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-rw-r--r--sound/pci/hda/alc260_quirks.c968
-rw-r--r--sound/pci/hda/alc_quirks.c301
-rw-r--r--sound/pci/hda/hda_codec.c21
-rw-r--r--sound/pci/hda/hda_codec.h1
-rw-r--r--sound/pci/hda/hda_intel.c28
-rw-r--r--sound/pci/hda/hda_jack.c16
-rw-r--r--sound/pci/hda/hda_jack.h13
-rw-r--r--sound/pci/hda/hda_local.h3
-rw-r--r--sound/pci/hda/patch_analog.c66
-rw-r--r--sound/pci/hda/patch_conexant.c36
-rw-r--r--sound/pci/hda/patch_realtek.c221
-rw-r--r--sound/pci/hda/patch_sigmatel.c29
-rw-r--r--sound/pci/hda/patch_via.c28
13 files changed, 238 insertions, 1493 deletions
diff --git a/sound/pci/hda/alc260_quirks.c b/sound/pci/hda/alc260_quirks.c
deleted file mode 100644
index 3b5170b9700..00000000000
--- a/sound/pci/hda/alc260_quirks.c
+++ /dev/null
@@ -1,968 +0,0 @@
-/*
- * ALC260 quirk models
- * included by patch_realtek.c
- */
-
-/* ALC260 models */
-enum {
- ALC260_AUTO,
- ALC260_BASIC,
- ALC260_FUJITSU_S702X,
- ALC260_ACER,
- ALC260_WILL,
- ALC260_REPLACER_672V,
- ALC260_FAVORIT100,
-#ifdef CONFIG_SND_DEBUG
- ALC260_TEST,
-#endif
- ALC260_MODEL_LAST /* last tag */
-};
-
-static const hda_nid_t alc260_dac_nids[1] = {
- /* front */
- 0x02,
-};
-
-static const hda_nid_t alc260_adc_nids[1] = {
- /* ADC0 */
- 0x04,
-};
-
-static const hda_nid_t alc260_adc_nids_alt[1] = {
- /* ADC1 */
- 0x05,
-};
-
-/* NIDs used when simultaneous access to both ADCs makes sense. Note that
- * alc260_capture_mixer assumes ADC0 (nid 0x04) is the first ADC.
- */
-static const hda_nid_t alc260_dual_adc_nids[2] = {
- /* ADC0, ADC1 */
- 0x04, 0x05
-};
-
-#define ALC260_DIGOUT_NID 0x03
-#define ALC260_DIGIN_NID 0x06
-
-static const struct hda_input_mux alc260_capture_source = {
- .num_items = 4,
- .items = {
- { "Mic", 0x0 },
- { "Front Mic", 0x1 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- },
-};
-
-/* On Fujitsu S702x laptops capture only makes sense from Mic/LineIn jack,
- * headphone jack and the internal CD lines since these are the only pins at
- * which audio can appear. For flexibility, also allow the option of
- * recording the mixer output on the second ADC (ADC0 doesn't have a
- * connection to the mixer output).
- */
-static const struct hda_input_mux alc260_fujitsu_capture_sources[2] = {
- {
- .num_items = 3,
- .items = {
- { "Mic/Line", 0x0 },
- { "CD", 0x4 },
- { "Headphone", 0x2 },
- },
- },
- {
- .num_items = 4,
- .items = {
- { "Mic/Line", 0x0 },
- { "CD", 0x4 },
- { "Headphone", 0x2 },
- { "Mixer", 0x5 },
- },
- },
-
-};
-
-/* Acer TravelMate(/Extensa/Aspire) notebooks have similar configuration to
- * the Fujitsu S702x, but jacks are marked differently.
- */
-static const struct hda_input_mux alc260_acer_capture_sources[2] = {
- {
- .num_items = 4,
- .items = {
- { "Mic", 0x0 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- { "Headphone", 0x5 },
- },
- },
- {
- .num_items = 5,
- .items = {
- { "Mic", 0x0 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- { "Headphone", 0x6 },
- { "Mixer", 0x5 },
- },
- },
-};
-
-/* Maxdata Favorit 100XS */
-static const struct hda_input_mux alc260_favorit100_capture_sources[2] = {
- {
- .num_items = 2,
- .items = {
- { "Line/Mic", 0x0 },
- { "CD", 0x4 },
- },
- },
- {
- .num_items = 3,
- .items = {
- { "Line/Mic", 0x0 },
- { "CD", 0x4 },
- { "Mixer", 0x5 },
- },
- },
-};
-
-/*
- * This is just place-holder, so there's something for alc_build_pcms to look
- * at when it calculates the maximum number of channels. ALC260 has no mixer
- * element which allows changing the channel mode, so the verb list is
- * never used.
- */
-static const struct hda_channel_mode alc260_modes[1] = {
- { 2, NULL },
-};
-
-
-/* Mixer combinations
- *
- * basic: base_output + input + pc_beep + capture
- * fujitsu: fujitsu + capture
- * acer: acer + capture
- */
-
-static const struct snd_kcontrol_new alc260_base_output_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc260_input_mixer[] = {
- HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x07, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x07, 0x01, HDA_INPUT),
- { } /* end */
-};
-
-/* Fujitsu S702x series laptops. ALC260 pin usage: Mic/Line jack = 0x12,
- * HP jack = 0x14, CD audio = 0x16, internal speaker = 0x10.
- */
-static const struct snd_kcontrol_new alc260_fujitsu_mixer[] = {
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Headphone Playback Switch", 0x08, 2, HDA_INPUT),
- ALC_PIN_MODE("Headphone Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic/Line Playback Volume", 0x07, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic/Line Playback Switch", 0x07, 0x0, HDA_INPUT),
- ALC_PIN_MODE("Mic/Line Jack Mode", 0x12, ALC_PIN_DIR_IN),
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Speaker Playback Switch", 0x09, 2, HDA_INPUT),
- { } /* end */
-};
-
-/* Mixer for Acer TravelMate(/Extensa/Aspire) notebooks. Note that current
- * versions of the ALC260 don't act on requests to enable mic bias from NID
- * 0x0f (used to drive the headphone jack in these laptops). The ALC260
- * datasheet doesn't mention this restriction. At this stage it's not clear
- * whether this behaviour is intentional or is a hardware bug in chip
- * revisions available in early 2006. Therefore for now allow the
- * "Headphone Jack Mode" control to span all choices, but if it turns out
- * that the lack of mic bias for this NID is intentional we could change the
- * mode from ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS.
- *
- * In addition, Acer TravelMate(/Extensa/Aspire) notebooks in early 2006
- * don't appear to make the mic bias available from the "line" jack, even
- * though the NID used for this jack (0x14) can supply it. The theory is
- * that perhaps Acer have included blocking capacitors between the ALC260
- * and the output jack. If this turns out to be the case for all such
- * models the "Line Jack Mode" mode could be changed from ALC_PIN_DIR_INOUT
- * to ALC_PIN_DIR_INOUT_NOMICBIAS.
- *
- * The C20x Tablet series have a mono internal speaker which is controlled
- * via the chip's Mono sum widget and pin complex, so include the necessary
- * controls for such models. On models without a "mono speaker" the control
- * won't do anything.
- */
-static const struct snd_kcontrol_new alc260_acer_mixer[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT),
- ALC_PIN_MODE("Headphone Jack Mode", 0x0f, ALC_PIN_DIR_INOUT),
- HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0,
- HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Speaker Playback Switch", 0x0a, 1, 2,
- HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
- ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
- ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
- { } /* end */
-};
-
-/* Maxdata Favorit 100XS: one output and one input (0x12) jack
- */
-static const struct snd_kcontrol_new alc260_favorit100_mixer[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT),
- ALC_PIN_MODE("Output Jack Mode", 0x0f, ALC_PIN_DIR_INOUT),
- HDA_CODEC_VOLUME("Line/Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Line/Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
- ALC_PIN_MODE("Line/Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
- { } /* end */
-};
-
-/* Packard bell V7900 ALC260 pin usage: HP = 0x0f, Mic jack = 0x12,
- * Line In jack = 0x14, CD audio = 0x16, pc beep = 0x17.
- */
-static const struct snd_kcontrol_new alc260_will_mixer[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Master Playback Switch", 0x08, 0x2, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
- ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
- ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
- { } /* end */
-};
-
-/* Replacer 672V ALC260 pin usage: Mic jack = 0x12,
- * Line In jack = 0x14, ATAPI Mic = 0x13, speaker = 0x0f.
