture Switch", 0x1a, 2, HDA_OUTPUT),
/* The below appears problematic on some hardwares */
/*HDA_CODEC_VOLUME("PCBEEP Playback Volume", 0x1d, 0x0, HDA_INPUT),*/
HDA_CODEC_VOLUME("PCM-IN1 Capture Volume", 0x23, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("PCM-IN1 Capture Switch", 0x23, 2, HDA_OUTPUT),
HDA_CODEC_VOLUME("PCM-IN2 Capture Volume", 0x24, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("PCM-IN2 Capture Switch", 0x24, 2, HDA_OUTPUT),
/* Modes for retasking pin widgets */
ALC_PIN_MODE("LINE-OUT pin mode", 0x14, ALC_PIN_DIR_INOUT),
ALC_PIN_MODE("HP-OUT pin mode", 0x15, ALC_PIN_DIR_INOUT),
ALC_PIN_MODE("MIC1 pin mode", 0x18, ALC_PIN_DIR_INOUT),
ALC_PIN_MODE("LINE1 pin mode", 0x1a, ALC_PIN_DIR_INOUT),
/* 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 SPDIF output pin to be enabled.
* The ALC268 does not have an SPDIF input.
*/
ALC_SPDIF_CTRL_SWITCH("SPDIF Playback Switch", 0x06, 0x01),
/* A switch allowing EAPD to be enabled. Some laptops seem to use
* this output to turn on an external amplifier.
*/
ALC_EAPD_CTRL_SWITCH("LINE-OUT EAPD Enable Switch", 0x0f, 0x02),
ALC_EAPD_CTRL_SWITCH("HP-OUT EAPD Enable Switch", 0x10, 0x02),
{ } /* end */
};
#endif
/*
* configuration and preset
*/
static const char * const alc268_models[ALC268_MODEL_LAST] = {
[ALC267_QUANTA_IL1] = "quanta-il1",
[ALC268_3ST] = "3stack",
[ALC268_TOSHIBA] = "toshiba",
[ALC268_ACER] = "acer",
[ALC268_ACER_DMIC] = "acer-dmic",
[ALC268_ACER_ASPIRE_ONE] = "acer-aspire",
[ALC268_DELL] = "dell",
[ALC268_ZEPTO] = "zepto",
#ifdef CONFIG_SND_DEBUG
[ALC268_TEST] = "test",
#endif
[ALC268_AUTO] = "auto",
};
static const struct snd_pci_quirk alc268_cfg_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x011e, "Acer Aspire 5720z", ALC268_ACER),
SND_PCI_QUIRK(0x1025, 0x0126, "Acer", ALC268_ACER),
SND_PCI_QUIRK(0x1025, 0x012e, "Acer Aspire 5310", ALC268_ACER),
SND_PCI_QUIRK(0x1025, 0x0130, "Acer Extensa 5210", ALC268_ACER),
SND_PCI_QUIRK(0x1025, 0x0136, "Acer Aspire 5315", ALC268_ACER),
SND_PCI_QUIRK(0x1025, 0x015b, "Acer Aspire One",
ALC268_ACER_ASPIRE_ONE),
SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL),
SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron 910", ALC268_AUTO),
SND_PCI_QUIRK_MASK(0x1028, 0xfff0, 0x02b0,
"Dell Inspiron Mini9/Vostro A90", ALC268_DELL),
/* almost compatible with toshiba but with optional digital outs;
* auto-probing seems working fine
*/
SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP TX25xx series",
ALC268_AUTO),
SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO),
SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA),
SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1),
{}
};
/* Toshiba laptops have no unique PCI SSID but only codec SSID */
static const struct snd_pci_quirk alc268_ssid_cfg_tbl[] = {
SND_PCI_QUIRK(0x1179, 0xff0a, "TOSHIBA X-200", ALC268_AUTO),
SND_PCI_QUIRK(0x1179, 0xff0e, "TOSHIBA X-200 HDMI", ALC268_AUTO),
SND_PCI_QUIRK_MASK(0x1179, 0xff00, 0xff00, "TOSHIBA A/Lx05",
ALC268_TOSHIBA),
{}
};
static const struct alc_config_preset alc268_presets[] = {
[ALC267_QUANTA_IL1] = {
.mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer,
alc268_capture_nosrc_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc267_quanta_il1_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
.unsol_event = alc_sku_unsol_event,
.setup = alc267_quanta_il1_setup,
.init_hook = alc_inithook,
},
[ALC268_3ST] = {
.mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
alc268_beep_mixer },
.init_verbs = { alc268_base_init_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
.capsrc_nids = alc268_capsrc_nids,
.hp_nid = 0x03,
.dig_out_nid = ALC268_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
.input_mux = &alc268_capture_source,
},
[ALC268_TOSHIBA] = {
.mixers = { alc268_toshiba_mixer, alc268_capture_alt_mixer,
alc268_beep_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_toshiba_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
.capsrc_nids = alc268_capsrc_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
.input_mux = &alc268_capture_source,
.unsol_event = alc_sku_unsol_event,
.setup = alc268_toshiba_setup,
.init_hook = alc_inithook,
},
[ALC268_ACER] = {
.mixers = { alc268_acer_mixer, alc268_capture_alt_mixer,
alc268_beep_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_acer_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
.capsrc_nids = alc268_capsrc_nids,
.hp_nid = 0x02,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
.input_mux = &alc268_acer_capture_source,
.unsol_event = alc_sku_unsol_event,
.setup = alc268_acer_setup,
.init_hook = alc_inithook,
},
[ALC268_ACER_DMIC] = {
.mixers = { alc268_acer_dmic_mixer, alc268_capture_alt_mixer,
alc268_beep_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_acer_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
.capsrc_nids = alc268_capsrc_nids,
.hp_nid = 0x02,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
.input_mux = &alc268_acer_dmic_capture_source,
.unsol_event = alc_sku_unsol_event,
.setup = alc268_acer_setup,
.init_hook = alc_inithook,
},
[ALC268_ACER_ASPIRE_ONE] = {
.mixers = { alc268_acer_aspire_one_mixer,
alc268_beep_mixer,
alc268_capture_nosrc_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_acer_aspire_one_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
.capsrc_nids = alc268_capsrc_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
.unsol_event = alc_sku_unsol_event,
.setup = alc268_acer_lc_setup,
.init_hook = alc_inithook,
},
[ALC268_DELL] = {
.mixers = { alc268_dell_mixer, alc268_beep_mixer,
alc268_capture_nosrc_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_dell_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
.capsrc_nids = alc268_capsrc_nids,
.hp_nid = 0x02,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
.unsol_event = alc_sku_unsol_event,
.setup = alc268_dell_setup,
.init_hook = alc_inithook,
},
[ALC268_ZEPTO] = {
.mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
alc268_beep_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_toshiba_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
.capsrc_nids = alc268_capsrc_nids,
.hp_nid = 0x03,
.dig_out_nid = ALC268_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
.input_mux = &alc268_capture_source,
.unsol_event = alc_sku_unsol_event,
.setup = alc268_toshiba_setup,
.init_hook = alc_inithook,
},
#ifdef CONFIG_SND_DEBUG
[ALC268_TEST] = {
.mixers = { alc268_test_mixer, alc268_capture_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_volume_init_verbs,
alc268_beep_init_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
.capsrc_nids = alc268_capsrc_nids,
.hp_nid = 0x03,
.dig_out_nid = ALC268_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
.input_mux = &alc268_capture_source,
},
#endif
};

