diff options
| author | Linus Torvalds <torvalds@g5.osdl.org> | 2006-09-23 17:21:12 -0700 |
|---|---|---|
| committer | Linus Torvalds <torvalds@g5.osdl.org> | 2006-09-23 17:21:12 -0700 |
| commit | f7425b160db500520c33f241edb066fc5c413f03 (patch) | |
| tree | f1f50b935fa49a273f8df685b5fb2fcf6a0f07a6 /sound/pci/hda/patch_analog.c | |
| parent | 9f261e011340bcd22c1dd48b465153bd78caa8c8 (diff) | |
| parent | f0063c4489a00ed5395378ef80a7edea4272f20b (diff) | |
Merge branch 'linus' of master.kernel.org:/pub/scm/linux/kernel/git/perex/alsa
* 'linus' of master.kernel.org:/pub/scm/linux/kernel/git/perex/alsa: (148 commits)
[ALSA] intel8x0m - Free irq in suspend
[ALSA] Move CONFIG_SND_AC97_POWER_SAVE to pci/Kconfig
[ALSA] usb-audio: add mixer control names for the Aureon 5.1 MkII
[ALSA] ES1938: remove duplicate field initialization
[ALSA] usb-audio: increase number of packets per URB
[ALSA] hda-codec - Fix headphone auto-toggle on sigmatel codec
[ALSA] hda-intel - A slight cleanup of timeout check in azx_get_response()
[ALSA] hda-codec - Fix mic input with STAC92xx codecs
[ALSA] mixart: Use SEEK_{SET,CUR,END} instead of hardcoded values
[ALSA] gus: Use SEEK_{SET,CUR,END} instead of hardcoded values
[ALSA] opl4: Use SEEK_{SET,CUR,END} instead of hardcoded values
[ALSA] sound core: Use SEEK_{SET,CUR,END} instead of hardcoded values
[ALSA] hda-codec - Support multiple headphone pins
[ALSA] hda_intel prefer 24bit instead of 20bit
[ALSA] hda-codec - Add vendor ids for Motorola and Conexant
[ALSA] hda-codec - Add device id for Motorola si3054-compatible codec
[ALSA] Add missing compat ioctls for ALSA control API
[ALSA] powermac - Fix Oops when conflicting with aoa driver
[ALSA] aoa: add locking to tas codec
[ALSA] hda-intel - Fix suspend/resume with MSI
...
Diffstat (limited to 'sound/pci/hda/patch_analog.c')
| -rw-r--r-- | sound/pci/hda/patch_analog.c | 21 |
1 files changed, 16 insertions, 5 deletions
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 6823f2bc10b3..511df07fa2a3 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -488,9 +488,13 @@ static struct snd_kcontrol_new ad1986a_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "PCM Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, .info = ad1986a_pcm_amp_vol_info, .get = ad1986a_pcm_amp_vol_get, .put = ad1986a_pcm_amp_vol_put, + .tlv = { .c = snd_hda_mixer_amp_tlv }, .private_value = HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT) }, { @@ -637,6 +641,7 @@ static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = { .info = snd_hda_mixer_amp_volume_info, .get = snd_hda_mixer_amp_volume_get, .put = ad1986a_laptop_master_vol_put, + .tlv = { .c = snd_hda_mixer_amp_tlv }, .private_value = HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT), }, { @@ -791,6 +796,8 @@ static struct hda_board_config ad1986a_cfg_tbl[] = { .config = AD1986A_3STACK }, /* ASUS A8N-VM CSM */ { .pci_subvendor = 0x1043, .pci_subdevice = 0x81b3, .config = AD1986A_3STACK }, /* ASUS P5RD2-VM / P5GPL-X SE */ + { .pci_subvendor = 0x1043, .pci_subdevice = 0x81cb, + .config = AD1986A_3STACK }, /* ASUS M2NPV-VM */ { .modelname = "laptop", .config = AD1986A_LAPTOP }, { .pci_subvendor = 0x144d, .pci_subdevice = 0xc01e, .config = AD1986A_LAPTOP }, /* FSC V2060 */ @@ -803,6 +810,8 @@ static struct hda_board_config ad1986a_cfg_tbl[] = { .config = AD1986A_LAPTOP_EAPD }, /* Samsung X60 Chane */ { .pci_subvendor = 0x144d, .pci_subdevice = 0xc024, .config = AD1986A_LAPTOP_EAPD }, /* Samsung R65-T2300 Charis */ + { .pci_subvendor = 0x144d, .pci_subdevice = 0xc026, + .config = AD1986A_LAPTOP_EAPD }, /* Samsung X10-T2300 Culesa */ { .pci_subvendor = 0x1043, .pci_subdevice = 0x1153, .config = AD1986A_LAPTOP_EAPD }, /* ASUS M9 */ { .pci_subvendor = 0x1043, .pci_subdevice = 0x1213, @@ -1626,10 +1635,12 @@ static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct ad198x_spec *spec = codec->spec; - if (spec->need_dac_fix) + int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode, + spec->num_channel_mode, + &spec->multiout.max_channels); + if (! err && spec->need_dac_fix) spec->multiout.num_dacs = spec->multiout.max_channels / 2; - return snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode, - spec->num_channel_mode, &spec->multiout.max_channels); + return err; } /* 6-stack mode */ @@ -2460,7 +2471,7 @@ static void ad1988_auto_init_extra_out(struct hda_codec *codec) pin = spec->autocfg.speaker_pins[0]; if (pin) /* connect to front */ ad1988_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); - pin = spec->autocfg.hp_pin; + pin = spec->autocfg.hp_pins[0]; if (pin) /* connect to front */ ad1988_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); } @@ -2512,7 +2523,7 @@ static int ad1988_parse_auto_config(struct hda_codec *codec) (err = ad1988_auto_create_extra_out(codec, spec->autocfg.speaker_pins[0], "Speaker")) < 0 || - (err = ad1988_auto_create_extra_out(codec, spec->autocfg.hp_pin, + (err = ad1988_auto_create_extra_out(codec, spec->autocfg.hp_pins[0], "Headphone")) < 0 || (err = ad1988_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0) return err; |