- */
-static const struct snd_kcontrol_new alc260_replacer_672v_mixer[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Master Playback Switch", 0x08, 0x2, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
- ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
- HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x07, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("ATATI Mic Playback Switch", 0x07, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
- ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
- { } /* end */
-};
-
-/*
- * initialization verbs
- */
-static const struct hda_verb alc260_init_verbs[] = {
- /* Line In pin widget for input */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- /* CD pin widget for input */
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- /* Mic1 (rear panel) pin widget for input and vref at 80% */
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- /* Mic2 (front panel) pin widget for input and vref at 80% */
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- /* LINE-2 is used for line-out in rear */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* select line-out */
- {0x0e, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* LINE-OUT pin */
- {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* enable HP */
- {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- /* enable Mono */
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* mute capture amp left and right */
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* set connection select to line in (default select for this ADC) */
- {0x04, AC_VERB_SET_CONNECT_SEL, 0x02},
- /* mute capture amp left and right */
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* set connection select to line in (default select for this ADC) */
- {0x05, AC_VERB_SET_CONNECT_SEL, 0x02},
- /* set vol=0 Line-Out mixer amp left and right */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* unmute pin widget amp left and right (no gain on this amp) */
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* set vol=0 HP mixer amp left and right */
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* unmute pin widget amp left and right (no gain on this amp) */
- {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* set vol=0 Mono mixer amp left and right */
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* unmute pin widget amp left and right (no gain on this amp) */
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* unmute LINE-2 out pin */
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 &
- * Line In 2 = 0x03
- */
- /* mute analog inputs */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
- /* mute Front out path */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* mute Headphone out path */
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* mute Mono out path */
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- { }
-};
-
-/* Initialisation sequence for ALC260 as configured in Fujitsu S702x
- * laptops. ALC260 pin usage: Mic/Line jack = 0x12, HP jack = 0x14, CD
- * audio = 0x16, internal speaker = 0x10.
- */
-static const struct hda_verb alc260_fujitsu_init_verbs[] = {
- /* Disable all GPIOs */
- {0x01, AC_VERB_SET_GPIO_MASK, 0},
- /* Internal speaker is connected to headphone pin */
- {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- /* Headphone/Line-out jack connects to Line1 pin; make it an output */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* Mic/Line-in jack is connected to mic1 pin, so make it an input */
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- /* Ensure all other unused pins are disabled and muted. */
- {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-
- /* Disable digital (SPDIF) pins */
- {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
- {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
-
- /* Ensure Line1 pin widget takes its input from the OUT1 sum bus
- * when acting as an output.
- */
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0},
-
- /* Start with output sum widgets muted and their output gains at min */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
- /* Unmute HP pin widget amp left and right (no equiv mixer ctrl) */
- {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Unmute Line1 pin widget output buffer since it starts as an output.
- * If the pin mode is changed by the user the pin mode control will
- * take care of enabling the pin's input/output buffers as needed.
- * Therefore there's no need to enable the input buffer at this
- * stage.
- */
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Unmute input buffer of pin widget used for Line-in (no equiv
- * mixer ctrl)
- */
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
- /* Mute capture amp left and right */
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- /* Set ADC connection select to match default mixer setting - line
- * in (on mic1 pin)
- */
- {0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- /* Do the same for the second ADC: mute capture input amp and
- * set ADC connection to line in (on mic1 pin)
- */
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- /* Mute all inputs to mixer widget (even unconnected ones) */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
-
- { }
-};
-
-/* Initialisation sequence for ALC260 as configured in Acer TravelMate and
- * similar laptops (adapted from Fujitsu init verbs).
- */
-static const struct hda_verb alc260_acer_init_verbs[] = {
- /* On TravelMate laptops, GPIO 0 enables the internal speaker and
- * the headphone jack. Turn this on and rely on the standard mute
- * methods whenever the user wants to turn these outputs off.
- */
- {0x01, AC_VERB_SET_GPIO_MASK, 0x01},
- {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
- {0x01, AC_VERB_SET_GPIO_DATA, 0x01},
- /* Internal speaker/Headphone jack is connected to Line-out pin */
- {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- /* Internal microphone/Mic jack is connected to Mic1 pin */
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
- /* Line In jack is connected to Line1 pin */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- /* Some Acers (eg: C20x Tablets) use Mono pin for internal speaker */
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- /* Ensure all other unused pins are disabled and muted. */
- {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- /* Disable digital (SPDIF) pins */
- {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
- {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
-
- /* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum
- * bus when acting as outputs.
- */
- {0x0b, AC_VERB_SET_CONNECT_SEL, 0},
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0},
-
- /* Start with output sum widgets muted and their output gains at min */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
- /* Unmute Line-out pin widget amp left and right
- * (no equiv mixer ctrl)
- */
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Unmute mono pin widget amp output (no equiv mixer ctrl) */
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Unmute Mic1 and Line1 pin widget input buffers since they start as
- * inputs. If the pin mode is changed by the user the pin mode control
- * will take care of enabling the pin's input/output buffers as needed.
- * Therefore there's no need to enable the input buffer at this
- * stage.
- */
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
- /* Mute capture amp left and right */
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- /* Set ADC connection select to match default mixer setting - mic
- * (on mic1 pin)
- */
- {0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- /* Do similar with the second ADC: mute capture input amp and
- * set ADC connection to mic to match ALSA's default state.
- */
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- /* Mute all inputs to mixer widget (even unconnected ones) */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
-
- { }
-};
-
-/* Initialisation sequence for Maxdata Favorit 100XS
- * (adapted from Acer init verbs).
- */
-static const struct hda_verb alc260_favorit100_init_verbs[] = {
- /* GPIO 0 enables the output jack.
- * Turn this on and rely on the standard mute
- * methods whenever the user wants to turn these outputs off.
- */
- {0x01, AC_VERB_SET_GPIO_MASK, 0x01},
- {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
- {0x01, AC_VERB_SET_GPIO_DATA, 0x01},
- /* Line/Mic input jack is connected to Mic1 pin */
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
- /* Ensure all other unused pins are disabled and muted. */
- {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- /* Disable digital (SPDIF) pins */
- {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
- {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
-
- /* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum
- * bus when acting as outputs.
- */
- {0x0b, AC_VERB_SET_CONNECT_SEL, 0},
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0},
-
- /* Start with output sum widgets muted and their output gains at min */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
- /* Unmute Line-out pin widget amp left and right
- * (no equiv mixer ctrl)
- */
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Unmute Mic1 and Line1 pin widget input buffers since they start as
- * inputs. If the pin mode is changed by the user the pin mode control
- * will take care of enabling the pin's input/output buffers as needed.
- * Therefore there's no need to enable the input buffer at this
- * stage.
- */
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
- /* Mute capture amp left and right */
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- /* Set ADC connection select to match default mixer setting - mic
- * (on mic1 pin)
- */
- {0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- /* Do similar with the second ADC: mute capture input amp and
- * set ADC connection to mic to match ALSA's default state.
- */
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- /* Mute all inputs to mixer widget (even unconnected ones) */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
-
- { }
-};
-
-static const struct hda_verb alc260_will_verbs[] = {
- {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x0b, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x0f, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
- {0x1a, AC_VERB_SET_COEF_INDEX, 0x07},
- {0x1a, AC_VERB_SET_PROC_COEF, 0x3040},
- {}
-};
-
-static const struct hda_verb alc260_replacer_672v_verbs[] = {
- {0x0f, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
- {0x1a, AC_VERB_SET_COEF_INDEX, 0x07},
- {0x1a, AC_VERB_SET_PROC_COEF, 0x3050},
-
- {0x01, AC_VERB_SET_GPIO_MASK, 0x01},
- {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
- {0x01, AC_VERB_SET_GPIO_DATA, 0x00},
-
- {0x0f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
- {}
-};
-
-/* toggle speaker-output according to the hp-jack state */
-static void alc260_replacer_672v_automute(struct hda_codec *codec)
-{
- unsigned int present;
-
- /* speaker --> GPIO Data 0, hp or spdif --> GPIO data 1 */
- present = snd_hda_jack_detect(codec, 0x0f);
- if (present) {
- snd_hda_codec_write_cache(codec, 0x01, 0,
- AC_VERB_SET_GPIO_DATA, 1);
- snd_hda_codec_write_cache(codec, 0x0f, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- PIN_HP);
- } else {
- snd_hda_codec_write_cache(codec, 0x01, 0,
- AC_VERB_SET_GPIO_DATA, 0);
- snd_hda_codec_write_cache(codec, 0x0f, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- PIN_OUT);
- }
-}
-
-static void alc260_replacer_672v_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- if ((res >> 26) == ALC_HP_EVENT)
- alc260_replacer_672v_automute(codec);
-}
-
-static const struct hda_verb alc260_hp_dc7600_verbs[] = {
- {0x05, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x10, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
- {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
- {}
-};
-
-/* Test configuration for debugging, modelled after the ALC880 test
- * configuration.
- */
-#ifdef CONFIG_SND_DEBUG
-static const hda_nid_t alc260_test_dac_nids[1] = {
- 0x02,
-};
-static const hda_nid_t alc260_test_adc_nids[2] = {
- 0x04, 0x05,
-};
-/* For testing the ALC260, each input MUX needs its own definition since
- * the signal assignments are different. This assumes that the first ADC
- * is NID 0x04.
