Gitweb:
http://git.kernel.org/git/?p=linux/kernel/git/torvalds/linux-2.6.git;a=commit;h=2134ea4f37d36addbe86d4901f6c67a22a5db006
Commit: 2134ea4f37d36addbe86d4901f6c67a22a5db006
Parent: 3b0a5f22d4649433a5842ffc7313803292e95718
Author: Takashi Iwai <[EMAIL PROTECTED]>
AuthorDate: Thu Jan 10 16:53:55 2008 +0100
Committer: Jaroslav Kysela <[EMAIL PROTECTED]>
CommitDate: Thu Jan 31 17:29:54 2008 +0100
[ALSA] hda-codec - Add virtual master controls
Add master controls using vmaster to codecs that have no real hardware
master volume registers.
Signed-off-by: Takashi Iwai <[EMAIL PROTECTED]>
Signed-off-by: Jaroslav Kysela <[EMAIL PROTECTED]>
---
sound/pci/hda/hda_codec.c | 60 +++++++++++++++++++++
sound/pci/hda/hda_local.h | 7 +++
sound/pci/hda/patch_analog.c | 69 +++++++++++++++++++++++++
sound/pci/hda/patch_realtek.c | 111 +++++++++++++++++++++++++++++++++-------
sound/pci/hda/patch_sigmatel.c | 48 +++++++++++++++++
5 files changed, 276 insertions(+), 19 deletions(-)
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index a2b40dc..caacc58 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1012,6 +1012,66 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol,
int op_flag,
return 0;
}
+/*
+ * set (static) TLV for virtual master volume; recalculated as max 0dB
+ */
+void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir,
+ unsigned int *tlv)
+{
+ u32 caps;
+ int nums, step;
+
+ caps = query_amp_caps(codec, nid, dir);
+ nums = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT;
+ step = (caps & AC_AMPCAP_STEP_SIZE) >> AC_AMPCAP_STEP_SIZE_SHIFT;
+ step = (step + 1) * 25;
+ tlv[0] = SNDRV_CTL_TLVT_DB_SCALE;
+ tlv[1] = 2 * sizeof(unsigned int);
+ tlv[2] = -nums * step;
+ tlv[3] = step;
+}
+
+/* find a mixer control element with the given name */
+struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec,
+ const char *name)
+{
+ struct snd_ctl_elem_id id;
+ memset(&id, 0, sizeof(id));
+ id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ strcpy(id.name, name);
+ return snd_ctl_find_id(codec->bus->card, &id);
+}
+
+/* create a virtual master control and add slaves */
+int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
+ unsigned int *tlv, const char **slaves)
+{
+ struct snd_kcontrol *kctl;
+ const char **s;
+ int err;
+
+ kctl = snd_ctl_make_virtual_master(name, tlv);
+ if (!kctl)
+ return -ENOMEM;
+ err = snd_ctl_add(codec->bus->card, kctl);
+ if (err < 0)
+ return err;
+
+ for (s = slaves; *s; s++) {
+ struct snd_kcontrol *sctl;
+
+ sctl = snd_hda_find_mixer_ctl(codec, *s);
+ if (!sctl) {
+ snd_printdd("Cannot find slave %s, skipped\n", *s);
+ continue;
+ }
+ err = snd_ctl_add_slave(kctl, sctl);
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
/* switch */
int snd_hda_mixer_amp_switch_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index e09f41b..ddc61a1 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -90,6 +90,13 @@ int snd_hda_codec_amp_stereo(struct hda_codec *codec,
hda_nid_t nid,
void snd_hda_codec_resume_amp(struct hda_codec *codec);
#endif
+void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir,
+ unsigned int *tlv);
+struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec,
+ const char *name);
+int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
+ unsigned int *tlv, const char **slaves);
+
/* amp value bits */
#define HDA_AMP_MUTE 0x80
#define HDA_AMP_UNMUTE 0x00
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 6664a06..b075540 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -78,6 +78,11 @@ struct ad198x_spec {
#ifdef CONFIG_SND_HDA_POWER_SAVE
struct hda_loopback_check loopback;
#endif
+ /* for virtual master */
+ hda_nid_t vmaster_nid;
+ u32 vmaster_tlv[4];
+ const char **slave_vols;
+ const char **slave_sws;
};
/*
@@ -125,6 +130,28 @@ static int ad198x_init(struct hda_codec *codec)
return 0;
}
+static const char *ad_slave_vols[] = {
+ "Front Playback Volume",
+ "Surround Playback Volume",
+ "Center Playback Volume",
+ "LFE Playback Volume",
+ "Side Playback Volume",
+ "Headphone Playback Volume",
+ "Mono Playback Volume",
+ NULL
+};
+
+static const char *ad_slave_sws[] = {
+ "Front Playback Switch",
+ "Surround Playback Switch",
+ "Center Playback Switch",
+ "LFE Playback Switch",
+ "Side Playback Switch",
+ "Headphone Playback Switch",
+ "Mono Playback Switch",
+ NULL
+};
+
static int ad198x_build_controls(struct hda_codec *codec)
{
struct ad198x_spec *spec = codec->spec;
@@ -146,6 +173,27 @@ static int ad198x_build_controls(struct hda_codec *codec)
if (err < 0)
return err;
}
+
+ /* if we have no master control, let's create it */
+ if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) {
+ snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid,
+ HDA_OUTPUT, spec->vmaster_tlv);
+ err = snd_hda_add_vmaster(codec, "Master Playback Volume",
+ spec->vmaster_tlv,
+ (spec->slave_vols ?
