From 27346166a9b3b9eee586bce212502cddf9685a07 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 12 Jan 2006 18:28:44 +0100
Subject: [ALSA] hda-intel - Add single_cmd option for debugging

Modules: Documentation,HDA Intel driver

Added single_cmd module option for debugging in the case CORB/RIRB
doesn't work well (e.g. due to wrong irq routings).

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 Documentation/sound/alsa/ALSA-Configuration.txt | 13 +++++++++++++
 1 file changed, 13 insertions(+)

(limited to 'Documentation')

diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 36b511c7cade..cc8a70187199 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -671,6 +671,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     model	- force the model name
     position_fix - Fix DMA pointer (0 = auto, 1 = none, 2 = POSBUF, 3 = FIFO size)
+    single_cmd  - Use single immediate commands to communicate with
+		codecs (for debugging only)
 
     This module supports one card and autoprobe.
 
@@ -723,6 +725,17 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	    (Usually SD_LPLIB register is more accurate than the
 	    position buffer.)
 
+    NB: If you get many "azx_get_response timeout" messages at
+    loading, it's likely a problem of interrupts (e.g. ACPI irq
+    routing).  Try to boot with options like "pci=noacpi".  Also, you
+    can try "single_cmd=1" module option.  This will switch the
+    communication method between HDA controller and codecs to the
+    single immediate commands instead of CORB/RIRB.  Basically, the
+    single command mode is provided only for BIOS, and you won't get
+    unsolicited events, too.  But, at least, this works independently
+    from the irq.  Remember this is a last resort, and should be
+    avoided as much as possible...
+    
     The power-management is supported.
 
   Module snd-hdsp
-- 
cgit v1.2.3


From 7a6c8ff1ef83df4ce44b586999e54966d8e5bda8 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 13 Jan 2006 13:56:33 +0100
Subject: [ALSA] Update description of ice1724 driver

Modules: Documentation

Updated the description of ice1724 driver.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 Documentation/sound/alsa/ALSA-Configuration.txt | 8 +++++---
 1 file changed, 5 insertions(+), 3 deletions(-)

(limited to 'Documentation')

diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index cc8a70187199..b45f7be47816 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -815,6 +815,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
   ------------------
 
     Module for Envy24HT (VT/ICE1724), Envy24PT (VT1720) based PCI sound cards.
+			* MidiMan M Audio Revolution 5.1
 			* MidiMan M Audio Revolution 7.1
 			* AMP Ltd AUDIO2000
 			* TerraTec Aureon 5.1 Sky
@@ -823,6 +824,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 			* TerraTec Phase 22
 			* TerraTec Phase 28
 			* AudioTrak Prodigy 7.1
+			* AudioTrak Prodigy 7.1LT
 			* AudioTrak Prodigy 192
 			* Pontis MS300
 			* Albatron K8X800 Pro II 
@@ -833,9 +835,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 			* Shuttle SN25P
 
     model       - Use the given board model, one of the following:
-		  revo71, amp2000, prodigy71, prodigy192, aureon51,
-		  aureon71, universe, k8x800, phase22, phase28, ms300,
-		  av710
+		  revo51, revo71, amp2000, prodigy71, prodigy71lt,
+		  prodigy192, aureon51, aureon71, universe,
+		  k8x800, phase22, phase28, ms300, av710
 
     This module supports multiple cards and autoprobe.
 
-- 
cgit v1.2.3


From d6ec894b6d6bf12885a34a4667bccb7f67e2916c Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 20 Jan 2006 14:05:06 +0100
Subject: [ALSA] Add the notes on PM to ens1370/ens1371 sections

Modules: Documentation

Add the notes on PM to ens1370/ens1371 sections.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 Documentation/sound/alsa/ALSA-Configuration.txt | 4 ++++
 1 file changed, 4 insertions(+)

(limited to 'Documentation')

diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index b45f7be47816..f28c45ca8395 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -513,6 +513,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     This module supports multiple cards and autoprobe.
     
+    The power-management is supported.
+
   Module snd-ens1371
   ------------------
 
@@ -526,6 +528,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     This module supports multiple cards and autoprobe.
     
+    The power-management is supported.
+
   Module snd-es968
   ----------------
 
-- 
cgit v1.2.3


From ed345f8f8630d6a3f52d899e181968e2bf8e4be8 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 1 Mar 2006 14:16:53 +0100
Subject: [ALSA] Update description of hda-intel models

Modules: Documentation

Updated the description of hda-intel models for realtek codecs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 Documentation/sound/alsa/ALSA-Configuration.txt | 20 ++++++++++++++++++++
 1 file changed, 20 insertions(+)

(limited to 'Documentation')

diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index f28c45ca8395..0daba0a8c4fd 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -703,10 +703,30 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  test		for testing/debugging purpose, almost all controls can be
 			adjusted.  Appearing only when compiled with
 			$CONFIG_SND_DEBUG=y
+	  auto		auto-config reading BIOS (default)
 
 	ALC260
 	  hp		HP machines
 	  fujitsu	Fujitsu S7020
+	  acer		Acer TravelMate
+	  basic		fixed pin assignment (old default model)
+	  auto		auto-config reading BIOS (default)
+
+	ALC262
+	  fujitsu	Fujitsu Laptop
+	  basic		fixed pin assignment w/o SPDIF
+	  auto		auto-config reading BIOS (default)
+
+	ALC882/883/885
+	  3stack-dig	3-jack with SPDIF I/O
+	  6stck-dig	6-jack digital with SPDIF I/O
+	  auto		auto-config reading BIOS (default)
+
+	ALC861
+	  3stack	3-jack
+	  3stack-dig	3-jack with SPDIF I/O
+	  6stack-dig	6-jack with SPDIF I/O
+	  auto		auto-config reading BIOS (default)
 
 	CMI9880
 	  minimal	3-jack in back
-- 
cgit v1.2.3


From 0b7bed4ec2a16434336e018505b66bd51bb35560 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 2 Mar 2006 15:35:55 +0100
Subject: [ALSA] Fix typos in document

Modules: Documentation

Fixed typos in document.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'Documentation')

diff --git a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
index 4251085d38d3..6dc9d9f622ca 100644
--- a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
+++ b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
@@ -1834,7 +1834,7 @@
           mychip_set_sample_format(chip, runtime->format);
           mychip_set_sample_rate(chip, runtime->rate);
           mychip_set_channels(chip, runtime->channels);
-          mychip_set_dma_setup(chip, runtime->dma_area,
+          mychip_set_dma_setup(chip, runtime->dma_addr,
                                chip->buffer_size,
                                chip->period_size);
           return 0;
@@ -3388,7 +3388,7 @@ struct _snd_pcm_runtime {
           .name = "PCM Playback Switch",
           .index = 0,
           .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
-          .private_values = 0xffff,
+          .private_value = 0xffff,
           .info = my_control_info,
           .get = my_control_get,
           .put = my_control_put
@@ -3449,7 +3449,7 @@ struct _snd_pcm_runtime {
       </para>
 
       <para>
-        The <structfield>private_values</structfield> field contains
+        The <structfield>private_value</structfield> field contains
       an arbitrary long integer value for this record. When using
       generic <structfield>info</structfield>,
       <structfield>get</structfield> and
-- 
cgit v1.2.3


From ae6b813a4dbba2713df497c032798b845289653f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 3 Mar 2006 16:47:17 +0100
Subject: [ALSA] hda-codec - Add lg model for LG laptop

Modules: Documentation,HDA Codec driver

Added a new model 'lg' for LG laptop (m1 express dual) with ALC880 codec.
Also clean up the initialization/unsol_event hooks in patch_realtek.c.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 Documentation/sound/alsa/ALSA-Configuration.txt |   1 +
 sound/pci/hda/patch_realtek.c                   | 220 +++++++++++++++++++++---
 2 files changed, 193 insertions(+), 28 deletions(-)