- */
-static const struct hda_input_mux alc260_test_capture_sources[2] = {
- {
- .num_items = 7,
- .items = {
- { "MIC1 pin", 0x0 },
- { "MIC2 pin", 0x1 },
- { "LINE1 pin", 0x2 },
- { "LINE2 pin", 0x3 },
- { "CD pin", 0x4 },
- { "LINE-OUT pin", 0x5 },
- { "HP-OUT pin", 0x6 },
- },
- },
- {
- .num_items = 8,
- .items = {
- { "MIC1 pin", 0x0 },
- { "MIC2 pin", 0x1 },
- { "LINE1 pin", 0x2 },
- { "LINE2 pin", 0x3 },
- { "CD pin", 0x4 },
- { "Mixer", 0x5 },
- { "LINE-OUT pin", 0x6 },
- { "HP-OUT pin", 0x7 },
- },
- },
-};
-static const struct snd_kcontrol_new alc260_test_mixer[] = {
- /* Output driver widgets */
- HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x09, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("LOUT2 Playback Switch", 0x09, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x08, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("LOUT1 Playback Switch", 0x08, 2, HDA_INPUT),
-
- /* Modes for retasking pin widgets
- * Note: the ALC260 doesn't seem to act on requests to enable mic
- * bias from NIDs 0x0f and 0x10. The ALC260 datasheet doesn't
- * mention this restriction. At this stage it's not clear whether
- * this behaviour is intentional or is a hardware bug in chip
- * revisions available at least up until early 2006. Therefore for
- * now allow the "HP-OUT" and "LINE-OUT" Mode controls to span all
- * choices, but if it turns out that the lack of mic bias for these
- * NIDs is intentional we could change their modes from
- * ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS.
- */
- ALC_PIN_MODE("HP-OUT pin mode", 0x10, ALC_PIN_DIR_INOUT),
- ALC_PIN_MODE("LINE-OUT pin mode", 0x0f, ALC_PIN_DIR_INOUT),
- ALC_PIN_MODE("LINE2 pin mode", 0x15, ALC_PIN_DIR_INOUT),
- ALC_PIN_MODE("LINE1 pin mode", 0x14, ALC_PIN_DIR_INOUT),
- ALC_PIN_MODE("MIC2 pin mode", 0x13, ALC_PIN_DIR_INOUT),
- ALC_PIN_MODE("MIC1 pin mode", 0x12, ALC_PIN_DIR_INOUT),
-
- /* Loopback mixer controls */
- HDA_CODEC_VOLUME("MIC1 Playback Volume", 0x07, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("MIC1 Playback Switch", 0x07, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("MIC2 Playback Volume", 0x07, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("MIC2 Playback Switch", 0x07, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("LINE1 Playback Volume", 0x07, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("LINE1 Playback Switch", 0x07, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("LINE2 Playback Volume", 0x07, 0x03, HDA_INPUT),
- HDA_CODEC_MUTE("LINE2 Playback Switch", 0x07, 0x03, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("LINE-OUT loopback Playback Volume", 0x07, 0x06, HDA_INPUT),
- HDA_CODEC_MUTE("LINE-OUT loopback Playback Switch", 0x07, 0x06, HDA_INPUT),
- HDA_CODEC_VOLUME("HP-OUT loopback Playback Volume", 0x07, 0x7, HDA_INPUT),
- HDA_CODEC_MUTE("HP-OUT loopback Playback Switch", 0x07, 0x7, HDA_INPUT),
-
- /* Controls for GPIO pins, assuming they are configured as outputs */
- ALC_GPIO_DATA_SWITCH("GPIO pin 0", 0x01, 0x01),
- ALC_GPIO_DATA_SWITCH("GPIO pin 1", 0x01, 0x02),
- ALC_GPIO_DATA_SWITCH("GPIO pin 2", 0x01, 0x04),
- ALC_GPIO_DATA_SWITCH("GPIO pin 3", 0x01, 0x08),
-
- /* Switches to allow the digital IO pins to be enabled. The datasheet
- * is ambigious as to which NID is which; testing on laptops which
- * make this output available should provide clarification.
- */
- ALC_SPDIF_CTRL_SWITCH("SPDIF Playback Switch", 0x03, 0x01),
- ALC_SPDIF_CTRL_SWITCH("SPDIF Capture Switch", 0x06, 0x01),
-
- /* A switch allowing EAPD to be enabled. Some laptops seem to use
- * this output to turn on an external amplifier.
- */
- ALC_EAPD_CTRL_SWITCH("LINE-OUT EAPD Enable Switch", 0x0f, 0x02),
- ALC_EAPD_CTRL_SWITCH("HP-OUT EAPD Enable Switch", 0x10, 0x02),
-
- { } /* end */
-};
-static const struct hda_verb alc260_test_init_verbs[] = {
- /* Enable all GPIOs as outputs with an initial value of 0 */
- {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x0f},
- {0x01, AC_VERB_SET_GPIO_DATA, 0x00},
- {0x01, AC_VERB_SET_GPIO_MASK, 0x0f},
-
- /* Enable retasking pins as output, initially without power amp */
- {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-
- /* Disable digital (SPDIF) pins initially, but users can enable
- * them via a mixer switch. In the case of SPDIF-out, this initverb
- * payload also sets the generation to 0, output to be in "consumer"
- * PCM format, copyright asserted, no pre-emphasis and no validity
- * control.
- */
- {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
- {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
-
- /* Ensure mic1, mic2, line1 and line2 pin widgets take input from the
- * OUT1 sum bus when acting as an output.
- */
- {0x0b, AC_VERB_SET_CONNECT_SEL, 0},
- {0x0c, AC_VERB_SET_CONNECT_SEL, 0},
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0},
- {0x0e, AC_VERB_SET_CONNECT_SEL, 0},
-
- /* Start with output sum widgets muted and their output gains at min */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
- /* Unmute retasking pin widget output buffers since the default
- * state appears to be output. As the pin mode is changed by the
- * user the pin mode control will take care of enabling the pin's
- * input/output buffers as needed.
- */
- {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Also unmute the mono-out pin widget */
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- /* Mute capture amp left and right */
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- /* Set ADC connection select to match default mixer setting (mic1
- * pin)
- */
- {0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- /* Do the same for the second ADC: mute capture input amp and
- * set ADC connection to mic1 pin
- */
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- /* Mute all inputs to mixer widget (even unconnected ones) */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
-
- { }
-};
-#endif
-
-/*
- * ALC260 configurations
- */
-static const char * const alc260_models[ALC260_MODEL_LAST] = {
- [ALC260_BASIC] = "basic",
- [ALC260_FUJITSU_S702X] = "fujitsu",
- [ALC260_ACER] = "acer",
- [ALC260_WILL] = "will",
- [ALC260_REPLACER_672V] = "replacer",
- [ALC260_FAVORIT100] = "favorit100",
-#ifdef CONFIG_SND_DEBUG
- [ALC260_TEST] = "test",
-#endif
- [ALC260_AUTO] = "auto",
-};
-
-static const struct snd_pci_quirk alc260_cfg_tbl[] = {
- SND_PCI_QUIRK(0x1025, 0x007b, "Acer C20x", ALC260_ACER),
- SND_PCI_QUIRK(0x1025, 0x007f, "Acer", ALC260_WILL),
- SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER),
- SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100),
- SND_PCI_QUIRK(0x104d, 0x81bb, "Sony VAIO", ALC260_BASIC),
- SND_PCI_QUIRK(0x104d, 0x81cc, "Sony VAIO", ALC260_BASIC),
- SND_PCI_QUIRK(0x104d, 0x81cd, "Sony VAIO", ALC260_BASIC),
- SND_PCI_QUIRK(0x10cf, 0x1326, "Fujitsu S702X", ALC260_FUJITSU_S702X),
- SND_PCI_QUIRK(0x152d, 0x0729, "CTL U553W", ALC260_BASIC),
- SND_PCI_QUIRK(0x161f, 0x2057, "Replacer 672V", ALC260_REPLACER_672V),
- SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_WILL),
- {}
-};
-
-static const struct alc_config_preset alc260_presets[] = {
- [ALC260_BASIC] = {
- .mixers = { alc260_base_output_mixer,
- alc260_input_mixer },
- .init_verbs = { alc260_init_verbs },
- .num_dacs = ARRAY_SIZE(alc260_dac_nids),
- .dac_nids = alc260_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
- .adc_nids = alc260_dual_adc_nids,
- .num_channel_mode = ARRAY_SIZE(alc260_modes),
- .channel_mode = alc260_modes,
- .input_mux = &alc260_capture_source,
- },
- [ALC260_FUJITSU_S702X] = {
- .mixers = { alc260_fujitsu_mixer },
- .init_verbs = { alc260_fujitsu_init_verbs },
- .num_dacs = ARRAY_SIZE(alc260_dac_nids),
- .dac_nids = alc260_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
- .adc_nids = alc260_dual_adc_nids,
- .num_channel_mode = ARRAY_SIZE(alc260_modes),
- .channel_mode = alc260_modes,
- .num_mux_defs = ARRAY_SIZE(alc260_fujitsu_capture_sources),
- .input_mux = alc260_fujitsu_capture_sources,