681
sound/pci/hda/alc269_quirks.c Normal file
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@ -0,0 +1,681 @@
/*
* ALC269/ALC270/ALC275/ALC276 quirk models
* included by patch_realtek.c
*/
/* ALC269 models */
enum {
ALC269_AUTO,
ALC269_BASIC,
ALC269_QUANTA_FL1,
ALC269_AMIC,
ALC269_DMIC,
ALC269VB_AMIC,
ALC269VB_DMIC,
ALC269_FUJITSU,
ALC269_LIFEBOOK,
ALC271_ACER,
ALC269_MODEL_LAST /* last tag */
};
/*
* ALC269 channel source setting (2 channel)
*/
#define ALC269_DIGOUT_NID ALC880_DIGOUT_NID
#define alc269_dac_nids alc260_dac_nids
static const hda_nid_t alc269_adc_nids[1] = {
/* ADC1 */
0x08,
};
static const hda_nid_t alc269_capsrc_nids[1] = {
0x23,
};
static const hda_nid_t alc269vb_adc_nids[1] = {
/* ADC1 */
0x09,
};
static const hda_nid_t alc269vb_capsrc_nids[1] = {
0x22,
};
#define alc269_modes alc260_modes
#define alc269_capture_source alc880_lg_lw_capture_source
static const struct snd_kcontrol_new alc269_base_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = {
/* output mixer control */
HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Master Playback Switch",
.subdevice = HDA_SUBDEV_AMP_FLAG,
.info = snd_hda_mixer_amp_switch_info,
.get = snd_hda_mixer_amp_switch_get,
.put = alc268_acer_master_sw_put,
.private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
},
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
{ }
};
static const struct snd_kcontrol_new alc269_lifebook_mixer[] = {
/* output mixer control */
HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Master Playback Switch",
.subdevice = HDA_SUBDEV_AMP_FLAG,
.info = snd_hda_mixer_amp_switch_info,
.get = snd_hda_mixer_amp_switch_get,
.put = alc268_acer_master_sw_put,
.private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
},
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x0b, 0x03, HDA_INPUT),
HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x0b, 0x03, HDA_INPUT),
HDA_CODEC_VOLUME("Dock Mic Boost Volume", 0x1b, 0, HDA_INPUT),
{ }
};
static const struct snd_kcontrol_new alc269_laptop_mixer[] = {
HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc269vb_laptop_mixer[] = {
HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc269_asus_mixer[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Master Playback Switch", 0x0c, 0x0, HDA_INPUT),
{ } /* end */
};
/* capture mixer elements */
static const struct snd_kcontrol_new alc269_laptop_analog_capture_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc269_laptop_digital_capture_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc269vb_laptop_analog_capture_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc269vb_laptop_digital_capture_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
{ } /* end */
};
/* FSC amilo */
#define alc269_fujitsu_mixer alc269_laptop_mixer
static const struct hda_verb alc269_quanta_fl1_verbs[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{ }
};
static const struct hda_verb alc269_lifebook_verbs[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{ }
};
/* toggle speaker-output according to the hp-jack state */
static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec)
{
alc_hp_automute(codec);
snd_hda_codec_write(codec, 0x20, 0,
AC_VERB_SET_COEF_INDEX, 0x0c);
snd_hda_codec_write(codec, 0x20, 0,
AC_VERB_SET_PROC_COEF, 0x680);
snd_hda_codec_write(codec, 0x20, 0,
AC_VERB_SET_COEF_INDEX, 0x0c);
snd_hda_codec_write(codec, 0x20, 0,
AC_VERB_SET_PROC_COEF, 0x480);
}
#define alc269_lifebook_speaker_automute \
alc269_quanta_fl1_speaker_automute
static void alc269_lifebook_mic_autoswitch(struct hda_codec *codec)
{
unsigned int present_laptop;
unsigned int present_dock;
present_laptop = snd_hda_jack_detect(codec, 0x18);
present_dock = snd_hda_jack_detect(codec, 0x1b);
/* Laptop mic port overrides dock mic port, design decision */
if (present_dock)
snd_hda_codec_write(codec, 0x23, 0,
AC_VERB_SET_CONNECT_SEL, 0x3);
if (present_laptop)
snd_hda_codec_write(codec, 0x23, 0,
AC_VERB_SET_CONNECT_SEL, 0x0);
if (!present_dock && !present_laptop)
snd_hda_codec_write(codec, 0x23, 0,
AC_VERB_SET_CONNECT_SEL, 0x1);
}
static void alc269_quanta_fl1_unsol_event(struct hda_codec *codec,
unsigned int res)
{
switch (res >> 26) {
case ALC_HP_EVENT:
alc269_quanta_fl1_speaker_automute(codec);
break;
case ALC_MIC_EVENT:
alc_mic_automute(codec);
break;
}
}
static void alc269_lifebook_unsol_event(struct hda_codec *codec,
unsigned int res)
{
if ((res >> 26) == ALC_HP_EVENT)
alc269_lifebook_speaker_automute(codec);
if ((res >> 26) == ALC_MIC_EVENT)
alc269_lifebook_mic_autoswitch(codec);
}
static void alc269_quanta_fl1_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->automute_mixer_nid[0] = 0x0c;
spec->automute = 1;
spec->automute_mode = ALC_AUTOMUTE_MIXER;
spec->ext_mic_pin = 0x18;
spec->int_mic_pin = 0x19;
spec->auto_mic = 1;
}
static void alc269_quanta_fl1_init_hook(struct hda_codec *codec)
{
alc269_quanta_fl1_speaker_automute(codec);
alc_mic_automute(codec);
}
static void alc269_lifebook_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.hp_pins[1] = 0x1a;
spec->autocfg.speaker_pins[0] = 0x14;
spec->automute_mixer_nid[0] = 0x0c;
spec->automute = 1;
spec->automute_mode = ALC_AUTOMUTE_MIXER;
}
static void alc269_lifebook_init_hook(struct hda_codec *codec)
{
alc269_lifebook_speaker_automute(codec);
alc269_lifebook_mic_autoswitch(codec);
}
static const struct hda_verb alc269_laptop_dmic_init_verbs[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x23, AC_VERB_SET_CONNECT_SEL, 0x05},
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 },
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7019 | (0x00 << 8))},
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
{}
};
static const struct hda_verb alc269_laptop_amic_init_verbs[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x23, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 },
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x701b | (0x00 << 8))},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
{}
};
static const struct hda_verb alc269vb_laptop_dmic_init_verbs[] = {
{0x21, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x22, AC_VERB_SET_CONNECT_SEL, 0x06},
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 },
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7019 | (0x00 << 8))},
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
{}
};
static const struct hda_verb alc269vb_laptop_amic_init_verbs[] = {
{0x21, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x22, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 },
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7019 | (0x00 << 8))},
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
{}
};
static const struct hda_verb alc271_acer_dmic_verbs[] = {
{0x20, AC_VERB_SET_COEF_INDEX, 0x0d},
{0x20, AC_VERB_SET_PROC_COEF, 0x4000},
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x21, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
{0x22, AC_VERB_SET_CONNECT_SEL, 6},
{ }
};
static void alc269_laptop_amic_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->automute_mixer_nid[0] = 0x0c;
spec->automute = 1;
spec->automute_mode = ALC_AUTOMUTE_MIXER;
spec->ext_mic_pin = 0x18;
spec->int_mic_pin = 0x19;
spec->auto_mic = 1;
}
static void alc269_laptop_dmic_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->automute_mixer_nid[0] = 0x0c;
spec->automute = 1;
spec->automute_mode = ALC_AUTOMUTE_MIXER;
spec->ext_mic_pin = 0x18;
spec->int_mic_pin = 0x12;
spec->auto_mic = 1;
}
static void alc269vb_laptop_amic_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x21;
spec->autocfg.speaker_pins[0] = 0x14;
spec->automute_mixer_nid[0] = 0x0c;
spec->automute = 1;
spec->automute_mode = ALC_AUTOMUTE_MIXER;
spec->ext_mic_pin = 0x18;
spec->int_mic_pin = 0x19;
spec->auto_mic = 1;
}
static void alc269vb_laptop_dmic_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x21;
spec->autocfg.speaker_pins[0] = 0x14;
spec->automute_mixer_nid[0] = 0x0c;
spec->automute = 1;
spec->automute_mode = ALC_AUTOMUTE_MIXER;
spec->ext_mic_pin = 0x18;
spec->int_mic_pin = 0x12;
spec->auto_mic = 1;
}
/*
* generic initialization of ADC, input mixers and output mixers
*/
static const struct hda_verb alc269_init_verbs[] = {
/*
* Unmute ADC0 and set the default input to mic-in
*/
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/*
* Set up output mixers (0x02 - 0x03)
*/
/* set vol=0 to output mixers */
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* set up input amps for analog loopback */
/* Amp Indices: DAC = 0, mixer = 1 */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* FIXME: use Mux-type input source selection */
/* Mixer elements: 0x18, 19, 1a, 1b, 1d, 0b */
/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
{0x23, AC_VERB_SET_CONNECT_SEL, 0x00},
/* set EAPD */
{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
{ }
};
static const struct hda_verb alc269vb_init_verbs[] = {
/*
* Unmute ADC0 and set the default input to mic-in
*/
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/*
* Set up output mixers (0x02 - 0x03)
*/
/* set vol=0 to output mixers */
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* set up input amps for analog loopback */
/* Amp Indices: DAC = 0, mixer = 1 */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* FIXME: use Mux-type input source selection */
/* Mixer elements: 0x18, 19, 1a, 1b, 1d, 0b */
/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
{0x22, AC_VERB_SET_CONNECT_SEL, 0x00},
/* set EAPD */
{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
{ }
};
/*
* configuration and preset
*/
static const char * const alc269_models[ALC269_MODEL_LAST] = {
[ALC269_BASIC] = "basic",
[ALC269_QUANTA_FL1] = "quanta",
[ALC269_AMIC] = "laptop-amic",
[ALC269_DMIC] = "laptop-dmic",
[ALC269_FUJITSU] = "fujitsu",
[ALC269_LIFEBOOK] = "lifebook",
[ALC269_AUTO] = "auto",
};
static const struct snd_pci_quirk alc269_cfg_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_QUANTA_FL1),
SND_PCI_QUIRK(0x1025, 0x047c, "ACER ZGA", ALC271_ACER),
SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A",
ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1013, "ASUS N61Da", ALC269VB_AMIC),
SND_PCI_QUIRK(0x1043, 0x1113, "ASUS N63Jn", ALC269VB_AMIC),
SND_PCI_QUIRK(0x1043, 0x1143, "ASUS B53f", ALC269VB_AMIC),
SND_PCI_QUIRK(0x1043, 0x1133, "ASUS UJ20ft", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1183, "ASUS K72DR", ALC269VB_AMIC),
SND_PCI_QUIRK(0x1043, 0x11b3, "ASUS K52DR", ALC269VB_AMIC),
SND_PCI_QUIRK(0x1043, 0x11e3, "ASUS U33Jc", ALC269VB_AMIC),
SND_PCI_QUIRK(0x1043, 0x1273, "ASUS UL80Jt", ALC269VB_AMIC),
SND_PCI_QUIRK(0x1043, 0x1283, "ASUS U53Jc", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS N82JV", ALC269VB_AMIC),
SND_PCI_QUIRK(0x1043, 0x12d3, "ASUS N61Jv", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x13a3, "ASUS UL30Vt", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1373, "ASUS G73JX", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1383, "ASUS UJ30Jc", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x13d3, "ASUS N61JA", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1413, "ASUS UL50", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1443, "ASUS UL30", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1453, "ASUS M60Jv", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1483, "ASUS UL80", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x14f3, "ASUS F83Vf", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x14e3, "ASUS UL20", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1513, "ASUS UX30", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1593, "ASUS N51Vn", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x15a3, "ASUS N60Jv", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x15b3, "ASUS N60Dp", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x15c3, "ASUS N70De", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x15e3, "ASUS F83T", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1643, "ASUS M60J", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1693, "ASUS F50N", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_DMIC),
SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x831a, "ASUS Eeepc P901",
ALC269_DMIC),
SND_PCI_QUIRK(0x1043, 0x834a, "ASUS Eeepc S101",
ALC269_DMIC),
SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_DMIC),
SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_DMIC),
SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_AUTO),
SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK),
SND_PCI_QUIRK(0x152d, 0x1778, "Quanta ON1", ALC269_DMIC),
SND_PCI_QUIRK(0x1734, 0x115d, "FSC Amilo", ALC269_FUJITSU),
SND_PCI_QUIRK(0x17aa, 0x3be9, "Quanta Wistron", ALC269_AMIC),
SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_AMIC),
SND_PCI_QUIRK(0x17ff, 0x059a, "Quanta EL3", ALC269_DMIC),
SND_PCI_QUIRK(0x17ff, 0x059b, "Quanta JR1", ALC269_DMIC),
{}
};
static const struct alc_config_preset alc269_presets[] = {
[ALC269_BASIC] = {
.mixers = { alc269_base_mixer },
.init_verbs = { alc269_init_verbs },
.num_dacs = ARRAY_SIZE(alc269_dac_nids),
.dac_nids = alc269_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc269_modes),
.channel_mode = alc269_modes,
.input_mux = &alc269_capture_source,
},
[ALC269_QUANTA_FL1] = {
.mixers = { alc269_quanta_fl1_mixer },
.init_verbs = { alc269_init_verbs, alc269_quanta_fl1_verbs },
.num_dacs = ARRAY_SIZE(alc269_dac_nids),
.dac_nids = alc269_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc269_modes),
.channel_mode = alc269_modes,
.input_mux = &alc269_capture_source,
.unsol_event = alc269_quanta_fl1_unsol_event,
.setup = alc269_quanta_fl1_setup,
.init_hook = alc269_quanta_fl1_init_hook,
},
[ALC269_AMIC] = {
.mixers = { alc269_laptop_mixer },
.cap_mixer = alc269_laptop_analog_capture_mixer,
.init_verbs = { alc269_init_verbs,
alc269_laptop_amic_init_verbs },
.num_dacs = ARRAY_SIZE(alc269_dac_nids),
.dac_nids = alc269_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc269_modes),
.channel_mode = alc269_modes,
.unsol_event = alc_sku_unsol_event,
.setup = alc269_laptop_amic_setup,
.init_hook = alc_inithook,
},
[ALC269_DMIC] = {
.mixers = { alc269_laptop_mixer },
.cap_mixer = alc269_laptop_digital_capture_mixer,
.init_verbs = { alc269_init_verbs,
alc269_laptop_dmic_init_verbs },
.num_dacs = ARRAY_SIZE(alc269_dac_nids),
.dac_nids = alc269_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc269_modes),
.channel_mode = alc269_modes,
.unsol_event = alc_sku_unsol_event,
.setup = alc269_laptop_dmic_setup,
.init_hook = alc_inithook,
},
[ALC269VB_AMIC] = {
.mixers = { alc269vb_laptop_mixer },
.cap_mixer = alc269vb_laptop_analog_capture_mixer,
.init_verbs = { alc269vb_init_verbs,
alc269vb_laptop_amic_init_verbs },
.num_dacs = ARRAY_SIZE(alc269_dac_nids),
.dac_nids = alc269_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc269_modes),
.channel_mode = alc269_modes,
.unsol_event = alc_sku_unsol_event,
.setup = alc269vb_laptop_amic_setup,
.init_hook = alc_inithook,
},
[ALC269VB_DMIC] = {
.mixers = { alc269vb_laptop_mixer },
.cap_mixer = alc269vb_laptop_digital_capture_mixer,
.init_verbs = { alc269vb_init_verbs,
alc269vb_laptop_dmic_init_verbs },
.num_dacs = ARRAY_SIZE(alc269_dac_nids),
.dac_nids = alc269_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc269_modes),
.channel_mode = alc269_modes,
.unsol_event = alc_sku_unsol_event,
.setup = alc269vb_laptop_dmic_setup,
.init_hook = alc_inithook,
},
[ALC269_FUJITSU] = {
.mixers = { alc269_fujitsu_mixer },
.cap_mixer = alc269_laptop_digital_capture_mixer,
.init_verbs = { alc269_init_verbs,
alc269_laptop_dmic_init_verbs },
.num_dacs = ARRAY_SIZE(alc269_dac_nids),
.dac_nids = alc269_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc269_modes),
.channel_mode = alc269_modes,
.unsol_event = alc_sku_unsol_event,
.setup = alc269_laptop_dmic_setup,
.init_hook = alc_inithook,
},
[ALC269_LIFEBOOK] = {
.mixers = { alc269_lifebook_mixer },
.init_verbs = { alc269_init_verbs, alc269_lifebook_verbs },
.num_dacs = ARRAY_SIZE(alc269_dac_nids),
.dac_nids = alc269_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc269_modes),
.channel_mode = alc269_modes,
.input_mux = &alc269_capture_source,
.unsol_event = alc269_lifebook_unsol_event,
.setup = alc269_lifebook_setup,
.init_hook = alc269_lifebook_init_hook,
},
[ALC271_ACER] = {
.mixers = { alc269_asus_mixer },
.cap_mixer = alc269vb_laptop_digital_capture_mixer,
.init_verbs = { alc269_init_verbs, alc271_acer_dmic_verbs },
.num_dacs = ARRAY_SIZE(alc269_dac_nids),
.dac_nids = alc269_dac_nids,
.adc_nids = alc262_dmic_adc_nids,
.num_adc_nids = ARRAY_SIZE(alc262_dmic_adc_nids),
.capsrc_nids = alc262_dmic_capsrc_nids,
.num_channel_mode = ARRAY_SIZE(alc269_modes),
.channel_mode = alc269_modes,
.input_mux = &alc269_capture_source,
.dig_out_nid = ALC880_DIGOUT_NID,
.unsol_event = alc_sku_unsol_event,
.setup = alc269vb_laptop_dmic_setup,
.init_hook = alc_inithook,
},
};

