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-rw-r--r--sound/pci/Kconfig22
-rw-r--r--sound/pci/Makefile1
-rw-r--r--sound/pci/ac97/ac97_patch.c46
-rw-r--r--sound/pci/ac97/ac97_pcm.c1
-rw-r--r--sound/pci/ad1889.c6
-rw-r--r--sound/pci/ali5451/ali5451.c32
-rw-r--r--sound/pci/als300.c4
-rw-r--r--sound/pci/atiixp.c2
-rw-r--r--sound/pci/atiixp_modem.c2
-rw-r--r--sound/pci/au88x0/au88x0.c2
-rw-r--r--sound/pci/au88x0/au88x0_pcm.c10
-rw-r--r--sound/pci/aw2/Makefile3
-rw-r--r--sound/pci/aw2/aw2-alsa.c794
-rw-r--r--sound/pci/aw2/aw2-saa7146.c465
-rw-r--r--sound/pci/aw2/aw2-saa7146.h105
-rw-r--r--sound/pci/aw2/aw2-tsl.c110
-rw-r--r--sound/pci/aw2/saa7146.h168
-rw-r--r--sound/pci/azt3328.c7
-rw-r--r--sound/pci/ca0106/ca0106_main.c21
-rw-r--r--sound/pci/ca0106/ca0106_mixer.c59
-rw-r--r--sound/pci/cmipci.c13
-rw-r--r--sound/pci/cs46xx/cs46xx_lib.c6
-rw-r--r--sound/pci/echoaudio/echoaudio.c7
-rw-r--r--sound/pci/emu10k1/emu10k1_main.c15
-rw-r--r--sound/pci/emu10k1/emu10k1x.c30
-rw-r--r--sound/pci/emu10k1/emuproc.c2
-rw-r--r--sound/pci/ens1370.c9
-rw-r--r--sound/pci/es1938.c5
-rw-r--r--sound/pci/es1968.c42
-rw-r--r--sound/pci/fm801.c8
-rw-r--r--sound/pci/hda/Makefile2
-rw-r--r--sound/pci/hda/hda_codec.c201
-rw-r--r--sound/pci/hda/hda_codec.h13
-rw-r--r--sound/pci/hda/hda_generic.c4
-rw-r--r--sound/pci/hda/hda_intel.c459
-rw-r--r--sound/pci/hda/hda_local.h20
-rw-r--r--sound/pci/hda/hda_patch.h28
-rw-r--r--sound/pci/hda/patch_analog.c581
-rw-r--r--sound/pci/hda/patch_atihdmi.c8
-rw-r--r--sound/pci/hda/patch_cmedia.c13
-rw-r--r--sound/pci/hda/patch_conexant.c68
-rw-r--r--sound/pci/hda/patch_realtek.c1248
-rw-r--r--sound/pci/hda/patch_si3054.c4
-rw-r--r--sound/pci/hda/patch_sigmatel.c391
-rw-r--r--sound/pci/hda/patch_via.c14
-rw-r--r--sound/pci/hda/vmaster.c364
-rw-r--r--sound/pci/ice1712/delta.c22
-rw-r--r--sound/pci/ice1712/delta.h2
-rw-r--r--sound/pci/ice1712/ews.c15
-rw-r--r--sound/pci/ice1712/ews.h4
-rw-r--r--sound/pci/ice1712/hoontech.c21
-rw-r--r--sound/pci/ice1712/ice1712.c45
-rw-r--r--sound/pci/ice1712/ice1712.h17
-rw-r--r--sound/pci/ice1712/ice1724.c431
-rw-r--r--sound/pci/ice1712/juli.c486
-rw-r--r--sound/pci/ice1712/pontis.c4
-rw-r--r--sound/pci/ice1712/prodigy192.c37
-rw-r--r--sound/pci/ice1712/revo.c55
-rw-r--r--sound/pci/intel8x0.c33
-rw-r--r--sound/pci/intel8x0m.c9
-rw-r--r--sound/pci/korg1212/korg1212.c1
-rw-r--r--sound/pci/maestro3.c38
-rw-r--r--sound/pci/nm256/nm256.c4
-rw-r--r--sound/pci/oxygen/cs4362a.h69
-rw-r--r--sound/pci/oxygen/cs4398.h69
-rw-r--r--sound/pci/oxygen/hifier.c36
-rw-r--r--sound/pci/oxygen/oxygen.c129
-rw-r--r--sound/pci/oxygen/oxygen.h23
-rw-r--r--sound/pci/oxygen/oxygen_io.c23
-rw-r--r--sound/pci/oxygen/oxygen_lib.c113
-rw-r--r--sound/pci/oxygen/oxygen_mixer.c217
-rw-r--r--sound/pci/oxygen/oxygen_pcm.c78
-rw-r--r--sound/pci/oxygen/pcm1796.h58
-rw-r--r--sound/pci/oxygen/virtuoso.c594
-rw-r--r--sound/pci/oxygen/wm8785.h45
-rw-r--r--sound/pci/pcxhr/pcxhr.c7
-rw-r--r--sound/pci/pcxhr/pcxhr_core.c33
-rw-r--r--sound/pci/riptide/riptide.c14
-rw-r--r--sound/pci/rme32.c3
-rw-r--r--sound/pci/rme96.c3
-rw-r--r--sound/pci/rme9652/hdsp.c54
-rw-r--r--sound/pci/rme9652/hdspm.c19
-rw-r--r--sound/pci/sis7019.c7
-rw-r--r--sound/pci/trident/trident_main.c4
-rw-r--r--sound/pci/via82xx.c2
-rw-r--r--sound/pci/via82xx_modem.c2
-rw-r--r--sound/pci/ymfpci/ymfpci_main.c4
87 files changed, 6137 insertions, 2108 deletions
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 812085d521f..581debf37dc 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -122,6 +122,21 @@ config SND_AU8830
To compile this driver as a module, choose M here: the module
will be called snd-au8830.
+config SND_AW2
+ tristate "Emagic Audiowerk 2"
+ depends on SND
+ help
+ Say Y here to include support for Emagic Audiowerk 2 soundcards.
+
+ Supported features: Analog and SPDIF output. Analog or SPDIF input.
+ Note: Switch between analog and digital input does not always work.
+ It can produce continuous noise. The workaround is to switch again
+ (and again) between digital and analog input until it works.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-aw2.
+
+
config SND_AZT3328
tristate "Aztech AZF3328 / PCI168 (EXPERIMENTAL)"
depends on SND && EXPERIMENTAL
@@ -162,6 +177,7 @@ config SND_CA0106
depends on SND
select SND_AC97_CODEC
select SND_RAWMIDI
+ select SND_VMASTER
help
Say Y here to include support for the Sound Blaster Audigy LS
and Live 24bit.
@@ -517,6 +533,7 @@ config SND_HDA_INTEL
tristate "Intel HD Audio"
depends on SND
select SND_PCM
+ select SND_VMASTER
help
Say Y here to include support for Intel "High Definition
Audio" (Azalia) motherboard devices.
@@ -680,6 +697,7 @@ config SND_ICE1724
depends on SND
select SND_MPU401_UART
select SND_AC97_CODEC
+ select SND_VMASTER
help
Say Y here to include support for soundcards based on
ICE/VT1724/1720 (Envy24HT/PT) chips.
@@ -896,12 +914,12 @@ config SND_VIA82XX_MODEM
will be called snd-via82xx-modem.
config SND_VIRTUOSO
- tristate "Asus Virtuoso 200 (Xonar)"
+ tristate "Asus Virtuoso 100/200 (Xonar)"
depends on SND
select SND_OXYGEN_LIB
help
Say Y here to include support for sound cards based on the
- Asus AV200 chip, i.e., Xonar D2 and Xonar D2X.
+ Asus AV100/AV200 chips, i.e., Xonar D2, DX and D2X.
To compile this driver as a module, choose M here: the module
will be called snd-virtuoso.
diff --git a/sound/pci/Makefile b/sound/pci/Makefile
index 2d42fd28f4e..85ef14bc805 100644
--- a/sound/pci/Makefile
+++ b/sound/pci/Makefile
@@ -58,6 +58,7 @@ obj-$(CONFIG_SND) += \
ac97/ \
ali5451/ \
au88x0/ \
+ aw2/ \
ca0106/ \
cs46xx/ \
cs5535audio/ \
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 50c637e55ff..39198e505b1 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -114,10 +114,9 @@ static int ac97_surround_jack_mode_put(struct snd_kcontrol *kcontrol, struct snd
static int ac97_channel_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
{
- static const char *texts[] = { "2ch", "4ch", "6ch" };
- if (kcontrol->private_value)
- return ac97_enum_text_info(kcontrol, uinfo, texts, 2); /* 4ch only */
- return ac97_enum_text_info(kcontrol, uinfo, texts, 3);
+ static const char *texts[] = { "2ch", "4ch", "6ch", "8ch" };
+ return ac97_enum_text_info(kcontrol, uinfo, texts,
+ kcontrol->private_value);
}
static int ac97_channel_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -133,13 +132,8 @@ static int ac97_channel_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e
struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
unsigned char mode = ucontrol->value.enumerated.item[0];
- if (kcontrol->private_value) {
- if (mode >= 2)
- return -EINVAL;
- } else {
- if (mode >= 3)
- return -EINVAL;
- }
+ if (mode >= kcontrol->private_value)
+ return -EINVAL;
if (mode != ac97->channel_mode) {
ac97->channel_mode = mode;
@@ -158,6 +152,7 @@ static int ac97_channel_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e
.get = ac97_surround_jack_mode_get, \
.put = ac97_surround_jack_mode_put, \
}
+/* 6ch */
#define AC97_CHANNEL_MODE_CTL \
{ \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
@@ -165,7 +160,9 @@ static int ac97_channel_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e
.info = ac97_channel_mode_info, \
.get = ac97_channel_mode_get, \
.put = ac97_channel_mode_put, \
+ .private_value = 3, \
}
+/* 4ch */
#define AC97_CHANNEL_MODE_4CH_CTL \
{ \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
@@ -173,7 +170,17 @@ static int ac97_channel_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e
.info = ac97_channel_mode_info, \
.get = ac97_channel_mode_get, \
.put = ac97_channel_mode_put, \
- .private_value = 1, \
+ .private_value = 2, \
+ }
+/* 8ch */
+#define AC97_CHANNEL_MODE_8CH_CTL \
+ { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = "Channel Mode", \
+ .info = ac97_channel_mode_info, \
+ .get = ac97_channel_mode_get, \
+ .put = ac97_channel_mode_put, \
+ .private_value = 4, \
}
static inline int is_surround_on(struct snd_ac97 *ac97)
@@ -210,6 +217,10 @@ static inline int is_shared_micin(struct snd_ac97 *ac97)
return !ac97->indep_surround && !is_clfe_on(ac97);
}
+static inline int alc850_is_aux_back_surround(struct snd_ac97 *ac97)
+{
+ return is_surround_on(ac97);
+}
/* The following snd_ac97_ymf753_... items added by David Shust (dshust@shustring.com) */
/* Modified for YMF743 by Keita Maehara <maehara@debian.org> */
@@ -2816,10 +2827,12 @@ static int patch_alc655(struct snd_ac97 * ac97)
#define AC97_ALC850_JACK_SELECT 0x76
#define AC97_ALC850_MISC1 0x7a
+#define AC97_ALC850_MULTICH 0x6a
static void alc850_update_jacks(struct snd_ac97 *ac97)
{
int shared;
+ int aux_is_back_surround;
/* shared Line-In / Surround Out */
shared = is_shared_surrout(ac97);
@@ -2837,13 +2850,18 @@ static void alc850_update_jacks(struct snd_ac97 *ac97)
/* MIC-IN = 1, CENTER-LFE = 5 */
snd_ac97_update_bits(ac97, AC97_ALC850_JACK_SELECT, 7 << 4,
shared ? (5<<4) : (1<<4));
+
+ aux_is_back_surround = alc850_is_aux_back_surround(ac97);
+ /* Aux is Back Surround */
+ snd_ac97_update_bits(ac97, AC97_ALC850_MULTICH, 1 << 10,
+ aux_is_back_surround ? (1<<10) : (0<<10));
}
static const struct snd_kcontrol_new snd_ac97_controls_alc850[] = {
AC97_PAGE_SINGLE("Duplicate Front", AC97_ALC650_MULTICH, 0, 1, 0, 0),
AC97_SINGLE("Mic Front Input Switch", AC97_ALC850_JACK_SELECT, 15, 1, 1),
AC97_SURROUND_JACK_MODE_CTL,
- AC97_CHANNEL_MODE_CTL,
+ AC97_CHANNEL_MODE_8CH_CTL,
};
static int patch_alc850_specific(struct snd_ac97 *ac97)
@@ -2869,6 +2887,7 @@ static int patch_alc850(struct snd_ac97 *ac97)
ac97->build_ops = &patch_alc850_ops;
ac97->spec.dev_flags = 0; /* for IEC958 playback route - ALC655 compatible */
+ ac97->flags |= AC97_HAS_8CH;
/* assume only page 0 for writing cache */
snd_ac97_update_bits(ac97, AC97_INT_PAGING, AC97_PAGE_MASK, AC97_PAGE_VENDOR);
@@ -2878,6 +2897,7 @@ static int patch_alc850(struct snd_ac97 *ac97)
spdif-in monitor off, spdif-in PCM off
center on mic off, surround on line-in off
duplicate front off
+ NB default bit 10=0 = Aux is Capture, not Back Surround
*/
snd_ac97_write_cache(ac97, AC97_ALC650_MULTICH, 1<<15);
/* SURR_OUT: on, Surr 1kOhm: on, Surr Amp: off, Front 1kOhm: off
diff --git a/sound/pci/ac97/ac97_pcm.c b/sound/pci/ac97/ac97_pcm.c
index 3674f35c4a7..48cbda9378c 100644
--- a/sound/pci/ac97/ac97_pcm.c
+++ b/sound/pci/ac97/ac97_pcm.c
@@ -574,7 +574,6 @@ int snd_ac97_pcm_open(struct ac97_pcm *pcm, unsigned int rate,
r = rate > 48000;
bus = pcm->bus;
if (cfg == AC97_PCM_CFG_SPDIF) {
- int err;
for (cidx = 0; cidx < 4; cidx++)
if (bus->codec[cidx] && (bus->codec[cidx]->ext_id & AC97_EI_SPDIF)) {
err = set_spdif_rate(bus->codec[cidx], rate);
diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c
index a66d5150bb7..39ec55b57b1 100644
--- a/sound/pci/ad1889.c
+++ b/sound/pci/ad1889.c
@@ -264,10 +264,10 @@ snd_ad1889_ac97_ready(struct snd_ad1889 *chip)
mdelay(1);
if (!retry) {
snd_printk(KERN_ERR PFX "[%s] Link is not ready.\n",
- __FUNCTION__);
+ __func__);
return -EIO;
}
- ad1889_debug("[%s] ready after %d ms\n", __FUNCTION__, 400 - retry);
+ ad1889_debug("[%s] ready after %d ms\n", __func__, 400 - retry);
return 0;
}
@@ -854,8 +854,6 @@ snd_ad1889_free(struct snd_ad1889 *chip)
spin_unlock_irq(&chip->lock);
- synchronize_irq(chip->irq);
-
if (chip->irq >= 0)
free_irq(chip->irq, chip);
diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
index 6a905ed9cbd..1a0fd65ec28 100644
--- a/sound/pci/ali5451/ali5451.c
+++ b/sound/pci/ali5451/ali5451.c
@@ -1809,26 +1809,26 @@ static int snd_ali5451_spdif_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_ali *codec = kcontrol->private_data;
- unsigned int enable;
+ unsigned int spdif_enable;
- enable = ucontrol->value.integer.value[0] ? 1 : 0;
+ spdif_enable = ucontrol->value.integer.value[0] ? 1 : 0;
spin_lock_irq(&codec->reg_lock);
switch (kcontrol->private_value) {
case 0:
- enable = (codec->spdif_mask & 0x02) ? 1 : 0;
+ spdif_enable = (codec->spdif_mask & 0x02) ? 1 : 0;
break;
case 1:
- enable = ((codec->spdif_mask & 0x02) &&
+ spdif_enable = ((codec->spdif_mask & 0x02) &&
(codec->spdif_mask & 0x04)) ? 1 : 0;
break;
case 2:
- enable = (codec->spdif_mask & 0x01) ? 1 : 0;
+ spdif_enable = (codec->spdif_mask & 0x01) ? 1 : 0;
break;
default:
break;
}
- ucontrol->value.integer.value[0] = enable;
+ ucontrol->value.integer.value[0] = spdif_enable;
spin_unlock_irq(&codec->reg_lock);
return 0;
}
@@ -1837,17 +1837,17 @@ static int snd_ali5451_spdif_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_ali *codec = kcontrol->private_data;
- unsigned int change = 0, enable = 0;
+ unsigned int change = 0, spdif_enable = 0;
- enable = ucontrol->value.integer.value[0] ? 1 : 0;
+ spdif_enable = ucontrol->value.integer.value[0] ? 1 : 0;
spin_lock_irq(&codec->reg_lock);
switch (kcontrol->private_value) {
case 0:
change = (codec->spdif_mask & 0x02) ? 1 : 0;
- change = change ^ enable;
+ change = change ^ spdif_enable;
if (change) {
- if (enable) {
+ if (spdif_enable) {
codec->spdif_mask |= 0x02;
snd_ali_enable_spdif_out(codec);
} else {
@@ -1859,9 +1859,9 @@ static int snd_ali5451_spdif_put(struct snd_kcontrol *kcontrol,
break;
case 1:
change = (codec->spdif_mask & 0x04) ? 1 : 0;
- change = change ^ enable;
+ change = change ^ spdif_enable;
if (change && (codec->spdif_mask & 0x02)) {
- if (enable) {
+ if (spdif_enable) {
codec->spdif_mask |= 0x04;
snd_ali_enable_spdif_chnout(codec);
} else {
@@ -1872,9 +1872,9 @@ static int snd_ali5451_spdif_put(struct snd_kcontrol *kcontrol,
break;
case 2:
change = (codec->spdif_mask & 0x01) ? 1 : 0;
- change = change ^ enable;
+ change = change ^ spdif_enable;
if (change) {
- if (enable) {
+ if (spdif_enable) {
codec->spdif_mask |= 0x01;
snd_ali_enable_spdif_in(codec);
} else {
@@ -2047,10 +2047,8 @@ static int snd_ali_free(struct snd_ali * codec)
{
if (codec->hw_initialized)
snd_ali_disable_address_interrupt(codec);
- if (codec->irq >= 0) {
- synchronize_irq(codec->irq);
+ if (codec->irq >= 0)
free_irq(codec->irq, codec);
- }
if (codec->port)
pci_release_regions(codec->pci);
pci_disable_device(codec->pci);
diff --git a/sound/pci/als300.c b/sound/pci/als300.c
index 0e990a73582..8df6824b51c 100644
--- a/sound/pci/als300.c
+++ b/sound/pci/als300.c
@@ -92,8 +92,8 @@
#if DEBUG_CALLS
#define snd_als300_dbgcalls(format, args...) printk(format, ##args)
-#define snd_als300_dbgcallenter() printk(KERN_ERR "--> %s\n", __FUNCTION__)
-#define snd_als300_dbgcallleave() printk(KERN_ERR "<-- %s\n", __FUNCTION__)
+#define snd_als300_dbgcallenter() printk(KERN_ERR "--> %s\n", __func__)
+#define snd_als300_dbgcallleave() printk(KERN_ERR "<-- %s\n", __func__)
#else
#define snd_als300_dbgcalls(format, args...)
#define snd_als300_dbgcallenter()
diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c
index 4594186b83e..457228fb22a 100644
--- a/sound/pci/atiixp.c
+++ b/sound/pci/atiixp.c
@@ -1553,7 +1553,7 @@ static int snd_atiixp_free(struct atiixp *chip)
if (chip->irq < 0)
goto __hw_end;
snd_atiixp_chip_stop(chip);
- synchronize_irq(chip->irq);
+
__hw_end:
if (chip->irq >= 0)
free_irq(chip->irq, chip);
diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c
index a67a869180d..d457a32a793 100644
--- a/sound/pci/atiixp_modem.c
+++ b/sound/pci/atiixp_modem.c
@@ -1197,7 +1197,7 @@ static int snd_atiixp_free(struct atiixp_modem *chip)
if (chip->irq < 0)
goto __hw_end;
snd_atiixp_chip_stop(chip);
- synchronize_irq(chip->irq);
+
__hw_end:
if (chip->irq >= 0)
free_irq(chip->irq, chip);
diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c
index 26819e2f576..68368e49007 100644
--- a/sound/pci/au88x0/au88x0.c
+++ b/sound/pci/au88x0/au88x0.c
@@ -126,7 +126,6 @@ static int snd_vortex_dev_free(struct snd_device *device)
vortex_gameport_unregister(vortex);
vortex_core_shutdown(vortex);
// Take down PCI interface.
- synchronize_irq(vortex->irq);
free_irq(vortex->irq, vortex);
iounmap(vortex->mmio);
pci_release_regions(vortex->pci_dev);
@@ -220,7 +219,6 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip)
return 0;
alloc_out:
- synchronize_irq(chip->irq);
free_irq(chip->irq, chip);
irq_out:
vortex_core_shutdown(chip);
diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c
index 526c6c5ecf7..f9a58b4a30e 100644
--- a/sound/pci/au88x0/au88x0_pcm.c
+++ b/sound/pci/au88x0/au88x0_pcm.c
@@ -498,14 +498,14 @@ static struct snd_kcontrol_new snd_vortex_mixer_spdif[] __devinitdata = {
};
/* create a pcm device */
-static int __devinit snd_vortex_new_pcm(vortex_t * chip, int idx, int nr)
+static int __devinit snd_vortex_new_pcm(vortex_t *chip, int idx, int nr)
{
struct snd_pcm *pcm;
struct snd_kcontrol *kctl;
int i;
int err, nr_capt;
- if ((chip == 0) || (idx < 0) || (idx >= VORTEX_PCM_LAST))
+ if (!chip || idx < 0 || idx >= VORTEX_PCM_LAST)
return -ENODEV;
/* idx indicates which kind of PCM device. ADB, SPDIF, I2S and A3D share the
@@ -514,9 +514,9 @@ static int __devinit snd_vortex_new_pcm(vortex_t * chip, int idx, int nr)
nr_capt = nr;
else
nr_capt = 0;
- if ((err =
- snd_pcm_new(chip->card, vortex_pcm_prettyname[idx], idx, nr,
- nr_capt, &pcm)) < 0)
+ err = snd_pcm_new(chip->card, vortex_pcm_prettyname[idx], idx, nr,
+ nr_capt, &pcm);
+ if (err < 0)
return err;
strcpy(pcm->name, vortex_pcm_name[idx]);
chip->pcm[idx] = pcm;
diff --git a/sound/pci/aw2/Makefile b/sound/pci/aw2/Makefile
new file mode 100644
index 00000000000..842335d3b73
--- /dev/null
+++ b/sound/pci/aw2/Makefile
@@ -0,0 +1,3 @@
+snd-aw2-objs := aw2-alsa.o aw2-saa7146.o
+
+obj-$(CONFIG_SND_AW2) += snd-aw2.o
diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c
new file mode 100644
index 00000000000..56f87cd33c1
--- /dev/null
+++ b/sound/pci/aw2/aw2-alsa.c
@@ -0,0 +1,794 @@
+/*****************************************************************************
+ *
+ * Copyright (C) 2008 Cedric Bregardis <cedric.bregardis@free.fr> and
+ * Jean-Christian Hassler <jhassler@free.fr>
+ *
+ * This file is part of the Audiowerk2 ALSA driver
+ *
+ * The Audiowerk2 ALSA driver is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2.
+ *
+ * The Audiowerk2 ALSA driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with the Audiowerk2 ALSA driver; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301,
+ * USA.
+ *
+ *****************************************************************************/
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/dma-mapping.h>
+#include <linux/slab.h>
+#include <linux/interrupt.h>
+#include <linux/delay.h>
+#include <asm/io.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/control.h>
+
+#include "saa7146.h"
+#include "aw2-saa7146.h"
+
+MODULE_AUTHOR("Cedric Bregardis <cedric.bregardis@free.fr>, "
+ "Jean-Christian Hassler <jhassler@free.fr>");
+MODULE_DESCRIPTION("Emagic Audiowerk 2 sound driver");
+MODULE_LICENSE("GPL");
+
+/*********************************
+ * DEFINES
+ ********************************/
+#define PCI_VENDOR_ID_SAA7146 0x1131
+#define PCI_DEVICE_ID_SAA7146 0x7146
+
+#define CTL_ROUTE_ANALOG 0
+#define CTL_ROUTE_DIGITAL 1
+
+/*********************************
+ * TYPEDEFS
+ ********************************/
+ /* hardware definition */
+static struct snd_pcm_hardware snd_aw2_playback_hw = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_44100,
+ .rate_min = 44100,
+ .rate_max = 44100,
+ .channels_min = 2,
+ .channels_max = 4,
+ .buffer_bytes_max = 32768,
+ .period_bytes_min = 4096,
+ .period_bytes_max = 32768,
+ .periods_min = 1,
+ .periods_max = 1024,
+};
+
+static struct snd_pcm_hardware snd_aw2_capture_hw = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_44100,
+ .rate_min = 44100,
+ .rate_max = 44100,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 32768,
+ .period_bytes_min = 4096,
+ .period_bytes_max = 32768,
+ .periods_min = 1,
+ .periods_max = 1024,
+};
+
+struct aw2_pcm_device {
+ struct snd_pcm *pcm;
+ unsigned int stream_number;
+ struct aw2 *chip;
+};
+
+struct aw2 {
+ struct snd_aw2_saa7146 saa7146;
+
+ struct pci_dev *pci;
+ int irq;
+ spinlock_t reg_lock;
+ struct mutex mtx;
+
+ unsigned long iobase_phys;
+ void __iomem *iobase_virt;
+
+ struct snd_card *card;
+
+ struct aw2_pcm_device device_playback[NB_STREAM_PLAYBACK];
+ struct aw2_pcm_device device_capture[NB_STREAM_CAPTURE];
+};
+
+/*********************************
+ * FUNCTION DECLARATIONS
+ ********************************/
+static int __init alsa_card_aw2_init(void);
+static void __exit alsa_card_aw2_exit(void);
+static int snd_aw2_dev_free(struct snd_device *device);
+static int __devinit snd_aw2_create(struct snd_card *card,
+ struct pci_dev *pci, struct aw2 **rchip);
+static int __devinit snd_aw2_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id);
+static void __devexit snd_aw2_remove(struct pci_dev *pci);
+static int snd_aw2_pcm_playback_open(struct snd_pcm_substream *substream);
+static int snd_aw2_pcm_playback_close(struct snd_pcm_substream *substream);
+static int snd_aw2_pcm_capture_open(struct snd_pcm_substream *substream);
+static int snd_aw2_pcm_capture_close(struct snd_pcm_substream *substream);
+static int snd_aw2_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params);
+static int snd_aw2_pcm_hw_free(struct snd_pcm_substream *substream);
+static int snd_aw2_pcm_prepare_playback(struct snd_pcm_substream *substream);
+static int snd_aw2_pcm_prepare_capture(struct snd_pcm_substream *substream);
+static int snd_aw2_pcm_trigger_playback(struct snd_pcm_substream *substream,
+ int cmd);
+static int snd_aw2_pcm_trigger_capture(struct snd_pcm_substream *substream,
+ int cmd);
+static snd_pcm_uframes_t snd_aw2_pcm_pointer_playback(struct snd_pcm_substream
+ *substream);
+static snd_pcm_uframes_t snd_aw2_pcm_pointer_capture(struct snd_pcm_substream
+ *substream);
+static int __devinit snd_aw2_new_pcm(struct aw2 *chip);
+
+static int snd_aw2_control_switch_capture_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo);
+static int snd_aw2_control_switch_capture_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value
+ *ucontrol);
+static int snd_aw2_control_switch_capture_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value
+ *ucontrol);
+
+/*********************************
+ * VARIABLES
+ ********************************/
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for Audiowerk2 soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for the Audiowerk2 soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable Audiowerk2 soundcard.");
+
+static struct pci_device_id snd_aw2_ids[] = {
+ {PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, PCI_ANY_ID, PCI_ANY_ID,
+ 0, 0, 0},
+ {0}
+};
+
+MODULE_DEVICE_TABLE(pci, snd_aw2_ids);
+
+/* pci_driver definition */
+static struct pci_driver driver = {
+ .name = "Emagic Audiowerk 2",
+ .id_table = snd_aw2_ids,
+ .probe = snd_aw2_probe,
+ .remove = __devexit_p(snd_aw2_remove),
+};
+
+/* operators for playback PCM alsa interface */
+static struct snd_pcm_ops snd_aw2_playback_ops = {
+ .open = snd_aw2_pcm_playback_open,
+ .close = snd_aw2_pcm_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_aw2_pcm_hw_params,
+ .hw_free = snd_aw2_pcm_hw_free,
+ .prepare = snd_aw2_pcm_prepare_playback,
+ .trigger = snd_aw2_pcm_trigger_playback,
+ .pointer = snd_aw2_pcm_pointer_playback,
+};
+
+/* operators for capture PCM alsa interface */
+static struct snd_pcm_ops snd_aw2_capture_ops = {
+ .open = snd_aw2_pcm_capture_open,
+ .close = snd_aw2_pcm_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_aw2_pcm_hw_params,
+ .hw_free = snd_aw2_pcm_hw_free,
+ .prepare = snd_aw2_pcm_prepare_capture,
+ .trigger = snd_aw2_pcm_trigger_capture,
+ .pointer = snd_aw2_pcm_pointer_capture,
+};
+
+static struct snd_kcontrol_new aw2_control __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "PCM Capture Route",
+ .index = 0,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .private_value = 0xffff,
+ .info = snd_aw2_control_switch_capture_info,
+ .get = snd_aw2_control_switch_capture_get,
+ .put = snd_aw2_control_switch_capture_put
+};
+
+/*********************************
+ * FUNCTION IMPLEMENTATIONS
+ ********************************/
+
+/* initialization of the module */
+static int __init alsa_card_aw2_init(void)
+{
+ snd_printdd(KERN_DEBUG "aw2: Load aw2 module\n");
+ return pci_register_driver(&driver);
+}
+
+/* clean up the module */
+static void __exit alsa_card_aw2_exit(void)
+{
+ snd_printdd(KERN_DEBUG "aw2: Unload aw2 module\n");
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_aw2_init);
+module_exit(alsa_card_aw2_exit);
+
+/* component-destructor */
+static int snd_aw2_dev_free(struct snd_device *device)
+{
+ struct aw2 *chip = device->device_data;
+
+ /* Free hardware */
+ snd_aw2_saa7146_free(&chip->saa7146);
+
+ /* release the irq */
+ if (chip->irq >= 0)
+ free_irq(chip->irq, (void *)chip);
+ /* release the i/o ports & memory */
+ if (chip->iobase_virt)
+ iounmap(chip->iobase_virt);
+
+ pci_release_regions(chip->pci);
+ /* disable the PCI entry */
+ pci_disable_device(chip->pci);
+ /* release the data */
+ kfree(chip);
+
+ return 0;
+}
+
+/* chip-specific constructor */
+static int __devinit snd_aw2_create(struct snd_card *card,
+ struct pci_dev *pci, struct aw2 **rchip)
+{
+ struct aw2 *chip;
+ int err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_aw2_dev_free,
+ };
+
+ *rchip = NULL;
+
+ /* initialize the PCI entry */
+ err = pci_enable_device(pci);
+ if (err < 0)
+ return err;
+ pci_set_master(pci);
+
+ /* check PCI availability (32bit DMA) */
+ if ((pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0) ||
+ (pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0)) {
+ printk(KERN_ERR "aw2: Impossible to set 32bit mask DMA\n");
+ pci_disable_device(pci);
+ return -ENXIO;
+ }
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+
+ /* initialize the stuff */
+ chip->card = card;
+ chip->pci = pci;
+ chip->irq = -1;
+
+ /* (1) PCI resource allocation */
+ err = pci_request_regions(pci, "Audiowerk2");
+ if (err < 0) {
+ pci_disable_device(pci);
+ kfree(chip);
+ return err;
+ }
+ chip->iobase_phys = pci_resource_start(pci, 0);
+ chip->iobase_virt =
+ ioremap_nocache(chip->iobase_phys,
+ pci_resource_len(pci, 0));
+
+ if (chip->iobase_virt == NULL) {
+ printk(KERN_ERR "aw2: unable to remap memory region");
+ pci_release_regions(pci);
+ pci_disable_device(pci);
+ kfree(chip);
+ return -ENOMEM;
+ }
+
+
+ if (request_irq(pci->irq, snd_aw2_saa7146_interrupt,
+ IRQF_SHARED, "Audiowerk2", chip)) {
+ printk(KERN_ERR "aw2: Cannot grab irq %d\n", pci->irq);
+
+ iounmap(chip->iobase_virt);
+ pci_release_regions(chip->pci);
+ pci_disable_device(chip->pci);
+ kfree(chip);
+ return -EBUSY;
+ }
+ chip->irq = pci->irq;
+
+ /* (2) initialization of the chip hardware */
+ snd_aw2_saa7146_setup(&chip->saa7146, chip->iobase_virt);
+ err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
+ if (err < 0) {
+ free_irq(chip->irq, (void *)chip);
+ iounmap(chip->iobase_virt);
+ pci_release_regions(chip->pci);
+ pci_disable_device(chip->pci);
+ kfree(chip);
+ return err;
+ }
+
+ snd_card_set_dev(card, &pci->dev);
+ *rchip = chip;
+
+ printk(KERN_INFO
+ "Audiowerk 2 sound card (saa7146 chipset) detected and "
+ "managed\n");
+ return 0;
+}
+
+/* constructor */
+static int __devinit snd_aw2_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct aw2 *chip;
+ int err;
+
+ /* (1) Continue if device is not enabled, else inc dev */
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ /* (2) Create card instance */
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ /* (3) Create main component */
+ err = snd_aw2_create(card, pci, &chip);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ /* initialize mutex */
+ mutex_init(&chip->mtx);
+ /* init spinlock */
+ spin_lock_init(&chip->reg_lock);
+ /* (4) Define driver ID and name string */
+ strcpy(card->driver, "aw2");
+ strcpy(card->shortname, "Audiowerk2");
+
+ sprintf(card->longname, "%s with SAA7146 irq %i",
+ card->shortname, chip->irq);
+
+ /* (5) Create other components */
+ snd_aw2_new_pcm(chip);
+
+ /* (6) Register card instance */
+ err = snd_card_register(card);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ /* (7) Set PCI driver data */
+ pci_set_drvdata(pci, card);
+
+ dev++;
+ return 0;
+}
+
+/* destructor */
+static void __devexit snd_aw2_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+/* open callback */
+static int snd_aw2_pcm_playback_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ snd_printdd(KERN_DEBUG "aw2: Playback_open \n");
+ runtime->hw = snd_aw2_playback_hw;
+ return 0;
+}
+
+/* close callback */
+static int snd_aw2_pcm_playback_close(struct snd_pcm_substream *substream)
+{
+ return 0;
+
+}
+
+static int snd_aw2_pcm_capture_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ snd_printdd(KERN_DEBUG "aw2: Capture_open \n");
+ runtime->hw = snd_aw2_capture_hw;
+ return 0;
+}
+
+/* close callback */
+static int snd_aw2_pcm_capture_close(struct snd_pcm_substream *substream)
+{
+ /* TODO: something to do ? */
+ return 0;
+}
+
+ /* hw_params callback */
+static int snd_aw2_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+}
+
+/* hw_free callback */
+static int snd_aw2_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+/* prepare callback for playback */
+static int snd_aw2_pcm_prepare_playback(struct snd_pcm_substream *substream)
+{
+ struct aw2_pcm_device *pcm_device = snd_pcm_substream_chip(substream);
+ struct aw2 *chip = pcm_device->chip;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned long period_size, buffer_size;
+
+ mutex_lock(&chip->mtx);
+
+ period_size = snd_pcm_lib_period_bytes(substream);
+ buffer_size = snd_pcm_lib_buffer_bytes(substream);
+
+ snd_aw2_saa7146_pcm_init_playback(&chip->saa7146,
+ pcm_device->stream_number,
+ runtime->dma_addr, period_size,
+ buffer_size);
+
+ /* Define Interrupt callback */
+ snd_aw2_saa7146_define_it_playback_callback(pcm_device->stream_number,
+ (snd_aw2_saa7146_it_cb)
+ snd_pcm_period_elapsed,
+ (void *)substream);
+
+ mutex_unlock(&chip->mtx);
+
+ return 0;
+}
+
+/* prepare callback for capture */
+static int snd_aw2_pcm_prepare_capture(struct snd_pcm_substream *substream)
+{
+ struct aw2_pcm_device *pcm_device = snd_pcm_substream_chip(substream);
+ struct aw2 *chip = pcm_device->chip;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned long period_size, buffer_size;
+
+ mutex_lock(&chip->mtx);
+
+ period_size = snd_pcm_lib_period_bytes(substream);
+ buffer_size = snd_pcm_lib_buffer_bytes(substream);
+
+ snd_aw2_saa7146_pcm_init_capture(&chip->saa7146,
+ pcm_device->stream_number,
+ runtime->dma_addr, period_size,
+ buffer_size);
+
+ /* Define Interrupt callback */
+ snd_aw2_saa7146_define_it_capture_callback(pcm_device->stream_number,
+ (snd_aw2_saa7146_it_cb)
+ snd_pcm_period_elapsed,
+ (void *)substream);
+
+ mutex_unlock(&chip->mtx);
+
+ return 0;
+}
+
+/* playback trigger callback */
+static int snd_aw2_pcm_trigger_playback(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ int status = 0;
+ struct aw2_pcm_device *pcm_device = snd_pcm_substream_chip(substream);
+ struct aw2 *chip = pcm_device->chip;
+ spin_lock(&chip->reg_lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ snd_aw2_saa7146_pcm_trigger_start_playback(&chip->saa7146,
+ pcm_device->
+ stream_number);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ snd_aw2_saa7146_pcm_trigger_stop_playback(&chip->saa7146,
+ pcm_device->
+ stream_number);
+ break;
+ default:
+ status = -EINVAL;
+ }
+ spin_unlock(&chip->reg_lock);
+ return status;
+}
+
+/* capture trigger callback */
+static int snd_aw2_pcm_trigger_capture(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ int status = 0;
+ struct aw2_pcm_device *pcm_device = snd_pcm_substream_chip(substream);
+ struct aw2 *chip = pcm_device->chip;
+ spin_lock(&chip->reg_lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ snd_aw2_saa7146_pcm_trigger_start_capture(&chip->saa7146,
+ pcm_device->
+ stream_number);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ snd_aw2_saa7146_pcm_trigger_stop_capture(&chip->saa7146,
+ pcm_device->
+ stream_number);
+ break;
+ default:
+ status = -EINVAL;
+ }
+ spin_unlock(&chip->reg_lock);
+ return status;
+}
+
+/* playback pointer callback */
+static snd_pcm_uframes_t snd_aw2_pcm_pointer_playback(struct snd_pcm_substream
+ *substream)
+{
+ struct aw2_pcm_device *pcm_device = snd_pcm_substream_chip(substream);
+ struct aw2 *chip = pcm_device->chip;
+ unsigned int current_ptr;
+
+ /* get the current hardware pointer */
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ current_ptr =
+ snd_aw2_saa7146_get_hw_ptr_playback(&chip->saa7146,
+ pcm_device->stream_number,
+ runtime->dma_area,
+ runtime->buffer_size);
+
+ return bytes_to_frames(substream->runtime, current_ptr);
+}
+
+/* capture pointer callback */
+static snd_pcm_uframes_t snd_aw2_pcm_pointer_capture(struct snd_pcm_substream
+ *substream)
+{
+ struct aw2_pcm_device *pcm_device = snd_pcm_substream_chip(substream);
+ struct aw2 *chip = pcm_device->chip;
+ unsigned int current_ptr;
+
+ /* get the current hardware pointer */
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ current_ptr =
+ snd_aw2_saa7146_get_hw_ptr_capture(&chip->saa7146,
+ pcm_device->stream_number,
+ runtime->dma_area,
+ runtime->buffer_size);
+
+ return bytes_to_frames(substream->runtime, current_ptr);
+}
+
+/* create a pcm device */
+static int __devinit snd_aw2_new_pcm(struct aw2 *chip)
+{
+ struct snd_pcm *pcm_playback_ana;
+ struct snd_pcm *pcm_playback_num;
+ struct snd_pcm *pcm_capture;
+ struct aw2_pcm_device *pcm_device;
+ int err = 0;
+
+ /* Create new Alsa PCM device */
+
+ err = snd_pcm_new(chip->card, "Audiowerk2 analog playback", 0, 1, 0,
+ &pcm_playback_ana);
+ if (err < 0) {
+ printk(KERN_ERR "aw2: snd_pcm_new error (0x%X)\n", err);
+ return err;
+ }
+
+ /* Creation ok */
+ pcm_device = &chip->device_playback[NUM_STREAM_PLAYBACK_ANA];
+
+ /* Set PCM device name */
+ strcpy(pcm_playback_ana->name, "Analog playback");
+ /* Associate private data to PCM device */
+ pcm_playback_ana->private_data = pcm_device;
+ /* set operators of PCM device */
+ snd_pcm_set_ops(pcm_playback_ana, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_aw2_playback_ops);
+ /* store PCM device */
+ pcm_device->pcm = pcm_playback_ana;
+ /* give base chip pointer to our internal pcm device
+ structure */
+ pcm_device->chip = chip;
+ /* Give stream number to PCM device */
+ pcm_device->stream_number = NUM_STREAM_PLAYBACK_ANA;
+
+ /* pre-allocation of buffers */
+ /* Preallocate continuous pages. */
+ err = snd_pcm_lib_preallocate_pages_for_all(pcm_playback_ana,
+ SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data
+ (chip->pci),
+ 64 * 1024, 64 * 1024);
+ if (err)
+ printk(KERN_ERR "aw2: snd_pcm_lib_preallocate_pages_for_all "
+ "error (0x%X)\n", err);
+
+ err = snd_pcm_new(chip->card, "Audiowerk2 digital playback", 1, 1, 0,
+ &pcm_playback_num);
+
+ if (err < 0) {
+ printk(KERN_ERR "aw2: snd_pcm_new error (0x%X)\n", err);
+ return err;
+ }
+ /* Creation ok */
+ pcm_device = &chip->device_playback[NUM_STREAM_PLAYBACK_DIG];
+
+ /* Set PCM device name */
+ strcpy(pcm_playback_num->name, "Digital playback");
+ /* Associate private data to PCM device */
+ pcm_playback_num->private_data = pcm_device;
+ /* set operators of PCM device */
+ snd_pcm_set_ops(pcm_playback_num, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_aw2_playback_ops);
+ /* store PCM device */
+ pcm_device->pcm = pcm_playback_num;
+ /* give base chip pointer to our internal pcm device
+ structure */
+ pcm_device->chip = chip;
+ /* Give stream number to PCM device */
+ pcm_device->stream_number = NUM_STREAM_PLAYBACK_DIG;
+
+ /* pre-allocation of buffers */
+ /* Preallocate continuous pages. */
+ err = snd_pcm_lib_preallocate_pages_for_all(pcm_playback_num,
+ SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data
+ (chip->pci),
+ 64 * 1024, 64 * 1024);
+ if (err)
+ printk(KERN_ERR
+ "aw2: snd_pcm_lib_preallocate_pages_for_all error "
+ "(0x%X)\n", err);
+
+
+
+ err = snd_pcm_new(chip->card, "Audiowerk2 capture", 2, 0, 1,
+ &pcm_capture);
+
+ if (err < 0) {
+ printk(KERN_ERR "aw2: snd_pcm_new error (0x%X)\n", err);
+ return err;
+ }
+
+ /* Creation ok */
+ pcm_device = &chip->device_capture[NUM_STREAM_CAPTURE_ANA];
+
+ /* Set PCM device name */
+ strcpy(pcm_capture->name, "Capture");
+ /* Associate private data to PCM device */
+ pcm_capture->private_data = pcm_device;
+ /* set operators of PCM device */
+ snd_pcm_set_ops(pcm_capture, SNDRV_PCM_STREAM_CAPTURE,
+ &snd_aw2_capture_ops);
+ /* store PCM device */
+ pcm_device->pcm = pcm_capture;
+ /* give base chip pointer to our internal pcm device
+ structure */
+ pcm_device->chip = chip;
+ /* Give stream number to PCM device */
+ pcm_device->stream_number = NUM_STREAM_CAPTURE_ANA;
+
+ /* pre-allocation of buffers */
+ /* Preallocate continuous pages. */
+ err = snd_pcm_lib_preallocate_pages_for_all(pcm_capture,
+ SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data
+ (chip->pci),
+ 64 * 1024, 64 * 1024);
+ if (err)
+ printk(KERN_ERR
+ "aw2: snd_pcm_lib_preallocate_pages_for_all error "
+ "(0x%X)\n", err);
+
+
+ /* Create control */
+ err = snd_ctl_add(chip->card, snd_ctl_new1(&aw2_control, chip));
+ if (err < 0) {
+ printk(KERN_ERR "aw2: snd_ctl_add error (0x%X)\n", err);
+ return err;
+ }
+
+ return 0;
+}
+
+static int snd_aw2_control_switch_capture_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[2] = {
+ "Analog", "Digital"
+ };
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) {
+ uinfo->value.enumerated.item =
+ uinfo->value.enumerated.items - 1;
+ }
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_aw2_control_switch_capture_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value
+ *ucontrol)
+{
+ struct aw2 *chip = snd_kcontrol_chip(kcontrol);
+ if (snd_aw2_saa7146_is_using_digital_input(&chip->saa7146))
+ ucontrol->value.enumerated.item[0] = CTL_ROUTE_DIGITAL;
+ else
+ ucontrol->value.enumerated.item[0] = CTL_ROUTE_ANALOG;
+ return 0;
+}
+
+static int snd_aw2_control_switch_capture_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value
+ *ucontrol)
+{
+ struct aw2 *chip = snd_kcontrol_chip(kcontrol);
+ int changed = 0;
+ int is_disgital =
+ snd_aw2_saa7146_is_using_digital_input(&chip->saa7146);
+
+ if (((ucontrol->value.integer.value[0] == CTL_ROUTE_DIGITAL)
+ && !is_disgital)
+ || ((ucontrol->value.integer.value[0] == CTL_ROUTE_ANALOG)
+ && is_disgital)) {
+ snd_aw2_saa7146_use_digital_input(&chip->saa7146, !is_disgital);
+ changed = 1;
+ }
+ return changed;
+}
diff --git a/sound/pci/aw2/aw2-saa7146.c b/sound/pci/aw2/aw2-saa7146.c
new file mode 100644
index 00000000000..6a3891ab69d
--- /dev/null
+++ b/sound/pci/aw2/aw2-saa7146.c
@@ -0,0 +1,465 @@
+/*****************************************************************************
+ *
+ * Copyright (C) 2008 Cedric Bregardis <cedric.bregardis@free.fr> and
+ * Jean-Christian Hassler <jhassler@free.fr>
+ *
+ * This file is part of the Audiowerk2 ALSA driver
+ *
+ * The Audiowerk2 ALSA driver is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2.
+ *
+ * The Audiowerk2 ALSA driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with the Audiowerk2 ALSA driver; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301,
+ * USA.
+ *
+ *****************************************************************************/
+
+#define AW2_SAA7146_M
+
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/interrupt.h>
+#include <linux/delay.h>
+#include <asm/system.h>
+#include <asm/io.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+
+#include "saa7146.h"
+#include "aw2-saa7146.h"
+
+#include "aw2-tsl.c"
+
+#define WRITEREG(value, addr) writel((value), chip->base_addr + (addr))
+#define READREG(addr) readl(chip->base_addr + (addr))
+
+static struct snd_aw2_saa7146_cb_param
+ arr_substream_it_playback_cb[NB_STREAM_PLAYBACK];
+static struct snd_aw2_saa7146_cb_param
+ arr_substream_it_capture_cb[NB_STREAM_CAPTURE];
+
+static int snd_aw2_saa7146_get_limit(int size);
+
+/* chip-specific destructor */
+int snd_aw2_saa7146_free(struct snd_aw2_saa7146 *chip)
+{
+ /* disable all irqs */
+ WRITEREG(0, IER);
+
+ /* reset saa7146 */
+ WRITEREG((MRST_N << 16), MC1);
+
+ /* Unset base addr */
+ chip->base_addr = NULL;
+
+ return 0;
+}
+
+void snd_aw2_saa7146_setup(struct snd_aw2_saa7146 *chip,
+ void __iomem *pci_base_addr)
+{
+ /* set PCI burst/threshold
+
+ Burst length definition
+ VALUE BURST LENGTH
+ 000 1 Dword
+ 001 2 Dwords
+ 010 4 Dwords
+ 011 8 Dwords
+ 100 16 Dwords
+ 101 32 Dwords
+ 110 64 Dwords
+ 111 128 Dwords
+
+ Threshold definition
+ VALUE WRITE MODE READ MODE
+ 00 1 Dword of valid data 1 empty Dword
+ 01 4 Dwords of valid data 4 empty Dwords
+ 10 8 Dwords of valid data 8 empty Dwords
+ 11 16 Dwords of valid data 16 empty Dwords */
+
+ unsigned int acon2;
+ unsigned int acon1 = 0;
+ int i;
+
+ /* Set base addr */
+ chip->base_addr = pci_base_addr;
+
+ /* disable all irqs */
+ WRITEREG(0, IER);
+
+ /* reset saa7146 */
+ WRITEREG((MRST_N << 16), MC1);
+
+ /* enable audio interface */
+#ifdef __BIG_ENDIAN
+ acon1 |= A1_SWAP;
+ acon1 |= A2_SWAP;
+#endif
+ /* WS0_CTRL, WS0_SYNC: input TSL1, I2S */
+
+ /* At initialization WS1 and WS2 are disbaled (configured as input */
+ acon1 |= 0 * WS1_CTRL;
+ acon1 |= 0 * WS2_CTRL;
+
+ /* WS4 is not used. So it must not restart A2.
+ This is why it is configured as output (force to low) */
+ acon1 |= 3 * WS4_CTRL;
+
+ /* WS3_CTRL, WS3_SYNC: output TSL2, I2S */
+ acon1 |= 2 * WS3_CTRL;
+
+ /* A1 and A2 are active and asynchronous */
+ acon1 |= 3 * AUDIO_MODE;
+ WRITEREG(acon1, ACON1);
+
+ /* The following comes from original windows driver.
+ It is needed to have a correct behavior of input and output
+ simultenously, but I don't know why ! */
+ WRITEREG(3 * (BurstA1_in) + 3 * (ThreshA1_in) +
+ 3 * (BurstA1_out) + 3 * (ThreshA1_out) +
+ 3 * (BurstA2_out) + 3 * (ThreshA2_out), PCI_BT_A);
+
+ /* enable audio port pins */
+ WRITEREG((EAP << 16) | EAP, MC1);
+
+ /* enable I2C */
+ WRITEREG((EI2C << 16) | EI2C, MC1);
+ /* enable interrupts */
+ WRITEREG(A1_out | A2_out | A1_in | IIC_S | IIC_E, IER);
+
+ /* audio configuration */
+ acon2 = A2_CLKSRC | BCLK1_OEN;
+ WRITEREG(acon2, ACON2);
+
+ /* By default use analog input */
+ snd_aw2_saa7146_use_digital_input(chip, 0);
+
+ /* TSL setup */
+ for (i = 0; i < 8; ++i) {
+ WRITEREG(tsl1[i], TSL1 + (i * 4));
+ WRITEREG(tsl2[i], TSL2 + (i * 4));
+ }
+
+}
+
+void snd_aw2_saa7146_pcm_init_playback(struct snd_aw2_saa7146 *chip,
+ int stream_number,
+ unsigned long dma_addr,
+ unsigned long period_size,
+ unsigned long buffer_size)
+{
+ unsigned long dw_page, dw_limit;
+
+ /* Configure DMA for substream
+ Configuration informations: ALSA has allocated continuous memory
+ pages. So we don't need to use MMU of saa7146.
+ */
+
+ /* No MMU -> nothing to do with PageA1, we only configure the limit of
+ PageAx_out register */
+ /* Disable MMU */
+ dw_page = (0L << 11);
+
+ /* Configure Limit for DMA access.
+ The limit register defines an address limit, which generates
+ an interrupt if passed by the actual PCI address pointer.
+ '0001' means an interrupt will be generated if the lower
+ 6 bits (64 bytes) of the PCI address are zero. '0010'
+ defines a limit of 128 bytes, '0011' one of 256 bytes, and
+ so on up to 1 Mbyte defined by '1111'. This interrupt range
+ can be calculated as follows:
+ Range = 2^(5 + Limit) bytes.
+ */
+ dw_limit = snd_aw2_saa7146_get_limit(period_size);
+ dw_page |= (dw_limit << 4);
+
+ if (stream_number == 0) {
+ WRITEREG(dw_page, PageA2_out);
+
+ /* Base address for DMA transfert. */
+ /* This address has been reserved by ALSA. */
+ /* This is a physical address */
+ WRITEREG(dma_addr, BaseA2_out);
+
+ /* Define upper limit for DMA access */
+ WRITEREG(dma_addr + buffer_size, ProtA2_out);
+
+ } else if (stream_number == 1) {
+ WRITEREG(dw_page, PageA1_out);
+
+ /* Base address for DMA transfert. */
+ /* This address has been reserved by ALSA. */
+ /* This is a physical address */
+ WRITEREG(dma_addr, BaseA1_out);
+
+ /* Define upper limit for DMA access */
+ WRITEREG(dma_addr + buffer_size, ProtA1_out);
+ } else {
+ printk(KERN_ERR
+ "aw2: snd_aw2_saa7146_pcm_init_playback: "
+ "Substream number is not 0 or 1 -> not managed\n");
+ }
+}
+
+void snd_aw2_saa7146_pcm_init_capture(struct snd_aw2_saa7146 *chip,
+ int stream_number, unsigned long dma_addr,
+ unsigned long period_size,
+ unsigned long buffer_size)
+{
+ unsigned long dw_page, dw_limit;
+
+ /* Configure DMA for substream
+ Configuration informations: ALSA has allocated continuous memory
+ pages. So we don't need to use MMU of saa7146.
+ */
+
+ /* No MMU -> nothing to do with PageA1, we only configure the limit of
+ PageAx_out register */
+ /* Disable MMU */
+ dw_page = (0L << 11);
+
+ /* Configure Limit for DMA access.
+ The limit register defines an address limit, which generates
+ an interrupt if passed by the actual PCI address pointer.
+ '0001' means an interrupt will be generated if the lower
+ 6 bits (64 bytes) of the PCI address are zero. '0010'
+ defines a limit of 128 bytes, '0011' one of 256 bytes, and
+ so on up to 1 Mbyte defined by '1111'. This interrupt range
+ can be calculated as follows:
+ Range = 2^(5 + Limit) bytes.
+ */
+ dw_limit = snd_aw2_saa7146_get_limit(period_size);
+ dw_page |= (dw_limit << 4);
+
+ if (stream_number == 0) {
+ WRITEREG(dw_page, PageA1_in);
+
+ /* Base address for DMA transfert. */
+ /* This address has been reserved by ALSA. */
+ /* This is a physical address */
+ WRITEREG(dma_addr, BaseA1_in);
+
+ /* Define upper limit for DMA access */
+ WRITEREG(dma_addr + buffer_size, ProtA1_in);
+ } else {
+ printk(KERN_ERR
+ "aw2: snd_aw2_saa7146_pcm_init_capture: "
+ "Substream number is not 0 -> not managed\n");
+ }
+}
+
+void snd_aw2_saa7146_define_it_playback_callback(unsigned int stream_number,
+ snd_aw2_saa7146_it_cb
+ p_it_callback,
+ void *p_callback_param)
+{
+ if (stream_number < NB_STREAM_PLAYBACK) {
+ arr_substream_it_playback_cb[stream_number].p_it_callback =
+ (snd_aw2_saa7146_it_cb) p_it_callback;
+ arr_substream_it_playback_cb[stream_number].p_callback_param =
+ (void *)p_callback_param;
+ }
+}
+
+void snd_aw2_saa7146_define_it_capture_callback(unsigned int stream_number,
+ snd_aw2_saa7146_it_cb
+ p_it_callback,
+ void *p_callback_param)
+{
+ if (stream_number < NB_STREAM_CAPTURE) {
+ arr_substream_it_capture_cb[stream_number].p_it_callback =
+ (snd_aw2_saa7146_it_cb) p_it_callback;
+ arr_substream_it_capture_cb[stream_number].p_callback_param =
+ (void *)p_callback_param;
+ }
+}
+
+void snd_aw2_saa7146_pcm_trigger_start_playback(struct snd_aw2_saa7146 *chip,
+ int stream_number)
+{
+ unsigned int acon1 = 0;
+ /* In aw8 driver, dma transfert is always active. It is
+ started and stopped in a larger "space" */
+ acon1 = READREG(ACON1);
+ if (stream_number == 0) {
+ WRITEREG((TR_E_A2_OUT << 16) | TR_E_A2_OUT, MC1);
+
+ /* WS2_CTRL, WS2_SYNC: output TSL2, I2S */
+ acon1 |= 2 * WS2_CTRL;
+ WRITEREG(acon1, ACON1);
+
+ } else if (stream_number == 1) {
+ WRITEREG((TR_E_A1_OUT << 16) | TR_E_A1_OUT, MC1);
+
+ /* WS1_CTRL, WS1_SYNC: output TSL1, I2S */
+ acon1 |= 1 * WS1_CTRL;
+ WRITEREG(acon1, ACON1);
+ }
+}
+
+void snd_aw2_saa7146_pcm_trigger_stop_playback(struct snd_aw2_saa7146 *chip,
+ int stream_number)
+{
+ unsigned int acon1 = 0;
+ acon1 = READREG(ACON1);
+ if (stream_number == 0) {
+ /* WS2_CTRL, WS2_SYNC: output TSL2, I2S */
+ acon1 &= ~(3 * WS2_CTRL);
+ WRITEREG(acon1, ACON1);
+
+ WRITEREG((TR_E_A2_OUT << 16), MC1);
+ } else if (stream_number == 1) {
+ /* WS1_CTRL, WS1_SYNC: output TSL1, I2S */
+ acon1 &= ~(3 * WS1_CTRL);
+ WRITEREG(acon1, ACON1);
+
+ WRITEREG((TR_E_A1_OUT << 16), MC1);
+ }
+}
+
+void snd_aw2_saa7146_pcm_trigger_start_capture(struct snd_aw2_saa7146 *chip,
+ int stream_number)
+{
+ /* In aw8 driver, dma transfert is always active. It is
+ started and stopped in a larger "space" */
+ if (stream_number == 0)
+ WRITEREG((TR_E_A1_IN << 16) | TR_E_A1_IN, MC1);
+}
+
+void snd_aw2_saa7146_pcm_trigger_stop_capture(struct snd_aw2_saa7146 *chip,
+ int stream_number)
+{
+ if (stream_number == 0)
+ WRITEREG((TR_E_A1_IN << 16), MC1);
+}
+
+irqreturn_t snd_aw2_saa7146_interrupt(int irq, void *dev_id)
+{
+ unsigned int isr;
+ unsigned int iicsta;
+ struct snd_aw2_saa7146 *chip = dev_id;
+
+ isr = READREG(ISR);
+ if (!isr)
+ return IRQ_NONE;
+
+ WRITEREG(isr, ISR);
+
+ if (isr & (IIC_S | IIC_E)) {
+ iicsta = READREG(IICSTA);
+ WRITEREG(0x100, IICSTA);
+ }
+
+ if (isr & A1_out) {
+ if (arr_substream_it_playback_cb[1].p_it_callback != NULL) {
+ arr_substream_it_playback_cb[1].
+ p_it_callback(arr_substream_it_playback_cb[1].
+ p_callback_param);
+ }
+ }
+ if (isr & A2_out) {
+ if (arr_substream_it_playback_cb[0].p_it_callback != NULL) {
+ arr_substream_it_playback_cb[0].
+ p_it_callback(arr_substream_it_playback_cb[0].
+ p_callback_param);
+ }
+
+ }
+ if (isr & A1_in) {
+ if (arr_substream_it_capture_cb[0].p_it_callback != NULL) {
+ arr_substream_it_capture_cb[0].
+ p_it_callback(arr_substream_it_capture_cb[0].
