Merge branch 'topic/hda' into for-linus

Documentation/sound/alsa/HD-Audio-Controls.txt

+This file explains the codec-specific mixer controls.
+
+Realtek codecs
+--------------
+
+* Channel Mode
+  This is an enum control to change the surround-channel setup,
+  appears only when the surround channels are available.
+  It gives the number of channels to be used, "2ch", "4ch", "6ch",
+  and "8ch".  According to the configuration, this also controls the
+  jack-retasking of multi-I/O jacks.
+
+* Auto-Mute Mode
+  This is an enum control to change the auto-mute behavior of the
+  headphone and line-out jacks.  If built-in speakers and headphone
+  and/or line-out jacks are available on a machine, this controls
+  appears.
+  When there are only either headphones or line-out jacks, it gives
+  "Disabled" and "Enabled" state.  When enabled, the speaker is muted
+  automatically when a jack is plugged.
+
+  When both headphone and line-out jacks are present, it gives
+  "Disabled", "Speaker Only" and "Line-Out+Speaker".  When
+  speaker-only is chosen, plugging into a headphone or a line-out jack
+  mutes the speakers, but not line-outs.  When line-out+speaker is
+  selected, plugging to a headphone jack mutes both speakers and
+  line-outs.
+
+
+IDT/Sigmatel codecs
+-------------------
+
+* Analog Loopback
+  This control enables/disables the analog-loopback circuit.  This
+  appears only when "loopback" is set to true in a codec hint
+  (see HD-Audio.txt).  Note that on some codecs the analog-loopback
+  and the normal PCM playback are exclusive, i.e. when this is on, you
+  won't hear any PCM stream.
+
+* Swap Center/LFE
+  Swaps the center and LFE channel order.  Normally, the left
+  corresponds to the center and the right to the LFE.  When this is
+  ON, the left to the LFE and the right to the center.
+
+* Headphone as Line Out
+  When this control is ON, treat the headphone jacks as line-out
+  jacks.  That is, the headphone won't auto-mute the other line-outs,
+  and no HP-amp is set to the pins.
+
+* Mic Jack Mode, Line Jack Mode, etc
+  These enum controls the direction and the bias of the input jack
+  pins.  Depending on the jack type, it can set as "Mic In" and "Line 
+  In", for determining the input bias, or it can be set to "Line Out"
+  when the pin is a multi-I/O jack for surround channels.
+
+
+VIA codecs
+----------
+
+* Smart 5.1
+  An enum control to re-task the multi-I/O jacks for surround outputs.
+  When it's ON, the corresponding input jacks (usually a line-in and a
+  mic-in) are switched as the surround and the CLFE output jacks.
+
+* Independent HP
+  When this enum control is enabled, the headphone output is routed
+  from an individual stream (the third PCM such as hw:0,2) instead of
+  the primary stream.  In the case the headphone DAC is shared with a
+  side or a CLFE-channel DAC, the DAC is switched to the headphone
+  automatically.
+
+* Loopback Mixing
+  An enum control to determine whether the analog-loopback route is
+  enabled or not.  When it's enabled, the analog-loopback is mixed to
+  the front-channel.  Also, the same route is used for the headphone
+  and speaker outputs.  As a side-effect, when this mode is set, the
+  individual volume controls will be no longer available for
+  headphones and speakers because there is only one DAC connected to a
+  mixer widget.
+
+* Dynamic Power-Control
+  This control determines whether the dynamic power-control per jack
+  detection is enabled or not.  When enabled, the widgets power state
+  (D0/D3) are changed dynamically depending on the jack plugging
+  state for saving power consumptions.  However, if your system
+  doesn't provide a proper jack-detection, this won't work; in such a
+  case, turn this control OFF.
+
+* Jack Detect
+  This control is provided only for VT1708 codec which gives no proper
+  unsolicited event per jack plug.  When this is on, the driver polls
+  the jack detection so that the headphone auto-mute can work, while 
+  turning this off would reduce the power consumption.
+
+
+Conexant codecs
+---------------
+
+* Auto-Mute Mode
+  See Reatek codecs.