- },
- [ALC260_ACER] = {
- .mixers = { alc260_acer_mixer },
- .init_verbs = { alc260_acer_init_verbs },
- .num_dacs = ARRAY_SIZE(alc260_dac_nids),
- .dac_nids = alc260_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
- .adc_nids = alc260_dual_adc_nids,
- .num_channel_mode = ARRAY_SIZE(alc260_modes),
- .channel_mode = alc260_modes,
- .num_mux_defs = ARRAY_SIZE(alc260_acer_capture_sources),
- .input_mux = alc260_acer_capture_sources,
- },
- [ALC260_FAVORIT100] = {
- .mixers = { alc260_favorit100_mixer },
- .init_verbs = { alc260_favorit100_init_verbs },
- .num_dacs = ARRAY_SIZE(alc260_dac_nids),
- .dac_nids = alc260_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
- .adc_nids = alc260_dual_adc_nids,
- .num_channel_mode = ARRAY_SIZE(alc260_modes),
- .channel_mode = alc260_modes,
- .num_mux_defs = ARRAY_SIZE(alc260_favorit100_capture_sources),
- .input_mux = alc260_favorit100_capture_sources,
- },
- [ALC260_WILL] = {
- .mixers = { alc260_will_mixer },
- .init_verbs = { alc260_init_verbs, alc260_will_verbs },
- .num_dacs = ARRAY_SIZE(alc260_dac_nids),
- .dac_nids = alc260_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc260_adc_nids),
- .adc_nids = alc260_adc_nids,
- .dig_out_nid = ALC260_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc260_modes),
- .channel_mode = alc260_modes,
- .input_mux = &alc260_capture_source,
- },
- [ALC260_REPLACER_672V] = {
- .mixers = { alc260_replacer_672v_mixer },
- .init_verbs = { alc260_init_verbs, alc260_replacer_672v_verbs },
- .num_dacs = ARRAY_SIZE(alc260_dac_nids),
- .dac_nids = alc260_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc260_adc_nids),
- .adc_nids = alc260_adc_nids,
- .dig_out_nid = ALC260_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc260_modes),
- .channel_mode = alc260_modes,
- .input_mux = &alc260_capture_source,
- .unsol_event = alc260_replacer_672v_unsol_event,
- .init_hook = alc260_replacer_672v_automute,
- },
-#ifdef CONFIG_SND_DEBUG
- [ALC260_TEST] = {
- .mixers = { alc260_test_mixer },
- .init_verbs = { alc260_test_init_verbs },
- .num_dacs = ARRAY_SIZE(alc260_test_dac_nids),
- .dac_nids = alc260_test_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc260_test_adc_nids),
- .adc_nids = alc260_test_adc_nids,
- .num_channel_mode = ARRAY_SIZE(alc260_modes),
- .channel_mode = alc260_modes,
- .num_mux_defs = ARRAY_SIZE(alc260_test_capture_sources),
- .input_mux = alc260_test_capture_sources,
- },
-#endif
-};
-
diff --git a/sound/pci/hda/alc_quirks.c b/sound/pci/hda/alc_quirks.c
index a18952ed431..b344603ac06 100644
--- a/sound/pci/hda/alc_quirks.c
+++ b/sound/pci/hda/alc_quirks.c
@@ -74,307 +74,6 @@ static int alc_ch_mode_put(struct snd_kcontrol *kcontrol,
return err;
}
-/*
- * Control the mode of pin widget settings via the mixer. "pc" is used
- * instead of "%" to avoid consequences of accidentally treating the % as
- * being part of a format specifier. Maximum allowed length of a value is
- * 63 characters plus NULL terminator.
- *
- * Note: some retasking pin complexes seem to ignore requests for input
- * states other than HiZ (eg: PIN_VREFxx) and revert to HiZ if any of these
- * are requested. Therefore order this list so that this behaviour will not
- * cause problems when mixer clients move through the enum sequentially.
- * NIDs 0x0f and 0x10 have been observed to have this behaviour as of
- * March 2006.
- */
-static const char * const alc_pin_mode_names[] = {
- "Mic 50pc bias", "Mic 80pc bias",
- "Line in", "Line out", "Headphone out",
-};
-static const unsigned char alc_pin_mode_values[] = {
- PIN_VREF50, PIN_VREF80, PIN_IN, PIN_OUT, PIN_HP,
-};
-/* The control can present all 5 options, or it can limit the options based
- * in the pin being assumed to be exclusively an input or an output pin. In
- * addition, "input" pins may or may not process the mic bias option
- * depending on actual widget capability (NIDs 0x0f and 0x10 don't seem to
- * accept requests for bias as of chip versions up to March 2006) and/or
- * wiring in the computer.
- */
-#define ALC_PIN_DIR_IN 0x00
-#define ALC_PIN_DIR_OUT 0x01
-#define ALC_PIN_DIR_INOUT 0x02
-#define ALC_PIN_DIR_IN_NOMICBIAS 0x03
-#define ALC_PIN_DIR_INOUT_NOMICBIAS 0x04
-
-/* Info about the pin modes supported by the different pin direction modes.
- * For each direction the minimum and maximum values are given.
- */
-static const signed char alc_pin_mode_dir_info[5][2] = {
- { 0, 2 }, /* ALC_PIN_DIR_IN */
- { 3, 4 }, /* ALC_PIN_DIR_OUT */
- { 0, 4 }, /* ALC_PIN_DIR_INOUT */
- { 2, 2 }, /* ALC_PIN_DIR_IN_NOMICBIAS */
- { 2, 4 }, /* ALC_PIN_DIR_INOUT_NOMICBIAS */
-};
-#define alc_pin_mode_min(_dir) (alc_pin_mode_dir_info[_dir][0])
-#define alc_pin_mode_max(_dir) (alc_pin_mode_dir_info[_dir][1])
-#define alc_pin_mode_n_items(_dir) \
- (alc_pin_mode_max(_dir)-alc_pin_mode_min(_dir)+1)
-
-static int alc_pin_mode_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- unsigned int item_num = uinfo->value.enumerated.item;
- unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
-
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = alc_pin_mode_n_items(dir);
-
- if (item_num<alc_pin_mode_min(dir) || item_num>alc_pin_mode_max(dir))
- item_num = alc_pin_mode_min(dir);
- strcpy(uinfo->value.enumerated.name, alc_pin_mode_names[item_num]);
- return 0;
-}
-
-static int alc_pin_mode_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- unsigned int i;
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = kcontrol->private_value & 0xffff;
- unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
- long *valp = ucontrol->value.integer.value;
- unsigned int pinctl = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_PIN_WIDGET_CONTROL,
- 0x00);
-
- /* Find enumerated value for current pinctl setting */
- i = alc_pin_mode_min(dir);
- while (i <= alc_pin_mode_max(dir) && alc_pin_mode_values[i] != pinctl)
- i++;
- *valp = i <= alc_pin_mode_max(dir) ? i: alc_pin_mode_min(dir);
- return 0;
-}
-
-static int alc_pin_mode_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- signed int change;
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = kcontrol->private_value & 0xffff;
- unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
- long val = *ucontrol->value.integer.value;
- unsigned int pinctl = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_PIN_WIDGET_CONTROL,
- 0x00);
-
- if (val < alc_pin_mode_min(dir) || val > alc_pin_mode_max(dir))
- val = alc_pin_mode_min(dir);
-
- change = pinctl != alc_pin_mode_values[val];
- if (change) {
- /* Set pin mode to that requested */
- snd_hda_codec_write_cache(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- alc_pin_mode_values[val]);
-
- /* Also enable the retasking pin's input/output as required
- * for the requested pin mode. Enum values of 2 or less are
- * input modes.
- *
- * Dynamically switching the input/output buffers probably
- * reduces noise slightly (particularly on input) so we'll
- * do it. However, having both input and output buffers
- * enabled simultaneously doesn't seem to be problematic if
- * this turns out to be necessary in the future.
- */
- if (val <= 2) {
- snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, HDA_AMP_MUTE);
- snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0,
- HDA_AMP_MUTE, 0);
- } else {
- snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0,
- HDA_AMP_MUTE, HDA_AMP_MUTE);
- snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, 0);
- }
- }
- return change;
-}
-
-#define ALC_PIN_MODE(xname, nid, dir) \
- { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
- .subdevice = HDA_SUBDEV_NID_FLAG | nid, \
- .info = alc_pin_mode_info, \
- .get = alc_pin_mode_get, \
- .put = alc_pin_mode_put, \
- .private_value = nid | (dir<<16) }
-
-/* A switch control for ALC260 GPIO pins. Multiple GPIOs can be ganged
- * together using a mask with more than one bit set. This control is
- * currently used only by the ALC260 test model. At this stage they are not
- * needed for any "production" models.