(limited to 'Documentation')

diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 0daba0a8c4fd..c12dab05176a 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -700,6 +700,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  asus		3-jack
 	  uniwill	3-jack
 	  F1734		2-jack
+	  lg		LG laptop (m1 express dual)
 	  test		for testing/debugging purpose, almost all controls can be
 			adjusted.  Appearing only when compiled with
 			$CONFIG_SND_DEBUG=y
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 5de754a51fc7..fcab766862d4 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -51,6 +51,7 @@ enum {
 	ALC880_UNIWILL_DIG,
 	ALC880_CLEVO,
 	ALC880_TCL_S700,
+	ALC880_LG,
 #ifdef CONFIG_SND_DEBUG
 	ALC880_TEST,
 #endif
@@ -147,6 +148,10 @@ struct alc_spec {
 	struct hda_input_mux private_imux;
 	hda_nid_t private_dac_nids[5];
 
+	/* hooks */
+	void (*init_hook)(struct hda_codec *codec);
+	void (*unsol_event)(struct hda_codec *codec, unsigned int res);
+
 	/* for pin sensing */
 	unsigned int sense_updated: 1;
 	unsigned int jack_present: 1;
@@ -168,6 +173,8 @@ struct alc_config_preset {
 	unsigned int num_channel_mode;
 	const struct hda_channel_mode *channel_mode;
 	const struct hda_input_mux *input_mux;
+	void (*unsol_event)(struct hda_codec *, unsigned int);
+	void (*init_hook)(struct hda_codec *);
 };
 
 
@@ -481,6 +488,9 @@ static void setup_preset(struct alc_spec *spec, const struct alc_config_preset *
 	spec->num_adc_nids = preset->num_adc_nids;
 	spec->adc_nids = preset->adc_nids;
 	spec->dig_in_nid = preset->dig_in_nid;
+
+	spec->unsol_event = preset->unsol_event;
+	spec->init_hook = preset->init_hook;
 }
 
 /*
@@ -1283,6 +1293,141 @@ static struct hda_verb alc880_pin_tcl_S700_init_verbs[] = {
 };
 
 /*
+ * LG m1 express dual
+ *
+ * Pin assignment:
+ *   Rear Line-In/Out (blue): 0x14
+ *   Build-in Mic-In: 0x15
+ *   Speaker-out: 0x17
+ *   HP-Out (green): 0x1b
+ *   Mic-In/Out (red): 0x19
+ *   SPDIF-Out: 0x1e
+ */
+
+/* To make 5.1 output working (green=Front, blue=Surr, red=CLFE) */
+static hda_nid_t alc880_lg_dac_nids[3] = {
+	0x05, 0x02, 0x03
+};
+
+/* seems analog CD is not working */
+static struct hda_input_mux alc880_lg_capture_source = {
+	.num_items = 3,
+	.items = {
+		{ "Mic", 0x1 },
+		{ "Line", 0x5 },
+		{ "Internal Mic", 0x6 },
+	},
+};
+
+/* 2,4,6 channel modes */
+static struct hda_verb alc880_lg_ch2_init[] = {
+	/* set line-in and mic-in to input */
+	{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+	{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+	{ }
+};
+
+static struct hda_verb alc880_lg_ch4_init[] = {
+	/* set line-in to out and mic-in to input */
+	{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
+	{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+	{ }
+};
+
+static struct hda_verb alc880_lg_ch6_init[] = {
+	/* set line-in and mic-in to output */
+	{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
+	{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
+	{ }
+};
+
+static struct hda_channel_mode alc880_lg_ch_modes[3] = {
+	{ 2, alc880_lg_ch2_init },
+	{ 4, alc880_lg_ch4_init },
+	{ 6, alc880_lg_ch6_init },
+};
+
+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("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),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT),
+	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x06, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x06, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x07, HDA_INPUT),
+	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x07, HDA_INPUT),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Channel Mode",
+		.info = alc_ch_mode_info,
+		.get = alc_ch_mode_get,
+		.put = alc_ch_mode_put,
+	},
+	{ } /* end */
+};
+
+static struct hda_verb alc880_lg_init_verbs[] = {
+	/* set capture source to mic-in */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	/* mute all amp mixer inputs */
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(6)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(7)},
+	/* line-in to input */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* built-in mic */
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* speaker-out */
+	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* mic-in to input */
+	{0x11, AC_VERB_SET_CONNECT_SEL, 0x01},
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* HP-out */
+	{0x13, AC_VERB_SET_CONNECT_SEL, 0x03},
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* jack sense */
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | 0x1},
+	{ }
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc880_lg_automute(struct hda_codec *codec)
+{
+	unsigned int present;
+
+	present = snd_hda_codec_read(codec, 0x1b, 0,
+				     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+	snd_hda_codec_amp_update(codec, 0x17, 0, HDA_OUTPUT, 0,
+				 0x80, present ? 0x80 : 0);
+	snd_hda_codec_amp_update(codec, 0x17, 1, HDA_OUTPUT, 0,
+				 0x80, present ? 0x80 : 0);
+}
+
+static void alc880_lg_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+	/* Looks like the unsol event is incompatible with the standard
+	 * definition.  4bit tag is placed at 28 bit!
+	 */
+	if ((res >> 28) == 0x01)
+		alc880_lg_automute(codec);
+}
+
+/*
+ * Common callbacks
  */
 
 static int alc_init(struct hda_codec *codec)
@@ -1292,9 +1437,21 @@ static int alc_init(struct hda_codec *codec)
 
 	for (i = 0; i < spec->num_init_verbs; i++)
 		snd_hda_sequence_write(codec, spec->init_verbs[i]);
+
+	if (spec->init_hook)
+		spec->init_hook(codec);
+
 	return 0;
 }
 
+static void alc_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+	struct alc_spec *spec = codec->spec;
+
+	if (spec->unsol_event)
+		spec->unsol_event(codec, res);
+}
+
 #ifdef CONFIG_PM
 /*
  * resume
@@ -1531,6 +1688,7 @@ static struct hda_codec_ops alc_patch_ops = {
 	.build_pcms = alc_build_pcms,
 	.init = alc_init,
 	.free = alc_free,
+	.unsol_event = alc_unsol_event,
 #ifdef CONFIG_PM
 	.resume = alc_resume,
 #endif
@@ -1549,13 +1707,15 @@ static hda_nid_t alc880_test_dac_nids[4] = {
 };
 
 static struct hda_input_mux alc880_test_capture_source = {
-	.num_items = 5,
+	.num_items = 7,
 	.items = {
 		{ "In-1", 0x0 },
 		{ "In-2", 0x1 },
 		{ "In-3", 0x2 },
 		{ "In-4", 0x3 },
 		{ "CD", 0x4 },
+		{ "Front", 0x5 },
+		{ "Surround", 0x6 },
 	},
 };
 
@@ -1911,6 +2071,9 @@ static struct hda_board_config alc880_cfg_tbl[] = {
 	{ .pci_subvendor = 0x1734, .pci_subdevice = 0x107c, .config = ALC880_F1734 },
 	{ .pci_subvendor = 0x1584, .pci_subdevice = 0x9054, .config = ALC880_F1734 },
 
+	{ .modelname = "lg", .config = ALC880_LG },
+	{ .pci_subvendor = 0x1854, .pci_subdevice = 0x003b, .config = ALC880_LG },
+
 #ifdef CONFIG_SND_DEBUG
 	{ .modelname = "test", .config = ALC880_TEST },
 #endif
@@ -2088,6 +2251,19 @@ static struct alc_config_preset alc880_presets[] = {
 		.channel_mode = alc880_threestack_modes,
 		.input_mux = &alc880_capture_source,
 	},
+	[ALC880_LG] = {
+		.mixers = { alc880_lg_mixer },
+		.init_verbs = { alc880_volume_init_verbs,
+				alc880_lg_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc880_lg_dac_nids),
+		.dac_nids = alc880_lg_dac_nids,
+		.dig_out_nid = ALC880_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc880_lg_ch_modes),
+		.channel_mode = alc880_lg_ch_modes,
+		.input_mux = &alc880_lg_capture_source,
+		.unsol_event = alc880_lg_unsol_event,
+		.init_hook = alc880_lg_automute,
+	},
 #ifdef CONFIG_SND_DEBUG
 	[ALC880_TEST] = {
 		.mixers = { alc880_test_mixer },
@@ -2427,14 +2603,12 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
 	return 1;
 }
 
