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[alsa-devel] [PATCH] hda: Added STAC92HD73 support.
by Matthew Ranostay 13 Dec '07

13 Dec '07

Added support for new STAC92HD73 family of codecs. Additionally added features for multiple analog loopbacks, and multiple dmux mixers. Regression testing for the analog loopback changes for STAC9205 and STAC9274D completed with any issues, as well for the dmux changes. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r eb09731e73d7 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Mon Dec 03 17:08:40 2007 +0100 +++ b/pci/hda/patch_sigmatel.c Tue Dec 04 11:28:40 2007 -0500 @@ -62,6 +62,11 @@ enum { }; enum { + STAC_92HD73XX_REF, + STAC_92HD73XX_MODELS +}; + +enum { STAC_92HD71BXX_REF, STAC_92HD71BXX_MODELS }; @@ -118,6 +123,7 @@ struct sigmatel_spec { unsigned int gpio_mute: 1; unsigned int gpio_mask, gpio_data; + unsigned int aloopback_mask; /* playback */ struct hda_multi_out multiout; @@ -130,7 +136,7 @@ struct sigmatel_spec { unsigned int num_muxes; hda_nid_t *dmic_nids; unsigned int num_dmics; - hda_nid_t dmux_nid; + hda_nid_t *dmux_nids; hda_nid_t dig_in_nid; /* pin widgets */ @@ -145,7 +151,7 @@ struct sigmatel_spec { /* capture source */ struct hda_input_mux *dinput_mux; - unsigned int cur_dmux; + unsigned int cur_dmux[2]; struct hda_input_mux *input_mux; unsigned int cur_mux[3]; @@ -176,12 +182,38 @@ static hda_nid_t stac9200_dac_nids[1] = 0x02, }; +static hda_nid_t stac92hd73xx_adc_nids[2] = { + 0x1a, 0x1b +}; + +#define STAC92HD73XX_NUM_DMICS 2 +static hda_nid_t stac92hd73xx_dmic_nids[STAC92HD73XX_NUM_DMICS + 1] = { + 0x13, 0x14, 0 +}; + +#define STAC92HD73_DAC_COUNT 5 +static hda_nid_t stac92hd73xx_dac_nids[STAC92HD73_DAC_COUNT] = { + 0x15, 0x16, 0x17, 0x18, 0x19, +}; + +static hda_nid_t stac92hd73xx_mux_nids[4] = { + 0x28, 0x29, 0x2a, 0x2b, +}; + +static hda_nid_t stac92hd73xx_dmux_nids[2] = { + 0x20, 0x21, +}; + static hda_nid_t stac92hd71bxx_adc_nids[2] = { 0x12, 0x13, }; static hda_nid_t stac92hd71bxx_mux_nids[2] = { 0x1a, 0x1b +}; + +static hda_nid_t stac92hd71bxx_dmux_nids[1] = { + 0x1c, }; static hda_nid_t stac92hd71bxx_dac_nids[2] = { @@ -226,6 +258,10 @@ static hda_nid_t stac927x_mux_nids[3] = 0x15, 0x16, 0x17 }; +static hda_nid_t stac927x_dmux_nids[1] = { + 0x1b, +}; + #define STAC927X_NUM_DMICS 2 static hda_nid_t stac927x_dmic_nids[STAC927X_NUM_DMICS + 1] = { 0x13, 0x14, 0 @@ -239,6 +275,10 @@ static hda_nid_t stac9205_mux_nids[2] = 0x19, 0x1a }; +static hda_nid_t stac9205_dmux_nids[1] = { + 0x1d, +}; + #define STAC9205_NUM_DMICS 2 static hda_nid_t stac9205_dmic_nids[STAC9205_NUM_DMICS + 1] = { 0x17, 0x18, 0 @@ -257,6 +297,12 @@ static hda_nid_t stac922x_pin_nids[10] = static hda_nid_t stac922x_pin_nids[10] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x15, 0x1b, +}; + +static hda_nid_t stac92hd73xx_pin_nids[12] = { + 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, + 0x0f, 0x10, 0x11, 0x12, 0x13, + 0x14, 0x22 }; static hda_nid_t stac92hd71bxx_pin_nids[10] = { @@ -289,8 +335,9 @@ static int stac92xx_dmux_enum_get(struct { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; - - ucontrol->value.enumerated.item[0] = spec->cur_dmux; + unsigned int dmux_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + + ucontrol->value.enumerated.item[0] = spec->cur_dmux[dmux_idx]; return 0; } @@ -299,9 +346,10 @@ static int stac92xx_dmux_enum_put(struct { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; + unsigned int dmux_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); return snd_hda_input_mux_put(codec, spec->dinput_mux, ucontrol, - spec->dmux_nid, &spec->cur_dmux); + spec->dmux_nids[dmux_idx], &spec->cur_dmux[dmux_idx]); } static int stac92xx_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -337,9 +385,11 @@ static int stac92xx_aloopback_get(struct struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - - ucontrol->value.integer.value[0] = spec->aloopback; + unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + struct sigmatel_spec *spec = codec->spec; + + ucontrol->value.integer.value[0] = spec->aloopback & + (spec->aloopback_mask << idx); return 0; } @@ -348,24 +398,29 @@ static int stac92xx_aloopback_put(struct { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; + unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); unsigned int dac_mode; - unsigned