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authorSteven Whitehouse <swhiteho@redhat.com>2006-09-25 12:26:59 -0400
committerSteven Whitehouse <swhiteho@redhat.com>2006-09-25 12:26:59 -0400
commit363e065c02b1273364d5356711a83e7f548fc0c8 (patch)
tree0df0e65da403ade33ade580c2770c97437b1b1af /sound/pci
parent907b9bceb41fa46beae93f79cc4a2247df502c0f (diff)
parent7c250413e5b7c3dfae89354725b70c76d7621395 (diff)
[GFS2] Fix up merge of Linus' kernel into GFS2
This fixes up a couple of conflicts when merging up with Linus' latest kernel. This will hopefully allow GFS2 to be more easily merged into forthcoming -mm and FC kernels due to the "one line per header" format now used for the kernel headers. Signed-off-by: Steven Whitehouse <swhiteho@redhat.com> Conflicts: include/linux/Kbuild include/linux/kernel.h
Diffstat (limited to 'sound/pci')
-rw-r--r--sound/pci/Kconfig14
-rw-r--r--sound/pci/ac97/ac97_codec.c334
-rw-r--r--sound/pci/ac97/ac97_patch.c98
-rw-r--r--sound/pci/ac97/ac97_patch.h1
-rw-r--r--sound/pci/ac97/ac97_pcm.c18
-rw-r--r--sound/pci/ac97/ac97_proc.c18
-rw-r--r--sound/pci/ac97/ak4531_codec.c49
-rw-r--r--sound/pci/ca0106/ca0106_mixer.c10
-rw-r--r--sound/pci/cs4281.c5
-rw-r--r--sound/pci/cs46xx/dsp_spos.c52
-rw-r--r--sound/pci/cs46xx/dsp_spos_scb_lib.c2
-rw-r--r--sound/pci/cs5535audio/Makefile2
-rw-r--r--sound/pci/emu10k1/emu10k1.c2
-rw-r--r--sound/pci/emu10k1/emu10k1_main.c1
-rw-r--r--sound/pci/emu10k1/emu10k1x.c7
-rw-r--r--sound/pci/emu10k1/emufx.c12
-rw-r--r--sound/pci/emu10k1/p16v.c5
-rw-r--r--sound/pci/es1938.c104
-rw-r--r--sound/pci/es1968.c40
-rw-r--r--sound/pci/fm801.c63
-rw-r--r--sound/pci/hda/hda_codec.c76
-rw-r--r--sound/pci/hda/hda_codec.h2
-rw-r--r--sound/pci/hda/hda_generic.c199
-rw-r--r--sound/pci/hda/hda_intel.c132
-rw-r--r--sound/pci/hda/hda_local.h8
-rw-r--r--sound/pci/hda/hda_proc.c12
-rw-r--r--sound/pci/hda/patch_analog.c21
-rw-r--r--sound/pci/hda/patch_realtek.c330
-rw-r--r--sound/pci/hda/patch_si3054.c1
-rw-r--r--sound/pci/hda/patch_sigmatel.c904
-rw-r--r--sound/pci/ice1712/aureon.c104
-rw-r--r--sound/pci/ice1712/ice1712.c14
-rw-r--r--sound/pci/ice1712/phase.c39
-rw-r--r--sound/pci/ice1712/pontis.c9
-rw-r--r--sound/pci/ice1712/prodigy192.c14
-rw-r--r--sound/pci/ice1712/revo.c68
-rw-r--r--sound/pci/ice1712/revo.h2
-rw-r--r--sound/pci/intel8x0.c14
-rw-r--r--sound/pci/intel8x0m.c5
-rw-r--r--sound/pci/mixart/mixart.c12
-rw-r--r--sound/pci/mixart/mixart_mixer.c14
-rw-r--r--sound/pci/pcxhr/pcxhr_mixer.c16
-rw-r--r--sound/pci/riptide/riptide.c10
-rw-r--r--sound/pci/rme9652/hdsp.c48
-rw-r--r--sound/pci/trident/trident_main.c10
-rw-r--r--sound/pci/via82xx.c23
-rw-r--r--sound/pci/vx222/vx222.c7
-rw-r--r--sound/pci/vx222/vx222_ops.c9
-rw-r--r--sound/pci/ymfpci/ymfpci_main.c7
49 files changed, 2346 insertions, 601 deletions
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index e49c0fe21b0..8a6b1803c76 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -475,6 +475,7 @@ config SND_FM801_TEA575X
depends on SND_FM801_TEA575X_BOOL
default SND_FM801
select VIDEO_V4L1
+ select VIDEO_DEV
config SND_HDA_INTEL
tristate "Intel HD Audio"
@@ -743,4 +744,17 @@ config SND_YMFPCI
To compile this driver as a module, choose M here: the module
will be called snd-ymfpci.
+config SND_AC97_POWER_SAVE
+ bool "AC97 Power-Saving Mode"
+ depends on SND_AC97_CODEC && EXPERIMENTAL
+ default n
+ help
+ Say Y here to enable the aggressive power-saving support of
+ AC97 codecs. In this mode, the power-mode is dynamically
+ controlled at each open/close.
+
+ The mode is activated by passing power_save=1 option to
+ snd-ac97-codec driver. You can toggle it dynamically over
+ sysfs, too.
+
endmenu
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index 51e83d7a839..a79e91850ba 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -31,6 +31,7 @@
#include <linux/mutex.h>
#include <sound/core.h>
#include <sound/pcm.h>
+#include <sound/tlv.h>
#include <sound/ac97_codec.h>
#include <sound/asoundef.h>
#include <sound/initval.h>
@@ -47,6 +48,11 @@ static int enable_loopback;
module_param(enable_loopback, bool, 0444);
MODULE_PARM_DESC(enable_loopback, "Enable AC97 ADC/DAC Loopback Control");
+#ifdef CONFIG_SND_AC97_POWER_SAVE
+static int power_save;
+module_param(power_save, bool, 0644);
+MODULE_PARM_DESC(power_save, "Enable AC97 power-saving control");
+#endif
/*
*/
@@ -151,7 +157,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = {
{ 0x4e534300, 0xffffffff, "LM4540,43,45,46,48", NULL, NULL }, // only guess --jk
{ 0x4e534331, 0xffffffff, "LM4549", NULL, NULL },
{ 0x4e534350, 0xffffffff, "LM4550", patch_lm4550, NULL }, // volume wrap fix
-{ 0x50534304, 0xffffffff, "UCB1400", NULL, NULL },
+{ 0x50534304, 0xffffffff, "UCB1400", patch_ucb1400, NULL },
{ 0x53494c20, 0xffffffe0, "Si3036,8", mpatch_si3036, mpatch_si3036, AC97_MODEM_PATCH },
{ 0x54524102, 0xffffffff, "TR28022", NULL, NULL },
{ 0x54524106, 0xffffffff, "TR28026", NULL, NULL },
@@ -187,6 +193,8 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = {
};
+static void update_power_regs(struct snd_ac97 *ac97);
+
/*
* I/O routines
*/
@@ -554,6 +562,18 @@ int snd_ac97_put_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value
}
err = snd_ac97_update_bits(ac97, reg, val_mask, val);
snd_ac97_page_restore(ac97, page_save);
+#ifdef CONFIG_SND_AC97_POWER_SAVE
+ /* check analog mixer power-down */
+ if ((val_mask & 0x8000) &&
+ (kcontrol->private_value & (1<<30))) {
+ if (val & 0x8000)
+ ac97->power_up &= ~(1 << (reg>>1));
+ else
+ ac97->power_up |= 1 << (reg>>1);
+ if (power_save)
+ update_power_regs(ac97);
+ }
+#endif
return err;
}
@@ -962,6 +982,10 @@ static int snd_ac97_bus_dev_free(struct snd_device *device)
static int snd_ac97_free(struct snd_ac97 *ac97)
{
if (ac97) {
+#ifdef CONFIG_SND_AC97_POWER_SAVE
+ if (ac97->power_workq)
+ destroy_workqueue(ac97->power_workq);
+#endif
snd_ac97_proc_done(ac97);
if (ac97->bus)
ac97->bus->codec[ac97->num] = NULL;
@@ -1117,7 +1141,9 @@ struct snd_kcontrol *snd_ac97_cnew(const struct snd_kcontrol_new *_template, str
/*
* create mute switch(es) for normal stereo controls
*/
-static int snd_ac97_cmute_new_stereo(struct snd_card *card, char *name, int reg, int check_stereo, struct snd_ac97 *ac97)
+static int snd_ac97_cmute_new_stereo(struct snd_card *card, char *name, int reg,
+ int check_stereo, int check_amix,
+ struct snd_ac97 *ac97)
{
struct snd_kcontrol *kctl;
int err;
@@ -1137,10 +1163,14 @@ static int snd_ac97_cmute_new_stereo(struct snd_card *card, char *name, int reg,
}
if (mute_mask == 0x8080) {
struct snd_kcontrol_new tmp = AC97_DOUBLE(name, reg, 15, 7, 1, 1);
+ if (check_amix)
+ tmp.private_value |= (1 << 30);
tmp.index = ac97->num;
kctl = snd_ctl_new1(&tmp, ac97);
} else {
struct snd_kcontrol_new tmp = AC97_SINGLE(name, reg, 15, 1, 1);
+ if (check_amix)
+ tmp.private_value |= (1 << 30);
tmp.index = ac97->num;
kctl = snd_ctl_new1(&tmp, ac97);
}
@@ -1153,6 +1183,32 @@ static int snd_ac97_cmute_new_stereo(struct snd_card *card, char *name, int reg,
}
/*
+ * set dB information
+ */
+static DECLARE_TLV_DB_SCALE(db_scale_4bit, -4500, 300, 0);
+static DECLARE_TLV_DB_SCALE(db_scale_5bit, -4650, 150, 0);
+static DECLARE_TLV_DB_SCALE(db_scale_6bit, -9450, 150, 0);
+static DECLARE_TLV_DB_SCALE(db_scale_5bit_12db_max, -3450, 150, 0);
+static DECLARE_TLV_DB_SCALE(db_scale_rec_gain, 0, 150, 0);
+
+static unsigned int *find_db_scale(unsigned int maxval)
+{
+ switch (maxval) {
+ case 0x0f: return db_scale_4bit;
+ case 0x1f: return db_scale_5bit;
+ case 0x3f: return db_scale_6bit;
+ }
+ return NULL;
+}
+
+static void set_tlv_db_scale(struct snd_kcontrol *kctl, unsigned int *tlv)
+{
+ kctl->tlv.p = tlv;
+ if (tlv)
+ kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ;
+}
+
+/*
* create a volume for normal stereo/mono controls
*/
static int snd_ac97_cvol_new(struct snd_card *card, char *name, int reg, unsigned int lo_max,
@@ -1174,6 +1230,10 @@ static int snd_ac97_cvol_new(struct snd_card *card, char *name, int reg, unsigne
tmp.index = ac97->num;
kctl = snd_ctl_new1(&tmp, ac97);
}
+ if (reg >= AC97_PHONE && reg <= AC97_PCM)
+ set_tlv_db_scale(kctl, db_scale_5bit_12db_max);
+ else
+ set_tlv_db_scale(kctl, find_db_scale(lo_max));
err = snd_ctl_add(card, kctl);
if (err < 0)
return err;
@@ -1186,7 +1246,9 @@ static int snd_ac97_cvol_new(struct snd_card *card, char *name, int reg, unsigne
/*
* create a mute-switch and a volume for normal stereo/mono controls
*/
-static int snd_ac97_cmix_new_stereo(struct snd_card *card, const char *pfx, int reg, int check_stereo, struct snd_ac97 *ac97)
+static int snd_ac97_cmix_new_stereo(struct snd_card *card, const char *pfx,
+ int reg, int check_stereo, int check_amix,
+ struct snd_ac97 *ac97)
{
int err;
char name[44];
@@ -1197,7 +1259,9 @@ static int snd_ac97_cmix_new_stereo(struct snd_card *card, const char *pfx, int
if (snd_ac97_try_bit(ac97, reg, 15)) {
sprintf(name, "%s Switch", pfx);
- if ((err = snd_ac97_cmute_new_stereo(card, name, reg, check_stereo, ac97)) < 0)
+ if ((err = snd_ac97_cmute_new_stereo(card, name, reg,
+ check_stereo, check_amix,
+ ac97)) < 0)
return err;
}
check_volume_resolution(ac97, reg, &lo_max, &hi_max);
@@ -1209,8 +1273,10 @@ static int snd_ac97_cmix_new_stereo(struct snd_card *card, const char *pfx, int
return 0;
}
-#define snd_ac97_cmix_new(card, pfx, reg, ac97) snd_ac97_cmix_new_stereo(card, pfx, reg, 0, ac97)
-#define snd_ac97_cmute_new(card, name, reg, ac97) snd_ac97_cmute_new_stereo(card, name, reg, 0, ac97)
+#define snd_ac97_cmix_new(card, pfx, reg, acheck, ac97) \
+ snd_ac97_cmix_new_stereo(card, pfx, reg, 0, acheck, ac97)
+#define snd_ac97_cmute_new(card, name, reg, acheck, ac97) \
+ snd_ac97_cmute_new_stereo(card, name, reg, 0, acheck, ac97)
static unsigned int snd_ac97_determine_spdif_rates(struct snd_ac97 *ac97);
@@ -1226,9 +1292,11 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97)
/* AD claims to remove this control from AD1887, although spec v2.2 does not allow this */
if (snd_ac97_try_volume_mix(ac97, AC97_MASTER)) {
if (ac97->flags & AC97_HAS_NO_MASTER_VOL)
- err = snd_ac97_cmute_new(card, "Master Playback Switch", AC97_MASTER, ac97);
+ err = snd_ac97_cmute_new(card, "Master Playback Switch",
+ AC97_MASTER, 0, ac97);
else
- err = snd_ac97_cmix_new(card, "Master Playback", AC97_MASTER, ac97);
+ err = snd_ac97_cmix_new(card, "Master Playback",
+ AC97_MASTER, 0, ac97);
if (err < 0)
return err;
}
@@ -1245,6 +1313,7 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97)
snd_ac97_change_volume_params2(ac97, AC97_CENTER_LFE_MASTER, 0, &max);
kctl->private_value &= ~(0xff << 16);
kctl->private_value |= (int)max << 16;
+ set_tlv_db_scale(kctl, find_db_scale(max));
snd_ac97_write_cache(ac97, AC97_CENTER_LFE_MASTER, ac97->regs[AC97_CENTER_LFE_MASTER] | max);
}
@@ -1258,6 +1327,7 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97)
snd_ac97_change_volume_params2(ac97, AC97_CENTER_LFE_MASTER, 8, &max);
kctl->private_value &= ~(0xff << 16);
kctl->private_value |= (int)max << 16;
+ set_tlv_db_scale(kctl, find_db_scale(max));
snd_ac97_write_cache(ac97, AC97_CENTER_LFE_MASTER, ac97->regs[AC97_CENTER_LFE_MASTER] | max << 8);
}
@@ -1265,19 +1335,23 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97)
if ((snd_ac97_try_volume_mix(ac97, AC97_SURROUND_MASTER))
&& !(ac97->flags & AC97_AD_MULTI)) {
/* Surround Master (0x38) is with stereo mutes */
- if ((err = snd_ac97_cmix_new_stereo(card, "Surround Playback", AC97_SURROUND_MASTER, 1, ac97)) < 0)
+ if ((err = snd_ac97_cmix_new_stereo(card, "Surround Playback",
+ AC97_SURROUND_MASTER, 1, 0,
+ ac97)) < 0)
return err;
}
/* build headphone controls */
if (snd_ac97_try_volume_mix(ac97, AC97_HEADPHONE)) {
- if ((err = snd_ac97_cmix_new(card, "Headphone Playback", AC97_HEADPHONE, ac97)) < 0)
+ if ((err = snd_ac97_cmix_new(card, "Headphone Playback",
+ AC97_HEADPHONE, 0, ac97)) < 0)
return err;
}
/* build master mono controls */
if (snd_ac97_try_volume_mix(ac97, AC97_MASTER_MONO)) {
- if ((err = snd_ac97_cmix_new(card, "Master Mono Playback", AC97_MASTER_MONO, ac97)) < 0)
+ if ((err = snd_ac97_cmix_new(card, "Master Mono Playback",
+ AC97_MASTER_MONO, 0, ac97)) < 0)
return err;
}
@@ -1301,8 +1375,9 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97)
((ac97->flags & AC97_HAS_PC_BEEP) ||
snd_ac97_try_volume_mix(ac97, AC97_PC_BEEP))) {
for (idx = 0; idx < 2; idx++)
- if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_pc_beep[idx], ac97))) < 0)
+ if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_pc_beep[idx], ac97))) < 0)
return err;
+ set_tlv_db_scale(kctl, db_scale_4bit);
snd_ac97_write_cache(ac97, AC97_PC_BEEP,
snd_ac97_read(ac97, AC97_PC_BEEP) | 0x801e);
}
@@ -1310,7 +1385,8 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97)
/* build Phone controls */
if (!(ac97->flags & AC97_HAS_NO_PHONE)) {
if (snd_ac97_try_volume_mix(ac97, AC97_PHONE)) {
- if ((err = snd_ac97_cmix_new(card, "Phone Playback", AC97_PHONE, ac97)) < 0)
+ if ((err = snd_ac97_cmix_new(card, "Phone Playback",
+ AC97_PHONE, 1, ac97)) < 0)
return err;
}
}
@@ -1318,7 +1394,8 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97)
/* build MIC controls */
if (!(ac97->flags & AC97_HAS_NO_MIC)) {
if (snd_ac97_try_volume_mix(ac97, AC97_MIC)) {
- if ((err = snd_ac97_cmix_new(card, "Mic Playback", AC97_MIC, ac97)) < 0)
+ if ((err = snd_ac97_cmix_new(card, "Mic Playback",
+ AC97_MIC, 1, ac97)) < 0)
return err;
if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_mic_boost, ac97))) < 0)
return err;
@@ -1327,14 +1404,16 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97)
/* build Line controls */
if (snd_ac97_try_volume_mix(ac97, AC97_LINE)) {
- if ((err = snd_ac97_cmix_new(card, "Line Playback", AC97_LINE, ac97)) < 0)
+ if ((err = snd_ac97_cmix_new(card, "Line Playback",
+ AC97_LINE, 1, ac97)) < 0)
return err;
}
/* build CD controls */
if (!(ac97->flags & AC97_HAS_NO_CD)) {
if (snd_ac97_try_volume_mix(ac97, AC97_CD)) {
- if ((err = snd_ac97_cmix_new(card, "CD Playback", AC97_CD, ac97)) < 0)
+ if ((err = snd_ac97_cmix_new(card, "CD Playback",
+ AC97_CD, 1, ac97)) < 0)
return err;
}
}
@@ -1342,7 +1421,8 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97)
/* build Video controls */
if (!(ac97->flags & AC97_HAS_NO_VIDEO)) {
if (snd_ac97_try_volume_mix(ac97, AC97_VIDEO)) {
- if ((err = snd_ac97_cmix_new(card, "Video Playback", AC97_VIDEO, ac97)) < 0)
+ if ((err = snd_ac97_cmix_new(card, "Video Playback",
+ AC97_VIDEO, 1, ac97)) < 0)
return err;
}
}
@@ -1350,7 +1430,8 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97)
/* build Aux controls */
if (!(ac97->flags & AC97_HAS_NO_AUX)) {
if (snd_ac97_try_volume_mix(ac97, AC97_AUX)) {
- if ((err = snd_ac97_cmix_new(card, "Aux Playback", AC97_AUX, ac97)) < 0)
+ if ((err = snd_ac97_cmix_new(card, "Aux Playback",
+ AC97_AUX, 1, ac97)) < 0)
return err;
}
}
@@ -1363,31 +1444,38 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97)
else
init_val = 0x9f1f;
for (idx = 0; idx < 2; idx++)
- if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_ad18xx_pcm[idx], ac97))) < 0)
+ if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_ad18xx_pcm[idx], ac97))) < 0)
return err;
+ set_tlv_db_scale(kctl, db_scale_5bit);
ac97->spec.ad18xx.pcmreg[0] = init_val;
if (ac97->scaps & AC97_SCAP_SURROUND_DAC) {
for (idx = 0; idx < 2; idx++)
- if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_ad18xx_surround[idx], ac97))) < 0)
+ if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_ad18xx_surround[idx], ac97))) < 0)
return err;
+ set_tlv_db_scale(kctl, db_scale_5bit);
ac97->spec.ad18xx.pcmreg[1] = init_val;
}
if (ac97->scaps & AC97_SCAP_CENTER_LFE_DAC) {
for (idx = 0; idx < 2; idx++)
- if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_ad18xx_center[idx], ac97))) < 0)
+ if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_ad18xx_center[idx], ac97))) < 0)
return err;
+ set_tlv_db_scale(kctl, db_scale_5bit);
for (idx = 0; idx < 2; idx++)
- if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_ad18xx_lfe[idx], ac97))) < 0)
+ if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_ad18xx_lfe[idx], ac97))) < 0)
return err;
+ set_tlv_db_scale(kctl, db_scale_5bit);
ac97->spec.ad18xx.pcmreg[2] = init_val;
}
snd_ac97_write_cache(ac97, AC97_PCM, init_val);
} else {
if (!(ac97->flags & AC97_HAS_NO_STD_PCM)) {
if (ac97->flags & AC97_HAS_NO_PCM_VOL)
- err = snd_ac97_cmute_new(card, "PCM Playback Switch", AC97_PCM, ac97);
+ err = snd_ac97_cmute_new(card,
+ "PCM Playback Switch",
+ AC97_PCM, 0, ac97);
else
- err = snd_ac97_cmix_new(card, "PCM Playback", AC97_PCM, ac97);
+ err = snd_ac97_cmix_new(card, "PCM Playback",
+ AC97_PCM, 0, ac97);
if (err < 0)
return err;
}
@@ -1398,19 +1486,23 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97)
if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_control_capture_src, ac97))) < 0)
return err;
if (snd_ac97_try_bit(ac97, AC97_REC_GAIN, 15)) {
- if ((err = snd_ac97_cmute_new(card, "Capture Switch", AC97_REC_GAIN, ac97)) < 0)
+ err = snd_ac97_cmute_new(card, "Capture Switch",
+ AC97_REC_GAIN, 0, ac97);
+ if (err < 0)
return err;
}
- if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_control_capture_vol, ac97))) < 0)
+ if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_control_capture_vol, ac97))) < 0)
return err;
+ set_tlv_db_scale(kctl, db_scale_rec_gain);
snd_ac97_write_cache(ac97, AC97_REC_SEL, 0x0000);
snd_ac97_write_cache(ac97, AC97_REC_GAIN, 0x0000);
}
/* build MIC Capture controls */
if (snd_ac97_try_volume_mix(ac97, AC97_REC_GAIN_MIC)) {
for (idx = 0; idx < 2; idx++)
- if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_mic_capture[idx], ac97))) < 0)
+ if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_mic_capture[idx], ac97))) < 0)
return err;
+ set_tlv_db_scale(kctl, db_scale_rec_gain);
snd_ac97_write_cache(ac97, AC97_REC_GAIN_MIC, 0x0000);
}
@@ -1481,6 +1573,12 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97)
}
/* build S/PDIF controls */
+
+ /* Hack for ASUS P5P800-VM, which does not indicate S/PDIF capability */
+ if (ac97->subsystem_vendor == 0x1043 &&
+ ac97->subsystem_device == 0x810f)
+ ac97->ext_id |= AC97_EI_SPDIF;
+
if ((ac97->ext_id & AC97_EI_SPDIF) && !(ac97->scaps & AC97_SCAP_NO_SPDIF)) {
if (ac97->build_ops->build_spdif) {
if ((err = ac97->build_ops->build_spdif(ac97)) < 0)
@@ -1817,18 +1915,25 @@ static int snd_ac97_dev_register(struct snd_device *device)
return 0;
}
-/* unregister ac97 codec */
-static int snd_ac97_dev_unregister(struct snd_device *device)
+/* disconnect ac97 codec */
+static int snd_ac97_dev_disconnect(struct snd_device *device)
{
struct snd_ac97 *ac97 = device->device_data;
if (ac97->dev.bus)
device_unregister(&ac97->dev);
- return snd_ac97_free(ac97);
+ return 0;
}
/* build_ops to do nothing */
static struct snd_ac97_build_ops null_build_ops;
+#ifdef CONFIG_SND_AC97_POWER_SAVE
+static void do_update_power(void *data)
+{
+ update_power_regs(data);
+}
+#endif
+
/**
* snd_ac97_mixer - create an Codec97 component
* @bus: the AC97 bus which codec is attached to
@@ -1860,7 +1965,7 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template,
static struct snd_device_ops ops = {
.dev_free = snd_ac97_dev_free,
.dev_register = snd_ac97_dev_register,
- .dev_unregister = snd_ac97_dev_unregister,
+ .dev_disconnect = snd_ac97_dev_disconnect,
};
snd_assert(rac97 != NULL, return -EINVAL);
@@ -1883,6 +1988,10 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template,
bus->codec[ac97->num] = ac97;
mutex_init(&ac97->reg_mutex);
mutex_init(&ac97->page_mutex);
+#ifdef CONFIG_SND_AC97_POWER_SAVE
+ ac97->power_workq = create_workqueue("ac97");
+ INIT_WORK(&ac97->power_work, do_update_power, ac97);
+#endif
#ifdef CONFIG_PCI
if (ac97->pci) {
@@ -2117,15 +2226,8 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template,
return -ENOMEM;
}
}
- /* make sure the proper powerdown bits are cleared */
- if (ac97->scaps && ac97_is_audio(ac97)) {
- reg = snd_ac97_read(ac97, AC97_EXTENDED_STATUS);
- if (ac97->scaps & AC97_SCAP_SURROUND_DAC)
- reg &= ~AC97_EA_PRJ;
- if (ac97->scaps & AC97_SCAP_CENTER_LFE_DAC)
- reg &= ~(AC97_EA_PRI | AC97_EA_PRK);
- snd_ac97_write_cache(ac97, AC97_EXTENDED_STATUS, reg);
- }
+ if (ac97_is_audio(ac97))
+ update_power_regs(ac97);
snd_ac97_proc_init(ac97);
if ((err = snd_device_new(card, SNDRV_DEV_CODEC, ac97, &ops)) < 0) {
snd_ac97_free(ac97);
@@ -2153,19 +2255,152 @@ static void snd_ac97_powerdown(struct snd_ac97 *ac97)
snd_ac97_write(ac97, AC97_HEADPHONE, 0x9f9f);
}
- power = ac97->regs[AC97_POWERDOWN] | 0x8000; /* EAPD */
- power |= 0x4000; /* Headphone amplifier powerdown */
- power |= 0x0300; /* ADC & DAC powerdown */
- snd_ac97_write(ac97, AC97_POWERDOWN, power);
- udelay(100);
- power |= 0x0400; /* Analog Mixer powerdown (Vref on) */
+ /* surround, CLFE, mic powerdown */
+ power = ac97->regs[AC97_EXTENDED_STATUS];
+ if (ac97->scaps & AC97_SCAP_SURROUND_DAC)
+ power |= AC97_EA_PRJ;
+ if (ac97->scaps & AC97_SCAP_CENTER_LFE_DAC)
+ power |= AC97_EA_PRI | AC97_EA_PRK;
+ power |= AC97_EA_PRL;
+ snd_ac97_write(ac97, AC97_EXTENDED_STATUS, power);
+
+ /* powerdown external amplifier */
+ if (ac97->scaps & AC97_SCAP_INV_EAPD)
+ power = ac97->regs[AC97_POWERDOWN] & ~AC97_PD_EAPD;
+ else if (! (ac97->scaps & AC97_SCAP_EAPD_LED))
+ power = ac97->regs[AC97_POWERDOWN] | AC97_PD_EAPD;
+ power |= AC97_PD_PR6; /* Headphone amplifier powerdown */
+ power |= AC97_PD_PR0 | AC97_PD_PR1; /* ADC & DAC powerdown */
snd_ac97_write(ac97, AC97_POWERDOWN, power);
udelay(100);
-#if 0
- /* FIXME: this causes click noises on some boards at resume */
- power |= 0x3800; /* AC-link powerdown, internal Clk disable */
+ power |= AC97_PD_PR2 | AC97_PD_PR3; /* Analog Mixer powerdown */
snd_ac97_write(ac97, AC97_POWERDOWN, power);
+#ifdef CONFIG_SND_AC97_POWER_SAVE
+ if (power_save) {
+ udelay(100);
+ /* AC-link powerdown, internal Clk disable */
+ /* FIXME: this may cause click noises on some boards */
+ power |= AC97_PD_PR4 | AC97_PD_PR5;
+ snd_ac97_write(ac97, AC97_POWERDOWN, power);
+ }
+#endif
+}
+
+
+struct ac97_power_reg {
+ unsigned short reg;
+ unsigned short power_reg;
+ unsigned short mask;
+};
+
+enum { PWIDX_ADC, PWIDX_FRONT, PWIDX_CLFE, PWIDX_SURR, PWIDX_MIC, PWIDX_SIZE };
+
+static struct ac97_power_reg power_regs[PWIDX_SIZE] = {
+ [PWIDX_ADC] = { AC97_PCM_LR_ADC_RATE, AC97_POWERDOWN, AC97_PD_PR0},
+ [PWIDX_FRONT] = { AC97_PCM_FRONT_DAC_RATE, AC97_POWERDOWN, AC97_PD_PR1},
+ [PWIDX_CLFE] = { AC97_PCM_LFE_DAC_RATE, AC97_EXTENDED_STATUS,
+ AC97_EA_PRI | AC97_EA_PRK},
+ [PWIDX_SURR] = { AC97_PCM_SURR_DAC_RATE, AC97_EXTENDED_STATUS,
+ AC97_EA_PRJ},
+ [PWIDX_MIC] = { AC97_PCM_MIC_ADC_RATE, AC97_EXTENDED_STATUS,
+ AC97_EA_PRL},
+};
+
+#ifdef CONFIG_SND_AC97_POWER_SAVE
+/**
+ * snd_ac97_update_power - update the powerdown register
+ * @ac97: the codec instance
+ * @reg: the rate register, e.g. AC97_PCM_FRONT_DAC_RATE
+ * @powerup: non-zero when power up the part
+ *
+ * Update the AC97 powerdown register bits of the given part.
