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: Takashi Iwai <tiwai@suse.de>
diff --git a/sound/pci/hda/alc680_quirks.c b/sound/pci/hda/alc680_quirks.c
new file mode 100644
index 0000000..0eeb227
--- /dev/null
+++ b/sound/pci/hda/alc680_quirks.c
@@ -0,0 +1,222 @@
+/*
+ * 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,
+
+ },
+};