int val; - + unsigned int val, idx_val; + + idx_val = (spec->aloopback_mask & 0xFF) << idx; val = !!ucontrol->value.integer.value[0]; - if (spec->aloopback == val) + if ((spec->aloopback | idx) == val) return 0; - spec->aloopback = val; - + spec->aloopback ^= idx_val; + + /* Only return the bits defined by the shift value of the + * first two bytes of the mask */ dac_mode = snd_hda_codec_read(codec, codec->afg, 0, - kcontrol->private_value & 0xFFFF, 0x0); - - if (spec->aloopback) { + kcontrol->private_value & 0xFFFF, 0x0) >> + ((spec->aloopback_mask & 0xFF00) >> 8); + + if (spec->aloopback & idx_val) { snd_hda_power_up(codec); - dac_mode |= 0x40; + dac_mode |= idx_val; } else { snd_hda_power_down(codec); - dac_mode &= ~0x40; + dac_mode &= ~idx_val; } snd_hda_codec_write_cache(codec, codec->afg, 0, @@ -384,6 +439,86 @@ static struct hda_verb stac9200_eapd_ini /* set dac0mux for dac converter */ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x08, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, + {} +}; + +static struct hda_verb stac92hd73xx_6ch_core_init[] = { + /* set master volume and direct control */ + { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* setup audio connections */ + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00}, + { 0x10, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x11, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* setup adcs to point to mixer */ + { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Front Mic */ + { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Mic */ + { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Line In */ + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* setup import muxs */ + { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x00}, + {} +}; + +static struct hda_verb stac92hd73xx_8ch_core_init[] = { + /* set master volume and direct control */ + { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* setup audio connections */ + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00}, + { 0x10, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x11, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* connect hp ports to dac3 */ + { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x03}, + { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x03}, + /* setup adcs to point to mixer */ + { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Front Mic */ + { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Mic */ + { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Line In */ + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* setup import muxs */ + { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x03}, + {} +}; + +static struct hda_verb stac92hd73xx_10ch_core_init[] = { + /* set master volume and direct control */ + { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* setup audio connections */ + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 }, + { 0x10, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { 0x11, AC_VERB_SET_CONNECT_SEL, 0x02 }, + /* dac3 is connected to import3 mux */ + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0xb07f}, + /* connect hp ports to dac4 */ + { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x04}, + { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x04}, + /* setup adcs to point to mixer */ + { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Front Mic */ + { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Mic */ + { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Line In */ + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* setup import muxs */ + { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x03}, {} }; @@ -460,11 +595,11 @@ static struct hda_verb stac9205_core_ini .put = stac92xx_mux_enum_put, \ } -#define STAC_ANALOG_LOOPBACK(verb_read,verb_write) \ +#define STAC_ANALOG_LOOPBACK(verb_read, verb_write, cnt) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = "Analog Loopback", \ - .count = 1, \ + .count = cnt, \ .info = stac92xx_aloopback_info, \ .get = stac92xx_aloopback_get, \ .put = stac92xx_aloopback_put, \ @@ -481,6 +616,99 @@ static struct snd_kcontrol_new stac9200_ { } /* end */ }; +static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = { + STAC_DIGITAL_INPUT_SOURCE(2), + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), + + /* hardware