1408
sound/pci/hda/alc662_quirks.c Normal file
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222
sound/pci/hda/alc680_quirks.c Normal file
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/*
* ALC680 quirk models
* included by patch_realtek.c
*/
/* ALC680 models */
enum {
ALC680_AUTO,
ALC680_BASE,
ALC680_MODEL_LAST,
};
#define ALC680_DIGIN_NID ALC880_DIGIN_NID
#define ALC680_DIGOUT_NID ALC880_DIGOUT_NID
#define alc680_modes alc260_modes
static const hda_nid_t alc680_dac_nids[3] = {
/* Lout1, Lout2, hp */
0x02, 0x03, 0x04
};
static const hda_nid_t alc680_adc_nids[3] = {
/* ADC0-2 */
/* DMIC, MIC, Line-in*/
0x07, 0x08, 0x09
};
/*
* Analog capture ADC cgange
*/
static hda_nid_t alc680_get_cur_adc(struct hda_codec *codec)
{
static hda_nid_t pins[] = {0x18, 0x19};
static hda_nid_t adcs[] = {0x08, 0x09};
int i;
for (i = 0; i < ARRAY_SIZE(pins); i++) {
if (!is_jack_detectable(codec, pins[i]))
continue;
if (snd_hda_jack_detect(codec, pins[i]))
return adcs[i];
}
return 0x07;
}
static void alc680_rec_autoswitch(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
hda_nid_t nid = alc680_get_cur_adc(codec);
if (spec->cur_adc && nid != spec->cur_adc) {
__snd_hda_codec_cleanup_stream(codec, spec->cur_adc, 1);
spec->cur_adc = nid;
snd_hda_codec_setup_stream(codec, nid,
spec->cur_adc_stream_tag, 0,
spec->cur_adc_format);
}
}
static int alc680_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
unsigned int stream_tag,
unsigned int format,
struct snd_pcm_substream *substream)
{
struct alc_spec *spec = codec->spec;
hda_nid_t nid = alc680_get_cur_adc(codec);
spec->cur_adc = nid;
spec->cur_adc_stream_tag = stream_tag;
spec->cur_adc_format = format;
snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format);
return 0;
}
static int alc680_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
struct alc_spec *spec = codec->spec;
snd_hda_codec_cleanup_stream(codec, spec->cur_adc);
spec->cur_adc = 0;
return 0;
}
static const struct hda_pcm_stream alc680_pcm_analog_auto_capture = {
.substreams = 1, /* can be overridden */
.channels_min = 2,
.channels_max = 2,
/* NID is set in alc_build_pcms */
.ops = {
.prepare = alc680_capture_pcm_prepare,
.cleanup = alc680_capture_pcm_cleanup
},
};
static const struct snd_kcontrol_new alc680_base_mixer[] = {
/* output mixer control */
HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x4, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x16, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x12, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Line In Boost Volume", 0x19, 0, HDA_INPUT),
{ }
};
static const struct hda_bind_ctls alc680_bind_cap_vol = {
.ops = &snd_hda_bind_vol,
.values = {
HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT),
HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
0
},
};
static const struct hda_bind_ctls alc680_bind_cap_switch = {
.ops = &snd_hda_bind_sw,
.values = {
HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT),
HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
0
},
};
static const struct snd_kcontrol_new alc680_master_capture_mixer[] = {
HDA_BIND_VOL("Capture Volume", &alc680_bind_cap_vol),
HDA_BIND_SW("Capture Switch", &alc680_bind_cap_switch),
{ } /* end */
};
/*
* generic initialization of ADC, input mixers and output mixers
*/
static const struct hda_verb alc680_init_verbs[] = {
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x16, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN},
{0x19, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN},
{ }
};
/* toggle speaker-output according to the hp-jack state */
static void alc680_base_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x16;
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x15;
spec->autocfg.num_inputs = 2;
spec->autocfg.inputs[0].pin = 0x18;
spec->autocfg.inputs[0].type = AUTO_PIN_MIC;
spec->autocfg.inputs[1].pin = 0x19;
spec->autocfg.inputs[1].type = AUTO_PIN_LINE_IN;
spec->automute = 1;
spec->automute_mode = ALC_AUTOMUTE_AMP;
}
static void alc680_unsol_event(struct hda_codec *codec,
unsigned int res)
{
if ((res >> 26) == ALC_HP_EVENT)
alc_hp_automute(codec);
if ((res >> 26) == ALC_MIC_EVENT)
alc680_rec_autoswitch(codec);
}
static void alc680_inithook(struct hda_codec *codec)
{
alc_hp_automute(codec);
alc680_rec_autoswitch(codec);
}
/*
* configuration and preset
*/
static const char * const alc680_models[ALC680_MODEL_LAST] = {
[ALC680_BASE] = "base",
[ALC680_AUTO] = "auto",
};
static const struct snd_pci_quirk alc680_cfg_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x12f3, "ASUS NX90", ALC680_BASE),
{}
};
static const struct alc_config_preset alc680_presets[] = {
[ALC680_BASE] = {
.mixers = { alc680_base_mixer },
.cap_mixer = alc680_master_capture_mixer,
.init_verbs = { alc680_init_verbs },
.num_dacs = ARRAY_SIZE(alc680_dac_nids),
.dac_nids = alc680_dac_nids,
.dig_out_nid = ALC680_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc680_modes),
.channel_mode = alc680_modes,
.unsol_event = alc680_unsol_event,
.setup = alc680_base_setup,
.init_hook = alc680_inithook,
},
};

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sound/pci/hda/alc861_quirks.c Normal file
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/*
* ALC660/ALC861 quirk models
* included by patch_realtek.c
*/
/* ALC861 models */
enum {
ALC861_AUTO,
ALC861_3ST,
ALC660_3ST,
ALC861_3ST_DIG,
ALC861_6ST_DIG,
ALC861_UNIWILL_M31,
ALC861_TOSHIBA,
ALC861_ASUS,
ALC861_ASUS_LAPTOP,
ALC861_MODEL_LAST,
};
/*
* ALC861 channel source setting (2/6 channel selection for 3-stack)
*/
/*
* set the path ways for 2 channel output
* need to set the codec line out and mic 1 pin widgets to inputs
*/
static const struct hda_verb alc861_threestack_ch2_init[] = {
/* set pin widget 1Ah (line in) for input */
{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* set pin widget 18h (mic1/2) for input, for mic also enable
* the vref
*/
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c },
#if 0
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8)) }, /*line-in*/
#endif
{ } /* end */
};
/*
* 6ch mode
* need to set the codec line out and mic 1 pin widgets to outputs
*/
static const struct hda_verb alc861_threestack_ch6_init[] = {
/* set pin widget 1Ah (line in) for output (Back Surround)*/
{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
/* set pin widget 18h (mic1) for output (CLFE)*/
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x0c, AC_VERB_SET_CONNECT_SEL, 0x00 },
{ 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00 },
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
#if 0
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8)) }, /*line in*/
#endif
{ } /* end */
};
static const struct hda_channel_mode alc861_threestack_modes[2] = {
{ 2, alc861_threestack_ch2_init },
{ 6, alc861_threestack_ch6_init },
};
/* Set mic1 as input and unmute the mixer */
static const struct hda_verb alc861_uniwill_m31_ch2_init[] = {
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/
{ } /* end */
};
/* Set mic1 as output and mute mixer */
static const struct hda_verb alc861_uniwill_m31_ch4_init[] = {
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
{ } /* end */
};
static const struct hda_channel_mode alc861_uniwill_m31_modes[2] = {
{ 2, alc861_uniwill_m31_ch2_init },
{ 4, alc861_uniwill_m31_ch4_init },
};
/* Set mic1 and line-in as input and unmute the mixer */
static const struct hda_verb alc861_asus_ch2_init[] = {
/* set pin widget 1Ah (line in) for input */
{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* set pin widget 18h (mic1/2) for input, for mic also enable
* the vref
*/
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c },
#if 0
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8)) }, /*line-in*/
#endif
{ } /* end */
};
/* Set mic1 nad line-in as output and mute mixer */
static const struct hda_verb alc861_asus_ch6_init[] = {
/* set pin widget 1Ah (line in) for output (Back Surround)*/
{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
/* { 0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, */
/* set pin widget 18h (mic1) for output (CLFE)*/
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
/* { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, */
{ 0x0c, AC_VERB_SET_CONNECT_SEL, 0x00 },
{ 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00 },
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
#if 0
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8)) }, /*line in*/
#endif
{ } /* end */
};
static const struct hda_channel_mode alc861_asus_modes[2] = {
{ 2, alc861_asus_ch2_init },
{ 6, alc861_asus_ch6_init },
};
/* patch-ALC861 */
static const struct snd_kcontrol_new alc861_base_mixer[] = {
/* output mixer control */
HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT),
/*Input mixer control */
/* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */
HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc861_3ST_mixer[] = {
/* output mixer control */
HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
/*HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT), */
/* Input mixer control */
/* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */
HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
.private_value = ARRAY_SIZE(alc861_threestack_modes),
},
{ } /* end */
};
static const struct snd_kcontrol_new alc861_toshiba_mixer[] = {
/* output mixer control */
HDA_CODEC_MUTE("Master Playback Switch", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc861_uniwill_m31_mixer[] = {
/* output mixer control */
HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
/*HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT), */
/* Input mixer control */
/* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */
HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
.private_value = ARRAY_SIZE(alc861_uniwill_m31_modes),
},
{ } /* end */
};
static const struct snd_kcontrol_new alc861_asus_mixer[] = {
/* output mixer control */
HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT),
/* Input mixer control */
HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_OUTPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
.private_value = ARRAY_SIZE(alc861_asus_modes),
},
{ }
};
/* additional mixer */
static const struct snd_kcontrol_new alc861_asus_laptop_mixer[] = {
HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
{ }
};
/*
* generic initialization of ADC, input mixers and output mixers
*/
static const struct hda_verb alc861_base_init_verbs[] = {
/*
* Unmute ADC0 and set the default input to mic-in
*/
/* port-A for surround (rear panel) */
{ 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x0e, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-B for mic-in (rear panel) with vref */
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* port-C for line-in (rear panel) */
{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* port-D for Front */
{ 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-E for HP out (front panel) */
{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
/* route front PCM to HP */
{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-F for mic-in (front panel) with vref */
{ 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* port-G for CLFE (rear panel) */
{ 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x1f, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-H for side (rear panel) */
{ 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x20, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* CD-in */
{ 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* route front mic to ADC1*/
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Unmute DAC0~3 & spdif out*/
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute Mixer 14 (mic) 1c (Line in)*/
{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* Unmute Stereo Mixer 15 */
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* hp used DAC 3 (Front) */
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{ }
};
static const struct hda_verb alc861_threestack_init_verbs[] = {
/*
* Unmute ADC0 and set the default input to mic-in
*/
/* port-A for surround (rear panel) */
{ 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
/* port-B for mic-in (rear panel) with vref */
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* port-C for line-in (rear panel) */
{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* port-D for Front */
{ 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-E for HP out (front panel) */
{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
/* route front PCM to HP */
{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-F for mic-in (front panel) with vref */
{ 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* port-G for CLFE (rear panel) */
{ 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
/* port-H for side (rear panel) */
{ 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
/* CD-in */
{ 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* route front mic to ADC1*/
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Unmute DAC0~3 & spdif out*/
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute Mixer 14 (mic) 1c (Line in)*/
{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* Unmute Stereo Mixer 15 */
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* hp used DAC 3 (Front) */
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{ }
};
static const struct hda_verb alc861_uniwill_m31_init_verbs[] = {
/*
* Unmute ADC0 and set the default input to mic-in
*/
/* port-A for surround (rear panel) */
{ 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
/* port-B for mic-in (rear panel) with vref */
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* port-C for line-in (rear panel) */
{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* port-D for Front */
{ 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-E for HP out (front panel) */
/* this has to be set to VREF80 */
{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* route front PCM to HP */
{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-F for mic-in (front panel) with vref */
{ 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* port-G for CLFE (rear panel) */
{ 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
/* port-H for side (rear panel) */
{ 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
/* CD-in */
{ 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* route front mic to ADC1*/
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Unmute DAC0~3 & spdif out*/
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute Mixer 14 (mic) 1c (Line in)*/
{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* Unmute Stereo Mixer 15 */
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* hp used DAC 3 (Front) */
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{ }
};
static const struct hda_verb alc861_asus_init_verbs[] = {
/*
* Unmute ADC0 and set the default input to mic-in
*/
/* port-A for surround (rear panel)
* according to codec#0 this is the HP jack
*/
{ 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, /* was 0x00 */
/* route front PCM to HP */
{ 0x0e, AC_VERB_SET_CONNECT_SEL, 0x01 },
/* port-B for mic-in (rear panel) with vref */
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* port-C for line-in (rear panel) */
{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* port-D for Front */
{ 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-E for HP out (front panel) */
/* this has to be set to VREF80 */
{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* route front PCM to HP */
{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-F for mic-in (front panel) with vref */
{ 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* port-G for CLFE (rear panel) */
{ 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
/* port-H for side (rear panel) */
{ 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
/* CD-in */
{ 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* route front mic to ADC1*/
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Unmute DAC0~3 & spdif out*/
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute Mixer 14 (mic) 1c (Line in)*/
{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* Unmute Stereo Mixer 15 */
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* hp used DAC 3 (Front) */
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{ }
};
/* additional init verbs for ASUS laptops */
static const struct hda_verb alc861_asus_laptop_init_verbs[] = {
{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x45 }, /* HP-out */
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2) }, /* mute line-in */
{ }
};
static const struct hda_verb alc861_toshiba_init_verbs[] = {
{0x0f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
{ }
};
/* toggle speaker-output according to the hp-jack state */
static void alc861_toshiba_automute(struct hda_codec *codec)
{
unsigned int present = snd_hda_jack_detect(codec, 0x0f);
snd_hda_codec_amp_stereo(codec, 0x16, HDA_INPUT, 0,
HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
snd_hda_codec_amp_stereo(codec, 0x1a, HDA_INPUT, 3,
HDA_AMP_MUTE, present ? 0 : HDA_AMP_MUTE);
}
static void alc861_toshiba_unsol_event(struct hda_codec *codec,
unsigned int res)
{
if ((res >> 26) == ALC_HP_EVENT)
alc861_toshiba_automute(codec);
}
#define ALC861_DIGOUT_NID 0x07
static const struct hda_channel_mode alc861_8ch_modes[1] = {
{ 8, NULL }
};
static const hda_nid_t alc861_dac_nids[4] = {
/* front, surround, clfe, side */
0x03, 0x06, 0x05, 0x04
};
static const hda_nid_t alc660_dac_nids[3] = {
/* front, clfe, surround */
0x03, 0x05, 0x06
};
static const hda_nid_t alc861_adc_nids[1] = {
/* ADC0-2 */
0x08,
};
static const struct hda_input_mux alc861_capture_source = {
.num_items = 5,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x3 },
{ "Line", 0x1 },
{ "CD", 0x4 },
{ "Mixer", 0x5 },
},
};
/*
* configuration and preset
*/
static const char * const alc861_models[ALC861_MODEL_LAST] = {
[ALC861_3ST] = "3stack",
[ALC660_3ST] = "3stack-660",
[ALC861_3ST_DIG] = "3stack-dig",
[ALC861_6ST_DIG] = "6stack-dig",
[ALC861_UNIWILL_M31] = "uniwill-m31",
[ALC861_TOSHIBA] = "toshiba",
[ALC861_ASUS] = "asus",
[ALC861_ASUS_LAPTOP] = "asus-laptop",
[ALC861_AUTO] = "auto",
};
static const struct snd_pci_quirk alc861_cfg_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC861_3ST),
SND_PCI_QUIRK(0x1043, 0x1335, "ASUS F2/3", ALC861_ASUS_LAPTOP),
SND_PCI_QUIRK(0x1043, 0x1338, "ASUS F2/3", ALC861_ASUS_LAPTOP),
SND_PCI_QUIRK(0x1043, 0x1393, "ASUS", ALC861_ASUS),
SND_PCI_QUIRK(0x1043, 0x13d7, "ASUS A9rp", ALC861_ASUS_LAPTOP),
SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS P1-AH2", ALC861_3ST_DIG),
SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba", ALC861_TOSHIBA),
/* FIXME: the entry below breaks Toshiba A100 (model=auto works!)
* Any other models that need this preset?
*/
/* SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba", ALC861_TOSHIBA), */
SND_PCI_QUIRK(0x1462, 0x7254, "HP dx2200 (MSI MS-7254)", ALC861_3ST),
SND_PCI_QUIRK(0x1462, 0x7297, "HP dx2250 (MSI MS-7297)", ALC861_3ST),
SND_PCI_QUIRK(0x1584, 0x2b01, "Uniwill X40AIx", ALC861_UNIWILL_M31),
SND_PCI_QUIRK(0x1584, 0x9072, "Uniwill m31", ALC861_UNIWILL_M31),
SND_PCI_QUIRK(0x1584, 0x9075, "Airis Praxis N1212", ALC861_ASUS_LAPTOP),
/* FIXME: the below seems conflict */
/* SND_PCI_QUIRK(0x1584, 0x9075, "Uniwill", ALC861_UNIWILL_M31), */
SND_PCI_QUIRK(0x1849, 0x0660, "Asrock 939SLI32", ALC660_3ST),
SND_PCI_QUIRK(0x8086, 0xd600, "Intel", ALC861_3ST),
{}
};
static const struct alc_config_preset alc861_presets[] = {
[ALC861_3ST] = {
.mixers = { alc861_3ST_mixer },
.init_verbs = { alc861_threestack_init_verbs },
.num_dacs = ARRAY_SIZE(alc861_dac_nids),
.dac_nids = alc861_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
.channel_mode = alc861_threestack_modes,
.need_dac_fix = 1,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
},
[ALC861_3ST_DIG] = {
.mixers = { alc861_base_mixer },
.init_verbs = { alc861_threestack_init_verbs },
.num_dacs = ARRAY_SIZE(alc861_dac_nids),
.dac_nids = alc861_dac_nids,
.dig_out_nid = ALC861_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
.channel_mode = alc861_threestack_modes,
.need_dac_fix = 1,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
},
[ALC861_6ST_DIG] = {
.mixers = { alc861_base_mixer },
.init_verbs = { alc861_base_init_verbs },
.num_dacs = ARRAY_SIZE(alc861_dac_nids),
.dac_nids = alc861_dac_nids,
.dig_out_nid = ALC861_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc861_8ch_modes),
.channel_mode = alc861_8ch_modes,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
},
[ALC660_3ST] = {
.mixers = { alc861_3ST_mixer },
.init_verbs = { alc861_threestack_init_verbs },
.num_dacs = ARRAY_SIZE(alc660_dac_nids),
.dac_nids = alc660_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
.channel_mode = alc861_threestack_modes,
.need_dac_fix = 1,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
},
[ALC861_UNIWILL_M31] = {
.mixers = { alc861_uniwill_m31_mixer },
.init_verbs = { alc861_uniwill_m31_init_verbs },
.num_dacs = ARRAY_SIZE(alc861_dac_nids),
.dac_nids = alc861_dac_nids,
.dig_out_nid = ALC861_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc861_uniwill_m31_modes),
.channel_mode = alc861_uniwill_m31_modes,
.need_dac_fix = 1,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
},
[ALC861_TOSHIBA] = {
.mixers = { alc861_toshiba_mixer },
.init_verbs = { alc861_base_init_verbs,
alc861_toshiba_init_verbs },
.num_dacs = ARRAY_SIZE(alc861_dac_nids),
.dac_nids = alc861_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
.unsol_event = alc861_toshiba_unsol_event,
.init_hook = alc861_toshiba_automute,
},
[ALC861_ASUS] = {
.mixers = { alc861_asus_mixer },
.init_verbs = { alc861_asus_init_verbs },
.num_dacs = ARRAY_SIZE(alc861_dac_nids),
.dac_nids = alc861_dac_nids,
.dig_out_nid = ALC861_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc861_asus_modes),
.channel_mode = alc861_asus_modes,
.need_dac_fix = 1,
.hp_nid = 0x06,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
},
[ALC861_ASUS_LAPTOP] = {
.mixers = { alc861_toshiba_mixer, alc861_asus_laptop_mixer },
.init_verbs = { alc861_asus_init_verbs,
alc861_asus_laptop_init_verbs },
.num_dacs = ARRAY_SIZE(alc861_dac_nids),
.dac_nids = alc861_dac_nids,
.dig_out_nid = ALC861_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.need_dac_fix = 1,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
},
};