-/* init callback for auto-configuration model -- overriding the default init */
-static int alc880_auto_init(struct hda_codec *codec)
+/* additional initialization for auto-configuration model */
+static void alc880_auto_init(struct hda_codec *codec)
 {
-	alc_init(codec);
 	alc880_auto_init_multi_out(codec);
 	alc880_auto_init_extra_out(codec);
 	alc880_auto_init_analog_input(codec);
-	return 0;
 }
 
 /*
@@ -2501,7 +2675,7 @@ static int patch_alc880(struct hda_codec *codec)
 
 	codec->patch_ops = alc_patch_ops;
 	if (board_config == ALC880_AUTO)
-		codec->patch_ops.init = alc880_auto_init;
+		spec->init_hook = alc880_auto_init;
 
 	return 0;
 }
@@ -3456,13 +3630,11 @@ static int alc260_parse_auto_config(struct hda_codec *codec)
 	return 1;
 }
 
-/* init callback for auto-configuration model -- overriding the default init */
-static int alc260_auto_init(struct hda_codec *codec)
+/* additional initialization for auto-configuration model */
+static void alc260_auto_init(struct hda_codec *codec)
 {
-	alc_init(codec);
 	alc260_auto_init_multi_out(codec);
 	alc260_auto_init_analog_input(codec);
-	return 0;
 }
 
 /*
@@ -3614,7 +3786,7 @@ static int patch_alc260(struct hda_codec *codec)
 
 	codec->patch_ops = alc_patch_ops;
 	if (board_config == ALC260_AUTO)
-		codec->patch_ops.init = alc260_auto_init;
+		spec->init_hook = alc260_auto_init;
 
 	return 0;
 }
@@ -4089,14 +4261,12 @@ static int alc882_parse_auto_config(struct hda_codec *codec)
 	return err;
 }
 
-/* init callback for auto-configuration model -- overriding the default init */
-static int alc882_auto_init(struct hda_codec *codec)
+/* additional initialization for auto-configuration model */
+static void alc882_auto_init(struct hda_codec *codec)
 {
-	alc_init(codec);
 	alc882_auto_init_multi_out(codec);
 	alc882_auto_init_hp_out(codec);
 	alc882_auto_init_analog_input(codec);
-	return 0;
 }
 
 /*
@@ -4163,7 +4333,7 @@ static int patch_alc882(struct hda_codec *codec)
 
 	codec->patch_ops = alc_patch_ops;
 	if (board_config == ALC882_AUTO)
-		codec->patch_ops.init = alc882_auto_init;
+		spec->init_hook = alc882_auto_init;
 
 	return 0;
 }
@@ -4583,13 +4753,11 @@ static int alc262_parse_auto_config(struct hda_codec *codec)
 
 
 /* init callback for auto-configuration model -- overriding the default init */
-static int alc262_auto_init(struct hda_codec *codec)
+static void alc262_auto_init(struct hda_codec *codec)
 {
-	alc_init(codec);
 	alc262_auto_init_multi_out(codec);
 	alc262_auto_init_hp_out(codec);
 	alc262_auto_init_analog_input(codec);
-	return 0;
 }
 
 /*
@@ -4624,6 +4792,7 @@ static struct alc_config_preset alc262_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc262_modes),
 		.channel_mode = alc262_modes,
 		.input_mux = &alc262_fujitsu_capture_source,
+		.unsol_event = alc262_fujitsu_unsol_event,
 	},
 };
 
@@ -4698,9 +4867,7 @@ static int patch_alc262(struct hda_codec *codec)
 
 	codec->patch_ops = alc_patch_ops;
 	if (board_config == ALC262_AUTO)
-		codec->patch_ops.init = alc262_auto_init;
-	if (board_config == ALC262_FUJITSU)
-		codec->patch_ops.unsol_event = alc262_fujitsu_unsol_event;
+		spec->init_hook = alc262_auto_init;
 		
 	return 0;
 }
@@ -5262,15 +5429,12 @@ static int alc861_parse_auto_config(struct hda_codec *codec)
 	return 1;
 }
 
-/* init callback for auto-configuration model -- overriding the default init */
-static int alc861_auto_init(struct hda_codec *codec)
+/* additional initialization for auto-configuration model */
+static void alc861_auto_init(struct hda_codec *codec)
 {
-	alc_init(codec);
 	alc861_auto_init_multi_out(codec);
 	alc861_auto_init_hp_out(codec);
 	alc861_auto_init_analog_input(codec);
-
-	return 0;
 }
 
 
@@ -5368,7 +5532,7 @@ static int patch_alc861(struct hda_codec *codec)
 
 	codec->patch_ops = alc_patch_ops;
 	if (board_config == ALC861_AUTO)
-		codec->patch_ops.init = alc861_auto_init;
+		spec->init_hook = alc861_auto_init;
 		
 	return 0;
 }
-- 
cgit v1.2.3


From 9230d2148a0c53188c216b446cf17ea213ebca8a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 13 Mar 2006 13:49:49 +0100
Subject: [ALSA] hda-codec - Fix support of laptops with AD1986A codec

Modules: Documentation,HDA Codec driver

Fix the support of laptops with AD1986A HD-audio codec.
Added new models '3stack' and 'laptop'.  Currently, fixed for FSC V2060
and Samsung M50.

Also fixed the description of missing models in ALSA-Configuration.txt.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 Documentation/sound/alsa/ALSA-Configuration.txt |  18 ++++
 sound/pci/hda/patch_analog.c                    | 131 ++++++++++++++++++++++++
 2 files changed, 149 insertions(+)

(limited to 'Documentation')

diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index c12dab05176a..1065beed8d75 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -737,6 +737,24 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  allout	5-jack in back, 2-jack in front, SPDIF out
 	  auto		auto-config reading BIOS (default)
 
+	AD1981
+	  basic		3-jack (default)
+	  hp		HP nx6320
+
+	AD1986A
+	  6stack	6-jack, separate surrounds (default)
+	  3stack	3-stack, shared surrounds
+	  laptop	2-channel only (FSC V2060, Samsung M50)
+
+	AD1988
+	  6stack	6-jack
+	  6stack-dig	ditto with SPDIF
+	  3stack	3-jack
+	  3stack-dig	ditto with SPDIF
+	  laptop	3-jack with hp-jack automute
+	  laptop-dig	ditto with SPDIF
+	  auto		auto-confgi reading BIOS (default)
+
     If the default configuration doesn't work and one of the above
     matches with your device, report it together with the PCI
     subsystem ID (output of "lspci -nv") to ALSA BTS or alsa-devel
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 4d363cfa6c81..cdcc815a9320 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -379,6 +379,14 @@ static int ad198x_eapd_put(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
+static int ad198x_ch_mode_info(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_info *uinfo);
+static int ad198x_ch_mode_get(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol);
+static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol);
+
+
 /*
  * AD1986A specific
  */
@@ -527,6 +535,52 @@ static struct snd_kcontrol_new ad1986a_mixers[] = {
 	{ } /* end */
 };
 
+/* additional mixers for 3stack mode */
+static struct snd_kcontrol_new ad1986a_3st_mixers[] = {
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Channel Mode",
+		.info = ad198x_ch_mode_info,
+		.get = ad198x_ch_mode_get,
+		.put = ad198x_ch_mode_put,
+	},
+	{ } /* end */
+};
+
+/* laptop model - 2ch only */
+static hda_nid_t ad1986a_laptop_dac_nids[1] = { AD1986A_FRONT_DAC };
+
+static struct snd_kcontrol_new ad1986a_laptop_mixers[] = {
+	HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Master Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Master Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+	/* HDA_CODEC_VOLUME("Headphone Playback Volume", 0x1a, 0x0, HDA_OUTPUT),
+	   HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x0, HDA_OUTPUT), */
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Aux Playback Volume", 0x16, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Aux Playback Switch", 0x16, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
+	/* HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x18, 0x0, HDA_OUTPUT),
+	   HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x18, 0x0, HDA_OUTPUT),
+	   HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT),
+	   HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT), */
+	HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Capture Source",
+		.info = ad198x_mux_enum_info,
+		.get = ad198x_mux_enum_get,
+		.put = ad198x_mux_enum_put,
+	},
+	{ } /* end */
+};
+
 /*
  * initialization verbs
  */
@@ -585,10 +639,68 @@ static struct hda_verb ad1986a_init_verbs[] = {
 	{ } /* end */
 };
 