gain controls */ + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), + + HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT), + + HDA_CODEC_VOLUME("Line In Mixer Capture Volume", 0x1d, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("Line In Mixer Capture Switch", 0x1d, 0x2, HDA_INPUT), + + HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT), + + HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT), + { } /* end */ +}; + +static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = { + STAC_DIGITAL_INPUT_SOURCE(2), + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 4), + + /* hardware gain controls */ + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), + + HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT), + + HDA_CODEC_VOLUME("Line In Mixer Capture Volume", 0x1d, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("Line In Mixer Capture Switch", 0x1d, 0x2, HDA_INPUT), + + HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT), + + HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT), + { } /* end */ +}; + +static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = { + STAC_DIGITAL_INPUT_SOURCE(2), + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5), + + /* hardware gain controls */ + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), + + HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT), + + HDA_CODEC_VOLUME("Line In Mixer Capture Volume", 0x1d, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("Line In Mixer Capture Switch", 0x1d, 0x2, HDA_INPUT), + + HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT), + + HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT), + { } /* end */ +}; + static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { STAC_DIGITAL_INPUT_SOURCE(1), STAC_INPUT_SOURCE(2), @@ -513,7 +741,7 @@ static struct snd_kcontrol_new stac9205_ static struct snd_kcontrol_new stac9205_mixer[] = { STAC_DIGITAL_INPUT_SOURCE(1), STAC_INPUT_SOURCE(2), - STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0), + STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0, 1), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1b, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1d, 0x0, HDA_OUTPUT), @@ -543,7 +771,7 @@ static struct snd_kcontrol_new stac927x_ static struct snd_kcontrol_new stac927x_mixer[] = { STAC_DIGITAL_INPUT_SOURCE(1), STAC_INPUT_SOURCE(3), - STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB), + STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB, 1), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x18, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1b, 0x0, HDA_OUTPUT), @@ -851,6 +1079,27 @@ static struct snd_pci_quirk stac925x_cfg SND_PCI_QUIRK(0x107b, 0x0461, "Gateway NX560XL", STAC_MA6), SND_PCI_QUIRK(0x107b, 0x0681, "Gateway NX860", STAC_PA6), SND_PCI_QUIRK(0x1002, 0x437b, "Gateway MX6453", STAC_M2_2), + {} /* terminator */ +}; + +static unsigned int ref92hd73xx_pin_configs[12] = { + 0x02214030, 0x02a19040, 0x01a19020, 0x02214030, + 0x0181302e, 0x01014010, 0x01014020, 0x01014030, + 0x02319040, 0x90a000f0, 0x90a000f0, 0x01452050, +}; + +static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = { + [STAC_92HD73XX_REF] = ref92hd73xx_pin_configs, +}; + +static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = { + [STAC_92HD73XX_REF] = "ref", +}; + +static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { + /* SigmaTel reference board */ + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, + "DFI LanParty", STAC_92HD73XX_REF), {} /* terminator */ }; @@ -2030,7 +2279,7 @@ static int stac92xx_auto_create_dmic_inp continue; num_cons = snd_hda_get_connections(codec, - spec->dmux_nid, + spec->dmux_nids[0], con_lst, HDA_MAX_NUM_INPUTS); for (j = 0; j < num_cons; j++) @@ -2211,7 +2460,8 @@ static int stac92xx_parse_auto_config(st spec->mixers[spec->num_mixers++] = spec->kctl_alloc; spec->input_mux = &spec->private_imux; - spec->dinput_mux = &spec->private_dimux; + if (!spec->dinput_mux) + spec->dinput_mux = &spec->private_dimux; return 1; } @@ -2686,6 +2936,111 @@ static int patch_stac925x(struct hda_cod } err = -EINVAL; } + if (err < 0) { + stac92xx_free(codec); + return err; + } + + codec->patch_ops = stac92xx_patch_ops; + + return 0; +} + +static struct hda_input_mux stac92hd73xx_dmux = { + .num_items = 4, + .items = { + { "Analog Inputs", 0x0b }, + { "CD", 0x08 }, + { "Digital Mic 1", 0x09 }, + { "Digital Mic 2", 0x0a }, + } +}; + +static int patch_stac92hd73xx(struct hda_codec *codec) +{ + struct sigmatel_spec *spec; + hda_nid_t conn[STAC92HD73_DAC_COUNT + 2]; + int err = 0; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + spec->num_pins = ARRAY_SIZE(stac92hd73xx_pin_nids); + spec->pin_nids = stac92hd73xx_pin_nids; + spec->board_config = snd_hda_check_board_config(codec, + STAC_92HD73XX_MODELS, + stac92hd73xx_models, + stac92hd73xx_cfg_tbl); +again: + if (spec->board_config < 0) { + snd_printdd(KERN_INFO "hda_codec: Unknown model for" + " STAC92HD73XX, using BIOS defaults\n"); + err = stac92xx_save_bios_config_regs(codec); + if (err < 0) { + stac92xx_free(codec); + return err; + } + spec->pin_configs = spec->bios_pin_configs; + } else { + spec->pin_configs = stac92hd73xx_brd_tbl[spec->board_config]; + stac92xx_set_config_regs(codec); + } + + spec->multiout.num_dacs = snd_hda_get_connections(codec, 0x0a, + conn, STAC92HD73_DAC_COUNT + 2) - 1; + + if (spec->multiout.num_dacs < 0) { + printk(KERN_WARNING "hda_codec: Could not determine " + "number of channels defaulting to DAC count\n"); + spec->multiout.num_dacs = STAC92HD73_DAC_COUNT; + } + + switch (spec->multiout.num_dacs) { + case 0x3: /* 6 Channel */ + spec->mixer = stac92hd73xx_6ch_mixer; + spec->init = stac92hd73xx_6ch_core_init; + break; + case 0x4: /* 8 Channel */ + spec->multiout.hp_nid = 0x18; + spec->mixer = stac92hd73xx_8ch_mixer; + spec->init = stac92hd73xx_8ch_core_init; + break; + case 0x5: /* 10 Channel */ + spec->multiout.hp_nid = 0x19; + spec->mixer = stac92hd73xx_10ch_mixer; + spec->init = stac92hd73xx_10ch_core_init; + }; + + spec->multiout.dac_nids = stac92hd73xx_dac_nids; + spec->aloopback_mask = 0x801; + + spec->mux_nids = stac92hd73xx_mux_nids; + spec->adc_nids = stac92hd73xx_adc_nids; + spec->dmic_nids = stac92hd73xx_dmic_nids; + spec->dmux_nids = stac92hd73xx_dmux_nids; + + spec->num_muxes = ARRAY_SIZE(stac92hd73xx_mux_nids); + spec->num_adcs = ARRAY_SIZE(stac92hd73xx_adc_nids); + spec->num_dmics = STAC92HD73XX_NUM_DMICS; + spec->dinput_mux = &stac92hd73xx_dmux; + /* GPIO0 High = Enable EAPD */ + spec->gpio_mask = spec->gpio_data = 0x000001; + stac92xx_enable_gpio_mask(codec); + + err = stac92xx_parse_auto_config(codec, 0x22, 0x24); + + if (!err) { + if (spec->board_config < 0) { + printk(KERN_WARNING "hda_codec: No auto-config is " + "available, default to model=ref\n"); + spec->board_config = STAC_92HD73XX_REF; + goto again; + } + err = -EINVAL; + } + if (err < 0) { stac92xx_free(codec); return err; @@ -2736,7 +3091,7 @@ again: spec->mux_nids = stac92hd71bxx_mux_nids; spec->adc_nids = stac92hd71bxx_adc_nids; spec->dmic_nids = stac92hd71bxx_dmic_nids; - spec->dmux_nid = 0x1c; + spec->dmux_nids = stac92hd71bxx_dmux_nids; spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids); @@ -2931,7 +3286,7 @@ static int patch_stac927x(struct hda_cod case 0x10280209: spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; - spec->dmux_nid = 0x1b; + spec->dmux_nids = stac927x_dmux_nids; /* Enable DMIC0 */ stac92xx_set_config_reg(codec, 0x13, 0x90a60040); @@ -2947,6 +3302,7 @@ static int patch_stac927x(struct hda_cod } spec->multiout.dac_nids = spec->dac_nids; + spec->aloopback_mask = 0x40; stac92xx_enable_gpio_mask(codec); err = stac92xx_parse_auto_config(codec, 0x1e, 0x20); @@ -3004,11 +3360,12 @@ static int patch_stac9205(struct hda_cod spec->num_muxes = ARRAY_SIZE(stac9205_mux_nids); spec->dmic_nids = stac9205_dmic_nids; spec->num_dmics = STAC9205_NUM_DMICS; - spec->dmux_nid = 0x1d; + spec->dmux_nids = stac9205_dmux_nids; spec->init = stac9205_core_init; spec->mixer = stac9205_mixer; + spec->aloopback_mask = 0x40; spec->multiout.dac_nids = spec->dac_nids; switch (spec->board_config){ @@ -3337,6 +3694,8 @@ struct hda_codec_preset snd_hda_preset_s { .id = 0x838476a5, .name = "STAC9255D", .patch = patch_stac9205 }, { .id = 0x838476a6, .name = "STAC9254", .patch = patch_stac9205 }, { .id = 0x838476a7, .name = "STAC9254D", .patch = patch_stac9205 }, + { .id = 0x111d7676, .name = "92HD73E1X5", .patch = patch_stac92hd73xx }, + { .id = 0x111d7675, .name = "92HD73D1X5", .patch = patch_stac92hd73xx }, { .id = 0x111d76b0, .name = "92HD71BXX", .patch = patch_stac92hd71bxx }, {} /* terminator */ };