+ p_callback_param);
+ }
+ }
+ return IRQ_HANDLED;
+}
+
+unsigned int snd_aw2_saa7146_get_hw_ptr_playback(struct snd_aw2_saa7146 *chip,
+ int stream_number,
+ unsigned char *start_addr,
+ unsigned int buffer_size)
+{
+ long pci_adp = 0;
+ size_t ptr = 0;
+
+ if (stream_number == 0) {
+ pci_adp = READREG(PCI_ADP3);
+ ptr = pci_adp - (long)start_addr;
+
+ if (ptr == buffer_size)
+ ptr = 0;
+ }
+ if (stream_number == 1) {
+ pci_adp = READREG(PCI_ADP1);
+ ptr = pci_adp - (size_t) start_addr;
+
+ if (ptr == buffer_size)
+ ptr = 0;
+ }
+ return ptr;
+}
+
+unsigned int snd_aw2_saa7146_get_hw_ptr_capture(struct snd_aw2_saa7146 *chip,
+ int stream_number,
+ unsigned char *start_addr,
+ unsigned int buffer_size)
+{
+ size_t pci_adp = 0;
+ size_t ptr = 0;
+ if (stream_number == 0) {
+ pci_adp = READREG(PCI_ADP2);
+ ptr = pci_adp - (size_t) start_addr;
+
+ if (ptr == buffer_size)
+ ptr = 0;
+ }
+ return ptr;
+}
+
+void snd_aw2_saa7146_use_digital_input(struct snd_aw2_saa7146 *chip,
+ int use_digital)
+{
+ /* FIXME: switch between analog and digital input does not always work.
+ It can produce a kind of white noise. It seams that received data
+ are inverted sometime (endian inversion). Why ? I don't know, maybe
+ a problem of synchronization... However for the time being I have
+ not found the problem. Workaround: switch again (and again) between
+ digital and analog input until it works. */
+ if (use_digital)
+ WRITEREG(0x40, GPIO_CTRL);
+ else
+ WRITEREG(0x50, GPIO_CTRL);
+}
+
+int snd_aw2_saa7146_is_using_digital_input(struct snd_aw2_saa7146 *chip)
+{
+ unsigned int reg_val = READREG(GPIO_CTRL);
+ if ((reg_val & 0xFF) == 0x40)
+ return 1;
+ else
+ return 0;
+}
+
+
+static int snd_aw2_saa7146_get_limit(int size)
+{
+ int limitsize = 32;
+ int limit = 0;
+ while (limitsize < size) {
+ limitsize *= 2;
+ limit++;
+ }
+ return limit;
+}
diff --git a/sound/pci/aw2/aw2-saa7146.h b/sound/pci/aw2/aw2-saa7146.h
new file mode 100644
index 00000000000..5b35e358937
--- /dev/null
+++ b/sound/pci/aw2/aw2-saa7146.h
@@ -0,0 +1,105 @@
+/*****************************************************************************
+ *
+ * Copyright (C) 2008 Cedric Bregardis <cedric.bregardis@free.fr> and
+ * Jean-Christian Hassler <jhassler@free.fr>
+ *
+ * This file is part of the Audiowerk2 ALSA driver
+ *
+ * The Audiowerk2 ALSA driver is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2.
+ *
+ * The Audiowerk2 ALSA driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with the Audiowerk2 ALSA driver; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301,
+ * USA.
+ *
+ *****************************************************************************/
+
+#ifndef AW2_SAA7146_H
+#define AW2_SAA7146_H
+
+#define NB_STREAM_PLAYBACK 2
+#define NB_STREAM_CAPTURE 1
+
+#define NUM_STREAM_PLAYBACK_ANA 0
+#define NUM_STREAM_PLAYBACK_DIG 1
+
+#define NUM_STREAM_CAPTURE_ANA 0
+
+typedef void (*snd_aw2_saa7146_it_cb) (void *);
+
+struct snd_aw2_saa7146_cb_param {
+ snd_aw2_saa7146_it_cb p_it_callback;
+ void *p_callback_param;
+};
+
+/* definition of the chip-specific record */
+
+struct snd_aw2_saa7146 {
+ void __iomem *base_addr;
+};
+
+extern void snd_aw2_saa7146_setup(struct snd_aw2_saa7146 *chip,
+ void __iomem *pci_base_addr);
+extern int snd_aw2_saa7146_free(struct snd_aw2_saa7146 *chip);
+
+extern void snd_aw2_saa7146_pcm_init_playback(struct snd_aw2_saa7146 *chip,
+ int stream_number,
+ unsigned long dma_addr,
+ unsigned long period_size,
+ unsigned long buffer_size);
+extern void snd_aw2_saa7146_pcm_init_capture(struct snd_aw2_saa7146 *chip,
+ int stream_number,
+ unsigned long dma_addr,
+ unsigned long period_size,
+ unsigned long buffer_size);
+extern void snd_aw2_saa7146_define_it_playback_callback(unsigned int
+ stream_number,
+ snd_aw2_saa7146_it_cb
+ p_it_callback,
+ void *p_callback_param);
+extern void snd_aw2_saa7146_define_it_capture_callback(unsigned int
+ stream_number,
+ snd_aw2_saa7146_it_cb
+ p_it_callback,
+ void *p_callback_param);
+extern void snd_aw2_saa7146_pcm_trigger_start_capture(struct snd_aw2_saa7146
+ *chip, int stream_number);
+extern void snd_aw2_saa7146_pcm_trigger_stop_capture(struct snd_aw2_saa7146
+ *chip, int stream_number);
+
+extern void snd_aw2_saa7146_pcm_trigger_start_playback(struct snd_aw2_saa7146
+ *chip,
+ int stream_number);
+extern void snd_aw2_saa7146_pcm_trigger_stop_playback(struct snd_aw2_saa7146
+ *chip, int stream_number);
+
+extern irqreturn_t snd_aw2_saa7146_interrupt(int irq, void *dev_id);
+extern unsigned int snd_aw2_saa7146_get_hw_ptr_playback(struct snd_aw2_saa7146
+ *chip,
+ int stream_number,
+ unsigned char
+ *start_addr,
+ unsigned int
+ buffer_size);
+extern unsigned int snd_aw2_saa7146_get_hw_ptr_capture(struct snd_aw2_saa7146
+ *chip,
+ int stream_number,
+ unsigned char
+ *start_addr,
+ unsigned int
+ buffer_size);
+
+extern void snd_aw2_saa7146_use_digital_input(struct snd_aw2_saa7146 *chip,
+ int use_digital);
+
+extern int snd_aw2_saa7146_is_using_digital_input(struct snd_aw2_saa7146
+ *chip);
+
+#endif
diff --git a/sound/pci/aw2/aw2-tsl.c b/sound/pci/aw2/aw2-tsl.c
new file mode 100644
index 00000000000..459b0311ea3
--- /dev/null
+++ b/sound/pci/aw2/aw2-tsl.c
@@ -0,0 +1,110 @@
+/*****************************************************************************
+ *
+ * Copyright (C) 2008 Cedric Bregardis <cedric.bregardis@free.fr> and
+ * Jean-Christian Hassler <jhassler@free.fr>
+ * Copyright 1998 Emagic Soft- und Hardware GmbH
+ * Copyright 2002 Martijn Sipkema
+ *
+ * This file is part of the Audiowerk2 ALSA driver
+ *
+ * The Audiowerk2 ALSA driver is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2.
+ *
+ * The Audiowerk2 ALSA driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with the Audiowerk2 ALSA driver; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301,
+ * USA.
+ *
+ *****************************************************************************/
+
+#define TSL_WS0 (1UL << 31)
+#define TSL_WS1 (1UL << 30)
+#define TSL_WS2 (1UL << 29)
+#define TSL_WS3 (1UL << 28)
+#define TSL_WS4 (1UL << 27)
+#define TSL_DIS_A1 (1UL << 24)
+#define TSL_SDW_A1 (1UL << 23)
+#define TSL_SIB_A1 (1UL << 22)
+#define TSL_SF_A1 (1UL << 21)
+#define TSL_LF_A1 (1UL << 20)
+#define TSL_BSEL_A1 (1UL << 17)
+#define TSL_DOD_A1 (1UL << 15)
+#define TSL_LOW_A1 (1UL << 14)
+#define TSL_DIS_A2 (1UL << 11)
+#define TSL_SDW_A2 (1UL << 10)
+#define TSL_SIB_A2 (1UL << 9)
+#define TSL_SF_A2 (1UL << 8)
+#define TSL_LF_A2 (1UL << 7)
+#define TSL_BSEL_A2 (1UL << 4)
+#define TSL_DOD_A2 (1UL << 2)
+#define TSL_LOW_A2 (1UL << 1)
+#define TSL_EOS (1UL << 0)
+
+ /* Audiowerk8 hardware setup: */
+ /* WS0, SD4, TSL1 - Analog/ digital in */
+ /* WS1, SD0, TSL1 - Analog out #1, digital out */
+ /* WS2, SD2, TSL1 - Analog out #2 */
+ /* WS3, SD1, TSL2 - Analog out #3 */
+ /* WS4, SD3, TSL2 - Analog out #4 */
+
+ /* Audiowerk8 timing: */
+ /* Timeslot: | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | ... */
+
+ /* A1_INPUT: */
+ /* SD4: <_ADC-L_>-------<_ADC-R_>-------< */
+ /* WS0: _______________/---------------\_ */
+
+ /* A1_OUTPUT: */
+ /* SD0: <_1-L___>-------<_1-R___>-------< */
+ /* WS1: _______________/---------------\_ */
+ /* SD2: >-------<_2-L___>-------<_2-R___> */
+ /* WS2: -------\_______________/--------- */
+
+ /* A2_OUTPUT: */
+ /* SD1: <_3-L___>-------<_3-R___>-------< */
+ /* WS3: _______________/---------------\_ */
+ /* SD3: >-------<_4-L___>-------<_4-R___> */
+ /* WS4: -------\_______________/--------- */
+
+static int tsl1[8] = {
+ 1 * TSL_SDW_A1 | 3 * TSL_BSEL_A1 |
+ 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1 | TSL_LF_A1,
+
+ 1 * TSL_SDW_A1 | 2 * TSL_BSEL_A1 |
+ 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1,
+
+ 0 * TSL_SDW_A1 | 3 * TSL_BSEL_A1 |
+ 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1,
+
+ 0 * TSL_SDW_A1 | 2 * TSL_BSEL_A1 |
+ 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1,
+
+ 1 * TSL_SDW_A1 | 1 * TSL_BSEL_A1 |
+ 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1 | TSL_WS1 | TSL_WS0,
+
+ 1 * TSL_SDW_A1 | 0 * TSL_BSEL_A1 |
+ 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1 | TSL_WS1 | TSL_WS0,
+
+ 0 * TSL_SDW_A1 | 1 * TSL_BSEL_A1 |
+ 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1 | TSL_WS1 | TSL_WS0,
+
+ 0 * TSL_SDW_A1 | 0 * TSL_BSEL_A1 | 0 * TSL_DIS_A1 |
+ 0 * TSL_DOD_A1 | TSL_WS1 | TSL_WS0 | TSL_SF_A1 | TSL_EOS,
+};
+
+static int tsl2[8] = {
+ 0 * TSL_SDW_A2 | 3 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_LF_A2,
+ 0 * TSL_SDW_A2 | 2 * TSL_BSEL_A2 | 2 * TSL_DOD_A2,
+ 0 * TSL_SDW_A2 | 3 * TSL_BSEL_A2 | 2 * TSL_DOD_A2,
+ 0 * TSL_SDW_A2 | 2 * TSL_BSEL_A2 | 2 * TSL_DOD_A2,
+ 0 * TSL_SDW_A2 | 1 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_WS2,
+ 0 * TSL_SDW_A2 | 0 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_WS2,
+ 0 * TSL_SDW_A2 | 1 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_WS2,
+ 0 * TSL_SDW_A2 | 0 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_WS2 | TSL_EOS
+};
diff --git a/sound/pci/aw2/saa7146.h b/sound/pci/aw2/saa7146.h
new file mode 100644
index 00000000000..ce0ab5f9ee9
--- /dev/null
+++ b/sound/pci/aw2/saa7146.h
@@ -0,0 +1,168 @@
+/*****************************************************************************
+ *
+ * Copyright (C) 2008 Cedric Bregardis <cedric.bregardis@free.fr> and
+ * Jean-Christian Hassler <jhassler@free.fr>
+ *
+ * This file is part of the Audiowerk2 ALSA driver
+ *
+ * The Audiowerk2 ALSA driver is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2.
+ *
+ * The Audiowerk2 ALSA driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with the Audiowerk2 ALSA driver; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301,
+ * USA.
+ *
+ *****************************************************************************/
+
+/* SAA7146 registers */
+#define PCI_BT_A 0x4C
+#define IICTFR 0x8C
+#define IICSTA 0x90
+#define BaseA1_in 0x94
+#define ProtA1_in 0x98
+#define PageA1_in 0x9C
+#define BaseA1_out 0xA0
+#define ProtA1_out 0xA4
+#define PageA1_out 0xA8
+#define BaseA2_in 0xAC
+#define ProtA2_in 0xB0
+#define PageA2_in 0xB4
+#define BaseA2_out 0xB8
+#define ProtA2_out 0xBC
+#define PageA2_out 0xC0
+#define IER 0xDC
+#define GPIO_CTRL 0xE0
+#define ACON1 0xF4
+#define ACON2 0xF8
+#define MC1 0xFC
+#define MC2 0x100
+#define ISR 0x10C
+#define PSR 0x110
+#define SSR 0x114
+#define PCI_ADP1 0x12C
+#define PCI_ADP2 0x130
+#define PCI_ADP3 0x134
+#define PCI_ADP4 0x138
+#define LEVEL_REP 0x140
+#define FB_BUFFER1 0x144
+#define FB_BUFFER2 0x148
+#define TSL1 0x180
+#define TSL2 0x1C0
+
+#define ME (1UL << 11)
+#define LIMIT (1UL << 4)
+#define PV (1UL << 3)
+
+/* PSR/ISR/IER */
+#define PPEF (1UL << 31)
+#define PABO (1UL << 30)
+#define IIC_S (1UL << 17)
+#define IIC_E (1UL << 16)
+#define A2_in (1UL << 15)
+#define A2_out (1UL << 14)
+#define A1_in (1UL << 13)
+#define A1_out (1UL << 12)
+#define AFOU (1UL << 11)
+#define PIN3 (1UL << 6)
+#define PIN2 (1UL << 5)
+#define PIN1 (1UL << 4)
+#define PIN0 (1UL << 3)
+#define ECS (1UL << 2)
+#define EC3S (1UL << 1)
+#define EC0S (1UL << 0)
+
+/* SSR */
+#define PRQ (1UL << 31)
+#define PMA (1UL << 30)
+#define IIC_EA (1UL << 21)
+#define IIC_EW (1UL << 20)
+#define IIC_ER (1UL << 19)
+#define IIC_EL (1UL << 18)
+#define IIC_EF (1UL << 17)
+#define AF2_in (1UL << 10)
+#define AF2_out (1UL << 9)
+#define AF1_in (1UL << 8)
+#define AF1_out (1UL << 7)
+#define EC5S (1UL << 3)
+#define EC4S (1UL << 2)
+#define EC2S (1UL << 1)
+#define EC1S (1UL << 0)
+
+/* PCI_BT_A */
+#define BurstA1_in (1UL << 26)
+#define ThreshA1_in (1UL << 24)
+#define BurstA1_out (1UL << 18)
+#define ThreshA1_out (1UL << 16)
+#define BurstA2_in (1UL << 10)
+#define ThreshA2_in (1UL << 8)
+#define BurstA2_out (1UL << 2)
+#define ThreshA2_out (1UL << 0)
+
+/* MC1 */
+#define MRST_N (1UL << 15)
+#define EAP (1UL << 9)
+#define EI2C (1UL << 8)
+#define TR_E_A2_OUT (1UL << 3)
+#define TR_E_A2_IN (1UL << 2)
+#define TR_E_A1_OUT (1UL << 1)
+#define TR_E_A1_IN (1UL << 0)
+
+/* MC2 */
+#define UPLD_IIC (1UL << 0)
+
+/* ACON1 */
+#define AUDIO_MODE (1UL << 29)
+#define MAXLEVEL (1UL << 22)
+#define A1_SWAP (1UL << 21)
+#define A2_SWAP (1UL << 20)
+#define WS0_CTRL (1UL << 18)
+#define WS0_SYNC (1UL << 16)
+#define WS1_CTRL (1UL << 14)
+#define WS1_SYNC (1UL << 12)
+#define WS2_CTRL (1UL << 10)
+#define WS2_SYNC (1UL << 8)
+#define WS3_CTRL (1UL << 6)
+#define WS3_SYNC (1UL << 4)
+#define WS4_CTRL (1UL << 2)
+#define WS4_SYNC (1UL << 0)
+
+/* ACON2 */
+#define A1_CLKSRC (1UL << 27)
+#define A2_CLKSRC (1UL << 22)
+#define INVERT_BCLK1 (1UL << 21)
+#define INVERT_BCLK2 (1UL << 20)
+#define BCLK1_OEN (1UL << 19)
+#define BCLK2_OEN (1UL << 18)
+
+/* IICSTA */
+#define IICCC (1UL << 8)
+#define ABORT (1UL << 7)
+#define SPERR (1UL << 6)
+#define APERR (1UL << 5)
+#define DTERR (1UL << 4)
+#define DRERR (1UL << 3)
+#define AL (1UL << 2)
+#define ERR (1UL << 1)
+#define BUSY (1UL << 0)
+
+/* IICTFR */
+#define BYTE2 (1UL << 24)
+#define BYTE1 (1UL << 16)
+#define BYTE0 (1UL << 8)
+#define ATRR2 (1UL << 6)
+#define ATRR1 (1UL << 4)
+#define ATRR0 (1UL << 2)
+#define ERR (1UL << 1)
+#define BUSY (1UL << 0)
+
+#define START 3
+#define CONT 2
+#define STOP 1
+#define NOP 0
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 4e71a55120a..5f63af6b88a 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -157,8 +157,8 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}");
#if DEBUG_CALLS
#define snd_azf3328_dbgcalls(format, args...) printk(format, ##args)
-#define snd_azf3328_dbgcallenter() printk(KERN_ERR "--> %s\n", __FUNCTION__)
-#define snd_azf3328_dbgcallleave() printk(KERN_ERR "<-- %s\n", __FUNCTION__)
+#define snd_azf3328_dbgcallenter() printk(KERN_ERR "--> %s\n", __func__)
+#define snd_azf3328_dbgcallleave() printk(KERN_ERR "<-- %s\n", __func__)
#else
#define snd_azf3328_dbgcalls(format, args...)
#define snd_azf3328_dbgcallenter()
@@ -1514,7 +1514,8 @@ snd_azf3328_free(struct snd_azf3328 *chip)
/* well, at least we know how to disable the timer IRQ */
snd_azf3328_codec_outb(chip, IDX_IO_TIMER_VALUE + 3, 0x00);
- synchronize_irq(chip->irq);
+ if (chip->irq >= 0)
+ synchronize_irq(chip->irq);
__end_hw:
snd_azf3328_free_joystick(chip);
if (chip->irq >= 0)
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index 176e0f0e805..ecbe79b67e4 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -435,22 +435,22 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu,
static void snd_ca0106_intr_enable(struct snd_ca0106 *emu, unsigned int intrenb)
{
unsigned long flags;
- unsigned int enable;
-
+ unsigned int intr_enable;
+
spin_lock_irqsave(&emu->emu_lock, flags);
- enable = inl(emu->port + INTE) | intrenb;
- outl(enable, emu->port + INTE);
+ intr_enable = inl(emu->port + INTE) | intrenb;
+ outl(intr_enable, emu->port + INTE);
spin_unlock_irqrestore(&emu->emu_lock, flags);
}
static void snd_ca0106_intr_disable(struct snd_ca0106 *emu, unsigned int intrenb)
{
unsigned long flags;
- unsigned int enable;
-
+ unsigned int intr_enable;
+
spin_lock_irqsave(&emu->emu_lock, flags);
- enable = inl(emu->port + INTE) & ~intrenb;
- outl(enable, emu->port + INTE);
+ intr_enable = inl(emu->port + INTE) & ~intrenb;
+ outl(intr_enable, emu->port + INTE);
spin_unlock_irqrestore(&emu->emu_lock, flags);
}
@@ -1114,6 +1114,8 @@ static int snd_ca0106_free(struct snd_ca0106 *chip)
* So we can fix: snd-malloc: Memory leak? pages not freed = 8
*/
}
+ if (chip->irq >= 0)
+ free_irq(chip->irq, chip);
// release the data
#if 1
if (chip->buffer.area)
@@ -1123,9 +1125,6 @@ static int snd_ca0106_free(struct snd_ca0106 *chip)
// release the i/o port
release_and_free_resource(chip->res_port);
- // release the irq
- if (chip->irq >= 0)
- free_irq(chip->irq, chip);
pci_disable_device(chip->pci);
kfree(chip);
return 0;
diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c
index af736869d9b..3025ed1b6e1 100644
--- a/sound/pci/ca0106/ca0106_mixer.c
+++ b/sound/pci/ca0106/ca0106_mixer.c
@@ -650,19 +650,55 @@ static int __devinit rename_ctl(struct snd_card *card, const char *src, const ch
#define ADD_CTLS(emu, ctls) \
do { \
- int i, err; \
+ int i, _err; \
for (i = 0; i < ARRAY_SIZE(ctls); i++) { \
- err = snd_ctl_add(card, snd_ctl_new1(&ctls[i], emu)); \
- if (err < 0) \
- return err; \
+ _err = snd_ctl_add(card, snd_ctl_new1(&ctls[i], emu)); \
+ if (_err < 0) \
+ return _err; \
} \
} while (0)
+static __devinitdata
+DECLARE_TLV_DB_SCALE(snd_ca0106_master_db_scale, -6375, 50, 1);
+
+static char *slave_vols[] __devinitdata = {
+ "Analog Front Playback Volume",
+ "Analog Rear Playback Volume",
+ "Analog Center/LFE Playback Volume",
+ "Analog Side Playback Volume",
+ "IEC958 Front Playback Volume",
+ "IEC958 Rear Playback Volume",
+ "IEC958 Center/LFE Playback Volume",
+ "IEC958 Unknown Playback Volume",
+ "CAPTURE feedback Playback Volume",
+ NULL
+};
+
+static char *slave_sws[] __devinitdata = {
+ "Analog Front Playback Switch",
+ "Analog Rear Playback Switch",
+ "Analog Center/LFE Playback Switch",
+ "Analog Side Playback Switch",
+ "IEC958 Playback Switch",
+ NULL
+};
+
+static void __devinit add_slaves(struct snd_card *card,
+ struct snd_kcontrol *master, char **list)
+{
+ for (; *list; list++) {
+ struct snd_kcontrol *slave = ctl_find(card, *list);
+ if (slave)
+ snd_ctl_add_slave(master, slave);
+ }
+}
+
int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu)
{
int err;
struct snd_card *card = emu->card;
char **c;
+ struct snd_kcontrol *vmaster;
static char *ca0106_remove_ctls[] = {
"Master Mono Playback Switch",
"Master Mono Playback Volume",
@@ -719,6 +755,21 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu)
}
if (emu->details->spi_dac == 1)
ADD_CTLS(emu, snd_ca0106_volume_spi_dac_ctls);
+
+ /* Create virtual master controls */
+ vmaster = snd_ctl_make_virtual_master("Master Playback Volume",
+ snd_ca0106_master_db_scale);
+ if (!vmaster)
+ return -ENOMEM;
+ add_slaves(card, vmaster, slave_vols);
+
+ if (emu->details->spi_dac == 1) {
+ vmaster = snd_ctl_make_virtual_master("Master Playback Switch",
+ NULL);
+ if (!vmaster)
+ return -ENOMEM;
+ add_slaves(card, vmaster, slave_sws);
+ }
return 0;
}
diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index 135f3086075..9971b5b7735 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -2744,12 +2744,13 @@ static int __devinit snd_cmipci_mixer_new(struct cmipci *cm, int pcm_spdif_devic
}
for (idx = 0; idx < CM_SAVED_MIXERS; idx++) {
- struct snd_ctl_elem_id id;
+ struct snd_ctl_elem_id elem_id;
struct snd_kcontrol *ctl;
- memset(&id, 0, sizeof(id));
- id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
- strcpy(id.name, cm_saved_mixer[idx].name);
- if ((ctl = snd_ctl_find_id(cm->card, &id)) != NULL)
+ memset(&elem_id, 0, sizeof(elem_id));
+ elem_id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ strcpy(elem_id.name, cm_saved_mixer[idx].name);
+ ctl = snd_ctl_find_id(cm->card, &elem_id);
+ if (ctl)
cm->mixer_res_ctl[idx] = ctl;
}
@@ -2932,8 +2933,6 @@ static int snd_cmipci_free(struct cmipci *cm)
/* reset mixer */
snd_cmipci_mixer_write(cm, 0, 0);
- synchronize_irq(cm->irq);
-
free_irq(cm->irq, cm);
}
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index 87ddffcd9d8..e214e567dec 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -2772,6 +2772,9 @@ static int snd_cs46xx_free(struct snd_cs46xx *chip)
if (chip->irq >= 0)
free_irq(chip->irq, chip);
+ if (chip->active_ctrl)
+ chip->active_ctrl(chip, -chip->amplifier);
+
for (idx = 0; idx < 5; idx++) {
struct snd_cs46xx_region *region = &chip->region.idx[idx];
if (region->remap_addr)
@@ -2779,9 +2782,6 @@ static int snd_cs46xx_free(struct snd_cs46xx *chip)
release_and_free_resource(region->resource);
}
- if (chip->active_ctrl)
- chip->active_ctrl(chip, -chip->amplifier);
-
#ifdef CONFIG_SND_CS46XX_NEW_DSP
if (chip->dsp_spos_instance) {
cs46xx_dsp_spos_destroy(chip);
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index 90ec090792b..e16dc92e82f 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -1852,15 +1852,16 @@ static irqreturn_t snd_echo_interrupt(int irq, void *dev_id)
static int snd_echo_free(struct echoaudio *chip)
{
DE_INIT(("Stop DSP...\n"));
- if (chip->comm_page) {
+ if (chip->comm_page)
rest_in_peace(chip);
- snd_dma_free_pages(&chip->commpage_dma_buf);
- }
DE_INIT(("Stopped.\n"));
if (chip->irq >= 0)
free_irq(chip->irq, chip);
+ if (chip->comm_page)
+ snd_dma_free_pages(&chip->commpage_dma_buf);
+
if (chip->dsp_registers)
iounmap(chip->dsp_registers);
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 9a9b977d3cf..abde5b90188 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -1249,11 +1249,6 @@ static int snd_emu10k1_free(struct snd_emu10k1 *emu)
if (emu->port) { /* avoid access to already used hardware */
snd_emu10k1_fx8010_tram_setup(emu, 0);
snd_emu10k1_done(emu);
- /* remove reserved page */
- if (emu->reserved_page) {
- snd_emu10k1_synth_free(emu, (struct snd_util_memblk *)emu->reserved_page);
- emu->reserved_page = NULL;
- }
snd_emu10k1_free_efx(emu);
}
if (emu->card_capabilities->emu_model == EMU_MODEL_EMU1010) {
@@ -1262,6 +1257,14 @@ static int snd_emu10k1_free(struct snd_emu10k1 *emu)
}
if (emu->emu1010.firmware_thread)
kthread_stop(emu->emu1010.firmware_thread);
+ if (emu->irq >= 0)
+ free_irq(emu->irq, emu);
+ /* remove reserved page */
+ if (emu->reserved_page) {
+ snd_emu10k1_synth_free(emu,
+ (struct snd_util_memblk *)emu->reserved_page);
+ emu->reserved_page = NULL;
+ }
if (emu->memhdr)
snd_util_memhdr_free(emu->memhdr);
if (emu->silent_page.area)
@@ -1273,8 +1276,6 @@ static int snd_emu10k1_free(struct snd_emu10k1 *emu)
#ifdef CONFIG_PM
free_pm_buffer(emu);
#endif
- if (emu->irq >= 0)
- free_irq(emu->irq, emu);
if (emu->port)
pci_release_regions(emu->pci);
if (emu->card_capabilities->ca0151_chip) /* P16V */
diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c
index 5512abd98bd..491a4a50f86 100644
--- a/sound/pci/emu10k1/emu10k1x.c
+++ b/sound/pci/emu10k1/emu10k1x.c
@@ -327,22 +327,22 @@ static void snd_emu10k1x_ptr_write(struct emu10k1x *emu,
static void snd_emu10k1x_intr_enable(struct emu10k1x *emu, unsigned int intrenb)
{
unsigned long flags;
- unsigned int enable;
-
+ unsigned int intr_enable;
+
spin_lock_irqsave(&emu->emu_lock, flags);
- enable = inl(emu->port + INTE) | intrenb;
- outl(enable, emu->port + INTE);
+ intr_enable = inl(emu->port + INTE) | intrenb;
+ outl(intr_enable, emu->port + INTE);
spin_unlock_irqrestore(&emu->emu_lock, flags);
}
static void snd_emu10k1x_intr_disable(struct emu10k1x *emu, unsigned int intrenb)
{
unsigned long flags;
- unsigned int enable;
-
+ unsigned int intr_enable;
+
spin_lock_irqsave(&emu->emu_lock, flags);
- enable = inl(emu->port + INTE) & ~intrenb;
- outl(enable, emu->port + INTE);
+ intr_enable = inl(emu->port + INTE) & ~intrenb;
+ outl(intr_enable, emu->port + INTE);
spin_unlock_irqrestore(&emu->emu_lock, flags);
}
@@ -754,13 +754,13 @@ static int snd_emu10k1x_free(struct emu10k1x *chip)
// disable audio
outl(HCFG_LOCKSOUNDCACHE, chip->port + HCFG);
- // release the i/o port
- release_and_free_resource(chip->res_port);
-
- // release the irq
+ /* release the irq */
if (chip->irq >= 0)
free_irq(chip->irq, chip);
+ // release the i/o port
+ release_and_free_resource(chip->res_port);
+
// release the DMA
if (chip->dma_buffer.area) {
snd_dma_free_pages(&chip->dma_buffer);
@@ -795,9 +795,9 @@ static irqreturn_t snd_emu10k1x_interrupt(int irq, void *dev_id)
// capture interrupt
if (status & (IPR_CAP_0_LOOP | IPR_CAP_0_HALF_LOOP)) {
- struct emu10k1x_voice *pvoice = &chip->capture_voice;
- if (pvoice->use)
- snd_emu10k1x_pcm_interrupt(chip, pvoice);
+ struct emu10k1x_voice *cap_voice = &chip->capture_voice;
+ if (cap_voice->use)
+ snd_emu10k1x_pcm_interrupt(chip, cap_voice);
else
snd_emu10k1x_intr_disable(chip,
INTE_CAP_0_LOOP |
diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c
index f3caa3f890c..216f9748aff 100644
--- a/sound/pci/emu10k1/emuproc.c
+++ b/sound/pci/emu10k1/emuproc.c
@@ -412,7 +412,7 @@ static void snd_emu_proc_emu1010_reg_read(struct snd_info_entry *entry,
struct snd_info_buffer *buffer)
{
struct snd_emu10k1 *emu = entry->private_data;
- int value;
+ u32 value;
unsigned long flags;
int i;
snd_iprintf(buffer, "EMU1010 Registers:\n\n");
diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c
index 72d85a5ae6a..fbf1124f7c7 100644
--- a/sound/pci/ens1370.c
+++ b/sound/pci/ens1370.c
@@ -1635,20 +1635,20 @@ static int __devinit snd_ensoniq_1371_mixer(struct ensoniq *ensoniq,
if (has_spdif > 0 ||
(!has_spdif && es1371_quirk_lookup(ensoniq, es1371_spdif_present))) {
struct snd_kcontrol *kctl;
- int i, index = 0;
+ int i, is_spdif = 0;
ensoniq->spdif_default = ensoniq->spdif_stream =
SNDRV_PCM_DEFAULT_CON_SPDIF;
outl(ensoniq->spdif_default, ES_REG(ensoniq, CHANNEL_STATUS));
if (ensoniq->u.es1371.ac97->ext_id & AC97_EI_SPDIF)
- index++;
+ is_spdif++;
for (i = 0; i < ARRAY_SIZE(snd_es1371_mixer_spdif); i++) {
kctl = snd_ctl_new1(&snd_es1371_mixer_spdif[i], ensoniq);
if (!kctl)
return -ENOMEM;
- kctl->id.index = index;
+ kctl->id.index = is_spdif;
err = snd_ctl_add(card, kctl);
if (err < 0)
return err;
@@ -1910,7 +1910,8 @@ static int snd_ensoniq_free(struct ensoniq *ensoniq)
outl(0, ES_REG(ensoniq, CONTROL)); /* switch everything off */
outl(0, ES_REG(ensoniq, SERIAL)); /* clear serial interface */
#endif
- synchronize_irq(ensoniq->irq);
+ if (ensoniq->irq >= 0)
+ synchronize_irq(ensoniq->irq);
pci_set_power_state(ensoniq->pci, 3);
__hw_end:
#ifdef CHIP1370
diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c
index 1a314fa99c4..84fac1fbf10 100644
--- a/sound/pci/es1938.c
+++ b/sound/pci/es1938.c
@@ -1488,7 +1488,6 @@ static int es1938_suspend(struct pci_dev *pci, pm_message_t state)
outb(0x00, SLIO_REG(chip, IRQCONTROL)); /* disable irqs */
if (chip->irq >= 0) {
- synchronize_irq(chip->irq);
free_irq(chip->irq, chip);
chip->irq = -1;
}
@@ -1578,10 +1577,8 @@ static int snd_es1938_free(struct es1938 *chip)
snd_es1938_free_gameport(chip);
- if (chip->irq >= 0) {
- synchronize_irq(chip->irq);
+ if (chip->irq >= 0)
free_irq(chip->irq, chip);
- }
pci_release_regions(chip->pci);
pci_disable_device(chip->pci);
kfree(chip);
diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c
index 7d911a18c08..1bf298d214b 100644
--- a/sound/pci/es1968.c
+++ b/sound/pci/es1968.c
@@ -1827,6 +1827,22 @@ snd_es1968_pcm(struct es1968 *chip, int device)
return 0;
}
+/*
+ * suppress jitter on some maestros when playing stereo
+ */
+static void snd_es1968_suppress_jitter(struct es1968 *chip, struct esschan *es)
+{
+ unsigned int cp1;
+ unsigned int cp2;
+ unsigned int diff;
+
+ cp1 = __apu_get_register(chip, 0, 5);
+ cp2 = __apu_get_register(chip, 1, 5);
+ diff = (cp1 > cp2 ? cp1 - cp2 : cp2 - cp1);
+
+ if (diff > 1)
+ __maestro_write(chip, IDR0_DATA_PORT, cp1);
+}
/*
* update pointer
@@ -1948,8 +1964,11 @@ static irqreturn_t snd_es1968_interrupt(int irq, void *dev_id)
struct esschan *es;
spin_lock(&chip->substream_lock);
list_for_each_entry(es, &chip->substream_list, list) {
- if (es->running)
+ if (es->running) {
snd_es1968_update_pcm(chip, es);
+ if (es->fmt & ESS_FMT_STEREO)
+ snd_es1968_suppress_jitter(chip, es);
+ }
}
spin_unlock(&chip->substream_lock);
if (chip->in_measurement) {
@@ -1972,7 +1991,7 @@ snd_es1968_mixer(struct es1968 *chip)
{
struct snd_ac97_bus *pbus;
struct snd_ac97_template ac97;
- struct snd_ctl_elem_id id;
+ struct snd_ctl_elem_id elem_id;
int err;
static struct snd_ac97_bus_ops ops = {
.write = snd_es1968_ac97_write,
@@ -1989,14 +2008,14 @@ snd_es1968_mixer(struct es1968 *chip)
return err;
/* attach master switch / volumes for h/w volume control */
- memset(&id, 0, sizeof(id));
- id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
- strcpy(id.name, "Master Playback Switch");
- chip->master_switch = snd_ctl_find_id(chip->card, &id);
- memset(&id, 0, sizeof(id));
- id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
- strcpy(id.name, "Master Playback Volume");
- chip->master_volume = snd_ctl_find_id(chip->card, &id);
+ memset(&elem_id, 0, sizeof(elem_id));
+ elem_id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ strcpy(elem_id.name, "Master Playback Switch");
+ chip->master_switch = snd_ctl_find_id(chip->card, &elem_id);
+ memset(&elem_id, 0, sizeof(elem_id));
+ elem_id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ strcpy(elem_id.name, "Master Playback Volume");
+ chip->master_volume = snd_ctl_find_id(chip->card, &elem_id);
return 0;
}
@@ -2456,7 +2475,8 @@ static inline void snd_es1968_free_gameport(struct es1968 *chip) { }
static int snd_es1968_free(struct es1968 *chip)
{
if (chip->io_port) {
- synchronize_irq(chip->irq);
+ if (chip->irq >= 0)
+ synchronize_irq(chip->irq);
outw(1, chip->io_port + 0x04); /* clear WP interrupts */
outw(0, chip->io_port + ESM_PORT_HOST_IRQ); /* disable IRQ */
}
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index 4c300e6149f..c129f9e2072 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -1285,7 +1285,6 @@ static int wait_for_codec(struct fm801 *chip, unsigned int codec_id,
static int snd_fm801_chip_init(struct fm801 *chip, int resume)
{
- int id;
unsigned short cmdw;
if (chip->tea575x_tuner & 0x0010)
@@ -1310,13 +1309,14 @@ static int snd_fm801_chip_init(struct fm801 *chip, int resume)
} else {
/* my card has the secondary codec */
/* at address #3, so the loop is inverted */
- for (id = 3; id > 0; id--) {
- if (! wait_for_codec(chip, id, AC97_VENDOR_ID1,
+ int i;
+ for (i = 3; i > 0; i--) {
+ if (!wait_for_codec(chip, i, AC97_VENDOR_ID1,
msecs_to_jiffies(50))) {
cmdw = inw(FM801_REG(chip, AC97_DATA));
if (cmdw != 0xffff && cmdw != 0) {
chip->secondary = 1;
- chip->secondary_addr = id;
+ chip->secondary_addr = i;
break;
}
}
diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile
index 9e0d8a1268a..ab0c726d648 100644
--- a/sound/pci/hda/Makefile
+++ b/sound/pci/hda/Makefile
@@ -2,7 +2,7 @@ snd-hda-intel-y := hda_intel.o
# since snd-hda-intel is the only driver using hda-codec,
# merge it into a single module although it was originally
# designed to be individual modules
-snd-hda-intel-y += hda_codec.o vmaster.o
+snd-hda-intel-y += hda_codec.o
snd-hda-intel-$(CONFIG_PROC_FS) += hda_proc.o
snd-hda-intel-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o
snd-hda-intel-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 37c413923db..a6be6e3e871 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -31,6 +31,7 @@
#include <sound/initval.h>
#include "hda_local.h"
#include <sound/hda_hwdep.h>
+#include "hda_patch.h" /* codec presets */
#ifdef CONFIG_SND_HDA_POWER_SAVE
/* define this option here to hide as static */
@@ -51,21 +52,50 @@ struct hda_vendor_id {
/* codec vendor labels */
static struct hda_vendor_id hda_vendor_ids[] = {
- { 0x10ec, "Realtek" },
+ { 0x1002, "ATI" },
{ 0x1057, "Motorola" },
+ { 0x1095, "Silicon Image" },
+ { 0x10ec, "Realtek" },
{ 0x1106, "VIA" },
{ 0x111d, "IDT" },
+ { 0x11c1, "LSI" },
{ 0x11d4, "Analog Devices" },
{ 0x13f6, "C-Media" },
{ 0x14f1, "Conexant" },
+ { 0x17e8, "Chrontel" },
+ { 0x1854, "LG" },
{ 0x434d, "C-Media" },
{ 0x8384, "SigmaTel" },
{} /* terminator */
};
-/* codec presets */
-#include "hda_patch.h"
-
+static const struct hda_codec_preset *hda_preset_tables[] = {
+#ifdef CONFIG_SND_HDA_CODEC_REALTEK
+ snd_hda_preset_realtek,
+#endif
+#ifdef CONFIG_SND_HDA_CODEC_CMEDIA
+ snd_hda_preset_cmedia,
+#endif
+#ifdef CONFIG_SND_HDA_CODEC_ANALOG
+ snd_hda_preset_analog,
+#endif
+#ifdef CONFIG_SND_HDA_CODEC_SIGMATEL
+ snd_hda_preset_sigmatel,
+#endif
+#ifdef CONFIG_SND_HDA_CODEC_SI3054
+ snd_hda_preset_si3054,
+#endif
+#ifdef CONFIG_SND_HDA_CODEC_ATIHDMI
+ snd_hda_preset_atihdmi,
+#endif
+#ifdef CONFIG_SND_HDA_CODEC_CONEXANT
+ snd_hda_preset_conexant,
+#endif
+#ifdef CONFIG_SND_HDA_CODEC_VIA
+ snd_hda_preset_via,
+#endif
+ NULL
+};
#ifdef CONFIG_SND_HDA_POWER_SAVE
static void hda_power_work(struct work_struct *work);
@@ -690,6 +720,19 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, format);
}
+void snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid)
+{
+ if (!nid)
+ return;
+
+ snd_printdd("hda_codec_cleanup_stream: NID=0x%x\n", nid);
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0);
+#if 0 /* keep the format */
+ msleep(1);
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0);
+#endif
+}
+
/*
* amp access functions
*/
@@ -1037,16 +1080,24 @@ void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir,
}
/* find a mixer control element with the given name */
-struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec,
- const char *name)
+static struct snd_kcontrol *
+_snd_hda_find_mixer_ctl(struct hda_codec *codec,
+ const char *name, int idx)
{
struct snd_ctl_elem_id id;
memset(&id, 0, sizeof(id));
id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ id.index = idx;
strcpy(id.name, name);
return snd_ctl_find_id(codec->bus->card, &id);
}
+struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec,
+ const char *name)
+{
+ return _snd_hda_find_mixer_ctl(codec, name, 0);
+}
+
/* create a virtual master control and add slaves */
int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
unsigned int *tlv, const char **slaves)
@@ -1481,6 +1532,8 @@ static struct snd_kcontrol_new dig_mixes[] = {
{ } /* end */
};
+#define SPDIF_MAX_IDX 4 /* 4 instances should be enough to probe */
+
/**
* snd_hda_create_spdif_out_ctls - create Output SPDIF-related controls
* @codec: the HDA codec
@@ -1496,9 +1549,20 @@ int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid)
int err;
struct snd_kcontrol *kctl;
struct snd_kcontrol_new *dig_mix;
+ int idx;
+ for (idx = 0; idx < SPDIF_MAX_IDX; idx++) {
+ if (!_snd_hda_find_mixer_ctl(codec, "IEC958 Playback Switch",
+ idx))
+ break;
+ }
+ if (idx >= SPDIF_MAX_IDX) {
+ printk(KERN_ERR "hda_codec: too many IEC958 outputs\n");
+ return -EBUSY;
+ }
for (dig_mix = dig_mixes; dig_mix->name; dig_mix++) {
kctl = snd_ctl_new1(dig_mix, codec);
+ kctl->id.index = idx;
kctl->private_value = nid;
err = snd_ctl_add(codec->bus->card, kctl);
if (err < 0)
@@ -1512,6 +1576,43 @@ int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid)
}
/*
+ * SPDIF sharing with analog output
+ */
+static int spdif_share_sw_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_multi_out *mout = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.integer.value[0] = mout->share_spdif;
+ return 0;
+}
+
+static int spdif_share_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_multi_out *mout = snd_kcontrol_chip(kcontrol);
+ mout->share_spdif = !!ucontrol->value.integer.value[0];
+ return 0;
+}
+
+static struct snd_kcontrol_new spdif_share_sw = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "IEC958 Default PCM Playback Switch",
+ .info = snd_ctl_boolean_mono_info,
+ .get = spdif_share_sw_get,
+ .put = spdif_share_sw_put,
+};
+
+int snd_hda_create_spdif_share_sw(struct hda_codec *codec,
+ struct hda_multi_out *mout)
+{
+ if (!mout->dig_out_nid)
+ return 0;
+ /* ATTENTION: here mout is passed as private_data, instead of codec */
+ return snd_ctl_add(codec->bus->card,
+ snd_ctl_new1(&spdif_share_sw, mout));
+}
+
+/*
* SPDIF input
*/
@@ -1595,7 +1696,17 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid)
int err;
struct snd_kcontrol *kctl;
struct snd_kcontrol_new *dig_mix;
+ int idx;
+ for (idx = 0; idx < SPDIF_MAX_IDX; idx++) {
+ if (!_snd_hda_find_mixer_ctl(codec, "IEC958 Capture Switch",
+ idx))
+ break;
+ }
+ if (idx >= SPDIF_MAX_IDX) {
+ printk(KERN_ERR "hda_codec: too many IEC958 inputs\n");
+ return -EBUSY;
+ }
for (dig_mix = dig_in_ctls; dig_mix->name; dig_mix++) {
kctl = snd_ctl_new1(dig_mix, codec);
kctl->private_value = nid;
@@ -2106,7 +2217,7 @@ static int hda_pcm_default_cleanup(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
- snd_hda_codec_setup_stream(codec, hinfo->nid, 0, 0, 0);
+ snd_hda_codec_cleanup_stream(codec, hinfo->nid);
return 0;
}
@@ -2491,7 +2602,7 @@ int snd_hda_multi_out_dig_open(struct hda_codec *codec,
mutex_lock(&codec->spdif_mutex);
if (mout->dig_out_used == HDA_DIG_ANALOG_DUP)
/* already opened as analog dup; reset it once */
- snd_hda_codec_setup_stream(codec, mout->dig_out_nid, 0, 0, 0);
+ snd_hda_codec_cleanup_stream(codec, mout->dig_out_nid);
mout->dig_out_used = HDA_DIG_EXCLUSIVE;
mutex_unlock(&codec->spdif_mutex);
return 0;
@@ -2526,9 +2637,36 @@ int snd_hda_multi_out_dig_close(struct hda_codec *codec,
*/
int snd_hda_multi_out_analog_open(struct hda_codec *codec,
struct hda_multi_out *mout,
- struct snd_pcm_substream *substream)
-{
- substream->runtime->hw.channels_max = mout->max_channels;
+ struct snd_pcm_substream *substream,
+ struct hda_pcm_stream *hinfo)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ runtime->hw.channels_max = mout->max_channels;
+ if (mout->dig_out_nid) {
+ if (!mout->analog_rates) {
+ mout->analog_rates = hinfo->rates;
+ mout->analog_formats = hinfo->formats;
+ mout->analog_maxbps = hinfo->maxbps;
+ } else {
+ runtime->hw.rates = mout->analog_rates;
+ runtime->hw.formats = mout->analog_formats;
+ hinfo->maxbps = mout->analog_maxbps;
+ }
+ if (!mout->spdif_rates) {
+ snd_hda_query_supported_pcm(codec, mout->dig_out_nid,
+ &mout->spdif_rates,
+ &mout->spdif_formats,
+ &mout->spdif_maxbps);
+ }
+ mutex_lock(&codec->spdif_mutex);
+ if (mout->share_spdif) {
+ runtime->hw.rates &= mout->spdif_rates;
+ runtime->hw.formats &= mout->spdif_formats;
+ if (mout->spdif_maxbps < hinfo->maxbps)
+ hinfo->maxbps = mout->spdif_maxbps;
+ }
+ mutex_unlock(&codec->spdif_mutex);
+ }
return snd_pcm_hw_constraint_step(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_CHANNELS, 2);
}
@@ -2548,7 +2686,8 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec,
int i;
mutex_lock(&codec->spdif_mutex);
- if (mout->dig_out_nid && mout->dig_out_used != HDA_DIG_EXCLUSIVE) {
+ if (mout->dig_out_nid && mout->share_spdif &&
+ mout->dig_out_used != HDA_DIG_EXCLUSIVE) {
if (chs == 2 &&
snd_hda_is_supported_format(codec, mout->dig_out_nid,
format) &&
@@ -2558,8 +2697,7 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec,
stream_tag, format);
} else {
mout->dig_out_used = 0;
- snd_hda_codec_setup_stream(codec, mout->dig_out_nid,
- 0, 0, 0);
+ snd_hda_codec_cleanup_stream(codec, mout->dig_out_nid);
}
}
mutex_unlock(&codec->spdif_mutex);
@@ -2601,17 +2739,16 @@ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec,
int i;
for (i = 0; i < mout->num_dacs; i++)
- snd_hda_codec_setup_stream(codec, nids[i], 0, 0, 0);
+ snd_hda_codec_cleanup_stream(codec, nids[i]);
if (mout->hp_nid)
- snd_hda_codec_setup_stream(codec, mout->hp_nid, 0, 0, 0);
+ snd_hda_codec_cleanup_stream(codec, mout->hp_nid);
for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++)
if (mout->extra_out_nid[i])
- snd_hda_codec_setup_stream(codec,
- mout->extra_out_nid[i],
- 0, 0, 0);
+ snd_hda_codec_cleanup_stream(codec,
+ mout->extra_out_nid[i]);
mutex_lock(&codec->spdif_mutex);
if (mout->dig_out_nid && mout->dig_out_used == HDA_DIG_ANALOG_DUP) {
- snd_hda_codec_setup_stream(codec, mout->dig_out_nid, 0, 0, 0);
+ snd_hda_codec_cleanup_stream(codec, mout->dig_out_nid);
mout->dig_out_used = 0;
}
mutex_unlock(&codec->spdif_mutex);
@@ -2790,6 +2927,30 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
}
}
+ /* FIX-UP:
+ * If no line-out is defined but multiple HPs are found,
+ * some of them might be the real line-outs.
