From 239fb862e8e26287212bdefabb021ef8c5ef1b53 Mon Sep 17 00:00:00 2001
From: Huacai Chen <chenhc@lemote.com>
Date: Fri, 5 Oct 2012 21:25:09 +0800
Subject: ALSA: HDA: Make hda sound card usable for Loongson

Lemote A1004(Laptop) and A1205(All-In-One) use Conexant's hda codec,
this patch modify patch_conexant.c to add Lemote specific code.

Both A1004 and A1205 use the same pin configurations, but A1004 need
to increase the default boost of internal mic.

Signed-off-by: Jie Chen <chenj@lemote.com>
Signed-off-by: Huacai Chen <chenhc@lemote.com>
Signed-off-by: Hongliang Tao <taohl@lemote.com>
Signed-off-by: Hua Yan <yanh@lemote.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 44 ++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 44 insertions(+)

(limited to 'sound/pci/hda/patch_conexant.c')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 03b1dc317ff0..252d4197f221 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -4405,7 +4405,10 @@ static const struct hda_codec_ops cx_auto_patch_ops = {
 enum {
 	CXT_PINCFG_LENOVO_X200,
 	CXT_PINCFG_LENOVO_TP410,
+	CXT_PINCFG_LEMOTE_A1004,
+	CXT_PINCFG_LEMOTE_A1205,
 	CXT_FIXUP_STEREO_DMIC,
+	CXT_FIXUP_INC_MIC_BOOST,
 };
 
 static void cxt_fixup_stereo_dmic(struct hda_codec *codec,
@@ -4415,6 +4418,19 @@ static void cxt_fixup_stereo_dmic(struct hda_codec *codec,
 	spec->fixup_stereo_dmic = 1;
 }
 
+static void cxt5066_increase_mic_boost(struct hda_codec *codec,
+				   const struct hda_fixup *fix, int action)
+{
+	if (action != HDA_FIXUP_ACT_PRE_PROBE)
+		return;
+
+	snd_hda_override_amp_caps(codec, 0x17, HDA_OUTPUT,
+				  (0x3 << AC_AMPCAP_OFFSET_SHIFT) |
+				  (0x4 << AC_AMPCAP_NUM_STEPS_SHIFT) |
+				  (0x27 << AC_AMPCAP_STEP_SIZE_SHIFT) |
+				  (0 << AC_AMPCAP_MUTE_SHIFT));
+}
+
 /* ThinkPad X200 & co with cxt5051 */
 static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = {
 	{ 0x16, 0x042140ff }, /* HP (seq# overridden) */
@@ -4432,6 +4448,18 @@ static const struct hda_pintbl cxt_pincfg_lenovo_tp410[] = {
 	{}
 };
 
+/* Lemote A1004/A1205 with cxt5066 */
+static const struct hda_pintbl cxt_pincfg_lemote[] = {
+	{ 0x1a, 0x90a10020 }, /* Internal mic */
+	{ 0x1b, 0x03a11020 }, /* External mic */
+	{ 0x1d, 0x400101f0 }, /* Not used */
+	{ 0x1e, 0x40a701f0 }, /* Not used */
+	{ 0x20, 0x404501f0 }, /* Not used */
+	{ 0x22, 0x404401f0 }, /* Not used */
+	{ 0x23, 0x40a701f0 }, /* Not used */
+	{}
+};
+
 static const struct hda_fixup cxt_fixups[] = {
 	[CXT_PINCFG_LENOVO_X200] = {
 		.type = HDA_FIXUP_PINS,
@@ -4441,10 +4469,24 @@ static const struct hda_fixup cxt_fixups[] = {
 		.type = HDA_FIXUP_PINS,
 		.v.pins = cxt_pincfg_lenovo_tp410,
 	},
+	[CXT_PINCFG_LEMOTE_A1004] = {
+		.type = HDA_FIXUP_PINS,
+		.chained = true,
+		.chain_id = CXT_FIXUP_INC_MIC_BOOST,
+		.v.pins = cxt_pincfg_lemote,
+	},
+	[CXT_PINCFG_LEMOTE_A1205] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = cxt_pincfg_lemote,
+	},
 	[CXT_FIXUP_STEREO_DMIC] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = cxt_fixup_stereo_dmic,
 	},
+	[CXT_FIXUP_INC_MIC_BOOST] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = cxt5066_increase_mic_boost,
+	},
 };
 
