[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 <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

5 files changed, 276 insertions, 19 deletions

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index a2b40dc372c9..caacc58c0813 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 e09f41bd6b2a..ddc61a1d1153 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 6664a0688ef5..b0755407be9d 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 9184586c9721..4bc7f3daeab0 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 a0af8680dd0d..190e112f2f8e 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;	
 }