605
sound/pci/hda/alc861vd_quirks.c Normal file
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/*
* ALC660-VD/ALC861-VD quirk models
* included by patch_realtek.c
*/
/* ALC861-VD models */
enum {
ALC861VD_AUTO,
ALC660VD_3ST,
ALC660VD_3ST_DIG,
ALC660VD_ASUS_V1S,
ALC861VD_3ST,
ALC861VD_3ST_DIG,
ALC861VD_6ST_DIG,
ALC861VD_LENOVO,
ALC861VD_DALLAS,
ALC861VD_HP,
ALC861VD_MODEL_LAST,
};
#define ALC861VD_DIGOUT_NID 0x06
static const hda_nid_t alc861vd_dac_nids[4] = {
/* front, surr, clfe, side surr */
0x02, 0x03, 0x04, 0x05
};
/* dac_nids for ALC660vd are in a different order - according to
* Realtek's driver.
* This should probably result in a different mixer for 6stack models
* of ALC660vd codecs, but for now there is only 3stack mixer
* - and it is the same as in 861vd.
* adc_nids in ALC660vd are (is) the same as in 861vd
*/
static const hda_nid_t alc660vd_dac_nids[3] = {
/* front, rear, clfe, rear_surr */
0x02, 0x04, 0x03
};
static const hda_nid_t alc861vd_adc_nids[1] = {
/* ADC0 */
0x09,
};
static const hda_nid_t alc861vd_capsrc_nids[1] = { 0x22 };
/* input MUX */
/* FIXME: should be a matrix-type input source selection */
static const struct hda_input_mux alc861vd_capture_source = {
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x1 },
{ "Line", 0x2 },
{ "CD", 0x4 },
},
};
static const struct hda_input_mux alc861vd_dallas_capture_source = {
.num_items = 2,
.items = {
{ "Mic", 0x0 },
{ "Internal Mic", 0x1 },
},
};
static const struct hda_input_mux alc861vd_hp_capture_source = {
.num_items = 2,
.items = {
{ "Front Mic", 0x0 },
{ "ATAPI Mic", 0x1 },
},
};
/*
* 2ch mode
*/
static const struct hda_channel_mode alc861vd_3stack_2ch_modes[1] = {
{ 2, NULL }
};
/*
* 6ch mode
*/
static const struct hda_verb alc861vd_6stack_ch6_init[] = {
{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ } /* end */
};
/*
* 8ch mode
*/
static const struct hda_verb alc861vd_6stack_ch8_init[] = {
{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ } /* end */
};
static const struct hda_channel_mode alc861vd_6stack_modes[2] = {
{ 6, alc861vd_6stack_ch6_init },
{ 8, alc861vd_6stack_ch8_init },
};
static const struct snd_kcontrol_new alc861vd_chmode_mixer[] = {
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
},
{ } /* end */
};
/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
* Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
*/
static const struct snd_kcontrol_new alc861vd_6st_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0,
HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0,
HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Side Playback Volume", 0x05, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc861vd_3st_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc861vd_lenovo_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
/*HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),*/
HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
{ } /* end */
};
/* Pin assignment: Speaker=0x14, HP = 0x15,
* Mic=0x18, Internal Mic = 0x19, CD = 0x1c, PC Beep = 0x1d
*/
static const struct snd_kcontrol_new alc861vd_dallas_mixer[] = {
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, 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 Boost Volume", 0x19, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
/* Pin assignment: Speaker=0x14, Line-out = 0x15,
* Front Mic=0x18, ATAPI Mic = 0x19,
*/
static const struct snd_kcontrol_new alc861vd_hp_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
/*
* generic initialization of ADC, input mixers and output mixers
*/
static const struct hda_verb alc861vd_volume_init_verbs[] = {
/*
* Unmute ADC0 and set the default input to mic-in
*/
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of
* the analog-loopback mixer widget
*/
/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* Capture mixer: unmute Mic, F-Mic, Line, CD inputs */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
/*
* Set up output mixers (0x02 - 0x05)
*/
/* set vol=0 to output mixers */
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* set up input amps for analog loopback */
/* Amp Indices: DAC = 0, mixer = 1 */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{ }
};
/*
* 3-stack pin configuration:
* front = 0x14, mic/clfe = 0x18, HP = 0x19, line/surr = 0x1a, f-mic = 0x1b
*/
static const struct hda_verb alc861vd_3stack_init_verbs[] = {
/*
* Set pin mode and muting
*/
/* set front pin widgets 0x14 for output */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Mic (rear) pin: input vref at 80% */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Front Mic pin: input vref at 80% */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Line In pin: input */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Line-2 In: Headphone output (output 0 - 0x0c) */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
/* CD pin widget for input */
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{ }
};
/*
* 6-stack pin configuration:
*/
static const struct hda_verb alc861vd_6stack_init_verbs[] = {
/*
* Set pin mode and muting
*/
/* set front pin widgets 0x14 for output */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Rear Pin: output 1 (0x0d) */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
/* CLFE Pin: output 2 (0x0e) */
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
/* Side Pin: output 3 (0x0f) */
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
/* Mic (rear) pin: input vref at 80% */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Front Mic pin: input vref at 80% */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Line In pin: input */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Line-2 In: Headphone output (output 0 - 0x0c) */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
/* CD pin widget for input */
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{ }
};
static const struct hda_verb alc861vd_eapd_verbs[] = {
{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
{ }
};
static const struct hda_verb alc861vd_lenovo_unsol_verbs[] = {
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
{}
};
static void alc861vd_lenovo_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x14;
spec->automute = 1;
spec->automute_mode = ALC_AUTOMUTE_AMP;
}
static void alc861vd_lenovo_init_hook(struct hda_codec *codec)
{
alc_hp_automute(codec);
alc88x_simple_mic_automute(codec);
}
static void alc861vd_lenovo_unsol_event(struct hda_codec *codec,
unsigned int res)
{
switch (res >> 26) {
case ALC_MIC_EVENT:
alc88x_simple_mic_automute(codec);
break;
default:
alc_sku_unsol_event(codec, res);
break;
}
}
static const struct hda_verb alc861vd_dallas_verbs[] = {
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
{ } /* end */
};
/* toggle speaker-output according to the hp-jack state */
static void alc861vd_dallas_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->automute = 1;
spec->automute_mode = ALC_AUTOMUTE_AMP;
}
/*
* configuration and preset
*/
static const char * const alc861vd_models[ALC861VD_MODEL_LAST] = {
[ALC660VD_3ST] = "3stack-660",
[ALC660VD_3ST_DIG] = "3stack-660-digout",
[ALC660VD_ASUS_V1S] = "asus-v1s",
[ALC861VD_3ST] = "3stack",
[ALC861VD_3ST_DIG] = "3stack-digout",
[ALC861VD_6ST_DIG] = "6stack-digout",
[ALC861VD_LENOVO] = "lenovo",
[ALC861VD_DALLAS] = "dallas",
[ALC861VD_HP] = "hp",
[ALC861VD_AUTO] = "auto",
};
static const struct snd_pci_quirk alc861vd_cfg_tbl[] = {
SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST),
SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP),
SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST),
/*SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST),*/ /* auto */
SND_PCI_QUIRK(0x1043, 0x1633, "Asus V1Sn", ALC660VD_ASUS_V1S),
SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG),
SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST),
SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO),
/*SND_PCI_QUIRK(0x1179, 0xff00, "DALLAS", ALC861VD_DALLAS),*/ /*lenovo*/
SND_PCI_QUIRK(0x1179, 0xff01, "Toshiba A135", ALC861VD_LENOVO),
SND_PCI_QUIRK(0x1179, 0xff03, "Toshiba P205", ALC861VD_LENOVO),
SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba L30-149", ALC861VD_DALLAS),
SND_PCI_QUIRK(0x1565, 0x820d, "Biostar NF61S SE", ALC861VD_6ST_DIG),
SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", ALC861VD_LENOVO),
SND_PCI_QUIRK(0x1849, 0x0862, "ASRock K8NF6G-VSTA", ALC861VD_6ST_DIG),
{}
};
static const struct alc_config_preset alc861vd_presets[] = {
[ALC660VD_3ST] = {
.mixers = { alc861vd_3st_mixer },
.init_verbs = { alc861vd_volume_init_verbs,
alc861vd_3stack_init_verbs },
.num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
.dac_nids = alc660vd_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_capture_source,
},
[ALC660VD_3ST_DIG] = {
.mixers = { alc861vd_3st_mixer },
.init_verbs = { alc861vd_volume_init_verbs,
alc861vd_3stack_init_verbs },
.num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
.dac_nids = alc660vd_dac_nids,
.dig_out_nid = ALC861VD_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_capture_source,
},
[ALC861VD_3ST] = {
.mixers = { alc861vd_3st_mixer },
.init_verbs = { alc861vd_volume_init_verbs,
alc861vd_3stack_init_verbs },
.num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
.dac_nids = alc861vd_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_capture_source,
},
[ALC861VD_3ST_DIG] = {
.mixers = { alc861vd_3st_mixer },
.init_verbs = { alc861vd_volume_init_verbs,
alc861vd_3stack_init_verbs },
.num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
.dac_nids = alc861vd_dac_nids,
.dig_out_nid = ALC861VD_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_capture_source,
},
[ALC861VD_6ST_DIG] = {
.mixers = { alc861vd_6st_mixer, alc861vd_chmode_mixer },
.init_verbs = { alc861vd_volume_init_verbs,
alc861vd_6stack_init_verbs },
.num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
.dac_nids = alc861vd_dac_nids,
.dig_out_nid = ALC861VD_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc861vd_6stack_modes),
.channel_mode = alc861vd_6stack_modes,
.input_mux = &alc861vd_capture_source,
},
[ALC861VD_LENOVO] = {
.mixers = { alc861vd_lenovo_mixer },
.init_verbs = { alc861vd_volume_init_verbs,
alc861vd_3stack_init_verbs,
alc861vd_eapd_verbs,
alc861vd_lenovo_unsol_verbs },
.num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
.dac_nids = alc660vd_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_capture_source,
.unsol_event = alc861vd_lenovo_unsol_event,
.setup = alc861vd_lenovo_setup,
.init_hook = alc861vd_lenovo_init_hook,
},
[ALC861VD_DALLAS] = {
.mixers = { alc861vd_dallas_mixer },
.init_verbs = { alc861vd_dallas_verbs },
.num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
.dac_nids = alc861vd_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_dallas_capture_source,
.unsol_event = alc_sku_unsol_event,
.setup = alc861vd_dallas_setup,
.init_hook = alc_hp_automute,
},
[ALC861VD_HP] = {
.mixers = { alc861vd_hp_mixer },
.init_verbs = { alc861vd_dallas_verbs, alc861vd_eapd_verbs },
.num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
.dac_nids = alc861vd_dac_nids,
.dig_out_nid = ALC861VD_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_hp_capture_source,
.unsol_event = alc_sku_unsol_event,
.setup = alc861vd_dallas_setup,
.init_hook = alc_hp_automute,
},
[ALC660VD_ASUS_V1S] = {
.mixers = { alc861vd_lenovo_mixer },
.init_verbs = { alc861vd_volume_init_verbs,
alc861vd_3stack_init_verbs,
alc861vd_eapd_verbs,
alc861vd_lenovo_unsol_verbs },
.num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
.dac_nids = alc660vd_dac_nids,
.dig_out_nid = ALC861VD_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_capture_source,
.unsol_event = alc861vd_lenovo_unsol_event,
.setup = alc861vd_lenovo_setup,
.init_hook = alc861vd_lenovo_init_hook,
},
};