 static const struct snd_pci_quirk cxt5051_fixups[] = {
@@ -4461,6 +4503,8 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
 	SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC),
+	SND_PCI_QUIRK(0x1c06, 0x2011, "Lemote A1004", CXT_PINCFG_LEMOTE_A1004),
+	SND_PCI_QUIRK(0x1c06, 0x2012, "Lemote A1205", CXT_PINCFG_LEMOTE_A1205),
 	{}
 };
 
-- 
cgit v1.2.3


From f37bc7a88d374448a1f4bba9267d308606d78bf2 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 8 Nov 2012 15:59:23 +0100
Subject: ALSA: hda - Give standard "Bass Speaker" mixer for 2.1 speakers

When two built-in speakers are found on the machine, we can suppose
it's rather a 2.1 speaker system with a bass output instead of
front/surround channels.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_cirrus.c   |  8 +++++++-
 sound/pci/hda/patch_conexant.c | 28 +++++++++++++++++++++++++---
 sound/pci/hda/patch_sigmatel.c |  8 ++++----
 sound/pci/hda/patch_via.c      |  6 +++---
 4 files changed, 39 insertions(+), 11 deletions(-)

(limited to 'sound/pci/hda/patch_conexant.c')

diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index b9039dbd704d..794b0da11212 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -68,6 +68,7 @@ struct cs_spec {
 
 	unsigned int hp_detect:1;
 	unsigned int mic_detect:1;
+	unsigned int speaker_2_1:1;
 	/* CS421x */
 	unsigned int spdif_detect:1;
 	unsigned int sense_b:1;
@@ -443,6 +444,9 @@ static int parse_output(struct hda_codec *codec)
 	spec->multiout.dac_nids = spec->dac_nid;
 	spec->multiout.max_channels = i * 2;
 
+	if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT && i == 2)
+		spec->speaker_2_1 = 1; /* assume 2.1 speakers */
+
 	/* add HP and speakers */
 	extra_nids = 0;
 	for (i = 0; i < cfg->hp_outs; i++) {
@@ -632,7 +636,9 @@ static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx,
 		index = idx;
 		break;
 	case AUTO_PIN_SPEAKER_OUT:
-		if (num_ctls > 1)
+		if (spec->speaker_2_1)
+			name = idx ? "Bass Speaker" : "Speaker";
+		else if (num_ctls > 1)
 			name = speakers[idx];
 		else
 			name = "Speaker";
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 252d4197f221..f8e9ff493dc7 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -472,7 +472,7 @@ static const struct snd_kcontrol_new cxt_beep_mixer[] = {
 #endif
 
 static const char * const slave_pfxs[] = {
-	"Headphone", "Speaker", "Front", "Surround", "CLFE",
+	"Headphone", "Speaker", "Bass Speaker", "Front", "Surround", "CLFE",
 	NULL
 };
 
@@ -4116,11 +4116,26 @@ static int try_add_pb_volume(struct hda_codec *codec, hda_nid_t dac,
 	return 0;
 }
 