- */
-#ifdef CONFIG_SND_DEBUG
-#define alc_gpio_data_info snd_ctl_boolean_mono_info
-
-static int alc_gpio_data_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = kcontrol->private_value & 0xffff;
- unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
- long *valp = ucontrol->value.integer.value;
- unsigned int val = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_GPIO_DATA, 0x00);
-
- *valp = (val & mask) != 0;
- return 0;
-}
-static int alc_gpio_data_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- signed int change;
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = kcontrol->private_value & 0xffff;
- unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
- long val = *ucontrol->value.integer.value;
- unsigned int gpio_data = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_GPIO_DATA,
- 0x00);
-
- /* Set/unset the masked GPIO bit(s) as needed */
- change = (val == 0 ? 0 : mask) != (gpio_data & mask);
- if (val == 0)
- gpio_data &= ~mask;
- else
- gpio_data |= mask;
- snd_hda_codec_write_cache(codec, nid, 0,
- AC_VERB_SET_GPIO_DATA, gpio_data);
-
- return change;
-}
-#define ALC_GPIO_DATA_SWITCH(xname, nid, mask) \
- { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
- .subdevice = HDA_SUBDEV_NID_FLAG | nid, \
- .info = alc_gpio_data_info, \
- .get = alc_gpio_data_get, \
- .put = alc_gpio_data_put, \
- .private_value = nid | (mask<<16) }
-#endif /* CONFIG_SND_DEBUG */
-
-/* A switch control to allow the enabling of the digital IO pins on the
- * ALC260. This is incredibly simplistic; the intention of this control is
- * to provide something in the test model allowing digital outputs to be
- * identified if present. If models are found which can utilise these
- * outputs a more complete mixer control can be devised for those models if
- * necessary.
- */
-#ifdef CONFIG_SND_DEBUG
-#define alc_spdif_ctrl_info snd_ctl_boolean_mono_info
-
-static int alc_spdif_ctrl_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = kcontrol->private_value & 0xffff;
- unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
- long *valp = ucontrol->value.integer.value;
- unsigned int val = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_DIGI_CONVERT_1, 0x00);
-
- *valp = (val & mask) != 0;
- return 0;
-}
-static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- signed int change;
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = kcontrol->private_value & 0xffff;
- unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
- long val = *ucontrol->value.integer.value;
- unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_DIGI_CONVERT_1,
- 0x00);
-
- /* Set/unset the masked control bit(s) as needed */
- change = (val == 0 ? 0 : mask) != (ctrl_data & mask);
- if (val==0)
- ctrl_data &= ~mask;
- else
- ctrl_data |= mask;
- snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
- ctrl_data);
-
- return change;
-}
-#define ALC_SPDIF_CTRL_SWITCH(xname, nid, mask) \
- { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
- .subdevice = HDA_SUBDEV_NID_FLAG | nid, \
- .info = alc_spdif_ctrl_info, \
- .get = alc_spdif_ctrl_get, \
- .put = alc_spdif_ctrl_put, \
- .private_value = nid | (mask<<16) }
-#endif /* CONFIG_SND_DEBUG */
-
-/* A switch control to allow the enabling EAPD digital outputs on the ALC26x.
- * Again, this is only used in the ALC26x test models to help identify when
- * the EAPD line must be asserted for features to work.
- */
-#ifdef CONFIG_SND_DEBUG
-#define alc_eapd_ctrl_info snd_ctl_boolean_mono_info
-
-static int alc_eapd_ctrl_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = kcontrol->private_value & 0xffff;
- unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
- long *valp = ucontrol->value.integer.value;
- unsigned int val = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_EAPD_BTLENABLE, 0x00);
-
- *valp = (val & mask) != 0;
- return 0;
-}
-
-static int alc_eapd_ctrl_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- int change;
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = kcontrol->private_value & 0xffff;
- unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
- long val = *ucontrol->value.integer.value;
- unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_EAPD_BTLENABLE,
- 0x00);
-
- /* Set/unset the masked control bit(s) as needed */
- change = (!val ? 0 : mask) != (ctrl_data & mask);
- if (!val)
- ctrl_data &= ~mask;
- else
- ctrl_data |= mask;
- snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE,
- ctrl_data);
-
- return change;
-}
-
-#define ALC_EAPD_CTRL_SWITCH(xname, nid, mask) \
- { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
- .subdevice = HDA_SUBDEV_NID_FLAG | nid, \
- .info = alc_eapd_ctrl_info, \
- .get = alc_eapd_ctrl_get, \
- .put = alc_eapd_ctrl_put, \
- .private_value = nid | (mask<<16) }
-#endif /* CONFIG_SND_DEBUG */
-
static void alc_fixup_autocfg_pin_nums(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index c2c65f63bf0..65c01798d84 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2300,7 +2300,7 @@ typedef int (*map_slave_func_t)(void *, struct snd_kcontrol *);
/* apply the function to all matching slave ctls in the mixer list */
static int map_slaves(struct hda_codec *codec, const char * const *slaves,
- map_slave_func_t func, void *data)
+ const char *suffix, map_slave_func_t func, void *data)
{
struct hda_nid_item *items;
const char * const *s;
@@ -2313,7 +2313,14 @@ static int map_slaves(struct hda_codec *codec, const char * const *slaves,
sctl->id.iface != SNDRV_CTL_ELEM_IFACE_MIXER)
continue;
for (s = slaves; *s; s++) {
- if (!strcmp(sctl->id.name, *s)) {
+ char tmpname[sizeof(sctl->id.name)];
+ const char *name = *s;
+ if (suffix) {
+ snprintf(tmpname, sizeof(tmpname), "%s %s",
+ name, suffix);
+ name = tmpname;
+ }
+ if (!strcmp(sctl->id.name, name)) {
err = func(data, sctl);
if (err)
return err;
@@ -2335,6 +2342,7 @@ static int check_slave_present(void *data, struct snd_kcontrol *sctl)
* @name: vmaster control name
* @tlv: TLV data (optional)
* @slaves: slave control names (optional)
+ * @suffix: suffix string to each slave name (optional)
*
* Create a virtual master control with the given name. The TLV data
* must be either NULL or a valid data.
@@ -2346,12 +2354,13 @@ static int check_slave_present(void *data, struct snd_kcontrol *sctl)
* This function returns zero if successful or a negative error code.
