From 29fdbec2dcb1ce364812778271056aa9516ff3ed Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 20 Jan 2009 13:07:55 +0100 Subject: ALSA: hda - Add extra volume offset to standard volume amp macros Added the volume offset to base for the standard volume controls to handle elements with too big volume scales like -96dB..0dB. For such elements, you can set the base volume to reduce the range. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 45 +++++++++++++++++++++++++++++++++++++-------- sound/pci/hda/hda_local.h | 5 ++++- 2 files changed, 41 insertions(+), 9 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b7bba7dc7cf1..0cf2424ada6a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1119,6 +1119,7 @@ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol, u16 nid = get_amp_nid(kcontrol); u8 chs = get_amp_channels(kcontrol); int dir = get_amp_direction(kcontrol); + unsigned int ofs = get_amp_offset(kcontrol); u32 caps; caps = query_amp_caps(codec, nid, dir); @@ -1130,6 +1131,8 @@ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol, kcontrol->id.name); return -EINVAL; } + if (ofs < caps) + caps -= ofs; uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = chs == 3 ? 2 : 1; uinfo->value.integer.min = 0; @@ -1138,6 +1141,32 @@ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_volume_info); + +static inline unsigned int +read_amp_value(struct hda_codec *codec, hda_nid_t nid, + int ch, int dir, int idx, unsigned int ofs) +{ + unsigned int val; + val = snd_hda_codec_amp_read(codec, nid, ch, dir, idx); + val &= HDA_AMP_VOLMASK; + if (val >= ofs) + val -= ofs; + else + val = 0; + return val; +} + +static inline int +update_amp_value(struct hda_codec *codec, hda_nid_t nid, + int ch, int dir, int idx, unsigned int ofs, + unsigned int val) +{ + if (val > 0) + val += ofs; + return snd_hda_codec_amp_update(codec, nid, ch, dir, idx, + HDA_AMP_VOLMASK, val); +} + int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1146,14 +1175,13 @@ int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol, int chs = get_amp_channels(kcontrol); int dir = get_amp_direction(kcontrol); int idx = get_amp_index(kcontrol); + unsigned int ofs = get_amp_offset(kcontrol); long *valp = ucontrol->value.integer.value; if (chs & 1) - *valp++ = snd_hda_codec_amp_read(codec, nid, 0, dir, idx) - & HDA_AMP_VOLMASK; + *valp++ = read_amp_value(codec, nid, 0, dir, idx, ofs); if (chs & 2) - *valp = snd_hda_codec_amp_read(codec, nid, 1, dir, idx) - & HDA_AMP_VOLMASK; + *valp = read_amp_value(codec, nid, 1, dir, idx, ofs); return 0; } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_volume_get); @@ -1166,18 +1194,17 @@ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol, int chs = get_amp_channels(kcontrol); int dir = get_amp_direction(kcontrol); int idx = get_amp_index(kcontrol); + unsigned int ofs = get_amp_offset(kcontrol); long *valp = ucontrol->value.integer.value; int change = 0; snd_hda_power_up(codec); if (chs & 1) { - change = snd_hda_codec_amp_update(codec, nid, 0, dir, idx, - 0x7f, *valp); + change = update_amp_value(codec, nid, 0, dir, idx, ofs, *valp); valp++; } if (chs & 2) - change |= snd_hda_codec_amp_update(codec, nid, 1, dir, idx, - 0x7f, *valp); + change |= update_amp_value(codec, nid, 1, dir, idx, ofs, *valp); snd_hda_power_down(codec); return change; } @@ -1189,6 +1216,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = get_amp_nid(kcontrol); int dir = get_amp_direction(kcontrol); + unsigned int ofs = get_amp_offset(kcontrol); u32 caps, val1, val2; if (size < 4 * sizeof(unsigned int)) @@ -1197,6 +1225,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, val2 = (caps & AC_AMPCAP_STEP_SIZE) >> AC_AMPCAP_STEP_SIZE_SHIFT; val2 = (val2 + 1) * 25; val1 = -((caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT); + val1 += ofs; val1 = ((int)val1) * ((int)val2); if (put_user(SNDRV_CTL_TLVT_DB_SCALE, _tlv)) return -EFAULT; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 1dd8716c387f..d53ce1f85419 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -26,8 +26,10 @@ /* * for mixer controls */ +#define HDA_COMPOSE_AMP_VAL_OFS(nid,chs,idx,dir,ofs) \ + ((nid) | ((chs)<<16) | ((dir)<<18) | ((idx)<<19) | ((ofs)<<23)) #define HDA_COMPOSE_AMP_VAL(nid,chs,idx,dir) \ - ((nid) | ((chs)<<16) | ((dir)<<18) | ((idx)<<19)) + HDA_COMPOSE_AMP_VAL_OFS(nid, chs, idx, dir, 0) /* mono volume with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ @@ -456,6 +458,7 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, #define get_amp_channels(kc) (((kc)->private_value >> 16) & 0x3) #define get_amp_direction(kc) (((kc)->private_value >> 18) & 0x1) #define get_amp_index(kc) (((kc)->private_value >> 19) & 0xf) +#define get_amp_offset(kc) (((kc)->private_value >> 23) & 0x3f) /* * CEA Short Audio Descriptor data -- cgit v1.2.3-59-g8ed1b From 7c7767ebe2fa847c91a0dd5551ca422aba359473 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 20 Jan 2009 15:28:38 +0100 Subject: ALSA: hda - Halve too large volume scales for STAC/IDT codecs STAC/IDT codecs have often too large volume scales such as -96dB, and exposing this as is results in too large scale in percentage representation. This patch adds the check of the volume scale and halves the volume range if it's too large automatically. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 41 +++++++++++++++++++++++++++++++++-------- 1 file changed, 33 insertions(+), 8 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a4d4afe6b4fc..c2d4abee3b09 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -166,6 +166,7 @@ struct sigmatel_spec { unsigned int alt_switch: 1; unsigned int hp_detect: 1; unsigned int spdif_mute: 1; + unsigned int check_volume_offset:1; /* gpio lines */ unsigned int eapd_mask; @@ -202,6 +203,8 @@ struct sigmatel_spec { hda_nid_t hp_dacs[5]; hda_nid_t speaker_dacs[5]; + int volume_offset; + /* capture */ hda_nid_t *adc_nids; unsigned int num_adcs; @@ -1297,6 +1300,8 @@ static int stac92xx_build_controls(struct hda_codec *codec) unsigned int vmaster_tlv[4]; snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0], HDA_OUTPUT, vmaster_tlv); + /* correct volume offset */ + vmaster_tlv[2] += vmaster_tlv[3] * spec->volume_offset; err = snd_hda_add_vmaster(codec, "Master Playback Volume", vmaster_tlv, slave_vols); if (err < 0) @@ -2980,14 +2985,34 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec) } /* create volume control/switch for the given prefx type */ -static int create_controls(struct sigmatel_spec *spec, const char *pfx, hda_nid_t nid, int chs) +static int create_controls(struct hda_codec *codec, const char *pfx, + hda_nid_t nid, int chs) { + struct sigmatel_spec *spec = codec->spec; char name[32]; int err; + if (!spec->check_volume_offset) { + unsigned int caps, step, nums, db_scale; + caps = query_amp_caps(codec, nid, HDA_OUTPUT); + step = (caps & AC_AMPCAP_STEP_SIZE) >> + AC_AMPCAP_STEP_SIZE_SHIFT; + step = (step + 1) * 25; /* in .01dB unit */ + nums = (caps & AC_AMPCAP_NUM_STEPS) >> + AC_AMPCAP_NUM_STEPS_SHIFT; + db_scale = nums * step; + /* if dB scale is over -64dB, and finer enough, + * let's reduce it to half + */ + if (db_scale > 6400 && nums >= 0x1f) + spec->volume_offset = nums / 2; + spec->check_volume_offset = 1; + } + sprintf(name, "%s Playback Volume", pfx); err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT)); + HDA_COMPOSE_AMP_VAL_OFS(nid, chs, 0, HDA_OUTPUT, + spec->volume_offset)); if (err < 0) return err; sprintf(name, "%s Playback Switch", pfx); @@ -3053,10 +3078,10 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, nid = spec->multiout.dac_nids[i]; if (i == 2) { /* Center/LFE */ - err = create_controls(spec, "Center", nid, 1); + err = create_controls(codec, "Center", nid, 1); if (err < 0) return err; - err = create_controls(spec, "LFE", nid, 2); + err = create_controls(codec, "LFE", nid, 2); if (err < 0) return err; @@ -3084,7 +3109,7 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, break; } } - err = create_controls(spec, name, nid, 3); + err = create_controls(codec, name, nid, 3); if (err < 0) return err; } @@ -3139,7 +3164,7 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, nid = spec->hp_dacs[i]; if (!nid) continue; - err = create_controls(spec, pfxs[nums++], nid, 3); + err = create_controls(codec, pfxs[nums++], nid, 3); if (err < 0) return err; } @@ -3153,7 +3178,7 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, nid = spec->speaker_dacs[i]; if (!nid) continue; - err = create_controls(spec, pfxs[nums++], nid, 3); + err = create_controls(codec, pfxs[nums++], nid, 3); if (err < 0) return err; } @@ -3729,7 +3754,7 @@ static int stac9200_auto_create_lfe_ctls(struct hda_codec *codec, } if (lfe_pin) { - err = create_controls(spec, "LFE", lfe_pin, 1); + err = create_controls(codec, "LFE", lfe_pin, 1); if (err < 0) return err; } -- cgit v1.2.3-59-g8ed1b From 89ce9e87083216389d2ff5740cc60f835537d8d0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 20 Jan 2009 17:15:57 +0100 Subject: ALSA: hda - Add debug prints for digital I/O pin detections Add the debug prints for digital I/O pin detections in snd_hda_parse_pin_def_config() function. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b7bba7dc7cf1..c03de0bc3999 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3499,6 +3499,8 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, cfg->hp_pins[1], cfg->hp_pins[2], cfg->hp_pins[3], cfg->hp_pins[4]); snd_printd(" mono: mono_out=0x%x\n", cfg->mono_out_pin); + if (cfg->dig_out_pin) + snd_printd(" dig-out=0x%x\n", cfg->dig_out_pin); snd_printd(" inputs: mic=0x%x, fmic=0x%x, line=0x%x, fline=0x%x," " cd=0x%x, aux=0x%x\n", cfg->input_pins[AUTO_PIN_MIC], @@ -3507,6 +3509,8 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, cfg->input_pins[AUTO_PIN_FRONT_LINE], cfg->input_pins[AUTO_PIN_CD], cfg->input_pins[AUTO_PIN_AUX]); + if (cfg->dig_out_pin) + snd_printd(" dig-in=0x%x\n", cfg->dig_in_pin); return 0; } -- cgit v1.2.3-59-g8ed1b From 1b52ae701fedf97f9984e73b6a1fe2444230871b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 20 Jan 2009 17:17:29 +0100 Subject: ALSA: hda - Detect non-SPDIF digital I/O Accept non-SPDIF digital I/O pins as the digital pins. These are usually corresponding to HDMI I/O. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c03de0bc3999..2d6f72ca0140 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3390,9 +3390,11 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, cfg->input_pins[AUTO_PIN_AUX] = nid; break; case AC_JACK_SPDIF_OUT: + case AC_JACK_DIG_OTHER_OUT: cfg->dig_out_pin = nid; break; case AC_JACK_SPDIF_IN: + case AC_JACK_DIG_OTHER_IN: cfg->dig_in_pin = nid; break; } -- cgit v1.2.3-59-g8ed1b From caa10b6e808a4d65eb0306f0006308244f2b8d79 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 20 Jan 2009 17:19:01 +0100 Subject: ALSA: hda - Improve auto-probing of STAC9872 codec Use the standard STAC/IDT auto-probing routine for non-static STAC9872 codec probing. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 58 ++++++++++++++++++++++++++++++++++-------- 1 file changed, 48 insertions(+), 10 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a4d4afe6b4fc..b6e797d1c218 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5511,24 +5511,62 @@ static struct snd_pci_quirk stac9872_cfg_tbl[] = { {} }; +static struct snd_kcontrol_new stac9872_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT), + STAC_INPUT_SOURCE(1), + { } /* end */ +}; + +static hda_nid_t stac9872_pin_nids[] = { + 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, + 0x11, 0x13, 0x14, +}; + +static hda_nid_t stac9872_adc_nids[] = { + 0x8 /*,0x6*/ +}; + +static hda_nid_t stac9872_mux_nids[] = { + 0x15 +}; + static int patch_stac9872(struct hda_codec *codec) { struct sigmatel_spec *spec; - int board_config; - board_config = snd_hda_check_board_config(codec, STAC_9872_MODELS, - stac9872_models, - stac9872_cfg_tbl); - if (board_config < 0) - /* unknown config, let generic-parser do its job... */ - return snd_hda_parse_generic_codec(codec); - spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - switch (board_config) { + + spec->board_config = snd_hda_check_board_config(codec, STAC_9872_MODELS, + stac9872_models, + stac9872_cfg_tbl); + if (spec->board_config < 0) { + int err; + + spec->num_pins = ARRAY_SIZE(stac9872_pin_nids); + spec->pin_nids = stac9872_pin_nids; + spec->multiout.dac_nids = spec->dac_nids; + spec->num_adcs = ARRAY_SIZE(stac9872_adc_nids); + spec->adc_nids = stac9872_adc_nids; + spec->num_muxes = ARRAY_SIZE(stac9872_mux_nids); + spec->mux_nids = stac9872_mux_nids; + spec->mixer = stac9872_mixer; + spec->init = vaio_init; + + err = stac92xx_parse_auto_config(codec, 0x10, 0x12); + if (err < 0) { + stac92xx_free(codec); + return -EINVAL; + } + spec->input_mux = &spec->private_imux; + codec->patch_ops = stac92xx_patch_ops; + return 0; + } + + switch (spec->board_config) { case CXD9872RD_VAIO: case STAC9872AK_VAIO: case STAC9872K_VAIO: -- cgit v1.2.3-59-g8ed1b From 41b5b01afb71226653282951965d5efa9d7b843d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 20 Jan 2009 18:21:23 +0100 Subject: ALSA: hda - Don't break the PCM creation loop Don't break the loop in snd_hda_codec_build_pcms() even if the item has no substreams. It's possible that it's an empty item and the next item containing the valid substreams (e.g. realtek codecs may create the analog and alt-analog but no digitl streams). Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 2d6f72ca0140..0129e95672ae 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2613,7 +2613,7 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec) int dev; if (!cpcm->stream[0].substreams && !cpcm->stream[1].substreams) - return 0; /* no substreams assigned */ + continue; /* no substreams assigned */ if (!cpcm->pcm) { dev = get_empty_pcm_device(codec->bus, cpcm->pcm_type); -- cgit v1.2.3-59-g8ed1b From 2297bd6e526ce1469279284ffda9140f8d60ea84 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 20 Jan 2009 18:24:13 +0100 Subject: ALSA: hda - Check HDMI jack types in the auto configuration Add dig_out_type and dig_in_type fields to autocfg struct. A proper HDA_PCM_TYPE_* value is assigned to these fields according to the pin-jack location type value. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 8 ++++++++ sound/pci/hda/hda_local.h | 2 ++ 2 files changed, 10 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0129e95672ae..dd419ce43d92 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3392,10 +3392,18 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, case AC_JACK_SPDIF_OUT: case AC_JACK_DIG_OTHER_OUT: cfg->dig_out_pin = nid; + if (loc == AC_JACK_LOC_HDMI) + cfg->dig_out_type = HDA_PCM_TYPE_HDMI; + else + cfg->dig_out_type = HDA_PCM_TYPE_SPDIF; break; case AC_JACK_SPDIF_IN: case AC_JACK_DIG_OTHER_IN: cfg->dig_in_pin = nid; + if (loc == AC_JACK_LOC_HDMI) + cfg->dig_in_type = HDA_PCM_TYPE_HDMI; + else + cfg->dig_in_type = HDA_PCM_TYPE_SPDIF; break; } } diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 1dd8716c387f..a4ecd77a451a 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -355,6 +355,8 @@ struct auto_pin_cfg { hda_nid_t dig_out_pin; hda_nid_t dig_in_pin; hda_nid_t mono_out_pin; + int dig_out_type; /* HDA_PCM_TYPE_XXX */ + int dig_in_type; /* HDA_PCM_TYPE_XXX */ }; #define get_defcfg_connect(cfg) \ -- cgit v1.2.3-59-g8ed1b From 8c441982fdc00f77b7aa609061c6411f47bcceda Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 20 Jan 2009 18:30:20 +0100 Subject: ALSA: hda - Assign proper digital I/O type for STAC/IDT Assign the proper PCM digital I/O type (HDA_PCM_TYPE_*) for the digital I/O on STAC/IDT codecs. HDA_PCM_TYPE_HDMI is assigned for the HDMI I/O. A similar framework is implemented to patch_realtek.c, but it's not set up and still using only HDA_PCM_TYPE_SPDIF yet. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 6 +++++- sound/pci/hda/patch_sigmatel.c | 2 +- 2 files changed, 6 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5d249a547fbf..4fdae06162ed 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -269,6 +269,7 @@ struct alc_spec { * dig_out_nid and hp_nid are optional */ hda_nid_t alt_dac_nid; + int dig_out_type; /* capture */ unsigned int num_adc_nids; @@ -3087,7 +3088,10 @@ static int alc_build_pcms(struct hda_codec *codec) codec->num_pcms = 2; info = spec->pcm_rec + 1; info->name = spec->stream_name_digital; - info->pcm_type = HDA_PCM_TYPE_SPDIF; + if (spec->dig_out_type) + info->pcm_type = spec->dig_out_type; + else + info->pcm_type = HDA_PCM_TYPE_SPDIF; if (spec->multiout.dig_out_nid && spec->stream_digital_playback) { info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_digital_playback); diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index b6e797d1c218..1dd448e85bc1 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2553,7 +2553,7 @@ static int stac92xx_build_pcms(struct hda_codec *codec) codec->num_pcms++; info++; info->name = "STAC92xx Digital"; - info->pcm_type = HDA_PCM_TYPE_SPDIF; + info->pcm_type = spec->autocfg.dig_out_type; if (spec->multiout.dig_out_nid) { info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_digital_playback; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid; -- cgit v1.2.3-59-g8ed1b From e64f14f4e570d6ec5bc88abac92a3a27150756d7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 20 Jan 2009 18:32:55 +0100 Subject: ALSA: hda - Allow digital-only I/O on ALC262 codec Some laptops like VAIO have multiple codecs and uses ALC262 only for the SPIDF output without analog I/O. So far, the codec-parser assumes the presence of analog I/O and returned an error for such a case. This patch adds some hacks to allow the digital-only configuration for ALC262. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 43 +++++++++++++++++++++++++++++++++---------- 1 file changed, 33 insertions(+), 10 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4fdae06162ed..4cfa78c54398 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -306,6 +306,9 @@ struct alc_spec { unsigned int jack_present: 1; unsigned int master_sw: 1; + /* other flags */ + unsigned int no_analog :1; /* digital I/O only */ + /* for virtual master */ hda_nid_t vmaster_nid; #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -2019,11 +2022,13 @@ static int alc_build_controls(struct hda_codec *codec) spec->multiout.dig_out_nid); if (err < 0) return err; - err = snd_hda_create_spdif_share_sw(codec, - &spec->multiout); - if (err < 0) - return err; - spec->multiout.share_spdif = 1; + if (!spec->no_analog) { + err = snd_hda_create_spdif_share_sw(codec, + &spec->multiout); + if (err < 0) + return err; + spec->multiout.share_spdif = 1; + } } if (spec->dig_in_nid) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); @@ -2032,7 +2037,8 @@ static int alc_build_controls(struct hda_codec *codec) } /* if we have no master control, let's create it */ - if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { + if (!spec->no_analog && + !snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { unsigned int vmaster_tlv[4]; snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, HDA_OUTPUT, vmaster_tlv); @@ -2041,7 +2047,8 @@ static int alc_build_controls(struct hda_codec *codec) if (err < 0) return err; } - if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { + 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); if (err < 0) @@ -3060,6 +3067,9 @@ static int alc_build_pcms(struct hda_codec *codec) codec->num_pcms = 1; codec->pcm_info = info; + if (spec->no_analog) + goto skip_analog; + info->name = spec->stream_name_analog; if (spec->stream_analog_playback) { if (snd_BUG_ON(!spec->multiout.dac_nids)) @@ -3083,6 +3093,7 @@ static int alc_build_pcms(struct hda_codec *codec) } } + skip_analog: /* SPDIF for stream index #1 */ if (spec->multiout.dig_out_nid || spec->dig_in_nid) { codec->num_pcms = 2; @@ -3106,6 +3117,9 @@ static int alc_build_pcms(struct hda_codec *codec) codec->spdif_status_reset = 1; } + if (spec->no_analog) + return 0; + /* If the use of more than one ADC is requested for the current * model, configure a second analog capture-only PCM. */ @@ -10468,8 +10482,14 @@ static int alc262_parse_auto_config(struct hda_codec *codec) alc262_ignore); if (err < 0) return err; - if (!spec->autocfg.line_outs) + if (!spec->autocfg.line_outs) { + if (spec->autocfg.dig_out_pin || spec->autocfg.dig_in_pin) { + spec->multiout.max_channels = 2; + spec->no_analog = 1; + goto dig_only; + } return 0; /* can't find valid BIOS pin config */ + } err = alc262_auto_create_multi_out_ctls(spec, &spec->autocfg); if (err < 0) return err; @@ -10479,8 +10499,11 @@ static int alc262_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + dig_only: + if (spec->autocfg.dig_out_pin) { spec->multiout.dig_out_nid = ALC262_DIGOUT_NID; + spec->dig_out_type = spec->autocfg.dig_out_type; + } if (spec->autocfg.dig_in_pin) spec->dig_in_nid = ALC262_DIGIN_NID; @@ -10875,7 +10898,7 @@ static int patch_alc262(struct hda_codec *codec) spec->capsrc_nids = alc262_capsrc_nids; } } - if (!spec->cap_mixer) + if (!spec->cap_mixer && !spec->no_analog) set_capture_mixer(spec); spec->vmaster_nid = 0x0c; -- cgit v1.2.3-59-g8ed1b From 1e137f929bb490ff615ea475ac3904d58b0cdd5e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 21 Jan 2009 07:41:22 +0100 Subject: ALSA: hda - Clean up old VAIO hack codes for STAC9872 Get rid of old VAIO static hack codes for STAC9872 and use the BIOS auto-parser for all models. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 238 ++++------------------------------------- 1 file changed, 21 insertions(+), 217 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 775f8581906e..dbe8b1201eff 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5351,172 +5351,12 @@ static int patch_stac9205(struct hda_codec *codec) * STAC9872 hack */ -/* static config for Sony VAIO FE550G and Sony VAIO AR */ -static hda_nid_t vaio_dacs[] = { 0x2 }; -#define VAIO_HP_DAC 0x5 -static hda_nid_t vaio_adcs[] = { 0x8 /*,0x6*/ }; -static hda_nid_t vaio_mux_nids[] = { 0x15 }; - -static struct hda_input_mux vaio_mux = { - .num_items = 3, - .items = { - /* { "HP", 0x0 }, */ - { "Mic Jack", 0x1 }, - { "Internal Mic", 0x2 }, - { "PCM", 0x3 }, - } -}; - -static struct hda_verb vaio_init[] = { - {0x0a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP <- 0x2 */ - {0x0a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | STAC_HP_EVENT}, - {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Speaker <- 0x5 */ - {0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? (<- 0x2) */ - {0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mic-sel: 0a,0d,14,02 */ - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */ - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */ - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* capture sw/vol -> 0x8 */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* CD-in -> 0x6 */ - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Mic-in -> 0x9 */ - {} -}; - -static struct hda_verb vaio_ar_init[] = { - {0x0a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP <- 0x2 */ - {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Speaker <- 0x5 */ - {0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? (<- 0x2) */ - {0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */ -/* {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },*/ /* Optical Out */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */ +static struct hda_verb stac9872_core_init[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mic-sel: 0a,0d,14,02 */ - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */ - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */ -/* {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},*/ /* Optical Out */ - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* capture sw/vol -> 0x8 */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* CD-in -> 0x6 */ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Mic-in -> 0x9 */ {} }; -static struct snd_kcontrol_new vaio_mixer[] = { - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x02, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x02, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x05, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x05, 0, HDA_OUTPUT), - /* HDA_CODEC_VOLUME("CD Capture Volume", 0x07, 0, HDA_INPUT), */ - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .count = 1, - .info = stac92xx_mux_enum_info, - .get = stac92xx_mux_enum_get, - .put = stac92xx_mux_enum_put, - }, - {} -}; - -static struct snd_kcontrol_new vaio_ar_mixer[] = { - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x02, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x02, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x05, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x05, 0, HDA_OUTPUT), - /* HDA_CODEC_VOLUME("CD Capture Volume", 0x07, 0, HDA_INPUT), */ - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT), - /*HDA_CODEC_MUTE("Optical Out Switch", 0x10, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Optical Out Volume", 0x10, 0, HDA_OUTPUT),*/ - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .count = 1, - .info = stac92xx_mux_enum_info, - .get = stac92xx_mux_enum_get, - .put = stac92xx_mux_enum_put, - }, - {} -}; - -static struct hda_codec_ops stac9872_patch_ops = { - .build_controls = stac92xx_build_controls, - .build_pcms = stac92xx_build_pcms, - .init = stac92xx_init, - .free = stac92xx_free, -#ifdef SND_HDA_NEEDS_RESUME - .resume = stac92xx_resume, -#endif -}; - -static int stac9872_vaio_init(struct hda_codec *codec) -{ - int err; - - err = stac92xx_init(codec); - if (err < 0) - return err; - if (codec->patch_ops.unsol_event) - codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); - return 0; -} - -static void stac9872_vaio_hp_detect(struct hda_codec *codec, unsigned int res) -{ - if (get_pin_presence(codec, 0x0a)) { - stac92xx_reset_pinctl(codec, 0x0f, AC_PINCTL_OUT_EN); - stac92xx_set_pinctl(codec, 0x0a, AC_PINCTL_OUT_EN); - } else { - stac92xx_reset_pinctl(codec, 0x0a, AC_PINCTL_OUT_EN); - stac92xx_set_pinctl(codec, 0x0f, AC_PINCTL_OUT_EN); - } -} - -static void stac9872_vaio_unsol_event(struct hda_codec *codec, unsigned int res) -{ - switch (res >> 26) { - case STAC_HP_EVENT: - stac9872_vaio_hp_detect(codec, res); - break; - } -} - -static struct hda_codec_ops stac9872_vaio_patch_ops = { - .build_controls = stac92xx_build_controls, - .build_pcms = stac92xx_build_pcms, - .init = stac9872_vaio_init, - .free = stac92xx_free, - .unsol_event = stac9872_vaio_unsol_event, -#ifdef CONFIG_PM - .resume = stac92xx_resume, -#endif -}; - -enum { /* FE and SZ series. id=0x83847661 and subsys=0x104D0700 or 104D1000. */ - CXD9872RD_VAIO, - /* Unknown. id=0x83847662 and subsys=0x104D1200 or 104D1000. */ - STAC9872AK_VAIO, - /* Unknown. id=0x83847661 and subsys=0x104D1200. */ - STAC9872K_VAIO, - /* AR Series. id=0x83847664 and subsys=104D1300 */ - CXD9872AKD_VAIO, - STAC_9872_MODELS, -}; - -static const char *stac9872_models[STAC_9872_MODELS] = { - [CXD9872RD_VAIO] = "vaio", - [CXD9872AKD_VAIO] = "vaio-ar", -}; - -static struct snd_pci_quirk stac9872_cfg_tbl[] = { - SND_PCI_QUIRK(0x104d, 0x81e6, "Sony VAIO F/S", CXD9872RD_VAIO), - SND_PCI_QUIRK(0x104d, 0x81ef, "Sony VAIO F/S", CXD9872RD_VAIO), - SND_PCI_QUIRK(0x104d, 0x81fd, "Sony VAIO AR", CXD9872AKD_VAIO), - SND_PCI_QUIRK(0x104d, 0x8205, "Sony VAIO AR", CXD9872AKD_VAIO), - {} -}; - static struct snd_kcontrol_new stac9872_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT), @@ -5540,72 +5380,36 @@ static hda_nid_t stac9872_mux_nids[] = { static int patch_stac9872(struct hda_codec *codec) { struct sigmatel_spec *spec; + int err; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) return -ENOMEM; codec->spec = spec; +#if 0 /* no model right now */ spec->board_config = snd_hda_check_board_config(codec, STAC_9872_MODELS, stac9872_models, stac9872_cfg_tbl); - if (spec->board_config < 0) { - int err; - - spec->num_pins = ARRAY_SIZE(stac9872_pin_nids); - spec->pin_nids = stac9872_pin_nids; - spec->multiout.dac_nids = spec->dac_nids; - spec->num_adcs = ARRAY_SIZE(stac9872_adc_nids); - spec->adc_nids = stac9872_adc_nids; - spec->num_muxes = ARRAY_SIZE(stac9872_mux_nids); - spec->mux_nids = stac9872_mux_nids; - spec->mixer = stac9872_mixer; - spec->init = vaio_init; - - err = stac92xx_parse_auto_config(codec, 0x10, 0x12); - if (err < 0) { - stac92xx_free(codec); - return -EINVAL; - } - spec->input_mux = &spec->private_imux; - codec->patch_ops = stac92xx_patch_ops; - return 0; - } - - switch (spec->board_config) { - case CXD9872RD_VAIO: - case STAC9872AK_VAIO: - case STAC9872K_VAIO: - spec->mixer = vaio_mixer; - spec->init = vaio_init; - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = ARRAY_SIZE(vaio_dacs); - spec->multiout.dac_nids = vaio_dacs; - spec->multiout.hp_nid = VAIO_HP_DAC; - spec->num_adcs = ARRAY_SIZE(vaio_adcs); - spec->adc_nids = vaio_adcs; - spec->num_pwrs = 0; - spec->input_mux = &vaio_mux; - spec->mux_nids = vaio_mux_nids; - codec->patch_ops = stac9872_vaio_patch_ops; - break; - - case CXD9872AKD_VAIO: - spec->mixer = vaio_ar_mixer; - spec->init = vaio_ar_init; - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = ARRAY_SIZE(vaio_dacs); - spec->multiout.dac_nids = vaio_dacs; - spec->multiout.hp_nid = VAIO_HP_DAC; - spec->num_adcs = ARRAY_SIZE(vaio_adcs); - spec->num_pwrs = 0; - spec->adc_nids = vaio_adcs; - spec->input_mux = &vaio_mux; - spec->mux_nids = vaio_mux_nids; - codec->patch_ops = stac9872_patch_ops; - break; - } +#endif + spec->num_pins = ARRAY_SIZE(stac9872_pin_nids); + spec->pin_nids = stac9872_pin_nids; + spec->multiout.dac_nids = spec->dac_nids; + spec->num_adcs = ARRAY_SIZE(stac9872_adc_nids); + spec->adc_nids = stac9872_adc_nids; + spec->num_muxes = ARRAY_SIZE(stac9872_mux_nids); + spec->mux_nids = stac9872_mux_nids; + spec->mixer = stac9872_mixer; + spec->init = stac9872_core_init; + + err = stac92xx_parse_auto_config(codec, 0x10, 0x12); + if (err < 0) { + stac92xx_free(codec); + return -EINVAL; + } + spec->input_mux = &spec->private_imux; + codec->patch_ops = stac92xx_patch_ops; return 0; } -- cgit v1.2.3-59-g8ed1b From 8ce8419829998c91b33200894a0db5e1441d6952 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 22 Jan 2009 16:59:20 +0100 Subject: ALSA: hda - Avoid to set the pin control again if already set Check the present pin control bit and avoid the write if it's already set in patch_sigmatel.c. This will reduce the number of verb execs at jack plugging. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 20 +++++++++++++------- 1 file changed, 13 insertions(+), 7 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 0fa6c593d1d3..11634a4478ea 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4126,7 +4126,9 @@ static void stac92xx_free(struct hda_codec *codec) static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid, unsigned int flag) { - unsigned int pin_ctl = snd_hda_codec_read(codec, nid, + unsigned int old_ctl, pin_ctl; + + pin_ctl = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0x00); if (pin_ctl & AC_PINCTL_IN_EN) { @@ -4140,14 +4142,17 @@ static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid, return; } + old_ctl = pin_ctl; /* if setting pin direction bits, clear the current direction bits first */ if (flag & (AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN)) pin_ctl &= ~(AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN); - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - pin_ctl | flag); + pin_ctl |= flag; + if (old_ctl != pin_ctl) + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + pin_ctl); } static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid, @@ -4155,9 +4160,10 @@ static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid, { unsigned int pin_ctl = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0x00); - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - pin_ctl & ~flag); + if (pin_ctl & flag) + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + pin_ctl & ~flag); } static int get_pin_presence(struct hda_codec *codec, hda_nid_t nid) -- cgit v1.2.3-59-g8ed1b From d9a4268ee92ba1a2355c892a3add1fa66856b510 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 22 Jan 2009 17:40:18 +0100 Subject: ALSA: hda - Add quirk for Gateway %1616 laptop Gateway T1616 laptop needs EAPD always on while the current STAC9205 code turns off per HP plug. Added a new model "eapd" to keep it on. Reference: Novell bnc#467597 https://bugzilla.novell.com/show_bug.cgi?id=467597 Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_sigmatel.c | 10 +++++++++- 2 files changed, 10 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 75914bcdce72..ef6b22e25412 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -285,6 +285,7 @@ STAC9205/9254 dell-m42 Dell (unknown) dell-m43 Dell Precision dell-m44 Dell Inspiron + eapd Keep EAPD on (e.g. Gateway T1616) STAC9220/9221 ============= diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3f85731055c0..ed2fa431b03f 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -66,6 +66,7 @@ enum { STAC_9205_DELL_M42, STAC_9205_DELL_M43, STAC_9205_DELL_M44, + STAC_9205_EAPD, STAC_9205_MODELS }; @@ -2240,6 +2241,7 @@ static unsigned int *stac9205_brd_tbl[STAC_9205_MODELS] = { [STAC_9205_DELL_M42] = dell_9205_m42_pin_configs, [STAC_9205_DELL_M43] = dell_9205_m43_pin_configs, [STAC_9205_DELL_M44] = dell_9205_m44_pin_configs, + [STAC_9205_EAPD] = NULL, }; static const char *stac9205_models[STAC_9205_MODELS] = { @@ -2247,12 +2249,14 @@ static const char *stac9205_models[STAC_9205_MODELS] = { [STAC_9205_DELL_M42] = "dell-m42", [STAC_9205_DELL_M43] = "dell-m43", [STAC_9205_DELL_M44] = "dell-m44", + [STAC_9205_EAPD] = "eapd", }; static struct snd_pci_quirk stac9205_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_9205_REF), + /* Dell */ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f1, "unknown Dell", STAC_9205_DELL_M42), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f2, @@ -2283,6 +2287,8 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = { "Dell Inspiron", STAC_9205_DELL_M44), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0228, "Dell Vostro 1500", STAC_9205_DELL_M42), + /* Gateway */ + SND_PCI_QUIRK(0x107b, 0x0565, "Gateway T1616", STAC_9205_EAPD), {} /* terminator */ }; @@ -5320,7 +5326,9 @@ static int patch_stac9205(struct hda_codec *codec) spec->aloopback_mask = 0x40; spec->aloopback_shift = 0; - spec->eapd_switch = 1; + /* Turn on/off EAPD per HP plugging */ + if (spec->board_config != STAC_9205_EAPD) + spec->eapd_switch = 1; spec->multiout.dac_nids = spec->dac_nids; switch (spec->board_config){ -- cgit v1.2.3-59-g8ed1b From 577aa2c195045599275b54356969ae19f34e7a66 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Thu, 22 Jan 2009 22:55:44 -0500 Subject: ALSA: hda: add reference board SND_PCI_QUIRK Add another LanParty reference board SND_PCI_QUIRK to quirk lists of all codec families. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 212d8c09a67b..3fbe22053b3f 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1517,6 +1517,8 @@ static struct snd_pci_quirk stac9200_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_REF), /* Dell laptops have BIOS problem */ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01a8, "unknown Dell", STAC_9200_DELL_D21), @@ -1666,6 +1668,7 @@ static struct snd_pci_quirk stac925x_codec_id_cfg_tbl[] = { static struct snd_pci_quirk stac925x_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_REF), SND_PCI_QUIRK(0x8384, 0x7632, "Stac9202 Reference Board", STAC_REF), /* Default table for unknown ID */ @@ -1709,6 +1712,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_92HD73XX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_92HD73XX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0254, "Dell Studio 1535", STAC_DELL_M6_DMIC), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0255, @@ -1753,6 +1758,8 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_92HD83XXX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_92HD83XXX_REF), {} /* terminator */ }; @@ -1802,6 +1809,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_92HD71BXX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_92HD71BXX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f2, "HP dv5", STAC_HP_M4), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f4, @@ -1992,6 +2001,8 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_D945_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_D945_REF), /* Intel 945G based systems */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0101, "Intel D945G", STAC_D945GTP3), @@ -2148,6 +2159,8 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_D965_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_D965_REF), /* Intel 946 based systems */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x3d01, "Intel D946", STAC_D965_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0xa301, "Intel D946", STAC_D965_3ST), @@ -2259,6 +2272,8 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_9205_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_9205_REF), /* Dell */ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f1, "unknown Dell", STAC_9205_DELL_M42), -- cgit v1.2.3-59-g8ed1b From 8056d47e77a0f7e3c99c114deab4859d31496075 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 23 Jan 2009 09:09:43 +0100 Subject: ALSA: hda - Add model=ref for Intel board with STAC9221 An intel board (8086:0204) works only with model=ref. Reference: Novell bug #406529 https://bugzilla.novell.com/show_bug.cgi?id=406529 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3fbe22053b3f..4ee9f7fc7724 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2056,6 +2056,9 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = { "Intel D945P", STAC_D945GTP3), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0707, "Intel D945P", STAC_D945GTP5), + /* other intel */ + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0204, + "Intel D945", STAC_D945_REF), /* other systems */ /* Apple Intel Mac (Mac Mini, MacBook, MacBook Pro...) */ SND_PCI_QUIRK(0x8384, 0x7680, -- cgit v1.2.3-59-g8ed1b From e3c75964666a27cec46d2cccf2d9806336becd48 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 23 Jan 2009 11:57:22 +0100 Subject: ALSA: hda - Create "Input Source" control dynamically for STAC/IDT Instead of fixed kcontrol_new element, build "Input Source" controls dynamically. If the number of input-source items is 0 or 1, we don't need to create such a control. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 78 ++++++++++++++++++++++++++++-------------- 1 file changed, 53 insertions(+), 25 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index b3c3a02a4222..80a4c288b319 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -958,16 +958,6 @@ static struct hda_verb stac9205_core_init[] = { .private_value = HDA_COMPOSE_AMP_VAL(nid, chs, idx, dir) \ } -#define STAC_INPUT_SOURCE(cnt) \ - { \ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = "Input Source", \ - .count = cnt, \ - .info = stac92xx_mux_enum_info, \ - .get = stac92xx_mux_enum_get, \ - .put = stac92xx_mux_enum_put, \ - } - #define STAC_ANALOG_LOOPBACK(verb_read, verb_write, cnt) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ @@ -982,7 +972,6 @@ static struct hda_verb stac9205_core_init[] = { static struct snd_kcontrol_new stac9200_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0xb, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT), - STAC_INPUT_SOURCE(1), HDA_CODEC_VOLUME("Capture Volume", 0x0a, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0a, 0, HDA_OUTPUT), { } /* end */ @@ -1098,7 +1087,6 @@ static struct snd_kcontrol_new stac92hd83xxx_mixer[] = { }; static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { - STAC_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -1127,7 +1115,6 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { }; static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { - STAC_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -1141,14 +1128,12 @@ static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { static struct snd_kcontrol_new stac925x_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x0e, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0x0e, 0, HDA_OUTPUT), - STAC_INPUT_SOURCE(1), HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x14, 0, HDA_OUTPUT), { } /* end */ }; static struct snd_kcontrol_new stac9205_mixer[] = { - STAC_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0, 1), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1b, 0x0, HDA_INPUT), @@ -1161,7 +1146,6 @@ static struct snd_kcontrol_new stac9205_mixer[] = { /* This needs to be generated dynamically based on sequence */ static struct snd_kcontrol_new stac922x_mixer[] = { - STAC_INPUT_SOURCE(2), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x17, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x17, 0x0, HDA_INPUT), @@ -1172,7 +1156,6 @@ static struct snd_kcontrol_new stac922x_mixer[] = { static struct snd_kcontrol_new stac927x_mixer[] = { - STAC_INPUT_SOURCE(3), STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB, 1), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x18, 0x0, HDA_INPUT), @@ -2777,22 +2760,37 @@ static struct snd_kcontrol_new stac92xx_control_templates[] = { }; /* add dynamic controls */ -static int stac92xx_add_control_temp(struct sigmatel_spec *spec, - struct snd_kcontrol_new *ktemp, - int idx, const char *name, - unsigned long val) +static struct snd_kcontrol_new * +stac_control_new(struct sigmatel_spec *spec, + struct snd_kcontrol_new *ktemp, + const char *name) { struct snd_kcontrol_new *knew; snd_array_init(&spec->kctls, sizeof(*knew), 32); knew = snd_array_new(&spec->kctls); if (!knew) - return -ENOMEM; + return NULL; *knew = *ktemp; - knew->index = idx; knew->name = kstrdup(name, GFP_KERNEL); - if (!knew->name) + if (!knew->name) { + /* roolback */ + memset(knew, 0, sizeof(*knew)); + spec->kctls.alloced--; + return NULL; + } + return knew; +} + +static int stac92xx_add_control_temp(struct sigmatel_spec *spec, + struct snd_kcontrol_new *ktemp, + int idx, const char *name, + unsigned long val) +{ + struct snd_kcontrol_new *knew = stac_control_new(spec, ktemp, name); + if (!knew) return -ENOMEM; + knew->index = idx; knew->private_value = val; return 0; } @@ -2814,6 +2812,29 @@ static inline int stac92xx_add_control(struct sigmatel_spec *spec, int type, return stac92xx_add_control_idx(spec, type, 0, name, val); } +static struct snd_kcontrol_new stac_input_src_temp = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input Source", + .info = stac92xx_mux_enum_info, + .get = stac92xx_mux_enum_get, + .put = stac92xx_mux_enum_put, +}; + +static int stac92xx_add_input_source(struct sigmatel_spec *spec) +{ + struct snd_kcontrol_new *knew; + struct hda_input_mux *imux = &spec->private_imux; + + if (!spec->num_adcs || imux->num_items <= 1) + return 0; /* no need for input source control */ + knew = stac_control_new(spec, &stac_input_src_temp, + stac_input_src_temp.name); + if (!knew) + return -ENOMEM; + knew->count = spec->num_adcs; + return 0; +} + /* check whether the line-input can be used as line-out */ static hda_nid_t check_line_out_switch(struct hda_codec *codec) { @@ -3699,6 +3720,10 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out return err; } + err = stac92xx_add_input_source(spec); + if (err < 0) + return err; + spec->multiout.max_channels = spec->multiout.num_dacs * 2; if (spec->multiout.max_channels > 2) spec->surr_switch = 1; @@ -3812,6 +3837,10 @@ static int stac9200_parse_auto_config(struct hda_codec *codec) return err; } + err = stac92xx_add_input_source(spec); + if (err < 0) + return err; + if (spec->autocfg.dig_out_pin) spec->multiout.dig_out_nid = 0x05; if (spec->autocfg.dig_in_pin) @@ -5426,7 +5455,6 @@ static struct hda_verb stac9872_core_init[] = { static struct snd_kcontrol_new stac9872_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT), - STAC_INPUT_SOURCE(1), { } /* end */ }; -- cgit v1.2.3-59-g8ed1b From 6d6e17de4f64131e9c976fd524d73aaec268178f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 23 Jan 2009 12:33:54 +0100 Subject: ALSA: hda - Fix initial verbs for mic-boosts on AD1981HD The mic boosts (NID 0x08 and 0x18) are input-amps, not output-amps. Fix the initial verbs for them. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 2e7371ec2e23..9a902c2f05a2 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1407,8 +1407,8 @@ static struct hda_verb ad1981_init_verbs[] = { {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, /* Mic boost: 0dB */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* Record selector: Front mic */ {0x15, AC_VERB_SET_CONNECT_SEL, 0x0}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, -- cgit v1.2.3-59-g8ed1b From 19a2d3e9b99ffa264adf1138bd8d8aef8909dca9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 23 Jan 2009 12:35:25 +0100 Subject: ALSA: hda - Remove invalid amp initializations for AD1988* codecs The ADC widgets on AD1988* codecs have no amp controls. Remove invalid initialization verbs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 12 ------------ 1 file changed, 12 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 9a902c2f05a2..52bc85dd6f54 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2288,10 +2288,6 @@ static struct hda_verb ad1988_capture_init_verbs[] = { {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1}, {0x0d, AC_VERB_SET_CONNECT_SEL, 0x1}, {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* ADCs; muted */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, { } }; @@ -2399,10 +2395,6 @@ static struct hda_verb ad1988_3stack_init_verbs[] = { {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1}, {0x0d, AC_VERB_SET_CONNECT_SEL, 0x1}, {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* ADCs; muted */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Analog Mix output amp */ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ { } @@ -2474,10 +2466,6 @@ static struct hda_verb ad1988_laptop_init_verbs[] = { {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1}, {0x0d, AC_VERB_SET_CONNECT_SEL, 0x1}, {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* ADCs; muted */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Analog Mix output amp */ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ { } -- cgit v1.2.3-59-g8ed1b From 60e388e89c9e258a51a0995ddd9e18fdebcdbe12 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 23 Jan 2009 12:37:09 +0100 Subject: ALSA: hda - Fix invalid verbs for mic-boosts on AD1884* The mic-boosts (0x14 and 0x15) on AD1884* codecs are input-amps, not output-amps. Fix the invalid initialization verbs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 52bc85dd6f54..a7298d28a0d4 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3183,10 +3183,10 @@ static struct hda_verb ad1884_init_verbs[] = { {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1}, /* Port-B (front mic) pin */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* Port-C (rear mic) pin */ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* Analog mixer; mute as default */ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, @@ -3601,10 +3601,10 @@ static struct hda_verb ad1884a_init_verbs[] = { {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Port-B (front mic) pin */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* Port-C (rear line-in) pin */ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* Port-E (rear mic) pin */ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, -- cgit v1.2.3-59-g8ed1b From 4cfb91c6d764b18e81bfb6e6779e07bcecbb197c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 23 Jan 2009 12:53:09 +0100 Subject: ALSA: hda - Fix invalid amp init for ALC268 codec Fix some invalid AMP initializations for ALC268 codecs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 18 ++---------------- 1 file changed, 2 insertions(+), 16 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4cfa78c54398..863ab957204b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11279,19 +11279,13 @@ static void alc267_quanta_il1_unsol_event(struct hda_codec *codec, static struct hda_verb alc268_base_init_verbs[] = { /* Unmute DAC0-1 and set vol = 0 */ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, /* * Set up output mixers (0x0c - 0x0e) */ /* set vol=0 to output mixers */ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x0e, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -11310,9 +11304,7 @@ static struct hda_verb alc268_base_init_verbs[] = { {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* set PCBEEP vol = 0, mute connections */ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -11334,10 +11326,8 @@ static struct hda_verb alc268_base_init_verbs[] = { */ static struct hda_verb alc268_volume_init_verbs[] = { /* set output DAC */ - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, @@ -11345,16 +11335,12 @@ static struct hda_verb alc268_volume_init_verbs[] = { {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* set PCBEEP vol = 0, mute connections */ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, -- cgit v1.2.3-59-g8ed1b From 70040c07402ef5a3fad2133daffb7ee61b0d4641 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 23 Jan 2009 14:18:11 +0100 Subject: ALSA: hda - Fix wrong initial verb for AD1984 thinkpad model The docking mic-boost (0x25) has no mute bit. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index a7298d28a0d4..e934e2c187d0 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3337,7 +3337,7 @@ static struct hda_verb ad1984_thinkpad_init_verbs[] = { {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* docking mic boost */ - {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* Analog mixer - docking mic; mute as default */ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* enable EAPD bit */ -- cgit v1.2.3-59-g8ed1b From ca8d33fc9fafe373362d35107f01fba1e73fb966 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Mon, 26 Jan 2009 09:33:52 -0500 Subject: ALSA: hda: 92hd71xxx disable unmute support for codecs that don't have input amps Some revisions of the 92hd71xxx codec families don't have input amps on ports 0xa, 0xd and 0xf, so probe the widget caps on port 0xa and check for support, if found run snd_hda_sequence_write_cache() on the stac92hd71xxx_unmute_core_init verb list. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 20 ++++++++++++-------- 1 file changed, 12 insertions(+), 8 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 80a4c288b319..03b26426611b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -858,26 +858,25 @@ static struct hda_verb stac92hd83xxx_core_init[] = { static struct hda_verb stac92hd71bxx_core_init[] = { /* set master volume and direct control */ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ - { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {} }; -#define HD_DISABLE_PORTF 2 +#define HD_DISABLE_PORTF 1 static struct hda_verb stac92hd71bxx_analog_core_init[] = { /* start of config #1 */ /* connect port 0f to audio mixer */ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, - /* unmute right and left channels for node 0x0f */ - { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* start of config #2 */ /* set master volume and direct control */ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* unmute right and left channels for nodes 0x0a, 0xd */ + {} +}; + +static struct hda_verb stac92hd71bxx_unmute_core_init[] = { + /* unmute right and left channels for nodes 0x0f, 0xa, 0x0d */ + { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {} @@ -4942,6 +4941,7 @@ static struct hda_input_mux stac92hd71bxx_dmux = { static int patch_stac92hd71bxx(struct hda_codec *codec) { struct sigmatel_spec *spec; + struct hda_verb *unmute_init = stac92hd71bxx_unmute_core_init; int err = 0; spec = kzalloc(sizeof(*spec), GFP_KERNEL); @@ -5015,6 +5015,7 @@ again: /* disable VSW */ spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF]; + unmute_init++; stac_change_pin_config(codec, 0xf, 0x40f000f0); break; case 0x111d7603: /* 6 Port with Analog Mixer */ @@ -5031,6 +5032,9 @@ again: codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; } + if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP) + snd_hda_sequence_write_cache(codec, unmute_init); + spec->aloopback_mask = 0x50; spec->aloopback_shift = 0; -- cgit v1.2.3-59-g8ed1b From 61b9b9b109217b2bfb128c3ca24d8f8c839a425f Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Wed, 28 Jan 2009 09:16:33 -0200 Subject: ALSA: hda - Consider additional capture source/selector in ALC889 Currently code for capture source support in ALC889 only considers capture mixers. This change adds additional support for ADC+selector present in ALC889, taking into account also the presence of an additional DMIC connection item in the selector. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 105 +++++++++++++++++++++++++++++++----------- 1 file changed, 77 insertions(+), 28 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 863ab957204b..d81cb5eb8c5f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -238,6 +238,13 @@ enum { ALC883_MODEL_LAST, }; +/* styles of capture selection */ +enum { + CAPT_MUX = 0, /* only mux based */ + CAPT_MIX, /* only mixer based */ + CAPT_1MUX_MIX, /* first mux and other mixers */ +}; + /* for GPIO Poll */ #define GPIO_MASK 0x03 @@ -276,7 +283,7 @@ struct alc_spec { hda_nid_t *adc_nids; hda_nid_t *capsrc_nids; hda_nid_t dig_in_nid; /* digital-in NID; optional */ - unsigned char is_mix_capture; /* matrix-style capture (non-mux) */ + int capture_style; /* capture style (CAPT_*) */ /* capture source */ unsigned int num_mux_defs; @@ -294,7 +301,7 @@ struct alc_spec { /* dynamic controls, init_verbs and input_mux */ struct auto_pin_cfg autocfg; struct snd_array kctls; - struct hda_input_mux private_imux; + struct hda_input_mux private_imux[3]; hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; /* hooks */ @@ -396,7 +403,8 @@ static int alc_mux_enum_put(struct snd_kcontrol *kcontrol, mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx; imux = &spec->input_mux[mux_idx]; - if (spec->is_mix_capture) { + if (spec->capture_style && + !(spec->capture_style == CAPT_1MUX_MIX && !adc_idx)) { /* Matrix-mixer style (e.g. ALC882) */ unsigned int *cur_val = &spec->cur_mux[adc_idx]; unsigned int i, idx; @@ -4130,7 +4138,7 @@ static int new_analog_input(struct alc_spec *spec, hda_nid_t pin, static int alc880_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, err, idx; for (i = 0; i < AUTO_PIN_LAST; i++) { @@ -4279,7 +4287,7 @@ static int alc880_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc880_volume_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; store_pin_configs(codec); return 1; @@ -5487,7 +5495,7 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec, static int alc260_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, err, idx; for (i = 0; i < AUTO_PIN_LAST; i++) { @@ -5647,7 +5655,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc260_volume_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; store_pin_configs(codec); return 1; @@ -7087,7 +7095,7 @@ static int patch_alc882(struct hda_codec *codec) spec->stream_digital_playback = &alc882_pcm_digital_playback; spec->stream_digital_capture = &alc882_pcm_digital_capture; - spec->is_mix_capture = 1; /* matrix-style capture */ + spec->capture_style = CAPT_MIX; /* matrix-style capture */ if (!spec->adc_nids && spec->input_mux) { /* check whether NID 0x07 is valid */ unsigned int wcap = get_wcaps(codec, 0x07); @@ -7155,10 +7163,14 @@ static hda_nid_t alc883_adc_nids_rev[2] = { 0x09, 0x08 }; +#define alc889_adc_nids alc880_adc_nids + static hda_nid_t alc883_capsrc_nids[2] = { 0x23, 0x22 }; static hda_nid_t alc883_capsrc_nids_rev[2] = { 0x22, 0x23 }; +#define alc889_capsrc_nids alc882_capsrc_nids + /* input MUX */ /* FIXME: should be a matrix-type input source selection */ @@ -8977,6 +8989,8 @@ static int alc883_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; int err = alc880_parse_auto_config(codec); + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; if (err < 0) return err; @@ -8990,6 +9004,26 @@ static int alc883_parse_auto_config(struct hda_codec *codec) /* hack - override the init verbs */ spec->init_verbs[0] = alc883_auto_init_verbs; + /* setup input_mux for ALC889 */ + if (codec->vendor_id == 0x10ec0889) { + /* digital-mic input pin is excluded in alc880_auto_create..() + * because it's under 0x18 + */ + if (cfg->input_pins[AUTO_PIN_MIC] == 0x12 || + cfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) { + struct hda_input_mux *imux = &spec->private_imux[0]; + for (i = 1; i < 3; i++) + memcpy(&spec->private_imux[i], + &spec->private_imux[0], + sizeof(spec->private_imux[0])); + imux->items[imux->num_items].label = "Int DMic"; + imux->items[imux->num_items].index = 0x0b; + imux->num_items++; + spec->num_mux_defs = 3; + spec->input_mux = spec->private_imux; + } + } + return 1; /* config found */ } @@ -9053,14 +9087,36 @@ static int patch_alc883(struct hda_codec *codec) spec->stream_name_analog = "ALC888 Analog"; spec->stream_name_digital = "ALC888 Digital"; } + if (!spec->num_adc_nids) { + spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); + spec->adc_nids = alc883_adc_nids; + } + if (!spec->capsrc_nids) + spec->capsrc_nids = alc883_capsrc_nids; + spec->capture_style = CAPT_MIX; /* matrix-style capture */ break; case 0x10ec0889: spec->stream_name_analog = "ALC889 Analog"; spec->stream_name_digital = "ALC889 Digital"; + if (!spec->num_adc_nids) { + spec->num_adc_nids = ARRAY_SIZE(alc889_adc_nids); + spec->adc_nids = alc889_adc_nids; + } + if (!spec->capsrc_nids) + spec->capsrc_nids = alc889_capsrc_nids; + spec->capture_style = CAPT_1MUX_MIX; /* 1mux/Nmix-style + capture */ break; default: spec->stream_name_analog = "ALC883 Analog"; spec->stream_name_digital = "ALC883 Digital"; + if (!spec->num_adc_nids) { + spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); + spec->adc_nids = alc883_adc_nids; + } + if (!spec->capsrc_nids) + spec->capsrc_nids = alc883_capsrc_nids; + spec->capture_style = CAPT_MIX; /* matrix-style capture */ break; } @@ -9071,13 +9127,6 @@ static int patch_alc883(struct hda_codec *codec) spec->stream_digital_playback = &alc883_pcm_digital_playback; spec->stream_digital_capture = &alc883_pcm_digital_capture; - if (!spec->num_adc_nids) { - spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); - spec->adc_nids = alc883_adc_nids; - } - if (!spec->capsrc_nids) - spec->capsrc_nids = alc883_capsrc_nids; - spec->is_mix_capture = 1; /* matrix-style capture */ if (!spec->cap_mixer) set_capture_mixer(spec); @@ -10512,7 +10561,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc262_volume_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; err = alc_auto_add_mic_boost(codec); if (err < 0) @@ -10881,7 +10930,7 @@ static int patch_alc262(struct hda_codec *codec) spec->stream_digital_playback = &alc262_pcm_digital_playback; spec->stream_digital_capture = &alc262_pcm_digital_capture; - spec->is_mix_capture = 1; + spec->capture_style = CAPT_MIX; if (!spec->adc_nids && spec->input_mux) { /* check whether NID 0x07 is valid */ unsigned int wcap = get_wcaps(codec, 0x07); @@ -11539,7 +11588,7 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec, static int alc268_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, idx1; for (i = 0; i < AUTO_PIN_LAST; i++) { @@ -11657,7 +11706,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc268_volume_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; err = alc_auto_add_mic_boost(codec); if (err < 0) @@ -12511,7 +12560,7 @@ static int alc269_auto_create_analog_input_ctls(struct alc_spec *spec, */ if (cfg->input_pins[AUTO_PIN_MIC] == 0x12 || cfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; imux->items[imux->num_items].label = "Int Mic"; imux->items[imux->num_items].index = 0x05; imux->num_items++; @@ -12567,7 +12616,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc269_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; /* set default input source */ snd_hda_codec_write_cache(codec, alc269_capsrc_nids[0], 0, AC_VERB_SET_CONNECT_SEL, @@ -13483,7 +13532,7 @@ static int alc861_auto_create_hp_ctls(struct alc_spec *spec, hda_nid_t pin) static int alc861_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, err, idx, idx1; for (i = 0; i < AUTO_PIN_LAST; i++) { @@ -13620,7 +13669,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc861_auto_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; spec->adc_nids = alc861_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc861_adc_nids); @@ -14724,7 +14773,7 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc861vd_volume_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; err = alc_auto_add_mic_boost(codec); if (err < 0) @@ -14803,7 +14852,7 @@ static int patch_alc861vd(struct hda_codec *codec) spec->adc_nids = alc861vd_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids); spec->capsrc_nids = alc861vd_capsrc_nids; - spec->is_mix_capture = 1; + spec->capture_style = CAPT_MIX; set_capture_mixer(spec); @@ -16397,7 +16446,7 @@ static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin, static int alc662_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, err, idx; for (i = 0; i < AUTO_PIN_LAST; i++) { @@ -16528,7 +16577,7 @@ static int alc662_parse_auto_config(struct hda_codec *codec) add_mixer(spec, spec->kctls.list); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; add_verb(spec, alc662_auto_init_verbs); if (codec->vendor_id == 0x10ec0663) @@ -16613,7 +16662,7 @@ static int patch_alc662(struct hda_codec *codec) spec->adc_nids = alc662_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids); spec->capsrc_nids = alc662_capsrc_nids; - spec->is_mix_capture = 1; + spec->capture_style = CAPT_MIX; if (!spec->cap_mixer) set_capture_mixer(spec); -- cgit v1.2.3-59-g8ed1b From b98b7b347eed333d6fa2f74770beb8106e576cc6 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Thu, 29 Jan 2009 13:18:31 -0200 Subject: ALSA: hda - make alc882_auto_init_input_src aware of selectors In the case of having a selector instead of mixer while initing input sources, the case that happens with ALC889, we must select instead of muting input. Problem was found while testing with hda-emu. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 21 ++++++++++++++++++--- 1 file changed, 18 insertions(+), 3 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d81cb5eb8c5f..3666cc5dc3bc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6924,18 +6924,21 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec) static void alc882_auto_init_input_src(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - const struct hda_input_mux *imux = spec->input_mux; int c; for (c = 0; c < spec->num_adc_nids; c++) { hda_nid_t conn_list[HDA_MAX_NUM_INPUTS]; hda_nid_t nid = spec->capsrc_nids[c]; + unsigned int mux_idx; + const struct hda_input_mux *imux; int conns, mute, idx, item; conns = snd_hda_get_connections(codec, nid, conn_list, ARRAY_SIZE(conn_list)); if (conns < 0) continue; + mux_idx = c >= spec->num_mux_defs ? 0 : c; + imux = &spec->input_mux[mux_idx]; for (idx = 0; idx < conns; idx++) { /* if the current connection is the selected one, * unmute it as default - otherwise mute it @@ -6948,8 +6951,20 @@ static void alc882_auto_init_input_src(struct hda_codec *codec) break; } } - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, mute); + /* check if we have a selector or mixer + * we could check for the widget type instead, but + * just check for Amp-In presence (in case of mixer + * without amp-in there is something wrong, this + * function shouldn't be used or capsrc nid is wrong) + */ + if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + mute); + else if (mute != AMP_IN_MUTE(idx)) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, + idx); } } } -- cgit v1.2.3-59-g8ed1b From ba340e825f4b892782779abd0f93bcff7e763567 Mon Sep 17 00:00:00 2001 From: Tony Vroon Date: Mon, 2 Feb 2009 19:01:30 +0000 Subject: ALSA: hda - Add tyan model for Realtek ALC262 The Realtek ALC262 on the Tyan Thunder n6650W (S2915-E) mainboard has a rather odd configuration template. As a result, the white AUX connector can not be used. This rewrites the default config to more accurately reflect the connector layout, colour and function. Unfortunately the black CD_IN connector, which is suspected to be widget 0x1c refuses to switch into input (0x20), instead opting to remain on 0x0. As such, no mixer controls are exposed for it. Autoswitching is implemented between the front headphone output and back line output. Signed-off-by: Tony Vroon Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 77 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 77 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0c81d92c3d75..bd9ef3363890 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -103,6 +103,7 @@ enum { ALC262_NEC, ALC262_TOSHIBA_S06, ALC262_TOSHIBA_RX1, + ALC262_TYAN, ALC262_AUTO, ALC262_MODEL_LAST /* last tag */ }; @@ -9509,6 +9510,67 @@ static struct snd_kcontrol_new alc262_benq_t31_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc262_tyan_mixer[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Aux Playback Volume", 0x0b, 0x06, HDA_INPUT), + HDA_CODEC_MUTE("Aux Playback Switch", 0x0b, 0x06, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + { } /* end */ +}; + +static struct hda_verb alc262_tyan_verbs[] = { + /* Headphone automute */ + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* P11 AUX_IN, white 4-pin connector */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, 0xe1}, + {0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, 0x93}, + {0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, 0x19}, + + {} +}; + +/* unsolicited event for HP jack sensing */ +static void alc262_tyan_automute(struct hda_codec *codec) +{ + unsigned int mute; + unsigned int present; + + snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0); + present = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0); + present = (present & 0x80000000) != 0; + if (present) { + /* mute line output on ATX panel */ + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, + HDA_AMP_MUTE, HDA_AMP_MUTE); + } else { + /* unmute line output if necessary */ + mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0); + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); + } +} + +static void alc262_tyan_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) != ALC880_HP_EVENT) + return; + alc262_tyan_automute(codec); +} + #define alc262_capture_mixer alc882_capture_mixer #define alc262_capture_alt_mixer alc882_capture_alt_mixer @@ -10626,6 +10688,7 @@ static const char *alc262_models[ALC262_MODEL_LAST] = { [ALC262_ULTRA] = "ultra", [ALC262_LENOVO_3000] = "lenovo-3000", [ALC262_NEC] = "nec", + [ALC262_TYAN] = "tyan", [ALC262_AUTO] = "auto", }; @@ -10666,6 +10729,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff7b, "Toshiba S06", ALC262_TOSHIBA_S06), SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU), + SND_PCI_QUIRK(0x10f1, 0x2915, "Tyan Thunder n6650W", ALC262_TYAN), SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA), SND_PCI_QUIRK(0x144d, 0xc039, "Samsung Q1U EL", ALC262_ULTRA), SND_PCI_QUIRK(0x144d, 0xc510, "Samsung Q45", ALC262_HIPPO), @@ -10884,6 +10948,19 @@ static struct alc_config_preset alc262_presets[] = { .unsol_event = alc262_hippo_unsol_event, .init_hook = alc262_hippo_automute, }, + [ALC262_TYAN] = { + .mixers = { alc262_tyan_mixer }, + .init_verbs = { alc262_init_verbs, alc262_tyan_verbs}, + .num_dacs = ARRAY_SIZE(alc262_dac_nids), + .dac_nids = alc262_dac_nids, + .hp_nid = 0x02, + .dig_out_nid = ALC262_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc262_modes), + .channel_mode = alc262_modes, + .input_mux = &alc262_capture_source, + .unsol_event = alc262_tyan_unsol_event, + .init_hook = alc262_tyan_automute, + }, }; static int patch_alc262(struct hda_codec *codec) -- cgit v1.2.3-59-g8ed1b From 680cd53652d8bfb2b97d8c0248d1afb82de6b61d Mon Sep 17 00:00:00 2001 From: Kusanagi Kouichi Date: Thu, 5 Feb 2009 00:00:58 +0900 Subject: ALSA: hda: Add digital beep generator support for Realtek codecs. A digital beep generator can be used via input layer. Signed-off-by: Kusanagi Kouichi Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.h | 2 +- sound/pci/hda/patch_realtek.c | 62 +++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 63 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index b9679f081cae..51bf6a5daf39 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -39,7 +39,7 @@ struct hda_beep { int snd_hda_attach_beep_device(struct hda_codec *codec, int nid); void snd_hda_detach_beep_device(struct hda_codec *codec); #else -#define snd_hda_attach_beep_device(...) +#define snd_hda_attach_beep_device(...) 0 #define snd_hda_detach_beep_device(...) #endif #endif diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bd9ef3363890..0faa41bfc8be 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -30,6 +30,7 @@ #include #include "hda_codec.h" #include "hda_local.h" +#include "hda_beep.h" #define ALC880_FRONT_EVENT 0x01 #define ALC880_DCVOL_EVENT 0x02 @@ -3187,6 +3188,7 @@ static void alc_free(struct hda_codec *codec) alc_free_kctls(codec); kfree(spec); + snd_hda_detach_beep_device(codec); codec->spec = NULL; /* to be sure */ } @@ -4355,6 +4357,12 @@ static int patch_alc880(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC880_AUTO) setup_preset(spec, &alc880_presets[board_config]); @@ -5882,6 +5890,12 @@ static int patch_alc260(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC260_AUTO) setup_preset(spec, &alc260_presets[board_config]); @@ -7093,6 +7107,12 @@ static int patch_alc882(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC882_AUTO) setup_preset(spec, &alc882_presets[board_config]); @@ -9093,6 +9113,12 @@ static int patch_alc883(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC883_AUTO) setup_preset(spec, &alc883_presets[board_config]); @@ -11013,6 +11039,12 @@ static int patch_alc262(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC262_AUTO) setup_preset(spec, &alc262_presets[board_config]); @@ -12051,6 +12083,12 @@ static int patch_alc268(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC268_AUTO) setup_preset(spec, &alc268_presets[board_config]); @@ -12885,6 +12923,12 @@ static int patch_alc269(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC269_AUTO) setup_preset(spec, &alc269_presets[board_config]); @@ -13978,6 +14022,12 @@ static int patch_alc861(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x23); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC861_AUTO) setup_preset(spec, &alc861_presets[board_config]); @@ -14924,6 +14974,12 @@ static int patch_alc861vd(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x23); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC861VD_AUTO) setup_preset(spec, &alc861vd_presets[board_config]); @@ -16733,6 +16789,12 @@ static int patch_alc662(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC662_AUTO) setup_preset(spec, &alc662_presets[board_config]); -- cgit v1.2.3-59-g8ed1b From 616f89e74cd954e04ae4f8bad6a3dc8730a4a47a Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Wed, 4 Feb 2009 11:23:19 -0500 Subject: ALSA: hda - Additional pin nids for STAC92HD71Bx and STAC92HD75Bx codecs Current code for STAC92HD71Bx and STAC92HD75Bx doesn't consider pin complexes 0x20 and 0x27. Also for 4 port models, nids 0x0e and 0x0f are vendor reserved. This commit changes code so it'll consider the additional pin complexes for models that have it, and avoid reserved nids to be touched on 4 port models. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 59 ++++++++++++++++++++++++++++++------------ 1 file changed, 43 insertions(+), 16 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a7df81efed23..58c9ff9d27f5 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -481,10 +481,17 @@ static hda_nid_t stac92hd83xxx_pin_nids[14] = { 0x0f, 0x10, 0x11, 0x12, 0x13, 0x1d, 0x1e, 0x1f, 0x20 }; -static hda_nid_t stac92hd71bxx_pin_nids[11] = { + +#define STAC92HD71BXX_NUM_PINS 13 +static hda_nid_t stac92hd71bxx_pin_nids_4port[STAC92HD71BXX_NUM_PINS] = { + 0x0a, 0x0b, 0x0c, 0x0d, 0x00, + 0x00, 0x14, 0x18, 0x19, 0x1e, + 0x1f, 0x20, 0x27 +}; +static hda_nid_t stac92hd71bxx_pin_nids_6port[STAC92HD71BXX_NUM_PINS] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x14, 0x18, 0x19, 0x1e, - 0x1f, + 0x1f, 0x20, 0x27 }; static hda_nid_t stac927x_pin_nids[14] = { @@ -1745,28 +1752,32 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { {} /* terminator */ }; -static unsigned int ref92hd71bxx_pin_configs[11] = { +static unsigned int ref92hd71bxx_pin_configs[STAC92HD71BXX_NUM_PINS] = { 0x02214030, 0x02a19040, 0x01a19020, 0x01014010, 0x0181302e, 0x01014010, 0x01019020, 0x90a000f0, - 0x90a000f0, 0x01452050, 0x01452050, + 0x90a000f0, 0x01452050, 0x01452050, 0x00000000, + 0x00000000 }; -static unsigned int dell_m4_1_pin_configs[11] = { +static unsigned int dell_m4_1_pin_configs[STAC92HD71BXX_NUM_PINS] = { 0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110, 0x23a1902e, 0x23014250, 0x40f000f0, 0x90a000f0, - 0x40f000f0, 0x4f0000f0, 0x4f0000f0, + 0x40f000f0, 0x4f0000f0, 0x4f0000f0, 0x00000000, + 0x00000000 }; -static unsigned int dell_m4_2_pin_configs[11] = { +static unsigned int dell_m4_2_pin_configs[STAC92HD71BXX_NUM_PINS] = { 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110, 0x23a1902e, 0x23014250, 0x40f000f0, 0x40f000f0, - 0x40f000f0, 0x044413b0, 0x044413b0, + 0x40f000f0, 0x044413b0, 0x044413b0, 0x00000000, + 0x00000000 }; -static unsigned int dell_m4_3_pin_configs[11] = { +static unsigned int dell_m4_3_pin_configs[STAC92HD71BXX_NUM_PINS] = { 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110, 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x90a000f0, - 0x40f000f0, 0x044413b0, 0x044413b0, + 0x40f000f0, 0x044413b0, 0x044413b0, 0x00000000, + 0x00000000 }; static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { @@ -2311,7 +2322,9 @@ static int stac92xx_save_bios_config_regs(struct hda_codec *codec) for (i = 0; i < spec->num_pins; i++) { hda_nid_t nid = spec->pin_nids[i]; unsigned int pin_cfg; - + + if (!nid) + continue; pin_cfg = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0x00); snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x bios pin config %8.8x\n", @@ -2354,8 +2367,9 @@ static void stac92xx_set_config_regs(struct hda_codec *codec) return; for (i = 0; i < spec->num_pins; i++) - stac92xx_set_config_reg(codec, spec->pin_nids[i], - spec->pin_configs[i]); + if (spec->pin_nids[i] && spec->pin_configs[i]) + stac92xx_set_config_reg(codec, spec->pin_nids[i], + spec->pin_configs[i]); } static int stac_save_pin_cfgs(struct hda_codec *codec, unsigned int *pins) @@ -4952,9 +4966,21 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) codec->spec = spec; codec->patch_ops = stac92xx_patch_ops; - spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids); + spec->num_pins = STAC92HD71BXX_NUM_PINS; + switch (codec->vendor_id) { + case 0x111d76b6: + case 0x111d76b7: + spec->pin_nids = stac92hd71bxx_pin_nids_4port; + break; + case 0x111d7603: + case 0x111d7608: + /* On 92HD75Bx 0x27 isn't a pin nid */ + spec->num_pins--; + /* fallthrough */ + default: + spec->pin_nids = stac92hd71bxx_pin_nids_6port; + } spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); - spec->pin_nids = stac92hd71bxx_pin_nids; memcpy(&spec->private_dimux, &stac92hd71bxx_dmux, sizeof(stac92hd71bxx_dmux)); spec->board_config = snd_hda_check_board_config(codec, @@ -5018,7 +5044,8 @@ again: /* disable VSW */ spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF]; unmute_init++; - stac_change_pin_config(codec, 0xf, 0x40f000f0); + stac_change_pin_config(codec, 0x0f, 0x40f000f0); + stac_change_pin_config(codec, 0x19, 0x40f000f3); break; case 0x111d7603: /* 6 Port with Analog Mixer */ if ((codec->revision_id & 0xf) == 1) -- cgit v1.2.3-59-g8ed1b From 6df703aefc81252447c69d24d2863007de2338e9 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Wed, 4 Feb 2009 11:34:22 -0500 Subject: ALSA: hda - Dynamic detection of dmics/dmuxes/smuxes in stac92hd71bxx Detect the number of connected ports and number of smuxes dynamically, looking at pin configs, using new introduced functions stac92hd71bxx_connected_ports and stac92hd71bxx_connected_smuxes. Also use proper input mux configuration for 4port and 5port models. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 99 +++++++++++++++++++++++++++++++++++++----- 1 file changed, 87 insertions(+), 12 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 58c9ff9d27f5..c36c1c0f9574 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4944,7 +4944,16 @@ again: return 0; } -static struct hda_input_mux stac92hd71bxx_dmux = { +static struct hda_input_mux stac92hd71bxx_dmux_nomixer = { + .num_items = 3, + .items = { + { "Analog Inputs", 0x00 }, + { "Digital Mic 1", 0x02 }, + { "Digital Mic 2", 0x03 }, + } +}; + +static struct hda_input_mux stac92hd71bxx_dmux_amixer = { .num_items = 4, .items = { { "Analog Inputs", 0x00 }, @@ -4954,11 +4963,57 @@ static struct hda_input_mux stac92hd71bxx_dmux = { } }; +static int stac92hd71bxx_connected_ports(struct hda_codec *codec, + hda_nid_t *nids, int num_nids) +{ + struct sigmatel_spec *spec = codec->spec; + int idx, num; + unsigned int def_conf; + + for (num = 0; num < num_nids; num++) { + for (idx = 0; idx < spec->num_pins; idx++) + if (spec->pin_nids[idx] == nids[num]) + break; + if (idx >= spec->num_pins) + break; + def_conf = get_defcfg_connect(spec->pin_configs[idx]); + if (def_conf == AC_JACK_PORT_NONE) + break; + } + return num; +} + +static int stac92hd71bxx_connected_smuxes(struct hda_codec *codec, + hda_nid_t dig0pin) +{ + struct sigmatel_spec *spec = codec->spec; + int idx; + + for (idx = 0; idx < spec->num_pins; idx++) + if (spec->pin_nids[idx] == dig0pin) + break; + if ((idx + 2) >= spec->num_pins) + return 0; + + /* dig1pin case */ + if (get_defcfg_connect(spec->pin_configs[idx+1]) != AC_JACK_PORT_NONE) + return 2; + + /* dig0pin + dig2pin case */ + if (get_defcfg_connect(spec->pin_configs[idx+2]) != AC_JACK_PORT_NONE) + return 2; + if (get_defcfg_connect(spec->pin_configs[idx]) != AC_JACK_PORT_NONE) + return 1; + else + return 0; +} + static int patch_stac92hd71bxx(struct hda_codec *codec) { struct sigmatel_spec *spec; struct hda_verb *unmute_init = stac92hd71bxx_unmute_core_init; int err = 0; + unsigned int ndmic_nids = 0; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -4981,8 +5036,6 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) spec->pin_nids = stac92hd71bxx_pin_nids_6port; } spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); - memcpy(&spec->private_dimux, &stac92hd71bxx_dmux, - sizeof(stac92hd71bxx_dmux)); spec->board_config = snd_hda_check_board_config(codec, STAC_92HD71BXX_MODELS, stac92hd71bxx_models, @@ -5007,16 +5060,32 @@ again: spec->gpio_data = 0x01; } + spec->dmic_nids = stac92hd71bxx_dmic_nids; + spec->dmux_nids = stac92hd71bxx_dmux_nids; + switch (codec->vendor_id) { case 0x111d76b6: /* 4 Port without Analog Mixer */ case 0x111d76b7: case 0x111d76b4: /* 6 Port without Analog Mixer */ case 0x111d76b5: + memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_nomixer, + sizeof(stac92hd71bxx_dmux_nomixer)); spec->mixer = stac92hd71bxx_mixer; spec->init = stac92hd71bxx_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; + spec->num_dmics = stac92hd71bxx_connected_ports(codec, + stac92hd71bxx_dmic_nids, + STAC92HD71BXX_NUM_DMICS); + if (spec->num_dmics) { + spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); + spec->dinput_mux = &spec->private_dimux; + ndmic_nids = ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1; + } break; case 0x111d7608: /* 5 Port with Analog Mixer */ + memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_amixer, + sizeof(stac92hd71bxx_dmux_amixer)); + spec->private_dimux.num_items--; switch (spec->board_config) { case STAC_HP_M4: /* Enable VREF power saving on GPIO1 detect */ @@ -5046,6 +5115,12 @@ again: unmute_init++; stac_change_pin_config(codec, 0x0f, 0x40f000f0); stac_change_pin_config(codec, 0x19, 0x40f000f3); + stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS - 1] = 0; + spec->num_dmics = stac92hd71bxx_connected_ports(codec, + stac92hd71bxx_dmic_nids, + STAC92HD71BXX_NUM_DMICS - 1); + spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); + ndmic_nids = ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 2; break; case 0x111d7603: /* 6 Port with Analog Mixer */ if ((codec->revision_id & 0xf) == 1) @@ -5055,10 +5130,17 @@ again: spec->num_pwrs = 0; /* fallthru */ default: + memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_amixer, + sizeof(stac92hd71bxx_dmux_amixer)); spec->dinput_mux = &spec->private_dimux; spec->mixer = stac92hd71bxx_analog_mixer; spec->init = stac92hd71bxx_analog_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; + spec->num_dmics = stac92hd71bxx_connected_ports(codec, + stac92hd71bxx_dmic_nids, + STAC92HD71BXX_NUM_DMICS); + spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); + ndmic_nids = ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1; } if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP) @@ -5071,13 +5153,12 @@ again: spec->digbeep_nid = 0x26; spec->mux_nids = stac92hd71bxx_mux_nids; spec->adc_nids = stac92hd71bxx_adc_nids; - spec->dmic_nids = stac92hd71bxx_dmic_nids; - spec->dmux_nids = stac92hd71bxx_dmux_nids; spec->smux_nids = stac92hd71bxx_smux_nids; spec->pwr_nids = stac92hd71bxx_pwr_nids; spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids); + spec->num_smuxes = stac92hd71bxx_connected_smuxes(codec, 0x1e); switch (spec->board_config) { case STAC_HP_M4: @@ -5097,17 +5178,11 @@ again: spec->num_smuxes = 0; spec->num_dmuxes = 0; break; - default: - spec->num_dmics = STAC92HD71BXX_NUM_DMICS; - spec->num_smuxes = ARRAY_SIZE(stac92hd71bxx_smux_nids); - spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); }; spec->multiout.dac_nids = spec->dac_nids; if (spec->dinput_mux) - spec->private_dimux.num_items += - spec->num_dmics - - (ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1); + spec->private_dimux.num_items += spec->num_dmics - ndmic_nids; err = stac92xx_parse_auto_config(codec, 0x21, 0x23); if (!err) { -- cgit v1.2.3-59-g8ed1b From 29d4ab4d6e996ef4c71910c915611151c34f1c75 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Wed, 4 Feb 2009 11:37:27 -0500 Subject: ALSA: hda - Don't call stac92xx_parse_auto_config with wrong dig_in Don't use uneeded/wrong third parameter for stac92xx_parse_auto_config in patch_stac92hd71bxx (no SPDIF in). Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index c36c1c0f9574..0b00110a5a02 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5184,7 +5184,7 @@ again: if (spec->dinput_mux) spec->private_dimux.num_items += spec->num_dmics - ndmic_nids; - err = stac92xx_parse_auto_config(codec, 0x21, 0x23); + err = stac92xx_parse_auto_config(codec, 0x21, 0); if (!err) { if (spec->board_config < 0) { printk(KERN_WARNING "hda_codec: No auto-config is " -- cgit v1.2.3-59-g8ed1b From 45c1d85bcc6438454d104966c30fd2497ae1cdd7 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Wed, 4 Feb 2009 17:49:41 -0500 Subject: ALSA: hda: Added stac378x digital slave out struct Added the ADATOut nid to a slave digital outs struct to allow output via the DigOut pin. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 0b00110a5a02..85dc642d1130 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -404,6 +404,10 @@ static hda_nid_t stac922x_mux_nids[2] = { 0x12, 0x13, }; +static hda_nid_t stac927x_slave_dig_outs[2] = { + 0x1f, 0, +}; + static hda_nid_t stac927x_adc_nids[3] = { 0x07, 0x08, 0x09 }; @@ -5320,6 +5324,7 @@ static int patch_stac927x(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + codec->slave_dig_outs = stac927x_slave_dig_outs; spec->num_pins = ARRAY_SIZE(stac927x_pin_nids); spec->pin_nids = stac927x_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_927X_MODELS, -- cgit v1.2.3-59-g8ed1b From 67f7857ab12e9f8005ef988f0b667396e07622c2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Feb 2009 12:14:52 +0100 Subject: ALSA: hda - Add quirk for HP zenith laptop Added model=laptop for another HP laptop (103c:3072) with AD1984A codec. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index e934e2c187d0..6e348d03b716 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3890,6 +3890,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE), SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE), + SND_PCI_QUIRK(0x103c, 0x3072, "HP", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30e6, "HP 6730b", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30e7, "HP EliteBook 8530p", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x3614, "HP 6730s", AD1884A_LAPTOP), -- cgit v1.2.3-59-g8ed1b From 632da7321b7e9fa5375956280f8a0f380836c22d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Feb 2009 15:02:06 +0100 Subject: ALSA: hda - Add quirk for another HP laptop Add model=laptop entry for another HP laptop (103c:3077) with AD1984A. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 6e348d03b716..30399cbf8193 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3891,6 +3891,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE), SND_PCI_QUIRK(0x103c, 0x3072, "HP", AD1884A_LAPTOP), + SND_PCI_QUIRK(0x103c, 0x3077, "HP", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30e6, "HP 6730b", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30e7, "HP EliteBook 8530p", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x3614, "HP 6730s", AD1884A_LAPTOP), -- cgit v1.2.3-59-g8ed1b From 939778aedd9386e13051a9e1d57c14cba2b6ae13 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Feb 2009 15:57:55 +0100 Subject: ALSA: hda - Add missing KERN_* prefix to printk ... and disable the annoying debug message. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5218118f01b0..d2812ab729cc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8265,7 +8265,7 @@ static void alc888_6st_dell_unsol_event(struct hda_codec *codec, { switch (res >> 26) { case ALC880_HP_EVENT: - printk("hp_event\n"); + /* printk(KERN_DEBUG "hp_event\n"); */ alc888_6st_dell_front_automute(codec); break; } @@ -16564,7 +16564,7 @@ static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin, if (alc880_is_fixed_pin(pin)) { nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin)); - /* printk("DAC nid=%x\n",nid); */ + /* printk(KERN_DEBUG "DAC nid=%x\n",nid); */ /* specify the DAC as the extra output */ if (!spec->multiout.hp_nid) spec->multiout.hp_nid = nid; -- cgit v1.2.3-59-g8ed1b From b25c9da19889e33bb4ee2dff369fc46caa4543b0 Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Fri, 6 Feb 2009 15:02:27 +0800 Subject: ALSA: enable concurrent digital outputs for ALC1200 Add the SPDIF pin as slave digital out to enable concurrent HDMI/SPDIF outputs for ASUS M3A-H/HDMI with ALC1200 codec. Tested-by: Thomas Schneider Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_local.h | 1 + sound/pci/hda/patch_realtek.c | 8 ++++++++ 2 files changed, 9 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index ec687b206c0a..4086491ed33a 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -229,6 +229,7 @@ struct hda_multi_out { hda_nid_t hp_nid; /* optional DAC for HP, 0 when not exists */ hda_nid_t extra_out_nid[3]; /* optional DACs, 0 when not exists */ hda_nid_t dig_out_nid; /* digital out audio widget */ + hda_nid_t *slave_dig_outs; int max_channels; /* currently supported analog channels */ int dig_out_used; /* current usage of digital out (HDA_DIG_XXX) */ int no_share_stream; /* don't share a stream with multiple pins */ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d2812ab729cc..5194a58fafaa 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -349,6 +349,7 @@ struct alc_config_preset { hda_nid_t *dac_nids; hda_nid_t dig_out_nid; /* optional */ hda_nid_t hp_nid; /* optional */ + hda_nid_t *slave_dig_outs; unsigned int num_adc_nids; hda_nid_t *adc_nids; hda_nid_t *capsrc_nids; @@ -824,6 +825,7 @@ static void setup_preset(struct alc_spec *spec, spec->multiout.num_dacs = preset->num_dacs; spec->multiout.dac_nids = preset->dac_nids; spec->multiout.dig_out_nid = preset->dig_out_nid; + spec->multiout.slave_dig_outs = preset->slave_dig_outs; spec->multiout.hp_nid = preset->hp_nid; spec->num_mux_defs = preset->num_mux_defs; @@ -3107,6 +3109,7 @@ static int alc_build_pcms(struct hda_codec *codec) /* SPDIF for stream index #1 */ if (spec->multiout.dig_out_nid || spec->dig_in_nid) { codec->num_pcms = 2; + codec->slave_dig_outs = spec->multiout.slave_dig_outs; info = spec->pcm_rec + 1; info->name = spec->stream_name_digital; if (spec->dig_out_type) @@ -8603,6 +8606,10 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { {} }; +static hda_nid_t alc1200_slave_dig_outs[] = { + ALC883_DIGOUT_NID, 0, +}; + static struct alc_config_preset alc883_presets[] = { [ALC883_3ST_2ch_DIG] = { .mixers = { alc883_3ST_2ch_mixer }, @@ -8943,6 +8950,7 @@ static struct alc_config_preset alc883_presets[] = { .dac_nids = alc883_dac_nids, .dig_out_nid = ALC1200_DIGOUT_NID, .dig_in_nid = ALC883_DIGIN_NID, + .slave_dig_outs = alc1200_slave_dig_outs, .