+ */
+ if (!cfg->line_outs && cfg->hp_outs > 1) {
+ int i = 0;
+ while (i < cfg->hp_outs) {
+ /* The real HPs should have the sequence 0x0f */
+ if ((sequences_hp[i] & 0x0f) == 0x0f) {
+ i++;
+ continue;
+ }
+ /* Move it to the line-out table */
+ cfg->line_out_pins[cfg->line_outs] = cfg->hp_pins[i];
+ sequences_line_out[cfg->line_outs] = sequences_hp[i];
+ cfg->line_outs++;
+ cfg->hp_outs--;
+ memmove(cfg->hp_pins + i, cfg->hp_pins + i + 1,
+ sizeof(cfg->hp_pins[0]) * (cfg->hp_outs - i));
+ memmove(sequences_hp + i - 1, sequences_hp + i,
+ sizeof(sequences_hp[0]) * (cfg->hp_outs - i));
+ }
+ }
+
/* sort by sequence */
sort_pins_by_sequence(cfg->line_out_pins, sequences_line_out,
cfg->line_outs);
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index f14871151be..dcd390b2bba 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -590,11 +590,21 @@ struct hda_pcm_stream {
struct hda_pcm_ops ops;
};
+/* PCM types */
+enum {
+ HDA_PCM_TYPE_AUDIO,
+ HDA_PCM_TYPE_SPDIF,
+ HDA_PCM_TYPE_HDMI,
+ HDA_PCM_TYPE_MODEM,
+ HDA_PCM_NTYPES
+};
+
/* for PCM creation */
struct hda_pcm {
char *name;
struct hda_pcm_stream stream[2];
- unsigned int is_modem; /* modem codec? */
+ unsigned int pcm_type; /* HDA_PCM_TYPE_XXX */
+ int device; /* assigned device number */
};
/* codec information */
@@ -712,6 +722,7 @@ int snd_hda_build_pcms(struct hda_bus *bus);
void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
u32 stream_tag,
int channel_id, int format);
+void snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid);
unsigned int snd_hda_calc_stream_format(unsigned int rate,
unsigned int channels,
unsigned int format,
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index f9de7c467c2..59e4389c94a 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -1007,8 +1007,8 @@ static int generic_pcm2_cleanup(struct hda_pcm_stream *hinfo,
{
struct hda_gspec *spec = codec->spec;
- snd_hda_codec_setup_stream(codec, hinfo->nid, 0, 0, 0);
- snd_hda_codec_setup_stream(codec, spec->dac_node[1]->nid, 0, 0, 0);
+ snd_hda_codec_cleanup_stream(codec, hinfo->nid);
+ snd_hda_codec_cleanup_stream(codec, spec->dac_node[1]->nid);
return 0;
}
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 4be36c84b36..b3a618eb42c 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -39,6 +39,7 @@
#include <linux/interrupt.h>
#include <linux/kernel.h>
#include <linux/module.h>
+#include <linux/dma-mapping.h>
#include <linux/moduleparam.h>
#include <linux/init.h>
#include <linux/slab.h>
@@ -185,35 +186,28 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
/* max number of SDs */
/* ICH, ATI and VIA have 4 playback and 4 capture */
-#define ICH6_CAPTURE_INDEX 0
#define ICH6_NUM_CAPTURE 4
-#define ICH6_PLAYBACK_INDEX 4
#define ICH6_NUM_PLAYBACK 4
/* ULI has 6 playback and 5 capture */
-#define ULI_CAPTURE_INDEX 0
#define ULI_NUM_CAPTURE 5
-#define ULI_PLAYBACK_INDEX 5
#define ULI_NUM_PLAYBACK 6
/* ATI HDMI has 1 playback and 0 capture */
-#define ATIHDMI_CAPTURE_INDEX 0
#define ATIHDMI_NUM_CAPTURE 0
-#define ATIHDMI_PLAYBACK_INDEX 0
#define ATIHDMI_NUM_PLAYBACK 1
/* this number is statically defined for simplicity */
#define MAX_AZX_DEV 16
/* max number of fragments - we may use more if allocating more pages for BDL */
-#define BDL_SIZE PAGE_ALIGN(8192)
-#define AZX_MAX_FRAG (BDL_SIZE / (MAX_AZX_DEV * 16))
+#define BDL_SIZE 4096
+#define AZX_MAX_BDL_ENTRIES (BDL_SIZE / 16)
+#define AZX_MAX_FRAG 32
/* max buffer size - no h/w limit, you can increase as you like */
#define AZX_MAX_BUF_SIZE (1024*1024*1024)
/* max number of PCM devics per card */
-#define AZX_MAX_AUDIO_PCMS 6
-#define AZX_MAX_MODEM_PCMS 2
-#define AZX_MAX_PCMS (AZX_MAX_AUDIO_PCMS + AZX_MAX_MODEM_PCMS)
+#define AZX_MAX_PCMS 8
/* RIRB int mask: overrun[2], response[0] */
#define RIRB_INT_RESPONSE 0x01
@@ -227,6 +221,9 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
/* SD_CTL bits */
#define SD_CTL_STREAM_RESET 0x01 /* stream reset bit */
#define SD_CTL_DMA_START 0x02 /* stream DMA start bit */
+#define SD_CTL_STRIPE (3 << 16) /* stripe control */
+#define SD_CTL_TRAFFIC_PRIO (1 << 18) /* traffic priority */
+#define SD_CTL_DIR (1 << 19) /* bi-directional stream */
#define SD_CTL_STREAM_TAG_MASK (0xf << 20)
#define SD_CTL_STREAM_TAG_SHIFT 20
@@ -284,12 +281,10 @@ enum {
*/
struct azx_dev {
- u32 *bdl; /* virtual address of the BDL */
- dma_addr_t bdl_addr; /* physical address of the BDL */
+ struct snd_dma_buffer bdl; /* BDL buffer */
u32 *posbuf; /* position buffer pointer */
unsigned int bufsize; /* size of the play buffer in bytes */
- unsigned int fragsize; /* size of each period in bytes */
unsigned int frags; /* number for period in the play buffer */
unsigned int fifo_size; /* FIFO size */
@@ -350,7 +345,6 @@ struct azx {
struct azx_dev *azx_dev;
/* PCM */
- unsigned int pcm_devs;
struct snd_pcm *pcm[AZX_MAX_PCMS];
/* HD codec */
@@ -361,8 +355,7 @@ struct azx {
struct azx_rb corb;
struct azx_rb rirb;
- /* BDL, CORB/RIRB and position buffers */
- struct snd_dma_buffer bdl;
+ /* CORB/RIRB and position buffers */
struct snd_dma_buffer rb;
struct snd_dma_buffer posbuf;
@@ -546,8 +539,9 @@ static void azx_update_rirb(struct azx *chip)
if (res_ex & ICH6_RIRB_EX_UNSOL_EV)
snd_hda_queue_unsol_event(chip->bus, res, res_ex);
else if (chip->rirb.cmds) {
- chip->rirb.cmds--;
chip->rirb.res = res;
+ smp_wmb();
+ chip->rirb.cmds--;
}
}
}
@@ -566,8 +560,10 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec)
azx_update_rirb(chip);
spin_unlock_irq(&chip->reg_lock);
}
- if (!chip->rirb.cmds)
+ if (!chip->rirb.cmds) {
+ smp_rmb();
return chip->rirb.res; /* the last value */
+ }
if (time_after(jiffies, timeout))
break;
if (codec->bus->needs_damn_long_delay)
@@ -965,30 +961,57 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id)
/*
* set up BDL entries
*/
-static void azx_setup_periods(struct azx_dev *azx_dev)
+static int azx_setup_periods(struct snd_pcm_substream *substream,
+ struct azx_dev *azx_dev)
{
- u32 *bdl = azx_dev->bdl;
- dma_addr_t dma_addr = azx_dev->substream->runtime->dma_addr;
- int idx;
+ struct snd_sg_buf *sgbuf = snd_pcm_substream_sgbuf(substream);
+ u32 *bdl;
+ int i, ofs, periods, period_bytes;
/* reset BDL address */
azx_sd_writel(azx_dev, SD_BDLPL, 0);
azx_sd_writel(azx_dev, SD_BDLPU, 0);
+ period_bytes = snd_pcm_lib_period_bytes(substream);
+ periods = azx_dev->bufsize / period_bytes;
+
/* program the initial BDL entries */
- for (idx = 0; idx < azx_dev->frags; idx++) {
- unsigned int off = idx << 2; /* 4 dword step */
- dma_addr_t addr = dma_addr + idx * azx_dev->fragsize;
- /* program the address field of the BDL entry */
- bdl[off] = cpu_to_le32((u32)addr);
- bdl[off+1] = cpu_to_le32(upper_32bit(addr));
-
- /* program the size field of the BDL entry */
- bdl[off+2] = cpu_to_le32(azx_dev->fragsize);
-
- /* program the IOC to enable interrupt when buffer completes */
- bdl[off+3] = cpu_to_le32(0x01);
+ bdl = (u32 *)azx_dev->bdl.area;
+ ofs = 0;
+ azx_dev->frags = 0;
+ for (i = 0; i < periods; i++) {
+ int size, rest;
+ if (i >= AZX_MAX_BDL_ENTRIES) {
+ snd_printk(KERN_ERR "Too many BDL entries: "
+ "buffer=%d, period=%d\n",
+ azx_dev->bufsize, period_bytes);
+ /* reset */
+ azx_sd_writel(azx_dev, SD_BDLPL, 0);
+ azx_sd_writel(azx_dev, SD_BDLPU, 0);
+ return -EINVAL;
+ }
+ rest = period_bytes;
+ do {
+ dma_addr_t addr = snd_pcm_sgbuf_get_addr(sgbuf, ofs);
+ /* program the address field of the BDL entry */
+ bdl[0] = cpu_to_le32((u32)addr);
+ bdl[1] = cpu_to_le32(upper_32bit(addr));
+ /* program the size field of the BDL entry */
+ size = PAGE_SIZE - (ofs % PAGE_SIZE);
+ if (rest < size)
+ size = rest;
+ bdl[2] = cpu_to_le32(size);
+ /* program the IOC to enable interrupt
+ * only when the whole fragment is processed
+ */
+ rest -= size;
+ bdl[3] = rest ? 0 : cpu_to_le32(0x01);
+ bdl += 4;
+ azx_dev->frags++;
+ ofs += size;
+ } while (rest > 0);
}
+ return 0;
}
/*
@@ -1037,14 +1060,17 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev)
/* program the BDL address */
/* lower BDL address */
- azx_sd_writel(azx_dev, SD_BDLPL, (u32)azx_dev->bdl_addr);
+ azx_sd_writel(azx_dev, SD_BDLPL, (u32)azx_dev->bdl.addr);
/* upper BDL address */
- azx_sd_writel(azx_dev, SD_BDLPU, upper_32bit(azx_dev->bdl_addr));
+ azx_sd_writel(azx_dev, SD_BDLPU, upper_32bit(azx_dev->bdl.addr));
/* enable the position buffer */
- if (!(azx_readl(chip, DPLBASE) & ICH6_DPLBASE_ENABLE))
- azx_writel(chip, DPLBASE,
- (u32)chip->posbuf.addr |ICH6_DPLBASE_ENABLE);
+ if (chip->position_fix == POS_FIX_POSBUF ||
+ chip->position_fix == POS_FIX_AUTO) {
+ if (!(azx_readl(chip, DPLBASE) & ICH6_DPLBASE_ENABLE))
+ azx_writel(chip, DPLBASE,
+ (u32)chip->posbuf.addr | ICH6_DPLBASE_ENABLE);
+ }
/* set the interrupt enable bits in the descriptor control register */
azx_sd_writel(azx_dev, SD_CTL,
@@ -1157,7 +1183,8 @@ static struct snd_pcm_hardware azx_pcm_hw = {
SNDRV_PCM_INFO_MMAP_VALID |
/* No full-resume yet implemented */
/* SNDRV_PCM_INFO_RESUME |*/
- SNDRV_PCM_INFO_PAUSE),
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_SYNC_START),
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.rates = SNDRV_PCM_RATE_48000,
.rate_min = 48000,
@@ -1219,6 +1246,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
spin_unlock_irqrestore(&chip->reg_lock, flags);
runtime->private_data = azx_dev;
+ snd_pcm_set_sync(substream);
mutex_unlock(&chip->open_mutex);
return 0;
}
@@ -1275,8 +1303,6 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream)
struct snd_pcm_runtime *runtime = substream->runtime;
azx_dev->bufsize = snd_pcm_lib_buffer_bytes(substream);
- azx_dev->fragsize = snd_pcm_lib_period_bytes(substream);
- azx_dev->frags = azx_dev->bufsize / azx_dev->fragsize;
azx_dev->format_val = snd_hda_calc_stream_format(runtime->rate,
runtime->channels,
runtime->format,
@@ -1288,10 +1314,10 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream)
return -EINVAL;
}
- snd_printdd("azx_pcm_prepare: bufsize=0x%x, fragsize=0x%x, "
- "format=0x%x\n",
- azx_dev->bufsize, azx_dev->fragsize, azx_dev->format_val);
- azx_setup_periods(azx_dev);
+ snd_printdd("azx_pcm_prepare: bufsize=0x%x, format=0x%x\n",
+ azx_dev->bufsize, azx_dev->format_val);
+ if (azx_setup_periods(substream, azx_dev) < 0)
+ return -EINVAL;
azx_setup_controller(chip, azx_dev);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
azx_dev->fifo_size = azx_sd_readw(azx_dev, SD_FIFOSIZE) + 1;
@@ -1305,37 +1331,94 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream)
static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
{
struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
- struct azx_dev *azx_dev = get_azx_dev(substream);
struct azx *chip = apcm->chip;
- int err = 0;
+ struct azx_dev *azx_dev;
+ struct snd_pcm_substream *s;
+ int start, nsync = 0, sbits = 0;
+ int nwait, timeout;
- spin_lock(&chip->reg_lock);
switch (cmd) {
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_START:
- azx_stream_start(chip, azx_dev);
- azx_dev->running = 1;
+ start = 1;
break;
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_STOP:
- azx_stream_stop(chip, azx_dev);
- azx_dev->running = 0;
+ start = 0;
break;
default:
- err = -EINVAL;
+ return -EINVAL;
+ }
+
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (s->pcm->card != substream->pcm->card)
+ continue;
+ azx_dev = get_azx_dev(s);
+ sbits |= 1 << azx_dev->index;
+ nsync++;
+ snd_pcm_trigger_done(s, substream);
+ }
+
+ spin_lock(&chip->reg_lock);
+ if (nsync > 1) {
+ /* first, set SYNC bits of corresponding streams */
+ azx_writel(chip, SYNC, azx_readl(chip, SYNC) | sbits);
+ }
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (s->pcm->card != substream->pcm->card)
+ continue;
+ azx_dev = get_azx_dev(s);
+ if (start)
+ azx_stream_start(chip, azx_dev);
+ else
+ azx_stream_stop(chip, azx_dev);
+ azx_dev->running = start;
}
spin_unlock(&chip->reg_lock);
- if (cmd == SNDRV_PCM_TRIGGER_PAUSE_PUSH ||
- cmd == SNDRV_PCM_TRIGGER_SUSPEND ||
- cmd == SNDRV_PCM_TRIGGER_STOP) {
- int timeout = 5000;
- while ((azx_sd_readb(azx_dev, SD_CTL) & SD_CTL_DMA_START) &&
- --timeout)
- ;
+ if (start) {
+ if (nsync == 1)
+ return 0;
+ /* wait until all FIFOs get ready */
+ for (timeout = 5000; timeout; timeout--) {
+ nwait = 0;
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (s->pcm->card != substream->pcm->card)
+ continue;
+ azx_dev = get_azx_dev(s);
+ if (!(azx_sd_readb(azx_dev, SD_STS) &
+ SD_STS_FIFO_READY))
+ nwait++;
+ }
+ if (!nwait)
+ break;
+ cpu_relax();
+ }
+ } else {
+ /* wait until all RUN bits are cleared */
+ for (timeout = 5000; timeout; timeout--) {
+ nwait = 0;
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (s->pcm->card != substream->pcm->card)
+ continue;
+ azx_dev = get_azx_dev(s);
+ if (azx_sd_readb(azx_dev, SD_CTL) &
+ SD_CTL_DMA_START)
+ nwait++;
+ }
+ if (!nwait)
+ break;
+ cpu_relax();
+ }
}
- return err;
+ if (nsync > 1) {
+ spin_lock(&chip->reg_lock);
+ /* reset SYNC bits */
+ azx_writel(chip, SYNC, azx_readl(chip, SYNC) & ~sbits);
+ spin_unlock(&chip->reg_lock);
+ }
+ return 0;
}
static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream)
@@ -1378,6 +1461,7 @@ static struct snd_pcm_ops azx_pcm_ops = {
.prepare = azx_pcm_prepare,
.trigger = azx_pcm_trigger,
.pointer = azx_pcm_pointer,
+ .page = snd_pcm_sgbuf_ops_page,
};
static void azx_pcm_free(struct snd_pcm *pcm)
@@ -1386,7 +1470,7 @@ static void azx_pcm_free(struct snd_pcm *pcm)
}
static int __devinit create_codec_pcm(struct azx *chip, struct hda_codec *codec,
- struct hda_pcm *cpcm, int pcm_dev)
+ struct hda_pcm *cpcm)
{
int err;
struct snd_pcm *pcm;
@@ -1400,7 +1484,7 @@ static int __devinit create_codec_pcm(struct azx *chip, struct hda_codec *codec,
snd_assert(cpcm->name, return -EINVAL);
- err = snd_pcm_new(chip->card, cpcm->name, pcm_dev,
+ err = snd_pcm_new(chip->card, cpcm->name, cpcm->device,
cpcm->stream[0].substreams,
cpcm->stream[1].substreams,
&pcm);
@@ -1420,62 +1504,70 @@ static int __devinit create_codec_pcm(struct azx *chip, struct hda_codec *codec,
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &azx_pcm_ops);
if (cpcm->stream[1].substreams)
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &azx_pcm_ops);
- snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG,
snd_dma_pci_data(chip->pci),
1024 * 64, 1024 * 1024);
- chip->pcm[pcm_dev] = pcm;
- if (chip->pcm_devs < pcm_dev + 1)
- chip->pcm_devs = pcm_dev + 1;
-
+ chip->pcm[cpcm->device] = pcm;
return 0;
}
static int __devinit azx_pcm_create(struct azx *chip)
{
+ static const char *dev_name[HDA_PCM_NTYPES] = {
+ "Audio", "SPDIF", "HDMI", "Modem"
+ };
+ /* starting device index for each PCM type */
+ static int dev_idx[HDA_PCM_NTYPES] = {
+ [HDA_PCM_TYPE_AUDIO] = 0,
+ [HDA_PCM_TYPE_SPDIF] = 1,
+ [HDA_PCM_TYPE_HDMI] = 3,
+ [HDA_PCM_TYPE_MODEM] = 6
+ };
+ /* normal audio device indices; not linear to keep compatibility */
+ static int audio_idx[4] = { 0, 2, 4, 5 };
struct hda_codec *codec;
int c, err;
- int pcm_dev;
+ int num_devs[HDA_PCM_NTYPES];
err = snd_hda_build_pcms(chip->bus);
if (err < 0)
return err;
/* create audio PCMs */
- pcm_dev = 0;
- list_for_each_entry(codec, &chip->bus->codec_list, list) {
- for (c = 0; c < codec->num_pcms; c++) {
- if (codec->pcm_info[c].is_modem)
- continue; /* create later */
- if (pcm_dev >= AZX_MAX_AUDIO_PCMS) {
- snd_printk(KERN_ERR SFX
- "Too many audio PCMs\n");
- return -EINVAL;
- }
- err = create_codec_pcm(chip, codec,
- &codec->pcm_info[c], pcm_dev);
- if (err < 0)
- return err;
- pcm_dev++;
- }
- }
-
- /* create modem PCMs */
- pcm_dev = AZX_MAX_AUDIO_PCMS;
+ memset(num_devs, 0, sizeof(num_devs));
list_for_each_entry(codec, &chip->bus->codec_list, list) {
for (c = 0; c < codec->num_pcms; c++) {
- if (!codec->pcm_info[c].is_modem)
- continue; /* already created */
- if (pcm_dev >= AZX_MAX_PCMS) {
- snd_printk(KERN_ERR SFX
- "Too many modem PCMs\n");
- return -EINVAL;
+ struct hda_pcm *cpcm = &codec->pcm_info[c];
+ int type = cpcm->pcm_type;
+ switch (type) {
+ case HDA_PCM_TYPE_AUDIO:
+ if (num_devs[type] >= ARRAY_SIZE(audio_idx)) {
+ snd_printk(KERN_WARNING
+ "Too many audio devices\n");
+ continue;
+ }
+ cpcm->device = audio_idx[num_devs[type]];
+ break;
+ case HDA_PCM_TYPE_SPDIF:
+ case HDA_PCM_TYPE_HDMI:
+ case HDA_PCM_TYPE_MODEM:
+ if (num_devs[type]) {
+ snd_printk(KERN_WARNING
+ "%s already defined\n",
+ dev_name[type]);
+ continue;
+ }
+ cpcm->device = dev_idx[type];
+ break;
+ default:
+ snd_printk(KERN_WARNING
+ "Invalid PCM type %d\n", type);
+ continue;
}
- err = create_codec_pcm(chip, codec,
- &codec->pcm_info[c], pcm_dev);
+ num_devs[type]++;
+ err = create_codec_pcm(chip, codec, cpcm);
if (err < 0)
return err;
- chip->pcm[pcm_dev]->dev_class = SNDRV_PCM_CLASS_MODEM;
- pcm_dev++;
}
}
return 0;
@@ -1502,10 +1594,7 @@ static int __devinit azx_init_stream(struct azx *chip)
* and initialize
*/
for (i = 0; i < chip->num_streams; i++) {
- unsigned int off = sizeof(u32) * (i * AZX_MAX_FRAG * 4);
struct azx_dev *azx_dev = &chip->azx_dev[i];
- azx_dev->bdl = (u32 *)(chip->bdl.area + off);
- azx_dev->bdl_addr = chip->bdl.addr + off;
azx_dev->posbuf = (u32 __iomem *)(chip->posbuf.area + i * 8);
/* offset: SDI0=0x80, SDI1=0xa0, ... SDO3=0x160 */
azx_dev->sd_addr = chip->remap_addr + (0x20 * i + 0x80);
@@ -1587,13 +1676,12 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state)
int i;
snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
- for (i = 0; i < chip->pcm_devs; i++)
+ for (i = 0; i < AZX_MAX_PCMS; i++)
snd_pcm_suspend_all(chip->pcm[i]);
if (chip->initialized)
snd_hda_suspend(chip->bus, state);
azx_stop_chip(chip);
if (chip->irq >= 0) {
- synchronize_irq(chip->irq);
free_irq(chip->irq, chip);
chip->irq = -1;
}
@@ -1641,24 +1729,26 @@ static int azx_resume(struct pci_dev *pci)
*/
static int azx_free(struct azx *chip)
{
+ int i;
+
if (chip->initialized) {
- int i;
for (i = 0; i < chip->num_streams; i++)
azx_stream_stop(chip, &chip->azx_dev[i]);
azx_stop_chip(chip);
}
- if (chip->irq >= 0) {
- synchronize_irq(chip->irq);
+ if (chip->irq >= 0)
free_irq(chip->irq, (void*)chip);
- }
if (chip->msi)
pci_disable_msi(chip->pci);
if (chip->remap_addr)
iounmap(chip->remap_addr);
- if (chip->bdl.area)
- snd_dma_free_pages(&chip->bdl);
+ if (chip->azx_dev) {
+ for (i = 0; i < chip->num_streams; i++)
+ if (chip->azx_dev[i].bdl.area)
+ snd_dma_free_pages(&chip->azx_dev[i].bdl);
+ }
if (chip->rb.area)
snd_dma_free_pages(&chip->rb);
if (chip->posbuf.area)
@@ -1682,6 +1772,7 @@ static int azx_dev_free(struct snd_device *device)
static struct snd_pci_quirk position_fix_list[] __devinitdata = {
SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_NONE),
SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_NONE),
+ SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_NONE),
{}
};
@@ -1740,7 +1831,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
struct azx **rchip)
{
struct azx *chip;
- int err;
+ int i, err;
unsigned short gcap;
static struct snd_device_ops ops = {
.dev_free = azx_dev_free,
@@ -1812,38 +1903,35 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
gcap = azx_readw(chip, GCAP);
snd_printdd("chipset global capabilities = 0x%x\n", gcap);
- if (gcap) {
- /* read number of streams from GCAP register instead of using
- * hardcoded value
- */
- chip->playback_streams = (gcap & (0xF << 12)) >> 12;
- chip->capture_streams = (gcap & (0xF << 8)) >> 8;
- chip->playback_index_offset = chip->capture_streams;
- chip->capture_index_offset = 0;
- } else {
+ /* allow 64bit DMA address if supported by H/W */
+ if ((gcap & 0x01) && !pci_set_dma_mask(pci, DMA_64BIT_MASK))
+ pci_set_consistent_dma_mask(pci, DMA_64BIT_MASK);
+
+ /* read number of streams from GCAP register instead of using
+ * hardcoded value
+ */
+ chip->capture_streams = (gcap >> 8) & 0x0f;
+ chip->playback_streams = (gcap >> 12) & 0x0f;
+ if (!chip->playback_streams && !chip->capture_streams) {
/* gcap didn't give any info, switching to old method */
switch (chip->driver_type) {
case AZX_DRIVER_ULI:
chip->playback_streams = ULI_NUM_PLAYBACK;
chip->capture_streams = ULI_NUM_CAPTURE;
- chip->playback_index_offset = ULI_PLAYBACK_INDEX;
- chip->capture_index_offset = ULI_CAPTURE_INDEX;
break;
case AZX_DRIVER_ATIHDMI:
chip->playback_streams = ATIHDMI_NUM_PLAYBACK;
chip->capture_streams = ATIHDMI_NUM_CAPTURE;
- chip->playback_index_offset = ATIHDMI_PLAYBACK_INDEX;
- chip->capture_index_offset = ATIHDMI_CAPTURE_INDEX;
break;
default:
chip->playback_streams = ICH6_NUM_PLAYBACK;
chip->capture_streams = ICH6_NUM_CAPTURE;
- chip->playback_index_offset = ICH6_PLAYBACK_INDEX;
- chip->capture_index_offset = ICH6_CAPTURE_INDEX;
break;
}
}
+ chip->capture_index_offset = 0;
+ chip->playback_index_offset = chip->capture_streams;
chip->num_streams = chip->playback_streams + chip->capture_streams;
chip->azx_dev = kcalloc(chip->num_streams, sizeof(*chip->azx_dev),
GFP_KERNEL);
@@ -1852,13 +1940,15 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
goto errout;
}
- /* allocate memory for the BDL for each stream */
- err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV,
- snd_dma_pci_data(chip->pci),
- BDL_SIZE, &chip->bdl);
- if (err < 0) {
- snd_printk(KERN_ERR SFX "cannot allocate BDL\n");
- goto errout;
+ for (i = 0; i < chip->num_streams; i++) {
+ /* allocate memory for the BDL for each stream */
+ err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci),
+ BDL_SIZE, &chip->azx_dev[i].bdl);
+ if (err < 0) {
+ snd_printk(KERN_ERR SFX "cannot allocate BDL\n");
+ goto errout;
+ }
}
/* allocate memory for the position buffer */
err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV,
@@ -1994,48 +2084,63 @@ static void __devexit azx_remove(struct pci_dev *pci)
/* PCI IDs */
static struct pci_device_id azx_ids[] = {
- { 0x8086, 0x2668, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH6 */
- { 0x8086, 0x27d8, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH7 */
- { 0x8086, 0x269a, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ESB2 */
- { 0x8086, 0x284b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH8 */
- { 0x8086, 0x293e, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH9 */
- { 0x8086, 0x293f, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH9 */
- { 0x8086, 0x3a3e, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH10 */
- { 0x8086, 0x3a6e, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH10 */
- { 0x8086, 0x811b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_SCH }, /* SCH*/
- { 0x1002, 0x437b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATI }, /* ATI SB450 */
- { 0x1002, 0x4383, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATI }, /* ATI SB600 */
- { 0x1002, 0x793b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RS600 HDMI */
- { 0x1002, 0x7919, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RS690 HDMI */
- { 0x1002, 0x960f, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RS780 HDMI */
- { 0x1002, 0xaa00, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI R600 HDMI */
- { 0x1002, 0xaa08, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV630 HDMI */
- { 0x1002, 0xaa10, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV610 HDMI */
- { 0x1002, 0xaa18, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV670 HDMI */
- { 0x1002, 0xaa20, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV635 HDMI */
- { 0x1002, 0xaa28, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV620 HDMI */
- { 0x1002, 0xaa30, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV770 HDMI */
- { 0x1106, 0x3288, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_VIA }, /* VIA VT8251/VT8237A */
- { 0x1039, 0x7502, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_SIS }, /* SIS966 */
- { 0x10b9, 0x5461, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ULI }, /* ULI M5461 */
- { 0x10de, 0x026c, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP51 */
- { 0x10de, 0x0371, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP55 */
- { 0x10de, 0x03e4, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP61 */
- { 0x10de, 0x03f0, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP61 */
- { 0x10de, 0x044a, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP65 */
- { 0x10de, 0x044b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP65 */
- { 0x10de, 0x055c, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP67 */
- { 0x10de, 0x055d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP67 */
- { 0x10de, 0x07fc, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP73 */
- { 0x10de, 0x07fd, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP73 */
- { 0x10de, 0x0774, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */
- { 0x10de, 0x0775, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */
- { 0x10de, 0x0776, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */
- { 0x10de, 0x0777, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */
- { 0x10de, 0x0ac0, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP79 */
- { 0x10de, 0x0ac1, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP79 */
- { 0x10de, 0x0ac2, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP79 */
- { 0x10de, 0x0ac3, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP79 */
+ /* ICH 6..10 */
+ { PCI_DEVICE(0x8086, 0x2668), .driver_data = AZX_DRIVER_ICH },
+ { PCI_DEVICE(0x8086, 0x27d8), .driver_data = AZX_DRIVER_ICH },
+ { PCI_DEVICE(0x8086, 0x269a), .driver_data = AZX_DRIVER_ICH },
+ { PCI_DEVICE(0x8086, 0x284b), .driver_data = AZX_DRIVER_ICH },
+ { PCI_DEVICE(0x8086, 0x293e), .driver_data = AZX_DRIVER_ICH },
+ { PCI_DEVICE(0x8086, 0x293f), .driver_data = AZX_DRIVER_ICH },
+ { PCI_DEVICE(0x8086, 0x3a3e), .driver_data = AZX_DRIVER_ICH },
+ { PCI_DEVICE(0x8086, 0x3a6e), .driver_data = AZX_DRIVER_ICH },
+ /* SCH */
+ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH },
+ /* ATI SB 450/600 */
+ { PCI_DEVICE(0x1002, 0x437b), .driver_data = AZX_DRIVER_ATI },
+ { PCI_DEVICE(0x1002, 0x4383), .driver_data = AZX_DRIVER_ATI },
+ /* ATI HDMI */
+ { PCI_DEVICE(0x1002, 0x793b), .driver_data = AZX_DRIVER_ATIHDMI },
+ { PCI_DEVICE(0x1002, 0x7919), .driver_data = AZX_DRIVER_ATIHDMI },
+ { PCI_DEVICE(0x1002, 0x960f), .driver_data = AZX_DRIVER_ATIHDMI },
+ { PCI_DEVICE(0x1002, 0xaa00), .driver_data = AZX_DRIVER_ATIHDMI },
+ { PCI_DEVICE(0x1002, 0xaa08), .driver_data = AZX_DRIVER_ATIHDMI },
+ { PCI_DEVICE(0x1002, 0xaa10), .driver_data = AZX_DRIVER_ATIHDMI },
+ { PCI_DEVICE(0x1002, 0xaa18), .driver_data = AZX_DRIVER_ATIHDMI },
+ { PCI_DEVICE(0x1002, 0xaa20), .driver_data = AZX_DRIVER_ATIHDMI },
+ { PCI_DEVICE(0x1002, 0xaa28), .driver_data = AZX_DRIVER_ATIHDMI },
+ { PCI_DEVICE(0x1002, 0xaa30), .driver_data = AZX_DRIVER_ATIHDMI },
+ { PCI_DEVICE(0x1002, 0xaa38), .driver_data = AZX_DRIVER_ATIHDMI },
+ { PCI_DEVICE(0x1002, 0xaa40), .driver_data = AZX_DRIVER_ATIHDMI },
+ { PCI_DEVICE(0x1002, 0xaa48), .driver_data = AZX_DRIVER_ATIHDMI },
+ /* VIA VT8251/VT8237A */
+ { PCI_DEVICE(0x1106, 0x3288), .driver_data = AZX_DRIVER_VIA },
+ /* SIS966 */
+ { PCI_DEVICE(0x1039, 0x7502), .driver_data = AZX_DRIVER_SIS },
+ /* ULI M5461 */
+ { PCI_DEVICE(0x10b9, 0x5461), .driver_data = AZX_DRIVER_ULI },
+ /* NVIDIA MCP */
+ { PCI_DEVICE(0x10de, 0x026c), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0371), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x03e4), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x03f0), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x044a), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x044b), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x055c), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x055d), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0774), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0775), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0776), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0777), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x07fc), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x07fd), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0ac0), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0ac1), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0ac2), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0ac3), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0bd4), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0bd5), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0bd6), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0bd7), .driver_data = AZX_DRIVER_NVIDIA },
{ 0, }
};
MODULE_DEVICE_TABLE(pci, azx_ids);
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index ad0014ab71f..5c9e578f7f2 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -228,8 +228,18 @@ struct hda_multi_out {
int max_channels; /* currently supported analog channels */
int dig_out_used; /* current usage of digital out (HDA_DIG_XXX) */
int no_share_stream; /* don't share a stream with multiple pins */
+ int share_spdif; /* share SPDIF pin */
+ /* PCM information for both analog and SPDIF DACs */
+ unsigned int analog_rates;
+ unsigned int analog_maxbps;
+ u64 analog_formats;
+ unsigned int spdif_rates;
+ unsigned int spdif_maxbps;
+ u64 spdif_formats;
};
+int snd_hda_create_spdif_share_sw(struct hda_codec *codec,
+ struct hda_multi_out *mout);
int snd_hda_multi_out_dig_open(struct hda_codec *codec,
struct hda_multi_out *mout);
int snd_hda_multi_out_dig_close(struct hda_codec *codec,
@@ -241,7 +251,8 @@ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec,
struct snd_pcm_substream *substream);
int snd_hda_multi_out_analog_open(struct hda_codec *codec,
struct hda_multi_out *mout,
- struct snd_pcm_substream *substream);
+ struct snd_pcm_substream *substream,
+ struct hda_pcm_stream *hinfo);
int snd_hda_multi_out_analog_prepare(struct hda_codec *codec,
struct hda_multi_out *mout,
unsigned int stream_tag,
@@ -407,11 +418,4 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec,
hda_nid_t nid);
#endif /* CONFIG_SND_HDA_POWER_SAVE */
-/*
- * virtual master control
- */
-struct snd_kcontrol *snd_ctl_make_virtual_master(char *name,
- const unsigned int *tlv);
-int snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave);
-
#endif /* __SOUND_HDA_LOCAL_H */
diff --git a/sound/pci/hda/hda_patch.h b/sound/pci/hda/hda_patch.h
index f5c23bb16d7..2fdf2358dbc 100644
--- a/sound/pci/hda/hda_patch.h
+++ b/sound/pci/hda/hda_patch.h
@@ -18,31 +18,3 @@ extern struct hda_codec_preset snd_hda_preset_atihdmi[];
extern struct hda_codec_preset snd_hda_preset_conexant[];
/* VIA codecs */
extern struct hda_codec_preset snd_hda_preset_via[];
-
-static const struct hda_codec_preset *hda_preset_tables[] = {
-#ifdef CONFIG_SND_HDA_CODEC_REALTEK
- snd_hda_preset_realtek,
-#endif
-#ifdef CONFIG_SND_HDA_CODEC_CMEDIA
- snd_hda_preset_cmedia,
-#endif
-#ifdef CONFIG_SND_HDA_CODEC_ANALOG
- snd_hda_preset_analog,
-#endif
-#ifdef CONFIG_SND_HDA_CODEC_SIGMATEL
- snd_hda_preset_sigmatel,
-#endif
-#ifdef CONFIG_SND_HDA_CODEC_SI3054
- snd_hda_preset_si3054,
-#endif
-#ifdef CONFIG_SND_HDA_CODEC_ATIHDMI
- snd_hda_preset_atihdmi,
-#endif
-#ifdef CONFIG_SND_HDA_CODEC_CONEXANT
- snd_hda_preset_conexant,
-#endif
-#ifdef CONFIG_SND_HDA_CODEC_VIA
- snd_hda_preset_via,
-#endif
- NULL
-};
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index c8649282c2c..e0a605adde4 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -28,6 +28,7 @@
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"
+#include "hda_patch.h"
struct ad198x_spec {
struct snd_kcontrol_new *mixers[5];
@@ -80,7 +81,6 @@ struct ad198x_spec {
#endif
/* for virtual master */
hda_nid_t vmaster_nid;
- u32 vmaster_tlv[4];
const char **slave_vols;
const char **slave_sws;
};
@@ -171,6 +171,11 @@ static int ad198x_build_controls(struct hda_codec *codec)
err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
if (err < 0)
return err;
+ err = snd_hda_create_spdif_share_sw(codec,
+ &spec->multiout);
+ if (err < 0)
+ return err;
+ spec->multiout.share_spdif = 1;
}
if (spec->dig_in_nid) {
err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
@@ -180,10 +185,11 @@ static int ad198x_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->vmaster_nid,
- HDA_OUTPUT, spec->vmaster_tlv);
+ HDA_OUTPUT, vmaster_tlv);
err = snd_hda_add_vmaster(codec, "Master Playback Volume",
- spec->vmaster_tlv,
+ vmaster_tlv,
(spec->slave_vols ?
spec->slave_vols : ad_slave_vols));
if (err < 0)
@@ -217,7 +223,8 @@ static int ad198x_playback_pcm_open(struct hda_pcm_stream *hinfo,
struct snd_pcm_substream *substream)
{
struct ad198x_spec *spec = codec->spec;
- return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream);
+ return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
+ hinfo);
}
static int ad198x_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
@@ -289,8 +296,7 @@ static int ad198x_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
struct snd_pcm_substream *substream)
{
struct ad198x_spec *spec = codec->spec;
- snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number],
- 0, 0, 0);
+ snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]);
return 0;
}
@@ -359,6 +365,7 @@ static int ad198x_build_pcms(struct hda_codec *codec)
info++;
codec->num_pcms++;
info->name = "AD198x Digital";
+ info->pcm_type = HDA_PCM_TYPE_SPDIF;
info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad198x_pcm_digital_playback;
info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid;
if (spec->dig_in_nid) {
@@ -611,13 +618,19 @@ static struct hda_input_mux ad1986a_laptop_eapd_capture_source = {
},
};
+static struct hda_input_mux ad1986a_automic_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic", 0x0 },
+ { "Mix", 0x5 },
+ },
+};
+
static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = {
HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol),
HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw),
HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT),
@@ -641,6 +654,33 @@ static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = {
{ } /* end */
};
+/* re-connect the mic boost input according to the jack sensing */
+static void ad1986a_automic(struct hda_codec *codec)
+{
+ unsigned int present;
+ present = snd_hda_codec_read(codec, 0x1f, 0, AC_VERB_GET_PIN_SENSE, 0);
+ /* 0 = 0x1f, 2 = 0x1d, 4 = mixed */
+ snd_hda_codec_write(codec, 0x0f, 0, AC_VERB_SET_CONNECT_SEL,
+ (present & AC_PINSENSE_PRESENCE) ? 0 : 2);
+}
+
+#define AD1986A_MIC_EVENT 0x36
+
+static void ad1986a_automic_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) != AD1986A_MIC_EVENT)
+ return;
+ ad1986a_automic(codec);
+}
+
+static int ad1986a_automic_init(struct hda_codec *codec)
+{
+ ad198x_init(codec);
+ ad1986a_automic(codec);
+ return 0;
+}
+
/* laptop-automute - 2ch only */
static void ad1986a_update_hp(struct hda_codec *codec)
@@ -844,6 +884,15 @@ static struct hda_verb ad1986a_eapd_init_verbs[] = {
{}
};
+static struct hda_verb ad1986a_automic_verbs[] = {
+ {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ /*{0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},*/
+ {0x0f, AC_VERB_SET_CONNECT_SEL, 0x0},
+ {0x1f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1986A_MIC_EVENT},
+ {}
+};
+
/* Ultra initialization */
static struct hda_verb ad1986a_ultra_init[] = {
/* eapd initialization */
@@ -986,14 +1035,17 @@ static int patch_ad1986a(struct hda_codec *codec)
break;
case AD1986A_LAPTOP_EAPD:
spec->mixers[0] = ad1986a_laptop_eapd_mixers;
- spec->num_init_verbs = 2;
+ spec->num_init_verbs = 3;
spec->init_verbs[1] = ad1986a_eapd_init_verbs;
+ spec->init_verbs[2] = ad1986a_automic_verbs;
spec->multiout.max_channels = 2;
spec->multiout.num_dacs = 1;
spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
if (!is_jack_available(codec, 0x25))
spec->multiout.dig_out_nid = 0;
- spec->input_mux = &ad1986a_laptop_eapd_capture_source;
+ spec->input_mux = &ad1986a_automic_capture_source;
+ codec->patch_ops.unsol_event = ad1986a_automic_unsol_event;
+ codec->patch_ops.init = ad1986a_automic_init;
break;
case AD1986A_LAPTOP_AUTOMUTE:
spec->mixers[0] = ad1986a_laptop_automute_mixers;
@@ -1365,7 +1417,10 @@ static int ad1981_hp_master_sw_put(struct snd_kcontrol *kcontrol,
if (! ad198x_eapd_put(kcontrol, ucontrol))
return 0;
-
+ /* change speaker pin appropriately */
+ snd_hda_codec_write(codec, 0x05, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ spec->cur_eapd ? PIN_OUT : 0);
/* toggle HP mute appropriately */
snd_hda_codec_amp_stereo(codec, 0x06, HDA_OUTPUT, 0,
HDA_AMP_MUTE,
@@ -2087,6 +2142,10 @@ static struct snd_kcontrol_new ad1988_spdif_in_mixers[] = {
{ } /* end */
};
+static struct snd_kcontrol_new ad1989_spdif_out_mixers[] = {
+ HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
/*
* initialization verbs
@@ -2187,6 +2246,13 @@ static struct hda_verb ad1988_spdif_init_verbs[] = {
{ }
};
+/* AD1989 has no ADC -> SPDIF route */
+static struct hda_verb ad1989_spdif_init_verbs[] = {
+ /* SPDIF out pin */
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
+ { }
+};
+
/*
* verbs for 3stack (+dig)
*/
@@ -2894,10 +2960,19 @@ static int patch_ad1988(struct hda_codec *codec)
spec->mixers[spec->num_mixers++] = ad1988_capture_mixers;
spec->init_verbs[spec->num_init_verbs++] = ad1988_capture_init_verbs;
if (spec->multiout.dig_out_nid) {
- spec->mixers[spec->num_mixers++] = ad1988_spdif_out_mixers;
- spec->init_verbs[spec->num_init_verbs++] = ad1988_spdif_init_verbs;
+ if (codec->vendor_id >= 0x11d4989a) {
+ spec->mixers[spec->num_mixers++] =
+ ad1989_spdif_out_mixers;
+ spec->init_verbs[spec->num_init_verbs++] =
+ ad1989_spdif_init_verbs;
+ } else {
+ spec->mixers[spec->num_mixers++] =
+ ad1988_spdif_out_mixers;
+ spec->init_verbs[spec->num_init_verbs++] =
+ ad1988_spdif_init_verbs;
+ }
}
- if (spec->dig_in_nid)
+ if (spec->dig_in_nid && codec->vendor_id < 0x11d4989a)
spec->mixers[spec->num_mixers++] = ad1988_spdif_in_mixers;
codec->patch_ops = ad198x_patch_ops;
@@ -3133,11 +3208,12 @@ static int patch_ad1884(struct hda_codec *codec)
* Lenovo Thinkpad T61/X61
*/
static struct hda_input_mux ad1984_thinkpad_capture_source = {
- .num_items = 3,
+ .num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Internal Mic", 0x1 },
{ "Mix", 0x3 },
+ { "Docking-Station", 0x4 },
},
};
@@ -3268,8 +3344,7 @@ static int ad1984_pcm_dmic_cleanup(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
- snd_hda_codec_setup_stream(codec, 0x05 + substream->number,
- 0, 0, 0);
+ snd_hda_codec_cleanup_stream(codec, 0x05 + substream->number);
return 0;
}
@@ -3356,6 +3431,472 @@ static int patch_ad1984(struct hda_codec *codec)
/*
+ * AD1883 / AD1884A / AD1984A / AD1984B
+ *
+ * port-B (0x14) - front mic-in
+ * port-E (0x1c) - rear mic-in
+ * port-F (0x16) - CD / ext out
+ * port-C (0x15) - rear line-in
+ * port-D (0x12) - rear line-out
+ * port-A (0x11) - front hp-out
+ *
+ * AD1984A = AD1884A + digital-mic
+ * AD1883 = equivalent with AD1984A
+ * AD1984B = AD1984A + extra SPDIF-out
+ *
+ * FIXME:
+ * We share the single DAC for both HP and line-outs (see AD1884/1984).
+ */
+
+static hda_nid_t ad1884a_dac_nids[1] = {
+ 0x03,
+};
+
+#define ad1884a_adc_nids ad1884_adc_nids
+#define ad1884a_capsrc_nids ad1884_capsrc_nids
+
+#define AD1884A_SPDIF_OUT 0x02
+
+static struct hda_input_mux ad1884a_capture_source = {
+ .num_items = 5,
+ .items = {
+ { "Front Mic", 0x0 },
+ { "Mic", 0x4 },
+ { "Line", 0x1 },
+ { "CD", 0x2 },
+ { "Mix", 0x3 },
+ },
+};
+
+static struct snd_kcontrol_new ad1884a_base_mixers[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
+ HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT),
+ HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Boost", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x25, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = ad198x_mux_enum_info,
+ .get = ad198x_mux_enum_get,
+ .put = ad198x_mux_enum_put,
+ },
+ /* SPDIF controls */
+ HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
+ /* identical with ad1983 */
+ .info = ad1983_spdif_route_info,
+ .get = ad1983_spdif_route_get,
+ .put = ad1983_spdif_route_put,
+ },
+ { } /* end */
+};
+
+/*
+ * initialization verbs
+ */
+static struct hda_verb ad1884a_init_verbs[] = {
+ /* DACs; unmute as default */
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
+ /* Port-A (HP) mixer - route only from analog mixer */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* Port-A pin */
+ {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Port-D (Line-out) mixer - route only from analog mixer */
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* Port-D pin */
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Mono-out mixer - route only from analog mixer */
+ {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* Mono-out pin */
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Port-B (front mic) pin */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Port-C (rear line-in) pin */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Port-E (rear mic) pin */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* no boost */
+ /* Port-F (CD) pin */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Analog mixer; mute as default */
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, /* aux */
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+ /* Analog Mix output amp */
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* capture sources */
+ {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* SPDIF output amp */
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
+ { } /* end */
+};
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list ad1884a_loopbacks[] = {
+ { 0x20, HDA_INPUT, 0 }, /* Front Mic */
+ { 0x20, HDA_INPUT, 1 }, /* Mic */
+ { 0x20, HDA_INPUT, 2 }, /* CD */
+ { 0x20, HDA_INPUT, 4 }, /* Docking */
+ { } /* end */
+};
+#endif
+
+/*
+ * Laptop model
+ *
+ * Port A: Headphone jack
+ * Port B: MIC jack
+ * Port C: Internal MIC
+ * Port D: Dock Line Out (if enabled)
+ * Port E: Dock Line In (if enabled)
+ * Port F: Internal speakers
+ */
+
+static struct hda_input_mux ad1884a_laptop_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 }, /* port-B */
+ { "Internal Mic", 0x1 }, /* port-C */
+ { "Dock Mic", 0x4 }, /* port-E */
+ { "Mix", 0x3 },
+ },
+};
+
+static struct snd_kcontrol_new ad1884a_laptop_mixers[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Dock Playback Switch", 0x12, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
+ HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x20, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x20, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT),
+ HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Boost", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Dock Mic Boost", 0x25, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = ad198x_mux_enum_info,
+ .get = ad198x_mux_enum_get,
+ .put = ad198x_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct hda_input_mux ad1884a_mobile_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic", 0x1 }, /* port-C */
+ { "Mix", 0x3 },
+ },
+};
+
+static struct snd_kcontrol_new ad1884a_mobile_mixers[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
+ HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT),
+ HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = ad198x_mux_enum_info,
+ .get = ad198x_mux_enum_get,
+ .put = ad198x_mux_enum_put,
+ },
+ { } /* end */
+};
+
+/* mute internal speaker if HP is plugged */
+static void ad1884a_hp_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x11, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_EAPD_BTLENABLE,
+ present ? 0x00 : 0x02);
+}
+
+#define AD1884A_HP_EVENT 0x37
+
+/* unsolicited event for HP jack sensing */
+static void ad1884a_hp_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ if ((res >> 26) != AD1884A_HP_EVENT)
+ return;
+ ad1884a_hp_automute(codec);
+}
+
+/* initialize jack-sensing, too */
+static int ad1884a_hp_init(struct hda_codec *codec)
+{
+ ad198x_init(codec);
+ ad1884a_hp_automute(codec);
+ return 0;
+}
+
+/* additional verbs for laptop model */
+static struct hda_verb ad1884a_laptop_verbs[] = {
+ /* Port-A (HP) pin - always unmuted */
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Port-F (int speaker) mixer - route only from analog mixer */
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* Port-F pin */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* analog mix */
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* unsolicited event for pin-sense */
+ {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
+ { } /* end */
+};
+
+/*
+ * Thinkpad X300
+ * 0x11 - HP
+ * 0x12 - speaker
+ * 0x14 - mic-in
+ * 0x17 - built-in mic
+ */
+
+static struct hda_verb ad1984a_thinkpad_verbs[] = {
+ /* HP unmute */
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* analog mix */
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* turn on EAPD */
+ {0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
+ /* unsolicited event for pin-sense */
+ {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
+ /* internal mic - dmic */
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ /* set magic COEFs for dmic */
+ {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7},
+ {0x01, AC_VERB_SET_PROC_COEF, 0x08},
+ { } /* end */
+};
+
+static struct snd_kcontrol_new ad1984a_thinkpad_mixers[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
+ HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT),
+ HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Boost", 0x17, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = ad198x_mux_enum_info,
+ .get = ad198x_mux_enum_get,
+ .put = ad198x_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct hda_input_mux ad1984a_thinkpad_capture_source = {
+ .num_items = 3,
+ .items = {
+ { "Mic", 0x0 },
+ { "Internal Mic", 0x5 },
+ { "Mix", 0x3 },
+ },
+};
+
+/* mute internal speaker if HP is plugged */
+static void ad1984a_thinkpad_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x11, 0, AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+ snd_hda_codec_amp_stereo(codec, 0x12, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+}
+
+/* unsolicited event for HP jack sensing */
+static void ad1984a_thinkpad_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) != AD1884A_HP_EVENT)
+ return;
+ ad1984a_thinkpad_automute(codec);
+}
+
+/* initialize jack-sensing, too */
+static int ad1984a_thinkpad_init(struct hda_codec *codec)
+{
+ ad198x_init(codec);
+ ad1984a_thinkpad_automute(codec);
+ return 0;
+}
+
+/*
+ */
+
+enum {
+ AD1884A_DESKTOP,
+ AD1884A_LAPTOP,
+ AD1884A_MOBILE,
+ AD1884A_THINKPAD,
+ AD1884A_MODELS
+};
+
+static const char *ad1884a_models[AD1884A_MODELS] = {
+ [AD1884A_DESKTOP] = "desktop",
+ [AD1884A_LAPTOP] = "laptop",
+ [AD1884A_MOBILE] = "mobile",
+ [AD1884A_THINKPAD] = "thinkpad",
+};
+
+static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE),
+ SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD),
+ {}
+};
+
+static int patch_ad1884a(struct hda_codec *codec)
+{
+ struct ad198x_spec *spec;
+ int board_config;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ mutex_init(&spec->amp_mutex);
+ codec->spec = spec;
+
+ spec->multiout.max_channels = 2;
+ spec->multiout.num_dacs = ARRAY_SIZE(ad1884a_dac_nids);
+ spec->multiout.dac_nids = ad1884a_dac_nids;
+ spec->multiout.dig_out_nid = AD1884A_SPDIF_OUT;
+ spec->num_adc_nids = ARRAY_SIZE(ad1884a_adc_nids);
+ spec->adc_nids = ad1884a_adc_nids;
+ spec->capsrc_nids = ad1884a_capsrc_nids;
+ spec->input_mux = &ad1884a_capture_source;
+ spec->num_mixers = 1;
+ spec->mixers[0] = ad1884a_base_mixers;
+ spec->num_init_verbs = 1;
+ spec->init_verbs[0] = ad1884a_init_verbs;
+ spec->spdif_route = 0;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = ad1884a_loopbacks;
+#endif
+ codec->patch_ops = ad198x_patch_ops;
+
+ /* override some parameters */
+ board_config = snd_hda_check_board_config(codec, AD1884A_MODELS,
+ ad1884a_models,
+ ad1884a_cfg_tbl);
+ switch (board_config) {
+ case AD1884A_LAPTOP:
+ spec->mixers[0] = ad1884a_laptop_mixers;
+ spec->init_verbs[spec->num_init_verbs++] = ad1884a_laptop_verbs;
+ spec->multiout.dig_out_nid = 0;
+ spec->input_mux = &ad1884a_laptop_capture_source;
+ codec->patch_ops.unsol_event = ad1884a_hp_unsol_event;
+ codec->patch_ops.init = ad1884a_hp_init;
+ break;
+ case AD1884A_MOBILE:
+ spec->mixers[0] = ad1884a_mobile_mixers;
+ spec->init_verbs[spec->num_init_verbs++] = ad1884a_laptop_verbs;
+ spec->multiout.dig_out_nid = 0;
+ spec->input_mux = &ad1884a_mobile_capture_source;
+ codec->patch_ops.unsol_event = ad1884a_hp_unsol_event;
+ codec->patch_ops.init = ad1884a_hp_init;
+ break;
+ case AD1884A_THINKPAD:
+ spec->mixers[0] = ad1984a_thinkpad_mixers;
+ spec->init_verbs[spec->num_init_verbs++] =
+ ad1984a_thinkpad_verbs;
+ spec->multiout.dig_out_nid = 0;
+ spec->input_mux = &ad1984a_thinkpad_capture_source;
+ codec->patch_ops.unsol_event = ad1984a_thinkpad_unsol_event;
+ codec->patch_ops.init = ad1984a_thinkpad_init;
+ break;
+ }
+
+ return 0;
+}
+
+
+/*
* AD1882
*
* port-A - front hp-out
@@ -3654,13 +4195,19 @@ static int patch_ad1882(struct hda_codec *codec)
* patch entries
*/
struct hda_codec_preset snd_hda_preset_analog[] = {
+ { .id = 0x11d4184a, .name = "AD1884A", .patch = patch_ad1884a },
{ .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 },
+ { .id = 0x11d41883, .name = "AD1883", .patch = patch_ad1884a },
{ .id = 0x11d41884, .name = "AD1884", .patch = patch_ad1884 },
+ { .id = 0x11d4194a, .name = "AD1984A", .patch = patch_ad1884a },
+ { .id = 0x11d4194b, .name = "AD1984B", .patch = patch_ad1884a },
{ .id = 0x11d41981, .name = "AD1981", .patch = patch_ad1981 },
{ .id = 0x11d41983, .name = "AD1983", .patch = patch_ad1983 },
{ .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1984 },
{ .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a },
{ .id = 0x11d41988, .name = "AD1988", .patch = patch_ad1988 },
{ .id = 0x11d4198b, .name = "AD1988B", .patch = patch_ad1988 },
+ { .id = 0x11d4989a, .name = "AD1989A", .patch = patch_ad1988 },
+ { .id = 0x11d4989b, .name = "AD1989B", .patch = patch_ad1988 },
{} /* terminator */
};
diff --git a/sound/pci/hda/patch_atihdmi.c b/sound/pci/hda/patch_atihdmi.c
index 9a8bb4ce3f8..12272508b11 100644
--- a/sound/pci/hda/patch_atihdmi.c
+++ b/sound/pci/hda/patch_atihdmi.c
@@ -27,6 +27,7 @@
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"
+#include "hda_patch.h"
struct atihdmi_spec {
struct hda_multi_out multiout;
@@ -58,6 +59,10 @@ static int atihdmi_build_controls(struct hda_codec *codec)
static int atihdmi_init(struct hda_codec *codec)
{
snd_hda_sequence_write(codec, atihdmi_basic_init);
+ /* SI codec requires to unmute the pin */
+ if (get_wcaps(codec, 0x03) & AC_WCAP_OUT_AMP)
+ snd_hda_codec_write(codec, 0x03, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_OUT_UNMUTE);
return 0;
}
@@ -112,6 +117,7 @@ static int atihdmi_build_pcms(struct hda_codec *codec)
codec->pcm_info = info;
info->name = "ATI HDMI";
+ info->pcm_type = HDA_PCM_TYPE_HDMI;
info->stream[SNDRV_PCM_STREAM_PLAYBACK] = atihdmi_pcm_digital_playback;
return 0;
@@ -158,5 +164,7 @@ struct hda_codec_preset snd_hda_preset_atihdmi[] = {
{ .id = 0x10027919, .name = "ATI RS600 HDMI", .patch = patch_atihdmi },
{ .id = 0x1002791a, .name = "ATI RS690/780 HDMI", .patch = patch_atihdmi },
{ .id = 0x1002aa01, .name = "ATI R6xx HDMI", .patch = patch_atihdmi },
+ { .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_atihdmi },
+ { .id = 0x17e80047, .name = "Chrontel HDMI", .patch = patch_atihdmi },
{} /* terminator */
};
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index 3d6097ba1d6..c73ce074a6e 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -28,6 +28,7 @@
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"
+#include "hda_patch.h"
#define NUM_PINS 11
@@ -329,6 +330,11 @@ static int cmi9880_build_controls(struct hda_codec *codec)
err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
if (err < 0)
return err;
+ err = snd_hda_create_spdif_share_sw(codec,
+ &spec->multiout);
+ if (err < 0)
+ return err;
+ spec->multiout.share_spdif = 1;
}
if (spec->dig_in_nid) {
err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
@@ -432,7 +438,8 @@ static int cmi9880_playback_pcm_open(struct hda_pcm_stream *hinfo,
struct snd_pcm_substream *substream)
{
struct cmi_spec *spec = codec->spec;
- return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream);
+ return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
+ hinfo);
}
static int cmi9880_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
@@ -506,7 +513,7 @@ static int cmi9880_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
{
struct cmi_spec *spec = codec->spec;
- snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], 0, 0, 0);
+ snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]);
return 0;
}
@@ -571,6 +578,7 @@ static int cmi9880_build_pcms(struct hda_codec *codec)
codec->num_pcms++;
info++;
info->name = "CMI9880 Digital";
+ info->pcm_type = HDA_PCM_TYPE_SPDIF;
if (spec->multiout.dig_out_nid) {
info->stream[SNDRV_PCM_STREAM_PLAYBACK] = cmi9880_pcm_digital_playback;
info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid;
@@ -603,6 +611,7 @@ static const char *cmi9880_models[CMI_MODELS] = {
static struct snd_pci_quirk cmi9880_cfg_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", CMI_FULL_DIG),
+ SND_PCI_QUIRK(0x1854, 0x0032, "LG", CMI_FULL_DIG),
{} /* terminator */
};
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 7206b30cbf9..36fd8526003 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -27,6 +27,7 @@
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"
+#include "hda_patch.h"
#define CXT_PIN_DIR_IN 0x00
#define CXT_PIN_DIR_OUT 0x01
@@ -98,7 +99,8 @@ static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo,
struct snd_pcm_substream *substream)
{
struct conexant_spec *spec = codec->spec;
- return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream);
+ return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
+ hinfo);
}
static int conexant_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
@@ -172,8 +174,7 @@ static int conexant_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
struct snd_pcm_substream *substream)
{
struct conexant_spec *spec = codec->spec;
- snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number],
- 0, 0, 0);
+ snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]);
return 0;
}
@@ -241,7 +242,7 @@ static int cx5051_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
struct snd_pcm_substream *substream)
{
struct conexant_spec *spec = codec->spec;
- snd_hda_codec_setup_stream(codec, spec->cur_adc, 0, 0, 0);
+ snd_hda_codec_cleanup_stream(codec, spec->cur_adc);
spec->cur_adc = 0;
return 0;
}
@@ -284,6 +285,7 @@ static int conexant_build_pcms(struct hda_codec *codec)
info++;
codec->num_pcms++;
info->name = "Conexant Digital";
+ info->pcm_type = HDA_PCM_TYPE_SPDIF;
info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
conexant_pcm_digital_playback;
info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid =
@@ -371,6 +373,11 @@ static int conexant_build_controls(struct hda_codec *codec)
spec->multiout.dig_out_nid);
if (err < 0)
return err;
+ err = snd_hda_create_spdif_share_sw(codec,
+ &spec->multiout);
+ if (err < 0)
+ return err;
+ spec->multiout.share_spdif = 1;
}
if (spec->dig_in_nid) {
err = snd_hda_create_spdif_in_ctls(codec,spec->dig_in_nid);
@@ -511,6 +518,14 @@ static struct hda_input_mux cxt5045_capture_source_benq = {
}
};
+static struct hda_input_mux cxt5045_capture_source_hp530 = {
+ .num_items = 2,
+ .items = {
+ { "ExtMic", 0x1 },
+ { "IntMic", 0x2 },
+ }
+};
+
/* turn on/off EAPD (+ mute HP) as a master switch */
static int cxt5045_hp_master_sw_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -639,6 +654,37 @@ static struct snd_kcontrol_new cxt5045_benq_mixers[] = {
{}
};
+static struct snd_kcontrol_new cxt5045_mixers_hp530[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = conexant_mux_enum_info,
+ .get = conexant_mux_enum_get,
+ .put = conexant_mux_enum_put
+ },
+ HDA_CODEC_VOLUME("Int Mic Capture Volume", 0x1a, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Int Mic Capture Switch", 0x1a, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Ext Mic Capture Volume", 0x1a, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Ext Mic Capture Switch", 0x1a, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x17, 0x2, HDA_INPUT),
+ HDA_CODEC_MUTE("Int Mic Playback Switch", 0x17, 0x2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Ext Mic Playback Volume", 0x17, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Ext Mic Playback Switch", 0x17, 0x1, HDA_INPUT),
+ HDA_BIND_VOL("Master Playback Volume", &cxt5045_hp_bind_master_vol),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = cxt_eapd_info,
+ .get = cxt_eapd_get,
+ .put = cxt5045_hp_master_sw_put,
+ .private_value = 0x10,
+ },
+
+ {}
+};
+
static struct hda_verb cxt5045_init_verbs[] = {
/* Line in, Mic */
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
@@ -833,6 +879,7 @@ enum {
CXT5045_LAPTOP_MICSENSE,
CXT5045_LAPTOP_HPMICSENSE,
CXT5045_BENQ,
+ CXT5045_LAPTOP_HP530,
#ifdef CONFIG_SND_DEBUG
CXT5045_TEST,
#endif
@@ -844,6 +891,7 @@ static const char *cxt5045_models[CXT5045_MODELS] = {
[CXT5045_LAPTOP_MICSENSE] = "laptop-micsense",
[CXT5045_LAPTOP_HPMICSENSE] = "laptop-hpmicsense",
[CXT5045_BENQ] = "benq",
+ [CXT5045_LAPTOP_HP530] = "laptop-hp530",
#ifdef CONFIG_SND_DEBUG
[CXT5045_TEST] = "test",
#endif
@@ -857,7 +905,7 @@ static struct snd_pci_quirk cxt5045_cfg_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x30bb, "HP DV8000", CXT5045_LAPTOP_HPSENSE),
SND_PCI_QUIRK(0x103c, 0x30cd, "HP DV Series", CXT5045_LAPTOP_HPSENSE),
SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV9533EG", CXT5045_LAPTOP_HPSENSE),
- SND_PCI_QUIRK(0x103c, 0x30d5, "HP 530", CXT5045_LAPTOP_HPSENSE),
+ SND_PCI_QUIRK(0x103c, 0x30d5, "HP 530", CXT5045_LAPTOP_HP530),
SND_PCI_QUIRK(0x103c, 0x30d9, "HP Spartan", CXT5045_LAPTOP_HPSENSE),
SND_PCI_QUIRK(0x152d, 0x0753, "Benq R55E", CXT5045_BENQ),
SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_LAPTOP_MICSENSE),
@@ -941,6 +989,14 @@ static int patch_cxt5045(struct hda_codec *codec)
spec->num_mixers = 2;
codec->patch_ops.init = cxt5045_init;
break;
+ case CXT5045_LAPTOP_HP530:
+ codec->patch_ops.unsol_event = cxt5045_hp_unsol_event;
+ spec->input_mux = &cxt5045_capture_source_hp530;
+ spec->num_init_verbs = 2;
+ spec->init_verbs[1] = cxt5045_hp_sense_init_verbs;
+ spec->mixers[0] = cxt5045_mixers_hp530;
+ codec->patch_ops.init = cxt5045_init;
+ break;
#ifdef CONFIG_SND_DEBUG
case CXT5045_TEST:
spec->input_mux = &cxt5045_test_capture_source;
@@ -1537,7 +1593,7 @@ static void cxt5051_portc_automic(struct hda_codec *codec)
new_adc = spec->adc_nids[spec->cur_adc_idx];
if (spec->cur_adc && spec->cur_adc != new_adc) {
/* stream is running, let's swap the current ADC */
- snd_hda_codec_setup_stream(codec, spec->cur_adc, 0, 0, 0);
+ snd_hda_codec_cleanup_stream(codec, spec->cur_adc);
spec->cur_adc = new_adc;
snd_hda_codec_setup_stream(codec, new_adc,
spec->cur_adc_stream_tag, 0,
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 33282f9c01c..cdda64b02f4 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -30,6 +30,7 @@
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"
+#include "hda_patch.h"
#define ALC880_FRONT_EVENT 0x01
#define ALC880_DCVOL_EVENT 0x02
@@ -97,16 +98,19 @@ enum {
ALC262_SONY_ASSAMD,
ALC262_BENQ_T31,
ALC262_ULTRA,
+ ALC262_LENOVO_3000,
ALC262_AUTO,
ALC262_MODEL_LAST /* last tag */
};
/* ALC268 models */
enum {
+ ALC267_QUANTA_IL1,
ALC268_3ST,
ALC268_TOSHIBA,
ALC268_ACER,
ALC268_DELL,
+ ALC268_ZEPTO,
#ifdef CONFIG_SND_DEBUG
ALC268_TEST,
#endif
@@ -195,10 +199,11 @@ enum {
ALC883_LENOVO_NB0763,
ALC888_LENOVO_MS7195_DIG,
ALC883_HAIER_W66,
- ALC888_6ST_HP,
ALC888_3ST_HP,
ALC888_6ST_DELL,
ALC883_MITAC,
+ ALC883_CLEVO_M720,
+ ALC883_FUJITSU_PI2515,
ALC883_AUTO,
ALC883_MODEL_LAST,
};
@@ -237,6 +242,7 @@ struct alc_spec {
/* capture */
unsigned int num_adc_nids;
hda_nid_t *adc_nids;
+ hda_nid_t *capsrc_nids;
hda_nid_t dig_in_nid; /* digital-in NID; optional */
/* capture source */
@@ -270,7 +276,6 @@ struct alc_spec {
/* for virtual master */
hda_nid_t vmaster_nid;
- u32 vmaster_tlv[4];
#ifdef CONFIG_SND_HDA_POWER_SAVE
struct hda_loopback_check loopback;
#endif
@@ -290,6 +295,7 @@ struct alc_config_preset {
hda_nid_t hp_nid; /* optional */
unsigned int num_adc_nids;
hda_nid_t *adc_nids;
+ hda_nid_t *capsrc_nids;
hda_nid_t dig_in_nid;
unsigned int num_channel_mode;
const struct hda_channel_mode *channel_mode;
@@ -336,9 +342,10 @@ static int alc_mux_enum_put(struct snd_kcontrol *kcontrol,
struct alc_spec *spec = codec->spec;
unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
unsigned int mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx;
+ hda_nid_t nid = spec->capsrc_nids ?