2 3

[alsa-devel] [patch 13/18] alsa: nopage
by npiggin＠suse.de 13 Dec '07

13 Dec '07

Convert ALSA from nopage to fault. Switch from OOM to SIGBUS if the resource is not available. Signed-off-by: Nick Piggin <npiggin(a)suse.de> Cc: perex(a)perex.cz Cc: tiwai(a)suse.de Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- sound/core/pcm_native.c | 59 ++++++++++++++++++++---------------------------- 1 file changed, 25 insertions(+), 34 deletions(-) Index: linux-2.6/sound/core/pcm_native.c =================================================================== --- linux-2.6.orig/sound/core/pcm_native.c +++ linux-2.6/sound/core/pcm_native.c @@ -3018,26 +3018,23 @@ static unsigned int snd_pcm_capture_poll /* * mmap status record */ -static struct page * snd_pcm_mmap_status_nopage(struct vm_area_struct *area, - unsigned long address, int *type) +static int snd_pcm_mmap_status_fault(struct vm_area_struct *area, + struct vm_fault *vmf) { struct snd_pcm_substream *substream = area->vm_private_data; struct snd_pcm_runtime *runtime; - struct page * page; if (substream == NULL) - return NOPAGE_SIGBUS; + return VM_FAULT_SIGBUS; runtime = substream->runtime; - page = virt_to_page(runtime->status); - get_page(page); - if (type) - *type = VM_FAULT_MINOR; - return page; + vmf->page = virt_to_page(runtime->status); + get_page(vmf->page); + return 0; } static struct vm_operations_struct snd_pcm_vm_ops_status = { - .nopage = snd_pcm_mmap_status_nopage, + .fault = snd_pcm_mmap_status_fault, }; static int snd_pcm_mmap_status(struct snd_pcm_substream *substream, struct file *file, @@ -3061,26 +3058,23 @@ static int snd_pcm_mmap_status(struct sn /* * mmap control record */ -static struct page * snd_pcm_mmap_control_nopage(struct vm_area_struct *area, - unsigned long address, int *type) +static int snd_pcm_mmap_control_fault(struct vm_area_struct *area, + struct vm_fault *vmf) { struct snd_pcm_substream *substream = area->vm_private_data; struct snd_pcm_runtime *runtime; - struct page * page; if (substream == NULL) - return NOPAGE_SIGBUS; + return VM_FAULT_SIGBUS; runtime = substream->runtime; - page = virt_to_page(runtime->control); - get_page(page); - if (type) - *type = VM_FAULT_MINOR; - return page; + vmf->page = virt_to_page(runtime->control); + get_page(vmf->page); + return 0; } static struct vm_operations_struct snd_pcm_vm_ops_control = { - .nopage = snd_pcm_mmap_control_nopage, + .fault = snd_pcm_mmap_control_fault, }; static int snd_pcm_mmap_control(struct snd_pcm_substream *substream, struct file *file, @@ -3117,10 +3111,10 @@ static int snd_pcm_mmap_control(struct s #endif /* coherent mmap */ /* - * nopage callback for mmapping a RAM page + * fault callback for mmapping a RAM page */ -static struct page *snd_pcm_mmap_data_nopage(struct vm_area_struct *area, - unsigned long address, int *type) +static int snd_pcm_mmap_data_fault(struct vm_area_struct *area, + struct vm_fault *vmf) { struct snd_pcm_substream *substream = area->vm_private_data; struct snd_pcm_runtime *runtime; @@ -3130,33 +3124,30 @@ static struct page *snd_pcm_mmap_data_no size_t dma_bytes; if (substream == NULL) - return NOPAGE_SIGBUS; + return VM_FAULT_SIGBUS; runtime = substream->runtime; - offset = area->vm_pgoff << PAGE_SHIFT; - offset += address - area->vm_start; - snd_assert((offset % PAGE_SIZE) == 0, return NOPAGE_SIGBUS); + offset = vmf->pgoff << PAGE_SHIFT; dma_bytes = PAGE_ALIGN(runtime->dma_bytes); if (offset > dma_bytes - PAGE_SIZE) - return NOPAGE_SIGBUS; + return VM_FAULT_SIGBUS; if (substream->ops->page) { page = substream->ops->page(substream, offset); - if (! page) - return NOPAGE_OOM; /* XXX: is this really due to OOM? */ + if (!page) + return VM_FAULT_SIGBUS; } else { vaddr = runtime->dma_area + offset; page = virt_to_page(vaddr); } get_page(page); - if (type) - *type = VM_FAULT_MINOR; - return page; + vmf->page = page; + return 0; } static struct vm_operations_struct snd_pcm_vm_ops_data = { .open = snd_pcm_mmap_data_open, .close = snd_pcm_mmap_data_close, - .nopage = snd_pcm_mmap_data_nopage, + .fault = snd_pcm_mmap_data_fault, }; /* --

2 1

[alsa-devel] [PATCH] Initial support of the Mitac 8252D (based on ALC883)
by Andy Shevchenko 13 Dec '07

13 Dec '07

Hi! The attached patch adds initial support of the Mitac 8252D (http://www.mitac-mtc.com.tw/English/products/8252Dspec.htm). Working: - Front speakers (volume + mute) - Center/LFE speakers (volume + mute) - HP out (with Front Volume) - HP individual mute switch - HP Jack sense - Front Mic and its volume Not tested: - external mic and its volume Not working while now: - Mic Jack sense Questionable: - is Mic have Jack sense? - one or two Mic volume controls? - CD/Line-in: presense in the mixer Signed-off-by: Andy Shevchenko <andy(a)smile.org.ua> --- alsa-kernel/pci/hda/patch_realtek.c.orig 2007-12-05 17:44:28.000000000 +0300 +++ alsa-kernel/pci/hda/patch_realtek.c 2007-12-05 20:16:46.000000000 +0300 @@ -185,6 +185,7 @@ ALC883_HAIER_W66, ALC888_6ST_HP, ALC888_3ST_HP, + ALC883_MITAC, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -6342,6 +6343,36 @@ { } /* end */ }; +static struct snd_kcontrol_new alc883_mitac_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + 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, + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc883_mux_enum_info, + .get = alc883_mux_enum_get, + .put = alc883_mux_enum_put, + }, + { } /* end */ +}; + static struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -6774,6 +6805,67 @@ { } }; +/* toggle speaker-output according to the hp-jack state */ +static void alc883_mitac_hp_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); +} + +/* auto-toggle front mic */ +/* +static void alc883_mitac_mic_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_codec_read(codec, 0x18, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits); +} +*/ + +static void alc883_mitac_automute(struct hda_codec *codec) +{ + alc883_mitac_hp_automute(codec); + /* alc883_mitac_mic_automute(codec); */ +} + +static void alc883_mitac_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc883_mitac_hp_automute(codec); + break; + case ALC880_MIC_EVENT: + /* alc883_mitac_mic_automute(codec); */ + break; + } +} + +static struct hda_verb alc883_mitac_verbs[] = { + /* HP */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Subwoofer */ + {0x17, AC_VERB_SET_CONNECT_SEL, 0x02}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + + /* enable unsolicited event */ + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + /* {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN}, */ + + { } /* end */ +}; + static struct hda_verb alc883_tagra_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -7156,6 +7246,7 @@ [ALC883_HAIER_W66] = "haier-w66", [ALC888_6ST_HP] = "6stack-hp", [ALC888_3ST_HP] = "3stack-hp", + [ALC883_MITAC] = "mitac", [ALC883_AUTO] = "auto", }; @@ -7190,6 +7281,7 @@ SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), + SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch), SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo 101e", ALC883_LENOVO_101E_2ch), @@ -7437,6 +7529,19 @@ .need_dac_fix = 1, .input_mux = &alc883_capture_source, }, + [ALC883_MITAC] = { + .mixers = { alc883_mitac_mixer }, + .init_verbs = { alc883_init_verbs, alc883_mitac_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .input_mux = &alc883_capture_source, + .unsol_event = alc883_mitac_unsol_event, + .init_hook = alc883_mitac_automute, + }, }; -- With best regards, Andy Shevchenko. mailto: andy(a)smile.org.ua