1898
sound/pci/hda/alc880_quirks.c Normal file
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3755
sound/pci/hda/alc882_quirks.c Normal file
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467
sound/pci/hda/alc_quirks.c Normal file
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@ -0,0 +1,467 @@
/*
* Common codes for Realtek codec quirks
* included by patch_realtek.c
*/
/*
* configuration template - to be copied to the spec instance
*/
struct alc_config_preset {
const struct snd_kcontrol_new *mixers[5]; /* should be identical size
* with spec
*/
const struct snd_kcontrol_new *cap_mixer; /* capture mixer */
const struct hda_verb *init_verbs[5];
unsigned int num_dacs;
const hda_nid_t *dac_nids;
hda_nid_t dig_out_nid; /* optional */
hda_nid_t hp_nid; /* optional */
const hda_nid_t *slave_dig_outs;
unsigned int num_adc_nids;
const hda_nid_t *adc_nids;
const hda_nid_t *capsrc_nids;
hda_nid_t dig_in_nid;
unsigned int num_channel_mode;
const struct hda_channel_mode *channel_mode;
int need_dac_fix;
int const_channel_count;
unsigned int num_mux_defs;
const struct hda_input_mux *input_mux;
void (*unsol_event)(struct hda_codec *, unsigned int);
void (*setup)(struct hda_codec *);
void (*init_hook)(struct hda_codec *);
#ifdef CONFIG_SND_HDA_POWER_SAVE
const struct hda_amp_list *loopbacks;
void (*power_hook)(struct hda_codec *codec);
#endif
};
/*
* channel mode setting
*/
static int alc_ch_mode_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
return snd_hda_ch_mode_info(codec, uinfo, spec->channel_mode,
spec->num_channel_mode);
}
static int alc_ch_mode_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode,
spec->num_channel_mode,
spec->ext_channel_count);
}
static int alc_ch_mode_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode,
spec->num_channel_mode,
&spec->ext_channel_count);
if (err >= 0 && !spec->const_channel_count) {
spec->multiout.max_channels = spec->ext_channel_count;
if (spec->need_dac_fix)
spec->multiout.num_dacs = spec->multiout.max_channels / 2;
}
return err;
}
/*
* Control the mode of pin widget settings via the mixer. "pc" is used
* instead of "%" to avoid consequences of accidentally 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 const char * const alc_pin_mode_names[] = {
"Mic 50pc bias", "Mic 80pc bias",
"Line in", "Line out", "Headphone out",
};
static const 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.
*/
#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 const 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)
{
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.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_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;
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);
/* Find enumerated value for current pinctl setting */
i = alc_pin_mode_min(dir);
while (i <= alc_pin_mode_max(dir) && alc_pin_mode_values[i] != pinctl)
i++;
*valp = i <= alc_pin_mode_max(dir) ? i: alc_pin_mode_min(dir);
return 0;
}
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;
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);
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_cache(codec, nid, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
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_amp_stereo(codec, nid, HDA_OUTPUT, 0,
HDA_AMP_MUTE, HDA_AMP_MUTE);
snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0,
HDA_AMP_MUTE, 0);
} else {
snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0,
HDA_AMP_MUTE, HDA_AMP_MUTE);
snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
HDA_AMP_MUTE, 0);
}
}
return change;
}
#define ALC_PIN_MODE(xname, nid, dir) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
.subdevice = HDA_SUBDEV_NID_FLAG | nid, \
.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
#define alc_gpio_data_info snd_ctl_boolean_mono_info
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_cache(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, \
.subdevice = HDA_SUBDEV_NID_FLAG | nid, \
.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
#define alc_spdif_ctrl_info snd_ctl_boolean_mono_info
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_1, 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_1,
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_cache(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, \
.subdevice = HDA_SUBDEV_NID_FLAG | nid, \
.info = alc_spdif_ctrl_info, \
.get = alc_spdif_ctrl_get, \
.put = alc_spdif_ctrl_put, \
.private_value = nid | (mask<<16) }
#endif /* CONFIG_SND_DEBUG */
/* A switch control to allow the enabling EAPD digital outputs on the ALC26x.
* Again, this is only used in the ALC26x test models to help identify when
* the EAPD line must be asserted for features to work.
*/
#ifdef CONFIG_SND_DEBUG
#define alc_eapd_ctrl_info snd_ctl_boolean_mono_info
static int alc_eapd_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_EAPD_BTLENABLE, 0x00);
*valp = (val & mask) != 0;
return 0;
}
static int alc_eapd_ctrl_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
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_EAPD_BTLENABLE,
0x00);
/* Set/unset the masked control bit(s) as needed */
change = (!val ? 0 : mask) != (ctrl_data & mask);
if (!val)
ctrl_data &= ~mask;
else
ctrl_data |= mask;
snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE,
ctrl_data);
return change;
}
#define ALC_EAPD_CTRL_SWITCH(xname, nid, mask) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
.subdevice = HDA_SUBDEV_NID_FLAG | nid, \
.info = alc_eapd_ctrl_info, \
.get = alc_eapd_ctrl_get, \
.put = alc_eapd_ctrl_put, \
.private_value = nid | (mask<<16) }
#endif /* CONFIG_SND_DEBUG */
static void alc_fixup_autocfg_pin_nums(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
if (!cfg->line_outs) {
while (cfg->line_outs < AUTO_CFG_MAX_OUTS &&
cfg->line_out_pins[cfg->line_outs])
cfg->line_outs++;
}
if (!cfg->speaker_outs) {
while (cfg->speaker_outs < AUTO_CFG_MAX_OUTS &&
cfg->speaker_pins[cfg->speaker_outs])
cfg->speaker_outs++;
}
if (!cfg->hp_outs) {
while (cfg->hp_outs < AUTO_CFG_MAX_OUTS &&
cfg->hp_pins[cfg->hp_outs])
cfg->hp_outs++;
}
}
/*
* set up from the preset table
*/
static void setup_preset(struct hda_codec *codec,
const struct alc_config_preset *preset)
{
struct alc_spec *spec = codec->spec;
int i;
for (i = 0; i < ARRAY_SIZE(preset->mixers) && preset->mixers[i]; i++)
add_mixer(spec, preset->mixers[i]);
spec->cap_mixer = preset->cap_mixer;
for (i = 0; i < ARRAY_SIZE(preset->init_verbs) && preset->init_verbs[i];
i++)
add_verb(spec, preset->init_verbs[i]);
spec->channel_mode = preset->channel_mode;
spec->num_channel_mode = preset->num_channel_mode;
spec->need_dac_fix = preset->need_dac_fix;
spec->const_channel_count = preset->const_channel_count;
if (preset->const_channel_count)
spec->multiout.max_channels = preset->const_channel_count;
else
spec->multiout.max_channels = spec->channel_mode[0].channels;
spec->ext_channel_count = spec->channel_mode[0].channels;
spec->multiout.num_dacs = preset->num_dacs;
spec->multiout.dac_nids = preset->dac_nids;
spec->multiout.dig_out_nid = preset->dig_out_nid;
spec->multiout.slave_dig_outs = preset->slave_dig_outs;
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->capsrc_nids = preset->capsrc_nids;
spec->dig_in_nid = preset->dig_in_nid;
spec->unsol_event = preset->unsol_event;
spec->init_hook = preset->init_hook;
#ifdef CONFIG_SND_HDA_POWER_SAVE
spec->power_hook = preset->power_hook;
spec->loopback.amplist = preset->loopbacks;
#endif
if (preset->setup)
preset->setup(codec);
alc_fixup_autocfg_pin_nums(codec);
}
/* auto-toggle front mic */
static void alc88x_simple_mic_automute(struct hda_codec *codec)
{
unsigned int present;
unsigned char bits;
present = snd_hda_jack_detect(codec, 0x18);
bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits);
}