+ spec->capsrc_nids[adc_idx] : spec->adc_nids[adc_idx];
return snd_hda_input_mux_put(codec, &spec->input_mux[mux_idx], ucontrol,
- spec->adc_nids[adc_idx],
- &spec->cur_mux[adc_idx]);
+ nid, &spec->cur_mux[adc_idx]);
}
@@ -707,6 +714,7 @@ static void setup_preset(struct alc_spec *spec,
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;
@@ -741,7 +749,6 @@ static struct hda_verb alc_gpio3_init_verbs[] = {
static void alc_sku_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- unsigned int mute;
unsigned int present;
unsigned int hp_nid = spec->autocfg.hp_pins[0];
unsigned int sp_nid = spec->autocfg.speaker_pins[0];
@@ -751,16 +758,8 @@ static void alc_sku_automute(struct hda_codec *codec)
present = snd_hda_codec_read(codec, hp_nid, 0,
AC_VERB_GET_PIN_SENSE, 0);
spec->jack_present = (present & 0x80000000) != 0;
- if (spec->jack_present) {
- /* mute internal speaker */
- snd_hda_codec_amp_stereo(codec, sp_nid, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, HDA_AMP_MUTE);
- } else {
- /* unmute internal speaker if necessary */
- mute = snd_hda_codec_amp_read(codec, hp_nid, 0, HDA_OUTPUT, 0);
- snd_hda_codec_amp_stereo(codec, sp_nid, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, mute);
- }
+ snd_hda_codec_write(codec, sp_nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ spec->jack_present ? 0 : PIN_OUT);
}
/* unsolicited event for HP jack sensing */
@@ -1319,11 +1318,19 @@ static struct snd_kcontrol_new alc880_f1734_mixer[] = {
HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
+static struct hda_input_mux alc880_f1734_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic", 0x1 },
+ { "CD", 0x4 },
+ },
+};
+
/*
* ALC880 ASUS model
@@ -1516,6 +1523,11 @@ static int alc_build_controls(struct hda_codec *codec)
spec->multiout.dig_out_nid);
if (err < 0)
return err;
+ err = snd_hda_create_spdif_share_sw(codec,
+ &spec->multiout);
+ if (err < 0)
+ return err;
+ spec->multiout.share_spdif = 1;
}
if (spec->dig_in_nid) {
err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
@@ -1525,10 +1537,11 @@ static int alc_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->vmaster_nid,
- HDA_OUTPUT, spec->vmaster_tlv);
+ HDA_OUTPUT, vmaster_tlv);
err = snd_hda_add_vmaster(codec, "Master Playback Volume",
- spec->vmaster_tlv, alc_slave_vols);
+ vmaster_tlv, alc_slave_vols);
if (err < 0)
return err;
}
@@ -1882,7 +1895,7 @@ static void alc880_uniwill_p53_hp_automute(struct hda_codec *codec)
present = snd_hda_codec_read(codec, 0x14, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
bits = present ? HDA_AMP_MUTE : 0;
- snd_hda_codec_amp_stereo(codec, 0x15, HDA_INPUT, 0, HDA_AMP_MUTE, bits);
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits);
}
static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec)
@@ -1915,6 +1928,7 @@ static void alc880_uniwill_p53_unsol_event(struct hda_codec *codec,
* HP = 0x14, speaker-out = 0x15, mic = 0x18
*/
static struct hda_verb alc880_pin_f1734_init_verbs[] = {
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x10, AC_VERB_SET_CONNECT_SEL, 0x02},
{0x11, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x12, AC_VERB_SET_CONNECT_SEL, 0x01},
@@ -1927,7 +1941,7 @@ static struct hda_verb alc880_pin_f1734_init_verbs[] = {
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
@@ -1935,6 +1949,9 @@ static struct hda_verb alc880_pin_f1734_init_verbs[] = {
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_HP_EVENT},
+ {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_DCVOL_EVENT},
+
{ }
};
@@ -2318,7 +2335,8 @@ static int alc880_playback_pcm_open(struct hda_pcm_stream *hinfo,
struct snd_pcm_substream *substream)
{
struct alc_spec *spec = codec->spec;
- return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream);
+ return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
+ hinfo);
}
static int alc880_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
@@ -2392,8 +2410,8 @@ static int alc880_alt_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
{
struct alc_spec *spec = codec->spec;
- snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number + 1],
- 0, 0, 0);
+ snd_hda_codec_cleanup_stream(codec,
+ spec->adc_nids[substream->number + 1]);
return 0;
}
@@ -2498,6 +2516,7 @@ static int alc_build_pcms(struct hda_codec *codec)
codec->num_pcms = 2;
info = spec->pcm_rec + 1;
info->name = spec->stream_name_digital;
+ info->pcm_type = HDA_PCM_TYPE_SPDIF;
if (spec->multiout.dig_out_nid &&
spec->stream_digital_playback) {
info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_digital_playback);
@@ -2560,6 +2579,7 @@ static void alc_free(struct hda_codec *codec)
kfree(spec->kctl_alloc);
}
kfree(spec);
+ codec->spec = NULL; /* to be sure */
}
/*
@@ -3057,7 +3077,9 @@ static struct alc_config_preset alc880_presets[] = {
.hp_nid = 0x02,
.num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
.channel_mode = alc880_2_jack_modes,
- .input_mux = &alc880_capture_source,
+ .input_mux = &alc880_f1734_capture_source,
+ .unsol_event = alc880_uniwill_p53_unsol_event,
+ .init_hook = alc880_uniwill_p53_hp_automute,
},
[ALC880_ASUS] = {
.mixers = { alc880_asus_mixer },
@@ -3467,15 +3489,21 @@ static int alc880_auto_create_analog_input_ctls(struct alc_spec *spec,
return 0;
}
-static void alc880_auto_set_output_and_unmute(struct hda_codec *codec,
- hda_nid_t nid, int pin_type,
- int dac_idx)
+static void alc_set_pin_output(struct hda_codec *codec, hda_nid_t nid,
+ unsigned int pin_type)
{
- /* set as output */
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
pin_type);
+ /* unmute pin */
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
AMP_OUT_UNMUTE);
+}
+
+static void alc880_auto_set_output_and_unmute(struct hda_codec *codec,
+ hda_nid_t nid, int pin_type,
+ int dac_idx)
+{
+ alc_set_pin_output(codec, nid, pin_type);
/* need the manual connection? */
if (alc880_is_multi_pin(nid)) {
struct alc_spec *spec = codec->spec;
@@ -3597,9 +3625,12 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
/* additional initialization for auto-configuration model */
static void alc880_auto_init(struct hda_codec *codec)
{
+ struct alc_spec *spec = codec->spec;
alc880_auto_init_multi_out(codec);
alc880_auto_init_extra_out(codec);
alc880_auto_init_analog_input(codec);
+ if (spec->unsol_event)
+ alc_sku_automute(codec);
}
/*
@@ -4795,11 +4826,7 @@ static void alc260_auto_set_output_and_unmute(struct hda_codec *codec,
hda_nid_t nid, int pin_type,
int sel_idx)
{
- /* set as output */
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
- pin_type);
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
- AMP_OUT_UNMUTE);
+ alc_set_pin_output(codec, nid, pin_type);
/* need the manual connection? */
if (nid >= 0x12) {
int idx = nid - 0x12;
@@ -4929,7 +4956,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec)
/* check whether NID 0x04 is valid */
wcap = get_wcaps(codec, 0x04);
wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; /* get type */
- if (wcap != AC_WID_AUD_IN) {
+ if (wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) {
spec->adc_nids = alc260_adc_nids_alt;
spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt);
spec->mixers[spec->num_mixers] = alc260_capture_alt_mixer;
@@ -4946,8 +4973,11 @@ static int alc260_parse_auto_config(struct hda_codec *codec)
/* additional initialization for auto-configuration model */
static void alc260_auto_init(struct hda_codec *codec)
{
+ struct alc_spec *spec = codec->spec;
alc260_auto_init_multi_out(codec);
alc260_auto_init_analog_input(codec);
+ if (spec->unsol_event)
+ alc_sku_automute(codec);
}
#ifdef CONFIG_SND_HDA_POWER_SAVE
@@ -5204,6 +5234,9 @@ static hda_nid_t alc882_dac_nids[4] = {
#define alc882_adc_nids alc880_adc_nids
#define alc882_adc_nids_alt alc880_adc_nids_alt
+static hda_nid_t alc882_capsrc_nids[3] = { 0x24, 0x23, 0x22 };
+static hda_nid_t alc882_capsrc_nids_alt[2] = { 0x23, 0x22 };
+
/* input MUX */
/* FIXME: should be a matrix-type input source selection */
@@ -5226,15 +5259,11 @@ static int alc882_mux_enum_put(struct snd_kcontrol *kcontrol,
struct alc_spec *spec = codec->spec;
const struct hda_input_mux *imux = spec->input_mux;
unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
- static hda_nid_t capture_mixers[3] = { 0x24, 0x23, 0x22 };
- hda_nid_t nid;
+ hda_nid_t nid = spec->capsrc_nids ?
+ spec->capsrc_nids[adc_idx] : spec->adc_nids[adc_idx];
unsigned int *cur_val = &spec->cur_mux[adc_idx];
unsigned int i, idx;
- if (spec->num_adc_nids < 3)
- nid = capture_mixers[adc_idx + 1];
- else
- nid = capture_mixers[adc_idx];
idx = ucontrol->value.enumerated.item[0];
if (idx >= imux->num_items)
idx = imux->num_items - 1;
@@ -6111,6 +6140,7 @@ static struct alc_config_preset alc882_presets[] = {
.dig_out_nid = ALC882_DIGOUT_NID,
.num_adc_nids = ARRAY_SIZE(alc882_adc_nids),
.adc_nids = alc882_adc_nids,
+ .capsrc_nids = alc882_capsrc_nids,
.num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes),
.channel_mode = alc882_3ST_6ch_modes,
.need_dac_fix = 1,
@@ -6127,6 +6157,7 @@ static struct alc_config_preset alc882_presets[] = {
.dig_out_nid = ALC882_DIGOUT_NID,
.num_adc_nids = ARRAY_SIZE(alc882_adc_nids),
.adc_nids = alc882_adc_nids,
+ .capsrc_nids = alc882_capsrc_nids,
.num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes),
.channel_mode = alc882_3ST_6ch_modes,
.need_dac_fix = 1,
@@ -6182,15 +6213,11 @@ static void alc882_auto_set_output_and_unmute(struct hda_codec *codec,
struct alc_spec *spec = codec->spec;
int idx;
+ alc_set_pin_output(codec, nid, pin_type);
if (spec->multiout.dac_nids[dac_idx] == 0x25)
idx = 4;
else
idx = spec->multiout.dac_nids[dac_idx] - 2;
-
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
- pin_type);
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
- AMP_OUT_UNMUTE);
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx);
}
@@ -6219,6 +6246,9 @@ static void alc882_auto_init_hp_out(struct hda_codec *codec)
if (pin) /* connect to front */
/* use dac 0 */
alc882_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
+ pin = spec->autocfg.speaker_pins[0];
+ if (pin)
+ alc882_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
}
#define alc882_is_input_pin(nid) alc880_is_input_pin(nid)
@@ -6231,16 +6261,21 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec)
for (i = 0; i < AUTO_PIN_LAST; i++) {
hda_nid_t nid = spec->autocfg.input_pins[i];
- if (alc882_is_input_pin(nid)) {
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- i <= AUTO_PIN_FRONT_MIC ?
- PIN_VREF80 : PIN_IN);
- if (nid != ALC882_PIN_CD_NID)
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_AMP_GAIN_MUTE,
- AMP_OUT_MUTE);
+ unsigned int vref;
+ if (!nid)
+ continue;
+ vref = PIN_IN;
+ if (1 /*i <= AUTO_PIN_FRONT_MIC*/) {
+ if (snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP) &
+ AC_PINCAP_VREF_80)
+ vref = PIN_VREF80;
}
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, vref);
+ if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_OUT_MUTE);
}
}
@@ -6294,11 +6329,16 @@ static int alc882_parse_auto_config(struct hda_codec *codec)
/* additional initialization for auto-configuration model */
static void alc882_auto_init(struct hda_codec *codec)
{
+ struct alc_spec *spec = codec->spec;
alc882_auto_init_multi_out(codec);
alc882_auto_init_hp_out(codec);
alc882_auto_init_analog_input(codec);
+ if (spec->unsol_event)
+ alc_sku_automute(codec);
}
+static int patch_alc883(struct hda_codec *codec); /* called in patch_alc882() */
+
static int patch_alc882(struct hda_codec *codec)
{
struct alc_spec *spec;
@@ -6328,6 +6368,11 @@ static int patch_alc882(struct hda_codec *codec)
board_config = ALC885_MBP3;
break;
default:
+ /* ALC889A is handled better as ALC888-compatible */
+ if (codec->revision_id == 0x100103) {
+ alc_free(codec);
+ return patch_alc883(codec);
+ }
printk(KERN_INFO "hda_codec: Unknown model for ALC882, "
"trying auto-probe from BIOS...\n");
board_config = ALC882_AUTO;
@@ -6372,12 +6417,14 @@ static int patch_alc882(struct hda_codec *codec)
if (wcap != AC_WID_AUD_IN) {
spec->adc_nids = alc882_adc_nids_alt;
spec->num_adc_nids = ARRAY_SIZE(alc882_adc_nids_alt);
+ spec->capsrc_nids = alc882_capsrc_nids_alt;
spec->mixers[spec->num_mixers] =
alc882_capture_alt_mixer;
spec->num_mixers++;
} else {
spec->adc_nids = alc882_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(alc882_adc_nids);
+ spec->capsrc_nids = alc882_capsrc_nids;
spec->mixers[spec->num_mixers] = alc882_capture_mixer;
spec->num_mixers++;
}
@@ -6412,7 +6459,7 @@ static int patch_alc882(struct hda_codec *codec)
static hda_nid_t alc883_dac_nids[4] = {
/* front, rear, clfe, rear_surr */
- 0x02, 0x04, 0x03, 0x05
+ 0x02, 0x03, 0x04, 0x05
};
static hda_nid_t alc883_adc_nids[2] = {
@@ -6420,6 +6467,8 @@ static hda_nid_t alc883_adc_nids[2] = {
0x08, 0x09,
};
+static hda_nid_t alc883_capsrc_nids[2] = { 0x23, 0x22 };
+
/* input MUX */
/* FIXME: should be a matrix-type input source selection */
@@ -6451,35 +6500,18 @@ static struct hda_input_mux alc883_lenovo_nb0763_capture_source = {
},
};
+static struct hda_input_mux alc883_fujitsu_pi2515_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic", 0x0 },
+ { "Int Mic", 0x1 },
+ },
+};
+
#define alc883_mux_enum_info alc_mux_enum_info
#define alc883_mux_enum_get alc_mux_enum_get
-
-static int alc883_mux_enum_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;
- const struct hda_input_mux *imux = spec->input_mux;
- unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
- static hda_nid_t capture_mixers[2] = { 0x23, 0x22 };
- hda_nid_t nid = capture_mixers[adc_idx];
- unsigned int *cur_val = &spec->cur_mux[adc_idx];
- unsigned int i, idx;
-
- idx = ucontrol->value.enumerated.item[0];
- if (idx >= imux->num_items)
- idx = imux->num_items - 1;
- if (*cur_val == idx)
- return 0;
- for (i = 0; i < imux->num_items; i++) {
- unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE;
- snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT,
- imux->items[i].index,
- HDA_AMP_MUTE, v);
- }
- *cur_val = idx;
- return 1;
-}
+/* ALC883 has the ALC882-type input selection */
+#define alc883_mux_enum_put alc882_mux_enum_put
/*
* 2ch mode
@@ -6638,6 +6670,60 @@ static struct snd_kcontrol_new alc883_mitac_mixer[] = {
{ } /* end */
};
+static struct snd_kcontrol_new alc883_clevo_m720_mixer[] = {
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = alc883_mux_enum_info,
+ .get = alc883_mux_enum_get,
+ .put = alc883_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc883_2ch_fujitsu_pi2515_mixer[] = {
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = alc883_mux_enum_info,
+ .get = alc883_mux_enum_get,
+ .put = alc883_mux_enum_put,
+ },
+ { } /* end */
+};
+
static struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
@@ -6787,6 +6873,9 @@ static struct snd_kcontrol_new alc883_tagra_2ch_mixer[] = {
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
@@ -6878,124 +6967,6 @@ static struct snd_kcontrol_new alc883_medion_md2_mixer[] = {
{ } /* end */
};
-static struct snd_kcontrol_new alc888_6st_hp_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", 0x0e, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("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", 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", 0x19, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
- HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- /* .name = "Capture Source", */
- .name = "Input Source",
- .count = 2,
- .info = alc883_mux_enum_info,
- .get = alc883_mux_enum_get,
- .put = alc883_mux_enum_put,
- },
- { } /* end */
-};
-
-static struct snd_kcontrol_new alc888_3st_hp_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", 0x0e, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT),
- HDA_CODEC_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", 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", 0x19, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
- HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- /* .name = "Capture Source", */
- .name = "Input Source",
- .count = 2,
- .info = alc883_mux_enum_info,
- .get = alc883_mux_enum_get,
- .put = alc883_mux_enum_put,
- },
- { } /* end */
-};
-
-static struct snd_kcontrol_new alc888_6st_dell_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", 0x0e, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("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", 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", 0x19, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
- HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- /* .name = "Capture Source", */
- .name = "Input Source",
- .count = 2,
- .info = alc883_mux_enum_info,
- .get = alc883_mux_enum_get,
- .put = alc883_mux_enum_put,
- },
- { } /* end */
-};
-
static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
@@ -7171,6 +7142,35 @@ static struct hda_verb alc883_mitac_verbs[] = {
{ } /* end */
};
+static struct hda_verb alc883_clevo_m720_verbs[] = {
+ /* HP */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ /* Int speaker */
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+ /* enable unsolicited event */
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN},
+
+ { } /* end */
+};
+
+static struct hda_verb alc883_2ch_fujitsu_pi2515_verbs[] = {
+ /* HP */
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ /* Subwoofer */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+ /* enable unsolicited event */
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+
+ { } /* end */
+};
+
static struct hda_verb alc883_tagra_verbs[] = {
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
@@ -7227,26 +7227,14 @@ static struct hda_verb alc883_haier_w66_verbs[] = {
{ } /* end */
};
-static struct hda_verb alc888_6st_hp_verbs[] = {
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Rear : output 2 (0x0e) */
- {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* CLFE : output 1 (0x0d) */
- {0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, /* Side : output 3 (0x0f) */
- { }
-};
-
static struct hda_verb alc888_3st_hp_verbs[] = {
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */
- {0x18, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */
- {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CLFE : output 2 (0x0e) */
+ {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */
+ {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CLFE : output 2 (0x0e) */
{ }
};
static struct hda_verb alc888_6st_dell_verbs[] = {
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Rear : output 1 (0x0e) */
- {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* CLFE : output 2 (0x0d) */
- {0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, /* Side : output 3 (0x0f) */
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{ }
};
@@ -7354,6 +7342,68 @@ static void alc883_tagra_unsol_event(struct hda_codec *codec, unsigned int res)
alc883_tagra_automute(codec);
}
+/* toggle speaker-output according to the hp-jack state */
+static void alc883_clevo_m720_hp_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+}
+
+static void alc883_clevo_m720_mic_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x18, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+}
+
+static void alc883_clevo_m720_automute(struct hda_codec *codec)
+{
+ alc883_clevo_m720_hp_automute(codec);
+ alc883_clevo_m720_mic_automute(codec);
+}
+
+static void alc883_clevo_m720_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case ALC880_HP_EVENT:
+ alc883_clevo_m720_hp_automute(codec);
+ break;
+ case ALC880_MIC_EVENT:
+ alc883_clevo_m720_mic_automute(codec);
+ break;
+ }
+}
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc883_2ch_fujitsu_pi2515_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+}
+
+static void alc883_2ch_fujitsu_pi2515_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc883_2ch_fujitsu_pi2515_automute(codec);
+}
+
static void alc883_haier_w66_automute(struct hda_codec *codec)
{
unsigned int present;
@@ -7587,10 +7637,11 @@ static const char *alc883_models[ALC883_MODEL_LAST] = {
[ALC883_LENOVO_NB0763] = "lenovo-nb0763",
[ALC888_LENOVO_MS7195_DIG] = "lenovo-ms7195-dig",
[ALC883_HAIER_W66] = "haier-w66",
- [ALC888_6ST_HP] = "6stack-hp",
[ALC888_3ST_HP] = "3stack-hp",
[ALC888_6ST_DELL] = "6stack-dell",
[ALC883_MITAC] = "mitac",
+ [ALC883_CLEVO_M720] = "clevo-m720",
+ [ALC883_FUJITSU_PI2515] = "fujitsu-pi2515",
[ALC883_AUTO] = "auto",
};
@@ -7604,7 +7655,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP),
SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP),
- SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC888_6ST_HP),
+ SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG),
SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC),
@@ -7614,7 +7665,9 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG),
SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG),
SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG),
+ SND_PCI_QUIRK(0x1462, 0x2fb3, "MSI", ALC883_TARGA_2ch_DIG),
SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG),
+ SND_PCI_QUIRK(0x1462, 0x3783, "NEC S970", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x3b7f, "MSI", ALC883_TARGA_2ch_DIG),
SND_PCI_QUIRK(0x1462, 0x3ef9, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x3fc1, "MSI", ALC883_TARGA_DIG),
@@ -7627,13 +7680,17 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0x7250, "MSI", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x1462, 0x7267, "MSI", ALC883_3ST_6ch_DIG),
SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG),
SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720),
+ SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720),
SND_PCI_QUIRK(0x1558, 0, "Clevo laptop", ALC883_LAPTOP_EAPD),
SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch),
SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION),
+ SND_PCI_QUIRK(0x1734, 0x1108, "Fujitsu AMILO Pi2515", ALC883_FUJITSU_PI2515),
SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo 101e", ALC883_LENOVO_101E_2ch),
SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763),
SND_PCI_QUIRK(0x17aa, 0x3bfc, "Lenovo NB0763", ALC883_LENOVO_NB0763),
@@ -7652,8 +7709,6 @@ static struct alc_config_preset alc883_presets[] = {
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
- .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
- .adc_nids = alc883_adc_nids,
.dig_in_nid = ALC883_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
@@ -7665,8 +7720,6 @@ static struct alc_config_preset alc883_presets[] = {
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
- .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
- .adc_nids = alc883_adc_nids,
.dig_in_nid = ALC883_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
.channel_mode = alc883_3ST_6ch_modes,
@@ -7678,8 +7731,6 @@ static struct alc_config_preset alc883_presets[] = {
.init_verbs = { alc883_init_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
- .adc_nids = alc883_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
.channel_mode = alc883_3ST_6ch_modes,
.need_dac_fix = 1,
@@ -7691,8 +7742,6 @@ static struct alc_config_preset alc883_presets[] = {
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
- .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
- .adc_nids = alc883_adc_nids,
.dig_in_nid = ALC883_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
.channel_mode = alc883_sixstack_modes,
@@ -7704,8 +7753,6 @@ static struct alc_config_preset alc883_presets[] = {
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
- .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
- .adc_nids = alc883_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
.channel_mode = alc883_3ST_6ch_modes,
.need_dac_fix = 1,
@@ -7719,8 +7766,6 @@ static struct alc_config_preset alc883_presets[] = {
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
- .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
- .adc_nids = alc883_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
@@ -7737,8 +7782,6 @@ static struct alc_config_preset alc883_presets[] = {
.init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
- .adc_nids = alc883_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
@@ -7749,8 +7792,6 @@ static struct alc_config_preset alc883_presets[] = {
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
- .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
- .adc_nids = alc883_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
@@ -7764,8 +7805,6 @@ static struct alc_config_preset alc883_presets[] = {
alc883_medion_eapd_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
- .adc_nids = alc883_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
.channel_mode = alc883_sixstack_modes,
.input_mux = &alc883_capture_source,
@@ -7776,8 +7815,6 @@ static struct alc_config_preset alc883_presets[] = {
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
- .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
- .adc_nids = alc883_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
@@ -7789,19 +7826,27 @@ static struct alc_config_preset alc883_presets[] = {
.init_verbs = { alc883_init_verbs, alc882_eapd_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
- .adc_nids = alc883_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
},
+ [ALC883_CLEVO_M720] = {
+ .mixers = { alc883_clevo_m720_mixer },
+ .init_verbs = { alc883_init_verbs, alc883_clevo_m720_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .input_mux = &alc883_capture_source,
+ .unsol_event = alc883_clevo_m720_unsol_event,
+ .init_hook = alc883_clevo_m720_automute,
+ },
[ALC883_LENOVO_101E_2ch] = {
.mixers = { alc883_lenovo_101e_2ch_mixer},
.init_verbs = { alc883_init_verbs, alc883_lenovo_101e_verbs},
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
- .adc_nids = alc883_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_lenovo_101e_capture_source,
@@ -7813,8 +7858,6 @@ static struct alc_config_preset alc883_presets[] = {
.init_verbs = { alc883_init_verbs, alc883_lenovo_nb0763_verbs},
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
- .adc_nids = alc883_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.need_dac_fix = 1,
@@ -7828,8 +7871,6 @@ static struct alc_config_preset alc883_presets[] = {
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
- .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
- .adc_nids = alc883_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
.channel_mode = alc883_3ST_6ch_modes,
.need_dac_fix = 1,
@@ -7843,47 +7884,28 @@ static struct alc_config_preset alc883_presets[] = {
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
- .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
- .adc_nids = alc883_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
.unsol_event = alc883_haier_w66_unsol_event,
.init_hook = alc883_haier_w66_automute,
- },
- [ALC888_6ST_HP] = {
- .mixers = { alc888_6st_hp_mixer, alc883_chmode_mixer },
- .init_verbs = { alc883_init_verbs, alc888_6st_hp_verbs },
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .dig_out_nid = ALC883_DIGOUT_NID,
- .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
- .adc_nids = alc883_adc_nids,
- .dig_in_nid = ALC883_DIGIN_NID,
- .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
- .channel_mode = alc883_sixstack_modes,
- .input_mux = &alc883_capture_source,
},
[ALC888_3ST_HP] = {
- .mixers = { alc888_3st_hp_mixer, alc883_chmode_mixer },
+ .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs, alc888_3st_hp_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
- .adc_nids = alc883_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc888_3st_hp_modes),
.channel_mode = alc888_3st_hp_modes,
.need_dac_fix = 1,
.input_mux = &alc883_capture_source,
},
[ALC888_6ST_DELL] = {
- .mixers = { alc888_6st_dell_mixer, alc883_chmode_mixer },
+ .mixers = { alc883_base_mixer, alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs, alc888_6st_dell_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
- .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
- .adc_nids = alc883_adc_nids,
.dig_in_nid = ALC883_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
.channel_mode = alc883_sixstack_modes,
@@ -7896,14 +7918,25 @@ static struct alc_config_preset alc883_presets[] = {
.init_verbs = { alc883_init_verbs, alc883_mitac_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
- .adc_nids = alc883_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
.unsol_event = alc883_mitac_unsol_event,
.init_hook = alc883_mitac_automute,
},
+ [ALC883_FUJITSU_PI2515] = {
+ .mixers = { alc883_2ch_fujitsu_pi2515_mixer },
+ .init_verbs = { alc883_init_verbs,
+ alc883_2ch_fujitsu_pi2515_verbs},
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .input_mux = &alc883_fujitsu_pi2515_capture_source,
+ .unsol_event = alc883_2ch_fujitsu_pi2515_unsol_event,
+ .init_hook = alc883_2ch_fujitsu_pi2515_automute,
+ },
};
@@ -7918,15 +7951,11 @@ static void alc883_auto_set_output_and_unmute(struct hda_codec *codec,
struct alc_spec *spec = codec->spec;
int idx;
+ alc_set_pin_output(codec, nid, pin_type);
if (spec->multiout.dac_nids[dac_idx] == 0x25)
idx = 4;
else
idx = spec->multiout.dac_nids[dac_idx] - 2;
-
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
- pin_type);
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
- AMP_OUT_UNMUTE);
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx);
}
@@ -7955,6 +7984,9 @@ static void alc883_auto_init_hp_out(struct hda_codec *codec)
if (pin) /* connect to front */
/* use dac 0 */
alc883_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
+ pin = spec->autocfg.speaker_pins[0];
+ if (pin)
+ alc883_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
}
#define alc883_is_input_pin(nid) alc880_is_input_pin(nid)
@@ -8006,9 +8038,12 @@ static int alc883_parse_auto_config(struct hda_codec *codec)
/* additional initialization for auto-configuration model */
static void alc883_auto_init(struct hda_codec *codec)
{
+ struct alc_spec *spec = codec->spec;
alc883_auto_init_multi_out(codec);
alc883_auto_init_hp_out(codec);
alc883_auto_init_analog_input(codec);
+ if (spec->unsol_event)
+ alc_sku_automute(codec);
}
static int patch_alc883(struct hda_codec *codec)
@@ -8057,10 +8092,9 @@ static int patch_alc883(struct hda_codec *codec)
spec->stream_digital_playback = &alc883_pcm_digital_playback;
spec->stream_digital_capture = &alc883_pcm_digital_capture;
- if (!spec->adc_nids && spec->input_mux) {
- spec->adc_nids = alc883_adc_nids;
- spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids);
- }
+ spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids);
+ spec->adc_nids = alc883_adc_nids;
+ spec->capsrc_nids = alc883_capsrc_nids;
spec->vmaster_nid = 0x0c;
@@ -8085,6 +8119,8 @@ static int patch_alc883(struct hda_codec *codec)
#define alc262_dac_nids alc260_dac_nids
#define alc262_adc_nids alc882_adc_nids
#define alc262_adc_nids_alt alc882_adc_nids_alt
+#define alc262_capsrc_nids alc882_capsrc_nids
+#define alc262_capsrc_nids_alt alc882_capsrc_nids_alt
#define alc262_modes alc260_modes
#define alc262_capture_source alc882_capture_source
@@ -8585,7 +8621,8 @@ static void alc262_hippo1_unsol_event(struct hda_codec *codec,
/*
* fujitsu model
- * 0x14 = headphone/spdif-out, 0x15 = internal speaker
+ * 0x14 = headphone/spdif-out, 0x15 = internal speaker,
+ * 0x1b = port replicator headphone out
*/
#define ALC_HP_EVENT 0x37
@@ -8593,6 +8630,14 @@ static void alc262_hippo1_unsol_event(struct hda_codec *codec,
static struct hda_verb alc262_fujitsu_unsol_verbs[] = {
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {}
+};
+
+static struct hda_verb alc262_lenovo_3000_unsol_verbs[] = {
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{}
};
@@ -8633,12 +8678,16 @@ static void alc262_fujitsu_automute(struct hda_codec *codec, int force)
unsigned int mute;
if (force || !spec->sense_updated) {
- unsigned int present;
+ unsigned int present_int_hp, present_dock_hp;
/* need to execute and sync at first */
snd_hda_codec_read(codec, 0x14, 0, AC_VERB_SET_PIN_SENSE, 0);
- present = snd_hda_codec_read(codec, 0x14, 0,
- AC_VERB_GET_PIN_SENSE, 0);
- spec->jack_present = (present & 0x80000000) != 0;
+ present_int_hp = snd_hda_codec_read(codec, 0x14, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ snd_hda_codec_read(codec, 0x1B, 0, AC_VERB_SET_PIN_SENSE, 0);
+ present_dock_hp = snd_hda_codec_read(codec, 0x1b, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ spec->jack_present = (present_int_hp & 0x80000000) != 0;
+ spec->jack_present |= (present_dock_hp & 0x80000000) != 0;
spec->sense_updated = 1;
}
if (spec->jack_present) {
@@ -8672,6 +8721,46 @@ static struct hda_bind_ctls alc262_fujitsu_bind_master_vol = {
},
};
+/* mute/unmute internal speaker according to the hp jack and mute state */
+static void alc262_lenovo_3000_automute(struct hda_codec *codec, int force)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int mute;
+
+ if (force || !spec->sense_updated) {
+ unsigned int present_int_hp;
+ /* need to execute and sync at first */
+ snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0);
+ present_int_hp = snd_hda_codec_read(codec, 0x1b, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ spec->jack_present = (present_int_hp & 0x80000000) != 0;
+ spec->sense_updated = 1;
+ }
+ if (spec->jack_present) {
+ /* mute internal speaker */
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, HDA_AMP_MUTE);
+ snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, HDA_AMP_MUTE);
+ } else {
+ /* unmute internal speaker if necessary */
+ mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0);
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
+ snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
+ }
+}
+
+/* unsolicited event for HP jack sensing */
+static void alc262_lenovo_3000_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) != ALC_HP_EVENT)
+ return;
+ alc262_lenovo_3000_automute(codec, 1);
+}
+
/* bind hp and internal speaker mute (with plug check) */
static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -8680,12 +8769,13 @@ static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol,
long *valp = ucontrol->value.integer.value;
int change;
- change = snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
- HDA_AMP_MUTE,
- valp[0] ? 0 : HDA_AMP_MUTE);
- change |= snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
- HDA_AMP_MUTE,
- valp[1] ? 0 : HDA_AMP_MUTE);
+ change = snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE,
+ valp ? 0 : HDA_AMP_MUTE);
+ change |= snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE,
+ valp ? 0 : HDA_AMP_MUTE);
+
if (change)
alc262_fujitsu_automute(codec, 0);
return change;
@@ -8703,6 +8793,46 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = {
},
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("PC Speaker Volume", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Speaker Switch", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+/* bind hp and internal speaker mute (with plug check) */
+static int alc262_lenovo_3000_master_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ long *valp = ucontrol->value.integer.value;
+ int change;
+
+ change = snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE,
+ valp ? 0 : HDA_AMP_MUTE);
+
+ if (change)
+ alc262_lenovo_3000_automute(codec, 0);
+ return change;
+}
+
+static struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = {
+ HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = snd_hda_mixer_amp_switch_info,
+ .get = snd_hda_mixer_amp_switch_get,
+ .put = alc262_lenovo_3000_master_sw_put,
+ .private_value = HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
+ },
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
@@ -8730,59 +8860,72 @@ static struct hda_verb alc262_benq_t31_EAPD_verbs[] = {
/* Samsung Q1 Ultra Vista model setup */
static struct snd_kcontrol_new alc262_ultra_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Mic Boost", 0x15, 0, HDA_INPUT),
{ } /* end */
};
static struct hda_verb alc262_ultra_verbs[] = {
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ /* output mixer */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ /* speaker */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* HP */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
- /* Mic is on Node 0x19 */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
- {0x22, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
- {0x23, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
- {0x24, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ /* internal mic */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* ADC, choose mic */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(8)},
{}
};
-static struct hda_input_mux alc262_ultra_capture_source = {
- .num_items = 1,
- .items = {
- { "Mic", 0x1 },
- },
-};
-
/* mute/unmute internal speaker according to the hp jack and mute state */
static void alc262_ultra_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
unsigned int mute;
- unsigned int present;
- /* need to execute and sync at first */
- snd_hda_codec_read(codec, 0x15, 0, AC_VERB_SET_PIN_SENSE, 0);
- present = snd_hda_codec_read(codec, 0x15, 0,
- AC_VERB_GET_PIN_SENSE, 0);
- spec->jack_present = (present & 0x80000000) != 0;
- if (spec->jack_present) {
- /* mute internal speaker */
- snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, HDA_AMP_MUTE);
- } else {
- /* unmute internal speaker if necessary */
- mute = snd_hda_codec_amp_read(codec, 0x15, 0, HDA_OUTPUT, 0);
- snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, mute);
+ mute = 0;
+ /* auto-mute only when HP is used as HP */
+ if (!spec->cur_mux[0]) {
+ unsigned int present;
+ /* need to execute and sync at first */
+ snd_hda_codec_read(codec, 0x15, 0, AC_VERB_SET_PIN_SENSE, 0);
+ present = snd_hda_codec_read(codec, 0x15, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0;
+ if (spec->jack_present)
+ mute = HDA_AMP_MUTE;
}
+ /* mute/unmute internal speaker */
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
+ /* mute/unmute HP */
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute ? 0 : HDA_AMP_MUTE);
}
/* unsolicited event for HP jack sensing */
@@ -8794,6 +8937,45 @@ static void alc262_ultra_unsol_event(struct hda_codec *codec,
alc262_ultra_automute(codec);
}
+static struct hda_input_mux alc262_ultra_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic", 0x1 },
+ { "Headphone", 0x7 },
+ },
+};
+
+static int alc262_ultra_mux_enum_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 ret;
+
+ ret = alc882_mux_enum_put(kcontrol, ucontrol);
+ if (!ret)
+ return 0;
+ /* reprogram the HP pin as mic or HP according to the input source */
+ snd_hda_codec_write_cache(codec, 0x15, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ spec->cur_mux[0] ? PIN_VREF80 : PIN_HP);
+ alc262_ultra_automute(codec); /* mute/unmute HP */
+ return ret;
+}
+
+static struct snd_kcontrol_new alc262_ultra_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = alc882_mux_enum_info,
+ .get = alc882_mux_enum_get,
+ .put = alc262_ultra_mux_enum_put,
+ },
+ { } /* end */
+};
+
/* add playback controls from the parsed DAC table */
static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec,
const struct auto_pin_cfg *cfg)
@@ -9185,9 +9367,12 @@ static int alc262_parse_auto_config(struct hda_codec *codec)
/* init callback for auto-configuration model -- overriding the default init */
static void alc262_auto_init(struct hda_codec *codec)
{
+ struct alc_spec *spec = codec->spec;
alc262_auto_init_multi_out(codec);
alc262_auto_init_hp_out(codec);
alc262_auto_init_analog_input(codec);
+ if (spec->unsol_event)
+ alc_sku_automute(codec);
}
/*
@@ -9206,6 +9391,7 @@ static const char *alc262_models[ALC262_MODEL_LAST] = {
[ALC262_BENQ_T31] = "benq-t31",
[ALC262_SONY_ASSAMD] = "sony-assamd",
[ALC262_ULTRA] = "ultra",
+ [ALC262_LENOVO_3000] = "lenovo-3000",
[ALC262_AUTO] = "auto",
};
@@ -9241,6 +9427,8 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU),
SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU),
SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA),
+ SND_PCI_QUIRK(0x144d, 0xc039, "Samsung Q1U EL", ALC262_ULTRA),
+ SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000 y410", ALC262_LENOVO_3000),
SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8),
SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_BENQ_T31),
SND_PCI_QUIRK(0x17ff, 0x058f, "Benq Hippo", ALC262_HIPPO_1),
@@ -9390,18 +9578,32 @@ static struct alc_config_preset alc262_presets[] = {
.init_hook = alc262_hippo_automute,
},
[ALC262_ULTRA] = {
- .mixers = { alc262_ultra_mixer },
- .init_verbs = { alc262_init_verbs, alc262_ultra_verbs },
+ .mixers = { alc262_ultra_mixer, alc262_ultra_capture_mixer },
+ .init_verbs = { alc262_ultra_verbs },
.num_dacs = ARRAY_SIZE(alc262_dac_nids),
.dac_nids = alc262_dac_nids,
- .hp_nid = 0x03,
- .dig_out_nid = ALC262_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc262_modes),
.channel_mode = alc262_modes,
.input_mux = &alc262_ultra_capture_source,
+ .adc_nids = alc262_adc_nids, /* ADC0 */
+ .capsrc_nids = alc262_capsrc_nids,
+ .num_adc_nids = 1, /* single ADC */
.unsol_event = alc262_ultra_unsol_event,
.init_hook = alc262_ultra_automute,
},
+ [ALC262_LENOVO_3000] = {
+ .mixers = { alc262_lenovo_3000_mixer },
+ .init_verbs = { alc262_init_verbs, alc262_EAPD_verbs,
+ alc262_lenovo_3000_unsol_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x03,
+ .dig_out_nid = ALC262_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_fujitsu_capture_source,
+ .unsol_event = alc262_lenovo_3000_unsol_event,
+ },
};
static int patch_alc262(struct hda_codec *codec)
@@ -9472,12 +9674,14 @@ static int patch_alc262(struct hda_codec *codec)
if (wcap != AC_WID_AUD_IN) {
spec->adc_nids = alc262_adc_nids_alt;
spec->num_adc_nids = ARRAY_SIZE(alc262_adc_nids_alt);
+ spec->capsrc_nids = alc262_capsrc_nids_alt;
spec->mixers[spec->num_mixers] =
alc262_capture_alt_mixer;
spec->num_mixers++;
} else {
spec->adc_nids = alc262_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(alc262_adc_nids);
+ spec->capsrc_nids = alc262_capsrc_nids;
spec->mixers[spec->num_mixers] = alc262_capture_mixer;
spec->num_mixers++;
}
@@ -9517,6 +9721,8 @@ static hda_nid_t alc268_adc_nids_alt[1] = {
0x08
};
+static hda_nid_t alc268_capsrc_nids[2] = { 0x23, 0x24 };
+
static struct snd_kcontrol_new alc268_base_mixer[] = {
/* output mixer control */
HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
@@ -9529,6 +9735,22 @@ static struct snd_kcontrol_new alc268_base_mixer[] = {
{ }
};
+/* bind Beep switches of both NID 0x0f and 0x10 */
+static struct hda_bind_ctls alc268_bind_beep_sw = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x0f, 3, 1, HDA_INPUT),
+ HDA_COMPOSE_AMP_VAL(0x10, 3, 1, HDA_INPUT),
+ 0
+ },
+};
+
+static struct snd_kcontrol_new alc268_beep_mixer[] = {
+ HDA_CODEC_VOLUME("Beep Playback Volume", 0x1d, 0x0, HDA_INPUT),
+ HDA_BIND_SW("Beep Playback Switch", &alc268_bind_beep_sw),
+ { }
+};
+
static struct hda_verb alc268_eapd_verbs[] = {
{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
@@ -9613,8 +9835,12 @@ static struct snd_kcontrol_new alc268_acer_mixer[] = {
};
static struct hda_verb alc268_acer_verbs[] = {
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* internal dmic? */
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{ }
@@ -9685,6 +9911,64 @@ static void alc268_dell_unsol_event(struct hda_codec *codec,
#define alc268_dell_init_hook alc268_dell_automute
+static struct snd_kcontrol_new alc267_quanta_il1_mixer[] = {
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x2, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Capture Volume", 0x23, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Mic Capture Switch", 0x23, 2, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Ext Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
+ { }
+};
+
+static struct hda_verb alc267_quanta_il1_verbs[] = {
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN},
+ { }
+};
+
+static void alc267_quanta_il1_hp_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+ snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ present ? 0 : PIN_OUT);
+}
+
+static void alc267_quanta_il1_mic_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x18, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_write(codec, 0x23, 0,
+ AC_VERB_SET_CONNECT_SEL,
+ present ? 0x00 : 0x01);
+}
+
+static void alc267_quanta_il1_automute(struct hda_codec *codec)
+{
+ alc267_quanta_il1_hp_automute(codec);
+ alc267_quanta_il1_mic_automute(codec);
+}
+
+static void alc267_quanta_il1_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case ALC880_HP_EVENT:
+ alc267_quanta_il1_hp_automute(codec);
+ break;
+ case ALC880_MIC_EVENT:
+ alc267_quanta_il1_mic_automute(codec);
+ break;
+ }
+}
+
/*
* generic initialization of ADC, input mixers and output mixers
*/
@@ -9725,7 +10009,11 @@ static struct hda_verb alc268_base_init_verbs[] = {
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+
+ /* set PCBEEP vol = 0, mute connections */
+ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Unmute Selector 23h,24h and set the default input to mic-in */
@@ -9764,29 +10052,17 @@ static struct hda_verb alc268_volume_init_verbs[] = {
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* set PCBEEP vol = 0 */
- {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, (0xb000 | (0x00 << 8))},
+ /* set PCBEEP vol = 0, mute connections */
+ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{ }
};
#define alc268_mux_enum_info alc_mux_enum_info
#define alc268_mux_enum_get alc_mux_enum_get
-
-static int alc268_mux_enum_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;
-
- unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
- static hda_nid_t capture_mixers[3] = { 0x23, 0x24 };
- hda_nid_t nid = capture_mixers[adc_idx];
-
- return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
- nid,
- &spec->cur_mux[adc_idx]);
-}
+#define alc268_mux_enum_put alc_mux_enum_put
static struct snd_kcontrol_new alc268_capture_alt_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
@@ -9836,13 +10112,17 @@ static struct hda_input_mux alc268_capture_source = {
},
};
+static struct hda_input_mux alc268_acer_capture_source = {
+ .num_items = 3,
+ .items = {
+ { "Mic", 0x0 },
+ { "Internal Mic", 0x6 },
+ { "Line", 0x2 },
+ },
+};
+
#ifdef CONFIG_SND_DEBUG
static struct snd_kcontrol_new alc268_test_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
-
/* Volume widgets */
HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x03, 0x0, HDA_OUTPUT),
@@ -9981,6 +10261,10 @@ static int alc268_auto_create_analog_input_ctls(struct alc_spec *spec,
case 0x1c:
idx1 = 3; /* CD */
break;
+ case 0x12:
+ case 0x13:
+ idx1 = 6; /* digital mics */
+ break;
default:
continue;
}
@@ -10073,6 +10357,9 @@ static int alc268_parse_auto_config(struct hda_codec *codec)
if (spec->kctl_alloc)
spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+ if (spec->autocfg.speaker_pins[0] != 0x1d)
+ spec->mixers[spec->num_mixers++] = alc268_beep_mixer;
+
spec->init_verbs[spec->num_init_verbs++] = alc268_volume_init_verbs;
spec->num_mux_defs = 1;
spec->input_mux = &spec->private_imux;
@@ -10091,20 +10378,25 @@ static int alc268_parse_auto_config(struct hda_codec *codec)
/* init callback for auto-configuration model -- overriding the default init */
static void alc268_auto_init(struct hda_codec *codec)
{
+ struct alc_spec *spec = codec->spec;
alc268_auto_init_multi_out(codec);
alc268_auto_init_hp_out(codec);
alc268_auto_init_mono_speaker_out(codec);
alc268_auto_init_analog_input(codec);
+ if (spec->unsol_event)
+ alc_sku_automute(codec);
}
/*
* configuration and preset
*/
static const char *alc268_models[ALC268_MODEL_LAST] = {
+ [ALC267_QUANTA_IL1] = "quanta-il1",
[ALC268_3ST] = "3stack",
[ALC268_TOSHIBA] = "toshiba",
[ALC268_ACER] = "acer",
[ALC268_DELL] = "dell",
+ [ALC268_ZEPTO] = "zepto",
#ifdef CONFIG_SND_DEBUG
[ALC268_TEST] = "test",
#endif
@@ -10112,6 +10404,7 @@ static const char *alc268_models[ALC268_MODEL_LAST] = {
};
static struct snd_pci_quirk alc268_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x1025, 0x011e, "Acer Aspire 5720z", ALC268_ACER),
SND_PCI_QUIRK(0x1025, 0x0126, "Acer", ALC268_ACER),
SND_PCI_QUIRK(0x1025, 0x012e, "Acer Aspire 5310", ALC268_ACER),
SND_PCI_QUIRK(0x1025, 0x0130, "Acer Extensa 5210", ALC268_ACER),
@@ -10122,17 +10415,36 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = {
SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA),
SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA),
SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER),
+ SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1),
+ SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO),
{}
};
static struct alc_config_preset alc268_presets[] = {
+ [ALC267_QUANTA_IL1] = {
+ .mixers = { alc267_quanta_il1_mixer },
+ .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+ alc267_quanta_il1_verbs },
+ .num_dacs = ARRAY_SIZE(alc268_dac_nids),
+ .dac_nids = alc268_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+ .adc_nids = alc268_adc_nids_alt,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc268_modes),
+ .channel_mode = alc268_modes,
+ .input_mux = &alc268_capture_source,
+ .unsol_event = alc267_quanta_il1_unsol_event,
+ .init_hook = alc267_quanta_il1_automute,
+ },
[ALC268_3ST] = {
- .mixers = { alc268_base_mixer, alc268_capture_alt_mixer },
+ .mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
+ alc268_beep_mixer },
.init_verbs = { alc268_base_init_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
+ .capsrc_nids = alc268_capsrc_nids,
.hp_nid = 0x03,
.dig_out_nid = ALC268_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
@@ -10140,13 +10452,15 @@ static struct alc_config_preset alc268_presets[] = {
.input_mux = &alc268_capture_source,
},
[ALC268_TOSHIBA] = {
- .mixers = { alc268_base_mixer, alc268_capture_alt_mixer },
+ .mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
+ alc268_beep_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_toshiba_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
+ .capsrc_nids = alc268_capsrc_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
@@ -10155,22 +10469,24 @@ static struct alc_config_preset alc268_presets[] = {
.init_hook = alc268_toshiba_automute,
},
[ALC268_ACER] = {
- .mixers = { alc268_acer_mixer, alc268_capture_alt_mixer },
+ .mixers = { alc268_acer_mixer, alc268_capture_alt_mixer,
+ alc268_beep_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_acer_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
+ .capsrc_nids = alc268_capsrc_nids,
.hp_nid = 0x02,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
- .input_mux = &alc268_capture_source,
+ .input_mux = &alc268_acer_capture_source,
.unsol_event = alc268_acer_unsol_event,
.init_hook = alc268_acer_init_hook,
},
[ALC268_DELL] = {
- .mixers = { alc268_dell_mixer },
+ .mixers = { alc268_dell_mixer, alc268_beep_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_dell_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -10182,6 +10498,24 @@ static struct alc_config_preset alc268_presets[] = {
.init_hook = alc268_dell_init_hook,
.input_mux = &alc268_capture_source,
},
+ [ALC268_ZEPTO] = {
+ .mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
+ alc268_beep_mixer },
+ .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+ alc268_toshiba_verbs },
+ .num_dacs = ARRAY_SIZE(alc268_dac_nids),
+ .dac_nids = alc268_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+ .adc_nids = alc268_adc_nids_alt,
+ .capsrc_nids = alc268_capsrc_nids,
+ .hp_nid = 0x03,
+ .dig_out_nid = ALC268_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc268_modes),
+ .channel_mode = alc268_modes,
+ .input_mux = &alc268_capture_source,
+ .unsol_event = alc268_toshiba_unsol_event,
+ .init_hook = alc268_toshiba_automute
+ },
#ifdef CONFIG_SND_DEBUG
[ALC268_TEST] = {
.mixers = { alc268_test_mixer, alc268_capture_mixer },
@@ -10191,6 +10525,7 @@ static struct alc_config_preset alc268_presets[] = {
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
+ .capsrc_nids = alc268_capsrc_nids,
.hp_nid = 0x03,
.dig_out_nid = ALC268_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
@@ -10247,13 +10582,22 @@ static int patch_alc268(struct hda_codec *codec)
spec->stream_name_digital = "ALC268 Digital";
spec->stream_digital_playback = &alc268_pcm_digital_playback;
+ if (!query_amp_caps(codec, 0x1d, HDA_INPUT))
+ /* override the amp caps for beep generator */
+ snd_hda_override_amp_caps(codec, 0x1d, HDA_INPUT,
+ (0x0c << AC_AMPCAP_OFFSET_SHIFT) |
+ (0x0c << AC_AMPCAP_NUM_STEPS_SHIFT) |
+ (0x07 << AC_AMPCAP_STEP_SIZE_SHIFT) |
+ (0 << AC_AMPCAP_MUTE_SHIFT));
+
if (!spec->adc_nids && spec->input_mux) {
/* check whether NID 0x07 is valid */
unsigned int wcap = get_wcaps(codec, 0x07);
+ int i;
/* get type */
wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
- if (wcap != AC_WID_AUD_IN) {
+ if (wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) {
spec->adc_nids = alc268_adc_nids_alt;
spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt);
spec->mixers[spec->num_mixers] =
@@ -10266,6 +10610,12 @@ static int patch_alc268(struct hda_codec *codec)
alc268_capture_mixer;
spec->num_mixers++;
}
+ spec->capsrc_nids = alc268_capsrc_nids;
+ /* set default input source */
+ for (i = 0; i < spec->num_adc_nids; i++)
+ snd_hda_codec_write_cache(codec, alc268_capsrc_nids[i],
+ 0, AC_VERB_SET_CONNECT_SEL,
+ spec->input_mux->items[0].index);
}
spec->vmaster_nid = 0x02;
@@ -10539,9 +10889,12 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
/* init callback for auto-configuration model -- overriding the default init */
static void alc269_auto_init(struct hda_codec *codec)
{
+ struct alc_spec *spec = codec->spec;
alc269_auto_init_multi_out(codec);
alc269_auto_init_hp_out(codec);
alc269_auto_init_analog_input(codec);
+ if (spec->unsol_event)
+ alc_sku_automute(codec);
}
/*
@@ -11463,13 +11816,7 @@ static void alc861_auto_set_output_and_unmute(struct hda_codec *codec,
hda_nid_t nid,
int pin_type, int dac_idx)
{
- /* set as output */
-
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
- pin_type);
- snd_hda_codec_write(codec, dac_idx, 0, AC_VERB_SET_AMP_GAIN_MUTE,
- AMP_OUT_UNMUTE);
-
+ alc_set_pin_output(codec, nid, pin_type);
}
static void alc861_auto_init_multi_out(struct hda_codec *codec)
@@ -11496,6 +11843,9 @@ static void alc861_auto_init_hp_out(struct hda_codec *codec)
if (pin) /* connect to front */
alc861_auto_set_output_and_unmute(codec, pin, PIN_HP,
spec->multiout.dac_nids[0]);
+ pin = spec->autocfg.speaker_pins[0];
+ if (pin)
+ alc861_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
}
static void alc861_auto_init_analog_input(struct hda_codec *codec)
@@ -11568,9 +11918,12 @@ static int alc861_parse_auto_config(struct hda_codec *codec)
/* additional initialization for auto-configuration model */
static void alc861_auto_init(struct hda_codec *codec)
{
+ struct alc_spec *spec = codec->spec;
alc861_auto_init_multi_out(codec);
alc861_auto_init_hp_out(codec);
alc861_auto_init_analog_input(codec);
+ if (spec->unsol_event)
+ alc_sku_automute(codec);
}
#ifdef CONFIG_SND_HDA_POWER_SAVE
@@ -11822,6 +12175,8 @@ static hda_nid_t alc861vd_adc_nids[1] = {
0x09,
};
+static hda_nid_t alc861vd_capsrc_nids[1] = { 0x22 };
+
/* input MUX */
/* FIXME: should be a matrix-type input source selection */
static struct hda_input_mux alc861vd_capture_source = {
@@ -11835,11 +12190,10 @@ static struct hda_input_mux alc861vd_capture_source = {
};
static struct hda_input_mux alc861vd_dallas_capture_source = {
- .num_items = 3,
+ .num_items = 2,
.items = {
- { "Front Mic", 0x0 },
- { "ATAPI Mic", 0x1 },
- { "Line In", 0x5 },
+ { "Ext Mic", 0x0 },
+ { "Int Mic", 0x1 },
},
};
@@ -11853,33 +12207,8 @@ static struct hda_input_mux alc861vd_hp_capture_source = {
#define alc861vd_mux_enum_info alc_mux_enum_info
#define alc861vd_mux_enum_get alc_mux_enum_get
-
-static int alc861vd_mux_enum_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;
- const struct hda_input_mux *imux = spec->input_mux;
- unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
- static hda_nid_t capture_mixers[1] = { 0x22 };
- hda_nid_t nid = capture_mixers[adc_idx];
- unsigned int *cur_val = &spec->cur_mux[adc_idx];
- unsigned int i, idx;
-
- idx = ucontrol->value.enumerated.item[0];
- if (idx >= imux->num_items)
- idx = imux->num_items - 1;
- if (*cur_val == idx)
- return 0;
- for (i = 0; i < imux->num_items; i++) {
- unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE;
- snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT,
- imux->items[i].index,
- HDA_AMP_MUTE, v);
- }
- *cur_val = idx;
- return 1;
-}
+/* ALC861VD has the ALC882-type input selection (but has only one ADC) */
+#define alc861vd_mux_enum_put alc882_mux_enum_put
/*
* 2ch mode
@@ -12034,20 +12363,22 @@ static struct snd_kcontrol_new alc861vd_lenovo_mixer[] = {
{ } /* end */
};
-/* Pin assignment: Front=0x14, HP = 0x15,
- * Front Mic=0x18, ATAPI Mic = 0x19, Line In = 0x1d
+/* Pin assignment: Speaker=0x14, HP = 0x15,
+ * Ext Mic=0x18, Int Mic = 0x19, CD = 0x1c, PC Beep = 0x1d
*/
static struct snd_kcontrol_new alc861vd_dallas_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x05, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_VOLUME("Ext Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Ext Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Ext Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("PC Beep Volume", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Beep Switch", 0x0b, 0x05, HDA_INPUT),
{ } /* end */
};
@@ -12348,6 +12679,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = {
/*SND_PCI_QUIRK(0x1179, 0xff00, "DALLAS", ALC861VD_DALLAS),*/ /*lenovo*/
SND_PCI_QUIRK(0x1179, 0xff01, "DALLAS", ALC861VD_DALLAS),
SND_PCI_QUIRK(0x1179, 0xff03, "Toshiba P205", ALC861VD_LENOVO),
+ SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba L30-149", ALC861VD_DALLAS),
SND_PCI_QUIRK(0x1565, 0x820d, "Biostar NF61S SE", ALC861VD_6ST_DIG),
SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo", ALC861VD_LENOVO),
SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo 3000 C200", ALC861VD_LENOVO),
@@ -12362,8 +12694,6 @@ static struct alc_config_preset alc861vd_presets[] = {
alc861vd_3stack_init_verbs },
.num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
.dac_nids = alc660vd_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids),
- .adc_nids = alc861vd_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_capture_source,
@@ -12375,8 +12705,6 @@ static struct alc_config_preset alc861vd_presets[] = {