3 5

[alsa-devel] [patch] Device ID for MSI L745
by zhejiang 13 Dec '07

13 Dec '07

Hi, Please refer to the [0003641] on bugtracking system. The user found that "model=targa-2ch-dig" works and provided the ID. diff -Nur a/alsa-driver-hg20071211/alsa-kernel/pci/hda/patch_realtek.c b/alsa-driver-hg20071211/alsa-kernel/pci/hda/patch_realtek.c --- a/alsa-driver-hg20071211/alsa-kernel/pci/hda/patch_realtek.c 2007-12-04 09:00:07.000000000 +0800 +++ b/alsa-driver-hg20071211/alsa-kernel/pci/hda/patch_realtek.c 2007-12-11 15:30:20.000000000 +0800 @@ -7173,6 +7173,7 @@ SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG), + SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3ef9, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3b7f, "MSI", ALC883_TARGA_2ch_DIG),

2 1

[alsa-devel] [PATCH] 1616(M) cardbus improvements
by Ctirad Fertr 13 Dec '07

13 Dec '07

Hello, this patch improves E-Mu 1616(M) cardbus support. It adds definitions of the new Microdock and 1010 cardbus registers (thanks again for descriptions James) and improves mixer for this card. Now you can use S/PDIF and ADAT on Mirodock and also use headpohone output on host cardbus card as another independent output. Signed-off-by: Ctirad Fertr <c.fertr(a)gmail.com> ------ Ctirad

3 7

[alsa-devel] [PATCH] hda-codec - Fix definition of AC_KNBCAP_DELTA to match datasheet.
by Andrew Paprocki 13 Dec '07

13 Dec '07

hda-codec - Fix definition of AC_KNBCAP_DELTA to match datasheet. AC_KNBCAP_DELTA is incorrectly defined as (1<<8). According to the ALC883 datasheet rev 1.3, this is bit 7 after AC_KNBCAP_NUM_STEPS which is a 0x7f mask. Signed-off-by: Andrew Paprocki <andrew(a)ishiboo.com> diff -r da002fb740a42b4fd6dcbd45693f9bee3c828812 -r 65b577e690e95e4ace950c3f986270dd3e992192 pci/hda/hda_codec.h --- a/pci/hda/hda_codec.h Mon Dec 03 22:52:58 2007 -0500 +++ b/pci/hda/hda_codec.h Mon Dec 03 23:02:51 2007 -0500 @@ -238,7 +238,7 @@ enum { /* Volume knobs capabilities */ #define AC_KNBCAP_NUM_STEPS (0x7f<<0) -#define AC_KNBCAP_DELTA (1<<8) +#define AC_KNBCAP_DELTA (1<<7) /* * Control Parameters

2 1

[alsa-devel] [PATCH] hda-codec - Add missing #defines (and 1 rename) in hda_codec.h.
by Andrew Paprocki 13 Dec '07