+static bool is_2_1_speaker(struct conexant_spec *spec)
+{
+	int i, type, num_spk = 0;
+
+	for (i = 0; i < spec->dac_info_filled; i++) {
+		type = spec->dac_info[i].type;
+		if (type == AUTO_PIN_LINE_OUT)
+			type = spec->autocfg.line_out_type;
+		if (type == AUTO_PIN_SPEAKER_OUT)
+			num_spk++;
+	}
+	return (num_spk == 2 && spec->autocfg.line_out_type != AUTO_PIN_LINE_OUT);
+}
+
 static int cx_auto_build_output_controls(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
 	int i, err;
 	int num_line = 0, num_hp = 0, num_spk = 0;
+	bool speaker_2_1;
 	static const char * const texts[3] = { "Front", "Surround", "CLFE" };
 
 	if (spec->dac_info_filled == 1)
@@ -4128,6 +4143,8 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
 					 spec->dac_info[0].pin,
 					 "Master", 0);
 
+	speaker_2_1 = is_2_1_speaker(spec);
+
 	for (i = 0; i < spec->dac_info_filled; i++) {
 		const char *label;
 		int idx, type;
@@ -4146,8 +4163,13 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
 			idx = num_hp++;
 			break;
 		case AUTO_PIN_SPEAKER_OUT:
-			label = "Speaker";
-			idx = num_spk++;
+			if (speaker_2_1) {
+				label = num_spk++ ? "Bass Speaker" : "Speaker";
+				idx = 0;
+			} else {
+				label = "Speaker";
+				idx = num_spk++;
+			}
 			break;
 		}
 		err = try_add_pb_volume(codec, dac,
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 962a948f4f10..f799406f4404 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1081,7 +1081,7 @@ static struct snd_kcontrol_new stac_smux_mixer = {
 
 static const char * const slave_pfxs[] = {
 	"Front", "Surround", "Center", "LFE", "Side",
-	"Headphone", "Speaker", "IEC958", "PCM",
+	"Headphone", "Speaker", "Bass Speaker", "IEC958", "PCM",
 	NULL
 };
 
@@ -3269,9 +3269,9 @@ static int create_multi_out_ctls(struct hda_codec *codec, int num_outs,
 				idx = i;
 				break;
 			case AUTO_PIN_SPEAKER_OUT:
-				if (num_outs <= 1) {
-					name = "Speaker";
-					idx = i;
+				if (num_outs <= 2) {
+					name = i ? "Bass Speaker" : "Speaker";
+					idx = 0;
 					break;
 				}
 				/* Fall through in case of multi speaker outs */
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 019e1a00414a..0e9b0747adc6 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1454,7 +1454,7 @@ static const struct hda_pcm_stream via_pcm_digital_capture = {
  */
 static const char * const via_slave_pfxs[] = {
 	"Front", "Surround", "Center", "LFE", "Side",
-	"Headphone", "Speaker",
+	"Headphone", "Speaker", "Bass Speaker",
 	NULL,
 };
 
@@ -1969,8 +1969,8 @@ static int via_auto_create_multi_out_ctls(struct hda_codec *codec)
 		} else {
 			const char *pfx = chname[i];
 			if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT &&
-			    cfg->line_outs == 1)
-				pfx = "Speaker";
+			    cfg->line_outs <= 2)
+				pfx = i ? "Bass Speaker" : "Speaker";
 			err = create_ch_ctls(codec, pfx, 3, true, path);
 			if (err < 0)
 				return err;
-- 
cgit v1.2.3


From ee81abb623cb5e03c182d16871bb4fb34fdc9b4f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 8 Nov 2012 17:12:10 +0100
Subject: ALSA: hda - Apply a proper chmap for built-in 2.1 speakers

When 2.1 speakers are detected, use the corresponding channel map
instead of the standard map with front+rear surrounds.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c      | 18 ++++++++++++++++--
 sound/pci/hda/hda_codec.h      |  3 +++
 sound/pci/hda/patch_cirrus.c   |  3 +++
 sound/pci/hda/patch_conexant.c |  5 +++++
 sound/pci/hda/patch_realtek.c  |  4 ++++
 sound/pci/hda/patch_sigmatel.c |  5 +++++
 sound/pci/hda/patch_via.c      |  4 ++++
 7 files changed, 40 insertions(+), 2 deletions(-)