sound/pci/hda/Kconfig

@@ -14,6 +14,19 @@ menuconfig SND_HDA_INTEL
 
 if SND_HDA_INTEL
 
+config SND_HDA_PREALLOC_SIZE
+	int "Pre-allocated buffer size for HD-audio driver"
+	range 0 32768
+	default 64
+	help
+	  Specifies the default pre-allocated buffer-size in kB for the
+	  HD-audio driver.  A larger buffer (e.g. 2048) is preferred
+	  for systems using PulseAudio.  The default 64 is chosen just
+	  for compatibility reasons.
+
+	  Note that the pre-allocation size can be changed dynamically
+	  via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too.
+
 config SND_HDA_HWDEP
 	bool "Build hwdep interface for HD-audio driver"
 	select SND_HWDEP
@@ -83,6 +96,19 @@ config SND_HDA_CODEC_REALTEK
 	  snd-hda-codec-realtek.
 	  This module is automatically loaded at probing.
 
+config SND_HDA_ENABLE_REALTEK_QUIRKS
+	bool "Build static quirks for Realtek codecs"
+	depends on SND_HDA_CODEC_REALTEK
+	default y
+	help
+	  Say Y here to build the static quirks codes for Realtek codecs.
+	  If you need the "model" preset that the default BIOS auto-parser
+	  can't handle, turn this option on.
+
+	  If your device works with model=auto option, basically you don't
+	  need the quirk code.  By turning this off, you can reduce the
+	  module size quite a lot.
+
 config SND_HDA_CODEC_ANALOG
 	bool "Build Analog Device HD-audio codec support"
 	default y
@@ -171,6 +197,19 @@ config SND_HDA_CODEC_CA0110
 	  snd-hda-codec-ca0110.
 	  This module is automatically loaded at probing.
 
+config SND_HDA_CODEC_CA0132
+	bool "Build Creative CA0132 codec support"
+	depends on SND_HDA_INTEL
+	default y
+	help
+	  Say Y here to include Creative CA0132 codec support in
+	  snd-hda-intel driver.
+
+	  When the HD-audio driver is built as a module, the codec
+	  support code is also built as another module,
+	  snd-hda-codec-ca0132.
+	  This module is automatically loaded at probing.
+
 config SND_HDA_CODEC_CMEDIA
 	bool "Build C-Media HD-audio codec support"
 	default y

sound/pci/hda/Makefile

@@ -13,6 +13,7 @@ snd-hda-codec-idt-objs :=	patch_sigmatel.o
 snd-hda-codec-si3054-objs :=	patch_si3054.o
 snd-hda-codec-cirrus-objs :=	patch_cirrus.o
 snd-hda-codec-ca0110-objs :=	patch_ca0110.o
+snd-hda-codec-ca0132-objs :=	patch_ca0132.o
 snd-hda-codec-conexant-objs :=	patch_conexant.o
 snd-hda-codec-via-objs :=	patch_via.o
 snd-hda-codec-hdmi-objs :=	patch_hdmi.o hda_eld.o
@@ -42,6 +43,9 @@ endif
 ifdef CONFIG_SND_HDA_CODEC_CA0110
 obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-ca0110.o
 endif
+ifdef CONFIG_SND_HDA_CODEC_CA0132
+obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-ca0132.o
+endif
 ifdef CONFIG_SND_HDA_CODEC_CONEXANT
 obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-conexant.o
 endif

