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Merge branch 'topic/hda' into for-linus

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This commit is contained in:
Takashi Iwai 2012-03-18 18:22:30 +01:00
commit dbf117cbb9
21 changed files with 1831 additions and 5016 deletions
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8
Documentation/sound/alsa/ALSA-Configuration.txt
View File

@ -860,7 +860,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
[Multiple options for each card instance]
model - force the model name
position_fix - Fix DMA pointer (0 = auto, 1 = use LPIB, 2 = POSBUF)
position_fix - Fix DMA pointer (0 = auto, 1 = use LPIB, 2 = POSBUF,
3 = VIACOMBO, 4 = COMBO)
probe_mask - Bitmask to probe codecs (default = -1, meaning all slots)
When the bit 8 (0x100) is set, the lower 8 bits are used
as the "fixed" codec slots; i.e. the driver probes the
@ -925,6 +926,11 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
(Usually SD_LPIB register is more accurate than the
position buffer.)
position_fix=3 is specific to VIA devices. The position
of the capture stream is checked from both LPIB and POSBUF
values. position_fix=4 is a combination mode, using LPIB
for playback and POSBUF for capture.
NB: If you get many "azx_get_response timeout" messages at
loading, it's likely a problem of interrupts (e.g. ACPI irq
routing). Try to boot with options like "pci=noacpi". Also, you

79
Documentation/sound/alsa/HD-Audio-Models.txt
View File

@ -8,37 +8,10 @@ ALC880
5stack-digout 5-jack in back, 2-jack in front, a SPDIF out
6stack 6-jack in back, 2-jack in front
6stack-digout 6-jack with a SPDIF out
w810 3-jack
z71v 3-jack (HP shared SPDIF)
asus 3-jack (ASUS Mobo)
asus-w1v ASUS W1V
asus-dig ASUS with SPDIF out
asus-dig2 ASUS with SPDIF out (using GPIO2)
uniwill 3-jack
fujitsu Fujitsu Laptops (Pi1536)
F1734 2-jack
lg LG laptop (m1 express dual)
lg-lw LG LW20/LW25 laptop
tcl TCL S700
clevo Clevo laptops (m520G, m665n)
medion Medion Rim 2150
test for testing/debugging purpose, almost all controls can be
adjusted. Appearing only when compiled with
$CONFIG_SND_DEBUG=y
auto auto-config reading BIOS (default)
ALC260
======
fujitsu Fujitsu S7020
acer Acer TravelMate
will Will laptops (PB V7900)
replacer Replacer 672V
favorit100 Maxdata Favorit 100XS
basic fixed pin assignment (old default model)
test for testing/debugging purpose, almost all controls can
adjusted. Appearing only when compiled with
$CONFIG_SND_DEBUG=y
auto auto-config reading BIOS (default)
N/A
ALC262
======
@ -70,55 +43,7 @@ ALC680
ALC882/883/885/888/889
======================
3stack-dig 3-jack with SPDIF I/O
6stack-dig 6-jack digital with SPDIF I/O
arima Arima W820Di1
targa Targa T8, MSI-1049 T8
asus-a7j ASUS A7J
asus-a7m ASUS A7M
macpro MacPro support
mb5 Macbook 5,1
macmini3 Macmini 3,1
mba21 Macbook Air 2,1
mbp3 Macbook Pro rev3
imac24 iMac 24'' with jack detection
imac91 iMac 9,1
w2jc ASUS W2JC
3stack-2ch-dig 3-jack with SPDIF I/O (ALC883)
alc883-6stack-dig 6-jack digital with SPDIF I/O (ALC883)
3stack-6ch 3-jack 6-channel
3stack-6ch-dig 3-jack 6-channel with SPDIF I/O
6stack-dig-demo 6-jack digital for Intel demo board
acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc)
acer-aspire Acer Aspire 9810
acer-aspire-4930g Acer Aspire 4930G
acer-aspire-6530g Acer Aspire 6530G
acer-aspire-7730g Acer Aspire 7730G
acer-aspire-8930g Acer Aspire 8930G
medion Medion Laptops
targa-dig Targa/MSI
targa-2ch-dig Targa/MSI with 2-channel
targa-8ch-dig Targa/MSI with 8-channel (MSI GX620)
laptop-eapd 3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE)
lenovo-101e Lenovo 101E
lenovo-nb0763 Lenovo NB0763
lenovo-ms7195-dig Lenovo MS7195
lenovo-sky Lenovo Sky
haier-w66 Haier W66
3stack-hp HP machines with 3stack (Lucknow, Samba boards)
6stack-dell Dell machines with 6stack (Inspiron 530)
mitac Mitac 8252D
clevo-m540r Clevo M540R (6ch + digital)
clevo-m720 Clevo M720 laptop series
fujitsu-pi2515 Fujitsu AMILO Pi2515
fujitsu-xa3530 Fujitsu AMILO XA3530
3stack-6ch-intel Intel DG33* boards
intel-alc889a Intel IbexPeak with ALC889A
intel-x58 Intel DX58 with ALC889
asus-p5q ASUS P5Q-EM boards
mb31 MacBook 3,1
sony-vaio-tt Sony VAIO TT
auto auto-config reading BIOS (default)
N/A
ALC861/660
==========

7
Documentation/sound/alsa/HD-Audio.txt
View File

@ -59,7 +59,12 @@ a case, you can change the default method via `position_fix` option.
`position_fix=1` means to use LPIB method explicitly.
`position_fix=2` means to use the position-buffer.
`position_fix=3` means to use a combination of both methods, needed
for some VIA and ATI controllers. 0 is the default value for all other
for some VIA controllers. The capture stream position is corrected
by comparing both LPIB and position-buffer values.
`position_fix=4` is another combination available for all controllers,
and uses LPIB for the playback and the position-buffer for the capture
streams.
0 is the default value for all other
controllers, the automatic check and fallback to LPIB as described in
the above. If you get a problem of repeated sounds, this option might
help.

5
include/sound/control.h
View File

@ -227,6 +227,11 @@ snd_ctl_add_slave_uncached(struct snd_kcontrol *master,
return _snd_ctl_add_slave(master, slave, SND_CTL_SLAVE_NEED_UPDATE);
}
int snd_ctl_add_vmaster_hook(struct snd_kcontrol *kctl,
void (*hook)(void *private_data, int),
void *private_data);
void snd_ctl_sync_vmaster_hook(struct snd_kcontrol *kctl);
/*
* Helper functions for jack-detection controls
*/

46
sound/core/vmaster.c
View File

@ -37,6 +37,8 @@ struct link_master {
struct link_ctl_info info;
int val; /* the master value */
unsigned int tlv[4];
void (*hook)(void *private_data, int);
void *hook_private_data;
};
/*
@ -126,7 +128,9 @@ static int master_init(struct link_master *master)
master->info.count = 1; /* always mono */
/* set full volume as default (= no attenuation) */
master->val = master->info.max_val;
return 0;
if (master->hook)
master->hook(master->hook_private_data, master->val);
return 1;
}
return -ENOENT;
}
@ -329,6 +333,8 @@ static int master_put(struct snd_kcontrol *kcontrol,
slave_put_val(slave, uval);
}
kfree(uval);
if (master->hook && !err)
master->hook(master->hook_private_data, master->val);
return 1;
}
@ -408,3 +414,41 @@ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name,
return kctl;
}
EXPORT_SYMBOL(snd_ctl_make_virtual_master);
/**
* snd_ctl_add_vmaster_hook - Add a hook to a vmaster control
* @kcontrol: vmaster kctl element
* @hook: the hook function
*
* Adds the given hook to the vmaster control element so that it's called
* at each time when the value is changed.
*/
int snd_ctl_add_vmaster_hook(struct snd_kcontrol *kcontrol,
void (*hook)(void *private_data, int),
void *private_data)
{
struct link_master *master = snd_kcontrol_chip(kcontrol);
master->hook = hook;
master->hook_private_data = private_data;
return 0;
}
EXPORT_SYMBOL_GPL(snd_ctl_add_vmaster_hook);
/**
* snd_ctl_sync_vmaster_hook - Sync the vmaster hook
* @kcontrol: vmaster kctl element
*
* Call the hook function to synchronize with the current value of the given
* vmaster element. NOP when NULL is passed to @kcontrol or the hook doesn't
* exist.
*/
void snd_ctl_sync_vmaster_hook(struct snd_kcontrol *kcontrol)
{
struct link_master *master;
if (!kcontrol)
return;
master = snd_kcontrol_chip(kcontrol);
if (master->hook)
master->hook(master->hook_private_data, master->val);
}
EXPORT_SYMBOL_GPL(snd_ctl_sync_vmaster_hook);