.num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
.dac_nids = alc660vd_dac_nids,
.dig_out_nid = ALC861VD_DIGOUT_NID,
- .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids),
- .adc_nids = alc861vd_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_capture_source,
@@ -12421,8 +12749,6 @@ static struct alc_config_preset alc861vd_presets[] = {
alc861vd_lenovo_unsol_verbs },
.num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
.dac_nids = alc660vd_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids),
- .adc_nids = alc861vd_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_capture_source,
@@ -12434,8 +12760,6 @@ static struct alc_config_preset alc861vd_presets[] = {
.init_verbs = { alc861vd_dallas_verbs },
.num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
.dac_nids = alc861vd_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids),
- .adc_nids = alc861vd_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_dallas_capture_source,
@@ -12447,9 +12771,7 @@ static struct alc_config_preset alc861vd_presets[] = {
.init_verbs = { alc861vd_dallas_verbs, alc861vd_eapd_verbs },
.num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
.dac_nids = alc861vd_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids),
.dig_out_nid = ALC861VD_DIGOUT_NID,
- .adc_nids = alc861vd_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_hp_capture_source,
@@ -12464,11 +12786,7 @@ static struct alc_config_preset alc861vd_presets[] = {
static void alc861vd_auto_set_output_and_unmute(struct hda_codec *codec,
hda_nid_t nid, int pin_type, int dac_idx)
{
- /* set as output */
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type);
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE);
+ alc_set_pin_output(codec, nid, pin_type);
}
static void alc861vd_auto_init_multi_out(struct hda_codec *codec)
@@ -12495,6 +12813,9 @@ static void alc861vd_auto_init_hp_out(struct hda_codec *codec)
pin = spec->autocfg.hp_pins[0];
if (pin) /* connect to front and use dac 0 */
alc861vd_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
+ pin = spec->autocfg.speaker_pins[0];
+ if (pin)
+ alc861vd_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
}
#define alc861vd_is_input_pin(nid) alc880_is_input_pin(nid)
@@ -12698,9 +13019,12 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec)
/* additional initialization for auto-configuration model */
static void alc861vd_auto_init(struct hda_codec *codec)
{
+ struct alc_spec *spec = codec->spec;
alc861vd_auto_init_multi_out(codec);
alc861vd_auto_init_hp_out(codec);
alc861vd_auto_init_analog_input(codec);
+ if (spec->unsol_event)
+ alc_sku_automute(codec);
}
static int patch_alc861vd(struct hda_codec *codec)
@@ -12751,6 +13075,7 @@ static int patch_alc861vd(struct hda_codec *codec)
spec->adc_nids = alc861vd_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids);
+ spec->capsrc_nids = alc861vd_capsrc_nids;
spec->mixers[spec->num_mixers] = alc861vd_capture_mixer;
spec->num_mixers++;
@@ -12792,9 +13117,11 @@ static hda_nid_t alc662_adc_nids[1] = {
/* ADC1-2 */
0x09,
};
+
+static hda_nid_t alc662_capsrc_nids[1] = { 0x22 };
+
/* input MUX */
/* FIXME: should be a matrix-type input source selection */
-
static struct hda_input_mux alc662_capture_source = {
.num_items = 4,
.items = {
@@ -12823,33 +13150,8 @@ static struct hda_input_mux alc662_eeepc_capture_source = {
#define alc662_mux_enum_info alc_mux_enum_info
#define alc662_mux_enum_get alc_mux_enum_get
+#define alc662_mux_enum_put alc882_mux_enum_put
-static int alc662_mux_enum_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;
- const struct hda_input_mux *imux = spec->input_mux;
- unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
- static hda_nid_t capture_mixers[2] = { 0x23, 0x22 };
- hda_nid_t nid = capture_mixers[adc_idx];
- unsigned int *cur_val = &spec->cur_mux[adc_idx];
- unsigned int i, idx;
-
- idx = ucontrol->value.enumerated.item[0];
- if (idx >= imux->num_items)
- idx = imux->num_items - 1;
- if (*cur_val == idx)
- return 0;
- for (i = 0; i < imux->num_items; i++) {
- unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE;
- snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT,
- imux->items[i].index,
- HDA_AMP_MUTE, v);
- }
- *cur_val = idx;
- return 1;
-}
/*
* 2ch mode
*/
@@ -12918,13 +13220,13 @@ static struct hda_channel_mode alc662_5stack_modes[2] = {
static struct snd_kcontrol_new alc662_base_mixer[] = {
/* output mixer control */
HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x3, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Surround Playback Switch", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Surround Playback Switch", 0x0d, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x04, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x04, 2, 2, HDA_INPUT),
+ HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
/*Input mixer control */
@@ -12941,7 +13243,7 @@ static struct snd_kcontrol_new alc662_base_mixer[] = {
static struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x02, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, 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),
@@ -12958,13 +13260,13 @@ static struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = {
static struct snd_kcontrol_new alc662_3ST_6ch_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x02, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x03, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Surround Playback Switch", 0x0d, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x04, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x04, 2, 2, HDA_INPUT),
+ HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x0, 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),
@@ -13313,6 +13615,7 @@ static const char *alc662_models[ALC662_MODEL_LAST] = {
};
static struct snd_pci_quirk alc662_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x1043, 0x8290, "ASUS P5GC-MX", ALC662_3ST_6ch_DIG),
SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701),
SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20),
SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E),
@@ -13326,8 +13629,6 @@ static struct alc_config_preset alc662_presets[] = {
.num_dacs = ARRAY_SIZE(alc662_dac_nids),
.dac_nids = alc662_dac_nids,
.dig_out_nid = ALC662_DIGOUT_NID,
- .num_adc_nids = ARRAY_SIZE(alc662_adc_nids),
- .adc_nids = alc662_adc_nids,
.dig_in_nid = ALC662_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.channel_mode = alc662_3ST_2ch_modes,
@@ -13340,8 +13641,6 @@ static struct alc_config_preset alc662_presets[] = {
.num_dacs = ARRAY_SIZE(alc662_dac_nids),
.dac_nids = alc662_dac_nids,
.dig_out_nid = ALC662_DIGOUT_NID,
- .num_adc_nids = ARRAY_SIZE(alc662_adc_nids),
- .adc_nids = alc662_adc_nids,
.dig_in_nid = ALC662_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
.channel_mode = alc662_3ST_6ch_modes,
@@ -13354,8 +13653,6 @@ static struct alc_config_preset alc662_presets[] = {
.init_verbs = { alc662_init_verbs },
.num_dacs = ARRAY_SIZE(alc662_dac_nids),
.dac_nids = alc662_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc662_adc_nids),
- .adc_nids = alc662_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
.channel_mode = alc662_3ST_6ch_modes,
.need_dac_fix = 1,
@@ -13368,8 +13665,6 @@ static struct alc_config_preset alc662_presets[] = {
.num_dacs = ARRAY_SIZE(alc662_dac_nids),
.dac_nids = alc662_dac_nids,
.dig_out_nid = ALC662_DIGOUT_NID,
- .num_adc_nids = ARRAY_SIZE(alc662_adc_nids),
- .adc_nids = alc662_adc_nids,
.dig_in_nid = ALC662_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc662_5stack_modes),
.channel_mode = alc662_5stack_modes,
@@ -13380,8 +13675,6 @@ static struct alc_config_preset alc662_presets[] = {
.init_verbs = { alc662_init_verbs, alc662_sue_init_verbs },
.num_dacs = ARRAY_SIZE(alc662_dac_nids),
.dac_nids = alc662_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc662_adc_nids),
- .adc_nids = alc662_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.channel_mode = alc662_3ST_2ch_modes,
.input_mux = &alc662_lenovo_101e_capture_source,
@@ -13394,8 +13687,6 @@ static struct alc_config_preset alc662_presets[] = {
alc662_eeepc_sue_init_verbs },
.num_dacs = ARRAY_SIZE(alc662_dac_nids),
.dac_nids = alc662_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids),
- .adc_nids = alc662_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.channel_mode = alc662_3ST_2ch_modes,
.input_mux = &alc662_eeepc_capture_source,
@@ -13409,8 +13700,6 @@ static struct alc_config_preset alc662_presets[] = {
alc662_eeepc_ep20_sue_init_verbs },
.num_dacs = ARRAY_SIZE(alc662_dac_nids),
.dac_nids = alc662_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc662_adc_nids),
- .adc_nids = alc662_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
.channel_mode = alc662_3ST_6ch_modes,
.input_mux = &alc662_lenovo_101e_capture_source,
@@ -13556,11 +13845,7 @@ static void alc662_auto_set_output_and_unmute(struct hda_codec *codec,
hda_nid_t nid, int pin_type,
int dac_idx)
{
- /* set as output */
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type);
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE);
+ alc_set_pin_output(codec, nid, pin_type);
/* need the manual connection? */
if (alc880_is_multi_pin(nid)) {
struct alc_spec *spec = codec->spec;
@@ -13595,6 +13880,9 @@ static void alc662_auto_init_hp_out(struct hda_codec *codec)
if (pin) /* connect to front */
/* use dac 0 */
alc662_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
+ pin = spec->autocfg.speaker_pins[0];
+ if (pin)
+ alc662_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
}
#define alc662_is_input_pin(nid) alc880_is_input_pin(nid)
@@ -13672,9 +13960,12 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
/* additional initialization for auto-configuration model */
static void alc662_auto_init(struct hda_codec *codec)
{
+ struct alc_spec *spec = codec->spec;
alc662_auto_init_multi_out(codec);
alc662_auto_init_hp_out(codec);
alc662_auto_init_analog_input(codec);
+ if (spec->unsol_event)
+ alc_sku_automute(codec);
}
static int patch_alc662(struct hda_codec *codec)
@@ -13722,10 +14013,9 @@ static int patch_alc662(struct hda_codec *codec)
spec->stream_digital_playback = &alc662_pcm_digital_playback;
spec->stream_digital_capture = &alc662_pcm_digital_capture;
- if (!spec->adc_nids && spec->input_mux) {
- spec->adc_nids = alc662_adc_nids;
- spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids);
- }
+ spec->adc_nids = alc662_adc_nids;
+ spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids);
+ spec->capsrc_nids = alc662_capsrc_nids;
spec->vmaster_nid = 0x02;
@@ -13761,6 +14051,8 @@ struct hda_codec_preset snd_hda_preset_realtek[] = {
{ .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 },
{ .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 },
{ .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc883 },
+ { .id = 0x10ec0885, .rev = 0x100103, .name = "ALC889A",
+ .patch = patch_alc882 }, /* should be patch_alc883() in future */
{ .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 },
{ .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc883 },
{ .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc883 },
diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c
index d22f5a6b850..9332b63e406 100644
--- a/sound/pci/hda/patch_si3054.c
+++ b/sound/pci/hda/patch_si3054.c
@@ -28,7 +28,7 @@
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"
-
+#include "hda_patch.h"
/* si3054 verbs */
#define SI3054_VERB_READ_NODE 0x900
@@ -206,7 +206,7 @@ static int si3054_build_pcms(struct hda_codec *codec)
info->name = "Si3054 Modem";
info->stream[SNDRV_PCM_STREAM_PLAYBACK] = si3054_pcm;
info->stream[SNDRV_PCM_STREAM_CAPTURE] = si3054_pcm;
- info->is_modem = 1;
+ info->pcm_type = HDA_PCM_TYPE_MODEM;
return 0;
}
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index caf48edaa92..b3a15d61687 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -32,6 +32,7 @@
#include <sound/asoundef.h>
#include "hda_codec.h"
#include "hda_local.h"
+#include "hda_patch.h"
#define NUM_CONTROL_ALLOC 32
#define STAC_PWR_EVENT 0x20
@@ -39,6 +40,7 @@
enum {
STAC_REF,
+ STAC_9200_OQO,
STAC_9200_DELL_D21,
STAC_9200_DELL_D22,
STAC_9200_DELL_D23,
@@ -50,6 +52,7 @@ enum {
STAC_9200_DELL_M26,
STAC_9200_DELL_M27,
STAC_9200_GATEWAY,
+ STAC_9200_PANASONIC,
STAC_9200_MODELS
};
@@ -63,11 +66,14 @@ enum {
enum {
STAC_92HD73XX_REF,
+ STAC_DELL_M6,
STAC_92HD73XX_MODELS
};
enum {
STAC_92HD71BXX_REF,
+ STAC_DELL_M4_1,
+ STAC_DELL_M4_2,
STAC_92HD71BXX_MODELS
};
@@ -123,6 +129,7 @@ struct sigmatel_spec {
unsigned int hp_detect: 1;
/* gpio lines */
+ unsigned int eapd_mask;
unsigned int gpio_mask;
unsigned int gpio_dir;
unsigned int gpio_data;
@@ -135,6 +142,7 @@ struct sigmatel_spec {
/* power management */
unsigned int num_pwrs;
hda_nid_t *pwr_nids;
+ hda_nid_t *dac_list;
/* playback */
struct hda_input_mux *mono_mux;
@@ -173,6 +181,7 @@ struct sigmatel_spec {
/* i/o switches */
unsigned int io_switch[2];
unsigned int clfe_swap;
+ unsigned int hp_switch;
unsigned int aloopback;
struct hda_pcm pcm_rec[2]; /* PCM information */
@@ -184,9 +193,6 @@ struct sigmatel_spec {
struct hda_input_mux private_dimux;
struct hda_input_mux private_imux;
struct hda_input_mux private_mono_mux;
-
- /* virtual master */
- unsigned int vmaster_tlv[4];
};
static hda_nid_t stac9200_adc_nids[1] = {
@@ -244,7 +250,7 @@ static hda_nid_t stac92hd71bxx_dmux_nids[1] = {
0x1c,
};
-static hda_nid_t stac92hd71bxx_dac_nids[2] = {
+static hda_nid_t stac92hd71bxx_dac_nids[1] = {
0x10, /*0x11, */
};
@@ -290,6 +296,10 @@ static hda_nid_t stac927x_mux_nids[3] = {
0x15, 0x16, 0x17
};
+static hda_nid_t stac927x_dac_nids[6] = {
+ 0x02, 0x03, 0x04, 0x05, 0x06, 0
+};
+
static hda_nid_t stac927x_dmux_nids[1] = {
0x1b,
};
@@ -331,10 +341,10 @@ static hda_nid_t stac922x_pin_nids[10] = {
0x0f, 0x10, 0x11, 0x15, 0x1b,
};
-static hda_nid_t stac92hd73xx_pin_nids[12] = {
+static hda_nid_t stac92hd73xx_pin_nids[13] = {
0x0a, 0x0b, 0x0c, 0x0d, 0x0e,
0x0f, 0x10, 0x11, 0x12, 0x13,
- 0x14, 0x22
+ 0x14, 0x1e, 0x22
};
static hda_nid_t stac92hd71bxx_pin_nids[10] = {
@@ -527,6 +537,43 @@ static struct hda_verb stac92hd73xx_6ch_core_init[] = {
{}
};
+static struct hda_verb dell_eq_core_init[] = {
+ /* set master volume to max value without distortion
+ * and direct control */
+ { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xec},
+ /* setup audio connections */
+ { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00},
+ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01},
+ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x02},
+ /* setup adcs to point to mixer */
+ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b},
+ { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b},
+ /* setup import muxs */
+ { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01},
+ { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01},
+ { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01},
+ { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {}
+};
+
+static struct hda_verb dell_m6_core_init[] = {
+ /* set master volume and direct control */
+ { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
+ /* setup audio connections */
+ { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00},
+ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01},
+ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x02},
+ /* setup adcs to point to mixer */
+ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b},
+ { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b},
+ /* setup import muxs */
+ { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01},
+ { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01},
+ { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01},
+ { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {}
+};
+
static struct hda_verb stac92hd73xx_8ch_core_init[] = {
/* set master volume and direct control */
{ 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
@@ -910,6 +957,11 @@ static int stac92xx_build_controls(struct hda_codec *codec)
err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
if (err < 0)
return err;
+ err = snd_hda_create_spdif_share_sw(codec,
+ &spec->multiout);
+ if (err < 0)
+ return err;
+ spec->multiout.share_spdif = 1;
}
if (spec->dig_in_nid) {
err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
@@ -919,10 +971,11 @@ 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, spec->vmaster_tlv);
+ HDA_OUTPUT, vmaster_tlv);
err = snd_hda_add_vmaster(codec, "Master Playback Volume",
- spec->vmaster_tlv, slave_vols);
+ vmaster_tlv, slave_vols);
if (err < 0)
return err;
}
@@ -1052,9 +1105,15 @@ static unsigned int dell9200_m27_pin_configs[8] = {
0x90170310, 0x04a11020, 0x90170310, 0x40f003fc,
};
+static unsigned int oqo9200_pin_configs[8] = {
+ 0x40c000f0, 0x404000f1, 0x0221121f, 0x02211210,
+ 0x90170111, 0x90a70120, 0x400000f2, 0x400000f3,
+};
+
static unsigned int *stac9200_brd_tbl[STAC_9200_MODELS] = {
[STAC_REF] = ref9200_pin_configs,
+ [STAC_9200_OQO] = oqo9200_pin_configs,
[STAC_9200_DELL_D21] = dell9200_d21_pin_configs,
[STAC_9200_DELL_D22] = dell9200_d22_pin_configs,
[STAC_9200_DELL_D23] = dell9200_d23_pin_configs,
@@ -1065,10 +1124,12 @@ static unsigned int *stac9200_brd_tbl[STAC_9200_MODELS] = {
[STAC_9200_DELL_M25] = dell9200_m25_pin_configs,
[STAC_9200_DELL_M26] = dell9200_m26_pin_configs,
[STAC_9200_DELL_M27] = dell9200_m27_pin_configs,
+ [STAC_9200_PANASONIC] = ref9200_pin_configs,
};
static const char *stac9200_models[STAC_9200_MODELS] = {
[STAC_REF] = "ref",
+ [STAC_9200_OQO] = "oqo",
[STAC_9200_DELL_D21] = "dell-d21",
[STAC_9200_DELL_D22] = "dell-d22",
[STAC_9200_DELL_D23] = "dell-d23",
@@ -1080,6 +1141,7 @@ static const char *stac9200_models[STAC_9200_MODELS] = {
[STAC_9200_DELL_M26] = "dell-m26",
[STAC_9200_DELL_M27] = "dell-m27",
[STAC_9200_GATEWAY] = "gateway",
+ [STAC_9200_PANASONIC] = "panasonic",
};
static struct snd_pci_quirk stac9200_cfg_tbl[] = {
@@ -1146,13 +1208,15 @@ static struct snd_pci_quirk stac9200_cfg_tbl[] = {
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f6,
"unknown Dell", STAC_9200_DELL_M26),
/* Panasonic */
- SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-74", STAC_REF),
+ SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-74", STAC_9200_PANASONIC),
/* Gateway machines needs EAPD to be set on resume */
SND_PCI_QUIRK(0x107b, 0x0205, "Gateway S-7110M", STAC_9200_GATEWAY),
SND_PCI_QUIRK(0x107b, 0x0317, "Gateway MT3423, MX341*",
STAC_9200_GATEWAY),
SND_PCI_QUIRK(0x107b, 0x0318, "Gateway ML3019, MT3707",
STAC_9200_GATEWAY),
+ /* OQO Mobile */
+ SND_PCI_QUIRK(0x1106, 0x3288, "OQO Model 2", STAC_9200_OQO),
{} /* terminator */
};
@@ -1202,24 +1266,48 @@ static struct snd_pci_quirk stac925x_cfg_tbl[] = {
{} /* terminator */
};
-static unsigned int ref92hd73xx_pin_configs[12] = {
+static unsigned int ref92hd73xx_pin_configs[13] = {
0x02214030, 0x02a19040, 0x01a19020, 0x02214030,
0x0181302e, 0x01014010, 0x01014020, 0x01014030,
0x02319040, 0x90a000f0, 0x90a000f0, 0x01452050,
+ 0x01452050,
+};
+
+static unsigned int dell_m6_pin_configs[13] = {
+ 0x0321101f, 0x4f00000f, 0x4f0000f0, 0x90170110,
+ 0x03a11020, 0x0321101f, 0x4f0000f0, 0x4f0000f0,
+ 0x4f0000f0, 0x90a60160, 0x4f0000f0, 0x4f0000f0,
+ 0x4f0000f0,
};
static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = {
- [STAC_92HD73XX_REF] = ref92hd73xx_pin_configs,
+ [STAC_92HD73XX_REF] = ref92hd73xx_pin_configs,
+ [STAC_DELL_M6] = dell_m6_pin_configs,
};
static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = {
[STAC_92HD73XX_REF] = "ref",
+ [STAC_DELL_M6] = "dell-m6",
};
static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
/* SigmaTel reference board */
SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
- "DFI LanParty", STAC_92HD73XX_REF),
+ "DFI LanParty", STAC_92HD73XX_REF),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0254,
+ "unknown Dell", STAC_DELL_M6),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0255,
+ "unknown Dell", STAC_DELL_M6),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0256,
+ "unknown Dell", STAC_DELL_M6),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0257,
+ "unknown Dell", STAC_DELL_M6),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x025e,
+ "unknown Dell", STAC_DELL_M6),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x025f,
+ "unknown Dell", STAC_DELL_M6),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0271,
+ "unknown Dell", STAC_DELL_M6),
{} /* terminator */
};
@@ -1229,18 +1317,56 @@ static unsigned int ref92hd71bxx_pin_configs[10] = {
0x90a000f0, 0x01452050,
};
+static unsigned int dell_m4_1_pin_configs[13] = {
+ 0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110,
+ 0x23a1902e, 0x23014250, 0x40f000f0, 0x90a000f0,
+ 0x40f000f0, 0x4f0000f0,
+};
+
+static unsigned int dell_m4_2_pin_configs[13] = {
+ 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110,
+ 0x23a1902e, 0x23014250, 0x40f000f0, 0x40f000f0,
+ 0x40f000f0, 0x044413b0,
+};
+
static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = {
[STAC_92HD71BXX_REF] = ref92hd71bxx_pin_configs,
+ [STAC_DELL_M4_1] = dell_m4_1_pin_configs,
+ [STAC_DELL_M4_2] = dell_m4_2_pin_configs,
};
static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = {
[STAC_92HD71BXX_REF] = "ref",
+ [STAC_DELL_M4_1] = "dell-m4-1",
+ [STAC_DELL_M4_2] = "dell-m4-2",
};
static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
/* SigmaTel reference board */
SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
"DFI LanParty", STAC_92HD71BXX_REF),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233,
+ "unknown Dell", STAC_DELL_M4_1),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0234,
+ "unknown Dell", STAC_DELL_M4_1),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0250,
+ "unknown Dell", STAC_DELL_M4_1),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x024f,
+ "unknown Dell", STAC_DELL_M4_1),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x024d,
+ "unknown Dell", STAC_DELL_M4_1),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0251,
+ "unknown Dell", STAC_DELL_M4_1),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0277,
+ "unknown Dell", STAC_DELL_M4_1),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0263,
+ "unknown Dell", STAC_DELL_M4_2),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0265,
+ "unknown Dell", STAC_DELL_M4_2),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0262,
+ "unknown Dell", STAC_DELL_M4_2),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0264,
+ "unknown Dell", STAC_DELL_M4_2),
{} /* terminator */
};
@@ -1733,7 +1859,8 @@ static int stac92xx_playback_pcm_open(struct hda_pcm_stream *hinfo,
struct snd_pcm_substream *substream)
{
struct sigmatel_spec *spec = codec->spec;
- return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream);
+ return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
+ hinfo);
}
static int stac92xx_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
@@ -1807,7 +1934,7 @@ static int stac92xx_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
{
struct sigmatel_spec *spec = codec->spec;
- snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], 0, 0, 0);
+ snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]);
return 0;
}
@@ -1889,6 +2016,7 @@ static int stac92xx_build_pcms(struct hda_codec *codec)
codec->num_pcms++;
info++;
info->name = "STAC92xx Digital";
+ info->pcm_type = HDA_PCM_TYPE_SPDIF;
if (spec->multiout.dig_out_nid) {
info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_digital_playback;
info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid;
@@ -1925,6 +2053,34 @@ static void stac92xx_auto_set_pinctl(struct hda_codec *codec, hda_nid_t nid, int
AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type);
}
+#define stac92xx_hp_switch_info snd_ctl_boolean_mono_info
+
+static int stac92xx_hp_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+
+ ucontrol->value.integer.value[0] = spec->hp_switch;
+ return 0;
+}
+
+static int stac92xx_hp_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+
+ spec->hp_switch = ucontrol->value.integer.value[0];
+
+ /* check to be sure that the ports are upto date with
+ * switch changes
+ */
+ codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26);
+
+ return 1;
+}
+
#define stac92xx_io_switch_info snd_ctl_boolean_mono_info
static int stac92xx_io_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -1996,6 +2152,15 @@ static int stac92xx_clfe_switch_put(struct snd_kcontrol *kcontrol,
return 1;
}
+#define STAC_CODEC_HP_SWITCH(xname) \
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = 0, \
+ .info = stac92xx_hp_switch_info, \
+ .get = stac92xx_hp_switch_get, \
+ .put = stac92xx_hp_switch_put, \
+ }
+
#define STAC_CODEC_IO_SWITCH(xname, xpval) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
.name = xname, \
@@ -2020,6 +2185,7 @@ enum {
STAC_CTL_WIDGET_VOL,
STAC_CTL_WIDGET_MUTE,
STAC_CTL_WIDGET_MONO_MUX,
+ STAC_CTL_WIDGET_HP_SWITCH,
STAC_CTL_WIDGET_IO_SWITCH,
STAC_CTL_WIDGET_CLFE_SWITCH
};
@@ -2028,6 +2194,7 @@ static struct snd_kcontrol_new stac92xx_control_templates[] = {
HDA_CODEC_VOLUME(NULL, 0, 0, 0),
HDA_CODEC_MUTE(NULL, 0, 0, 0),
STAC_MONO_MUX,
+ STAC_CODEC_HP_SWITCH(NULL),
STAC_CODEC_IO_SWITCH(NULL, 0),
STAC_CODEC_CLFE_SWITCH(NULL, 0),
};
@@ -2222,6 +2389,29 @@ static int create_controls(struct sigmatel_spec *spec, const char *pfx, hda_nid_
return 0;
}
+static int add_spec_dacs(struct sigmatel_spec *spec, hda_nid_t nid)
+{
+ if (!spec->multiout.hp_nid)
+ spec->multiout.hp_nid = nid;
+ else if (spec->multiout.num_dacs > 4) {
+ printk(KERN_WARNING "stac92xx: No space for DAC 0x%x\n", nid);
+ return 1;
+ } else {
+ spec->multiout.dac_nids[spec->multiout.num_dacs] = nid;
+ spec->multiout.num_dacs++;
+ }
+ return 0;
+}
+
+static int check_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid)
+{
+ if (is_in_dac_nids(spec, nid))
+ return 1;
+ if (spec->multiout.hp_nid == nid)
+ return 1;
+ return 0;
+}
+
/* add playback controls from the parsed DAC table */
static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec,
const struct auto_pin_cfg *cfg)
@@ -2236,7 +2426,7 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec,
unsigned int wid_caps, pincap;
- for (i = 0; i < cfg->line_outs; i++) {
+ for (i = 0; i < cfg->line_outs && i < spec->multiout.num_dacs; i++) {
if (!spec->multiout.dac_nids[i])
continue;
@@ -2269,6 +2459,14 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec,
}
}
+ if (cfg->hp_outs > 1) {
+ err = stac92xx_add_control(spec,
+ STAC_CTL_WIDGET_HP_SWITCH,
+ "Headphone as Line Out Switch", 0);
+ if (err < 0)
+ return err;
+ }
+
if (spec->line_switch) {
nid = cfg->input_pins[AUTO_PIN_LINE];
pincap = snd_hda_param_read(codec, nid,
@@ -2284,10 +2482,11 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec,
if (spec->mic_switch) {
unsigned int def_conf;
- nid = cfg->input_pins[AUTO_PIN_MIC];
+ unsigned int mic_pin = AUTO_PIN_MIC;
+again:
+ nid = cfg->input_pins[mic_pin];
def_conf = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_CONFIG_DEFAULT, 0);
-
/* some laptops have an internal analog microphone
* which can't be used as a output */
if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) {
@@ -2297,38 +2496,22 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec,
err = stac92xx_add_control(spec,
STAC_CTL_WIDGET_IO_SWITCH,
"Mic as Output Switch", (nid << 8) | 1);
+ nid = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_CONNECT_LIST, 0) & 0xff;
+ if (!check_in_dac_nids(spec, nid))
+ add_spec_dacs(spec, nid);
if (err < 0)
return err;
}
+ } else if (mic_pin == AUTO_PIN_MIC) {
+ mic_pin = AUTO_PIN_FRONT_MIC;
+ goto again;
}
}
return 0;
}
-static int check_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid)
-{
- if (is_in_dac_nids(spec, nid))
- return 1;
- if (spec->multiout.hp_nid == nid)
- return 1;
- return 0;
-}
-
-static int add_spec_dacs(struct sigmatel_spec *spec, hda_nid_t nid)
-{
- if (!spec->multiout.hp_nid)
- spec->multiout.hp_nid = nid;
- else if (spec->multiout.num_dacs > 4) {
- printk(KERN_WARNING "stac92xx: No space for DAC 0x%x\n", nid);
- return 1;
- } else {
- spec->multiout.dac_nids[spec->multiout.num_dacs] = nid;
- spec->multiout.num_dacs++;
- }
- return 0;
-}
-
/* add playback controls for Speaker and HP outputs */
static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec,
struct auto_pin_cfg *cfg)
@@ -2378,12 +2561,8 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec,
return err;
}
if (spec->multiout.hp_nid) {
- const char *pfx;
- if (old_num_dacs == spec->multiout.num_dacs)
- pfx = "Master";
- else
- pfx = "Headphone";
- err = create_controls(spec, pfx, spec->multiout.hp_nid, 3);
+ err = create_controls(spec, "Headphone",
+ spec->multiout.hp_nid, 3);
if (err < 0)
return err;
}
@@ -2745,7 +2924,7 @@ static int stac9200_auto_create_lfe_ctls(struct hda_codec *codec,
*/
for (i = 0; i < spec->autocfg.speaker_outs && lfe_pin == 0x0; i++) {
hda_nid_t pin = spec->autocfg.speaker_pins[i];
- unsigned long wcaps = get_wcaps(codec, pin);
+ unsigned int wcaps = get_wcaps(codec, pin);
wcaps &= (AC_WCAP_STEREO | AC_WCAP_OUT_AMP);
if (wcaps == AC_WCAP_OUT_AMP)
/* found a mono speaker with an amp, must be lfe */
@@ -2756,12 +2935,12 @@ static int stac9200_auto_create_lfe_ctls(struct hda_codec *codec,
if (lfe_pin == 0 && spec->autocfg.speaker_outs == 0) {
for (i = 0; i < spec->autocfg.line_outs && lfe_pin == 0x0; i++) {
hda_nid_t pin = spec->autocfg.line_out_pins[i];
- unsigned long cfg;
- cfg = snd_hda_codec_read(codec, pin, 0,
+ unsigned int defcfg;
+ defcfg = snd_hda_codec_read(codec, pin, 0,
AC_VERB_GET_CONFIG_DEFAULT,
0x00);
- if (get_defcfg_device(cfg) == AC_JACK_SPEAKER) {
- unsigned long wcaps = get_wcaps(codec, pin);
+ if (get_defcfg_device(defcfg) == AC_JACK_SPEAKER) {
+ unsigned int wcaps = get_wcaps(codec, pin);
wcaps &= (AC_WCAP_STEREO | AC_WCAP_OUT_AMP);
if (wcaps == AC_WCAP_OUT_AMP)
/* found a mono speaker with an amp,
@@ -2866,6 +3045,19 @@ static int is_nid_hp_pin(struct auto_pin_cfg *cfg, hda_nid_t nid)
return 0; /* nid is not a HP-Out */
};
+static void stac92xx_power_down(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec = codec->spec;
+
+ /* power down inactive DACs */
+ hda_nid_t *dac;
+ for (dac = spec->dac_list; *dac; dac++)
+ if (!is_in_dac_nids(spec, *dac) &&
+ spec->multiout.hp_nid != *dac)
+ snd_hda_codec_write_cache(codec, *dac, 0,
+ AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+}
+
static int stac92xx_init(struct hda_codec *codec)
{
struct sigmatel_spec *spec = codec->spec;
@@ -2909,16 +3101,21 @@ static int stac92xx_init(struct hda_codec *codec)
? STAC_HP_EVENT : STAC_PWR_EVENT;
int pinctl = snd_hda_codec_read(codec, spec->pwr_nids[i],
0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ int def_conf = snd_hda_codec_read(codec, spec->pwr_nids[i],
+ 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
/* outputs are only ports capable of power management
* any attempts on powering down a input port cause the
* referenced VREF to act quirky.
*/
if (pinctl & AC_PINCTL_IN_EN)
continue;
+ if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED)
+ continue;
enable_pin_detect(codec, spec->pwr_nids[i], event | i);
codec->patch_ops.unsol_event(codec, (event | i) << 26);
}
-
+ if (spec->dac_list)
+ stac92xx_power_down(codec);
if (cfg->dig_out_pin)
stac92xx_auto_set_pinctl(codec, cfg->dig_out_pin,
AC_PINCTL_OUT_EN);
@@ -3014,6 +3211,7 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res)
{
struct sigmatel_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
+ int nid = cfg->hp_pins[cfg->hp_outs - 1];
int i, presence;
presence = 0;
@@ -3024,26 +3222,42 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res)
for (i = 0; i < cfg->hp_outs; i++) {
if (presence)
break;
+ if (spec->hp_switch && cfg->hp_pins[i] == nid)
+ break;
presence = get_hp_pin_presence(codec, cfg->hp_pins[i]);
}
if (presence) {
/* disable lineouts, enable hp */
+ if (spec->hp_switch)
+ stac92xx_reset_pinctl(codec, nid, AC_PINCTL_OUT_EN);
for (i = 0; i < cfg->line_outs; i++)
stac92xx_reset_pinctl(codec, cfg->line_out_pins[i],
AC_PINCTL_OUT_EN);
for (i = 0; i < cfg->speaker_outs; i++)
stac92xx_reset_pinctl(codec, cfg->speaker_pins[i],
AC_PINCTL_OUT_EN);
+ if (spec->eapd_mask)
+ stac_gpio_set(codec, spec->gpio_mask,
+ spec->gpio_dir, spec->gpio_data &
+ ~spec->eapd_mask);
} else {
/* enable lineouts, disable hp */
+ if (spec->hp_switch)
+ stac92xx_set_pinctl(codec, nid, AC_PINCTL_OUT_EN);
for (i = 0; i < cfg->line_outs; i++)
stac92xx_set_pinctl(codec, cfg->line_out_pins[i],
AC_PINCTL_OUT_EN);
for (i = 0; i < cfg->speaker_outs; i++)
stac92xx_set_pinctl(codec, cfg->speaker_pins[i],
AC_PINCTL_OUT_EN);
+ if (spec->eapd_mask)
+ stac_gpio_set(codec, spec->gpio_mask,
+ spec->gpio_dir, spec->gpio_data |
+ spec->eapd_mask);
}
+ if (!spec->hp_switch && cfg->hp_outs > 1 && presence)
+ stac92xx_set_pinctl(codec, nid, AC_PINCTL_OUT_EN);
}
static void stac92xx_pin_sense(struct hda_codec *codec, int idx)
@@ -3091,6 +3305,9 @@ static int stac92xx_resume(struct hda_codec *codec)
spec->gpio_dir, spec->gpio_data);
snd_hda_codec_resume_amp(codec);
snd_hda_codec_resume_cache(codec);
+ /* power down inactive DACs */
+ if (spec->dac_list)
+ stac92xx_power_down(codec);
/* invoke unsolicited event to reset the HP state */
if (spec->hp_detect)
codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26);
@@ -3147,12 +3364,18 @@ static int patch_stac9200(struct hda_codec *codec)
spec->num_adcs = 1;
spec->num_pwrs = 0;
- if (spec->board_config == STAC_9200_GATEWAY)
+ if (spec->board_config == STAC_9200_GATEWAY ||
+ spec->board_config == STAC_9200_OQO)
spec->init = stac9200_eapd_init;
else
spec->init = stac9200_core_init;
spec->mixer = stac9200_mixer;
+ if (spec->board_config == STAC_9200_PANASONIC) {
+ spec->gpio_mask = spec->gpio_dir = 0x09;
+ spec->gpio_data = 0x00;
+ }
+
err = stac9200_parse_auto_config(codec);
if (err < 0) {
stac92xx_free(codec);
@@ -3293,6 +3516,7 @@ again:
switch (spec->multiout.num_dacs) {
case 0x3: /* 6 Channel */
+ spec->multiout.hp_nid = 0x17;
spec->mixer = stac92hd73xx_6ch_mixer;
spec->init = stac92hd73xx_6ch_core_init;
break;
@@ -3318,13 +3542,42 @@ again:
spec->num_muxes = ARRAY_SIZE(stac92hd73xx_mux_nids);
spec->num_adcs = ARRAY_SIZE(stac92hd73xx_adc_nids);
- spec->num_dmics = STAC92HD73XX_NUM_DMICS;
spec->num_dmuxes = ARRAY_SIZE(stac92hd73xx_dmux_nids);
spec->dinput_mux = &stac92hd73xx_dmux;
/* GPIO0 High = Enable EAPD */
- spec->gpio_mask = spec->gpio_dir = 0x1;
+ spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1;
spec->gpio_data = 0x01;
+ switch (spec->board_config) {
+ case STAC_DELL_M6:
+ spec->init = dell_eq_core_init;
+ switch (codec->subsystem_id) {
+ case 0x1028025e: /* Analog Mics */
+ case 0x1028025f:
+ stac92xx_set_config_reg(codec, 0x0b, 0x90A70170);
+ spec->num_dmics = 0;
+ break;
+ case 0x10280271: /* Digital Mics */
+ case 0x10280272:
+ spec->init = dell_m6_core_init;
+ /* fall-through */
+ case 0x10280254:
+ case 0x10280255:
+ stac92xx_set_config_reg(codec, 0x13, 0x90A60160);
+ spec->num_dmics = 1;
+ break;
+ case 0x10280256: /* Both */
+ case 0x10280057:
+ stac92xx_set_config_reg(codec, 0x0b, 0x90A70170);
+ stac92xx_set_config_reg(codec, 0x13, 0x90A60160);
+ spec->num_dmics = 1;
+ break;
+ }
+ break;
+ default:
+ spec->num_dmics = STAC92HD73XX_NUM_DMICS;
+ }
+
spec->num_pwrs = ARRAY_SIZE(stac92hd73xx_pwr_nids);
spec->pwr_nids = stac92hd73xx_pwr_nids;
@@ -3398,7 +3651,10 @@ again:
spec->aloopback_shift = 0;
/* GPIO0 High = EAPD */
- spec->gpio_mask = spec->gpio_dir = spec->gpio_data = 0x1;
+ spec->gpio_mask = 0x01;
+ spec->gpio_dir = 0x01;
+ spec->gpio_mask = 0x01;
+ spec->gpio_data = 0x01;
spec->mux_nids = stac92hd71bxx_mux_nids;
spec->adc_nids = stac92hd71bxx_adc_nids;
@@ -3413,7 +3669,7 @@ again:
spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids);
spec->pwr_nids = stac92hd71bxx_pwr_nids;
- spec->multiout.num_dacs = 2;
+ spec->multiout.num_dacs = 1;
spec->multiout.hp_nid = 0x11;
spec->multiout.dac_nids = stac92hd71bxx_dac_nids;
@@ -3577,13 +3833,14 @@ static int patch_stac927x(struct hda_codec *codec)
spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids);
spec->mux_nids = stac927x_mux_nids;
spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids);
+ spec->dac_list = stac927x_dac_nids;
spec->multiout.dac_nids = spec->dac_nids;
switch (spec->board_config) {
case STAC_D965_3ST:
case STAC_D965_5ST:
/* GPIO0 High = Enable EAPD */
- spec->gpio_mask = spec->gpio_dir = 0x01;
+ spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x01;
spec->gpio_data = 0x01;
spec->num_dmics = 0;
@@ -3591,14 +3848,23 @@ static int patch_stac927x(struct hda_codec *codec)
spec->mixer = stac927x_mixer;
break;
case STAC_DELL_BIOS:
+ switch (codec->subsystem_id) {
+ case 0x10280209:
+ case 0x1028022e:
+ /* correct the device field to SPDIF out */
+ stac92xx_set_config_reg(codec, 0x21, 0x01442070);
+ break;
+ };
+ /* configure the analog microphone on some laptops */
+ stac92xx_set_config_reg(codec, 0x0c, 0x90a79130);
/* correct the front output jack as a hp out */
- stac92xx_set_config_reg(codec, 0x0f, 0x02270110);
+ stac92xx_set_config_reg(codec, 0x0f, 0x0227011f);
/* correct the front input jack as a mic */
stac92xx_set_config_reg(codec, 0x0e, 0x02a79130);
/* fallthru */
case STAC_DELL_3ST:
/* GPIO2 High = Enable EAPD */
- spec->gpio_mask = spec->gpio_dir = 0x04;
+ spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x04;
spec->gpio_data = 0x04;
spec->dmic_nids = stac927x_dmic_nids;
spec->num_dmics = STAC927X_NUM_DMICS;
@@ -3610,7 +3876,7 @@ static int patch_stac927x(struct hda_codec *codec)
break;
default:
/* GPIO0 High = Enable EAPD */
- spec->gpio_mask = spec->gpio_dir = 0x1;
+ spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1;
spec->gpio_data = 0x01;
spec->num_dmics = 0;
@@ -3714,6 +3980,7 @@ static int patch_stac9205(struct hda_codec *codec)
(AC_USRSP_EN | STAC_HP_EVENT));
spec->gpio_dir = 0x0b;
+ spec->eapd_mask = 0x01;
spec->gpio_mask = 0x1b;
spec->gpio_mute = 0x10;
/* GPIO0 High = EAPD, GPIO1 Low = Headphone Mute,
@@ -3723,7 +3990,7 @@ static int patch_stac9205(struct hda_codec *codec)
break;
default:
/* GPIO0 High = EAPD */
- spec->gpio_mask = spec->gpio_dir = 0x1;
+ spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1;
spec->gpio_data = 0x01;
break;
}
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 4e5dd4cf36f..52b1d81a26f 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -39,7 +39,7 @@
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"
-
+#include "hda_patch.h"
/* amp values */
#define AMP_VAL_IDX_SHIFT 19
@@ -357,7 +357,8 @@ static int via_playback_pcm_open(struct hda_pcm_stream *hinfo,
struct snd_pcm_substream *substream)
{
struct via_spec *spec = codec->spec;
- return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream);
+ return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
+ hinfo);
}
static int via_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
@@ -430,8 +431,7 @@ static int via_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
struct snd_pcm_substream *substream)
{
struct via_spec *spec = codec->spec;
- snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number],
- 0, 0, 0);
+ snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]);
return 0;
}
@@ -493,6 +493,11 @@ static int via_build_controls(struct hda_codec *codec)
spec->multiout.dig_out_nid);
if (err < 0)
return err;
+ err = snd_hda_create_spdif_share_sw(codec,
+ &spec->multiout);
+ if (err < 0)
+ return err;
+ spec->multiout.share_spdif = 1;
}
if (spec->dig_in_nid) {
err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
@@ -523,6 +528,7 @@ static int via_build_pcms(struct hda_codec *codec)
codec->num_pcms++;
info++;
info->name = spec->stream_name_digital;
+ info->pcm_type = HDA_PCM_TYPE_SPDIF;
if (spec->multiout.dig_out_nid) {
info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
*(spec->stream_digital_playback);
diff --git a/sound/pci/hda/vmaster.c b/sound/pci/hda/vmaster.c
deleted file mode 100644
index 2da49d20a1f..00000000000
--- a/sound/pci/hda/vmaster.c
+++ /dev/null
@@ -1,364 +0,0 @@
-/*
- * Virtual master and slave controls
- *
- * Copyright (c) 2008 by Takashi Iwai <tiwai@suse.de>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License as
- * published by the Free Software Foundation, version 2.
- *
- */
-
-#include <linux/slab.h>
-#include <sound/core.h>
-#include <sound/control.h>
-
-/*
- * a subset of information returned via ctl info callback
- */
-struct link_ctl_info {
- int type; /* value type */
- int count; /* item count */
- int min_val, max_val; /* min, max values */
-};
-
-/*
- * link master - this contains a list of slave controls that are
- * identical types, i.e. info returns the same value type and value
- * ranges, but may have different number of counts.
- *
- * The master control is so far only mono volume/switch for simplicity.
- * The same value will be applied to all slaves.
- */
-struct link_master {
- struct list_head slaves;
- struct link_ctl_info info;
- int val; /* the master value */
-};
-
-/*
- * link slave - this contains a slave control element
- *
- * It fakes the control callbacsk with additional attenuation by the
- * master control. A slave may have either one or two channels.
- */
-
-struct link_slave {
- struct list_head list;
- struct link_master *master;
- struct link_ctl_info info;
- int vals[2]; /* current values */
- struct snd_kcontrol slave; /* the copy of original control entry */
-};
-
-/* get the slave ctl info and save the initial values */
-static int slave_init(struct link_slave *slave)
-{
- struct snd_ctl_elem_info *uinfo;
- struct snd_ctl_elem_value *uctl;
- int err, ch;
-
- if (slave->info.count)
- return 0; /* already initialized */
-
- uinfo = kmalloc(sizeof(*uinfo), GFP_KERNEL);
- if (!uinfo)
- return -ENOMEM;
- uinfo->id = slave->slave.id;
- err = slave->slave.info(&slave->slave, uinfo);
- if (err < 0) {
- kfree(uinfo);
- return err;
- }
- slave->info.type = uinfo->type;
- slave->info.count = uinfo->count;
- if (slave->info.count > 2 ||
- (slave->info.type != SNDRV_CTL_ELEM_TYPE_INTEGER &&
- slave->info.type != SNDRV_CTL_ELEM_TYPE_BOOLEAN)) {
- snd_printk(KERN_ERR "invalid slave element\n");
- kfree(uinfo);
- return -EINVAL;
- }
- slave->info.min_val = uinfo->value.integer.min;
- slave->info.max_val = uinfo->value.integer.max;
- kfree(uinfo);
-
- uctl = kmalloc(sizeof(*uctl), GFP_KERNEL);
- if (!uctl)
- return -ENOMEM;
- uctl->id = slave->slave.id;
- err = slave->slave.get(&slave->slave, uctl);
- for (ch = 0; ch < slave->info.count; ch++)
- slave->vals[ch] = uctl->value.integer.value[ch];
- kfree(uctl);
- return 0;
-}
-
-/* initialize master volume */
-static int master_init(struct link_master *master)
-{
- struct link_slave *slave;
-
- if (master->info.count)
- return 0; /* already initialized */
-
- list_for_each_entry(slave, &master->slaves, list) {
- int err = slave_init(slave);
- if (err < 0)
- return err;
- master->info = slave->info;
- master->info.count = 1; /* always mono */
- /* set full volume as default (= no attenuation) */
- master->val = master->info.max_val;
- return 0;
- }
- return -ENOENT;
-}
-
-static int slave_get_val(struct link_slave *slave,
- struct snd_ctl_elem_value *ucontrol)
-{
- int err, ch;
-
- err = slave_init(slave);
- if (err < 0)
- return err;
- for (ch = 0; ch < slave->info.count; ch++)
- ucontrol->value.integer.value[ch] = slave->vals[ch];
- return 0;
-}
-
-static int slave_put_val(struct link_slave *slave,
- struct snd_ctl_elem_value *ucontrol)
-{
- int err, ch, vol;
-
- err = master_init(slave->master);
- if (err < 0)
- return err;
-
- switch (slave->info.type) {
- case SNDRV_CTL_ELEM_TYPE_BOOLEAN:
- for (ch = 0; ch < slave->info.count; ch++)
- ucontrol->value.integer.value[ch] &=
- !!slave->master->val;
- break;
- case SNDRV_CTL_ELEM_TYPE_INTEGER:
- for (ch = 0; ch < slave->info.count; ch++) {
- /* max master volume is supposed to be 0 dB */
- vol = ucontrol->value.integer.value[ch];
- vol += slave->master->val - slave->master->info.max_val;
- if (vol < slave->info.min_val)
- vol = slave->info.min_val;
- else if (vol > slave->info.max_val)
- vol = slave->info.max_val;
- ucontrol->value.integer.value[ch] = vol;
- }
- break;
- }
- return slave->slave.put(&slave->slave, ucontrol);
-}
-
-/*
- * ctl callbacks for slaves
- */
-static int slave_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- struct link_slave *slave = snd_kcontrol_chip(kcontrol);
- return slave->slave.info(&slave->slave, uinfo);
-}
-
-static int slave_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct link_slave *slave = snd_kcontrol_chip(kcontrol);
- return slave_get_val(slave, ucontrol);
-}
-
-static int slave_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct link_slave *slave = snd_kcontrol_chip(kcontrol);
- int err, ch, changed = 0;
-
- err = slave_init(slave);
- if (err < 0)
- return err;
- for (ch = 0; ch < slave->info.count; ch++) {
- if (slave->vals[ch] != ucontrol->value.integer.value[ch]) {
- changed = 1;
- slave->vals[ch] = ucontrol->value.integer.value[ch];
- }
- }
- if (!changed)
- return 0;
- return slave_put_val(slave, ucontrol);
-}
-
-static int slave_tlv_cmd(struct snd_kcontrol *kcontrol,
- int op_flag, unsigned int size,
- unsigned int __user *tlv)
-{
- struct link_slave *slave = snd_kcontrol_chip(kcontrol);
- /* FIXME: this assumes that the max volume is 0 dB */
- return slave->slave.tlv.c(&slave->slave, op_flag, size, tlv);
-}
-
-static void slave_free(struct snd_kcontrol *kcontrol)
-{
- struct link_slave *slave = snd_kcontrol_chip(kcontrol);
- if (slave->slave.private_free)
- slave->slave.private_free(&slave->slave);
- if (slave->master)
- list_del(&slave->list);
- kfree(slave);
-}
-
-/*
- * Add a slave control to the group with the given master control
- *
- * All slaves must be the same type (returning the same information
- * via info callback). The fucntion doesn't check it, so it's your
- * responsibility.
- *
- * Also, some additional limitations:
- * - at most two channels
- * - logarithmic volume control (dB level), no linear volume
- * - master can only attenuate the volume, no gain
- */
-int snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave)
-{
- struct link_master *master_link = snd_kcontrol_chip(master);
- struct link_slave *srec;
-
- srec = kzalloc(sizeof(*srec) +
- slave->count * sizeof(*slave->vd), GFP_KERNEL);
- if (!srec)
- return -ENOMEM;
- srec->slave = *slave;
- memcpy(srec->slave.vd, slave->vd, slave->count * sizeof(*slave->vd));
- srec->master = master_link;
-
- /* override callbacks */
- slave->info = slave_info;
- slave->get = slave_get;
- slave->put = slave_put;
- if (slave->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK)
- slave->tlv.c = slave_tlv_cmd;
- slave->private_data = srec;
- slave->private_free = slave_free;
-
- list_add_tail(&srec->list, &master_link->slaves);
- return 0;
-}
-
-/*
- * ctl callbacks for master controls
- */
-static int master_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- struct link_master *master = snd_kcontrol_chip(kcontrol);
- int ret;
-
- ret = master_init(master);
- if (ret < 0)
- return ret;
- uinfo->type = master->info.type;
- uinfo->count = master->info.count;
- uinfo->value.integer.min = master->info.min_val;
- uinfo->value.integer.max = master->info.max_val;
- return 0;
-}
-
-static int master_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct link_master *master = snd_kcontrol_chip(kcontrol);
- int err = master_init(master);
- if (err < 0)
- return err;
- ucontrol->value.integer.value[0] = master->val;
- return 0;
-}
-
-static int master_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct link_master *master = snd_kcontrol_chip(kcontrol);
- struct link_slave *slave;
- struct snd_ctl_elem_value *uval;
- int err, old_val;
-
- err = master_init(master);
- if (err < 0)
- return err;
- old_val = master->val;
- if (ucontrol->value.integer.value[0] == old_val)
- return 0;
-
- uval = kmalloc(sizeof(*uval), GFP_KERNEL);
- if (!uval)
- return -ENOMEM;
- list_for_each_entry(slave, &master->slaves, list) {
- master->val = old_val;
- uval->id = slave->slave.id;
- slave_get_val(slave, uval);
- master->val = ucontrol->value.integer.value[0];
- slave_put_val(slave, uval);
- }
- kfree(uval);
- return 1;
-}
-
-static void master_free(struct snd_kcontrol *kcontrol)
-{
- struct link_master *master = snd_kcontrol_chip(kcontrol);
- struct link_slave *slave;
-
- list_for_each_entry(slave, &master->slaves, list)
- slave->master = NULL;
- kfree(master);
-}
-
-
-/*
- * Create a virtual master control with the given name
- */
-struct snd_kcontrol *snd_ctl_make_virtual_master(char *name,
- const unsigned int *tlv)
-{
- struct link_master *master;
- struct snd_kcontrol *kctl;
- struct snd_kcontrol_new knew;
-
- memset(&knew, 0, sizeof(knew));
- knew.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
- knew.name = name;
- knew.info = master_info;
-
- master = kzalloc(sizeof(*master), GFP_KERNEL);
- if (!master)
- return NULL;
- INIT_LIST_HEAD(&master->slaves);
-
- kctl = snd_ctl_new1(&knew, master);
- if (!kctl) {
- kfree(master);
- return NULL;
- }
- /* override some callbacks */
- kctl->info = master_info;
- kctl->get = master_get;
- kctl->put = master_put;
- kctl->private_free = master_free;
-
- /* additional (constant) TLV read */
- if (tlv) {
- /* FIXME: this assumes that the max volume is 0 dB */
- kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ;
- kctl->tlv.p = tlv;
- }
- return kctl;
-}
diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c
index efd180b40e5..0ed96c17805 100644
--- a/sound/pci/ice1712/delta.c
+++ b/sound/pci/ice1712/delta.c
@@ -1,8 +1,8 @@
/*
* ALSA driver for ICEnsemble ICE1712 (Envy24)
*
- * Lowlevel functions for M-Audio Delta 1010, 44, 66, Dio2496, Audiophile
- * Digigram VX442
+ * Lowlevel functions for M-Audio Delta 1010, 1010E, 44, 66, 66E, Dio2496,
+ * Audiophile, Digigram VX442
*
* Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
*
@@ -86,6 +86,7 @@ static unsigned char ap_cs8427_codec_select(struct snd_ice1712 *ice)
unsigned char tmp;
tmp = snd_ice1712_read(ice, ICE1712_IREG_GPIO_DATA);
switch (ice->eeprom.subvendor) {
+ case ICE1712_SUBDEVICE_DELTA1010E:
case ICE1712_SUBDEVICE_DELTA1010LT:
tmp &= ~ICE1712_DELTA_1010LT_CS;
tmp |= ICE1712_DELTA_1010LT_CCLK | ICE1712_DELTA_1010LT_CS_CS8427;
@@ -109,6 +110,7 @@ static unsigned char ap_cs8427_codec_select(struct snd_ice1712 *ice)
static void ap_cs8427_codec_deassert(struct snd_ice1712 *ice, unsigned char tmp)
{
switch (ice->eeprom.subvendor) {
+ case ICE1712_SUBDEVICE_DELTA1010E:
case ICE1712_SUBDEVICE_DELTA1010LT:
tmp &= ~ICE1712_DELTA_1010LT_CS;
tmp |= ICE1712_DELTA_1010LT_CS_NONE;
@@ -534,6 +536,14 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice)
int err;
struct snd_akm4xxx *ak;
+ if (ice->eeprom.subvendor == ICE1712_SUBDEVICE_DELTA1010 &&
+ ice->eeprom.gpiodir == 0x7b)
+ ice->eeprom.subvendor = ICE1712_SUBDEVICE_DELTA1010E;
+
+ if (ice->eeprom.subvendor == ICE1712_SUBDEVICE_DELTA66 &&
+ ice->eeprom.gpiodir == 0xfb)
+ ice->eeprom.subvendor = ICE1712_SUBDEVICE_DELTA66E;
+
/* determine I2C, DACs and ADCs */
switch (ice->eeprom.subvendor) {
case ICE1712_SUBDEVICE_AUDIOPHILE:
@@ -550,6 +560,7 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice)
ice->num_total_adcs = ice->omni ? 8 : 4;
break;
case ICE1712_SUBDEVICE_DELTA1010:
+ case ICE1712_SUBDEVICE_DELTA1010E:
case ICE1712_SUBDEVICE_DELTA1010LT:
case ICE1712_SUBDEVICE_MEDIASTATION:
ice->num_total_dacs = 8;
@@ -559,6 +570,7 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice)
ice->num_total_dacs = 4; /* two AK4324 codecs */
break;
case ICE1712_SUBDEVICE_VX442:
+ case ICE1712_SUBDEVICE_DELTA66E: /* omni not suported yet */
ice->num_total_dacs = 4;
ice->num_total_adcs = 4;
break;
@@ -568,8 +580,10 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice)
switch (ice->eeprom.subvendor) {
case ICE1712_SUBDEVICE_AUDIOPHILE:
case ICE1712_SUBDEVICE_DELTA410:
+ case ICE1712_SUBDEVICE_DELTA1010E:
case ICE1712_SUBDEVICE_DELTA1010LT:
case ICE1712_SUBDEVICE_VX442:
+ case ICE1712_SUBDEVICE_DELTA66E:
if ((err = snd_i2c_bus_create(ice->card, "ICE1712 GPIO 1", NULL, &ice->i2c)) < 0) {
snd_printk(KERN_ERR "unable to create I2C bus\n");
return err;
@@ -601,6 +615,7 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice)
/* no analog? */
switch (ice->eeprom.subvendor) {
case ICE1712_SUBDEVICE_DELTA1010:
+ case ICE1712_SUBDEVICE_DELTA1010E:
case ICE1712_SUBDEVICE_DELTADIO2496:
case ICE1712_SUBDEVICE_MEDIASTATION:
return 0;
@@ -627,6 +642,7 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice)
err = snd_ice1712_akm4xxx_init(ak, &akm_delta44, &akm_delta44_priv, ice);
break;
case ICE1712_SUBDEVICE_VX442:
+ case ICE1712_SUBDEVICE_DELTA66E:
err = snd_ice1712_akm4xxx_init(ak, &akm_vx442, &akm_vx442_priv, ice);
break;
default:
@@ -674,6 +690,7 @@ static int __devinit snd_ice1712_delta_add_controls(struct snd_ice1712 *ice)
if (err < 0)
return err;
break;
+ case ICE1712_SUBDEVICE_DELTA1010E:
case ICE1712_SUBDEVICE_DELTA1010LT:
err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_delta1010lt_wordclock_select, ice));
if (err < 0)
@@ -716,6 +733,7 @@ static int __devinit snd_ice1712_delta_add_controls(struct snd_ice1712 *ice)
case ICE1712_SUBDEVICE_DELTA44:
case ICE1712_SUBDEVICE_DELTA66:
case ICE1712_SUBDEVICE_VX442:
+ case ICE1712_SUBDEVICE_DELTA66E:
err = snd_ice1712_akm4xxx_build_controls(ice);
if (err < 0)
return err;
diff --git a/sound/pci/ice1712/delta.h b/sound/pci/ice1712/delta.h
index 26ea05a32f5..ea7116c304c 100644
--- a/sound/pci/ice1712/delta.h
+++ b/sound/pci/ice1712/delta.h
@@ -36,8 +36,10 @@
"{Lionstracs,Mediastation},"
#define ICE1712_SUBDEVICE_DELTA1010 0x121430d6
+#define ICE1712_SUBDEVICE_DELTA1010E 0xff1430d6
#define ICE1712_SUBDEVICE_DELTADIO2496 0x121431d6
#define ICE1712_SUBDEVICE_DELTA66 0x121432d6
+#define ICE1712_SUBDEVICE_DELTA66E 0xff1432d6
#define ICE1712_SUBDEVICE_DELTA44 0x121433d6
#define ICE1712_SUBDEVICE_AUDIOPHILE 0x121434d6
#define ICE1712_SUBDEVICE_DELTA410 0x121438d6
diff --git a/sound/pci/ice1712/ews.c b/sound/pci/ice1712/ews.c
index 064760d2a02..013fc4f0482 100644
--- a/sound/pci/ice1712/ews.c
+++ b/sound/pci/ice1712/ews.c
@@ -238,6 +238,7 @@ static void snd_ice1712_ews_cs8404_spdif_write(struct snd_ice1712 *ice, unsigned
case ICE1712_SUBDEVICE_EWS88MT:
case ICE1712_SUBDEVICE_EWS88MT_NEW:
case ICE1712_SUBDEVICE_PHASE88:
+ case ICE1712_SUBDEVICE_TS88:
if (snd_i2c_sendbytes(spec->i2cdevs[EWS_I2C_CS8404], &bits, 1)
!= 1)
goto _error;
@@ -433,6 +434,7 @@ static int __devinit snd_ice1712_ews_init(struct snd_ice1712 *ice)
case ICE1712_SUBDEVICE_EWS88MT:
case ICE1712_SUBDEVICE_EWS88MT_NEW:
case ICE1712_SUBDEVICE_PHASE88:
+ case ICE1712_SUBDEVICE_TS88:
ice->num_total_dacs = 8;
ice->num_total_adcs = 8;
break;
@@ -475,6 +477,8 @@ static int __devinit snd_ice1712_ews_init(struct snd_ice1712 *ice)
case ICE1712_SUBDEVICE_EWS88MT:
case ICE1712_SUBDEVICE_EWS88MT_NEW:
case ICE1712_SUBDEVICE_PHASE88:
+ case ICE1712_SUBDEVICE_TS88:
+
err = snd_i2c_device_create(ice->i2c, "CS8404",
ICE1712_EWS88MT_CS8404_ADDR,
&spec->i2cdevs[EWS_I2C_CS8404]);
@@ -518,6 +522,7 @@ static int __devinit snd_ice1712_ews_init(struct snd_ice1712 *ice)
case ICE1712_SUBDEVICE_EWS88MT:
case ICE1712_SUBDEVICE_EWS88MT_NEW:
case ICE1712_SUBDEVICE_PHASE88:
+ case ICE1712_SUBDEVICE_TS88:
case ICE1712_SUBDEVICE_EWS88D:
/* set up CS8404 */
ice->spdif.ops.open = ews88_open_spdif;
@@ -547,6 +552,7 @@ static int __devinit snd_ice1712_ews_init(struct snd_ice1712 *ice)
case ICE1712_SUBDEVICE_EWS88MT:
case ICE1712_SUBDEVICE_EWS88MT_NEW:
case ICE1712_SUBDEVICE_PHASE88:
+ case ICE1712_SUBDEVICE_TS88:
err = snd_ice1712_akm4xxx_init(ak, &akm_ews88mt, &akm_ews88mt_priv, ice);
break;
case ICE1712_SUBDEVICE_EWX2496:
@@ -973,6 +979,7 @@ static int __devinit snd_ice1712_ews_add_controls(struct snd_ice1712 *ice)
case ICE1712_SUBDEVICE_EWS88MT:
case ICE1712_SUBDEVICE_EWS88MT_NEW:
case ICE1712_SUBDEVICE_PHASE88:
+ case ICE1712_SUBDEVICE_TS88:
case ICE1712_SUBDEVICE_DMX6FIRE:
err = snd_ice1712_akm4xxx_build_controls(ice);
if (err < 0)
@@ -992,6 +999,7 @@ static int __devinit snd_ice1712_ews_add_controls(struct snd_ice1712 *ice)
case ICE1712_SUBDEVICE_EWS88MT:
case ICE1712_SUBDEVICE_EWS88MT_NEW:
case ICE1712_SUBDEVICE_PHASE88:
+ case ICE1712_SUBDEVICE_TS88:
err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_ews88mt_input_sense, ice));
if (err < 0)
return err;
@@ -1049,6 +1057,13 @@ struct snd_ice1712_card_info snd_ice1712_ews_cards[] __devinitdata = {
.build_controls = snd_ice1712_ews_add_controls,
},
{
+ .subvendor = ICE1712_SUBDEVICE_TS88,
+ .name = "terrasoniq TS88",
+ .model = "phase88",
+ .chip_init = snd_ice1712_ews_init,
+ .build_controls = snd_ice1712_ews_add_controls,
+ },
+ {
.subvendor = ICE1712_SUBDEVICE_EWS88D,
.name = "TerraTec EWS88D",
.model = "ews88d",
diff --git a/sound/pci/ice1712/ews.h b/sound/pci/ice1712/ews.h
index e4ed1b475b0..1c443718af0 100644
--- a/sound/pci/ice1712/ews.h
+++ b/sound/pci/ice1712/ews.h
@@ -30,7 +30,8 @@
"{TerraTec,EWS 88MT},"\
"{TerraTec,EWS 88D},"\
"{TerraTec,DMX 6Fire},"\
- "{TerraTec,Phase 88},"
+ "{TerraTec,Phase 88}," \
+ "{terrasoniq,TS 88},"
#define ICE1712_SUBDEVICE_EWX2496 0x3b153011
#define ICE1712_SUBDEVICE_EWS88MT 0x3b151511
@@ -38,6 +39,7 @@
#define ICE1712_SUBDEVICE_EWS88D 0x3b152b11
#define ICE1712_SUBDEVICE_DMX6FIRE 0x3b153811
#define ICE1712_SUBDEVICE_PHASE88 0x3b155111
+#define ICE1712_SUBDEVICE_TS88 0x3b157c11
/* entry point */
extern struct snd_ice1712_card_info snd_ice1712_ews_cards[];
diff --git a/sound/pci/ice1712/hoontech.c b/sound/pci/ice1712/hoontech.c
index cf5c7c0898f..6914189073a 100644
--- a/sound/pci/ice1712/hoontech.c
+++ b/sound/pci/ice1712/hoontech.c
@@ -208,6 +208,19 @@ static int __devinit snd_ice1712_hoontech_init(struct snd_ice1712 *ice)
/* ICE1712_STDSP24_MUTE |
ICE1712_STDSP24_INSEL |
ICE1712_STDSP24_DAREAR; */
+ /* These boxconfigs have caused problems in the past.
+ * The code is not optimal, but should now enable a working config to
+ * be achieved.
+ * ** MIDI IN can only be configured on one box **
+ * ICE1712_STDSP24_BOX_MIDI1 needs to be set for that box.
+ * Tests on a ADAC2000 box suggest the box config flags do not
+ * work as would be expected, and the inputs are crossed.
+ * Setting ICE1712_STDSP24_BOX_MIDI1 and ICE1712_STDSP24_BOX_MIDI2
+ * on the same box connects MIDI-In to both 401 uarts; both outputs
+ * are then active on all boxes.
+ * The default config here sets up everything on the first box.
