ALSA: hda - Split quirk codes from patch_realtek.c

Put the all static quirk codes out of patch_realtek.c, split into the
file for each codec model.  For controlling the build of quirk codes,
a new Kconfig, CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS is introduced.
By setting this off, all quirk codes won't be built, thus you can save
lots of memory.

The codes in patch_realtek.c are also shuffled and more comments are
given, but the contents aren't changed.  This is just a refactoring.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

sound/pci/hda/Kconfig

@@ -83,6 +83,19 @@ 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

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,
+
+	},
+};