968
sound/pci/hda/alc260_quirks.c
View File

@ -1,968 +0,0 @@
/*
* ALC260 quirk models
* included by patch_realtek.c
*/
/* ALC260 models */
enum {
ALC260_AUTO,
ALC260_BASIC,
ALC260_FUJITSU_S702X,
ALC260_ACER,
ALC260_WILL,
ALC260_REPLACER_672V,
ALC260_FAVORIT100,
#ifdef CONFIG_SND_DEBUG
ALC260_TEST,
#endif
ALC260_MODEL_LAST /* last tag */
};
static const hda_nid_t alc260_dac_nids[1] = {
/* front */
0x02,
};
static const hda_nid_t alc260_adc_nids[1] = {
/* ADC0 */
0x04,
};
static const hda_nid_t alc260_adc_nids_alt[1] = {
/* ADC1 */
0x05,
};
/* NIDs used when simultaneous access to both ADCs makes sense. Note that
* alc260_capture_mixer assumes ADC0 (nid 0x04) is the first ADC.
*/
static const hda_nid_t alc260_dual_adc_nids[2] = {
/* ADC0, ADC1 */
0x04, 0x05
};
#define ALC260_DIGOUT_NID 0x03
#define ALC260_DIGIN_NID 0x06
static const struct hda_input_mux alc260_capture_source = {
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x1 },
{ "Line", 0x2 },
{ "CD", 0x4 },
},
};
/* On Fujitsu S702x laptops capture only makes sense from Mic/LineIn jack,
* headphone jack and the internal CD lines since these are the only pins at
* which audio can appear. For flexibility, also allow the option of
* recording the mixer output on the second ADC (ADC0 doesn't have a
* connection to the mixer output).
*/
static const struct hda_input_mux alc260_fujitsu_capture_sources[2] = {
{
.num_items = 3,
.items = {
{ "Mic/Line", 0x0 },
{ "CD", 0x4 },
{ "Headphone", 0x2 },
},
},
{
.num_items = 4,
.items = {
{ "Mic/Line", 0x0 },
{ "CD", 0x4 },
{ "Headphone", 0x2 },
{ "Mixer", 0x5 },
},
},
};
/* Acer TravelMate(/Extensa/Aspire) notebooks have similar configuration to
* the Fujitsu S702x, but jacks are marked differently.
*/
static const struct hda_input_mux alc260_acer_capture_sources[2] = {
{
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Line", 0x2 },
{ "CD", 0x4 },
{ "Headphone", 0x5 },
},
},
{
.num_items = 5,
.items = {
{ "Mic", 0x0 },
{ "Line", 0x2 },
{ "CD", 0x4 },
{ "Headphone", 0x6 },
{ "Mixer", 0x5 },
},
},
};
/* Maxdata Favorit 100XS */
static const struct hda_input_mux alc260_favorit100_capture_sources[2] = {
{
.num_items = 2,
.items = {
{ "Line/Mic", 0x0 },
{ "CD", 0x4 },
},
},
{
.num_items = 3,
.items = {
{ "Line/Mic", 0x0 },
{ "CD", 0x4 },
{ "Mixer", 0x5 },
},
},
};
/*
* This is just place-holder, so there's something for alc_build_pcms to look
* at when it calculates the maximum number of channels. ALC260 has no mixer
* element which allows changing the channel mode, so the verb list is
* never used.
*/
static const struct hda_channel_mode alc260_modes[1] = {
{ 2, NULL },
};
/* Mixer combinations
*
* basic: base_output + input + pc_beep + capture
* fujitsu: fujitsu + capture
* acer: acer + capture
*/
static const struct snd_kcontrol_new alc260_base_output_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc260_input_mixer[] = {
HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x07, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x07, 0x01, HDA_INPUT),
{ } /* end */
};
/* Fujitsu S702x series laptops. ALC260 pin usage: Mic/Line jack = 0x12,
* HP jack = 0x14, CD audio = 0x16, internal speaker = 0x10.
*/
static const struct snd_kcontrol_new alc260_fujitsu_mixer[] = {
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Headphone Playback Switch", 0x08, 2, HDA_INPUT),
ALC_PIN_MODE("Headphone Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Mic/Line Playback Volume", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic/Line Playback Switch", 0x07, 0x0, HDA_INPUT),
ALC_PIN_MODE("Mic/Line Jack Mode", 0x12, ALC_PIN_DIR_IN),
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Speaker Playback Switch", 0x09, 2, HDA_INPUT),
{ } /* end */
};
/* Mixer for Acer TravelMate(/Extensa/Aspire) notebooks. Note that current
* versions of the ALC260 don't act on requests to enable mic bias from NID
* 0x0f (used to drive the headphone jack in these laptops). The ALC260
* datasheet doesn't mention this restriction. At this stage it's not clear
* whether this behaviour is intentional or is a hardware bug in chip
* revisions available in early 2006. Therefore for now allow the
* "Headphone Jack Mode" control to span all choices, but if it turns out
* that the lack of mic bias for this NID is intentional we could change the
* mode from ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS.
*
* In addition, Acer TravelMate(/Extensa/Aspire) notebooks in early 2006
* don't appear to make the mic bias available from the "line" jack, even
* though the NID used for this jack (0x14) can supply it. The theory is
* that perhaps Acer have included blocking capacitors between the ALC260
* and the output jack. If this turns out to be the case for all such
* models the "Line Jack Mode" mode could be changed from ALC_PIN_DIR_INOUT
* to ALC_PIN_DIR_INOUT_NOMICBIAS.
*
* The C20x Tablet series have a mono internal speaker which is controlled
* via the chip's Mono sum widget and pin complex, so include the necessary
* controls for such models. On models without a "mono speaker" the control
* won't do anything.
*/
static const struct snd_kcontrol_new alc260_acer_mixer[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT),
ALC_PIN_MODE("Headphone Jack Mode", 0x0f, ALC_PIN_DIR_INOUT),
HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0,
HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Speaker Playback Switch", 0x0a, 1, 2,
HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
{ } /* end */
};
/* Maxdata Favorit 100XS: one output and one input (0x12) jack
*/
static const struct snd_kcontrol_new alc260_favorit100_mixer[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT),
ALC_PIN_MODE("Output Jack Mode", 0x0f, ALC_PIN_DIR_INOUT),
HDA_CODEC_VOLUME("Line/Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Line/Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
ALC_PIN_MODE("Line/Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
{ } /* end */
};
/* Packard bell V7900 ALC260 pin usage: HP = 0x0f, Mic jack = 0x12,
* Line In jack = 0x14, CD audio = 0x16, pc beep = 0x17.
*/
static const struct snd_kcontrol_new alc260_will_mixer[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Master Playback Switch", 0x08, 0x2, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
{ } /* end */
};
/* Replacer 672V ALC260 pin usage: Mic jack = 0x12,
* Line In jack = 0x14, ATAPI Mic = 0x13, speaker = 0x0f.
*/
static const struct snd_kcontrol_new alc260_replacer_672v_mixer[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Master Playback Switch", 0x08, 0x2, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x07, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("ATATI Mic Playback Switch", 0x07, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
{ } /* end */
};
/*
* initialization verbs
*/
static const struct hda_verb alc260_init_verbs[] = {
/* Line In pin widget for input */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* CD pin widget for input */
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* Mic1 (rear panel) pin widget for input and vref at 80% */
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
/* Mic2 (front panel) pin widget for input and vref at 80% */
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
/* LINE-2 is used for line-out in rear */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* select line-out */
{0x0e, AC_VERB_SET_CONNECT_SEL, 0x00},
/* LINE-OUT pin */
{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* enable HP */
{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
/* enable Mono */
{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* mute capture amp left and right */
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* set connection select to line in (default select for this ADC) */
{0x04, AC_VERB_SET_CONNECT_SEL, 0x02},
/* mute capture amp left and right */
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* set connection select to line in (default select for this ADC) */
{0x05, AC_VERB_SET_CONNECT_SEL, 0x02},
/* set vol=0 Line-Out mixer amp left and right */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* unmute pin widget amp left and right (no gain on this amp) */
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* set vol=0 HP mixer amp left and right */
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* unmute pin widget amp left and right (no gain on this amp) */
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* set vol=0 Mono mixer amp left and right */
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* unmute pin widget amp left and right (no gain on this amp) */
{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* unmute LINE-2 out pin */
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 &
* Line In 2 = 0x03
*/
/* mute analog inputs */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
/* mute Front out path */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* mute Headphone out path */
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* mute Mono out path */
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{ }
};
/* Initialisation sequence for ALC260 as configured in Fujitsu S702x
* laptops. ALC260 pin usage: Mic/Line jack = 0x12, HP jack = 0x14, CD
* audio = 0x16, internal speaker = 0x10.
*/
static const struct hda_verb alc260_fujitsu_init_verbs[] = {
/* Disable all GPIOs */
{0x01, AC_VERB_SET_GPIO_MASK, 0},
/* Internal speaker is connected to headphone pin */
{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
/* Headphone/Line-out jack connects to Line1 pin; make it an output */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* Mic/Line-in jack is connected to mic1 pin, so make it an input */
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* Ensure all other unused pins are disabled and muted. */
{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
/* Disable digital (SPDIF) pins */
{0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
{0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
/* Ensure Line1 pin widget takes its input from the OUT1 sum bus
* when acting as an output.
*/
{0x0d, AC_VERB_SET_CONNECT_SEL, 0},
/* Start with output sum widgets muted and their output gains at min */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* Unmute HP pin widget amp left and right (no equiv mixer ctrl) */
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute Line1 pin widget output buffer since it starts as an output.
* If the pin mode is changed by the user the pin mode control will
* take care of enabling the pin's input/output buffers as needed.
* Therefore there's no need to enable the input buffer at this
* stage.
*/
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute input buffer of pin widget used for Line-in (no equiv
* mixer ctrl)
*/
{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Mute capture amp left and right */
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
/* Set ADC connection select to match default mixer setting - line
* in (on mic1 pin)
*/
{0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Do the same for the second ADC: mute capture input amp and
* set ADC connection to line in (on mic1 pin)
*/
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Mute all inputs to mixer widget (even unconnected ones) */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
{ }
};
/* Initialisation sequence for ALC260 as configured in Acer TravelMate and
* similar laptops (adapted from Fujitsu init verbs).
*/
static const struct hda_verb alc260_acer_init_verbs[] = {
/* On TravelMate laptops, GPIO 0 enables the internal speaker and
* the headphone jack. Turn this on and rely on the standard mute
* methods whenever the user wants to turn these outputs off.
*/
{0x01, AC_VERB_SET_GPIO_MASK, 0x01},
{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
{0x01, AC_VERB_SET_GPIO_DATA, 0x01},
/* Internal speaker/Headphone jack is connected to Line-out pin */
{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
/* Internal microphone/Mic jack is connected to Mic1 pin */
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
/* Line In jack is connected to Line1 pin */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* Some Acers (eg: C20x Tablets) use Mono pin for internal speaker */
{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
/* Ensure all other unused pins are disabled and muted. */
{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
/* Disable digital (SPDIF) pins */
{0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
{0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
/* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum
* bus when acting as outputs.
*/
{0x0b, AC_VERB_SET_CONNECT_SEL, 0},
{0x0d, AC_VERB_SET_CONNECT_SEL, 0},
/* Start with output sum widgets muted and their output gains at min */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* Unmute Line-out pin widget amp left and right
* (no equiv mixer ctrl)
*/
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute mono pin widget amp output (no equiv mixer ctrl) */
{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute Mic1 and Line1 pin widget input buffers since they start as
* inputs. If the pin mode is changed by the user the pin mode control
* will take care of enabling the pin's input/output buffers as needed.
* Therefore there's no need to enable the input buffer at this
* stage.
*/
{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Mute capture amp left and right */
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
/* Set ADC connection select to match default mixer setting - mic
* (on mic1 pin)
*/
{0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Do similar with the second ADC: mute capture input amp and
* set ADC connection to mic to match ALSA's default state.
*/
