October 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 16/31] HDA patch_via.c: Remove via_playback_pcm_prepare/cleanup
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Remove via_playback_pcm_prepare/cleanup replaced with via_playback_multi_pcm_prepare/cleanup to support multi-stream operations Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:10:49.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:10:59.000000000 +0800 @@ -989,26 +989,6 @@ hinfo); } -static int via_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct via_spec *spec = codec->spec; - return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, - stream_tag, format, substream); -} - -static int via_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct via_spec *spec = codec->spec; - return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); -} - - static void playback_multi_pcm_prep_0(struct hda_codec *codec, unsigned int stream_tag, unsigned int format, @@ -1217,7 +1197,7 @@ }; static struct hda_pcm_stream vt1708_pcm_analog_s16_playback = { - .substreams = 1, + .substreams = 2, .channels_min = 2, .channels_max = 8, .nid = 0x10, /* NID to query formats and rates */ @@ -1228,8 +1208,8 @@ .formats = SNDRV_PCM_FMTBIT_S16_LE, .ops = { .open = via_playback_pcm_open, - .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup }, }; @@ -2023,8 +2003,8 @@ .nid = 0x10, /* NID to query formats and rates */ .ops = { .open = via_playback_pcm_open, - .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, }, }; @@ -2035,8 +2015,8 @@ .nid = 0x10, /* NID to query formats and rates */ .ops = { .open = via_playback_pcm_open, - .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, }, }; @@ -3127,8 +3107,8 @@ .nid = 0x10, /* NID to query formats and rates */ .ops = { .open = via_playback_pcm_open, - .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup, + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, .close = via_pcm_open_close }, };

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[alsa-devel] [PATCH 15/31] HDA patch_via.c: Modify vt1708_set_pinconfig_connect function.
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Modify vt1708_set_pinconfig_connect function. like seqassoc 0xff, seqassoc 0xf0 of vt1708 should override Port Connectivity field into 'AC_JACK_PORT_COMPLEX' Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:10:36.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:10:49.000000000 +0800 @@ -1725,11 +1725,10 @@ 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_set_pincfg(codec, nid, def_conf); - } + if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE + && (seqassoc == 0xf0 || seqassoc == 0xff)) { + def_conf = def_conf & (~(AC_JACK_PORT_BOTH << 30)); + snd_hda_codec_set_pincfg(codec, nid, def_conf); } return;

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[alsa-devel] [PATCH 12/31] HDA patch_via.c: Add VIA_JACK_EVENT process in via_unsol_event.
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Add VIA_JACK_EVENT process in via_unsol_event. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:10:05.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:10:13.000000000 +0800 @@ -124,6 +124,7 @@ #define VIA_HP_EVENT 0x01 #define VIA_GPIO_EVENT 0x02 +#define VIA_JACK_EVENT 0x04 enum { VIA_CTL_WIDGET_VOL, @@ -1408,10 +1409,12 @@ unsigned int res) { res >>= 26; - if (res == VIA_HP_EVENT) + if (res & VIA_HP_EVENT) via_hp_automute(codec); - else if (res == VIA_GPIO_EVENT) + if (res & VIA_GPIO_EVENT) via_gpio_control(codec); + if (res & VIA_JACK_EVENT) + set_jack_power_state(codec); } static int via_init(struct hda_codec *codec) @@ -1873,7 +1876,8 @@ }; static struct hda_verb vt1709_uniwill_init_verbs[] = { - {0x20, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT}, + {0x20, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, { } }; @@ -2509,7 +2513,15 @@ }; static struct hda_verb vt1708B_uniwill_init_verbs[] = { - {0x1D, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT}, + {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x22, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x23, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, { } }; @@ -3004,7 +3016,15 @@ }; static struct hda_verb vt1708S_uniwill_init_verbs[] = { - {0x1D, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT}, + {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x22, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x23, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, { } }; @@ -3443,8 +3463,12 @@ }; static struct hda_verb vt1702_uniwill_init_verbs[] = { - {0x01, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_GPIO_EVENT}, - {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT}, + {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, { } };