+ * Alan Horstmann 5.2.2008
+ */
spec->boxconfig[0] = ICE1712_STDSP24_BOX_CHN1 |
ICE1712_STDSP24_BOX_CHN2 |
ICE1712_STDSP24_BOX_CHN3 |
@@ -223,14 +236,14 @@ static int __devinit snd_ice1712_hoontech_init(struct snd_ice1712 *ice)
(spec->config & ICE1712_STDSP24_MUTE) ? 1 : 0);
snd_ice1712_stdsp24_insel(ice,
(spec->config & ICE1712_STDSP24_INSEL) ? 1 : 0);
- for (box = 0; box < 1; box++) {
+ for (box = 0; box < 4; box++) {
if (spec->boxconfig[box] & ICE1712_STDSP24_BOX_MIDI2)
snd_ice1712_stdsp24_midi2(ice, 1);
for (chn = 0; chn < 4; chn++)
snd_ice1712_stdsp24_box_channel(ice, box, chn,
(spec->boxconfig[box] & (1 << chn)) ? 1 : 0);
- snd_ice1712_stdsp24_box_midi(ice, box,
- (spec->boxconfig[box] & ICE1712_STDSP24_BOX_MIDI1) ? 1 : 0);
+ if (spec->boxconfig[box] & ICE1712_STDSP24_BOX_MIDI1)
+ snd_ice1712_stdsp24_box_midi(ice, box, 1);
}
return 0;
@@ -322,6 +335,8 @@ struct snd_ice1712_card_info snd_ice1712_hoontech_cards[] __devinitdata = {
.name = "Hoontech SoundTrack Audio DSP24",
.model = "dsp24",
.chip_init = snd_ice1712_hoontech_init,
+ .mpu401_1_name = "MIDI-1 Hoontech/STA DSP24",
+ .mpu401_2_name = "MIDI-2 Hoontech/STA DSP24",
},
{
.subvendor = ICE1712_SUBDEVICE_STDSP24_VALUE, /* a dummy id */
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index df292af6738..29d449d73c9 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -1297,11 +1297,14 @@ static void snd_ice1712_update_volume(struct snd_ice1712 *ice, int index)
static int snd_ice1712_pro_mixer_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
- int index = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + kcontrol->private_value;
+ int priv_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) +
+ kcontrol->private_value;
spin_lock_irq(&ice->reg_lock);
- ucontrol->value.integer.value[0] = !((ice->pro_volumes[index] >> 15) & 1);
- ucontrol->value.integer.value[1] = !((ice->pro_volumes[index] >> 31) & 1);
+ ucontrol->value.integer.value[0] =
+ !((ice->pro_volumes[priv_idx] >> 15) & 1);
+ ucontrol->value.integer.value[1] =
+ !((ice->pro_volumes[priv_idx] >> 31) & 1);
spin_unlock_irq(&ice->reg_lock);
return 0;
}
@@ -1309,16 +1312,17 @@ static int snd_ice1712_pro_mixer_switch_get(struct snd_kcontrol *kcontrol, struc
static int snd_ice1712_pro_mixer_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
- int index = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + kcontrol->private_value;
+ int priv_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) +
+ kcontrol->private_value;
unsigned int nval, change;
nval = (ucontrol->value.integer.value[0] ? 0 : 0x00008000) |
(ucontrol->value.integer.value[1] ? 0 : 0x80000000);
spin_lock_irq(&ice->reg_lock);
- nval |= ice->pro_volumes[index] & ~0x80008000;
- change = nval != ice->pro_volumes[index];
- ice->pro_volumes[index] = nval;
- snd_ice1712_update_volume(ice, index);
+ nval |= ice->pro_volumes[priv_idx] & ~0x80008000;
+ change = nval != ice->pro_volumes[priv_idx];
+ ice->pro_volumes[priv_idx] = nval;
+ snd_ice1712_update_volume(ice, priv_idx);
spin_unlock_irq(&ice->reg_lock);
return change;
}
@@ -1335,11 +1339,14 @@ static int snd_ice1712_pro_mixer_volume_info(struct snd_kcontrol *kcontrol, stru
static int snd_ice1712_pro_mixer_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
- int index = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + kcontrol->private_value;
+ int priv_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) +
+ kcontrol->private_value;
spin_lock_irq(&ice->reg_lock);
- ucontrol->value.integer.value[0] = (ice->pro_volumes[index] >> 0) & 127;
- ucontrol->value.integer.value[1] = (ice->pro_volumes[index] >> 16) & 127;
+ ucontrol->value.integer.value[0] =
+ (ice->pro_volumes[priv_idx] >> 0) & 127;
+ ucontrol->value.integer.value[1] =
+ (ice->pro_volumes[priv_idx] >> 16) & 127;
spin_unlock_irq(&ice->reg_lock);
return 0;
}
@@ -1347,16 +1354,17 @@ static int snd_ice1712_pro_mixer_volume_get(struct snd_kcontrol *kcontrol, struc
static int snd_ice1712_pro_mixer_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
- int index = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + kcontrol->private_value;
+ int priv_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) +
+ kcontrol->private_value;
unsigned int nval, change;
nval = (ucontrol->value.integer.value[0] & 127) |
((ucontrol->value.integer.value[1] & 127) << 16);
spin_lock_irq(&ice->reg_lock);
- nval |= ice->pro_volumes[index] & ~0x007f007f;
- change = nval != ice->pro_volumes[index];
- ice->pro_volumes[index] = nval;
- snd_ice1712_update_volume(ice, index);
+ nval |= ice->pro_volumes[priv_idx] & ~0x007f007f;
+ change = nval != ice->pro_volumes[priv_idx];
+ ice->pro_volumes[priv_idx] = nval;
+ snd_ice1712_update_volume(ice, priv_idx);
spin_unlock_irq(&ice->reg_lock);
return change;
}
@@ -2482,10 +2490,9 @@ static int snd_ice1712_free(struct snd_ice1712 *ice)
outb(0xff, ICEREG(ice, IRQMASK));
/* --- */
__hw_end:
- if (ice->irq >= 0) {
- synchronize_irq(ice->irq);
+ if (ice->irq >= 0)
free_irq(ice->irq, ice);
- }
+
if (ice->port)
pci_release_regions(ice->pci);
snd_ice1712_akm4xxx_free(ice);
diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h
index 303cffe08bd..3208901c740 100644
--- a/sound/pci/ice1712/ice1712.h
+++ b/sound/pci/ice1712/ice1712.h
@@ -367,6 +367,15 @@ struct snd_ice1712 {
/* other board-specific data */
void *spec;
+
+ /* VT172x specific */
+ int pro_rate_default;
+ int (*is_spdif_master)(struct snd_ice1712 *ice);
+ unsigned int (*get_rate)(struct snd_ice1712 *ice);
+ void (*set_rate)(struct snd_ice1712 *ice, unsigned int rate);
+ unsigned char (*set_mclk)(struct snd_ice1712 *ice, unsigned int rate);
+ void (*set_spdif_clock)(struct snd_ice1712 *ice);
+
};
@@ -429,10 +438,14 @@ int snd_ice1712_gpio_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_valu
static inline void snd_ice1712_gpio_write_bits(struct snd_ice1712 *ice,
unsigned int mask, unsigned int bits)
{
+ unsigned val;
+
ice->gpio.direction |= mask;
snd_ice1712_gpio_set_dir(ice, ice->gpio.direction);
- snd_ice1712_gpio_set_mask(ice, ~mask);
- snd_ice1712_gpio_write(ice, mask & bits);
+ val = snd_ice1712_gpio_read(ice);
+ val &= ~mask;
+ val |= mask & bits;
+ snd_ice1712_gpio_write(ice, val);
}
static inline int snd_ice1712_gpio_read_bits(struct snd_ice1712 *ice,
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index f533850ec6e..4490422fb93 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -106,15 +106,19 @@ static unsigned int PRO_RATE_DEFAULT = 44100;
* Basic I/O
*/
+/*
+ * default rates, default clock routines
+ */
+
/* check whether the clock mode is spdif-in */
-static inline int is_spdif_master(struct snd_ice1712 *ice)
+static inline int stdclock_is_spdif_master(struct snd_ice1712 *ice)
{
return (inb(ICEMT1724(ice, RATE)) & VT1724_SPDIF_MASTER) ? 1 : 0;
}
static inline int is_pro_rate_locked(struct snd_ice1712 *ice)
{
- return is_spdif_master(ice) || PRO_RATE_LOCKED;
+ return ice->is_spdif_master(ice) || PRO_RATE_LOCKED;
}
/*
@@ -219,6 +223,32 @@ static unsigned int snd_vt1724_get_gpio_data(struct snd_ice1712 *ice)
}
/*
+ * MPU401 accessor
+ */
+static unsigned char snd_vt1724_mpu401_read(struct snd_mpu401 *mpu,
+ unsigned long addr)
+{
+ /* fix status bits to the standard position */
+ /* only RX_EMPTY and TX_FULL are checked */
+ if (addr == MPU401C(mpu))
+ return (inb(addr) & 0x0c) << 4;
+ else
+ return inb(addr);
+}
+
+static void snd_vt1724_mpu401_write(struct snd_mpu401 *mpu,
+ unsigned char data, unsigned long addr)
+{
+ if (addr == MPU401C(mpu)) {
+ if (data == MPU401_ENTER_UART)
+ outb(0x01, addr);
+ /* what else? */
+ } else
+ outb(data, addr);
+}
+
+
+/*
* Interrupt handler
*/
@@ -226,24 +256,53 @@ static irqreturn_t snd_vt1724_interrupt(int irq, void *dev_id)
{
struct snd_ice1712 *ice = dev_id;
unsigned char status;
+ unsigned char status_mask =
+ VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX | VT1724_IRQ_MTPCM;
int handled = 0;
+#ifdef CONFIG_SND_DEBUG
+ int timeout = 0;
+#endif
while (1) {
status = inb(ICEREG1724(ice, IRQSTAT));
+ status &= status_mask;
if (status == 0)
break;
-
+#ifdef CONFIG_SND_DEBUG
+ if (++timeout > 10) {
+ printk(KERN_ERR
+ "ice1724: Too long irq loop, status = 0x%x\n",
+ status);
+ break;
+ }
+#endif
handled = 1;
- /* these should probably be separated at some point,
- * but as we don't currently have MPU support on the board
- * I will leave it
- */
- if ((status & VT1724_IRQ_MPU_RX)||(status & VT1724_IRQ_MPU_TX)) {
+ if (status & VT1724_IRQ_MPU_TX) {
+ if (ice->rmidi[0])
+ snd_mpu401_uart_interrupt_tx(irq,
+ ice->rmidi[0]->private_data);
+ else /* disable TX to be sure */
+ outb(inb(ICEREG1724(ice, IRQMASK)) |
+ VT1724_IRQ_MPU_TX,
+ ICEREG1724(ice, IRQMASK));
+ /* Due to mysterical reasons, MPU_TX is always
+ * generated (and can't be cleared) when a PCM
+ * playback is going. So let's ignore at the
+ * next loop.
+ */
+ status_mask &= ~VT1724_IRQ_MPU_TX;
+ }
+ if (status & VT1724_IRQ_MPU_RX) {
if (ice->rmidi[0])
- snd_mpu401_uart_interrupt(irq, ice->rmidi[0]->private_data);
- outb(status & (VT1724_IRQ_MPU_RX|VT1724_IRQ_MPU_TX), ICEREG1724(ice, IRQSTAT));
- status &= ~(VT1724_IRQ_MPU_RX|VT1724_IRQ_MPU_TX);
+ snd_mpu401_uart_interrupt(irq,
+ ice->rmidi[0]->private_data);
+ else /* disable RX to be sure */
+ outb(inb(ICEREG1724(ice, IRQMASK)) |
+ VT1724_IRQ_MPU_RX,
+ ICEREG1724(ice, IRQMASK));
}
+ /* ack MPU irq */
+ outb(status, ICEREG1724(ice, IRQSTAT));
if (status & VT1724_IRQ_MTPCM) {
/*
* Multi-track PCM
@@ -391,51 +450,61 @@ static int snd_vt1724_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
#define DMA_PAUSES (VT1724_RDMA0_PAUSE|VT1724_PDMA0_PAUSE|VT1724_RDMA1_PAUSE|\
VT1724_PDMA1_PAUSE|VT1724_PDMA2_PAUSE|VT1724_PDMA3_PAUSE|VT1724_PDMA4_PAUSE)
-static int get_max_rate(struct snd_ice1712 *ice)
+static const unsigned int stdclock_rate_list[16] = {
+ 48000, 24000, 12000, 9600, 32000, 16000, 8000, 96000, 44100,
+ 22050, 11025, 88200, 176400, 0, 192000, 64000
+};
+
+static unsigned int stdclock_get_rate(struct snd_ice1712 *ice)
{
+ unsigned int rate;
+ rate = stdclock_rate_list[inb(ICEMT1724(ice, RATE)) & 15];
+ return rate;
+}
+
+static void stdclock_set_rate(struct snd_ice1712 *ice, unsigned int rate)
+{
+ int i;
+ for (i = 0; i < ARRAY_SIZE(stdclock_rate_list); i++) {
+ if (stdclock_rate_list[i] == rate) {
+ outb(i, ICEMT1724(ice, RATE));
+ return;
+ }
+ }
+}
+
+static unsigned char stdclock_set_mclk(struct snd_ice1712 *ice,
+ unsigned int rate)
+{
+ unsigned char val, old;
+ /* check MT02 */
if (ice->eeprom.data[ICE_EEP2_ACLINK] & VT1724_CFG_PRO_I2S) {
- if ((ice->eeprom.data[ICE_EEP2_I2S] & 0x08) && !ice->vt1720)
- return 192000;
+ val = old = inb(ICEMT1724(ice, I2S_FORMAT));
+ if (rate > 96000)
+ val |= VT1724_MT_I2S_MCLK_128X; /* 128x MCLK */
else
- return 96000;
- } else
- return 48000;
+ val &= ~VT1724_MT_I2S_MCLK_128X; /* 256x MCLK */
+ if (val != old) {
+ outb(val, ICEMT1724(ice, I2S_FORMAT));
+ /* master clock changed */
+ return 1;
+ }
+ }
+ /* no change in master clock */
+ return 0;
}
static void snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate,
int force)
{
unsigned long flags;
- unsigned char val, old;
- unsigned int i, mclk_change;
+ unsigned char mclk_change;
+ unsigned int i, old_rate;
- if (rate > get_max_rate(ice))
+ if (rate > ice->hw_rates->list[ice->hw_rates->count - 1])
return;
-
- switch (rate) {
- case 8000: val = 6; break;
- case 9600: val = 3; break;
- case 11025: val = 10; break;
- case 12000: val = 2; break;
- case 16000: val = 5; break;
- case 22050: val = 9; break;
- case 24000: val = 1; break;
- case 32000: val = 4; break;
- case 44100: val = 8; break;
- case 48000: val = 0; break;
- case 64000: val = 15; break;
- case 88200: val = 11; break;
- case 96000: val = 7; break;
- case 176400: val = 12; break;
- case 192000: val = 14; break;
- default:
- snd_BUG();
- val = 0;
- break;
- }
-
spin_lock_irqsave(&ice->reg_lock, flags);
- if ((inb(ICEMT1724(ice, DMA_CONTROL)) & DMA_STARTS) ||
+ if ((inb(ICEMT1724(ice, DMA_CONTROL)) & DMA_STARTS) ||
(inb(ICEMT1724(ice, DMA_PAUSE)) & DMA_PAUSES)) {
/* running? we cannot change the rate now... */
spin_unlock_irqrestore(&ice->reg_lock, flags);
@@ -446,9 +515,9 @@ static void snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate,
return;
}
- old = inb(ICEMT1724(ice, RATE));
- if (force || old != val)
- outb(val, ICEMT1724(ice, RATE));
+ old_rate = ice->get_rate(ice);
+ if (force || (old_rate != rate))
+ ice->set_rate(ice, rate);
else if (rate == ice->cur_rate) {
spin_unlock_irqrestore(&ice->reg_lock, flags);
return;
@@ -456,19 +525,9 @@ static void snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate,
ice->cur_rate = rate;
- /* check MT02 */
- mclk_change = 0;
- if (ice->eeprom.data[ICE_EEP2_ACLINK] & VT1724_CFG_PRO_I2S) {
- val = old = inb(ICEMT1724(ice, I2S_FORMAT));
- if (rate > 96000)
- val |= VT1724_MT_I2S_MCLK_128X; /* 128x MCLK */
- else
- val &= ~VT1724_MT_I2S_MCLK_128X; /* 256x MCLK */
- if (val != old) {
- outb(val, ICEMT1724(ice, I2S_FORMAT));
- mclk_change = 1;
- }
- }
+ /* setting master clock */
+ mclk_change = ice->set_mclk(ice, rate);
+
spin_unlock_irqrestore(&ice->reg_lock, flags);
if (mclk_change && ice->gpio.i2s_mclk_changed)
@@ -727,43 +786,32 @@ static const struct snd_pcm_hardware snd_vt1724_2ch_stereo =
/*
* set rate constraints
*/
-static int set_rate_constraints(struct snd_ice1712 *ice,
- struct snd_pcm_substream *substream)
+static void set_std_hw_rates(struct snd_ice1712 *ice)
{
- struct snd_pcm_runtime *runtime = substream->runtime;
- if (ice->hw_rates) {
- /* hardware specific */
- runtime->hw.rate_min = ice->hw_rates->list[0];
- runtime->hw.rate_max = ice->hw_rates->list[ice->hw_rates->count - 1];
- runtime->hw.rates = SNDRV_PCM_RATE_KNOT;
- return snd_pcm_hw_constraint_list(runtime, 0,
- SNDRV_PCM_HW_PARAM_RATE,
- ice->hw_rates);
- }
if (ice->eeprom.data[ICE_EEP2_ACLINK] & VT1724_CFG_PRO_I2S) {
/* I2S */
/* VT1720 doesn't support more than 96kHz */
if ((ice->eeprom.data[ICE_EEP2_I2S] & 0x08) && !ice->vt1720)
- return snd_pcm_hw_constraint_list(runtime, 0,
- SNDRV_PCM_HW_PARAM_RATE,
- &hw_constraints_rates_192);
- else {
- runtime->hw.rates = SNDRV_PCM_RATE_KNOT |
- SNDRV_PCM_RATE_8000_96000;
- runtime->hw.rate_max = 96000;
- return snd_pcm_hw_constraint_list(runtime, 0,
- SNDRV_PCM_HW_PARAM_RATE,
- &hw_constraints_rates_96);
- }
- } else if (ice->ac97) {
+ ice->hw_rates = &hw_constraints_rates_192;
+ else
+ ice->hw_rates = &hw_constraints_rates_96;
+ } else {
/* ACLINK */
- runtime->hw.rate_max = 48000;
- runtime->hw.rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000;
- return snd_pcm_hw_constraint_list(runtime, 0,
- SNDRV_PCM_HW_PARAM_RATE,
- &hw_constraints_rates_48);
+ ice->hw_rates = &hw_constraints_rates_48;
}
- return 0;
+}
+
+static int set_rate_constraints(struct snd_ice1712 *ice,
+ struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ runtime->hw.rate_min = ice->hw_rates->list[0];
+ runtime->hw.rate_max = ice->hw_rates->list[ice->hw_rates->count - 1];
+ runtime->hw.rates = SNDRV_PCM_RATE_KNOT;
+ return snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ ice->hw_rates);
}
/* multi-channel playback needs alignment 8x32bit regardless of the channels
@@ -824,7 +872,7 @@ static int snd_vt1724_playback_pro_close(struct snd_pcm_substream *substream)
struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
if (PRO_RATE_RESET)
- snd_vt1724_set_pro_rate(ice, PRO_RATE_DEFAULT, 0);
+ snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 0);
ice->playback_pro_substream = NULL;
return 0;
@@ -835,7 +883,7 @@ static int snd_vt1724_capture_pro_close(struct snd_pcm_substream *substream)
struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
if (PRO_RATE_RESET)
- snd_vt1724_set_pro_rate(ice, PRO_RATE_DEFAULT, 0);
+ snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 0);
ice->capture_pro_substream = NULL;
return 0;
}
@@ -970,6 +1018,8 @@ static int snd_vt1724_playback_spdif_open(struct snd_pcm_substream *substream)
VT1724_BUFFER_ALIGN);
snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
VT1724_BUFFER_ALIGN);
+ if (ice->spdif.ops.open)
+ ice->spdif.ops.open(ice, substream);
return 0;
}
@@ -978,8 +1028,10 @@ static int snd_vt1724_playback_spdif_close(struct snd_pcm_substream *substream)
struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
if (PRO_RATE_RESET)
- snd_vt1724_set_pro_rate(ice, PRO_RATE_DEFAULT, 0);
+ snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 0);
ice->playback_con_substream = NULL;
+ if (ice->spdif.ops.close)
+ ice->spdif.ops.close(ice, substream);
return 0;
}
@@ -1002,6 +1054,8 @@ static int snd_vt1724_capture_spdif_open(struct snd_pcm_substream *substream)
VT1724_BUFFER_ALIGN);
snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
VT1724_BUFFER_ALIGN);
+ if (ice->spdif.ops.open)
+ ice->spdif.ops.open(ice, substream);
return 0;
}
@@ -1010,8 +1064,10 @@ static int snd_vt1724_capture_spdif_close(struct snd_pcm_substream *substream)
struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
if (PRO_RATE_RESET)
- snd_vt1724_set_pro_rate(ice, PRO_RATE_DEFAULT, 0);
+ snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 0);
ice->capture_con_substream = NULL;
+ if (ice->spdif.ops.close)
+ ice->spdif.ops.close(ice, substream);
return 0;
}
@@ -1154,7 +1210,7 @@ static int snd_vt1724_playback_indep_close(struct snd_pcm_substream *substream)
struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
if (PRO_RATE_RESET)
- snd_vt1724_set_pro_rate(ice, PRO_RATE_DEFAULT, 0);
+ snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 0);
ice->playback_con_substream_ds[substream->number] = NULL;
ice->pcm_reserved[substream->number] = NULL;
@@ -1572,50 +1628,18 @@ int snd_ice1712_gpio_put(struct snd_kcontrol *kcontrol,
static int snd_vt1724_pro_internal_clock_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static const char * const texts_1724[] = {
- "8000", /* 0: 6 */
- "9600", /* 1: 3 */
- "11025", /* 2: 10 */
- "12000", /* 3: 2 */
- "16000", /* 4: 5 */
- "22050", /* 5: 9 */
- "24000", /* 6: 1 */
- "32000", /* 7: 4 */
- "44100", /* 8: 8 */
- "48000", /* 9: 0 */
- "64000", /* 10: 15 */
- "88200", /* 11: 11 */
- "96000", /* 12: 7 */
- "176400", /* 13: 12 */
- "192000", /* 14: 14 */
- "IEC958 Input", /* 15: -- */
- };
- static const char * const texts_1720[] = {
- "8000", /* 0: 6 */
- "9600", /* 1: 3 */
- "11025", /* 2: 10 */
- "12000", /* 3: 2 */
- "16000", /* 4: 5 */
- "22050", /* 5: 9 */
- "24000", /* 6: 1 */
- "32000", /* 7: 4 */
- "44100", /* 8: 8 */
- "48000", /* 9: 0 */
- "64000", /* 10: 15 */
- "88200", /* 11: 11 */
- "96000", /* 12: 7 */
- "IEC958 Input", /* 13: -- */
- };
struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
- uinfo->value.enumerated.items = ice->vt1720 ? 14 : 16;
+ uinfo->value.enumerated.items = ice->hw_rates->count + 1;
if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
- strcpy(uinfo->value.enumerated.name,
- ice->vt1720 ? texts_1720[uinfo->value.enumerated.item] :
- texts_1724[uinfo->value.enumerated.item]);
+ if (uinfo->value.enumerated.item == uinfo->value.enumerated.items - 1)
+ strcpy(uinfo->value.enumerated.name, "IEC958 Input");
+ else
+ sprintf(uinfo->value.enumerated.name, "%d",
+ ice->hw_rates->list[uinfo->value.enumerated.item]);
return 0;
}
@@ -1623,68 +1647,79 @@ static int snd_vt1724_pro_internal_clock_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
- static const unsigned char xlate[16] = {
- 9, 6, 3, 1, 7, 4, 0, 12, 8, 5, 2, 11, 13, 255, 14, 10
- };
- unsigned char val;
+ unsigned int i, rate;
spin_lock_irq(&ice->reg_lock);
- if (is_spdif_master(ice)) {
- ucontrol->value.enumerated.item[0] = ice->vt1720 ? 13 : 15;
+ if (ice->is_spdif_master(ice)) {
+ ucontrol->value.enumerated.item[0] = ice->hw_rates->count;
} else {
- val = xlate[inb(ICEMT1724(ice, RATE)) & 15];
- if (val == 255) {
- snd_BUG();
- val = 0;
+ rate = ice->get_rate(ice);
+ ucontrol->value.enumerated.item[0] = 0;
+ for (i = 0; i < ice->hw_rates->count; i++) {
+ if (ice->hw_rates->list[i] == rate) {
+ ucontrol->value.enumerated.item[0] = i;
+ break;
+ }
}
- ucontrol->value.enumerated.item[0] = val;
}
spin_unlock_irq(&ice->reg_lock);
return 0;
}
+/* setting clock to external - SPDIF */
+static void stdclock_set_spdif_clock(struct snd_ice1712 *ice)
+{
+ unsigned char oval;
+ unsigned char i2s_oval;
+ oval = inb(ICEMT1724(ice, RATE));
+ outb(oval | VT1724_SPDIF_MASTER, ICEMT1724(ice, RATE));
+ /* setting 256fs */
+ i2s_oval = inb(ICEMT1724(ice, I2S_FORMAT));
+ outb(i2s_oval & ~VT1724_MT_I2S_MCLK_128X, ICEMT1724(ice, I2S_FORMAT));
+}
+
static int snd_vt1724_pro_internal_clock_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
- unsigned char oval;
- int rate;
- int change = 0;
- int spdif = ice->vt1720 ? 13 : 15;
+ unsigned int old_rate, new_rate;
+ unsigned int item = ucontrol->value.enumerated.item[0];
+ unsigned int spdif = ice->hw_rates->count;
+
+ if (item > spdif)
+ return -EINVAL;
spin_lock_irq(&ice->reg_lock);
- oval = inb(ICEMT1724(ice, RATE));
- if (ucontrol->value.enumerated.item[0] == spdif) {
- unsigned char i2s_oval;
- outb(oval | VT1724_SPDIF_MASTER, ICEMT1724(ice, RATE));
- /* setting 256fs */
- i2s_oval = inb(ICEMT1724(ice, I2S_FORMAT));
- outb(i2s_oval & ~VT1724_MT_I2S_MCLK_128X,
- ICEMT1724(ice, I2S_FORMAT));
+ if (ice->is_spdif_master(ice))
+ old_rate = 0;
+ else
+ old_rate = ice->get_rate(ice);
+ if (item == spdif) {
+ /* switching to external clock via SPDIF */
+ ice->set_spdif_clock(ice);
+ new_rate = 0;
} else {
- rate = rates[ucontrol->value.integer.value[0] % 15];
- if (rate <= get_max_rate(ice)) {
- PRO_RATE_DEFAULT = rate;
- spin_unlock_irq(&ice->reg_lock);
- snd_vt1724_set_pro_rate(ice, PRO_RATE_DEFAULT, 1);
- spin_lock_irq(&ice->reg_lock);
- }
+ /* internal on-card clock */
+ new_rate = ice->hw_rates->list[item];
+ ice->pro_rate_default = new_rate;
+ spin_unlock_irq(&ice->reg_lock);
+ snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 1);
+ spin_lock_irq(&ice->reg_lock);
}
- change = inb(ICEMT1724(ice, RATE)) != oval;
spin_unlock_irq(&ice->reg_lock);
- if ((oval & VT1724_SPDIF_MASTER) !=
- (inb(ICEMT1724(ice, RATE)) & VT1724_SPDIF_MASTER)) {
+ /* the first reset to the SPDIF master mode? */
+ if (old_rate != new_rate && !new_rate) {
/* notify akm chips as well */
- if (is_spdif_master(ice)) {
- unsigned int i;
- for (i = 0; i < ice->akm_codecs; i++) {
- if (ice->akm[i].ops.set_rate_val)
- ice->akm[i].ops.set_rate_val(&ice->akm[i], 0);
- }
+ unsigned int i;
+ if (ice->gpio.set_pro_rate)
+ ice->gpio.set_pro_rate(ice, 0);
+ for (i = 0; i < ice->akm_codecs; i++) {
+ if (ice->akm[i].ops.set_rate_val)
+ ice->akm[i].ops.set_rate_val(&ice->akm[i], 0);
}
}
- return change;
+ return old_rate != new_rate;
}
static struct snd_kcontrol_new snd_vt1724_pro_internal_clock __devinitdata = {
@@ -2065,12 +2100,16 @@ static int __devinit snd_vt1724_read_eeprom(struct snd_ice1712 *ice,
-static int __devinit snd_vt1724_chip_init(struct snd_ice1712 *ice)
+static void __devinit snd_vt1724_chip_reset(struct snd_ice1712 *ice)
{
outb(VT1724_RESET , ICEREG1724(ice, CONTROL));
- udelay(200);
+ msleep(10);
outb(0, ICEREG1724(ice, CONTROL));
- udelay(200);
+ msleep(10);
+}
+
+static int __devinit snd_vt1724_chip_init(struct snd_ice1712 *ice)
+{
outb(ice->eeprom.data[ICE_EEP2_SYSCONF], ICEREG1724(ice, SYS_CFG));
outb(ice->eeprom.data[ICE_EEP2_ACLINK], ICEREG1724(ice, AC97_CFG));
outb(ice->eeprom.data[ICE_EEP2_I2S], ICEREG1724(ice, I2S_FEATURES));
@@ -2169,10 +2208,8 @@ static int snd_vt1724_free(struct snd_ice1712 *ice)
outb(0xff, ICEREG1724(ice, IRQMASK));
/* --- */
__hw_end:
- if (ice->irq >= 0) {
- synchronize_irq(ice->irq);
+ if (ice->irq >= 0)
free_irq(ice->irq, ice);
- }
pci_release_regions(ice->pci);
snd_ice1712_akm4xxx_free(ice);
pci_disable_device(ice->pci);
@@ -2243,6 +2280,7 @@ static int __devinit snd_vt1724_create(struct snd_card *card,
ice->irq = pci->irq;
+ snd_vt1724_chip_reset(ice);
if (snd_vt1724_read_eeprom(ice, modelname) < 0) {
snd_vt1724_free(ice);
return -EIO;
@@ -2253,10 +2291,7 @@ static int __devinit snd_vt1724_create(struct snd_card *card,
}
/* unmask used interrupts */
- if (! (ice->eeprom.data[ICE_EEP2_SYSCONF] & VT1724_CFG_MPU401))
- mask = VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX;
- else
- mask = 0;
+ mask = VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX;
outb(mask, ICEREG1724(ice, IRQMASK));
/* don't handle FIFO overrun/underruns (just yet),
* since they cause machine lockups
@@ -2335,6 +2370,19 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci,
* was called so in ice1712 driver, and vt1724 driver is derived from
* ice1712 driver.
*/
+ ice->pro_rate_default = PRO_RATE_DEFAULT;
+ if (!ice->is_spdif_master)
+ ice->is_spdif_master = stdclock_is_spdif_master;
+ if (!ice->get_rate)
+ ice->get_rate = stdclock_get_rate;
+ if (!ice->set_rate)
+ ice->set_rate = stdclock_set_rate;
+ if (!ice->set_mclk)
+ ice->set_mclk = stdclock_set_mclk;
+ if (!ice->set_spdif_clock)
+ ice->set_spdif_clock = stdclock_set_spdif_clock;
+ if (!ice->hw_rates)
+ set_std_hw_rates(ice);
if ((err = snd_vt1724_pcm_profi(ice, pcm_dev++)) < 0) {
snd_card_free(card);
@@ -2377,14 +2425,29 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci,
if (! c->no_mpu401) {
if (ice->eeprom.data[ICE_EEP2_SYSCONF] & VT1724_CFG_MPU401) {
+ struct snd_mpu401 *mpu;
if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712,
ICEREG1724(ice, MPU_CTRL),
- MPU401_INFO_INTEGRATED,
+ (MPU401_INFO_INTEGRATED |
+ MPU401_INFO_TX_IRQ),
ice->irq, 0,
&ice->rmidi[0])) < 0) {
snd_card_free(card);
return err;
}
+ mpu = ice->rmidi[0]->private_data;
+ mpu->read = snd_vt1724_mpu401_read;
+ mpu->write = snd_vt1724_mpu401_write;
+ /* unmask MPU RX/TX irqs */
+ outb(inb(ICEREG1724(ice, IRQMASK)) &
+ ~(VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX),
+ ICEREG1724(ice, IRQMASK));
+#if 0 /* for testing */
+ /* set watermarks */
+ outb(VT1724_MPU_RX_FIFO | 0x1,
+ ICEREG1724(ice, MPU_FIFO_WM));
+ outb(0x1, ICEREG1724(ice, MPU_FIFO_WM));
+#endif
}
}
diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c
index e8038c0ceb7..b4e0c16852a 100644
--- a/sound/pci/ice1712/juli.c
+++ b/sound/pci/ice1712/juli.c
@@ -4,6 +4,8 @@
* Lowlevel functions for ESI Juli@ cards
*
* Copyright (c) 2004 Jaroslav Kysela <perex@perex.cz>
+ * 2008 Pavel Hofman <dustin@seznam.cz>
+ *
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
@@ -27,11 +29,11 @@
#include <linux/init.h>
#include <linux/slab.h>
#include <sound/core.h>
+#include <sound/tlv.h>
#include "ice1712.h"
#include "envy24ht.h"
#include "juli.h"
-
struct juli_spec {
struct ak4114 *ak4114;
unsigned int analog: 1;
@@ -44,6 +46,32 @@ struct juli_spec {
#define AK4358_ADDR 0x22 /* DAC */
/*
+ * Juli does not use the standard ICE1724 clock scheme. Juli's ice1724 chip is
+ * supplied by external clock provided by Xilinx array and MK73-1 PLL frequency
+ * multiplier. Actual frequency is set by ice1724 GPIOs hooked to the Xilinx.
+ *
+ * The clock circuitry is supplied by the two ice1724 crystals. This
+ * arrangement allows to generate independent clock signal for AK4114's input
+ * rate detection circuit. As a result, Juli, unlike most other
+ * ice1724+ak4114-based cards, detects spdif input rate correctly.
+ * This fact is applied in the driver, allowing to modify PCM stream rate
+ * parameter according to the actual input rate.
+ *
+ * Juli uses the remaining three stereo-channels of its DAC to optionally
+ * monitor analog input, digital input, and digital output. The corresponding
+ * I2S signals are routed by Xilinx, controlled by GPIOs.
+ *
+ * The master mute is implemented using output muting transistors (GPIO) in
+ * combination with smuting the DAC.
+ *
+ * The card itself has no HW master volume control, implemented using the
+ * vmaster control.
+ *
+ * TODO:
+ * researching and fixing the input monitors
+ */
+
+/*
* GPIO pins
*/
#define GPIO_FREQ_MASK (3<<0)
@@ -55,17 +83,82 @@ struct juli_spec {
#define GPIO_MULTI_2X (1<<2)
#define GPIO_MULTI_1X (2<<2) /* also external */
#define GPIO_MULTI_HALF (3<<2)
-#define GPIO_INTERNAL_CLOCK (1<<4)
+#define GPIO_INTERNAL_CLOCK (1<<4) /* 0 = external, 1 = internal */
+#define GPIO_CLOCK_MASK (1<<4)
#define GPIO_ANALOG_PRESENT (1<<5) /* RO only: 0 = present */
#define GPIO_RXMCLK_SEL (1<<7) /* must be 0 */
#define GPIO_AK5385A_CKS0 (1<<8)
-#define GPIO_AK5385A_DFS0 (1<<9) /* swapped with DFS1 according doc? */
-#define GPIO_AK5385A_DFS1 (1<<10)
+#define GPIO_AK5385A_DFS1 (1<<9)
+#define GPIO_AK5385A_DFS0 (1<<10)
#define GPIO_DIGOUT_MONITOR (1<<11) /* 1 = active */
#define GPIO_DIGIN_MONITOR (1<<12) /* 1 = active */
#define GPIO_ANAIN_MONITOR (1<<13) /* 1 = active */
-#define GPIO_AK5385A_MCLK (1<<14) /* must be 0 */
-#define GPIO_MUTE_CONTROL (1<<15) /* 0 = off, 1 = on */
+#define GPIO_AK5385A_CKS1 (1<<14) /* must be 0 */
+#define GPIO_MUTE_CONTROL (1<<15) /* output mute, 1 = muted */
+
+#define GPIO_RATE_MASK (GPIO_FREQ_MASK | GPIO_MULTI_MASK | \
+ GPIO_CLOCK_MASK)
+#define GPIO_AK5385A_MASK (GPIO_AK5385A_CKS0 | GPIO_AK5385A_DFS0 | \
+ GPIO_AK5385A_DFS1 | GPIO_AK5385A_CKS1)
+
+#define JULI_PCM_RATE (SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \
+ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \
+ SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000)
+
+#define GPIO_RATE_16000 (GPIO_FREQ_32KHZ | GPIO_MULTI_HALF | \
+ GPIO_INTERNAL_CLOCK)
+#define GPIO_RATE_22050 (GPIO_FREQ_44KHZ | GPIO_MULTI_HALF | \
+ GPIO_INTERNAL_CLOCK)
+#define GPIO_RATE_24000 (GPIO_FREQ_48KHZ | GPIO_MULTI_HALF | \
+ GPIO_INTERNAL_CLOCK)
+#define GPIO_RATE_32000 (GPIO_FREQ_32KHZ | GPIO_MULTI_1X | \
+ GPIO_INTERNAL_CLOCK)
+#define GPIO_RATE_44100 (GPIO_FREQ_44KHZ | GPIO_MULTI_1X | \
+ GPIO_INTERNAL_CLOCK)
+#define GPIO_RATE_48000 (GPIO_FREQ_48KHZ | GPIO_MULTI_1X | \
+ GPIO_INTERNAL_CLOCK)
+#define GPIO_RATE_64000 (GPIO_FREQ_32KHZ | GPIO_MULTI_2X | \
+ GPIO_INTERNAL_CLOCK)
+#define GPIO_RATE_88200 (GPIO_FREQ_44KHZ | GPIO_MULTI_2X | \
+ GPIO_INTERNAL_CLOCK)
+#define GPIO_RATE_96000 (GPIO_FREQ_48KHZ | GPIO_MULTI_2X | \
+ GPIO_INTERNAL_CLOCK)
+#define GPIO_RATE_176400 (GPIO_FREQ_44KHZ | GPIO_MULTI_4X | \
+ GPIO_INTERNAL_CLOCK)
+#define GPIO_RATE_192000 (GPIO_FREQ_48KHZ | GPIO_MULTI_4X | \
+ GPIO_INTERNAL_CLOCK)
+
+/*
+ * Initial setup of the conversion array GPIO <-> rate
+ */
+static unsigned int juli_rates[] = {
+ 16000, 22050, 24000, 32000,
+ 44100, 48000, 64000, 88200,
+ 96000, 176400, 192000,
+};
+
+static unsigned int gpio_vals[] = {
+ GPIO_RATE_16000, GPIO_RATE_22050, GPIO_RATE_24000, GPIO_RATE_32000,
+ GPIO_RATE_44100, GPIO_RATE_48000, GPIO_RATE_64000, GPIO_RATE_88200,
+ GPIO_RATE_96000, GPIO_RATE_176400, GPIO_RATE_192000,
+};
+
+static struct snd_pcm_hw_constraint_list juli_rates_info = {
+ .count = ARRAY_SIZE(juli_rates),
+ .list = juli_rates,
+ .mask = 0,
+};
+
+static int get_gpio_val(int rate)
+{
+ int i;
+ for (i = 0; i < ARRAY_SIZE(juli_rates); i++)
+ if (juli_rates[i] == rate)
+ return gpio_vals[i];
+ return 0;
+}
static void juli_ak4114_write(void *private_data, unsigned char reg, unsigned char val)
{
@@ -78,6 +171,27 @@ static unsigned char juli_ak4114_read(void *private_data, unsigned char reg)
}
/*
+ * If SPDIF capture and slaved to SPDIF-IN, setting runtime rate
+ * to the external rate
+ */
+static void juli_spdif_in_open(struct snd_ice1712 *ice,
+ struct snd_pcm_substream *substream)
+{
+ struct juli_spec *spec = ice->spec;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int rate;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ||
+ !ice->is_spdif_master(ice))
+ return;
+ rate = snd_ak4114_external_rate(spec->ak4114);
+ if (rate >= runtime->hw.rate_min && rate <= runtime->hw.rate_max) {
+ runtime->hw.rate_min = rate;
+ runtime->hw.rate_max = rate;
+ }
+}
+
+/*
* AK4358 section
*/
@@ -99,57 +213,285 @@ static void juli_akm_write(struct snd_akm4xxx *ak, int chip,
}
/*
- * change the rate of envy24HT, AK4358
+ * change the rate of envy24HT, AK4358, AK5385
*/
static void juli_akm_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate)
{
- unsigned char old, tmp, dfs;
+ unsigned char old, tmp, ak4358_dfs;
+ unsigned int ak5385_pins, old_gpio, new_gpio;
+ struct snd_ice1712 *ice = ak->private_data[0];
+ struct juli_spec *spec = ice->spec;
- if (rate == 0) /* no hint - S/PDIF input is master, simply return */
+ if (rate == 0) /* no hint - S/PDIF input is master or the new spdif
+ input rate undetected, simply return */
return;
-
+
/* adjust DFS on codecs */
- if (rate > 96000)
- dfs = 2;
- else if (rate > 48000)
- dfs = 1;
- else
- dfs = 0;
-
+ if (rate > 96000) {
+ ak4358_dfs = 2;
+ ak5385_pins = GPIO_AK5385A_DFS1 | GPIO_AK5385A_CKS0;
+ } else if (rate > 48000) {
+ ak4358_dfs = 1;
+ ak5385_pins = GPIO_AK5385A_DFS0;
+ } else {
+ ak4358_dfs = 0;
+ ak5385_pins = 0;
+ }
+ /* AK5385 first, since it requires cold reset affecting both codecs */
+ old_gpio = ice->gpio.get_data(ice);
+ new_gpio = (old_gpio & ~GPIO_AK5385A_MASK) | ak5385_pins;
+ /* printk(KERN_DEBUG "JULI - ak5385 set_rate_val: new gpio 0x%x\n",
+ new_gpio); */
+ ice->gpio.set_data(ice, new_gpio);
+
+ /* cold reset */
+ old = inb(ICEMT1724(ice, AC97_CMD));
+ outb(old | VT1724_AC97_COLD, ICEMT1724(ice, AC97_CMD));
+ udelay(1);
+ outb(old & ~VT1724_AC97_COLD, ICEMT1724(ice, AC97_CMD));
+
+ /* AK4358 */
+ /* set new value, reset DFS */
tmp = snd_akm4xxx_get(ak, 0, 2);
- old = (tmp >> 4) & 0x03;
- if (old == dfs)
- return;
- /* reset DFS */
snd_akm4xxx_reset(ak, 1);
tmp = snd_akm4xxx_get(ak, 0, 2);
tmp &= ~(0x03 << 4);
- tmp |= dfs << 4;
+ tmp |= ak4358_dfs << 4;
snd_akm4xxx_set(ak, 0, 2, tmp);
snd_akm4xxx_reset(ak, 0);
+
+ /* reinit ak4114 */
+ snd_ak4114_reinit(spec->ak4114);
}
+#define AK_DAC(xname, xch) { .name = xname, .num_channels = xch }
+#define PCM_VOLUME "PCM Playback Volume"
+#define MONITOR_AN_IN_VOLUME "Monitor Analog In Volume"
+#define MONITOR_DIG_IN_VOLUME "Monitor Digital In Volume"
+#define MONITOR_DIG_OUT_VOLUME "Monitor Digital Out Volume"
+
+static const struct snd_akm4xxx_dac_channel juli_dac[] = {
+ AK_DAC(PCM_VOLUME, 2),
+ AK_DAC(MONITOR_AN_IN_VOLUME, 2),
+ AK_DAC(MONITOR_DIG_OUT_VOLUME, 2),
+ AK_DAC(MONITOR_DIG_IN_VOLUME, 2),
+};
+
+
static struct snd_akm4xxx akm_juli_dac __devinitdata = {
.type = SND_AK4358,
- .num_dacs = 2,
+ .num_dacs = 8, /* DAC1 - analog out
+ DAC2 - analog in monitor
+ DAC3 - digital out monitor
+ DAC4 - digital in monitor
+ */
.ops = {
.lock = juli_akm_lock,
.unlock = juli_akm_unlock,
.write = juli_akm_write,
.set_rate_val = juli_akm_set_rate_val
+ },
+ .dac_info = juli_dac,
+};
+
+#define juli_mute_info snd_ctl_boolean_mono_info
+
+static int juli_mute_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ val = ice->gpio.get_data(ice) & (unsigned int) kcontrol->private_value;
+ if (kcontrol->private_value == GPIO_MUTE_CONTROL)
+ /* val 0 = signal on */
+ ucontrol->value.integer.value[0] = (val) ? 0 : 1;
+ else
+ /* val 1 = signal on */
+ ucontrol->value.integer.value[0] = (val) ? 1 : 0;
+ return 0;
+}
+
+static int juli_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned int old_gpio, new_gpio;
+ old_gpio = ice->gpio.get_data(ice);
+ if (ucontrol->value.integer.value[0]) {
+ /* unmute */
+ if (kcontrol->private_value == GPIO_MUTE_CONTROL) {
+ /* 0 = signal on */
+ new_gpio = old_gpio & ~GPIO_MUTE_CONTROL;
+ /* un-smuting DAC */
+ snd_akm4xxx_write(ice->akm, 0, 0x01, 0x01);
+ } else
+ /* 1 = signal on */
+ new_gpio = old_gpio |
+ (unsigned int) kcontrol->private_value;
+ } else {
+ /* mute */
+ if (kcontrol->private_value == GPIO_MUTE_CONTROL) {
+ /* 1 = signal off */
+ new_gpio = old_gpio | GPIO_MUTE_CONTROL;
+ /* smuting DAC */
+ snd_akm4xxx_write(ice->akm, 0, 0x01, 0x03);
+ } else
+ /* 0 = signal off */
+ new_gpio = old_gpio &
+ ~((unsigned int) kcontrol->private_value);
+ }
+ /* printk("JULI - mute/unmute: control_value: 0x%x, old_gpio: 0x%x, \
+ new_gpio 0x%x\n",
+ (unsigned int)ucontrol->value.integer.value[0], old_gpio,
+ new_gpio); */
+ if (old_gpio != new_gpio) {
+ ice->gpio.set_data(ice, new_gpio);
+ return 1;
+ }
+ /* no change */
+ return 0;
+}
+
+static struct snd_kcontrol_new juli_mute_controls[] __devinitdata = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = juli_mute_info,
+ .get = juli_mute_get,
+ .put = juli_mute_put,
+ .private_value = GPIO_MUTE_CONTROL,
+ },
+ /* Although the following functionality respects the succint NDA'd
+ * documentation from the card manufacturer, and the same way of
+ * operation is coded in OSS Juli driver, only Digital Out monitor
+ * seems to work. Surprisingly, Analog input monitor outputs Digital
+ * output data. The two are independent, as enabling both doubles
+ * volume of the monitor sound.
+ *
+ * Checking traces on the board suggests the functionality described
+ * by the manufacturer is correct - I2S from ADC and AK4114
+ * go to ICE as well as to Xilinx, I2S inputs of DAC2,3,4 (the monitor
+ * inputs) are fed from Xilinx.
+ *
+ * I even checked traces on board and coded a support in driver for
+ * an alternative possiblity - the unused I2S ICE output channels
+ * switched to HW-IN/SPDIF-IN and providing the monitoring signal to
+ * the DAC - to no avail. The I2S outputs seem to be unconnected.
+ *
+ * The windows driver supports the monitoring correctly.
+ */
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Monitor Analog In Switch",
+ .info = juli_mute_info,
+ .get = juli_mute_get,
+ .put = juli_mute_put,
+ .private_value = GPIO_ANAIN_MONITOR,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Monitor Digital Out Switch",
+ .info = juli_mute_info,
+ .get = juli_mute_get,
+ .put = juli_mute_put,
+ .private_value = GPIO_DIGOUT_MONITOR,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Monitor Digital In Switch",
+ .info = juli_mute_info,
+ .get = juli_mute_get,
+ .put = juli_mute_put,
+ .private_value = GPIO_DIGIN_MONITOR,
+ },
+};
+
+
+static void ak4358_proc_regs_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_ice1712 *ice = (struct snd_ice1712 *)entry->private_data;
+ int reg, val;
+ for (reg = 0; reg <= 0xf; reg++) {
+ val = snd_akm4xxx_get(ice->akm, 0, reg);
+ snd_iprintf(buffer, "0x%02x = 0x%02x\n", reg, val);
}
+}
+
+static void ak4358_proc_init(struct snd_ice1712 *ice)
+{
+ struct snd_info_entry *entry;
+ if (!snd_card_proc_new(ice->card, "ak4358_codec", &entry))
+ snd_info_set_text_ops(entry, ice, ak4358_proc_regs_read);
+}
+
+static char *slave_vols[] __devinitdata = {
+ PCM_VOLUME,
+ MONITOR_AN_IN_VOLUME,
+ MONITOR_DIG_IN_VOLUME,
+ MONITOR_DIG_OUT_VOLUME,
+ NULL
};
+static __devinitdata
+DECLARE_TLV_DB_SCALE(juli_master_db_scale, -6350, 50, 1);
+
+static struct snd_kcontrol __devinit *ctl_find(struct snd_card *card,
+ const char *name)
+{
+ struct snd_ctl_elem_id sid;
+ memset(&sid, 0, sizeof(sid));
+ /* FIXME: strcpy is bad. */
+ strcpy(sid.name, name);
+ sid.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ return snd_ctl_find_id(card, &sid);
+}
+
+static void __devinit add_slaves(struct snd_card *card,
+ struct snd_kcontrol *master, char **list)
+{
+ for (; *list; list++) {
+ struct snd_kcontrol *slave = ctl_find(card, *list);
+ /* printk(KERN_DEBUG "add_slaves - %s\n", *list); */
+ if (slave) {
+ /* printk(KERN_DEBUG "slave %s found\n", *list); */
+ snd_ctl_add_slave(master, slave);
+ }
+ }
+}
+
static int __devinit juli_add_controls(struct snd_ice1712 *ice)
{
struct juli_spec *spec = ice->spec;
int err;
+ unsigned int i;
+ struct snd_kcontrol *vmaster;
+
err = snd_ice1712_akm4xxx_build_controls(ice);
if (err < 0)
return err;
+
+ for (i = 0; i < ARRAY_SIZE(juli_mute_controls); i++) {
+ err = snd_ctl_add(ice->card,
+ snd_ctl_new1(&juli_mute_controls[i], ice));
+ if (err < 0)
+ return err;
+ }
+ /* Create virtual master control */
+ vmaster = snd_ctl_make_virtual_master("Master Playback Volume",
+ juli_master_db_scale);
+ if (!vmaster)
+ return -ENOMEM;
+ add_slaves(ice->card, vmaster, slave_vols);
+ err = snd_ctl_add(ice->card, vmaster);
+ if (err < 0)
+ return err;
+
/* only capture SPDIF over AK4114 */
err = snd_ak4114_build(spec->ak4114, NULL,
- ice->pcm_pro->streams[SNDRV_PCM_STREAM_CAPTURE].substream);
+ ice->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream);
+
+ ak4358_proc_init(ice);
if (err < 0)
return err;
return 0;
@@ -158,6 +500,74 @@ static int __devinit juli_add_controls(struct snd_ice1712 *ice)
/*
* initialize the chip
*/
+
+static inline int juli_is_spdif_master(struct snd_ice1712 *ice)
+{
+ return (ice->gpio.get_data(ice) & GPIO_INTERNAL_CLOCK) ? 0 : 1;
+}
+
+static unsigned int juli_get_rate(struct snd_ice1712 *ice)
+{
+ int i;
+ unsigned char result;
+
+ result = ice->gpio.get_data(ice) & GPIO_RATE_MASK;
+ for (i = 0; i < ARRAY_SIZE(gpio_vals); i++)
+ if (gpio_vals[i] == result)
+ return juli_rates[i];
+ return 0;
+}
+
+/* setting new rate */
+static void juli_set_rate(struct snd_ice1712 *ice, unsigned int rate)
+{
+ unsigned int old, new;
+ unsigned char val;
+
+ old = ice->gpio.get_data(ice);
+ new = (old & ~GPIO_RATE_MASK) | get_gpio_val(rate);
+ /* printk(KERN_DEBUG "JULI - set_rate: old %x, new %x\n",
+ old & GPIO_RATE_MASK,
+ new & GPIO_RATE_MASK); */
+
+ ice->gpio.set_data(ice, new);
+ /* switching to external clock - supplied by external circuits */
+ val = inb(ICEMT1724(ice, RATE));
+ outb(val | VT1724_SPDIF_MASTER, ICEMT1724(ice, RATE));
+}
+
+static inline unsigned char juli_set_mclk(struct snd_ice1712 *ice,
+ unsigned int rate)
+{
+ /* no change in master clock */
+ return 0;
+}
+
+/* setting clock to external - SPDIF */
+static void juli_set_spdif_clock(struct snd_ice1712 *ice)
+{
+ unsigned int old;
+ old = ice->gpio.get_data(ice);
+ /* external clock (= 0), multiply 1x, 48kHz */
+ ice->gpio.set_data(ice, (old & ~GPIO_RATE_MASK) | GPIO_MULTI_1X |
+ GPIO_FREQ_48KHZ);
+}
+
+/* Called when ak4114 detects change in the input SPDIF stream */
+static void juli_ak4114_change(struct ak4114 *ak4114, unsigned char c0,
+ unsigned char c1)
+{
+ struct snd_ice1712 *ice = ak4114->change_callback_private;
+ int rate;
+ if (ice->is_spdif_master(ice) && c1) {
+ /* only for SPDIF master mode, rate was changed */
+ rate = snd_ak4114_external_rate(ak4114);
+ /* printk(KERN_DEBUG "ak4114 - input rate changed to %d\n",
+ rate); */
+ juli_akm_set_rate_val(ice->akm, rate);
+ }
+}
+
static int __devinit juli_init(struct snd_ice1712 *ice)
{
static const unsigned char ak4114_init_vals[] = {
@@ -187,6 +597,11 @@ static int __devinit juli_init(struct snd_ice1712 *ice)
ice, &spec->ak4114);
if (err < 0)
return err;
+ /* callback for codecs rate setting */
+ spec->ak4114->change_callback = juli_ak4114_change;
+ spec->ak4114->change_callback_private = ice;
+ /* AK4114 in Juli can detect external rate correctly */
+ spec->ak4114->check_flags = 0;
#if 0
/* it seems that the analog doughter board detection does not work
@@ -210,6 +625,15 @@ static int __devinit juli_init(struct snd_ice1712 *ice)
return err;
}
+ /* juli is clocked by Xilinx array */
+ ice->hw_rates = &juli_rates_info;
+ ice->is_spdif_master = juli_is_spdif_master;
+ ice->get_rate = juli_get_rate;
+ ice->set_rate = juli_set_rate;
+ ice->set_mclk = juli_set_mclk;
+ ice->set_spdif_clock = juli_set_spdif_clock;
+
+ ice->spdif.ops.open = juli_spdif_in_open;
return 0;
}
@@ -220,18 +644,20 @@ static int __devinit juli_init(struct snd_ice1712 *ice)
*/
static unsigned char juli_eeprom[] __devinitdata = {
- [ICE_EEP2_SYSCONF] = 0x20, /* clock 512, mpu401, 1xADC, 1xDACs */
+ [ICE_EEP2_SYSCONF] = 0x2b, /* clock 512, mpu401, 1xADC, 1xDACs,
+ SPDIF in */
[ICE_EEP2_ACLINK] = 0x80, /* I2S */
[ICE_EEP2_I2S] = 0xf8, /* vol, 96k, 24bit, 192k */
[ICE_EEP2_SPDIF] = 0xc3, /* out-en, out-int, spdif-in */
- [ICE_EEP2_GPIO_DIR] = 0x9f,
+ [ICE_EEP2_GPIO_DIR] = 0x9f, /* 5, 6:inputs; 7, 4-0 outputs*/
[ICE_EEP2_GPIO_DIR1] = 0xff,
[ICE_EEP2_GPIO_DIR2] = 0x7f,
- [ICE_EEP2_GPIO_MASK] = 0x9f,
- [ICE_EEP2_GPIO_MASK1] = 0xff,
+ [ICE_EEP2_GPIO_MASK] = 0x60, /* 5, 6: locked; 7, 4-0 writable */
+ [ICE_EEP2_GPIO_MASK1] = 0x00, /* 0-7 writable */
[ICE_EEP2_GPIO_MASK2] = 0x7f,
- [ICE_EEP2_GPIO_STATE] = 0x16, /* internal clock, multiple 1x, 48kHz */
- [ICE_EEP2_GPIO_STATE1] = 0x80, /* mute */
+ [ICE_EEP2_GPIO_STATE] = GPIO_FREQ_48KHZ | GPIO_MULTI_1X |
+ GPIO_INTERNAL_CLOCK, /* internal clock, multiple 1x, 48kHz*/
+ [ICE_EEP2_GPIO_STATE1] = 0x00, /* unmuted */
[ICE_EEP2_GPIO_STATE2] = 0x00,
};
diff --git a/sound/pci/ice1712/pontis.c b/sound/pci/ice1712/pontis.c
index 4945c81e8a9..203cdc1bf8d 100644
--- a/sound/pci/ice1712/pontis.c
+++ b/sound/pci/ice1712/pontis.c
@@ -246,7 +246,7 @@ static int wm_adc_mux_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val
wm_put(ice, WM_ADC_MUX, nval);
}
mutex_unlock(&ice->gpio_mutex);
- return 0;
+ return change;
}
/*
@@ -450,7 +450,7 @@ static int cs_source_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_valu
change = 1;
}
mutex_unlock(&ice->gpio_mutex);
- return 0;
+ return change;
}
diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c
index 48cf40a8f32..48d3679292a 100644
--- a/sound/pci/ice1712/prodigy192.c
+++ b/sound/pci/ice1712/prodigy192.c
@@ -319,12 +319,11 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol,
/*
* Handler for setting correct codec rate - called when rate change is detected
*/
-static void stac9460_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate)
+static void stac9460_set_rate_val(struct snd_ice1712 *ice, unsigned int rate)
{
unsigned char old, new;
int idx;
unsigned char changed[7];
- struct snd_ice1712 *ice = ak->private_data[0];
struct prodigy192_spec *spec = ice->spec;
if (rate == 0) /* no hint - S/PDIF input is master, simply return */
@@ -357,16 +356,6 @@ static void stac9460_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate)
mutex_unlock(&spec->mute_mutex);
}
-/* using akm infrastructure for setting rate of the codec */
-static struct snd_akm4xxx akmlike_stac9460 __devinitdata = {
- .type = NON_AKM, /* special value */
- .num_adcs = 6, /* not used in any way, just for completeness */
- .num_dacs = 2,
- .ops = {
- .set_rate_val = stac9460_set_rate_val
- }
-};
-
static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0);
static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0);
@@ -642,12 +631,19 @@ static int prodigy192_ak4114_init(struct snd_ice1712 *ice)
0x41, 0x02, 0x2c, 0x00, 0x00
};
struct prodigy192_spec *spec = ice->spec;
+ int err;
- return snd_ak4114_create(ice->card,
+ err = snd_ak4114_create(ice->card,
prodigy192_ak4114_read,
prodigy192_ak4114_write,
ak4114_init_vals, ak4114_init_txcsb,
ice, &spec->ak4114);
+ if (err < 0)
+ return err;
+ /* AK4114 in Prodigy192 cannot detect external rate correctly.
+ * No reason to stop capture stream due to incorrect checks */
+ spec->ak4114->check_flags = AK4114_CHECK_NO_RATE;
+ return 0;
}
static void stac9460_proc_regs_read(struct snd_info_entry *entry,
@@ -743,7 +739,6 @@ static int __devinit prodigy192_init(struct snd_ice1712 *ice)
};
const unsigned short *p;
int err = 0;
- struct snd_akm4xxx *ak;
struct prodigy192_spec *spec;
/* prodigy 192 */
@@ -761,15 +756,7 @@ static int __devinit prodigy192_init(struct snd_ice1712 *ice)
p = stac_inits_prodigy;
for (; *p != (unsigned short)-1; p += 2)
stac9460_put(ice, p[0], p[1]);
- /* reusing the akm codecs infrastructure,
- * for setting rate on stac9460 */
- ak = ice->akm = kmalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL);
- if (!ak)
- return -ENOMEM;
- ice->akm_codecs = 1;
- err = snd_ice1712_akm4xxx_init(ak, &akmlike_stac9460, NULL, ice);
- if (err < 0)
- return err;
+ ice->gpio.set_pro_rate = stac9460_set_rate_val;
/* MI/ODI/O add on card with AK4114 */
if (prodigy192_miodio_exists(ice)) {
@@ -825,10 +812,6 @@ struct snd_ice1712_card_info snd_vt1724_prodigy192_cards[] __devinitdata = {
.build_controls = prodigy192_add_controls,
.eeprom_size = sizeof(prodigy71_eeprom),
.eeprom_data = prodigy71_eeprom,
- /* the current MPU401 code loops infinitely
- * when opening midi device
- */
- .no_mpu401 = 1,
},
{ } /* terminator */
};
diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c
index 301bf929acd..4d2631434dc 100644
--- a/sound/pci/ice1712/revo.c
+++ b/sound/pci/ice1712/revo.c
@@ -322,17 +322,23 @@ static struct snd_pt2258 ptc_revo51_volume;
static void ap192_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate)
{
struct snd_ice1712 *ice = ak->private_data[0];
+ int dfs;
revo_set_rate_val(ak, rate);
-#if 1 /* FIXME: do we need this procedure? */
- /* reset DFS pin of AK5385A for ADC, too */
- /* DFS0 (pin 18) -- GPIO10 pin 77 */
- snd_ice1712_save_gpio_status(ice);
- snd_ice1712_gpio_write_bits(ice, 1 << 10,
- rate > 48000 ? (1 << 10) : 0);
- snd_ice1712_restore_gpio_status(ice);
-#endif
+ /* reset CKS */
+ snd_ice1712_gpio_write_bits(ice, 1 << 8, rate > 96000 ? 1 << 8 : 0);
+ /* reset DFS pins of AK5385A for ADC, too */
+ if (rate > 96000)
+ dfs = 2;
+ else if (rate > 48000)
+ dfs = 1;
+ else
+ dfs = 0;
+ snd_ice1712_gpio_write_bits(ice, 3 << 9, dfs << 9);
+ /* reset ADC */
+ snd_ice1712_gpio_write_bits(ice, 1 << 11, 0);
+ snd_ice1712_gpio_write_bits(ice, 1 << 11, 1 << 11);
}
static const struct snd_akm4xxx_dac_channel ap192_dac[] = {
@@ -353,28 +359,20 @@ static struct snd_ak4xxx_private akm_ap192_priv __devinitdata = {
.cif = 0,
.data_mask = VT1724_REVO_CDOUT,
.clk_mask = VT1724_REVO_CCLK,
- .cs_mask = VT1724_REVO_CS0 | VT1724_REVO_CS3,
- .cs_addr = VT1724_REVO_CS3,
- .cs_none = VT1724_REVO_CS0 | VT1724_REVO_CS3,
+ .cs_mask = VT1724_REVO_CS0 | VT1724_REVO_CS1,
+ .cs_addr = VT1724_REVO_CS1,
+ .cs_none = VT1724_REVO_CS0 | VT1724_REVO_CS1,
.add_flags = VT1724_REVO_CCLK, /* high at init */
.mask_flags = 0,
};
-#if 0
-/* FIXME: ak4114 makes the sound much lower due to some confliction,
- * so let's disable it right now...
