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Commit 60db6b53 authored by Kailang Yang's avatar Kailang Yang Committed by Jaroslav Kysela
Browse files

ALSA: hda - Add support of Quanta FL1



Added the support of Quanta FL1 with ALC269 code chip.
Also a bit space clean-ups.

Signed-off-by: default avatarKailang Yang <kailang@realtek.com>
Signed-off-by: default avatarTakashi Iwai <tiwai@suse.de>
Signed-off-by: default avatarJaroslav Kysela <perex@perex.cz>
parent 8ef355da
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+175 −75
Original line number Diff line number Diff line
@@ -126,6 +126,7 @@ enum {
/* ALC269 models */
enum {
	ALC269_BASIC,
	ALC269_QUANTA_FL1,
	ALC269_ASUS_EEEPC_P703,
	ALC269_ASUS_EEEPC_P901,
	ALC269_AUTO,
@@ -11378,6 +11379,28 @@ static struct snd_kcontrol_new alc269_base_mixer[] = {
	{ } /* end */
};

static struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = {
	/* output mixer control */
	HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
	{
		.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 = alc268_acer_master_sw_put,
		.private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
	},
	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
	HDA_CODEC_VOLUME("Internal Mic Boost", 0x19, 0, HDA_INPUT),
	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x04, HDA_INPUT),
	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x04, HDA_INPUT),
	{ }
};

/* bind volumes of both NID 0x0c and 0x0d */
static struct hda_bind_ctls alc269_epc_bind_vol = {
	.ops = &snd_hda_bind_vol,
@@ -11428,75 +11451,65 @@ static struct snd_kcontrol_new alc269_beep_mixer[] = {
	{ } /* end */
};

/*
 * generic initialization of ADC, input mixers and output mixers
 */
static struct hda_verb alc269_init_verbs[] = {
	/*
	 * Unmute ADC0 and set the default input to mic-in
	 */
	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
static struct hda_verb alc269_quanta_fl1_verbs[] = {
	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
	{ }
};

	/* Mute input amps (PCBeep, Line In, Mic 1 & Mic 2) of the
	 * analog-loopback mixer widget
	 * Note: PASD motherboards uses the Line In 2 as the input for
	 * front panel mic (mic 2)
	 */
	/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* toggle speaker-output according to the hp-jack state */
static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec)
{
	unsigned int present;
	unsigned char bits;

	/*
	 * Set up output mixers (0x0c - 0x0e)
	 */
	/* set vol=0 to output mixers */
	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
	present = snd_hda_codec_read(codec, 0x15, 0,
			AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
	bits = present ? AMP_IN_MUTE(0) : 0;
	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
			AMP_IN_MUTE(0), bits);
	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
			AMP_IN_MUTE(0), bits);

	/* set up input amps for analog loopback */
	/* Amp Indices: DAC = 0, mixer = 1 */
	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
	snd_hda_codec_write(codec, 0x20, 0,
			AC_VERB_SET_COEF_INDEX, 0x0c);
	snd_hda_codec_write(codec, 0x20, 0,
			AC_VERB_SET_PROC_COEF, 0x680);

	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
	snd_hda_codec_write(codec, 0x20, 0,
			AC_VERB_SET_COEF_INDEX, 0x0c);
	snd_hda_codec_write(codec, 0x20, 0,
			AC_VERB_SET_PROC_COEF, 0x480);
}

	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
	{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},
	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
static void alc269_quanta_fl1_mic_automute(struct hda_codec *codec)
{
	unsigned int present;

	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
	present = snd_hda_codec_read(codec, 0x18, 0,
				AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
	snd_hda_codec_write(codec, 0x23, 0,
			    AC_VERB_SET_CONNECT_SEL, present ? 0x0 : 0x1);
}

	/* FIXME: use matrix-type input source selection */
	/* Mixer elements: 0x18, 19, 1a, 1b, 1d, 0b */
	/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
static void alc269_quanta_fl1_unsol_event(struct hda_codec *codec,
				    unsigned int res)
{
	if ((res >> 26) == ALC880_HP_EVENT)
		alc269_quanta_fl1_speaker_automute(codec);
	if ((res >> 26) == ALC880_MIC_EVENT)
		alc269_quanta_fl1_mic_automute(codec);
}