(limited to 'sound/pci/hda/patch_conexant.c')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 569bc05aad6c..a8e7b00fe90d 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3747,13 +3747,14 @@ static int add_std_chmaps(struct hda_codec *codec)
 			struct hda_pcm_stream *hinfo =
 				&codec->pcm_info[i].stream[str];
 			struct snd_pcm_chmap *chmap;
+			const struct snd_pcm_chmap_elem *elem;
 
 			if (codec->pcm_info[i].own_chmap)
 				continue;
 			if (!pcm || !hinfo->substreams)
 				continue;
-			err = snd_pcm_add_chmap_ctls(pcm, str,
-						     snd_pcm_std_chmaps,
+			elem = hinfo->chmap ? hinfo->chmap : snd_pcm_std_chmaps;
+			err = snd_pcm_add_chmap_ctls(pcm, str, elem,
 						     hinfo->channels_max,
 						     0, &chmap);
 			if (err < 0)
@@ -3764,6 +3765,19 @@ static int add_std_chmaps(struct hda_codec *codec)
 	return 0;
 }
 
+/* default channel maps for 2.1 speakers;
+ * since HD-audio supports only stereo, odd number channels are omitted
+ */
+const struct snd_pcm_chmap_elem snd_pcm_2_1_chmaps[] = {
+	{ .channels = 2,
+	  .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } },
+	{ .channels = 4,
+	  .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR,
+		   SNDRV_CHMAP_LFE, SNDRV_CHMAP_LFE } },
+	{ }
+};
+EXPORT_SYMBOL_GPL(snd_pcm_2_1_chmaps);
+
 int snd_hda_codec_build_controls(struct hda_codec *codec)
 {
 	int err = 0;
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 62d4229c7b95..baad7bf6133b 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -757,6 +757,7 @@ struct hda_pcm_stream {
 	u32 rates;	/* supported rates */
 	u64 formats;	/* supported formats (SNDRV_PCM_FMTBIT_) */
 	unsigned int maxbps;	/* supported max. bit per sample */
+	const struct snd_pcm_chmap_elem *chmap; /* chmap to override */
 	struct hda_pcm_ops ops;
 };
 
@@ -1026,6 +1027,8 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate,
 int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid,
 				unsigned int format);
 
+extern const struct snd_pcm_chmap_elem snd_pcm_2_1_chmaps[];
+
 /*
  * Misc
  */
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 794b0da11212..f99cbf942b8d 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -344,6 +344,9 @@ static int cs_build_pcms(struct hda_codec *codec)
 	info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dac_nid[0];
 	info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max =
 		spec->multiout.max_channels;
+	if (spec->speaker_2_1)
+		info->stream[SNDRV_PCM_STREAM_PLAYBACK].chmap =
+			snd_pcm_2_1_chmaps;
 	info->stream[SNDRV_PCM_STREAM_CAPTURE] = cs_pcm_analog_capture;
 	info->stream[SNDRV_PCM_STREAM_CAPTURE].nid =
 		spec->adc_nid[spec->cur_input];
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index f8e9ff493dc7..3401a087de80 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -337,6 +337,8 @@ static const struct hda_pcm_stream cx5051_pcm_analog_capture = {
 	},
 };
 