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Mute all inputs to mixer widget (even unconnected ones) */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
{ }
};
/* Initialisation sequence for Maxdata Favorit 100XS
* (adapted from Acer init verbs).
*/
static const struct hda_verb alc260_favorit100_init_verbs[] = {
/* GPIO 0 enables the output jack.
* Turn this on and rely on the standard mute
* methods whenever the user wants to turn these outputs off.
*/
{0x01, AC_VERB_SET_GPIO_MASK, 0x01},
{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
{0x01, AC_VERB_SET_GPIO_DATA, 0x01},
/* Line/Mic input jack is connected to Mic1 pin */
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
/* Ensure all other unused pins are disabled and muted. */
{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
/* Disable digital (SPDIF) pins */
{0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
{0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
/* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum
* bus when acting as outputs.
*/
{0x0b, AC_VERB_SET_CONNECT_SEL, 0},
{0x0d, AC_VERB_SET_CONNECT_SEL, 0},
/* Start with output sum widgets muted and their output gains at min */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* Unmute Line-out pin widget amp left and right
* (no equiv mixer ctrl)
*/
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute Mic1 and Line1 pin widget input buffers since they start as
* inputs. If the pin mode is changed by the user the pin mode control
* will take care of enabling the pin's input/output buffers as needed.
* Therefore there's no need to enable the input buffer at this
* stage.
*/
{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Mute capture amp left and right */
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
/* Set ADC connection select to match default mixer setting - mic
* (on mic1 pin)
*/
{0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Do similar with the second ADC: mute capture input amp and
* set ADC connection to mic to match ALSA's default state.
*/
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Mute all inputs to mixer widget (even unconnected ones) */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
{ }
};
static const struct hda_verb alc260_will_verbs[] = {
{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x0b, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x0d, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x0f, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
{0x1a, AC_VERB_SET_COEF_INDEX, 0x07},
{0x1a, AC_VERB_SET_PROC_COEF, 0x3040},
{}
};
static const struct hda_verb alc260_replacer_672v_verbs[] = {
{0x0f, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
{0x1a, AC_VERB_SET_COEF_INDEX, 0x07},
{0x1a, AC_VERB_SET_PROC_COEF, 0x3050},
{0x01, AC_VERB_SET_GPIO_MASK, 0x01},
{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
{0x01, AC_VERB_SET_GPIO_DATA, 0x00},
{0x0f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
{}
};
/* toggle speaker-output according to the hp-jack state */
static void alc260_replacer_672v_automute(struct hda_codec *codec)
{
unsigned int present;
/* speaker --> GPIO Data 0, hp or spdif --> GPIO data 1 */
present = snd_hda_jack_detect(codec, 0x0f);
if (present) {
snd_hda_codec_write_cache(codec, 0x01, 0,
AC_VERB_SET_GPIO_DATA, 1);
snd_hda_codec_write_cache(codec, 0x0f, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
PIN_HP);
} else {
snd_hda_codec_write_cache(codec, 0x01, 0,
AC_VERB_SET_GPIO_DATA, 0);
snd_hda_codec_write_cache(codec, 0x0f, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
PIN_OUT);
}
}
static void alc260_replacer_672v_unsol_event(struct hda_codec *codec,
unsigned int res)
{
if ((res >> 26) == ALC_HP_EVENT)
alc260_replacer_672v_automute(codec);
}
static const struct hda_verb alc260_hp_dc7600_verbs[] = {
{0x05, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x10, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
{}
};
/* Test configuration for debugging, modelled after the ALC880 test
* configuration.
*/
#ifdef CONFIG_SND_DEBUG
static const hda_nid_t alc260_test_dac_nids[1] = {
0x02,
};
static const hda_nid_t alc260_test_adc_nids[2] = {
0x04, 0x05,
};
/* For testing the ALC260, each input MUX needs its own definition since
* the signal assignments are different. This assumes that the first ADC
* is NID 0x04.
*/
static const struct hda_input_mux alc260_test_capture_sources[2] = {
{
.num_items = 7,
.items = {
{ "MIC1 pin", 0x0 },
{ "MIC2 pin", 0x1 },
{ "LINE1 pin", 0x2 },
{ "LINE2 pin", 0x3 },
{ "CD pin", 0x4 },
{ "LINE-OUT pin", 0x5 },
{ "HP-OUT pin", 0x6 },
},
},
{
.num_items = 8,
.items = {
{ "MIC1 pin", 0x0 },
{ "MIC2 pin", 0x1 },
{ "LINE1 pin", 0x2 },
{ "LINE2 pin", 0x3 },
{ "CD pin", 0x4 },
{ "Mixer", 0x5 },
{ "LINE-OUT pin", 0x6 },
{ "HP-OUT pin", 0x7 },
},
},
};
static const struct snd_kcontrol_new alc260_test_mixer[] = {
/* Output driver widgets */
HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT),
HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x09, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("LOUT2 Playback Switch", 0x09, 2, HDA_INPUT),
HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x08, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("LOUT1 Playback Switch", 0x08, 2, HDA_INPUT),
/* Modes for retasking pin widgets
* Note: the ALC260 doesn't seem to act on requests to enable mic
* bias from NIDs 0x0f and 0x10. The ALC260 datasheet doesn't
* mention this restriction. At this stage it's not clear whether
* this behaviour is intentional or is a hardware bug in chip
* revisions available at least up until early 2006. Therefore for
* now allow the "HP-OUT" and "LINE-OUT" Mode controls to span all
* choices, but if it turns out that the lack of mic bias for these
* NIDs is intentional we could change their modes from
* ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS.
*/
ALC_PIN_MODE("HP-OUT pin mode", 0x10, ALC_PIN_DIR_INOUT),
ALC_PIN_MODE("LINE-OUT pin mode", 0x0f, ALC_PIN_DIR_INOUT),
ALC_PIN_MODE("LINE2 pin mode", 0x15, ALC_PIN_DIR_INOUT),
ALC_PIN_MODE("LINE1 pin mode", 0x14, ALC_PIN_DIR_INOUT),
ALC_PIN_MODE("MIC2 pin mode", 0x13, ALC_PIN_DIR_INOUT),
ALC_PIN_MODE("MIC1 pin mode", 0x12, ALC_PIN_DIR_INOUT),
/* Loopback mixer controls */
HDA_CODEC_VOLUME("MIC1 Playback Volume", 0x07, 0x00, HDA_INPUT),
HDA_CODEC_MUTE("MIC1 Playback Switch", 0x07, 0x00, HDA_INPUT),
HDA_CODEC_VOLUME("MIC2 Playback Volume", 0x07, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("MIC2 Playback Switch", 0x07, 0x01, HDA_INPUT),
HDA_CODEC_VOLUME("LINE1 Playback Volume", 0x07, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("LINE1 Playback Switch", 0x07, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("LINE2 Playback Volume", 0x07, 0x03, HDA_INPUT),
HDA_CODEC_MUTE("LINE2 Playback Switch", 0x07, 0x03, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("LINE-OUT loopback Playback Volume", 0x07, 0x06, HDA_INPUT),
HDA_CODEC_MUTE("LINE-OUT loopback Playback Switch", 0x07, 0x06, HDA_INPUT),
HDA_CODEC_VOLUME("HP-OUT loopback Playback Volume", 0x07, 0x7, HDA_INPUT),
HDA_CODEC_MUTE("HP-OUT loopback Playback Switch", 0x07, 0x7, HDA_INPUT),
/* Controls for GPIO pins, assuming they are configured as outputs */
ALC_GPIO_DATA_SWITCH("GPIO pin 0", 0x01, 0x01),
ALC_GPIO_DATA_SWITCH("GPIO pin 1", 0x01, 0x02),
ALC_GPIO_DATA_SWITCH("GPIO pin 2", 0x01, 0x04),
ALC_GPIO_DATA_SWITCH("GPIO pin 3", 0x01, 0x08),
/* Switches to allow the digital IO pins to be enabled. The datasheet
* is ambigious as to which NID is which; testing on laptops which
* make this output available should provide clarification.
*/
ALC_SPDIF_CTRL_SWITCH("SPDIF Playback Switch", 0x03, 0x01),
ALC_SPDIF_CTRL_SWITCH("SPDIF Capture Switch", 0x06, 0x01),
/* 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 */
};
static const struct hda_verb alc260_test_init_verbs[] = {
/* Enable all GPIOs as outputs with an initial value of 0 */
{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x0f},
{0x01, AC_VERB_SET_GPIO_DATA, 0x00},
{0x01, AC_VERB_SET_GPIO_MASK, 0x0f},
/* Enable retasking pins as output, initially without power amp */
{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* Disable digital (SPDIF) pins initially, but users can enable
* them via a mixer switch. In the case of SPDIF-out, this initverb
* payload also sets the generation to 0, output to be in "consumer"
* PCM format, copyright asserted, no pre-emphasis and no validity
* control.
*/
{0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
{0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
/* Ensure mic1, mic2, line1 and line2 pin widgets take input from the
* OUT1 sum bus when acting as an output.
*/
{0x0b, AC_VERB_SET_CONNECT_SEL, 0},
{0x0c, AC_VERB_SET_CONNECT_SEL, 0},
{0x0d, AC_VERB_SET_CONNECT_SEL, 0},
{0x0e, AC_VERB_SET_CONNECT_SEL, 0},
/* Start with output sum widgets muted and their output gains at min */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* Unmute retasking pin widget output buffers since the default
* state appears to be output. As the pin mode is changed by the
* user the pin mode control will take care of enabling the pin's
* input/output buffers as needed.
*/
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Also unmute the mono-out pin widget */
{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Mute capture amp left and right */
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
/* Set ADC connection select to match default mixer setting (mic1
* pin)
*/
{0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Do the same for the second ADC: mute capture input amp and
* set ADC connection to mic1 pin
*/
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Mute all inputs to mixer widget (even unconnected ones) */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
{ }
};
#endif
/*
* ALC260 configurations
*/
static const char * const alc260_models[ALC260_MODEL_LAST] = {
[ALC260_BASIC] = "basic",
[ALC260_FUJITSU_S702X] = "fujitsu",
[ALC260_ACER] = "acer",
[ALC260_WILL] = "will",
[ALC260_REPLACER_672V] = "replacer",
[ALC260_FAVORIT100] = "favorit100",
#ifdef CONFIG_SND_DEBUG
[ALC260_TEST] = "test",
#endif
[ALC260_AUTO] = "auto",
};
static const struct snd_pci_quirk alc260_cfg_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x007b, "Acer C20x", ALC260_ACER),
SND_PCI_QUIRK(0x1025, 0x007f, "Acer", ALC260_WILL),
SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER),
SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100),
SND_PCI_QUIRK(0x104d, 0x81bb, "Sony VAIO", ALC260_BASIC),
SND_PCI_QUIRK(0x104d, 0x81cc, "Sony VAIO", ALC260_BASIC),
SND_PCI_QUIRK(0x104d, 0x81cd, "Sony VAIO", ALC260_BASIC),
SND_PCI_QUIRK(0x10cf, 0x1326, "Fujitsu S702X", ALC260_FUJITSU_S702X),
SND_PCI_QUIRK(0x152d, 0x0729, "CTL U553W", ALC260_BASIC),
SND_PCI_QUIRK(0x161f, 0x2057, "Replacer 672V", ALC260_REPLACER_672V),
SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_WILL),
{}
};
static const struct alc_config_preset alc260_presets[] = {
[ALC260_BASIC] = {
.mixers = { alc260_base_output_mixer,
alc260_input_mixer },
.init_verbs = { alc260_init_verbs },
.num_dacs = ARRAY_SIZE(alc260_dac_nids),
.dac_nids = alc260_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
.adc_nids = alc260_dual_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
.input_mux = &alc260_capture_source,
},
[ALC260_FUJITSU_S702X] = {
.mixers = { alc260_fujitsu_mixer },
.init_verbs = { alc260_fujitsu_init_verbs },
.num_dacs = ARRAY_SIZE(alc260_dac_nids),
.dac_nids = alc260_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
.adc_nids = alc260_dual_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
.num_mux_defs = ARRAY_SIZE(alc260_fujitsu_capture_sources),
.input_mux = alc260_fujitsu_capture_sources,
},
[ALC260_ACER] = {
.mixers = { alc260_acer_mixer },
.init_verbs = { alc260_acer_init_verbs },
.num_dacs = ARRAY_SIZE(alc260_dac_nids),
.dac_nids = alc260_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
.adc_nids = alc260_dual_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
.num_mux_defs = ARRAY_SIZE(alc260_acer_capture_sources),
.input_mux = alc260_acer_capture_sources,
},
[ALC260_FAVORIT100] = {
.mixers = { alc260_favorit100_mixer },
.init_verbs = { alc260_favorit100_init_verbs },
.num_dacs = ARRAY_SIZE(alc260_dac_nids),
.dac_nids = alc260_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
.adc_nids = alc260_dual_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
.num_mux_defs = ARRAY_SIZE(alc260_favorit100_capture_sources),
.input_mux = alc260_favorit100_capture_sources,
},
[ALC260_WILL] = {
.mixers = { alc260_will_mixer },
.init_verbs = { alc260_init_verbs, alc260_will_verbs },
.num_dacs = ARRAY_SIZE(alc260_dac_nids),
.dac_nids = alc260_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc260_adc_nids),
.adc_nids = alc260_adc_nids,
.dig_out_nid = ALC260_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
.input_mux = &alc260_capture_source,
},
[ALC260_REPLACER_672V] = {
.mixers = { alc260_replacer_672v_mixer },
.init_verbs = { alc260_init_verbs, alc260_replacer_672v_verbs },
.num_dacs = ARRAY_SIZE(alc260_dac_nids),
.dac_nids = alc260_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc260_adc_nids),
.adc_nids = alc260_adc_nids,
.dig_out_nid = ALC260_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
.input_mux = &alc260_capture_source,
.unsol_event = alc260_replacer_672v_unsol_event,
.init_hook = alc260_replacer_672v_automute,
},
#ifdef CONFIG_SND_DEBUG
[ALC260_TEST] = {
.mixers = { alc260_test_mixer },
.init_verbs = { alc260_test_init_verbs },
.num_dacs = ARRAY_SIZE(alc260_test_dac_nids),
.dac_nids = alc260_test_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc260_test_adc_nids),
.adc_nids = alc260_test_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
.num_mux_defs = ARRAY_SIZE(alc260_test_capture_sources),
.input_mux = alc260_test_capture_sources,
},
#endif
};

