diff options
author | Takashi Iwai <tiwai@suse.de> | 2005-05-25 18:11:40 +0200 |
---|---|---|
committer | Jaroslav Kysela <perex@suse.cz> | 2005-05-29 10:14:56 +0200 |
commit | 71fe7b82db923e0f71b05deeb54fce75d5a4d08d (patch) | |
tree | 3067daf15929d6f51f964ee929935f327e3d8a75 /sound | |
parent | 033a1f10923bdc3646750eef471a618f9d4c074b (diff) | |
download | kernel_samsung_crespo-71fe7b82db923e0f71b05deeb54fce75d5a4d08d.zip kernel_samsung_crespo-71fe7b82db923e0f71b05deeb54fce75d5a4d08d.tar.gz kernel_samsung_crespo-71fe7b82db923e0f71b05deeb54fce75d5a4d08d.tar.bz2 |
[ALSA] Fix ALC880 capture problems
HDA Codec driver
Fixed the capture on some ALC880 devices.
The node 0x07 is sometimes invalid as an ADC. Also, the intialization
of ADC nodes are fixed on some models.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Diffstat (limited to 'sound')
-rw-r--r-- | sound/pci/hda/patch_realtek.c | 86 |
1 files changed, 67 insertions, 19 deletions
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9f12d1f..ee1c4cd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -97,10 +97,20 @@ static hda_nid_t alc880_z71v_dac_nids[1] = { 0x02 }; +#if 0 +/* The datasheet says the node 0x07 is connected from inputs, + * but it shows zero connection in the real implementation. + */ static hda_nid_t alc880_adc_nids[3] = { /* ADC0-2 */ 0x07, 0x08, 0x09, }; +#else +static hda_nid_t alc880_adc_nids[2] = { + /* ADC1-2 */ + 0x08, 0x09, +}; +#endif #define ALC880_DIGOUT_NID 0x06 #define ALC880_DIGIN_NID 0x0a @@ -380,10 +390,11 @@ static snd_kcontrol_new_t alc880_base_mixer[] = { HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x19, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT), + /* We don't use NID 0x07 - see above */ + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer @@ -434,10 +445,11 @@ static snd_kcontrol_new_t alc880_five_stack_mixer[] = { HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x19, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT), + /* We don't use NID 0x07 - see above */ + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer @@ -506,6 +518,8 @@ static snd_kcontrol_new_t alc880_z71v_mixer[] = { HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x09, 0x0, HDA_INPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer @@ -514,7 +528,7 @@ static snd_kcontrol_new_t alc880_z71v_mixer[] = { */ /* .name = "Capture Source", */ .name = "Input Source", - .count = 2, + .count = 3, .info = alc_mux_enum_info, .get = alc_mux_enum_get, .put = alc_mux_enum_put, @@ -564,8 +578,16 @@ static struct hda_verb alc880_init_verbs_three_stack[] = { {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, /* unmute amp left and right */ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, - /* set connection select to line in (default select for this ADC) */ - {0x07, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* set connection select to mic in */ + {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* unmute amp left and right */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, + /* set connection select to mic in */ + {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* unmute amp left and right */ + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, + /* set connection select to mic in */ + {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, /* unmute front mixer amp left (volume = 0) */ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, /* mute pin widget amp left and right (no gain on this amp) */ @@ -639,8 +661,16 @@ static struct hda_verb alc880_init_verbs_five_stack[] = { {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, /* unmute amp left and right */ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, - /* set connection select to line in (default select for this ADC) */ - {0x07, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* set connection select to mic in */ + {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* unmute amp left and right */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, + /* set connection select to mic in */ + {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* unmute amp left and right */ + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, + /* set connection select to mic in */ + {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, /* unmute front mixer amp left and right (volume = 0) */ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, /* mute pin widget amp left and right (no gain on this amp) */ @@ -798,9 +828,16 @@ static struct hda_verb alc880_z71v_init_verbs[] = { {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, /* unmute amp left and right */ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, - /* set connection select to line in (default select for this ADC) */ - {0x07, AC_VERB_SET_CONNECT_SEL, 0x02}, - + /* set connection select to mic in */ + {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* unmute amp left and right */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, + /* set connection select to mic in */ + {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* unmute amp left and right */ + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, + /* set connection select to mic in */ + {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) for mixer * widget(nid=0x0B) to support the input path of analog loopback */ @@ -941,7 +978,9 @@ static struct hda_pcm_stream alc880_pcm_analog_capture = { .substreams = 2, .channels_min = 2, .channels_max = 2, - .nid = 0x07, /* NID to query formats and rates */ + .nid = 0x08, /* NID to query formats and rates + * (0x07 might be broken on some devices) + */ .ops = { .prepare = alc880_capture_pcm_prepare, .cleanup = alc880_capture_pcm_cleanup @@ -1261,6 +1300,13 @@ static struct hda_verb alc880_test_init_verbs[] = { {0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + /* ADC set up */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, + {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, + {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, + {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, { } }; #endif @@ -1822,11 +1868,13 @@ static struct hda_verb alc882_init_verbs[] = { {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, /* ADC1: unmute amp left and right */ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, + {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, /* ADC2: unmute amp left and right */ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, + {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, /* ADC3: unmute amp left and right */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, - + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, + {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Unmute front loopback */ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, /* Unmute rear loopback */ |