+ */
+int snd_ac97_update_power(struct snd_ac97 *ac97, int reg, int powerup)
+{
+ int i;
+
+ if (! ac97)
+ return 0;
+
+ if (reg) {
+ /* SPDIF requires DAC power, too */
+ if (reg == AC97_SPDIF)
+ reg = AC97_PCM_FRONT_DAC_RATE;
+ for (i = 0; i < PWIDX_SIZE; i++) {
+ if (power_regs[i].reg == reg) {
+ if (powerup)
+ ac97->power_up |= (1 << i);
+ else
+ ac97->power_up &= ~(1 << i);
+ break;
+ }
+ }
+ }
+
+ if (! power_save)
+ return 0;
+
+ if (! powerup && ac97->power_workq)
+ /* adjust power-down bits after two seconds delay
+ * (for avoiding loud click noises for many (OSS) apps
+ * that open/close frequently)
+ */
+ queue_delayed_work(ac97->power_workq, &ac97->power_work, HZ*2);
+ else
+ update_power_regs(ac97);
+
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_ac97_update_power);
+#endif /* CONFIG_SND_AC97_POWER_SAVE */
+
+static void update_power_regs(struct snd_ac97 *ac97)
+{
+ unsigned int power_up, bits;
+ int i;
+
+#ifdef CONFIG_SND_AC97_POWER_SAVE
+ if (power_save)
+ power_up = ac97->power_up;
+ else {
#endif
+ power_up = (1 << PWIDX_FRONT) | (1 << PWIDX_ADC);
+ power_up |= (1 << PWIDX_MIC);
+ if (ac97->scaps & AC97_SCAP_SURROUND_DAC)
+ power_up |= (1 << PWIDX_SURR);
+ if (ac97->scaps & AC97_SCAP_CENTER_LFE_DAC)
+ power_up |= (1 << PWIDX_CLFE);
+#ifdef CONFIG_SND_AC97_POWER_SAVE
+ }
+#endif
+ if (power_up) {
+ if (ac97->regs[AC97_POWERDOWN] & AC97_PD_PR2) {
+ /* needs power-up analog mix and vref */
+ snd_ac97_update_bits(ac97, AC97_POWERDOWN,
+ AC97_PD_PR3, 0);
+ msleep(1);
+ snd_ac97_update_bits(ac97, AC97_POWERDOWN,
+ AC97_PD_PR2, 0);
+ }
+ }
+ for (i = 0; i < PWIDX_SIZE; i++) {
+ if (power_up & (1 << i))
+ bits = 0;
+ else
+ bits = power_regs[i].mask;
+ snd_ac97_update_bits(ac97, power_regs[i].power_reg,
+ power_regs[i].mask, bits);
+ }
+ if (! power_up) {
+ if (! (ac97->regs[AC97_POWERDOWN] & AC97_PD_PR2)) {
+ /* power down analog mix and vref */
+ snd_ac97_update_bits(ac97, AC97_POWERDOWN,
+ AC97_PD_PR2, AC97_PD_PR2);
+ snd_ac97_update_bits(ac97, AC97_POWERDOWN,
+ AC97_PD_PR3, AC97_PD_PR3);
+ }
+ }
}
@@ -2484,6 +2719,7 @@ static int tune_mute_led(struct snd_ac97 *ac97)
msw->put = master_mute_sw_put;
snd_ac97_remove_ctl(ac97, "External Amplifier", NULL);
snd_ac97_update_bits(ac97, AC97_POWERDOWN, 0x8000, 0x8000); /* mute LED on */
+ ac97->scaps |= AC97_SCAP_EAPD_LED;
return 0;
}
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 094cfc1f3a1..dc28b111a06 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -32,6 +32,7 @@
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/control.h>
+#include <sound/tlv.h>
#include <sound/ac97_codec.h>
#include "ac97_patch.h"
#include "ac97_id.h"
@@ -51,6 +52,20 @@ static int patch_build_controls(struct snd_ac97 * ac97, const struct snd_kcontro
return 0;
}
+/* replace with a new TLV */
+static void reset_tlv(struct snd_ac97 *ac97, const char *name,
+ unsigned int *tlv)
+{
+ struct snd_ctl_elem_id sid;
+ struct snd_kcontrol *kctl;
+ memset(&sid, 0, sizeof(sid));
+ strcpy(sid.name, name);
+ sid.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ kctl = snd_ctl_find_id(ac97->bus->card, &sid);
+ if (kctl && kctl->tlv.p)
+ kctl->tlv.p = tlv;
+}
+
/* set to the page, update bits and restore the page */
static int ac97_update_bits_page(struct snd_ac97 *ac97, unsigned short reg, unsigned short mask, unsigned short value, unsigned short page)
{
@@ -466,7 +481,7 @@ int patch_wolfson05(struct snd_ac97 * ac97)
ac97->build_ops = &patch_wolfson_wm9705_ops;
#ifdef CONFIG_TOUCHSCREEN_WM9705
/* WM9705 touchscreen uses AUX and VIDEO for touch */
- ac97->flags |=3D AC97_HAS_NO_VIDEO | AC97_HAS_NO_AUX;
+ ac97->flags |= AC97_HAS_NO_VIDEO | AC97_HAS_NO_AUX;
#endif
return 0;
}
@@ -1380,6 +1395,17 @@ static void ad1888_resume(struct snd_ac97 *ac97)
#endif
+static const struct snd_ac97_res_table ad1819_restbl[] = {
+ { AC97_PHONE, 0x9f1f },
+ { AC97_MIC, 0x9f1f },
+ { AC97_LINE, 0x9f1f },
+ { AC97_CD, 0x9f1f },
+ { AC97_VIDEO, 0x9f1f },
+ { AC97_AUX, 0x9f1f },
+ { AC97_PCM, 0x9f1f },
+ { } /* terminator */
+};
+
int patch_ad1819(struct snd_ac97 * ac97)
{
unsigned short scfg;
@@ -1387,6 +1413,7 @@ int patch_ad1819(struct snd_ac97 * ac97)
// patch for Analog Devices
scfg = snd_ac97_read(ac97, AC97_AD_SERIAL_CFG);
snd_ac97_write_cache(ac97, AC97_AD_SERIAL_CFG, scfg | 0x7000); /* select all codecs */
+ ac97->res_table = ad1819_restbl;
return 0;
}
@@ -1522,12 +1549,16 @@ static const struct snd_kcontrol_new snd_ac97_controls_ad1885[] = {
AC97_SINGLE("Line Jack Sense", AC97_AD_JACK_SPDIF, 8, 1, 1), /* inverted */
};
+static DECLARE_TLV_DB_SCALE(db_scale_6bit_6db_max, -8850, 150, 0);
+
static int patch_ad1885_specific(struct snd_ac97 * ac97)
{
int err;
if ((err = patch_build_controls(ac97, snd_ac97_controls_ad1885, ARRAY_SIZE(snd_ac97_controls_ad1885))) < 0)
return err;
+ reset_tlv(ac97, "Headphone Playback Volume",
+ db_scale_6bit_6db_max);
return 0;
}
@@ -1551,12 +1582,27 @@ int patch_ad1885(struct snd_ac97 * ac97)
return 0;
}
+static int patch_ad1886_specific(struct snd_ac97 * ac97)
+{
+ reset_tlv(ac97, "Headphone Playback Volume",
+ db_scale_6bit_6db_max);
+ return 0;
+}
+
+static struct snd_ac97_build_ops patch_ad1886_build_ops = {
+ .build_specific = &patch_ad1886_specific,
+#ifdef CONFIG_PM
+ .resume = ad18xx_resume
+#endif
+};
+
int patch_ad1886(struct snd_ac97 * ac97)
{
patch_ad1881(ac97);
/* Presario700 workaround */
/* for Jack Sense/SPDIF Register misetting causing */
snd_ac97_write_cache(ac97, AC97_AD_JACK_SPDIF, 0x0010);
+ ac97->build_ops = &patch_ad1886_build_ops;
return 0;
}
@@ -2015,6 +2061,8 @@ static const struct snd_kcontrol_new snd_ac97_spdif_controls_alc650[] = {
/* AC97_SINGLE("IEC958 Input Monitor", AC97_ALC650_MULTICH, 13, 1, 0), */
};
+static DECLARE_TLV_DB_SCALE(db_scale_5bit_3db_max, -4350, 150, 0);
+
static int patch_alc650_specific(struct snd_ac97 * ac97)
{
int err;
@@ -2025,6 +2073,9 @@ static int patch_alc650_specific(struct snd_ac97 * ac97)
if ((err = patch_build_controls(ac97, snd_ac97_spdif_controls_alc650, ARRAY_SIZE(snd_ac97_spdif_controls_alc650))) < 0)
return err;
}
+ if (ac97->id != AC97_ID_ALC650F)
+ reset_tlv(ac97, "Master Playback Volume",
+ db_scale_5bit_3db_max);
return 0;
}
@@ -2208,7 +2259,8 @@ int patch_alc655(struct snd_ac97 * ac97)
val &= ~(1 << 1); /* Pin 47 is spdif input pin */
else { /* ALC655 */
if (ac97->subsystem_vendor == 0x1462 &&
- ac97->subsystem_device == 0x0131) /* MSI S270 laptop */
+ (ac97->subsystem_device == 0x0131 || /* MSI S270 laptop */
+ ac97->subsystem_device == 0x0161)) /* LG K1 Express */
val &= ~(1 << 1); /* Pin 47 is EAPD (for internal speaker) */
else
val |= (1 << 1); /* Pin 47 is spdif input pin */
@@ -2759,6 +2811,10 @@ int patch_vt1616(struct snd_ac97 * ac97)
*/
int patch_vt1617a(struct snd_ac97 * ac97)
{
+ /* bring analog power consumption to normal, like WinXP driver
+ * for EPIA SP
+ */
+ snd_ac97_write_cache(ac97, 0x5c, 0x20);
ac97->ext_id |= AC97_EI_SPDIF; /* force the detection of spdif */
ac97->rates[AC97_RATES_SPDIF] = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000;
return 0;
@@ -2872,3 +2928,41 @@ int patch_lm4550(struct snd_ac97 *ac97)
ac97->res_table = lm4550_restbl;
return 0;
}
+
+/*
+ * UCB1400 codec (http://www.semiconductors.philips.com/acrobat_download/datasheets/UCB1400-02.pdf)
+ */
+static const struct snd_kcontrol_new snd_ac97_controls_ucb1400[] = {
+/* enable/disable headphone driver which allows direct connection to
+ stereo headphone without the use of external DC blocking
+ capacitors */
+AC97_SINGLE("Headphone Driver", 0x6a, 6, 1, 0),
+/* Filter used to compensate the DC offset is added in the ADC to remove idle
+ tones from the audio band. */
+AC97_SINGLE("DC Filter", 0x6a, 4, 1, 0),
+/* Control smart-low-power mode feature. Allows automatic power down
+ of unused blocks in the ADC analog front end and the PLL. */
+AC97_SINGLE("Smart Low Power Mode", 0x6c, 4, 3, 0),
+};
+
+static int patch_ucb1400_specific(struct snd_ac97 * ac97)
+{
+ int idx, err;
+ for (idx = 0; idx < ARRAY_SIZE(snd_ac97_controls_ucb1400); idx++)
+ if ((err = snd_ctl_add(ac97->bus->card, snd_ctl_new1(&snd_ac97_controls_ucb1400[idx], ac97))) < 0)
+ return err;
+ return 0;
+}
+
+static struct snd_ac97_build_ops patch_ucb1400_ops = {
+ .build_specific = patch_ucb1400_specific,
+};
+
+int patch_ucb1400(struct snd_ac97 * ac97)
+{
+ ac97->build_ops = &patch_ucb1400_ops;
+ /* enable headphone driver and smart low power mode by default */
+ snd_ac97_write(ac97, 0x6a, 0x0050);
+ snd_ac97_write(ac97, 0x6c, 0x0030);
+ return 0;
+}
diff --git a/sound/pci/ac97/ac97_patch.h b/sound/pci/ac97/ac97_patch.h
index adcaa04586c..74197921720 100644
--- a/sound/pci/ac97/ac97_patch.h
+++ b/sound/pci/ac97/ac97_patch.h
@@ -58,5 +58,6 @@ int patch_cm9780(struct snd_ac97 * ac97);
int patch_vt1616(struct snd_ac97 * ac97);
int patch_vt1617a(struct snd_ac97 * ac97);
int patch_it2646(struct snd_ac97 * ac97);
+int patch_ucb1400(struct snd_ac97 * ac97);
int mpatch_si3036(struct snd_ac97 * ac97);
int patch_lm4550(struct snd_ac97 * ac97);
diff --git a/sound/pci/ac97/ac97_pcm.c b/sound/pci/ac97/ac97_pcm.c
index f684aa2c006..3758d07182f 100644
--- a/sound/pci/ac97/ac97_pcm.c
+++ b/sound/pci/ac97/ac97_pcm.c
@@ -269,6 +269,7 @@ int snd_ac97_set_rate(struct snd_ac97 *ac97, int reg, unsigned int rate)
return -EINVAL;
}
+ snd_ac97_update_power(ac97, reg, 1);
switch (reg) {
case AC97_PCM_MIC_ADC_RATE:
if ((ac97->regs[AC97_EXTENDED_STATUS] & AC97_EA_VRM) == 0) /* MIC VRA */
@@ -606,6 +607,7 @@ int snd_ac97_pcm_open(struct ac97_pcm *pcm, unsigned int rate,
goto error;
}
}
+ pcm->cur_dbl = r;
spin_unlock_irq(&pcm->bus->bus_lock);
for (i = 3; i < 12; i++) {
if (!(slots & (1 << i)))
@@ -651,6 +653,21 @@ int snd_ac97_pcm_close(struct ac97_pcm *pcm)
unsigned short slots = pcm->aslots;
int i, cidx;
+#ifdef CONFIG_SND_AC97_POWER_SAVE
+ int r = pcm->cur_dbl;
+ for (i = 3; i < 12; i++) {
+ if (!(slots & (1 << i)))
+ continue;
+ for (cidx = 0; cidx < 4; cidx++) {
+ if (pcm->r[r].rslots[cidx] & (1 << i)) {
+ int reg = get_slot_reg(pcm, cidx, i, r);
+ snd_ac97_update_power(pcm->r[r].codec[cidx],
+ reg, 0);
+ }
+ }
+ }
+#endif
+
bus = pcm->bus;
spin_lock_irq(&pcm->bus->bus_lock);
for (i = 3; i < 12; i++) {
@@ -660,6 +677,7 @@ int snd_ac97_pcm_close(struct ac97_pcm *pcm)
bus->used_slots[pcm->stream][cidx] &= ~(1 << i);
}
pcm->aslots = 0;
+ pcm->cur_dbl = 0;
spin_unlock_irq(&pcm->bus->bus_lock);
return 0;
}
diff --git a/sound/pci/ac97/ac97_proc.c b/sound/pci/ac97/ac97_proc.c
index 2118df50b9d..a3fdd7da911 100644
--- a/sound/pci/ac97/ac97_proc.c
+++ b/sound/pci/ac97/ac97_proc.c
@@ -457,14 +457,10 @@ void snd_ac97_proc_init(struct snd_ac97 * ac97)
void snd_ac97_proc_done(struct snd_ac97 * ac97)
{
- if (ac97->proc_regs) {
- snd_info_unregister(ac97->proc_regs);
- ac97->proc_regs = NULL;
- }
- if (ac97->proc) {
- snd_info_unregister(ac97->proc);
- ac97->proc = NULL;
- }
+ snd_info_free_entry(ac97->proc_regs);
+ ac97->proc_regs = NULL;
+ snd_info_free_entry(ac97->proc);
+ ac97->proc = NULL;
}
void snd_ac97_bus_proc_init(struct snd_ac97_bus * bus)
@@ -485,8 +481,6 @@ void snd_ac97_bus_proc_init(struct snd_ac97_bus * bus)
void snd_ac97_bus_proc_done(struct snd_ac97_bus * bus)
{
- if (bus->proc) {
- snd_info_unregister(bus->proc);
- bus->proc = NULL;
- }
+ snd_info_free_entry(bus->proc);
+ bus->proc = NULL;
}
diff --git a/sound/pci/ac97/ak4531_codec.c b/sound/pci/ac97/ak4531_codec.c
index 94c26ec0588..c153cb79c51 100644
--- a/sound/pci/ac97/ak4531_codec.c
+++ b/sound/pci/ac97/ak4531_codec.c
@@ -27,6 +27,7 @@
#include <sound/core.h>
#include <sound/ak4531_codec.h>
+#include <sound/tlv.h>
MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
MODULE_DESCRIPTION("Universal routines for AK4531 codec");
@@ -63,6 +64,14 @@ static void snd_ak4531_dump(struct snd_ak4531 *ak4531)
.info = snd_ak4531_info_single, \
.get = snd_ak4531_get_single, .put = snd_ak4531_put_single, \
.private_value = reg | (shift << 16) | (mask << 24) | (invert << 22) }
+#define AK4531_SINGLE_TLV(xname, xindex, reg, shift, mask, invert, xtlv) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \
+ .name = xname, .index = xindex, \
+ .info = snd_ak4531_info_single, \
+ .get = snd_ak4531_get_single, .put = snd_ak4531_put_single, \
+ .private_value = reg | (shift << 16) | (mask << 24) | (invert << 22), \
+ .tlv = { .p = (xtlv) } }
static int snd_ak4531_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
{
@@ -122,6 +131,14 @@ static int snd_ak4531_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_e
.info = snd_ak4531_info_double, \
.get = snd_ak4531_get_double, .put = snd_ak4531_put_double, \
.private_value = left_reg | (right_reg << 8) | (left_shift << 16) | (right_shift << 19) | (mask << 24) | (invert << 22) }
+#define AK4531_DOUBLE_TLV(xname, xindex, left_reg, right_reg, left_shift, right_shift, mask, invert, xtlv) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \
+ .name = xname, .index = xindex, \
+ .info = snd_ak4531_info_double, \
+ .get = snd_ak4531_get_double, .put = snd_ak4531_put_double, \
+ .private_value = left_reg | (right_reg << 8) | (left_shift << 16) | (right_shift << 19) | (mask << 24) | (invert << 22), \
+ .tlv = { .p = (xtlv) } }
static int snd_ak4531_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
{
@@ -250,50 +267,62 @@ static int snd_ak4531_put_input_sw(struct snd_kcontrol *kcontrol, struct snd_ctl
return change;
}
+static DECLARE_TLV_DB_SCALE(db_scale_master, -6200, 200, 0);
+static DECLARE_TLV_DB_SCALE(db_scale_mono, -2800, 400, 0);
+static DECLARE_TLV_DB_SCALE(db_scale_input, -5000, 200, 0);
+
static struct snd_kcontrol_new snd_ak4531_controls[] = {
-AK4531_DOUBLE("Master Playback Switch", 0, AK4531_LMASTER, AK4531_RMASTER, 7, 7, 1, 1),
+AK4531_DOUBLE_TLV("Master Playback Switch", 0,
+ AK4531_LMASTER, AK4531_RMASTER, 7, 7, 1, 1,
+ db_scale_master),
AK4531_DOUBLE("Master Playback Volume", 0, AK4531_LMASTER, AK4531_RMASTER, 0, 0, 0x1f, 1),
-AK4531_SINGLE("Master Mono Playback Switch", 0, AK4531_MONO_OUT, 7, 1, 1),
+AK4531_SINGLE_TLV("Master Mono Playback Switch", 0, AK4531_MONO_OUT, 7, 1, 1,
+ db_scale_mono),
AK4531_SINGLE("Master Mono Playback Volume", 0, AK4531_MONO_OUT, 0, 0x07, 1),
AK4531_DOUBLE("PCM Switch", 0, AK4531_LVOICE, AK4531_RVOICE, 7, 7, 1, 1),
-AK4531_DOUBLE("PCM Volume", 0, AK4531_LVOICE, AK4531_RVOICE, 0, 0, 0x1f, 1),
+AK4531_DOUBLE_TLV("PCM Volume", 0, AK4531_LVOICE, AK4531_RVOICE, 0, 0, 0x1f, 1,
+ db_scale_input),
AK4531_DOUBLE("PCM Playback Switch", 0, AK4531_OUT_SW2, AK4531_OUT_SW2, 3, 2, 1, 0),
AK4531_DOUBLE("PCM Capture Switch", 0, AK4531_LIN_SW2, AK4531_RIN_SW2, 2, 2, 1, 0),
AK4531_DOUBLE("PCM Switch", 1, AK4531_LFM, AK4531_RFM, 7, 7, 1, 1),
-AK4531_DOUBLE("PCM Volume", 1, AK4531_LFM, AK4531_RFM, 0, 0, 0x1f, 1),
+AK4531_DOUBLE_TLV("PCM Volume", 1, AK4531_LFM, AK4531_RFM, 0, 0, 0x1f, 1,
+ db_scale_input),
AK4531_DOUBLE("PCM Playback Switch", 1, AK4531_OUT_SW1, AK4531_OUT_SW1, 6, 5, 1, 0),
AK4531_INPUT_SW("PCM Capture Route", 1, AK4531_LIN_SW1, AK4531_RIN_SW1, 6, 5),
AK4531_DOUBLE("CD Switch", 0, AK4531_LCD, AK4531_RCD, 7, 7, 1, 1),
-AK4531_DOUBLE("CD Volume", 0, AK4531_LCD, AK4531_RCD, 0, 0, 0x1f, 1),
+AK4531_DOUBLE_TLV("CD Volume", 0, AK4531_LCD, AK4531_RCD, 0, 0, 0x1f, 1,
+ db_scale_input),
AK4531_DOUBLE("CD Playback Switch", 0, AK4531_OUT_SW1, AK4531_OUT_SW1, 2, 1, 1, 0),
AK4531_INPUT_SW("CD Capture Route", 0, AK4531_LIN_SW1, AK4531_RIN_SW1, 2, 1),
AK4531_DOUBLE("Line Switch", 0, AK4531_LLINE, AK4531_RLINE, 7, 7, 1, 1),
-AK4531_DOUBLE("Line Volume", 0, AK4531_LLINE, AK4531_RLINE, 0, 0, 0x1f, 1),
+AK4531_DOUBLE_TLV("Line Volume", 0, AK4531_LLINE, AK4531_RLINE, 0, 0, 0x1f, 1,
+ db_scale_input),
AK4531_DOUBLE("Line Playback Switch", 0, AK4531_OUT_SW1, AK4531_OUT_SW1, 4, 3, 1, 0),
AK4531_INPUT_SW("Line Capture Route", 0, AK4531_LIN_SW1, AK4531_RIN_SW1, 4, 3),
AK4531_DOUBLE("Aux Switch", 0, AK4531_LAUXA, AK4531_RAUXA, 7, 7, 1, 1),
-AK4531_DOUBLE("Aux Volume", 0, AK4531_LAUXA, AK4531_RAUXA, 0, 0, 0x1f, 1),
+AK4531_DOUBLE_TLV("Aux Volume", 0, AK4531_LAUXA, AK4531_RAUXA, 0, 0, 0x1f, 1,
+ db_scale_input),
AK4531_DOUBLE("Aux Playback Switch", 0, AK4531_OUT_SW2, AK4531_OUT_SW2, 5, 4, 1, 0),
AK4531_INPUT_SW("Aux Capture Route", 0, AK4531_LIN_SW2, AK4531_RIN_SW2, 4, 3),
AK4531_SINGLE("Mono Switch", 0, AK4531_MONO1, 7, 1, 1),
-AK4531_SINGLE("Mono Volume", 0, AK4531_MONO1, 0, 0x1f, 1),
+AK4531_SINGLE_TLV("Mono Volume", 0, AK4531_MONO1, 0, 0x1f, 1, db_scale_input),
AK4531_SINGLE("Mono Playback Switch", 0, AK4531_OUT_SW2, 0, 1, 0),
AK4531_DOUBLE("Mono Capture Switch", 0, AK4531_LIN_SW2, AK4531_RIN_SW2, 0, 0, 1, 0),
AK4531_SINGLE("Mono Switch", 1, AK4531_MONO2, 7, 1, 1),
-AK4531_SINGLE("Mono Volume", 1, AK4531_MONO2, 0, 0x1f, 1),
+AK4531_SINGLE_TLV("Mono Volume", 1, AK4531_MONO2, 0, 0x1f, 1, db_scale_input),
AK4531_SINGLE("Mono Playback Switch", 1, AK4531_OUT_SW2, 1, 1, 0),
AK4531_DOUBLE("Mono Capture Switch", 1, AK4531_LIN_SW2, AK4531_RIN_SW2, 1, 1, 1, 0),
-AK4531_SINGLE("Mic Volume", 0, AK4531_MIC, 0, 0x1f, 1),
+AK4531_SINGLE_TLV("Mic Volume", 0, AK4531_MIC, 0, 0x1f, 1, db_scale_input),
AK4531_SINGLE("Mic Switch", 0, AK4531_MIC, 7, 1, 1),
AK4531_SINGLE("Mic Playback Switch", 0, AK4531_OUT_SW1, 0, 1, 0),
AK4531_DOUBLE("Mic Capture Switch", 0, AK4531_LIN_SW1, AK4531_RIN_SW1, 0, 0, 1, 0),
diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c
index 146eed70dce..9855f528ea7 100644
--- a/sound/pci/ca0106/ca0106_mixer.c
+++ b/sound/pci/ca0106/ca0106_mixer.c
@@ -70,9 +70,13 @@
#include <sound/pcm.h>
#include <sound/ac97_codec.h>
#include <sound/info.h>
+#include <sound/tlv.h>
#include "ca0106.h"
+static DECLARE_TLV_DB_SCALE(snd_ca0106_db_scale1, -5175, 25, 1);
+static DECLARE_TLV_DB_SCALE(snd_ca0106_db_scale2, -10350, 50, 1);
+
static int snd_ca0106_shared_spdif_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
@@ -469,18 +473,24 @@ static int snd_ca0106_i2c_volume_put(struct snd_kcontrol *kcontrol,
#define CA_VOLUME(xname,chid,reg) \
{ \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ, \
.info = snd_ca0106_volume_info, \
.get = snd_ca0106_volume_get, \
.put = snd_ca0106_volume_put, \
+ .tlv = { .p = snd_ca0106_db_scale1 }, \
.private_value = ((chid) << 8) | (reg) \
}
#define I2C_VOLUME(xname,chid) \
{ \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ, \
.info = snd_ca0106_i2c_volume_info, \
.get = snd_ca0106_i2c_volume_get, \
.put = snd_ca0106_i2c_volume_put, \
+ .tlv = { .p = snd_ca0106_db_scale2 }, \
.private_value = chid \
}
diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c
index 9631456ec3d..1990430a21c 100644
--- a/sound/pci/cs4281.c
+++ b/sound/pci/cs4281.c
@@ -33,6 +33,7 @@
#include <sound/pcm.h>
#include <sound/rawmidi.h>
#include <sound/ac97_codec.h>
+#include <sound/tlv.h>
#include <sound/opl3.h>
#include <sound/initval.h>
@@ -1054,6 +1055,8 @@ static int snd_cs4281_put_volume(struct snd_kcontrol *kcontrol,
return change;
}
+static DECLARE_TLV_DB_SCALE(db_scale_dsp, -4650, 150, 0);
+
static struct snd_kcontrol_new snd_cs4281_fm_vol =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1062,6 +1065,7 @@ static struct snd_kcontrol_new snd_cs4281_fm_vol =
.get = snd_cs4281_get_volume,
.put = snd_cs4281_put_volume,
.private_value = ((BA0_FMLVC << 16) | BA0_FMRVC),
+ .tlv = { .p = db_scale_dsp },
};
static struct snd_kcontrol_new snd_cs4281_pcm_vol =
@@ -1072,6 +1076,7 @@ static struct snd_kcontrol_new snd_cs4281_pcm_vol =
.get = snd_cs4281_get_volume,
.put = snd_cs4281_put_volume,
.private_value = ((BA0_PPLVC << 16) | BA0_PPRVC),
+ .tlv = { .p = db_scale_dsp },
};
static void snd_cs4281_mixer_free_ac97_bus(struct snd_ac97_bus *bus)
diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c
index 5c9711c0265..89c402770a1 100644
--- a/sound/pci/cs46xx/dsp_spos.c
+++ b/sound/pci/cs46xx/dsp_spos.c
@@ -868,35 +868,23 @@ int cs46xx_dsp_proc_done (struct snd_cs46xx *chip)
struct dsp_spos_instance * ins = chip->dsp_spos_instance;
int i;
- if (ins->proc_sym_info_entry) {
- snd_info_unregister(ins->proc_sym_info_entry);
- ins->proc_sym_info_entry = NULL;
- }
-
- if (ins->proc_modules_info_entry) {
- snd_info_unregister(ins->proc_modules_info_entry);
- ins->proc_modules_info_entry = NULL;
- }
-
- if (ins->proc_parameter_dump_info_entry) {
- snd_info_unregister(ins->proc_parameter_dump_info_entry);
- ins->proc_parameter_dump_info_entry = NULL;
- }
-
- if (ins->proc_sample_dump_info_entry) {
- snd_info_unregister(ins->proc_sample_dump_info_entry);
- ins->proc_sample_dump_info_entry = NULL;
- }
-
- if (ins->proc_scb_info_entry) {
- snd_info_unregister(ins->proc_scb_info_entry);
- ins->proc_scb_info_entry = NULL;
- }
-
- if (ins->proc_task_info_entry) {
- snd_info_unregister(ins->proc_task_info_entry);
- ins->proc_task_info_entry = NULL;
- }
+ snd_info_free_entry(ins->proc_sym_info_entry);
+ ins->proc_sym_info_entry = NULL;
+
+ snd_info_free_entry(ins->proc_modules_info_entry);
+ ins->proc_modules_info_entry = NULL;
+
+ snd_info_free_entry(ins->proc_parameter_dump_info_entry);
+ ins->proc_parameter_dump_info_entry = NULL;
+
+ snd_info_free_entry(ins->proc_sample_dump_info_entry);
+ ins->proc_sample_dump_info_entry = NULL;
+
+ snd_info_free_entry(ins->proc_scb_info_entry);
+ ins->proc_scb_info_entry = NULL;
+
+ snd_info_free_entry(ins->proc_task_info_entry);
+ ins->proc_task_info_entry = NULL;
mutex_lock(&chip->spos_mutex);
for (i = 0; i < ins->nscb; ++i) {
@@ -905,10 +893,8 @@ int cs46xx_dsp_proc_done (struct snd_cs46xx *chip)
}
mutex_unlock(&chip->spos_mutex);
- if (ins->proc_dsp_dir) {
- snd_info_unregister (ins->proc_dsp_dir);
- ins->proc_dsp_dir = NULL;
- }
+ snd_info_free_entry(ins->proc_dsp_dir);
+ ins->proc_dsp_dir = NULL;
return 0;
}
diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c
index 232b337852f..343f51d5311 100644
--- a/sound/pci/cs46xx/dsp_spos_scb_lib.c
+++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c
@@ -233,7 +233,7 @@ void cs46xx_dsp_proc_free_scb_desc (struct dsp_scb_descriptor * scb)
snd_printdd("cs46xx_dsp_proc_free_scb_desc: freeing %s\n",scb->scb_name);
- snd_info_unregister(scb->proc_info);
+ snd_info_free_entry(scb->proc_info);
scb->proc_info = NULL;