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[alsa-devel] [PATCH 11/31] HDA patch_via.c: When changing input source, update jack power state.
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: When changing input source, update jack power state. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:09:55.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:10:05.000000000 +0800 @@ -548,6 +548,14 @@ if (!spec->mux_nids[adc_idx]) return -EINVAL; + /* switch to D0 beofre change index */ + if (snd_hda_codec_read(codec, spec->mux_nids[adc_idx], 0, + AC_VERB_GET_POWER_STATE, 0x00) != AC_PWRST_D0) + snd_hda_codec_write(codec, spec->mux_nids[adc_idx], 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + /* update jack power state */ + set_jack_power_state(codec); + return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, spec->mux_nids[adc_idx], &spec->cur_mux[adc_idx]);

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[alsa-devel] [PATCH 10/31] HDA patch_via.c: Add smart5.1 function.
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Add smart5.1 function. Smart 5.1 is for 3-jacks model, to reuse input pins as outputs. While off, they act as "line out" / "line in" / "mic in". While on, they acts as "line out" / "back left/right" / "center/lfe". Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:09:46.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:09:55.000000000 +0800 @@ -211,7 +211,7 @@ struct via_spec { /* codec parameterization */ - struct snd_kcontrol_new *mixers[3]; + struct snd_kcontrol_new *mixers[4]; unsigned int num_mixers; struct hda_verb *init_verbs[5]; @@ -253,7 +253,7 @@ const struct hda_input_mux *hp_mux; unsigned int hp_independent_mode; unsigned int hp_independent_mode_index; - + unsigned int smart51_enabled; enum VIA_HDA_CODEC codec_type; #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -390,6 +390,8 @@ } } +static int is_smart51_pins(struct via_spec *spec, hda_nid_t pin); + static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid, unsigned int *affected_parm) { @@ -400,9 +402,10 @@ & AC_DEFCFG_MISC_NO_PRESENCE; /* do not support pin sense */ unsigned present = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0) >> 31; - - if ((no_presence || present) && get_defcfg_connect(def_conf) - != AC_JACK_PORT_NONE) { + struct via_spec *spec = codec->spec; + if ((spec->smart51_enabled && is_smart51_pins(spec, nid)) + || ((no_presence || present) + && get_defcfg_connect(def_conf) != AC_JACK_PORT_NONE)) { *affected_parm = AC_PWRST_D0; /* if it's connected */ parm = AC_PWRST_D0; } else @@ -649,6 +652,171 @@ { } /* end */ }; +static void notify_aa_path_ctls(struct hda_codec *codec) +{ + struct snd_card *card = codec->bus->card; + struct snd_kcontrol *kctl; + list_for_each_entry(kctl, &card->controls, list) { + int i; + static char *labels[] = {"Mic", "Front Mic", "Line"}; + for (i = 0; i < ARRAY_SIZE(labels); i++) { + char name[32]; + sprintf(name, "%s Playback Volume", labels[i]); + if (strcmp(kctl->id.name, name) == 0) { + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &kctl->id); + break; + } + } + } +} + +static void mute_aa_path(struct hda_codec *codec, int mute) +{ + struct via_spec *spec = codec->spec; + hda_nid_t nid_mixer; + int start_idx; + int end_idx; + int i; + /* get nid of MW0 and start & end index */ + switch (spec->codec_type) { + case VT1708: + nid_mixer = 0x17; + start_idx = 2; + end_idx = 4; + break; + case VT1709_10CH: + case VT1709_6CH: + nid_mixer = 0x18; + start_idx = 2; + end_idx = 4; + break; + case VT1708B_8CH: + case VT1708B_4CH: + case VT1708S: + nid_mixer = 0x16; + start_idx = 2; + end_idx = 4; + break; + default: + return; + } + /* check AA path's mute status */ + for (i = start_idx; i <= end_idx; i++) { + int val = mute ? HDA_AMP_MUTE : HDA_AMP_UNMUTE; + snd_hda_codec_amp_stereo(codec, nid_mixer, HDA_INPUT, i, + HDA_AMP_MUTE, val); + } +} +static