+ spec->slave_vols : ad_slave_vols));
+ if (err < 0)
+ return err;
+ }
+ if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
+ err = snd_hda_add_vmaster(codec, "Master Playback Switch",
+ NULL,
+ (spec->slave_sws ?
+ spec->slave_sws : ad_slave_sws));
+ if (err < 0)
+ return err;
+ }
+
return 0;
}
@@ -899,6 +947,7 @@ static int patch_ad1986a(struct hda_codec *codec)
#ifdef CONFIG_SND_HDA_POWER_SAVE
spec->loopback.amplist = ad1986a_loopbacks;
#endif
+ spec->vmaster_nid = 0x1b;
codec->patch_ops = ad198x_patch_ops;
@@ -1141,6 +1190,7 @@ static int patch_ad1983(struct hda_codec *codec)
#ifdef CONFIG_SND_HDA_POWER_SAVE
spec->loopback.amplist = ad1983_loopbacks;
#endif
+ spec->vmaster_nid = 0x05;
codec->patch_ops = ad198x_patch_ops;
@@ -1537,6 +1587,7 @@ static int patch_ad1981(struct hda_codec *codec)
#ifdef CONFIG_SND_HDA_POWER_SAVE
spec->loopback.amplist = ad1981_loopbacks;
#endif
+ spec->vmaster_nid = 0x05;
codec->patch_ops = ad198x_patch_ops;
@@ -2850,6 +2901,7 @@ static int patch_ad1988(struct hda_codec *codec)
#ifdef CONFIG_SND_HDA_POWER_SAVE
spec->loopback.amplist = ad1988_loopbacks;
#endif
+ spec->vmaster_nid = 0x04;
return 0;
}
@@ -3016,6 +3068,19 @@ static struct hda_amp_list ad1884_loopbacks[] = {
};
#endif
+static const char *ad1884_slave_vols[] = {
+ "PCM Playback Volume",
+ "Mic Playback Volume",
+ "Mono Playback Volume",
+ "Front Mic Playback Volume",
+ "Mic Playback Volume",
+ "CD Playback Volume",
+ "Internal Mic Playback Volume",
+ "Docking Mic Playback Volume"
+ "Beep Playback Volume",
+ NULL
+};
+
static int patch_ad1884(struct hda_codec *codec)
{
struct ad198x_spec *spec;
@@ -3043,6 +3108,9 @@ static int patch_ad1884(struct hda_codec *codec)
#ifdef CONFIG_SND_HDA_POWER_SAVE
spec->loopback.amplist = ad1884_loopbacks;
#endif
+ spec->vmaster_nid = 0x04;
+ /* we need to cover all playback volumes */
+ spec->slave_vols = ad1884_slave_vols;
codec->patch_ops = ad198x_patch_ops;
@@ -3485,6 +3553,7 @@ static int patch_ad1882(struct hda_codec *codec)
#ifdef CONFIG_SND_HDA_POWER_SAVE
spec->loopback.amplist = ad1882_loopbacks;
#endif
+ spec->vmaster_nid = 0x04;
codec->patch_ops = ad198x_patch_ops;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 9184586..4bc7f3d 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -262,6 +262,9 @@ struct alc_spec {
unsigned int sense_updated: 1;
unsigned int jack_present: 1;
+ /* for virtual master */
+ hda_nid_t vmaster_nid;
+ u32 vmaster_tlv[4];
#ifdef CONFIG_SND_HDA_POWER_SAVE
struct hda_loopback_check loopback;
#endif
@@ -1309,8 +1312,8 @@ static hda_nid_t alc880_f1734_dac_nids[1] = {
static struct snd_kcontrol_new alc880_f1734_mixer[] = {
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Speaker Playback Volume", 0x0d, 0x0,
HDA_OUTPUT),
- HDA_BIND_MUTE("Internal Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
@@ -1408,10 +1411,10 @@ static struct snd_kcontrol_new alc880_tcl_s700_mixer[]
= {
/* Uniwill */
static struct snd_kcontrol_new alc880_uniwill_mixer[] = {
- HDA_CODEC_VOLUME("HPhone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("HPhone Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("iSpeaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("iSpeaker Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0,
HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
@@ -1451,16 +1454,50 @@ static struct snd_kcontrol_new alc880_fujitsu_mixer[] =
{
};
static struct snd_kcontrol_new alc880_uniwill_p53_mixer[] = {
- HDA_CODEC_VOLUME("HPhone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("HPhone Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("iSpeaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("iSpeaker Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