361
sound/pci/hda/hda_codec.c
View file

@ -243,7 +243,8 @@ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid,
{
unsigned cmd = make_codec_cmd(codec, nid, direct, verb, parm);
unsigned int res;
codec_exec_verb(codec, cmd, &res);
if (codec_exec_verb(codec, cmd, &res))
return -1;
return res;
}
EXPORT_SYMBOL_HDA(snd_hda_codec_read);
@ -307,14 +308,65 @@ int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid,
}
EXPORT_SYMBOL_HDA(snd_hda_get_sub_nodes);
static int _hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
hda_nid_t *conn_list, int max_conns);
static bool add_conn_list(struct snd_array *array, hda_nid_t nid);
static int copy_conn_list(hda_nid_t nid, hda_nid_t *dst, int max_dst,
hda_nid_t *src, int len);
/* look up the cached results */
static hda_nid_t *lookup_conn_list(struct snd_array *array, hda_nid_t nid)
{
int i, len;
for (i = 0; i < array->used; ) {
hda_nid_t *p = snd_array_elem(array, i);
if (nid == *p)
return p;
len = p[1];
i += len + 2;
}
return NULL;
}
/**
* snd_hda_get_connections - get connection list
* snd_hda_get_conn_list - get connection list
* @codec: the HDA codec
* @nid: NID to parse
* @listp: the pointer to store NID list
*
* Parses the connection list of the given widget and stores the list
* of NIDs.
*
* Returns the number of connections, or a negative error code.
*/
int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid,
const hda_nid_t **listp)
{
struct snd_array *array = &codec->conn_lists;
int len, err;
hda_nid_t list[HDA_MAX_CONNECTIONS];
hda_nid_t *p;
bool added = false;
again:
/* if the connection-list is already cached, read it */
p = lookup_conn_list(array, nid);
if (p) {
if (listp)
*listp = p + 2;
return p[1];
}
if (snd_BUG_ON(added))
return -EINVAL;
/* read the connection and add to the cache */
len = snd_hda_get_raw_connections(codec, nid, list, HDA_MAX_CONNECTIONS);
if (len < 0)
return len;
err = snd_hda_override_conn_list(codec, nid, len, list);
if (err < 0)
return err;
added = true;
goto again;
}
EXPORT_SYMBOL_HDA(snd_hda_get_conn_list);
/**
* snd_hda_get_connections - copy connection list
* @codec: the HDA codec
* @nid: NID to parse
* @conn_list: connection list array
@ -328,42 +380,35 @@ static int copy_conn_list(hda_nid_t nid, hda_nid_t *dst, int max_dst,
int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
hda_nid_t *conn_list, int max_conns)
{
struct snd_array *array = &codec->conn_lists;
int i, len, old_used;
hda_nid_t list[HDA_MAX_CONNECTIONS];
const hda_nid_t *list;
int len = snd_hda_get_conn_list(codec, nid, &list);
/* look up the cached results */
for (i = 0; i < array->used; ) {
hda_nid_t *p = snd_array_elem(array, i);
len = p[1];
if (nid == *p)
return copy_conn_list(nid, conn_list, max_conns,
p + 2, len);
i += len + 2;
}
len = _hda_get_connections(codec, nid, list, HDA_MAX_CONNECTIONS);
if (len < 0)
if (len <= 0)
return len;
/* add to the cache */
old_used = array->used;
if (!add_conn_list(array, nid) || !add_conn_list(array, len))
goto error_add;
for (i = 0; i < len; i++)
if (!add_conn_list(array, list[i]))
goto error_add;
return copy_conn_list(nid, conn_list, max_conns, list, len);
error_add:
array->used = old_used;
return -ENOMEM;
if (len > max_conns) {
snd_printk(KERN_ERR "hda_codec: "
"Too many connections %d for NID 0x%x\n",
len, nid);
return -EINVAL;
}
memcpy(conn_list, list, len * sizeof(hda_nid_t));
return len;
}
EXPORT_SYMBOL_HDA(snd_hda_get_connections);
static int _hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
hda_nid_t *conn_list, int max_conns)
/**
* snd_hda_get_raw_connections - copy connection list without cache
* @codec: the HDA codec
* @nid: NID to parse
* @conn_list: connection list array
* @max_conns: max. number of connections to store
*
* Like snd_hda_get_connections(), copy the connection list but without
* checking through the connection-list cache.
* Currently called only from hda_proc.c, so not exported.
*/
int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid,
hda_nid_t *conn_list, int max_conns)
{
unsigned int parm;
int i, conn_len, conns;
@ -376,11 +421,8 @@ static int _hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
wcaps = get_wcaps(codec, nid);
if (!(wcaps & AC_WCAP_CONN_LIST) &&
get_wcaps_type(wcaps) != AC_WID_VOL_KNB) {
snd_printk(KERN_WARNING "hda_codec: "
"connection list not available for 0x%x\n", nid);
return -EINVAL;
}
get_wcaps_type(wcaps) != AC_WID_VOL_KNB)
return 0;
parm = snd_hda_param_read(codec, nid, AC_PAR_CONNLIST_LEN);
if (parm & AC_CLIST_LONG) {
@ -470,18 +512,77 @@ static bool add_conn_list(struct snd_array *array, hda_nid_t nid)
return true;
}
static int copy_conn_list(hda_nid_t nid, hda_nid_t *dst, int max_dst,
hda_nid_t *src, int len)
/**
* snd_hda_override_conn_list - add/modify the connection-list to cache
* @codec: the HDA codec
* @nid: NID to parse
* @len: number of connection list entries
* @list: the list of connection entries
*
* Add or modify the given connection-list to the cache. If the corresponding
* cache already exists, invalidate it and append a new one.
*
* Returns zero or a negative error code.
*/
int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int len,
const hda_nid_t *list)
{
if (len > max_dst) {
snd_printk(KERN_ERR "hda_codec: "
"Too many connections %d for NID 0x%x\n",
len, nid);
return -EINVAL;
}
memcpy(dst, src, len * sizeof(hda_nid_t));
return len;
struct snd_array *array = &codec->conn_lists;
hda_nid_t *p;
int i, old_used;
p = lookup_conn_list(array, nid);
if (p)
*p = -1; /* invalidate the old entry */
old_used = array->used;
if (!add_conn_list(array, nid) || !add_conn_list(array, len))
goto error_add;
for (i = 0; i < len; i++)
if (!add_conn_list(array, list[i]))
goto error_add;
return 0;
error_add:
array->used = old_used;
return -ENOMEM;
}
EXPORT_SYMBOL_HDA(snd_hda_override_conn_list);
/**
* snd_hda_get_conn_index - get the connection index of the given NID
* @codec: the HDA codec
* @mux: NID containing the list
* @nid: NID to select
* @recursive: 1 when searching NID recursively, otherwise 0
*
* Parses the connection list of the widget @mux and checks whether the
* widget @nid is present. If it is, return the connection index.
* Otherwise it returns -1.
*/
int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
hda_nid_t nid, int recursive)
{
hda_nid_t conn[HDA_MAX_NUM_INPUTS];
int i, nums;
nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn));
for (i = 0; i < nums; i++)
if (conn[i] == nid)
return i;
if (!recursive)
return -1;
if (recursive > 5) {
snd_printd("hda_codec: too deep connection for 0x%x\n", nid);
return -1;
}
recursive++;
for (i = 0; i < nums; i++)
if (snd_hda_get_conn_index(codec, conn[i], nid, recursive) >= 0)
return i;
return -1;
}
EXPORT_SYMBOL_HDA(snd_hda_get_conn_index);
/**
* snd_hda_queue_unsol_event - add an unsolicited event to queue
@ -1083,6 +1184,7 @@ static void snd_hda_codec_free(struct hda_codec *codec)
snd_array_free(&codec->mixers);
snd_array_free(&codec->nids);
snd_array_free(&codec->conn_lists);
snd_array_free(&codec->spdif_out);
codec->bus->caddr_tbl[codec->addr] = NULL;
if (codec->patch_ops.free)
codec->patch_ops.free(codec);
@ -1144,6 +1246,7 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus,
snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16);
snd_array_init(&codec->cvt_setups, sizeof(struct hda_cvt_setup), 8);
snd_array_init(&codec->conn_lists, sizeof(hda_nid_t), 64);
snd_array_init(&codec->spdif_out, sizeof(struct hda_spdif_out), 16);
if (codec->bus->modelname) {
codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL);
if (!codec->modelname) {
@ -2555,11 +2658,13 @@ static int snd_hda_spdif_default_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
int idx = kcontrol->private_value;
struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
ucontrol->value.iec958.status[0] = codec->spdif_status & 0xff;
ucontrol->value.iec958.status[1] = (codec->spdif_status >> 8) & 0xff;
ucontrol->value.iec958.status[2] = (codec->spdif_status >> 16) & 0xff;
ucontrol->value.iec958.status[3] = (codec->spdif_status >> 24) & 0xff;
ucontrol->value.iec958.status[0] = spdif->status & 0xff;
ucontrol->value.iec958.status[1] = (spdif->status >> 8) & 0xff;
ucontrol->value.iec958.status[2] = (spdif->status >> 16) & 0xff;
ucontrol->value.iec958.status[3] = (spdif->status >> 24) & 0xff;
return 0;
}
@ -2644,23 +2749,23 @@ static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value;
int idx = kcontrol->private_value;
struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
hda_nid_t nid = spdif->nid;
unsigned short val;
int change;
mutex_lock(&codec->spdif_mutex);
codec->spdif_status = ucontrol->value.iec958.status[0] |
spdif->status = ucontrol->value.iec958.status[0] |
((unsigned int)ucontrol->value.iec958.status[1] << 8) |
((unsigned int)ucontrol->value.iec958.status[2] << 16) |
((unsigned int)ucontrol->value.iec958.status[3] << 24);
val = convert_from_spdif_status(codec->spdif_status);
val |= codec->spdif_ctls & 1;
change = codec->spdif_ctls != val;
codec->spdif_ctls = val;
if (change)
val = convert_from_spdif_status(spdif->status);
val |= spdif->ctls & 1;
change = spdif->ctls != val;
spdif->ctls = val;
if (change && nid != (u16)-1)
set_dig_out_convert(codec, nid, val & 0xff, (val >> 8) & 0xff);
mutex_unlock(&codec->spdif_mutex);
return change;
}
@ -2671,33 +2776,42 @@ static int snd_hda_spdif_out_switch_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
int idx = kcontrol->private_value;
struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
ucontrol->value.integer.value[0] = codec->spdif_ctls & AC_DIG1_ENABLE;
ucontrol->value.integer.value[0] = spdif->ctls & AC_DIG1_ENABLE;
return 0;
}
static inline void set_spdif_ctls(struct hda_codec *codec, hda_nid_t nid,
int dig1, int dig2)
{
set_dig_out_convert(codec, nid, dig1, dig2);
/* unmute amp switch (if any) */
if ((get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) &&
(dig1 & AC_DIG1_ENABLE))
snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
HDA_AMP_MUTE, 0);
}
static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value;
int idx = kcontrol->private_value;
struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
hda_nid_t nid = spdif->nid;
unsigned short val;
int change;
mutex_lock(&codec->spdif_mutex);
val = codec->spdif_ctls & ~AC_DIG1_ENABLE;
val = spdif->ctls & ~AC_DIG1_ENABLE;
if (ucontrol->value.integer.value[0])
val |= AC_DIG1_ENABLE;
change = codec->spdif_ctls != val;
if (change) {
codec->spdif_ctls = val;
set_dig_out_convert(codec, nid, val & 0xff, -1);
/* unmute amp switch (if any) */
if ((get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) &&
(val & AC_DIG1_ENABLE))
snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
HDA_AMP_MUTE, 0);
}
change = spdif->ctls != val;
spdif->ctls = val;
if (change && nid != (u16)-1)
set_spdif_ctls(codec, nid, val & 0xff, -1);
mutex_unlock(&codec->spdif_mutex);
return change;
}
@ -2744,36 +2858,79 @@ static struct snd_kcontrol_new dig_mixes[] = {
*
* Returns 0 if successful, or a negative error code.
*/
int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid)
int snd_hda_create_spdif_out_ctls(struct hda_codec *codec,
hda_nid_t associated_nid,
hda_nid_t cvt_nid)
{
int err;
struct snd_kcontrol *kctl;
struct snd_kcontrol_new *dig_mix;
int idx;
struct hda_spdif_out *spdif;
idx = find_empty_mixer_ctl_idx(codec, "IEC958 Playback Switch");
if (idx < 0) {
printk(KERN_ERR "hda_codec: too many IEC958 outputs\n");
return -EBUSY;
}
spdif = snd_array_new(&codec->spdif_out);
for (dig_mix = dig_mixes; dig_mix->name; dig_mix++) {
kctl = snd_ctl_new1(dig_mix, codec);
if (!kctl)
return -ENOMEM;
kctl->id.index = idx;
kctl->private_value = nid;
err = snd_hda_ctl_add(codec, nid, kctl);
kctl->private_value = codec->spdif_out.used - 1;
err = snd_hda_ctl_add(codec, associated_nid, kctl);
if (err < 0)
return err;
}
codec->spdif_ctls =
snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_DIGI_CONVERT_1, 0);
codec->spdif_status = convert_to_spdif_status(codec->spdif_ctls);
spdif->nid = cvt_nid;
spdif->ctls = snd_hda_codec_read(codec, cvt_nid, 0,
AC_VERB_GET_DIGI_CONVERT_1, 0);
spdif->status = convert_to_spdif_status(spdif->ctls);
return 0;
}
EXPORT_SYMBOL_HDA(snd_hda_create_spdif_out_ctls);
struct hda_spdif_out *snd_hda_spdif_out_of_nid(struct hda_codec *codec,
hda_nid_t nid)
{
int i;
for (i = 0; i < codec->spdif_out.used; i++) {
struct hda_spdif_out *spdif =
snd_array_elem(&codec->spdif_out, i);
if (spdif->nid == nid)
return spdif;
}
return NULL;
}
EXPORT_SYMBOL_HDA(snd_hda_spdif_out_of_nid);
void snd_hda_spdif_ctls_unassign(struct hda_codec *codec, int idx)
{
struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
mutex_lock(&codec->spdif_mutex);
spdif->nid = (u16)-1;
mutex_unlock(&codec->spdif_mutex);
}
EXPORT_SYMBOL_HDA(snd_hda_spdif_ctls_unassign);
void snd_hda_spdif_ctls_assign(struct hda_codec *codec, int idx, hda_nid_t nid)
{
struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
unsigned short val;
mutex_lock(&codec->spdif_mutex);
if (spdif->nid != nid) {
spdif->nid = nid;
val = spdif->ctls;
set_spdif_ctls(codec, nid, val & 0xff, (val >> 8) & 0xff);
}
mutex_unlock(&codec->spdif_mutex);
}
EXPORT_SYMBOL_HDA(snd_hda_spdif_ctls_assign);
/*
* SPDIF sharing with analog output
*/
@ -3356,7 +3513,7 @@ static unsigned int query_stream_param(struct hda_codec *codec, hda_nid_t nid)
*
* Returns 0 if successful, otherwise a negative error code.
*/
static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
u32 *ratesp, u64 *formatsp, unsigned int *bpsp)
{
unsigned int i, val, wcaps;
@ -3448,6 +3605,7 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
return 0;
}
EXPORT_SYMBOL_HDA(snd_hda_query_supported_pcm);
/**
* snd_hda_is_supported_format - Check the validity of the format
@ -4177,10 +4335,12 @@ EXPORT_SYMBOL_HDA(snd_hda_input_mux_put);
static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid,
unsigned int stream_tag, unsigned int format)
{
struct hda_spdif_out *spdif = snd_hda_spdif_out_of_nid(codec, nid);
/* turn off SPDIF once; otherwise the IEC958 bits won't be updated */
if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE))
if (codec->spdif_status_reset && (spdif->ctls & AC_DIG1_ENABLE))
set_dig_out_convert(codec, nid,
codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff,
spdif->ctls & ~AC_DIG1_ENABLE & 0xff,
-1);
snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format);
if (codec->slave_dig_outs) {
@ -4190,9 +4350,9 @@ static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid,
format);
}
/* turn on again (if needed) */
if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE))
if (codec->spdif_status_reset && (spdif->ctls & AC_DIG1_ENABLE))
set_dig_out_convert(codec, nid,
codec->spdif_ctls & 0xff, -1);
spdif->ctls & 0xff, -1);
}
static void cleanup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid)
@ -4348,6 +4508,8 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec,
{
const hda_nid_t *nids = mout->dac_nids;
int chs = substream->runtime->channels;
struct hda_spdif_out *spdif =
snd_hda_spdif_out_of_nid(codec, mout->dig_out_nid);
int i;
mutex_lock(&codec->spdif_mutex);
@ -4356,7 +4518,7 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec,
if (chs == 2 &&
snd_hda_is_supported_format(codec, mout->dig_out_nid,
format) &&
!(codec->spdif_status & IEC958_AES0_NONAUDIO)) {
!(spdif->status & IEC958_AES0_NONAUDIO)) {
mout->dig_out_used = HDA_DIG_ANALOG_DUP;
setup_dig_out_stream(codec, mout->dig_out_nid,
stream_tag, format);
@ -4528,7 +4690,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
unsigned int wid_caps = get_wcaps(codec, nid);
unsigned int wid_type = get_wcaps_type(wid_caps);
unsigned int def_conf;
short assoc, loc;
short assoc, loc, conn, dev;
/* read all default configuration for pin complex */
if (wid_type != AC_WID_PIN)
@ -4538,10 +4700,19 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
continue;
def_conf = snd_hda_codec_get_pincfg(codec, nid);
if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE)
conn = get_defcfg_connect(def_conf);
if (conn == AC_JACK_PORT_NONE)
continue;
loc = get_defcfg_location(def_conf);
switch (get_defcfg_device(def_conf)) {
dev = get_defcfg_device(def_conf);
/* workaround for buggy BIOS setups */
if (dev == AC_JACK_LINE_OUT) {
if (conn == AC_JACK_PORT_FIXED)
dev = AC_JACK_SPEAKER;
}
switch (dev) {
case AC_JACK_LINE_OUT:
seq = get_defcfg_sequence(def_conf);
assoc = get_defcfg_association(def_conf);