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), .channel_mode = alc883_sixstack_modes, .input_mux = &alc883_capture_source, -- cgit v1.2.3-59-g8ed1b From 45bdd1c1bbac56876cb9c71649300013281e4b22 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Feb 2009 16:11:25 +0100 Subject: ALSA: hda - Create beep mixer controls dynamically for Realtek codecs Create beep mixer controls dynamically for Realtek codecs instead of static arrays. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 147 ++++++++++++++---------------------------- 1 file changed, 47 insertions(+), 100 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5194a58fafaa..3b3b483e2a91 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -255,6 +255,7 @@ struct alc_spec { struct snd_kcontrol_new *mixers[5]; /* mixer arrays */ unsigned int num_mixers; struct snd_kcontrol_new *cap_mixer; /* capture mixer */ + unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */ const struct hda_verb *init_verbs[5]; /* initialization verbs * don't forget NULL @@ -937,7 +938,7 @@ static void alc_mic_automute(struct hda_codec *codec) HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); } #else -#define alc_mic_automute(codec) /* NOP */ +#define alc_mic_automute(codec) do {} while(0) /* NOP */ #endif /* disabled */ /* unsolicited event for HP jack sensing */ @@ -1389,8 +1390,6 @@ static struct snd_kcontrol_new alc888_base_mixer[] = { 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("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -1497,8 +1496,6 @@ static struct snd_kcontrol_new alc880_three_stack_mixer[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x3, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x19, 0x0, HDA_OUTPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1720,8 +1717,6 @@ static struct snd_kcontrol_new alc880_six_stack_mixer[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Channel Mode", @@ -1898,13 +1893,6 @@ static struct snd_kcontrol_new alc880_asus_w1v_mixer[] = { { } /* end */ }; -/* additional mixers to alc880_asus_mixer */ -static struct snd_kcontrol_new alc880_pcbeep_mixer[] = { - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - { } /* end */ -}; - /* TCL S700 */ static struct snd_kcontrol_new alc880_tcl_s700_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -1937,8 +1925,6 @@ static struct snd_kcontrol_new alc880_uniwill_mixer[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Channel Mode", @@ -2013,6 +1999,13 @@ static const char *alc_slave_sws[] = { static void alc_free_kctls(struct hda_codec *codec); +/* additional beep mixers; the actual parameters are overwritten at build */ +static struct snd_kcontrol_new alc_beep_mixer[] = { + HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_INPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0, 0, HDA_INPUT), + { } /* end */ +}; + static int alc_build_controls(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -2048,6 +2041,21 @@ static int alc_build_controls(struct hda_codec *codec) return err; } + /* create beep controls if needed */ + if (spec->beep_amp) { + struct snd_kcontrol_new *knew; + for (knew = alc_beep_mixer; knew->name; knew++) { + struct snd_kcontrol *kctl; + kctl = snd_ctl_new1(knew, codec); + if (!kctl) + return -ENOMEM; + kctl->private_value = spec->beep_amp; + err = snd_hda_ctl_add(codec, kctl); + if (err < 0) + return err; + } + } + /* if we have no master control, let's create it */ if (!spec->no_analog && !snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { @@ -3812,7 +3820,7 @@ static struct alc_config_preset alc880_presets[] = { .input_mux = &alc880_capture_source, }, [ALC880_UNIWILL_DIG] = { - .mixers = { alc880_asus_mixer, alc880_pcbeep_mixer }, + .mixers = { alc880_asus_mixer }, .init_verbs = { alc880_volume_init_verbs, alc880_pin_asus_init_verbs }, .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids), @@ -3850,8 +3858,7 @@ static struct alc_config_preset alc880_presets[] = { .init_hook = alc880_uniwill_p53_hp_automute, }, [ALC880_FUJITSU] = { - .mixers = { alc880_fujitsu_mixer, - alc880_pcbeep_mixer, }, + .mixers = { alc880_fujitsu_mixer }, .init_verbs = { alc880_volume_init_verbs, alc880_uniwill_p53_init_verbs, alc880_beep_init_verbs }, @@ -4310,10 +4317,6 @@ static void alc880_auto_init(struct hda_codec *codec) alc_inithook(codec); } -/* - * OK, here we have finally the patch for ALC880 - */ - static void set_capture_mixer(struct alc_spec *spec) { static struct snd_kcontrol_new *caps[3] = { @@ -4325,6 +4328,13 @@ static void set_capture_mixer(struct alc_spec *spec) spec->cap_mixer = caps[spec->num_adc_nids - 1]; } +#define set_beep_amp(spec, nid, idx, dir) \ + ((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir)) + +/* + * OK, here we have finally the patch for ALC880 + */ + static int patch_alc880(struct hda_codec *codec) { struct alc_spec *spec; @@ -4392,6 +4402,7 @@ static int patch_alc880(struct hda_codec *codec) } } set_capture_mixer(spec); + set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x0c; @@ -4541,12 +4552,6 @@ static struct snd_kcontrol_new alc260_input_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc260_pc_beep_mixer[] = { - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x07, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x07, 0x05, HDA_INPUT), - { } /* end */ -}; - /* update HP, line and mono out pins according to the master switch */ static void alc260_hp_master_update(struct hda_codec *codec, hda_nid_t hp, hda_nid_t line, @@ -4738,8 +4743,6 @@ static struct snd_kcontrol_new alc260_fujitsu_mixer[] = { 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("Beep Playback Volume", 0x07, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT), HDA_CODEC_VOLUME("Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Speaker Playback Switch", 0x09, 2, HDA_INPUT), { } /* end */ @@ -4784,8 +4787,6 @@ static struct snd_kcontrol_new alc260_acer_mixer[] = { 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("Beep Playback Volume", 0x07, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT), { } /* end */ }; @@ -4803,8 +4804,6 @@ static struct snd_kcontrol_new alc260_will_mixer[] = { 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), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT), { } /* end */ }; @@ -5308,8 +5307,6 @@ static struct snd_kcontrol_new alc260_test_mixer[] = { 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("Beep Playback Volume", 0x07, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, 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), @@ -5737,8 +5734,7 @@ static struct snd_pci_quirk alc260_cfg_tbl[] = { static struct alc_config_preset alc260_presets[] = { [ALC260_BASIC] = { .mixers = { alc260_base_output_mixer, - alc260_input_mixer, - alc260_pc_beep_mixer }, + alc260_input_mixer }, .init_verbs = { alc260_init_verbs }, .num_dacs = ARRAY_SIZE(alc260_dac_nids), .dac_nids = alc260_dac_nids, @@ -5924,6 +5920,7 @@ static int patch_alc260(struct hda_codec *codec) } } set_capture_mixer(spec); + set_beep_amp(spec, 0x07, 0x05, HDA_INPUT); spec->vmaster_nid = 0x08; @@ -6095,8 +6092,6 @@ static struct snd_kcontrol_new alc882_base_mixer[] = { 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("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -6123,8 +6118,6 @@ static struct snd_kcontrol_new alc882_w2jc_mixer[] = { 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("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -6176,8 +6169,6 @@ static struct snd_kcontrol_new alc882_asus_a7m_mixer[] = { 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("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -6286,8 +6277,10 @@ static struct snd_kcontrol_new alc882_macpro_mixer[] = { HDA_CODEC_MUTE("Headphone Playback Switch", 0x18, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT), + /* FIXME: this looks suspicious... HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x02, HDA_INPUT), + */ { } /* end */ }; @@ -7153,6 +7146,7 @@ static int patch_alc882(struct hda_codec *codec) } } set_capture_mixer(spec); + set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x0c; @@ -7429,8 +7423,6 @@ static struct snd_kcontrol_new alc883_base_mixer[] = { 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("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -7493,8 +7485,6 @@ static struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = { 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("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -7518,8 +7508,6 @@ static struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = { 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("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -7544,8 +7532,6 @@ static struct snd_kcontrol_new alc883_3ST_6ch_intel_mixer[] = { HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -7569,8 +7555,6 @@ static struct snd_kcontrol_new alc883_fivestack_mixer[] = { 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("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -9183,6 +9167,7 @@ static int patch_alc883(struct hda_codec *codec) if (!spec->cap_mixer) set_capture_mixer(spec); + set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x0c; @@ -9235,8 +9220,6 @@ static struct snd_kcontrol_new alc262_base_mixer[] = { HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), - /* HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT), */ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), @@ -9257,8 +9240,6 @@ static struct snd_kcontrol_new alc262_hippo1_mixer[] = { HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), - /* HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT), */ /*HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT),*/ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), { } /* end */ @@ -9367,8 +9348,6 @@ static struct snd_kcontrol_new alc262_HP_BPC_mixer[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_VOLUME("AUX IN Playback Volume", 0x0b, 0x06, HDA_INPUT), HDA_CODEC_MUTE("AUX IN Playback Switch", 0x0b, 0x06, HDA_INPUT), { } /* end */ @@ -9397,8 +9376,6 @@ static struct snd_kcontrol_new alc262_HP_BPC_WildWest_mixer[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -10073,8 +10050,6 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = { }, HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Switch", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), @@ -11085,6 +11060,7 @@ static int patch_alc262(struct hda_codec *codec) } if (!spec->cap_mixer && !spec->no_analog) set_capture_mixer(spec); + set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x0c; @@ -12205,8 +12181,6 @@ static struct snd_kcontrol_new alc269_base_mixer[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x0b, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x0b, 0x4, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), @@ -12233,8 +12207,6 @@ static struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = { HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Boost", 0x19, 0, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x04, HDA_INPUT), { } }; @@ -12258,8 +12230,6 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = { HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x0b, 0x03, HDA_INPUT), HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x0b, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("Dock Mic Boost", 0x1b, 0, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x04, HDA_INPUT), { } }; @@ -12296,13 +12266,6 @@ static struct snd_kcontrol_new alc269_fujitsu_mixer[] = { { } /* end */ }; -/* beep control */ -static struct snd_kcontrol_new alc269_beep_mixer[] = { - HDA_CODEC_VOLUME("Beep Playback Volume", 0x0b, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x0b, 0x4, HDA_INPUT), - { } /* end */ -}; - static struct hda_verb alc269_quanta_fl1_verbs[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, @@ -12749,13 +12712,6 @@ static int alc269_parse_auto_config(struct hda_codec *codec) if (spec->kctls.list) add_mixer(spec, spec->kctls.list); - /* create a beep mixer control if the pin 0x1d isn't assigned */ - for (i = 0; i < ARRAY_SIZE(spec->autocfg.input_pins); i++) - if (spec->autocfg.input_pins[i] == 0x1d) - break; - if (i >= ARRAY_SIZE(spec->autocfg.input_pins)) - add_mixer(spec, alc269_beep_mixer); - add_verb(spec, alc269_init_verbs); spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux[0]; @@ -12868,7 +12824,7 @@ static struct alc_config_preset alc269_presets[] = { .init_hook = alc269_eeepc_dmic_inithook, }, [ALC269_FUJITSU] = { - .mixers = { alc269_fujitsu_mixer, alc269_beep_mixer }, + .mixers = { alc269_fujitsu_mixer }, .cap_mixer = alc269_epc_capture_mixer, .init_verbs = { alc269_init_verbs, alc269_eeepc_dmic_init_verbs }, @@ -12955,6 +12911,7 @@ static int patch_alc269(struct hda_codec *codec) spec->capsrc_nids = alc269_capsrc_nids; if (!spec->cap_mixer) set_capture_mixer(spec); + set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); codec->patch_ops = alc_patch_ops; if (board_config == ALC269_AUTO) @@ -13205,8 +13162,6 @@ static struct snd_kcontrol_new alc861_asus_mixer[] = { static struct snd_kcontrol_new alc861_asus_laptop_mixer[] = { HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x23, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PC Beep Playback Switch", 0x23, 0x0, HDA_OUTPUT), { } }; @@ -14049,6 +14004,8 @@ static int patch_alc861(struct hda_codec *codec) spec->stream_digital_playback = &alc861_pcm_digital_playback; spec->stream_digital_capture = &alc861_pcm_digital_capture; + set_beep_amp(spec, 0x23, 0, HDA_OUTPUT); + spec->vmaster_nid = 0x03; codec->patch_ops = alc_patch_ops; @@ -14205,9 +14162,6 @@ static struct snd_kcontrol_new alc861vd_6st_mixer[] = { HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - { } /* end */ }; @@ -14231,9 +14185,6 @@ static struct snd_kcontrol_new alc861vd_3st_mixer[] = { HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - { } /* end */ }; @@ -14272,8 +14223,6 @@ static struct snd_kcontrol_new alc861vd_dallas_mixer[] = { HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Beep Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Beep Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -15015,6 +14964,7 @@ static int patch_alc861vd(struct hda_codec *codec) spec->capture_style = CAPT_MIX; set_capture_mixer(spec); + set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x02; @@ -15203,8 +15153,6 @@ static struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -15226,8 +15174,6 @@ static struct snd_kcontrol_new alc662_3ST_6ch_mixer[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -16832,6 +16778,7 @@ static int patch_alc662(struct hda_codec *codec) if (!spec->cap_mixer) set_capture_mixer(spec); + set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x02; -- cgit v1.2.3-59-g8ed1b From c8dcdf829ca1827a802eae841dd04de8c9d6653f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Feb 2009 16:21:20 +0100 Subject: ALSA: hda - Add missing NULL check in snd_hda_create_spdif_in_ctls() Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c9158799ccb6..93412f335dc4 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1984,6 +1984,8 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid) } for (dig_mix = dig_in_ctls; dig_mix->name; dig_mix++) { kctl = snd_ctl_new1(dig_mix, codec); + if (!kctl) + return -ENOMEM; kctl->private_value = nid; err = snd_hda_ctl_add(codec, kctl); if (err < 0) -- cgit v1.2.3-59-g8ed1b From c44765b8c8bfc883c9868ab7aef37d27b5b14be8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Feb 2009 16:48:10 +0100 Subject: ALSA: hda - Clear codec->beep at release Clear codec->beep field in snd_hda_detach_beep_device() to be sure. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 960fd7970384..4de5bacd3929 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -138,6 +138,7 @@ void snd_hda_detach_beep_device(struct hda_codec *codec) input_unregister_device(beep->dev); kfree(beep); + codec->beep = NULL; } } EXPORT_SYMBOL_HDA(snd_hda_detach_beep_device); -- cgit v1.2.3-59-g8ed1b From a4ddeba9c8896cba8c6ce7a98c0b5c755c15a746 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Feb 2009 17:21:09 +0100 Subject: ALSA: hda - Remove superfluous code in patch_realtek.c codec->spec is reset in the caller side. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 76934bc8b484..3d933e307b19 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3202,7 +3202,6 @@ static void alc_free(struct hda_codec *codec) alc_free_kctls(codec); kfree(spec); snd_hda_detach_beep_device(codec); - codec->spec = NULL; /* to be sure */ } #ifdef SND_HDA_NEEDS_RESUME -- cgit v1.2.3-59-g8ed1b From c5a4bcd0cac546c5d776af881c5e913ba4a9922d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Feb 2009 17:22:05 +0100 Subject: ALSA: hda - Use digital beep for AD codecs Use digital beep instead of analog pc-beep for AD codecs. Create the beep mixer controls dynamically on demand. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 140 +++++++++++++++++++++++++++---------------- 1 file changed, 88 insertions(+), 52 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 30399cbf8193..cc02f2df2510 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -27,11 +27,12 @@ #include #include "hda_codec.h" #include "hda_local.h" +#include "hda_beep.h" struct ad198x_spec { struct snd_kcontrol_new *mixers[5]; int num_mixers; - + unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */ const struct hda_verb *init_verbs[5]; /* initialization verbs * don't forget NULL termination! */ @@ -154,6 +155,16 @@ static const char *ad_slave_sws[] = { static void ad198x_free_kctls(struct hda_codec *codec); +/* additional beep mixers; the actual parameters are overwritten at build */ +static struct snd_kcontrol_new ad_beep_mixer[] = { + HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0, 0, HDA_OUTPUT), + { } /* end */ +}; + +#define set_beep_amp(spec, nid, idx, dir) \ + ((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir)) /* mono */ + static int ad198x_build_controls(struct hda_codec *codec) { struct ad198x_spec *spec = codec->spec; @@ -181,6 +192,21 @@ static int ad198x_build_controls(struct hda_codec *codec) return err; } + /* create beep controls if needed */ + if (spec->beep_amp) { + struct snd_kcontrol_new *knew; + for (knew = ad_beep_mixer; knew->name; knew++) { + struct snd_kcontrol *kctl; + kctl = snd_ctl_new1(knew, codec); + if (!kctl) + return -ENOMEM; + kctl->private_value = spec->beep_amp; + err = snd_hda_ctl_add(codec, kctl); + if (err < 0) + return err; + } + } + /* if we have no master control, let's create it */ if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { unsigned int vmaster_tlv[4]; @@ -397,7 +423,8 @@ static void ad198x_free(struct hda_codec *codec) return; ad198x_free_kctls(codec); - kfree(codec->spec); + kfree(spec); + snd_hda_detach_beep_device(codec); } static struct hda_codec_ops ad198x_patch_ops = { @@ -536,8 +563,6 @@ static struct snd_kcontrol_new ad1986a_mixers[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x18, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x18, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT), @@ -601,8 +626,7 @@ static struct snd_kcontrol_new ad1986a_laptop_mixers[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT), - /* HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x18, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x18, 0x0, HDA_OUTPUT), + /* HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT), */ HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT), @@ -800,8 +824,6 @@ static struct snd_kcontrol_new ad1986a_laptop_automute_mixers[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x18, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x18, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT), { @@ -1026,7 +1048,7 @@ static int is_jack_available(struct hda_codec *codec, hda_nid_t nid) static int patch_ad1986a(struct hda_codec *codec) { struct ad198x_spec *spec; - int board_config; + int err, board_config; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -1034,6 +1056,13 @@ static int patch_ad1986a(struct hda_codec *codec) codec->spec = spec; + err = snd_hda_attach_beep_device(codec, 0x19); + if (err < 0) { + ad198x_free(codec); + return err; + } + set_beep_amp(spec, 0x18, 0, HDA_OUTPUT); + spec->multiout.max_channels = 6; spec->multiout.num_dacs = ARRAY_SIZE(ad1986a_dac_nids); spec->multiout.dac_nids = ad1986a_dac_nids; @@ -1213,8 +1242,6 @@ static struct snd_kcontrol_new ad1983_mixers[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("PC Speaker Playback Volume", 0x10, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("PC Speaker Playback Switch", 0x10, 1, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Boost", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT), @@ -1285,6 +1312,7 @@ static struct hda_amp_list ad1983_loopbacks[] = { static int patch_ad1983(struct hda_codec *codec) { struct ad198x_spec *spec; + int err; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -1292,6 +1320,13 @@ static int patch_ad1983(struct hda_codec *codec) codec->spec = spec; + err = snd_hda_attach_beep_device(codec, 0x10); + if (err < 0) { + ad198x_free(codec); + return err; + } + set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); + spec->multiout.max_channels = 2; spec->multiout.num_dacs = ARRAY_SIZE(ad1983_dac_nids); spec->multiout.dac_nids = ad1983_dac_nids; @@ -1361,8 +1396,6 @@ static struct snd_kcontrol_new ad1981_mixers[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("PC Speaker Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("PC Speaker Playback Switch", 0x0d, 1, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x08, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT), @@ -1685,7 +1718,7 @@ static struct snd_pci_quirk ad1981_cfg_tbl[] = { static int patch_ad1981(struct hda_codec *codec) { struct ad198x_spec *spec; - int board_config; + int err, board_config; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -1693,6 +1726,13 @@ static int patch_ad1981(struct hda_codec *codec) codec->spec = spec; + err = snd_hda_attach_beep_device(codec, 0x10); + if (err < 0) { + ad198x_free(codec); + return err; + } + set_beep_amp(spec, 0x0d, 0, HDA_OUTPUT); + spec->multiout.max_channels = 2; spec->multiout.num_dacs = ARRAY_SIZE(ad1981_dac_nids); spec->multiout.dac_nids = ad1981_dac_nids; @@ -1979,9 +2019,6 @@ static struct snd_kcontrol_new ad1988_6stack_mixers2[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), @@ -2025,9 +2062,6 @@ static struct snd_kcontrol_new ad1988_3stack_mixers2[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), @@ -2057,9 +2091,6 @@ static struct snd_kcontrol_new ad1988_laptop_mixers[] = { HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), @@ -2919,7 +2950,7 @@ static struct snd_pci_quirk ad1988_cfg_tbl[] = { static int patch_ad1988(struct hda_codec *codec) { struct ad198x_spec *spec; - int board_config; + int err, board_config; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -2939,7 +2970,7 @@ static int patch_ad1988(struct hda_codec *codec) if (board_config == AD1988_AUTO) { /* automatic parse from the BIOS config */ - int err = ad1988_parse_auto_config(codec); + err = ad1988_parse_auto_config(codec); if (err < 0) { ad198x_free(codec); return err; @@ -2949,6 +2980,13 @@ static int patch_ad1988(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x10); + if (err < 0) { + ad198x_free(codec); + return err; + } + set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); + switch (board_config) { case AD1988_6STACK: case AD1988_6STACK_DIG: @@ -3105,12 +3143,6 @@ static struct snd_kcontrol_new ad1884_base_mixers[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT), - /* - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_VOLUME("Digital Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Digital Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT), - */ HDA_CODEC_VOLUME("Mic Boost", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -3219,7 +3251,7 @@ static const char *ad1884_slave_vols[] = { "CD Playback Volume", "Internal Mic Playback Volume", "Docking Mic Playback Volume" - "Beep Playback Volume", + /* "Beep Playback Volume", */ "IEC958 Playback Volume", NULL }; @@ -3227,6 +3259,7 @@ static const char *ad1884_slave_vols[] = { static int patch_ad1884(struct hda_codec *codec) { struct ad198x_spec *spec; + int err; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -3234,6 +3267,13 @@ static int patch_ad1884(struct hda_codec *codec) codec->spec = spec; + err = snd_hda_attach_beep_device(codec, 0x10); + if (err < 0) { + ad198x_free(codec); + return err; + } + set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); + spec->multiout.max_channels = 2; spec->multiout.num_dacs = ARRAY_SIZE(ad1884_dac_nids); spec->multiout.dac_nids = ad1884_dac_nids; @@ -3300,8 +3340,6 @@ static struct snd_kcontrol_new ad1984_thinkpad_mixers[] = { HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Boost", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Docking Mic Boost", 0x25, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), @@ -3358,10 +3396,6 @@ static struct snd_kcontrol_new ad1984_dell_desktop_mixers[] = { HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT), HDA_CODEC_VOLUME("Line-In Playback Volume", 0x20, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Line-In Playback Switch", 0x20, 0x01, HDA_INPUT), - /* - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x20, 0x03, HDA_INPUT), - */ HDA_CODEC_VOLUME("Line-In Boost", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -3540,8 +3574,6 @@ static struct snd_kcontrol_new ad1884a_base_mixers[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Line Boost", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x25, 0x0, HDA_OUTPUT), @@ -3674,8 +3706,6 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = { HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x20, 0x04, HDA_INPUT), HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Boost", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Dock Mic Boost", 0x25, 0x0, HDA_OUTPUT), @@ -3703,8 +3733,6 @@ static struct snd_kcontrol_new ad1884a_mobile_mixers[] = { HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("Mic Capture Volume", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -3815,8 +3843,6 @@ static struct snd_kcontrol_new ad1984a_thinkpad_mixers[] = { HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Boost", 0x17, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -3902,7 +3928,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = { static int patch_ad1884a(struct hda_codec *codec) { struct ad198x_spec *spec; - int board_config; + int err, board_config; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -3910,6 +3936,13 @@ static int patch_ad1884a(struct hda_codec *codec) codec->spec = spec; + err = snd_hda_attach_beep_device(codec, 0x10); + if (err < 0) { + ad198x_free(codec); + return err; + } + set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); + spec->multiout.max_channels = 2; spec->multiout.num_dacs = ARRAY_SIZE(ad1884a_dac_nids); spec->multiout.dac_nids = ad1884a_dac_nids; @@ -4064,8 +4097,6 @@ static struct snd_kcontrol_new ad1882_loopback_mixers[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x07, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x07, HDA_INPUT), { } /* end */ }; @@ -4078,8 +4109,6 @@ static struct snd_kcontrol_new ad1882a_loopback_mixers[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x07, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x07, HDA_INPUT), HDA_CODEC_VOLUME("Digital Mic Boost", 0x1f, 0x0, HDA_INPUT), { } /* end */ }; @@ -4238,7 +4267,7 @@ static const char *ad1882_models[AD1986A_MODELS] = { static int patch_ad1882(struct hda_codec *codec) { struct ad198x_spec *spec; - int board_config; + int err, board_config; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -4246,6 +4275,13 @@ static int patch_ad1882(struct hda_codec *codec) codec->spec = spec; + err = snd_hda_attach_beep_device(codec, 0x10); + if (err < 0) { + ad198x_free(codec); + return err; + } + set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); + spec->multiout.max_channels = 6; spec->multiout.num_dacs = 3; spec->multiout.dac_nids = ad1882_dac_nids; -- cgit v1.2.3-59-g8ed1b From cfb9fb5517faa9e61c7e874fc89ef9c9253a0902 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Feb 2009 17:34:03 +0100 Subject: ALSA: hda - Fix unused variable compile warning MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit sound/pci/hda/patch_realtek.c:12693: warning: unused variable ‘i’ Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3d933e307b19..f594a0960290 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12690,7 +12690,7 @@ static int alc269_auto_create_analog_input_ctls(struct alc_spec *spec, static int alc269_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - int i, err; + int err; static hda_nid_t alc269_ignore[] = { 0x1d, 0 }; err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, -- cgit v1.2.3-59-g8ed1b From 8663ae55f39e99c25242adb6242a191258a4eca1 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Sun, 8 Feb 2009 19:50:34 -0200 Subject: ALSA: hda - Bind new ecs mobo id (1019:2950) to model=ecs202 This adds a new sound quirk entry (model=ecs202) for an ecs motherboard with IDT STAC9221 codec (1019:2950). Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 85dc642d1130..d16d5c60eecd 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2108,6 +2108,8 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = { "ECS/PC chips", STAC_ECS_202), SND_PCI_QUIRK(0x1019, 0x2820, "ECS/PC chips", STAC_ECS_202), + SND_PCI_QUIRK(0x1019, 0x2950, + "ECS/PC chips", STAC_ECS_202), {} /* terminator */ }; -- cgit v1.2.3-59-g8ed1b From 23c7b521c250b261dd97a7a06d5a2e74b56233d5 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Sun, 8 Feb 2009 19:51:28 -0200 Subject: ALSA: hda - Don't touch non-existent port f on 4-port 92hd71bxx codecs When checking for input amps on pins 0x0a, 0x0d and 0x0f, and initializing them for 92hd71xxx codec models, we must skip nid 0x0f for 4-port models too like with 5-port models, as it is unused (nid 0x0f is vendor reserved in 4-port models). Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d16d5c60eecd..2f4e090b0557 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5072,6 +5072,8 @@ again: switch (codec->vendor_id) { case 0x111d76b6: /* 4 Port without Analog Mixer */ case 0x111d76b7: + unmute_init++; + /* fallthru */ case 0x111d76b4: /* 6 Port without Analog Mixer */ case 0x111d76b5: memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_nomixer, -- cgit v1.2.3-59-g8ed1b From dea0a5095b5e21306a81c496567043798fac7815 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 9 Feb 2009 17:14:52 +0100 Subject: ALSA: hda - Clean up quirk lists Clean up quirk lists with bit masks. Also, sorted in numerical order for alc662_cfg_tbl[]. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 10 ++--- sound/pci/hda/patch_conexant.c | 20 +++------ sound/pci/hda/patch_realtek.c | 97 ++++++++++++++++++++---------------------- sound/pci/hda/patch_sigmatel.c | 61 ++++---------------------- 4 files changed, 65 insertions(+), 123 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index cc02f2df2510..6106dfe8ec04 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1015,10 +1015,8 @@ static struct snd_pci_quirk ad1986a_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba", AD1986A_LAPTOP_EAPD), SND_PCI_QUIRK(0x144d, 0xb03c, "Samsung R55", AD1986A_3STACK), SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_LAPTOP), - SND_PCI_QUIRK(0x144d, 0xc023, "Samsung X60", AD1986A_SAMSUNG), - SND_PCI_QUIRK(0x144d, 0xc024, "Samsung R65", AD1986A_SAMSUNG), - SND_PCI_QUIRK(0x144d, 0xc026, "Samsung X11", AD1986A_SAMSUNG), SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_ULTRA), + SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc000, "Samsung", AD1986A_SAMSUNG), SND_PCI_QUIRK(0x144d, 0xc504, "Samsung Q35", AD1986A_3STACK), SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_LAPTOP), SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_3STACK), @@ -1706,10 +1704,10 @@ static struct snd_pci_quirk ad1981_cfg_tbl[] = { SND_PCI_QUIRK(0x1014, 0x0597, "Lenovo Z60", AD1981_THINKPAD), SND_PCI_QUIRK(0x1014, 0x05b7, "Lenovo Z60m", AD1981_THINKPAD), /* All HP models */ - SND_PCI_QUIRK(0x103c, 0, "HP nx", AD1981_HP), + SND_PCI_QUIRK_VENDOR(0x103c, "HP nx", AD1981_HP), SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba U205", AD1981_TOSHIBA), /* Lenovo Thinkpad T60/X60/Z6xx */ - SND_PCI_QUIRK(0x17aa, 0, "Lenovo Thinkpad", AD1981_THINKPAD), + SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1981_THINKPAD), /* HP nx6320 (reversed SSID, H/W bug) */ SND_PCI_QUIRK(0x30b0, 0x103c, "HP nx6320", AD1981_HP), {} @@ -3481,7 +3479,7 @@ static const char *ad1984_models[AD1984_MODELS] = { static struct snd_pci_quirk ad1984_cfg_tbl[] = { /* Lenovo Thinkpad T61/X61 */ - SND_PCI_QUIRK(0x17aa, 0, "Lenovo Thinkpad", AD1984_THINKPAD), + SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1984_THINKPAD), SND_PCI_QUIRK(0x1028, 0x0214, "Dell T3400", AD1984_DELL_DESKTOP), {} }; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 0177ef8f4c9e..fdf876be712d 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1002,15 +1002,9 @@ static const char *cxt5045_models[CXT5045_MODELS] = { }; static struct snd_pci_quirk cxt5045_cfg_tbl[] = { - SND_PCI_QUIRK(0x103c, 0x30a5, "HP", CXT5045_LAPTOP_HPSENSE), - SND_PCI_QUIRK(0x103c, 0x30b2, "HP DV Series", CXT5045_LAPTOP_HPSENSE), - SND_PCI_QUIRK(0x103c, 0x30b5, "HP DV2120", CXT5045_LAPTOP_HPSENSE), - SND_PCI_QUIRK(0x103c, 0x30b7, "HP DV6000Z", CXT5045_LAPTOP_HPSENSE), - SND_PCI_QUIRK(0x103c, 0x30bb, "HP DV8000", CXT5045_LAPTOP_HPSENSE), - SND_PCI_QUIRK(0x103c, 0x30cd, "HP DV Series", CXT5045_LAPTOP_HPSENSE), - SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV9533EG", CXT5045_LAPTOP_HPSENSE), SND_PCI_QUIRK(0x103c, 0x30d5, "HP 530", CXT5045_LAPTOP_HP530), - SND_PCI_QUIRK(0x103c, 0x30d9, "HP Spartan", CXT5045_LAPTOP_HPSENSE), + SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP DV Series", + CXT5045_LAPTOP_HPSENSE), SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba P105", CXT5045_LAPTOP_MICSENSE), SND_PCI_QUIRK(0x152d, 0x0753, "Benq R55E", CXT5045_BENQ), SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_LAPTOP_MICSENSE), @@ -1020,8 +1014,8 @@ static struct snd_pci_quirk cxt5045_cfg_tbl[] = { SND_PCI_QUIRK(0x1509, 0x1e40, "FIC", CXT5045_LAPTOP_HPMICSENSE), SND_PCI_QUIRK(0x1509, 0x2f05, "FIC", CXT5045_LAPTOP_HPMICSENSE), SND_PCI_QUIRK(0x1509, 0x2f06, "FIC", CXT5045_LAPTOP_HPMICSENSE), - SND_PCI_QUIRK(0x1631, 0xc106, "Packard Bell", CXT5045_LAPTOP_HPMICSENSE), - SND_PCI_QUIRK(0x1631, 0xc107, "Packard Bell", CXT5045_LAPTOP_HPMICSENSE), + SND_PCI_QUIRK_MASK(0x1631, 0xff00, 0xc100, "Packard Bell", + CXT5045_LAPTOP_HPMICSENSE), SND_PCI_QUIRK(0x8086, 0x2111, "Conexant Reference board", CXT5045_LAPTOP_HPSENSE), {} }; @@ -1571,11 +1565,9 @@ static const char *cxt5047_models[CXT5047_MODELS] = { }; static struct snd_pci_quirk cxt5047_cfg_tbl[] = { - SND_PCI_QUIRK(0x103c, 0x30a0, "HP DV1000", CXT5047_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30a5, "HP DV5200T/DV8000T", CXT5047_LAPTOP_HP), - SND_PCI_QUIRK(0x103c, 0x30b2, "HP DV2000T/DV3000T", CXT5047_LAPTOP), - SND_PCI_QUIRK(0x103c, 0x30b5, "HP DV2000Z", CXT5047_LAPTOP), - SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6700", CXT5047_LAPTOP), + SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP DV Series", + CXT5047_LAPTOP), SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba P100", CXT5047_LAPTOP_EAPD), {} }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f594a0960290..7ae8fad0189f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3598,7 +3598,7 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x8181, "ASUS P4GPL", ALC880_ASUS_DIG), SND_PCI_QUIRK(0x1043, 0x8196, "ASUS P5GD1", ALC880_6ST), SND_PCI_QUIRK(0x1043, 0x81b4, "ASUS", ALC880_6ST), - SND_PCI_QUIRK(0x1043, 0, "ASUS", ALC880_ASUS), /* default ASUS */ + SND_PCI_QUIRK_VENDOR(0x1043, "ASUS", ALC880_ASUS), /* default ASUS */ SND_PCI_QUIRK(0x104d, 0x81a0, "Sony", ALC880_3ST), SND_PCI_QUIRK(0x104d, 0x81d6, "Sony", ALC880_3ST), SND_PCI_QUIRK(0x107b, 0x3032, "Gateway", ALC880_5ST), @@ -3641,7 +3641,8 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x8086, 0xe400, "Intel mobo", ALC880_5ST_DIG), SND_PCI_QUIRK(0x8086, 0xe401, "Intel mobo", ALC880_5ST_DIG), SND_PCI_QUIRK(0x8086, 0xe402, "Intel mobo", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x8086, 0, "Intel mobo", ALC880_3ST), /* default Intel */ + /* default Intel */ + SND_PCI_QUIRK_VENDOR(0x8086, "Intel mobo", ALC880_3ST), SND_PCI_QUIRK(0xa0a0, 0x0560, "AOpen i915GMm-HFS", ALC880_5ST_DIG), SND_PCI_QUIRK(0xe803, 0x1019, NULL, ALC880_6ST_DIG), {} @@ -8521,7 +8522,8 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G", ALC888_ACER_ASPIRE_4930G), - SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), /* default Acer */ + /* default Acer */ + SND_PCI_QUIRK_VENDOR(0x1025, "Acer laptop", ALC883_ACER), SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL), SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG), SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP), @@ -8566,7 +8568,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720), SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720), - SND_PCI_QUIRK(0x1558, 0, "Clevo laptop", ALC883_LAPTOP_EAPD), + SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), SND_PCI_QUIRK(0x1734, 0x1107, "FSC AMILO Xi2550", @@ -10707,14 +10709,10 @@ static const char *alc262_models[ALC262_MODEL_LAST] = { static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x1002, 0x437b, "Hippo", ALC262_HIPPO), SND_PCI_QUIRK(0x1033, 0x8895, "NEC Versa S9100", ALC262_NEC), - SND_PCI_QUIRK(0x103c, 0x12fe, "HP xw9400", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x12ff, "HP xw4550", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x1306, "HP xw8600", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x1307, "HP xw6600", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x1308, "HP xw4600", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x1309, "HP xw4*00", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x130a, "HP xw6*00", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x130b, "HP xw8*00", ALC262_HP_BPC), + SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1200, "HP xw series", + ALC262_HP_BPC), + SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1300, "HP xw series", + ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL), SND_PCI_QUIRK(0x103c, 0x2801, "HP D7000", ALC262_HP_BPC_D7000_WF), SND_PCI_QUIRK(0x103c, 0x2802, "HP D7000", ALC262_HP_BPC_D7000_WL), @@ -10742,8 +10740,8 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU), SND_PCI_QUIRK(0x10f1, 0x2915, "Tyan Thunder n6650W", ALC262_TYAN), - SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA), - SND_PCI_QUIRK(0x144d, 0xc039, "Samsung Q1U EL", ALC262_ULTRA), + SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc032, "Samsung Q1", + ALC262_ULTRA), SND_PCI_QUIRK(0x144d, 0xc510, "Samsung Q45", ALC262_HIPPO), SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000 y410", ALC262_LENOVO_3000), SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8), @@ -14534,9 +14532,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff03, "Toshiba P205", ALC861VD_LENOVO), SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba L30-149", ALC861VD_DALLAS), SND_PCI_QUIRK(0x1565, 0x820d, "Biostar NF61S SE", ALC861VD_6ST_DIG), - SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo", ALC861VD_LENOVO), - SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo 3000 C200", ALC861VD_LENOVO), - SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000 N200", ALC861VD_LENOVO), + SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", ALC861VD_LENOVO), SND_PCI_QUIRK(0x1849, 0x0862, "ASRock K8NF6G-VSTA", ALC861VD_6ST_DIG), {} }; @@ -16150,56 +16146,55 @@ static const char *alc662_models[ALC662_MODEL_LAST] = { }; static struct snd_pci_quirk alc662_cfg_tbl[] = { - SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M51VA", ALC663_ASUS_M51VA), - SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS G50V", ALC663_ASUS_G50V), - SND_PCI_QUIRK(0x1043, 0x8290, "ASUS P5GC-MX", ALC662_3ST_6ch_DIG), - SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701), - SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20), - SND_PCI_QUIRK(0x1043, 0x1903, "ASUS F5GL", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M50Vr", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_ECS), SND_PCI_QUIRK(0x1043, 0x1000, "ASUS N50Vm", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS F7Z", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x1953, "ASUS NB", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS NB", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC663_ASUS_MODE3), + SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS M70V", ALC663_ASUS_MODE3), SND_PCI_QUIRK(0x1043, 0x11d3, "ASUS NB", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_ASUS_MODE2), SND_PCI_QUIRK(0x1043, 0x1203, "ASUS NB", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x19e3, "ASUS NB", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x1993, "ASUS N20", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x19c3, "ASUS F5Z/F6x", ALC662_ASUS_MODE2), SND_PCI_QUIRK(0x1043, 0x1339, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1913, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1843, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x16c3, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1753, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1763, "ASUS NB", ALC663_ASUS_MODE6), + SND_PCI_QUIRK(0x1043, 0x1765, "ASUS NB", ALC663_ASUS_MODE6), + SND_PCI_QUIRK(0x1043, 0x1783, "ASUS NB", ALC662_ASUS_MODE2), SND_PCI_QUIRK(0x1043, 0x1813, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1876, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1823, "ASUS NB", ALC663_ASUS_MODE5), + SND_PCI_QUIRK(0x1043, 0x1833, "ASUS NB", ALC663_ASUS_MODE6), + SND_PCI_QUIRK(0x1043, 0x1843, "ASUS NB", ALC662_ASUS_MODE2), SND_PCI_QUIRK(0x1043, 0x1864, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1783, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1753, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x16c3, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1933, "ASUS F80Q", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1876, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M51VA", ALC663_ASUS_M51VA), + /*SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M50Vr", ALC663_ASUS_MODE1),*/ SND_PCI_QUIRK(0x1043, 0x1893, "ASUS M50Vm", ALC663_ASUS_MODE3), - SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS M70V", ALC663_ASUS_MODE3), - SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71C", ALC663_ASUS_MODE3), SND_PCI_QUIRK(0x1043, 0x1894, "ASUS X55", ALC663_ASUS_MODE3), - SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC663_ASUS_MODE3), + SND_PCI_QUIRK(0x1043, 0x1903, "ASUS F5GL", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1913, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1933, "ASUS F80Q", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1953, "ASUS NB", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71C", ALC663_ASUS_MODE3), + SND_PCI_QUIRK(0x1043, 0x1993, "ASUS N20", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS G50V", ALC663_ASUS_G50V), + /*SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS NB", ALC663_ASUS_MODE1),*/ + SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS F7Z", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x19c3, "ASUS F5Z/F6x", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x19e3, "ASUS NB", ALC663_ASUS_MODE1), SND_PCI_QUIRK(0x1043, 0x19f3, "ASUS NB", ALC663_ASUS_MODE4), - SND_PCI_QUIRK(0x1043, 0x1823, "ASUS NB", ALC663_ASUS_MODE5), - SND_PCI_QUIRK(0x1043, 0x1833, "ASUS NB", ALC663_ASUS_MODE6), - SND_PCI_QUIRK(0x1043, 0x1763, "ASUS NB", ALC663_ASUS_MODE6), - SND_PCI_QUIRK(0x1043, 0x1765, "ASUS NB", ALC663_ASUS_MODE6), + SND_PCI_QUIRK(0x1043, 0x8290, "ASUS P5GC-MX", ALC662_3ST_6ch_DIG), + SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701), + SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20), + SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS), SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K", ALC662_3ST_6ch_DIG), - SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E), - SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_ECS), - SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L", ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x1565, 0x820f, "Biostar TA780G M2+", ALC662_3ST_6ch_DIG), + SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E), SND_PCI_QUIRK(0x1849, 0x3662, "ASROCK K10N78FullHD-hSLI R3.0", ALC662_3ST_6ch_DIG), - SND_PCI_QUIRK(0x1854, 0x2000, "ASUS H13-2000", ALC663_ASUS_H13), - SND_PCI_QUIRK(0x1854, 0x2001, "ASUS H13-2001", ALC663_ASUS_H13), - SND_PCI_QUIRK(0x1854, 0x2002, "ASUS H13-2002", ALC663_ASUS_H13), + SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x", + ALC663_ASUS_H13), {} }; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 2f4e090b0557..12b30884843b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2082,33 +2082,7 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01d7, "Dell XPS M1210", STAC_922X_DELL_M82), /* ECS/PC Chips boards */ - SND_PCI_QUIRK(0x1019, 0x2144, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2608, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2633, - "ECS/PC chips P17G/1333", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2811, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2812, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2813, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2814, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2815, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2816, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2817, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2818, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2819, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2820, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2950, + SND_PCI_QUIRK_MASK(0x1019, 0xf000, 0x2000, "ECS/PC chips", STAC_ECS_202), {} /* terminator */ }; @@ -2169,22 +2143,10 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x3d01, "Intel D946", STAC_D965_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0xa301, "Intel D946", STAC_D965_3ST), /* 965 based 3 stack systems */ - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2116, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2115, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2114, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2113, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2112, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2111, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2110, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2009, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2008, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2007, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2006, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2005, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2004, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2003, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2002, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2001, "Intel D965", STAC_D965_3ST), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2100, + "Intel D965", STAC_D965_3ST), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2000, + "Intel D965", STAC_D965_3ST), /* Dell 3 stack systems */ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f7, "Dell XPS M1730", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01dd, "Dell Dimension E520", STAC_DELL_3ST), @@ -2200,15 +2162,10 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02ff, "Dell ", STAC_DELL_BIOS), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0209, "Dell XPS 1330", STAC_DELL_BIOS), /* 965 based 5 stack systems */ - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2301, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2302, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2303, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2304, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2305, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2501, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2502, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2503, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2504, "Intel D965", STAC_D965_5ST), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2300, + "Intel D965", STAC_D965_5ST), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2500, + "Intel D965", STAC_D965_5ST), {} /* terminator */ }; -- cgit v1.2.3-59-g8ed1b From 22971e3a77f193579be525a12f3ab91dbf241517 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Feb 2009 11:56:44 +0100 Subject: ALSA: hda - add digital beep support for ALC268 Added the digital beep support for ALC268. It was missing in the last patches. However, ALC268 has a strange pin use for widget 0x1d, which could be used as another purpose. So, the patch adds a check of the beep control before creating the hook for input layer. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 28 ++++++++++++++++++++++------ 1 file changed, 22 insertions(+), 6 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7ae8fad0189f..97eaf3b1d97f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11885,7 +11885,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { static struct alc_config_preset alc268_presets[] = { [ALC267_QUANTA_IL1] = { - .mixers = { alc267_quanta_il1_mixer }, + .mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer }, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc267_quanta_il1_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -11967,7 +11967,8 @@ static struct alc_config_preset alc268_presets[] = { }, [ALC268_ACER_ASPIRE_ONE] = { .mixers = { alc268_acer_aspire_one_mixer, - alc268_capture_alt_mixer }, + alc268_beep_mixer, + alc268_capture_alt_mixer }, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_acer_aspire_one_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -12036,7 +12037,7 @@ static int patch_alc268(struct hda_codec *codec) { struct alc_spec *spec; int board_config; - int err; + int i, has_beep, err; spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -12091,13 +12092,28 @@ static int patch_alc268(struct hda_codec *codec) spec->stream_digital_playback = &alc268_pcm_digital_playback; - if (!query_amp_caps(codec, 0x1d, HDA_INPUT)) - /* override the amp caps for beep generator */ - snd_hda_override_amp_caps(codec, 0x1d, HDA_INPUT, + has_beep = 0; + for (i = 0; i < spec->num_mixers; i++) { + if (spec->mixers[i] == alc268_beep_mixer) { + has_beep = 1; + break; + } + } + + if (has_beep) { + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (!query_amp_caps(codec, 0x1d, HDA_INPUT)) + /* override the amp caps for beep generator */ + snd_hda_override_amp_caps(codec, 0x1d, HDA_INPUT, (0x0c << AC_AMPCAP_OFFSET_SHIFT) | (0x0c << AC_AMPCAP_NUM_STEPS_SHIFT) | (0x07 << AC_AMPCAP_STEP_SIZE_SHIFT) | (0 << AC_AMPCAP_MUTE_SHIFT)); + } if (!spec->adc_nids && spec->input_mux) { /* check whether NID 0x07 is valid */ -- cgit v1.2.3-59-g8ed1b From 32d2c7fa1344ddf51886eddf31e228d139501dc6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Feb 2009 11:33:13 +0100 Subject: ALSA: hda - Fix a wrong pin check in snd_hda_parse_pin_def_config() Fixed a wrong pin check (a typo) for debug print of digital input pin. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 93412f335dc4..95f10aec7a06 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3551,7 +3551,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, cfg->input_pins[AUTO_PIN_FRONT_LINE], cfg->input_pins[AUTO_PIN_CD], cfg->input_pins[AUTO_PIN_AUX]); - if (cfg->dig_out_pin) + if (cfg->dig_in_pin) snd_printd(" dig-in=0x%x\n", cfg->dig_in_pin); return 0; -- cgit v1.2.3-59-g8ed1b From 0852d7a654f75d22a3c09fd7da4a3551bbb37740 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Feb 2009 11:35:15 +0100 Subject: ALSA: hda - Detect multiple digital-out pins Detect multiple digital-out pins in snd_hda_parse_pin_defconfig(). The dig_out_pin and dig_out_type fields become arrays. The codec parser still doesn't use this multiple pins detection, though. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 17 ++++++++++------- sound/pci/hda/hda_local.h | 5 +++-- sound/pci/hda/patch_analog.c | 2 +- sound/pci/hda/patch_realtek.c | 20 ++++++++++---------- sound/pci/hda/patch_sigmatel.c | 10 +++++----- sound/pci/hda/patch_via.c | 10 +++++----- 6 files changed, 34 insertions(+), 30 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 95f10aec7a06..29eeb748561d 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3423,11 +3423,13 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, break; case AC_JACK_SPDIF_OUT: case AC_JACK_DIG_OTHER_OUT: - cfg->dig_out_pin = nid; - if (loc == AC_JACK_LOC_HDMI) - cfg->dig_out_type = HDA_PCM_TYPE_HDMI; - else - cfg->dig_out_type = HDA_PCM_TYPE_SPDIF; + if (cfg->dig_outs >= ARRAY_SIZE(cfg->dig_out_pins)) + continue; + cfg->dig_out_pins[cfg->dig_outs] = nid; + cfg->dig_out_type[cfg->dig_outs] = + (loc == AC_JACK_LOC_HDMI) ? + HDA_PCM_TYPE_HDMI : HDA_PCM_TYPE_SPDIF; + cfg->dig_outs++; break; case AC_JACK_SPDIF_IN: case AC_JACK_DIG_OTHER_IN: @@ -3541,8 +3543,9 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, cfg->hp_pins[1], cfg->hp_pins[2], cfg->hp_pins[3], cfg->hp_pins[4]); snd_printd(" mono: mono_out=0x%x\n", cfg->mono_out_pin); - if (cfg->dig_out_pin) - snd_printd(" dig-out=0x%x\n", cfg->dig_out_pin); + if (cfg->dig_outs) + snd_printd(" dig-out=0x%x/0x%x\n", + cfg->dig_out_pins[0], cfg->dig_out_pins[1]); snd_printd(" inputs: mic=0x%x, fmic=0x%x, line=0x%x, fline=0x%x," " cd=0x%x, aux=0x%x\n", cfg->input_pins[AUTO_PIN_MIC], diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 4086491ed33a..2ae6b53a4628 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -355,10 +355,11 @@ struct auto_pin_cfg { int line_out_type; /* AUTO_PIN_XXX_OUT */ hda_nid_t hp_pins[AUTO_CFG_MAX_OUTS]; hda_nid_t input_pins[AUTO_PIN_LAST]; - hda_nid_t dig_out_pin; + int dig_outs; + hda_nid_t dig_out_pins[2]; hda_nid_t dig_in_pin; hda_nid_t mono_out_pin; - int dig_out_type; /* HDA_PCM_TYPE_XXX */ + int dig_out_type[2]; /* HDA_PCM_TYPE_XXX */ int dig_in_type; /* HDA_PCM_TYPE_XXX */ }; diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 6106dfe8ec04..d58c32b5b433 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2898,7 +2898,7 @@ static int ad1988_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = AD1988_SPDIF_OUT; if (spec->autocfg.dig_in_pin) spec->dig_in_nid = AD1988_SPDIF_IN; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1db99df79502..e46251bceb9d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4291,7 +4291,7 @@ static int alc880_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = ALC880_DIGOUT_NID; if (spec->autocfg.dig_in_pin) spec->dig_in_nid = ALC880_DIGIN_NID; @@ -5658,7 +5658,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = ALC260_DIGOUT_NID; if (spec->kctls.list) add_mixer(spec, spec->kctls.list); @@ -10626,7 +10626,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; if (!spec->autocfg.line_outs) { - if (spec->autocfg.dig_out_pin || spec->autocfg.dig_in_pin) { + if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) { spec->multiout.max_channels = 2; spec->no_analog = 1; goto dig_only; @@ -10643,9 +10643,9 @@ static int alc262_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; dig_only: - if (spec->autocfg.dig_out_pin) { + if (spec->autocfg.dig_outs) { spec->multiout.dig_out_nid = ALC262_DIGOUT_NID; - spec->dig_out_type = spec->autocfg.dig_out_type; + spec->dig_out_type = spec->autocfg.dig_out_type[0]; } if (spec->autocfg.dig_in_pin) spec->dig_in_nid = ALC262_DIGIN_NID; @@ -11807,7 +11807,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = 2; /* digital only support output */ - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = ALC268_DIGOUT_NID; if (spec->kctls.list) @@ -12722,7 +12722,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = ALC269_DIGOUT_NID; if (spec->kctls.list) @@ -13779,7 +13779,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = ALC861_DIGOUT_NID; if (spec->kctls.list) @@ -14881,7 +14881,7 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = ALC861VD_DIGOUT_NID; if (spec->kctls.list) @@ -16689,7 +16689,7 @@ static int alc662_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = ALC880_DIGOUT_NID; if (spec->kctls.list) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 12b30884843b..1882c5735879 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2546,7 +2546,7 @@ static int stac92xx_build_pcms(struct hda_codec *codec) codec->num_pcms++; info++; info->name = "STAC92xx Digital"; - info->pcm_type = spec->autocfg.dig_out_type; + info->pcm_type = spec->autocfg.dig_out_type[0]; if (spec->multiout.dig_out_nid) { info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_digital_playback; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid; @@ -3706,7 +3706,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if (spec->multiout.max_channels > 2) spec->surr_switch = 1; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = dig_out; if (dig_in && spec->autocfg.dig_in_pin) spec->dig_in_nid = dig_in; @@ -3819,7 +3819,7 @@ static int stac9200_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = 0x05; if (spec->autocfg.dig_in_pin) spec->dig_in_nid = 0x04; @@ -4069,8 +4069,8 @@ static int stac92xx_init(struct hda_codec *codec) for (i = 0; i < spec->num_dmics; i++) stac92xx_auto_set_pinctl(codec, spec->dmic_nids[i], AC_PINCTL_IN_EN); - if (cfg->dig_out_pin) - stac92xx_auto_set_pinctl(codec, cfg->dig_out_pin, + if (cfg->dig_out_pins[0]) + stac92xx_auto_set_pinctl(codec, cfg->dig_out_pins[0], AC_PINCTL_OUT_EN); if (cfg->dig_in_pin) stac92xx_auto_set_pinctl(codec, cfg->dig_in_pin, diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index c761394cbe84..639b2ff510a6 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1354,7 +1354,7 @@ static int vt1708_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = VT1708_DIGOUT_NID; if (spec->autocfg.dig_in_pin) spec->dig_in_nid = VT1708_DIGIN_NID; @@ -1827,7 +1827,7 @@ static int vt1709_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = VT1709_DIGOUT_NID; if (spec->autocfg.dig_in_pin) spec->dig_in_nid = VT1709_DIGIN_NID; @@ -2371,7 +2371,7 @@ static int vt1708B_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = VT1708B_DIGOUT_NID; if (spec->autocfg.dig_in_pin) spec->dig_in_nid = VT1708B_DIGIN_NID; @@ -2836,7 +2836,7 @@ static int vt1708S_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = VT1708S_DIGOUT_NID; spec->extra_dig_out_nid = 0x15; @@ -3155,7 +3155,7 @@ static int vt1702_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = VT1702_DIGOUT_NID; spec->extra_dig_out_nid = 0x1B; -- cgit v1.2.3-59-g8ed1b From c98041f7d71890ac6aa2257d78ef175db44d2cd3 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Wed, 11 Feb 2009 20:33:15 -0200 Subject: ALSA: hda - Cleanup setting of pin_configs in patch_stac927x After commit "ALSA: hda - Fix restore of pin configs at resume for STAC/IDT codecs", the introduced stac_save_pin_cfgs function checks already for pins == NULL case, saving then default pin configs from machine with stac92xx_save_bios_config_regs. So we can remove the extra checks when stac927x_brd_tbl[spec->board_config] == NULL. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 1882c5735879..3c84817ccd2e 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5292,10 +5292,9 @@ static int patch_stac927x(struct hda_codec *codec) stac927x_models, stac927x_cfg_tbl); again: - if (spec->board_config < 0 || !stac927x_brd_tbl[spec->board_config]) { - if (spec->board_config < 0) - snd_printdd(KERN_INFO "hda_codec: Unknown model for" - "STAC927x, using BIOS defaults\n"); + if (spec->board_config < 0) { + snd_printdd(KERN_INFO "hda_codec: Unknown model for" + "STAC927x, using BIOS defaults\n"); err = stac92xx_save_bios_config_regs(codec); } else err = stac_save_pin_cfgs(codec, -- cgit v1.2.3-59-g8ed1b From f1eaaeec11982c6b529d4255987fdf507a5fa69e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Feb 2009 08:16:55 +0100 Subject: ALSA: hda - Allow fixed codec-probe mask Some devices have broken BIOS and they don't set the codec probe-bit properly after cleared by the driver. This makes the driver skipping the necessary codec slots. Since BIOS update isn't always easy, now the semantics of probe_mask option is changed a bit. When it contains the bit 8 (0x100), the lower bits are used to probe that slots regardless of codec-probe bits returned by the hardware. For example, probe_mask=0x103 will force to probe the codec slot #0 and #1. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 24 +++++++++++++++++------- 1 file changed, 17 insertions(+), 7 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 11e791b965f6..19886e4bc829 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -381,6 +381,7 @@ struct azx { /* HD codec */ unsigned short codec_mask; + int codec_probe_mask; /* copied from probe_mask option */ struct hda_bus *bus; /* CORB/RIRB */ @@ -1228,7 +1229,6 @@ static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] __devinitdata = { }; static int __devinit azx_codec_create(struct azx *chip, const char *model, - unsigned int codec_probe_mask, int no_init) { struct hda_bus_template bus_temp; @@ -1261,7 +1261,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model, /* First try to probe all given codec slots */ for (c = 0; c < max_slots; c++) { - if ((chip->codec_mask & (1 << c)) & codec_probe_mask) { + if ((chip->codec_mask & (1 << c)) & chip->codec_probe_mask) { if (probe_codec(chip, c) < 0) { /* Some BIOSen give you wrong codec addresses * that don't exist @@ -1285,7 +1285,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model, /* Then create codec instances */ for (c = 0; c < max_slots; c++) { - if ((chip->codec_mask & (1 << c)) & codec_probe_mask) { + if ((chip->codec_mask & (1 << c)) & chip->codec_probe_mask) { struct hda_codec *codec; err = snd_hda_codec_new(chip->bus, c, !no_init, &codec); if (err < 0) @@ -2101,20 +2101,31 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = { {} }; +#define AZX_FORCE_CODEC_MASK 0x100 + static void __devinit check_probe_mask(struct azx *chip, int dev) { const struct snd_pci_quirk *q; - if (probe_mask[dev] == -1) { + chip->codec_probe_mask = probe_mask[dev]; + if (chip->codec_probe_mask == -1) { q = snd_pci_quirk_lookup(chip->pci, probe_mask_list); if (q) { printk(KERN_INFO "hda_intel: probe_mask set to 0x%x " "for device %04x:%04x\n", q->value, q->subvendor, q->subdevice); - probe_mask[dev] = q->value; + chip->codec_probe_mask = q->value; } } + + /* check forced option */ + if (chip->codec_probe_mask != -1 && + (chip->codec_probe_mask & AZX_FORCE_CODEC_MASK)) { + chip->codec_mask = chip->codec_probe_mask & 0xff; + printk(KERN_INFO "hda_intel: codec_mask forced to 0x%x\n", + chip->codec_mask); + } } @@ -2347,8 +2358,7 @@ static int __devinit azx_probe(struct pci_dev *pci, card->private_data = chip; /* create codec instances */ - err = azx_codec_create(chip, model[dev], probe_mask[dev], - probe_only[dev]); + err = azx_codec_create(chip, model[dev], probe_only[dev]); if (err < 0) goto out_free; -- cgit v1.2.3-59-g8ed1b From 20db7cb0acd0ba5a3b12f686148d670294a69366 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Feb 2009 08:18:48 +0100 Subject: ALSA: hda - Add forced codec-slots for ASUS W5F ASUS W5F needs the fixed codec-slots to probe to override the BIOS problem. Tested-by: Giovanni Moser Frainer Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 19886e4bc829..e853e4a8bde9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2098,6 +2098,8 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = { SND_PCI_QUIRK(0x1028, 0x20ac, "Dell Studio Desktop", 0x01), /* including bogus ALC268 in slot#2 that conflicts with ALC888 */ SND_PCI_QUIRK(0x17c0, 0x4085, "Medion MD96630", 0x01), + /* forced codec slots */ + SND_PCI_QUIRK(0x1046, 0x1262, "ASUS W5F", 0x103), {} }; -- cgit v1.2.3-59-g8ed1b From 8bb0ac5573ff0879fef511e1a80a4a4db0316daa Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Thu, 12 Feb 2009 16:50:01 -0500 Subject: ALSA: hda: Add STAC_DELL_S14 quirk Add STAC_DELL_S14 quirk for new laptop series. Removed un-needed pins in pin_nids for stac92hd83xxx. Also reorganized connection selection code for the respective ports per quirk define. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 52 ++++++++++++++++++++++++++++-------------- 1 file changed, 35 insertions(+), 17 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3c84817ccd2e..1ebb36ca2e00 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -83,6 +83,7 @@ enum { enum { STAC_92HD83XXX_REF, STAC_92HD83XXX_PWR_REF, + STAC_DELL_S14, STAC_92HD83XXX_MODELS }; @@ -480,10 +481,9 @@ static hda_nid_t stac92hd73xx_pin_nids[13] = { 0x14, 0x22, 0x23 }; -static hda_nid_t stac92hd83xxx_pin_nids[14] = { +static hda_nid_t stac92hd83xxx_pin_nids[10] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, - 0x0f, 0x10, 0x11, 0x12, 0x13, - 0x1d, 0x1e, 0x1f, 0x20 + 0x0f, 0x10, 0x11, 0x1f, 0x20, }; #define STAC92HD71BXX_NUM_PINS 13 @@ -857,9 +857,9 @@ static struct hda_verb stac92hd73xx_10ch_core_init[] = { }; static struct hda_verb stac92hd83xxx_core_init[] = { - { 0xa, AC_VERB_SET_CONNECT_SEL, 0x0}, - { 0xb, AC_VERB_SET_CONNECT_SEL, 0x0}, - { 0xd, AC_VERB_SET_CONNECT_SEL, 0x1}, + { 0xa, AC_VERB_SET_CONNECT_SEL, 0x1}, + { 0xb, AC_VERB_SET_CONNECT_SEL, 0x1}, + { 0xd, AC_VERB_SET_CONNECT_SEL, 0x0}, /* power state controls amps */ { 0x01, AC_VERB_SET_EAPD, 1 << 2}, @@ -1730,21 +1730,28 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { {} /* terminator */ }; -static unsigned int ref92hd83xxx_pin_configs[14] = { +static unsigned int ref92hd83xxx_pin_configs[10] = { 0x02214030, 0x02211010, 0x02a19020, 0x02170130, 0x01014050, 0x01819040, 0x01014020, 0x90a3014e, - 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x01451160, 0x98560170, }; +static unsigned int dell_s14_pin_configs[10] = { + 0x02214030, 0x02211010, 0x02a19020, 0x01014050, + 0x40f000f0, 0x01819040, 0x40f000f0, 0x90a60160, + 0x40f000f0, 0x40f000f0, +}; + static unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_REF] = ref92hd83xxx_pin_configs, [STAC_92HD83XXX_PWR_REF] = ref92hd83xxx_pin_configs, + [STAC_DELL_S14] = dell_s14_pin_configs, }; static const char *stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_REF] = "ref", [STAC_92HD83XXX_PWR_REF] = "mic-ref", + [STAC_DELL_S14] = "dell-s14", }; static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { @@ -1753,6 +1760,8 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { "DFI LanParty", STAC_92HD83XXX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_92HD83XXX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02ba, + "unknown Dell", STAC_DELL_S14), {} /* terminator */ }; @@ -4822,6 +4831,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) hda_nid_t conn[STAC92HD83_DAC_COUNT + 1]; int err; int num_dacs; + hda_nid_t nid; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -4840,15 +4850,6 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids); spec->multiout.dac_nids = spec->dac_nids; - - /* set port 0xe to select the last DAC - */ - num_dacs = snd_hda_get_connections(codec, 0x0e, - conn, STAC92HD83_DAC_COUNT + 1) - 1; - - snd_hda_codec_write_cache(codec, 0xe, 0, - AC_VERB_SET_CONNECT_SEL, num_dacs); - spec->init = stac92hd83xxx_core_init; spec->mixer = stac92hd83xxx_mixer; spec->num_pins = ARRAY_SIZE(stac92hd83xxx_pin_nids); @@ -4900,6 +4901,23 @@ again: return err; } + switch (spec->board_config) { + case STAC_DELL_S14: + nid = 0xf; + break; + default: + nid = 0xe; + break; + } + + num_dacs = snd_hda_get_connections(codec, nid, + conn, STAC92HD83_DAC_COUNT + 1) - 1; + + /* set port X to select the last DAC + */ + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, num_dacs); + codec->patch_ops = stac92xx_patch_ops; codec->proc_widget_hook = stac92hd_proc_hook; -- cgit v1.2.3-59-g8ed1b From 27e089888fb1a3d1d13892262f9d522b03985044 Mon Sep 17 00:00:00 2001 From: Aristeu Sergio Rozanski Filho Date: Thu, 12 Feb 2009 17:50:37 -0500 Subject: ALSA: hda: add quirk for Lenovo X200 laptop dock Currently the HP connector on X200 dock doesn't detect when a HP is connected nor allows sound to be played using it. This patch fixes the problem by adding a quirk for this specific model. It's possible that others have the same NID (0x19) to report when dock HP is connected, but I don't have access to any. Please Cc me in the reply since I'm not subscribed to alsa-devel@. Signed-off-by: Aristeu Rozanski Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_conexant.c | 40 ++++++++++++++++++++++++++++ 2 files changed, 41 insertions(+) (limited to 'sound/pci') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 8f40999a456e..0e52d273ce96 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -262,6 +262,7 @@ Conexant 5051 ============= laptop Basic Laptop config (default) hp HP Spartan laptop + lenovo-x200 Lenovo X200 laptop STAC9200 ======== diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index fdf876be712d..b8de73ecfdeb 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1798,6 +1798,40 @@ static struct hda_verb cxt5051_init_verbs[] = { { } /* end */ }; +static struct hda_verb cxt5051_lenovo_x200_init_verbs[] = { + /* Line in, Mic */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, + /* SPK */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* HP, Amp */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* Docking HP */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* DAC1 */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Record selector: Int mic */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44}, + /* SPDIF route: PCM */ + {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* EAPD */ + {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ + {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, + {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTC_EVENT}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, + { } /* end */ +}; + /* initialize jack-sensing, too */ static int cxt5051_init(struct hda_codec *codec) { @@ -1815,18 +1849,21 @@ static int cxt5051_init(struct hda_codec *codec) enum { CXT5051_LAPTOP, /* Laptops w/ EAPD support */ CXT5051_HP, /* no docking */ + CXT5051_LENOVO_X200, /* Lenovo X200 laptop */ CXT5051_MODELS }; static const char *cxt5051_models[CXT5051_MODELS] = { [CXT5051_LAPTOP] = "laptop", [CXT5051_HP] = "hp", + [CXT5051_LENOVO_X200] = "lenovo-x200", }; static struct snd_pci_quirk cxt5051_cfg_tbl[] = { SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", CXT5051_LAPTOP), SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), + SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT5051_LENOVO_X200), {} }; @@ -1867,6 +1904,9 @@ static int patch_cxt5051(struct hda_codec *codec) codec->patch_ops.unsol_event = cxt5051_hp_unsol_event; spec->mixers[0] = cxt5051_hp_mixers; break; + case CXT5051_LENOVO_X200: + spec->init_verbs[0] = cxt5051_lenovo_x200_init_verbs; + /* fallthru */ default: case CXT5051_LAPTOP: codec->patch_ops.unsol_event = cxt5051_hp_unsol_event; -- cgit v1.2.3-59-g8ed1b From 946835074e026f4bbe9f3c2b091dca6346bd1474 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Feb 2009 09:31:20 +0100 Subject: ALSA: hda - Add quirk for Acer AX1700-U3700A Force model=auto for Acer AX1700-U3700A with ALC888 codec. Since Acer devices are handlded as model=acer as default, the auto parsing has to be specified explicitly. (Maybe it's better rather to remove this default model=acer handling, though.) Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e46251bceb9d..2306cca1b69f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8520,6 +8520,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x013f, "Acer Aspire 5930G", ALC888_ACER_ASPIRE_4930G), + SND_PCI_QUIRK(0x1025, 0x0158, "Acer AX1700-U3700A", ALC883_AUTO), SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G", ALC888_ACER_ASPIRE_4930G), /* default Acer */ -- cgit v1.2.3-59-g8ed1b From 9b5f12e5a4029c1cd03784754687faef6d9e54fa Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Feb 2009 11:47:37 +0100 Subject: ALSA: hda - Add proper cleanup for multiout-dig for ALC codecs The recent patch_realtek.c contains the slave digital-out support as well. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2306cca1b69f..ef9b7ee34100 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2979,6 +2979,14 @@ static int alc880_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, stream_tag, format, substream); } +static int alc880_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct alc_spec *spec = codec->spec; + return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout); +} + static int alc880_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) @@ -3062,7 +3070,8 @@ static struct hda_pcm_stream alc880_pcm_digital_playback = { .ops = { .open = alc880_dig_playback_pcm_open, .close = alc880_dig_playback_pcm_close, - .prepare = alc880_dig_playback_pcm_prepare + .prepare = alc880_dig_playback_pcm_prepare, + .cleanup = alc880_dig_playback_pcm_cleanup }, }; -- cgit v1.2.3-59-g8ed1b From 6a05ac4afa90ac9c38fedd3f6940fe8da5d1fcf6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Feb 2009 11:19:09 +0100 Subject: ALSA: hda - Support multiple digital outs with auto-probing Added the support of multiple digital outputs via auto-probing for Realtek ALC88x codecs. The multiple outputs are handled as slave streams, so only one PCM stream (and the corresponding IEC958* elements) is provided to control both digital outputs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 22 +++++++++++++++++++--- 1 file changed, 19 insertions(+), 3 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ef9b7ee34100..244de597c5be 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -279,6 +279,7 @@ struct alc_spec { * dig_out_nid and hp_nid are optional */ hda_nid_t alt_dac_nid; + hda_nid_t slave_dig_outs[3]; /* optional - for auto-parsing */ int dig_out_type; /* capture */ @@ -4269,7 +4270,7 @@ static void alc880_auto_init_analog_input(struct hda_codec *codec) static int alc880_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - int err; + int i, err; static hda_nid_t alc880_ignore[] = { 0x1d, 0 }; err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, @@ -4300,8 +4301,23 @@ static int alc880_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_outs) - spec->multiout.dig_out_nid = ALC880_DIGOUT_NID; + /* check multiple SPDIF-out (for recent codecs) */ + for (i = 0; i < spec->autocfg.dig_outs; i++) { + hda_nid_t dig_nid; + err = snd_hda_get_connections(codec, + spec->autocfg.dig_out_pins[i], + &dig_nid, 1); + if (err < 0) + continue; + if (!i) + spec->multiout.dig_out_nid = dig_nid; + else { + spec->multiout.slave_dig_outs = spec->slave_dig_outs; + spec->slave_dig_outs[i - 1] = dig_nid; + if (i == ARRAY_SIZE(spec->slave_dig_outs) - 1) + break; + } + } if (spec->autocfg.dig_in_pin) spec->dig_in_nid = ALC880_DIGIN_NID; -- cgit v1.2.3-59-g8ed1b From e2ea57a8df6da45f5f63ab7b56528a552f36fb72 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Mon, 16 Feb 2009 10:23:00 +0100 Subject: ALSA: hda - Fix speaker output on HP DV4 1155-SE Force speaker pin config with model=hp-dv5 model for cases when bios doesn't set it up properly. All reported hp laptops using model=hp-dv5 model have speaker at pin 0x0d with same config, so it's safe to add this within hp-dv5 model. Reference: alsa-devel mailing list thread on http://mailman.alsa-project.org/pipermail/alsa-devel/2009-February/014390.html Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index aeb5d2126dae..7320059b7135 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1823,6 +1823,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "HP dv7", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f7, "HP dv4", STAC_HP_DV5), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fb, + "HP dv7", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fc, "HP dv7", STAC_HP_M4), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3600, @@ -5170,6 +5172,10 @@ again: spec->num_smuxes = 0; spec->num_dmuxes = 0; break; + case STAC_HP_DV5: + stac_change_pin_config(codec, 0x0d, 0x90170010); + stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN); + break; }; spec->multiout.dac_nids = spec->dac_nids; -- cgit v1.2.3-59-g8ed1b From 2ae466f8cc522843fa9a456e46007dd98b052b13 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 Feb 2009 14:16:36 +0100 Subject: ALSA: hda - Cleanup IDT92HD7x HP quirks Clean up IDT92HD7x quirks for HP laptops with SND_PCI_QUIRK_MASK(). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 20 +++++--------------- 1 file changed, 5 insertions(+), 15 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7320059b7135..d00a211a813b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1817,22 +1817,12 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "DFI LanParty", STAC_92HD71BXX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_92HD71BXX_REF), - SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f2, - "HP dv5", STAC_HP_M4), - SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f4, - "HP dv7", STAC_HP_DV5), - SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f7, - "HP dv4", STAC_HP_DV5), - SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fb, - "HP dv7", STAC_HP_DV5), - SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fc, - "HP dv7", STAC_HP_M4), - SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3600, - "HP dv5", STAC_HP_DV5), - SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3603, - "HP dv5", STAC_HP_DV5), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x30f0, + "HP dv4-7", STAC_HP_DV5), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3600, + "HP dv4-7", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361a, - "unknown HP", STAC_HP_M4), + "HP mini 1000", STAC_HP_M4), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233, "unknown Dell", STAC_DELL_M4_1), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0234, -- cgit v1.2.3-59-g8ed1b From c23127566c7a54c8413bf1b99becea76072f467e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 Feb 2009 15:20:41 +0100 Subject: ALSA: hda - Clean up quirks for HP laptops with AD1984A Clean up quirks for HP laptops with AD1984A using SND_PCI_QUIRK_MASK() Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index af6b0035e2e8..2c58d7b05aba 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3923,8 +3923,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE), SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE), - SND_PCI_QUIRK(0x103c, 0x3072, "HP", AD1884A_LAPTOP), - SND_PCI_QUIRK(0x103c, 0x3077, "HP", AD1884A_LAPTOP), + SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x3070, "HP", AD1884A_MOBILE), SND_PCI_QUIRK(0x103c, 0x30e6, "HP 6730b", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30e7, "HP EliteBook 8530p", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x3614, "HP 6730s", AD1884A_LAPTOP), -- cgit v1.2.3-59-g8ed1b From b3bdb30b6d1989129e297641fec791e9e555e4d8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 18 Feb 2009 13:16:26 +0100 Subject: ALSA: hda - Add quirk for Acer X3200 Acer X3200 needs model=auto, otherwise model=acer is pre-selected. Reference: Novell bnc#476268 https://bugzilla.novell.com/show_bug.cgi?id=476268 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 244de597c5be..192c92a5af38 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8545,6 +8545,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x013f, "Acer Aspire 5930G", ALC888_ACER_ASPIRE_4930G), + SND_PCI_QUIRK(0x1025, 0x0157, "Acer X3200", ALC883_AUTO), SND_PCI_QUIRK(0x1025, 0x0158, "Acer AX1700-U3700A", ALC883_AUTO), SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G", ALC888_ACER_ASPIRE_4930G), -- cgit v1.2.3-59-g8ed1b From 07eba61dd68678e30b24b4776f59798f625e089d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Feb 2009 08:06:35 +0100 Subject: ALSA: hda - Don't enable beep for digital-only ALC262 When ALC262 codec is configured as digital-only, it's meaningless to add the digital beep input. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 192c92a5af38..91da92259c80 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11051,10 +11051,12 @@ static int patch_alc262(struct hda_codec *codec) } } - err = snd_hda_attach_beep_device(codec, 0x1); - if (err < 0) { - alc_free(codec); - return err; + if (!spec->no_analog) { + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } } if (board_config != ALC262_AUTO) @@ -11087,7 +11089,8 @@ static int patch_alc262(struct hda_codec *codec) } if (!spec->cap_mixer && !spec->no_analog) set_capture_mixer(spec); - set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + if (!spec->no_analog) + set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x0c; -- cgit v1.2.3-59-g8ed1b From ab9fec099b796b002b6996c4c5845167d8fe6dbd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Feb 2009 08:13:26 +0100 Subject: ALSA: hda - Avoid doubly beep attachment in patch_alc268() Remove the doubly attachment in patch_alc268(). The input beep is attached conditionally only when needed. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 91da92259c80..df32f9353e71 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12100,12 +12100,6 @@ static int patch_alc268(struct hda_codec *codec) } } - err = snd_hda_attach_beep_device(codec, 0x1); - if (err < 0) { - alc_free(codec); - return err; - } - if (board_config != ALC268_AUTO) setup_preset(spec, &alc268_presets[board_config]); -- cgit v1.2.3-59-g8ed1b From 7e0e44d430281d398769f1d7864e161203252760 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Feb 2009 08:15:49 +0100 Subject: ALSA: hda - Add digital-only mode for ALC268 ALC268 can be configured as digital-only, e.g. for HDMI, on some machines. Allow the parser to set up the digital-only mode. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 19 +++++++++++++------ 1 file changed, 13 insertions(+), 6 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index df32f9353e71..169b3837af52 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11824,9 +11824,14 @@ static int alc268_parse_auto_config(struct hda_codec *codec) alc268_ignore); if (err < 0) return err; - if (!spec->autocfg.line_outs) + if (!spec->autocfg.line_outs) { + if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) { + spec->multiout.max_channels = 2; + spec->no_analog = 1; + goto dig_only; + } return 0; /* can't find valid BIOS pin config */ - + } err = alc268_auto_create_multi_out_ctls(spec, &spec->autocfg); if (err < 0) return err; @@ -11836,10 +11841,12 @@ static int alc268_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = 2; + dig_only: /* digital only support output */ - if (spec->autocfg.dig_outs) + if (spec->autocfg.dig_outs) { spec->multiout.dig_out_nid = ALC268_DIGOUT_NID; - + spec->dig_out_type = spec->autocfg.dig_out_type[0]; + } if (spec->kctls.list) add_mixer(spec, spec->kctls.list); @@ -12140,7 +12147,7 @@ static int patch_alc268(struct hda_codec *codec) (0 << AC_AMPCAP_MUTE_SHIFT)); } - if (!spec->adc_nids && spec->input_mux) { + if (!spec->no_analog && !spec->adc_nids && spec->input_mux) { /* check whether NID 0x07 is valid */ unsigned int wcap = get_wcaps(codec, 0x07); int i; @@ -12764,7 +12771,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - if (!spec->cap_mixer) + if (!spec->cap_mixer && !spec->no_analog) set_capture_mixer(spec); store_pin_configs(codec); -- cgit v1.2.3-59-g8ed1b From 3be141494a080a9189b51fa78154c975ad8d9806 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Feb 2009 14:11:16 +0100 Subject: ALSA: hda - Add generic pincfg initialization Added the generic pincfg cache and save/restore functions. Also introduced the pin-overriding via hwdep sysfs. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 151 +++++++++++++++++++++++++++++++++++++++++++--- sound/pci/hda/hda_codec.h | 15 +++++ sound/pci/hda/hda_hwdep.c | 66 ++++++++++++++++++++ 3 files changed, 223 insertions(+), 9 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 98884bc8f35f..6fa871f66a72 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -682,11 +682,132 @@ static int read_widget_caps(struct hda_codec *codec, hda_nid_t fg_node) return 0; } +/* read all pin default configurations and save codec->init_pins */ +static int read_pin_defaults(struct hda_codec *codec) +{ + int i; + hda_nid_t nid = codec->start_nid; + + for (i = 0; i < codec->num_nodes; i++, nid++) { + struct hda_pincfg *pin; + unsigned int wcaps = get_wcaps(codec, nid); + unsigned int wid_type = (wcaps & AC_WCAP_TYPE) >> + AC_WCAP_TYPE_SHIFT; + if (wid_type != AC_WID_PIN) + continue; + pin = snd_array_new(&codec->init_pins); + if (!pin) + return -ENOMEM; + pin->nid = nid; + pin->cfg = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONFIG_DEFAULT, 0); + } + return 0; +} + +/* look up the given pin config list and return the item matching with NID */ +static struct hda_pincfg *look_up_pincfg(struct hda_codec *codec, + struct snd_array *array, + hda_nid_t nid) +{ + int i; + for (i = 0; i < array->used; i++) { + struct hda_pincfg *pin = snd_array_elem(array, i); + if (pin->nid == nid) + return pin; + } + return NULL; +} + +/* write a config value for the given NID */ +static void set_pincfg(struct hda_codec *codec, hda_nid_t nid, + unsigned int cfg) +{ + int i; + for (i = 0; i < 4; i++) { + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 + i, + cfg & 0xff); + cfg >>= 8; + } +} + +/* set the current pin config value for the given NID. + * the value is cached, and read via snd_hda_codec_get_pincfg() + */ +int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list, + hda_nid_t nid, unsigned int cfg) +{ + struct hda_pincfg *pin; + + pin = look_up_pincfg(codec, list, nid); + if (!pin) { + pin = snd_array_new(list); + if (!pin) + return -ENOMEM; + pin->nid = nid; + } + pin->cfg = cfg; + set_pincfg(codec, nid, cfg); + return 0; +} + +int snd_hda_codec_set_pincfg(struct hda_codec *codec, + hda_nid_t nid, unsigned int cfg) +{ + return snd_hda_add_pincfg(codec, &codec->cur_pins, nid, cfg); +} +EXPORT_SYMBOL_HDA(snd_hda_codec_set_pincfg); + +/* get the current pin config value of the given pin NID */ +unsigned int snd_hda_codec_get_pincfg(struct hda_codec *codec, hda_nid_t nid) +{ + struct hda_pincfg *pin; + + pin = look_up_pincfg(codec, &codec->cur_pins, nid); + if (pin) + return pin->cfg; +#ifdef CONFIG_SND_HDA_HWDEP + pin = look_up_pincfg(codec, &codec->override_pins, nid); + if (pin) + return pin->cfg; +#endif + pin = look_up_pincfg(codec, &codec->init_pins, nid); + if (pin) + return pin->cfg; + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_codec_get_pincfg); + +/* restore all current pin configs */ +static void restore_pincfgs(struct hda_codec *codec) +{ + int i; + for (i = 0; i < codec->init_pins.used; i++) { + struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); + set_pincfg(codec, pin->nid, + snd_hda_codec_get_pincfg(codec, pin->nid)); + } +} static void init_hda_cache(struct hda_cache_rec *cache, unsigned int record_size); static void free_hda_cache(struct hda_cache_rec *cache); +/* restore the initial pin cfgs and release all pincfg lists */ +static void restore_init_pincfgs(struct hda_codec *codec) +{ + /* first free cur_pins and override_pins, then call restore_pincfg + * so that only the values in init_pins are restored + */ + snd_array_free(&codec->cur_pins); +#ifdef CONFIG_SND_HDA_HWDEP + snd_array_free(&codec->override_pins); +#endif + restore_pincfgs(codec); + snd_array_free(&codec->init_pins); +} + /* * codec destructor */ @@ -694,6 +815,7 @@ static void snd_hda_codec_free(struct hda_codec *codec) { if (!codec) return; + restore_init_pincfgs(codec); #ifdef CONFIG_SND_HDA_POWER_SAVE cancel_delayed_work(&codec->power_work); flush_workqueue(codec->bus->workq); @@ -751,6 +873,8 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); snd_array_init(&codec->mixers, sizeof(struct snd_kcontrol *), 32); + snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); + snd_array_init(&codec->cur_pins, sizeof(struct hda_pincfg), 16); if (codec->bus->modelname) { codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL); if (!codec->modelname) { @@ -787,15 +911,18 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr setup_fg_nodes(codec); if (!codec->afg && !codec->mfg) { snd_printdd("hda_codec: no AFG or MFG node found\n"); - snd_hda_codec_free(codec); - return -ENODEV; + err = -ENODEV; + goto error; } - if (read_widget_caps(codec, codec->afg ? codec->afg : codec->mfg) < 0) { + err = read_widget_caps(codec, codec->afg ? codec->afg : codec->mfg); + if (err < 0) { snd_printk(KERN_ERR "hda_codec: cannot malloc\n"); - snd_hda_codec_free(codec); - return -ENOMEM; + goto error; } + err = read_pin_defaults(codec); + if (err < 0) + goto error; if (!codec->subsystem_id) { hda_nid_t nid = codec->afg ? codec->afg : codec->mfg; @@ -808,10 +935,8 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr if (do_init) { err = snd_hda_codec_configure(codec); - if (err < 0) { - snd_hda_codec_free(codec); - return err; - } + if (err < 0) + goto error; } snd_hda_codec_proc_new(codec); @@ -824,6 +949,10 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr if (codecp) *codecp = codec; return 0; + + error: + snd_hda_codec_free(codec); + return err; } EXPORT_SYMBOL_HDA(snd_hda_codec_new); @@ -1334,6 +1463,9 @@ void snd_hda_codec_reset(struct hda_codec *codec) free_hda_cache(&codec->cmd_cache); init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); + /* free only cur_pins so that init_pins + override_pins are restored */ + snd_array_free(&codec->cur_pins); + restore_pincfgs(codec); codec->num_pcms = 0; codec->pcm_info = NULL; codec->preset = NULL; @@ -2175,6 +2307,7 @@ static void hda_call_codec_resume(struct hda_codec *codec) hda_set_power_state(codec, codec->afg ? codec->afg : codec->mfg, AC_PWRST_D0); + restore_pincfgs(codec); /* restore all current pin configs */ hda_exec_init_verbs(codec); if (codec->patch_ops.resume) codec->patch_ops.resume(codec); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 09a332ada0c6..6d01a8058f0a 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -778,11 +778,14 @@ struct hda_codec { unsigned short spdif_ctls; /* SPDIF control bits */ unsigned int spdif_in_enable; /* SPDIF input enable? */ hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */ + struct snd_array init_pins; /* initial (BIOS) pin configurations */ + struct snd_array cur_pins; /* current pin configurations */ #ifdef CONFIG_SND_HDA_HWDEP struct snd_hwdep *hwdep; /* assigned hwdep device */ struct snd_array init_verbs; /* additional init verbs */ struct snd_array hints; /* additional hints */ + struct snd_array override_pins; /* default pin configs to override */ #endif /* misc flags */ @@ -855,6 +858,18 @@ void snd_hda_codec_resume_cache(struct hda_codec *codec); #define snd_hda_sequence_write_cache snd_hda_sequence_write #endif +/* the struct for codec->pin_configs */ +struct hda_pincfg { + hda_nid_t nid; + unsigned int cfg; +}; + +unsigned int snd_hda_codec_get_pincfg(struct hda_codec *codec, hda_nid_t nid); +int snd_hda_codec_set_pincfg(struct hda_codec *codec, hda_nid_t nid, + unsigned int cfg); +int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list, + hda_nid_t nid, unsigned int cfg); /* for hwdep */ + /* * Mixer */ diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 4ae51dcb81af..71039a6dec28 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -109,6 +109,7 @@ static void clear_hwdep_elements(struct hda_codec *codec) for (i = 0; i < codec->hints.used; i++, head++) kfree(*head); snd_array_free(&codec->hints); + snd_array_free(&codec->override_pins); } static void hwdep_free(struct snd_hwdep *hwdep) @@ -141,6 +142,7 @@ int /*__devinit*/ snd_hda_create_hwdep(struct hda_codec *codec) snd_array_init(&codec->init_verbs, sizeof(struct hda_verb), 32); snd_array_init(&codec->hints, sizeof(char *), 32); + snd_array_init(&codec->override_pins, sizeof(struct hda_pincfg), 16); return 0; } @@ -316,6 +318,67 @@ static ssize_t hints_store(struct device *dev, return count; } +static ssize_t pin_configs_show(struct hda_codec *codec, + struct snd_array *list, + char *buf) +{ + int i, len = 0; + for (i = 0; i < list->used; i++) { + struct hda_pincfg *pin = snd_array_elem(list, i); + len += sprintf(buf + len, "0x%02x 0x%08x\n", + pin->nid, pin->cfg); + } + return len; +} + +static ssize_t init_pin_configs_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + return pin_configs_show(codec, &codec->init_pins, buf); +} + +static ssize_t override_pin_configs_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + return pin_configs_show(codec, &codec->override_pins, buf); +} + +static ssize_t cur_pin_configs_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + return pin_configs_show(codec, &codec->cur_pins, buf); +} + +#define MAX_PIN_CONFIGS 32 + +static ssize_t override_pin_configs_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + int nid, cfg; + int err; + + if (sscanf(buf, "%i %i", &nid, &cfg) != 2) + return -EINVAL; + if (!nid) + return -EINVAL; + err = snd_hda_add_pincfg(codec, &codec->override_pins, nid, cfg); + if (err < 0) + return err; + return count; +} + #define CODEC_ATTR_RW(type) \ __ATTR(type, 0644, type##_show, type##_store) #define CODEC_ATTR_RO(type) \ @@ -333,6 +396,9 @@ static struct device_attribute codec_attrs[] = { CODEC_ATTR_RW(modelname), CODEC_ATTR_WO(init_verbs), CODEC_ATTR_WO(hints), + CODEC_ATTR_RO(init_pin_configs), + CODEC_ATTR_RW(override_pin_configs), + CODEC_ATTR_RO(cur_pin_configs), CODEC_ATTR_WO(reconfig), CODEC_ATTR_WO(clear), }; -- cgit v1.2.3-59-g8ed1b From 0e8a21b59d48a63f45b3e6d2aca7fb91c5aec882 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Feb 2009 14:13:06 +0100 Subject: ALSA: hda - Remove realtek codec-specific pin save/restore functions Now it's done in the common code. Also use the common access functions for pin defaults. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 78 ++----------------------------------------- 1 file changed, 3 insertions(+), 75 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 169b3837af52..d7f255e3b916 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -329,13 +329,6 @@ struct alc_spec { /* for PLL fix */ hda_nid_t pll_nid; unsigned int pll_coef_idx, pll_coef_bit; - -#ifdef SND_HDA_NEEDS_RESUME -#define ALC_MAX_PINS 16 - unsigned int num_pins; - hda_nid_t pin_nids[ALC_MAX_PINS]; - unsigned int pin_cfgs[ALC_MAX_PINS]; -#endif }; /* @@ -1009,8 +1002,7 @@ static void alc_subsystem_id(struct hda_codec *codec, nid = 0x1d; if (codec->vendor_id == 0x10ec0260) nid = 0x17; - ass = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + ass = snd_hda_codec_get_pincfg(codec, nid); if (!(ass & 1) && !(ass & 0x100000)) return; if ((ass >> 30) != 1) /* no physical connection */ @@ -1184,16 +1176,8 @@ static void alc_fix_pincfg(struct hda_codec *codec, return; cfg = pinfix[quirk->value]; - for (; cfg->nid; cfg++) { - int i; - u32 val = cfg->val; - for (i = 0; i < 4; i++) { - snd_hda_codec_write(codec, cfg->nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 + i, - val & 0xff); - val >>= 8; - } - } + for (; cfg->nid; cfg++) + snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); } /* @@ -3215,61 +3199,13 @@ static void alc_free(struct hda_codec *codec) } #ifdef SND_HDA_NEEDS_RESUME -static void store_pin_configs(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - hda_nid_t nid, end_nid; - - end_nid = codec->start_nid + codec->num_nodes; - for (nid = codec->start_nid; nid < end_nid; nid++) { - unsigned int wid_caps = get_wcaps(codec, nid); - unsigned int wid_type = - (wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; - if (wid_type != AC_WID_PIN) - continue; - if (spec->num_pins >= ARRAY_SIZE(spec->pin_nids)) - break; - spec->pin_nids[spec->num_pins] = nid; - spec->pin_cfgs[spec->num_pins] = - snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); - spec->num_pins++; - } -} - -static void resume_pin_configs(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - int i; - - for (i = 0; i < spec->num_pins; i++) { - hda_nid_t pin_nid = spec->pin_nids[i]; - unsigned int pin_config = spec->pin_cfgs[i]; - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_0, - pin_config & 0x000000ff); - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, - (pin_config & 0x0000ff00) >> 8); - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, - (pin_config & 0x00ff0000) >> 16); - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, - pin_config >> 24); - } -} - static int alc_resume(struct hda_codec *codec) { - resume_pin_configs(codec); codec->patch_ops.init(codec); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); return 0; } -#else -#define store_pin_configs(codec) #endif /* @@ -4329,7 +4265,6 @@ static int alc880_parse_auto_config(struct hda_codec *codec) spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux[0]; - store_pin_configs(codec); return 1; } @@ -5693,7 +5628,6 @@ static int alc260_parse_auto_config(struct hda_codec *codec) spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux[0]; - store_pin_configs(codec); return 1; } @@ -10688,7 +10622,6 @@ static int alc262_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - store_pin_configs(codec); return 1; } @@ -11861,7 +11794,6 @@ static int alc268_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - store_pin_configs(codec); return 1; } @@ -12774,7 +12706,6 @@ static int alc269_parse_auto_config(struct hda_codec *codec) if (!spec->cap_mixer && !spec->no_analog) set_capture_mixer(spec); - store_pin_configs(codec); return 1; } @@ -13825,7 +13756,6 @@ static int alc861_parse_auto_config(struct hda_codec *codec) spec->num_adc_nids = ARRAY_SIZE(alc861_adc_nids); set_capture_mixer(spec); - store_pin_configs(codec); return 1; } @@ -14927,7 +14857,6 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - store_pin_configs(codec); return 1; } @@ -16737,7 +16666,6 @@ static int alc662_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - store_pin_configs(codec); return 1; } -- cgit v1.2.3-59-g8ed1b From 330ee9957910826a072c2ad5d4045182335f9963 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Feb 2009 14:33:36 +0100 Subject: ALSA: hda - Remove IDT codec-specific pin save/restore functions Removed its own save/restore functions and replaced with the common code. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 255 +++++++++++------------------------------ 1 file changed, 65 insertions(+), 190 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d00a211a813b..da48d8c0b295 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -229,7 +229,6 @@ struct sigmatel_spec { /* pin widgets */ hda_nid_t *pin_nids; unsigned int num_pins; - unsigned int *pin_configs; /* codec specific stuff */ struct hda_verb *init; @@ -2272,101 +2271,19 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = { {} /* terminator */ }; -static int stac92xx_save_bios_config_regs(struct hda_codec *codec) +static void stac92xx_set_config_regs(struct hda_codec *codec, + unsigned int *pincfgs) { int i; struct sigmatel_spec *spec = codec->spec; - - kfree(spec->pin_configs); - spec->pin_configs = kcalloc(spec->num_pins, sizeof(*spec->pin_configs), - GFP_KERNEL); - if (!spec->pin_configs) - return -ENOMEM; - - for (i = 0; i < spec->num_pins; i++) { - hda_nid_t nid = spec->pin_nids[i]; - unsigned int pin_cfg; - - if (!nid) - continue; - pin_cfg = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0x00); - snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x bios pin config %8.8x\n", - nid, pin_cfg); - spec->pin_configs[i] = pin_cfg; - } - - return 0; -} -static void stac92xx_set_config_reg(struct hda_codec *codec, - hda_nid_t pin_nid, unsigned int pin_config) -{ - int i; - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_0, - pin_config & 0x000000ff); - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, - (pin_config & 0x0000ff00) >> 8); - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, - (pin_config & 0x00ff0000) >> 16); - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, - pin_config >> 24); - i = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, - 0x00); - snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x pin config %8.8x\n", - pin_nid, i); -} - -static void stac92xx_set_config_regs(struct hda_codec *codec) -{ - int i; - struct sigmatel_spec *spec = codec->spec; - - if (!spec->pin_configs) - return; + if (!pincfgs) + return; for (i = 0; i < spec->num_pins; i++) - if (spec->pin_nids[i] && spec->pin_configs[i]) - stac92xx_set_config_reg(codec, spec->pin_nids[i], - spec->pin_configs[i]); -} - -static int stac_save_pin_cfgs(struct hda_codec *codec, unsigned int *pins) -{ - struct sigmatel_spec *spec = codec->spec; - - if (!pins) - return stac92xx_save_bios_config_regs(codec); - - kfree(spec->pin_configs); - spec->pin_configs = kmemdup(pins, - spec->num_pins * sizeof(*pins), - GFP_KERNEL); - if (!spec->pin_configs) - return -ENOMEM; - - stac92xx_set_config_regs(codec); - return 0; -} - -static void stac_change_pin_config(struct hda_codec *codec, hda_nid_t nid, - unsigned int cfg) -{ - struct sigmatel_spec *spec = codec->spec; - int i; - - for (i = 0; i < spec->num_pins; i++) { - if (spec->pin_nids[i] == nid) { - spec->pin_configs[i] = cfg; - stac92xx_set_config_reg(codec, nid, cfg); - break; - } - } + if (spec->pin_nids[i] && pincfgs[i]) + snd_hda_codec_set_pincfg(codec, spec->pin_nids[i], + pincfgs[i]); } /* @@ -2853,8 +2770,7 @@ static hda_nid_t check_mic_out_switch(struct hda_codec *codec) mic_pin = AUTO_PIN_MIC; for (;;) { hda_nid_t nid = cfg->input_pins[mic_pin]; - def_conf = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + def_conf = snd_hda_codec_get_pincfg(codec, nid); /* some laptops have an internal analog microphone * which can't be used as a output */ if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) { @@ -3426,11 +3342,7 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, unsigned int wcaps; unsigned int def_conf; - def_conf = snd_hda_codec_read(codec, - spec->dmic_nids[i], - 0, - AC_VERB_GET_CONFIG_DEFAULT, - 0); + def_conf = snd_hda_codec_get_pincfg(codec, spec->dmic_nids[i]); if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) continue; @@ -3779,9 +3691,7 @@ static int stac9200_auto_create_lfe_ctls(struct hda_codec *codec, for (i = 0; i < spec->autocfg.line_outs && lfe_pin == 0x0; i++) { hda_nid_t pin = spec->autocfg.line_out_pins[i]; unsigned int defcfg; - defcfg = snd_hda_codec_read(codec, pin, 0, - AC_VERB_GET_CONFIG_DEFAULT, - 0x00); + defcfg = snd_hda_codec_get_pincfg(codec, pin); if (get_defcfg_device(defcfg) == AC_JACK_SPEAKER) { unsigned int wcaps = get_wcaps(codec, pin); wcaps &= (AC_WCAP_STEREO | AC_WCAP_OUT_AMP); @@ -3885,8 +3795,7 @@ static int stac92xx_add_jack(struct hda_codec *codec, #ifdef CONFIG_SND_JACK struct sigmatel_spec *spec = codec->spec; struct sigmatel_jack *jack; - int def_conf = snd_hda_codec_read(codec, nid, - 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + int def_conf = snd_hda_codec_get_pincfg(codec, nid); int connectivity = get_defcfg_connect(def_conf); char name[32]; @@ -4066,8 +3975,7 @@ static int stac92xx_init(struct hda_codec *codec) pinctl); } } - conf = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + conf = snd_hda_codec_get_pincfg(codec, nid); if (get_defcfg_connect(conf) != AC_JACK_PORT_FIXED) { enable_pin_detect(codec, nid, STAC_INSERT_EVENT); @@ -4108,8 +4016,7 @@ static int stac92xx_init(struct hda_codec *codec) stac_toggle_power_map(codec, nid, 1); continue; } - def_conf = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + def_conf = snd_hda_codec_get_pincfg(codec, nid); def_conf = get_defcfg_connect(def_conf); /* skip any ports that don't have jacks since presence * detection is useless */ @@ -4163,7 +4070,6 @@ static void stac92xx_free(struct hda_codec *codec) if (! spec) return; - kfree(spec->pin_configs); stac92xx_free_jacks(codec); snd_array_free(&spec->events); @@ -4474,7 +4380,6 @@ static int stac92xx_resume(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - stac92xx_set_config_regs(codec); stac92xx_init(codec); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); @@ -4523,16 +4428,11 @@ static int patch_stac9200(struct hda_codec *codec) spec->board_config = snd_hda_check_board_config(codec, STAC_9200_MODELS, stac9200_models, stac9200_cfg_tbl); - if (spec->board_config < 0) { + if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9200, using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + else + stac92xx_set_config_regs(codec, stac9200_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; - } spec->multiout.max_channels = 2; spec->multiout.num_dacs = 1; @@ -4600,17 +4500,12 @@ static int patch_stac925x(struct hda_codec *codec) stac925x_models, stac925x_cfg_tbl); again: - if (spec->board_config < 0) { + if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC925x," "using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + else + stac92xx_set_config_regs(codec, stac925x_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; - } spec->multiout.max_channels = 2; spec->multiout.num_dacs = 1; @@ -4688,17 +4583,12 @@ static int patch_stac92hd73xx(struct hda_codec *codec) stac92hd73xx_models, stac92hd73xx_cfg_tbl); again: - if (spec->board_config < 0) { + 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); - } else - err = stac_save_pin_cfgs(codec, + else + stac92xx_set_config_regs(codec, stac92hd73xx_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; - } num_dacs = snd_hda_get_connections(codec, 0x0a, conn, STAC92HD73_DAC_COUNT + 2) - 1; @@ -4758,18 +4648,18 @@ again: spec->init = dell_m6_core_init; switch (spec->board_config) { case STAC_DELL_M6_AMIC: /* Analog Mics */ - stac92xx_set_config_reg(codec, 0x0b, 0x90A70170); + snd_hda_codec_set_pincfg(codec, 0x0b, 0x90A70170); spec->num_dmics = 0; spec->private_dimux.num_items = 1; break; case STAC_DELL_M6_DMIC: /* Digital Mics */ - stac92xx_set_config_reg(codec, 0x13, 0x90A60160); + snd_hda_codec_set_pincfg(codec, 0x13, 0x90A60160); spec->num_dmics = 1; spec->private_dimux.num_items = 2; break; case STAC_DELL_M6_BOTH: /* Both */ - stac92xx_set_config_reg(codec, 0x0b, 0x90A70170); - stac92xx_set_config_reg(codec, 0x13, 0x90A60160); + snd_hda_codec_set_pincfg(codec, 0x0b, 0x90A70170); + snd_hda_codec_set_pincfg(codec, 0x13, 0x90A60160); spec->num_dmics = 1; spec->private_dimux.num_items = 2; break; @@ -4865,17 +4755,12 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) stac92hd83xxx_models, stac92hd83xxx_cfg_tbl); again: - if (spec->board_config < 0) { + if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: Unknown model for" " STAC92HD83XXX, using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + else + stac92xx_set_config_regs(codec, stac92hd83xxx_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; - } switch (codec->vendor_id) { case 0x111d7604: @@ -4945,6 +4830,16 @@ static struct hda_input_mux stac92hd71bxx_dmux_amixer = { } }; +/* get the pin connection (fixed, none, etc) */ +static unsigned int stac_get_defcfg_connect(struct hda_codec *codec, int idx) +{ + struct sigmatel_spec *spec = codec->spec; + unsigned int cfg; + + cfg = snd_hda_codec_get_pincfg(codec, spec->pin_nids[idx]); + return get_defcfg_connect(cfg); +} + static int stac92hd71bxx_connected_ports(struct hda_codec *codec, hda_nid_t *nids, int num_nids) { @@ -4958,7 +4853,7 @@ static int stac92hd71bxx_connected_ports(struct hda_codec *codec, break; if (idx >= spec->num_pins) break; - def_conf = get_defcfg_connect(spec->pin_configs[idx]); + def_conf = stac_get_defcfg_connect(codec, idx); if (def_conf == AC_JACK_PORT_NONE) break; } @@ -4978,13 +4873,13 @@ static int stac92hd71bxx_connected_smuxes(struct hda_codec *codec, return 0; /* dig1pin case */ - if (get_defcfg_connect(spec->pin_configs[idx+1]) != AC_JACK_PORT_NONE) + if (stac_get_defcfg_connect(codec, idx + 1) != AC_JACK_PORT_NONE) return 2; /* dig0pin + dig2pin case */ - if (get_defcfg_connect(spec->pin_configs[idx+2]) != AC_JACK_PORT_NONE) + if (stac_get_defcfg_connect(codec, idx + 2) != AC_JACK_PORT_NONE) return 2; - if (get_defcfg_connect(spec->pin_configs[idx]) != AC_JACK_PORT_NONE) + if (stac_get_defcfg_connect(codec, idx) != AC_JACK_PORT_NONE) return 1; else return 0; @@ -5023,17 +4918,12 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) stac92hd71bxx_models, stac92hd71bxx_cfg_tbl); again: - if (spec->board_config < 0) { + if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: Unknown model for" " STAC92HD71BXX, using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + else + stac92xx_set_config_regs(codec, stac92hd71bxx_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; - } if (spec->board_config > STAC_92HD71BXX_REF) { /* GPIO0 = EAPD */ @@ -5097,8 +4987,8 @@ again: /* disable VSW */ spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF]; unmute_init++; - stac_change_pin_config(codec, 0x0f, 0x40f000f0); - stac_change_pin_config(codec, 0x19, 0x40f000f3); + snd_hda_codec_set_pincfg(codec, 0x0f, 0x40f000f0); + snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3); stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS - 1] = 0; spec->num_dmics = stac92hd71bxx_connected_ports(codec, stac92hd71bxx_dmic_nids, @@ -5147,7 +5037,7 @@ again: switch (spec->board_config) { case STAC_HP_M4: /* enable internal microphone */ - stac_change_pin_config(codec, 0x0e, 0x01813040); + snd_hda_codec_set_pincfg(codec, 0x0e, 0x01813040); stac92xx_auto_set_pinctl(codec, 0x0e, AC_PINCTL_IN_EN | AC_PINCTL_VREF_80); /* fallthru */ @@ -5163,7 +5053,7 @@ again: spec->num_dmuxes = 0; break; case STAC_HP_DV5: - stac_change_pin_config(codec, 0x0d, 0x90170010); + snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010); stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN); break; }; @@ -5247,17 +5137,12 @@ static int patch_stac922x(struct hda_codec *codec) } again: - if (spec->board_config < 0) { + if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC922x, " "using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + else + stac92xx_set_config_regs(codec, stac922x_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; - } spec->adc_nids = stac922x_adc_nids; spec->mux_nids = stac922x_mux_nids; @@ -5315,17 +5200,12 @@ static int patch_stac927x(struct hda_codec *codec) stac927x_models, stac927x_cfg_tbl); again: - if (spec->board_config < 0) { + if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: Unknown model for" "STAC927x, using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + else + stac92xx_set_config_regs(codec, stac927x_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; - } spec->digbeep_nid = 0x23; spec->adc_nids = stac927x_adc_nids; @@ -5354,15 +5234,15 @@ static int patch_stac927x(struct hda_codec *codec) case 0x10280209: case 0x1028022e: /* correct the device field to SPDIF out */ - stac_change_pin_config(codec, 0x21, 0x01442070); + snd_hda_codec_set_pincfg(codec, 0x21, 0x01442070); break; }; /* configure the analog microphone on some laptops */ - stac_change_pin_config(codec, 0x0c, 0x90a79130); + snd_hda_codec_set_pincfg(codec, 0x0c, 0x90a79130); /* correct the front output jack as a hp out */ - stac_change_pin_config(codec, 0x0f, 0x0227011f); + snd_hda_codec_set_pincfg(codec, 0x0f, 0x0227011f); /* correct the front input jack as a mic */ - stac_change_pin_config(codec, 0x0e, 0x02a79130); + snd_hda_codec_set_pincfg(codec, 0x0e, 0x02a79130); /* fallthru */ case STAC_DELL_3ST: /* GPIO2 High = Enable EAPD */ @@ -5447,16 +5327,11 @@ static int patch_stac9205(struct hda_codec *codec) stac9205_models, stac9205_cfg_tbl); again: - if (spec->board_config < 0) { + if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9205, using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + else + stac92xx_set_config_regs(codec, stac9205_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; - } spec->digbeep_nid = 0x23; spec->adc_nids = stac9205_adc_nids; @@ -5484,8 +5359,8 @@ static int patch_stac9205(struct hda_codec *codec) switch (spec->board_config){ case STAC_9205_DELL_M43: /* Enable SPDIF in/out */ - stac_change_pin_config(codec, 0x1f, 0x01441030); - stac_change_pin_config(codec, 0x20, 0x1c410030); + snd_hda_codec_set_pincfg(codec, 0x1f, 0x01441030); + snd_hda_codec_set_pincfg(codec, 0x20, 0x1c410030); /* Enable unsol response for GPIO4/Dock HP connection */ err = stac_add_event(spec, codec->afg, STAC_VREF_EVENT, 0x01); -- cgit v1.2.3-59-g8ed1b From 2f334f92cfb44d17b9f24a43f8998cca03f9a3dd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Feb 2009 14:37:42 +0100 Subject: ALSA: hda - Remove codec-specific pin save/restore functions Replace the accessor to pin defaults with the common code for caching. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 3 +-- sound/pci/hda/patch_cmedia.c | 12 ++++++------ sound/pci/hda/patch_via.c | 7 ++----- 3 files changed, 9 insertions(+), 13 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 2c58d7b05aba..53d0edaf04c7 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1047,8 +1047,7 @@ static struct hda_amp_list ad1986a_loopbacks[] = { static int is_jack_available(struct hda_codec *codec, hda_nid_t nid) { - unsigned int conf = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + unsigned int conf = snd_hda_codec_get_pincfg(codec, nid); return get_defcfg_connect(conf) != AC_JACK_PORT_NONE; } diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index f3ebe837f2d5..c921264bbd71 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -680,13 +680,13 @@ static int patch_cmi9880(struct hda_codec *codec) struct auto_pin_cfg cfg; /* collect pin default configuration */ - port_e = snd_hda_codec_read(codec, 0x0f, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); - port_f = snd_hda_codec_read(codec, 0x10, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + port_e = snd_hda_codec_get_pincfg(codec, 0x0f); + port_f = snd_hda_codec_get_pincfg(codec, 0x10); spec->front_panel = 1; if (get_defcfg_connect(port_e) == AC_JACK_PORT_NONE || get_defcfg_connect(port_f) == AC_JACK_PORT_NONE) { - port_g = snd_hda_codec_read(codec, 0x1f, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); - port_h = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + port_g = snd_hda_codec_get_pincfg(codec, 0x1f); + port_h = snd_hda_codec_get_pincfg(codec, 0x20); spec->channel_modes = cmi9880_channel_modes; /* no front panel */ if (get_defcfg_connect(port_g) == AC_JACK_PORT_NONE || @@ -703,8 +703,8 @@ static int patch_cmi9880(struct hda_codec *codec) spec->multiout.max_channels = cmi9880_channel_modes[0].channels; } else { spec->input_mux = &cmi9880_basic_mux; - port_spdifi = snd_hda_codec_read(codec, 0x13, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); - port_spdifo = snd_hda_codec_read(codec, 0x12, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + port_spdifi = snd_hda_codec_get_pincfg(codec, 0x13); + port_spdifo = snd_hda_codec_get_pincfg(codec, 0x12); if (get_defcfg_connect(port_spdifo) != AC_JACK_PORT_NONE) spec->multiout.dig_out_nid = CMI_DIG_OUT_NID; if (get_defcfg_connect(port_spdifi) != AC_JACK_PORT_NONE) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 639b2ff510a6..b25a5cc637d6 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1308,16 +1308,13 @@ static void vt1708_set_pinconfig_connect(struct hda_codec *codec, hda_nid_t nid) unsigned int def_conf; unsigned char seqassoc; - def_conf = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + def_conf = snd_hda_codec_get_pincfg(codec, nid); seqassoc = (unsigned char) get_defcfg_association(def_conf); seqassoc = (seqassoc << 4) | get_defcfg_sequence(def_conf); if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) { if (seqassoc == 0xff) { def_conf = def_conf & (~(AC_JACK_PORT_BOTH << 30)); - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, - def_conf >> 24); + snd_hda_codec_set_pincfg(codec, nid, def_conf); } } -- cgit v1.2.3-59-g8ed1b From cc95948972576c3efa43c9ed05b4a265805a4c54 Mon Sep 17 00:00:00 2001 From: Michael Schwingen Date: Sun, 22 Feb 2009 18:58:45 +0100 Subject: ALSA: hda - add support for "Maxdata Favorit 100XS" (Intel HDA/ALC260) Signed-off-by: Michael Schwingen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 130 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 130 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 169b3837af52..abddabc1efa8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -78,6 +78,7 @@ enum { ALC260_ACER, ALC260_WILL, ALC260_REPLACER_672V, + ALC260_FAVORIT100, #ifdef CONFIG_SND_DEBUG ALC260_TEST, #endif @@ -4537,6 +4538,26 @@ static struct hda_input_mux alc260_acer_capture_sources[2] = { }, }, }; + +/* Maxdata Favorit 100XS */ +static 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 @@ -4817,6 +4838,18 @@ static struct snd_kcontrol_new alc260_acer_mixer[] = { { } /* end */ }; +/* Maxdata Favorit 100XS: one output and one input (0x12) jack + */ +static 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. */ @@ -5188,6 +5221,89 @@ static struct hda_verb alc260_acer_init_verbs[] = { { } }; +/* Initialisation sequence for Maxdata Favorit 100XS + * (adapted from Acer init verbs). + */ +static 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 struct hda_verb alc260_will_verbs[] = { {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x0b, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -5730,6 +5846,7 @@ static const char *alc260_models[ALC260_MODEL_LAST] = { [ALC260_ACER] = "acer", [ALC260_WILL] = "will", [ALC260_REPLACER_672V] = "replacer", + [ALC260_FAVORIT100] = "favorit100", #ifdef CONFIG_SND_DEBUG [ALC260_TEST] = "test", #endif @@ -5739,6 +5856,7 @@ static const char *alc260_models[ALC260_MODEL_LAST] = { static struct snd_pci_quirk alc260_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x007b, "Acer C20x", ALC260_ACER), SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER), + SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100), SND_PCI_QUIRK(0x103c, 0x2808, "HP d5700", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x3010, "HP", ALC260_HP_3013), @@ -5840,6 +5958,18 @@ static struct alc_config_preset alc260_presets[] = { .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 }, -- cgit v1.2.3-59-g8ed1b From c17a1abae2f29047a0f57324240b01609489261b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Feb 2009 09:28:12 +0100 Subject: ALSA: hda - Use snd_hda_codec_get_pincfg() in the rest places Replace with snd_hda_codec_get_pincfg() in the places where available. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 3 +-- sound/pci/hda/hda_generic.c | 2 +- 2 files changed, 2 insertions(+), 3 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 6fa871f66a72..8ec2dfca9a6c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3488,8 +3488,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, if (ignore_nids && is_in_nid_list(nid, ignore_nids)) continue; - def_conf = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + def_conf = snd_hda_codec_get_pincfg(codec, nid); if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) continue; loc = get_defcfg_location(def_conf); diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 65745e96dc70..2c81a683e8f8 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -146,7 +146,7 @@ static int add_new_node(struct hda_codec *codec, struct hda_gspec *spec, hda_nid if (node->type == AC_WID_PIN) { node->pin_caps = snd_hda_param_read(codec, node->nid, AC_PAR_PIN_CAP); node->pin_ctl = snd_hda_codec_read(codec, node->nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - node->def_cfg = snd_hda_codec_read(codec, node->nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + node->def_cfg = snd_hda_codec_get_pincfg(codec, node->nid); } if (node->wid_caps & AC_WCAP_OUT_AMP) { -- cgit v1.2.3-59-g8ed1b From 346ff70fdbe9093947b9494fe714c89cafcceade Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Feb 2009 09:42:57 +0100 Subject: ALSA: hda - Rename {override,cur}_pin with {user,driver}_pin Rename from override_pin and cur_pin with user_pin and driver_pin, respectively, to be a bit more intuitive. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio.txt | 12 ++++++------ sound/pci/hda/hda_codec.c | 18 +++++++++--------- sound/pci/hda/hda_codec.h | 4 ++-- sound/pci/hda/hda_hwdep.c | 32 ++++++++++++++++---------------- 4 files changed, 33 insertions(+), 33 deletions(-) (limited to 'sound/pci') diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index 9c51e104546f..f590850c149f 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -371,16 +371,16 @@ hints:: not used. init_pin_configs:: Shows the initial pin default config values set by BIOS. -override_pin_configs:: - Shows the pin default config values to override the BIOS setup. - Writing this (with two numbers, NID and value) appends the new - value. The given will be used instead of the initial BIOS value at - the next reconfiguration time. -cur_pin_configs:: +driver_pin_configs:: Shows the pin default values set by the codec parser explicitly. This doesn't show all pin values but only the changed values by the parser. That is, if the parser doesn't change the pin default config values by itself, this will contain nothing. +user_pin_configs:: + Shows the pin default config values to override the BIOS setup. + Writing this (with two numbers, NID and value) appends the new + value. The given will be used instead of the initial BIOS value at + the next reconfiguration time. reconfig:: Triggers the codec re-configuration. When any value is written to this file, the driver re-initialize and parses the codec tree diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 8ec2dfca9a6c..df9453d0122e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -755,7 +755,7 @@ int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list, int snd_hda_codec_set_pincfg(struct hda_codec *codec, hda_nid_t nid, unsigned int cfg) { - return snd_hda_add_pincfg(codec, &codec->cur_pins, nid, cfg); + return snd_hda_add_pincfg(codec, &codec->driver_pins, nid, cfg); } EXPORT_SYMBOL_HDA(snd_hda_codec_set_pincfg); @@ -764,11 +764,11 @@ unsigned int snd_hda_codec_get_pincfg(struct hda_codec *codec, hda_nid_t nid) { struct hda_pincfg *pin; - pin = look_up_pincfg(codec, &codec->cur_pins, nid); + pin = look_up_pincfg(codec, &codec->driver_pins, nid); if (pin) return pin->cfg; #ifdef CONFIG_SND_HDA_HWDEP - pin = look_up_pincfg(codec, &codec->override_pins, nid); + pin = look_up_pincfg(codec, &codec->user_pins, nid); if (pin) return pin->cfg; #endif @@ -797,12 +797,12 @@ static void free_hda_cache(struct hda_cache_rec *cache); /* restore the initial pin cfgs and release all pincfg lists */ static void restore_init_pincfgs(struct hda_codec *codec) { - /* first free cur_pins and override_pins, then call restore_pincfg + /* first free driver_pins and user_pins, then call restore_pincfg * so that only the values in init_pins are restored */ - snd_array_free(&codec->cur_pins); + snd_array_free(&codec->driver_pins); #ifdef CONFIG_SND_HDA_HWDEP - snd_array_free(&codec->override_pins); + snd_array_free(&codec->user_pins); #endif restore_pincfgs(codec); snd_array_free(&codec->init_pins); @@ -874,7 +874,7 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); snd_array_init(&codec->mixers, sizeof(struct snd_kcontrol *), 32); snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); - snd_array_init(&codec->cur_pins, sizeof(struct hda_pincfg), 16); + snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16); if (codec->bus->modelname) { codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL); if (!codec->modelname) { @@ -1463,8 +1463,8 @@ void snd_hda_codec_reset(struct hda_codec *codec) free_hda_cache(&codec->cmd_cache); init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); - /* free only cur_pins so that init_pins + override_pins are restored */ - snd_array_free(&codec->cur_pins); + /* free only driver_pins so that init_pins + user_pins are restored */ + snd_array_free(&codec->driver_pins); restore_pincfgs(codec); codec->num_pcms = 0; codec->pcm_info = NULL; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 6d01a8058f0a..2ea628478a91 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -779,13 +779,13 @@ struct hda_codec { unsigned int spdif_in_enable; /* SPDIF input enable? */ hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */ struct snd_array init_pins; /* initial (BIOS) pin configurations */ - struct snd_array cur_pins; /* current pin configurations */ + struct snd_array driver_pins; /* pin configs set by codec parser */ #ifdef CONFIG_SND_HDA_HWDEP struct snd_hwdep *hwdep; /* assigned hwdep device */ struct snd_array init_verbs; /* additional init verbs */ struct snd_array hints; /* additional hints */ - struct snd_array override_pins; /* default pin configs to override */ + struct snd_array user_pins; /* default pin configs to override */ #endif /* misc flags */ diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 71039a6dec28..c660383ef381 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -109,7 +109,7 @@ static void clear_hwdep_elements(struct hda_codec *codec) for (i = 0; i < codec->hints.used; i++, head++) kfree(*head); snd_array_free(&codec->hints); - snd_array_free(&codec->override_pins); + snd_array_free(&codec->user_pins); } static void hwdep_free(struct snd_hwdep *hwdep) @@ -142,7 +142,7 @@ int /*__devinit*/ snd_hda_create_hwdep(struct hda_codec *codec) snd_array_init(&codec->init_verbs, sizeof(struct hda_verb), 32); snd_array_init(&codec->hints, sizeof(char *), 32); - snd_array_init(&codec->override_pins, sizeof(struct hda_pincfg), 16); + snd_array_init(&codec->user_pins, sizeof(struct hda_pincfg), 16); return 0; } @@ -340,29 +340,29 @@ static ssize_t init_pin_configs_show(struct device *dev, return pin_configs_show(codec, &codec->init_pins, buf); } -static ssize_t override_pin_configs_show(struct device *dev, - struct device_attribute *attr, - char *buf) +static ssize_t user_pin_configs_show(struct device *dev, + struct device_attribute *attr, + char *buf) { struct snd_hwdep *hwdep = dev_get_drvdata(dev); struct hda_codec *codec = hwdep->private_data; - return pin_configs_show(codec, &codec->override_pins, buf); + return pin_configs_show(codec, &codec->user_pins, buf); } -static ssize_t cur_pin_configs_show(struct device *dev, - struct device_attribute *attr, - char *buf) +static ssize_t driver_pin_configs_show(struct device *dev, + struct device_attribute *attr, + char *buf) { struct snd_hwdep *hwdep = dev_get_drvdata(dev); struct hda_codec *codec = hwdep->private_data; - return pin_configs_show(codec, &codec->cur_pins, buf); + return pin_configs_show(codec, &codec->driver_pins, buf); } #define MAX_PIN_CONFIGS 32 -static ssize_t override_pin_configs_store(struct device *dev, - struct device_attribute *attr, - const char *buf, size_t count) +static ssize_t user_pin_configs_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) { struct snd_hwdep *hwdep = dev_get_drvdata(dev); struct hda_codec *codec = hwdep->private_data; @@ -373,7 +373,7 @@ static ssize_t override_pin_configs_store(struct device *dev, return -EINVAL; if (!nid) return -EINVAL; - err = snd_hda_add_pincfg(codec, &codec->override_pins, nid, cfg); + err = snd_hda_add_pincfg(codec, &codec->user_pins, nid, cfg); if (err < 0) return err; return count; @@ -397,8 +397,8 @@ static struct device_attribute codec_attrs[] = { CODEC_ATTR_WO(init_verbs), CODEC_ATTR_WO(hints), CODEC_ATTR_RO(init_pin_configs), - CODEC_ATTR_RW(override_pin_configs), - CODEC_ATTR_RO(cur_pin_configs), + CODEC_ATTR_RW(user_pin_configs), + CODEC_ATTR_RO(driver_pin_configs), CODEC_ATTR_WO(reconfig), CODEC_ATTR_WO(clear), }; -- cgit v1.2.3-59-g8ed1b From 5e7b8e0d87091ae21b291588817b5359a5e00795 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Feb 2009 09:45:59 +0100 Subject: ALSA: hda - Make user_pin overriding the driver setup Make user_pin overriding even the driver pincfg, e.g. the static / fixed pin config table in patch_sigmatel.c. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio.txt | 3 ++- sound/pci/hda/hda_codec.c | 16 ++++++++++++---- 2 files changed, 14 insertions(+), 5 deletions(-) (limited to 'sound/pci') diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index f590850c149f..a4e5ef87af6b 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -380,7 +380,8 @@ user_pin_configs:: Shows the pin default config values to override the BIOS setup. Writing this (with two numbers, NID and value) appends the new value. The given will be used instead of the initial BIOS value at - the next reconfiguration time. + the next reconfiguration time. Note that this config will override + even the driver pin configs, too. reconfig:: Triggers the codec re-configuration. When any value is written to this file, the driver re-initialize and parses the codec tree diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index df9453d0122e..a13480fa8e74 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -739,7 +739,9 @@ int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list, hda_nid_t nid, unsigned int cfg) { struct hda_pincfg *pin; + unsigned int oldcfg; + oldcfg = snd_hda_codec_get_pincfg(codec, nid); pin = look_up_pincfg(codec, list, nid); if (!pin) { pin = snd_array_new(list); @@ -748,7 +750,13 @@ int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list, pin->nid = nid; } pin->cfg = cfg; - set_pincfg(codec, nid, cfg); + + /* change only when needed; e.g. if the pincfg is already present + * in user_pins[], don't write it + */ + cfg = snd_hda_codec_get_pincfg(codec, nid); + if (oldcfg != cfg) + set_pincfg(codec, nid, cfg); return 0; } @@ -764,14 +772,14 @@ unsigned int snd_hda_codec_get_pincfg(struct hda_codec *codec, hda_nid_t nid) { struct hda_pincfg *pin; - pin = look_up_pincfg(codec, &codec->driver_pins, nid); - if (pin) - return pin->cfg; #ifdef CONFIG_SND_HDA_HWDEP pin = look_up_pincfg(codec, &codec->user_pins, nid); if (pin) return pin->cfg; #endif + pin = look_up_pincfg(codec, &codec->driver_pins, nid); + if (pin) + return pin->cfg; pin = look_up_pincfg(codec, &codec->init_pins, nid); if (pin) return pin->cfg; -- cgit v1.2.3-59-g8ed1b From 13c989beba166b470b1e6b0fa117148bcbfa3dd1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Feb 2009 11:33:34 +0100 Subject: ALSA: hda - Don't give over 0dB volume for AD1984A HP laptops Set the upper limit 0dB to the volume of mixer amp 0x20 for AD1984A HP laptops. The overloaded volume may damage the internal speaker. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 2c58d7b05aba..b16802841468 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3986,6 +3986,14 @@ static int patch_ad1884a(struct hda_codec *codec) spec->multiout.dig_out_nid = 0; codec->patch_ops.unsol_event = ad1884a_hp_unsol_event; codec->patch_ops.init = ad1884a_hp_init; + /* set the upper-limit for mixer amp to 0dB for avoiding the + * possible damage by overloading + */ + snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT, + (0x17 << AC_AMPCAP_OFFSET_SHIFT) | + (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (1 << AC_AMPCAP_MUTE_SHIFT)); break; case AD1884A_THINKPAD: spec->mixers[0] = ad1984a_thinkpad_mixers; -- cgit v1.2.3-59-g8ed1b From a65d629ceb4cff5e7d5edadfd6bf1f64c370a517 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Feb 2009 16:57:04 +0100 Subject: ALSA: hda - Add pseudo device-locking for clear/reconfig Added the pseudo device-locking using card->shutdown flag to avoid the crash via clear/reconfig during operations. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 54 +++++++++++++++++++++++++++++++++++++++++++---- sound/pci/hda/hda_hwdep.c | 15 +++++++++++-- sound/pci/hda/hda_local.h | 2 +- 3 files changed, 64 insertions(+), 7 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index a13480fa8e74..5dceee8a113b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1445,9 +1445,52 @@ void snd_hda_ctls_clear(struct hda_codec *codec) snd_array_free(&codec->mixers); } -void snd_hda_codec_reset(struct hda_codec *codec) +/* pseudo device locking + * toggle card->shutdown to allow/disallow the device access (as a hack) + */ +static int hda_lock_devices(struct snd_card *card) { - int i; + spin_lock(&card->files_lock); + if (card->shutdown) { + spin_unlock(&card->files_lock); + return -EINVAL; + } + card->shutdown = 1; + spin_unlock(&card->files_lock); + return 0; +} + +static void hda_unlock_devices(struct snd_card *card) +{ + spin_lock(&card->files_lock); + card->shutdown = 0; + spin_unlock(&card->files_lock); +} + +int snd_hda_codec_reset(struct hda_codec *codec) +{ + struct snd_card *card = codec->bus->card; + int i, pcm; + + if (hda_lock_devices(card) < 0) + return -EBUSY; + /* check whether the codec isn't used by any mixer or PCM streams */ + if (!list_empty(&card->ctl_files)) { + hda_unlock_devices(card); + return -EBUSY; + } + for (pcm = 0; pcm < codec->num_pcms; pcm++) { + struct hda_pcm *cpcm = &codec->pcm_info[pcm]; + if (!cpcm->pcm) + continue; + if (cpcm->pcm->streams[0].substream_opened || + cpcm->pcm->streams[1].substream_opened) { + hda_unlock_devices(card); + return -EBUSY; + } + } + + /* OK, let it free */ #ifdef CONFIG_SND_HDA_POWER_SAVE cancel_delayed_work(&codec->power_work); @@ -1457,8 +1500,7 @@ void snd_hda_codec_reset(struct hda_codec *codec) /* relase PCMs */ for (i = 0; i < codec->num_pcms; i++) { if (codec->pcm_info[i].pcm) { - snd_device_free(codec->bus->card, - codec->pcm_info[i].pcm); + snd_device_free(card, codec->pcm_info[i].pcm); clear_bit(codec->pcm_info[i].device, codec->bus->pcm_dev_bits); } @@ -1479,6 +1521,10 @@ void snd_hda_codec_reset(struct hda_codec *codec) codec->preset = NULL; module_put(codec->owner); codec->owner = NULL; + + /* allow device access again */ + hda_unlock_devices(card); + return 0; } #endif /* CONFIG_SND_HDA_RECONFIG */ diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index c660383ef381..4af484b8240c 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -155,7 +155,13 @@ int /*__devinit*/ snd_hda_create_hwdep(struct hda_codec *codec) static int clear_codec(struct hda_codec *codec) { - snd_hda_codec_reset(codec); + int err; + + err = snd_hda_codec_reset(codec); + if (err < 0) { + snd_printk(KERN_ERR "The codec is being used, can't free.\n"); + return err; + } clear_hwdep_elements(codec); return 0; } @@ -165,7 +171,12 @@ static int reconfig_codec(struct hda_codec *codec) int err; snd_printk(KERN_INFO "hda-codec: reconfiguring\n"); - snd_hda_codec_reset(codec); + err = snd_hda_codec_reset(codec); + if (err < 0) { + snd_printk(KERN_ERR + "The codec is being used, can't reconfigure.\n"); + return err; + } err = snd_hda_codec_configure(codec); if (err < 0) return err; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 84e2cf644fd7..4bd82a37a4c8 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -98,7 +98,7 @@ 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 **slaves); -void snd_hda_codec_reset(struct hda_codec *codec); +int snd_hda_codec_reset(struct hda_codec *codec); int snd_hda_codec_configure(struct hda_codec *codec); /* amp value bits */ -- cgit v1.2.3-59-g8ed1b From 873dc78a8676b7ba6260b1d74c50d8ea5025ecbe Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Feb 2009 18:12:13 +0100 Subject: ALSA: hda - Clean up / fix quirks for HP laptops with AD1984A Use SND_PCI_QUIRK_MASK() to clean up / support better HP laptops with AD1984A codec. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 0253cb93aa7f..5bb48ee8b6c6 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3923,9 +3923,8 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE), SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x3070, "HP", AD1884A_MOBILE), - SND_PCI_QUIRK(0x103c, 0x30e6, "HP 6730b", AD1884A_LAPTOP), - SND_PCI_QUIRK(0x103c, 0x30e7, "HP EliteBook 8530p", AD1884A_LAPTOP), - SND_PCI_QUIRK(0x103c, 0x3614, "HP 6730s", AD1884A_LAPTOP), + SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30e0, "HP laptop", AD1884A_LAPTOP), + SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP), SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD), {} }; -- cgit v1.2.3-59-g8ed1b From f872a9194cb006994d47a58efc875218594e6072 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Feb 2009 00:57:01 +0100 Subject: ALSA: hda - Clean up / fix quirk for Sony laptops with ALC262 Clean up / fix quirk entries for Sony laptops with ALC262 codec using NSD_PCI_QUIRK_MASK(). This also fixes the kernel bug #12780 http://bugme.linux-foundation.org/show_bug.cgi?id=12780 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 50ae8f33af5f..d670d33cfa17 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10824,10 +10824,8 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x8203, "Sony UX-90", ALC262_HIPPO), SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD), - SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD), - SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD), - SND_PCI_QUIRK(0x104d, 0x9033, "Sony VAIO VGN-SR19XN", - ALC262_SONY_ASSAMD), + SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO", + ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", ALC262_TOSHIBA_RX1), SND_PCI_QUIRK(0x1179, 0xff7b, "Toshiba S06", ALC262_TOSHIBA_S06), -- cgit v1.2.3-59-g8ed1b From 5d9b6c07831456b7a7d90eac31c853d60eaf8ab6 Mon Sep 17 00:00:00 2001 From: Hannes Eder Date: Wed, 25 Feb 2009 22:28:45 +0100 Subject: ALSA: sound/pci/hda: fix sparse warning: different signedness Fix this sparse warning: sound/pci/hda/hda_codec.c:1544:19: warning: incorrect type in assignment (different signedness) sound/pci/hda/hda_codec.c:1544:19: expected unsigned long *vals sound/pci/hda/hda_codec.c:1544:19: got long * Signed-off-by: Hannes Eder Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_local.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 4bd82a37a4c8..03ee9dd04913 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -136,7 +136,7 @@ extern struct hda_ctl_ops snd_hda_bind_sw; /* for bind-switch */ struct hda_bind_ctls { struct hda_ctl_ops *ops; - long values[]; + unsigned long values[]; }; int snd_hda_mixer_bind_ctls_info(struct snd_kcontrol *kcontrol, -- cgit v1.2.3-59-g8ed1b From 23f0c048ba59ad5c2f3fd85ed98360b631dbf6f8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Feb 2009 13:03:58 +0100 Subject: ALSA: hda - Clean up the input pin setup in automatic mode Clean up the input-pin setup in automatic mode in patch_realtek.c. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 62 ++++++++++++++++++------------------------- 1 file changed, 26 insertions(+), 36 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d670d33cfa17..b3406302d069 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -760,6 +760,24 @@ static int alc_eapd_ctrl_put(struct snd_kcontrol *kcontrol, .private_value = nid | (mask<<16) } #endif /* CONFIG_SND_DEBUG */ +/* + * set up the input pin config (depending on the given auto-pin type) + */ +static void alc_set_input_pin(struct hda_codec *codec, hda_nid_t nid, + int auto_pin_type) +{ + unsigned int val = PIN_IN; + + if (auto_pin_type <= AUTO_PIN_FRONT_MIC) { + unsigned int pincap; + pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT; + if (pincap & AC_PINCAP_VREF_80) + val = PIN_VREF80; + } + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, val); +} + /* */ static void add_mixer(struct alc_spec *spec, struct snd_kcontrol_new *mix) @@ -4188,10 +4206,7 @@ static void alc880_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; if (alc880_is_input_pin(nid)) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - i <= AUTO_PIN_FRONT_MIC ? - PIN_VREF80 : PIN_IN); + alc_set_input_pin(codec, nid, i); if (nid != ALC880_PIN_CD_NID) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, @@ -5657,10 +5672,7 @@ static void alc260_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; if (nid >= 0x12) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - i <= AUTO_PIN_FRONT_MIC ? - PIN_VREF80 : PIN_IN); + alc_set_input_pin(codec, nid, i); if (nid != ALC260_PIN_CD_NID) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, @@ -7006,16 +7018,7 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec) unsigned int vref; if (!nid) continue; - vref = PIN_IN; - if (1 /*i <= AUTO_PIN_FRONT_MIC*/) { - unsigned int pincap; - pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); - if ((pincap >> AC_PINCAP_VREF_SHIFT) & - AC_PINCAP_VREF_80) - vref = PIN_VREF80; - } - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, vref); + alc_set_input_pin(codec, nid, AUTO_PIN_FRONT_MIC /*i*/); if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, @@ -9100,10 +9103,7 @@ static void alc883_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; if (alc883_is_input_pin(nid)) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - (i <= AUTO_PIN_FRONT_MIC ? - PIN_VREF80 : PIN_IN)); + alc_set_input_pin(codec, nid, i); if (nid != ALC883_PIN_CD_NID) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, @@ -13831,12 +13831,8 @@ static void alc861_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; - if (nid >= 0x0c && nid <= 0x11) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - i <= AUTO_PIN_FRONT_MIC ? - PIN_VREF80 : PIN_IN); - } + if (nid >= 0x0c && nid <= 0x11) + alc_set_input_pin(codec, nid, i); } } @@ -14803,10 +14799,7 @@ static void alc861vd_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; if (alc861vd_is_input_pin(nid)) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - i <= AUTO_PIN_FRONT_MIC ? - PIN_VREF80 : PIN_IN); + alc_set_input_pin(codec, nid, i); if (nid != ALC861VD_PIN_CD_NID) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, @@ -16732,10 +16725,7 @@ static void alc662_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; if (alc662_is_input_pin(nid)) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - (i <= AUTO_PIN_FRONT_MIC ? - PIN_VREF80 : PIN_IN)); + alc_set_input_pin(codec, nid, i); if (nid != ALC662_PIN_CD_NID) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, -- cgit v1.2.3-59-g8ed1b From 1607b8ea0a4cc20752978fadb027daafc8a2d93c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Feb 2009 16:50:43 +0100 Subject: ALSA: hda - Add model=auto for STAC/IDT codecs Added the model=auto to STAC/IDT codecs to use the BIOS default setup explicitly. It can be used to disable the device-specific model quirk in the driver. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 8 ++++++++ sound/pci/hda/patch_sigmatel.c | 16 ++++++++++++++++ 2 files changed, 24 insertions(+) (limited to 'sound/pci') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 0e52d273ce96..a448bbefd488 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -280,6 +280,7 @@ STAC9200 gateway-m4 Gateway laptops with EAPD control gateway-m4-2 Gateway laptops with EAPD control panasonic Panasonic CF-74 + auto BIOS setup (default) STAC9205/9254 ============= @@ -288,6 +289,7 @@ STAC9205/9254 dell-m43 Dell Precision dell-m44 Dell Inspiron eapd Keep EAPD on (e.g. Gateway T1616) + auto BIOS setup (default) STAC9220/9221 ============= @@ -311,6 +313,7 @@ STAC9220/9221 dell-d82 Dell (unknown) dell-m81 Dell (unknown) dell-m82 Dell XPS M1210 + auto BIOS setup (default) STAC9202/9250/9251 ================== @@ -322,6 +325,7 @@ STAC9202/9250/9251 m3 Some Gateway MX series laptops m5 Some Gateway MX series laptops (MP6954) m6 Some Gateway NX series laptops + auto BIOS setup (default) STAC9227/9228/9229/927x ======================= @@ -331,6 +335,7 @@ STAC9227/9228/9229/927x 5stack D965 5stack + SPDIF dell-3stack Dell Dimension E520 dell-bios Fixes with Dell BIOS setup + auto BIOS setup (default) STAC92HD71B* ============ @@ -339,6 +344,7 @@ STAC92HD71B* dell-m4-2 Dell desktops dell-m4-3 Dell desktops hp-m4 HP dv laptops + auto BIOS setup (default) STAC92HD73* =========== @@ -348,11 +354,13 @@ STAC92HD73* dell-m6-dmic Dell desktops/laptops with digital mics dell-m6 Dell desktops/laptops with both type of mics dell-eq Dell desktops/laptops + auto BIOS setup (default) STAC92HD83* =========== ref Reference board mic-ref Reference board with power managment for ports + auto BIOS setup (default) STAC9872 ======== diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index da48d8c0b295..37ffd96a9ff2 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -43,6 +43,7 @@ enum { }; enum { + STAC_AUTO, STAC_REF, STAC_9200_OQO, STAC_9200_DELL_D21, @@ -62,6 +63,7 @@ enum { }; enum { + STAC_9205_AUTO, STAC_9205_REF, STAC_9205_DELL_M42, STAC_9205_DELL_M43, @@ -71,6 +73,7 @@ enum { }; enum { + STAC_92HD73XX_AUTO, STAC_92HD73XX_NO_JD, /* no jack-detection */ STAC_92HD73XX_REF, STAC_DELL_M6_AMIC, @@ -81,6 +84,7 @@ enum { }; enum { + STAC_92HD83XXX_AUTO, STAC_92HD83XXX_REF, STAC_92HD83XXX_PWR_REF, STAC_DELL_S14, @@ -88,6 +92,7 @@ enum { }; enum { + STAC_92HD71BXX_AUTO, STAC_92HD71BXX_REF, STAC_DELL_M4_1, STAC_DELL_M4_2, @@ -98,6 +103,7 @@ enum { }; enum { + STAC_925x_AUTO, STAC_925x_REF, STAC_M1, STAC_M1_2, @@ -110,6 +116,7 @@ enum { }; enum { + STAC_922X_AUTO, STAC_D945_REF, STAC_D945GTP3, STAC_D945GTP5, @@ -137,6 +144,7 @@ enum { }; enum { + STAC_927X_AUTO, STAC_D965_REF_NO_JD, /* no jack-detection */ STAC_D965_REF, STAC_D965_3ST, @@ -1488,6 +1496,7 @@ static unsigned int *stac9200_brd_tbl[STAC_9200_MODELS] = { }; static const char *stac9200_models[STAC_9200_MODELS] = { + [STAC_AUTO] = "auto", [STAC_REF] = "ref", [STAC_9200_OQO] = "oqo", [STAC_9200_DELL_D21] = "dell-d21", @@ -1633,6 +1642,7 @@ static unsigned int *stac925x_brd_tbl[STAC_925x_MODELS] = { }; static const char *stac925x_models[STAC_925x_MODELS] = { + [STAC_925x_AUTO] = "auto", [STAC_REF] = "ref", [STAC_M1] = "m1", [STAC_M1_2] = "m1-2", @@ -1692,6 +1702,7 @@ static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = { }; static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = { + [STAC_92HD73XX_AUTO] = "auto", [STAC_92HD73XX_NO_JD] = "no-jd", [STAC_92HD73XX_REF] = "ref", [STAC_DELL_M6_AMIC] = "dell-m6-amic", @@ -1748,6 +1759,7 @@ static unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = { }; static const char *stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { + [STAC_92HD83XXX_AUTO] = "auto", [STAC_92HD83XXX_REF] = "ref", [STAC_92HD83XXX_PWR_REF] = "mic-ref", [STAC_DELL_S14] = "dell-s14", @@ -1802,6 +1814,7 @@ static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { }; static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { + [STAC_92HD71BXX_AUTO] = "auto", [STAC_92HD71BXX_REF] = "ref", [STAC_DELL_M4_1] = "dell-m4-1", [STAC_DELL_M4_2] = "dell-m4-2", @@ -1973,6 +1986,7 @@ static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = { }; static const char *stac922x_models[STAC_922X_MODELS] = { + [STAC_922X_AUTO] = "auto", [STAC_D945_REF] = "ref", [STAC_D945GTP5] = "5stack", [STAC_D945GTP3] = "3stack", @@ -2125,6 +2139,7 @@ static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = { }; static const char *stac927x_models[STAC_927X_MODELS] = { + [STAC_927X_AUTO] = "auto", [STAC_D965_REF_NO_JD] = "ref-no-jd", [STAC_D965_REF] = "ref", [STAC_D965_3ST] = "3stack", @@ -2222,6 +2237,7 @@ static unsigned int *stac9205_brd_tbl[STAC_9205_MODELS] = { }; static const char *stac9205_models[STAC_9205_MODELS] = { + [STAC_9205_AUTO] = "auto", [STAC_9205_REF] = "ref", [STAC_9205_DELL_M42] = "dell-m42", [STAC_9205_DELL_M43] = "dell-m43", -- cgit v1.2.3-59-g8ed1b From bedfcebb4fb33fc9ebd395462e72afa103db0bec Mon Sep 17 00:00:00 2001 From: peerchen Date: Fri, 27 Feb 2009 17:03:19 +0800 Subject: ALSA: hda - Add the Device IDs for MCP89 and remove IDs of MCP7B Added the Device IDs for MCP89 HD audio controller. Removed the IDs of MCP7B cause this chipset had been cancelled. Signed-off-by: Peer Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c5a5dc5698a2..47a5833feb7a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2454,10 +2454,10 @@ static struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x10de, 0x0ac1), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0ac2), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0ac3), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0bd4), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0bd5), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0bd6), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0bd7), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0d94), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0d95), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0d96), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0d97), .driver_data = AZX_DRIVER_NVIDIA }, /* Teradici */ { PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA }, /* AMD Generic, PCI class code and Vendor ID for HD Audio */ -- cgit v1.2.3-59-g8ed1b From 53eff7e1e0de1cde8e8cbe619f401d2578dde946 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Feb 2009 17:49:44 +0100 Subject: ALSA: hda - Match all 103c:17xx devices for HP BPC model Use SND_PCI_QUIRK_MASK() to match all devices with 103c:17xx for HP BPC model. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e72b74efc697..0b4afa0a3519 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10807,7 +10807,8 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { ALC262_HP_BPC), SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1300, "HP xw series", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x170b, "HP xw*", ALC262_HP_BPC), + SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1700, "HP xw series", + ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL), SND_PCI_QUIRK(0x103c, 0x2801, "HP D7000", ALC262_HP_BPC_D7000_WF), SND_PCI_QUIRK(0x103c, 0x2802, "HP D7000", ALC262_HP_BPC_D7000_WL), -- cgit v1.2.3-59-g8ed1b From c82c8abdeef53eb0bb0504becb4e91bbccceaee8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Feb 2009 17:52:22 +0100 Subject: ALSA: hda - Fix an "unused variable" compile warning MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Forgot to remove an unused variable. sound/pci/hda/patch_realtek.c: In function ‘alc882_auto_init_analog_input’: sound/pci/hda/patch_realtek.c:7018: warning: unused variable ‘vref’ Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0b4afa0a3519..1cc31ac03522 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7015,7 +7015,6 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; - unsigned int vref; if (!nid) continue; alc_set_input_pin(codec, nid, AUTO_PIN_FRONT_MIC /*i*/); -- cgit v1.2.3-59-g8ed1b From 892981ffbe9a5c4cbc9d75f423b145f32c765f9c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Mar 2009 08:04:35 +0100 Subject: ALSA: hda - Don't create a beep control for digital-only ALC268 When an ALC268 codec is set up as the digital-only (as found in Toshiba laptops), it shouldn't contain any beep control that conflict with the primary codec. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1cc31ac03522..c60c86acd9ba 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11915,7 +11915,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec) if (spec->kctls.list) add_mixer(spec, spec->kctls.list); - if (spec->autocfg.speaker_pins[0] != 0x1d) + if (!spec->no_analog && spec->autocfg.speaker_pins[0] != 0x1d) add_mixer(spec, alc268_beep_mixer); add_verb(spec, alc268_volume_init_verbs); -- cgit v1.2.3-59-g8ed1b From 4c4531d64dd0442813c7307b860bf40a2aec51bc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Mar 2009 08:06:11 +0100 Subject: ALSA: hda - Remove Toshiba probe_mask quirk Revert the Toshiba probe_mask quirk for 2.6.29 kernel (commit 38f1df27e3191d76e983cb9c6b4392582fd32fda). In the current tree, the digital-only codec is handled properly so no codec conflict should occur. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 68a128fb487d..47a5833feb7a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2095,8 +2095,6 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = { SND_PCI_QUIRK(0x1028, 0x20ac, "Dell Studio Desktop", 0x01), /* including bogus ALC268 in slot#2 that conflicts with ALC888 */ SND_PCI_QUIRK(0x17c0, 0x4085, "Medion MD96630", 0x01), - /* conflict of ALC268 in slot#3 (digital I/O); a temporary fix */ - SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba laptop", 0x03), /* forced codec slots */ SND_PCI_QUIRK(0x1046, 0x1262, "ASUS W5F", 0x103), {} -- cgit v1.2.3-59-g8ed1b From d1f1af2dbf8207db590853a59bec465c4f68cfdc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Mar 2009 10:35:29 +0100 Subject: ALSA: hda - Intialize more codec fields in snd_hda_codec_reset() Initiailize forgotten fields in snd_hda_codec_reset(). Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5dceee8a113b..3b44c789f23b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1519,6 +1519,9 @@ int snd_hda_codec_reset(struct hda_codec *codec) codec->num_pcms = 0; codec->pcm_info = NULL; codec->preset = NULL; + memset(&codec->patch_ops, 0, sizeof(codec->patch_ops)); + codec->slave_dig_outs = NULL; + codec->spdif_status_reset = 0; module_put(codec->owner); codec->owner = NULL; -- cgit v1.2.3-59-g8ed1b From f93d461bcde6ac3db542361c00a7e4167f88176d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Mar 2009 10:44:15 +0100 Subject: ALSA: hda - Revert the codec probe at control-creation errors Revert the codec probe instead of returning the error to the driver when any error occurs at creating the control elements. The control element conflict can be non-fatal in many cases, especially if it comes from the digital-only codec. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3b44c789f23b..1be34ed9c0eb 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1434,7 +1434,6 @@ int snd_hda_ctl_add(struct hda_codec *codec, struct snd_kcontrol *kctl) } EXPORT_SYMBOL_HDA(snd_hda_ctl_add); -#ifdef CONFIG_SND_HDA_RECONFIG /* Clear all controls assigned to the given codec */ void snd_hda_ctls_clear(struct hda_codec *codec) { @@ -1529,7 +1528,6 @@ int snd_hda_codec_reset(struct hda_codec *codec) hda_unlock_devices(card); return 0; } -#endif /* CONFIG_SND_HDA_RECONFIG */ /* create a virtual master control and add slaves */ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, @@ -2392,8 +2390,16 @@ int /*__devinit*/ snd_hda_build_controls(struct hda_bus *bus) list_for_each_entry(codec, &bus->codec_list, list) { int err = snd_hda_codec_build_controls(codec); - if (err < 0) - return err; + if (err < 0) { + printk(KERN_ERR "hda_codec: cannot build controls" + "for #%d (error %d)\n", codec->addr, err); + err = snd_hda_codec_reset(codec); + if (err < 0) { + printk(KERN_ERR + "hda_codec: cannot revert codec\n"); + return err; + } + } } return 0; } -- cgit v1.2.3-59-g8ed1b From 6e655bf21697d2594243098a14e0699e8d4a4059 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Mar 2009 10:46:03 +0100 Subject: ALSA: hda - Don't return a fatal error at PCM-creation errors Don't return a fatal error to the driver but continue to probe when any error occurs at creating PCM streams for each codec. It's often non-fatal and keeping it would help debugging. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 22 +++++++++++++++++----- 1 file changed, 17 insertions(+), 5 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 1be34ed9c0eb..7c9ef5c18e77 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2833,8 +2833,16 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec) if (!codec->patch_ops.build_pcms) return 0; err = codec->patch_ops.build_pcms(codec); - if (err < 0) - return err; + if (err < 0) { + printk(KERN_ERR "hda_codec: cannot build PCMs" + "for #%d (error %d)\n", codec->addr, err); + err = snd_hda_codec_reset(codec); + if (err < 0) { + printk(KERN_ERR + "hda_codec: cannot revert codec\n"); + return err; + } + } } for (pcm = 0; pcm < codec->num_pcms; pcm++) { struct hda_pcm *cpcm = &codec->pcm_info[pcm]; @@ -2846,11 +2854,15 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec) if (!cpcm->pcm) { dev = get_empty_pcm_device(codec->bus, cpcm->pcm_type); if (dev < 0) - return 0; + continue; /* no fatal error */ cpcm->device = dev; err = snd_hda_attach_pcm(codec, cpcm); - if (err < 0) - return err; + if (err < 0) { + printk(KERN_ERR "hda_codec: cannot attach " + "PCM stream %d for codec #%d\n", + dev, codec->addr); + continue; /* no fatal error */ + } } } return 0; -- cgit v1.2.3-59-g8ed1b From 43b62713f67d9f0655f3a61f5bd14d6297ddd3ce Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Mar 2009 14:25:17 +0100 Subject: ALSA: hda - Add hint string helper functions Added snd_hda_get_hint() and snd_hda_get_bool_hint() helper functions to retrieve a hint value. Internally, the hint is stored in a pair of two strings, key and val. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_hwdep.c | 112 ++++++++++++++++++++++++++++++++++++++++------ sound/pci/hda/hda_local.h | 17 +++++++ 2 files changed, 116 insertions(+), 13 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 4af484b8240c..5e554de9cd9b 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -30,6 +30,12 @@ #include #include +/* hint string pair */ +struct hda_hint { + const char *key; + const char *val; /* contained in the same alloc as key */ +}; + /* * write/read an out-of-bound verb */ @@ -99,15 +105,15 @@ static int hda_hwdep_open(struct snd_hwdep *hw, struct file *file) static void clear_hwdep_elements(struct hda_codec *codec) { - char **head; int i; /* clear init verbs */ snd_array_free(&codec->init_verbs); /* clear hints */ - head = codec->hints.list; - for (i = 0; i < codec->hints.used; i++, head++) - kfree(*head); + for (i = 0; i < codec->hints.used; i++) { + struct hda_hint *hint = snd_array_elem(&codec->hints, i); + kfree(hint->key); /* we don't need to free hint->val */ + } snd_array_free(&codec->hints); snd_array_free(&codec->user_pins); } @@ -141,7 +147,7 @@ int /*__devinit*/ snd_hda_create_hwdep(struct hda_codec *codec) #endif snd_array_init(&codec->init_verbs, sizeof(struct hda_verb), 32); - snd_array_init(&codec->hints, sizeof(char *), 32); + snd_array_init(&codec->hints, sizeof(struct hda_hint), 32); snd_array_init(&codec->user_pins, sizeof(struct hda_pincfg), 16); return 0; @@ -306,26 +312,81 @@ static ssize_t init_verbs_store(struct device *dev, return count; } +static struct hda_hint *get_hint(struct hda_codec *codec, const char *key) +{ + int i; + + for (i = 0; i < codec->hints.used; i++) { + struct hda_hint *hint = snd_array_elem(&codec->hints, i); + if (!strcmp(hint->key, key)) + return hint; + } + return NULL; +} + +static void remove_trail_spaces(char *str) +{ + char *p; + if (!*str) + return; + p = str + strlen(str) - 1; + for (; isspace(*p); p--) { + *p = 0; + if (p == str) + return; + } +} + +#define MAX_HINTS 1024 + static ssize_t hints_store(struct device *dev, struct device_attribute *attr, const char *buf, size_t count) { struct snd_hwdep *hwdep = dev_get_drvdata(dev); struct hda_codec *codec = hwdep->private_data; - char *p; - char **hint; + char *key, *val; + struct hda_hint *hint; - if (!*buf || isspace(*buf) || *buf == '#' || *buf == '\n') + while (isspace(*buf)) + buf++; + if (!*buf || *buf == '#' || *buf == '\n') return count; - p = kstrndup_noeol(buf, 1024); - if (!p) + if (*buf == '=') + return -EINVAL; + key = kstrndup_noeol(buf, 1024); + if (!key) return -ENOMEM; - hint = snd_array_new(&codec->hints); + /* extract key and val */ + val = strchr(key, '='); + if (!val) { + kfree(key); + return -EINVAL; + } + *val++ = 0; + while (isspace(*val)) + val++; + remove_trail_spaces(key); + remove_trail_spaces(val); + hint = get_hint(codec, key); + if (hint) { + /* replace */ + kfree(hint->key); + hint->key = key; + hint->val = val; + return count; + } + /* allocate a new hint entry */ + if (codec->hints.used >= MAX_HINTS) + hint = NULL; + else + hint = snd_array_new(&codec->hints); if (!hint) { - kfree(p); + kfree(key); return -ENOMEM; } - *hint = p; + hint->key = key; + hint->val = val; return count; } @@ -428,4 +489,29 @@ int snd_hda_hwdep_add_sysfs(struct hda_codec *codec) return 0; } +/* + * Look for hint string + */ +const char *snd_hda_get_hint(struct hda_codec *codec, const char *key) +{ + struct hda_hint *hint = get_hint(codec, key); + return hint ? hint->val : NULL; +} +EXPORT_SYMBOL_HDA(snd_hda_get_hint); + +int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key) +{ + const char *p = snd_hda_get_hint(codec, key); + if (!p || !*p) + return -ENOENT; + switch (toupper(*p)) { + case 'T': /* true */ + case 'Y': /* yes */ + case '1': + return 1; + } + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_get_bool_hint); + #endif /* CONFIG_SND_HDA_RECONFIG */ diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 03ee9dd04913..27428c718fd7 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -433,6 +433,23 @@ static inline int snd_hda_hwdep_add_sysfs(struct hda_codec *codec) } #endif +#ifdef CONFIG_SND_HDA_RECONFIG +const char *snd_hda_get_hint(struct hda_codec *codec, const char *key); +int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key); +#else +static inline +const char *snd_hda_get_hint(struct hda_codec *codec, const char *key) +{ + return NULL; +} + +static inline +int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key) +{ + return -ENOENT; +} +#endif + /* * power-management */ -- cgit v1.2.3-59-g8ed1b From ab1726f920275b52991b2eff7538ac6d313bf9a2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Mar 2009 17:09:25 +0100 Subject: ALSA: hda - Add show for init_verbs and hints sysfs entries Added the show method for init_verbs and hints hwdep sysfs entries. They show the current values. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_hwdep.c | 35 +++++++++++++++++++++++++++++++++-- 1 file changed, 33 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 5e554de9cd9b..1e3ccc740afc 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -290,6 +290,22 @@ static ssize_t type##_store(struct device *dev, \ CODEC_ACTION_STORE(reconfig); CODEC_ACTION_STORE(clear); +static ssize_t init_verbs_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + int i, len = 0; + for (i = 0; i < codec->init_verbs.used; i++) { + struct hda_verb *v = snd_array_elem(&codec->init_verbs, i); + len += snprintf(buf + len, PAGE_SIZE - len, + "0x%02x 0x%03x 0x%04x\n", + v->nid, v->verb, v->param); + } + return len; +} + static ssize_t init_verbs_store(struct device *dev, struct device_attribute *attr, const char *buf, size_t count) @@ -312,6 +328,21 @@ static ssize_t init_verbs_store(struct device *dev, return count; } +static ssize_t hints_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + int i, len = 0; + for (i = 0; i < codec->hints.used; i++) { + struct hda_hint *hint = snd_array_elem(&codec->hints, i); + len += snprintf(buf + len, PAGE_SIZE - len, + "%s = %s\n", hint->key, hint->val); + } + return len; +} + static struct hda_hint *get_hint(struct hda_codec *codec, const char *key) { int i; @@ -466,8 +497,8 @@ static struct device_attribute codec_attrs[] = { CODEC_ATTR_RO(mfg), CODEC_ATTR_RW(name), CODEC_ATTR_RW(modelname), - CODEC_ATTR_WO(init_verbs), - CODEC_ATTR_WO(hints), + CODEC_ATTR_RW(init_verbs), + CODEC_ATTR_RW(hints), CODEC_ATTR_RO(init_pin_configs), CODEC_ATTR_RW(user_pin_configs), CODEC_ATTR_RO(driver_pin_configs), -- cgit v1.2.3-59-g8ed1b From d78d7a90adf793943cc29a414b6f4364a700aad5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Mar 2009 14:26:25 +0100 Subject: ALSA: hda - Create "Analog Loopback" controls optionally Don't create "Analog Loopback" controls as default since these controls are usually more harmful than useful for normal users. Only created when "loopback = yes" hint is given. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 56 ++++++++++++++++++++++++++++++++---------- 1 file changed, 43 insertions(+), 13 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 13056429aa64..7381325b98ff 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -190,6 +190,7 @@ struct sigmatel_spec { unsigned int stream_delay; /* analog loopback */ + struct snd_kcontrol_new *aloopback_ctl; unsigned char aloopback_mask; unsigned char aloopback_shift; @@ -1013,8 +1014,6 @@ static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = { HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT), HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT), - STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), @@ -1024,9 +1023,22 @@ static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = { +static struct snd_kcontrol_new stac92hd73xx_6ch_loopback[] = { + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), + {} +}; + +static struct snd_kcontrol_new stac92hd73xx_8ch_loopback[] = { STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 4), + {} +}; +static struct snd_kcontrol_new stac92hd73xx_10ch_loopback[] = { + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5), + {} +}; + +static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), @@ -1051,8 +1063,6 @@ static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = { }; static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = { - STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), @@ -1104,8 +1114,6 @@ static struct snd_kcontrol_new stac92hd83xxx_mixer[] = { }; static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { - STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -1131,9 +1139,11 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { - STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), +static struct snd_kcontrol_new stac92hd71bxx_loopback[] = { + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2) +}; +static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -1151,8 +1161,6 @@ static struct snd_kcontrol_new stac925x_mixer[] = { }; static struct snd_kcontrol_new stac9205_mixer[] = { - 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), @@ -1161,6 +1169,11 @@ static struct snd_kcontrol_new stac9205_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new stac9205_loopback[] = { + STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0, 1), + {} +}; + /* This needs to be generated dynamically based on sequence */ static struct snd_kcontrol_new stac922x_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x17, 0x0, HDA_INPUT), @@ -1173,8 +1186,6 @@ static struct snd_kcontrol_new stac922x_mixer[] = { static struct snd_kcontrol_new stac927x_mixer[] = { - 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), @@ -1186,6 +1197,11 @@ static struct snd_kcontrol_new stac927x_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new stac927x_loopback[] = { + STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB, 1), + {} +}; + static struct snd_kcontrol_new stac_dmux_mixer = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Digital Input Source", @@ -1312,6 +1328,13 @@ static int stac92xx_build_controls(struct hda_codec *codec) return err; } + if (spec->aloopback_ctl && + snd_hda_get_bool_hint(codec, "loopback") == 1) { + err = snd_hda_add_new_ctls(codec, spec->aloopback_ctl); + if (err < 0) + return err; + } + stac92xx_free_kctls(codec); /* no longer needed */ /* create jack input elements */ @@ -4618,14 +4641,18 @@ again: case 0x3: /* 6 Channel */ spec->mixer = stac92hd73xx_6ch_mixer; spec->init = stac92hd73xx_6ch_core_init; + spec->aloopback_ctl = stac92hd73xx_6ch_loopback; break; case 0x4: /* 8 Channel */ spec->mixer = stac92hd73xx_8ch_mixer; spec->init = stac92hd73xx_8ch_core_init; + spec->aloopback_ctl = stac92hd73xx_8ch_loopback; break; case 0x5: /* 10 Channel */ spec->mixer = stac92hd73xx_10ch_mixer; spec->init = stac92hd73xx_10ch_core_init; + spec->aloopback_ctl = stac92hd73xx_10ch_loopback; + break; } spec->multiout.dac_nids = spec->dac_nids; @@ -5036,6 +5063,7 @@ again: if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP) snd_hda_sequence_write_cache(codec, unmute_init); + spec->aloopback_ctl = stac92hd71bxx_loopback; spec->aloopback_mask = 0x50; spec->aloopback_shift = 0; @@ -5285,6 +5313,7 @@ static int patch_stac927x(struct hda_codec *codec) } spec->num_pwrs = 0; + spec->aloopback_ctl = stac927x_loopback; spec->aloopback_mask = 0x40; spec->aloopback_shift = 0; spec->eapd_switch = 1; @@ -5364,6 +5393,7 @@ static int patch_stac9205(struct hda_codec *codec) spec->init = stac9205_core_init; spec->mixer = stac9205_mixer; + spec->aloopback_ctl = stac9205_loopback; spec->aloopback_mask = 0x40; spec->aloopback_shift = 0; -- cgit v1.2.3-59-g8ed1b From 6565e4faca257fc51a4c55199d72e2701ba7e819 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Mar 2009 14:38:35 +0100 Subject: ALSA: hda - Add more hint options for IDT/Sigmatel codecs Allow more options to be set/reset via hwdep hint entry. hp_detect, gpio_mask, gpio_dir, gpio_data, eapd_mask and eapd_switch can be checked. For example, to disable hp_detect on the fly, # echo "hp_detect=0" > /sys/class/sound/hwC0D0/hints Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 33 +++++++++++++++++++++++++++++++++ 1 file changed, 33 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7381325b98ff..e9331561a484 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3949,6 +3949,36 @@ static void stac92xx_power_down(struct hda_codec *codec) static void stac_toggle_power_map(struct hda_codec *codec, hda_nid_t nid, int enable); +/* override some hints from the hwdep entry */ +static void stac_store_hints(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + const char *p; + int val; + + val = snd_hda_get_bool_hint(codec, "hp_detect"); + if (val >= 0) + spec->hp_detect = val; + p = snd_hda_get_hint(codec, "gpio_mask"); + if (p) { + spec->gpio_mask = simple_strtoul(p, NULL, 0); + spec->eapd_mask = spec->gpio_dir = spec->gpio_data = + spec->gpio_mask; + } + p = snd_hda_get_hint(codec, "gpio_dir"); + if (p) + spec->gpio_dir = simple_strtoul(p, NULL, 0) & spec->gpio_mask; + p = snd_hda_get_hint(codec, "gpio_data"); + if (p) + spec->gpio_data = simple_strtoul(p, NULL, 0) & spec->gpio_mask; + p = snd_hda_get_hint(codec, "eapd_mask"); + if (p) + spec->eapd_mask = simple_strtoul(p, NULL, 0) & spec->gpio_mask; + val = snd_hda_get_bool_hint(codec, "eapd_switch"); + if (val >= 0) + spec->eapd_switch = val; +} + static int stac92xx_init(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -3965,6 +3995,9 @@ static int stac92xx_init(struct hda_codec *codec) spec->adc_nids[i], 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + /* override some hints */ + stac_store_hints(codec); + /* set up GPIO */ gpio = spec->gpio_data; /* turn on EAPD statically when spec->eapd_switch isn't set. -- cgit v1.2.3-59-g8ed1b From 82ad39f9391fca1d3177bd9f6a5264eff5b5346a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 3 Mar 2009 15:00:35 +0100 Subject: ALSA: hda - Fix gcc compile warning MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit It's false positive, but annoying. sound/pci/hda/hda_codec.c: In function ‘get_empty_pcm_device’: sound/pci/hda/hda_codec.c:2772: warning: ‘dev’ may be used uninitialized in this function Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 7c9ef5c18e77..04cb1251e3e7 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2776,13 +2776,10 @@ static int get_empty_pcm_device(struct hda_bus *bus, int type) for (i = 0; i < ARRAY_SIZE(audio_idx); i++) { dev = audio_idx[i]; if (!test_bit(dev, bus->pcm_dev_bits)) - break; - } - if (i >= ARRAY_SIZE(audio_idx)) { - snd_printk(KERN_WARNING "Too many audio devices\n"); - return -EAGAIN; + goto ok; } - break; + snd_printk(KERN_WARNING "Too many audio devices\n"); + return -EAGAIN; case HDA_PCM_TYPE_SPDIF: case HDA_PCM_TYPE_HDMI: case HDA_PCM_TYPE_MODEM: @@ -2797,6 +2794,7 @@ static int get_empty_pcm_device(struct hda_bus *bus, int type) snd_printk(KERN_WARNING "Invalid PCM type %d\n", type); return -EINVAL; } + ok: set_bit(dev, bus->pcm_dev_bits); return dev; } -- cgit v1.2.3-59-g8ed1b From 79d7d5333b598e9a559bf27833f0ad2b8bf6ad2c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 4 Mar 2009 09:03:50 +0100 Subject: ALSA: hda - Fix HP dv6736 mic input Fix the mic input of HP dv6736 with Conexant 5051 codec chip. This laptop seems have no mic-switching per jack connection. A new model hp-dv6736 is introduced to match with the h/w implementation. Reference: Novell bnc#480753 https://bugzilla.novell.com/show_bug.cgi?id=480753 Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_conexant.c | 63 +++++++++++++++++++++++++--- 2 files changed, 59 insertions(+), 5 deletions(-) (limited to 'sound/pci') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index a448bbefd488..80b796e4a80e 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -262,6 +262,7 @@ Conexant 5051 ============= laptop Basic Laptop config (default) hp HP Spartan laptop + hp-dv6736 HP dv6736 lenovo-x200 Lenovo X200 laptop STAC9200 diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index b8de73ecfdeb..1938e92e1f03 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -72,6 +72,7 @@ struct conexant_spec { */ unsigned int cur_eapd; unsigned int hp_present; + unsigned int no_auto_mic; unsigned int need_dac_fix; /* capture */ @@ -1665,8 +1666,11 @@ static int cxt5051_hp_master_sw_put(struct snd_kcontrol *kcontrol, /* toggle input of built-in and mic jack appropriately */ static void cxt5051_portb_automic(struct hda_codec *codec) { + struct conexant_spec *spec = codec->spec; unsigned int present; + if (spec->no_auto_mic) + return; present = snd_hda_codec_read(codec, 0x17, 0, AC_VERB_GET_PIN_SENSE, 0) & AC_PINSENSE_PRESENCE; @@ -1682,6 +1686,8 @@ static void cxt5051_portc_automic(struct hda_codec *codec) unsigned int present; hda_nid_t new_adc; + if (spec->no_auto_mic) + return; present = snd_hda_codec_read(codec, 0x18, 0, AC_VERB_GET_PIN_SENSE, 0) & AC_PINSENSE_PRESENCE; @@ -1768,6 +1774,22 @@ static struct snd_kcontrol_new cxt5051_hp_mixers[] = { {} }; +static struct snd_kcontrol_new cxt5051_hp_dv6736_mixers[] = { + HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Mic Switch", 0x14, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = cxt_eapd_info, + .get = cxt_eapd_get, + .put = cxt5051_hp_master_sw_put, + .private_value = 0x1a, + }, + + {} +}; + static struct hda_verb cxt5051_init_verbs[] = { /* Line in, Mic */ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, @@ -1798,6 +1820,32 @@ static struct hda_verb cxt5051_init_verbs[] = { { } /* end */ }; +static struct hda_verb cxt5051_hp_dv6736_init_verbs[] = { + /* Line in, Mic */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, + /* SPK */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* HP, Amp */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* DAC1 */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Record selector: Int mic */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* SPDIF route: PCM */ + {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* EAPD */ + {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ + {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, + {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT}, + { } /* end */ +}; + static struct hda_verb cxt5051_lenovo_x200_init_verbs[] = { /* Line in, Mic */ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, @@ -1849,6 +1897,7 @@ static int cxt5051_init(struct hda_codec *codec) enum { CXT5051_LAPTOP, /* Laptops w/ EAPD support */ CXT5051_HP, /* no docking */ + CXT5051_HP_DV6736, /* HP without mic switch */ CXT5051_LENOVO_X200, /* Lenovo X200 laptop */ CXT5051_MODELS }; @@ -1856,10 +1905,12 @@ enum { static const char *cxt5051_models[CXT5051_MODELS] = { [CXT5051_LAPTOP] = "laptop", [CXT5051_HP] = "hp", + [CXT5051_HP_DV6736] = "hp-dv6736", [CXT5051_LENOVO_X200] = "lenovo-x200", }; static struct snd_pci_quirk cxt5051_cfg_tbl[] = { + SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6736", CXT5051_HP_DV6736), SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", CXT5051_LAPTOP), SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), @@ -1896,20 +1947,22 @@ static int patch_cxt5051(struct hda_codec *codec) spec->cur_adc = 0; spec->cur_adc_idx = 0; + codec->patch_ops.unsol_event = cxt5051_hp_unsol_event; + board_config = snd_hda_check_board_config(codec, CXT5051_MODELS, cxt5051_models, cxt5051_cfg_tbl); switch (board_config) { case CXT5051_HP: - codec->patch_ops.unsol_event = cxt5051_hp_unsol_event; spec->mixers[0] = cxt5051_hp_mixers; break; + case CXT5051_HP_DV6736: + spec->init_verbs[0] = cxt5051_hp_dv6736_init_verbs; + spec->mixers[0] = cxt5051_hp_dv6736_mixers; + spec->no_auto_mic = 1; + break; case CXT5051_LENOVO_X200: spec->init_verbs[0] = cxt5051_lenovo_x200_init_verbs; - /* fallthru */ - default: - case CXT5051_LAPTOP: - codec->patch_ops.unsol_event = cxt5051_hp_unsol_event; break; } -- cgit v1.2.3-59-g8ed1b From bd6afe3f34d41ed81e0c62a5a2181bb7bd51aebf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 4 Mar 2009 11:30:25 +0100 Subject: ALSA: hda - Fix conflict of mixer controls on Sony VAIO VGN-AR71S The recent update enabled the model=sony-assamd for all ALC262 with PCI SSID 104d:90xx. But this includes the VAIO VGN-AR* that has the primary codec of STAC92xx and the secondary ALC262 as a slave digital-only codec. For this device, the model=auto must be chosen to work properly. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c60c86acd9ba..8c02f789e4fc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10825,6 +10825,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x8203, "Sony UX-90", ALC262_HIPPO), SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD), + SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */ SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", -- cgit v1.2.3-59-g8ed1b From 3ea0d7cf472c6118bb8c0842d606f5436251e179 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Wed, 4 Mar 2009 14:22:50 -0300 Subject: ALSA: hda - Add 4 channel mode for 3stack-hp model (ALC888) Add additional 4 channel mode for 3stack-hp models. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 27 ++++++++++++++++++++++++--- 1 file changed, 24 insertions(+), 3 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8c02f789e4fc..3696ff31838f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8068,24 +8068,45 @@ static struct hda_verb alc888_6st_dell_verbs[] = { { } }; +/* + * 2ch mode + */ static struct hda_verb alc888_3st_hp_2ch_init[] = { { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { } + { } /* end */ }; +/* + * 4ch mode + */ +static struct hda_verb alc888_3st_hp_4ch_init[] = { + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x16, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +/* + * 6ch mode + */ static struct hda_verb alc888_3st_hp_6ch_init[] = { { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 }, { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { } + { 0x16, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ }; -static struct hda_channel_mode alc888_3st_hp_modes[2] = { +static struct hda_channel_mode alc888_3st_hp_modes[3] = { { 2, alc888_3st_hp_2ch_init }, + { 4, alc888_3st_hp_4ch_init }, { 6, alc888_3st_hp_6ch_init }, }; -- cgit v1.2.3-59-g8ed1b From 8718b700ccbcc3c6016d38a75e005293c3660f1c Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Wed, 4 Mar 2009 14:22:51 -0300 Subject: ALSA: hda - Add headphone automute support for 3stack-hp model (ALC888) Mute speaker outputs on headphone insertion for machines that use 3stack-hp model. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 38 +++++++++++++++++++++++++++++++++----- 1 file changed, 33 insertions(+), 5 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3696ff31838f..251647d8b5b9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8056,16 +8056,42 @@ static struct hda_verb alc888_lenovo_sky_verbs[] = { { } /* end */ }; +static struct hda_verb alc888_6st_dell_verbs[] = { + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + { } +}; + +static void alc888_3st_hp_front_automute(struct hda_codec *codec) +{ + unsigned int present, bits; + + present = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); + snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); + snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); +} + +static void alc888_3st_hp_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc888_3st_hp_front_automute(codec); + break; + } +} + static struct hda_verb alc888_3st_hp_verbs[] = { {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */ {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CLFE : output 2 (0x0e) */ - { } -}; - -static struct hda_verb alc888_6st_dell_verbs[] = { {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, - { } + { } /* end */ }; /* @@ -8950,6 +8976,8 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc888_3st_hp_modes, .need_dac_fix = 1, .input_mux = &alc883_capture_source, + .unsol_event = alc888_3st_hp_unsol_event, + .init_hook = alc888_3st_hp_front_automute, }, [ALC888_6ST_DELL] = { .mixers = { alc883_base_mixer, alc883_chmode_mixer }, -- cgit v1.2.3-59-g8ed1b From 7ec30f0e7768985ab2ef6334840e3fc8fa253421 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Wed, 4 Mar 2009 14:22:52 -0300 Subject: ALSA: hda - Map 3stack-hp model (ALC888) for HP Educ.ar Added model=3stack-hp for HP Educ.ar desktop machine (103c:2a72). Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 251647d8b5b9..91ef9f27b120 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8673,6 +8673,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG), SND_PCI_QUIRK(0x103c, 0x2a66, "HP Acacia", ALC888_3ST_HP), + SND_PCI_QUIRK(0x103c, 0x2a72, "HP Educ.ar", ALC888_3ST_HP), SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V), SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x1043, 0x8284, "Asus Z37E", ALC883_6ST_DIG), -- cgit v1.2.3-59-g8ed1b From 37db623ae2a7bde234a8ed683d0d13d6f939199c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Mar 2009 09:40:16 +0100 Subject: ALSA: hda - Fix check of ALC888S-VC in alc888_coef_init() Fixed the wrong bits check to identify ALC888S-VC model in alc888_coef_init(). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 91ef9f27b120..6325ea43cf08 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -982,7 +982,7 @@ static void alc888_coef_init(struct hda_codec *codec) snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0); tmp = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_PROC_COEF, 0); snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 7); - if ((tmp & 0xf0) == 2) + if ((tmp & 0xf0) == 0x20) /* alc888S-VC */ snd_hda_codec_read(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, 0x830); -- cgit v1.2.3-59-g8ed1b From f03d3115a6bcb814019d945c50c2ef91e5f14477 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Mar 2009 14:18:16 +0100 Subject: ALSA: Fix sample rate of Lenovo Ideapad to 44.1kHz Noises can be heard on analog outputs of (some model of) Lenovo Ideapad due to the hardware problem, and the only workaround right now is to fix the sample rate to 44.1kHz. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 34 +++++++++++++++++++++++++++++++--- 1 file changed, 31 insertions(+), 3 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6325ea43cf08..b794cba494c3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12845,6 +12845,27 @@ static int alc269_auto_create_analog_input_ctls(struct alc_spec *spec, #define alc269_pcm_digital_playback alc880_pcm_digital_playback #define alc269_pcm_digital_capture alc880_pcm_digital_capture +static struct hda_pcm_stream alc269_44k_pcm_analog_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_44100, /* fixed rate */ + /* NID is set in alc_build_pcms */ + .ops = { + .open = alc880_playback_pcm_open, + .prepare = alc880_playback_pcm_prepare, + .cleanup = alc880_playback_pcm_cleanup + }, +}; + +static struct hda_pcm_stream alc269_44k_pcm_analog_capture = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_44100, /* fixed rate */ + /* NID is set in alc_build_pcms */ +}; + /* * BIOS auto configuration */ @@ -13060,9 +13081,16 @@ static int patch_alc269(struct hda_codec *codec) setup_preset(spec, &alc269_presets[board_config]); spec->stream_name_analog = "ALC269 Analog"; - spec->stream_analog_playback = &alc269_pcm_analog_playback; - spec->stream_analog_capture = &alc269_pcm_analog_capture; - + if (codec->subsystem_id == 0x17aa3bf8) { + /* Due to a hardware problem on Lenovo Ideadpad, we need to + * fix the sample rate of analog I/O to 44.1kHz + */ + spec->stream_analog_playback = &alc269_44k_pcm_analog_playback; + spec->stream_analog_capture = &alc269_44k_pcm_analog_capture; + } else { + spec->stream_analog_playback = &alc269_pcm_analog_playback; + spec->stream_analog_capture = &alc269_pcm_analog_capture; + } spec->stream_name_digital = "ALC269 Digital"; spec->stream_digital_playback = &alc269_pcm_digital_playback; spec->stream_digital_capture = &alc269_pcm_digital_capture; -- cgit v1.2.3-59-g8ed1b From dc04d1b4d2043e2fca2d94d6d5542b930f2bc5b3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Mar 2009 10:00:05 +0100 Subject: ALSA: hda - Create output controls according to pin types for IDT/STAC Improve the parser to pick up more intuitive control names for the outputs judging from the pin type, instead of fixed names assigned to channels. Also, revive the multi-HP workaround since this change fixes the problem with the multi-HP detection. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 141 +++++++++++++++++++++-------------------- 1 file changed, 72 insertions(+), 69 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 2e0a599f8c12..edd2ed7ebb49 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3039,35 +3039,33 @@ static int add_spec_extra_dacs(struct sigmatel_spec *spec, hda_nid_t nid) return 1; } -static int is_unique_dac(struct sigmatel_spec *spec, hda_nid_t nid) -{ - int i; - - if (spec->autocfg.line_outs != 1) - return 0; - if (spec->multiout.hp_nid == nid) - return 0; - for (i = 0; i < ARRAY_SIZE(spec->multiout.extra_out_nid); i++) - if (spec->multiout.extra_out_nid[i] == nid) - return 0; - return 1; -} - -/* add playback controls from the parsed DAC table */ -static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, - const struct auto_pin_cfg *cfg) +/* Create output controls + * The mixer elements are named depending on the given type (AUTO_PIN_XXX_OUT) + */ +static int create_multi_out_ctls(struct hda_codec *codec, int num_outs, + const hda_nid_t *pins, + const hda_nid_t *dac_nids, + int type) { struct sigmatel_spec *spec = codec->spec; static const char *chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; - hda_nid_t nid = 0; + static const char *hp_pfxs[] = { + "Headphone", "Headphone2", "Headphone3", "Headphone4" + }; + static const char *speaker_pfxs[] = { + "Speaker", "External Speaker", "Speaker2", "Speaker3" + }; + hda_nid_t nid; int i, err; unsigned int wid_caps; - for (i = 0; i < cfg->line_outs && spec->multiout.dac_nids[i]; i++) { - nid = spec->multiout.dac_nids[i]; - if (i == 2) { + for (i = 0; i < num_outs && i < ARRAY_SIZE(chname); i++) { + nid = dac_nids[i]; + if (!nid) + continue; + if (type != AUTO_PIN_HP_OUT && i == 2) { /* Center/LFE */ err = create_controls(codec, "Center", nid, 1); if (err < 0) @@ -3088,23 +3086,43 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, } } else { - const char *name = chname[i]; - /* if it's a single DAC, assign a better name */ - if (!i && is_unique_dac(spec, nid)) { - switch (cfg->line_out_type) { - case AUTO_PIN_HP_OUT: - name = "Headphone"; - break; - case AUTO_PIN_SPEAKER_OUT: - name = "Speaker"; - break; - } + const char *name; + switch (type) { + case AUTO_PIN_HP_OUT: + name = hp_pfxs[i]; + break; + case AUTO_PIN_SPEAKER_OUT: + name = speaker_pfxs[i]; + break; + default: + name = chname[i]; + break; } err = create_controls(codec, name, nid, 3); if (err < 0) return err; + if (type == AUTO_PIN_HP_OUT && !spec->hp_detect) { + wid_caps = get_wcaps(codec, pins[i]); + if (wid_caps & AC_WCAP_UNSOL_CAP) + spec->hp_detect = 1; + } } } + return 0; +} + +/* add playback controls from the parsed DAC table */ +static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, + const struct auto_pin_cfg *cfg) +{ + struct sigmatel_spec *spec = codec->spec; + int err; + + err = create_multi_out_ctls(codec, cfg->line_outs, cfg->line_out_pins, + spec->multiout.dac_nids, + cfg->line_out_type); + if (err < 0) + return err; if (cfg->hp_outs > 1 && cfg->line_out_type == AUTO_PIN_LINE_OUT) { err = stac92xx_add_control(spec, @@ -3139,40 +3157,18 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, struct auto_pin_cfg *cfg) { struct sigmatel_spec *spec = codec->spec; - hda_nid_t nid; - int i, err, nums; + int err; + + err = create_multi_out_ctls(codec, cfg->hp_outs, cfg->hp_pins, + spec->hp_dacs, AUTO_PIN_HP_OUT); + if (err < 0) + return err; + + err = create_multi_out_ctls(codec, cfg->speaker_outs, cfg->speaker_pins, + spec->speaker_dacs, AUTO_PIN_SPEAKER_OUT); + if (err < 0) + return err; - nums = 0; - for (i = 0; i < cfg->hp_outs; i++) { - static const char *pfxs[] = { - "Headphone", "Headphone2", "Headphone3", - }; - unsigned int wid_caps = get_wcaps(codec, cfg->hp_pins[i]); - if (wid_caps & AC_WCAP_UNSOL_CAP) - spec->hp_detect = 1; - if (nums >= ARRAY_SIZE(pfxs)) - continue; - nid = spec->hp_dacs[i]; - if (!nid) - continue; - err = create_controls(codec, pfxs[nums++], nid, 3); - if (err < 0) - return err; - } - nums = 0; - for (i = 0; i < cfg->speaker_outs; i++) { - static const char *pfxs[] = { - "Speaker", "External Speaker", "Speaker2", - }; - if (nums >= ARRAY_SIZE(pfxs)) - continue; - nid = spec->speaker_dacs[i]; - if (!nid) - continue; - err = create_controls(codec, pfxs[nums++], nid, 3); - if (err < 0) - return err; - } return 0; } @@ -3505,6 +3501,7 @@ static void stac92xx_auto_init_hp_out(struct hda_codec *codec) static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out, hda_nid_t dig_in) { struct sigmatel_spec *spec = codec->spec; + int hp_swap = 0; int err; if ((err = snd_hda_parse_pin_def_config(codec, @@ -3514,7 +3511,6 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if (! spec->autocfg.line_outs) return 0; /* can't find valid pin config */ -#if 0 /* FIXME: temporarily disabled */ /* If we have no real line-out pin and multiple hp-outs, HPs should * be set up as multi-channel outputs. */ @@ -3533,8 +3529,8 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out spec->autocfg.line_outs = spec->autocfg.hp_outs; spec->autocfg.line_out_type = AUTO_PIN_HP_OUT; spec->autocfg.hp_outs = 0; + hp_swap = 1; } -#endif /* FIXME: temporarily disabled */ if (spec->autocfg.mono_out_pin) { int dir = get_wcaps(codec, spec->autocfg.mono_out_pin) & (AC_WCAP_OUT_AMP | AC_WCAP_IN_AMP); @@ -3627,12 +3623,19 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out #endif err = stac92xx_auto_create_hp_ctls(codec, &spec->autocfg); - if (err < 0) return err; - err = stac92xx_auto_create_analog_input_ctls(codec, &spec->autocfg); + /* All output parsing done, now restore the swapped hp pins */ + if (hp_swap) { + memcpy(spec->autocfg.hp_pins, spec->autocfg.line_out_pins, + sizeof(spec->autocfg.hp_pins)); + spec->autocfg.hp_outs = spec->autocfg.line_outs; + spec->autocfg.line_out_type = AUTO_PIN_HP_OUT; + spec->autocfg.line_outs = 0; + } + err = stac92xx_auto_create_analog_input_ctls(codec, &spec->autocfg); if (err < 0) return err; -- cgit v1.2.3-59-g8ed1b From 7a411ee01bf3114ba2a2ae013eaae4e3c41f8eb5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Mar 2009 10:08:14 +0100 Subject: ALSA: hda - Allow slave controls with non-zero indices Fix snd_hda_add_vmaster() to check the non-zero indices of slave controls. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 21 +++++++++++++-------- 1 file changed, 13 insertions(+), 8 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 04cb1251e3e7..1885e7649101 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1552,15 +1552,20 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, for (s = slaves; *s; s++) { struct snd_kcontrol *sctl; - - sctl = snd_hda_find_mixer_ctl(codec, *s); - if (!sctl) { - snd_printdd("Cannot find slave %s, skipped\n", *s); - continue; + int i = 0; + for (;;) { + sctl = _snd_hda_find_mixer_ctl(codec, *s, i); + if (!sctl) { + if (!i) + snd_printdd("Cannot find slave %s, " + "skipped\n", *s); + break; + } + err = snd_ctl_add_slave(kctl, sctl); + if (err < 0) + return err; + i++; } - err = snd_ctl_add_slave(kctl, sctl); - if (err < 0) - return err; } return 0; } -- cgit v1.2.3-59-g8ed1b From 668b9652be33510a2a42b290dd335d34d38e2068 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Mar 2009 10:13:24 +0100 Subject: ALSA: hda - Create multiple HP / speaker controls with index Create multiple "Headphone" and "Speaker" controls with non-zero index numbers instead of "Headphone2", etc. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 33 ++++++++++++++------------------- 1 file changed, 14 insertions(+), 19 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index edd2ed7ebb49..d19090fd2d13 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1227,10 +1227,7 @@ static const char *slave_vols[] = { "LFE Playback Volume", "Side Playback Volume", "Headphone Playback Volume", - "Headphone2 Playback Volume", "Speaker Playback Volume", - "External Speaker Playback Volume", - "Speaker2 Playback Volume", NULL }; @@ -1241,10 +1238,7 @@ static const char *slave_sws[] = { "LFE Playback Switch", "Side Playback Switch", "Headphone Playback Switch", - "Headphone2 Playback Switch", "Speaker Playback Switch", - "External Speaker Playback Switch", - "Speaker2 Playback Switch", "IEC958 Playback Switch", NULL }; @@ -2976,8 +2970,8 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec) } /* create volume control/switch for the given prefx type */ -static int create_controls(struct hda_codec *codec, const char *pfx, - hda_nid_t nid, int chs) +static int create_controls_idx(struct hda_codec *codec, const char *pfx, + int idx, hda_nid_t nid, int chs) { struct sigmatel_spec *spec = codec->spec; char name[32]; @@ -3001,19 +2995,22 @@ static int create_controls(struct hda_codec *codec, const char *pfx, } sprintf(name, "%s Playback Volume", pfx); - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, name, + err = stac92xx_add_control_idx(spec, STAC_CTL_WIDGET_VOL, idx, name, HDA_COMPOSE_AMP_VAL_OFS(nid, chs, 0, HDA_OUTPUT, spec->volume_offset)); if (err < 0) return err; sprintf(name, "%s Playback Switch", pfx); - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, name, + err = stac92xx_add_control_idx(spec, STAC_CTL_WIDGET_MUTE, idx, name, HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT)); if (err < 0) return err; return 0; } +#define create_controls(codec, pfx, nid, chs) \ + create_controls_idx(codec, pfx, 0, nid, chs) + static int add_spec_dacs(struct sigmatel_spec *spec, hda_nid_t nid) { if (spec->multiout.num_dacs > 4) { @@ -3051,12 +3048,6 @@ static int create_multi_out_ctls(struct hda_codec *codec, int num_outs, static const char *chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; - static const char *hp_pfxs[] = { - "Headphone", "Headphone2", "Headphone3", "Headphone4" - }; - static const char *speaker_pfxs[] = { - "Speaker", "External Speaker", "Speaker2", "Speaker3" - }; hda_nid_t nid; int i, err; unsigned int wid_caps; @@ -3087,18 +3078,22 @@ static int create_multi_out_ctls(struct hda_codec *codec, int num_outs, } else { const char *name; + int idx; switch (type) { case AUTO_PIN_HP_OUT: - name = hp_pfxs[i]; + name = "Headphone"; + idx = i; break; case AUTO_PIN_SPEAKER_OUT: - name = speaker_pfxs[i]; + name = "Speaker"; + idx = i; break; default: name = chname[i]; + idx = 0; break; } - err = create_controls(codec, name, nid, 3); + err = create_controls_idx(codec, name, idx, nid, 3); if (err < 0) return err; if (type == AUTO_PIN_HP_OUT && !spec->hp_detect) { -- cgit v1.2.3-59-g8ed1b From ee58a7ca21b2acf0d7ad0e1eb2f8d916ecf9fadc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Mar 2009 12:00:24 +0100 Subject: ALSA: hda - Connect to primary DAC if no individual DAC is available In stac92xx_auto_fill_dac_nids[], connect to the primary DAC if no individual DAC is available for each pin. This ensures that the pin works somehow at least. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d19090fd2d13..ee119259183c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2871,6 +2871,16 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid) return conn[j]; } } + /* if all DACs are already assigned, connect to the primary DAC */ + if (conn_len > 1) { + for (j = 0; j < conn_len; j++) { + if (conn[j] == spec->multiout.dac_nids[0]) { + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, j); + break; + } + } + } return 0; } -- cgit v1.2.3-59-g8ed1b From 139e071b0ff37800ed0a68b10c4bb325f51786eb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Mar 2009 12:10:41 +0100 Subject: ALSA: hda - Assign HP and speaker DACs before mic/line-in Assign DACs to HP and speaker before mic-in/line-in shared outputs. This improves the usability as it results in more intuitive mixer names. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 40 ++++++++++++++++++++-------------------- 1 file changed, 20 insertions(+), 20 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index ee119259183c..123bcf7c3b24 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2921,6 +2921,26 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec) add_spec_dacs(spec, dac); } + for (i = 0; i < cfg->hp_outs; i++) { + nid = cfg->hp_pins[i]; + dac = get_unassigned_dac(codec, nid); + if (dac) { + if (!spec->multiout.hp_nid) + spec->multiout.hp_nid = dac; + else + add_spec_extra_dacs(spec, dac); + } + spec->hp_dacs[i] = dac; + } + + for (i = 0; i < cfg->speaker_outs; i++) { + nid = cfg->speaker_pins[i]; + dac = get_unassigned_dac(codec, nid); + if (dac) + add_spec_extra_dacs(spec, dac); + spec->speaker_dacs[i] = dac; + } + /* add line-in as output */ nid = check_line_out_switch(codec); if (nid) { @@ -2948,26 +2968,6 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec) } } - for (i = 0; i < cfg->hp_outs; i++) { - nid = cfg->hp_pins[i]; - dac = get_unassigned_dac(codec, nid); - if (dac) { - if (!spec->multiout.hp_nid) - spec->multiout.hp_nid = dac; - else - add_spec_extra_dacs(spec, dac); - } - spec->hp_dacs[i] = dac; - } - - for (i = 0; i < cfg->speaker_outs; i++) { - nid = cfg->speaker_pins[i]; - dac = get_unassigned_dac(codec, nid); - if (dac) - add_spec_extra_dacs(spec, dac); - spec->speaker_dacs[i] = dac; - } - snd_printd("stac92xx: dac_nids=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n", spec->multiout.num_dacs, spec->multiout.dac_nids[0], -- cgit v1.2.3-59-g8ed1b From ae6241fbf5c8863631532e8069037bae460607be Mon Sep 17 00:00:00 2001 From: Christoph Plattner Date: Sun, 8 Mar 2009 23:19:05 +0100 Subject: ALSA: hda - Added HP HDX16/HDX18 notebook support for HDA codecs (82HD71) Added codec recognition of HP HDX platforms and added support of the MUTE LED (orange/white). For this feature the CONFIG_SND_HDA_POWER_SAVE is needed to use event handling for mute control. Signed-off-by: Christoph Plattner Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 57 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 57 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 123bcf7c3b24..fb9f4ccba885 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -99,6 +99,7 @@ enum { STAC_DELL_M4_3, STAC_HP_M4, STAC_HP_DV5, + STAC_HP_HDX, STAC_92HD71BXX_MODELS }; @@ -1828,6 +1829,7 @@ static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { [STAC_DELL_M4_3] = dell_m4_3_pin_configs, [STAC_HP_M4] = NULL, [STAC_HP_DV5] = NULL, + [STAC_HP_HDX] = NULL, }; static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { @@ -1838,6 +1840,7 @@ static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { [STAC_DELL_M4_3] = "dell-m4-3", [STAC_HP_M4] = "hp-m4", [STAC_HP_DV5] = "hp-dv5", + [STAC_HP_HDX] = "hp-hdx", }; static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { @@ -1852,6 +1855,10 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "HP dv4-7", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361a, "HP mini 1000", STAC_HP_M4), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361b, + "HP HDX", STAC_HP_HDX), /* HDX16 */ + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3610, + "HP HDX", STAC_HP_HDX), /* HDX18 */ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233, "unknown Dell", STAC_DELL_M4_1), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0234, @@ -4472,6 +4479,41 @@ static int stac92xx_resume(struct hda_codec *codec) return 0; } + +/* + * using power check for controlling mute led of HP HDX notebooks + * check for mute state only on Speakers (nid = 0x10) + * + * For this feature CONFIG_SND_HDA_POWER_SAVE is needed, otherwise + * the LED is NOT working properly ! + */ + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static int stac92xx_check_power_status (struct hda_codec * codec, hda_nid_t nid) +{ + struct sigmatel_spec *spec = codec->spec; + + /* only handle on HP HDX */ + if (spec->board_config != STAC_HP_HDX) + return 0; + + if (nid == 0x10) + { + if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & + HDA_AMP_MUTE) + spec->gpio_data &= ~0x08; /* orange */ + else + spec->gpio_data |= 0x08; /* white */ + + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, + spec->gpio_data); + } + + return 0; +} +#endif + static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state) { struct sigmatel_spec *spec = codec->spec; @@ -4493,6 +4535,9 @@ static struct hda_codec_ops stac92xx_patch_ops = { .suspend = stac92xx_suspend, .resume = stac92xx_resume, #endif +#ifdef CONFIG_SND_HDA_POWER_SAVE + .check_power_status = stac92xx_check_power_status, +#endif }; static int patch_stac9200(struct hda_codec *codec) @@ -5089,6 +5134,13 @@ again: /* no output amps */ spec->num_pwrs = 0; /* fallthru */ + case 0x111d76b2: /* Codec of HP HDX16/HDX18 */ + + /* orange/white mute led on GPIO3, orange=0, white=1 */ + spec->gpio_mask |= 0x08; + spec->gpio_dir |= 0x08; + spec->gpio_data |= 0x08; /* set to white */ + /* fallthru */ default: memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_amixer, sizeof(stac92hd71bxx_dmux_amixer)); @@ -5143,6 +5195,11 @@ again: snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010); stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN); break; + case STAC_HP_HDX: + spec->num_dmics = 1; + spec->num_dmuxes = 1; + spec->num_smuxes = 1; + break; }; spec->multiout.dac_nids = spec->dac_nids; -- cgit v1.2.3-59-g8ed1b From 443e26d014c242623dd70cda054cc6e5ebf7993d Mon Sep 17 00:00:00 2001 From: Christoph Plattner Date: Tue, 10 Mar 2009 00:05:56 +0100 Subject: ALSA: hda - Rework on patch_sigmatel.c for HP HDX16/HDX18 Code rework, comments of mail tiwai@suse.de (2009-03-09) incorporated. Code tested on HP HDX16 (not tested on HDX18 yet). Signed-off-by: Christoph Plattner Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 30 +++++++++++++++--------------- 1 file changed, 15 insertions(+), 15 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index fb9f4ccba885..d119feed42c9 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4489,14 +4489,10 @@ static int stac92xx_resume(struct hda_codec *codec) */ #ifdef CONFIG_SND_HDA_POWER_SAVE -static int stac92xx_check_power_status (struct hda_codec * codec, hda_nid_t nid) +static int stac92xx_hp_hdx_check_power_status (struct hda_codec * codec, hda_nid_t nid) { struct sigmatel_spec *spec = codec->spec; - /* only handle on HP HDX */ - if (spec->board_config != STAC_HP_HDX) - return 0; - if (nid == 0x10) { if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & @@ -4535,9 +4531,6 @@ static struct hda_codec_ops stac92xx_patch_ops = { .suspend = stac92xx_suspend, .resume = stac92xx_resume, #endif -#ifdef CONFIG_SND_HDA_POWER_SAVE - .check_power_status = stac92xx_check_power_status, -#endif }; static int patch_stac9200(struct hda_codec *codec) @@ -5134,13 +5127,6 @@ again: /* no output amps */ spec->num_pwrs = 0; /* fallthru */ - case 0x111d76b2: /* Codec of HP HDX16/HDX18 */ - - /* orange/white mute led on GPIO3, orange=0, white=1 */ - spec->gpio_mask |= 0x08; - spec->gpio_dir |= 0x08; - spec->gpio_data |= 0x08; /* set to white */ - /* fallthru */ default: memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_amixer, sizeof(stac92hd71bxx_dmux_amixer)); @@ -5199,6 +5185,20 @@ again: spec->num_dmics = 1; spec->num_dmuxes = 1; spec->num_smuxes = 1; + /* + * For controlling MUTE LED on HP HDX16/HDX18 notebooks, + * the CONFIG_SND_HDA_POWER_SAVE is needed to be set. + */ +#ifdef CONFIG_SND_HDA_POWER_SAVE + /* orange/white mute led on GPIO3, orange=0, white=1 */ + spec->gpio_mask |= 0x08; + spec->gpio_dir |= 0x08; + spec->gpio_data |= 0x08; /* set to white */ + + /* register check_power_status callback. */ + codec->patch_ops.check_power_status = + stac92xx_hp_hdx_check_power_status; +#endif break; }; -- cgit v1.2.3-59-g8ed1b From 6fce61aeaf0dc1dfa306092539397ab903a9afc4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2009 07:48:57 +0100 Subject: ALSA: hda - Fix coding style issues in last two patches Also re-ordered the quirk entries per SSID. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 26 +++++++++++++------------- 1 file changed, 13 insertions(+), 13 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d119feed42c9..72c87aa20bd9 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1853,12 +1853,12 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "HP dv4-7", STAC_HP_DV5), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3600, "HP dv4-7", STAC_HP_DV5), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3610, + "HP HDX", STAC_HP_HDX), /* HDX18 */ SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361a, "HP mini 1000", STAC_HP_M4), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361b, - "HP HDX", STAC_HP_HDX), /* HDX16 */ - SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3610, - "HP HDX", STAC_HP_HDX), /* HDX18 */ + "HP HDX", STAC_HP_HDX), /* HDX16 */ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233, "unknown Dell", STAC_DELL_M4_1), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0234, @@ -4489,20 +4489,20 @@ static int stac92xx_resume(struct hda_codec *codec) */ #ifdef CONFIG_SND_HDA_POWER_SAVE -static int stac92xx_hp_hdx_check_power_status (struct hda_codec * codec, hda_nid_t nid) +static int stac92xx_hp_hdx_check_power_status(struct hda_codec *codec, + hda_nid_t nid) { struct sigmatel_spec *spec = codec->spec; - - if (nid == 0x10) - { - if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & + + if (nid == 0x10) { + if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & HDA_AMP_MUTE) spec->gpio_data &= ~0x08; /* orange */ else spec->gpio_data |= 0x08; /* white */ - - stac_gpio_set(codec, spec->gpio_mask, - spec->gpio_dir, + + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, spec->gpio_data); } @@ -5185,7 +5185,7 @@ again: spec->num_dmics = 1; spec->num_dmuxes = 1; spec->num_smuxes = 1; - /* + /* * For controlling MUTE LED on HP HDX16/HDX18 notebooks, * the CONFIG_SND_HDA_POWER_SAVE is needed to be set. */ @@ -5196,7 +5196,7 @@ again: spec->gpio_data |= 0x08; /* set to white */ /* register check_power_status callback. */ - codec->patch_ops.check_power_status = + codec->patch_ops.check_power_status = stac92xx_hp_hdx_check_power_status; #endif break; -- cgit v1.2.3-59-g8ed1b From dd5746a85cb21ea5b3afca0b569586a05aa56846 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2009 14:30:40 +0100 Subject: ALSA: hda - Create vmaster for conexant codecs Instead of binding volumes, create a virtual master volume for Conexant codecs. This allows separate HP and speaker volume controls. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 47 ++++++++++++++++++++++++++++++++---------- 1 file changed, 36 insertions(+), 11 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 1938e92e1f03..e1476d6d8b39 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -58,6 +58,7 @@ struct conexant_spec { struct snd_kcontrol_new *mixers[5]; int num_mixers; + hda_nid_t vmaster_nid; const struct hda_verb *init_verbs[5]; /* initialization verbs * don't forget NULL @@ -462,6 +463,18 @@ static void conexant_free(struct hda_codec *codec) kfree(codec->spec); } +static const char *slave_vols[] = { + "Headphone Playback Volume", + "Speaker Playback Volume", + NULL +}; + +static const char *slave_sws[] = { + "Headphone Playback Switch", + "Speaker Playback Switch", + NULL +}; + static int conexant_build_controls(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; @@ -489,6 +502,26 @@ static int conexant_build_controls(struct hda_codec *codec) if (err < 0) return err; } + + /* if we have no master control, let's create it */ + if (spec->vmaster_nid && + !snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { + unsigned int vmaster_tlv[4]; + 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); + 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); + if (err < 0) + return err; + } + return 0; } @@ -1182,16 +1215,6 @@ static int cxt5047_hp_master_sw_put(struct snd_kcontrol *kcontrol, return 1; } -/* bind volumes of both NID 0x13 (Headphones) and 0x1d (Speakers) */ -static struct hda_bind_ctls cxt5047_bind_master_vol = { - .ops = &snd_hda_bind_vol, - .values = { - HDA_COMPOSE_AMP_VAL(0x13, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT), - 0 - }, -}; - /* mute internal speaker if HP is plugged */ static void cxt5047_hp_automute(struct hda_codec *codec) { @@ -1311,7 +1334,8 @@ static struct snd_kcontrol_new cxt5047_toshiba_mixers[] = { HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT), HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT), - HDA_BIND_VOL("Master Playback Volume", &cxt5047_bind_master_vol), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x00, HDA_OUTPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", @@ -1631,6 +1655,7 @@ static int patch_cxt5047(struct hda_codec *codec) codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; #endif } + spec->vmaster_nid = 0x13; return 0; } -- cgit v1.2.3-59-g8ed1b From b880c74adf7e79b97de710a152ea82f292f9abc7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2009 14:41:05 +0100 Subject: ALSA: hda - Create "Capture Source" control dynamically in patch_conexant.c Create "Capture Source" control dynamically for Conexant codecs. If only one capture item is available, don't create such a control since it's just useless. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 61 ++++++++++++------------------------------ 1 file changed, 17 insertions(+), 44 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e1476d6d8b39..d5d736ff7c6c 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -463,6 +463,17 @@ static void conexant_free(struct hda_codec *codec) kfree(codec->spec); } +static struct snd_kcontrol_new cxt_capture_mixers[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = conexant_mux_enum_info, + .get = conexant_mux_enum_get, + .put = conexant_mux_enum_put + }, + {} +}; + static const char *slave_vols[] = { "Headphone Playback Volume", "Speaker Playback Volume", @@ -522,6 +533,12 @@ static int conexant_build_controls(struct hda_codec *codec) return err; } + if (spec->input_mux) { + err = snd_hda_add_new_ctls(codec, cxt_capture_mixers); + if (err < 0) + return err; + } + return 0; } @@ -753,13 +770,6 @@ static void cxt5045_hp_unsol_event(struct hda_codec *codec, } static struct snd_kcontrol_new cxt5045_mixers[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = conexant_mux_enum_info, - .get = conexant_mux_enum_get, - .put = conexant_mux_enum_put - }, HDA_CODEC_VOLUME("Int Mic Capture Volume", 0x1a, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Int Mic Capture Switch", 0x1a, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Ext Mic Capture Volume", 0x1a, 0x02, HDA_INPUT), @@ -793,13 +803,6 @@ static struct snd_kcontrol_new cxt5045_benq_mixers[] = { }; static struct snd_kcontrol_new cxt5045_mixers_hp530[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = conexant_mux_enum_info, - .get = conexant_mux_enum_get, - .put = conexant_mux_enum_put - }, HDA_CODEC_VOLUME("Int Mic Capture Volume", 0x1a, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Int Mic Capture Switch", 0x1a, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Ext Mic Capture Volume", 0x1a, 0x01, HDA_INPUT), @@ -1170,20 +1173,6 @@ static struct hda_channel_mode cxt5047_modes[1] = { { 2, NULL }, }; -static struct hda_input_mux cxt5047_capture_source = { - .num_items = 1, - .items = { - { "Mic", 0x2 }, - } -}; - -static struct hda_input_mux cxt5047_hp_capture_source = { - .num_items = 1, - .items = { - { "ExtMic", 0x2 }, - } -}; - static struct hda_input_mux cxt5047_toshiba_capture_source = { .num_items = 2, .items = { @@ -1321,13 +1310,6 @@ static struct snd_kcontrol_new cxt5047_mixers[] = { }; static struct snd_kcontrol_new cxt5047_toshiba_mixers[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = conexant_mux_enum_info, - .get = conexant_mux_enum_get, - .put = conexant_mux_enum_put - }, HDA_CODEC_VOLUME("Mic Bypass Capture Volume", 0x19, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Mic Bypass Capture Switch", 0x19, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT), @@ -1349,13 +1331,6 @@ static struct snd_kcontrol_new cxt5047_toshiba_mixers[] = { }; static struct snd_kcontrol_new cxt5047_hp_mixers[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = conexant_mux_enum_info, - .get = conexant_mux_enum_get, - .put = conexant_mux_enum_put - }, HDA_CODEC_VOLUME("Mic Bypass Capture Volume", 0x19, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Mic Bypass Capture Switch", 0x19,0x02,HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT), @@ -1614,7 +1589,6 @@ static int patch_cxt5047(struct hda_codec *codec) spec->num_adc_nids = 1; spec->adc_nids = cxt5047_adc_nids; spec->capsrc_nids = cxt5047_capsrc_nids; - spec->input_mux = &cxt5047_capture_source; spec->num_mixers = 1; spec->mixers[0] = cxt5047_mixers; spec->num_init_verbs = 1; @@ -1633,7 +1607,6 @@ static int patch_cxt5047(struct hda_codec *codec) codec->patch_ops.unsol_event = cxt5047_hp2_unsol_event; break; case CXT5047_LAPTOP_HP: - spec->input_mux = &cxt5047_hp_capture_source; spec->num_init_verbs = 2; spec->init_verbs[1] = cxt5047_hp_init_verbs; spec->mixers[0] = cxt5047_hp_mixers; -- cgit v1.2.3-59-g8ed1b From 3b628867f328cfe1ad4811d63961579874f87041 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2009 14:53:54 +0100 Subject: ALSA: hda - Remove superfluous verbs for Cxt5047 laptop-eapd model Remove superfluous verbs from cxt5047_toshiba_init_verbs[]. Also fix comments and minor coding style issues. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index d5d736ff7c6c..e9e47574c613 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1377,12 +1377,9 @@ static struct hda_verb cxt5047_init_verbs[] = { /* configuration for Toshiba Laptops */ static struct hda_verb cxt5047_toshiba_init_verbs[] = { - {0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x0 }, /* default on */ - /* pin sensing on HP and Mic jacks */ - {0x13, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, + {0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x0}, /* default off */ /* Speaker routing */ - {0x1d, AC_VERB_SET_CONNECT_SEL,0x1}, + {0x1d, AC_VERB_SET_CONNECT_SEL, 0x1}, {} }; -- cgit v1.2.3-59-g8ed1b From 5b3a7440cbabdda07cfb3dcf4a07e0115a3dff9a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2009 15:10:55 +0100 Subject: ALSA: hda - Fix / clean up init verbs for Cxt5047 codec Fix the initial connections of output pins 0x13 and 0x1d for Conexant 5047 codec to point to the mixer amp properly. Removed unneeded (doubly) verbs from arrays, also removed the unneeded changing of widget 0x1c, which is now completely unused. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 36 +++--------------------------------- 1 file changed, 3 insertions(+), 33 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e9e47574c613..71822140294d 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1165,7 +1165,7 @@ static int patch_cxt5045(struct hda_codec *codec) /* Conexant 5047 specific */ #define CXT5047_SPDIF_OUT 0x11 -static hda_nid_t cxt5047_dac_nids[2] = { 0x10, 0x1c }; +static hda_nid_t cxt5047_dac_nids[1] = { 0x10 }; /* 0x1c */ static hda_nid_t cxt5047_adc_nids[1] = { 0x12 }; static hda_nid_t cxt5047_capsrc_nids[1] = { 0x1a }; @@ -1216,9 +1216,6 @@ static void cxt5047_hp_automute(struct hda_codec *codec) bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits); - /* Mute/Unmute PCM 2 for good measure - some systems need this */ - snd_hda_codec_amp_stereo(codec, 0x1c, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); } /* mute internal speaker if HP is plugged */ @@ -1233,9 +1230,6 @@ static void cxt5047_hp2_automute(struct hda_codec *codec) bits = spec->hp_present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits); - /* Mute/Unmute PCM 2 for good measure - some systems need this */ - snd_hda_codec_amp_stereo(codec, 0x1c, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); } /* toggle input of built-in and mic jack appropriately */ @@ -1299,8 +1293,6 @@ static struct snd_kcontrol_new cxt5047_mixers[] = { HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT), HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT), - HDA_CODEC_VOLUME("PCM-2 Volume", 0x1c, 0x00, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM-2 Switch", 0x1c, 0x00, HDA_OUTPUT), HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x00, HDA_OUTPUT), HDA_CODEC_MUTE("Speaker Playback Switch", 0x1d, 0x00, HDA_OUTPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT), @@ -1356,8 +1348,8 @@ static struct hda_verb cxt5047_init_verbs[] = { {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_50 }, /* HP, Speaker */ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, - {0x13, AC_VERB_SET_CONNECT_SEL,0x1}, - {0x1d, AC_VERB_SET_CONNECT_SEL,0x0}, + {0x13, AC_VERB_SET_CONNECT_SEL, 0x0}, /* mixer(0x19) */ + {0x1d, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mixer(0x19) */ /* Record selector: Mic */ {0x12, AC_VERB_SET_CONNECT_SEL,0x03}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, @@ -1378,26 +1370,6 @@ static struct hda_verb cxt5047_init_verbs[] = { /* configuration for Toshiba Laptops */ static struct hda_verb cxt5047_toshiba_init_verbs[] = { {0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x0}, /* default off */ - /* Speaker routing */ - {0x1d, AC_VERB_SET_CONNECT_SEL, 0x1}, - {} -}; - -/* configuration for HP Laptops */ -static struct hda_verb cxt5047_hp_init_verbs[] = { - /* pin sensing on HP jack */ - {0x13, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, - /* 0x13 is actually shared by both HP and speaker; - * setting the connection to 0 (=0x19) makes the master volume control - * working mysteriouslly... - */ - {0x13, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Record selector: Ext Mic */ - {0x12, AC_VERB_SET_CONNECT_SEL,0x03}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_INPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x17}, - /* Speaker routing */ - {0x1d, AC_VERB_SET_CONNECT_SEL,0x1}, {} }; @@ -1604,8 +1576,6 @@ static int patch_cxt5047(struct hda_codec *codec) codec->patch_ops.unsol_event = cxt5047_hp2_unsol_event; break; case CXT5047_LAPTOP_HP: - spec->num_init_verbs = 2; - spec->init_verbs[1] = cxt5047_hp_init_verbs; spec->mixers[0] = cxt5047_hp_mixers; codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; codec->patch_ops.init = cxt5047_hp_init; -- cgit v1.2.3-59-g8ed1b From df481e41b963b7fc3d7e3543a0c7bb140a682146 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2009 15:35:35 +0100 Subject: ALSA: hda - Clean up Cxt5047 parser Clean up Conexant 5047 pareser code: - Split mixer elements to separate arrays to reduce the duplicated entires - Fix mixer element names to the standard ones - Remove unneeded cxt5047_hp2_unsol_event; the normal unsol_event handler works fine. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 89 +++++++++--------------------------------- 1 file changed, 19 insertions(+), 70 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 71822140294d..d60ccb5bb127 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1218,20 +1218,6 @@ static void cxt5047_hp_automute(struct hda_codec *codec) HDA_AMP_MUTE, bits); } -/* mute internal speaker if HP is plugged */ -static void cxt5047_hp2_automute(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - unsigned int bits; - - spec->hp_present = snd_hda_codec_read(codec, 0x13, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - - bits = spec->hp_present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); -} - /* toggle input of built-in and mic jack appropriately */ static void cxt5047_hp_automic(struct hda_codec *codec) { @@ -1269,47 +1255,14 @@ static void cxt5047_hp_unsol_event(struct hda_codec *codec, } } -/* unsolicited event for HP jack sensing - non-EAPD systems */ -static void cxt5047_hp2_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - res >>= 26; - switch (res) { - case CONEXANT_HP_EVENT: - cxt5047_hp2_automute(codec); - break; - case CONEXANT_MIC_EVENT: - cxt5047_hp_automic(codec); - break; - } -} - -static struct snd_kcontrol_new cxt5047_mixers[] = { - HDA_CODEC_VOLUME("Mic Bypass Capture Volume", 0x19, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Mic Bypass Capture Switch", 0x19, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Gain Volume", 0x1a, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Mic Gain Switch", 0x1a, 0x0, HDA_OUTPUT), +static struct snd_kcontrol_new cxt5047_base_mixers[] = { + HDA_CODEC_VOLUME("Mic Playback Volume", 0x19, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x19, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x1a, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT), HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x00, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x1d, 0x00, HDA_OUTPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x13, 0x00, HDA_OUTPUT), - - {} -}; - -static struct snd_kcontrol_new cxt5047_toshiba_mixers[] = { - HDA_CODEC_VOLUME("Mic Bypass Capture Volume", 0x19, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Mic Bypass Capture Switch", 0x19, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT), - HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x00, HDA_OUTPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", @@ -1322,22 +1275,14 @@ static struct snd_kcontrol_new cxt5047_toshiba_mixers[] = { {} }; -static struct snd_kcontrol_new cxt5047_hp_mixers[] = { - HDA_CODEC_VOLUME("Mic Bypass Capture Volume", 0x19, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Mic Bypass Capture Switch", 0x19,0x02,HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT), - HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT), +static struct snd_kcontrol_new cxt5047_hp_spk_mixers[] = { + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x00, HDA_OUTPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT), + {} +}; + +static struct snd_kcontrol_new cxt5047_hp_only_mixers[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x13, 0x00, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = cxt_eapd_info, - .get = cxt_eapd_get, - .put = cxt5047_hp_master_sw_put, - .private_value = 0x13, - }, { } /* end */ }; @@ -1559,7 +1504,7 @@ static int patch_cxt5047(struct hda_codec *codec) spec->adc_nids = cxt5047_adc_nids; spec->capsrc_nids = cxt5047_capsrc_nids; spec->num_mixers = 1; - spec->mixers[0] = cxt5047_mixers; + spec->mixers[0] = cxt5047_base_mixers; spec->num_init_verbs = 1; spec->init_verbs[0] = cxt5047_init_verbs; spec->spdif_route = 0; @@ -1573,18 +1518,22 @@ static int patch_cxt5047(struct hda_codec *codec) cxt5047_cfg_tbl); switch (board_config) { case CXT5047_LAPTOP: - codec->patch_ops.unsol_event = cxt5047_hp2_unsol_event; + spec->num_mixers = 2; + spec->mixers[1] = cxt5047_hp_spk_mixers; + codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; break; case CXT5047_LAPTOP_HP: - spec->mixers[0] = cxt5047_hp_mixers; + spec->num_mixers = 2; + spec->mixers[1] = cxt5047_hp_only_mixers; codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; codec->patch_ops.init = cxt5047_hp_init; break; case CXT5047_LAPTOP_EAPD: spec->input_mux = &cxt5047_toshiba_capture_source; + spec->num_mixers = 2; + spec->mixers[1] = cxt5047_hp_spk_mixers; spec->num_init_verbs = 2; spec->init_verbs[1] = cxt5047_toshiba_init_verbs; - spec->mixers[0] = cxt5047_toshiba_mixers; codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; break; #ifdef CONFIG_SND_DEBUG -- cgit v1.2.3-59-g8ed1b From 5d75bc557859805f00eeddb09d7cc8ffc7e5334e Mon Sep 17 00:00:00 2001 From: Gregorio Guidi Date: Thu, 12 Mar 2009 16:41:51 +0100 Subject: ALSA: hda - fix headphone settings and master volume (Conexant CX20551) Update the places where the 0x1d widget is used for Conexant 5047, fixing mismatch introduced after changing the connection. Signed-off-by: Gregorio Guidi Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index d60ccb5bb127..6cb184e9c2f1 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1196,7 +1196,7 @@ static int cxt5047_hp_master_sw_put(struct snd_kcontrol *kcontrol, * the headphone jack */ bits = (!spec->hp_present && spec->cur_eapd) ? 0 : HDA_AMP_MUTE; - snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0, + snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0x01, HDA_AMP_MUTE, bits); bits = spec->cur_eapd ? 0 : HDA_AMP_MUTE; snd_hda_codec_amp_stereo(codec, 0x13, HDA_OUTPUT, 0, @@ -1214,7 +1214,7 @@ static void cxt5047_hp_automute(struct hda_codec *codec) AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0, + snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0x01, HDA_AMP_MUTE, bits); } @@ -1276,7 +1276,7 @@ static struct snd_kcontrol_new cxt5047_base_mixers[] = { }; static struct snd_kcontrol_new cxt5047_hp_spk_mixers[] = { - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x00, HDA_OUTPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x01, HDA_OUTPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT), {} }; -- cgit v1.2.3-59-g8ed1b From 3b7523fc828e41b2988feb400704e01b67859d78 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Mar 2009 16:45:01 +0100 Subject: ALSA: hda - Add comments for the previous fix for conexant codecs Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 6cb184e9c2f1..bc016fade192 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1196,6 +1196,10 @@ static int cxt5047_hp_master_sw_put(struct snd_kcontrol *kcontrol, * the headphone jack */ bits = (!spec->hp_present && spec->cur_eapd) ? 0 : HDA_AMP_MUTE; + /* NOTE: Conexat codec needs the index for *OUTPUT* amp of + * pin widgets unlike other codecs. In this case, we need to + * set index 0x01 for the volume from the mixer amp 0x19. + */ snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0x01, HDA_AMP_MUTE, bits); bits = spec->cur_eapd ? 0 : HDA_AMP_MUTE; @@ -1214,6 +1218,7 @@ static void cxt5047_hp_automute(struct hda_codec *codec) AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0; + /* See the note in cxt5047_hp_master_sw_put */ snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0x01, HDA_AMP_MUTE, bits); } @@ -1276,6 +1281,7 @@ static struct snd_kcontrol_new cxt5047_base_mixers[] = { }; static struct snd_kcontrol_new cxt5047_hp_spk_mixers[] = { + /* See the note in cxt5047_hp_master_sw_put */ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x01, HDA_OUTPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT), {} -- cgit v1.2.3-59-g8ed1b From 9421f9543b3a0a870499f64498406003de8214b4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Mar 2009 17:06:07 +0100 Subject: ALSA: hda - Print multiple out-amp values of pin widgets on Conext codecs Add a flag to work around the non-standard amp-value handling on Conexant codecs. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.h | 3 +++ sound/pci/hda/hda_proc.c | 10 ++++++++-- sound/pci/hda/patch_conexant.c | 3 +++ 3 files changed, 14 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 2ea628478a91..079e1ab718d4 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -793,6 +793,9 @@ struct hda_codec { * status change * (e.g. Realtek codecs) */ + unsigned int pin_amp_workaround:1; /* pin out-amp takes index + * (e.g. Conexant codecs) + */ #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_proc.c b/sound/pci/hda/hda_proc.c index 144b85276d5a..93b25ba4d00b 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -554,8 +554,14 @@ static void print_codec_info(struct snd_info_entry *entry, snd_iprintf(buffer, " Amp-Out caps: "); print_amp_caps(buffer, codec, nid, HDA_OUTPUT); snd_iprintf(buffer, " Amp-Out vals: "); - print_amp_vals(buffer, codec, nid, HDA_OUTPUT, - wid_caps & AC_WCAP_STEREO, 1); + if (wid_type == AC_WID_PIN && + codec->pin_amp_workaround) + print_amp_vals(buffer, codec, nid, HDA_OUTPUT, + wid_caps & AC_WCAP_STEREO, + conn_len); + else + print_amp_vals(buffer, codec, nid, HDA_OUTPUT, + wid_caps & AC_WCAP_STEREO, 1); } switch (wid_type) { diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index bc016fade192..1f2ad76ca94b 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1066,6 +1066,7 @@ static int patch_cxt5045(struct hda_codec *codec) if (!spec) return -ENOMEM; codec->spec = spec; + codec->pin_amp_workaround = 1; spec->multiout.max_channels = 2; spec->multiout.num_dacs = ARRAY_SIZE(cxt5045_dac_nids); @@ -1501,6 +1502,7 @@ static int patch_cxt5047(struct hda_codec *codec) if (!spec) return -ENOMEM; codec->spec = spec; + codec->pin_amp_workaround = 1; spec->multiout.max_channels = 2; spec->multiout.num_dacs = ARRAY_SIZE(cxt5047_dac_nids); @@ -1847,6 +1849,7 @@ static int patch_cxt5051(struct hda_codec *codec) if (!spec) return -ENOMEM; codec->spec = spec; + codec->pin_amp_workaround = 1; codec->patch_ops = conexant_patch_ops; codec->patch_ops.init = cxt5051_init; -- cgit v1.2.3-59-g8ed1b From 307282c8990c5658604b9fda8a64a9a07079b850 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Mar 2009 18:17:58 +0100 Subject: ALSA: hda - Add model=vaio for STAC9872 Add the default pin config for model=vaio (in case of broken BIOS). Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 3 ++- sound/pci/hda/patch_sigmatel.c | 33 ++++++++++++++++++++++++++-- 2 files changed, 33 insertions(+), 3 deletions(-) (limited to 'sound/pci') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index f9253ea3c19f..8eec05bc079e 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -368,4 +368,5 @@ STAC92HD83* STAC9872 ======== - N/A + vaio VAIO laptop without SPDIF + auto BIOS setup (default) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 72c87aa20bd9..e06fc7decd31 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -155,6 +155,12 @@ enum { STAC_927X_MODELS }; +enum { + STAC_9872_AUTO, + STAC_9872_VAIO, + STAC_9872_MODELS +}; + struct sigmatel_event { hda_nid_t nid; unsigned char type; @@ -5588,6 +5594,25 @@ static hda_nid_t stac9872_mux_nids[] = { 0x15 }; +static unsigned int stac9872_vaio_pin_configs[9] = { + 0x03211020, 0x411111f0, 0x411111f0, 0x03a15030, + 0x411111f0, 0x90170110, 0x411111f0, 0x411111f0, + 0x90a7013e +}; + +static const char *stac9872_models[STAC_9872_MODELS] = { + [STAC_9872_AUTO] = "auto", + [STAC_9872_VAIO] = "vaio", +}; + +static unsigned int *stac9872_brd_tbl[STAC_9872_MODELS] = { + [STAC_9872_VAIO] = stac9872_vaio_pin_configs, +}; + +static struct snd_pci_quirk stac9872_cfg_tbl[] = { + {} /* terminator */ +}; + static int patch_stac9872(struct hda_codec *codec) { struct sigmatel_spec *spec; @@ -5598,11 +5623,15 @@ static int patch_stac9872(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; -#if 0 /* no model right now */ spec->board_config = snd_hda_check_board_config(codec, STAC_9872_MODELS, stac9872_models, stac9872_cfg_tbl); -#endif + if (spec->board_config < 0) + snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9872, " + "using BIOS defaults\n"); + else + stac92xx_set_config_regs(codec, + stac9872_brd_tbl[spec->board_config]); spec->num_pins = ARRAY_SIZE(stac9872_pin_nids); spec->pin_nids = stac9872_pin_nids; -- cgit v1.2.3-59-g8ed1b From bb6ac72fb19c6676eb8bafa8e3b8bf970a2294a2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Mar 2009 09:02:42 +0100 Subject: ALSA: hda - power up before codec initialization Change the power state of each widget before starting the initialization work so that all verbs are executed properly. Also, keep power-up during hwdep reconfiguration. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 15 ++++++++------- sound/pci/hda/hda_hwdep.c | 14 +++++++++----- 2 files changed, 17 insertions(+), 12 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 1885e7649101..cf6339436de1 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -842,6 +842,9 @@ static void snd_hda_codec_free(struct hda_codec *codec) kfree(codec); } +static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state); + /** * snd_hda_codec_new - create a HDA codec * @bus: the bus to assign @@ -941,6 +944,11 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr if (bus->modelname) codec->modelname = kstrdup(bus->modelname, GFP_KERNEL); + /* power-up all before initialization */ + hda_set_power_state(codec, + codec->afg ? codec->afg : codec->mfg, + AC_PWRST_D0); + if (do_init) { err = snd_hda_codec_configure(codec); if (err < 0) @@ -2413,19 +2421,12 @@ EXPORT_SYMBOL_HDA(snd_hda_build_controls); int snd_hda_codec_build_controls(struct hda_codec *codec) { int err = 0; - /* fake as if already powered-on */ - hda_keep_power_on(codec); - /* then fire up */ - hda_set_power_state(codec, - codec->afg ? codec->afg : codec->mfg, - AC_PWRST_D0); hda_exec_init_verbs(codec); /* continue to initialize... */ if (codec->patch_ops.init) err = codec->patch_ops.init(codec); if (!err && codec->patch_ops.build_controls) err = codec->patch_ops.build_controls(codec); - snd_hda_power_down(codec); if (err < 0) return err; return 0; diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 1e3ccc740afc..1c57505c2874 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -176,25 +176,29 @@ static int reconfig_codec(struct hda_codec *codec) { int err; + snd_hda_power_up(codec); snd_printk(KERN_INFO "hda-codec: reconfiguring\n"); err = snd_hda_codec_reset(codec); if (err < 0) { snd_printk(KERN_ERR "The codec is being used, can't reconfigure.\n"); - return err; + goto error; } err = snd_hda_codec_configure(codec); if (err < 0) - return err; + goto error; /* rebuild PCMs */ err = snd_hda_codec_build_pcms(codec); if (err < 0) - return err; + goto error; /* rebuild mixers */ err = snd_hda_codec_build_controls(codec); if (err < 0) - return err; - return snd_card_register(codec->bus->card); + goto error; + err = snd_card_register(codec->bus->card); + error: + snd_hda_power_down(codec); + return err; } /* -- cgit v1.2.3-59-g8ed1b From 58d8395b74f78a2f4225c5faea8b5bffb8af1cf9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Mar 2009 17:04:34 +0100 Subject: ALSA: hda - Add another HP model with IDT92HD71bx codec HP laptops require GPIO0 on as EAPD. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index e06fc7decd31..4da72403fc87 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1855,6 +1855,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "DFI LanParty", STAC_92HD71BXX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_92HD71BXX_REF), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3080, + "HP", STAC_HP_DV5), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x30f0, "HP dv4-7", STAC_HP_DV5), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3600, -- cgit v1.2.3-59-g8ed1b From b8dbed0f095263b9ced5bd2e6d54743a7fa13f1b Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 16 Mar 2009 14:56:58 +0100 Subject: ALSA: snd-hda-intel: Fix ALC662/ALC663 Beep Amplifier Index ALC662/663 codecs have Beep Amplifier Index 0x04 not 0x05 in 0x0b NID. Confirmed by testing on real hardware. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b794cba494c3..672103d84ffc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -16951,7 +16951,7 @@ static int patch_alc662(struct hda_codec *codec) if (!spec->cap_mixer) set_capture_mixer(spec); - set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); spec->vmaster_nid = 0x02; -- cgit v1.2.3-59-g8ed1b From b9591448e5160ccd353d8547ade018cfdf2b3e09 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 Mar 2009 15:25:00 +0100 Subject: ALSA: hda - Fix ALC662 beep again The previous commit breaks the (digital-) beep on ALC662. ALC662 has the connection index 0x05 while ALC662 and ALC272 have the index 0x04 for the beep widget. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 672103d84ffc..5ad0f8d72ddb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -16951,7 +16951,10 @@ static int patch_alc662(struct hda_codec *codec) if (!spec->cap_mixer) set_capture_mixer(spec); - set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); + if (codec->vendor_id == 0x10ec0662) + set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + else + set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); spec->vmaster_nid = 0x02; -- cgit v1.2.3-59-g8ed1b From ee5047102cf632351c418060bfbe3b6eb5c42e7b Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 17 Mar 2009 14:30:31 +0100 Subject: ALSA: snd-hda-intel - add checks for invalid values to *query_supported_pcm() If ratesp or formatsp values are zero, wrong values are passed to ALSA's the PCM midlevel code. The bug is showed more later than expected. Also, clean a bit the code. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 33 +++++++++++++++++++++++++-------- 1 file changed, 25 insertions(+), 8 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index cf6339436de1..b90a2400f53d 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2539,12 +2539,11 @@ EXPORT_SYMBOL_HDA(snd_hda_calc_stream_format); static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, u32 *ratesp, u64 *formatsp, unsigned int *bpsp) { - int i; - unsigned int val, streams; + unsigned int i, val, wcaps; val = 0; - if (nid != codec->afg && - (get_wcaps(codec, nid) & AC_WCAP_FORMAT_OVRD)) { + wcaps = get_wcaps(codec, nid); + if (nid != codec->afg && (wcaps & AC_WCAP_FORMAT_OVRD)) { val = snd_hda_param_read(codec, nid, AC_PAR_PCM); if (val == -1) return -EIO; @@ -2558,15 +2557,20 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, if (val & (1 << i)) rates |= rate_bits[i].alsa_bits; } + if (rates == 0) { + snd_printk(KERN_ERR "hda_codec: rates == 0 " + "(nid=0x%x, val=0x%x, ovrd=%i)\n", + nid, val, + (wcaps & AC_WCAP_FORMAT_OVRD) ? 1 : 0); + return -EIO; + } *ratesp = rates; } if (formatsp || bpsp) { u64 formats = 0; - unsigned int bps; - unsigned int wcaps; + unsigned int streams, bps; - wcaps = get_wcaps(codec, nid); streams = snd_hda_param_read(codec, nid, AC_PAR_STREAM); if (streams == -1) return -EIO; @@ -2619,6 +2623,15 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, formats |= SNDRV_PCM_FMTBIT_U8; bps = 8; } + if (formats == 0) { + snd_printk(KERN_ERR "hda_codec: formats == 0 " + "(nid=0x%x, val=0x%x, ovrd=%i, " + "streams=0x%x)\n", + nid, val, + (wcaps & AC_WCAP_FORMAT_OVRD) ? 1 : 0, + streams); + return -EIO; + } if (formatsp) *formatsp = formats; if (bpsp) @@ -2734,12 +2747,16 @@ static int hda_pcm_default_cleanup(struct hda_pcm_stream *hinfo, static int set_pcm_default_values(struct hda_codec *codec, struct hda_pcm_stream *info) { + int err; + /* query support PCM information from the given NID */ if (info->nid && (!info->rates || !info->formats)) { - snd_hda_query_supported_pcm(codec, info->nid, + err = snd_hda_query_supported_pcm(codec, info->nid, info->rates ? NULL : &info->rates, info->formats ? NULL : &info->formats, info->maxbps ? NULL : &info->maxbps); + if (err < 0) + return err; } if (info->ops.open == NULL) info->ops.open = hda_pcm_default_open_close; -- cgit v1.2.3-59-g8ed1b From 97b71c94d691728b82052e9c4d6286fbc9965d7f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 18 Mar 2009 15:09:13 +0100 Subject: ALSA: hda - Don't reset BDL unnecessarily So far, the prepare callback is called multiple times, BDL entries are reset and re-programmed at each time. This patch adds the check to avoid the reset of BDL entries when the same parameters are used. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 46 ++++++++++++++++++++++++++++++++-------------- 1 file changed, 32 insertions(+), 14 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6bcf5af6edce..ba97795d89c4 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1076,8 +1076,7 @@ static int azx_setup_periods(struct azx *chip, azx_sd_writel(azx_dev, SD_BDLPL, 0); azx_sd_writel(azx_dev, SD_BDLPU, 0); - period_bytes = snd_pcm_lib_period_bytes(substream); - azx_dev->period_bytes = period_bytes; + period_bytes = azx_dev->period_bytes; periods = azx_dev->bufsize / period_bytes; /* program the initial BDL entries */ @@ -1124,9 +1123,6 @@ static int azx_setup_periods(struct azx *chip, error: snd_printk(KERN_ERR "Too many BDL entries: buffer=%d, period=%d\n", azx_dev->bufsize, period_bytes); - /* reset */ - azx_sd_writel(azx_dev, SD_BDLPL, 0); - azx_sd_writel(azx_dev, SD_BDLPU, 0); return -EINVAL; } @@ -1429,6 +1425,11 @@ static int azx_pcm_close(struct snd_pcm_substream *substream) static int azx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { + struct azx_dev *azx_dev = get_azx_dev(substream); + + azx_dev->bufsize = 0; + azx_dev->period_bytes = 0; + azx_dev->format_val = 0; return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); } @@ -1443,6 +1444,9 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream) azx_sd_writel(azx_dev, SD_BDLPL, 0); azx_sd_writel(azx_dev, SD_BDLPU, 0); azx_sd_writel(azx_dev, SD_CTL, 0); + azx_dev->bufsize = 0; + azx_dev->period_bytes = 0; + azx_dev->format_val = 0; hinfo->ops.cleanup(hinfo, apcm->codec, substream); @@ -1456,23 +1460,37 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) struct azx_dev *azx_dev = get_azx_dev(substream); struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int bufsize, period_bytes, format_val; + int err; - azx_dev->bufsize = snd_pcm_lib_buffer_bytes(substream); - azx_dev->format_val = snd_hda_calc_stream_format(runtime->rate, - runtime->channels, - runtime->format, - hinfo->maxbps); - if (!azx_dev->format_val) { + format_val = snd_hda_calc_stream_format(runtime->rate, + runtime->channels, + runtime->format, + hinfo->maxbps); + if (!format_val) { snd_printk(KERN_ERR SFX "invalid format_val, rate=%d, ch=%d, format=%d\n", runtime->rate, runtime->channels, runtime->format); return -EINVAL; } + bufsize = snd_pcm_lib_buffer_bytes(substream); + period_bytes = snd_pcm_lib_period_bytes(substream); + snd_printdd("azx_pcm_prepare: bufsize=0x%x, format=0x%x\n", - azx_dev->bufsize, azx_dev->format_val); - if (azx_setup_periods(chip, substream, azx_dev) < 0) - return -EINVAL; + bufsize, format_val); + + if (bufsize != azx_dev->bufsize || + period_bytes != azx_dev->period_bytes || + format_val != azx_dev->format_val) { + azx_dev->bufsize = bufsize; + azx_dev->period_bytes = period_bytes; + azx_dev->format_val = format_val; + err = azx_setup_periods(chip, substream, azx_dev); + if (err < 0) + return err; + } + azx_setup_controller(chip, azx_dev); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) azx_dev->fifo_size = azx_sd_readw(azx_dev, SD_FIFOSIZE) + 1; -- cgit v1.2.3-59-g8ed1b From 1dddab400b7ad028b21d7d5b060e4a068d6d3cd9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 18 Mar 2009 15:15:37 +0100 Subject: ALSA: hda - Don't reset stream at each prepare callback Don't reset the stream at each prepare callback but do it only once after the open. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 33 ++++++++++++++++++++++----------- 1 file changed, 22 insertions(+), 11 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ba97795d89c4..8b2e4160de8d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -859,13 +859,18 @@ static void azx_stream_start(struct azx *chip, struct azx_dev *azx_dev) SD_CTL_DMA_START | SD_INT_MASK); } -/* stop a stream */ -static void azx_stream_stop(struct azx *chip, struct azx_dev *azx_dev) +/* stop DMA */ +static void azx_stream_clear(struct azx *chip, struct azx_dev *azx_dev) { - /* stop DMA */ azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) & ~(SD_CTL_DMA_START | SD_INT_MASK)); azx_sd_writeb(azx_dev, SD_STS, SD_INT_MASK); /* to be sure */ +} + +/* stop a stream */ +static void azx_stream_stop(struct azx *chip, struct azx_dev *azx_dev) +{ + azx_stream_clear(chip, azx_dev); /* disable SIE */ azx_writeb(chip, INTCTL, azx_readb(chip, INTCTL) & ~(1 << azx_dev->index)); @@ -1126,18 +1131,14 @@ static int azx_setup_periods(struct azx *chip, return -EINVAL; } -/* - * set up the SD for streaming - */ -static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev) +/* reset stream */ +static void azx_stream_reset(struct azx *chip, struct azx_dev *azx_dev) { unsigned char val; int timeout; - /* make sure the run bit is zero for SD */ - azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) & - ~SD_CTL_DMA_START); - /* reset stream */ + azx_stream_clear(chip, azx_dev); + azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) | SD_CTL_STREAM_RESET); udelay(3); @@ -1154,7 +1155,15 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev) while (((val = azx_sd_readb(azx_dev, SD_CTL)) & SD_CTL_STREAM_RESET) && --timeout) ; +} +/* + * set up the SD for streaming + */ +static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev) +{ + /* make sure the run bit is zero for SD */ + azx_stream_clear(chip, azx_dev); /* program the stream_tag */ azx_sd_writel(azx_dev, SD_CTL, (azx_sd_readl(azx_dev, SD_CTL) & ~SD_CTL_STREAM_TAG_MASK)| @@ -1399,6 +1408,8 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) runtime->private_data = azx_dev; snd_pcm_set_sync(substream); mutex_unlock(&chip->open_mutex); + + azx_stream_reset(chip, azx_dev); return 0; } -- cgit v1.2.3-59-g8ed1b From 07a1e81355245ca65ab16c7b4ae2332e52ed7acd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Mar 2009 17:08:19 +0100 Subject: ALSA: hda - Don't show the current connection for power widgets The power-widgets have no connection selection, so skip the check in proc output, too. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_proc.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 93b25ba4d00b..639cf0edaa98 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -399,8 +399,10 @@ static void print_conn_list(struct snd_info_buffer *buffer, { int c, curr = -1; - if (conn_len > 1 && wid_type != AC_WID_AUD_MIX && - wid_type != AC_WID_VOL_KNB) + if (conn_len > 1 && + wid_type != AC_WID_AUD_MIX && + wid_type != AC_WID_VOL_KNB && + wid_type != AC_WID_POWER) curr = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0); snd_iprintf(buffer, " Connection: %d\n", conn_len); -- cgit v1.2.3-59-g8ed1b From 2d864c499a77129dc6aa4f7552ddf2885e4a9c47 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Mar 2009 12:52:47 +0100 Subject: ALSA: hda - Detect digital-mic inputs on ALC663 / ALC272 Fix the detection of digital-mic inputs on ALC663 / ALC272 codecs in the auto-detection mode. The automatic mic switch via plugging isn't implemented yet, though. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 63 ++++++++++++++++++++++++++++++++----------- 1 file changed, 47 insertions(+), 16 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5ad0f8d72ddb..b69d9864f6f3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -16725,26 +16725,58 @@ static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin, return 0; } +/* return the index of the src widget from the connection list of the nid. + * return -1 if not found + */ +static int alc662_input_pin_idx(struct hda_codec *codec, hda_nid_t nid, + hda_nid_t src) +{ + hda_nid_t conn_list[HDA_MAX_CONNECTIONS]; + int i, conns; + + conns = snd_hda_get_connections(codec, nid, conn_list, + ARRAY_SIZE(conn_list)); + if (conns < 0) + return -1; + for (i = 0; i < conns; i++) + if (conn_list[i] == src) + return i; + return -1; +} + +static int alc662_is_input_pin(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + return (pincap & AC_PINCAP_IN) != 0; +} + /* create playback/capture controls for input pins */ -static int alc662_auto_create_analog_input_ctls(struct alc_spec *spec, +static int alc662_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { + struct alc_spec *spec = codec->spec; struct hda_input_mux *imux = &spec->private_imux[0]; int i, err, idx; for (i = 0; i < AUTO_PIN_LAST; i++) { - if (alc880_is_input_pin(cfg->input_pins[i])) { - idx = alc880_input_pin_idx(cfg->input_pins[i]); - err = new_analog_input(spec, cfg->input_pins[i], - auto_pin_cfg_labels[i], - idx, 0x0b); - if (err < 0) - return err; - imux->items[imux->num_items].label = - auto_pin_cfg_labels[i]; - imux->items[imux->num_items].index = - alc880_input_pin_idx(cfg->input_pins[i]); - imux->num_items++; + if (alc662_is_input_pin(codec, cfg->input_pins[i])) { + idx = alc662_input_pin_idx(codec, 0x0b, + cfg->input_pins[i]); + if (idx >= 0) { + err = new_analog_input(spec, cfg->input_pins[i], + auto_pin_cfg_labels[i], + idx, 0x0b); + if (err < 0) + return err; + } + idx = alc662_input_pin_idx(codec, 0x22, + cfg->input_pins[i]); + if (idx >= 0) { + imux->items[imux->num_items].label = + auto_pin_cfg_labels[i]; + imux->items[imux->num_items].index = idx; + imux->num_items++; + } } } return 0; @@ -16794,7 +16826,6 @@ static void alc662_auto_init_hp_out(struct hda_codec *codec) alc662_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); } -#define alc662_is_input_pin(nid) alc880_is_input_pin(nid) #define ALC662_PIN_CD_NID ALC880_PIN_CD_NID static void alc662_auto_init_analog_input(struct hda_codec *codec) @@ -16804,7 +16835,7 @@ static void alc662_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; - if (alc662_is_input_pin(nid)) { + if (alc662_is_input_pin(codec, nid)) { alc_set_input_pin(codec, nid, i); if (nid != ALC662_PIN_CD_NID) snd_hda_codec_write(codec, nid, 0, @@ -16844,7 +16875,7 @@ static int alc662_parse_auto_config(struct hda_codec *codec) "Headphone"); if (err < 0) return err; - err = alc662_auto_create_analog_input_ctls(spec, &spec->autocfg); + err = alc662_auto_create_analog_input_ctls(codec, &spec->autocfg); if (err < 0) return err; -- cgit v1.2.3-59-g8ed1b From 234b4346a064f8a2a488da10b3c1e640fb778a17 Mon Sep 17 00:00:00 2001 From: Pascal de Bruijn Date: Mon, 23 Mar 2009 11:15:59 +0100 Subject: ALSA: hda - Add function id to proc output This patch does two things: Output Intel HDA Function Id in /proc/asound/cardX/codec#X Align Vendor/Subsystem/Revision Ids to 8 characters, front-padded with zeros Before: Vendor Id: 0x11d41884 Subsystem Id: 0x103c281a Revision Id: 0x100100 After: Function Id: 0x1 Vendor Id: 0x11d41884 Subsystem Id: 0x103c281a Revision Id: 0x0100100 As report on the Kernel Bugzilla #12888 Signed-off-by: Pascal de Bruijn Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 6 +++--- sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/hda_proc.c | 5 +++-- 3 files changed, 7 insertions(+), 5 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b90a2400f53d..1b5575ecb0a4 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -647,9 +647,9 @@ static void /*__devinit*/ setup_fg_nodes(struct hda_codec *codec) total_nodes = snd_hda_get_sub_nodes(codec, AC_NODE_ROOT, &nid); for (i = 0; i < total_nodes; i++, nid++) { - unsigned int func; - func = snd_hda_param_read(codec, nid, AC_PAR_FUNCTION_TYPE); - switch (func & 0xff) { + codec->function_id = snd_hda_param_read(codec, nid, + AC_PAR_FUNCTION_TYPE) & 0xff; + switch (codec->function_id) { case AC_GRP_AUDIO_FUNCTION: codec->afg = nid; break; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 079e1ab718d4..2fdecf4b0eb6 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -739,6 +739,7 @@ struct hda_codec { hda_nid_t mfg; /* MFG node id */ /* ids */ + u32 function_id; u32 vendor_id; u32 subsystem_id; u32 revision_id; diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 639cf0edaa98..93d7499350c6 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -469,8 +469,9 @@ static void print_codec_info(struct snd_info_entry *entry, snd_iprintf(buffer, "Codec: %s\n", codec->name ? codec->name : "Not Set"); snd_iprintf(buffer, "Address: %d\n", codec->addr); - snd_iprintf(buffer, "Vendor Id: 0x%x\n", codec->vendor_id); - snd_iprintf(buffer, "Subsystem Id: 0x%x\n", codec->subsystem_id); + snd_iprintf(buffer, "Function Id: 0x%x\n", codec->function_id); + snd_iprintf(buffer, "Vendor Id: 0x%08x\n", codec->vendor_id); + snd_iprintf(buffer, "Subsystem Id: 0x%08x\n", codec->subsystem_id); snd_iprintf(buffer, "Revision Id: 0x%x\n", codec->revision_id); if (codec->mfg) -- cgit v1.2.3-59-g8ed1b From 52ca15b7c0c711eb37f5e4b769e8488e5c516d43 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Mar 2009 12:51:55 +0100 Subject: ALSA: hda - Avoid output amp manipulation to digital mic pins Don't set amp-out values to pins without PINCAP_OUT capability, which are usually assigned for digital mics on ALC663/ALC272. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b69d9864f6f3..965a531d2fba 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -16750,6 +16750,12 @@ static int alc662_is_input_pin(struct hda_codec *codec, hda_nid_t nid) return (pincap & AC_PINCAP_IN) != 0; } +static int alc662_is_output_pin(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + return (pincap & AC_PINCAP_OUT) != 0; +} + /* create playback/capture controls for input pins */ static int alc662_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) @@ -16837,7 +16843,8 @@ static void alc662_auto_init_analog_input(struct hda_codec *codec) hda_nid_t nid = spec->autocfg.input_pins[i]; if (alc662_is_input_pin(codec, nid)) { alc_set_input_pin(codec, nid, i); - if (nid != ALC662_PIN_CD_NID) + if (nid != ALC662_PIN_CD_NID && + alc662_is_output_pin(codec, nid)) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); -- cgit v1.2.3-59-g8ed1b From 1327a32b878b5ed2113c63557b6f4f949f821857 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Mar 2009 13:07:47 +0100 Subject: ALSA: hda - Cache pin-cap values Added snd_hda_query_pin_caps() to read and cache pin-cap values to avoid too frequently issuing the same verbs. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 16 ++++++++++++++++ sound/pci/hda/hda_generic.c | 2 +- sound/pci/hda/hda_local.h | 1 + sound/pci/hda/patch_realtek.c | 6 +++--- sound/pci/hda/patch_sigmatel.c | 7 +++---- 5 files changed, 24 insertions(+), 8 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 1b5575ecb0a4..0f70d2d102e0 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1052,6 +1052,7 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_cleanup_stream); /* FIXME: more better hash key? */ #define HDA_HASH_KEY(nid,dir,idx) (u32)((nid) + ((idx) << 16) + ((dir) << 24)) +#define HDA_HASH_PINCAP_KEY(nid) (u32)((nid) + (0x02 << 24)) #define INFO_AMP_CAPS (1<<0) #define INFO_AMP_VOL(ch) (1 << (1 + (ch))) @@ -1142,6 +1143,21 @@ int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, } EXPORT_SYMBOL_HDA(snd_hda_override_amp_caps); +u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid) +{ + struct hda_amp_info *info; + + info = get_alloc_amp_hash(codec, HDA_HASH_PINCAP_KEY(nid)); + if (!info) + return 0; + if (!info->head.val) { + info->amp_caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + info->head.val |= INFO_AMP_CAPS; + } + return info->amp_caps; +} +EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps); + /* * read the current volume to info * if the cache exists, read the cache value. diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 2c81a683e8f8..1d5797a96682 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -144,7 +144,7 @@ static int add_new_node(struct hda_codec *codec, struct hda_gspec *spec, hda_nid node->type = (node->wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; if (node->type == AC_WID_PIN) { - node->pin_caps = snd_hda_param_read(codec, node->nid, AC_PAR_PIN_CAP); + node->pin_caps = snd_hda_query_pin_caps(codec, node->nid); node->pin_ctl = snd_hda_codec_read(codec, node->nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); node->def_cfg = snd_hda_codec_get_pincfg(codec, node->nid); } diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 27428c718fd7..83349013b4df 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -411,6 +411,7 @@ static inline u32 get_wcaps(struct hda_codec *codec, hda_nid_t nid) u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction); int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps); +u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid); int snd_hda_ctl_add(struct hda_codec *codec, struct snd_kcontrol *kctl); void snd_hda_ctls_clear(struct hda_codec *codec); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 965a531d2fba..bf7e64e2c468 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -770,7 +770,7 @@ static void alc_set_input_pin(struct hda_codec *codec, hda_nid_t nid, if (auto_pin_type <= AUTO_PIN_FRONT_MIC) { unsigned int pincap; - pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + pincap = snd_hda_query_pin_caps(codec, nid); pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT; if (pincap & AC_PINCAP_VREF_80) val = PIN_VREF80; @@ -16746,13 +16746,13 @@ static int alc662_input_pin_idx(struct hda_codec *codec, hda_nid_t nid, static int alc662_is_input_pin(struct hda_codec *codec, hda_nid_t nid) { - unsigned int pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + unsigned int pincap = snd_hda_query_pin_caps(codec, nid); return (pincap & AC_PINCAP_IN) != 0; } static int alc662_is_output_pin(struct hda_codec *codec, hda_nid_t nid) { - unsigned int pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + unsigned int pincap = snd_hda_query_pin_caps(codec, nid); return (pincap & AC_PINCAP_OUT) != 0; } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 4da72403fc87..b1c180a9e9be 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2537,8 +2537,7 @@ static int stac92xx_build_pcms(struct hda_codec *codec) static unsigned int stac92xx_get_vref(struct hda_codec *codec, hda_nid_t nid) { - unsigned int pincap = snd_hda_param_read(codec, nid, - AC_PAR_PIN_CAP); + unsigned int pincap = snd_hda_query_pin_caps(codec, nid); pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT; if (pincap & AC_PINCAP_VREF_100) return AC_PINCTL_VREF_100; @@ -2799,7 +2798,7 @@ static hda_nid_t check_line_out_switch(struct hda_codec *codec) if (cfg->line_out_type != AUTO_PIN_LINE_OUT) return 0; nid = cfg->input_pins[AUTO_PIN_LINE]; - pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + pincap = snd_hda_query_pin_caps(codec, nid); if (pincap & AC_PINCAP_OUT) return nid; return 0; @@ -2822,7 +2821,7 @@ static hda_nid_t check_mic_out_switch(struct hda_codec *codec) /* some laptops have an internal analog microphone * which can't be used as a output */ if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) { - pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + pincap = snd_hda_query_pin_caps(codec, nid); if (pincap & AC_PINCAP_OUT) return nid; } -- cgit v1.2.3-59-g8ed1b From e82c025b501a1ca62dec40989817dbb17c0b9167 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Mar 2009 15:17:38 +0100 Subject: ALSA: hda - Fix the wrong pin-cap check in patch_realtek.c The check for the amp-output must be done for widget-caps rather than pin-caps as implemented in the recent change... Simply a thinko. Also, add the similar checks to all places that put output-amp mutes in the initialization. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 20 +++++++++----------- 1 file changed, 9 insertions(+), 11 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bf7e64e2c468..8dcbb04e57b5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4207,7 +4207,8 @@ static void alc880_auto_init_analog_input(struct hda_codec *codec) hda_nid_t nid = spec->autocfg.input_pins[i]; if (alc880_is_input_pin(nid)) { alc_set_input_pin(codec, nid, i); - if (nid != ALC880_PIN_CD_NID) + if (nid != ALC880_PIN_CD_NID && + (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); @@ -5673,7 +5674,8 @@ static void alc260_auto_init_analog_input(struct hda_codec *codec) hda_nid_t nid = spec->autocfg.input_pins[i]; if (nid >= 0x12) { alc_set_input_pin(codec, nid, i); - if (nid != ALC260_PIN_CD_NID) + if (nid != ALC260_PIN_CD_NID && + (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); @@ -9153,7 +9155,8 @@ static void alc883_auto_init_analog_input(struct hda_codec *codec) hda_nid_t nid = spec->autocfg.input_pins[i]; if (alc883_is_input_pin(nid)) { alc_set_input_pin(codec, nid, i); - if (nid != ALC883_PIN_CD_NID) + if (nid != ALC883_PIN_CD_NID && + (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); @@ -14880,7 +14883,8 @@ static void alc861vd_auto_init_analog_input(struct hda_codec *codec) hda_nid_t nid = spec->autocfg.input_pins[i]; if (alc861vd_is_input_pin(nid)) { alc_set_input_pin(codec, nid, i); - if (nid != ALC861VD_PIN_CD_NID) + if (nid != ALC861VD_PIN_CD_NID && + (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); @@ -16750,12 +16754,6 @@ static int alc662_is_input_pin(struct hda_codec *codec, hda_nid_t nid) return (pincap & AC_PINCAP_IN) != 0; } -static int alc662_is_output_pin(struct hda_codec *codec, hda_nid_t nid) -{ - unsigned int pincap = snd_hda_query_pin_caps(codec, nid); - return (pincap & AC_PINCAP_OUT) != 0; -} - /* create playback/capture controls for input pins */ static int alc662_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) @@ -16844,7 +16842,7 @@ static void alc662_auto_init_analog_input(struct hda_codec *codec) if (alc662_is_input_pin(codec, nid)) { alc_set_input_pin(codec, nid, i); if (nid != ALC662_PIN_CD_NID && - alc662_is_output_pin(codec, nid)) + (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); -- cgit v1.2.3-59-g8ed1b From a23b688f4d5c2490a50677b30011a677d8edf3d0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Mar 2009 15:21:36 +0100 Subject: ALSA: hda - Don't create empty/single-item input source In patch_realtek.c, don't create empty or single-item "Input Source" control elements that are simply superfluous. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 47 ++++++++++++++++++++++++++++++++----------- 1 file changed, 35 insertions(+), 12 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8dcbb04e57b5..7a3c6db6d5be 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1595,8 +1595,7 @@ static int alc_cap_sw_put(struct snd_kcontrol *kcontrol, snd_hda_mixer_amp_switch_put); } -#define DEFINE_CAPMIX(num) \ -static struct snd_kcontrol_new alc_capture_mixer ## num[] = { \ +#define _DEFINE_CAPMIX(num) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = "Capture Switch", \ @@ -1617,7 +1616,9 @@ static struct snd_kcontrol_new alc_capture_mixer ## num[] = { \ .get = alc_cap_vol_get, \ .put = alc_cap_vol_put, \ .tlv = { .c = alc_cap_vol_tlv }, \ - }, \ + } + +#define _DEFINE_CAPSRC(num) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ /* .name = "Capture Source", */ \ @@ -1626,15 +1627,28 @@ static struct snd_kcontrol_new alc_capture_mixer ## num[] = { \ .info = alc_mux_enum_info, \ .get = alc_mux_enum_get, \ .put = alc_mux_enum_put, \ - }, \ - { } /* end */ \ + } + +#define DEFINE_CAPMIX(num) \ +static struct snd_kcontrol_new alc_capture_mixer ## num[] = { \ + _DEFINE_CAPMIX(num), \ + _DEFINE_CAPSRC(num), \ + { } /* end */ \ +} + +#define DEFINE_CAPMIX_NOSRC(num) \ +static struct snd_kcontrol_new alc_capture_mixer_nosrc ## num[] = { \ + _DEFINE_CAPMIX(num), \ + { } /* end */ \ } /* up to three ADCs */ DEFINE_CAPMIX(1); DEFINE_CAPMIX(2); DEFINE_CAPMIX(3); - +DEFINE_CAPMIX_NOSRC(1); +DEFINE_CAPMIX_NOSRC(2); +DEFINE_CAPMIX_NOSRC(3); /* * ALC880 5-stack model @@ -4298,13 +4312,22 @@ static void alc880_auto_init(struct hda_codec *codec) static void set_capture_mixer(struct alc_spec *spec) { - static struct snd_kcontrol_new *caps[3] = { - alc_capture_mixer1, - alc_capture_mixer2, - alc_capture_mixer3, + static struct snd_kcontrol_new *caps[2][3] = { + { alc_capture_mixer_nosrc1, + alc_capture_mixer_nosrc2, + alc_capture_mixer_nosrc3 }, + { alc_capture_mixer1, + alc_capture_mixer2, + alc_capture_mixer3 }, }; - if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3) - spec->cap_mixer = caps[spec->num_adc_nids - 1]; + if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3) { + int mux; + if (spec->input_mux && spec->input_mux->num_items > 1) + mux = 1; + else + mux = 0; + spec->cap_mixer = caps[mux][spec->num_adc_nids - 1]; + } } #define set_beep_amp(spec, nid, idx, dir) \ -- cgit v1.2.3-59-g8ed1b From 14bafe3278e5da952a6586a5a9a9d286566049ed Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Mar 2009 16:35:39 +0100 Subject: ALSA: hda - Use cached calls to get widget caps and pin caps Replace with the standard function calls to use caches for reading the widget caps and pin caps. hda_proc.c is still using the direct verbs to get raw values as much as possible. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 3 +-- sound/pci/hda/patch_sigmatel.c | 3 +-- 2 files changed, 2 insertions(+), 4 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0f70d2d102e0..a4e5e5952115 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2321,8 +2321,7 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, * don't power down the widget if it controls * eapd and EAPD_BTLENABLE is set. */ - pincap = snd_hda_param_read(codec, nid, - AC_PAR_PIN_CAP); + pincap = snd_hda_query_pin_caps(codec, nid); if (pincap & AC_PINCAP_EAPD) { int eapd = snd_hda_codec_read(codec, nid, 0, diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index b1c180a9e9be..b5e108aa8f63 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2869,8 +2869,7 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid) conn_len = snd_hda_get_connections(codec, nid, conn, HDA_MAX_CONNECTIONS); for (j = 0; j < conn_len; j++) { - wcaps = snd_hda_param_read(codec, conn[j], - AC_PAR_AUDIO_WIDGET_CAP); + wcaps = get_wcaps(codec, conn[j]); wtype = (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; /* we check only analog outputs */ if (wtype != AC_WID_AUD_OUT || (wcaps & AC_WCAP_DIGITAL)) -- cgit v1.2.3-59-g8ed1b From 9b6682ff4c69484b6955f89f7902e3dde2481bed Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Mar 2009 22:50:52 +0100 Subject: ALSA: hda - Add quirk for Acer Ferrari 5000 Add a quirk model=acer-aspire for Acer Ferrari 5000 with ALC883 codec. Note that model=auto doesn't work for this laptop because of broken BIOS (that doesn't set the subsystem id properly). Tested-by: Russ Dill Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7a3c6db6d5be..82097790f6f3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8677,6 +8677,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x0090, "Acer Aspire", ALC883_ACER_ASPIRE), + SND_PCI_QUIRK(0x1025, 0x010a, "Acer Ferrari 5000", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_ACER_ASPIRE), -- cgit v1.2.3-59-g8ed1b