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sound/pci/hda/alc880_quirks.c
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sound/pci/hda/alc882_quirks.c
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@ -1,866 +0,0 @@
/*
* ALC882/ALC883/ALC888/ALC889 quirk models
* included by patch_realtek.c
*/
/* ALC882 models */
enum {
ALC882_AUTO,
ALC885_MBA21,
ALC885_MBP3,
ALC885_MB5,
ALC885_MACMINI3,
ALC885_IMAC91,
ALC889A_MB31,
ALC882_MODEL_LAST,
};
#define ALC882_DIGOUT_NID 0x06
#define ALC882_DIGIN_NID 0x0a
#define ALC883_DIGOUT_NID ALC882_DIGOUT_NID
#define ALC883_DIGIN_NID ALC882_DIGIN_NID
#define ALC1200_DIGOUT_NID 0x10
static const struct hda_channel_mode alc882_ch_modes[1] = {
{ 8, NULL }
};
/* DACs */
static const hda_nid_t alc882_dac_nids[4] = {
/* front, rear, clfe, rear_surr */
0x02, 0x03, 0x04, 0x05
};
#define alc883_dac_nids alc882_dac_nids
/* ADCs */
#define alc882_adc_nids alc880_adc_nids
#define alc882_adc_nids_alt alc880_adc_nids_alt
#define alc883_adc_nids alc882_adc_nids_alt
static const hda_nid_t alc882_capsrc_nids_alt[2] = { 0x23, 0x22 };
#define alc883_capsrc_nids alc882_capsrc_nids_alt
/* input MUX */
/* FIXME: should be a matrix-type input source selection */
static const struct hda_input_mux alc882_capture_source = {
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x1 },
{ "Line", 0x2 },
{ "CD", 0x4 },
},
};
#define alc883_capture_source alc882_capture_source
static const struct hda_input_mux mb5_capture_source = {
.num_items = 3,
.items = {
{ "Mic", 0x1 },
{ "Line", 0x7 },
{ "CD", 0x4 },
},
};
static const struct hda_input_mux macmini3_capture_source = {
.num_items = 2,
.items = {
{ "Line", 0x2 },
{ "CD", 0x4 },
},
};
static const struct hda_input_mux alc883_3stack_6ch_intel = {
.num_items = 4,
.items = {
{ "Mic", 0x1 },
{ "Front Mic", 0x0 },
{ "Line", 0x2 },
{ "CD", 0x4 },
},
};
static const struct hda_input_mux alc889A_mb31_capture_source = {
.num_items = 2,
.items = {
{ "Mic", 0x0 },
/* Front Mic (0x01) unused */
{ "Line", 0x2 },
/* Line 2 (0x03) unused */
/* CD (0x04) unused? */
},
};
static const struct hda_input_mux alc889A_imac91_capture_source = {
.num_items = 2,
.items = {
{ "Mic", 0x01 },
{ "Line", 0x2 }, /* Not sure! */
},
};
/* Macbook Air 2,1 */
static const struct hda_channel_mode alc885_mba21_ch_modes[1] = {
{ 2, NULL },
};
/*
* macbook pro ALC885 can switch LineIn to LineOut without losing Mic
*/
/*
* 2ch mode
*/
static const struct hda_verb alc885_mbp_ch2_init[] = {
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{ } /* end */
};
/*
* 4ch mode
*/
static const struct hda_verb alc885_mbp_ch4_init[] = {
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{ } /* end */
};
static const struct hda_channel_mode alc885_mbp_4ch_modes[2] = {
{ 2, alc885_mbp_ch2_init },
{ 4, alc885_mbp_ch4_init },
};
/*
* 2ch
* Speakers/Woofer/HP = Front
* LineIn = Input
*/
static const struct hda_verb alc885_mb5_ch2_init[] = {
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{ } /* end */
};
/*
* 6ch mode
* Speakers/HP = Front
* Woofer = LFE
* LineIn = Surround
*/
static const struct hda_verb alc885_mb5_ch6_init[] = {
{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},
{ } /* end */
};
static const struct hda_channel_mode alc885_mb5_6ch_modes[2] = {
{ 2, alc885_mb5_ch2_init },
{ 6, alc885_mb5_ch6_init },
};
#define alc885_macmini3_6ch_modes alc885_mb5_6ch_modes
/* Macbook Air 2,1 same control for HP and internal Speaker */
static const struct snd_kcontrol_new alc885_mba21_mixer[] = {
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 0x02, HDA_OUTPUT),
{ }
};
static const struct snd_kcontrol_new alc885_mbp3_mixer[] = {
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE ("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0e, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE ("Headphone Playback Switch", 0x0e, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, 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, 0x00, HDA_INPUT),
HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT),
HDA_CODEC_VOLUME("Line Boost Volume", 0x1a, 0x00, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0x00, HDA_INPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc885_mb5_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x07, HDA_INPUT),
HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_VOLUME("Line Boost Volume", 0x15, 0x00, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x19, 0x00, HDA_INPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc885_macmini3_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x07, HDA_INPUT),
HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT),
HDA_CODEC_VOLUME("Line Boost Volume", 0x15, 0x00, HDA_INPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc885_imac91_mixer[] = {
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc882_chmode_mixer[] = {
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
},
{ } /* end */
};
static const struct hda_verb alc882_base_init_verbs[] = {
/* Front mixer: unmute input/output amp left and right (volume = 0) */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Rear mixer */
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* CLFE mixer */
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Side mixer */
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Front Pin: output 0 (0x0c) */
{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},
/* FIXME: use matrix-type input source selection */
/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
/* Input mixer2 */
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Input mixer3 */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* ADC2: mute amp left and right */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
/* ADC3: mute amp left and right */
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
{ }
};
#define alc883_init_verbs alc882_base_init_verbs
/* Macbook 5,1 */
static const struct hda_verb alc885_mb5_init_verbs[] = {
/* DACs */
{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},
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Front mixer */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Surround mixer */
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* LFE mixer */
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* HP mixer */
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Front Pin (0x0c) */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
/* LFE Pin (0x0e) */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1a, AC_VERB_SET_CONNECT_SEL, 0x02},
/* HP Pin (0x0f) */
{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, 0x03},
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
/* 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 */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0x1)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x7)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x4)},
{ }
};
/* Macmini 3,1 */
static const struct hda_verb alc885_macmini3_init_verbs[] = {
/* DACs */
{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},
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Front mixer */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Surround mixer */
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* LFE mixer */
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* HP mixer */
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Front Pin (0x0c) */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
/* LFE Pin (0x0e) */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1a, AC_VERB_SET_CONNECT_SEL, 0x02},
/* HP Pin (0x0f) */
{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, 0x03},
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
/* Line In pin */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
{ }
};
static const struct hda_verb alc885_mba21_init_verbs[] = {
/*Internal and HP Speaker Mixer*/
{0x0c, 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)},
/*Internal Speaker Pin (0x0c)*/
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) },
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
/* HP Pin: output 0 (0x0e) */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, (ALC_HP_EVENT | AC_USRSP_EN)},
/* Line in (is hp when jack connected)*/
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_VREF_50},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{ }
};
/* Macbook Pro rev3 */
static const struct hda_verb alc885_mbp3_init_verbs[] = {
/* Front mixer: unmute input/output amp left and right (volume = 0) */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Rear mixer */
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* HP mixer */
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Front Pin: output 0 (0x0c) */
{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},
/* HP Pin: output 0 (0x0e) */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x02},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
/* 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: use output 1 when in LineOut mode */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
/* FIXME: use matrix-type input source selection */
/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* Input mixer2 */
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* Input mixer3 */
{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)},
/* ADC1: mute amp left and right */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
/* ADC2: mute amp left and right */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
/* ADC3: mute amp left and right */
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
{ }
};
/* iMac 9,1 */
static const struct hda_verb alc885_imac91_init_verbs[] = {
/* Internal Speaker Pin (0x0c) */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) },
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) },
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
/* HP Pin: Rear */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, (ALC_HP_EVENT | AC_USRSP_EN)},
/* Line in Rear */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_VREF_50},
{0x15, 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},
/* Rear mixer */
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Line-Out mixer: unmute input/output amp left and right (volume = 0) */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* 0x24 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-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)},
/* 0x07 [Audio Input] wcaps 0x10011b: Stereo Amp-In */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
/* 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
/* 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In */
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
{ }
};
/* Toggle speaker-output according to the hp-jack state */
static void alc885_imac24_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x18;
spec->autocfg.speaker_pins[1] = 0x1a;
alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
#define alc885_mb5_setup alc885_imac24_setup
#define alc885_macmini3_setup alc885_imac24_setup
/* Macbook Air 2,1 */
static void alc885_mba21_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x18;
alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
static void alc885_mbp3_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
static void alc885_imac91_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x18;
spec->autocfg.speaker_pins[1] = 0x1a;
alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
/* 2ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:front) */
static const struct hda_verb alc889A_mb31_ch2_init[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP as front */
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Line as input */
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Line off */
{ } /* end */
};
/* 4ch mode (Speaker:front, Subwoofer:CLFE, Line:CLFE, Headphones:front) */
static const struct hda_verb alc889A_mb31_ch4_init[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP as front */
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Line as output */
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Line on */
{ } /* end */
};
/* 5ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:rear) */
static const struct hda_verb alc889A_mb31_ch5_init[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* HP as rear */
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Line as input */
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Line off */
{ } /* end */
};
/* 6ch mode (Speaker:front, Subwoofer:off, Line:CLFE, Headphones:Rear) */
static const struct hda_verb alc889A_mb31_ch6_init[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* HP as front */
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Subwoofer off */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Line as output */
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Line on */
{ } /* end */
};
static const struct hda_channel_mode alc889A_mb31_6ch_modes[4] = {
{ 2, alc889A_mb31_ch2_init },
{ 4, alc889A_mb31_ch4_init },
{ 5, alc889A_mb31_ch5_init },
{ 6, alc889A_mb31_ch6_init },
};
static const struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 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_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc889A_mb31_mixer[] = {
/* Output mixers */
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x00,
HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x02, HDA_INPUT),
/* Output switches */
HDA_CODEC_MUTE("Enable Speaker", 0x14, 0x00, HDA_OUTPUT),
HDA_CODEC_MUTE("Enable Headphones", 0x15, 0x00, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Enable LFE", 0x16, 2, 0x00, HDA_OUTPUT),
/* Boost mixers */
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0x00, HDA_INPUT),
HDA_CODEC_VOLUME("Line Boost Volume", 0x1a, 0x00, HDA_INPUT),
/* Input mixers */
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc883_chmode_mixer[] = {
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
},
{ } /* end */
};
static const struct hda_verb alc889A_mb31_verbs[] = {
/* Init rear pin (used as headphone output) */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4}, /* Apple Headphones */
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Connect to front */
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
/* Init line pin (used as output in 4ch and 6ch mode) */
{0x1a, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Connect to CLFE */
/* Init line 2 pin (used as headphone out by default) */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Use as input */
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Mute output */
{ } /* end */
};
/* Mute speakers according to the headphone jack state */
static void alc889A_mb31_automute(struct hda_codec *codec)
{
unsigned int present;
/* Mute only in 2ch or 4ch mode */
if (snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_CONNECT_SEL, 0)
== 0x00) {
present = snd_hda_jack_detect(codec, 0x15);
snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
}
}
static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res)
{
if ((res >> 26) == ALC_HP_EVENT)
alc889A_mb31_automute(codec);
}
static void alc882_unsol_event(struct hda_codec *codec, unsigned int res)
{
alc_exec_unsol_event(codec, res >> 26);
}
/*
* configuration and preset
*/
static const char * const alc882_models[ALC882_MODEL_LAST] = {
[ALC885_MB5] = "mb5",
[ALC885_MACMINI3] = "macmini3",
[ALC885_MBA21] = "mba21",
[ALC885_MBP3] = "mbp3",
[ALC885_IMAC91] = "imac91",
[ALC889A_MB31] = "mb31",
[ALC882_AUTO] = "auto",
};
/* codec SSID table for Intel Mac */
static const struct snd_pci_quirk alc882_ssid_cfg_tbl[] = {
SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC885_MBP3),
SND_PCI_QUIRK(0x106b, 0x00a1, "Macbook", ALC885_MBP3),
SND_PCI_QUIRK(0x106b, 0x00a4, "MacbookPro 4,1", ALC885_MBP3),
SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC885_MBP3),
SND_PCI_QUIRK(0x106b, 0x3000, "iMac", ALC889A_MB31),
SND_PCI_QUIRK(0x106b, 0x3400, "MacBookAir 1,1", ALC885_MBP3),
SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC885_MBA21),
SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889A_MB31),
SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC885_MBP3),
SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC885_IMAC91),
SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC885_MB5),
SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC885_MB5),
/* FIXME: HP jack sense seems not working for MBP 5,1 or 5,2,
* so apparently no perfect solution yet
*/
SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5),
SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC885_MB5),
SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC885_MACMINI3),
{} /* terminator */
};
static const struct alc_config_preset alc882_presets[] = {
[ALC885_MBA21] = {
.mixers = { alc885_mba21_mixer },
.init_verbs = { alc885_mba21_init_verbs, alc880_gpio1_init_verbs },
.num_dacs = 2,
.dac_nids = alc882_dac_nids,
.channel_mode = alc885_mba21_ch_modes,
.num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes),
.input_mux = &alc882_capture_source,
.unsol_event = alc882_unsol_event,
.setup = alc885_mba21_setup,
.init_hook = alc_hp_automute,
},
[ALC885_MBP3] = {
.mixers = { alc885_mbp3_mixer, alc882_chmode_mixer },
.init_verbs = { alc885_mbp3_init_verbs,
alc880_gpio1_init_verbs },
.num_dacs = 2,
.dac_nids = alc882_dac_nids,
.hp_nid = 0x04,
.channel_mode = alc885_mbp_4ch_modes,
.num_channel_mode = ARRAY_SIZE(alc885_mbp_4ch_modes),
.input_mux = &alc882_capture_source,
.dig_out_nid = ALC882_DIGOUT_NID,
.dig_in_nid = ALC882_DIGIN_NID,
.unsol_event = alc882_unsol_event,
.setup = alc885_mbp3_setup,
.init_hook = alc_hp_automute,
},
[ALC885_MB5] = {
.mixers = { alc885_mb5_mixer, alc882_chmode_mixer },
.init_verbs = { alc885_mb5_init_verbs,
alc880_gpio1_init_verbs },
.num_dacs = ARRAY_SIZE(alc882_dac_nids),
.dac_nids = alc882_dac_nids,
.channel_mode = alc885_mb5_6ch_modes,
.num_channel_mode = ARRAY_SIZE(alc885_mb5_6ch_modes),
.input_mux = &mb5_capture_source,
.dig_out_nid = ALC882_DIGOUT_NID,
.dig_in_nid = ALC882_DIGIN_NID,
.unsol_event = alc882_unsol_event,
.setup = alc885_mb5_setup,
.init_hook = alc_hp_automute,
},
[ALC885_MACMINI3] = {
.mixers = { alc885_macmini3_mixer, alc882_chmode_mixer },
.init_verbs = { alc885_macmini3_init_verbs,
alc880_gpio1_init_verbs },
.num_dacs = ARRAY_SIZE(alc882_dac_nids),
.dac_nids = alc882_dac_nids,
.channel_mode = alc885_macmini3_6ch_modes,
.num_channel_mode = ARRAY_SIZE(alc885_macmini3_6ch_modes),
.input_mux = &macmini3_capture_source,
.dig_out_nid = ALC882_DIGOUT_NID,
.dig_in_nid = ALC882_DIGIN_NID,
.unsol_event = alc882_unsol_event,
.setup = alc885_macmini3_setup,
.init_hook = alc_hp_automute,
},
[ALC885_IMAC91] = {
.mixers = {alc885_imac91_mixer},
.init_verbs = { alc885_imac91_init_verbs,
alc880_gpio1_init_verbs },
.num_dacs = ARRAY_SIZE(alc882_dac_nids),
.dac_nids = alc882_dac_nids,
.channel_mode = alc885_mba21_ch_modes,
.num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes),
.input_mux = &alc889A_imac91_capture_source,
.dig_out_nid = ALC882_DIGOUT_NID,
.dig_in_nid = ALC882_DIGIN_NID,
.unsol_event = alc882_unsol_event,
.setup = alc885_imac91_setup,
.init_hook = alc_hp_automute,
},
[ALC889A_MB31] = {
.mixers = { alc889A_mb31_mixer, alc883_chmode_mixer},
.init_verbs = { alc883_init_verbs, alc889A_mb31_verbs,
alc880_gpio1_init_verbs },
.adc_nids = alc883_adc_nids,
.num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
.capsrc_nids = alc883_capsrc_nids,
.dac_nids = alc883_dac_nids,
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.channel_mode = alc889A_mb31_6ch_modes,
.num_channel_mode = ARRAY_SIZE(alc889A_mb31_6ch_modes),
.input_mux = &alc889A_mb31_capture_source,
.dig_out_nid = ALC883_DIGOUT_NID,
.unsol_event = alc889A_mb31_unsol_event,
.init_hook = alc889A_mb31_automute,
},
};