13 Dec '07

hda-codec - Add missing #defines (and 1 rename) in hda_codec.h. Added AC_VERB_GET_DIGI_CONVERT_2 and renamed AC_VERB_GET_DIGI_CONVERT to AC_VERB_GET_DIGI_CONVERT_1 to stay consistent with the SET variants. Added AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, and AC_PINCAP_LR_SWAP. The missing fields were listed in the ALC883 datasheet rev 1.3. Signed-off-by: Andrew Paprocki <andrew(a)ishiboo.com> diff -r eb09731e73d7928c70ba776c1719149d112ee96a -r da002fb740a42b4fd6dcbd45693f9bee3c828812 pci/hda/hda_codec.c --- a/pci/hda/hda_codec.c Mon Dec 03 17:08:40 2007 +0100 +++ b/pci/hda/hda_codec.c Mon Dec 03 22:52:58 2007 -0500 @@ -1434,8 +1434,9 @@ int snd_hda_create_spdif_out_ctls(struct if (err < 0) return err; } - codec->spdif_ctls = - snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT, 0); + codec->spdif_ctls = + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DIGI_CONVERT_1, 0); codec->spdif_status = convert_to_spdif_status(codec->spdif_ctls); return 0; } @@ -1482,7 +1483,7 @@ static int snd_hda_spdif_in_status_get(s unsigned short val; unsigned int sbits; - val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT, 0); + val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT_1, 0); sbits = convert_to_spdif_status(val); ucontrol->value.iec958.status[0] = sbits; ucontrol->value.iec958.status[1] = sbits >> 8; @@ -1533,7 +1534,8 @@ int snd_hda_create_spdif_in_ctls(struct return err; } codec->spdif_in_enable = - snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT, 0) & + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DIGI_CONVERT_1, 0) & AC_DIG1_ENABLE; return 0; } diff -r eb09731e73d7928c70ba776c1719149d112ee96a -r da002fb740a42b4fd6dcbd45693f9bee3c828812 pci/hda/hda_codec.h --- a/pci/hda/hda_codec.h Mon Dec 03 17:08:40 2007 +0100 +++ b/pci/hda/hda_codec.h Mon Dec 03 22:52:58 2007 -0500 @@ -77,12 +77,14 @@ enum { #define AC_VERB_GET_PIN_SENSE 0x0f09 #define AC_VERB_GET_BEEP_CONTROL 0x0f0a #define AC_VERB_GET_EAPD_BTLENABLE 0x0f0c -#define AC_VERB_GET_DIGI_CONVERT 0x0f0d +#define AC_VERB_GET_DIGI_CONVERT_1 0x0f0d +#define AC_VERB_GET_DIGI_CONVERT_2 0x0f0e #define AC_VERB_GET_VOLUME_KNOB_CONTROL 0x0f0f /* f10-f1a: GPIO */ #define AC_VERB_GET_GPIO_DATA 0x0f15 #define AC_VERB_GET_GPIO_MASK 0x0f16 #define AC_VERB_GET_GPIO_DIRECTION 0x0f17 +#deifne AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK 0x0f19 #define AC_VERB_GET_CONFIG_DEFAULT 0x0f1c /* f20: AFG/MFG */ #define AC_VERB_GET_SUBSYSTEM_ID 0x0f20 @@ -110,6 +112,7 @@ enum { #define AC_VERB_SET_GPIO_DATA 0x715 #define AC_VERB_SET_GPIO_MASK 0x716 #define AC_VERB_SET_GPIO_DIRECTION 0x717 +#define AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK 0x719 #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 0x71c #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_1 0x71d #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_2 0x71e @@ -190,6 +193,7 @@ enum { #define AC_PINCAP_OUT (1<<4) /* output capable */ #define AC_PINCAP_IN (1<<5) /* input capable */ #define AC_PINCAP_BALANCE (1<<6) /* balanced I/O capable */ +#define AC_PINCAP_LR_SWAP (1<<7) /* L/R swap */ #define AC_PINCAP_VREF (0x37<<8) #define AC_PINCAP_VREF_SHIFT 8 #define AC_PINCAP_EAPD (1<<16) /* EAPD capable */ diff -r eb09731e73d7928c70ba776c1719149d112ee96a -r da002fb740a42b4fd6dcbd45693f9bee3c828812 pci/hda/patch_realtek.c --- a/pci/hda/patch_realtek.c Mon Dec 03 17:08:40 2007 +0100 +++ b/pci/hda/patch_realtek.c Mon Dec 03 22:52:58 2007 -0500 @@ -564,7 +564,7 @@ static int alc_spdif_ctrl_get(struct snd 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, 0x00); + AC_VERB_GET_DIGI_CONVERT_1, 0x00); *valp = (val & mask) != 0; return 0; @@ -578,7 +578,7 @@ static int alc_spdif_ctrl_put(struct snd 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, + AC_VERB_GET_DIGI_CONVERT_1, 0x00); /* Set/unset the masked control bit(s) as needed */

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[alsa-devel] M-audio Delta Audiophile 192 drivers
by Manel Clos 13 Dec '07

13 Dec '07

Hi all! We purchased a M-audio Delta Audiophile 192, which uses the ICE1724 chipset. Using Ubuntu gutsy 7.10 and alsa 1.0.14. PCM is working, but as soon as we start Rosegarden the complete system hangs (SysRq doesn't even work). Also, the input of the card is not shown in gnome-mixer. Is there anything we can do to help the development of the driver? We are able to compile kernel + patches, do C programming, test PCM, Midi... Thanks in advance!