{ } /* end */
};
/*
+ * virtual master controls
+ */
+
+/*
+ * slave controls for virtual master
+ */
+static const char *alc_slave_vols[] = {
+ "Front Playback Volume",
+ "Surround Playback Volume",
+ "Center Playback Volume",
+ "LFE Playback Volume",
+ "Side Playback Volume",
+ "Headphone Playback Volume",
+ "Speaker Playback Volume",
+ "Mono Playback Volume",
+ "iSpeaker Playback Volume",
+ "Line-Out Playback Volume",
+ NULL,
+};
+
+static const char *alc_slave_sws[] = {
+ "Front Playback Switch",
+ "Surround Playback Switch",
+ "Center Playback Switch",
+ "LFE Playback Switch",
+ "Side Playback Switch",
+ "Headphone Playback Switch",
+ "Speaker Playback Switch",
+ "Mono Playback Switch",
+ "iSpeaker Playback Switch",
+ NULL,
+};
+
+/*
* build control elements
*/
static int alc_build_controls(struct hda_codec *codec)
@@ -1486,6 +1523,23 @@ static int alc_build_controls(struct hda_codec *codec)
if (err < 0)
return err;
}
+
+ /* if we have no master control, let's create it */
+ if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) {
+ snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid,
+ HDA_OUTPUT, spec->vmaster_tlv);
+ err = snd_hda_add_vmaster(codec, "Master Playback Volume",
+ spec->vmaster_tlv, alc_slave_vols);
+ if (err < 0)
+ return err;
+ }
+ if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
+ err = snd_hda_add_vmaster(codec, "Master Playback Switch",
+ NULL, alc_slave_sws);
+ if (err < 0)
+ return err;
+ }
+
return 0;
}
@@ -2034,8 +2088,8 @@ static struct hda_channel_mode alc880_lg_ch_modes[3] = {
static struct snd_kcontrol_new alc880_lg_mixer[] = {
/* FIXME: it's not really "master" but front channels */
- HDA_CODEC_VOLUME("Master Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Master Playback Switch", 0x0f, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0f, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0,
HDA_OUTPUT),
@@ -3592,6 +3646,8 @@ static int patch_alc880(struct hda_codec *codec)
}
}
+ spec->vmaster_nid = 0x0c;
+
codec->patch_ops = alc_patch_ops;
if (board_config == ALC880_AUTO)
spec->init_hook = alc880_auto_init;
@@ -4969,6 +5025,8 @@ static int patch_alc260(struct hda_codec *codec)
spec->stream_digital_playback = &alc260_pcm_digital_playback;
spec->stream_digital_capture = &alc260_pcm_digital_capture;
+ spec->vmaster_nid = 0x08;
+
codec->patch_ops = alc_patch_ops;
if (board_config == ALC260_AUTO)
spec->init_hook = alc260_auto_init;
@@ -5169,15 +5227,15 @@ static struct snd_kcontrol_new alc882_base_mixer[] = {
};
static struct snd_kcontrol_new alc885_mbp3_mixer[] = {
- HDA_CODEC_VOLUME("Master Volume", 0x0c, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE ("Master Switch", 0x0c, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE ("Speaker Switch", 0x14, 0x00, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Line Out Volume", 0x0d,0x00, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Line In Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE ("Line In Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
+ HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE ("Speaker Playback Switch", 0x14, 0x00, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT),
HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Line In Boost", 0x1a, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Boost", 0x1a, 0x00, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x00, HDA_INPUT),
{ } /* end */
};
@@ -6181,6 +6239,8 @@ static int patch_alc882(struct hda_codec *codec)
}
}
+ spec->vmaster_nid = 0x0c;