480
sound/pci/hda/alc_quirks.c
View File

@ -1,480 +0,0 @@
/*
* 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);
}
static void alc_simple_setup_automute(struct alc_spec *spec, int mode)
{
int lo_pin = spec->autocfg.line_out_pins[0];
if (lo_pin == spec->autocfg.speaker_pins[0] ||
lo_pin == spec->autocfg.hp_pins[0])
lo_pin = 0;
spec->automute_mode = mode;
spec->detect_hp = !!spec->autocfg.hp_pins[0];
spec->detect_lo = !!lo_pin;
spec->automute_lo = spec->automute_lo_possible = !!lo_pin;
spec->automute_speaker = spec->automute_speaker_possible = !!spec->autocfg.speaker_pins[0];
}
/* 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);
}

192
sound/pci/hda/hda_codec.c
View File

@ -19,6 +19,7 @@
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include <linux/mm.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/slab.h>
@ -2304,7 +2305,7 @@ typedef int (*map_slave_func_t)(void *, struct snd_kcontrol *);
/* apply the function to all matching slave ctls in the mixer list */
static int map_slaves(struct hda_codec *codec, const char * const *slaves,
map_slave_func_t func, void *data)
const char *suffix, map_slave_func_t func, void *data)
{
struct hda_nid_item *items;
const char * const *s;
@ -2317,7 +2318,14 @@ static int map_slaves(struct hda_codec *codec, const char * const *slaves,
sctl->id.iface != SNDRV_CTL_ELEM_IFACE_MIXER)
continue;
for (s = slaves; *s; s++) {
if (!strcmp(sctl->id.name, *s)) {
char tmpname[sizeof(sctl->id.name)];
const char *name = *s;
if (suffix) {
snprintf(tmpname, sizeof(tmpname), "%s %s",
name, suffix);
name = tmpname;
}
if (!strcmp(sctl->id.name, name)) {
err = func(data, sctl);
if (err)
return err;
@ -2333,12 +2341,65 @@ static int check_slave_present(void *data, struct snd_kcontrol *sctl)
return 1;
}
/* guess the value corresponding to 0dB */
static int get_kctl_0dB_offset(struct snd_kcontrol *kctl)
{
int _tlv[4];
const int *tlv = NULL;
int val = -1;
if (kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) {
/* FIXME: set_fs() hack for obtaining user-space TLV data */
mm_segment_t fs = get_fs();
set_fs(get_ds());
if (!kctl->tlv.c(kctl, 0, sizeof(_tlv), _tlv))
tlv = _tlv;
set_fs(fs);
} else if (kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_READ)
tlv = kctl->tlv.p;
if (tlv && tlv[0] == SNDRV_CTL_TLVT_DB_SCALE)
val = -tlv[2] / tlv[3];
return val;
}
/* call kctl->put with the given value(s) */
static int put_kctl_with_value(struct snd_kcontrol *kctl, int val)
{
struct snd_ctl_elem_value *ucontrol;
ucontrol = kzalloc(sizeof(*ucontrol), GFP_KERNEL);
if (!ucontrol)
return -ENOMEM;
ucontrol->value.integer.value[0] = val;
ucontrol->value.integer.value[1] = val;
kctl->put(kctl, ucontrol);
kfree(ucontrol);
return 0;
}
/* initialize the slave volume with 0dB */
static int init_slave_0dB(void *data, struct snd_kcontrol *slave)
{
int offset = get_kctl_0dB_offset(slave);
if (offset > 0)
put_kctl_with_value(slave, offset);
return 0;
}
/* unmute the slave */
static int init_slave_unmute(void *data, struct snd_kcontrol *slave)
{
return put_kctl_with_value(slave, 1);
}
/**
* snd_hda_add_vmaster - create a virtual master control and add slaves
* @codec: HD-audio codec
* @name: vmaster control name
* @tlv: TLV data (optional)
* @slaves: slave control names (optional)
* @suffix: suffix string to each slave name (optional)
* @init_slave_vol: initialize slaves to unmute/0dB
* @ctl_ret: store the vmaster kcontrol in return
*
* Create a virtual master control with the given name. The TLV data
* must be either NULL or a valid data.
@ -2349,13 +2410,18 @@ static int check_slave_present(void *data, struct snd_kcontrol *sctl)
*
* This function returns zero if successful or a negative error code.
*/
int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
unsigned int *tlv, const char * const *slaves)
int __snd_hda_add_vmaster(struct hda_codec *codec, char *name,
unsigned int *tlv, const char * const *slaves,
const char *suffix, bool init_slave_vol,
struct snd_kcontrol **ctl_ret)
{
struct snd_kcontrol *kctl;
int err;
err = map_slaves(codec, slaves, check_slave_present, NULL);
if (ctl_ret)
*ctl_ret = NULL;
err = map_slaves(codec, slaves, suffix, check_slave_present, NULL);
if (err != 1) {
snd_printdd("No slave found for %s\n", name);
return 0;
@ -2367,13 +2433,119 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
if (err < 0)
return err;
err = map_slaves(codec, slaves, (map_slave_func_t)snd_ctl_add_slave,
kctl);
err = map_slaves(codec, slaves, suffix,
(map_slave_func_t)snd_ctl_add_slave, kctl);
if (err < 0)
return err;
/* init with master mute & zero volume */
put_kctl_with_value(kctl, 0);
if (init_slave_vol)
map_slaves(codec, slaves, suffix,
tlv ? init_slave_0dB : init_slave_unmute, kctl);
if (ctl_ret)
*ctl_ret = kctl;
return 0;
}
EXPORT_SYMBOL_HDA(snd_hda_add_vmaster);
EXPORT_SYMBOL_HDA(__snd_hda_add_vmaster);
/*
* mute-LED control using vmaster
*/
static int vmaster_mute_mode_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
static const char * const texts[] = {
"Off", "On", "Follow Master"
};
unsigned int index;
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = 3;
index = uinfo->value.enumerated.item;
if (index >= 3)
index = 2;
strcpy(uinfo->value.enumerated.name, texts[index]);
return 0;
}
static int vmaster_mute_mode_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_vmaster_mute_hook *hook = snd_kcontrol_chip(kcontrol);
ucontrol->value.enumerated.item[0] = hook->mute_mode;
return 0;
}
static int vmaster_mute_mode_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_vmaster_mute_hook *hook = snd_kcontrol_chip(kcontrol);
unsigned int old_mode = hook->mute_mode;
hook->mute_mode = ucontrol->value.enumerated.item[0];
if (hook->mute_mode > HDA_VMUTE_FOLLOW_MASTER)
hook->mute_mode = HDA_VMUTE_FOLLOW_MASTER;
if (old_mode == hook->mute_mode)
return 0;
snd_hda_sync_vmaster_hook(hook);
return 1;
}
static struct snd_kcontrol_new vmaster_mute_mode = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Mute-LED Mode",
.info = vmaster_mute_mode_info,
.get = vmaster_mute_mode_get,
.put = vmaster_mute_mode_put,
};
/*
* Add a mute-LED hook with the given vmaster switch kctl
* "Mute-LED Mode" control is automatically created and associated with
* the given hook.
*/
int snd_hda_add_vmaster_hook(struct hda_codec *codec,
struct hda_vmaster_mute_hook *hook,
bool expose_enum_ctl)
{
struct snd_kcontrol *kctl;
if (!hook->hook || !hook->sw_kctl)
return 0;
snd_ctl_add_vmaster_hook(hook->sw_kctl, hook->hook, codec);
hook->codec = codec;
hook->mute_mode = HDA_VMUTE_FOLLOW_MASTER;
if (!expose_enum_ctl)
return 0;
kctl = snd_ctl_new1(&vmaster_mute_mode, hook);
if (!kctl)
return -ENOMEM;
return snd_hda_ctl_add(codec, 0, kctl);
}
EXPORT_SYMBOL_HDA(snd_hda_add_vmaster_hook);
/*
* Call the hook with the current value for synchronization
* Should be called in init callback
*/
void snd_hda_sync_vmaster_hook(struct hda_vmaster_mute_hook *hook)
{
if (!hook->hook || !hook->codec)
return;
switch (hook->mute_mode) {
case HDA_VMUTE_FOLLOW_MASTER:
snd_ctl_sync_vmaster_hook(hook->sw_kctl);
break;
default:
hook->hook(hook->codec, hook->mute_mode);
break;
}
}
EXPORT_SYMBOL_HDA(snd_hda_sync_vmaster_hook);
/**
* snd_hda_mixer_amp_switch_info - Info callback for a standard AMP mixer switch
@ -5272,6 +5444,10 @@ int snd_hda_suspend(struct hda_bus *bus)
list_for_each_entry(codec, &bus->codec_list, list) {
if (hda_codec_is_power_on(codec))
hda_call_codec_suspend(codec);
else /* forcibly change the power to D3 even if not used */
hda_set_power_state(codec,
codec->afg ? codec->afg : codec->mfg,
AC_PWRST_D3);
if (codec->patch_ops.post_suspend)
codec->patch_ops.post_suspend(codec);
}

1
sound/pci/hda/hda_codec.h
View File

@ -855,6 +855,7 @@ struct hda_codec {
unsigned int pins_shutup:1; /* pins are shut up */
unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */
unsigned int ignore_misc_bit:1; /* ignore MISC_NO_PRESENCE bit */
unsigned int no_jack_detect:1; /* Machine has no jack-detection */
#ifdef CONFIG_SND_HDA_POWER_SAVE
unsigned int power_on :1; /* current (global) power-state */
unsigned int power_transition :1; /* power-state in transition */