sound/pci/hda/alc680_quirks.c

+/*
+ * ALC680 quirk models
+ * included by patch_realtek.c
+ */
+
+/* ALC680 models */
+enum {
+	ALC680_AUTO,
+	ALC680_BASE,
+	ALC680_MODEL_LAST,
+};
+
+#define ALC680_DIGIN_NID	ALC880_DIGIN_NID
+#define ALC680_DIGOUT_NID	ALC880_DIGOUT_NID
+#define alc680_modes		alc260_modes
+
+static const hda_nid_t alc680_dac_nids[3] = {
+	/* Lout1, Lout2, hp */
+	0x02, 0x03, 0x04
+};
+
+static const hda_nid_t alc680_adc_nids[3] = {
+	/* ADC0-2 */
+	/* DMIC, MIC, Line-in*/
+	0x07, 0x08, 0x09
+};
+
+/*
+ * Analog capture ADC cgange
+ */
+static hda_nid_t alc680_get_cur_adc(struct hda_codec *codec)
+{
+	static hda_nid_t pins[] = {0x18, 0x19};
+	static hda_nid_t adcs[] = {0x08, 0x09};
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(pins); i++) {
+		if (!is_jack_detectable(codec, pins[i]))
+			continue;
+		if (snd_hda_jack_detect(codec, pins[i]))
+			return adcs[i];
+	}
+	return 0x07;
+}
+
+static void alc680_rec_autoswitch(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	hda_nid_t nid = alc680_get_cur_adc(codec);
+	if (spec->cur_adc && nid != spec->cur_adc) {
+		__snd_hda_codec_cleanup_stream(codec, spec->cur_adc, 1);
+		spec->cur_adc = nid;
+		snd_hda_codec_setup_stream(codec, nid,
+					   spec->cur_adc_stream_tag, 0,
+					   spec->cur_adc_format);
+	}
+}
+
+static int alc680_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+				      struct hda_codec *codec,
+				      unsigned int stream_tag,
+				      unsigned int format,
+				      struct snd_pcm_substream *substream)
+{
+	struct alc_spec *spec = codec->spec;
+	hda_nid_t nid = alc680_get_cur_adc(codec);
+
+	spec->cur_adc = nid;
+	spec->cur_adc_stream_tag = stream_tag;
+	spec->cur_adc_format = format;
+	snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format);
+	return 0;
+}
+
+static int alc680_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+				      struct hda_codec *codec,
+				      struct snd_pcm_substream *substream)
+{
+	struct alc_spec *spec = codec->spec;
+	snd_hda_codec_cleanup_stream(codec, spec->cur_adc);
+	spec->cur_adc = 0;
+	return 0;
+}
+
+static const struct hda_pcm_stream alc680_pcm_analog_auto_capture = {
+	.substreams = 1, /* can be overridden */
+	.channels_min = 2,
+	.channels_max = 2,
+	/* NID is set in alc_build_pcms */
+	.ops = {
+		.prepare = alc680_capture_pcm_prepare,
+		.cleanup = alc680_capture_pcm_cleanup
+	},
+};
+
+static const struct snd_kcontrol_new alc680_base_mixer[] = {
+	/* output mixer control */
+	HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x4, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x16, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x12, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line In Boost Volume", 0x19, 0, HDA_INPUT),
+	{ }
+};
+
+static const struct hda_bind_ctls alc680_bind_cap_vol = {
+	.ops = &snd_hda_bind_vol,
+	.values = {
+		HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT),
+		HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
+		HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
+		0
+	},
+};
+
+static const struct hda_bind_ctls alc680_bind_cap_switch = {
+	.ops = &snd_hda_bind_sw,
+	.values = {
+		HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT),
+		HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
+		HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
+		0
+	},
+};
+
+static const struct snd_kcontrol_new alc680_master_capture_mixer[] = {
+	HDA_BIND_VOL("Capture Volume", &alc680_bind_cap_vol),
+	HDA_BIND_SW("Capture Switch", &alc680_bind_cap_switch),
+	{ } /* end */
+};
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static const struct hda_verb alc680_init_verbs[] = {
+	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+	{0x16, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT   | AC_USRSP_EN},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT  | AC_USRSP_EN},
+	{0x19, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT  | AC_USRSP_EN},
+
+	{ }
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc680_base_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	spec->autocfg.hp_pins[0] = 0x16;
+	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->autocfg.speaker_pins[1] = 0x15;
+	spec->autocfg.num_inputs = 2;
+	spec->autocfg.inputs[0].pin = 0x18;
+	spec->autocfg.inputs[0].type = AUTO_PIN_MIC;
+	spec->autocfg.inputs[1].pin = 0x19;
+	spec->autocfg.inputs[1].type = AUTO_PIN_LINE_IN;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+static void alc680_unsol_event(struct hda_codec *codec,
+					   unsigned int res)
+{
+	if ((res >> 26) == ALC_HP_EVENT)
+		alc_hp_automute(codec);
+	if ((res >> 26) == ALC_MIC_EVENT)
+		alc680_rec_autoswitch(codec);
+}
+
+static void alc680_inithook(struct hda_codec *codec)
+{
+	alc_hp_automute(codec);
+	alc680_rec_autoswitch(codec);
+}
+
+/*
+ * configuration and preset
+ */
+static const char * const alc680_models[ALC680_MODEL_LAST] = {
+	[ALC680_BASE]		= "base",
+	[ALC680_AUTO]		= "auto",
+};
+
+static const struct snd_pci_quirk alc680_cfg_tbl[] = {
+	SND_PCI_QUIRK(0x1043, 0x12f3, "ASUS NX90", ALC680_BASE),
+	{}
+};
+
+static const struct alc_config_preset alc680_presets[] = {
+	[ALC680_BASE] = {
+		.mixers = { alc680_base_mixer },
+		.cap_mixer =  alc680_master_capture_mixer,
+		.init_verbs = { alc680_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc680_dac_nids),
+		.dac_nids = alc680_dac_nids,
+		.dig_out_nid = ALC680_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc680_modes),
+		.channel_mode = alc680_modes,
+		.unsol_event = alc680_unsol_event,
+		.setup = alc680_base_setup,
+		.init_hook = alc680_inithook,
+
+	},
+};