30
sound/pci/hda/hda_codec.h
View file

@ -829,8 +829,7 @@ struct hda_codec {
struct mutex spdif_mutex;
struct mutex control_mutex;
unsigned int spdif_status; /* IEC958 status bits */
unsigned short spdif_ctls; /* SPDIF control bits */
struct snd_array spdif_out;
unsigned int spdif_in_enable; /* SPDIF input enable? */
const hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */
struct snd_array init_pins; /* initial (BIOS) pin configurations */
@ -904,6 +903,16 @@ int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid,
hda_nid_t *start_id);
int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
hda_nid_t *conn_list, int max_conns);
int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid,
hda_nid_t *conn_list, int max_conns);
int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid,
const hda_nid_t **listp);
int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int nums,
const hda_nid_t *list);
int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
hda_nid_t nid, int recursive);
int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
u32 *ratesp, u64 *formatsp, unsigned int *bpsp);
struct hda_verb {
hda_nid_t nid;
@ -947,6 +956,17 @@ int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list,
hda_nid_t nid, unsigned int cfg); /* for hwdep */
void snd_hda_shutup_pins(struct hda_codec *codec);
/* SPDIF controls */
struct hda_spdif_out {
hda_nid_t nid; /* Converter nid values relate to */
unsigned int status; /* IEC958 status bits */
unsigned short ctls; /* SPDIF control bits */
};
struct hda_spdif_out *snd_hda_spdif_out_of_nid(struct hda_codec *codec,
hda_nid_t nid);
void snd_hda_spdif_ctls_unassign(struct hda_codec *codec, int idx);
void snd_hda_spdif_ctls_assign(struct hda_codec *codec, int idx, hda_nid_t nid);
/*
* Mixer
*/
@ -997,17 +1017,15 @@ int snd_hda_suspend(struct hda_bus *bus);
int snd_hda_resume(struct hda_bus *bus);
#endif
#ifdef CONFIG_SND_HDA_POWER_SAVE
static inline
int hda_call_check_power_status(struct hda_codec *codec, hda_nid_t nid)
{
#ifdef CONFIG_SND_HDA_POWER_SAVE
if (codec->patch_ops.check_power_status)
return codec->patch_ops.check_power_status(codec, nid);
#endif
return 0;
}
#else
#define hda_call_check_power_status(codec, nid) 0
#endif
/*
* get widget information

46
sound/pci/hda/hda_eld.c
View file

@ -580,43 +580,45 @@ void snd_hda_eld_proc_free(struct hda_codec *codec, struct hdmi_eld *eld)
#endif /* CONFIG_PROC_FS */
/* update PCM info based on ELD */
void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm,
struct hda_pcm_stream *codec_pars)
void snd_hdmi_eld_update_pcm_info(struct hdmi_eld *eld,
struct hda_pcm_stream *hinfo)
{
u32 rates;
u64 formats;
unsigned int maxbps;
unsigned int channels_max;
int i;
/* assume basic audio support (the basic audio flag is not in ELD;
* however, all audio capable sinks are required to support basic
* audio) */
pcm->rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000;
pcm->formats = SNDRV_PCM_FMTBIT_S16_LE;
pcm->maxbps = 16;
pcm->channels_max = 2;
rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |
SNDRV_PCM_RATE_48000;
formats = SNDRV_PCM_FMTBIT_S16_LE;
maxbps = 16;
channels_max = 2;
for (i = 0; i < eld->sad_count; i++) {
struct cea_sad *a = &eld->sad[i];
pcm->rates |= a->rates;
if (a->channels > pcm->channels_max)
pcm->channels_max = a->channels;
rates |= a->rates;
if (a->channels > channels_max)
channels_max = a->channels;
if (a->format == AUDIO_CODING_TYPE_LPCM) {
if (a->sample_bits & AC_SUPPCM_BITS_20) {
pcm->formats |= SNDRV_PCM_FMTBIT_S32_LE;
if (pcm->maxbps < 20)
pcm->maxbps = 20;
formats |= SNDRV_PCM_FMTBIT_S32_LE;
if (maxbps < 20)
maxbps = 20;
}
if (a->sample_bits & AC_SUPPCM_BITS_24) {
pcm->formats |= SNDRV_PCM_FMTBIT_S32_LE;
if (pcm->maxbps < 24)
pcm->maxbps = 24;
formats |= SNDRV_PCM_FMTBIT_S32_LE;
if (maxbps < 24)
maxbps = 24;
}
}
}
if (!codec_pars)
return;
/* restrict the parameters by the values the codec provides */
pcm->rates &= codec_pars->rates;
pcm->formats &= codec_pars->formats;
pcm->channels_max = min(pcm->channels_max, codec_pars->channels_max);
pcm->maxbps = min(pcm->maxbps, codec_pars->maxbps);
hinfo->rates &= rates;
hinfo->formats &= formats;
hinfo->maxbps = min(hinfo->maxbps, maxbps);
hinfo->channels_max = min(hinfo->channels_max, channels_max);
}

76
sound/pci/hda/hda_intel.c
View file

@ -177,7 +177,8 @@ MODULE_DESCRIPTION("Intel HDA driver");
#define ICH6_REG_INTCTL 0x20
#define ICH6_REG_INTSTS 0x24
#define ICH6_REG_WALLCLK 0x30 /* 24Mhz source */
#define ICH6_REG_SYNC 0x34
#define ICH6_REG_OLD_SSYNC 0x34 /* SSYNC for old ICH */
#define ICH6_REG_SSYNC 0x38
#define ICH6_REG_CORBLBASE 0x40
#define ICH6_REG_CORBUBASE 0x44
#define ICH6_REG_CORBWP 0x48
@ -479,6 +480,7 @@ enum {
#define AZX_DCAPS_POSFIX_VIA (1 << 17) /* Use VIACOMBO as default */
#define AZX_DCAPS_NO_64BIT (1 << 18) /* No 64bit address */
#define AZX_DCAPS_SYNC_WRITE (1 << 19) /* sync each cmd write */
#define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */
/* quirks for ATI SB / AMD Hudson */
#define AZX_DCAPS_PRESET_ATI_SB \
@ -1706,13 +1708,16 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream)
struct snd_pcm_runtime *runtime = substream->runtime;
unsigned int bufsize, period_bytes, format_val, stream_tag;
int err;
struct hda_spdif_out *spdif =
snd_hda_spdif_out_of_nid(apcm->codec, hinfo->nid);
unsigned short ctls = spdif ? spdif->ctls : 0;
azx_stream_reset(chip, azx_dev);
format_val = snd_hda_calc_stream_format(runtime->rate,
runtime->channels,
runtime->format,
hinfo->maxbps,
apcm->codec->spdif_ctls);
ctls);
if (!format_val) {
snd_printk(KERN_ERR SFX
"invalid format_val, rate=%d, ch=%d, format=%d\n",
@ -1792,7 +1797,11 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
spin_lock(&chip->reg_lock);
if (nsync > 1) {
/* first, set SYNC bits of corresponding streams */
azx_writel(chip, SYNC, azx_readl(chip, SYNC) | sbits);
if (chip->driver_caps & AZX_DCAPS_OLD_SSYNC)
azx_writel(chip, OLD_SSYNC,
azx_readl(chip, OLD_SSYNC) | sbits);
else
azx_writel(chip, SSYNC, azx_readl(chip, SSYNC) | sbits);
}
snd_pcm_group_for_each_entry(s, substream) {
if (s->pcm->card != substream->pcm->card)
@ -1848,7 +1857,11 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
if (nsync > 1) {
spin_lock(&chip->reg_lock);
/* reset SYNC bits */
azx_writel(chip, SYNC, azx_readl(chip, SYNC) & ~sbits);
if (chip->driver_caps & AZX_DCAPS_OLD_SSYNC)
azx_writel(chip, OLD_SSYNC,
azx_readl(chip, OLD_SSYNC) & ~sbits);
else
azx_writel(chip, SSYNC, azx_readl(chip, SSYNC) & ~sbits);
spin_unlock(&chip->reg_lock);
}
return 0;
@ -1863,7 +1876,7 @@ static unsigned int azx_via_get_position(struct azx *chip,
unsigned int fifo_size;
link_pos = azx_sd_readl(azx_dev, SD_LPIB);
if (azx_dev->index >= 4) {
if (azx_dev->substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
/* Playback, no problem using link position */
return link_pos;
}
@ -1927,6 +1940,17 @@ static unsigned int azx_get_position(struct azx *chip,
default:
/* use the position buffer */
pos = le32_to_cpu(*azx_dev->posbuf);
if (chip->position_fix[stream] == POS_FIX_AUTO) {
if (!pos || pos == (u32)-1) {
printk(KERN_WARNING
"hda-intel: Invalid position buffer, "
"using LPIB read method instead.\n");
chip->position_fix[stream] = POS_FIX_LPIB;
pos = azx_sd_readl(azx_dev, SD_LPIB);
} else
chip->position_fix[stream] = POS_FIX_POSBUF;
}
break;
}
if (pos >= azx_dev->bufsize)
@ -1964,16 +1988,6 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev)
stream = azx_dev->substream->stream;
pos = azx_get_position(chip, azx_dev);
if (chip->position_fix[stream] == POS_FIX_AUTO) {
if (!pos) {
printk(KERN_WARNING
"hda-intel: Invalid position buffer, "
"using LPIB read method instead.\n");
chip->position_fix[stream] = POS_FIX_LPIB;
pos = azx_get_position(chip, azx_dev);
} else
chip->position_fix[stream] = POS_FIX_POSBUF;
}
if (WARN_ONCE(!azx_dev->period_bytes,
"hda-intel: zero azx_dev->period_bytes"))
@ -2061,6 +2075,8 @@ static void azx_pcm_free(struct snd_pcm *pcm)
}
}
#define MAX_PREALLOC_SIZE (32 * 1024 * 1024)
static int
azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec,
struct hda_pcm *cpcm)
@ -2069,6 +2085,7 @@ azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec,
struct snd_pcm *pcm;
struct azx_pcm *apcm;
int pcm_dev = cpcm->device;
unsigned int size;
int s, err;
if (pcm_dev >= HDA_MAX_PCMS) {
@ -2104,9 +2121,12 @@ azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec,
snd_pcm_set_ops(pcm, s, &azx_pcm_ops);
}
/* buffer pre-allocation */
size = CONFIG_SND_HDA_PREALLOC_SIZE * 1024;
if (size > MAX_PREALLOC_SIZE)
size = MAX_PREALLOC_SIZE;
snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG,
snd_dma_pci_data(chip->pci),
1024 * 64, 32 * 1024 * 1024);
size, MAX_PREALLOC_SIZE);
return 0;
}
@ -2347,28 +2367,20 @@ static int azx_dev_free(struct snd_device *device)
* white/black-listing for position_fix
*/
static struct snd_pci_quirk position_fix_list[] __devinitdata = {
SND_PCI_QUIRK(0x1025, 0x009f, "Acer Aspire 5110", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1025, 0x026f, "Acer Aspire 5538", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1028, 0x01f6, "Dell Latitude 131L", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1028, 0x0470, "Dell Inspiron 1120", POS_FIX_LPIB),
SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1043, 0x8410, "ASUS", POS_FIX_LPIB),
SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba A100-259", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1565, 0x820f, "Biostar Microtech", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1565, 0x8218, "Biostar Microtech", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1849, 0x0888, "775Dual-VSTA", POS_FIX_LPIB),
SND_PCI_QUIRK(0x8086, 0x2503, "DG965OT AAD63733-203", POS_FIX_LPIB),
SND_PCI_QUIRK(0x8086, 0xd601, "eMachines T5212", POS_FIX_LPIB),
{}
};
@ -2815,6 +2827,22 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
/* SCH */
{ PCI_DEVICE(0x8086, 0x811b),
.driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP },
{ PCI_DEVICE(0x8086, 0x2668),
.driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH6 */
{ PCI_DEVICE(0x8086, 0x27d8),
.driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH7 */
{ PCI_DEVICE(0x8086, 0x269a),
.driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ESB2 */
{ PCI_DEVICE(0x8086, 0x284b),
.driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH8 */
{ PCI_DEVICE(0x8086, 0x293e),
.driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH9 */
{ PCI_DEVICE(0x8086, 0x293f),
.driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH9 */
{ PCI_DEVICE(0x8086, 0x3a3e),
.driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH10 */
{ PCI_DEVICE(0x8086, 0x3a6e),
.driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH10 */
/* Generic Intel */
{ PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_ANY_ID),
.class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8,