snd_assert (scb_info != NULL, return);
diff --git a/sound/pci/cs5535audio/Makefile b/sound/pci/cs5535audio/Makefile
index 2911a8adc1f..ad947b4c04c 100644
--- a/sound/pci/cs5535audio/Makefile
+++ b/sound/pci/cs5535audio/Makefile
@@ -4,7 +4,7 @@
snd-cs5535audio-objs := cs5535audio.o cs5535audio_pcm.o
-ifdef CONFIG_PM
+ifeq ($(CONFIG_PM),y)
snd-cs5535audio-objs += cs5535audio_pm.o
endif
diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c
index 289bcd99c19..493ec0816bb 100644
--- a/sound/pci/emu10k1/emu10k1.c
+++ b/sound/pci/emu10k1/emu10k1.c
@@ -232,7 +232,7 @@ static int snd_emu10k1_suspend(struct pci_dev *pci, pm_message_t state)
return 0;
}
-int snd_emu10k1_resume(struct pci_dev *pci)
+static int snd_emu10k1_resume(struct pci_dev *pci)
{
struct snd_card *card = pci_get_drvdata(pci);
struct snd_emu10k1 *emu = card->private_data;
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 79f24cdf5fb..be65d4db8e2 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -927,6 +927,7 @@ static struct snd_emu_chip_details emu_chip_details[] = {
.ca0151_chip = 1,
.spk71 = 1,
.spdif_bug = 1,
+ .adc_1361t = 1, /* 24 bit capture instead of 16bit */
.ac97_chip = 1} ,
{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10051102,
.driver = "Audigy2", .name = "Audigy 2 EX [1005]",
diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c
index bda8bdf5993..da1610a571b 100644
--- a/sound/pci/emu10k1/emu10k1x.c
+++ b/sound/pci/emu10k1/emu10k1x.c
@@ -1626,12 +1626,7 @@ static struct pci_driver driver = {
// initialization of the module
static int __init alsa_card_emu10k1x_init(void)
{
- int err;
-
- if ((err = pci_register_driver(&driver)) > 0)
- return err;
-
- return 0;
+ return pci_register_driver(&driver);
}
// clean up the module
diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c
index dfba00230d4..13cd6ce8981 100644
--- a/sound/pci/emu10k1/emufx.c
+++ b/sound/pci/emu10k1/emufx.c
@@ -35,6 +35,7 @@
#include <linux/mutex.h>
#include <sound/core.h>
+#include <sound/tlv.h>
#include <sound/emu10k1.h>
#if 0 /* for testing purposes - digital out -> capture */
@@ -266,6 +267,7 @@ static const u32 treble_table[41][5] = {
{ 0x37c4448b, 0xa45ef51d, 0x262f3267, 0x081e36dc, 0xfd8f5d14 }
};
+/* dB gain = (float) 20 * log10( float(db_table_value) / 0x8000000 ) */
static const u32 db_table[101] = {
0x00000000, 0x01571f82, 0x01674b41, 0x01783a1b, 0x0189f540,
0x019c8651, 0x01aff763, 0x01c45306, 0x01d9a446, 0x01eff6b8,
@@ -290,6 +292,9 @@ static const u32 db_table[101] = {
0x7fffffff,
};
+/* EMU10k1/EMU10k2 DSP control db gain */
+static DECLARE_TLV_DB_SCALE(snd_emu10k1_db_scale1, -4000, 40, 1);
+
static const u32 onoff_table[2] = {
0x00000000, 0x00000001
};
@@ -755,6 +760,11 @@ static int snd_emu10k1_add_controls(struct snd_emu10k1 *emu,
knew.device = gctl->id.device;
knew.subdevice = gctl->id.subdevice;
knew.info = snd_emu10k1_gpr_ctl_info;
+ if (gctl->tlv.p) {
+ knew.tlv.p = gctl->tlv.p;
+ knew.access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ;
+ }
knew.get = snd_emu10k1_gpr_ctl_get;
knew.put = snd_emu10k1_gpr_ctl_put;
memset(nctl, 0, sizeof(*nctl));
@@ -1013,6 +1023,7 @@ snd_emu10k1_init_mono_control(struct snd_emu10k1_fx8010_control_gpr *ctl,
ctl->gpr[0] = gpr + 0; ctl->value[0] = defval;
ctl->min = 0;
ctl->max = 100;
+ ctl->tlv.p = snd_emu10k1_db_scale1;
ctl->translation = EMU10K1_GPR_TRANSLATION_TABLE100;
}
@@ -1027,6 +1038,7 @@ snd_emu10k1_init_stereo_control(struct snd_emu10k1_fx8010_control_gpr *ctl,
ctl->gpr[1] = gpr + 1; ctl->value[1] = defval;
ctl->min = 0;
ctl->max = 100;
+ ctl->tlv.p = snd_emu10k1_db_scale1;
ctl->translation = EMU10K1_GPR_TRANSLATION_TABLE100;
}
diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c
index 9905651935f..4e0f95438f4 100644
--- a/sound/pci/emu10k1/p16v.c
+++ b/sound/pci/emu10k1/p16v.c
@@ -100,6 +100,7 @@
#include <sound/pcm.h>
#include <sound/ac97_codec.h>
#include <sound/info.h>
+#include <sound/tlv.h>
#include <sound/emu10k1.h>
#include "p16v.h"
@@ -784,12 +785,16 @@ static int snd_p16v_capture_channel_put(struct snd_kcontrol *kcontrol,
}
return change;
}
+static DECLARE_TLV_DB_SCALE(snd_p16v_db_scale1, -5175, 25, 1);
#define P16V_VOL(xname,xreg,xhl) { \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ, \
.info = snd_p16v_volume_info, \
.get = snd_p16v_volume_get, \
.put = snd_p16v_volume_put, \
+ .tlv = { .p = snd_p16v_db_scale1 }, \
.private_value = ((xreg) | ((xhl) << 8)) \
}
diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c
index cc0f34f6818..3ce5a4e7e31 100644
--- a/sound/pci/es1938.c
+++ b/sound/pci/es1938.c
@@ -62,6 +62,7 @@
#include <sound/opl3.h>
#include <sound/mpu401.h>
#include <sound/initval.h>
+#include <sound/tlv.h>
#include <asm/io.h>
@@ -1164,6 +1165,14 @@ static int snd_es1938_reg_read(struct es1938 *chip, unsigned char reg)
return snd_es1938_read(chip, reg);
}
+#define ES1938_SINGLE_TLV(xname, xindex, reg, shift, mask, invert, xtlv) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ,\
+ .name = xname, .index = xindex, \
+ .info = snd_es1938_info_single, \
+ .get = snd_es1938_get_single, .put = snd_es1938_put_single, \
+ .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24), \
+ .tlv = { .p = xtlv } }
#define ES1938_SINGLE(xname, xindex, reg, shift, mask, invert) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
.info = snd_es1938_info_single, \
@@ -1217,6 +1226,14 @@ static int snd_es1938_put_single(struct snd_kcontrol *kcontrol,
return snd_es1938_reg_bits(chip, reg, mask, val) != val;
}
+#define ES1938_DOUBLE_TLV(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert, xtlv) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ,\
+ .name = xname, .index = xindex, \
+ .info = snd_es1938_info_double, \
+ .get = snd_es1938_get_double, .put = snd_es1938_put_double, \
+ .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22), \
+ .tlv = { .p = xtlv } }
#define ES1938_DOUBLE(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
.info = snd_es1938_info_double, \
@@ -1297,8 +1314,41 @@ static int snd_es1938_put_double(struct snd_kcontrol *kcontrol,
return change;
}
+static unsigned int db_scale_master[] = {
+ TLV_DB_RANGE_HEAD(2),
+ 0, 54, TLV_DB_SCALE_ITEM(-3600, 50, 1),
+ 54, 63, TLV_DB_SCALE_ITEM(-900, 100, 0),
+};
+
+static unsigned int db_scale_audio1[] = {
+ TLV_DB_RANGE_HEAD(2),
+ 0, 8, TLV_DB_SCALE_ITEM(-3300, 300, 1),
+ 8, 15, TLV_DB_SCALE_ITEM(-900, 150, 0),
+};
+
+static unsigned int db_scale_audio2[] = {
+ TLV_DB_RANGE_HEAD(2),
+ 0, 8, TLV_DB_SCALE_ITEM(-3450, 300, 1),
+ 8, 15, TLV_DB_SCALE_ITEM(-1050, 150, 0),
+};
+
+static unsigned int db_scale_mic[] = {
+ TLV_DB_RANGE_HEAD(2),
+ 0, 8, TLV_DB_SCALE_ITEM(-2400, 300, 1),
+ 8, 15, TLV_DB_SCALE_ITEM(0, 150, 0),
+};
+
+static unsigned int db_scale_line[] = {
+ TLV_DB_RANGE_HEAD(2),
+ 0, 8, TLV_DB_SCALE_ITEM(-3150, 300, 1),
+ 8, 15, TLV_DB_SCALE_ITEM(-750, 150, 0),
+};
+
+static DECLARE_TLV_DB_SCALE(db_scale_capture, 0, 150, 0);
+
static struct snd_kcontrol_new snd_es1938_controls[] = {
-ES1938_DOUBLE("Master Playback Volume", 0, 0x60, 0x62, 0, 0, 63, 0),
+ES1938_DOUBLE_TLV("Master Playback Volume", 0, 0x60, 0x62, 0, 0, 63, 0,
+ db_scale_master),
ES1938_DOUBLE("Master Playback Switch", 0, 0x60, 0x62, 6, 6, 1, 1),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1309,19 +1359,27 @@ ES1938_DOUBLE("Master Playback Switch", 0, 0x60, 0x62, 6, 6, 1, 1),
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READ |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "Hardware Master Playback Switch",
- .access = SNDRV_CTL_ELEM_ACCESS_READ,
.info = snd_es1938_info_hw_switch,
.get = snd_es1938_get_hw_switch,
+ .tlv = { .p = db_scale_master },
},
ES1938_SINGLE("Hardware Volume Split", 0, 0x64, 7, 1, 0),
-ES1938_DOUBLE("Line Playback Volume", 0, 0x3e, 0x3e, 4, 0, 15, 0),
+ES1938_DOUBLE_TLV("Line Playback Volume", 0, 0x3e, 0x3e, 4, 0, 15, 0,
+ db_scale_line),
ES1938_DOUBLE("CD Playback Volume", 0, 0x38, 0x38, 4, 0, 15, 0),
-ES1938_DOUBLE("FM Playback Volume", 0, 0x36, 0x36, 4, 0, 15, 0),
-ES1938_DOUBLE("Mono Playback Volume", 0, 0x6d, 0x6d, 4, 0, 15, 0),
-ES1938_DOUBLE("Mic Playback Volume", 0, 0x1a, 0x1a, 4, 0, 15, 0),
-ES1938_DOUBLE("Aux Playback Volume", 0, 0x3a, 0x3a, 4, 0, 15, 0),
-ES1938_DOUBLE("Capture Volume", 0, 0xb4, 0xb4, 4, 0, 15, 0),
+ES1938_DOUBLE_TLV("FM Playback Volume", 0, 0x36, 0x36, 4, 0, 15, 0,
+ db_scale_mic),
+ES1938_DOUBLE_TLV("Mono Playback Volume", 0, 0x6d, 0x6d, 4, 0, 15, 0,
+ db_scale_line),
+ES1938_DOUBLE_TLV("Mic Playback Volume", 0, 0x1a, 0x1a, 4, 0, 15, 0,
+ db_scale_mic),
+ES1938_DOUBLE_TLV("Aux Playback Volume", 0, 0x3a, 0x3a, 4, 0, 15, 0,
+ db_scale_line),
+ES1938_DOUBLE_TLV("Capture Volume", 0, 0xb4, 0xb4, 4, 0, 15, 0,
+ db_scale_capture),
ES1938_SINGLE("PC Speaker Volume", 0, 0x3c, 0, 7, 0),
ES1938_SINGLE("Record Monitor", 0, 0xa8, 3, 1, 0),
ES1938_SINGLE("Capture Switch", 0, 0x1c, 4, 1, 1),
@@ -1332,16 +1390,26 @@ ES1938_SINGLE("Capture Switch", 0, 0x1c, 4, 1, 1),
.get = snd_es1938_get_mux,
.put = snd_es1938_put_mux,
},
-ES1938_DOUBLE("Mono Input Playback Volume", 0, 0x6d, 0x6d, 4, 0, 15, 0),
-ES1938_DOUBLE("PCM Capture Volume", 0, 0x69, 0x69, 4, 0, 15, 0),
-ES1938_DOUBLE("Mic Capture Volume", 0, 0x68, 0x68, 4, 0, 15, 0),
-ES1938_DOUBLE("Line Capture Volume", 0, 0x6e, 0x6e, 4, 0, 15, 0),
-ES1938_DOUBLE("FM Capture Volume", 0, 0x6b, 0x6b, 4, 0, 15, 0),
-ES1938_DOUBLE("Mono Capture Volume", 0, 0x6f, 0x6f, 4, 0, 15, 0),
-ES1938_DOUBLE("CD Capture Volume", 0, 0x6a, 0x6a, 4, 0, 15, 0),
-ES1938_DOUBLE("Aux Capture Volume", 0, 0x6c, 0x6c, 4, 0, 15, 0),
-ES1938_DOUBLE("PCM Playback Volume", 0, 0x7c, 0x7c, 4, 0, 15, 0),
-ES1938_DOUBLE("PCM Playback Volume", 1, 0x14, 0x14, 4, 0, 15, 0),
+ES1938_DOUBLE_TLV("Mono Input Playback Volume", 0, 0x6d, 0x6d, 4, 0, 15, 0,
+ db_scale_line),
+ES1938_DOUBLE_TLV("PCM Capture Volume", 0, 0x69, 0x69, 4, 0, 15, 0,
+ db_scale_audio2),
+ES1938_DOUBLE_TLV("Mic Capture Volume", 0, 0x68, 0x68, 4, 0, 15, 0,
+ db_scale_mic),
+ES1938_DOUBLE_TLV("Line Capture Volume", 0, 0x6e, 0x6e, 4, 0, 15, 0,
+ db_scale_line),
+ES1938_DOUBLE_TLV("FM Capture Volume", 0, 0x6b, 0x6b, 4, 0, 15, 0,
+ db_scale_mic),
+ES1938_DOUBLE_TLV("Mono Capture Volume", 0, 0x6f, 0x6f, 4, 0, 15, 0,
+ db_scale_line),
+ES1938_DOUBLE_TLV("CD Capture Volume", 0, 0x6a, 0x6a, 4, 0, 15, 0,
+ db_scale_line),
+ES1938_DOUBLE_TLV("Aux Capture Volume", 0, 0x6c, 0x6c, 4, 0, 15, 0,
+ db_scale_line),
+ES1938_DOUBLE_TLV("PCM Playback Volume", 0, 0x7c, 0x7c, 4, 0, 15, 0,
+ db_scale_audio2),
+ES1938_DOUBLE_TLV("PCM Playback Volume", 1, 0x14, 0x14, 4, 0, 15, 0,
+ db_scale_audio1),
ES1938_SINGLE("3D Control - Level", 0, 0x52, 0, 63, 0),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c
index 3c5ab7c2e72..f3c40385c87 100644
--- a/sound/pci/es1968.c
+++ b/sound/pci/es1968.c
@@ -1905,7 +1905,7 @@ static void es1968_update_hw_volume(unsigned long private_data)
/* Figure out which volume control button was pushed,
based on differences from the default register
values. */
- x = inb(chip->io_port + 0x1c);
+ x = inb(chip->io_port + 0x1c) & 0xee;
/* Reset the volume control registers. */
outb(0x88, chip->io_port + 0x1c);
outb(0x88, chip->io_port + 0x1d);
@@ -1921,7 +1921,8 @@ static void es1968_update_hw_volume(unsigned long private_data)
/* FIXME: we can't call snd_ac97_* functions since here is in tasklet. */
spin_lock_irqsave(&chip->ac97_lock, flags);
val = chip->ac97->regs[AC97_MASTER];
- if (x & 1) {
+ switch (x) {
+ case 0x88:
/* mute */
val ^= 0x8000;
chip->ac97->regs[AC97_MASTER] = val;
@@ -1929,26 +1930,31 @@ static void es1968_update_hw_volume(unsigned long private_data)
outb(AC97_MASTER, chip->io_port + ESM_AC97_INDEX);
snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
&chip->master_switch->id);
- } else {
- val &= 0x7fff;
- if (((x>>1) & 7) > 4) {
- /* volume up */
- if ((val & 0xff) > 0)
- val--;
- if ((val & 0xff00) > 0)
- val -= 0x0100;
- } else {
- /* volume down */
- if ((val & 0xff) < 0x1f)
- val++;
- if ((val & 0xff00) < 0x1f00)
- val += 0x0100;
- }
+ break;
+ case 0xaa:
+ /* volume up */
+ if ((val & 0x7f) > 0)
+ val--;
+ if ((val & 0x7f00) > 0)
+ val -= 0x0100;
+ chip->ac97->regs[AC97_MASTER] = val;
+ outw(val, chip->io_port + ESM_AC97_DATA);
+ outb(AC97_MASTER, chip->io_port + ESM_AC97_INDEX);
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->master_volume->id);
+ break;
+ case 0x66:
+ /* volume down */
+ if ((val & 0x7f) < 0x1f)
+ val++;
+ if ((val & 0x7f00) < 0x1f00)
+ val += 0x0100;
chip->ac97->regs[AC97_MASTER] = val;
outw(val, chip->io_port + ESM_AC97_DATA);
outb(AC97_MASTER, chip->io_port + ESM_AC97_INDEX);
snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
&chip->master_volume->id);
+ break;
}
spin_unlock_irqrestore(&chip->ac97_lock, flags);
}
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index 13868c98512..bdfda1997d5 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -2,6 +2,7 @@
* The driver for the ForteMedia FM801 based soundcards
* Copyright (c) by Jaroslav Kysela <perex@suse.cz>
*
+ * Support FM only card by Andy Shevchenko <andy@smile.org.ua>
*
* 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
@@ -28,6 +29,7 @@
#include <linux/moduleparam.h>
#include <sound/core.h>
#include <sound/pcm.h>
+#include <sound/tlv.h>
#include <sound/ac97_codec.h>
#include <sound/mpu401.h>
#include <sound/opl3.h>
@@ -54,6 +56,7 @@ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card *
* 1 = MediaForte 256-PCS
* 2 = MediaForte 256-PCPR
* 3 = MediaForte 64-PCR
+ * 16 = setup tuner only (this is additional bit), i.e. SF-64-PCR FM card
* High 16-bits are video (radio) device number + 1
*/
static int tea575x_tuner[SNDRV_CARDS];
@@ -158,6 +161,7 @@ struct fm801 {
unsigned int multichannel: 1, /* multichannel support */
secondary: 1; /* secondary codec */
unsigned char secondary_addr; /* address of the secondary codec */
+ unsigned int tea575x_tuner; /* tuner flags */
unsigned short ply_ctrl; /* playback control */
unsigned short cap_ctrl; /* capture control */
@@ -318,10 +322,8 @@ static unsigned int channels[] = {
2, 4, 6
};
-#define CHANNELS sizeof(channels) / sizeof(channels[0])
-
static struct snd_pcm_hw_constraint_list hw_constraints_channels = {
- .count = CHANNELS,
+ .count = ARRAY_SIZE(channels),
.list = channels,
.mask = 0,
};
@@ -1052,6 +1054,13 @@ static int snd_fm801_put_single(struct snd_kcontrol *kcontrol,
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .info = snd_fm801_info_double, \
.get = snd_fm801_get_double, .put = snd_fm801_put_double, \
.private_value = reg | (shift_left << 8) | (shift_right << 12) | (mask << 16) | (invert << 24) }
+#define FM801_DOUBLE_TLV(xname, reg, shift_left, shift_right, mask, invert, xtlv) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \
+ .name = xname, .info = snd_fm801_info_double, \
+ .get = snd_fm801_get_double, .put = snd_fm801_put_double, \
+ .private_value = reg | (shift_left << 8) | (shift_right << 12) | (mask << 16) | (invert << 24), \
+ .tlv = { .p = (xtlv) } }
static int snd_fm801_info_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
@@ -1148,14 +1157,19 @@ static int snd_fm801_put_mux(struct snd_kcontrol *kcontrol,
return snd_fm801_update_bits(chip, FM801_REC_SRC, 7, val);
}
+static DECLARE_TLV_DB_SCALE(db_scale_dsp, -3450, 150, 0);
+
#define FM801_CONTROLS ARRAY_SIZE(snd_fm801_controls)
static struct snd_kcontrol_new snd_fm801_controls[] __devinitdata = {
-FM801_DOUBLE("Wave Playback Volume", FM801_PCM_VOL, 0, 8, 31, 1),
+FM801_DOUBLE_TLV("Wave Playback Volume", FM801_PCM_VOL, 0, 8, 31, 1,
+ db_scale_dsp),
FM801_SINGLE("Wave Playback Switch", FM801_PCM_VOL, 15, 1, 1),
-FM801_DOUBLE("I2S Playback Volume", FM801_I2S_VOL, 0, 8, 31, 1),
+FM801_DOUBLE_TLV("I2S Playback Volume", FM801_I2S_VOL, 0, 8, 31, 1,
+ db_scale_dsp),
FM801_SINGLE("I2S Playback Switch", FM801_I2S_VOL, 15, 1, 1),
-FM801_DOUBLE("FM Playback Volume", FM801_FM_VOL, 0, 8, 31, 1),
+FM801_DOUBLE_TLV("FM Playback Volume", FM801_FM_VOL, 0, 8, 31, 1,
+ db_scale_dsp),
FM801_SINGLE("FM Playback Switch", FM801_FM_VOL, 15, 1, 1),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1253,6 +1267,9 @@ static int snd_fm801_chip_init(struct fm801 *chip, int resume)
int id;
unsigned short cmdw;
+ if (chip->tea575x_tuner & 0x0010)
+ goto __ac97_ok;
+
/* codec cold reset + AC'97 warm reset */
outw((1<<5) | (1<<6), FM801_REG(chip, CODEC_CTRL));
inw(FM801_REG(chip, CODEC_CTRL)); /* flush posting data */
@@ -1290,6 +1307,8 @@ static int snd_fm801_chip_init(struct fm801 *chip, int resume)
wait_for_codec(chip, 0, AC97_VENDOR_ID1, msecs_to_jiffies(750));
}
+ __ac97_ok:
+
/* init volume */
outw(0x0808, FM801_REG(chip, PCM_VOL));
outw(0x9f1f, FM801_REG(chip, FM_VOL));
@@ -1298,9 +1317,12 @@ static int snd_fm801_chip_init(struct fm801 *chip, int resume)
/* I2S control - I2S mode */
outw(0x0003, FM801_REG(chip, I2S_MODE));
- /* interrupt setup - unmask MPU, PLAYBACK & CAPTURE */
+ /* interrupt setup */
cmdw = inw(FM801_REG(chip, IRQ_MASK));
- cmdw &= ~0x0083;
+ if (chip->irq < 0)
+ cmdw |= 0x00c3; /* mask everything, no PCM nor MPU */
+ else
+ cmdw &= ~0x0083; /* unmask MPU, PLAYBACK & CAPTURE */
outw(cmdw, FM801_REG(chip, IRQ_MASK));
/* interrupt clear */
@@ -1365,20 +1387,23 @@ static int __devinit snd_fm801_create(struct snd_card *card,
chip->card = card;
chip->pci = pci;
chip->irq = -1;
+ chip->tea575x_tuner = tea575x_tuner;
if ((err = pci_request_regions(pci, "FM801")) < 0) {
kfree(chip);
pci_disable_device(pci);
return err;
}
chip->port = pci_resource_start(pci, 0);
- if (request_irq(pci->irq, snd_fm801_interrupt, IRQF_DISABLED|IRQF_SHARED,
- "FM801", chip)) {
- snd_printk(KERN_ERR "unable to grab IRQ %d\n", chip->irq);
- snd_fm801_free(chip);
- return -EBUSY;
+ if ((tea575x_tuner & 0x0010) == 0) {
+ if (request_irq(pci->irq, snd_fm801_interrupt, IRQF_DISABLED|IRQF_SHARED,
+ "FM801", chip)) {
+ snd_printk(KERN_ERR "unable to grab IRQ %d\n", chip->irq);
+ snd_fm801_free(chip);
+ return -EBUSY;
+ }
+ chip->irq = pci->irq;
+ pci_set_master(pci);
}
- chip->irq = pci->irq;
- pci_set_master(pci);
pci_read_config_byte(pci, PCI_REVISION_ID, &rev);
if (rev >= 0xb1) /* FM801-AU */
@@ -1394,12 +1419,12 @@ static int __devinit snd_fm801_create(struct snd_card *card,
snd_card_set_dev(card, &pci->dev);
#ifdef TEA575X_RADIO
- if (tea575x_tuner > 0 && (tea575x_tuner & 0xffff) < 4) {
+ if (tea575x_tuner > 0 && (tea575x_tuner & 0x000f) < 4) {
chip->tea.dev_nr = tea575x_tuner >> 16;
chip->tea.card = card;
chip->tea.freq_fixup = 10700;
chip->tea.private_data = chip;
- chip->tea.ops = &snd_fm801_tea_ops[(tea575x_tuner & 0xffff) - 1];
+ chip->tea.ops = &snd_fm801_tea_ops[(tea575x_tuner & 0x000f) - 1];
snd_tea575x_init(&chip->tea);
}
#endif
@@ -1439,6 +1464,9 @@ static int __devinit snd_card_fm801_probe(struct pci_dev *pci,
sprintf(card->longname, "%s at 0x%lx, irq %i",
card->shortname, chip->port, chip->irq);
+ if (tea575x_tuner[dev] & 0x0010)
+ goto __fm801_tuner_only;
+
if ((err = snd_fm801_pcm(chip, 0, NULL)) < 0) {
snd_card_free(card);
return err;
@@ -1465,6 +1493,7 @@ static int __devinit snd_card_fm801_probe(struct pci_dev *pci,
return err;
}
+ __fm801_tuner_only:
if ((err = snd_card_register(card)) < 0) {
snd_card_free(card);
return err;
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 23201f3eeb1..9c3d7ac0806 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -29,6 +29,7 @@
#include <sound/core.h>
#include "hda_codec.h"
#include <sound/asoundef.h>
+#include <sound/tlv.h>
#include <sound/initval.h>
#include "hda_local.h"
@@ -50,8 +51,10 @@ struct hda_vendor_id {
/* codec vendor labels */
static struct hda_vendor_id hda_vendor_ids[] = {
{ 0x10ec, "Realtek" },
+ { 0x1057, "Motorola" },
{ 0x11d4, "Analog Devices" },
{ 0x13f6, "C-Media" },
+ { 0x14f1, "Conexant" },
{ 0x434d, "C-Media" },
{ 0x8384, "SigmaTel" },
{} /* terminator */
@@ -841,6 +844,31 @@ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e
return change;
}
+int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag,
+ unsigned int size, unsigned int __user *_tlv)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = get_amp_nid(kcontrol);
+ int dir = get_amp_direction(kcontrol);
+ u32 caps, val1, val2;
+
+ if (size < 4 * sizeof(unsigned int))
+ return -ENOMEM;
+ caps = query_amp_caps(codec, nid, dir);
+ val2 = (((caps & AC_AMPCAP_STEP_SIZE) >> AC_AMPCAP_STEP_SIZE_SHIFT) + 1) * 25;
+ val1 = -((caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT);
+ val1 = ((int)val1) * ((int)val2);
+ if (put_user(SNDRV_CTL_TLVT_DB_SCALE, _tlv))
+ return -EFAULT;
+ if (put_user(2 * sizeof(unsigned int), _tlv + 1))
+ return -EFAULT;
+ if (put_user(val1, _tlv + 2))
+ return -EFAULT;
+ if (put_user(val2, _tlv + 3))
+ return -EFAULT;
+ return 0;
+}
+
/* switch */
int snd_hda_mixer_amp_switch_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
{
@@ -1477,10 +1505,10 @@ int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
formats |= SNDRV_PCM_FMTBIT_S32_LE;
if (val & AC_SUPPCM_BITS_32)
bps = 32;
- else if (val & AC_SUPPCM_BITS_20)
- bps = 20;
else if (val & AC_SUPPCM_BITS_24)
bps = 24;
+ else if (val & AC_SUPPCM_BITS_20)
+ bps = 20;
}
}
else if (streams == AC_SUPFMT_FLOAT32) { /* should be exclusive */
@@ -1916,7 +1944,7 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, struct hda_multi_o
/* front */
snd_hda_codec_setup_stream(codec, nids[HDA_FRONT], stream_tag, 0, format);
- if (mout->hp_nid)
+ if (mout->hp_nid && mout->hp_nid != nids[HDA_FRONT])
/* headphone out will just decode front left/right (stereo) */
snd_hda_codec_setup_stream(codec, mout->hp_nid, stream_tag, 0, format);
/* extra outputs copied from front */
@@ -1984,7 +2012,7 @@ static int is_in_nid_list(hda_nid_t nid, hda_nid_t *list)
* in the order of front, rear, CLFE, side, ...
*
* If more extra outputs (speaker and headphone) are found, the pins are
- * assisnged to hp_pin and speaker_pins[], respectively. If no line-out jack
+ * assisnged to hp_pins[] and speaker_pins[], respectively. If no line-out jack
* is detected, one of speaker of HP pins is assigned as the primary
* output, i.e. to line_out_pins[0]. So, line_outs is always positive
* if any analog output exists.