*/
int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
- unsigned int *tlv, const char * const *slaves)
+ unsigned int *tlv, const char * const *slaves,
+ const char *suffix)
{
struct snd_kcontrol *kctl;
int err;
- err = map_slaves(codec, slaves, check_slave_present, NULL);
+ err = map_slaves(codec, slaves, suffix, check_slave_present, NULL);
if (err != 1) {
snd_printdd("No slave found for %s\n", name);
return 0;
@@ -2363,8 +2372,8 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
if (err < 0)
return err;
- err = map_slaves(codec, slaves, (map_slave_func_t)snd_ctl_add_slave,
- kctl);
+ err = map_slaves(codec, slaves, suffix,
+ (map_slave_func_t)snd_ctl_add_slave, kctl);
if (err < 0)
return err;
return 0;
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index e9f71dc0d46..654d2e41e25 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -852,6 +852,7 @@ struct hda_codec {
unsigned int pins_shutup:1; /* pins are shut up */
unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */
unsigned int ignore_misc_bit:1; /* ignore MISC_NO_PRESENCE bit */
+ unsigned int no_jack_detect:1; /* Machine has no jack-detection */
#ifdef CONFIG_SND_HDA_POWER_SAVE
unsigned int power_on :1; /* current (global) power-state */
unsigned int power_transition :1; /* power-state in transition */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 95dfb687494..e354c161654 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -94,7 +94,7 @@ MODULE_PARM_DESC(probe_only, "Only probing and no codec initialization.");
module_param(single_cmd, bool, 0444);
MODULE_PARM_DESC(single_cmd, "Use single command to communicate with codecs "
"(for debugging only).");
-module_param(enable_msi, int, 0444);
+module_param(enable_msi, bint, 0444);
MODULE_PARM_DESC(enable_msi, "Enable Message Signaled Interrupt (MSI)");
#ifdef CONFIG_SND_HDA_PATCH_LOADER
module_param_array(patch, charp, NULL, 0444);
@@ -121,8 +121,8 @@ module_param(power_save_controller, bool, 0644);
MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode.");
#endif
-static bool align_buffer_size = 1;
-module_param(align_buffer_size, bool, 0644);
+static int align_buffer_size = -1;
+module_param(align_buffer_size, bint, 0644);
MODULE_PARM_DESC(align_buffer_size,
"Force buffer and period sizes to be multiple of 128 bytes.");
@@ -148,6 +148,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6},"
"{Intel, PCH},"
"{Intel, CPT},"
"{Intel, PPT},"
+ "{Intel, LPT},"
"{Intel, PBG},"
"{Intel, SCH},"
"{ATI, SB450},"
@@ -515,6 +516,7 @@ enum {
#define AZX_DCAPS_SYNC_WRITE (1 << 19) /* sync each cmd write */
#define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */
#define AZX_DCAPS_BUFSIZE (1 << 21) /* no buffer size alignment */
+#define AZX_DCAPS_ALIGN_BUFSIZE (1 << 22) /* buffer size alignment */
/* quirks for ATI SB / AMD Hudson */
#define AZX_DCAPS_PRESET_ATI_SB \
@@ -527,7 +529,8 @@ enum {
/* quirks for Nvidia */
#define AZX_DCAPS_PRESET_NVIDIA \
- (AZX_DCAPS_NVIDIA_SNOOP | AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI)
+ (AZX_DCAPS_NVIDIA_SNOOP | AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI |\
+ AZX_DCAPS_ALIGN_BUFSIZE)
static char *driver_short_names[] __devinitdata = {
[AZX_DRIVER_ICH] = "HDA Intel",
@@ -2774,9 +2777,16 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
}
/* disable buffer size rounding to 128-byte multiples if supported */
- chip->align_buffer_size = align_buffer_size;
- if (chip->driver_caps & AZX_DCAPS_BUFSIZE)
- chip->align_buffer_size = 0;
+ if (align_buffer_size >= 0)
+ chip->align_buffer_size = !!align_buffer_size;
+ else {
+ if (chip->driver_caps & AZX_DCAPS_BUFSIZE)
+ chip->align_buffer_size = 0;
+ else if (chip->driver_caps & AZX_DCAPS_ALIGN_BUFSIZE)
+ chip->align_buffer_size = 1;
+ else
+ chip->align_buffer_size = 1;
+ }
/* allow 64bit DMA address if supported by H/W */
if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64)))
@@ -2992,6 +3002,10 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
{ PCI_DEVICE(0x8086, 0x1e20),
.driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
AZX_DCAPS_BUFSIZE},
+ /* Lynx Point */
+ { PCI_DEVICE(0x8086, 0x8c20),
+ .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
+ AZX_DCAPS_BUFSIZE},
/* SCH */
{ PCI_DEVICE(0x8086, 0x811b),
.driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP |
diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c
index 9d819c4b492..d68948499fb 100644
--- a/sound/pci/hda/hda_jack.c
+++ b/sound/pci/hda/hda_jack.c
@@ -19,6 +19,22 @@
#include "hda_local.h"
#include "hda_jack.h"
+bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid)
+{
+ if (codec->no_jack_detect)
+ return false;
+ if (!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT))
+ return false;
+ if (!codec->ignore_misc_bit &&
+ (get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) &
+ AC_DEFCFG_MISC_NO_PRESENCE))
+ return false;
+ if (!(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP))
+ return false;
+ return true;
+}
+EXPORT_SYMBOL_HDA(is_jack_detectable);
+
/* execute pin sense measurement */
static u32 read_pin_sense(struct hda_codec *codec, hda_nid_t nid)
{
diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h
index f8f97c71c9c..c66655cf413 100644
--- a/sound/pci/hda/hda_jack.h
+++ b/sound/pci/hda/hda_jack.h
@@ -62,18 +62,7 @@ int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid,
u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid);
int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid);
-static inline bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid)
-{
- if (!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT))
- return false;
- if (!codec->ignore_misc_bit &&
- (get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) &
- AC_DEFCFG_MISC_NO_PRESENCE))
- return false;
- if (!(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP))
- return false;
- return true;
-}
+bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid);
int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid,
const char *name, int idx);
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index aca8d3193b9..6094dea82bc 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -140,7 +140,8 @@ void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir,
struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec,
const char *name);
int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
- unsigned int *tlv, const char * const *slaves);
+ unsigned int *tlv, const char * const *slaves,
+ const char *suffix);
int snd_hda_codec_reset(struct hda_codec *codec);
/* amp value bits */
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 9cb14b42dff..9771b070245 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -137,51 +137,17 @@ static int ad198x_init(struct hda_codec *codec)
return 0;
}
-static const char * const ad_slave_vols[] = {
- "Front Playback Volume",
- "Surround Playback Volume",
- "Center Playback Volume",
- "LFE Playback Volume",
- "Side Playback Volume",
- "Headphone Playback Volume",
- "Mono Playback Volume",
- "Speaker Playback Volume",
- "IEC958 Playback Volume",
+static const char * const ad_slave_pfxs[] = {
+ "Front", "Surround", "Center", "LFE", "Side",
+ "Headphone", "Mono", "Speaker", "IEC958",
NULL
};
-static const char * const ad_slave_sws[] = {
- "Front Playback Switch",
- "Surround Playback Switch",
- "Center Playback Switch",
- "LFE Playback Switch",
- "Side Playback Switch",
- "Headphone Playback Switch",
- "Mono Playback Switch",
- "Speaker Playback Switch",
- "IEC958 Playback Switch",
+static const char * const ad1988_6stack_fp_slave_pfxs[] = {
+ "Front", "Surround", "Center", "LFE", "Side", "IEC958",
NULL
};
-static const char * const ad1988_6stack_fp_slave_vols[] = {
- "Front Playback Volume",
- "Surround Playback Volume",
- "Center Playback Volume",
- "LFE Playback Volume",
- "Side Playback Volume",
- "IEC958 Playback Volume",
- NULL
-};
-
-static const char * const ad1988_6stack_fp_slave_sws[] = {
- "Front Playback Switch",
- "Surround Playback Switch",
- "Center Playback Switch",
- "LFE Playback Switch",
- "Side Playback Switch",
- "IEC958 Playback Switch",
- NULL
-};
static void ad198x_free_kctls(struct hda_codec *codec);
#ifdef CONFIG_SND_HDA_INPUT_BEEP
@@ -260,7 +226,8 @@ static int ad198x_build_controls(struct hda_codec *codec)
err = snd_hda_add_vmaster(codec, "Master Playback Volume",
vmaster_tlv,
(spec->slave_vols ?
- spec->slave_vols : ad_slave_vols));
+ spec->slave_vols : ad_slave_pfxs),
+ "Playback Volume");
if (err < 0)
return err;
}
@@ -268,7 +235,8 @@ static int ad198x_build_controls(struct hda_codec *codec)
err = snd_hda_add_vmaster(codec, "Master Playback Switch",
NULL,
(spec->slave_sws ?
- spec->slave_sws : ad_slave_sws));
+ spec->slave_sws : ad_slave_pfxs),
+ "Playback Switch");
if (err < 0)
return err;
}
@@ -3385,8 +3353,8 @@ static int patch_ad1988(struct hda_codec *codec)
if (spec->autocfg.hp_pins[0]) {
spec->mixers[spec->num_mixers++] = ad1988_hp_mixers;
- spec->slave_vols = ad1988_6stack_fp_slave_vols;
- spec->slave_sws = ad1988_6stack_fp_slave_sws;
+ spec->slave_vols = ad1988_6stack_fp_slave_pfxs;
+ spec->slave_sws = ad1988_6stack_fp_slave_pfxs;
spec->alt_dac_nid = ad1988_alt_dac_nid;
spec->stream_analog_alt_playback =
&ad198x_pcm_analog_alt_playback;
@@ -3594,16 +3562,8 @@ static const struct hda_amp_list ad1884_loopbacks[] = {
#endif
static const char * const ad1884_slave_vols[] = {
- "PCM Playback Volume",
- "Mic Playback Volume",
- "Mono Playback Volume",
- "Front Mic Playback Volume",
- "Mic Playback Volume",
- "CD Playback Volume",
- "Internal Mic Playback Volume",
- "Docking Mic Playback Volume",
- /* "Beep Playback Volume", */
- "IEC958 Playback Volume",
+ "PCM", "Mic", "Mono", "Front Mic", "Mic", "CD",
+ "Internal Mic", "Docking Mic", /* "Beep", */ "IEC958",
NULL
};
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index a7a5733aa4d..266e5a68baf 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -465,21 +465,8 @@ static const struct snd_kcontrol_new cxt_beep_mixer[] = {
};
#endif
-static const char * const slave_vols[] = {
- "Headphone Playback Volume",
- "Speaker Playback Volume",
- "Front Playback Volume",
- "Surround Playback Volume",
- "CLFE Playback Volume",
- NULL
-};
-
-static const char * const slave_sws[] = {
- "Headphone Playback Switch",
- "Speaker Playback Switch",
- "Front Playback Switch",
- "Surround Playback Switch",
- "CLFE Playback Switch",
+static const char * const slave_pfxs[] = {
+ "Headphone", "Speaker", "Front", "Surround", "CLFE",
NULL
};
@@ -519,14 +506,16 @@ static int conexant_build_controls(struct hda_codec *codec)
snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid,
HDA_OUTPUT, vmaster_tlv);
err = snd_hda_add_vmaster(codec, "Master Playback Volume",
- vmaster_tlv, slave_vols);
+ vmaster_tlv, slave_pfxs,
+ "Playback Volume");
if (err < 0)
return err;
}
if (spec->vmaster_nid &&
!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
err = snd_hda_add_vmaster(codec, "Master Playback Switch",
- NULL, slave_sws);
+ NULL, slave_pfxs,
+ "Playback Switch");
if (err < 0)
return err;
}
@@ -3034,7 +3023,6 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS),
SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS),
SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO),
- SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */
SND_PCI_QUIRK(0x1b0a, 0x2092, "CyberpowerPC Gamer Xplorer N57001", CXT5066_AUTO),
{}
};
@@ -4414,6 +4402,18 @@ static int patch_conexant_auto(struct hda_codec *codec)
codec->patch_ops = cx_auto_patch_ops;
if (spec->beep_amp)
snd_hda_attach_beep_device(codec, spec->beep_amp);
+
+ /* Some laptops with Conexant chips show stalls in S3 resume,
+ * which falls into the single-cmd mode.