+/* additional verbs for 3-stack model */
+static struct hda_verb ad1986a_3st_init_verbs[] = {
+ 	/* Mic and line-in selectors */
+	{0x0f, AC_VERB_SET_CONNECT_SEL, 0x2},
+	{0x10, AC_VERB_SET_CONNECT_SEL, 0x1},
+ 	{ } /* end */
+};
+
+static struct hda_verb ad1986a_ch2_init[] = {
+	/* Surround out -> Line In */
+	{ 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+	{ 0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+	/* CLFE -> Mic in */
+	{ 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+	{ 0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+	{ } /* end */
+};
+
+static struct hda_verb ad1986a_ch4_init[] = {
+	/* Surround out -> Surround */
+	{ 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+	{ 0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+	/* CLFE -> Mic in */
+	{ 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+	{ 0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+	{ } /* end */
+};
+
+static struct hda_verb ad1986a_ch6_init[] = {
+	/* Surround out -> Surround out */
+	{ 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+	{ 0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+	/* CLFE -> CLFE */
+	{ 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+	{ 0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+	{ } /* end */
+};
+
+static struct hda_channel_mode ad1986a_modes[3] = {
+	{ 2, ad1986a_ch2_init },
+	{ 4, ad1986a_ch4_init },
+	{ 6, ad1986a_ch6_init },
+};
+
+/* models */
+enum { AD1986A_6STACK, AD1986A_3STACK, AD1986A_LAPTOP };
+
+static struct hda_board_config ad1986a_cfg_tbl[] = {
+	{ .modelname = "6stack",	.config = AD1986A_6STACK },
+	{ .modelname = "3stack",	.config = AD1986A_3STACK },
+	{ .modelname = "laptop",	.config = AD1986A_LAPTOP },
+	{ .pci_subvendor = 0x144d, .pci_subdevice = 0xc01e,
+	  .config = AD1986A_LAPTOP }, /* FSC V2060 */
+	{ .pci_subvendor = 0x17c0, .pci_subdevice = 0x2017,
+	  .config = AD1986A_LAPTOP }, /* Samsung M50 */
+	{}
+};
 
 static int patch_ad1986a(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
+	int board_config;
 
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
@@ -612,6 +724,25 @@ static int patch_ad1986a(struct hda_codec *codec)
 
 	codec->patch_ops = ad198x_patch_ops;
 
+	/* override some parameters */
+	board_config = snd_hda_check_board_config(codec, ad1986a_cfg_tbl);
+	switch (board_config) {
+	case AD1986A_3STACK:
+		spec->num_mixers = 2;
+		spec->mixers[1] = ad1986a_3st_mixers;
+		spec->num_init_verbs = 2;
+		spec->init_verbs[1] = ad1986a_3st_init_verbs;
+		spec->channel_mode = ad1986a_modes;
+		spec->num_channel_mode = ARRAY_SIZE(ad1986a_modes);
+		break;
+	case AD1986A_LAPTOP:
+		spec->mixers[0] = ad1986a_laptop_mixers;
+		spec->multiout.max_channels = 2;
+		spec->multiout.num_dacs = 1;
+		spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
+		break;
+	}
+
 	return 0;
 }
 
-- 
cgit v1.2.3


From e311334ee6bdd173d53be52f4fdffa5f39652e26 Mon Sep 17 00:00:00 2001
From: Thibault LE MEUR <Thibault.LeMeur@supelec.fr>
Date: Tue, 14 Mar 2006 11:44:53 +0100
Subject: [ALSA] Fixes audiophile usb analog capture with the new device_setup
 parameter

Modules: Documentation,USB generic driver

The patch adds the 'device_setup' module parameter and a specific
quirk to correctly initialize the audiophile usb device: this fixes
the distorted sound bug on the Analog capture port. Backward
compatibility is achieved by simply omitting the new parameter.

Signed-off-by: Thibault LE MEUR <Thibault.LeMeur@supelec.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 Documentation/sound/alsa/ALSA-Configuration.txt |   3 +
 Documentation/sound/alsa/Audiophile-Usb.txt     | 330 ++++++++++++++++++++++++
 sound/usb/usbaudio.c                            |  52 +++-
 3 files changed, 384 insertions(+), 1 deletion(-)
 create mode 100644 Documentation/sound/alsa/Audiophile-Usb.txt

(limited to 'Documentation')

diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 1065beed8d75..f947c4b04ab8 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -1411,6 +1411,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     vid             - Vendor ID for the device (optional)
     pid             - Product ID for the device (optional)
+    device_setup    - Device specific magic number (optional)
+                    - Influence depends on the device
+                    - Default: 0x0000 
 
     This module supports multiple devices, autoprobe and hotplugging.
 
diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt
new file mode 100644
index 000000000000..3ba45adbf040
--- /dev/null
+++ b/Documentation/sound/alsa/Audiophile-Usb.txt
@@ -0,0 +1,330 @@
+	Guide to using M-Audio Audiophile USB with ALSA and Jack	v1.1
+	========================================================
+
+	    Thibault Le Meur <Thibault.LeMeur@supelec.fr>
+
+This document is a guide to using the M-Audio Audiophile USB (tm) device with 
+ALSA and JACK.
+
+1 - Audiophile USB Specs and correct usage
+==========================================
+This part is a reminder of important facts about the functions and limitations 
+of the device.
+
+The device has 4 audio interfaces, and 2 MIDI ports:
+ * Analog Stereo Input (Ai)
+ * Analog Stereo Output (Ao)
+ * Digital Stereo Input (Di)
+ * Digital Stereo Output (Do)
+ * Midi In (Mi)
+ * Midi Out (Mo)
+
+The internal DAC/ADC has the following caracteristics:
+* sample depth of 16 or 24 bits
+* sample rate from 8kHz to 96kHz
+* Two ports can't use different sample depths at the same time.Moreover, the 
+Audiophile USB documentation gives the following Warning: "Please exit any 
+audio application running before switching between bit depths"
+
+Due to the USB 1.1 bandwidth limitation, a limited number of interfaces can be 
+activated at the same time depending on the audio mode selected:
+ * 16-bit/48kHz ==> 4 channels in/ 4 channels out
+   - Ai+Ao+Di+Do
+ * 24-bit/48kHz ==> 4 channels in/2 channels out, 
+                    or 2 channels in/4 channels out
+   - Ai+Ao+Do or Ai+Di+Ao or Ai+Di+Do or Di+Ao+Do
+ * 24-bit/96kHz ==> 2 channels in, or 2 channels out (half duplex only)
+   - Ai or Ao or Di or Do
+
+Important facts about the Digital interface:
+--------------------------------------------
+ * The Do port additionnaly supports surround-encoded AC-3 and DTS passthrough, 
+though I haven't tested it under linux
+   - Note that in this setup only the Do interface can be enabled
+ * Apart from recording an audio digital stream, enabling the Di port is a way 
+to syncrhonize the device to an external sample clock
+   - As a consequence, the Di port must be enable only if an active Digital 
+source is connected
+   - Enabling Di when no digital source is connected can result in a 
+synchronization error (for instance sound played at an odd sample rate)
+
+
+2 - Audiophile USB support in ALSA
+==================================
+
+2.1 - MIDI ports
+----------------
+The Audiophile USB MIDI ports will be automatically supported once the 
+following modules have been loaded:
+ * snd-usb-audio
+ * snd-seq
+ * snd-seq-midi
+
+No additionnal setting is required.
+
+2.2 - Audio ports
+-----------------
+
+Audio functions of the Audiophile USB device are handled by the snd-usb-audio 
+module. This module can work in a default mode (without any device-specific 
+parameter), or in an advanced mode with the device-specific parameter called 
+"device_setup".
+
+2.2.1 - Default Alsa driver mode
+
+The default behaviour of the snd-usb-audio driver is to parse the device 
+capabilities at startup and enable all functions inside the device (including 
+all ports at any sample rates and any sample depths supported). This approach 
+has the advantage to let the driver easily switch from sample rates/depths 
+automatically according to the need of the application claiming the device.
+
+In this case the Audiophile ports are mapped to alsa pcm devices in the 
+following way (I suppose the device's index is 1):
+ * hw:1,0 is Ao in playback and Di in capture
+ * hw:1,1 is Do in playback and Ai in capture
+ * hw:1,2 is Do in AC3/DTS passthrough mode
+
+You must note as well that the device uses Big Endian byte encoding so that 
+supported audio format are S16_BE  for 16-bit depth modes and S24_3BE for 
+24-bits depth mode. One exception is the hw:1,2 port which is Little Endian 
+compliant and thus uses S16_LE.
+
+Examples:
+ * playing a S24_3BE encoded raw file to the Ao port
+   % aplay -D hw:1,0 -c2 -t raw -r48000 -fS24_3BE test.raw
+ * recording a  S24_3BE encoded raw file from the Ai port
+   % arecord -D hw:1,1 -c2  -t raw -r48000 -fS24_3BE test.raw
+ * playing a S16_BE encoded raw file to the Do port
+   % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test.raw
+
+If you're happy with the default Alsa driver setup and don't experience any 
+issue with this mode, then you can skip the following chapter.
+
+2.2.2 - Advanced module setup
+
+Due to the hardware constraints described above, the device initialization made 
+by the Alsa driver in default mode may result in a corrupted state of the 
+device. For instance, a particularly annoying issue is that the sound captured 
+from the Ai port sounds distorted (as if boosted with an excessive high volume 
+gain).
+
+For people having this problem, the snd-usb-audio module has a new module 
+parameter called "device_setup".
+
+2.2.2.1 - Initializing the working mode of the Audiohile USB
+
+As far as the Audiohile USB device is concerned, this value let the user 
+specify:
+ * the sample depth
+ * the sample rate
+ * whether the Di port is used or not 
+
+Here is a list of supported device_setup values for this device:
+ * device_setup=0x00 (or omitted)
+   - Alsa driver default mode
+   - maintains backward compatibility with setups that do not use this 
+     parameter by not introducing any change
+   - results sometimes in corrupted sound as decribed earlier
+ * device_setup=0x01
+   - 16bits 48kHz mode with Di disabled
+   - Ai,Ao,Do can be used at the same time
+   - hw:1,0 is not available in capture mode
+   - hw:1,2 is not available
+ * device_setup=0x11
+   - 16bits 48kHz mode with Di enabled
+   - Ai,Ao,Di,Do can be used at the same time
+   - hw:1,0 is available in capture mode
+   - hw:1,2 is not available
+ * device_setup=0x09
+   - 24bits 48kHz mode with Di disabled
+   - Ai,Ao,Do can be used at the same time
+   - hw:1,0 is not available in capture mode
+   - hw:1,2 is not available
+ * device_setup=0x19
+   - 24bits 48kHz mode with Di enabled
+   - 3 ports from {Ai,Ao,Di,Do} can be used at the same time
+   - hw:1,0 is available in capture mode and an active digital source must be 
+     connected to Di
+   - hw:1,2 is not available
+ * device_setup=0x0D or 0x10
+   - 24bits 96kHz mode
+   - Di is enabled by default for this mode but does not need to be connected 
+     to an active source
+   - Only 1 port from {Ai,Ao,Di,Do} can be used at the same time
+   - hw:1,0 is available in captured mode
+   - hw:1,2 is not available
+ * device_setup=0x03
+   - 16bits 48kHz mode with only the Do port enabled 
+   - AC3 with DTS passthru (not tested)
+   - Caution with this setup the Do port is mapped to the pcm device hw:1,0
+
+2.2.2.2 - Setting and switching configurations with the device_setup parameter
+
+The parameter can be given:
+ * By manually probing the device (as root):
+   # modprobe -r snd-usb-audio
+   # modprobe snd-usb-audio index=1 device_setup=0x09
+ * Or while configuring the modules options in your modules configuration file
+   - For Fedora distributions, edit the /etc/modprobe.conf file:
+       alias snd-card-1 snd-usb-audio
+       options snd-usb-audio index=1 device_setup=0x09
+
+IMPORTANT NOTE WHEN SWITCHING CONFIGURATION:
+-------------------------------------------
+ * You may need to _first_ intialize the module with the correct device_setup 
+   parameter and _only_after_ turn on the Audiophile USB device
+ * This is especially true when switching the sample depth:
+   - first trun off the device
+   - de-register the snd-usb-audio module
+   - change the device_setup parameter (by either manually reprobing the module 
+     or changing modprobe.conf)
+   - turn on the device
+
+2.2.2.3 - Setting and switching configurations with the device_setup parameter
+
+If you want to understand the device_setup magic numbers for the Audiophile 
+USB, you need some very basic understanding of binary computation. However, 
+this is not required to use the parameter and you may skip thi section.
+
+The device_setup is one byte long and its structure is the following:
+
+       +---+---+---+---+---+---+---+---+
+       | b7| b6| b5| b4| b3| b2| b1| b0|
+       +---+---+---+---+---+---+---+---+
+       | 0 | 0 | 0 | Di|24B|96K|DTS|SET|
+       +---+---+---+---+---+---+---+---+
+
+Where:
+ * b0 is the "SET" bit
+   - it MUST be set if device_setup is initialized 
+ * b1 is the "DTS" bit
+   - it is set only for Digital output with DTS/AC3
+   - this setup is not tested
+ * b2 is the Rate selection flag
+   - When set to "1" the rate range is 48.1-96kHz
+   - Otherwise the sample rate range is 8-48kHz
+ * b3 is the bit depth selection flag
+   - When set to "1" samples are 24bits long
+   - Otherwise they are 16bits long
+   - Note that b2 implies b3 as the 96kHz mode is only supported for 24 bits 
+     samples
+ * b4 is the Digital input flag
+   - When set to "1" the device assumes that an active digital source is 
+     connected 
+   - You shouldn't enable Di if no source is seen on the port (this leads to 
+     synchronization issues)
+   - b4 is implied by b2 (since only one port is enabled at a time no synch 
+     error can occur) 
+ * b5 to b7 are reserved for future uses, and must be set to "0"
+   - might become Ao, Do, Ai, for b7, b6, b4 respectively
+
+Caution:
+ * there is no check on the value you will give to device_setup
+   - for instance choosing 0x05 (16bits 96kHz) will fail back to 0x09 since 
+     b2 implies b3. But _there_will_be_no_warning_ in /var/log/messages
+ * Hardware constraints due to the USB bus limitation aren't checked
+   - choosing b2 will prepare all interfaces for 24bits/96kHz but you'll
+     only be able to use one at the same time
+
+2.2.3 -  Technical Details for Audiophile Usb 
+
+You may safely skip this section if you're not interrested in driver 
+development.
+
+This section describes some internals aspect of the device and summarize the 
+data I got by usb-snooping the windows and linux drivers.
+
+The M-Audio Audiophile USB has 7 Usb Interfaces:
+a "USB interface":
+ * Usb Interface nb.0
+ * Usb Interface nb.1
+   - Audio Control function
+ * Usb Interface nb.2
+   - Analog Output
+ * Usb Interface nb.3
+   - Digital Output
+ * Usb Interface nb.4
+   - Analog Input
+ * Usb Interface nb.5
+   - Digital Input
+ * Usb Interface nb.6
+   - MIDI interface compliant with the MIDIMAN quirk 
+
+Each interface has 5 altsettings (AltSet 1,2,3,4,5) except:
+ * Interface 3 (Digital Out) has an extra Alset nb.6 
+ * Interface 5 (Digital In) does not have Alset nb.3 and 5 
+
+Here is a short description of the AltSettings capabilities:
+ * AltSettings 1 corresponds to
+  - 24-bit depth, 48.1-96kHz sample mode
+  - Adaptive playback (Ao and Do), Synch capture (Ai), or Asynch capture (Di)
+ * AltSettings 2 corresponds to
+  - 24-bit depth, 8-48kHz sample mode
+  - Asynch capture and playback  (Ao,Ai,Do,Di)
+ * AltSettings 3 corresponds to
+  - 24-bit depth, 8-48kHz sample mode
+  - Synch capture (Ai) and Adaptive playback (Ao,Do)
+ * AltSettings 4 corresponds to
+  - 16-bit depth, 8-48kHz sample mode
+  - Asynch capture and playback  (Ao,Ai,Do,Di)
+ * AltSettings 5 corresponds to
+  - 16-bit depth, 8-48kHz sample mode
+  - Synch capture (Ai) and Adaptive playback (Ao,Do)
+ * AltSettings 6 corresponds to
+  - 16-bit depth, 8-48kHz sample mode
+  - Synch playback (Do), audio format type III IEC1937_AC-3
+
+In order to ensure a correct intialization of the device, the driver 
+_must_know_ how the device will be used:
+ * if DTS is choosen, only Interface 2 with AltSet nb.6 must be
+   registered
+ * if 96KHz only AltSets nb.1 of each interface must be selected
+ * if samples are using 24bits/48KHz then AltSet 2 must me used if
+   Digital input is connected, and only AltSet nb.3 if Digital input
+   is not connected
+ * if samples are using 16bits/48KHz then AltSet 4 must me used if
+   Digital input is connected, and only AltSet nb.5 if Digital input
+   is not connected
+
+When device_setup is given as a parameter to the snd-usb-audio module, the 
+parse_audio_enpoint function uses a quirk called 
+"audiophile_skip_setting_quirk" in order to prevent AltSettings not 
+corresponding to device_setup from being registered in the driver.
+
+3 - Audiophile USB and Jack support
+===================================
+
+This section deals with support of the Audiophile USB device in Jack.
+The main issue regarding this support is that the device is Big Endian 
+compliant.
+
+3.1 - Using the plug alsa plugin
+--------------------------------
+
+Jack doesn't directly support big endian devices. Thus, one way to have support 
+for this device with Alsa is to use the Alsa "plug" converter.
+
+For instance here is one way to run Jack with 2 playback channels on Ao and 2 
+capture channels from Ai:
+  % jackd -R -dalsa -dplughw:1 -r48000 -p256 -n2 -D -Cplughw:1,1
+
+
+However you may see the following warning message:
+"You appear to be using the ALSA software "plug" layer, probably a result of 
+using the "default" ALSA device. This is less efficient than it could be. 
+Consider using a hardware device instead rather than using the plug layer."
+
+
+3.2 - Patching alsa to use direct pcm device
+-------------------------------------------
+A patch for Jack by Andreas Steinmetz adds support for Big Endian devices. 
+However it has not been included in the CVS tree.
+
+You can find it at the following URL:
+http://sourceforge.net/tracker/index.php?func=detail&aid=1289682&group_id=39687&
+atid=425939
+
+After having applied the patch you can run jackd with the following command 
+line:
+# /usr/local/bin/jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1
+
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index 6fad2c40c77c..4e614ac39f21 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -70,6 +70,7 @@ static int vid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 }; /* Vendor ID for
 static int pid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 }; /* Product ID for this card */
 static int nrpacks = 4;		/* max. number of packets per urb */
 static int async_unlink = 1;
+static int device_setup[SNDRV_CARDS]; /* device parameter for this card*/
 