52
sound/pci/hda/hda_intel.c
View File

@ -84,7 +84,7 @@ module_param_array(model, charp, NULL, 0444);
MODULE_PARM_DESC(model, "Use the given board model.");
module_param_array(position_fix, int, NULL, 0444);
MODULE_PARM_DESC(position_fix, "DMA pointer read method."
"(0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO).");
"(0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO).");
module_param_array(bdl_pos_adj, int, NULL, 0644);
MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset.");
module_param_array(probe_mask, int, NULL, 0444);
@ -94,7 +94,7 @@ MODULE_PARM_DESC(probe_only, "Only probing and no codec initialization.");
module_param(single_cmd, bool, 0444);
MODULE_PARM_DESC(single_cmd, "Use single command to communicate with codecs "
"(for debugging only).");
module_param(enable_msi, int, 0444);
module_param(enable_msi, bint, 0444);
MODULE_PARM_DESC(enable_msi, "Enable Message Signaled Interrupt (MSI)");
#ifdef CONFIG_SND_HDA_PATCH_LOADER
module_param_array(patch, charp, NULL, 0444);
@ -121,8 +121,8 @@ module_param(power_save_controller, bool, 0644);
MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode.");
#endif
static bool align_buffer_size = 1;
module_param(align_buffer_size, bool, 0644);
static int align_buffer_size = -1;
module_param(align_buffer_size, bint, 0644);
MODULE_PARM_DESC(align_buffer_size,
"Force buffer and period sizes to be multiple of 128 bytes.");
@ -148,6 +148,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6},"
"{Intel, PCH},"
"{Intel, CPT},"
"{Intel, PPT},"
"{Intel, LPT},"
"{Intel, PBG},"
"{Intel, SCH},"
"{ATI, SB450},"
@ -329,6 +330,7 @@ enum {
POS_FIX_LPIB,
POS_FIX_POSBUF,
POS_FIX_VIACOMBO,
POS_FIX_COMBO,
};
/* Defines for ATI HD Audio support in SB450 south bridge */
@ -515,6 +517,7 @@ enum {
#define AZX_DCAPS_SYNC_WRITE (1 << 19) /* sync each cmd write */
#define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */
#define AZX_DCAPS_BUFSIZE (1 << 21) /* no buffer size alignment */
#define AZX_DCAPS_ALIGN_BUFSIZE (1 << 22) /* buffer size alignment */
/* quirks for ATI SB / AMD Hudson */
#define AZX_DCAPS_PRESET_ATI_SB \
@ -527,7 +530,8 @@ enum {
/* quirks for Nvidia */
#define AZX_DCAPS_PRESET_NVIDIA \
(AZX_DCAPS_NVIDIA_SNOOP | AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI)
(AZX_DCAPS_NVIDIA_SNOOP | AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI |\
AZX_DCAPS_ALIGN_BUFSIZE)
static char *driver_short_names[] __devinitdata = {
[AZX_DRIVER_ICH] = "HDA Intel",
@ -2347,17 +2351,6 @@ static void azx_power_notify(struct hda_bus *bus)
* power management
*/
static int snd_hda_codecs_inuse(struct hda_bus *bus)
{
struct hda_codec *codec;
list_for_each_entry(codec, &bus->codec_list, list) {
if (snd_hda_codec_needs_resume(codec))
return 1;
}
return 0;
}
static int azx_suspend(struct pci_dev *pci, pm_message_t state)
{
struct snd_card *card = pci_get_drvdata(pci);
@ -2404,8 +2397,7 @@ static int azx_resume(struct pci_dev *pci)
return -EIO;
azx_init_pci(chip);
if (snd_hda_codecs_inuse(chip->bus))
azx_init_chip(chip, 1);
azx_init_chip(chip, 1);
snd_hda_resume(chip->bus);
snd_power_change_state(card, SNDRV_CTL_POWER_D0);
@ -2517,6 +2509,7 @@ static int __devinit check_position_fix(struct azx *chip, int fix)
case POS_FIX_LPIB:
case POS_FIX_POSBUF:
case POS_FIX_VIACOMBO:
case POS_FIX_COMBO:
return fix;
}
@ -2696,6 +2689,12 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
chip->position_fix[0] = chip->position_fix[1] =
check_position_fix(chip, position_fix[dev]);
/* combo mode uses LPIB for playback */
if (chip->position_fix[0] == POS_FIX_COMBO) {
chip->position_fix[0] = POS_FIX_LPIB;
chip->position_fix[1] = POS_FIX_AUTO;
}
check_probe_mask(chip, dev);
chip->single_cmd = single_cmd;
@ -2774,9 +2773,16 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
}
/* disable buffer size rounding to 128-byte multiples if supported */
chip->align_buffer_size = align_buffer_size;
if (chip->driver_caps & AZX_DCAPS_BUFSIZE)
chip->align_buffer_size = 0;
if (align_buffer_size >= 0)
chip->align_buffer_size = !!align_buffer_size;
else {
if (chip->driver_caps & AZX_DCAPS_BUFSIZE)
chip->align_buffer_size = 0;
else if (chip->driver_caps & AZX_DCAPS_ALIGN_BUFSIZE)
chip->align_buffer_size = 1;
else
chip->align_buffer_size = 1;
}
/* allow 64bit DMA address if supported by H/W */
if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64)))
@ -2992,6 +2998,10 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
{ PCI_DEVICE(0x8086, 0x1e20),
.driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
AZX_DCAPS_BUFSIZE},
/* Lynx Point */
{ PCI_DEVICE(0x8086, 0x8c20),
.driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
AZX_DCAPS_BUFSIZE},
/* SCH */
{ PCI_DEVICE(0x8086, 0x811b),
.driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP |

16
sound/pci/hda/hda_jack.c
View File

@ -19,6 +19,22 @@
#include "hda_local.h"
#include "hda_jack.h"
bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid)
{
if (codec->no_jack_detect)
return false;
if (!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT))
return false;
if (!codec->ignore_misc_bit &&
(get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) &
AC_DEFCFG_MISC_NO_PRESENCE))
return false;
if (!(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP))
return false;
return true;
}
EXPORT_SYMBOL_HDA(is_jack_detectable);
/* execute pin sense measurement */
static u32 read_pin_sense(struct hda_codec *codec, hda_nid_t nid)
{

13
sound/pci/hda/hda_jack.h
View File

@ -62,18 +62,7 @@ int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid,
u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid);
int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid);
static inline bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid)
{
if (!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT))
return false;
if (!codec->ignore_misc_bit &&
(get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) &
AC_DEFCFG_MISC_NO_PRESENCE))
return false;
if (!(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP))
return false;
return true;
}
bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid);
int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid,
const char *name, int idx);

30
sound/pci/hda/hda_local.h
View File

@ -139,10 +139,36 @@ void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir,
unsigned int *tlv);
struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec,
const char *name);
int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
unsigned int *tlv, const char * const *slaves);
int __snd_hda_add_vmaster(struct hda_codec *codec, char *name,
unsigned int *tlv, const char * const *slaves,
const char *suffix, bool init_slave_vol,
struct snd_kcontrol **ctl_ret);
#define snd_hda_add_vmaster(codec, name, tlv, slaves, suffix) \
__snd_hda_add_vmaster(codec, name, tlv, slaves, suffix, true, NULL)
int snd_hda_codec_reset(struct hda_codec *codec);
enum {
HDA_VMUTE_OFF,
HDA_VMUTE_ON,
HDA_VMUTE_FOLLOW_MASTER,
};
struct hda_vmaster_mute_hook {
/* below two fields must be filled by the caller of
* snd_hda_add_vmaster_hook() beforehand
*/
struct snd_kcontrol *sw_kctl;
void (*hook)(void *, int);
/* below are initialized automatically */
unsigned int mute_mode; /* HDA_VMUTE_XXX */
struct hda_codec *codec;
};
int snd_hda_add_vmaster_hook(struct hda_codec *codec,
struct hda_vmaster_mute_hook *hook,
bool expose_enum_ctl);
void snd_hda_sync_vmaster_hook(struct hda_vmaster_mute_hook *hook);
/* amp value bits */
#define HDA_AMP_MUTE 0x80
#define HDA_AMP_UNMUTE 0x00

72
sound/pci/hda/patch_analog.c
View File

@ -82,6 +82,7 @@ struct ad198x_spec {
unsigned int inv_jack_detect: 1;/* inverted jack-detection */
unsigned int inv_eapd: 1; /* inverted EAPD implementation */
unsigned int analog_beep: 1; /* analog beep input present */
unsigned int avoid_init_slave_vol:1;
#ifdef CONFIG_SND_HDA_POWER_SAVE
struct hda_loopback_check loopback;
@ -137,51 +138,17 @@ static int ad198x_init(struct hda_codec *codec)
return 0;
}
static const char * const ad_slave_vols[] = {
"Front Playback Volume",
"Surround Playback Volume",
"Center Playback Volume",
"LFE Playback Volume",
"Side Playback Volume",
"Headphone Playback Volume",
"Mono Playback Volume",
"Speaker Playback Volume",
"IEC958 Playback Volume",
static const char * const ad_slave_pfxs[] = {
"Front", "Surround", "Center", "LFE", "Side",
"Headphone", "Mono", "Speaker", "IEC958",
NULL
};
static const char * const ad_slave_sws[] = {
"Front Playback Switch",
"Surround Playback Switch",
"Center Playback Switch",
"LFE Playback Switch",
"Side Playback Switch",
"Headphone Playback Switch",
"Mono Playback Switch",
"Speaker Playback Switch",
"IEC958 Playback Switch",
static const char * const ad1988_6stack_fp_slave_pfxs[] = {
"Front", "Surround", "Center", "LFE", "Side", "IEC958",
NULL
};
static const char * const ad1988_6stack_fp_slave_vols[] = {
"Front Playback Volume",
"Surround Playback Volume",
"Center Playback Volume",
"LFE Playback Volume",
"Side Playback Volume",
"IEC958 Playback Volume",
NULL
};
static const char * const ad1988_6stack_fp_slave_sws[] = {
"Front Playback Switch",
"Surround Playback Switch",
"Center Playback Switch",
"LFE Playback Switch",
"Side Playback Switch",
"IEC958 Playback Switch",
NULL
};
static void ad198x_free_kctls(struct hda_codec *codec);
#ifdef CONFIG_SND_HDA_INPUT_BEEP
@ -257,10 +224,12 @@ static int ad198x_build_controls(struct hda_codec *codec)
unsigned int vmaster_tlv[4];
snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid,
HDA_OUTPUT, vmaster_tlv);
err = snd_hda_add_vmaster(codec, "Master Playback Volume",
err = __snd_hda_add_vmaster(codec, "Master Playback Volume",
vmaster_tlv,
(spec->slave_vols ?
spec->slave_vols : ad_slave_vols));
spec->slave_vols : ad_slave_pfxs),
"Playback Volume",
!spec->avoid_init_slave_vol, NULL);
if (err < 0)
return err;
}
@ -268,7 +237,8 @@ static int ad198x_build_controls(struct hda_codec *codec)
err = snd_hda_add_vmaster(codec, "Master Playback Switch",
NULL,
(spec->slave_sws ?
spec->slave_sws : ad_slave_sws));
spec->slave_sws : ad_slave_pfxs),
"Playback Switch");
if (err < 0)
return err;
}
@ -3385,8 +3355,8 @@ static int patch_ad1988(struct hda_codec *codec)
if (spec->autocfg.hp_pins[0]) {
spec->mixers[spec->num_mixers++] = ad1988_hp_mixers;
spec->slave_vols = ad1988_6stack_fp_slave_vols;
spec->slave_sws = ad1988_6stack_fp_slave_sws;
spec->slave_vols = ad1988_6stack_fp_slave_pfxs;
spec->slave_sws = ad1988_6stack_fp_slave_pfxs;
spec->alt_dac_nid = ad1988_alt_dac_nid;
spec->stream_analog_alt_playback =
&ad198x_pcm_analog_alt_playback;
@ -3594,16 +3564,8 @@ static const struct hda_amp_list ad1884_loopbacks[] = {
#endif
static const char * const ad1884_slave_vols[] = {
"PCM Playback Volume",
"Mic Playback Volume",
"Mono Playback Volume",
"Front Mic Playback Volume",
"Mic Playback Volume",
"CD Playback Volume",
"Internal Mic Playback Volume",
"Docking Mic Playback Volume",
/* "Beep Playback Volume", */
"IEC958 Playback Volume",
"PCM", "Mic", "Mono", "Front Mic", "Mic", "CD",
"Internal Mic", "Docking Mic", /* "Beep", */ "IEC958",
NULL
};
@ -3644,6 +3606,8 @@ static int patch_ad1884(struct hda_codec *codec)
spec->vmaster_nid = 0x04;
/* we need to cover all playback volumes */
spec->slave_vols = ad1884_slave_vols;
/* slaves may contain input volumes, so we can't raise to 0dB blindly */
spec->avoid_init_slave_vol = 1;
codec->patch_ops = ad198x_patch_ops;