+ * Better to make reset, then.
+ */
+ if (!codec->bus->sync_write) {
+ snd_printd("hda_codec: "
+ "Enable sync_write for stable communication\n");
+ codec->bus->sync_write = 1;
+ codec->bus->allow_bus_reset = 1;
+ }
+
return 0;
}
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 389a28a21fa..0ffccc17895 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1847,36 +1847,10 @@ DEFINE_CAPMIX_NOSRC(3);
/*
* slave controls for virtual master
*/
-static const char * const alc_slave_vols[] = {
- "Front Playback Volume",
- "Surround Playback Volume",
- "Center Playback Volume",
- "LFE Playback Volume",
- "Side Playback Volume",
- "Headphone Playback Volume",
- "Speaker Playback Volume",
- "Mono Playback Volume",
- "Line-Out Playback Volume",
- "CLFE Playback Volume",
- "Bass Speaker Playback Volume",
- "PCM Playback Volume",
- NULL,
-};
-
-static const char * const alc_slave_sws[] = {
- "Front Playback Switch",
- "Surround Playback Switch",
- "Center Playback Switch",
- "LFE Playback Switch",
- "Side Playback Switch",
- "Headphone Playback Switch",
- "Speaker Playback Switch",
- "Mono Playback Switch",
- "IEC958 Playback Switch",
- "Line-Out Playback Switch",
- "CLFE Playback Switch",
- "Bass Speaker Playback Switch",
- "PCM Playback Switch",
+static const char * const alc_slave_pfxs[] = {
+ "Front", "Surround", "Center", "LFE", "Side",
+ "Headphone", "Speaker", "Mono", "Line-Out",
+ "CLFE", "Bass Speaker", "PCM",
NULL,
};
@@ -1967,14 +1941,16 @@ static int __alc_build_controls(struct hda_codec *codec)
snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid,
HDA_OUTPUT, vmaster_tlv);
err = snd_hda_add_vmaster(codec, "Master Playback Volume",
- vmaster_tlv, alc_slave_vols);
+ vmaster_tlv, alc_slave_pfxs,
+ "Playback Volume");
if (err < 0)
return err;
}
if (!spec->no_analog &&
!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
err = snd_hda_add_vmaster(codec, "Master Playback Switch",
- NULL, alc_slave_sws);
+ NULL, alc_slave_pfxs,
+ "Playback Switch");
if (err < 0)
return err;
}
@@ -4236,34 +4212,111 @@ static const struct hda_amp_list alc260_loopbacks[] = {
* Pin config fixes
*/
enum {
- PINFIX_HP_DC5750,
+ ALC260_FIXUP_HP_DC5750,
+ ALC260_FIXUP_HP_PIN_0F,
+ ALC260_FIXUP_COEF,
+ ALC260_FIXUP_GPIO1,
+ ALC260_FIXUP_GPIO1_TOGGLE,
+ ALC260_FIXUP_REPLACER,
+ ALC260_FIXUP_HP_B1900,
};
+static void alc260_gpio1_automute(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
+ spec->hp_jack_present);
+}
+
+static void alc260_fixup_gpio1_toggle(struct hda_codec *codec,
+ const struct alc_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+ if (action == ALC_FIXUP_ACT_PROBE) {
+ /* although the machine has only one output pin, we need to
+ * toggle GPIO1 according to the jack state
+ */
+ spec->automute_hook = alc260_gpio1_automute;
+ spec->detect_hp = 1;
+ spec->automute_speaker = 1;
+ spec->autocfg.hp_pins[0] = 0x0f; /* copy it for automute */
+ snd_hda_jack_detect_enable(codec, 0x0f, ALC_HP_EVENT);
+ spec->unsol_event = alc_sku_unsol_event;
+ add_verb(codec->spec, alc_gpio1_init_verbs);
+ }
+}
+
static const struct alc_fixup alc260_fixups[] = {
- [PINFIX_HP_DC5750] = {
+ [ALC260_FIXUP_HP_DC5750] = {
.type = ALC_FIXUP_PINS,
.v.pins = (const struct alc_pincfg[]) {
{ 0x11, 0x90130110 }, /* speaker */
{ }
}
},
+ [ALC260_FIXUP_HP_PIN_0F] = {
+ .type = ALC_FIXUP_PINS,
+ .v.pins = (const struct alc_pincfg[]) {
+ { 0x0f, 0x01214000 }, /* HP */
+ { }
+ }
+ },
+ [ALC260_FIXUP_COEF] = {
+ .type = ALC_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 },
+ { 0x20, AC_VERB_SET_PROC_COEF, 0x3040 },
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC260_FIXUP_HP_PIN_0F,
+ },
+ [ALC260_FIXUP_GPIO1] = {
+ .type = ALC_FIXUP_VERBS,
+ .v.verbs = alc_gpio1_init_verbs,
+ },
+ [ALC260_FIXUP_GPIO1_TOGGLE] = {
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc260_fixup_gpio1_toggle,
+ .chained = true,
+ .chain_id = ALC260_FIXUP_HP_PIN_0F,
+ },
+ [ALC260_FIXUP_REPLACER] = {
+ .type = ALC_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 },
+ { 0x20, AC_VERB_SET_PROC_COEF, 0x3050 },
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC260_FIXUP_GPIO1_TOGGLE,
+ },
+ [ALC260_FIXUP_HP_B1900] = {
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc260_fixup_gpio1_toggle,
+ .chained = true,
+ .chain_id = ALC260_FIXUP_COEF,
+ }
};
static const struct snd_pci_quirk alc260_fixup_tbl[] = {
- SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", PINFIX_HP_DC5750),
+ SND_PCI_QUIRK(0x1025, 0x007b, "Acer C20x", ALC260_FIXUP_GPIO1),
+ SND_PCI_QUIRK(0x1025, 0x007f, "Acer Aspire 9500", ALC260_FIXUP_COEF),
+ SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_FIXUP_GPIO1),
+ SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", ALC260_FIXUP_HP_DC5750),
+ SND_PCI_QUIRK(0x103c, 0x30ba, "HP Presario B1900", ALC260_FIXUP_HP_B1900),
+ SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FIXUP_GPIO1),
+ SND_PCI_QUIRK(0x161f, 0x2057, "Replacer 672V", ALC260_FIXUP_REPLACER),
+ SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_FIXUP_COEF),
{}
};
/*
*/
-#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
-#include "alc260_quirks.c"
-#endif
-
static int patch_alc260(struct hda_codec *codec)
{
struct alc_spec *spec;
- int err, board_config;
+ int err;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
@@ -4273,38 +4326,13 @@ static int patch_alc260(struct hda_codec *codec)
spec->mixer_nid = 0x07;
- board_config = alc_board_config(codec, ALC260_MODEL_LAST,
- alc260_models, alc260_cfg_tbl);
- if (board_config < 0) {
- snd_printd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
- codec->chip_name);
- board_config = ALC_MODEL_AUTO;
- }
-
- if (board_config == ALC_MODEL_AUTO) {
- alc_pick_fixup(codec, NULL, alc260_fixup_tbl, alc260_fixups);
- alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
- }
-
- if (board_config == ALC_MODEL_AUTO) {
- /* automatic parse from the BIOS config */
- err = alc260_parse_auto_config(codec);
- if (err < 0)
- goto error;
-#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
- else if (!err) {
- printk(KERN_INFO
- "hda_codec: Cannot set up configuration "
- "from BIOS. Using base mode...\n");
- board_config = ALC260_BASIC;
- }
-#endif
- }
+ alc_pick_fixup(codec, NULL, alc260_fixup_tbl, alc260_fixups);
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
- if (board_config != ALC_MODEL_AUTO) {
- setup_preset(codec, &alc260_presets[board_config]);
- spec->vmaster_nid = 0x08;
- }
+ /* automatic parse from the BIOS config */
+ err = alc260_parse_auto_config(codec);
+ if (err < 0)
+ goto error;
if (!spec->no_analog && !spec->adc_nids) {
alc_auto_fill_adc_caps(codec);
@@ -4325,10 +4353,7 @@ static int patch_alc260(struct hda_codec *codec)
alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
codec->patch_ops = alc_patch_ops;
- if (board_config == ALC_MODEL_AUTO)
- spec->init_hook = alc_auto_init_std;
- else
- codec->patch_ops.build_controls = __alc_build_controls;
+ spec->init_hook = alc_auto_init_std;
spec->shutup = alc_eapd_shutup;
#ifdef CONFIG_SND_HDA_POWER_SAVE
if (!spec->loopback.amplist)
@@ -5399,7 +5424,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x3bf8, "Lenovo Ideapd", ALC269_FIXUP_PCM_44K),
SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
-#if 1
+#if 0
/* Below is a quirk table taken from the old code.