 module_param_array(index, int, NULL, 0444);
 MODULE_PARM_DESC(index, "Index value for the USB audio adapter.");
@@ -85,6 +86,8 @@ module_param(nrpacks, int, 0644);
 MODULE_PARM_DESC(nrpacks, "Max. number of packets per URB.");
 module_param(async_unlink, bool, 0444);
 MODULE_PARM_DESC(async_unlink, "Use async unlink mode.");
+module_param_array(device_setup, int, NULL, 0444);
+MODULE_PARM_DESC(device_setup, "Specific device setup (if needed).");
 
 
 /*
@@ -2547,6 +2550,8 @@ static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp
 	return 0;
 }
 
+static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip,
+					 int iface, int altno);
 static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
 {
 	struct usb_device *dev;
@@ -2581,6 +2586,12 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
 		stream = (get_endpoint(alts, 0)->bEndpointAddress & USB_DIR_IN) ?
 			SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
 		altno = altsd->bAlternateSetting;
+	
+		/* audiophile usb: skip altsets incompatible with device_setup
+		 */
+		if (chip->usb_id == USB_ID(0x0763, 0x2003) && 
+		    audiophile_skip_setting_quirk(chip, iface_no, altno))
+			continue;
 
 		/* get audio formats */
 		fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL);
@@ -2675,7 +2686,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
 			continue;
 		}
 
-		snd_printdd(KERN_INFO "%d:%u:%d: add audio endpoint 0x%x\n", dev->devnum, iface_no, i, fp->endpoint);
+		snd_printdd(KERN_INFO "%d:%u:%d: add audio endpoint 0x%x\n", dev->devnum, iface_no, altno, fp->endpoint);
 		err = add_audio_endpoint(chip, stream, fp);
 		if (err < 0) {
 			kfree(fp->rate_table);
@@ -3083,6 +3094,45 @@ static int snd_usb_audigy2nx_boot_quirk(struct usb_device *dev)
 	return 0;
 }
 
+/*
+ * Setup quirks
+ */
+#define AUDIOPHILE_SET			0x01 /* if set, parse device_setup */
+#define AUDIOPHILE_SET_DTS              0x02 /* if set, enable DTS Digital Output */
+#define AUDIOPHILE_SET_96K              0x04 /* 48-96KHz rate if set, 8-48KHz otherwise */
+#define AUDIOPHILE_SET_24B		0x08 /* 24bits sample if set, 16bits otherwise */
+#define AUDIOPHILE_SET_DI		0x10 /* if set, enable Digital Input */
+#define AUDIOPHILE_SET_MASK		0x1F /* bit mask for setup value */
+#define AUDIOPHILE_SET_24B_48K_DI	0x19 /* value for 24bits+48KHz+Digital Input */
+#define AUDIOPHILE_SET_24B_48K_NOTDI	0x09 /* value for 24bits+48KHz+No Digital Input */
+#define AUDIOPHILE_SET_16B_48K_DI	0x11 /* value for 16bits+48KHz+Digital Input */
+#define AUDIOPHILE_SET_16B_48K_NOTDI	0x01 /* value for 16bits+48KHz+No Digital Input */
+
+static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip,
+					 int iface, int altno)
+{
+	if (device_setup[chip->index] & AUDIOPHILE_SET) {
+		if ((device_setup[chip->index] & AUDIOPHILE_SET_DTS)
+		    && altno != 6)
+			return 1; /* skip this altsetting */
+		if ((device_setup[chip->index] & AUDIOPHILE_SET_96K)
+		    && altno != 1)
+			return 1; /* skip this altsetting */
+		if ((device_setup[chip->index] & AUDIOPHILE_SET_MASK) ==
+		    AUDIOPHILE_SET_24B_48K_DI && altno != 2)
+			return 1; /* skip this altsetting */
+		if ((device_setup[chip->index] & AUDIOPHILE_SET_MASK) ==
+		    AUDIOPHILE_SET_24B_48K_NOTDI && altno != 3)
+			return 1; /* skip this altsetting */
+		if ((device_setup[chip->index] & AUDIOPHILE_SET_MASK) ==
+		    AUDIOPHILE_SET_16B_48K_DI && altno != 4)
+			return 1; /* skip this altsetting */
+		if ((device_setup[chip->index] & AUDIOPHILE_SET_MASK) ==
+		    AUDIOPHILE_SET_16B_48K_NOTDI && altno != 5)
+			return 1; /* skip this altsetting */
+	}	
+	return 0; /* keep this altsetting */
+}
 