122
sound/pci/hda/patch_conexant.c
View File

@ -70,6 +70,8 @@ struct conexant_spec {
const struct snd_kcontrol_new *mixers[5];
int num_mixers;
hda_nid_t vmaster_nid;
struct hda_vmaster_mute_hook vmaster_mute;
bool vmaster_mute_led;
const struct hda_verb *init_verbs[5]; /* initialization verbs
* don't forget NULL
@ -465,21 +467,8 @@ static const struct snd_kcontrol_new cxt_beep_mixer[] = {
};
#endif
static const char * const slave_vols[] = {
"Headphone Playback Volume",
"Speaker Playback Volume",
"Front Playback Volume",
"Surround Playback Volume",
"CLFE Playback Volume",
NULL
};
static const char * const slave_sws[] = {
"Headphone Playback Switch",
"Speaker Playback Switch",
"Front Playback Switch",
"Surround Playback Switch",
"CLFE Playback Switch",
static const char * const slave_pfxs[] = {
"Headphone", "Speaker", "Front", "Surround", "CLFE",
NULL
};
@ -519,14 +508,17 @@ static int conexant_build_controls(struct hda_codec *codec)
snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid,
HDA_OUTPUT, vmaster_tlv);
err = snd_hda_add_vmaster(codec, "Master Playback Volume",
vmaster_tlv, slave_vols);
vmaster_tlv, slave_pfxs,
"Playback Volume");
if (err < 0)
return err;
}
if (spec->vmaster_nid &&
!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
err = snd_hda_add_vmaster(codec, "Master Playback Switch",
NULL, slave_sws);
err = __snd_hda_add_vmaster(codec, "Master Playback Switch",
NULL, slave_pfxs,
"Playback Switch", true,
&spec->vmaster_mute.sw_kctl);
if (err < 0)
return err;
}
@ -3034,7 +3026,6 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS),
SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS),
SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO),
SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */
SND_PCI_QUIRK(0x1b0a, 0x2092, "CyberpowerPC Gamer Xplorer N57001", CXT5066_AUTO),
{}
};
@ -3943,6 +3934,63 @@ static void enable_unsol_pins(struct hda_codec *codec, int num_pins,
snd_hda_jack_detect_enable(codec, pins[i], action);
}
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;
}
/* is the given NID found in any of autocfg items? */
static bool found_in_autocfg(struct auto_pin_cfg *cfg, hda_nid_t nid)
{
int i;
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) ||
found_in_nid_list(nid, cfg->dig_out_pins, cfg->dig_outs))
return true;
for (i = 0; i < cfg->num_inputs; i++)
if (cfg->inputs[i].pin == nid)
return true;
if (cfg->dig_in_pin == nid)
return true;
return false;
}
/* clear unsol-event tags on unused pins; Conexant codecs seem to leave
* invalid unsol tags by some reason
*/
static void clear_unsol_on_unused_pins(struct hda_codec *codec)
{
struct conexant_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
int i;
for (i = 0; i < codec->init_pins.used; i++) {
struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i);
if (!found_in_autocfg(cfg, pin->nid))
snd_hda_codec_write(codec, pin->nid, 0,
AC_VERB_SET_UNSOLICITED_ENABLE, 0);
}
}
/* turn on/off EAPD according to Master switch */
static void cx_auto_vmaster_hook(void *private_data, int enabled)
{
struct hda_codec *codec = private_data;
struct conexant_spec *spec = codec->spec;
if (enabled && spec->pin_eapd_ctrls) {
cx_auto_update_speakers(codec);
return;
}
cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, enabled);
}
static void cx_auto_init_output(struct hda_codec *codec)
{
struct conexant_spec *spec = codec->spec;
@ -3983,6 +4031,7 @@ static void cx_auto_init_output(struct hda_codec *codec)
/* turn on all EAPDs if no individual EAPD control is available */
if (!spec->pin_eapd_ctrls)
cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, true);
clear_unsol_on_unused_pins(codec);
}
static void cx_auto_init_input(struct hda_codec *codec)
@ -4046,11 +4095,13 @@ static void cx_auto_init_digital(struct hda_codec *codec)
static int cx_auto_init(struct hda_codec *codec)
{
struct conexant_spec *spec = codec->spec;
/*snd_hda_sequence_write(codec, cx_auto_init_verbs);*/
cx_auto_init_output(codec);
cx_auto_init_input(codec);
cx_auto_init_digital(codec);
snd_hda_jack_report_sync(codec);
snd_hda_sync_vmaster_hook(&spec->vmaster_mute);
return 0;
}
@ -4296,6 +4347,13 @@ static int cx_auto_build_controls(struct hda_codec *codec)
err = snd_hda_jack_add_kctls(codec, &spec->autocfg);
if (err < 0)
return err;
if (spec->vmaster_mute.sw_kctl) {
spec->vmaster_mute.hook = cx_auto_vmaster_hook;
err = snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute,
spec->vmaster_mute_led);
if (err < 0)
return err;
}
return 0;
}
@ -4320,7 +4378,6 @@ static int cx_auto_search_adcs(struct hda_codec *codec)
return 0;
}
static const struct hda_codec_ops cx_auto_patch_ops = {
.build_controls = cx_auto_build_controls,
.build_pcms = conexant_build_pcms,
@ -4368,6 +4425,7 @@ static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = {
{ 0x16, 0x042140ff }, /* HP (seq# overridden) */
{ 0x17, 0x21a11000 }, /* dock-mic */
{ 0x19, 0x2121103f }, /* dock-HP */
{ 0x1c, 0x21440100 }, /* dock SPDIF out */
{}
};
@ -4421,6 +4479,18 @@ static int patch_conexant_auto(struct hda_codec *codec)
apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl);
/* Show mute-led control only on HP laptops
* This is a sort of white-list: on HP laptops, EAPD corresponds
* only to the mute-LED without actualy amp function. Meanwhile,
* others may use EAPD really as an amp switch, so it might be
* not good to expose it blindly.
*/
switch (codec->subsystem_id >> 16) {
case 0x103c:
spec->vmaster_mute_led = 1;
break;
}
err = cx_auto_search_adcs(codec);
if (err < 0)
return err;
@ -4434,6 +4504,18 @@ static int patch_conexant_auto(struct hda_codec *codec)
codec->patch_ops = cx_auto_patch_ops;
if (spec->beep_amp)
snd_hda_attach_beep_device(codec, spec->beep_amp);
/* Some laptops with Conexant chips show stalls in S3 resume,
* which falls into the single-cmd mode.
* Better to make reset, then.
*/
if (!codec->bus->sync_write) {
snd_printd("hda_codec: "
"Enable sync_write for stable communication\n");
codec->bus->sync_write = 1;
codec->bus->allow_bus_reset = 1;
}
return 0;
}

2
sound/pci/hda/patch_hdmi.c
View File

@ -1912,6 +1912,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = {
{ .id = 0x80862804, .name = "IbexPeak HDMI", .patch = patch_generic_hdmi },
{ .id = 0x80862805, .name = "CougarPoint HDMI", .patch = patch_generic_hdmi },
{ .id = 0x80862806, .name = "PantherPoint HDMI", .patch = patch_generic_hdmi },
{ .id = 0x80862880, .name = "CedarTrail HDMI", .patch = patch_generic_hdmi },
{ .id = 0x808629fb, .name = "Crestline HDMI", .patch = patch_generic_hdmi },
{} /* terminator */
};
@ -1958,6 +1959,7 @@ MODULE_ALIAS("snd-hda-codec-id:80862803");
MODULE_ALIAS("snd-hda-codec-id:80862804");
MODULE_ALIAS("snd-hda-codec-id:80862805");
MODULE_ALIAS("snd-hda-codec-id:80862806");
MODULE_ALIAS("snd-hda-codec-id:80862880");
MODULE_ALIAS("snd-hda-codec-id:808629fb");
MODULE_LICENSE("GPL");