	/* set EAPD */
	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
	{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
	{ }
};
static void alc269_quanta_fl1_init_hook(struct hda_codec *codec)
{
	alc269_quanta_fl1_speaker_automute(codec);
	alc269_quanta_fl1_mic_automute(codec);
}

static struct hda_verb alc269_eeepc_dmic_init_verbs[] = {
	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
@@ -11523,7 +11536,7 @@ static struct hda_verb alc269_eeepc_amic_init_verbs[] = {
static void alc269_speaker_automute(struct hda_codec *codec)
{
	unsigned int present;
	unsigned int bits;
	unsigned char bits;

	present = snd_hda_codec_read(codec, 0x15, 0,
				AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
@@ -11538,22 +11551,22 @@ static void alc269_eeepc_dmic_automute(struct hda_codec *codec)
{
	unsigned int present;

	present = snd_hda_codec_read(codec, 0x18, 0, AC_VERB_GET_PIN_SENSE, 0)
		& AC_PINSENSE_PRESENCE;
	snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_CONNECT_SEL,
			    present ? 0 : 5);
	present = snd_hda_codec_read(codec, 0x18, 0,
				AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
	snd_hda_codec_write(codec, 0x23, 0,
				AC_VERB_SET_CONNECT_SEL,  (present ? 0 : 5));
}

static void alc269_eeepc_amic_automute(struct hda_codec *codec)
{
	unsigned int present;

	present = snd_hda_codec_read(codec, 0x18, 0, AC_VERB_GET_PIN_SENSE, 0)
		& AC_PINSENSE_PRESENCE;
	present = snd_hda_codec_read(codec, 0x18, 0,
				AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
	snd_hda_codec_write(codec, 0x24, 0, AC_VERB_SET_AMP_GAIN_MUTE,
			    present ? AMP_IN_UNMUTE(0) : AMP_IN_MUTE(0));
				0x7000 | (0x00 << 8) | (present ? 0 : 0x80));
	snd_hda_codec_write(codec, 0x24, 0, AC_VERB_SET_AMP_GAIN_MUTE,
			    present ? AMP_IN_MUTE(1) : AMP_IN_UNMUTE(1));
				0x7000 | (0x01 << 8) | (present ? 0x80 : 0));
}

/* unsolicited event for HP jack sensing */
@@ -11590,6 +11603,76 @@ static void alc269_eeepc_amic_inithook(struct hda_codec *codec)
	alc269_eeepc_amic_automute(codec);
}

/*
 * generic initialization of ADC, input mixers and output mixers
 */
static struct hda_verb alc269_init_verbs[] = {
	/*
	 * Unmute ADC0 and set the default input to mic-in
	 */
	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},

	/* Mute input amps (PCBeep, Line In, Mic 1 & Mic 2) of the
	 * analog-loopback mixer widget
	 * Note: PASD motherboards uses the Line In 2 as the input for
	 * front panel mic (mic 2)
	 */
	/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},

	/*
	 * Set up output mixers (0x0c - 0x0e)
	 */
	/* set vol=0 to output mixers */
	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},

	/* set up input amps for analog loopback */
	/* Amp Indices: DAC = 0, mixer = 1 */
	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},

	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},

	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
	{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},
	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},

	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},

	/* FIXME: use matrix-type input source selection */
	/* Mixer elements: 0x18, 19, 1a, 1b, 1d, 0b */
	/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},

	/* set EAPD */
	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
	{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
	{ }
};

/* add playback controls from the parsed DAC table */
static int alc269_auto_create_multi_out_ctls(struct alc_spec *spec,
					     const struct auto_pin_cfg *cfg)
@@ -11759,13 +11842,18 @@ static void alc269_auto_init(struct hda_codec *codec)
 */
static const char *alc269_models[ALC269_MODEL_LAST] = {
	[ALC269_BASIC]			= "basic",
	[ALC269_QUANTA_FL1]		= "Quanta",
	[ALC269_ASUS_EEEPC_P901]	= "Asus_Epc_Dmic"
};

static struct snd_pci_quirk alc269_cfg_tbl[] = {
	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_QUANTA_FL1),
	SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A",
		      ALC269_ASUS_EEEPC_P703),
	SND_PCI_QUIRK(0x1043, 0x831a, "ASUS Eeepc P901",
		      ALC269_ASUS_EEEPC_P901),
	SND_PCI_QUIRK(0x1043, 0x834a, "ASUS Eeepc S101",
		      ALC269_ASUS_EEEPC_P901),
	{}
};

@@ -11780,6 +11868,18 @@ static struct alc_config_preset alc269_presets[] = {
		.channel_mode = alc269_modes,
		.input_mux = &alc269_capture_source,
	},
	[ALC269_QUANTA_FL1] = {
		.mixers = { alc269_quanta_fl1_mixer },
		.init_verbs = { alc269_init_verbs, alc269_quanta_fl1_verbs },
		.num_dacs = ARRAY_SIZE(alc269_dac_nids),
		.dac_nids = alc269_dac_nids,
		.hp_nid = 0x03,
		.num_channel_mode = ARRAY_SIZE(alc269_modes),
		.channel_mode = alc269_modes,
		.input_mux = &alc269_capture_source,
		.unsol_event = alc269_quanta_fl1_unsol_event,
		.init_hook = alc269_quanta_fl1_init_hook,
	},
	[ALC269_ASUS_EEEPC_P703] = {
		.mixers = { alc269_eeepc_mixer, alc269_epc_capture_mixer },
		.init_verbs = { alc269_init_verbs,