@@ -2046,14 +2074,26 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *c
cfg->speaker_outs++;
break;
case AC_JACK_HP_OUT:
- cfg->hp_pin = nid;
+ if (cfg->hp_outs >= ARRAY_SIZE(cfg->hp_pins))
+ continue;
+ cfg->hp_pins[cfg->hp_outs] = nid;
+ cfg->hp_outs++;
break;
- case AC_JACK_MIC_IN:
- if (loc == AC_JACK_LOC_FRONT)
- cfg->input_pins[AUTO_PIN_FRONT_MIC] = nid;
- else
- cfg->input_pins[AUTO_PIN_MIC] = nid;
+ case AC_JACK_MIC_IN: {
+ int preferred, alt;
+ if (loc == AC_JACK_LOC_FRONT) {
+ preferred = AUTO_PIN_FRONT_MIC;
+ alt = AUTO_PIN_MIC;
+ } else {
+ preferred = AUTO_PIN_MIC;
+ alt = AUTO_PIN_FRONT_MIC;
+ }
+ if (!cfg->input_pins[preferred])
+ cfg->input_pins[preferred] = nid;
+ else if (!cfg->input_pins[alt])
+ cfg->input_pins[alt] = nid;
break;
+ }
case AC_JACK_LINE_IN:
if (loc == AC_JACK_LOC_FRONT)
cfg->input_pins[AUTO_PIN_FRONT_LINE] = nid;
@@ -2119,8 +2159,10 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *c
cfg->speaker_outs, cfg->speaker_pins[0],
cfg->speaker_pins[1], cfg->speaker_pins[2],
cfg->speaker_pins[3], cfg->speaker_pins[4]);
- snd_printd(" hp=0x%x, dig_out=0x%x, din_in=0x%x\n",
- cfg->hp_pin, cfg->dig_out_pin, cfg->dig_in_pin);
+ snd_printd(" hp_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n",
+ cfg->hp_outs, cfg->hp_pins[0],
+ cfg->hp_pins[1], cfg->hp_pins[2],
+ cfg->hp_pins[3], cfg->hp_pins[4]);
snd_printd(" inputs: mic=0x%x, fmic=0x%x, line=0x%x, fline=0x%x,"
" cd=0x%x, aux=0x%x\n",
cfg->input_pins[AUTO_PIN_MIC],
@@ -2141,10 +2183,12 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *c
sizeof(cfg->speaker_pins));
cfg->speaker_outs = 0;
memset(cfg->speaker_pins, 0, sizeof(cfg->speaker_pins));
- } else if (cfg->hp_pin) {
- cfg->line_outs = 1;
- cfg->line_out_pins[0] = cfg->hp_pin;
- cfg->hp_pin = 0;
+ } else if (cfg->hp_outs) {
+ cfg->line_outs = cfg->hp_outs;
+ memcpy(cfg->line_out_pins, cfg->hp_pins,
+ sizeof(cfg->hp_pins));
+ cfg->hp_outs = 0;
+ memset(cfg->hp_pins, 0, sizeof(cfg->hp_pins));
}
}
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 40520e9d5a4..c12bc4e8840 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -479,7 +479,7 @@ struct hda_codec_ops {
struct hda_amp_info {
u32 key; /* hash key */
u32 amp_caps; /* amp capabilities */
- u16 vol[2]; /* current volume & mute*/
+ u16 vol[2]; /* current volume & mute */
u16 status; /* update flag */
u16 next; /* next link */
};
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 85ad164ada5..97e9af130b7 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -46,11 +46,18 @@ struct hda_gnode {
};
/* patch-specific record */
+
+#define MAX_PCM_VOLS 2
+struct pcm_vol {
+ struct hda_gnode *node; /* Node for PCM volume */
+ unsigned int index; /* connection of PCM volume */
+};
+
struct hda_gspec {
struct hda_gnode *dac_node[2]; /* DAC node */
struct hda_gnode *out_pin_node[2]; /* Output pin (Line-Out) node */
- struct hda_gnode *pcm_vol_node[2]; /* Node for PCM volume */
- unsigned int pcm_vol_index[2]; /* connection of PCM volume */
+ struct pcm_vol pcm_vol[MAX_PCM_VOLS]; /* PCM volumes */
+ unsigned int pcm_vol_nodes; /* number of PCM volumes */
struct hda_gnode *adc_node; /* ADC node */
struct hda_gnode *cap_vol_node; /* Node for capture volume */
@@ -285,9 +292,11 @@ static int parse_output_path(struct hda_codec *codec, struct hda_gspec *spec,
return node == spec->dac_node[dac_idx];
}
spec->dac_node[dac_idx] = node;
- if (node->wid_caps & AC_WCAP_OUT_AMP) {
- spec->pcm_vol_node[dac_idx] = node;
- spec->pcm_vol_index[dac_idx] = 0;
+ if ((node->wid_caps & AC_WCAP_OUT_AMP) &&
+ spec->pcm_vol_nodes < MAX_PCM_VOLS) {
+ spec->pcm_vol[spec->pcm_vol_nodes].node = node;
+ spec->pcm_vol[spec->pcm_vol_nodes].index = 0;
+ spec->pcm_vol_nodes++;
}
return 1; /* found */
}
@@ -307,13 +316,16 @@ static int parse_output_path(struct hda_codec *codec, struct hda_gspec *spec,
select_input_connection(codec, node, i);
unmute_input(codec, node, i);
unmute_output(codec, node);
- if (! spec->pcm_vol_node[dac_idx]) {
- if (node->wid_caps & AC_WCAP_IN_AMP) {
- spec->pcm_vol_node[dac_idx] = node;
- spec->pcm_vol_index[dac_idx] = i;
- } else if (node->wid_caps & AC_WCAP_OUT_AMP) {
- spec->pcm_vol_node[dac_idx] = node;
- spec->pcm_vol_index[dac_idx] = 0;
+ if (spec->dac_node[dac_idx] &&
+ spec->pcm_vol_nodes < MAX_PCM_VOLS &&
+ !(spec->dac_node[dac_idx]->wid_caps &
+ AC_WCAP_OUT_AMP)) {
+ if ((node->wid_caps & AC_WCAP_IN_AMP) ||
+ (node->wid_caps & AC_WCAP_OUT_AMP)) {
+ int n = spec->pcm_vol_nodes;
+ spec->pcm_vol[n].node = node;
+ spec->pcm_vol[n].index = i;
+ spec->pcm_vol_nodes++;
}
}
return 1;
@@ -370,7 +382,9 @@ static struct hda_gnode *parse_output_jack(struct hda_codec *codec,
/* set PIN-Out enable */
snd_hda_codec_write(codec, node->nid, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
- AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN);
+ AC_PINCTL_OUT_EN |
+ ((node->pin_caps & AC_PINCAP_HP_DRV) ?
+ AC_PINCTL_HP_EN : 0));
return node;
}
}
@@ -461,14 +475,19 @@ static const char *get_input_type(struct hda_gnode *node, unsigned int *pinctl)
return "Front Line";
return "Line";
case AC_JACK_CD:
+#if 0
if (pinctl)
*pinctl |= AC_PINCTL_VREF_GRD;
+#endif
return "CD";
case AC_JACK_AUX:
if ((location & 0x0f) == AC_JACK_LOC_FRONT)
return "Front Aux";
return "Aux";
case AC_JACK_MIC_IN:
+ if (node->pin_caps &
+ (AC_PINCAP_VREF_80 << AC_PINCAP_VREF_SHIFT))
+ *pinctl |= AC_PINCTL_VREF_80;
if ((location & 0x0f) == AC_JACK_LOC_FRONT)
return "Front Mic";
return "Mic";
@@ -556,6 +575,29 @@ static int parse_adc_sub_nodes(struct hda_codec *codec, struct hda_gspec *spec,
return 1; /* found */
}
+/* add a capture source element */
+static void add_cap_src(struct hda_gspec *spec, int idx)
+{
+ struct hda_input_mux_item *csrc;
+ char *buf;
+ int num, ocap;
+
+ num = spec->input_mux.num_items;
+ csrc = &spec->input_mux.items[num];
+ buf = spec->cap_labels[num];
+ for (ocap = 0; ocap < num; ocap++) {
+ if (! strcmp(buf, spec->cap_labels[ocap])) {
+ /* same label already exists,
+ * put the index number to be unique
+ */
+ sprintf(buf, "%s %d", spec->cap_labels[ocap], num);
+ break;
+ }
+ }
+ csrc->index = idx;
+ spec->input_mux.num_items++;
+}
+
/*
* parse input
*/
@@ -576,28 +618,26 @@ static int parse_input_path(struct hda_codec *codec, struct hda_gnode *adc_node)
* if it reaches to a proper input PIN, add the path as the
* input path.
*/
+ /* first, check the direct connections to PIN widgets */
for (i = 0; i < adc_node->nconns; i++) {
node = hda_get_node(spec, adc_node->conn_list[i]);
- if (! node)
- continue;
- err = parse_adc_sub_nodes(codec, spec, node);
- if (err < 0)
- return err;
- else if (err > 0) {
- struct hda_input_mux_item *csrc = &spec->input_mux.items[spec->input_mux.num_items];
- char *buf = spec->cap_labels[spec->input_mux.num_items];
- int ocap;
- for (ocap = 0; ocap < spec->input_mux.num_items; ocap++) {
- if (! strcmp(buf, spec->cap_labels[ocap])) {
- /* same label already exists,
- * put the index number to be unique
- */
- sprintf(buf, "%s %d", spec->cap_labels[ocap],
- spec->input_mux.num_items);
- }
- }
- csrc->index = i;
- spec->input_mux.num_items++;
+ if (node && node->type == AC_WID_PIN) {
+ err = parse_adc_sub_nodes(codec, spec, node);
+ if (err < 0)
+ return err;
+ else if (err > 0)
+ add_cap_src(spec, i);
+ }
+ }
+ /* ... then check the rests, more complicated connections */
+ for (i = 0; i < adc_node->nconns; i++) {
+ node = hda_get_node(spec, adc_node->conn_list[i]);
+ if (node && node->type != AC_WID_PIN) {
+ err = parse_adc_sub_nodes(codec, spec, node);
+ if (err < 0)
+ return err;
+ else if (err > 0)
+ add_cap_src(spec, i);
}
}
@@ -647,9 +687,6 @@ static int parse_input(struct hda_codec *codec)
/*
* create mixer controls if possible
*/
-#define DIR_OUT 0x1
-#define DIR_IN 0x2
-
static int create_mixer(struct hda_codec *codec, struct hda_gnode *node,
unsigned int index, const char *type, const char *dir_sfx)
{
@@ -722,49 +759,97 @@ static int check_existing_control(struct hda_codec *codec, const char *type, con
/*
* build output mixer controls
*/
-static int build_output_controls(struct hda_codec *codec)
+static int create_output_mixers(struct hda_codec *codec, const char **names)
{
struct hda_gspec *spec = codec->spec;
- static const char *types[2] = { "Master", "Headphone" };
int i, err;
- for (i = 0; i < 2 && spec->pcm_vol_node[i]; i++) {
- err = create_mixer(codec, spec->pcm_vol_node[i],
- spec->pcm_vol_index[i],
- types[i], "Playback");
+ for (i = 0; i < spec->pcm_vol_nodes; i++) {
+ err = create_mixer(codec, spec->pcm_vol[i].node,
+ spec->pcm_vol[i].index,
+ names[i], "Playback");
if (err < 0)
return err;
}
return 0;
}
+static int build_output_controls(struct hda_codec *codec)
+{
+ struct hda_gspec *spec = codec->spec;
+ static const char *types_speaker[] = { "Speaker", "Headphone" };
+ static const char *types_line[] = { "Front", "Headphone" };
+
+ switch (spec->pcm_vol_nodes) {
+ case 1:
+ return create_mixer(codec, spec->pcm_vol[0].node,
+ spec->pcm_vol[0].index,
+ "Master", "Playback");
+ case 2:
+ if (defcfg_type(spec->out_pin_node[0]) == AC_JACK_SPEAKER)
+ return create_output_mixers(codec, types_speaker);
+ else
+ return create_output_mixers(codec, types_line);
+ }
+ return 0;
+}
+
/* create capture volume/switch */
static int build_input_controls(struct hda_codec *codec)
{
struct hda_gspec *spec = codec->spec;
struct hda_gnode *adc_node = spec->adc_node;
- int err;
-
- if (! adc_node)
+ int i, err;
+ static struct snd_kcontrol_new cap_sel = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = capture_source_info,
+ .get = capture_source_get,
+ .put = capture_source_put,
+ };
+
+ if (! adc_node || ! spec->input_mux.num_items)
return 0; /* not found */
+ spec->cur_cap_src = 0;
+ select_input_connection(codec, adc_node,
+ spec->input_mux.items[0].index);
+
/* create capture volume and switch controls if the ADC has an amp */
- err = create_mixer(codec, adc_node, 0, NULL, "Capture");
+ /* do we have only a single item? */
+ if (spec->input_mux.num_items == 1) {
+ err = create_mixer(codec, adc_node,
+ spec->input_mux.items[0].index,
+ NULL, "Capture");
+ if (err < 0)
+ return err;
+ return 0;
+ }
/* create input MUX if multiple sources are available */
- if (spec->input_mux.num_items > 1) {
- static struct snd_kcontrol_new cap_sel = {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Capture Source",
- .info = capture_source_info,
- .get = capture_source_get,
- .put = capture_source_put,
- };
- if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&cap_sel, codec))) < 0)
+ if ((err = snd_ctl_add(codec->bus->card,
+ snd_ctl_new1(&cap_sel, codec))) < 0)
+ return err;
+
+ /* no volume control? */
+ if (! (adc_node->wid_caps & AC_WCAP_IN_AMP) ||
+ ! (adc_node->amp_in_caps & AC_AMPCAP_NUM_STEPS))
+ return 0;
+
+ for (i = 0; i < spec->input_mux.num_items; i++) {
+ struct snd_kcontrol_new knew;
+ char name[32];
+ sprintf(name, "%s Capture Volume",
+ spec->input_mux.items[i].label);
+ knew = (struct snd_kcontrol_new)
+ HDA_CODEC_VOLUME(name, adc_node->nid,
+ spec->input_mux.items[i].index,
+ HDA_INPUT);
+ if ((err = snd_ctl_add(codec->bus->card,
+ snd_ctl_new1(&knew, codec))) < 0)
return err;
- spec->cur_cap_src = 0;
- select_input_connection(codec, adc_node, spec->input_mux.items[0].index);
}
+
return 0;
}
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 79d63c99f09..e9d4cb4d07e 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -55,6 +55,7 @@ static char *model;
static int position_fix;
static int probe_mask = -1;
static int single_cmd;
+static int disable_msi;
module_param(index, int, 0444);
MODULE_PARM_DESC(index, "Index value for Intel HD audio interface.");
@@ -68,6 +69,8 @@ module_param(probe_mask, int, 0444);
MODULE_PARM_DESC(probe_mask, "Bitmask to probe codecs (default = -1).");
module_param(single_cmd, bool, 0444);
MODULE_PARM_DESC(single_cmd, "Use single command to communicate with codecs (for debugging only).");
+module_param(disable_msi, int, 0);
+MODULE_PARM_DESC(disable_msi, "Disable Message Signaled Interrupt (MSI)");
/* just for backward compatibility */
@@ -252,7 +255,7 @@ enum {
struct azx_dev {
u32 *bdl; /* virtual address of the BDL */
dma_addr_t bdl_addr; /* physical address of the BDL */
- volatile u32 *posbuf; /* position buffer pointer */
+ u32 *posbuf; /* position buffer pointer */
unsigned int bufsize; /* size of the play buffer in bytes */
unsigned int fragsize; /* size of each period in bytes */
@@ -271,8 +274,8 @@ struct azx_dev {
/* for sanity check of position buffer */
unsigned int period_intr;
- unsigned int opened: 1;
- unsigned int running: 1;
+ unsigned int opened :1;
+ unsigned int running :1;
};
/* CORB/RIRB */
@@ -330,8 +333,9 @@ struct azx {
/* flags */
int position_fix;
- unsigned int initialized: 1;
- unsigned int single_cmd: 1;
+ unsigned int initialized :1;
+ unsigned int single_cmd :1;
+ unsigned int polling_mode :1;
};
/* driver types */
@@ -516,23 +520,36 @@ static void azx_update_rirb(struct azx *chip)
static unsigned int azx_rirb_get_response(struct hda_codec *codec)
{
struct azx *chip = codec->bus->private_data;
- int timeout = 50;
+ unsigned long timeout;
- while (chip->rirb.cmds) {
- if (! --timeout) {
- snd_printk(KERN_ERR
- "hda_intel: azx_get_response timeout, "
- "switching to single_cmd mode...\n");
- chip->rirb.rp = azx_readb(chip, RIRBWP);
- chip->rirb.cmds = 0;
- /* switch to single_cmd mode */
- chip->single_cmd = 1;
- azx_free_cmd_io(chip);
- return -1;
+ again:
+ timeout = jiffies + msecs_to_jiffies(1000);
+ do {
+ if (chip->polling_mode) {
+ spin_lock_irq(&chip->reg_lock);
+ azx_update_rirb(chip);
+ spin_unlock_irq(&chip->reg_lock);
}
- msleep(1);
+ if (! chip->rirb.cmds)
+ return chip->rirb.res; /* the last value */
+ schedule_timeout_interruptible(1);
+ } while (time_after_eq(timeout, jiffies));
+
+ if (!chip->polling_mode) {
+ snd_printk(KERN_WARNING "hda_intel: azx_get_response timeout, "
+ "switching to polling mode...\n");
+ chip->polling_mode = 1;
+ goto again;
}
- return chip->rirb.res; /* the last value */
+
+ snd_printk(KERN_ERR "hda_intel: azx_get_response timeout, "
+ "switching to single_cmd mode...\n");
+ chip->rirb.rp = azx_readb(chip, RIRBWP);
+ chip->rirb.cmds = 0;
+ /* switch to single_cmd mode */
+ chip->single_cmd = 1;
+ azx_free_cmd_io(chip);
+ return -1;
}
/*
@@ -642,14 +659,14 @@ static int azx_reset(struct azx *chip)
azx_writeb(chip, GCTL, azx_readb(chip, GCTL) | ICH6_GCTL_RESET);
count = 50;
- while (! azx_readb(chip, GCTL) && --count)
+ while (!azx_readb(chip, GCTL) && --count)
msleep(1);
- /* Brent Chartrand said to wait >= 540us for codecs to intialize */
+ /* Brent Chartrand said to wait >= 540us for codecs to initialize */
msleep(1);
/* check to see if controller is ready */
- if (! azx_readb(chip, GCTL)) {
+ if (!azx_readb(chip, GCTL)) {
snd_printd("azx_reset: controller not ready!\n");
return -EBUSY;
}
@@ -658,7 +675,7 @@ static int azx_reset(struct azx *chip)
azx_writel(chip, GCTL, azx_readl(chip, GCTL) | ICH6_GCTL_UREN);
/* detect codecs */
- if (! chip->codec_mask) {
+ if (!chip->codec_mask) {
chip->codec_mask = azx_readw(chip, STATESTS);
snd_printdd("codec_mask = 0x%x\n", chip->codec_mask);
}
@@ -766,7 +783,7 @@ static void azx_init_chip(struct azx *chip)
azx_int_enable(chip);
/* initialize the codec command I/O */
- if (! chip->single_cmd)
+ if (!chip->single_cmd)
azx_init_cmd_io(chip);
/* program the position buffer */
@@ -794,7 +811,7 @@ static void azx_init_chip(struct azx *chip)
/*
* interrupt handler
*/
-static irqreturn_t azx_interrupt(int irq, void* dev_id, struct pt_regs *regs)
+static irqreturn_t azx_interrupt(int irq, void *dev_id, struct pt_regs *regs)
{
struct azx *chip = dev_id;
struct azx_dev *azx_dev;
@@ -999,8 +1016,9 @@ static struct snd_pcm_hardware azx_pcm_hw = {
.info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_PAUSE /*|*/
- /*SNDRV_PCM_INFO_RESUME*/),
+ /* No full-resume yet implemented */
+ /* SNDRV_PCM_INFO_RESUME |*/
+ SNDRV_PCM_INFO_PAUSE),
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.rates = SNDRV_PCM_RATE_48000,
.rate_min = 48000,
@@ -1178,7 +1196,7 @@ static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream)
if (chip->position_fix == POS_FIX_POSBUF ||
chip->position_fix == POS_FIX_AUTO) {
/* use the position buffer */
- pos = *azx_dev->posbuf;
+ pos = le32_to_cpu(*azx_dev->posbuf);
if (chip->position_fix == POS_FIX_AUTO &&
azx_dev->period_intr == 1 && ! pos) {
printk(KERN_WARNING
@@ -1222,7 +1240,12 @@ static int __devinit create_codec_pcm(struct azx *chip, struct hda_codec *codec,
struct snd_pcm *pcm;
struct azx_pcm *apcm;
- snd_assert(cpcm->stream[0].substreams || cpcm->stream[1].substreams, return -EINVAL);
+ /* if no substreams are defined for both playback and capture,
+ * it's just a placeholder. ignore it.
+ */
+ if (!cpcm->stream[0].substreams && !cpcm->stream[1].substreams)
+ return 0;
+
snd_assert(cpcm->name, return -EINVAL);
err = snd_pcm_new(chip->card, cpcm->name, pcm_dev,
@@ -1248,7 +1271,8 @@ static int __devinit create_codec_pcm(struct azx *chip, struct hda_codec *codec,
snd_dma_pci_data(chip->pci),
1024 * 64, 1024 * 128);
chip->pcm[pcm_dev] = pcm;
- chip->pcm_devs = pcm_dev + 1;
+ if (chip->pcm_devs < pcm_dev + 1)
+ chip->pcm_devs = pcm_dev + 1;
return 0;
}
@@ -1326,7 +1350,7 @@ static int __devinit azx_init_stream(struct azx *chip)
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 = (volatile u32 *)(chip->posbuf.area + i * 8);
+ 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);
/* int mask: SDI0=0x01, SDI1=0x02, ... SDO3=0x80 */
@@ -1355,6 +1379,10 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state)
snd_pcm_suspend_all(chip->pcm[i]);
snd_hda_suspend(chip->bus, state);
azx_free_cmd_io(chip);
+ if (chip->irq >= 0)
+ free_irq(chip->irq, chip);
+ if (!disable_msi)
+ pci_disable_msi(chip->pci);
pci_disable_device(pci);
pci_save_state(pci);
return 0;
@@ -1367,6 +1395,12 @@ static int azx_resume(struct pci_dev *pci)
pci_restore_state(pci);
pci_enable_device(pci);
+ if (!disable_msi)
+ pci_enable_msi(pci);
+ /* FIXME: need proper error handling */
+ request_irq(pci->irq, azx_interrupt, IRQF_DISABLED|IRQF_SHARED,
+ "HDA Intel", chip);
+ chip->irq = pci->irq;
pci_set_master(pci);
azx_init_chip(chip);
snd_hda_resume(chip->bus);
@@ -1398,12 +1432,14 @@ static int azx_free(struct azx *chip)
azx_writel(chip, DPLBASE, 0);
azx_writel(chip, DPUBASE, 0);
- /* wait a little for interrupts to finish */
- msleep(1);
+ synchronize_irq(chip->irq);
}
- if (chip->irq >= 0)
+ if (chip->irq >= 0) {
free_irq(chip->irq, (void*)chip);
+ if (!disable_msi)
+ pci_disable_msi(chip->pci);
+ }
if (chip->remap_addr)
iounmap(chip->remap_addr);
@@ -1434,19 +1470,19 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
struct azx **rchip)
{
struct azx *chip;
- int err = 0;
+ int err;
static struct snd_device_ops ops = {
.dev_free = azx_dev_free,
};
*rchip = NULL;
- if ((err = pci_enable_device(pci)) < 0)
+ err = pci_enable_device(pci);
+ if (err < 0)
return err;
chip = kzalloc(sizeof(*chip), GFP_KERNEL);
-
- if (NULL == chip) {
+ if (!chip) {
snd_printk(KERN_ERR SFX "cannot allocate chip\n");
pci_disable_device(pci);
return -ENOMEM;
@@ -1472,13 +1508,14 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
}
#endif
- if ((err = pci_request_regions(pci, "ICH HD audio")) < 0) {
+ err = pci_request_regions(pci, "ICH HD audio");
+ if (err < 0) {
kfree(chip);
pci_disable_device(pci);
return err;
}
- chip->addr = pci_resource_start(pci,0);
+ chip->addr = pci_resource_start(pci, 0);
chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci,0));
if (chip->remap_addr == NULL) {
snd_printk(KERN_ERR SFX "ioremap error\n");
@@ -1486,6 +1523,9 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
goto errout;
}
+ if (!disable_msi)
+ pci_enable_msi(pci);
+
if (request_irq(pci->irq, azx_interrupt, IRQF_DISABLED|IRQF_SHARED,
"HDA Intel", (void*)chip)) {
snd_printk(KERN_ERR SFX "unable to grab IRQ %d\n", pci->irq);
@@ -1519,7 +1559,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
}
chip->num_streams = chip->playback_streams + chip->capture_streams;
chip->azx_dev = kcalloc(chip->num_streams, sizeof(*chip->azx_dev), GFP_KERNEL);
- if (! chip->azx_dev) {
+ if (!chip->azx_dev) {
snd_printk(KERN_ERR "cannot malloc azx_dev\n");
goto errout;
}
@@ -1550,7 +1590,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
chip->initialized = 1;
/* codec detection */
- if (! chip->codec_mask) {
+ if (!chip->codec_mask) {
snd_printk(KERN_ERR SFX "no codecs found!\n");
err = -ENODEV;
goto errout;
@@ -1577,16 +1617,16 @@ static int __devinit azx_probe(struct pci_dev *pci, const struct pci_device_id *
{
struct snd_card *card;
struct azx *chip;
- int err = 0;
+ int err;
card = snd_card_new(index, id, THIS_MODULE, 0);
- if (NULL == card) {
+ if (!card) {
snd_printk(KERN_ERR SFX "Error creating card!\n");
return -ENOMEM;
}
- if ((err = azx_create(card, pci, pci_id->driver_data,
- &chip)) < 0) {
+ err = azx_create(card, pci, pci_id->driver_data, &chip);
+ if (err < 0) {
snd_card_free(card);
return err;
}
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 14e8aa2806e..f9416c36396 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -30,9 +30,13 @@
/* mono volume with index (index=0,1,...) (channel=1,2) */
#define HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+ SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \
.info = snd_hda_mixer_amp_volume_info, \
.get = snd_hda_mixer_amp_volume_get, \
.put = snd_hda_mixer_amp_volume_put, \
+ .tlv = { .c = snd_hda_mixer_amp_tlv }, \
.private_value = HDA_COMPOSE_AMP_VAL(nid, channel, xindex, direction) }
/* stereo volume with index */
#define HDA_CODEC_VOLUME_IDX(xname, xcidx, nid, xindex, direction) \
@@ -63,6 +67,7 @@
int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo);
int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol);
int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol);
+int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *tlv);
int snd_hda_mixer_amp_switch_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo);
int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol);
int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol);
@@ -224,7 +229,8 @@ struct auto_pin_cfg {
hda_nid_t line_out_pins[5]; /* sorted in the order of Front/Surr/CLFE/Side */
int speaker_outs;
hda_nid_t speaker_pins[5];
- hda_nid_t hp_pin;
+ int hp_outs;
+ hda_nid_t hp_pins[5];
hda_nid_t input_pins[AUTO_PIN_LAST];
hda_nid_t dig_out_pin;
hda_nid_t dig_in_pin;
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index c2f0fe85bf3..d737f17695a 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -52,10 +52,9 @@ static void print_amp_caps(struct snd_info_buffer *buffer,
struct hda_codec *codec, hda_nid_t nid, int dir)
{
unsigned int caps;
- if (dir == HDA_OUTPUT)
- caps = snd_hda_param_read(codec, nid, AC_PAR_AMP_OUT_CAP);
- else
- caps = snd_hda_param_read(codec, nid, AC_PAR_AMP_IN_CAP);
+ caps = snd_hda_param_read(codec, nid,
+ dir == HDA_OUTPUT ?
+ AC_PAR_AMP_OUT_CAP : AC_PAR_AMP_IN_CAP);
if (caps == -1 || caps == 0) {
snd_iprintf(buffer, "N/A\n");
return;
@@ -74,10 +73,7 @@ static void print_amp_vals(struct snd_info_buffer *buffer,
unsigned int val;
int i;
- if (dir == HDA_OUTPUT)
- dir = AC_AMP_GET_OUTPUT;
- else
- dir = AC_AMP_GET_INPUT;
+ dir = dir == HDA_OUTPUT ? AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT;
for (i = 0; i < indices; i++) {
snd_iprintf(buffer, " [");
if (stereo) {
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 6823f2bc10b..511df07fa2a 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -488,9 +488,13 @@ static struct snd_kcontrol_new ad1986a_mixers[] = {
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "PCM Playback Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ |
+ SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK,
.info = ad1986a_pcm_amp_vol_info,
.get = ad1986a_pcm_amp_vol_get,
.put = ad1986a_pcm_amp_vol_put,
+ .tlv = { .c = snd_hda_mixer_amp_tlv },
.private_value = HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT)
},
{
@@ -637,6 +641,7 @@ static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = {
.info = snd_hda_mixer_amp_volume_info,
.get = snd_hda_mixer_amp_volume_get,
.put = ad1986a_laptop_master_vol_put,
+ .tlv = { .c = snd_hda_mixer_amp_tlv },
.private_value = HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT),
},
{
@@ -791,6 +796,8 @@ static struct hda_board_config ad1986a_cfg_tbl[] = {
.config = AD1986A_3STACK }, /* ASUS A8N-VM CSM */
{ .pci_subvendor = 0x1043, .pci_subdevice = 0x81b3,
.config = AD1986A_3STACK }, /* ASUS P5RD2-VM / P5GPL-X SE */
+ { .pci_subvendor = 0x1043, .pci_subdevice = 0x81cb,
+ .config = AD1986A_3STACK }, /* ASUS M2NPV-VM */
{ .modelname = "laptop", .config = AD1986A_LAPTOP },
{ .pci_subvendor = 0x144d, .pci_subdevice = 0xc01e,
.config = AD1986A_LAPTOP }, /* FSC V2060 */
@@ -803,6 +810,8 @@ static struct hda_board_config ad1986a_cfg_tbl[] = {
.config = AD1986A_LAPTOP_EAPD }, /* Samsung X60 Chane */
{ .pci_subvendor = 0x144d, .pci_subdevice = 0xc024,
.config = AD1986A_LAPTOP_EAPD }, /* Samsung R65-T2300 Charis */
+ { .pci_subvendor = 0x144d, .pci_subdevice = 0xc026,
+ .config = AD1986A_LAPTOP_EAPD }, /* Samsung X10-T2300 Culesa */
{ .pci_subvendor = 0x1043, .pci_subdevice = 0x1153,
.config = AD1986A_LAPTOP_EAPD }, /* ASUS M9 */
{ .pci_subvendor = 0x1043, .pci_subdevice = 0x1213,
@@ -1626,10 +1635,12 @@ static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol,
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct ad198x_spec *spec = codec->spec;
- if (spec->need_dac_fix)
+ int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode,
+ spec->num_channel_mode,
+ &spec->multiout.max_channels);
+ if (! err && spec->need_dac_fix)
spec->multiout.num_dacs = spec->multiout.max_channels / 2;
- return snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode,
- spec->num_channel_mode, &spec->multiout.max_channels);
+ return err;
}
/* 6-stack mode */
@@ -2460,7 +2471,7 @@ static void ad1988_auto_init_extra_out(struct hda_codec *codec)
pin = spec->autocfg.speaker_pins[0];
if (pin) /* connect to front */
ad1988_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
- pin = spec->autocfg.hp_pin;
+ pin = spec->autocfg.hp_pins[0];
if (pin) /* connect to front */
ad1988_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
}
@@ -2512,7 +2523,7 @@ static int ad1988_parse_auto_config(struct hda_codec *codec)
(err = ad1988_auto_create_extra_out(codec,
spec->autocfg.speaker_pins[0],
"Speaker")) < 0 ||
- (err = ad1988_auto_create_extra_out(codec, spec->autocfg.hp_pin,
+ (err = ad1988_auto_create_extra_out(codec, spec->autocfg.hp_pins[0],
"Headphone")) < 0 ||
(err = ad1988_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0)
return err;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 18d105263fe..d08d2e399c8 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -79,6 +79,7 @@ enum {
ALC262_BASIC,
ALC262_FUJITSU,
ALC262_HP_BPC,
+ ALC262_BENQ_ED8,
ALC262_AUTO,
ALC262_MODEL_LAST /* last tag */
};
@@ -89,6 +90,7 @@ enum {
ALC660_3ST,
ALC861_3ST_DIG,
ALC861_6ST_DIG,
+ ALC861_UNIWILL_M31,
ALC861_AUTO,
ALC861_MODEL_LAST,
};
@@ -97,6 +99,7 @@ enum {
enum {
ALC882_3ST_DIG,
ALC882_6ST_DIG,
+ ALC882_ARIMA,
ALC882_AUTO,
ALC882_MODEL_LAST,
};
@@ -108,6 +111,7 @@ enum {
ALC883_3ST_6ch,
ALC883_6ST_DIG,
ALC888_DEMO_BOARD,
+ ALC883_ACER,
ALC883_AUTO,
ALC883_MODEL_LAST,
};
@@ -153,6 +157,7 @@ struct alc_spec {
/* channel model */
const struct hda_channel_mode *channel_mode;
int num_channel_mode;
+ int need_dac_fix;
/* PCM information */
struct hda_pcm pcm_rec[3]; /* used in alc_build_pcms() */
@@ -190,6 +195,7 @@ struct alc_config_preset {
hda_nid_t dig_in_nid;
unsigned int num_channel_mode;
const struct hda_channel_mode *channel_mode;
+ int need_dac_fix;
unsigned int num_mux_defs;
const struct hda_input_mux *input_mux;
void (*unsol_event)(struct hda_codec *, unsigned int);
@@ -262,9 +268,12 @@ static int alc_ch_mode_put(struct snd_kcontrol *kcontrol,
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
- return snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode,
- spec->num_channel_mode,
- &spec->multiout.max_channels);
+ int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode,
+ spec->num_channel_mode,
+ &spec->multiout.max_channels);
+ if (! err && spec->need_dac_fix)
+ spec->multiout.num_dacs = spec->multiout.max_channels / 2;
+ return err;
}
/*
@@ -544,6 +553,7 @@ static void setup_preset(struct alc_spec *spec,
spec->channel_mode = preset->channel_mode;
spec->num_channel_mode = preset->num_channel_mode;
+ spec->need_dac_fix = preset->need_dac_fix;
spec->multiout.max_channels = spec->channel_mode[0].channels;
@@ -1348,6 +1358,10 @@ static struct hda_verb alc880_pin_clevo_init_verbs[] = {
};
static struct hda_verb alc880_pin_tcl_S700_init_verbs[] = {
+ /* change to EAPD mode */
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x3060},
+
/* Headphone output */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
/* Front output*/
@@ -1782,25 +1796,9 @@ static int alc_build_pcms(struct hda_codec *codec)
}
}
- /* If the use of more than one ADC is requested for the current
- * model, configure a second analog capture-only PCM.