+
codec->patch_ops = alc_patch_ops;
if (board_config == ALC882_AUTO)
spec->init_hook = alc882_auto_init;
@@ -7763,6 +7823,8 @@ static int patch_alc883(struct hda_codec *codec)
spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids);
}
+ spec->vmaster_nid = 0x0c;
+
codec->patch_ops = alc_patch_ops;
if (board_config == ALC883_AUTO)
spec->init_hook = alc883_auto_init;
@@ -9123,6 +9185,8 @@ static int patch_alc262(struct hda_codec *codec)
}
}
+ spec->vmaster_nid = 0x0c;
+
codec->patch_ops = alc_patch_ops;
if (board_config == ALC262_AUTO)
spec->init_hook = alc262_auto_init;
@@ -9848,6 +9912,9 @@ static int patch_alc268(struct hda_codec *codec)
}
}
}
+
+ spec->vmaster_nid = 0x02;
+
codec->patch_ops = alc_patch_ops;
if (board_config == ALC268_AUTO)
spec->init_hook = alc268_auto_init;
@@ -11358,6 +11425,8 @@ static int patch_alc861(struct hda_codec *codec)
spec->stream_digital_playback = &alc861_pcm_digital_playback;
spec->stream_digital_capture = &alc861_pcm_digital_capture;
+ spec->vmaster_nid = 0x03;
+
codec->patch_ops = alc_patch_ops;
if (board_config == ALC861_AUTO)
spec->init_hook = alc861_auto_init;
@@ -12334,6 +12403,8 @@ static int patch_alc861vd(struct hda_codec *codec)
spec->mixers[spec->num_mixers] = alc861vd_capture_mixer;
spec->num_mixers++;
+ spec->vmaster_nid = 0x02;
+
codec->patch_ops = alc_patch_ops;
if (board_config == ALC861VD_AUTO)
@@ -13305,6 +13376,8 @@ static int patch_alc662(struct hda_codec *codec)
spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids);
}
+ spec->vmaster_nid = 0x02;
+
codec->patch_ops = alc_patch_ops;
if (board_config == ALC662_AUTO)
spec->init_hook = alc662_auto_init;
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index a0af868..190e112 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -170,6 +170,9 @@ struct sigmatel_spec {
struct snd_kcontrol_new *kctl_alloc;
struct hda_input_mux private_dimux;
struct hda_input_mux private_imux;
+
+ /* virtual master */
+ unsigned int vmaster_tlv[4];
};
static hda_nid_t stac9200_adc_nids[1] = {
@@ -794,6 +797,34 @@ static struct snd_kcontrol_new stac_dmux_mixer = {
.put = stac92xx_dmux_enum_put,
};
+static const char *slave_vols[] = {
+ "Front Playback Volume",
+ "Surround Playback Volume",
+ "Center Playback Volume",
+ "LFE Playback Volume",
+ "Side Playback Volume",
+ "Headphone Playback Volume",
+ "Headphone Playback Volume",
+ "Speaker Playback Volume",
+ "External Speaker Playback Volume",
+ "Speaker2 Playback Volume",
+ NULL
+};
+
+static const char *slave_sws[] = {
+ "Front Playback Switch",
+ "Surround Playback Switch",
+ "Center Playback Switch",
+ "LFE Playback Switch",
+ "Side Playback Switch",
+ "Headphone Playback Switch",
+ "Headphone Playback Switch",
+ "Speaker Playback Switch",
+ "External Speaker Playback Switch",
+ "Speaker2 Playback Switch",
+ NULL
+};
+
static int stac92xx_build_controls(struct hda_codec *codec)
{
struct sigmatel_spec *spec = codec->spec;
@@ -827,6 +858,23 @@ static int stac92xx_build_controls(struct hda_codec *codec)
if (err < 0)
return err;
}
+
+ /* if we have no master control, let's create it */
+ if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) {
+ snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0],
+ HDA_OUTPUT, spec->vmaster_tlv);
+ err = snd_hda_add_vmaster(codec, "Master Playback Volume",
+ spec->vmaster_tlv, slave_vols);
+ if (err < 0)
+ return err;
+ }
+ if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
+ err = snd_hda_add_vmaster(codec, "Master Playback Switch",
+ NULL, slave_sws);
+ if (err < 0)
+ return err;
+ }
+
return 0;
}
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