Merge commit 'v3.3-rc6' into fbdev-next

1 files changed, 20 insertions, 7 deletions

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 1358987c49d8..f286bb8fda13 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -80,6 +80,8 @@ enum {
 	ALC_AUTOMUTE_MIXER,	/* mute/unmute mixer widget AMP */
 };
 
+#define MAX_VOL_NIDS	0x40
+
 struct alc_spec {
 	/* codec parameterization */
 	const struct snd_kcontrol_new *mixers[5];	/* mixer arrays */
@@ -118,8 +120,8 @@ struct alc_spec {
 	const hda_nid_t *capsrc_nids;
 	hda_nid_t dig_in_nid;		/* digital-in NID; optional */
 	hda_nid_t mixer_nid;		/* analog-mixer NID */
-	DECLARE_BITMAP(vol_ctls, 0x20 << 1);
-	DECLARE_BITMAP(sw_ctls, 0x20 << 1);
+	DECLARE_BITMAP(vol_ctls, MAX_VOL_NIDS << 1);
+	DECLARE_BITMAP(sw_ctls, MAX_VOL_NIDS << 1);
 
 	/* capture setup for dynamic dual-adc switch */
 	hda_nid_t cur_adc;
@@ -800,7 +802,7 @@ static int alc_automute_mode_info(struct snd_kcontrol *kcontrol,
 		"Disabled", "Enabled"
 	};
 	static const char * const texts3[] = {
-		"Disabled", "Speaker Only", "Line-Out+Speaker"
+		"Disabled", "Speaker Only", "Line Out+Speaker"
 	};
 	const char * const *texts;
 
@@ -1854,7 +1856,7 @@ static const char * const alc_slave_vols[] = {
 	"Headphone Playback Volume",
 	"Speaker Playback Volume",
 	"Mono Playback Volume",
-	"Line-Out Playback Volume",
+	"Line Out Playback Volume",
 	"CLFE Playback Volume",
 	"Bass Speaker Playback Volume",
 	"PCM Playback Volume",
@@ -1871,7 +1873,7 @@ static const char * const alc_slave_sws[] = {
 	"Speaker Playback Switch",
 	"Mono Playback Switch",
 	"IEC958 Playback Switch",
-	"Line-Out Playback Switch",
+	"Line Out Playback Switch",
 	"CLFE Playback Switch",
 	"Bass Speaker Playback Switch",
 	"PCM Playback Switch",
@@ -3149,7 +3151,10 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
 static inline unsigned int get_ctl_pos(unsigned int data)
 {
 	hda_nid_t nid = get_amp_nid_(data);
-	unsigned int dir = get_amp_direction_(data);
+	unsigned int dir;
+	if (snd_BUG_ON(nid >= MAX_VOL_NIDS))
+		return 0;
+	dir = get_amp_direction_(data);
 	return (nid << 1) | dir;
 }
 
@@ -3792,7 +3797,7 @@ static void alc_auto_init_input_src(struct hda_codec *codec)
 	else
 		nums = spec->num_adc_nids;
 	for (c = 0; c < nums; c++)
-		alc_mux_select(codec, 0, spec->cur_mux[c], true);
+		alc_mux_select(codec, c, spec->cur_mux[c], true);
 }
 
 /* add mic boosts if needed */
@@ -4436,12 +4441,20 @@ static void alc889_fixup_dac_route(struct hda_codec *codec,
 				   const struct alc_fixup *fix, int action)
 {
 	if (action == ALC_FIXUP_ACT_PRE_PROBE) {
+		/* fake the connections during parsing the tree */
 		hda_nid_t conn1[2] = { 0x0c, 0x0d };
 		hda_nid_t conn2[2] = { 0x0e, 0x0f };
 		snd_hda_override_conn_list(codec, 0x14, 2, conn1);
 		snd_hda_override_conn_list(codec, 0x15, 2, conn1);
 		snd_hda_override_conn_list(codec, 0x18, 2, conn2);
 		snd_hda_override_conn_list(codec, 0x1a, 2, conn2);
+	} else if (action == ALC_FIXUP_ACT_PROBE) {
+		/* restore the connections */
+		hda_nid_t conn[5] = { 0x0c, 0x0d, 0x0e, 0x0f, 0x26 };
+		snd_hda_override_conn_list(codec, 0x14, 5, conn);
+		snd_hda_override_conn_list(codec, 0x15, 5, conn);
+		snd_hda_override_conn_list(codec, 0x18, 5, conn);
+		snd_hda_override_conn_list(codec, 0x1a, 5, conn);
 	}
 }