- */
- if (spec->num_adc_nids > 1) {
- codec->num_pcms++;
- info++;
- info->name = spec->stream_name_analog;
- /* No playback stream for second PCM */
- info->stream[SNDRV_PCM_STREAM_PLAYBACK] = alc_pcm_null_playback;
- info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = 0;
- if (spec->stream_analog_capture) {
- snd_assert(spec->adc_nids, return -EINVAL);
- info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture);
- info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[1];
- }
- }
-
+ /* SPDIF for stream index #1 */
if (spec->multiout.dig_out_nid || spec->dig_in_nid) {
- codec->num_pcms++;
+ codec->num_pcms = 2;
info++;
info->name = spec->stream_name_digital;
if (spec->multiout.dig_out_nid &&
@@ -1815,6 +1813,24 @@ static int alc_build_pcms(struct hda_codec *codec)
}
}
+ /* If the use of more than one ADC is requested for the current
+ * model, configure a second analog capture-only PCM.
+ */
+ /* Additional Analaog capture for index #2 */
+ if (spec->num_adc_nids > 1 && spec->stream_analog_capture &&
+ spec->adc_nids) {
+ codec->num_pcms = 3;
+ info++;
+ info->name = spec->stream_name_analog;
+ /* No playback stream for second PCM */
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] = alc_pcm_null_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = 0;
+ if (spec->stream_analog_capture) {
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture);
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[1];
+ }
+ }
+
return 0;
}
@@ -2130,7 +2146,10 @@ static struct hda_board_config alc880_cfg_tbl[] = {
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe20f, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe210, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe211, .config = ALC880_3ST },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe212, .config = ALC880_3ST },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe213, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe214, .config = ALC880_3ST },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe234, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe302, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe303, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe304, .config = ALC880_3ST },
@@ -2145,6 +2164,7 @@ static struct hda_board_config alc880_cfg_tbl[] = {
{ .pci_subvendor = 0x107b, .pci_subdevice = 0x4040, .config = ALC880_3ST },
{ .pci_subvendor = 0x107b, .pci_subdevice = 0x4041, .config = ALC880_3ST },
/* TCL S700 */
+ { .modelname = "tcl", .config = ALC880_TCL_S700 },
{ .pci_subvendor = 0x19db, .pci_subdevice = 0x4188, .config = ALC880_TCL_S700 },
/* Back 3 jack, front 2 jack (Internal add Aux-In) */
@@ -2156,8 +2176,13 @@ static struct hda_board_config alc880_cfg_tbl[] = {
{ .modelname = "3stack-digout", .config = ALC880_3ST_DIG },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe308, .config = ALC880_3ST_DIG },
{ .pci_subvendor = 0x1025, .pci_subdevice = 0x0070, .config = ALC880_3ST_DIG },
- /* Clevo m520G NB */
- { .pci_subvendor = 0x1558, .pci_subdevice = 0x0520, .config = ALC880_CLEVO },
+
+ /* Clevo laptops */
+ { .modelname = "clevo", .config = ALC880_CLEVO },
+ { .pci_subvendor = 0x1558, .pci_subdevice = 0x0520,
+ .config = ALC880_CLEVO }, /* Clevo m520G NB */
+ { .pci_subvendor = 0x1558, .pci_subdevice = 0x0660,
+ .config = ALC880_CLEVO }, /* Clevo m665n */
/* Back 3 jack plus 1 SPDIF out jack, front 2 jack (Internal add Aux-In)*/
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe305, .config = ALC880_3ST_DIG },
@@ -2222,12 +2247,16 @@ static struct hda_board_config alc880_cfg_tbl[] = {
{ .pci_subvendor = 0x1043, .pci_subdevice = 0x1113, .config = ALC880_ASUS_DIG },
{ .pci_subvendor = 0x1043, .pci_subdevice = 0x1173, .config = ALC880_ASUS_DIG },
{ .pci_subvendor = 0x1043, .pci_subdevice = 0x1993, .config = ALC880_ASUS },
+ { .pci_subvendor = 0x1043, .pci_subdevice = 0x10c2, .config = ALC880_ASUS_DIG }, /* Asus W6A */
{ .pci_subvendor = 0x1043, .pci_subdevice = 0x10c3, .config = ALC880_ASUS_DIG },
{ .pci_subvendor = 0x1043, .pci_subdevice = 0x1133, .config = ALC880_ASUS },
{ .pci_subvendor = 0x1043, .pci_subdevice = 0x1123, .config = ALC880_ASUS_DIG },
{ .pci_subvendor = 0x1043, .pci_subdevice = 0x1143, .config = ALC880_ASUS },
+ { .modelname = "asus-w1v", .config = ALC880_ASUS_W1V },
{ .pci_subvendor = 0x1043, .pci_subdevice = 0x10b3, .config = ALC880_ASUS_W1V },
+ { .modelname = "asus-dig", .config = ALC880_ASUS_DIG },
{ .pci_subvendor = 0x1043, .pci_subdevice = 0x8181, .config = ALC880_ASUS_DIG }, /* ASUS P4GPL-X */
+ { .modelname = "asus-dig2", .config = ALC880_ASUS_DIG2 },
{ .pci_subvendor = 0x1558, .pci_subdevice = 0x5401, .config = ALC880_ASUS_DIG2 },
{ .modelname = "uniwill", .config = ALC880_UNIWILL_DIG },
@@ -2243,6 +2272,7 @@ static struct hda_board_config alc880_cfg_tbl[] = {
{ .modelname = "lg-lw", .config = ALC880_LG_LW },
{ .pci_subvendor = 0x1854, .pci_subdevice = 0x0018, .config = ALC880_LG_LW },
+ { .pci_subvendor = 0x1854, .pci_subdevice = 0x0077, .config = ALC880_LG_LW },
#ifdef CONFIG_SND_DEBUG
{ .modelname = "test", .config = ALC880_TEST },
@@ -2263,6 +2293,7 @@ static struct alc_config_preset alc880_presets[] = {
.dac_nids = alc880_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
.channel_mode = alc880_threestack_modes,
+ .need_dac_fix = 1,
.input_mux = &alc880_capture_source,
},
[ALC880_3ST_DIG] = {
@@ -2273,6 +2304,7 @@ static struct alc_config_preset alc880_presets[] = {
.dig_out_nid = ALC880_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
.channel_mode = alc880_threestack_modes,
+ .need_dac_fix = 1,
.input_mux = &alc880_capture_source,
},
[ALC880_TCL_S700] = {
@@ -2365,6 +2397,7 @@ static struct alc_config_preset alc880_presets[] = {
.dac_nids = alc880_asus_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
.channel_mode = alc880_asus_modes,
+ .need_dac_fix = 1,
.input_mux = &alc880_capture_source,
},
[ALC880_ASUS_DIG] = {
@@ -2376,6 +2409,7 @@ static struct alc_config_preset alc880_presets[] = {
.dig_out_nid = ALC880_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
.channel_mode = alc880_asus_modes,
+ .need_dac_fix = 1,
.input_mux = &alc880_capture_source,
},
[ALC880_ASUS_DIG2] = {
@@ -2387,6 +2421,7 @@ static struct alc_config_preset alc880_presets[] = {
.dig_out_nid = ALC880_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
.channel_mode = alc880_asus_modes,
+ .need_dac_fix = 1,
.input_mux = &alc880_capture_source,
},
[ALC880_ASUS_W1V] = {
@@ -2398,6 +2433,7 @@ static struct alc_config_preset alc880_presets[] = {
.dig_out_nid = ALC880_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
.channel_mode = alc880_asus_modes,
+ .need_dac_fix = 1,
.input_mux = &alc880_capture_source,
},
[ALC880_UNIWILL_DIG] = {
@@ -2408,6 +2444,7 @@ static struct alc_config_preset alc880_presets[] = {
.dig_out_nid = ALC880_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
.channel_mode = alc880_asus_modes,
+ .need_dac_fix = 1,
.input_mux = &alc880_capture_source,
},
[ALC880_CLEVO] = {
@@ -2419,6 +2456,7 @@ static struct alc_config_preset alc880_presets[] = {
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
.channel_mode = alc880_threestack_modes,
+ .need_dac_fix = 1,
.input_mux = &alc880_capture_source,
},
[ALC880_LG] = {
@@ -2430,6 +2468,7 @@ static struct alc_config_preset alc880_presets[] = {
.dig_out_nid = ALC880_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_lg_ch_modes),
.channel_mode = alc880_lg_ch_modes,
+ .need_dac_fix = 1,
.input_mux = &alc880_lg_capture_source,
.unsol_event = alc880_lg_unsol_event,
.init_hook = alc880_lg_automute,
@@ -2714,7 +2753,7 @@ static void alc880_auto_init_extra_out(struct hda_codec *codec)
pin = spec->autocfg.speaker_pins[0];
if (pin) /* connect to front */
alc880_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
- pin = spec->autocfg.hp_pin;
+ pin = spec->autocfg.hp_pins[0];
if (pin) /* connect to front */
alc880_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
}
@@ -2755,7 +2794,7 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
(err = alc880_auto_create_extra_out(spec,
spec->autocfg.speaker_pins[0],
"Speaker")) < 0 ||
- (err = alc880_auto_create_extra_out(spec, spec->autocfg.hp_pin,
+ (err = alc880_auto_create_extra_out(spec, spec->autocfg.hp_pins[0],
"Headphone")) < 0 ||
(err = alc880_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0)
return err;
@@ -3697,7 +3736,7 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec,
return err;
}
- nid = cfg->hp_pin;
+ nid = cfg->hp_pins[0];
if (nid) {
err = alc260_add_playback_controls(spec, nid, "Headphone");
if (err < 0)
@@ -3767,7 +3806,7 @@ static void alc260_auto_init_multi_out(struct hda_codec *codec)
if (nid)
alc260_auto_set_output_and_unmute(codec, nid, PIN_OUT, 0);
- nid = spec->autocfg.hp_pin;
+ nid = spec->autocfg.hp_pins[0];
if (nid)
alc260_auto_set_output_and_unmute(codec, nid, PIN_OUT, 0);
}
@@ -3900,7 +3939,8 @@ static struct hda_board_config alc260_cfg_tbl[] = {
{ .pci_subvendor = 0x152d, .pci_subdevice = 0x0729,
.config = ALC260_BASIC }, /* CTL Travel Master U553W */
{ .modelname = "hp", .config = ALC260_HP },
- { .pci_subvendor = 0x103c, .pci_subdevice = 0x3010, .config = ALC260_HP },
+ { .modelname = "hp-3013", .config = ALC260_HP_3013 },
+ { .pci_subvendor = 0x103c, .pci_subdevice = 0x3010, .config = ALC260_HP_3013 },
{ .pci_subvendor = 0x103c, .pci_subdevice = 0x3011, .config = ALC260_HP },
{ .pci_subvendor = 0x103c, .pci_subdevice = 0x3012, .config = ALC260_HP_3013 },
{ .pci_subvendor = 0x103c, .pci_subdevice = 0x3013, .config = ALC260_HP_3013 },
@@ -4266,6 +4306,13 @@ static struct hda_verb alc882_init_verbs[] = {
{ }
};
+static struct hda_verb alc882_eapd_verbs[] = {
+ /* change to EAPD mode */
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x3060},
+ { }
+};
+
/*
* generic initialization of ADC, input mixers and output mixers
*/
@@ -4397,6 +4444,9 @@ static struct hda_board_config alc882_cfg_tbl[] = {
.config = ALC882_6ST_DIG }, /* Foxconn */
{ .pci_subvendor = 0x1019, .pci_subdevice = 0x6668,
.config = ALC882_6ST_DIG }, /* ECS to Intel*/
+ { .modelname = "arima", .config = ALC882_ARIMA },
+ { .pci_subvendor = 0x161f, .pci_subdevice = 0x2054,
+ .config = ALC882_ARIMA }, /* Arima W820Di1 */
{ .modelname = "auto", .config = ALC882_AUTO },
{}
};
@@ -4411,6 +4461,7 @@ static struct alc_config_preset alc882_presets[] = {
.dig_in_nid = ALC882_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
.channel_mode = alc882_ch_modes,
+ .need_dac_fix = 1,
.input_mux = &alc882_capture_source,
},
[ALC882_6ST_DIG] = {
@@ -4424,6 +4475,15 @@ static struct alc_config_preset alc882_presets[] = {
.channel_mode = alc882_sixstack_modes,
.input_mux = &alc882_capture_source,
},
+ [ALC882_ARIMA] = {
+ .mixers = { alc882_base_mixer, alc882_chmode_mixer },
+ .init_verbs = { alc882_init_verbs, alc882_eapd_verbs },
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes),
+ .channel_mode = alc882_sixstack_modes,
+ .input_mux = &alc882_capture_source,
+ },
};
@@ -4466,7 +4526,7 @@ static void alc882_auto_init_hp_out(struct hda_codec *codec)
struct alc_spec *spec = codec->spec;
hda_nid_t pin;
- pin = spec->autocfg.hp_pin;
+ pin = spec->autocfg.hp_pins[0];
if (pin) /* connect to front */
alc882_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); /* use dac 0 */
}
@@ -4999,16 +5059,23 @@ static struct snd_kcontrol_new alc883_capture_mixer[] = {
*/
static struct hda_board_config alc883_cfg_tbl[] = {
{ .modelname = "3stack-dig", .config = ALC883_3ST_2ch_DIG },
+ { .modelname = "3stack-6ch-dig", .config = ALC883_3ST_6ch_DIG },
+ { .pci_subvendor = 0x1019, .pci_subdevice = 0x6668,
+ .config = ALC883_3ST_6ch_DIG }, /* ECS to Intel*/
+ { .modelname = "3stack-6ch", .config = ALC883_3ST_6ch },
+ { .pci_subvendor = 0x108e, .pci_subdevice = 0x534d,
+ .config = ALC883_3ST_6ch },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xd601,
+ .config = ALC883_3ST_6ch }, /* D102GGC */
{ .modelname = "6stack-dig", .config = ALC883_6ST_DIG },
- { .modelname = "6stack-dig-demo", .config = ALC888_DEMO_BOARD },
{ .pci_subvendor = 0x1462, .pci_subdevice = 0x6668,
.config = ALC883_6ST_DIG }, /* MSI */
{ .pci_subvendor = 0x105b, .pci_subdevice = 0x6668,
.config = ALC883_6ST_DIG }, /* Foxconn */
- { .pci_subvendor = 0x1019, .pci_subdevice = 0x6668,
- .config = ALC883_3ST_6ch_DIG }, /* ECS to Intel*/
- { .pci_subvendor = 0x108e, .pci_subdevice = 0x534d,
- .config = ALC883_3ST_6ch },
+ { .modelname = "6stack-dig-demo", .config = ALC888_DEMO_BOARD },
+ { .modelname = "acer", .config = ALC883_ACER },
+ { .pci_subvendor = 0x1025, .pci_subdevice = 0/*0x0102*/,
+ .config = ALC883_ACER },
{ .modelname = "auto", .config = ALC883_AUTO },
{}
};
@@ -5038,6 +5105,7 @@ static struct alc_config_preset alc883_presets[] = {
.dig_in_nid = ALC883_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
.channel_mode = alc883_3ST_6ch_modes,
+ .need_dac_fix = 1,
.input_mux = &alc883_capture_source,
},
[ALC883_3ST_6ch] = {
@@ -5049,6 +5117,7 @@ static struct alc_config_preset alc883_presets[] = {
.adc_nids = alc883_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
.channel_mode = alc883_3ST_6ch_modes,
+ .need_dac_fix = 1,
.input_mux = &alc883_capture_source,
},
[ALC883_6ST_DIG] = {
@@ -5077,6 +5146,23 @@ static struct alc_config_preset alc883_presets[] = {
.channel_mode = alc883_sixstack_modes,
.input_mux = &alc883_capture_source,
},
+ [ALC883_ACER] = {
+ .mixers = { alc883_base_mixer,
+ alc883_chmode_mixer },
+ /* On TravelMate laptops, GPIO 0 enables the internal speaker
+ * and the headphone jack. Turn this on and rely on the
+ * standard mute methods whenever the user wants to turn
+ * these outputs off.
+ */
+ .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,
+ },
};
@@ -5121,7 +5207,7 @@ static void alc883_auto_init_hp_out(struct hda_codec *codec)
struct alc_spec *spec = codec->spec;
hda_nid_t pin;
- pin = spec->autocfg.hp_pin;
+ pin = spec->autocfg.hp_pins[0];
if (pin) /* connect to front */
/* use dac 0 */
alc883_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
@@ -5217,8 +5303,10 @@ static int patch_alc883(struct hda_codec *codec)
spec->stream_digital_playback = &alc883_pcm_digital_playback;
spec->stream_digital_capture = &alc883_pcm_digital_capture;
- spec->adc_nids = alc883_adc_nids;
- spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids);
+ if (! spec->adc_nids && spec->input_mux) {
+ spec->adc_nids = alc883_adc_nids;
+ spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids);
+ }
codec->patch_ops = alc_patch_ops;
if (board_config == ALC883_AUTO)
@@ -5481,6 +5569,7 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = {
.info = snd_hda_mixer_amp_volume_info,
.get = snd_hda_mixer_amp_volume_get,
.put = alc262_fujitsu_master_vol_put,
+ .tlv = { .c = snd_hda_mixer_amp_tlv },
.private_value = HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT),
},
{
@@ -5499,6 +5588,13 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = {
{ } /* end */
};
+/* additional init verbs for Benq laptops */
+static struct hda_verb alc262_EAPD_verbs[] = {
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x3070},
+ {}
+};
+
/* 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)
{
@@ -5534,7 +5630,7 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, const struct
return err;
}
}
- nid = cfg->hp_pin;
+ nid = cfg->hp_pins[0];
if (nid) {
/* spec->multiout.hp_nid = 2; */
if (nid == 0x16) {
@@ -5769,6 +5865,7 @@ static struct hda_board_config alc262_cfg_tbl[] = {
{ .modelname = "fujitsu", .config = ALC262_FUJITSU },
{ .pci_subvendor = 0x10cf, .pci_subdevice = 0x1397,
.config = ALC262_FUJITSU },
+ { .modelname = "hp-bpc", .config = ALC262_HP_BPC },
{ .pci_subvendor = 0x103c, .pci_subdevice = 0x208c,
.config = ALC262_HP_BPC }, /* xw4400 */
{ .pci_subvendor = 0x103c, .pci_subdevice = 0x3014,
@@ -5777,6 +5874,9 @@ static struct hda_board_config alc262_cfg_tbl[] = {
.config = ALC262_HP_BPC }, /* xw8400 */
{ .pci_subvendor = 0x103c, .pci_subdevice = 0x12fe,
.config = ALC262_HP_BPC }, /* xw9400 */
+ { .modelname = "benq", .config = ALC262_BENQ_ED8 },
+ { .pci_subvendor = 0x17ff, .pci_subdevice = 0x0560,
+ .config = ALC262_BENQ_ED8 },
{ .modelname = "auto", .config = ALC262_AUTO },
{}
};
@@ -5814,6 +5914,16 @@ static struct alc_config_preset alc262_presets[] = {
.channel_mode = alc262_modes,
.input_mux = &alc262_HP_capture_source,
},
+ [ALC262_BENQ_ED8] = {
+ .mixers = { alc262_base_mixer },
+ .init_verbs = { alc262_init_verbs, alc262_EAPD_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_capture_source,
+ },
};
static int patch_alc262(struct hda_codec *codec)
@@ -5942,6 +6052,23 @@ static struct hda_channel_mode alc861_threestack_modes[2] = {
{ 2, alc861_threestack_ch2_init },
{ 6, alc861_threestack_ch6_init },
};
+/* Set mic1 as input and unmute the mixer */
+static struct hda_verb alc861_uniwill_m31_ch2_init[] = {
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/
+ { } /* end */
+};
+/* Set mic1 as output and mute mixer */
+static struct hda_verb alc861_uniwill_m31_ch4_init[] = {
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
+ { } /* end */
+};
+
+static struct hda_channel_mode alc861_uniwill_m31_modes[2] = {
+ { 2, alc861_uniwill_m31_ch2_init },
+ { 4, alc861_uniwill_m31_ch4_init },
+};
/* patch-ALC861 */
@@ -6020,6 +6147,47 @@ static struct snd_kcontrol_new alc861_3ST_mixer[] = {
},
{ } /* end */
};
+static struct snd_kcontrol_new alc861_uniwill_m31_mixer[] = {
+ /* output mixer control */
+ HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
+ /*HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT), */
+
+ /* Input mixer control */
+ /* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT),
+
+ /* Capture mixer control */
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .count = 1,
+ .info = alc_mux_enum_info,
+ .get = alc_mux_enum_get,
+ .put = alc_mux_enum_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc_ch_mode_info,
+ .get = alc_ch_mode_get,
+ .put = alc_ch_mode_put,
+ .private_value = ARRAY_SIZE(alc861_uniwill_m31_modes),
+ },
+ { } /* end */
+};
/*
* generic initialization of ADC, input mixers and output mixers
@@ -6148,6 +6316,67 @@ static struct hda_verb alc861_threestack_init_verbs[] = {
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{ }
};
+
+static struct hda_verb alc861_uniwill_m31_init_verbs[] = {
+ /*
+ * Unmute ADC0 and set the default input to mic-in
+ */
+ /* port-A for surround (rear panel) */
+ { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+ /* port-B for mic-in (rear panel) with vref */
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ /* port-C for line-in (rear panel) */
+ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ /* port-D for Front */
+ { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ /* port-E for HP out (front panel) */
+ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, // this has to be set to VREF80
+ /* route front PCM to HP */
+ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ /* port-F for mic-in (front panel) with vref */
+ { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ /* port-G for CLFE (rear panel) */
+ { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+ /* port-H for side (rear panel) */
+ { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+ /* CD-in */
+ { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ /* route front mic to ADC1*/
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ /* Unmute DAC0~3 & spdif out*/
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* Unmute Mixer 14 (mic) 1c (Line in)*/
+ {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ /* Unmute Stereo Mixer 15 */
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c }, //Output 0~12 step
+
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, // hp used DAC 3 (Front)
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ { }
+};
+
/*
* generic initialization of ADC, input mixers and output mixers
*/
@@ -6401,7 +6630,7 @@ static void alc861_auto_init_hp_out(struct hda_codec *codec)
struct alc_spec *spec = codec->spec;
hda_nid_t pin;
- pin = spec->autocfg.hp_pin;
+ pin = spec->autocfg.hp_pins[0];
if (pin) /* connect to front */
alc861_auto_set_output_and_unmute(codec, pin, PIN_HP, spec->multiout.dac_nids[0]);
}
@@ -6436,7 +6665,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec)
if ((err = alc861_auto_fill_dac_nids(spec, &spec->autocfg)) < 0 ||
(err = alc861_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 ||
- (err = alc861_auto_create_hp_ctls(spec, spec->autocfg.hp_pin)) < 0 ||
+ (err = alc861_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0])) < 0 ||
(err = alc861_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0)
return err;
@@ -6477,10 +6706,14 @@ static struct hda_board_config alc861_cfg_tbl[] = {
{ .modelname = "3stack", .config = ALC861_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xd600,
.config = ALC861_3ST },
+ { .modelname = "3stack-660", .config = ALC660_3ST },
{ .pci_subvendor = 0x1043, .pci_subdevice = 0x81e7,
.config = ALC660_3ST },
{ .modelname = "3stack-dig", .config = ALC861_3ST_DIG },
{ .modelname = "6stack-dig", .config = ALC861_6ST_DIG },
+ { .modelname = "uniwill-m31", .config = ALC861_UNIWILL_M31},
+ { .pci_subvendor = 0x1584, .pci_subdevice = 0x9072,
+ .config = ALC861_UNIWILL_M31 },
{ .modelname = "auto", .config = ALC861_AUTO },
{}
};
@@ -6493,6 +6726,7 @@ static struct alc_config_preset alc861_presets[] = {
.dac_nids = alc861_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
.channel_mode = alc861_threestack_modes,
+ .need_dac_fix = 1,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
@@ -6505,6 +6739,7 @@ static struct alc_config_preset alc861_presets[] = {
.dig_out_nid = ALC861_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
.channel_mode = alc861_threestack_modes,
+ .need_dac_fix = 1,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
@@ -6528,10 +6763,25 @@ static struct alc_config_preset alc861_presets[] = {
.dac_nids = alc660_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
.channel_mode = alc861_threestack_modes,
+ .need_dac_fix = 1,
+ .num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
+ .adc_nids = alc861_adc_nids,
+ .input_mux = &alc861_capture_source,
+ },
+ [ALC861_UNIWILL_M31] = {
+ .mixers = { alc861_uniwill_m31_mixer },
+ .init_verbs = { alc861_uniwill_m31_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc861_dac_nids),
+ .dac_nids = alc861_dac_nids,
+ .dig_out_nid = ALC861_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc861_uniwill_m31_modes),
+ .channel_mode = alc861_uniwill_m31_modes,
+ .need_dac_fix = 1,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
},
+
};
diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c
index 250242cd6c7..76ec3d75fa9 100644
--- a/sound/pci/hda/patch_si3054.c
+++ b/sound/pci/hda/patch_si3054.c
@@ -298,6 +298,7 @@ struct hda_codec_preset snd_hda_preset_si3054[] = {