1930
sound/pci/hda/patch_realtek.c
View File

File diff suppressed because it is too large Load Diff

203
sound/pci/hda/patch_sigmatel.c
View File

@ -99,6 +99,7 @@ enum {
STAC_DELL_VOSTRO_3500,
STAC_92HD83XXX_HP_cNB11_INTQUAD,
STAC_HP_DV7_4000,
STAC_HP_ZEPHYR,
STAC_92HD83XXX_MODELS
};
@ -309,6 +310,8 @@ struct sigmatel_spec {
unsigned long auto_capvols[MAX_ADCS_NUM];
unsigned auto_dmic_cnt;
hda_nid_t auto_dmic_nids[MAX_DMICS_NUM];
struct hda_vmaster_mute_hook vmaster_mute;
};
static const hda_nid_t stac9200_adc_nids[1] = {
@ -662,7 +665,6 @@ static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol,
return 0;
}
#ifdef CONFIG_SND_HDA_POWER_SAVE
static int stac_vrefout_set(struct hda_codec *codec,
hda_nid_t nid, unsigned int new_vref)
{
@ -686,7 +688,6 @@ static int stac_vrefout_set(struct hda_codec *codec,
return 1;
}
#endif
static unsigned int stac92xx_vref_set(struct hda_codec *codec,
hda_nid_t nid, unsigned int new_vref)
@ -894,6 +895,13 @@ static const struct hda_verb stac92hd83xxx_core_init[] = {
{}
};
static const struct hda_verb stac92hd83xxx_hp_zephyr_init[] = {
{ 0x22, 0x785, 0x43 },
{ 0x22, 0x782, 0xe0 },
{ 0x22, 0x795, 0x00 },
{}
};
static const struct hda_verb stac92hd71bxx_core_init[] = {
/* set master volume and direct control */
{ 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
@ -999,8 +1007,8 @@ static const struct hda_verb stac9205_core_init[] = {
}
static const struct snd_kcontrol_new stac9200_mixer[] = {
HDA_CODEC_VOLUME_MIN_MUTE("Master Playback Volume", 0xb, 0, HDA_OUTPUT),
HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MIN_MUTE("PCM Playback Volume", 0xb, 0, HDA_OUTPUT),
HDA_CODEC_MUTE("PCM Playback Switch", 0xb, 0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Capture Volume", 0x0a, 0, HDA_OUTPUT),
HDA_CODEC_MUTE("Capture Switch", 0x0a, 0, HDA_OUTPUT),
{ } /* end */
@ -1027,8 +1035,8 @@ static const struct snd_kcontrol_new stac92hd71bxx_loopback[] = {
};
static const struct snd_kcontrol_new stac925x_mixer[] = {
HDA_CODEC_VOLUME_MIN_MUTE("Master Playback Volume", 0xe, 0, HDA_OUTPUT),
HDA_CODEC_MUTE("Master Playback Switch", 0x0e, 0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MIN_MUTE("PCM Playback Volume", 0xe, 0, HDA_OUTPUT),
HDA_CODEC_MUTE("PCM Playback Switch", 0x0e, 0, HDA_OUTPUT),
{ } /* end */
};
@ -1060,34 +1068,25 @@ static struct snd_kcontrol_new stac_smux_mixer = {
.put = stac92xx_smux_enum_put,
};
static const char * const slave_vols[] = {
"Front Playback Volume",
"Surround Playback Volume",
"Center Playback Volume",
"LFE Playback Volume",
"Side Playback Volume",
"Headphone Playback Volume",
"Speaker Playback Volume",
static const char * const slave_pfxs[] = {
"Front", "Surround", "Center", "LFE", "Side",
"Headphone", "Speaker", "IEC958",
NULL
};
static const char * const slave_sws[] = {
"Front Playback Switch",
"Surround Playback Switch",
"Center Playback Switch",
"LFE Playback Switch",
"Side Playback Switch",
"Headphone Playback Switch",
"Speaker Playback Switch",
"IEC958 Playback Switch",
NULL
};
static void stac92xx_update_led_status(struct hda_codec *codec, int enabled);
static void stac92xx_vmaster_hook(void *private_data, int val)
{
stac92xx_update_led_status(private_data, val);
}
static void stac92xx_free_kctls(struct hda_codec *codec);
static int stac92xx_build_controls(struct hda_codec *codec)
{
struct sigmatel_spec *spec = codec->spec;
unsigned int vmaster_tlv[4];
int err;
int i;
@ -1144,22 +1143,28 @@ static int stac92xx_build_controls(struct hda_codec *codec)
}
/* if we have no master control, let's create it */
if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) {
unsigned int vmaster_tlv[4];
snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0],
HDA_OUTPUT, vmaster_tlv);
/* correct volume offset */
vmaster_tlv[2] += vmaster_tlv[3] * spec->volume_offset;
/* minimum value is actually mute */
vmaster_tlv[3] |= TLV_DB_SCALE_MUTE;
err = snd_hda_add_vmaster(codec, "Master Playback Volume",
vmaster_tlv, slave_vols);
if (err < 0)
return err;
}
if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
err = snd_hda_add_vmaster(codec, "Master Playback Switch",
NULL, slave_sws);
snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0],
HDA_OUTPUT, vmaster_tlv);
/* correct volume offset */
vmaster_tlv[2] += vmaster_tlv[3] * spec->volume_offset;
/* minimum value is actually mute */
vmaster_tlv[3] |= TLV_DB_SCALE_MUTE;
err = snd_hda_add_vmaster(codec, "Master Playback Volume",
vmaster_tlv, slave_pfxs,
"Playback Volume");
if (err < 0)
return err;
err = __snd_hda_add_vmaster(codec, "Master Playback Switch",
NULL, slave_pfxs,
"Playback Switch", true,
&spec->vmaster_mute.sw_kctl);
if (err < 0)
return err;
if (spec->gpio_led) {
spec->vmaster_mute.hook = stac92xx_vmaster_hook;
err = snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute, true);
if (err < 0)
return err;
}
@ -1636,6 +1641,12 @@ static const unsigned int hp_dv7_4000_pin_configs[10] = {
0x40f000f0, 0x40f000f0,
};
static const unsigned int hp_zephyr_pin_configs[10] = {
0x01813050, 0x0421201f, 0x04a1205e, 0x96130310,
0x96130310, 0x0101401f, 0x1111611f, 0xd5a30130,
0, 0,
};
static const unsigned int hp_cNB11_intquad_pin_configs[10] = {
0x40f000f0, 0x0221101f, 0x02a11020, 0x92170110,
0x40f000f0, 0x92170110, 0x40f000f0, 0xd5a30130,
@ -1649,6 +1660,7 @@ static const unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = {
[STAC_DELL_VOSTRO_3500] = dell_vostro_3500_pin_configs,
[STAC_92HD83XXX_HP_cNB11_INTQUAD] = hp_cNB11_intquad_pin_configs,
[STAC_HP_DV7_4000] = hp_dv7_4000_pin_configs,
[STAC_HP_ZEPHYR] = hp_zephyr_pin_configs,
};
static const char * const stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = {
@ -1659,6 +1671,7 @@ static const char * const stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = {
[STAC_DELL_VOSTRO_3500] = "dell-vostro-3500",
[STAC_92HD83XXX_HP_cNB11_INTQUAD] = "hp_cNB11_intquad",
[STAC_HP_DV7_4000] = "hp-dv7-4000",
[STAC_HP_ZEPHYR] = "hp-zephyr",
};
static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
@ -1711,6 +1724,14 @@ static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
"HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3593,
"HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3561,
"HP", STAC_HP_ZEPHYR),
{} /* terminator */
};
static const struct snd_pci_quirk stac92hd83xxx_codec_id_cfg_tbl[] = {
SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3561,
"HP", STAC_HP_ZEPHYR),
{} /* terminator */
};
@ -4410,8 +4431,7 @@ static int stac92xx_init(struct hda_codec *codec)
snd_hda_jack_report_sync(codec);
/* sync mute LED */
if (spec->gpio_led)
hda_call_check_power_status(codec, 0x01);
snd_hda_sync_vmaster_hook(&spec->vmaster_mute);
if (spec->dac_list)
stac92xx_power_down(codec);
return 0;
@ -4989,7 +5009,6 @@ static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state)
return 0;
}
#ifdef CONFIG_SND_HDA_POWER_SAVE
static int stac92xx_pre_resume(struct hda_codec *codec)
{
struct sigmatel_spec *spec = codec->spec;
@ -5024,83 +5043,41 @@ static void stac92xx_set_power_state(struct hda_codec *codec, hda_nid_t fg,
afg_power_state);
snd_hda_codec_set_power_to_all(codec, fg, power_state, true);
}
#else
#define stac92xx_suspend NULL
#define stac92xx_resume NULL
#define stac92xx_pre_resume NULL
#define stac92xx_set_power_state NULL
#endif /* CONFIG_PM */
/*
* For this feature CONFIG_SND_HDA_POWER_SAVE is needed
* as mute LED state is updated in check_power_status hook
*/
static int stac92xx_update_led_status(struct hda_codec *codec)
/* update mute-LED accoring to the master switch */
static void stac92xx_update_led_status(struct hda_codec *codec, int enabled)
{
struct sigmatel_spec *spec = codec->spec;
int i, num_ext_dacs, muted = 1;
unsigned int muted_lvl, notmtd_lvl;
hda_nid_t nid;
int muted = !enabled;
if (!spec->gpio_led)
return 0;
return;
/* LED state is inverted on these systems */
if (spec->gpio_led_polarity)
muted = !muted;
for (i = 0; i < spec->multiout.num_dacs; i++) {
nid = spec->multiout.dac_nids[i];
if (!(snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) &
HDA_AMP_MUTE)) {
muted = 0; /* something heard */
break;
}
}
if (muted && spec->multiout.hp_nid)
if (!(snd_hda_codec_amp_read(codec,
spec->multiout.hp_nid, 0, HDA_OUTPUT, 0) &
HDA_AMP_MUTE)) {
muted = 0; /* HP is not muted */
}
num_ext_dacs = ARRAY_SIZE(spec->multiout.extra_out_nid);
for (i = 0; muted && i < num_ext_dacs; i++) {
nid = spec->multiout.extra_out_nid[i];
if (nid == 0)
break;
if (!(snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) &
HDA_AMP_MUTE)) {
muted = 0; /* extra output is not muted */
}
}
/*polarity defines *not* muted state level*/
if (!spec->vref_mute_led_nid) {
if (muted)
spec->gpio_data &= ~spec->gpio_led; /* orange */
else
spec->gpio_data |= spec->gpio_led; /* white */
if (!spec->gpio_led_polarity) {
/* LED state is inverted on these systems */
spec->gpio_data ^= spec->gpio_led;
}
stac_gpio_set(codec, spec->gpio_mask,
spec->gpio_dir, spec->gpio_data);
} else {
notmtd_lvl = spec->gpio_led_polarity ?
AC_PINCTL_VREF_50 : AC_PINCTL_VREF_GRD;
muted_lvl = spec->gpio_led_polarity ?
AC_PINCTL_VREF_GRD : AC_PINCTL_VREF_50;
spec->vref_led = muted ? muted_lvl : notmtd_lvl;
spec->vref_led = muted ? AC_PINCTL_VREF_50 : AC_PINCTL_VREF_GRD;
stac_vrefout_set(codec, spec->vref_mute_led_nid,
spec->vref_led);
}
return 0;
}
/*
* use power check for controlling mute led of HP notebooks
*/
static int stac92xx_check_power_status(struct hda_codec *codec,
hda_nid_t nid)
{
stac92xx_update_led_status(codec);
return 0;
}
#endif /* CONFIG_SND_HDA_POWER_SAVE */
#endif /* CONFIG_PM */
static const struct hda_codec_ops stac92xx_patch_ops = {
.build_controls = stac92xx_build_controls,
.build_pcms = stac92xx_build_pcms,
@ -5580,6 +5557,12 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
STAC_92HD83XXX_MODELS,
stac92hd83xxx_models,
stac92hd83xxx_cfg_tbl);
/* check codec subsystem id if not found */
if (spec->board_config < 0)
spec->board_config =
snd_hda_check_board_codec_sid_config(codec,
STAC_92HD83XXX_MODELS, stac92hd83xxx_models,
stac92hd83xxx_codec_id_cfg_tbl);
again:
if (spec->board_config < 0)
snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
@ -5590,12 +5573,17 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
codec->patch_ops = stac92xx_patch_ops;
switch (spec->board_config) {
case STAC_HP_ZEPHYR:
spec->init = stac92hd83xxx_hp_zephyr_init;
break;
}
if (find_mute_led_cfg(codec, -1/*no default cfg*/))
snd_printd("mute LED gpio %d polarity %d\n",
spec->gpio_led,
spec->gpio_led_polarity);
#ifdef CONFIG_SND_HDA_POWER_SAVE
if (spec->gpio_led) {
if (!spec->vref_mute_led_nid) {
spec->gpio_mask |= spec->gpio_led;
@ -5605,11 +5593,10 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
codec->patch_ops.set_power_state =
stac92xx_set_power_state;
}
#ifdef CONFIG_PM
codec->patch_ops.pre_resume = stac92xx_pre_resume;
codec->patch_ops.check_power_status =
stac92xx_check_power_status;
#endif
}
#endif
err = stac92xx_parse_auto_config(codec);
if (!err) {
@ -5906,7 +5893,6 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
spec->gpio_led,
spec->gpio_led_polarity);
#ifdef CONFIG_SND_HDA_POWER_SAVE
if (spec->gpio_led) {
if (!spec->vref_mute_led_nid) {
spec->gpio_mask |= spec->gpio_led;
@ -5916,11 +5902,10 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
codec->patch_ops.set_power_state =
stac92xx_set_power_state;
}
#ifdef CONFIG_PM
codec->patch_ops.pre_resume = stac92xx_pre_resume;
codec->patch_ops.check_power_status =
stac92xx_check_power_status;
#endif
}
#endif
spec->multiout.dac_nids = spec->dac_nids;

48
sound/pci/hda/patch_via.c
View File

@ -550,7 +550,10 @@ static void via_auto_init_output(struct hda_codec *codec,
pin = path->path[path->depth - 1];
init_output_pin(codec, pin, pin_type);
caps = query_amp_caps(codec, pin, HDA_OUTPUT);
if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP)
caps = query_amp_caps(codec, pin, HDA_OUTPUT);
else
caps = 0;
if (caps & AC_AMPCAP_MUTE) {
unsigned int val;
val = (caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT;
@ -645,6 +648,10 @@ static void via_auto_init_analog_input(struct hda_codec *codec)
/* init ADCs */
for (i = 0; i < spec->num_adc_nids; i++) {
hda_nid_t nid = spec->adc_nids[i];
if (!(get_wcaps(codec, nid) & AC_WCAP_IN_AMP) ||
!(query_amp_caps(codec, nid, HDA_INPUT) & AC_AMPCAP_MUTE))
continue;
snd_hda_codec_write(codec, spec->adc_nids[i], 0,
AC_VERB_SET_AMP_GAIN_MUTE,
AMP_IN_UNMUTE(0));
@ -1445,25 +1452,9 @@ static const struct hda_pcm_stream via_pcm_digital_capture = {
/*
* slave controls for virtual master
*/
static const char * const via_slave_vols[] = {
"Front Playback Volume",
"Surround Playback Volume",
"Center Playback Volume",
"LFE Playback Volume",
"Side Playback Volume",
"Headphone Playback Volume",
"Speaker Playback Volume",
NULL,
};
static const char * const via_slave_sws[] = {
"Front Playback Switch",
"Surround Playback Switch",
"Center Playback Switch",
"LFE Playback Switch",
"Side Playback Switch",
"Headphone Playback Switch",
"Speaker Playback Switch",
static const char * const via_slave_pfxs[] = {
"Front", "Surround", "Center", "LFE", "Side",
"Headphone", "Speaker",
NULL,
};
@ -1508,13 +1499,15 @@ static int via_build_controls(struct hda_codec *codec)
snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0],
HDA_OUTPUT, vmaster_tlv);
err = snd_hda_add_vmaster(codec, "Master Playback Volume",
vmaster_tlv, via_slave_vols);
vmaster_tlv, via_slave_pfxs,
"Playback Volume");
if (err < 0)
return err;
}
if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
err = snd_hda_add_vmaster(codec, "Master Playback Switch",
NULL, via_slave_sws);
NULL, via_slave_pfxs,
"Playback Switch");
if (err < 0)
return err;
}
@ -1522,6 +1515,8 @@ static int via_build_controls(struct hda_codec *codec)
/* assign Capture Source enums to NID */
kctl = snd_hda_find_mixer_ctl(codec, "Input Source");
for (i = 0; kctl && i < kctl->count; i++) {
if (!spec->mux_nids[i])
continue;
err = snd_hda_add_nid(codec, kctl, i, spec->mux_nids[i]);
if (err < 0)
return err;
@ -2488,6 +2483,8 @@ static int create_mic_boost_ctls(struct hda_codec *codec)
{
struct via_spec *spec = codec->spec;
const struct auto_pin_cfg *cfg = &spec->autocfg;
const char *prev_label = NULL;
int type_idx = 0;
int i, err;
for (i = 0; i < cfg->num_inputs; i++) {
@ -2502,8 +2499,13 @@ static int create_mic_boost_ctls(struct hda_codec *codec)
if (caps == -1 || !(caps & AC_AMPCAP_NUM_STEPS))
continue;
label = hda_get_autocfg_input_label(codec, cfg, i);
if (prev_label && !strcmp(label, prev_label))
type_idx++;
else
type_idx = 0;
prev_label = label;
snprintf(name, sizeof(name), "%s Boost Volume", label);
err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
err = __via_add_control(spec, VIA_CTL_WIDGET_VOL, name, type_idx,
HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_INPUT));
if (err < 0)
return err;