* Basically the device should work as is without the fixup table.
* If BIOS doesn't give a proper info, enable the corresponding
@@ -5615,8 +5640,10 @@ static const struct hda_amp_list alc861_loopbacks[] = {
/* Pin config fixes */
enum {
- PINFIX_FSC_AMILO_PI1505,
- PINFIX_ASUS_A6RP,
+ ALC861_FIXUP_FSC_AMILO_PI1505,
+ ALC861_FIXUP_AMP_VREF_0F,
+ ALC861_FIXUP_NO_JACK_DETECT,
+ ALC861_FIXUP_ASUS_A6RP,
};
/* On some laptops, VREF of pin 0x0f is abused for controlling the main amp */
@@ -5638,8 +5665,16 @@ static void alc861_fixup_asus_amp_vref_0f(struct hda_codec *codec,
spec->keep_vref_in_automute = 1;
}
+/* suppress the jack-detection */
+static void alc_fixup_no_jack_detect(struct hda_codec *codec,
+ const struct alc_fixup *fix, int action)
+{
+ if (action == ALC_FIXUP_ACT_PRE_PROBE)
+ codec->no_jack_detect = 1;
+}
+
static const struct alc_fixup alc861_fixups[] = {
- [PINFIX_FSC_AMILO_PI1505] = {
+ [ALC861_FIXUP_FSC_AMILO_PI1505] = {
.type = ALC_FIXUP_PINS,
.v.pins = (const struct alc_pincfg[]) {
{ 0x0b, 0x0221101f }, /* HP */
@@ -5647,17 +5682,29 @@ static const struct alc_fixup alc861_fixups[] = {
{ }
}
},
- [PINFIX_ASUS_A6RP] = {
+ [ALC861_FIXUP_AMP_VREF_0F] = {
.type = ALC_FIXUP_FUNC,
.v.func = alc861_fixup_asus_amp_vref_0f,
},
+ [ALC861_FIXUP_NO_JACK_DETECT] = {
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc_fixup_no_jack_detect,
+ },
+ [ALC861_FIXUP_ASUS_A6RP] = {
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc861_fixup_asus_amp_vref_0f,
+ .chained = true,
+ .chain_id = ALC861_FIXUP_NO_JACK_DETECT,
+ }
};
static const struct snd_pci_quirk alc861_fixup_tbl[] = {
- SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", PINFIX_ASUS_A6RP),
- SND_PCI_QUIRK(0x1584, 0x0000, "Uniwill ECS M31EI", PINFIX_ASUS_A6RP),
- SND_PCI_QUIRK(0x1584, 0x2b01, "Haier W18", PINFIX_ASUS_A6RP),
- SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", PINFIX_FSC_AMILO_PI1505),
+ SND_PCI_QUIRK(0x1043, 0x1393, "ASUS A6Rp", ALC861_FIXUP_ASUS_A6RP),
+ SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", ALC861_FIXUP_AMP_VREF_0F),
+ SND_PCI_QUIRK(0x1462, 0x7254, "HP DX2200", ALC861_FIXUP_NO_JACK_DETECT),
+ SND_PCI_QUIRK(0x1584, 0x2b01, "Haier W18", ALC861_FIXUP_AMP_VREF_0F),
+ SND_PCI_QUIRK(0x1584, 0x0000, "Uniwill ECS M31EI", ALC861_FIXUP_AMP_VREF_0F),
+ SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", ALC861_FIXUP_FSC_AMILO_PI1505),
{}
};
@@ -5905,6 +5952,7 @@ enum {
ALC662_FIXUP_ASUS_MODE6,
ALC662_FIXUP_ASUS_MODE7,
ALC662_FIXUP_ASUS_MODE8,
+ ALC662_FIXUP_NO_JACK_DETECT,
};
static const struct alc_fixup alc662_fixups[] = {
@@ -6050,6 +6098,10 @@ static const struct alc_fixup alc662_fixups[] = {
.chained = true,
.chain_id = ALC662_FIXUP_SKU_IGNORE
},
+ [ALC662_FIXUP_NO_JACK_DETECT] = {
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc_fixup_no_jack_detect,
+ },
};
static const struct snd_pci_quirk alc662_fixup_tbl[] = {
@@ -6058,6 +6110,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800),
+ SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT),
SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_FIXUP_ASUS_MODE2),
SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 6345df131a0..4c769405d72 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1060,26 +1060,9 @@ static struct snd_kcontrol_new stac_smux_mixer = {
.put = stac92xx_smux_enum_put,
};
-static const char * const slave_vols[] = {
- "Front Playback Volume",
- "Surround Playback Volume",
- "Center Playback Volume",
- "LFE Playback Volume",
- "Side Playback Volume",
- "Headphone Playback Volume",
- "Speaker Playback Volume",
- NULL
-};
-
-static const char * const slave_sws[] = {
- "Front Playback Switch",
- "Surround Playback Switch",
- "Center Playback Switch",
- "LFE Playback Switch",
- "Side Playback Switch",
- "Headphone Playback Switch",
- "Speaker Playback Switch",
- "IEC958 Playback Switch",
+static const char * const slave_pfxs[] = {
+ "Front", "Surround", "Center", "LFE", "Side",
+ "Headphone", "Speaker", "IEC958",
NULL
};
@@ -1153,13 +1136,15 @@ static int stac92xx_build_controls(struct hda_codec *codec)
/* minimum value is actually mute */
vmaster_tlv[3] |= TLV_DB_SCALE_MUTE;
err = snd_hda_add_vmaster(codec, "Master Playback Volume",
- vmaster_tlv, slave_vols);
+ vmaster_tlv, slave_pfxs,
+ "Playback Volume");
if (err < 0)
return err;
}
if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
err = snd_hda_add_vmaster(codec, "Master Playback Switch",
- NULL, slave_sws);
+ NULL, slave_pfxs,
+ "Playback Switch");
if (err < 0)
return err;
}
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index dff9a00ee8f..c7eb4d7d05c 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1445,25 +1445,9 @@ static const struct hda_pcm_stream via_pcm_digital_capture = {
/*
* slave controls for virtual master
*/
-static const char * const via_slave_vols[] = {
- "Front Playback Volume",
- "Surround Playback Volume",
- "Center Playback Volume",
- "LFE Playback Volume",
- "Side Playback Volume",
- "Headphone Playback Volume",
- "Speaker Playback Volume",
- NULL,
-};
-
-static const char * const via_slave_sws[] = {
- "Front Playback Switch",
- "Surround Playback Switch",
- "Center Playback Switch",
- "LFE Playback Switch",
- "Side Playback Switch",
- "Headphone Playback Switch",
- "Speaker Playback Switch",
+static const char * const via_slave_pfxs[] = {
+ "Front", "Surround", "Center", "LFE", "Side",
+ "Headphone", "Speaker",
NULL,
};
@@ -1508,13 +1492,15 @@ static int via_build_controls(struct hda_codec *codec)
snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0],
HDA_OUTPUT, vmaster_tlv);
err = snd_hda_add_vmaster(codec, "Master Playback Volume",
- vmaster_tlv, via_slave_vols);
+ vmaster_tlv, via_slave_pfxs,
+ "Playback Volume");
if (err < 0)
return err;
}
if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
err = snd_hda_add_vmaster(codec, "Master Playback Switch",
- NULL, via_slave_sws);
+ NULL, via_slave_pfxs,
+ "Playback Switch");
if (err < 0)
return err;
}
generated by cgit v1.2.3-70-g09d2 (git 2.46.0) at 2025-02-13 00:22:29 +0000