{ .id = 0x163c3055, .name = "Si3054", .patch = patch_si3054 },
{ .id = 0x163c3155, .name = "Si3054", .patch = patch_si3054 },
{ .id = 0x11c13026, .name = "Si3054", .patch = patch_si3054 },
+ { .id = 0x10573057, .name = "Si3054", .patch = patch_si3054 },
{}
};
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index ea99083a102..731b7b97ee7 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -36,15 +36,15 @@
#define NUM_CONTROL_ALLOC 32
#define STAC_HP_EVENT 0x37
-#define STAC_UNSOL_ENABLE (AC_USRSP_EN | STAC_HP_EVENT)
#define STAC_REF 0
#define STAC_D945GTP3 1
#define STAC_D945GTP5 2
#define STAC_MACMINI 3
-#define STAC_D965_2112 4
-#define STAC_D965_284B 5
-#define STAC_922X_MODELS 6 /* number of 922x models */
+#define STAC_922X_MODELS 4 /* number of 922x models */
+#define STAC_D965_3ST 4
+#define STAC_D965_5ST 5
+#define STAC_927X_MODELS 6 /* number of 922x models */
struct sigmatel_spec {
struct snd_kcontrol_new *mixers[4];
@@ -73,6 +73,7 @@ struct sigmatel_spec {
hda_nid_t *pin_nids;
unsigned int num_pins;
unsigned int *pin_configs;
+ unsigned int *bios_pin_configs;
/* codec specific stuff */
struct hda_verb *init;
@@ -110,24 +111,10 @@ static hda_nid_t stac922x_adc_nids[2] = {
0x06, 0x07,
};
-static hda_nid_t stac9227_adc_nids[2] = {
- 0x07, 0x08,
-};
-
-#if 0
-static hda_nid_t d965_2112_dac_nids[3] = {
- 0x02, 0x03, 0x05,
-};
-#endif
-
static hda_nid_t stac922x_mux_nids[2] = {
0x12, 0x13,
};
-static hda_nid_t stac9227_mux_nids[2] = {
- 0x15, 0x16,
-};
-
static hda_nid_t stac927x_adc_nids[3] = {
0x07, 0x08, 0x09
};
@@ -136,8 +123,17 @@ static hda_nid_t stac927x_mux_nids[3] = {
0x15, 0x16, 0x17
};
+static hda_nid_t stac9205_adc_nids[2] = {
+ 0x12, 0x13
+};
+
+static hda_nid_t stac9205_mux_nids[2] = {
+ 0x19, 0x1a
+};
+
static hda_nid_t stac9200_pin_nids[8] = {
- 0x08, 0x09, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x12,
+ 0x08, 0x09, 0x0d, 0x0e,
+ 0x0f, 0x10, 0x11, 0x12,
};
static hda_nid_t stac922x_pin_nids[10] = {
@@ -151,6 +147,13 @@ static hda_nid_t stac927x_pin_nids[14] = {
0x14, 0x21, 0x22, 0x23,
};
+static hda_nid_t stac9205_pin_nids[12] = {
+ 0x0a, 0x0b, 0x0c, 0x0d, 0x0e,
+ 0x0f, 0x14, 0x16, 0x17, 0x18,
+ 0x21, 0x22,
+
+};
+
static int stac92xx_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
@@ -190,25 +193,23 @@ static struct hda_verb stac922x_core_init[] = {
{}
};
-static struct hda_verb stac9227_core_init[] = {
+static struct hda_verb d965_core_init[] = {
/* set master volume and direct control */
- { 0x16, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
+ { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
/* unmute node 0x1b */
{ 0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+ /* select node 0x03 as DAC */
+ { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x01},
{}
};
-static struct hda_verb d965_2112_core_init[] = {
+static struct hda_verb stac927x_core_init[] = {
/* set master volume and direct control */
- { 0x16, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
- /* unmute node 0x1b */
- { 0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- /* select node 0x03 as DAC */
- { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x01},
+ { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
{}
};
-static struct hda_verb stac927x_core_init[] = {
+static struct hda_verb stac9205_core_init[] = {
/* set master volume and direct control */
{ 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
{}
@@ -277,6 +278,21 @@ static snd_kcontrol_new_t stac927x_mixer[] = {
{ } /* end */
};
+static snd_kcontrol_new_t stac9205_mixer[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Input Source",
+ .count = 1,
+ .info = stac92xx_mux_enum_info,
+ .get = stac92xx_mux_enum_get,
+ .put = stac92xx_mux_enum_put,
+ },
+ HDA_CODEC_VOLUME("InMux Capture Volume", 0x19, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("InVol Capture Volume", 0x1b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("ADCMux Capture Switch", 0x1d, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
static int stac92xx_build_controls(struct hda_codec *codec)
{
struct sigmatel_spec *spec = codec->spec;
@@ -341,38 +357,67 @@ static unsigned int d945gtp5_pin_configs[10] = {
0x02a19320, 0x40000100,
};
-static unsigned int d965_2112_pin_configs[10] = {
- 0x0221401f, 0x40000100, 0x40000100, 0x01014011,
- 0x01a19021, 0x01813024, 0x01452130, 0x40000100,
- 0x02a19320, 0x40000100,
-};
-
static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = {
[STAC_REF] = ref922x_pin_configs,
[STAC_D945GTP3] = d945gtp3_pin_configs,
[STAC_D945GTP5] = d945gtp5_pin_configs,
[STAC_MACMINI] = d945gtp5_pin_configs,
- [STAC_D965_2112] = d965_2112_pin_configs,
};
static struct hda_board_config stac922x_cfg_tbl[] = {
+ { .modelname = "5stack", .config = STAC_D945GTP5 },
+ { .modelname = "3stack", .config = STAC_D945GTP3 },
{ .modelname = "ref",
.pci_subvendor = PCI_VENDOR_ID_INTEL,
.pci_subdevice = 0x2668, /* DFI LanParty */
.config = STAC_REF }, /* SigmaTel reference board */
+ /* Intel 945G based systems */
{ .pci_subvendor = PCI_VENDOR_ID_INTEL,
.pci_subdevice = 0x0101,
.config = STAC_D945GTP3 }, /* Intel D945GTP - 3 Stack */
{ .pci_subvendor = PCI_VENDOR_ID_INTEL,
.pci_subdevice = 0x0202,
- .config = STAC_D945GTP3 }, /* Intel D945GNT - 3 Stack, 9221 A1 */
+ .config = STAC_D945GTP3 }, /* Intel D945GNT - 3 Stack */
{ .pci_subvendor = PCI_VENDOR_ID_INTEL,
- .pci_subdevice = 0x0b0b,
- .config = STAC_D945GTP3 }, /* Intel D945PSN - 3 Stack, 9221 A1 */
+ .pci_subdevice = 0x0606,
+ .config = STAC_D945GTP3 }, /* Intel D945GTP - 3 Stack */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x0601,
+ .config = STAC_D945GTP3 }, /* Intel D945GTP - 3 Stack */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x0111,
+ .config = STAC_D945GTP3 }, /* Intel D945GZP - 3 Stack */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x1115,
+ .config = STAC_D945GTP3 }, /* Intel D945GPM - 3 Stack */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x1116,
+ .config = STAC_D945GTP3 }, /* Intel D945GBO - 3 Stack */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x1117,
+ .config = STAC_D945GTP3 }, /* Intel D945GPM - 3 Stack */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x1118,
+ .config = STAC_D945GTP3 }, /* Intel D945GPM - 3 Stack */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x1119,
+ .config = STAC_D945GTP3 }, /* Intel D945GPM - 3 Stack */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x8826,
+ .config = STAC_D945GTP3 }, /* Intel D945GPM - 3 Stack */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x5049,
+ .config = STAC_D945GTP3 }, /* Intel D945GCZ - 3 Stack */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x5055,
+ .config = STAC_D945GTP3 }, /* Intel D945GCZ - 3 Stack */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x5048,
+ .config = STAC_D945GTP3 }, /* Intel D945GPB - 3 Stack */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x0110,
+ .config = STAC_D945GTP3 }, /* Intel D945GLR - 3 Stack */
{ .pci_subvendor = PCI_VENDOR_ID_INTEL,
- .pci_subdevice = 0x0707,
- .config = STAC_D945GTP5 }, /* Intel D945PSV - 5 Stack */
- { .pci_subvendor = PCI_VENDOR_ID_INTEL,
.pci_subdevice = 0x0404,
.config = STAC_D945GTP5 }, /* Intel D945GTP - 5 Stack */
{ .pci_subvendor = PCI_VENDOR_ID_INTEL,
@@ -384,44 +429,214 @@ static struct hda_board_config stac922x_cfg_tbl[] = {
{ .pci_subvendor = PCI_VENDOR_ID_INTEL,
.pci_subdevice = 0x0417,
.config = STAC_D945GTP5 }, /* Intel D975XBK - 5 Stack */
+ /* Intel 945P based systems */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x0b0b,
+ .config = STAC_D945GTP3 }, /* Intel D945PSN - 3 Stack */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x0112,
+ .config = STAC_D945GTP3 }, /* Intel D945PLN - 3 Stack */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x0d0d,
+ .config = STAC_D945GTP3 }, /* Intel D945PLM - 3 Stack */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x0909,
+ .config = STAC_D945GTP3 }, /* Intel D945PAW - 3 Stack */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x0505,
+ .config = STAC_D945GTP3 }, /* Intel D945PLM - 3 Stack */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x0707,
+ .config = STAC_D945GTP5 }, /* Intel D945PSV - 5 Stack */
+ /* other systems */
{ .pci_subvendor = 0x8384,
.pci_subdevice = 0x7680,
.config = STAC_MACMINI }, /* Apple Mac Mini (early 2006) */
+ {} /* terminator */
+};
+
+static unsigned int ref927x_pin_configs[14] = {
+ 0x02214020, 0x02a19080, 0x0181304e, 0x01014010,
+ 0x01a19040, 0x01011012, 0x01016011, 0x0101201f,
+ 0x183301f0, 0x18a001f0, 0x18a001f0, 0x01442070,
+ 0x01c42190, 0x40000100,
+};
+
+static unsigned int d965_3st_pin_configs[14] = {
+ 0x0221401f, 0x02a19120, 0x40000100, 0x01014011,
+ 0x01a19021, 0x01813024, 0x40000100, 0x40000100,
+ 0x40000100, 0x40000100, 0x40000100, 0x40000100,
+ 0x40000100, 0x40000100
+};
+
+static unsigned int d965_5st_pin_configs[14] = {
+ 0x02214020, 0x02a19080, 0x0181304e, 0x01014010,
+ 0x01a19040, 0x01011012, 0x01016011, 0x40000100,
+ 0x40000100, 0x40000100, 0x40000100, 0x01442070,
+ 0x40000100, 0x40000100
+};
+
+static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = {
+ [STAC_REF] = ref927x_pin_configs,
+ [STAC_D965_3ST] = d965_3st_pin_configs,
+ [STAC_D965_5ST] = d965_5st_pin_configs,
+};
+
+static struct hda_board_config stac927x_cfg_tbl[] = {
+ { .modelname = "5stack", .config = STAC_D965_5ST },
+ { .modelname = "3stack", .config = STAC_D965_3ST },
+ { .modelname = "ref",
+ .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x2668, /* DFI LanParty */
+ .config = STAC_REF }, /* SigmaTel reference board */
+ /* Intel 946 based systems */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x3d01,
+ .config = STAC_D965_3ST }, /* D946 configuration */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0xa301,
+ .config = STAC_D965_3ST }, /* Intel D946GZT - 3 stack */
+ /* 965 based 3 stack systems */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x2116,
+ .config = STAC_D965_3ST }, /* Intel D965 3Stack config */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x2115,
+ .config = STAC_D965_3ST }, /* Intel DQ965WC - 3 Stack */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x2114,
+ .config = STAC_D965_3ST }, /* Intel D965 3Stack config */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x2113,
+ .config = STAC_D965_3ST }, /* Intel D965 3Stack config */
{ .pci_subvendor = PCI_VENDOR_ID_INTEL,
.pci_subdevice = 0x2112,
- .config = STAC_D965_2112 },
+ .config = STAC_D965_3ST }, /* Intel DG965MS - 3 Stack */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x2111,
+ .config = STAC_D965_3ST }, /* Intel D965 3Stack config */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x2110,
+ .config = STAC_D965_3ST }, /* Intel D965 3Stack config */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x2009,
+ .config = STAC_D965_3ST }, /* Intel D965 3Stack config */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x2008,
+ .config = STAC_D965_3ST }, /* Intel DQ965GF - 3 Stack */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x2007,
+ .config = STAC_D965_3ST }, /* Intel D965 3Stack config */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x2006,
+ .config = STAC_D965_3ST }, /* Intel D965 3Stack config */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x2005,
+ .config = STAC_D965_3ST }, /* Intel D965 3Stack config */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x2004,
+ .config = STAC_D965_3ST }, /* Intel D965 3Stack config */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x2003,
+ .config = STAC_D965_3ST }, /* Intel D965 3Stack config */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x2002,
+ .config = STAC_D965_3ST }, /* Intel D965 3Stack config */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x2001,
+ .config = STAC_D965_3ST }, /* Intel DQ965GF - 3 Stack */
+ /* 965 based 5 stack systems */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x2301,
+ .config = STAC_D965_5ST }, /* Intel DG965 - 5 Stack */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x2302,
+ .config = STAC_D965_5ST }, /* Intel DG965 - 5 Stack */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x2303,
+ .config = STAC_D965_5ST }, /* Intel DG965 - 5 Stack */
{ .pci_subvendor = PCI_VENDOR_ID_INTEL,
- .pci_subdevice = 0x284b,
- .config = STAC_D965_284B },
+ .pci_subdevice = 0x2304,
+ .config = STAC_D965_5ST }, /* Intel DG965 - 5 Stack */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x2305,
+ .config = STAC_D965_5ST }, /* Intel DG965 - 5 Stack */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x2501,
+ .config = STAC_D965_5ST }, /* Intel DG965MQ - 5 Stack */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x2502,
+ .config = STAC_D965_5ST }, /* Intel DG965 - 5 Stack */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x2503,
+ .config = STAC_D965_5ST }, /* Intel DG965 - 5 Stack */
+ { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+ .pci_subdevice = 0x2504,
+ .config = STAC_D965_5ST }, /* Intel DQ965GF - 5 Stack */
{} /* terminator */
};
-static unsigned int ref927x_pin_configs[14] = {
- 0x01813122, 0x01a19021, 0x01014010, 0x01016011,
- 0x01012012, 0x01011014, 0x40000100, 0x40000100,
- 0x40000100, 0x40000100, 0x40000100, 0x01441030,
- 0x01c41030, 0x40000100,
+static unsigned int ref9205_pin_configs[12] = {
+ 0x40000100, 0x40000100, 0x01016011, 0x01014010,
+ 0x01813122, 0x01a19021, 0x40000100, 0x40000100,
+ 0x40000100, 0x40000100, 0x01441030, 0x01c41030
};
-static unsigned int *stac927x_brd_tbl[] = {
- ref927x_pin_configs,
+static unsigned int *stac9205_brd_tbl[] = {
+ ref9205_pin_configs,
};
-static struct hda_board_config stac927x_cfg_tbl[] = {
+static struct hda_board_config stac9205_cfg_tbl[] = {
{ .modelname = "ref",
.pci_subvendor = PCI_VENDOR_ID_INTEL,
.pci_subdevice = 0x2668, /* DFI LanParty */
.config = STAC_REF }, /* SigmaTel reference board */
+ /* Dell laptops have BIOS problem */
+ { .pci_subvendor = PCI_VENDOR_ID_DELL, .pci_subdevice = 0x01b5,
+ .config = STAC_REF }, /* Dell Inspiron 630m */
+ { .pci_subvendor = PCI_VENDOR_ID_DELL, .pci_subdevice = 0x01c2,
+ .config = STAC_REF }, /* Dell Latitude D620 */
+ { .pci_subvendor = PCI_VENDOR_ID_DELL, .pci_subdevice = 0x01cb,
+ .config = STAC_REF }, /* Dell Latitude 120L */
{} /* terminator */
};
+static int stac92xx_save_bios_config_regs(struct hda_codec *codec)
+{
+ int i;
+ struct sigmatel_spec *spec = codec->spec;
+
+ if (! spec->bios_pin_configs) {
+ spec->bios_pin_configs = kcalloc(spec->num_pins,
+ sizeof(*spec->bios_pin_configs), GFP_KERNEL);
+ if (! spec->bios_pin_configs)
+ return -ENOMEM;
+ }
+
+ for (i = 0; i < spec->num_pins; i++) {
+ hda_nid_t nid = spec->pin_nids[i];
+ unsigned int pin_cfg;
+
+ pin_cfg = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_CONFIG_DEFAULT, 0x00);
+ snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x bios pin config %8.8x\n",
+ nid, pin_cfg);
+ spec->bios_pin_configs[i] = pin_cfg;
+ }
+
+ return 0;
+}
+
static void stac92xx_set_config_regs(struct hda_codec *codec)
{
int i;
struct sigmatel_spec *spec = codec->spec;
unsigned int pin_cfg;
- for (i=0; i < spec->num_pins; i++) {
+ if (! spec->pin_nids || ! spec->pin_configs)
+ return;
+
+ for (i = 0; i < spec->num_pins; i++) {
snd_hda_codec_write(codec, spec->pin_nids[i], 0,
AC_VERB_SET_CONFIG_DEFAULT_BYTES_0,
spec->pin_configs[i] & 0x000000ff);
@@ -795,11 +1010,29 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec,
return 0;
}
+/* create volume control/switch for the given prefx type */
+static int create_controls(struct sigmatel_spec *spec, const char *pfx, hda_nid_t nid, int chs)
+{
+ char name[32];
+ int err;
+
+ sprintf(name, "%s Playback Volume", pfx);
+ err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ sprintf(name, "%s Playback Switch", pfx);
+ err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ return 0;
+}
+
/* add playback controls from the parsed DAC table */
static int stac92xx_auto_create_multi_out_ctls(struct sigmatel_spec *spec,
const struct auto_pin_cfg *cfg)
{
- char name[32];
static const char *chname[4] = {
"Front", "Surround", NULL /*CLFE*/, "Side"
};
@@ -814,26 +1047,15 @@ static int stac92xx_auto_create_multi_out_ctls(struct sigmatel_spec *spec,
if (i == 2) {
/* Center/LFE */
- if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, "Center Playback Volume",
- HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT))) < 0)
- return err;
- if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, "LFE Playback Volume",
- HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT))) < 0)
- return err;
- if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, "Center Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT))) < 0)
+ err = create_controls(spec, "Center", nid, 1);
+ if (err < 0)
return err;
- if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, "LFE Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT))) < 0)
+ err = create_controls(spec, "LFE", nid, 2);
+ if (err < 0)
return err;
} else {
- sprintf(name, "%s Playback Volume", chname[i]);
- if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, name,
- HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0)
- return err;
- sprintf(name, "%s Playback Switch", chname[i]);
- if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, name,
- HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0)
+ err = create_controls(spec, chname[i], nid, 3);
+ if (err < 0)
return err;
}
}
@@ -849,39 +1071,85 @@ static int stac92xx_auto_create_multi_out_ctls(struct sigmatel_spec *spec,
return 0;
}
-/* add playback controls for HP output */
-static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, struct auto_pin_cfg *cfg)
+static int check_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid)
{
- struct sigmatel_spec *spec = codec->spec;
- hda_nid_t pin = cfg->hp_pin;
- hda_nid_t nid;
- int i, err;
- unsigned int wid_caps;
+ int i;
- if (! pin)
- return 0;
+ for (i = 0; i < spec->multiout.num_dacs; i++) {
+ if (spec->multiout.dac_nids[i] == nid)
+ return 1;
+ }
+ if (spec->multiout.hp_nid == nid)
+ return 1;
+ return 0;
+}
- wid_caps = get_wcaps(codec, pin);
- if (wid_caps & AC_WCAP_UNSOL_CAP)
- spec->hp_detect = 1;
+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;
+}
- nid = snd_hda_codec_read(codec, pin, 0, AC_VERB_GET_CONNECT_LIST, 0) & 0xff;
- for (i = 0; i < cfg->line_outs; i++) {
- if (! spec->multiout.dac_nids[i])
+/* add playback controls for Speaker and HP outputs */
+static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec,
+ struct auto_pin_cfg *cfg)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ hda_nid_t nid;
+ int i, old_num_dacs, err;
+
+ old_num_dacs = spec->multiout.num_dacs;
+ for (i = 0; i < cfg->hp_outs; i++) {
+ unsigned int wid_caps = get_wcaps(codec, cfg->hp_pins[i]);
+ if (wid_caps & AC_WCAP_UNSOL_CAP)
+ spec->hp_detect = 1;
+ nid = snd_hda_codec_read(codec, cfg->hp_pins[i], 0,
+ AC_VERB_GET_CONNECT_LIST, 0) & 0xff;
+ if (check_in_dac_nids(spec, nid))
+ nid = 0;
+ if (! nid)
continue;
- if (spec->multiout.dac_nids[i] == nid)
- return 0;
+ add_spec_dacs(spec, nid);
+ }
+ for (i = 0; i < cfg->speaker_outs; i++) {
+ nid = snd_hda_codec_read(codec, cfg->speaker_pins[0], 0,
+ AC_VERB_GET_CONNECT_LIST, 0) & 0xff;
+ if (check_in_dac_nids(spec, nid))
+ nid = 0;
+ if (check_in_dac_nids(spec, nid))
+ nid = 0;
+ if (! nid)
+ continue;
+ add_spec_dacs(spec, nid);
}
- spec->multiout.hp_nid = nid;
-
- /* control HP volume/switch on the output mixer amp */
- if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, "Headphone Playback Volume",
- HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0)
- return err;
- if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, "Headphone Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0)
- return err;
+ for (i = old_num_dacs; i < spec->multiout.num_dacs; i++) {
+ static const char *pfxs[] = {
+ "Speaker", "External Speaker", "Speaker2",
+ };
+ err = create_controls(spec, pfxs[i - old_num_dacs],
+ spec->multiout.dac_nids[i], 3);
+ if (err < 0)
+ 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);
+ if (err < 0)
+ return err;
+ }
return 0;
}
@@ -895,23 +1163,28 @@ static int stac92xx_auto_create_analog_input_ctls(struct hda_codec *codec, const
int i, j, k;
for (i = 0; i < AUTO_PIN_LAST; i++) {
- int index = -1;
- if (cfg->input_pins[i]) {
- imux->items[imux->num_items].label = auto_pin_cfg_labels[i];
-
- for (j=0; j<spec->num_muxes; j++) {
- int num_cons = snd_hda_get_connections(codec, spec->mux_nids[j], con_lst, HDA_MAX_NUM_INPUTS);
- for (k=0; k<num_cons; k++)
- if (con_lst[k] == cfg->input_pins[i]) {
- index = k;
- break;
- }
- if (index >= 0)
- break;
- }
- imux->items[imux->num_items].index = index;
- imux->num_items++;
+ int index;
+
+ if (!cfg->input_pins[i])
+ continue;
+ index = -1;
+ for (j = 0; j < spec->num_muxes; j++) {
+ int num_cons;
+ num_cons = snd_hda_get_connections(codec,
+ spec->mux_nids[j],
+ con_lst,
+ HDA_MAX_NUM_INPUTS);
+ for (k = 0; k < num_cons; k++)
+ if (con_lst[k] == cfg->input_pins[i]) {
+ index = k;
+ goto found;
+ }
}
+ continue;
+ found:
+ imux->items[imux->num_items].label = auto_pin_cfg_labels[i];
+ imux->items[imux->num_items].index = index;
+ imux->num_items++;
}
if (imux->num_items == 1) {
@@ -944,11 +1217,20 @@ static void stac92xx_auto_init_multi_out(struct hda_codec *codec)
static void stac92xx_auto_init_hp_out(struct hda_codec *codec)
{
struct sigmatel_spec *spec = codec->spec;
- hda_nid_t pin;
+ int i;
- pin = spec->autocfg.hp_pin;
- if (pin) /* connect to front */
- stac92xx_auto_set_pinctl(codec, pin, AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN);
+ for (i = 0; i < spec->autocfg.hp_outs; i++) {
+ hda_nid_t pin;
+ pin = spec->autocfg.hp_pins[i];
+ if (pin) /* connect to front */
+ stac92xx_auto_set_pinctl(codec, pin, AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN);
+ }
+ for (i = 0; i < spec->autocfg.speaker_outs; i++) {
+ hda_nid_t pin;
+ pin = spec->autocfg.speaker_pins[i];
+ if (pin) /* connect to front */
+ stac92xx_auto_set_pinctl(codec, pin, AC_PINCTL_OUT_EN);
+ }
}
static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out, hda_nid_t dig_in)
@@ -994,7 +1276,7 @@ static int stac9200_auto_create_hp_ctls(struct hda_codec *codec,
struct auto_pin_cfg *cfg)
{
struct sigmatel_spec *spec = codec->spec;
- hda_nid_t pin = cfg->hp_pin;
+ hda_nid_t pin = cfg->hp_pins[0];
unsigned int wid_caps;
if (! pin)
@@ -1007,6 +1289,57 @@ static int stac9200_auto_create_hp_ctls(struct hda_codec *codec,
return 0;
}
+/* add playback controls for LFE output */
+static int stac9200_auto_create_lfe_ctls(struct hda_codec *codec,
+ struct auto_pin_cfg *cfg)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ int err;
+ hda_nid_t lfe_pin = 0x0;
+ int i;
+
+ /*
+ * search speaker outs and line outs for a mono speaker pin
+ * with an amp. If one is found, add LFE controls
+ * for it.