 /*
  * audio-interface quirks
-- 
cgit v1.2.3


From db064e503419c32df463326a3891a973bb30582e Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 16 Mar 2006 16:04:58 +0100
Subject: [ALSA] hda-codec - Add support for VAIO FE550G and SZ110

Modules: Documentation,HDA Codec driver

Add support for VAIO FE550G and SZ110 laptops with Sigmatel codec (7661).
The new model 'vaio' is added.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 Documentation/sound/alsa/ALSA-Configuration.txt |   3 +
 sound/pci/hda/patch_sigmatel.c                  | 161 ++++++++++++++++++++++++
 2 files changed, 164 insertions(+)

(limited to 'Documentation')

diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index f947c4b04ab8..dcc9cd9633dd 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -755,6 +755,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  laptop-dig	ditto with SPDIF
 	  auto		auto-confgi reading BIOS (default)
 
+	STAC7661(?)
+	  vaio		Setup for VAIO FE550G/SZ110
+
     If the default configuration doesn't work and one of the above
     matches with your device, report it together with the PCI
     subsystem ID (output of "lspci -nv") to ALSA BTS or alsa-devel
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 45ddf548d6fb..240958df26ce 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1147,6 +1147,166 @@ static int patch_stac927x(struct hda_codec *codec)
 	return 0;
 }
 
+/*
+ * STAC 7661(?) hack
+ */
+
+/* static config for Sony VAIO FE550G */
+static hda_nid_t vaio_dacs[] = { 0x2 };
+#define VAIO_HP_DAC	0x5
+static hda_nid_t vaio_adcs[] = { 0x8 /*,0x6*/ };
+static hda_nid_t vaio_mux_nids[] = { 0x15 };
+
+static struct hda_input_mux vaio_mux = {
+	.num_items = 2,
+	.items = {
+		/* { "HP", 0x0 },
+		   { "Unknown", 0x1 }, */
+		{ "Mic", 0x2 },
+		{ "PCM", 0x3 },
+	}
+};
+
+static struct hda_verb vaio_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 */
+	{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 */
+	{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)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	long *valp = ucontrol->value.integer.value;
+	int change;
+
+	change = snd_hda_codec_amp_update(codec, 0x02, 0, HDA_OUTPUT, 0,
+					  0x7f, valp[0] & 0x7f);
+	change |= snd_hda_codec_amp_update(codec, 0x02, 1, HDA_OUTPUT, 0,
+					   0x7f, valp[1] & 0x7f);
+	snd_hda_codec_amp_update(codec, 0x05, 0, HDA_OUTPUT, 0,
+				 0x7f, valp[0] & 0x7f);
+	snd_hda_codec_amp_update(codec, 0x05, 1, HDA_OUTPUT, 0,
+				 0x7f, valp[1] & 0x7f);
+	return change;
+}
+
+/* bind volumes of both NID 0x02 and 0x05 */
+static int vaio_master_sw_put(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	long *valp = ucontrol->value.integer.value;
+	int change;
+
+	change = snd_hda_codec_amp_update(codec, 0x02, 0, HDA_OUTPUT, 0,
+					  0x80, valp[0] & 0x80);
+	change |= snd_hda_codec_amp_update(codec, 0x02, 1, HDA_OUTPUT, 0,
+					   0x80, valp[1] & 0x80);
+	snd_hda_codec_amp_update(codec, 0x05, 0, HDA_OUTPUT, 0,
+				 0x80, valp[0] & 0x80);
+	snd_hda_codec_amp_update(codec, 0x05, 1, HDA_OUTPUT, 0,
+				 0x80, valp[1] & 0x80);
+	return change;
+}
+
+static struct snd_kcontrol_new vaio_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),
+	},
+	{
+		.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 hda_codec_ops stac7661_patch_ops = {
+	.build_controls = stac92xx_build_controls,
+	.build_pcms = stac92xx_build_pcms,
+	.init = stac92xx_init,
+	.free = stac92xx_free,
+#ifdef CONFIG_PM
+	.resume = stac92xx_resume,
+#endif
+};
+
+enum { STAC7661_VAIO };
+
+static struct hda_board_config stac7661_cfg_tbl[] = {
+	{ .modelname = "vaio", .config = STAC7661_VAIO },
+	{ .pci_subvendor = 0x104d, .pci_subdevice = 0x81e6,
+	  .config = STAC7661_VAIO },
+	{ .pci_subvendor = 0x104d, .pci_subdevice = 0x81ef,
+	  .config = STAC7661_VAIO },
+	{}
+};
+
+static int patch_stac7661(struct hda_codec *codec)
+{
+	struct sigmatel_spec *spec;
+	int board_config;
+
+	board_config = snd_hda_check_board_config(codec, stac7661_cfg_tbl);
+	if (board_config < 0)
+		/* unknown config, let generic-parser do its job... */
+		return snd_hda_parse_generic_codec(codec);
+	
+	spec  = kzalloc(sizeof(*spec), GFP_KERNEL);
+	if (spec == NULL)
+		return -ENOMEM;
+
+	codec->spec = spec;
+	switch (board_config) {
+	case STAC7661_VAIO:
+		spec->mixer = vaio_mixer;
+		spec->init = vaio_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;
+	return 0;
+}
+
+
 /*
  * patch entries
  */
@@ -1168,5 +1328,6 @@ 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 },
 	{} /* terminator */
 };
-- 
cgit v1.2.3


From 825aa97241b46d2819c1db984c86a1a9df41b8e1 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 17 Mar 2006 10:50:49 +0100
Subject: [ALSA] hda-codec - Fix for Samsung R65 and ASUS A6J

Modules: Documentation,HDA Codec driver

Added a new model 'laptop-eapd' to AD1986A codec for Samsung R65 and
ASUS A6J laptops.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 Documentation/sound/alsa/ALSA-Configuration.txt |   1 +
 sound/pci/hda/patch_analog.c                    | 116 +++++++++++++++++++++++-
 2 files changed, 116 insertions(+), 1 deletion(-)

(limited to 'Documentation')

diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index dcc9cd9633dd..1def6049784c 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -745,6 +745,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  6stack	6-jack, separate surrounds (default)
 	  3stack	3-stack, shared surrounds
 	  laptop	2-channel only (FSC V2060, Samsung M50)
+	  laptop-eapd	2-channel with EAPD (Samsung R65, ASUS A6J)
 
 	AD1988
 	  6stack	6-jack
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index cdcc815a9320..2b14fa74a8fd 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -581,6 +581,97 @@ static struct snd_kcontrol_new ad1986a_laptop_mixers[] = {
 	{ } /* end */
 };
 