- */
-#define BUILD_AK4114_AP192
-#endif
-
-#ifdef BUILD_AK4114_AP192
/* AK4114 support on Audiophile 192 */
/* CDTO (pin 32) -- GPIO2 pin 52
* CDTI (pin 33) -- GPIO3 pin 53 (shared with AK4358)
* CCLK (pin 34) -- GPIO1 pin 51 (shared with AK4358)
* CSN (pin 35) -- GPIO7 pin 59
*/
-#define AK4114_ADDR 0x00
+#define AK4114_ADDR 0x02
static void write_data(struct snd_ice1712 *ice, unsigned int gpio,
unsigned int data, int idx)
@@ -428,7 +426,7 @@ static unsigned int ap192_4wire_start(struct snd_ice1712 *ice)
tmp = snd_ice1712_gpio_read(ice);
tmp |= VT1724_REVO_CCLK; /* high at init */
tmp |= VT1724_REVO_CS0;
- tmp &= ~VT1724_REVO_CS3;
+ tmp &= ~VT1724_REVO_CS1;
snd_ice1712_gpio_write(ice, tmp);
udelay(1);
return tmp;
@@ -436,7 +434,7 @@ static unsigned int ap192_4wire_start(struct snd_ice1712 *ice)
static void ap192_4wire_finish(struct snd_ice1712 *ice, unsigned int tmp)
{
- tmp |= VT1724_REVO_CS3;
+ tmp |= VT1724_REVO_CS1;
tmp |= VT1724_REVO_CS0;
snd_ice1712_gpio_write(ice, tmp);
udelay(1);
@@ -485,13 +483,17 @@ static int __devinit ap192_ak4114_init(struct snd_ice1712 *ice)
struct ak4114 *ak;
int err;
- return snd_ak4114_create(ice->card,
+ err = snd_ak4114_create(ice->card,
ap192_ak4114_read,
ap192_ak4114_write,
ak4114_init_vals, ak4114_init_txcsb,
ice, &ak);
+ /* AK4114 in Revo cannot detect external rate correctly.
+ * No reason to stop capture stream due to incorrect checks */
+ ak->check_flags = AK4114_CHECK_NO_RATE;
+
+ return 0; /* error ignored; it's no fatal error */
}
-#endif /* BUILD_AK4114_AP192 */
static int __devinit revo_init(struct snd_ice1712 *ice)
{
@@ -557,6 +559,9 @@ static int __devinit revo_init(struct snd_ice1712 *ice)
if (err < 0)
return err;
+ /* unmute all codecs */
+ snd_ice1712_gpio_write_bits(ice, VT1724_REVO_MUTE,
+ VT1724_REVO_MUTE);
break;
}
@@ -588,11 +593,9 @@ static int __devinit revo_add_controls(struct snd_ice1712 *ice)
err = snd_ice1712_akm4xxx_build_controls(ice);
if (err < 0)
return err;
-#ifdef BUILD_AK4114_AP192
err = ap192_ak4114_init(ice);
if (err < 0)
return err;
-#endif
break;
}
return 0;
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index c52abd0bf22..048d99e25ab 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -155,7 +155,8 @@ DEFINE_REGSET(SP, 0x60); /* SPDIF out */
#define ICH_PCM_SPDIF_69 0x80000000 /* s/pdif pcm on slots 6&9 */
#define ICH_PCM_SPDIF_1011 0xc0000000 /* s/pdif pcm on slots 10&11 */
#define ICH_PCM_20BIT 0x00400000 /* 20-bit samples (ICH4) */
-#define ICH_PCM_246_MASK 0x00300000 /* 6 channels (not all chips) */
+#define ICH_PCM_246_MASK 0x00300000 /* chan mask (not all chips) */
+#define ICH_PCM_8 0x00300000 /* 8 channels (not all chips) */
#define ICH_PCM_6 0x00200000 /* 6 channels (not all chips) */
#define ICH_PCM_4 0x00100000 /* 4 channels (not all chips) */
#define ICH_PCM_2 0x00000000 /* 2 channels (stereo) */
@@ -382,6 +383,7 @@ struct intel8x0 {
unsigned multi4: 1,
multi6: 1,
+ multi8 :1,
dra: 1,
smp20bit: 1;
unsigned in_ac97_init: 1,
@@ -997,6 +999,8 @@ static void snd_intel8x0_setup_pcm_out(struct intel8x0 *chip,
cnt |= ICH_PCM_4;
else if (runtime->channels == 6)
cnt |= ICH_PCM_6;
+ else if (runtime->channels == 8)
+ cnt |= ICH_PCM_8;
if (chip->device_type == DEVICE_NFORCE) {
/* reset to 2ch once to keep the 6 channel data in alignment,
* to start from Front Left always
@@ -1106,6 +1110,16 @@ static struct snd_pcm_hw_constraint_list hw_constraints_channels6 = {
.mask = 0,
};
+static unsigned int channels8[] = {
+ 2, 4, 6, 8,
+};
+
+static struct snd_pcm_hw_constraint_list hw_constraints_channels8 = {
+ .count = ARRAY_SIZE(channels8),
+ .list = channels8,
+ .mask = 0,
+};
+
static int snd_intel8x0_pcm_open(struct snd_pcm_substream *substream, struct ichdev *ichdev)
{
struct intel8x0 *chip = snd_pcm_substream_chip(substream);
@@ -1136,7 +1150,12 @@ static int snd_intel8x0_playback_open(struct snd_pcm_substream *substream)
if (err < 0)
return err;
- if (chip->multi6) {
+ if (chip->multi8) {
+ runtime->hw.channels_max = 8;
+ snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ &hw_constraints_channels8);
+ } else if (chip->multi6) {
runtime->hw.channels_max = 6;
snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
&hw_constraints_channels6);
@@ -2203,8 +2222,11 @@ static int __devinit snd_intel8x0_mixer(struct intel8x0 *chip, int ac97_clock,
}
if (pbus->pcms[0].r[0].slots & (1 << AC97_SLOT_PCM_SLEFT)) {
chip->multi4 = 1;
- if (pbus->pcms[0].r[0].slots & (1 << AC97_SLOT_LFE))
+ if (pbus->pcms[0].r[0].slots & (1 << AC97_SLOT_LFE)) {
chip->multi6 = 1;
+ if (chip->ac97[0]->flags & AC97_HAS_8CH)
+ chip->multi8 = 1;
+ }
}
if (pbus->pcms[0].r[1].rslots[0]) {
chip->dra = 1;
@@ -2446,7 +2468,7 @@ static int snd_intel8x0_free(struct intel8x0 *chip)
pci_write_config_dword(chip->pci, 0x4c, val);
}
/* --- */
- synchronize_irq(chip->irq);
+
__hw_end:
if (chip->irq >= 0)
free_irq(chip->irq, chip);
@@ -2495,7 +2517,6 @@ static int intel8x0_suspend(struct pci_dev *pci, pm_message_t state)
chip->sdm_saved = igetbyte(chip, ICHREG(SDM));
if (chip->irq >= 0) {
- synchronize_irq(chip->irq);
free_irq(chip->irq, chip);
chip->irq = -1;
}
@@ -2648,7 +2669,7 @@ static void __devinit intel8x0_measure_ac97_clock(struct intel8x0 *chip)
t = stop_time.tv_sec - start_time.tv_sec;
t *= 1000000;
t += stop_time.tv_usec - start_time.tv_usec;
- printk(KERN_INFO "%s: measured %lu usecs\n", __FUNCTION__, t);
+ printk(KERN_INFO "%s: measured %lu usecs\n", __func__, t);
if (t == 0) {
snd_printk(KERN_ERR "?? calculation error..\n");
return;
diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c
index cadda8d6b70..faf674e671a 100644
--- a/sound/pci/intel8x0m.c
+++ b/sound/pci/intel8x0m.c
@@ -985,17 +985,15 @@ static int snd_intel8x0_free(struct intel8x0m *chip)
/* reset channels */
for (i = 0; i < chip->bdbars_count; i++)
iputbyte(chip, ICH_REG_OFF_CR + chip->ichd[i].reg_offset, ICH_RESETREGS);
- /* --- */
- synchronize_irq(chip->irq);
- __hw_end:
+ __hw_end:
+ if (chip->irq >= 0)
+ free_irq(chip->irq, chip);
if (chip->bdbars.area)
snd_dma_free_pages(&chip->bdbars);
if (chip->addr)
pci_iounmap(chip->pci, chip->addr);
if (chip->bmaddr)
pci_iounmap(chip->pci, chip->bmaddr);
- if (chip->irq >= 0)
- free_irq(chip->irq, chip);
pci_release_regions(chip->pci);
pci_disable_device(chip->pci);
kfree(chip);
@@ -1017,7 +1015,6 @@ static int intel8x0m_suspend(struct pci_dev *pci, pm_message_t state)
snd_pcm_suspend_all(chip->pcm[i]);
snd_ac97_suspend(chip->ac97);
if (chip->irq >= 0) {
- synchronize_irq(chip->irq);
free_irq(chip->irq, chip);
chip->irq = -1;
}
diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c
index 10c713d9ac4..f4c85b52bde 100644
--- a/sound/pci/korg1212/korg1212.c
+++ b/sound/pci/korg1212/korg1212.c
@@ -2102,7 +2102,6 @@ snd_korg1212_free(struct snd_korg1212 *korg1212)
snd_korg1212_TurnOffIdleMonitor(korg1212);
if (korg1212->irq >= 0) {
- synchronize_irq(korg1212->irq);
snd_korg1212_DisableCardInterrupts(korg1212);
free_irq(korg1212->irq, korg1212);
korg1212->irq = -1;
diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c
index 04fa0a68416..a536c59fbea 100644
--- a/sound/pci/maestro3.c
+++ b/sound/pci/maestro3.c
@@ -2068,7 +2068,7 @@ static int __devinit snd_m3_mixer(struct snd_m3 *chip)
{
struct snd_ac97_bus *pbus;
struct snd_ac97_template ac97;
- struct snd_ctl_elem_id id;
+ struct snd_ctl_elem_id elem_id;
int err;
static struct snd_ac97_bus_ops ops = {
.write = snd_m3_ac97_write,
@@ -2088,14 +2088,14 @@ static int __devinit snd_m3_mixer(struct snd_m3 *chip)
schedule_timeout_uninterruptible(msecs_to_jiffies(100));
snd_ac97_write(chip->ac97, AC97_PCM, 0);
- memset(&id, 0, sizeof(id));
- id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
- strcpy(id.name, "Master Playback Switch");
- chip->master_switch = snd_ctl_find_id(chip->card, &id);
- memset(&id, 0, sizeof(id));
- id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
- strcpy(id.name, "Master Playback Volume");
- chip->master_volume = snd_ctl_find_id(chip->card, &id);
+ memset(&elem_id, 0, sizeof(elem_id));
+ elem_id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ strcpy(elem_id.name, "Master Playback Switch");
+ chip->master_switch = snd_ctl_find_id(chip->card, &elem_id);
+ memset(&elem_id, 0, sizeof(elem_id));
+ elem_id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ strcpy(elem_id.name, "Master Playback Volume");
+ chip->master_volume = snd_ctl_find_id(chip->card, &elem_id);
return 0;
}
@@ -2542,10 +2542,8 @@ static int snd_m3_free(struct snd_m3 *chip)
vfree(chip->suspend_mem);
#endif
- if (chip->irq >= 0) {
- synchronize_irq(chip->irq);
+ if (chip->irq >= 0)
free_irq(chip->irq, chip);
- }
if (chip->iobase)
pci_release_regions(chip->pci);
@@ -2569,7 +2567,7 @@ static int m3_suspend(struct pci_dev *pci, pm_message_t state)
{
struct snd_card *card = pci_get_drvdata(pci);
struct snd_m3 *chip = card->private_data;
- int i, index;
+ int i, dsp_index;
if (chip->suspend_mem == NULL)
return 0;
@@ -2583,12 +2581,12 @@ static int m3_suspend(struct pci_dev *pci, pm_message_t state)
snd_m3_assp_halt(chip);
/* save dsp image */
- index = 0;
+ dsp_index = 0;
for (i = REV_B_CODE_MEMORY_BEGIN; i <= REV_B_CODE_MEMORY_END; i++)
- chip->suspend_mem[index++] =
+ chip->suspend_mem[dsp_index++] =
snd_m3_assp_read(chip, MEMTYPE_INTERNAL_CODE, i);
for (i = REV_B_DATA_MEMORY_BEGIN ; i <= REV_B_DATA_MEMORY_END; i++)
- chip->suspend_mem[index++] =
+ chip->suspend_mem[dsp_index++] =
snd_m3_assp_read(chip, MEMTYPE_INTERNAL_DATA, i);
pci_disable_device(pci);
@@ -2601,7 +2599,7 @@ static int m3_resume(struct pci_dev *pci)
{
struct snd_card *card = pci_get_drvdata(pci);
struct snd_m3 *chip = card->private_data;
- int i, index;
+ int i, dsp_index;
if (chip->suspend_mem == NULL)
return 0;
@@ -2625,13 +2623,13 @@ static int m3_resume(struct pci_dev *pci)
snd_m3_ac97_reset(chip);
/* restore dsp image */
- index = 0;
+ dsp_index = 0;
for (i = REV_B_CODE_MEMORY_BEGIN; i <= REV_B_CODE_MEMORY_END; i++)
snd_m3_assp_write(chip, MEMTYPE_INTERNAL_CODE, i,
- chip->suspend_mem[index++]);
+ chip->suspend_mem[dsp_index++]);
for (i = REV_B_DATA_MEMORY_BEGIN ; i <= REV_B_DATA_MEMORY_END; i++)
snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA, i,
- chip->suspend_mem[index++]);
+ chip->suspend_mem[dsp_index++]);
/* tell the dma engine to restart itself */
snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c
index 7ac654e381d..7efb838d18a 100644
--- a/sound/pci/nm256/nm256.c
+++ b/sound/pci/nm256/nm256.c
@@ -1439,7 +1439,7 @@ static int snd_nm256_free(struct nm256 *chip)
snd_nm256_capture_stop(chip);
if (chip->irq >= 0)
- synchronize_irq(chip->irq);
+ free_irq(chip->irq, chip);
if (chip->cport)
iounmap(chip->cport);
@@ -1447,8 +1447,6 @@ static int snd_nm256_free(struct nm256 *chip)
iounmap(chip->buffer);
release_and_free_resource(chip->res_cport);
release_and_free_resource(chip->res_buffer);
- if (chip->irq >= 0)
- free_irq(chip->irq, chip);
pci_disable_device(chip->pci);
kfree(chip->ac97_regs);
diff --git a/sound/pci/oxygen/cs4362a.h b/sound/pci/oxygen/cs4362a.h
new file mode 100644
index 00000000000..6a4fedf5e1e
--- /dev/null
+++ b/sound/pci/oxygen/cs4362a.h
@@ -0,0 +1,69 @@
+/* register 01h */
+#define CS4362A_PDN 0x01
+#define CS4362A_DAC1_DIS 0x02
+#define CS4362A_DAC2_DIS 0x04
+#define CS4362A_DAC3_DIS 0x08
+#define CS4362A_MCLKDIV 0x20
+#define CS4362A_FREEZE 0x40
+#define CS4362A_CPEN 0x80
+/* register 02h */
+#define CS4362A_DIF_MASK 0x70
+#define CS4362A_DIF_LJUST 0x00
+#define CS4362A_DIF_I2S 0x10
+#define CS4362A_DIF_RJUST_16 0x20
+#define CS4362A_DIF_RJUST_24 0x30
+#define CS4362A_DIF_RJUST_20 0x40
+#define CS4362A_DIF_RJUST_18 0x50
+/* register 03h */
+#define CS4362A_MUTEC_MASK 0x03
+#define CS4362A_MUTEC_6 0x00
+#define CS4362A_MUTEC_1 0x01
+#define CS4362A_MUTEC_3 0x03
+#define CS4362A_AMUTE 0x04
+#define CS4362A_MUTEC_POL 0x08
+#define CS4362A_RMP_UP 0x10
+#define CS4362A_SNGLVOL 0x20
+#define CS4362A_ZERO_CROSS 0x40
+#define CS4362A_SOFT_RAMP 0x80
+/* register 04h */
+#define CS4362A_RMP_DN 0x01
+#define CS4362A_DEM_MASK 0x06
+#define CS4362A_DEM_NONE 0x00
+#define CS4362A_DEM_44100 0x02
+#define CS4362A_DEM_48000 0x04
+#define CS4362A_DEM_32000 0x06
+#define CS4362A_FILT_SEL 0x10
+/* register 05h */
+#define CS4362A_INV_A1 0x01
+#define CS4362A_INV_B1 0x02
+#define CS4362A_INV_A2 0x04
+#define CS4362A_INV_B2 0x08
+#define CS4362A_INV_A3 0x10
+#define CS4362A_INV_B3 0x20
+/* register 06h */
+#define CS4362A_FM_MASK 0x03
+#define CS4362A_FM_SINGLE 0x00
+#define CS4362A_FM_DOUBLE 0x01
+#define CS4362A_FM_QUAD 0x02
+#define CS4362A_FM_DSD 0x03
+#define CS4362A_ATAPI_MASK 0x7c
+#define CS4362A_ATAPI_B_MUTE 0x00
+#define CS4362A_ATAPI_B_R 0x04
+#define CS4362A_ATAPI_B_L 0x08
+#define CS4362A_ATAPI_B_LR 0x0c
+#define CS4362A_ATAPI_A_MUTE 0x00
+#define CS4362A_ATAPI_A_R 0x10
+#define CS4362A_ATAPI_A_L 0x20
+#define CS4362A_ATAPI_A_LR 0x30
+#define CS4362A_ATAPI_MIX_LR_VOL 0x40
+#define CS4362A_A_EQ_B 0x80
+/* register 07h */
+#define CS4362A_VOL_MASK 0x7f
+#define CS4362A_MUTE 0x80
+/* register 08h: like 07h */
+/* registers 09h..0Bh: like 06h..08h */
+/* registers 0Ch..0Eh: like 06h..08h */
+/* register 12h */
+#define CS4362A_REV_MASK 0x07
+#define CS4362A_PART_MASK 0xf8
+#define CS4362A_PART_CS4362A 0x50
diff --git a/sound/pci/oxygen/cs4398.h b/sound/pci/oxygen/cs4398.h
new file mode 100644
index 00000000000..5faf5efc882
--- /dev/null
+++ b/sound/pci/oxygen/cs4398.h
@@ -0,0 +1,69 @@
+/* register 1 */
+#define CS4398_REV_MASK 0x07
+#define CS4398_PART_MASK 0xf8
+#define CS4398_PART_CS4398 0x70
+/* register 2 */
+#define CS4398_FM_MASK 0x03
+#define CS4398_FM_SINGLE 0x00
+#define CS4398_FM_DOUBLE 0x01
+#define CS4398_FM_QUAD 0x02
+#define CS4398_FM_DSD 0x03
+#define CS4398_DEM_MASK 0x0c
+#define CS4398_DEM_NONE 0x00
+#define CS4398_DEM_44100 0x04
+#define CS4398_DEM_48000 0x08
+#define CS4398_DEM_32000 0x0c
+#define CS4398_DIF_MASK 0x70
+#define CS4398_DIF_LJUST 0x00
+#define CS4398_DIF_I2S 0x10
+#define CS4398_DIF_RJUST_16 0x20
+#define CS4398_DIF_RJUST_24 0x30
+#define CS4398_DIF_RJUST_20 0x40
+#define CS4398_DIF_RJUST_18 0x50
+#define CS4398_DSD_SRC 0x80
+/* register 3 */
+#define CS4398_ATAPI_MASK 0x1f
+#define CS4398_ATAPI_B_MUTE 0x00
+#define CS4398_ATAPI_B_R 0x01
+#define CS4398_ATAPI_B_L 0x02
+#define CS4398_ATAPI_B_LR 0x03
+#define CS4398_ATAPI_A_MUTE 0x00
+#define CS4398_ATAPI_A_R 0x04
+#define CS4398_ATAPI_A_L 0x08
+#define CS4398_ATAPI_A_LR 0x0c
+#define CS4398_ATAPI_MIX_LR_VOL 0x10
+#define CS4398_INVERT_B 0x20
+#define CS4398_INVERT_A 0x40
+#define CS4398_VOL_B_EQ_A 0x80
+/* register 4 */
+#define CS4398_MUTEP_MASK 0x03
+#define CS4398_MUTEP_AUTO 0x00
+#define CS4398_MUTEP_LOW 0x02
+#define CS4398_MUTEP_HIGH 0x03
+#define CS4398_MUTE_B 0x08
+#define CS4398_MUTE_A 0x10
+#define CS4398_MUTEC_A_EQ_B 0x20
+#define CS4398_DAMUTE 0x40
+#define CS4398_PAMUTE 0x80
+/* register 5 */
+#define CS4398_VOL_A_MASK 0xff
+/* register 6 */
+#define CS4398_VOL_B_MASK 0xff
+/* register 7 */
+#define CS4398_DIR_DSD 0x01
+#define CS4398_FILT_SEL 0x04
+#define CS4398_RMP_DN 0x10
+#define CS4398_RMP_UP 0x20
+#define CS4398_ZERO_CROSS 0x40
+#define CS4398_SOFT_RAMP 0x80
+/* register 8 */
+#define CS4398_MCLKDIV3 0x08
+#define CS4398_MCLKDIV2 0x10
+#define CS4398_FREEZE 0x20
+#define CS4398_CPEN 0x40
+#define CS4398_PDN 0x80
+/* register 9 */
+#define CS4398_DSD_PM_EN 0x01
+#define CS4398_DSD_PM_MODE 0x02
+#define CS4398_INVALID_DSD 0x04
+#define CS4398_STATIC_DSD 0x08
diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c
index 666f69a3312..090dd4354a2 100644
--- a/sound/pci/oxygen/hifier.c
+++ b/sound/pci/oxygen/hifier.c
@@ -66,12 +66,12 @@ static void hifier_init(struct oxygen *chip)
{
struct hifier_data *data = chip->model_data;
- data->ak4396_ctl2 = AK4396_DEM_OFF | AK4396_DFS_NORMAL;
+ data->ak4396_ctl2 = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL;
ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN);
ak4396_write(chip, AK4396_CONTROL_2, data->ak4396_ctl2);
ak4396_write(chip, AK4396_CONTROL_3, AK4396_PCM);
- ak4396_write(chip, AK4396_LCH_ATT, 0xff);
- ak4396_write(chip, AK4396_RCH_ATT, 0xff);
+ ak4396_write(chip, AK4396_LCH_ATT, 0);
+ ak4396_write(chip, AK4396_RCH_ATT, 0);
snd_component_add(chip->card, "AK4396");
snd_component_add(chip->card, "CS5340");
@@ -127,22 +127,8 @@ static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0);
static int hifier_control_filter(struct snd_kcontrol_new *template)
{
- if (!strcmp(template->name, "Master Playback Volume")) {
- template->access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ;
- template->tlv.p = ak4396_db_scale;
- } else if (!strcmp(template->name, "Stereo Upmixing")) {
+ if (!strcmp(template->name, "Stereo Upmixing"))
return 1; /* stereo only - we don't need upmixing */
- } else if (!strcmp(template->name,
- SNDRV_CTL_NAME_IEC958("", CAPTURE, MASK)) ||
- !strcmp(template->name,
- SNDRV_CTL_NAME_IEC958("", CAPTURE, DEFAULT))) {
- return 1; /* no digital input */
- }
- return 0;
-}
-
-static int hifier_mixer_init(struct oxygen *chip)
-{
return 0;
}
@@ -153,18 +139,20 @@ static const struct oxygen_model model_hifier = {
.owner = THIS_MODULE,
.init = hifier_init,
.control_filter = hifier_control_filter,
- .mixer_init = hifier_mixer_init,
.cleanup = hifier_cleanup,
.set_dac_params = set_ak4396_params,
.set_adc_params = set_cs5340_params,
.update_dac_volume = update_ak4396_volume,
.update_dac_mute = update_ak4396_mute,
+ .dac_tlv = ak4396_db_scale,
.model_data_size = sizeof(struct hifier_data),
+ .pcm_dev_cfg = PLAYBACK_0_TO_I2S |
+ PLAYBACK_1_TO_SPDIF |
+ CAPTURE_0_FROM_I2S_1,
.dac_channels = 2,
- .used_channels = OXYGEN_CHANNEL_A |
- OXYGEN_CHANNEL_SPDIF |
- OXYGEN_CHANNEL_MULTICH,
- .function_flags = 0,
+ .dac_volume_min = 0,
+ .dac_volume_max = 255,
+ .function_flags = OXYGEN_FUNCTION_SPI,
.dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
.adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
};
@@ -181,7 +169,7 @@ static int __devinit hifier_probe(struct pci_dev *pci,
++dev;
return -ENOENT;
}
- err = oxygen_pci_probe(pci, index[dev], id[dev], 0, &model_hifier);
+ err = oxygen_pci_probe(pci, index[dev], id[dev], &model_hifier);
if (err >= 0)
++dev;
return err;
diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c
index 9a9941bb046..63f185c1ed1 100644
--- a/sound/pci/oxygen/oxygen.c
+++ b/sound/pci/oxygen/oxygen.c
@@ -39,7 +39,7 @@
#include <sound/tlv.h>
#include "oxygen.h"
#include "ak4396.h"
-#include "cm9780.h"
+#include "wm8785.h"
MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
MODULE_DESCRIPTION("C-Media CMI8788 driver");
@@ -78,49 +78,6 @@ MODULE_DEVICE_TABLE(pci, oxygen_ids);
#define GPIO_AK5385_DFS_DOUBLE 0x0001
#define GPIO_AK5385_DFS_QUAD 0x0002
-#define GPIO_LINE_MUTE CM9780_GPO0
-
-#define WM8785_R0 0
-#define WM8785_R1 1
-#define WM8785_R2 2
-#define WM8785_R7 7
-
-/* R0 */
-#define WM8785_MCR_MASK 0x007
-#define WM8785_MCR_SLAVE 0x000
-#define WM8785_MCR_MASTER_128 0x001
-#define WM8785_MCR_MASTER_192 0x002
-#define WM8785_MCR_MASTER_256 0x003
-#define WM8785_MCR_MASTER_384 0x004
-#define WM8785_MCR_MASTER_512 0x005
-#define WM8785_MCR_MASTER_768 0x006
-#define WM8785_OSR_MASK 0x018
-#define WM8785_OSR_SINGLE 0x000
-#define WM8785_OSR_DOUBLE 0x008
-#define WM8785_OSR_QUAD 0x010
-#define WM8785_FORMAT_MASK 0x060
-#define WM8785_FORMAT_RJUST 0x000
-#define WM8785_FORMAT_LJUST 0x020
-#define WM8785_FORMAT_I2S 0x040
-#define WM8785_FORMAT_DSP 0x060
-/* R1 */
-#define WM8785_WL_MASK 0x003
-#define WM8785_WL_16 0x000
-#define WM8785_WL_20 0x001
-#define WM8785_WL_24 0x002
-#define WM8785_WL_32 0x003
-#define WM8785_LRP 0x004
-#define WM8785_BCLKINV 0x008
-#define WM8785_LRSWAP 0x010
-#define WM8785_DEVNO_MASK 0x0e0
-/* R2 */
-#define WM8785_HPFR 0x001
-#define WM8785_HPFL 0x002
-#define WM8785_SDODIS 0x004
-#define WM8785_PWRDNR 0x008
-#define WM8785_PWRDNL 0x010
-#define WM8785_TDM_MASK 0x1c0
-
struct generic_data {
u8 ak4396_ctl2;
};
@@ -155,7 +112,7 @@ static void ak4396_init(struct oxygen *chip)
struct generic_data *data = chip->model_data;
unsigned int i;
- data->ak4396_ctl2 = AK4396_DEM_OFF | AK4396_DFS_NORMAL;
+ data->ak4396_ctl2 = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL;
for (i = 0; i < 4; ++i) {
ak4396_write(chip, i,
AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN);
@@ -163,8 +120,8 @@ static void ak4396_init(struct oxygen *chip)
AK4396_CONTROL_2, data->ak4396_ctl2);
ak4396_write(chip, i,
AK4396_CONTROL_3, AK4396_PCM);
- ak4396_write(chip, i, AK4396_LCH_ATT, 0xff);
- ak4396_write(chip, i, AK4396_RCH_ATT, 0xff);
+ ak4396_write(chip, i, AK4396_LCH_ATT, 0);
+ ak4396_write(chip, i, AK4396_RCH_ATT, 0);
}
snd_component_add(chip->card, "AK4396");
}
@@ -185,23 +142,16 @@ static void wm8785_init(struct oxygen *chip)
snd_component_add(chip->card, "WM8785");
}
-static void cmi9780_init(struct oxygen *chip)
-{
- oxygen_ac97_clear_bits(chip, 0, CM9780_GPIO_STATUS, GPIO_LINE_MUTE);
-}
-
static void generic_init(struct oxygen *chip)
{
ak4396_init(chip);
wm8785_init(chip);
- cmi9780_init(chip);
}
static void meridian_init(struct oxygen *chip)
{
ak4396_init(chip);
ak5385_init(chip);
- cmi9780_init(chip);
}
static void generic_cleanup(struct oxygen *chip)
@@ -297,59 +247,32 @@ static void set_ak5385_params(struct oxygen *chip,
value, GPIO_AK5385_DFS_MASK);
}
-static void cmi9780_switch_hook(struct oxygen *chip, unsigned int codec,
- unsigned int reg, int mute)
-{
- if (codec != 0)
- return;
- switch (reg) {
- case AC97_LINE:
- oxygen_write_ac97_masked(chip, 0, CM9780_GPIO_STATUS,
- mute ? GPIO_LINE_MUTE : 0,
- GPIO_LINE_MUTE);
- break;
- case AC97_MIC:
- case AC97_CD:
- case AC97_AUX:
- if (!mute)
- oxygen_ac97_set_bits(chip, 0, CM9780_GPIO_STATUS,
- GPIO_LINE_MUTE);
- break;
- }
-}
-
static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0);
-static int ak4396_control_filter(struct snd_kcontrol_new *template)
-{
- if (!strcmp(template->name, "Master Playback Volume")) {
- template->access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ;
- template->tlv.p = ak4396_db_scale;
- }
- return 0;
-}
-
static const struct oxygen_model model_generic = {
.shortname = "C-Media CMI8788",
.longname = "C-Media Oxygen HD Audio",
.chip = "CMI8788",
.owner = THIS_MODULE,
.init = generic_init,
- .control_filter = ak4396_control_filter,
.cleanup = generic_cleanup,
.set_dac_params = set_ak4396_params,
.set_adc_params = set_wm8785_params,
.update_dac_volume = update_ak4396_volume,
.update_dac_mute = update_ak4396_mute,
- .ac97_switch_hook = cmi9780_switch_hook,
+ .dac_tlv = ak4396_db_scale,
.model_data_size = sizeof(struct generic_data),
+ .pcm_dev_cfg = PLAYBACK_0_TO_I2S |
+ PLAYBACK_1_TO_SPDIF |
+ PLAYBACK_2_TO_AC97_1 |
+ CAPTURE_0_FROM_I2S_1 |
+ CAPTURE_1_FROM_SPDIF |
+ CAPTURE_2_FROM_AC97_1,
.dac_channels = 8,
- .used_channels = OXYGEN_CHANNEL_A |
- OXYGEN_CHANNEL_C |
- OXYGEN_CHANNEL_SPDIF |
- OXYGEN_CHANNEL_MULTICH |
- OXYGEN_CHANNEL_AC97,
- .function_flags = OXYGEN_FUNCTION_ENABLE_SPI_4_5,
+ .dac_volume_min = 0,
+ .dac_volume_max = 255,
+ .function_flags = OXYGEN_FUNCTION_SPI |
+ OXYGEN_FUNCTION_ENABLE_SPI_4_5,
.dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
.adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
};
@@ -359,21 +282,25 @@ static const struct oxygen_model model_meridian = {
.chip = "CMI8788",
.owner = THIS_MODULE,
.init = meridian_init,
- .control_filter = ak4396_control_filter,
.cleanup = generic_cleanup,
.set_dac_params = set_ak4396_params,
.set_adc_params = set_ak5385_params,
.update_dac_volume = update_ak4396_volume,
.update_dac_mute = update_ak4396_mute,
- .ac97_switch_hook = cmi9780_switch_hook,
+ .dac_tlv = ak4396_db_scale,
.model_data_size = sizeof(struct generic_data),
+ .pcm_dev_cfg = PLAYBACK_0_TO_I2S |
+ PLAYBACK_1_TO_SPDIF |
+ PLAYBACK_2_TO_AC97_1 |
+ CAPTURE_0_FROM_I2S_2 |
+ CAPTURE_1_FROM_SPDIF |
+ CAPTURE_2_FROM_AC97_1,
.dac_channels = 8,
- .used_channels = OXYGEN_CHANNEL_B |
- OXYGEN_CHANNEL_C |
- OXYGEN_CHANNEL_SPDIF |
- OXYGEN_CHANNEL_MULTICH |
- OXYGEN_CHANNEL_AC97,
- .function_flags = OXYGEN_FUNCTION_ENABLE_SPI_4_5,
+ .dac_volume_min = 0,
+ .dac_volume_max = 255,
+ .misc_flags = OXYGEN_MISC_MIDI,
+ .function_flags = OXYGEN_FUNCTION_SPI |
+ OXYGEN_FUNCTION_ENABLE_SPI_4_5,
.dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
.adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
};
@@ -392,7 +319,7 @@ static int __devinit generic_oxygen_probe(struct pci_dev *pci,
return -ENOENT;
}
is_meridian = pci_id->driver_data;
- err = oxygen_pci_probe(pci, index[dev], id[dev], is_meridian,
+ err = oxygen_pci_probe(pci, index[dev], id[dev],
is_meridian ? &model_meridian : &model_generic);
if (err >= 0)
++dev;
diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h
index ad50fb8b206..a71c6e05926 100644
--- a/sound/pci/oxygen/oxygen.h
+++ b/sound/pci/oxygen/oxygen.h
@@ -16,6 +16,16 @@
#define PCM_AC97 5
#define PCM_COUNT 6
+/* model-specific configuration of outputs/inputs */
+#define PLAYBACK_0_TO_I2S 0x001
+#define PLAYBACK_1_TO_SPDIF 0x004
+#define PLAYBACK_2_TO_AC97_1 0x008
+#define CAPTURE_0_FROM_I2S_1 0x010
+#define CAPTURE_0_FROM_I2S_2 0x020
+#define CAPTURE_1_FROM_SPDIF 0x080
+#define CAPTURE_2_FROM_I2S_2 0x100
+#define CAPTURE_2_FROM_AC97_1 0x200
+
enum {
CONTROL_SPDIF_PCM,
CONTROL_SPDIF_INPUT_BITS,
@@ -87,12 +97,16 @@ struct oxygen_model {
struct snd_pcm_hw_params *params);
void (*update_dac_volume)(struct oxygen *chip);
void (*update_dac_mute)(struct oxygen *chip);
- void (*ac97_switch_hook)(struct oxygen *chip, unsigned int codec,
- unsigned int reg, int mute);
void (*gpio_changed)(struct oxygen *chip);
+ void (*ac97_switch)(struct oxygen *chip,
+ unsigned int reg, unsigned int mute);
+ const unsigned int *dac_tlv;
size_t model_data_size;
+ unsigned int pcm_dev_cfg;
u8 dac_channels;
- u8 used_channels;
+ u8 dac_volume_min;
+ u8 dac_volume_max;
+ u8 misc_flags;
u8 function_flags;
u16 dac_i2s_format;
u16 adc_i2s_format;
@@ -100,7 +114,7 @@ struct oxygen_model {
/* oxygen_lib.c */
-int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, int midi,
+int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
const struct oxygen_model *model);
void oxygen_pci_remove(struct pci_dev *pci);
@@ -137,6 +151,7 @@ void oxygen_write_ac97_masked(struct oxygen *chip, unsigned int codec,
unsigned int index, u16 data, u16 mask);
void oxygen_write_spi(struct oxygen *chip, u8 control, unsigned int data);
+void oxygen_write_i2c(struct oxygen *chip, u8 device, u8 map, u8 data);
static inline void oxygen_set_bits8(struct oxygen *chip,
unsigned int reg, u8 value)
diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c
index 74e23ef9c94..5569606ee87 100644
--- a/sound/pci/oxygen/oxygen_io.c
+++ b/sound/pci/oxygen/oxygen_io.c
@@ -190,12 +190,31 @@ void oxygen_write_spi(struct oxygen *chip, u8 control, unsigned int data)
--count;
}
- spin_lock_irq(&chip->reg_lock);
oxygen_write8(chip, OXYGEN_SPI_DATA1, data);
oxygen_write8(chip, OXYGEN_SPI_DATA2, data >> 8);
if (control & OXYGEN_SPI_DATA_LENGTH_3)
oxygen_write8(chip, OXYGEN_SPI_DATA3, data >> 16);
oxygen_write8(chip, OXYGEN_SPI_CONTROL, control);
- spin_unlock_irq(&chip->reg_lock);
}
EXPORT_SYMBOL(oxygen_write_spi);
+
+void oxygen_write_i2c(struct oxygen *chip, u8 device, u8 map, u8 data)
+{
+ unsigned long timeout;
+
+ /* should not need more than about 300 us */
+ timeout = jiffies + msecs_to_jiffies(1);
+ do {
+ if (!(oxygen_read16(chip, OXYGEN_2WIRE_BUS_STATUS)
+ & OXYGEN_2WIRE_BUSY))
+ break;
+ udelay(1);
+ cond_resched();
+ } while (time_after_eq(timeout, jiffies));
+
+ oxygen_write8(chip, OXYGEN_2WIRE_MAP, map);
+ oxygen_write8(chip, OXYGEN_2WIRE_DATA, data);
+ oxygen_write8(chip, OXYGEN_2WIRE_CONTROL,
+ device | OXYGEN_2WIRE_DIR_WRITE);
+}
+EXPORT_SYMBOL(oxygen_write_i2c);
diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c
index 78c21155218..897697d4350 100644
--- a/sound/pci/oxygen/oxygen_lib.c
+++ b/sound/pci/oxygen/oxygen_lib.c
@@ -221,7 +221,8 @@ static void oxygen_init(struct oxygen *chip)
chip->dac_routing = 1;
for (i = 0; i < 8; ++i)
- chip->dac_volume[i] = 0xff;
+ chip->dac_volume[i] = chip->model->dac_volume_min;
+ chip->dac_mute = 1;
chip->spdif_playback_enable = 1;
chip->spdif_bits = OXYGEN_SPDIF_C | OXYGEN_SPDIF_ORIGINAL |
(IEC958_AES1_CON_PCM_CODER << OXYGEN_SPDIF_CATEGORY_SHIFT);
@@ -240,12 +241,12 @@ static void oxygen_init(struct oxygen *chip)
chip->has_ac97_0 = (i & OXYGEN_AC97_CODEC_0) != 0;
chip->has_ac97_1 = (i & OXYGEN_AC97_CODEC_1) != 0;
- oxygen_set_bits8(chip, OXYGEN_FUNCTION,
- OXYGEN_FUNCTION_RESET_CODEC |
- chip->model->function_flags);
oxygen_write8_masked(chip, OXYGEN_FUNCTION,
- OXYGEN_FUNCTION_SPI,
- OXYGEN_FUNCTION_2WIRE_SPI_MASK);
+ OXYGEN_FUNCTION_RESET_CODEC |
+ chip->model->function_flags,
+ OXYGEN_FUNCTION_RESET_CODEC |
+ OXYGEN_FUNCTION_2WIRE_SPI_MASK |
+ OXYGEN_FUNCTION_ENABLE_SPI_4_5);
oxygen_write8(chip, OXYGEN_DMA_STATUS, 0);
oxygen_write8(chip, OXYGEN_DMA_PAUSE, 0);
oxygen_write8(chip, OXYGEN_PLAY_CHANNELS,
@@ -253,11 +254,13 @@ static void oxygen_init(struct oxygen *chip)
OXYGEN_DMA_A_BURST_8 |
OXYGEN_DMA_MULTICH_BURST_8);
oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, 0);
- oxygen_write8_masked(chip, OXYGEN_MISC, 0,
+ oxygen_write8_masked(chip, OXYGEN_MISC,
+ chip->model->misc_flags,
OXYGEN_MISC_WRITE_PCI_SUBID |
OXYGEN_MISC_REC_C_FROM_SPDIF |
OXYGEN_MISC_REC_B_FROM_AC97 |
- OXYGEN_MISC_REC_A_FROM_MULTICH);
+ OXYGEN_MISC_REC_A_FROM_MULTICH |
+ OXYGEN_MISC_MIDI);
oxygen_write8(chip, OXYGEN_REC_FORMAT,
(OXYGEN_FORMAT_16 << OXYGEN_REC_FORMAT_A_SHIFT) |
(OXYGEN_FORMAT_16 << OXYGEN_REC_FORMAT_B_SHIFT) |
@@ -267,35 +270,49 @@ static void oxygen_init(struct oxygen *chip)
(OXYGEN_FORMAT_16 << OXYGEN_MULTICH_FORMAT_SHIFT));
oxygen_write8(chip, OXYGEN_REC_CHANNELS, OXYGEN_REC_CHANNELS_2_2_2);
oxygen_write16(chip, OXYGEN_I2S_MULTICH_FORMAT,
- OXYGEN_RATE_48000 | OXYGEN_I2S_FORMAT_LJUST |
- OXYGEN_I2S_MCLK_128 | OXYGEN_I2S_BITS_16 |
- OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64);
- oxygen_write16(chip, OXYGEN_I2S_A_FORMAT,
- OXYGEN_RATE_48000 | OXYGEN_I2S_FORMAT_LJUST |
- OXYGEN_I2S_MCLK_128 | OXYGEN_I2S_BITS_16 |
- OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64);
- oxygen_write16(chip, OXYGEN_I2S_B_FORMAT,
- OXYGEN_RATE_48000 | OXYGEN_I2S_FORMAT_LJUST |
- OXYGEN_I2S_MCLK_128 | OXYGEN_I2S_BITS_16 |
+ OXYGEN_RATE_48000 | chip->model->dac_i2s_format |
+ OXYGEN_I2S_MCLK_256 | OXYGEN_I2S_BITS_16 |
OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64);
+ if (chip->model->pcm_dev_cfg & CAPTURE_0_FROM_I2S_1)
+ oxygen_write16(chip, OXYGEN_I2S_A_FORMAT,
+ OXYGEN_RATE_48000 | chip->model->adc_i2s_format |
+ OXYGEN_I2S_MCLK_256 | OXYGEN_I2S_BITS_16 |
+ OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64);
+ else
+ oxygen_write16(chip, OXYGEN_I2S_A_FORMAT,
+ OXYGEN_I2S_MASTER | OXYGEN_I2S_MUTE_MCLK);
+ if (chip->model->pcm_dev_cfg & (CAPTURE_0_FROM_I2S_2 |
+ CAPTURE_2_FROM_I2S_2))
+ oxygen_write16(chip, OXYGEN_I2S_B_FORMAT,
+ OXYGEN_RATE_48000 | chip->model->adc_i2s_format |
+ OXYGEN_I2S_MCLK_256 | OXYGEN_I2S_BITS_16 |
+ OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64);
+ else
+ oxygen_write16(chip, OXYGEN_I2S_B_FORMAT,
+ OXYGEN_I2S_MASTER | OXYGEN_I2S_MUTE_MCLK);
oxygen_write16(chip, OXYGEN_I2S_C_FORMAT,
- OXYGEN_RATE_48000 | OXYGEN_I2S_FORMAT_LJUST |
- OXYGEN_I2S_MCLK_128 | OXYGEN_I2S_BITS_16 |
- OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64);
- oxygen_write32_masked(chip, OXYGEN_SPDIF_CONTROL,
- OXYGEN_SPDIF_SENSE_MASK |
- OXYGEN_SPDIF_LOCK_MASK |
- OXYGEN_SPDIF_RATE_MASK |
- OXYGEN_SPDIF_LOCK_PAR |
- OXYGEN_SPDIF_IN_CLOCK_96,
- OXYGEN_SPDIF_OUT_ENABLE |
- OXYGEN_SPDIF_LOOPBACK |
- OXYGEN_SPDIF_SENSE_MASK |
- OXYGEN_SPDIF_LOCK_MASK |
- OXYGEN_SPDIF_RATE_MASK |
- OXYGEN_SPDIF_SENSE_PAR |
- OXYGEN_SPDIF_LOCK_PAR |
- OXYGEN_SPDIF_IN_CLOCK_MASK);
+ OXYGEN_I2S_MASTER | OXYGEN_I2S_MUTE_MCLK);
+ oxygen_clear_bits32(chip, OXYGEN_SPDIF_CONTROL,
+ OXYGEN_SPDIF_OUT_ENABLE |
+ OXYGEN_SPDIF_LOOPBACK);
+ if (chip->model->pcm_dev_cfg & CAPTURE_1_FROM_SPDIF)
+ oxygen_write32_masked(chip, OXYGEN_SPDIF_CONTROL,
+ OXYGEN_SPDIF_SENSE_MASK |
+ OXYGEN_SPDIF_LOCK_MASK |
+ OXYGEN_SPDIF_RATE_MASK |
+ OXYGEN_SPDIF_LOCK_PAR |
+ OXYGEN_SPDIF_IN_CLOCK_96,
+ OXYGEN_SPDIF_SENSE_MASK |
+ OXYGEN_SPDIF_LOCK_MASK |
+ OXYGEN_SPDIF_RATE_MASK |
+ OXYGEN_SPDIF_SENSE_PAR |
+ OXYGEN_SPDIF_LOCK_PAR |
+ OXYGEN_SPDIF_IN_CLOCK_MASK);
+ else
+ oxygen_clear_bits32(chip, OXYGEN_SPDIF_CONTROL,
+ OXYGEN_SPDIF_SENSE_MASK |
+ OXYGEN_SPDIF_LOCK_MASK |
+ OXYGEN_SPDIF_RATE_MASK);
oxygen_write32(chip, OXYGEN_SPDIF_OUTPUT_BITS, chip->spdif_bits);
oxygen_clear_bits8(chip, OXYGEN_MPU401_CONTROL, OXYGEN_MPU401_LOOPBACK);
oxygen_write8(chip, OXYGEN_GPI_INTERRUPT_MASK, 0);
@@ -318,9 +335,12 @@ static void oxygen_init(struct oxygen *chip)
(2 << OXYGEN_A_MONITOR_ROUTE_2_SHIFT) |
(3 << OXYGEN_A_MONITOR_ROUTE_3_SHIFT));
- oxygen_write8(chip, OXYGEN_AC97_INTERRUPT_MASK,
- OXYGEN_AC97_INT_READ_DONE |
- OXYGEN_AC97_INT_WRITE_DONE);
+ if (chip->has_ac97_0 | chip->has_ac97_1)
+ oxygen_write8(chip, OXYGEN_AC97_INTERRUPT_MASK,
+ OXYGEN_AC97_INT_READ_DONE |
+ OXYGEN_AC97_INT_WRITE_DONE);
+ else
+ oxygen_write8(chip, OXYGEN_AC97_INTERRUPT_MASK, 0);
oxygen_write32(chip, OXYGEN_AC97_OUT_CONFIG, 0);
oxygen_write32(chip, OXYGEN_AC97_IN_CONFIG, 0);
if (!(chip->has_ac97_0 | chip->has_ac97_1))
@@ -351,6 +371,8 @@ static void oxygen_init(struct oxygen *chip)
oxygen_write_ac97(chip, 0, AC97_REC_GAIN, 0x8000);
oxygen_write_ac97(chip, 0, AC97_CENTER_LFE_MASTER, 0x8080);
oxygen_write_ac97(chip, 0, AC97_SURROUND_MASTER, 0x8080);
+ oxygen_ac97_clear_bits(chip, 0, CM9780_GPIO_STATUS,
+ CM9780_GPO0);
/* power down unused ADCs and DACs */
oxygen_ac97_set_bits(chip, 0, AC97_POWERDOWN,
AC97_PD_PR0 | AC97_PD_PR1);
@@ -388,10 +410,8 @@ static void oxygen_card_free(struct snd_card *card)
oxygen_write16(chip, OXYGEN_DMA_STATUS, 0);
oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, 0);
spin_unlock_irq(&chip->reg_lock);
- if (chip->irq >= 0) {
+ if (chip->irq >= 0)
free_irq(chip->irq, chip);
- synchronize_irq(chip->irq);
- }
flush_scheduled_work();
chip->model->cleanup(chip);
mutex_destroy(&chip->mutex);
@@ -400,7 +420,7 @@ static void oxygen_card_free(struct snd_card *card)
}
int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
- int midi, const struct oxygen_model *model)
+ const struct oxygen_model *model)
{
struct snd_card *card;
struct oxygen *chip;
@@ -472,9 +492,7 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
if (err < 0)
goto err_card;
- oxygen_write8_masked(chip, OXYGEN_MISC,
- midi ? OXYGEN_MISC_MIDI : 0, OXYGEN_MISC_MIDI);
- if (midi) {
+ if (model->misc_flags & OXYGEN_MISC_MIDI) {
err = snd_mpu401_uart_new(card, 0, MPU401_HW_CMIPCI,
chip->addr + OXYGEN_MPU401,
MPU401_INFO_INTEGRATED, 0, 0,
@@ -486,7 +504,10 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
oxygen_proc_init(chip);
spin_lock_irq(&chip->reg_lock);
- chip->interrupt_mask |= OXYGEN_INT_SPDIF_IN_DETECT | OXYGEN_INT_AC97;
+ if (chip->model->pcm_dev_cfg & CAPTURE_1_FROM_SPDIF)
+ chip->interrupt_mask |= OXYGEN_INT_SPDIF_IN_DETECT;
+ if (chip->has_ac97_0 | chip->has_ac97_1)
+ chip->interrupt_mask |= OXYGEN_INT_AC97;
oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, chip->interrupt_mask);
spin_unlock_irq(&chip->reg_lock);
diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c
index a8e4623415d..cc0cddadd58 100644
--- a/sound/pci/oxygen/oxygen_mixer.c
+++ b/sound/pci/oxygen/oxygen_mixer.c
@@ -32,8 +32,8 @@ static int dac_volume_info(struct snd_kcontrol *ctl,
info->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
info->count = chip->model->dac_channels;
- info->value.integer.min = 0;
- info->value.integer.max = 0xff;
+ info->value.integer.min = chip->model->dac_volume_min;
+ info->value.integer.max = chip->model->dac_volume_max;
return 0;
}
@@ -446,6 +446,50 @@ static int spdif_loopback_put(struct snd_kcontrol *ctl,
return changed;
}
+static int monitor_volume_info(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_info *info)
+{
+ info->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ info->count = 1;
+ info->value.integer.min = 0;
+ info->value.integer.max = 1;
+ return 0;
+}
+
+static int monitor_get(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_value *value)
+{
+ struct oxygen *chip = ctl->private_data;
+ u8 bit = ctl->private_value;
+ int invert = ctl->private_value & (1 << 8);
+
+ value->value.integer.value[0] =
+ !!invert ^ !!(oxygen_read8(chip, OXYGEN_ADC_MONITOR) & bit);
+ return 0;
+}
+
+static int monitor_put(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_value *value)
+{
+ struct oxygen *chip = ctl->private_data;
+ u8 bit = ctl->private_value;
+ int invert = ctl->private_value & (1 << 8);
+ u8 oldreg, newreg;
+ int changed;
+
+ spin_lock_irq(&chip->reg_lock);
+ oldreg = oxygen_read8(chip, OXYGEN_ADC_MONITOR);
+ if ((!!value->value.integer.value[0] ^ !!invert) != 0)
+ newreg = oldreg | bit;
+ else
+ newreg = oldreg & ~bit;
+ changed = newreg != oldreg;
+ if (changed)
+ oxygen_write8(chip, OXYGEN_ADC_MONITOR, newreg);
+ spin_unlock_irq(&chip->reg_lock);
+ return changed;
+}
+
static int ac97_switch_get(struct snd_kcontrol *ctl,
struct snd_ctl_elem_value *value)
{
@@ -466,6 +510,21 @@ static int ac97_switch_get(struct snd_kcontrol *ctl,
return 0;
}
+static void mute_ac97_ctl(struct oxygen *chip, unsigned int control)
+{
+ unsigned int priv_idx = chip->controls[control]->private_value & 0xff;
+ u16 value;
+
+ value = oxygen_read_ac97(chip, 0, priv_idx);
+ if (!(value & 0x8000)) {
+ oxygen_write_ac97(chip, 0, priv_idx, value | 0x8000);
+ if (chip->model->ac97_switch)
+ chip->model->ac97_switch(chip, priv_idx, 0x8000);
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->controls[control]->id);
+ }
+}
+
static int ac97_switch_put(struct snd_kcontrol *ctl,
struct snd_ctl_elem_value *value)
{
@@ -487,9 +546,24 @@ static int ac97_switch_put(struct snd_kcontrol *ctl,
change = newreg != oldreg;
if (change) {
oxygen_write_ac97(chip, codec, index, newreg);
- if (bitnr == 15 && chip->model->ac97_switch_hook)
- chip->model->ac97_switch_hook(chip, codec, index,
- newreg & 0x8000);
+ if (codec == 0 && chip->model->ac97_switch)
+ chip->model->ac97_switch(chip, index, newreg & 0x8000);
+ if (index == AC97_LINE) {
+ oxygen_write_ac97_masked(chip, 0, CM9780_GPIO_STATUS,
+ newreg & 0x8000 ?