+ */
+ 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);
+ wcaps &= (AC_WCAP_STEREO | AC_WCAP_OUT_AMP);
+ if (wcaps == AC_WCAP_OUT_AMP)
+ /* found a mono speaker with an amp, must be lfe */
+ lfe_pin = pin;
+ }
+
+ /* if speaker_outs is 0, then speakers may be in line_outs */
+ 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,
+ AC_VERB_GET_CONFIG_DEFAULT,
+ 0x00);
+ if (get_defcfg_device(cfg) == AC_JACK_SPEAKER) {
+ unsigned long 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 */
+ lfe_pin = pin;
+ }
+ }
+ }
+
+ if (lfe_pin) {
+ err = create_controls(spec, "LFE", lfe_pin, 1);
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
static int stac9200_parse_auto_config(struct hda_codec *codec)
{
struct sigmatel_spec *spec = codec->spec;
@@ -1021,6 +1354,9 @@ static int stac9200_parse_auto_config(struct hda_codec *codec)
if ((err = stac9200_auto_create_hp_ctls(codec, &spec->autocfg)) < 0)
return err;
+ if ((err = stac9200_auto_create_lfe_ctls(codec, &spec->autocfg)) < 0)
+ return err;
+
if (spec->autocfg.dig_out_pin)
spec->multiout.dig_out_nid = 0x05;
if (spec->autocfg.dig_in_pin)
@@ -1073,6 +1409,15 @@ static void stac922x_gpio_mute(struct hda_codec *codec, int pin, int muted)
AC_VERB_SET_GPIO_DATA, gpiostate);
}
+static void enable_pin_detect(struct hda_codec *codec, hda_nid_t nid,
+ unsigned int event)
+{
+ if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP)
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_UNSOLICITED_ENABLE,
+ (AC_USRSP_EN | event));
+}
+
static int stac92xx_init(struct hda_codec *codec)
{
struct sigmatel_spec *spec = codec->spec;
@@ -1084,9 +1429,10 @@ static int stac92xx_init(struct hda_codec *codec)
/* set up pins */
if (spec->hp_detect) {
/* Enable unsolicited responses on the HP widget */
- snd_hda_codec_write(codec, cfg->hp_pin, 0,
- AC_VERB_SET_UNSOLICITED_ENABLE,
- STAC_UNSOL_ENABLE);
+ for (i = 0; i < cfg->hp_outs; i++)
+ enable_pin_detect(codec, cfg->hp_pins[i],
+ STAC_HP_EVENT);
+ stac92xx_auto_init_hp_out(codec);
/* fake event to set up pins */
codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26);
} else {
@@ -1131,6 +1477,9 @@ static void stac92xx_free(struct hda_codec *codec)
kfree(spec->kctl_alloc);
}
+ if (spec->bios_pin_configs)
+ kfree(spec->bios_pin_configs);
+
kfree(spec);
}
@@ -1139,6 +1488,8 @@ static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid,
{
unsigned int pin_ctl = snd_hda_codec_read(codec, nid,
0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0x00);
+ if (flag == AC_PINCTL_OUT_EN && (pin_ctl & AC_PINCTL_IN_EN))
+ return;
snd_hda_codec_write(codec, nid, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
pin_ctl | flag);
@@ -1154,33 +1505,57 @@ static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid,
pin_ctl & ~flag);
}
-static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
+static int get_pin_presence(struct hda_codec *codec, hda_nid_t nid)
+{
+ if (!nid)
+ return 0;
+ if (snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0x00)
+ & (1 << 31))
+ return 1;
+ return 0;
+}
+
+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 i, presence;
- if ((res >> 26) != STAC_HP_EVENT)
- return;
-
- presence = snd_hda_codec_read(codec, cfg->hp_pin, 0,
- AC_VERB_GET_PIN_SENSE, 0x00) >> 31;
+ presence = 0;
+ for (i = 0; i < cfg->hp_outs; i++) {
+ presence = get_pin_presence(codec, cfg->hp_pins[i]);
+ if (presence)
+ break;
+ }
if (presence) {
/* disable lineouts, enable hp */
for (i = 0; i < cfg->line_outs; i++)
stac92xx_reset_pinctl(codec, cfg->line_out_pins[i],
AC_PINCTL_OUT_EN);
- stac92xx_set_pinctl(codec, cfg->hp_pin, AC_PINCTL_OUT_EN);
+ for (i = 0; i < cfg->speaker_outs; i++)
+ stac92xx_reset_pinctl(codec, cfg->speaker_pins[i],
+ AC_PINCTL_OUT_EN);
} else {
/* enable lineouts, disable hp */
for (i = 0; i < cfg->line_outs; i++)
stac92xx_set_pinctl(codec, cfg->line_out_pins[i],
AC_PINCTL_OUT_EN);
- stac92xx_reset_pinctl(codec, cfg->hp_pin, AC_PINCTL_OUT_EN);
+ for (i = 0; i < cfg->speaker_outs; i++)
+ stac92xx_set_pinctl(codec, cfg->speaker_pins[i],
+ AC_PINCTL_OUT_EN);
}
}
+static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ switch (res >> 26) {
+ case STAC_HP_EVENT:
+ stac92xx_hp_detect(codec, res);
+ break;
+ }
+}
+
#ifdef CONFIG_PM
static int stac92xx_resume(struct hda_codec *codec)
{
@@ -1188,6 +1563,7 @@ static int stac92xx_resume(struct hda_codec *codec)
int i;
stac92xx_init(codec);
+ stac92xx_set_config_regs(codec);
for (i = 0; i < spec->num_mixers; i++)
snd_hda_resume_ctls(codec, spec->mixers[i]);
if (spec->multiout.dig_out_nid)
@@ -1220,12 +1596,18 @@ static int patch_stac9200(struct hda_codec *codec)
return -ENOMEM;
codec->spec = spec;
+ spec->num_pins = 8;
+ spec->pin_nids = stac9200_pin_nids;
spec->board_config = snd_hda_check_board_config(codec, stac9200_cfg_tbl);
- if (spec->board_config < 0)
- snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9200, using BIOS defaults\n");
- else {
- spec->num_pins = 8;
- spec->pin_nids = stac9200_pin_nids;
+ if (spec->board_config < 0) {
+ snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9200, using BIOS defaults\n");
+ err = stac92xx_save_bios_config_regs(codec);
+ if (err < 0) {
+ stac92xx_free(codec);
+ return err;
+ }
+ spec->pin_configs = spec->bios_pin_configs;
+ } else {
spec->pin_configs = stac9200_brd_tbl[spec->board_config];
stac92xx_set_config_regs(codec);
}
@@ -1261,13 +1643,19 @@ static int patch_stac922x(struct hda_codec *codec)
return -ENOMEM;
codec->spec = spec;
+ spec->num_pins = 10;
+ spec->pin_nids = stac922x_pin_nids;
spec->board_config = snd_hda_check_board_config(codec, stac922x_cfg_tbl);
- if (spec->board_config < 0)
- snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC922x, "
- "using BIOS defaults\n");
- else if (stac922x_brd_tbl[spec->board_config] != NULL) {
- spec->num_pins = 10;
- spec->pin_nids = stac922x_pin_nids;
+ if (spec->board_config < 0) {
+ snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC922x, "
+ "using BIOS defaults\n");
+ err = stac92xx_save_bios_config_regs(codec);
+ if (err < 0) {
+ stac92xx_free(codec);
+ return err;
+ }
+ spec->pin_configs = spec->bios_pin_configs;
+ } else if (stac922x_brd_tbl[spec->board_config] != NULL) {
spec->pin_configs = stac922x_brd_tbl[spec->board_config];
stac92xx_set_config_regs(codec);
}
@@ -1281,25 +1669,6 @@ static int patch_stac922x(struct hda_codec *codec)
spec->multiout.dac_nids = spec->dac_nids;
- switch (spec->board_config) {
- case STAC_D965_2112:
- spec->adc_nids = stac9227_adc_nids;
- spec->mux_nids = stac9227_mux_nids;
-#if 0
- spec->multiout.dac_nids = d965_2112_dac_nids;
- spec->multiout.num_dacs = ARRAY_SIZE(d965_2112_dac_nids);
-#endif
- spec->init = d965_2112_core_init;
- spec->mixer = stac9227_mixer;
- break;
- case STAC_D965_284B:
- spec->adc_nids = stac9227_adc_nids;
- spec->mux_nids = stac9227_mux_nids;
- spec->init = stac9227_core_init;
- spec->mixer = stac9227_mixer;
- break;
- }
-
err = stac92xx_parse_auto_config(codec, 0x08, 0x09);
if (err < 0) {
stac92xx_free(codec);
@@ -1324,26 +1693,94 @@ static int patch_stac927x(struct hda_codec *codec)
return -ENOMEM;
codec->spec = spec;
+ spec->num_pins = 14;
+ spec->pin_nids = stac927x_pin_nids;
spec->board_config = snd_hda_check_board_config(codec, stac927x_cfg_tbl);
- if (spec->board_config < 0)
+ if (spec->board_config < 0) {
snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC927x, using BIOS defaults\n");
- else {
- spec->num_pins = 14;
- spec->pin_nids = stac927x_pin_nids;
+ err = stac92xx_save_bios_config_regs(codec);
+ if (err < 0) {
+ stac92xx_free(codec);
+ return err;
+ }
+ spec->pin_configs = spec->bios_pin_configs;
+ } else if (stac927x_brd_tbl[spec->board_config] != NULL) {
spec->pin_configs = stac927x_brd_tbl[spec->board_config];
stac92xx_set_config_regs(codec);
}
- spec->adc_nids = stac927x_adc_nids;
- spec->mux_nids = stac927x_mux_nids;
+ switch (spec->board_config) {
+ case STAC_D965_3ST:
+ spec->adc_nids = stac927x_adc_nids;
+ spec->mux_nids = stac927x_mux_nids;
+ spec->num_muxes = 3;
+ spec->init = d965_core_init;
+ spec->mixer = stac9227_mixer;
+ break;
+ case STAC_D965_5ST:
+ spec->adc_nids = stac927x_adc_nids;
+ spec->mux_nids = stac927x_mux_nids;
+ spec->num_muxes = 3;
+ spec->init = d965_core_init;
+ spec->mixer = stac9227_mixer;
+ break;
+ default:
+ spec->adc_nids = stac927x_adc_nids;
+ spec->mux_nids = stac927x_mux_nids;
+ spec->num_muxes = 3;
+ spec->init = stac927x_core_init;
+ spec->mixer = stac927x_mixer;
+ }
+
+ spec->multiout.dac_nids = spec->dac_nids;
+
+ err = stac92xx_parse_auto_config(codec, 0x1e, 0x20);
+ if (err < 0) {
+ stac92xx_free(codec);
+ return err;
+ }
+
+ codec->patch_ops = stac92xx_patch_ops;
+
+ return 0;
+}
+
+static int patch_stac9205(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec;
+ int err;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+ spec->num_pins = 14;
+ spec->pin_nids = stac9205_pin_nids;
+ spec->board_config = snd_hda_check_board_config(codec, stac9205_cfg_tbl);
+ if (spec->board_config < 0) {
+ snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9205, using BIOS defaults\n");
+ err = stac92xx_save_bios_config_regs(codec);
+ if (err < 0) {
+ stac92xx_free(codec);
+ return err;
+ }
+ spec->pin_configs = spec->bios_pin_configs;
+ } else {
+ spec->pin_configs = stac9205_brd_tbl[spec->board_config];
+ stac92xx_set_config_regs(codec);
+ }
+
+ spec->adc_nids = stac9205_adc_nids;
+ spec->mux_nids = stac9205_mux_nids;
spec->num_muxes = 3;
- spec->init = stac927x_core_init;
- spec->mixer = stac927x_mixer;
+ spec->init = stac9205_core_init;
+ spec->mixer = stac9205_mixer;
spec->multiout.dac_nids = spec->dac_nids;
- err = stac92xx_parse_auto_config(codec, 0x1e, 0x20);
+ err = stac92xx_parse_auto_config(codec, 0x1f, 0x20);
if (err < 0) {
stac92xx_free(codec);
return err;
@@ -1355,10 +1792,10 @@ static int patch_stac927x(struct hda_codec *codec)
}
/*
- * STAC 7661(?) hack
+ * STAC9872 hack
*/
-/* static config for Sony VAIO FE550G */
+/* static config for Sony VAIO FE550G and Sony VAIO AR */
static hda_nid_t vaio_dacs[] = { 0x2 };
#define VAIO_HP_DAC 0x5
static hda_nid_t vaio_adcs[] = { 0x8 /*,0x6*/ };
@@ -1389,6 +1826,23 @@ static struct hda_verb vaio_init[] = {
{}
};
+static struct hda_verb vaio_ar_init[] = {
+ {0x0a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP <- 0x2 */
+ {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Speaker <- 0x5 */
+ {0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? (<- 0x2) */
+ {0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */
+/* {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },*/ /* Optical Out */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x2}, /* mic-sel: 0a,0d,14,02 */
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */
+/* {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},*/ /* Optical Out */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* capture sw/vol -> 0x8 */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* CD-in -> 0x6 */
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Mic-in -> 0x9 */
+ {}
+};
+
/* bind volumes of both NID 0x02 and 0x05 */
static int vaio_master_vol_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -1434,6 +1888,38 @@ static struct snd_kcontrol_new vaio_mixer[] = {
.info = snd_hda_mixer_amp_volume_info,
.get = snd_hda_mixer_amp_volume_get,
.put = vaio_master_vol_put,
+ .tlv = { .c = snd_hda_mixer_amp_tlv },
+ .private_value = HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = snd_hda_mixer_amp_switch_info,
+ .get = snd_hda_mixer_amp_switch_get,
+ .put = vaio_master_sw_put,
+ .private_value = HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
+ },
+ /* HDA_CODEC_VOLUME("CD Capture Volume", 0x07, 0, HDA_INPUT), */
+ HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .count = 1,
+ .info = stac92xx_mux_enum_info,
+ .get = stac92xx_mux_enum_get,
+ .put = stac92xx_mux_enum_put,
+ },
+ {}
+};
+
+static struct snd_kcontrol_new vaio_ar_mixer[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Volume",
+ .info = snd_hda_mixer_amp_volume_info,
+ .get = snd_hda_mixer_amp_volume_get,
+ .put = vaio_master_vol_put,
.private_value = HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
},
{
@@ -1447,6 +1933,8 @@ static struct snd_kcontrol_new vaio_mixer[] = {
/* HDA_CODEC_VOLUME("CD Capture Volume", 0x07, 0, HDA_INPUT), */
HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT),
+ /*HDA_CODEC_MUTE("Optical Out Switch", 0x10, 0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Optical Out Volume", 0x10, 0, HDA_OUTPUT),*/
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Capture Source",
@@ -1458,7 +1946,7 @@ static struct snd_kcontrol_new vaio_mixer[] = {
{}
};
-static struct hda_codec_ops stac7661_patch_ops = {
+static struct hda_codec_ops stac9872_patch_ops = {
.build_controls = stac92xx_build_controls,
.build_pcms = stac92xx_build_pcms,
.init = stac92xx_init,
@@ -1468,23 +1956,34 @@ static struct hda_codec_ops stac7661_patch_ops = {
#endif
};
-enum { STAC7661_VAIO };
-
-static struct hda_board_config stac7661_cfg_tbl[] = {
- { .modelname = "vaio", .config = STAC7661_VAIO },
+enum { /* FE and SZ series. id=0x83847661 and subsys=0x104D0700 or 104D1000. */
+ CXD9872RD_VAIO,
+ /* Unknown. id=0x83847662 and subsys=0x104D1200 or 104D1000. */
+ STAC9872AK_VAIO,
+ /* Unknown. id=0x83847661 and subsys=0x104D1200. */
+ STAC9872K_VAIO,
+ /* AR Series. id=0x83847664 and subsys=104D1300 */
+ CXD9872AKD_VAIO
+ };
+
+static struct hda_board_config stac9872_cfg_tbl[] = {
+ { .modelname = "vaio", .config = CXD9872RD_VAIO },
+ { .modelname = "vaio-ar", .config = CXD9872AKD_VAIO },
{ .pci_subvendor = 0x104d, .pci_subdevice = 0x81e6,
- .config = STAC7661_VAIO },
+ .config = CXD9872RD_VAIO },
{ .pci_subvendor = 0x104d, .pci_subdevice = 0x81ef,
- .config = STAC7661_VAIO },
+ .config = CXD9872RD_VAIO },
+ { .pci_subvendor = 0x104d, .pci_subdevice = 0x81fd,
+ .config = CXD9872AKD_VAIO },
{}
};
-static int patch_stac7661(struct hda_codec *codec)
+static int patch_stac9872(struct hda_codec *codec)
{
struct sigmatel_spec *spec;
int board_config;
- board_config = snd_hda_check_board_config(codec, stac7661_cfg_tbl);
+ board_config = snd_hda_check_board_config(codec, stac9872_cfg_tbl);
if (board_config < 0)
/* unknown config, let generic-parser do its job... */
return snd_hda_parse_generic_codec(codec);
@@ -1495,7 +1994,9 @@ static int patch_stac7661(struct hda_codec *codec)
codec->spec = spec;
switch (board_config) {
- case STAC7661_VAIO:
+ case CXD9872RD_VAIO:
+ case STAC9872AK_VAIO:
+ case STAC9872K_VAIO:
spec->mixer = vaio_mixer;
spec->init = vaio_init;
spec->multiout.max_channels = 2;
@@ -1507,9 +2008,22 @@ static int patch_stac7661(struct hda_codec *codec)
spec->input_mux = &vaio_mux;
spec->mux_nids = vaio_mux_nids;
break;
+
+ case CXD9872AKD_VAIO:
+ spec->mixer = vaio_ar_mixer;
+ spec->init = vaio_ar_init;
+ spec->multiout.max_channels = 2;
+ spec->multiout.num_dacs = ARRAY_SIZE(vaio_dacs);
+ spec->multiout.dac_nids = vaio_dacs;
+ spec->multiout.hp_nid = VAIO_HP_DAC;
+ spec->num_adcs = ARRAY_SIZE(vaio_adcs);
+ spec->adc_nids = vaio_adcs;
+ spec->input_mux = &vaio_mux;
+ spec->mux_nids = vaio_mux_nids;
+ break;
}
- codec->patch_ops = stac7661_patch_ops;
+ codec->patch_ops = stac9872_patch_ops;
return 0;
}
@@ -1525,12 +2039,12 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = {
{ .id = 0x83847681, .name = "STAC9220D/9223D A2", .patch = patch_stac922x },
{ .id = 0x83847682, .name = "STAC9221 A2", .patch = patch_stac922x },
{ .id = 0x83847683, .name = "STAC9221D A2", .patch = patch_stac922x },
- { .id = 0x83847618, .name = "STAC9227", .patch = patch_stac922x },
- { .id = 0x83847619, .name = "STAC9227", .patch = patch_stac922x },
- { .id = 0x83847616, .name = "STAC9228", .patch = patch_stac922x },
- { .id = 0x83847617, .name = "STAC9228", .patch = patch_stac922x },
- { .id = 0x83847614, .name = "STAC9229", .patch = patch_stac922x },
- { .id = 0x83847615, .name = "STAC9229", .patch = patch_stac922x },
+ { .id = 0x83847618, .name = "STAC9227", .patch = patch_stac927x },
+ { .id = 0x83847619, .name = "STAC9227", .patch = patch_stac927x },
+ { .id = 0x83847616, .name = "STAC9228", .patch = patch_stac927x },
+ { .id = 0x83847617, .name = "STAC9228", .patch = patch_stac927x },
+ { .id = 0x83847614, .name = "STAC9229", .patch = patch_stac927x },
+ { .id = 0x83847615, .name = "STAC9229", .patch = patch_stac927x },
{ .id = 0x83847620, .name = "STAC9274", .patch = patch_stac927x },
{ .id = 0x83847621, .name = "STAC9274D", .patch = patch_stac927x },
{ .id = 0x83847622, .name = "STAC9273X", .patch = patch_stac927x },
@@ -1541,6 +2055,20 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = {
{ .id = 0x83847627, .name = "STAC9271D", .patch = patch_stac927x },
{ .id = 0x83847628, .name = "STAC9274X5NH", .patch = patch_stac927x },
{ .id = 0x83847629, .name = "STAC9274D5NH", .patch = patch_stac927x },
- { .id = 0x83847661, .name = "STAC7661", .patch = patch_stac7661 },
+ /* The following does not take into account .id=0x83847661 when subsys =
+ * 104D0C00 which is STAC9225s. Because of this, some SZ Notebooks are
+ * currently not fully supported.
+ */
+ { .id = 0x83847661, .name = "CXD9872RD/K", .patch = patch_stac9872 },
+ { .id = 0x83847662, .name = "STAC9872AK", .patch = patch_stac9872 },
+ { .id = 0x83847664, .name = "CXD9872AKD", .patch = patch_stac9872 },
+ { .id = 0x838476a0, .name = "STAC9205", .patch = patch_stac9205 },
+ { .id = 0x838476a1, .name = "STAC9205D", .patch = patch_stac9205 },
+ { .id = 0x838476a2, .name = "STAC9204", .patch = patch_stac9205 },
+ { .id = 0x838476a3, .name = "STAC9204D", .patch = patch_stac9205 },
+ { .id = 0x838476a4, .name = "STAC9255", .patch = patch_stac9205 },
+ { .id = 0x838476a5, .name = "STAC9255D", .patch = patch_stac9205 },
+ { .id = 0x838476a6, .name = "STAC9254", .patch = patch_stac9205 },
+ { .id = 0x838476a7, .name = "STAC9254D", .patch = patch_stac9205 },
{} /* terminator */
};
diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c
index 9492f3d2455..9e76cebd2d2 100644
--- a/sound/pci/ice1712/aureon.c
+++ b/sound/pci/ice1712/aureon.c
@@ -60,6 +60,7 @@
#include "ice1712.h"
#include "envy24ht.h"
#include "aureon.h"
+#include <sound/tlv.h>
/* WM8770 registers */
#define WM_DAC_ATTEN 0x00 /* DAC1-8 analog attenuation */
@@ -660,6 +661,12 @@ static int aureon_ac97_mmute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e
return change;
}
+static DECLARE_TLV_DB_SCALE(db_scale_wm_dac, -12700, 100, 1);
+static DECLARE_TLV_DB_SCALE(db_scale_wm_pcm, -6400, 50, 1);
+static DECLARE_TLV_DB_SCALE(db_scale_wm_adc, -1200, 100, 0);
+static DECLARE_TLV_DB_SCALE(db_scale_ac97_master, -4650, 150, 0);
+static DECLARE_TLV_DB_SCALE(db_scale_ac97_gain, -3450, 150, 0);
+
/*
* Logarithmic volume values for WM8770
* Computed as 20 * Log10(255 / x)
@@ -1409,10 +1416,13 @@ static struct snd_kcontrol_new aureon_dac_controls[] __devinitdata = {
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "Master Playback Volume",
.info = wm_master_vol_info,
.get = wm_master_vol_get,
- .put = wm_master_vol_put
+ .put = wm_master_vol_put,
+ .tlv = { .p = db_scale_wm_dac }
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1424,11 +1434,14 @@ static struct snd_kcontrol_new aureon_dac_controls[] __devinitdata = {
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "Front Playback Volume",
.info = wm_vol_info,
.get = wm_vol_get,
.put = wm_vol_put,
- .private_value = (2 << 8) | 0
+ .private_value = (2 << 8) | 0,
+ .tlv = { .p = db_scale_wm_dac }
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1440,11 +1453,14 @@ static struct snd_kcontrol_new aureon_dac_controls[] __devinitdata = {
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "Rear Playback Volume",
.info = wm_vol_info,
.get = wm_vol_get,
.put = wm_vol_put,
- .private_value = (2 << 8) | 2
+ .private_value = (2 << 8) | 2,
+ .tlv = { .p = db_scale_wm_dac }
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1456,11 +1472,14 @@ static struct snd_kcontrol_new aureon_dac_controls[] __devinitdata = {
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "Center Playback Volume",
.info = wm_vol_info,
.get = wm_vol_get,
.put = wm_vol_put,
- .private_value = (1 << 8) | 4
+ .private_value = (1 << 8) | 4,
+ .tlv = { .p = db_scale_wm_dac }
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1472,11 +1491,14 @@ static struct snd_kcontrol_new aureon_dac_controls[] __devinitdata = {
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "LFE Playback Volume",
.info = wm_vol_info,
.get = wm_vol_get,
.put = wm_vol_put,
- .private_value = (1 << 8) | 5
+ .private_value = (1 << 8) | 5,
+ .tlv = { .p = db_scale_wm_dac }
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1488,11 +1510,14 @@ static struct snd_kcontrol_new aureon_dac_controls[] __devinitdata = {
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "Side Playback Volume",
.info = wm_vol_info,
.get = wm_vol_get,
.put = wm_vol_put,
- .private_value = (2 << 8) | 6
+ .private_value = (2 << 8) | 6,
+ .tlv = { .p = db_scale_wm_dac }
}
};
@@ -1506,10 +1531,13 @@ static struct snd_kcontrol_new wm_controls[] __devinitdata = {
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "PCM Playback Volume",
.info = wm_pcm_vol_info,
.get = wm_pcm_vol_get,
- .put = wm_pcm_vol_put
+ .put = wm_pcm_vol_put,
+ .tlv = { .p = db_scale_wm_pcm }
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1520,10 +1548,13 @@ static struct snd_kcontrol_new wm_controls[] __devinitdata = {
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "Capture Volume",
.info = wm_adc_vol_info,
.get = wm_adc_vol_get,
- .put = wm_adc_vol_put
+ .put = wm_adc_vol_put,
+ .tlv = { .p = db_scale_wm_adc }
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1567,11 +1598,14 @@ static struct snd_kcontrol_new ac97_controls[] __devinitdata = {
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "AC97 Playback Volume",
.info = aureon_ac97_vol_info,
.get = aureon_ac97_vol_get,
.put = aureon_ac97_vol_put,
- .private_value = AC97_MASTER|AUREON_AC97_STEREO
+ .private_value = AC97_MASTER|AUREON_AC97_STEREO,
+ .tlv = { .p = db_scale_ac97_master }
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1583,11 +1617,14 @@ static struct snd_kcontrol_new ac97_controls[] __devinitdata = {
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "CD Playback Volume",
.info = aureon_ac97_vol_info,
.get = aureon_ac97_vol_get,
.put = aureon_ac97_vol_put,
- .private_value = AC97_CD|AUREON_AC97_STEREO
+ .private_value = AC97_CD|AUREON_AC97_STEREO,
+ .tlv = { .p = db_scale_ac97_gain }
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1599,11 +1636,14 @@ static struct snd_kcontrol_new ac97_controls[] __devinitdata = {
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "Aux Playback Volume",
.info = aureon_ac97_vol_info,
.get = aureon_ac97_vol_get,
.put = aureon_ac97_vol_put,
- .private_value = AC97_AUX|AUREON_AC97_STEREO
+ .private_value = AC97_AUX|AUREON_AC97_STEREO,
+ .tlv = { .p = db_scale_ac97_gain }
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1615,11 +1655,14 @@ static struct snd_kcontrol_new ac97_controls[] __devinitdata = {
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "Line Playback Volume",
.info = aureon_ac97_vol_info,
.get = aureon_ac97_vol_get,
.put = aureon_ac97_vol_put,
- .private_value = AC97_LINE|AUREON_AC97_STEREO
+ .private_value = AC97_LINE|AUREON_AC97_STEREO,
+ .tlv = { .p = db_scale_ac97_gain }
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1631,11 +1674,14 @@ static struct snd_kcontrol_new ac97_controls[] __devinitdata = {
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "Mic Playback Volume",
.info = aureon_ac97_vol_info,
.get = aureon_ac97_vol_get,
.put = aureon_ac97_vol_put,
- .private_value = AC97_MIC
+ .private_value = AC97_MIC,
+ .tlv = { .p = db_scale_ac97_gain }
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1657,11 +1703,14 @@ static struct snd_kcontrol_new universe_ac97_controls[] __devinitdata = {
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "AC97 Playback Volume",
.info = aureon_ac97_vol_info,
.get = aureon_ac97_vol_get,
.put = aureon_ac97_vol_put,
- .private_value = AC97_MASTER|AUREON_AC97_STEREO
+ .private_value = AC97_MASTER|AUREON_AC97_STEREO,
+ .tlv = { .p = db_scale_ac97_master }
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1673,11 +1722,14 @@ static struct snd_kcontrol_new universe_ac97_controls[] __devinitdata = {
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "CD Playback Volume",
.info = aureon_ac97_vol_info,
.get = aureon_ac97_vol_get,
.put = aureon_ac97_vol_put,
- .private_value = AC97_AUX|AUREON_AC97_STEREO
+ .private_value = AC97_AUX|AUREON_AC97_STEREO,
+ .tlv = { .p = db_scale_ac97_gain }
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1685,15 +1737,18 @@ static struct snd_kcontrol_new universe_ac97_controls[] __devinitdata = {
.info = aureon_ac97_mute_info,
.get = aureon_ac97_mute_get,
.put = aureon_ac97_mute_put,
- .private_value = AC97_CD,
+ .private_value = AC97_CD
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "Phono Playback Volume",
.info = aureon_ac97_vol_info,
.get = aureon_ac97_vol_get,
.put = aureon_ac97_vol_put,
- .private_value = AC97_CD|AUREON_AC97_STEREO
+ .private_value = AC97_CD|AUREON_AC97_STEREO,
+ .tlv = { .p = db_scale_ac97_gain }
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1705,11 +1760,14 @@ static struct snd_kcontrol_new universe_ac97_controls[] __devinitdata = {
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "Line Playback Volume",
.info = aureon_ac97_vol_info,
.get = aureon_ac97_vol_get,
.put = aureon_ac97_vol_put,
- .private_value = AC97_LINE|AUREON_AC97_STEREO
+ .private_value = AC97_LINE|AUREON_AC97_STEREO,
+ .tlv = { .p = db_scale_ac97_gain }
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1721,11 +1779,14 @@ static struct snd_kcontrol_new universe_ac97_controls[] __devinitdata = {
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "Mic Playback Volume",
.info = aureon_ac97_vol_info,
.get = aureon_ac97_vol_get,
.put = aureon_ac97_vol_put,
- .private_value = AC97_MIC
+ .private_value = AC97_MIC,
+ .tlv = { .p = db_scale_ac97_gain }
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1744,11 +1805,14 @@ static struct snd_kcontrol_new universe_ac97_controls[] __devinitdata = {
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "Aux Playback Volume",
.info = aureon_ac97_vol_info,
.get = aureon_ac97_vol_get,
.put = aureon_ac97_vol_put,
- .private_value = AC97_VIDEO|AUREON_AC97_STEREO
+ .private_value = AC97_VIDEO|AUREON_AC97_STEREO,
+ .tlv = { .p = db_scale_ac97_gain }
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index bf20858d9f1..dc69392eafa 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -62,6 +62,7 @@
#include <sound/cs8427.h>
#include <sound/info.h>
#include <sound/initval.h>
+#include <sound/tlv.h>
#include <sound/asoundef.h>
@@ -1377,6 +1378,7 @@ static int snd_ice1712_pro_mixer_volume_put(struct snd_kcontrol *kcontrol, struc
return change;
}
+static DECLARE_TLV_DB_SCALE(db_scale_playback, -14400, 150, 0);
static struct snd_kcontrol_new snd_ice1712_multi_playback_ctrls[] __devinitdata = {
{
@@ -1390,12 +1392,15 @@ static struct snd_kcontrol_new snd_ice1712_multi_playback_ctrls[] __devinitdata
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "Multi Playback Volume",
.info = snd_ice1712_pro_mixer_volume_info,
.get = snd_ice1712_pro_mixer_volume_get,
.put = snd_ice1712_pro_mixer_volume_put,
.private_value = 0,
.count = 10,
+ .tlv = { .p = db_scale_playback }
},
};
@@ -1420,11 +1425,14 @@ static struct snd_kcontrol_new snd_ice1712_multi_capture_spdif_switch __devinitd
static struct snd_kcontrol_new snd_ice1712_multi_capture_analog_volume __devinitdata = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "H/W Multi Capture Volume",
.info = snd_ice1712_pro_mixer_volume_info,
.get = snd_ice1712_pro_mixer_volume_get,
.put = snd_ice1712_pro_mixer_volume_put,
.private_value = 10,
+ .tlv = { .p = db_scale_playback }
};
static struct snd_kcontrol_new snd_ice1712_multi_capture_spdif_volume __devinitdata = {
@@ -1857,7 +1865,7 @@ static int snd_ice1712_pro_internal_clock_put(struct snd_kcontrol *kcontrol,
{
struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
static unsigned int xrate[13] = {
- 8000, 9600, 11025, 12000, 1600, 22050, 24000,
+ 8000, 9600, 11025, 12000, 16000, 22050, 24000,
32000, 44100, 48000, 64000, 88200, 96000
};
unsigned char oval;
@@ -1924,7 +1932,7 @@ static int snd_ice1712_pro_internal_clock_default_get(struct snd_kcontrol *kcont
{
int val;
static unsigned int xrate[13] = {
- 8000, 9600, 11025, 12000, 1600, 22050, 24000,
+ 8000, 9600, 11025, 12000, 16000, 22050, 24000,
32000, 44100, 48000, 64000, 88200, 96000
};
@@ -1941,7 +1949,7 @@ static int snd_ice1712_pro_internal_clock_default_put(struct snd_kcontrol *kcont
struct snd_ctl_elem_value *ucontrol)
{
static unsigned int xrate[13] = {
- 8000, 9600, 11025, 12000, 1600, 22050, 24000,
+ 8000, 9600, 11025, 12000, 16000, 22050, 24000,
32000, 44100, 48000, 64000, 88200, 96000
};
unsigned char oval;
diff --git a/sound/pci/ice1712/phase.c b/sound/pci/ice1712/phase.c
index 502da1c8b5f..e08d73f4ff8 100644
--- a/sound/pci/ice1712/phase.c
+++ b/sound/pci/ice1712/phase.c
@@ -46,6 +46,7 @@
#include "ice1712.h"
#include "envy24ht.h"
#include "phase.h"
+#include <sound/tlv.h>
/* WM8770 registers */
#define WM_DAC_ATTEN 0x00 /* DAC1-8 analog attenuation */
@@ -696,6 +697,9 @@ static int phase28_oversampling_put(struct snd_kcontrol *kcontrol, struct snd_ct
return 0;
}
+static DECLARE_TLV_DB_SCALE(db_scale_wm_dac, -12700, 100, 1);
+static DECLARE_TLV_DB_SCALE(db_scale_wm_pcm, -6400, 50, 1);
+
static struct snd_kcontrol_new phase28_dac_controls[] __devinitdata = {
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -706,10 +710,13 @@ static struct snd_kcontrol_new phase28_dac_controls[] __devinitdata = {
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "Master Playback Volume",
.info = wm_master_vol_info,
.get = wm_master_vol_get,
- .put = wm_master_vol_put
+ .put = wm_master_vol_put,
+ .tlv = { .p = db_scale_wm_dac }
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -721,11 +728,14 @@ static struct snd_kcontrol_new phase28_dac_controls[] __devinitdata = {
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "Front Playback Volume",
.info = wm_vol_info,
.get = wm_vol_get,
.put = wm_vol_put,
- .private_value = (2 << 8) | 0
+ .private_value = (2 << 8) | 0,
+ .tlv = { .p = db_scale_wm_dac }
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -737,11 +747,14 @@ static struct snd_kcontrol_new phase28_dac_controls[] __devinitdata = {
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "Rear Playback Volume",
.info = wm_vol_info,
.get = wm_vol_get,
.put = wm_vol_put,
- .private_value = (2 << 8) | 2
+ .private_value = (2 << 8) | 2,
+ .tlv = { .p = db_scale_wm_dac }
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -753,11 +766,14 @@ static struct snd_kcontrol_new phase28_dac_controls[] __devinitdata = {
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "Center Playback Volume",
.info = wm_vol_info,
.get = wm_vol_get,
.put = wm_vol_put,
- .private_value = (1 << 8) | 4
+ .private_value = (1 << 8) | 4,
+ .tlv = { .p = db_scale_wm_dac }
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -769,11 +785,14 @@ static struct snd_kcontrol_new phase28_dac_controls[] __devinitdata = {
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "LFE Playback Volume",
.info = wm_vol_info,
.get = wm_vol_get,
.put = wm_vol_put,
- .private_value = (1 << 8) | 5
+ .private_value = (1 << 8) | 5,
+ .tlv = { .p = db_scale_wm_dac }
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -785,11 +804,14 @@ static struct snd_kcontrol_new phase28_dac_controls[] __devinitdata = {
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "Side Playback Volume",
.info = wm_vol_info,
.get = wm_vol_get,
.put = wm_vol_put,
- .private_value = (2 << 8) | 6
+ .private_value = (2 << 8) | 6,
+ .tlv = { .p = db_scale_wm_dac }
}
};
@@ -803,10 +825,13 @@ static struct snd_kcontrol_new wm_controls[] __devinitdata = {
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "PCM Playback Volume",
.info = wm_pcm_vol_info,
.get = wm_pcm_vol_get,
- .put = wm_pcm_vol_put
+ .put = wm_pcm_vol_put,
+ .tlv = { .p = db_scale_wm_pcm }
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
diff --git a/sound/pci/ice1712/pontis.c b/sound/pci/ice1712/pontis.c
index 0efcad9260a..6c74c2d2e7f 100644
--- a/sound/pci/ice1712/pontis.c
+++ b/sound/pci/ice1712/pontis.c
@@ -31,6 +31,7 @@
#include <sound/core.h>
#include <sound/info.h>
+#include <sound/tlv.h>
#include "ice1712.h"
#include "envy24ht.h"
@@ -564,6 +565,8 @@ static int pontis_gpio_data_put(struct snd_kcontrol *kcontrol, struct snd_ctl_el
return changed;
}
+static DECLARE_TLV_DB_SCALE(db_scale_volume, -6400, 50, 1);
+
/*
* mixers
*/
@@ -571,17 +574,23 @@ static int pontis_gpio_data_put(struct snd_kcontrol *kcontrol, struct snd_ctl_el
static struct snd_kcontrol_new pontis_controls[] __devinitdata = {
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "PCM Playback Volume",
.info = wm_dac_vol_info,
.get = wm_dac_vol_get,
.put = wm_dac_vol_put,
+ .tlv = { .p = db_scale_volume },
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "Capture Volume",
.info = wm_adc_vol_info,
.get = wm_adc_vol_get,
.put = wm_adc_vol_put,
+ .tlv = { .p = db_scale_volume },
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c
index fdb5cb8fac9..41b2605daa3 100644
--- a/sound/pci/ice1712/prodigy192.c
+++ b/sound/pci/ice1712/prodigy192.c
@@ -35,6 +35,7 @@
#include "envy24ht.h"
#include "prodigy192.h"
#include "stac946x.h"
+#include <sound/tlv.h>
static inline void stac9460_put(struct snd_ice1712 *ice, int reg, unsigned char val)
{
@@ -356,6 +357,9 @@ static int aureon_oversampling_put(struct snd_kcontrol *kcontrol, struct snd_ctl
}
#endif
+static DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0);
+static DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0);
+
/*
* mixers
*/
@@ -368,14 +372,18 @@ static struct snd_kcontrol_new stac_controls[] __devinitdata = {
.get = stac9460_dac_mute_get,
.put = stac9460_dac_mute_put,
.private_value = 1,
+ .tlv = { .p = db_scale_dac }
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "Master Playback Volume",
.info = stac9460_dac_vol_info,
.get = stac9460_dac_vol_get,
.put = stac9460_dac_vol_put,
.private_value = 1,
+ .tlv = { .p = db_scale_dac }
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -387,11 +395,14 @@ static struct snd_kcontrol_new stac_controls[] __devinitdata = {
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "DAC Volume",
.count = 6,
.info = stac9460_dac_vol_info,
.get = stac9460_dac_vol_get,
.put = stac9460_dac_vol_put,
+ .tlv = { .p = db_scale_dac }
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -404,11 +415,14 @@ static struct snd_kcontrol_new stac_controls[] __devinitdata = {
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "ADC Volume",
.count = 1,
.info = stac9460_adc_vol_info,
.get = stac9460_adc_vol_get,
.put = stac9460_adc_vol_put,
+ .tlv = { .p = db_scale_adc }
},
#if 0
{
diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c
index fec9440cb31..bf98ea34feb 100644
--- a/sound/pci/ice1712/revo.c
+++ b/sound/pci/ice1712/revo.c
@@ -87,16 +87,33 @@ static void revo_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate)
* initialize the chips on M-Audio Revolution cards
*/
-static unsigned int revo71_num_stereo_front[] = {2};
-static char *revo71_channel_names_front[] = {"PCM Playback Volume"};
+#define AK_DAC(xname,xch) { .name = xname, .num_channels = xch }
-static unsigned int revo71_num_stereo_surround[] = {1, 1, 2, 2};
-static char *revo71_channel_names_surround[] = {"PCM Center Playback Volume", "PCM LFE Playback Volume",
- "PCM Side Playback Volume", "PCM Rear Playback Volume"};
+static struct snd_akm4xxx_dac_channel revo71_front[] = {
+ AK_DAC("PCM Playback Volume", 2)
+};
+
+static struct snd_akm4xxx_dac_channel revo71_surround[] = {
+ AK_DAC("PCM Center Playback Volume", 1),
+ AK_DAC("PCM LFE Playback Volume", 1),
+ AK_DAC("PCM Side Playback Volume", 2),
+ AK_DAC("PCM Rear Playback Volume", 2),
+};
-static unsigned int revo51_num_stereo[] = {2, 1, 1, 2};
-static char *revo51_channel_names[] = {"PCM Playback Volume", "PCM Center Playback Volume",
- "PCM LFE Playback Volume", "PCM Rear Playback Volume"};
+static struct snd_akm4xxx_dac_channel revo51_dac[] = {
+ AK_DAC("PCM Playback Volume", 2),
+ AK_DAC("PCM Center Playback Volume", 1),
+ AK_DAC("PCM LFE Playback Volume", 1),
+ AK_DAC("PCM Rear Playback Volume", 2),
+};
+
+static struct snd_akm4xxx_adc_channel revo51_adc[] = {
+ {
+ .name = "PCM Capture Volume",
+ .switch_name = "PCM Capture Switch",
+ .num_channels = 2
+ },
+};
static struct snd_akm4xxx akm_revo_front __devinitdata = {
.type = SND_AK4381,
@@ -104,8 +121,7 @@ static struct snd_akm4xxx akm_revo_front __devinitdata = {
.ops = {
.set_rate_val = revo_set_rate_val
},
- .num_stereo = revo71_num_stereo_front,
- .channel_names = revo71_channel_names_front
+ .dac_info = revo71_front,
};
static struct snd_ak4xxx_private akm_revo_front_priv __devinitdata = {
@@ -127,8 +143,7 @@ static struct snd_akm4xxx akm_revo_surround __devinitdata = {
.ops = {
.set_rate_val = revo_set_rate_val
},
- .num_stereo = revo71_num_stereo_surround,
- .channel_names = revo71_channel_names_surround
+ .dac_info = revo71_surround,
};
static struct snd_ak4xxx_private akm_revo_surround_priv __devinitdata = {
@@ -149,8 +164,7 @@ static struct snd_akm4xxx akm_revo51 __devinitdata = {
.ops = {
.set_rate_val = revo_set_rate_val
},
- .num_stereo = revo51_num_stereo,
- .channel_names = revo51_channel_names
+ .dac_info = revo51_dac,
};
static struct snd_ak4xxx_private akm_revo51_priv __devinitdata = {
@@ -159,7 +173,25 @@ static struct snd_ak4xxx_private akm_revo51_priv __devinitdata = {
.data_mask = VT1724_REVO_CDOUT,
.clk_mask = VT1724_REVO_CCLK,
.cs_mask = VT1724_REVO_CS0 | VT1724_REVO_CS1 | VT1724_REVO_CS2,
- .cs_addr = 0,
+ .cs_addr = VT1724_REVO_CS1 | VT1724_REVO_CS2,
+ .cs_none = VT1724_REVO_CS0 | VT1724_REVO_CS1 | VT1724_REVO_CS2,
+ .add_flags = VT1724_REVO_CCLK, /* high at init */
+ .mask_flags = 0,
+};
+
+static struct snd_akm4xxx akm_revo51_adc __devinitdata = {
+ .type = SND_AK5365,
+ .num_adcs = 2,
+ .adc_info = revo51_adc,
+};
+
+static struct snd_ak4xxx_private akm_revo51_adc_priv __devinitdata = {
+ .caddr = 2,
+ .cif = 0,
+ .data_mask = VT1724_REVO_CDOUT,
+ .clk_mask = VT1724_REVO_CCLK,
+ .cs_mask = VT1724_REVO_CS0 | VT1724_REVO_CS1 | VT1724_REVO_CS2,
+ .cs_addr = VT1724_REVO_CS0 | VT1724_REVO_CS2,
.cs_none = VT1724_REVO_CS0 | VT1724_REVO_CS1 | VT1724_REVO_CS2,
.add_flags = VT1724_REVO_CCLK, /* high at init */
.mask_flags = 0,
@@ -202,9 +234,13 @@ static int __devinit revo_init(struct snd_ice1712 *ice)
snd_ice1712_gpio_write_bits(ice, VT1724_REVO_MUTE, VT1724_REVO_MUTE);
break;
case VT1724_SUBDEVICE_REVOLUTION51:
- ice->akm_codecs = 1;
+ ice->akm_codecs = 2;
if ((err = snd_ice1712_akm4xxx_init(ak, &akm_revo51, &akm_revo51_priv, ice)) < 0)
return err;
+ err = snd_ice1712_akm4xxx_init(ak + 1, &akm_revo51_adc,
+ &akm_revo51_adc_priv, ice);
+ if (err < 0)
+ return err;
/* unmute all codecs - needed! */
snd_ice1712_gpio_write_bits(ice, VT1724_REVO_MUTE, VT1724_REVO_MUTE);
break;
diff --git a/sound/pci/ice1712/revo.h b/sound/pci/ice1712/revo.h
index dea52ea219d..efbb86ec328 100644
--- a/sound/pci/ice1712/revo.h
+++ b/sound/pci/ice1712/revo.h
@@ -42,7 +42,7 @@ extern struct snd_ice1712_card_info snd_vt1724_revo_cards[];
#define VT1724_REVO_CCLK 0x02
#define VT1724_REVO_CDIN 0x04 /* not used */
#define VT1724_REVO_CDOUT 0x08
-#define VT1724_REVO_CS0 0x10 /* not used */
+#define VT1724_REVO_CS0 0x10 /* AK5365 chipselect for Rev. 5.1 */
#define VT1724_REVO_CS1 0x20 /* front AKM4381 chipselect */
#define VT1724_REVO_CS2 0x40 /* surround AKM4355 chipselect */
#define VT1724_REVO_MUTE (1<<22) /* 0 = all mute, 1 = normal operation */
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 6874263f168..72dbaedcbdf 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -2251,6 +2251,16 @@ static int snd_intel8x0_ich_chip_init(struct intel8x0 *chip, int probing)
/* ACLink on, 2 channels */
cnt = igetdword(chip, ICHREG(GLOB_CNT));
cnt &= ~(ICH_ACLINK | ICH_PCM_246_MASK);
+#ifdef CONFIG_SND_AC97_POWER_SAVE
+ /* do cold reset - the full ac97 powerdown may leave the controller
+ * in a warm state but actually it cannot communicate with the codec.