+/* laptop-eapd model - 2ch only */
+
+/* master controls both pins 0x1a and 0x1b */
+static int ad1986a_laptop_master_vol_put(struct snd_kcontrol *kcontrol,
+					 struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	long *valp = ucontrol->value.integer.value;
+	int change;
+
+	change = snd_hda_codec_amp_update(codec, 0x1a, 0, HDA_OUTPUT, 0,
+					  0x7f, valp[0] & 0x7f);
+	change |= snd_hda_codec_amp_update(codec, 0x1a, 1, HDA_OUTPUT, 0,
+					   0x7f, valp[1] & 0x7f);
+	snd_hda_codec_amp_update(codec, 0x1b, 0, HDA_OUTPUT, 0,
+				 0x7f, valp[0] & 0x7f);
+	snd_hda_codec_amp_update(codec, 0x1b, 1, HDA_OUTPUT, 0,
+				 0x7f, valp[1] & 0x7f);
+	return change;
+}
+
+static int ad1986a_laptop_master_sw_put(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	long *valp = ucontrol->value.integer.value;
+	int change;
+
+	change = snd_hda_codec_amp_update(codec, 0x1a, 0, HDA_OUTPUT, 0,
+					  0x80, valp[0] ? 0 : 0x80);
+	change |= snd_hda_codec_amp_update(codec, 0x1a, 1, HDA_OUTPUT, 0,
+					   0x80, valp[1] ? 0 : 0x80);
+	snd_hda_codec_amp_update(codec, 0x1b, 0, HDA_OUTPUT, 0,
+				 0x80, valp[0] ? 0 : 0x80);
+	snd_hda_codec_amp_update(codec, 0x1b, 1, HDA_OUTPUT, 0,
+				 0x80, valp[1] ? 0 : 0x80);
+	return change;
+}
+
+static struct hda_input_mux ad1986a_laptop_eapd_capture_source = {
+	.num_items = 3,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Internal Mic", 0x4 },
+		{ "Mix", 0x5 },
+	},
+};
+
+static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = {
+	{
+		.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 = ad1986a_laptop_master_vol_put,
+		.private_value = HDA_COMPOSE_AMP_VAL(0x1a, 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 = ad1986a_laptop_master_sw_put,
+		.private_value = HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT),
+	},
+	HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Capture Source",
+		.info = ad198x_mux_enum_info,
+		.get = ad198x_mux_enum_get,
+		.put = ad198x_mux_enum_put,
+	},
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "External Amplifier",
+		.info = ad198x_eapd_info,
+		.get = ad198x_eapd_get,
+		.put = ad198x_eapd_put,
+		.private_value = 0x1b | (1 << 8), /* port-D, inversed */
+	},
+	{ } /* end */
+};
+
 /*
  * initialization verbs
  */
@@ -683,8 +774,14 @@ static struct hda_channel_mode ad1986a_modes[3] = {
 	{ 6, ad1986a_ch6_init },
 };
 
+/* eapd initialization */
+static struct hda_verb ad1986a_eapd_init_verbs[] = {
+	{0x1b, AC_VERB_SET_EAPD_BTLENABLE, 0x00},
+	{}
+};
+
 /* models */
-enum { AD1986A_6STACK, AD1986A_3STACK, AD1986A_LAPTOP };
+enum { AD1986A_6STACK, AD1986A_3STACK, AD1986A_LAPTOP, AD1986A_LAPTOP_EAPD };
 
 static struct hda_board_config ad1986a_cfg_tbl[] = {
 	{ .modelname = "6stack",	.config = AD1986A_6STACK },
@@ -694,6 +791,13 @@ static struct hda_board_config ad1986a_cfg_tbl[] = {
 	  .config = AD1986A_LAPTOP }, /* FSC V2060 */
 	{ .pci_subvendor = 0x17c0, .pci_subdevice = 0x2017,
 	  .config = AD1986A_LAPTOP }, /* Samsung M50 */
+	{ .pci_subvendor = 0x1043, .pci_subdevice = 0x818f,
+	  .config = AD1986A_LAPTOP }, /* ASUS P5GV-MX */
+	{ .modelname = "laptop-eapd",	.config = AD1986A_LAPTOP_EAPD },
+	{ .pci_subvendor = 0x144d, .pci_subdevice = 0xc024,
+	  .config = AD1986A_LAPTOP_EAPD }, /* Samsung R65-T2300 Charis */
+	{ .pci_subvendor = 0x1043, .pci_subdevice = 0x1213,
+	  .config = AD1986A_LAPTOP_EAPD }, /* ASUS A6J */
 	{}
 };
 
@@ -741,6 +845,16 @@ static int patch_ad1986a(struct hda_codec *codec)
 		spec->multiout.num_dacs = 1;
 		spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
 		break;
+	case AD1986A_LAPTOP_EAPD:
+		spec->mixers[0] = ad1986a_laptop_eapd_mixers;
+		spec->num_init_verbs = 2;
+		spec->init_verbs[1] = ad1986a_eapd_init_verbs;
+		spec->multiout.max_channels = 2;
+		spec->multiout.num_dacs = 1;
+		spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
+		spec->multiout.dig_out_nid = 0;
+		spec->input_mux = &ad1986a_laptop_eapd_capture_source;
+		break;
 	}
 
 	return 0;
-- 
cgit v1.2.3


From 19739fef0203d2f3eecc9c4b1ef25b57d85f2b30 Mon Sep 17 00:00:00 2001
From: Thibault LE MEUR <Thibault.LeMeur@supelec.fr>
Date: Tue, 21 Mar 2006 11:06:40 +0100
Subject: [ALSA] Fixes typos in Audiophile-USB.txt

Modules: Documentation

Fixes typos in Audiophile-USB.txt.

Signed-off-by: Thibault LE MEUR <Thibault.LeMeur@supelec.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 Documentation/sound/alsa/Audiophile-Usb.txt | 29 ++++++++++++++++-------------
 1 file changed, 16 insertions(+), 13 deletions(-)

(limited to 'Documentation')

diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt
index 3ba45adbf040..4692c8e77dc1 100644
--- a/Documentation/sound/alsa/Audiophile-Usb.txt
+++ b/Documentation/sound/alsa/Audiophile-Usb.txt
@@ -1,4 +1,4 @@
-	Guide to using M-Audio Audiophile USB with ALSA and Jack	v1.1
+	Guide to using M-Audio Audiophile USB with ALSA and Jack	v1.2
 	========================================================
 
 	    Thibault Le Meur <Thibault.LeMeur@supelec.fr>
@@ -13,6 +13,9 @@ of the device.
 
 The device has 4 audio interfaces, and 2 MIDI ports:
  * Analog Stereo Input (Ai)
+   - This port supports 2 pairs of line-level audio inputs (1/4" TS and RCA) 
+   - When the 1/4" TS (jack) connectors are connected, the RCA connectors
+     are disabled
  * Analog Stereo Output (Ao)
  * Digital Stereo Input (Di)
  * Digital Stereo Output (Do)
@@ -42,7 +45,7 @@ Important facts about the Digital interface:
 though I haven't tested it under linux
    - Note that in this setup only the Do interface can be enabled
  * Apart from recording an audio digital stream, enabling the Di port is a way 
-to syncrhonize the device to an external sample clock
+to synchronize the device to an external sample clock
    - As a consequence, the Di port must be enable only if an active Digital 
 source is connected
    - Enabling Di when no digital source is connected can result in a 
@@ -180,7 +183,7 @@ IMPORTANT NOTE WHEN SWITCHING CONFIGURATION:
      or changing modprobe.conf)
    - turn on the device
 
-2.2.2.3 - Setting and switching configurations with the device_setup parameter
+2.2.2.3 - Audiophile USB's device_setup structure
 
 If you want to understand the device_setup magic numbers for the Audiophile 
 USB, you need some very basic understanding of binary computation. However, 
@@ -226,7 +229,7 @@ Caution:
    - choosing b2 will prepare all interfaces for 24bits/96kHz but you'll
      only be able to use one at the same time
 
-2.2.3 -  Technical Details for Audiophile Usb 
+2.2.3 -  USB implementation details for this device
 
 You may safely skip this section if you're not interrested in driver 
 development.
@@ -234,20 +237,20 @@ development.
 This section describes some internals aspect of the device and summarize the 
 data I got by usb-snooping the windows and linux drivers.
 
-The M-Audio Audiophile USB has 7 Usb Interfaces:
+The M-Audio Audiophile USB has 7 USB Interfaces:
 a "USB interface":
- * Usb Interface nb.0
- * Usb Interface nb.1
+ * USB Interface nb.0
+ * USB Interface nb.1
    - Audio Control function
- * Usb Interface nb.2
+ * USB Interface nb.2
    - Analog Output
- * Usb Interface nb.3
+ * USB Interface nb.3
    - Digital Output
- * Usb Interface nb.4
+ * USB Interface nb.4
    - Analog Input
- * Usb Interface nb.5
+ * USB Interface nb.5
    - Digital Input
- * Usb Interface nb.6
+ * USB Interface nb.6
    - MIDI interface compliant with the MIDIMAN quirk 
 
 Each interface has 5 altsettings (AltSet 1,2,3,4,5) except:
@@ -326,5 +329,5 @@ atid=425939
 
 After having applied the patch you can run jackd with the following command 
 line:
-# /usr/local/bin/jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1
+  % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1
 
-- 
cgit v1.2.3