+static bool is_2_1_speaker(struct conexant_spec *spec);
+
 static int conexant_build_pcms(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
@@ -351,6 +353,9 @@ static int conexant_build_pcms(struct hda_codec *codec)
 		spec->multiout.max_channels;
 	info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid =
 		spec->multiout.dac_nids[0];
+	if (is_2_1_speaker(spec))
+		info->stream[SNDRV_PCM_STREAM_PLAYBACK].chmap =
+			snd_pcm_2_1_chmaps;
 	if (spec->capture_stream)
 		info->stream[SNDRV_PCM_STREAM_CAPTURE] = *spec->capture_stream;
 	else {
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index bc71be32232e..89737ae49152 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2260,6 +2260,10 @@ static int alc_build_pcms(struct hda_codec *codec)
 		info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0];
 		info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max =
 			spec->multiout.max_channels;
+		if (spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT &&
+		    spec->autocfg.line_outs == 2)
+			info->stream[SNDRV_PCM_STREAM_PLAYBACK].chmap =
+				snd_pcm_2_1_chmaps;
 	}
 	if (spec->adc_nids) {
 		p = spec->stream_analog_capture;
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index f799406f4404..a6aeb6260719 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2516,6 +2516,11 @@ static int stac92xx_build_pcms(struct hda_codec *codec)
 	info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_analog_playback;
 	info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid =
 		spec->multiout.dac_nids[0];
+	if (spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT &&
+	    spec->autocfg.line_outs == 2)
+		info->stream[SNDRV_PCM_STREAM_PLAYBACK].chmap =
+			snd_pcm_2_1_chmaps;
+
 	info->stream[SNDRV_PCM_STREAM_CAPTURE] = stac92xx_pcm_analog_capture;
 	info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0];
 	info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_adcs;
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 0e9b0747adc6..9ae8cfcecf1d 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1555,6 +1555,10 @@ static int via_build_pcms(struct hda_codec *codec)
 				spec->multiout.dac_nids[0];
 			info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max =
 				spec->multiout.max_channels;
+			if (spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT
+			    && spec->autocfg.line_outs == 2)
+				info->stream[SNDRV_PCM_STREAM_PLAYBACK].chmap =
+					snd_pcm_2_1_chmaps;
 		}
 
 		if (!spec->stream_analog_capture) {
-- 
cgit v1.2.3


From dda415d41882449f841f88d829dd65b6ee1c374c Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 30 Nov 2012 18:34:38 +0100
Subject: ALSA: hda - Add a helper function for simple enum kcontrols

The same type of code is being used in multiple places in various
codec drivers, so put it as a core library.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c      | 28 ++++++++++++++++++++++++++++
 sound/pci/hda/hda_local.h      |  9 +++++++++
 sound/pci/hda/patch_conexant.c | 21 +++------------------
 sound/pci/hda/patch_realtek.c  | 21 +++------------------
 sound/pci/hda/patch_via.c      | 13 +------------
 5 files changed, 44 insertions(+), 48 deletions(-)

(limited to 'sound/pci/hda/patch_conexant.c')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 3634bfebc008..d05bcbb61077 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -4791,6 +4791,34 @@ int snd_hda_input_mux_put(struct hda_codec *codec,
 EXPORT_SYMBOL_HDA(snd_hda_input_mux_put);
 
 
+/*
+ * process kcontrol info callback of a simple string enum array
+ * when @num_items is 0 or @texts is NULL, assume a boolean enum array
+ */
+int snd_hda_enum_helper_info(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_info *uinfo,
+			     int num_items, const char * const *texts)
+{
+	static const char * const texts_default[] = {
+		"Disabled", "Enabled"
+	};
+
+	if (!texts || !num_items) {
+		num_items = 2;
+		texts = texts_default;
+	}
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = num_items;
+	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+	strcpy(uinfo->value.enumerated.name,
+	       texts[uinfo->value.enumerated.item]);
+	return 0;
+}
+EXPORT_SYMBOL_HDA(snd_hda_enum_helper_info);
+
 /*
  * Multi-channel / digital-out PCM helper functions
  */
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 8c43198b7f56..4b40a5e7a8f5 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -600,6 +600,15 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec,
 #define get_amp_offset(kc)	(((kc)->private_value >> 23) & 0x3f)
 #define get_amp_min_mute(kc)	(((kc)->private_value >> 29) & 0x1)
 