sound/pci/hda/hda_local.h

@@ -212,7 +212,9 @@ int snd_hda_mixer_bind_tlv(struct snd_kcontrol *kcontrol, int op_flag,
 /*
  * SPDIF I/O
  */
-int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid);
+int snd_hda_create_spdif_out_ctls(struct hda_codec *codec,
+				  hda_nid_t associated_nid,
+				  hda_nid_t cvt_nid);
 int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid);
 
 /*
@@ -563,7 +565,6 @@ int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key)
  * power-management
  */
 
-#ifdef CONFIG_SND_HDA_POWER_SAVE
 void snd_hda_schedule_power_save(struct hda_codec *codec);
 
 struct hda_amp_list {
@@ -580,7 +581,6 @@ struct hda_loopback_check {
 int snd_hda_check_amp_list_power(struct hda_codec *codec,
 				 struct hda_loopback_check *check,
 				 hda_nid_t nid);
-#endif /* CONFIG_SND_HDA_POWER_SAVE */
 
 /*
  * AMP control callbacks
@@ -639,8 +639,8 @@ struct hdmi_eld {
 int snd_hdmi_get_eld_size(struct hda_codec *codec, hda_nid_t nid);
 int snd_hdmi_get_eld(struct hdmi_eld *, struct hda_codec *, hda_nid_t);
 void snd_hdmi_show_eld(struct hdmi_eld *eld);
-void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm,
-			      struct hda_pcm_stream *codec_pars);
+void snd_hdmi_eld_update_pcm_info(struct hdmi_eld *eld,
+			      struct hda_pcm_stream *hinfo);
 
 #ifdef CONFIG_PROC_FS
 int snd_hda_eld_proc_new(struct hda_codec *codec, struct hdmi_eld *eld,

sound/pci/hda/hda_proc.c

@@ -636,7 +636,7 @@ static void print_codec_info(struct snd_info_entry *entry,
 			wid_caps |= AC_WCAP_CONN_LIST;
 
 		if (wid_caps & AC_WCAP_CONN_LIST)
-			conn_len = snd_hda_get_connections(codec, nid, conn,
+			conn_len = snd_hda_get_raw_connections(codec, nid, conn,
 							   HDA_MAX_CONNECTIONS);
 
 		if (wid_caps & AC_WCAP_IN_AMP) {

sound/pci/hda/patch_ca0110.c

@@ -240,7 +240,8 @@ static int ca0110_build_controls(struct hda_codec *codec)
 	}
 
 	if (spec->dig_out) {
-		err = snd_hda_create_spdif_out_ctls(codec, spec->dig_out);
+		err = snd_hda_create_spdif_out_ctls(codec, spec->dig_out,
+						    spec->dig_out);
 		if (err < 0)
 			return err;
 		err = snd_hda_create_spdif_share_sw(codec, &spec->multiout);