10
sound/pci/hda/hda_local.h
View file

@ -212,7 +212,9 @@ int snd_hda_mixer_bind_tlv(struct snd_kcontrol *kcontrol, int op_flag,
/*
* SPDIF I/O
*/
int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid);
int snd_hda_create_spdif_out_ctls(struct hda_codec *codec,
hda_nid_t associated_nid,
hda_nid_t cvt_nid);
int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid);
/*
@ -563,7 +565,6 @@ int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key)
* power-management
*/
#ifdef CONFIG_SND_HDA_POWER_SAVE
void snd_hda_schedule_power_save(struct hda_codec *codec);
struct hda_amp_list {
@ -580,7 +581,6 @@ struct hda_loopback_check {
int snd_hda_check_amp_list_power(struct hda_codec *codec,
struct hda_loopback_check *check,
hda_nid_t nid);
#endif /* CONFIG_SND_HDA_POWER_SAVE */
/*
* AMP control callbacks
@ -639,8 +639,8 @@ struct hdmi_eld {
int snd_hdmi_get_eld_size(struct hda_codec *codec, hda_nid_t nid);
int snd_hdmi_get_eld(struct hdmi_eld *, struct hda_codec *, hda_nid_t);
void snd_hdmi_show_eld(struct hdmi_eld *eld);
void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm,
struct hda_pcm_stream *codec_pars);
void snd_hdmi_eld_update_pcm_info(struct hdmi_eld *eld,
struct hda_pcm_stream *hinfo);
#ifdef CONFIG_PROC_FS
int snd_hda_eld_proc_new(struct hda_codec *codec, struct hdmi_eld *eld,

2
sound/pci/hda/hda_proc.c
View file

@ -636,7 +636,7 @@ static void print_codec_info(struct snd_info_entry *entry,
wid_caps |= AC_WCAP_CONN_LIST;
if (wid_caps & AC_WCAP_CONN_LIST)
conn_len = snd_hda_get_connections(codec, nid, conn,
conn_len = snd_hda_get_raw_connections(codec, nid, conn,
HDA_MAX_CONNECTIONS);
if (wid_caps & AC_WCAP_IN_AMP) {

7
sound/pci/hda/patch_analog.c
View file

@ -213,7 +213,9 @@ static int ad198x_build_controls(struct hda_codec *codec)
return err;
}
if (spec->multiout.dig_out_nid) {
err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
err = snd_hda_create_spdif_out_ctls(codec,
spec->multiout.dig_out_nid,
spec->multiout.dig_out_nid);
if (err < 0)
return err;
err = snd_hda_create_spdif_share_sw(codec,
@ -1920,7 +1922,8 @@ static int patch_ad1981(struct hda_codec *codec)
spec->mixers[0] = ad1981_hp_mixers;
spec->num_init_verbs = 2;
spec->init_verbs[1] = ad1981_hp_init_verbs;
spec->multiout.dig_out_nid = 0;
if (!is_jack_available(codec, 0x0a))
spec->multiout.dig_out_nid = 0;
spec->input_mux = &ad1981_hp_capture_source;
codec->patch_ops.init = ad1981_hp_init;

3
sound/pci/hda/patch_ca0110.c
View file

@ -240,7 +240,8 @@ static int ca0110_build_controls(struct hda_codec *codec)
}
if (spec->dig_out) {
err = snd_hda_create_spdif_out_ctls(codec, spec->dig_out);
err = snd_hda_create_spdif_out_ctls(codec, spec->dig_out,
spec->dig_out);
if (err < 0)
return err;
err = snd_hda_create_spdif_share_sw(codec, &spec->multiout);

1097
sound/pci/hda/patch_ca0132.c Normal file
View file

File diff suppressed because it is too large Load diff

19
sound/pci/hda/patch_cirrus.c
View file

@ -346,21 +346,15 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
nid = codec->start_nid;
for (i = 0; i < codec->num_nodes; i++, nid++) {
hda_nid_t pins[2];
unsigned int type;
int j, nums;
int idx;
type = get_wcaps_type(get_wcaps(codec, nid));
if (type != AC_WID_AUD_IN)
continue;
nums = snd_hda_get_connections(codec, nid, pins,
ARRAY_SIZE(pins));
if (nums <= 0)
continue;
for (j = 0; j < nums; j++) {
if (pins[j] == pin) {
*idxp = j;
return nid;
}
idx = snd_hda_get_conn_index(codec, nid, pin, 0);
if (idx >= 0) {
*idxp = idx;
return nid;
}
}
return 0;
@ -821,7 +815,8 @@ static int build_digital_output(struct hda_codec *codec)
if (!spec->multiout.dig_out_nid)
return 0;
err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid,
spec->multiout.dig_out_nid);
if (err < 0)
return err;
err = snd_hda_create_spdif_share_sw(codec, &spec->multiout);

17
sound/pci/hda/patch_cmedia.c
View file

@ -327,7 +327,9 @@ static int cmi9880_build_controls(struct hda_codec *codec)
return err;
}
if (spec->multiout.dig_out_nid) {
err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
err = snd_hda_create_spdif_out_ctls(codec,
spec->multiout.dig_out_nid,
spec->multiout.dig_out_nid);
if (err < 0)
return err;
err = snd_hda_create_spdif_share_sw(codec,
@ -396,12 +398,11 @@ static int cmi9880_fill_multi_init(struct hda_codec *codec, const struct auto_pi
{
struct cmi_spec *spec = codec->spec;
hda_nid_t nid;
int i, j, k, len;
int i, j, k;
/* clear the table, only one c-media dac assumed here */
memset(spec->multi_init, 0, sizeof(spec->multi_init));
for (j = 0, i = 0; i < cfg->line_outs; i++) {
hda_nid_t conn[4];
nid = cfg->line_out_pins[i];
/* set as output */
spec->multi_init[j].nid = nid;
@ -414,12 +415,10 @@ static int cmi9880_fill_multi_init(struct hda_codec *codec, const struct auto_pi
spec->multi_init[j].verb = AC_VERB_SET_CONNECT_SEL;
spec->multi_init[j].param = 0;
/* find the index in connect list */
len = snd_hda_get_connections(codec, nid, conn, 4);
for (k = 0; k < len; k++)
if (conn[k] == spec->dac_nids[i]) {
spec->multi_init[j].param = k;
break;
}
k = snd_hda_get_conn_index(codec, nid,
spec->dac_nids[i], 0);
if (k >= 0)
spec->multi_init[j].param = k;
j++;
}
}

71
sound/pci/hda/patch_conexant.c
View file

@ -155,6 +155,10 @@ struct conexant_spec {
unsigned int mic_boost; /* offset into cxt5066_analog_mic_boost */
unsigned int beep_amp;
/* extra EAPD pins */
unsigned int num_eapds;
hda_nid_t eapds[4];
};
static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo,
@ -510,6 +514,7 @@ static int conexant_build_controls(struct hda_codec *codec)
}
if (spec->multiout.dig_out_nid) {
err = snd_hda_create_spdif_out_ctls(codec,
spec->multiout.dig_out_nid,
spec->multiout.dig_out_nid);
if (err < 0)
return err;
@ -1123,10 +1128,8 @@ static int patch_cxt5045(struct hda_codec *codec)
board_config = snd_hda_check_board_config(codec, CXT5045_MODELS,
cxt5045_models,
cxt5045_cfg_tbl);
#if 0 /* use the old method just for safety */
if (board_config < 0)
board_config = CXT5045_AUTO;
#endif
board_config = CXT5045_AUTO; /* model=auto as default */
if (board_config == CXT5045_AUTO)
return patch_conexant_auto(codec);
@ -1564,10 +1567,8 @@ static int patch_cxt5047(struct hda_codec *codec)
board_config = snd_hda_check_board_config(codec, CXT5047_MODELS,
cxt5047_models,
cxt5047_cfg_tbl);
#if 0 /* not enabled as default, as BIOS often broken for this codec */
if (board_config < 0)
board_config = CXT5047_AUTO;
#endif
board_config = CXT5047_AUTO; /* model=auto as default */
if (board_config == CXT5047_AUTO)
return patch_conexant_auto(codec);
@ -1993,10 +1994,8 @@ static int patch_cxt5051(struct hda_codec *codec)
board_config = snd_hda_check_board_config(codec, CXT5051_MODELS,
cxt5051_models,
cxt5051_cfg_tbl);
#if 0 /* use the old method just for safety */
if (board_config < 0)
board_config = CXT5051_AUTO;
#endif
board_config = CXT5051_AUTO; /* model=auto as default */
if (board_config == CXT5051_AUTO)
return patch_conexant_auto(codec);
@ -3114,10 +3113,8 @@ static int patch_cxt5066(struct hda_codec *codec)
board_config = snd_hda_check_board_config(codec, CXT5066_MODELS,
cxt5066_models, cxt5066_cfg_tbl);
#if 0 /* use the old method just for safety */
if (board_config < 0)
board_config = CXT5066_AUTO;
#endif
board_config = CXT5066_AUTO; /* model=auto as default */
if (board_config == CXT5066_AUTO)
return patch_conexant_auto(codec);
@ -3308,19 +3305,8 @@ static const struct hda_pcm_stream cx_auto_pcm_analog_capture = {
static const hda_nid_t cx_auto_adc_nids[] = { 0x14 };
/* get the connection index of @nid in the widget @mux */
static int get_connection_index(struct hda_codec *codec, hda_nid_t mux,
hda_nid_t nid)
{
hda_nid_t conn[HDA_MAX_NUM_INPUTS];
int i, nums;
nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn));
for (i = 0; i < nums; i++)
if (conn[i] == nid)
return i;
return -1;
}
#define get_connection_index(codec, mux, nid)\
snd_hda_get_conn_index(codec, mux, nid, 0)
/* get an unassigned DAC from the given list.
* Return the nid if found and reduce the DAC list, or return zero if
@ -3919,6 +3905,38 @@ static void cx_auto_parse_beep(struct hda_codec *codec)
#define cx_auto_parse_beep(codec)
#endif
static bool found_in_nid_list(hda_nid_t nid, const hda_nid_t *list, int nums)
{
int i;
for (i = 0; i < nums; i++)
if (list[i] == nid)
return true;
return false;
}
/* parse extra-EAPD that aren't assigned to any pins */
static void cx_auto_parse_eapd(struct hda_codec *codec)
{
struct conexant_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
hda_nid_t nid, end_nid;
end_nid = codec->start_nid + codec->num_nodes;
for (nid = codec->start_nid; nid < end_nid; nid++) {
if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_PIN)
continue;
if (!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD))
continue;
if (found_in_nid_list(nid, cfg->line_out_pins, cfg->line_outs) ||
found_in_nid_list(nid, cfg->hp_pins, cfg->hp_outs) ||
found_in_nid_list(nid, cfg->speaker_pins, cfg->speaker_outs))
continue;
spec->eapds[spec->num_eapds++] = nid;
if (spec->num_eapds >= ARRAY_SIZE(spec->eapds))
break;
}
}
static int cx_auto_parse_auto_config(struct hda_codec *codec)
{
struct conexant_spec *spec = codec->spec;
@ -3932,6 +3950,7 @@ static int cx_auto_parse_auto_config(struct hda_codec *codec)
cx_auto_parse_input(codec);
cx_auto_parse_digital(codec);
cx_auto_parse_beep(codec);
cx_auto_parse_eapd(codec);
return 0;
}
@ -4019,6 +4038,8 @@ static void cx_auto_init_output(struct hda_codec *codec)
}
}
cx_auto_update_speakers(codec);
/* turn on/off extra EAPDs, too */
cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, true);
}
static void cx_auto_init_input(struct hda_codec *codec)

698
sound/pci/hda/patch_hdmi.c
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File diff suppressed because it is too large Load diff

19880
sound/pci/hda/patch_realtek.c
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File diff suppressed because it is too large Load diff

31
sound/pci/hda/patch_sigmatel.c
View file

@ -1112,7 +1112,9 @@ static int stac92xx_build_controls(struct hda_codec *codec)
}
if (spec->multiout.dig_out_nid) {
err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
err = snd_hda_create_spdif_out_ctls(codec,
spec->multiout.dig_out_nid,
spec->multiout.dig_out_nid);
if (err < 0)
return err;
err = snd_hda_create_spdif_share_sw(codec,
@ -3406,30 +3408,9 @@ static hda_nid_t get_connected_node(struct hda_codec *codec, hda_nid_t mux,
return 0;
}
static int get_connection_index(struct hda_codec *codec, hda_nid_t mux,
hda_nid_t nid)
{
hda_nid_t conn[HDA_MAX_NUM_INPUTS];
int i, nums;
if (!(get_wcaps(codec, mux) & AC_WCAP_CONN_LIST))
return -1;
nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn));
for (i = 0; i < nums; i++)
if (conn[i] == nid)
return i;
for (i = 0; i < nums; i++) {
unsigned int wid_caps = get_wcaps(codec, conn[i]);
unsigned int wid_type = get_wcaps_type(wid_caps);
if (wid_type != AC_WID_PIN && wid_type != AC_WID_AUD_MIX)
if (get_connection_index(codec, conn[i], nid) >= 0)
return i;
}
return -1;
}
/* look for NID recursively */
#define get_connection_index(codec, mux, nid) \
snd_hda_get_conn_index(codec, mux, nid, 1)
/* create a volume assigned to the given pin (only if supported) */
/* return 1 if the volume control is created */

6161
sound/pci/hda/patch_via.c
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