+/*
+ * enum control helper
+ */
+int snd_hda_enum_helper_info(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_info *uinfo,
+			     int num_entries, const char * const *texts);
+#define snd_hda_enum_bool_helper_info(kcontrol, uinfo) \
+	snd_hda_enum_helper_info(kcontrol, uinfo, 0, NULL)
+
 /*
  * CEA Short Audio Descriptor data
  */
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 3401a087de80..a3a2263d3b37 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3435,28 +3435,13 @@ static int cx_automute_mode_info(struct snd_kcontrol *kcontrol,
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct conexant_spec *spec = codec->spec;
-	static const char * const texts2[] = {
-		"Disabled", "Enabled"
-	};
 	static const char * const texts3[] = {
 		"Disabled", "Speaker Only", "Line Out+Speaker"
 	};
-	const char * const *texts;
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	if (spec->automute_hp_lo) {
-		uinfo->value.enumerated.items = 3;
-		texts = texts3;
-	} else {
-		uinfo->value.enumerated.items = 2;
-		texts = texts2;
-	}
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name,
-	       texts[uinfo->value.enumerated.item]);
-	return 0;
+	if (spec->automute_hp_lo)
+		return snd_hda_enum_helper_info(kcontrol, uinfo, 3, texts3);
+	return snd_hda_enum_bool_helper_info(kcontrol, uinfo);
 }
 
 static int cx_automute_mode_get(struct snd_kcontrol *kcontrol,
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 0062c9f401c3..5d8044d4dc52 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -815,28 +815,13 @@ static int alc_automute_mode_info(struct snd_kcontrol *kcontrol,
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct alc_spec *spec = codec->spec;
-	static const char * const texts2[] = {
-		"Disabled", "Enabled"
-	};
 	static const char * const texts3[] = {
 		"Disabled", "Speaker Only", "Line Out+Speaker"
 	};
-	const char * const *texts;
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	if (spec->automute_speaker_possible && spec->automute_lo_possible) {
-		uinfo->value.enumerated.items = 3;
-		texts = texts3;
-	} else {
-		uinfo->value.enumerated.items = 2;
-		texts = texts2;
-	}
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name,
-	       texts[uinfo->value.enumerated.item]);
-	return 0;
+	if (spec->automute_speaker_possible && spec->automute_lo_possible)
+		return snd_hda_enum_helper_info(kcontrol, uinfo, 3, texts3);
+	return snd_hda_enum_bool_helper_info(kcontrol, uinfo);
 }
 
 static int alc_automute_mode_get(struct snd_kcontrol *kcontrol,
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 274644f6bd48..dd8cd6af5d66 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -739,18 +739,7 @@ static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid,
 static int via_pin_power_ctl_info(struct snd_kcontrol *kcontrol,
 				  struct snd_ctl_elem_info *uinfo)
 {
-	static const char * const texts[] = {
-		"Disabled", "Enabled"
-	};
-
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 2;
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
-	strcpy(uinfo->value.enumerated.name,
-	       texts[uinfo->value.enumerated.item]);
-	return 0;
+	return snd_hda_enum_bool_helper_info(kcontrol, uinfo);
 }
 
 static int via_pin_power_ctl_get(struct snd_kcontrol *kcontrol,
-- 
cgit v1.2.3


From 63a077e27648b4043b1ca1b4e29f0c42d99616b6 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 12 Dec 2012 12:10:01 +0100
Subject: ALSA: hda - Add stereo-dmic fixup for Acer Aspire One 522

Acer Aspire One 522 has the infamous digital mic unit that needs the
phase inversion fixup for stereo.

Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=715737

Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci/hda/patch_conexant.c')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index a3a2263d3b37..60890bfecc19 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -4507,6 +4507,7 @@ static const struct snd_pci_quirk cxt5051_fixups[] = {
 };
 
 static const struct snd_pci_quirk cxt5066_fixups[] = {
+	SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410),
 	SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410),
 	SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410),
-- 
cgit v1.2.3