+ */
+ iputdword(chip, ICHREG(GLOB_CNT), cnt & ~ICH_AC97COLD);
+ cnt = igetdword(chip, ICHREG(GLOB_CNT));
+ udelay(10);
+ iputdword(chip, ICHREG(GLOB_CNT), cnt | ICH_AC97COLD);
+ msleep(1);
+#else
/* finish cold or do warm reset */
cnt |= (cnt & ICH_AC97COLD) == 0 ? ICH_AC97COLD : ICH_AC97WARM;
iputdword(chip, ICHREG(GLOB_CNT), cnt);
@@ -2265,6 +2275,7 @@ static int snd_intel8x0_ich_chip_init(struct intel8x0 *chip, int probing)
return -EIO;
__ok:
+#endif
if (probing) {
/* wait for any codec ready status.
* Once it becomes ready it should remain ready
@@ -2485,7 +2496,7 @@ static int intel8x0_resume(struct pci_dev *pci)
card->shortname, chip);
chip->irq = pci->irq;
synchronize_irq(chip->irq);
- snd_intel8x0_chip_init(chip, 1);
+ snd_intel8x0_chip_init(chip, 0);
/* re-initialize mixer stuff */
if (chip->device_type == DEVICE_INTEL_ICH4) {
@@ -2615,6 +2626,7 @@ static void __devinit intel8x0_measure_ac97_clock(struct intel8x0 *chip)
/* not 48000Hz, tuning the clock.. */
chip->ac97_bus->clock = (chip->ac97_bus->clock * 48000) / pos;
printk(KERN_INFO "intel8x0: clocking to %d\n", chip->ac97_bus->clock);
+ snd_ac97_update_power(chip->ac97[0], AC97_PCM_FRONT_DAC_RATE, 0);
}
#ifdef CONFIG_PROC_FS
diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c
index 91850281f89..268e2f7241e 100644
--- a/sound/pci/intel8x0m.c
+++ b/sound/pci/intel8x0m.c
@@ -1045,6 +1045,8 @@ static int intel8x0m_suspend(struct pci_dev *pci, pm_message_t state)
for (i = 0; i < chip->pcm_devs; i++)
snd_pcm_suspend_all(chip->pcm[i]);
snd_ac97_suspend(chip->ac97);
+ if (chip->irq >= 0)
+ free_irq(chip->irq, chip);
pci_disable_device(pci);
pci_save_state(pci);
return 0;
@@ -1058,6 +1060,9 @@ static int intel8x0m_resume(struct pci_dev *pci)
pci_restore_state(pci);
pci_enable_device(pci);
pci_set_master(pci);
+ request_irq(pci->irq, snd_intel8x0_interrupt, IRQF_DISABLED|IRQF_SHARED,
+ card->shortname, chip);
+ chip->irq = pci->irq;
snd_intel8x0_chip_init(chip, 0);
snd_ac97_resume(chip->ac97);
diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c
index cc43ecd6790..216aee5f93e 100644
--- a/sound/pci/mixart/mixart.c
+++ b/sound/pci/mixart/mixart.c
@@ -1109,13 +1109,13 @@ static long long snd_mixart_BA0_llseek(struct snd_info_entry *entry,
offset = offset & ~3; /* 4 bytes aligned */
switch(orig) {
- case 0: /* SEEK_SET */
+ case SEEK_SET:
file->f_pos = offset;
break;
- case 1: /* SEEK_CUR */
+ case SEEK_CUR:
file->f_pos += offset;
break;
- case 2: /* SEEK_END, offset is negative */
+ case SEEK_END: /* offset is negative */
file->f_pos = MIXART_BA0_SIZE + offset;
break;
default:
@@ -1135,13 +1135,13 @@ static long long snd_mixart_BA1_llseek(struct snd_info_entry *entry,
offset = offset & ~3; /* 4 bytes aligned */
switch(orig) {
- case 0: /* SEEK_SET */
+ case SEEK_SET:
file->f_pos = offset;
break;
- case 1: /* SEEK_CUR */
+ case SEEK_CUR:
file->f_pos += offset;
break;
- case 2: /* SEEK_END, offset is negative */
+ case SEEK_END: /* offset is negative */
file->f_pos = MIXART_BA1_SIZE + offset;
break;
default:
diff --git a/sound/pci/mixart/mixart_mixer.c b/sound/pci/mixart/mixart_mixer.c
index ed47b732c10..13de0f71d4b 100644
--- a/sound/pci/mixart/mixart_mixer.c
+++ b/sound/pci/mixart/mixart_mixer.c
@@ -31,6 +31,7 @@
#include "mixart_core.h"
#include "mixart_hwdep.h"
#include <sound/control.h>
+#include <sound/tlv.h>
#include "mixart_mixer.h"
static u32 mixart_analog_level[256] = {
@@ -388,12 +389,17 @@ static int mixart_analog_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e
return changed;
}
+static DECLARE_TLV_DB_SCALE(db_scale_analog, -9600, 50, 0);
+
static struct snd_kcontrol_new mixart_control_analog_level = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
/* name will be filled later */
.info = mixart_analog_vol_info,
.get = mixart_analog_vol_get,
.put = mixart_analog_vol_put,
+ .tlv = { .p = db_scale_analog },
};
/* shared */
@@ -866,14 +872,19 @@ static int mixart_pcm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem
return changed;
}
+static DECLARE_TLV_DB_SCALE(db_scale_digital, -10950, 50, 0);
+
static struct snd_kcontrol_new snd_mixart_pcm_vol =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
/* name will be filled later */
/* count will be filled later */
.info = mixart_digital_vol_info, /* shared */
.get = mixart_pcm_vol_get,
.put = mixart_pcm_vol_put,
+ .tlv = { .p = db_scale_digital },
};
@@ -984,10 +995,13 @@ static int mixart_monitor_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_
static struct snd_kcontrol_new mixart_control_monitor_vol = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "Monitoring Volume",
.info = mixart_digital_vol_info, /* shared */
.get = mixart_monitor_vol_get,
.put = mixart_monitor_vol_put,
+ .tlv = { .p = db_scale_digital },
};
/*
diff --git a/sound/pci/pcxhr/pcxhr_mixer.c b/sound/pci/pcxhr/pcxhr_mixer.c
index 94e63a1e90d..b133ad9e095 100644
--- a/sound/pci/pcxhr/pcxhr_mixer.c
+++ b/sound/pci/pcxhr/pcxhr_mixer.c
@@ -31,6 +31,7 @@
#include "pcxhr_hwdep.h"
#include "pcxhr_core.h"
#include <sound/control.h>
+#include <sound/tlv.h>
#include <sound/asoundef.h>
#include "pcxhr_mixer.h"
@@ -43,6 +44,9 @@
#define PCXHR_ANALOG_PLAYBACK_LEVEL_MAX 128 /* 0.0 dB */
#define PCXHR_ANALOG_PLAYBACK_ZERO_LEVEL 104 /* -24.0 dB ( 0.0 dB - fix level +24.0 dB ) */
+static DECLARE_TLV_DB_SCALE(db_scale_analog_capture, -9600, 50, 0);
+static DECLARE_TLV_DB_SCALE(db_scale_analog_playback, -12800, 100, 0);
+
static int pcxhr_update_analog_audio_level(struct snd_pcxhr *chip, int is_capture, int channel)
{
int err, vol;
@@ -130,10 +134,13 @@ static int pcxhr_analog_vol_put(struct snd_kcontrol *kcontrol,
static struct snd_kcontrol_new pcxhr_control_analog_level = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
/* name will be filled later */
.info = pcxhr_analog_vol_info,
.get = pcxhr_analog_vol_get,
.put = pcxhr_analog_vol_put,
+ /* tlv will be filled later */
};
/* shared */
@@ -188,6 +195,7 @@ static struct snd_kcontrol_new pcxhr_control_output_switch = {
#define PCXHR_DIGITAL_LEVEL_MAX 0x1ff /* +18 dB */
#define PCXHR_DIGITAL_ZERO_LEVEL 0x1b7 /* 0 dB */
+static DECLARE_TLV_DB_SCALE(db_scale_digital, -10950, 50, 0);
#define MORE_THAN_ONE_STREAM_LEVEL 0x000001
#define VALID_STREAM_PAN_LEVEL_MASK 0x800000
@@ -343,11 +351,14 @@ static int pcxhr_pcm_vol_put(struct snd_kcontrol *kcontrol,
static struct snd_kcontrol_new snd_pcxhr_pcm_vol =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
/* name will be filled later */
/* count will be filled later */
.info = pcxhr_digital_vol_info, /* shared */
.get = pcxhr_pcm_vol_get,
.put = pcxhr_pcm_vol_put,
+ .tlv = { .p = db_scale_digital },
};
@@ -433,10 +444,13 @@ static int pcxhr_monitor_vol_put(struct snd_kcontrol *kcontrol,
static struct snd_kcontrol_new pcxhr_control_monitor_vol = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "Monitoring Volume",
.info = pcxhr_digital_vol_info, /* shared */
.get = pcxhr_monitor_vol_get,
.put = pcxhr_monitor_vol_put,
+ .tlv = { .p = db_scale_digital },
};
/*
@@ -928,6 +942,7 @@ int pcxhr_create_mixer(struct pcxhr_mgr *mgr)
temp = pcxhr_control_analog_level;
temp.name = "Master Playback Volume";
temp.private_value = 0; /* playback */
+ temp.tlv.p = db_scale_analog_playback;
if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0)
return err;
/* output mute controls */
@@ -963,6 +978,7 @@ int pcxhr_create_mixer(struct pcxhr_mgr *mgr)
temp = pcxhr_control_analog_level;
temp.name = "Master Capture Volume";
temp.private_value = 1; /* capture */
+ temp.tlv.p = db_scale_analog_capture;
if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0)
return err;
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index f435fcd6dca..fe210c85344 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -673,9 +673,13 @@ static struct lbuspath lbus_rec_path = {
#define FIRMWARE_VERSIONS 1
static union firmware_version firmware_versions[] = {
{
- .firmware.ASIC = 3,.firmware.CODEC = 2,
- .firmware.AUXDSP = 3,.firmware.PROG = 773,
- },
+ .firmware = {
+ .ASIC = 3,
+ .CODEC = 2,
+ .AUXDSP = 3,
+ .PROG = 773,
+ },
+ },
};
static u32 atoh(unsigned char *in, unsigned int len)
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index e5a52da77b8..d3e07de433b 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -726,22 +726,36 @@ static int hdsp_get_iobox_version (struct hdsp *hdsp)
}
-static int hdsp_check_for_firmware (struct hdsp *hdsp, int show_err)
+#ifdef HDSP_FW_LOADER
+static int __devinit hdsp_request_fw_loader(struct hdsp *hdsp);
+#endif
+
+static int hdsp_check_for_firmware (struct hdsp *hdsp, int load_on_demand)
{
- if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return 0;
+ if (hdsp->io_type == H9652 || hdsp->io_type == H9632)
+ return 0;
if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) {
- snd_printk(KERN_ERR "Hammerfall-DSP: firmware not present.\n");
hdsp->state &= ~HDSP_FirmwareLoaded;
- if (! show_err)
+ if (! load_on_demand)
return -EIO;
+ snd_printk(KERN_ERR "Hammerfall-DSP: firmware not present.\n");
/* try to load firmware */
- if (hdsp->state & HDSP_FirmwareCached) {
- if (snd_hdsp_load_firmware_from_cache(hdsp) != 0)
- snd_printk(KERN_ERR "Hammerfall-DSP: Firmware loading from cache failed, please upload manually.\n");
- } else {
- snd_printk(KERN_ERR "Hammerfall-DSP: No firmware loaded nor cached, please upload firmware.\n");
+ if (! (hdsp->state & HDSP_FirmwareCached)) {
+#ifdef HDSP_FW_LOADER
+ if (! hdsp_request_fw_loader(hdsp))
+ return 0;
+#endif
+ snd_printk(KERN_ERR
+ "Hammerfall-DSP: No firmware loaded nor "
+ "cached, please upload firmware.\n");
+ return -EIO;
+ }
+ if (snd_hdsp_load_firmware_from_cache(hdsp) != 0) {
+ snd_printk(KERN_ERR
+ "Hammerfall-DSP: Firmware loading from "
+ "cache failed, please upload manually.\n");
+ return -EIO;
}
- return -EIO;
}
return 0;
}
@@ -3181,8 +3195,16 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
return;
}
} else {
- snd_iprintf(buffer, "No firmware loaded nor cached, please upload firmware.\n");
- return;
+ int err = -EINVAL;
+#ifdef HDSP_FW_LOADER
+ err = hdsp_request_fw_loader(hdsp);
+#endif
+ if (err < 0) {
+ snd_iprintf(buffer,
+ "No firmware loaded nor cached, "
+ "please upload firmware.\n");
+ return;
+ }
}
}
@@ -3851,7 +3873,7 @@ static int snd_hdsp_trigger(struct snd_pcm_substream *substream, int cmd)
if (hdsp_check_for_iobox (hdsp))
return -EIO;
- if (hdsp_check_for_firmware(hdsp, 1))
+ if (hdsp_check_for_firmware(hdsp, 0)) /* no auto-loading in trigger */
return -EIO;
spin_lock(&hdsp->lock);
diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c
index 4930cc6b054..ebbe12d78d8 100644
--- a/sound/pci/trident/trident_main.c
+++ b/sound/pci/trident/trident_main.c
@@ -40,6 +40,7 @@
#include <sound/core.h>
#include <sound/info.h>
#include <sound/control.h>
+#include <sound/tlv.h>
#include <sound/trident.h>
#include <sound/asoundef.h>
@@ -2627,6 +2628,8 @@ static int snd_trident_vol_control_get(struct snd_kcontrol *kcontrol,
return 0;
}
+static DECLARE_TLV_DB_SCALE(db_scale_gvol, -6375, 25, 0);
+
static int snd_trident_vol_control_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
@@ -2653,6 +2656,7 @@ static struct snd_kcontrol_new snd_trident_vol_music_control __devinitdata =
.get = snd_trident_vol_control_get,
.put = snd_trident_vol_control_put,
.private_value = 16,
+ .tlv = { .p = db_scale_gvol },
};
static struct snd_kcontrol_new snd_trident_vol_wave_control __devinitdata =
@@ -2663,6 +2667,7 @@ static struct snd_kcontrol_new snd_trident_vol_wave_control __devinitdata =
.get = snd_trident_vol_control_get,
.put = snd_trident_vol_control_put,
.private_value = 0,
+ .tlv = { .p = db_scale_gvol },
};
/*---------------------------------------------------------------------------
@@ -2730,6 +2735,7 @@ static struct snd_kcontrol_new snd_trident_pcm_vol_control __devinitdata =
.info = snd_trident_pcm_vol_control_info,
.get = snd_trident_pcm_vol_control_get,
.put = snd_trident_pcm_vol_control_put,
+ /* FIXME: no tlv yet */
};
/*---------------------------------------------------------------------------
@@ -2839,6 +2845,8 @@ static int snd_trident_pcm_rvol_control_put(struct snd_kcontrol *kcontrol,
return change;
}
+static DECLARE_TLV_DB_SCALE(db_scale_crvol, -3175, 25, 1);
+
static struct snd_kcontrol_new snd_trident_pcm_rvol_control __devinitdata =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -2848,6 +2856,7 @@ static struct snd_kcontrol_new snd_trident_pcm_rvol_control __devinitdata =
.info = snd_trident_pcm_rvol_control_info,
.get = snd_trident_pcm_rvol_control_get,
.put = snd_trident_pcm_rvol_control_put,
+ .tlv = { .p = db_scale_crvol },
};
/*---------------------------------------------------------------------------
@@ -2903,6 +2912,7 @@ static struct snd_kcontrol_new snd_trident_pcm_cvol_control __devinitdata =
.info = snd_trident_pcm_cvol_control_info,
.get = snd_trident_pcm_cvol_control_get,
.put = snd_trident_pcm_cvol_control_put,
+ .tlv = { .p = db_scale_crvol },
};
static void snd_trident_notify_pcm_change1(struct snd_card *card,
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index 08da9234efb..6db3d4cc4d8 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -59,6 +59,7 @@
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/info.h>
+#include <sound/tlv.h>
#include <sound/ac97_codec.h>
#include <sound/mpu401.h>
#include <sound/initval.h>
@@ -1277,7 +1278,18 @@ static int snd_via82xx_pcm_close(struct snd_pcm_substream *substream)
if (! ratep->used)
ratep->rate = 0;
spin_unlock_irq(&ratep->lock);
-
+ if (! ratep->rate) {
+ if (! viadev->direction) {
+ snd_ac97_update_power(chip->ac97,
+ AC97_PCM_FRONT_DAC_RATE, 0);
+ snd_ac97_update_power(chip->ac97,
+ AC97_PCM_SURR_DAC_RATE, 0);
+ snd_ac97_update_power(chip->ac97,
+ AC97_PCM_LFE_DAC_RATE, 0);
+ } else
+ snd_ac97_update_power(chip->ac97,
+ AC97_PCM_LR_ADC_RATE, 0);
+ }
viadev->substream = NULL;
return 0;
}
@@ -1687,21 +1699,29 @@ static int snd_via8233_pcmdxs_volume_put(struct snd_kcontrol *kcontrol,
return change;
}
+static DECLARE_TLV_DB_SCALE(db_scale_dxs, -9450, 150, 1);
+
static struct snd_kcontrol_new snd_via8233_pcmdxs_volume_control __devinitdata = {
.name = "PCM Playback Volume",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.info = snd_via8233_dxs_volume_info,
.get = snd_via8233_pcmdxs_volume_get,
.put = snd_via8233_pcmdxs_volume_put,
+ .tlv = { .p = db_scale_dxs }
};
static struct snd_kcontrol_new snd_via8233_dxs_volume_control __devinitdata = {
.name = "VIA DXS Playback Volume",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.count = 4,
.info = snd_via8233_dxs_volume_info,
.get = snd_via8233_dxs_volume_get,
.put = snd_via8233_dxs_volume_put,
+ .tlv = { .p = db_scale_dxs }
};
/*
@@ -2393,6 +2413,7 @@ static int __devinit check_dxs_list(struct pci_dev *pci, int revision)
{ .subvendor = 0x16f3, .subdevice = 0x6405, .action = VIA_DXS_SRC }, /* Jetway K8M8MS */
{ .subvendor = 0x1734, .subdevice = 0x1078, .action = VIA_DXS_SRC }, /* FSC Amilo L7300 */
{ .subvendor = 0x1734, .subdevice = 0x1093, .action = VIA_DXS_SRC }, /* FSC */
+ { .subvendor = 0x1734, .subdevice = 0x10ab, .action = VIA_DXS_SRC }, /* FSC */
{ .subvendor = 0x1849, .subdevice = 0x3059, .action = VIA_DXS_NO_VRA }, /* ASRock K7VM2 */
{ .subvendor = 0x1849, .subdevice = 0x9739, .action = VIA_DXS_SRC }, /* ASRock mobo(?) */
{ .subvendor = 0x1849, .subdevice = 0x9761, .action = VIA_DXS_SRC }, /* ASRock mobo(?) */
diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c
index 9c03c6b4e49..e7cd8acab59 100644
--- a/sound/pci/vx222/vx222.c
+++ b/sound/pci/vx222/vx222.c
@@ -26,6 +26,7 @@
#include <linux/moduleparam.h>
#include <sound/core.h>
#include <sound/initval.h>
+#include <sound/tlv.h>
#include "vx222.h"
#define CARD_NAME "VX222"
@@ -72,6 +73,9 @@ MODULE_DEVICE_TABLE(pci, snd_vx222_ids);
/*
*/
+static DECLARE_TLV_DB_SCALE(db_scale_old_vol, -11350, 50, 0);
+static DECLARE_TLV_DB_SCALE(db_scale_akm, -7350, 50, 0);
+
static struct snd_vx_hardware vx222_old_hw = {
.name = "VX222/Old",
@@ -81,6 +85,7 @@ static struct snd_vx_hardware vx222_old_hw = {
.num_ins = 1,
.num_outs = 1,
.output_level_max = VX_ANALOG_OUT_LEVEL_MAX,
+ .output_level_db_scale = db_scale_old_vol,
};
static struct snd_vx_hardware vx222_v2_hw = {
@@ -92,6 +97,7 @@ static struct snd_vx_hardware vx222_v2_hw = {
.num_ins = 1,
.num_outs = 1,
.output_level_max = VX2_AKM_LEVEL_MAX,
+ .output_level_db_scale = db_scale_akm,
};
static struct snd_vx_hardware vx222_mic_hw = {
@@ -103,6 +109,7 @@ static struct snd_vx_hardware vx222_mic_hw = {
.num_ins = 1,
.num_outs = 1,
.output_level_max = VX2_AKM_LEVEL_MAX,
+ .output_level_db_scale = db_scale_akm,
};
diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c
index 9b6d345b83a..5e51950e05f 100644
--- a/sound/pci/vx222/vx222_ops.c
+++ b/sound/pci/vx222/vx222_ops.c
@@ -28,6 +28,7 @@
#include <sound/core.h>
#include <sound/control.h>
+#include <sound/tlv.h>
#include <asm/io.h>
#include "vx222.h"
@@ -845,6 +846,8 @@ static void vx2_set_input_level(struct snd_vx222 *chip)
#define MIC_LEVEL_MAX 0xff
+static DECLARE_TLV_DB_SCALE(db_scale_mic, -6450, 50, 0);
+
/*
* controls API for input levels
*/
@@ -922,18 +925,24 @@ static int vx_mic_level_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v
static struct snd_kcontrol_new vx_control_input_level = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "Capture Volume",
.info = vx_input_level_info,
.get = vx_input_level_get,
.put = vx_input_level_put,
+ .tlv = { .p = db_scale_mic },
};
static struct snd_kcontrol_new vx_control_mic_level = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
.name = "Mic Capture Volume",
.info = vx_mic_level_info,
.get = vx_mic_level_get,
.put = vx_mic_level_put,
+ .tlv = { .p = db_scale_mic },
};
/*
diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c
index a55b5fd7da6..24f6fc52f89 100644
--- a/sound/pci/ymfpci/ymfpci_main.c
+++ b/sound/pci/ymfpci/ymfpci_main.c
@@ -36,6 +36,7 @@
#include <sound/core.h>
#include <sound/control.h>
#include <sound/info.h>
+#include <sound/tlv.h>
#include <sound/ymfpci.h>
#include <sound/asoundef.h>
#include <sound/mpu401.h>
@@ -1477,11 +1478,15 @@ static int snd_ymfpci_put_single(struct snd_kcontrol *kcontrol,
return change;
}
+static DECLARE_TLV_DB_LINEAR(db_scale_native, TLV_DB_GAIN_MUTE, 0);
+
#define YMFPCI_DOUBLE(xname, xindex, reg) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \
.info = snd_ymfpci_info_double, \
.get = snd_ymfpci_get_double, .put = snd_ymfpci_put_double, \
- .private_value = reg }
+ .private_value = reg, \
+ .tlv = { .p = db_scale_native } }
static int snd_ymfpci_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
{