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[alsa-devel] Question about pcm_formats[] array in sound/core/pcm_misc.c
by Timur Tabi 13 Dec '07

13 Dec '07

I'm confused about the entries for unsigned 24-bit sound in the pcm_formats[] array: [SNDRV_PCM_FORMAT_U24_BE] = { .width = 24, .phys = 32, .le = 0, .signd = 0, .silence = { 0x80, 0x00, 0x00 }, }, and [SNDRV_PCM_FORMAT_U24_3BE] = { .width = 24, .phys = 24, .le = 0, .signd = 0, .silence = { 0x80, 0x00, 0x00 }, }, SNDRV_PCM_FORMAT_U24_BE means a 24-bit sample packed into a 32-bit word. So I presume that if the 24-bit sample is: 0x123456 then when that sample is written to memory, it looks like this: X X + 1 X + 2 X + 3 0x00 0x12 0x34 0x56 (where X is some address in memory). If that's so, shouldn't the entry for SNDRV_PCM_FORMAT_U24_BE look like this: [SNDRV_PCM_FORMAT_U24_BE] = { .width = 24, .phys = 32, .le = 0, .signd = 0, .silence = { 0x00, 0x80, 0x00, 0x00 }, }, ^^^^^^ -- Timur Tabi Linux kernel developer at Freescale

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[alsa-devel] Sound problems in PCCHIPS A13G+
by alsa-devel.20.pris＠spamgourmet.com 11 Dec '07

11 Dec '07

My motherboard: PCCHIPS A13G+ http://www.pcchips.com.tw/PCCWebSite...uID=44&LanID=0 I'm using Ubuntu 7.10 - the Gutsy Gibbon, I was using Feisty before and had the same problems. The interesting thing: the system sounds all work. Some example .ogg files that came with Ubuntu also work. There is one example file however "fables_01_01_aesop.spx" which only gives a very low sound, almost inaudible. The real strange thing happens with ".avi" and mp3 files. They sound distorted and very low, almost inaudible. Some have better/louder sound, but still distorted, but I can understand what is being spoken, others not. Thanks for any help! If you need more information just ask. Technical details follow(including the Alsa Audio Debug): The sound hardware of my Mobo is in the NVIDIA MCP61S Single chipset. This chipset consists of: GeForce(R) 6100 GPU and MCP nForce(R) 405 codec(less /proc/asound/card0/codec#0): Codec: Realtek ALC660-VD Address: 0 Vendor Id: 0x10ec0660 Subsystem Id: 0x1019c601 Revision Id: 0x100001 The aplay command lists another codec: ALC861VD aplay -l **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: ALC861VD Analog [ALC861VD Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 Here is the Alsa Audio Debug: ALSA Audio Debug v0.1.0 - Mon Nov 12 00:00:12 BRST 2007 http://alsa.opensrc.org/aadebug http://www.gnu.org/licenses/gpl.txt Kernel ---------------------------------------------------- Linux 2.6.22-14-generic #1 SMP Sun Oct 14 23:05:12 GMT 2007 i686 GNU/Linux Loaded Modules -------------------------------------------- snd_hda_intel 263712 4 snd_pcm_oss 44672 0 snd_mixer_oss 17664 1 snd_pcm_oss snd_pcm 80388 3 snd_hda_intel,snd_pcm_oss snd_seq_dummy 4740 0 snd_seq_oss 33152 0 snd_seq_midi 9600 0 snd_rawmidi 25728 1 snd_seq_midi snd_seq_midi_event 8448 2 snd_seq_oss,snd_seq_midi snd_seq 53232 6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_timer 24324 3 snd_pcm,snd_seq snd_seq_device 9228 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd 54660 15 snd_hda_intel,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_oss,snd_rawmidi,snd_seq,snd_timer,snd_seq_device snd_page_alloc 11400 2 snd_hda_intel,snd_pcm Modprobe Conf --------------------------------------------- Warning: module config file does not exist This means any kernel modules will not be auto loaded See your linux distro docs on how to create this file Proc Asound ----------------------------------------------- Advanced Linux Sound Architecture Driver Version 1.0.14 (Thu May 31 09:03:25 2007 UTC). 0 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xfe028000 irq 16 2: : timer 3: : sequencer 4: [ 0- 0]: digital audio playback 5: [ 0- 0]: digital audio capture 6: [ 0] : control cat: /proc/asound/hwdep: No such file or directory 00-00: ALC861VD Analog : ALC861VD Analog : playback 1 : capture 2 Client info cur clients : 3 peak clients : 3 max clients : 192 Client 0 : "System" [Kernel] Port 0 : "Timer" (Rwe-) Port 1 : "Announce" (R-e-) Connecting To: 15:0 Client 14 : "Midi Through" [Kernel] Port 0 : "Midi Through Port-0" (RWe-) Client 15 : "OSS sequencer" [Kernel] Port 0 : "Receiver" (-we-) Connected From: 0:1 Dev Snd --------------------------------------------------- controlC0 pcmC0D0c pcmC0D0p seq timer CPU ------------------------------------------------------- model name : AMD Processor model unknown cpu MHz : 1900.000 RAM ------------------------------------------------------- MemTotal: 969524 kB SwapTotal: 1951888 kB Hardware -------------------------------------------------- 00:18.0 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] HyperTransport Technology Configuration 00:18.1 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Address Map 00:18.2 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM Controller 00:18.3 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Miscellaneous Control

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