+ CM9780_GPO0 : 0, CM9780_GPO0);
+ if (!(newreg & 0x8000)) {
+ mute_ac97_ctl(chip, CONTROL_MIC_CAPTURE_SWITCH);
+ mute_ac97_ctl(chip, CONTROL_CD_CAPTURE_SWITCH);
+ mute_ac97_ctl(chip, CONTROL_AUX_CAPTURE_SWITCH);
+ }
+ } else if ((index == AC97_MIC || index == AC97_CD ||
+ index == AC97_VIDEO || index == AC97_AUX) &&
+ bitnr == 15 && !(newreg & 0x8000)) {
+ mute_ac97_ctl(chip, CONTROL_LINE_CAPTURE_SWITCH);
+ oxygen_write_ac97_masked(chip, 0, CM9780_GPIO_STATUS,
+ CM9780_GPO0, CM9780_GPO0);
+ }
}
mutex_unlock(&chip->mutex);
return change;
@@ -608,6 +682,7 @@ static int ac97_fp_rec_volume_put(struct snd_kcontrol *ctl,
.private_value = ((codec) << 24) | (index), \
}
+static DECLARE_TLV_DB_SCALE(monitor_db_scale, -1000, 1000, 0);
static DECLARE_TLV_DB_SCALE(ac97_db_scale, -3450, 150, 0);
static DECLARE_TLV_DB_SCALE(ac97_rec_db_scale, 0, 150, 0);
@@ -667,6 +742,9 @@ static const struct snd_kcontrol_new controls[] = {
.get = spdif_pcm_get,
.put = spdif_pcm_put,
},
+};
+
+static const struct snd_kcontrol_new spdif_input_controls[] = {
{
.iface = SNDRV_CTL_ELEM_IFACE_PCM,
.device = 1,
@@ -692,11 +770,118 @@ static const struct snd_kcontrol_new controls[] = {
},
};
+static const struct {
+ unsigned int pcm_dev;
+ struct snd_kcontrol_new controls[2];
+} monitor_controls[] = {
+ {
+ .pcm_dev = CAPTURE_0_FROM_I2S_1,
+ .controls = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Analog Input Monitor Switch",
+ .info = snd_ctl_boolean_mono_info,
+ .get = monitor_get,
+ .put = monitor_put,
+ .private_value = OXYGEN_ADC_MONITOR_A,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Analog Input Monitor Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ,
+ .info = monitor_volume_info,
+ .get = monitor_get,
+ .put = monitor_put,
+ .private_value = OXYGEN_ADC_MONITOR_A_HALF_VOL
+ | (1 << 8),
+ .tlv = { .p = monitor_db_scale, },
+ },
+ },
+ },
+ {
+ .pcm_dev = CAPTURE_0_FROM_I2S_2,
+ .controls = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Analog Input Monitor Switch",
+ .info = snd_ctl_boolean_mono_info,
+ .get = monitor_get,
+ .put = monitor_put,
+ .private_value = OXYGEN_ADC_MONITOR_B,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Analog Input Monitor Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ,
+ .info = monitor_volume_info,
+ .get = monitor_get,
+ .put = monitor_put,
+ .private_value = OXYGEN_ADC_MONITOR_B_HALF_VOL
+ | (1 << 8),
+ .tlv = { .p = monitor_db_scale, },
+ },
+ },
+ },
+ {
+ .pcm_dev = CAPTURE_2_FROM_I2S_2,
+ .controls = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Analog Input Monitor Switch",
+ .index = 1,
+ .info = snd_ctl_boolean_mono_info,
+ .get = monitor_get,
+ .put = monitor_put,
+ .private_value = OXYGEN_ADC_MONITOR_B,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Analog Input Monitor Volume",
+ .index = 1,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ,
+ .info = monitor_volume_info,
+ .get = monitor_get,
+ .put = monitor_put,
+ .private_value = OXYGEN_ADC_MONITOR_B_HALF_VOL
+ | (1 << 8),
+ .tlv = { .p = monitor_db_scale, },
+ },
+ },
+ },
+ {
+ .pcm_dev = CAPTURE_1_FROM_SPDIF,
+ .controls = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Digital Input Monitor Switch",
+ .info = snd_ctl_boolean_mono_info,
+ .get = monitor_get,
+ .put = monitor_put,
+ .private_value = OXYGEN_ADC_MONITOR_C,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Digital Input Monitor Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ,
+ .info = monitor_volume_info,
+ .get = monitor_get,
+ .put = monitor_put,
+ .private_value = OXYGEN_ADC_MONITOR_C_HALF_VOL
+ | (1 << 8),
+ .tlv = { .p = monitor_db_scale, },
+ },
+ },
+ },
+};
+
static const struct snd_kcontrol_new ac97_controls[] = {
AC97_VOLUME("Mic Capture Volume", 0, AC97_MIC),
AC97_SWITCH("Mic Capture Switch", 0, AC97_MIC, 15, 1),
AC97_SWITCH("Mic Boost (+20dB)", 0, AC97_MIC, 6, 0),
- AC97_VOLUME("Line Capture Volume", 0, AC97_LINE),
AC97_SWITCH("Line Capture Switch", 0, AC97_LINE, 15, 1),
AC97_VOLUME("CD Capture Volume", 0, AC97_CD),
AC97_SWITCH("CD Capture Switch", 0, AC97_CD, 15, 1),
@@ -756,6 +941,11 @@ static int add_controls(struct oxygen *chip,
return err;
if (err == 1)
continue;
+ if (!strcmp(template.name, "Master Playback Volume") &&
+ chip->model->dac_tlv) {
+ template.tlv.p = chip->model->dac_tlv;
+ template.access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ;
+ }
ctl = snd_ctl_new1(&template, chip);
if (!ctl)
return -ENOMEM;
@@ -773,11 +963,26 @@ static int add_controls(struct oxygen *chip,
int oxygen_mixer_init(struct oxygen *chip)
{
+ unsigned int i;
int err;
err = add_controls(chip, controls, ARRAY_SIZE(controls));
if (err < 0)
return err;
+ if (chip->model->pcm_dev_cfg & CAPTURE_1_FROM_SPDIF) {
+ err = add_controls(chip, spdif_input_controls,
+ ARRAY_SIZE(spdif_input_controls));
+ if (err < 0)
+ return err;
+ }
+ for (i = 0; i < ARRAY_SIZE(monitor_controls); ++i) {
+ if (!(chip->model->pcm_dev_cfg & monitor_controls[i].pcm_dev))
+ continue;
+ err = add_controls(chip, monitor_controls[i].controls,
+ ARRAY_SIZE(monitor_controls[i].controls));
+ if (err < 0)
+ return err;
+ }
if (chip->has_ac97_0) {
err = add_controls(chip, ac97_controls,
ARRAY_SIZE(ac97_controls));
diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c
index b70046aca65..b17c405e069 100644
--- a/sound/pci/oxygen/oxygen_pcm.c
+++ b/sound/pci/oxygen/oxygen_pcm.c
@@ -119,7 +119,7 @@ static int oxygen_open(struct snd_pcm_substream *substream,
runtime->private_data = (void *)(uintptr_t)channel;
if (channel == PCM_B && chip->has_ac97_1 &&
- (chip->model->used_channels & OXYGEN_CHANNEL_AC97))
+ (chip->model->pcm_dev_cfg & CAPTURE_2_FROM_AC97_1))
runtime->hw = oxygen_ac97_hardware;
else
runtime->hw = *oxygen_hardware[channel];
@@ -365,7 +365,7 @@ static int oxygen_rec_b_hw_params(struct snd_pcm_substream *substream,
return err;
is_ac97 = chip->has_ac97_1 &&
- (chip->model->used_channels & OXYGEN_CHANNEL_AC97);
+ (chip->model->pcm_dev_cfg & CAPTURE_2_FROM_AC97_1);
spin_lock_irq(&chip->reg_lock);
oxygen_write8_masked(chip, OXYGEN_REC_FORMAT,
@@ -640,34 +640,39 @@ int oxygen_pcm_init(struct oxygen *chip)
int outs, ins;
int err;
- outs = 1; /* OXYGEN_CHANNEL_MULTICH is always used */
- ins = !!(chip->model->used_channels & (OXYGEN_CHANNEL_A |
- OXYGEN_CHANNEL_B));
- err = snd_pcm_new(chip->card, "Analog", 0, outs, ins, &pcm);
- if (err < 0)
- return err;
- snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &oxygen_multich_ops);
- if (chip->model->used_channels & OXYGEN_CHANNEL_A)
- snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
- &oxygen_rec_a_ops);
- else if (chip->model->used_channels & OXYGEN_CHANNEL_B)
- snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
- &oxygen_rec_b_ops);
- pcm->private_data = chip;
- pcm->private_free = oxygen_pcm_free;
- strcpy(pcm->name, "Analog");
- snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream,
- SNDRV_DMA_TYPE_DEV,
- snd_dma_pci_data(chip->pci),
- 512 * 1024, 2048 * 1024);
- if (ins)
- snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream,
- SNDRV_DMA_TYPE_DEV,
- snd_dma_pci_data(chip->pci),
- 128 * 1024, 256 * 1024);
-
- outs = !!(chip->model->used_channels & OXYGEN_CHANNEL_SPDIF);
- ins = !!(chip->model->used_channels & OXYGEN_CHANNEL_C);
+ outs = !!(chip->model->pcm_dev_cfg & PLAYBACK_0_TO_I2S);
+ ins = !!(chip->model->pcm_dev_cfg & (CAPTURE_0_FROM_I2S_1 |
+ CAPTURE_0_FROM_I2S_2));
+ if (outs | ins) {
+ err = snd_pcm_new(chip->card, "Analog", 0, outs, ins, &pcm);
+ if (err < 0)
+ return err;
+ if (outs)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &oxygen_multich_ops);
+ if (chip->model->pcm_dev_cfg & CAPTURE_0_FROM_I2S_1)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &oxygen_rec_a_ops);
+ else if (chip->model->pcm_dev_cfg & CAPTURE_0_FROM_I2S_2)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &oxygen_rec_b_ops);
+ pcm->private_data = chip;
+ pcm->private_free = oxygen_pcm_free;
+ strcpy(pcm->name, "Analog");
+ if (outs)
+ snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream,
+ SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci),
+ 512 * 1024, 2048 * 1024);
+ if (ins)
+ snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream,
+ SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci),
+ 128 * 1024, 256 * 1024);
+ }
+
+ outs = !!(chip->model->pcm_dev_cfg & PLAYBACK_1_TO_SPDIF);
+ ins = !!(chip->model->pcm_dev_cfg & CAPTURE_1_FROM_SPDIF);
if (outs | ins) {
err = snd_pcm_new(chip->card, "Digital", 1, outs, ins, &pcm);
if (err < 0)
@@ -686,12 +691,13 @@ int oxygen_pcm_init(struct oxygen *chip)
128 * 1024, 256 * 1024);
}
- outs = chip->has_ac97_1 &&
- (chip->model->used_channels & OXYGEN_CHANNEL_AC97);
- ins = outs ||
- (chip->model->used_channels & (OXYGEN_CHANNEL_A |
- OXYGEN_CHANNEL_B))
- == (OXYGEN_CHANNEL_A | OXYGEN_CHANNEL_B);
+ if (chip->has_ac97_1) {
+ outs = !!(chip->model->pcm_dev_cfg & PLAYBACK_2_TO_AC97_1);
+ ins = !!(chip->model->pcm_dev_cfg & CAPTURE_2_FROM_AC97_1);
+ } else {
+ outs = 0;
+ ins = !!(chip->model->pcm_dev_cfg & CAPTURE_2_FROM_I2S_2);
+ }
if (outs | ins) {
err = snd_pcm_new(chip->card, outs ? "AC97" : "Analog2",
2, outs, ins, &pcm);
diff --git a/sound/pci/oxygen/pcm1796.h b/sound/pci/oxygen/pcm1796.h
new file mode 100644
index 00000000000..698bf46c710
--- /dev/null
+++ b/sound/pci/oxygen/pcm1796.h
@@ -0,0 +1,58 @@
+#ifndef PCM1796_H_INCLUDED
+#define PCM1796_H_INCLUDED
+
+/* register 16 */
+#define PCM1796_ATL_MASK 0xff
+/* register 17 */
+#define PCM1796_ATR_MASK 0xff
+/* register 18 */
+#define PCM1796_MUTE 0x01
+#define PCM1796_DME 0x02
+#define PCM1796_DMF_MASK 0x0c
+#define PCM1796_DMF_DISABLED 0x00
+#define PCM1796_DMF_48 0x04
+#define PCM1796_DMF_441 0x08
+#define PCM1796_DMF_32 0x0c
+#define PCM1796_FMT_MASK 0x70
+#define PCM1796_FMT_16_RJUST 0x00
+#define PCM1796_FMT_20_RJUST 0x10
+#define PCM1796_FMT_24_RJUST 0x20
+#define PCM1796_FMT_24_LJUST 0x30
+#define PCM1796_FMT_16_I2S 0x40
+#define PCM1796_FMT_24_I2S 0x50
+#define PCM1796_ATLD 0x80
+/* register 19 */
+#define PCM1796_INZD 0x01
+#define PCM1796_FLT_MASK 0x02
+#define PCM1796_FLT_SHARP 0x00
+#define PCM1796_FLT_SLOW 0x02
+#define PCM1796_DFMS 0x04
+#define PCM1796_OPE 0x10
+#define PCM1796_ATS_MASK 0x60
+#define PCM1796_ATS_1 0x00
+#define PCM1796_ATS_2 0x20
+#define PCM1796_ATS_4 0x40
+#define PCM1796_ATS_8 0x60
+#define PCM1796_REV 0x80
+/* register 20 */
+#define PCM1796_OS_MASK 0x03
+#define PCM1796_OS_64 0x00
+#define PCM1796_OS_32 0x01
+#define PCM1796_OS_128 0x02
+#define PCM1796_CHSL_MASK 0x04
+#define PCM1796_CHSL_LEFT 0x00
+#define PCM1796_CHSL_RIGHT 0x04
+#define PCM1796_MONO 0x08
+#define PCM1796_DFTH 0x10
+#define PCM1796_DSD 0x20
+#define PCM1796_SRST 0x40
+/* register 21 */
+#define PCM1796_PCMZ 0x01
+#define PCM1796_DZ_MASK 0x06
+/* register 22 */
+#define PCM1796_ZFGL 0x01
+#define PCM1796_ZFGR 0x02
+/* register 23 */
+#define PCM1796_ID_MASK 0x1f
+
+#endif
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index d163397b85c..7f84fa5deca 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -18,6 +18,9 @@
*/
/*
+ * Xonar D2/D2X
+ * ------------
+ *
* CMI8788:
*
* SPI 0 -> 1st PCM1796 (front)
@@ -30,10 +33,33 @@
* GPIO 5 <- external power present (D2X only)
* GPIO 7 -> ALT
* GPIO 8 -> enable output to speakers
+ */
+
+/*
+ * Xonar DX
+ * --------
+ *
+ * CMI8788:
+ *
+ * I²C <-> CS4398 (front)
+ * <-> CS4362A (surround, center/LFE, back)
+ *
+ * GPI 0 <- external power present
*
- * CM9780:
+ * GPIO 0 -> enable output to speakers
+ * GPIO 1 -> enable front panel I/O
+ * GPIO 2 -> M0 of CS5361
+ * GPIO 3 -> M1 of CS5361
+ * GPIO 8 -> route input jack to line-in (0) or mic-in (1)
*
- * GPIO 0 -> enable AC'97 bypass (line in -> ADC)
+ * CS4398:
+ *
+ * AD0 <- 1
+ * AD1 <- 1
+ *
+ * CS4362A:
+ *
+ * AD0 <- 0
*/
#include <linux/pci.h>
@@ -47,11 +73,14 @@
#include <sound/tlv.h>
#include "oxygen.h"
#include "cm9780.h"
+#include "pcm1796.h"
+#include "cs4398.h"
+#include "cs4362a.h"
MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
-MODULE_DESCRIPTION("Asus AV200 driver");
+MODULE_DESCRIPTION("Asus AVx00 driver");
MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Asus,AV200}}");
+MODULE_SUPPORTED_DEVICE("{{Asus,AV100},{Asus,AV200}}");
static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
@@ -64,80 +93,44 @@ MODULE_PARM_DESC(id, "ID string");
module_param_array(enable, bool, NULL, 0444);
MODULE_PARM_DESC(enable, "enable card");
+enum {
+ MODEL_D2,
+ MODEL_D2X,
+ MODEL_DX,
+};
+
static struct pci_device_id xonar_ids[] __devinitdata = {
- { OXYGEN_PCI_SUBID(0x1043, 0x8269) }, /* Asus Xonar D2 */
- { OXYGEN_PCI_SUBID(0x1043, 0x82b7) }, /* Asus Xonar D2X */
+ { OXYGEN_PCI_SUBID(0x1043, 0x8269), .driver_data = MODEL_D2 },
+ { OXYGEN_PCI_SUBID(0x1043, 0x8275), .driver_data = MODEL_DX },
+ { OXYGEN_PCI_SUBID(0x1043, 0x82b7), .driver_data = MODEL_D2X },
{ }
};
MODULE_DEVICE_TABLE(pci, xonar_ids);
-#define GPIO_CS5381_M_MASK 0x000c
-#define GPIO_CS5381_M_SINGLE 0x0000
-#define GPIO_CS5381_M_DOUBLE 0x0004
-#define GPIO_CS5381_M_QUAD 0x0008
-#define GPIO_EXT_POWER 0x0020
-#define GPIO_ALT 0x0080
-#define GPIO_OUTPUT_ENABLE 0x0100
-
-#define GPIO_LINE_MUTE CM9780_GPO0
-
-/* register 16 */
-#define PCM1796_ATL_MASK 0xff
-/* register 17 */
-#define PCM1796_ATR_MASK 0xff
-/* register 18 */
-#define PCM1796_MUTE 0x01
-#define PCM1796_DME 0x02
-#define PCM1796_DMF_MASK 0x0c
-#define PCM1796_DMF_DISABLED 0x00
-#define PCM1796_DMF_48 0x04
-#define PCM1796_DMF_441 0x08
-#define PCM1796_DMF_32 0x0c
-#define PCM1796_FMT_MASK 0x70
-#define PCM1796_FMT_16_RJUST 0x00
-#define PCM1796_FMT_20_RJUST 0x10
-#define PCM1796_FMT_24_RJUST 0x20
-#define PCM1796_FMT_24_LJUST 0x30
-#define PCM1796_FMT_16_I2S 0x40
-#define PCM1796_FMT_24_I2S 0x50
-#define PCM1796_ATLD 0x80
-/* register 19 */
-#define PCM1796_INZD 0x01
-#define PCM1796_FLT_MASK 0x02
-#define PCM1796_FLT_SHARP 0x00
-#define PCM1796_FLT_SLOW 0x02
-#define PCM1796_DFMS 0x04
-#define PCM1796_OPE 0x10
-#define PCM1796_ATS_MASK 0x60
-#define PCM1796_ATS_1 0x00
-#define PCM1796_ATS_2 0x20
-#define PCM1796_ATS_4 0x40
-#define PCM1796_ATS_8 0x60
-#define PCM1796_REV 0x80
-/* register 20 */
-#define PCM1796_OS_MASK 0x03
-#define PCM1796_OS_64 0x00
-#define PCM1796_OS_32 0x01
-#define PCM1796_OS_128 0x02
-#define PCM1796_CHSL_MASK 0x04
-#define PCM1796_CHSL_LEFT 0x00
-#define PCM1796_CHSL_RIGHT 0x04
-#define PCM1796_MONO 0x08
-#define PCM1796_DFTH 0x10
-#define PCM1796_DSD 0x20
-#define PCM1796_SRST 0x40
-/* register 21 */
-#define PCM1796_PCMZ 0x01
-#define PCM1796_DZ_MASK 0x06
-/* register 22 */
-#define PCM1796_ZFGL 0x01
-#define PCM1796_ZFGR 0x02
-/* register 23 */
-#define PCM1796_ID_MASK 0x1f
+#define GPIO_CS53x1_M_MASK 0x000c
+#define GPIO_CS53x1_M_SINGLE 0x0000
+#define GPIO_CS53x1_M_DOUBLE 0x0004
+#define GPIO_CS53x1_M_QUAD 0x0008
+
+#define GPIO_D2X_EXT_POWER 0x0020
+#define GPIO_D2_ALT 0x0080
+#define GPIO_D2_OUTPUT_ENABLE 0x0100
+
+#define GPI_DX_EXT_POWER 0x01
+#define GPIO_DX_OUTPUT_ENABLE 0x0001
+#define GPIO_DX_FRONT_PANEL 0x0002
+#define GPIO_DX_INPUT_ROUTE 0x0100
+
+#define I2C_DEVICE_CS4398 0x9e /* 10011, AD1=1, AD0=1, /W=0 */
+#define I2C_DEVICE_CS4362A 0x30 /* 001100, AD0=0, /W=0 */
struct xonar_data {
- u8 is_d2x;
+ unsigned int anti_pop_delay;
+ u16 output_enable_bit;
+ u8 ext_power_reg;
+ u8 ext_power_int_reg;
+ u8 ext_power_bit;
u8 has_power;
};
@@ -156,62 +149,157 @@ static void pcm1796_write(struct oxygen *chip, unsigned int codec,
(reg << 8) | value);
}
-static void xonar_init(struct oxygen *chip)
+static void cs4398_write(struct oxygen *chip, u8 reg, u8 value)
+{
+ oxygen_write_i2c(chip, I2C_DEVICE_CS4398, reg, value);
+}
+
+static void cs4362a_write(struct oxygen *chip, u8 reg, u8 value)
+{
+ oxygen_write_i2c(chip, I2C_DEVICE_CS4362A, reg, value);
+}
+
+static void xonar_common_init(struct oxygen *chip)
+{
+ struct xonar_data *data = chip->model_data;
+
+ if (data->ext_power_reg) {
+ oxygen_set_bits8(chip, data->ext_power_int_reg,
+ data->ext_power_bit);
+ chip->interrupt_mask |= OXYGEN_INT_GPIO;
+ data->has_power = !!(oxygen_read8(chip, data->ext_power_reg)
+ & data->ext_power_bit);
+ }
+ oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_CS53x1_M_MASK);
+ oxygen_write16_masked(chip, OXYGEN_GPIO_DATA,
+ GPIO_CS53x1_M_SINGLE, GPIO_CS53x1_M_MASK);
+ oxygen_ac97_set_bits(chip, 0, CM9780_JACK, CM9780_FMIC2MIC);
+ msleep(data->anti_pop_delay);
+ oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, data->output_enable_bit);
+ oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit);
+}
+
+static void xonar_d2_init(struct oxygen *chip)
{
struct xonar_data *data = chip->model_data;
unsigned int i;
- data->is_d2x = chip->pci->subsystem_device == 0x82b7;
+ data->anti_pop_delay = 300;
+ data->output_enable_bit = GPIO_D2_OUTPUT_ENABLE;
for (i = 0; i < 4; ++i) {
- pcm1796_write(chip, i, 18, PCM1796_FMT_24_LJUST | PCM1796_ATLD);
+ pcm1796_write(chip, i, 18, PCM1796_MUTE | PCM1796_DMF_DISABLED |
+ PCM1796_FMT_24_LJUST | PCM1796_ATLD);
pcm1796_write(chip, i, 19, PCM1796_FLT_SHARP | PCM1796_ATS_1);
pcm1796_write(chip, i, 20, PCM1796_OS_64);
pcm1796_write(chip, i, 21, 0);
- pcm1796_write(chip, i, 16, 0xff); /* set ATL/ATR after ATLD */
- pcm1796_write(chip, i, 17, 0xff);
+ pcm1796_write(chip, i, 16, 0x0f); /* set ATL/ATR after ATLD */
+ pcm1796_write(chip, i, 17, 0x0f);
}
- oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL,
- GPIO_CS5381_M_MASK | GPIO_ALT);
- oxygen_write16_masked(chip, OXYGEN_GPIO_DATA,
- GPIO_CS5381_M_SINGLE,
- GPIO_CS5381_M_MASK | GPIO_ALT);
- if (data->is_d2x) {
- oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL,
- GPIO_EXT_POWER);
- oxygen_set_bits16(chip, OXYGEN_GPIO_INTERRUPT_MASK,
- GPIO_EXT_POWER);
- chip->interrupt_mask |= OXYGEN_INT_GPIO;
- data->has_power = !!(oxygen_read16(chip, OXYGEN_GPIO_DATA)
- & GPIO_EXT_POWER);
- }
- oxygen_ac97_set_bits(chip, 0, CM9780_JACK, CM9780_FMIC2MIC);
- oxygen_ac97_clear_bits(chip, 0, CM9780_GPIO_STATUS, GPIO_LINE_MUTE);
- msleep(300);
- oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_OUTPUT_ENABLE);
- oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, GPIO_OUTPUT_ENABLE);
+ oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_D2_ALT);
+ oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_D2_ALT);
+
+ xonar_common_init(chip);
snd_component_add(chip->card, "PCM1796");
snd_component_add(chip->card, "CS5381");
}
+static void xonar_d2x_init(struct oxygen *chip)
+{
+ struct xonar_data *data = chip->model_data;
+
+ data->ext_power_reg = OXYGEN_GPIO_DATA;
+ data->ext_power_int_reg = OXYGEN_GPIO_INTERRUPT_MASK;
+ data->ext_power_bit = GPIO_D2X_EXT_POWER;
+ oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_D2X_EXT_POWER);
+ xonar_d2_init(chip);
+}
+
+static void xonar_dx_init(struct oxygen *chip)
+{
+ struct xonar_data *data = chip->model_data;
+
+ data->anti_pop_delay = 800;
+ data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE;
+ data->ext_power_reg = OXYGEN_GPI_DATA;
+ data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK;
+ data->ext_power_bit = GPI_DX_EXT_POWER;
+
+ oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS,
+ OXYGEN_2WIRE_LENGTH_8 |
+ OXYGEN_2WIRE_INTERRUPT_MASK |
+ OXYGEN_2WIRE_SPEED_FAST);
+
+ /* set CPEN (control port mode) and power down */
+ cs4398_write(chip, 8, CS4398_CPEN | CS4398_PDN);
+ cs4362a_write(chip, 0x01, CS4362A_PDN | CS4362A_CPEN);
+ /* configure */
+ cs4398_write(chip, 2, CS4398_FM_SINGLE |
+ CS4398_DEM_NONE | CS4398_DIF_LJUST);
+ cs4398_write(chip, 3, CS4398_ATAPI_B_R | CS4398_ATAPI_A_L);
+ cs4398_write(chip, 4, CS4398_MUTEP_LOW | CS4398_PAMUTE);
+ cs4398_write(chip, 5, 0xfe);
+ cs4398_write(chip, 6, 0xfe);
+ cs4398_write(chip, 7, CS4398_RMP_DN | CS4398_RMP_UP |
+ CS4398_ZERO_CROSS | CS4398_SOFT_RAMP);
+ cs4362a_write(chip, 0x02, CS4362A_DIF_LJUST);
+ cs4362a_write(chip, 0x03, CS4362A_MUTEC_6 | CS4362A_AMUTE |
+ CS4362A_RMP_UP | CS4362A_ZERO_CROSS | CS4362A_SOFT_RAMP);
+ cs4362a_write(chip, 0x04, CS4362A_RMP_DN | CS4362A_DEM_NONE);
+ cs4362a_write(chip, 0x05, 0);
+ cs4362a_write(chip, 0x06, CS4362A_FM_SINGLE |
+ CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L);
+ cs4362a_write(chip, 0x07, 0x7f | CS4362A_MUTE);
+ cs4362a_write(chip, 0x08, 0x7f | CS4362A_MUTE);
+ cs4362a_write(chip, 0x09, CS4362A_FM_SINGLE |
+ CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L);
+ cs4362a_write(chip, 0x0a, 0x7f | CS4362A_MUTE);
+ cs4362a_write(chip, 0x0b, 0x7f | CS4362A_MUTE);
+ cs4362a_write(chip, 0x0c, CS4362A_FM_SINGLE |
+ CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L);
+ cs4362a_write(chip, 0x0d, 0x7f | CS4362A_MUTE);
+ cs4362a_write(chip, 0x0e, 0x7f | CS4362A_MUTE);
+ /* clear power down */
+ cs4398_write(chip, 8, CS4398_CPEN);
+ cs4362a_write(chip, 0x01, CS4362A_CPEN);
+
+ oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL,
+ GPIO_DX_FRONT_PANEL | GPIO_DX_INPUT_ROUTE);
+ oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA,
+ GPIO_DX_FRONT_PANEL | GPIO_DX_INPUT_ROUTE);
+
+ xonar_common_init(chip);
+
+ snd_component_add(chip->card, "CS4398");
+ snd_component_add(chip->card, "CS4362A");
+ snd_component_add(chip->card, "CS5361");
+}
+
static void xonar_cleanup(struct oxygen *chip)
{
- oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_OUTPUT_ENABLE);
+ struct xonar_data *data = chip->model_data;
+
+ oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit);
+}
+
+static void xonar_dx_cleanup(struct oxygen *chip)
+{
+ xonar_cleanup(chip);
+ cs4362a_write(chip, 0x01, CS4362A_PDN | CS4362A_CPEN);
+ oxygen_clear_bits8(chip, OXYGEN_FUNCTION, OXYGEN_FUNCTION_RESET_CODEC);
}
static void set_pcm1796_params(struct oxygen *chip,
struct snd_pcm_hw_params *params)
{
-#if 0
unsigned int i;
u8 value;
value = params_rate(params) >= 96000 ? PCM1796_OS_32 : PCM1796_OS_64;
for (i = 0; i < 4; ++i)
pcm1796_write(chip, i, 20, value);
-#endif
}
static void update_pcm1796_volume(struct oxygen *chip)
@@ -236,19 +324,73 @@ static void update_pcm1796_mute(struct oxygen *chip)
pcm1796_write(chip, i, 18, value);
}
-static void set_cs5381_params(struct oxygen *chip,
+static void set_cs53x1_params(struct oxygen *chip,
struct snd_pcm_hw_params *params)
{
unsigned int value;
if (params_rate(params) <= 54000)
- value = GPIO_CS5381_M_SINGLE;
+ value = GPIO_CS53x1_M_SINGLE;
else if (params_rate(params) <= 108000)
- value = GPIO_CS5381_M_DOUBLE;
+ value = GPIO_CS53x1_M_DOUBLE;
else
- value = GPIO_CS5381_M_QUAD;
+ value = GPIO_CS53x1_M_QUAD;
oxygen_write16_masked(chip, OXYGEN_GPIO_DATA,
- value, GPIO_CS5381_M_MASK);
+ value, GPIO_CS53x1_M_MASK);
+}
+
+static void set_cs43xx_params(struct oxygen *chip,
+ struct snd_pcm_hw_params *params)
+{
+ u8 fm_cs4398, fm_cs4362a;
+
+ fm_cs4398 = CS4398_DEM_NONE | CS4398_DIF_LJUST;
+ fm_cs4362a = CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L;
+ if (params_rate(params) <= 50000) {
+ fm_cs4398 |= CS4398_FM_SINGLE;
+ fm_cs4362a |= CS4362A_FM_SINGLE;
+ } else if (params_rate(params) <= 100000) {
+ fm_cs4398 |= CS4398_FM_DOUBLE;
+ fm_cs4362a |= CS4362A_FM_DOUBLE;
+ } else {
+ fm_cs4398 |= CS4398_FM_QUAD;
+ fm_cs4362a |= CS4362A_FM_QUAD;
+ }
+ cs4398_write(chip, 2, fm_cs4398);
+ cs4362a_write(chip, 0x06, fm_cs4362a);
+ cs4362a_write(chip, 0x09, fm_cs4362a);
+ cs4362a_write(chip, 0x0c, fm_cs4362a);
+}
+
+static void update_cs4362a_volumes(struct oxygen *chip)
+{
+ u8 mute;
+
+ mute = chip->dac_mute ? CS4362A_MUTE : 0;
+ cs4362a_write(chip, 7, (127 - chip->dac_volume[2]) | mute);
+ cs4362a_write(chip, 8, (127 - chip->dac_volume[3]) | mute);
+ cs4362a_write(chip, 10, (127 - chip->dac_volume[4]) | mute);
+ cs4362a_write(chip, 11, (127 - chip->dac_volume[5]) | mute);
+ cs4362a_write(chip, 13, (127 - chip->dac_volume[6]) | mute);
+ cs4362a_write(chip, 14, (127 - chip->dac_volume[7]) | mute);
+}
+
+static void update_cs43xx_volume(struct oxygen *chip)
+{
+ cs4398_write(chip, 5, (127 - chip->dac_volume[0]) * 2);
+ cs4398_write(chip, 6, (127 - chip->dac_volume[1]) * 2);
+ update_cs4362a_volumes(chip);
+}
+
+static void update_cs43xx_mute(struct oxygen *chip)
+{
+ u8 reg;
+
+ reg = CS4398_MUTEP_LOW | CS4398_PAMUTE;
+ if (chip->dac_mute)
+ reg |= CS4398_MUTE_B | CS4398_MUTE_A;
+ cs4398_write(chip, 4, reg);
+ update_cs4362a_volumes(chip);
}
static void xonar_gpio_changed(struct oxygen *chip)
@@ -256,10 +398,8 @@ static void xonar_gpio_changed(struct oxygen *chip)
struct xonar_data *data = chip->model_data;
u8 has_power;
- if (!data->is_d2x)
- return;
- has_power = !!(oxygen_read16(chip, OXYGEN_GPIO_DATA)
- & GPIO_EXT_POWER);
+ has_power = !!(oxygen_read8(chip, data->ext_power_reg)
+ & data->ext_power_bit);
if (has_power != data->has_power) {
data->has_power = has_power;
if (has_power) {
@@ -272,66 +412,13 @@ static void xonar_gpio_changed(struct oxygen *chip)
}
}
-static void mute_ac97_ctl(struct oxygen *chip, unsigned int control)
-{
- unsigned int index = chip->controls[control]->private_value & 0xff;
- u16 value;
-
- value = oxygen_read_ac97(chip, 0, index);
- if (!(value & 0x8000)) {
- oxygen_write_ac97(chip, 0, index, value | 0x8000);
- snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
- &chip->controls[control]->id);
- }
-}
-
-static void xonar_ac97_switch_hook(struct oxygen *chip, unsigned int codec,
- unsigned int reg, int mute)
-{
- if (codec != 0)
- return;
- /* line-in is exclusive */
- switch (reg) {
- case AC97_LINE:
- oxygen_write_ac97_masked(chip, 0, CM9780_GPIO_STATUS,
- mute ? GPIO_LINE_MUTE : 0,
- GPIO_LINE_MUTE);
- if (!mute) {
- mute_ac97_ctl(chip, CONTROL_MIC_CAPTURE_SWITCH);
- mute_ac97_ctl(chip, CONTROL_CD_CAPTURE_SWITCH);
- mute_ac97_ctl(chip, CONTROL_AUX_CAPTURE_SWITCH);
- }
- break;
- case AC97_MIC:
- case AC97_CD:
- case AC97_VIDEO:
- case AC97_AUX:
- if (!mute) {
- oxygen_ac97_set_bits(chip, 0, CM9780_GPIO_STATUS,
- GPIO_LINE_MUTE);
- mute_ac97_ctl(chip, CONTROL_LINE_CAPTURE_SWITCH);
- }
- break;
- }
-}
-
-static int pcm1796_volume_info(struct snd_kcontrol *ctl,
- struct snd_ctl_elem_info *info)
-{
- info->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
- info->count = 8;
- info->value.integer.min = 0x0f;
- info->value.integer.max = 0xff;
- return 0;
-}
-
static int alt_switch_get(struct snd_kcontrol *ctl,
struct snd_ctl_elem_value *value)
{
struct oxygen *chip = ctl->private_data;
value->value.integer.value[0] =
- !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) & GPIO_ALT);
+ !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) & GPIO_D2_ALT);
return 0;
}
@@ -345,9 +432,9 @@ static int alt_switch_put(struct snd_kcontrol *ctl,
spin_lock_irq(&chip->reg_lock);
old_bits = oxygen_read16(chip, OXYGEN_GPIO_DATA);
if (value->value.integer.value[0])
- new_bits = old_bits | GPIO_ALT;
+ new_bits = old_bits | GPIO_D2_ALT;
else
- new_bits = old_bits & ~GPIO_ALT;
+ new_bits = old_bits & ~GPIO_D2_ALT;
changed = new_bits != old_bits;
if (changed)
oxygen_write16(chip, OXYGEN_GPIO_DATA, new_bits);
@@ -363,20 +450,68 @@ static const struct snd_kcontrol_new alt_switch = {
.put = alt_switch_put,
};
+static int front_panel_get(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_value *value)
+{
+ struct oxygen *chip = ctl->private_data;
+
+ value->value.integer.value[0] =
+ !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) & GPIO_DX_FRONT_PANEL);
+ return 0;
+}
+
+static int front_panel_put(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_value *value)
+{
+ struct oxygen *chip = ctl->private_data;
+ u16 old_reg, new_reg;
+
+ spin_lock_irq(&chip->reg_lock);
+ old_reg = oxygen_read16(chip, OXYGEN_GPIO_DATA);
+ if (value->value.integer.value[0])
+ new_reg = old_reg | GPIO_DX_FRONT_PANEL;
+ else
+ new_reg = old_reg & ~GPIO_DX_FRONT_PANEL;
+ oxygen_write16(chip, OXYGEN_GPIO_DATA, new_reg);
+ spin_unlock_irq(&chip->reg_lock);
+ return old_reg != new_reg;
+}
+
+static const struct snd_kcontrol_new front_panel_switch = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Front Panel Switch",
+ .info = snd_ctl_boolean_mono_info,
+ .get = front_panel_get,
+ .put = front_panel_put,
+};
+
+static void xonar_dx_ac97_switch(struct oxygen *chip,
+ unsigned int reg, unsigned int mute)
+{
+ if (reg == AC97_LINE) {
+ spin_lock_irq(&chip->reg_lock);
+ oxygen_write16_masked(chip, OXYGEN_GPIO_DATA,
+ mute ? GPIO_DX_INPUT_ROUTE : 0,
+ GPIO_DX_INPUT_ROUTE);
+ spin_unlock_irq(&chip->reg_lock);
+ }
+}
+
static const DECLARE_TLV_DB_SCALE(pcm1796_db_scale, -12000, 50, 0);
+static const DECLARE_TLV_DB_SCALE(cs4362a_db_scale, -12700, 100, 0);
-static int xonar_control_filter(struct snd_kcontrol_new *template)
+static int xonar_d2_control_filter(struct snd_kcontrol_new *template)
{
- if (!strcmp(template->name, "Master Playback Volume")) {
- template->access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ;
- template->info = pcm1796_volume_info,
- template->tlv.p = pcm1796_db_scale;
- } else if (!strncmp(template->name, "CD Capture ", 11)) {
+ if (!strncmp(template->name, "CD Capture ", 11))
/* CD in is actually connected to the video in pin */
template->private_value ^= AC97_CD ^ AC97_VIDEO;
- } else if (!strcmp(template->name, "Line Capture Volume")) {
- return 1; /* line-in bypasses the AC'97 mixer */
- }
+ return 0;
+}
+
+static int xonar_dx_control_filter(struct snd_kcontrol_new *template)
+{
+ if (!strncmp(template->name, "CD Capture ", 11))
+ return 1; /* no CD input */
return 0;
}
@@ -385,30 +520,96 @@ static int xonar_mixer_init(struct oxygen *chip)
return snd_ctl_add(chip->card, snd_ctl_new1(&alt_switch, chip));
}
-static const struct oxygen_model model_xonar = {
- .shortname = "Asus AV200",
- .longname = "Asus Virtuoso 200",
- .chip = "AV200",
- .owner = THIS_MODULE,
- .init = xonar_init,
- .control_filter = xonar_control_filter,
- .mixer_init = xonar_mixer_init,
- .cleanup = xonar_cleanup,
- .set_dac_params = set_pcm1796_params,
- .set_adc_params = set_cs5381_params,
- .update_dac_volume = update_pcm1796_volume,
- .update_dac_mute = update_pcm1796_mute,
- .ac97_switch_hook = xonar_ac97_switch_hook,
- .gpio_changed = xonar_gpio_changed,
- .model_data_size = sizeof(struct xonar_data),
- .dac_channels = 8,
- .used_channels = OXYGEN_CHANNEL_B |
- OXYGEN_CHANNEL_C |
- OXYGEN_CHANNEL_SPDIF |
- OXYGEN_CHANNEL_MULTICH,
- .function_flags = OXYGEN_FUNCTION_ENABLE_SPI_4_5,
- .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
- .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
+static int xonar_dx_mixer_init(struct oxygen *chip)
+{
+ return snd_ctl_add(chip->card, snd_ctl_new1(&front_panel_switch, chip));
+}
+
+static const struct oxygen_model xonar_models[] = {
+ [MODEL_D2] = {
+ .shortname = "Xonar D2",
+ .longname = "Asus Virtuoso 200",
+ .chip = "AV200",
+ .owner = THIS_MODULE,
+ .init = xonar_d2_init,
+ .control_filter = xonar_d2_control_filter,
+ .mixer_init = xonar_mixer_init,
+ .cleanup = xonar_cleanup,
+ .set_dac_params = set_pcm1796_params,
+ .set_adc_params = set_cs53x1_params,
+ .update_dac_volume = update_pcm1796_volume,
+ .update_dac_mute = update_pcm1796_mute,
+ .dac_tlv = pcm1796_db_scale,
+ .model_data_size = sizeof(struct xonar_data),
+ .pcm_dev_cfg = PLAYBACK_0_TO_I2S |
+ PLAYBACK_1_TO_SPDIF |
+ CAPTURE_0_FROM_I2S_2 |
+ CAPTURE_1_FROM_SPDIF,
+ .dac_channels = 8,
+ .dac_volume_min = 0x0f,
+ .dac_volume_max = 0xff,
+ .misc_flags = OXYGEN_MISC_MIDI,
+ .function_flags = OXYGEN_FUNCTION_SPI |
+ OXYGEN_FUNCTION_ENABLE_SPI_4_5,
+ .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
+ .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
+ },
+ [MODEL_D2X] = {
+ .shortname = "Xonar D2X",
+ .longname = "Asus Virtuoso 200",
+ .chip = "AV200",
+ .owner = THIS_MODULE,
+ .init = xonar_d2x_init,
+ .control_filter = xonar_d2_control_filter,
+ .mixer_init = xonar_mixer_init,
+ .cleanup = xonar_cleanup,
+ .set_dac_params = set_pcm1796_params,
+ .set_adc_params = set_cs53x1_params,
+ .update_dac_volume = update_pcm1796_volume,
+ .update_dac_mute = update_pcm1796_mute,
+ .gpio_changed = xonar_gpio_changed,
+ .dac_tlv = pcm1796_db_scale,
+ .model_data_size = sizeof(struct xonar_data),
+ .pcm_dev_cfg = PLAYBACK_0_TO_I2S |
+ PLAYBACK_1_TO_SPDIF |
+ CAPTURE_0_FROM_I2S_2 |
+ CAPTURE_1_FROM_SPDIF,
+ .dac_channels = 8,
+ .dac_volume_min = 0x0f,
+ .dac_volume_max = 0xff,
+ .misc_flags = OXYGEN_MISC_MIDI,
+ .function_flags = OXYGEN_FUNCTION_SPI |
+ OXYGEN_FUNCTION_ENABLE_SPI_4_5,
+ .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
+ .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
+ },
+ [MODEL_DX] = {
+ .shortname = "Xonar DX",
+ .longname = "Asus Virtuoso 100",
+ .chip = "AV200",
+ .owner = THIS_MODULE,
+ .init = xonar_dx_init,
+ .control_filter = xonar_dx_control_filter,
+ .mixer_init = xonar_dx_mixer_init,
+ .cleanup = xonar_dx_cleanup,
+ .set_dac_params = set_cs43xx_params,
+ .set_adc_params = set_cs53x1_params,
+ .update_dac_volume = update_cs43xx_volume,
+ .update_dac_mute = update_cs43xx_mute,
+ .gpio_changed = xonar_gpio_changed,
+ .ac97_switch = xonar_dx_ac97_switch,
+ .dac_tlv = cs4362a_db_scale,
+ .model_data_size = sizeof(struct xonar_data),
+ .pcm_dev_cfg = PLAYBACK_0_TO_I2S |
+ PLAYBACK_1_TO_SPDIF |
+ CAPTURE_0_FROM_I2S_2,
+ .dac_channels = 8,
+ .dac_volume_min = 0,
+ .dac_volume_max = 127,
+ .function_flags = OXYGEN_FUNCTION_2WIRE,
+ .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
+ .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
+ },
};
static int __devinit xonar_probe(struct pci_dev *pci,
@@ -423,7 +624,8 @@ static int __devinit xonar_probe(struct pci_dev *pci,
++dev;
return -ENOENT;
}
- err = oxygen_pci_probe(pci, index[dev], id[dev], 1, &model_xonar);
+ err = oxygen_pci_probe(pci, index[dev], id[dev],
+ &xonar_models[pci_id->driver_data]);
if (err >= 0)
++dev;
return err;
diff --git a/sound/pci/oxygen/wm8785.h b/sound/pci/oxygen/wm8785.h
new file mode 100644
index 00000000000..8c23e315ae6
--- /dev/null
+++ b/sound/pci/oxygen/wm8785.h
@@ -0,0 +1,45 @@
+#ifndef WM8785_H_INCLUDED
+#define WM8785_H_INCLUDED
+
+#define WM8785_R0 0
+#define WM8785_R1 1
+#define WM8785_R2 2
+#define WM8785_R7 7
+
+/* R0 */
+#define WM8785_MCR_MASK 0x007
+#define WM8785_MCR_SLAVE 0x000
+#define WM8785_MCR_MASTER_128 0x001
+#define WM8785_MCR_MASTER_192 0x002
+#define WM8785_MCR_MASTER_256 0x003
+#define WM8785_MCR_MASTER_384 0x004
+#define WM8785_MCR_MASTER_512 0x005
+#define WM8785_MCR_MASTER_768 0x006
+#define WM8785_OSR_MASK 0x018
+#define WM8785_OSR_SINGLE 0x000
+#define WM8785_OSR_DOUBLE 0x008
+#define WM8785_OSR_QUAD 0x010
+#define WM8785_FORMAT_MASK 0x060
+#define WM8785_FORMAT_RJUST 0x000
+#define WM8785_FORMAT_LJUST 0x020
+#define WM8785_FORMAT_I2S 0x040
+#define WM8785_FORMAT_DSP 0x060
+/* R1 */
+#define WM8785_WL_MASK 0x003
+#define WM8785_WL_16 0x000
+#define WM8785_WL_20 0x001
+#define WM8785_WL_24 0x002
+#define WM8785_WL_32 0x003
+#define WM8785_LRP 0x004
+#define WM8785_BCLKINV 0x008
+#define WM8785_LRSWAP 0x010
+#define WM8785_DEVNO_MASK 0x0e0
+/* R2 */
+#define WM8785_HPFR 0x001
+#define WM8785_HPFL 0x002
+#define WM8785_SDODIS 0x004
+#define WM8785_PWRDNR 0x008
+#define WM8785_PWRDNL 0x010
+#define WM8785_TDM_MASK 0x1c0
+
+#endif
diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c
index 9d5bb76229a..7fdcdc8c6b6 100644
--- a/sound/pci/pcxhr/pcxhr.c
+++ b/sound/pci/pcxhr/pcxhr.c
@@ -458,7 +458,7 @@ static int pcxhr_update_r_buffer(struct pcxhr_stream *stream)
snd_printdd("pcxhr_update_r_buffer(pcm%c%d) : addr(%p) bytes(%zx) subs(%d)\n",
is_capture ? 'c' : 'p',
- chip->chip_idx, (void*)subs->runtime->dma_addr,
+ chip->chip_idx, (void *)(long)subs->runtime->dma_addr,
subs->runtime->dma_bytes, subs->number);
pcxhr_init_rmh(&rmh, CMD_UPDATE_R_BUFFERS);
@@ -626,7 +626,7 @@ static void pcxhr_trigger_tasklet(unsigned long arg)
#ifdef CONFIG_SND_DEBUG_DETECT
do_gettimeofday(&my_tv2);
snd_printdd("***TRIGGER TASKLET*** TIME = %ld (err = %x)\n",
- my_tv2.tv_usec - my_tv1.tv_usec, err);
+ (long)(my_tv2.tv_usec - my_tv1.tv_usec), err);
#endif
}
@@ -846,7 +846,6 @@ static int pcxhr_open(struct snd_pcm_substream *subs)
struct pcxhr_mgr *mgr = chip->mgr;
struct snd_pcm_runtime *runtime = subs->runtime;
struct pcxhr_stream *stream;
- int is_capture;
mutex_lock(&mgr->setup_mutex);
@@ -856,12 +855,10 @@ static int pcxhr_open(struct snd_pcm_substream *subs)
if( subs->stream == SNDRV_PCM_STREAM_PLAYBACK ) {
snd_printdd("pcxhr_open playback chip%d subs%d\n",
chip->chip_idx, subs->number);
- is_capture = 0;
stream = &chip->playback_stream[subs->number];
} else {
snd_printdd("pcxhr_open capture chip%d subs%d\n",
chip->chip_idx, subs->number);
- is_capture = 1;
if (mgr->mono_capture)
runtime->hw.channels_max = 1;
else
diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c
index c4e415d0738..78aa81feaa4 100644
--- a/sound/pci/pcxhr/pcxhr_core.c
+++ b/sound/pci/pcxhr/pcxhr_core.c
@@ -897,7 +897,7 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, int capture_m
#ifdef CONFIG_SND_DEBUG_DETECT
do_gettimeofday(&my_tv2);
snd_printdd("***SET PIPE STATE*** TIME = %ld (err = %x)\n",
- my_tv2.tv_usec - my_tv1.tv_usec, err);
+ (long)(my_tv2.tv_usec - my_tv1.tv_usec), err);
#endif
return 0;
}
@@ -1005,30 +1005,37 @@ void pcxhr_msg_tasklet(unsigned long arg)
int nb_stream = (prmh->stat[i] >> (2*FIELD_SIZE)) & MASK_FIRST_FIELD;
int pipe = prmh->stat[i] & MASK_FIRST_FIELD;
int is_capture = prmh->stat[i] & 0x400000;
- u32 err;
+ u32 err2;
if (prmh->stat[i] & 0x800000) { /* if BIT_END */
snd_printdd("TASKLET : End%sPipe %d\n",
is_capture ? "Record" : "Play", pipe);
}
i++;
- err = prmh->stat[i] ? prmh->stat[i] : prmh->stat[i+1];
- if (err)
- pcxhr_handle_async_err(mgr, err, PCXHR_ERR_PIPE,
+ err2 = prmh->stat[i] ? prmh->stat[i] : prmh->stat[i+1];
+ if (err2)
+ pcxhr_handle_async_err(mgr, err2,
+ PCXHR_ERR_PIPE,
pipe, is_capture);
i += 2;
for (j = 0; j < nb_stream; j++) {
- err = prmh->stat[i] ? prmh->stat[i] : prmh->stat[i+1];
- if (err)
- pcxhr_handle_async_err(mgr, err, PCXHR_ERR_STREAM,
- pipe, is_capture);
+ err2 = prmh->stat[i] ?
+ prmh->stat[i] : prmh->stat[i+1];
+ if (err2)
+ pcxhr_handle_async_err(mgr, err2,
+ PCXHR_ERR_STREAM,
+ pipe,
+ is_capture);
i += 2;
}
for (j = 0; j < nb_audio; j++) {
- err = prmh->stat[i] ? prmh->stat[i] : prmh->stat[i+1];
- if (err)
- pcxhr_handle_async_err(mgr, err, PCXHR_ERR_AUDIO,
- pipe, is_capture);
+ err2 = prmh->stat[i] ?
+ prmh->stat[i] : prmh->stat[i+1];
+ if (err2)
+ pcxhr_handle_async_err(mgr, err2,
+ PCXHR_ERR_AUDIO,
+ pipe,
+ is_capture);
i += 2;
}
}
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index 9408b1eeec4..979f7da641c 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -1630,14 +1630,14 @@ static int snd_riptide_playback_open(struct snd_pcm_substream *substream)
struct snd_riptide *chip = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
struct pcmhw *data;
- int index = substream->number;
+ int sub_num = substream->number;
- chip->playback_substream[index] = substream;
+ chip->playback_substream[sub_num] = substream;
runtime->hw = snd_riptide_playback;
data = kzalloc(sizeof(struct pcmhw), GFP_KERNEL);
- data->paths = lbus_play_paths[index];
- data->id = play_ids[index];
- data->source = play_sources[index];
+ data->paths = lbus_play_paths[sub_num];
+ data->id = play_ids[sub_num];
+ data->source = play_sources[sub_num];
data->intdec[0] = 0xff;
data->intdec[1] = 0xff;
data->state = ST_STOP;
@@ -1670,10 +1670,10 @@ static int snd_riptide_playback_close(struct snd_pcm_substream *substream)
{
struct snd_riptide *chip = snd_pcm_substream_chip(substream);
struct pcmhw *data = get_pcmhwdev(substream);
- int index = substream->number;
+ int sub_num = substream->number;
substream->runtime->private_data = NULL;
- chip->playback_substream[index] = NULL;
+ chip->playback_substream[sub_num] = NULL;
kfree(data);
return 0;
}
diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c
index df184aabce8..e7ef3a1a25a 100644
--- a/sound/pci/rme32.c
+++ b/sound/pci/rme32.c
@@ -1350,7 +1350,8 @@ static int __devinit snd_rme32_create(struct rme32 * rme32)
return err;
rme32->port = pci_resource_start(rme32->pci, 0);
- if ((rme32->iobase = ioremap_nocache(rme32->port, RME32_IO_SIZE)) == 0) {
+ rme32->iobase = ioremap_nocache(rme32->port, RME32_IO_SIZE);
+ if (!rme32->iobase) {
snd_printk(KERN_ERR "unable to remap memory region 0x%lx-0x%lx\n",
rme32->port, rme32->port + RME32_IO_SIZE - 1);
return -ENOMEM;
diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c
index fb0a4ee8bc0..3fdd488d097 100644
--- a/sound/pci/rme96.c
+++ b/sound/pci/rme96.c
@@ -1559,7 +1559,8 @@ snd_rme96_create(struct rme96 *rme96)
return err;
rme96->port = pci_resource_start(rme96->pci, 0);
- if ((rme96->iobase = ioremap_nocache(rme96->port, RME96_IO_SIZE)) == 0) {
+ rme96->iobase = ioremap_nocache(rme96->port, RME96_IO_SIZE);
+ if (!rme96->iobase) {
snd_printk(KERN_ERR "unable to remap memory region 0x%lx-0x%lx\n", rme96->port, rme96->port + RME96_IO_SIZE - 1);
return -ENOMEM;
}
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index 1be84f22d0d..4d6fbb36ab8 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -318,6 +318,10 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin");
#define HDSP_midi1IRQPending (1<<31)
#define HDSP_spdifFrequencyMask (HDSP_spdifFrequency0|HDSP_spdifFrequency1|HDSP_spdifFrequency2)
+#define HDSP_spdifFrequencyMask_9632 (HDSP_spdifFrequency0|\
+ HDSP_spdifFrequency1|\
+ HDSP_spdifFrequency2|\
+ HDSP_spdifFrequency3)
#define HDSP_spdifFrequency32KHz (HDSP_spdifFrequency0)
#define HDSP_spdifFrequency44_1KHz (HDSP_spdifFrequency1)
@@ -328,7 +332,9 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin");
#define HDSP_spdifFrequency96KHz (HDSP_spdifFrequency2|HDSP_spdifFrequency1)
/* This is for H9632 cards */
-#define HDSP_spdifFrequency128KHz HDSP_spdifFrequencyMask
+#define HDSP_spdifFrequency128KHz (HDSP_spdifFrequency0|\
+ HDSP_spdifFrequency1|\
+ HDSP_spdifFrequency2)
#define HDSP_spdifFrequency176_4KHz HDSP_spdifFrequency3
#define HDSP_spdifFrequency192KHz (HDSP_spdifFrequency3|HDSP_spdifFrequency0)
@@ -885,28 +891,15 @@ static int snd_hdsp_use_is_exclusive(struct hdsp *hdsp)
return ret;
}
-static int hdsp_external_sample_rate (struct hdsp *hdsp)
-{
- unsigned int status2 = hdsp_read(hdsp, HDSP_status2Register);
- unsigned int rate_bits = status2 & HDSP_systemFrequencyMask;
-
- switch (rate_bits) {
- case HDSP_systemFrequency32: return 32000;
- case HDSP_systemFrequency44_1: return 44100;
- case HDSP_systemFrequency48: return 48000;
- case HDSP_systemFrequency64: return 64000;
- case HDSP_systemFrequency88_2: return 88200;
- case HDSP_systemFrequency96: return 96000;
- default:
- return 0;
- }
-}
-
static int hdsp_spdif_sample_rate(struct hdsp *hdsp)
{
unsigned int status = hdsp_read(hdsp, HDSP_statusRegister);
unsigned int rate_bits = (status & HDSP_spdifFrequencyMask);
+ /* For the 9632, the mask is different */
+ if (hdsp->io_type == H9632)
+ rate_bits = (status & HDSP_spdifFrequencyMask_9632);
+
if (status & HDSP_SPDIFErrorFlag)
return 0;
@@ -933,6 +926,31 @@ static int hdsp_spdif_sample_rate(struct hdsp *hdsp)
return 0;
}
+static int hdsp_external_sample_rate(struct hdsp *hdsp)
+{
+ unsigned int status2 = hdsp_read(hdsp, HDSP_status2Register);
+ unsigned int rate_bits = status2 & HDSP_systemFrequencyMask;
+
+ /* For the 9632 card, there seems to be no bit for indicating external
+ * sample rate greater than 96kHz. The card reports the corresponding
+ * single speed. So the best means seems to get spdif rate when
+ * autosync reference is spdif */
+ if (hdsp->io_type == H9632 &&
+ hdsp_autosync_ref(hdsp) == HDSP_AUTOSYNC_FROM_SPDIF)
+ return hdsp_spdif_sample_rate(hdsp);
+
+ switch (rate_bits) {
+ case HDSP_systemFrequency32: return 32000;
+ case HDSP_systemFrequency44_1: return 44100;
+ case HDSP_systemFrequency48: return 48000;
+ case HDSP_systemFrequency64: return 64000;
+ case HDSP_systemFrequency88_2: return 88200;
+ case HDSP_systemFrequency96: return 96000;
+ default:
+ return 0;
+ }
+}
+
static void hdsp_compute_period_size(struct hdsp *hdsp)
{
hdsp->period_bytes = 1 << ((hdsp_decode_latency(hdsp->control_register) + 8));
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 9a19ae6a64d..ab423bc8234 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -540,7 +540,8 @@ static void hdspm_set_sgbuf(struct hdspm * hdspm, struct snd_sg_buf *sgbuf,
static inline int HDSPM_bit2freq(int n)
{
- static int bit2freq_tab[] = { 0, 32000, 44100, 48000, 64000, 88200,
+ static const int bit2freq_tab[] = {
+ 0, 32000, 44100, 48000, 64000, 88200,
96000, 128000, 176400, 192000 };
if (n < 1 || n > 9)
return 0;
@@ -582,7 +583,7 @@ static inline int hdspm_read_pb_gain(struct hdspm * hdspm, unsigned int chan,
return hdspm->mixer->ch[chan].pb[pb];
}
-static inline int hdspm_write_in_gain(struct hdspm * hdspm, unsigned int chan,
+static int hdspm_write_in_gain(struct hdspm *hdspm, unsigned int chan,
unsigned int in, unsigned short data)
{
if (chan >= HDSPM_MIXER_CHANNELS || in >= HDSPM_MIXER_CHANNELS)
@@ -595,7 +596,7 @@ static inline int hdspm_write_in_gain(struct hdspm * hdspm, unsigned int chan,
return 0;
}
-static inline int hdspm_write_pb_gain(struct hdspm * hdspm, unsigned int chan,
+static int hdspm_write_pb_gain(struct hdspm *hdspm, unsigned int chan,
unsigned int pb, unsigned short data)
{
if (chan >= HDSPM_MIXER_CHANNELS || pb >= HDSPM_MIXER_CHANNELS)
@@ -621,7 +622,7 @@ static inline void snd_hdspm_enable_out(struct hdspm * hdspm, int i, int v)
}
/* check if same process is writing and reading */
-static inline int snd_hdspm_use_is_exclusive(struct hdspm * hdspm)
+static int snd_hdspm_use_is_exclusive(struct hdspm *hdspm)
{
unsigned long flags;
int ret = 1;
@@ -636,7 +637,7 @@ static inline int snd_hdspm_use_is_exclusive(struct hdspm * hdspm)
}
/* check for external sample rate */
-static inline int hdspm_external_sample_rate(struct hdspm * hdspm)
+static int hdspm_external_sample_rate(struct hdspm *hdspm)
{
if (hdspm->is_aes32) {
unsigned int status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
@@ -787,7 +788,7 @@ static inline void hdspm_stop_audio(struct hdspm * s)
}
/* should I silence all or only opened ones ? doit all for first even is 4MB*/
-static inline void hdspm_silence_playback(struct hdspm * hdspm)
+static void hdspm_silence_playback(struct hdspm *hdspm)
{
int i;
int n = hdspm->period_bytes;
@@ -1028,9 +1029,9 @@ static inline void snd_hdspm_midi_write_byte (struct hdspm *hdspm, int id,
{
/* the hardware already does the relevant bit-mask with 0xff */
if (id)
- return hdspm_write(hdspm, HDSPM_midiDataOut1, val);
+ hdspm_write(hdspm, HDSPM_midiDataOut1, val);
else
- return hdspm_write(hdspm, HDSPM_midiDataOut0, val);
+ hdspm_write(hdspm, HDSPM_midiDataOut0, val);
}
static inline int snd_hdspm_midi_input_available (struct hdspm *hdspm, int id)
@@ -1057,7 +1058,7 @@ static inline int snd_hdspm_midi_output_possible (struct hdspm *hdspm, int id)
return 0;
}
-static inline void snd_hdspm_flush_midi_input (struct hdspm *hdspm, int id)
+static void snd_hdspm_flush_midi_input(struct hdspm *hdspm, int id)
{
while (snd_hdspm_midi_input_available (hdspm, id))
snd_hdspm_midi_read_byte (hdspm, id);
diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index dcd7cd01046..df2007e3be7 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -920,7 +920,7 @@ static unsigned short sis_ac97_rw(struct sis7019 *sis, int codec, u32 cmd)
u16 status;
u16 rdy;
int count;
- const static u16 codec_ready[3] = {
+ static const u16 codec_ready[3] = {
SIS_AC97_STATUS_CODEC_READY,
SIS_AC97_STATUS_CODEC2_READY,
SIS_AC97_STATUS_CODEC3_READY,
@@ -984,7 +984,7 @@ timeout:
static void sis_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
unsigned short val)
{
- const static u32 cmd[3] = {
+ static const u32 cmd[3] = {
SIS_AC97_CMD_CODEC_WRITE,
SIS_AC97_CMD_CODEC2_WRITE,
SIS_AC97_CMD_CODEC3_WRITE,
@@ -995,7 +995,7 @@ static void sis_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
static unsigned short sis_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
{
- const static u32 cmd[3] = {
+ static const u32 cmd[3] = {
SIS_AC97_CMD_CODEC_READ,
SIS_AC97_CMD_CODEC2_READ,
SIS_AC97_CMD_CODEC3_READ,
@@ -1194,7 +1194,6 @@ static int sis_suspend(struct pci_dev *pci, pm_message_t state)
/* snd_pcm_suspend_all() stopped all channels, so we're quiescent.
*/
if (sis->irq >= 0) {
- synchronize_irq(sis->irq);
free_irq(sis->irq, sis);
sis->irq = -1;
}
diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c
index 71138ff9b31..bbcee2c09ae 100644
--- a/sound/pci/trident/trident_main.c
+++ b/sound/pci/trident/trident_main.c
@@ -3676,6 +3676,8 @@ static int snd_trident_free(struct snd_trident *trident)
else if (trident->device == TRIDENT_DEVICE_ID_SI7018) {
outl(0, TRID_REG(trident, SI_SERIAL_INTF_CTRL));
}
+ if (trident->irq >= 0)
+ free_irq(trident->irq, trident);
if (trident->tlb.buffer.area) {
outl(0, TRID_REG(trident, NX_TLBC));
if (trident->tlb.memhdr)
@@ -3685,8 +3687,6 @@ static int snd_trident_free(struct snd_trident *trident)
vfree(trident->tlb.shadow_entries);
snd_dma_free_pages(&trident->tlb.buffer);
}
- if (trident->irq >= 0)
- free_irq(trident->irq, trident);
pci_release_regions(trident->pci);
pci_disable_device(trident->pci);
kfree(trident);
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index a756be661f9..b585cc3e4c4 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -2236,7 +2236,7 @@ static int snd_via82xx_free(struct via82xx *chip)
/* disable interrupts */
for (i = 0; i < chip->num_devs; i++)
snd_via82xx_channel_reset(chip, &chip->devs[i]);
- synchronize_irq(chip->irq);
+
if (chip->irq >= 0)
free_irq(chip->irq, chip);
__end_hw:
diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c
index f5df1c79bee..31f64ee3988 100644
--- a/sound/pci/via82xx_modem.c
+++ b/sound/pci/via82xx_modem.c
@@ -1075,7 +1075,7 @@ static int snd_via82xx_free(struct via82xx_modem *chip)
/* disable interrupts */
for (i = 0; i < chip->num_devs; i++)
snd_via82xx_channel_reset(chip, &chip->devs[i]);
- synchronize_irq(chip->irq);
+
__end_hw:
if (chip->irq >= 0)
free_irq(chip->irq, chip);
diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c
index 42c1eb7d35f..29b3056c510 100644
--- a/sound/pci/ymfpci/ymfpci_main.c
+++ b/sound/pci/ymfpci/ymfpci_main.c
@@ -2249,6 +2249,8 @@ static int snd_ymfpci_free(struct snd_ymfpci *chip)
#ifdef CONFIG_PM
vfree(chip->saved_regs);
#endif
+ if (chip->irq >= 0)
+ free_irq(chip->irq, chip);
release_and_free_resource(chip->mpu_res);
release_and_free_resource(chip->fm_res);
snd_ymfpci_free_gameport(chip);
@@ -2257,8 +2259,6 @@ static int snd_ymfpci_free(struct snd_ymfpci *chip)
if (chip->work_ptr.area)
snd_dma_free_pages(&chip->work_ptr);
- if (chip->irq >= 0)
- free_irq(chip->irq, chip);
release_and_free_resource(chip->res_reg_area);
pci_write_config_word(chip->pci, 0x40, chip->old_legacy_ctrl);