int is_smart51_pins(struct via_spec *spec, hda_nid_t pin) +{ + int res = 0; + int index; + for (index = AUTO_PIN_MIC; index < AUTO_PIN_FRONT_LINE; index++) { + if (pin == spec->autocfg.input_pins[index]) { + res = 1; + break; + } + } + return res; +} + +static int via_smart51_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int via_smart51_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + int index[] = { AUTO_PIN_MIC, AUTO_PIN_FRONT_MIC, AUTO_PIN_LINE }; + int on = 1; + int i; + + for (i = 0; i < ARRAY_SIZE(index); i++) { + hda_nid_t nid = spec->autocfg.input_pins[index[i]]; + if (nid) { + int ctl = + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, + 0); + if (i == AUTO_PIN_FRONT_MIC + && spec->hp_independent_mode) + continue; /* ignore FMic for independent HP */ + if (ctl & AC_PINCTL_IN_EN + && !(ctl & AC_PINCTL_OUT_EN)) + on = 0; + } + } + *ucontrol->value.integer.value = on; + return 0; +} + +static int via_smart51_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + int out_in = *ucontrol->value.integer.value + ? AC_PINCTL_OUT_EN : AC_PINCTL_IN_EN; + int index[] = { AUTO_PIN_MIC, AUTO_PIN_FRONT_MIC, AUTO_PIN_LINE }; + int i; + + for (i = 0; i < ARRAY_SIZE(index); i++) { + hda_nid_t nid = spec->autocfg.input_pins[index[i]]; + if (i == AUTO_PIN_FRONT_MIC + && spec->hp_independent_mode) + continue; /* don't retask FMic for independent HP */ + if (nid) { + unsigned int parm = snd_hda_codec_read( + codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + parm &= ~(AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN); + parm |= out_in; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + parm); + if (out_in == AC_PINCTL_OUT_EN) { + mute_aa_path(codec, 1); + notify_aa_path_ctls(codec); + } + } + if (i == AUTO_PIN_FRONT_MIC) { + if (spec->codec_type == VT1708S) { + /* input = index 1 (AOW3) */ + snd_hda_codec_write( + codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, 1); + snd_hda_codec_amp_stereo( + codec, nid, HDA_OUTPUT, + 0, HDA_AMP_MUTE, HDA_AMP_UNMUTE); + } + } + } + spec->smart51_enabled = *ucontrol->value.integer.value; + set_jack_power_state(codec); + return 1; +} + +static struct snd_kcontrol_new via_smart51_mixer[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Smart 5.1", + .count = 1, + .info = via_smart51_info, + .get = via_smart51_get, + .put = via_smart51_put, + }, + {} /* end */ +}; + /* capture mixer elements */ static struct snd_kcontrol_new vt1708_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_INPUT), @@ -1579,6 +1747,7 @@ if (spec->hp_mux) spec->mixers[spec->num_mixers++] = via_hp_mixer; + spec->mixers[spec->num_mixers++] = via_smart51_mixer; return 1; } @@ -2079,6 +2248,7 @@ if (spec->hp_mux) spec->mixers[spec->num_mixers++] = via_hp_mixer; + spec->mixers[spec->num_mixers++] = via_smart51_mixer; return 1; } @@ -2641,6 +2811,7 @@ if (spec->hp_mux) spec->mixers[spec->num_mixers++] = via_hp_mixer; + spec->mixers[spec->num_mixers++] = via_smart51_mixer; return 1; } @@ -3134,6 +3305,7 @@ if (spec->hp_mux) spec->mixers[spec->num_mixers++] = via_hp_mixer; + spec->mixers[spec->num_mixers++] = via_smart51_mixer; return 1; }

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[alsa-devel] [PATCH 9/31] HDA patch_via.c: Rewrite via_independent_hp_put
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Rewrite via_independent_hp_put Use hp_independent_mode_index to store hp index, and simplify function via_independent_hp_put with it. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:09:33.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:09:46.000000000 +0800 @@ -252,6 +252,7 @@ /* HP mode source */ const struct hda_input_mux *hp_mux; unsigned int hp_independent_mode; + unsigned int hp_independent_mode_index; enum VIA_HDA_CODEC codec_type; @@ -589,6 +590,30 @@ return 0; } +static int update_side_mute_status(struct hda_codec *codec) +{ + /* mute side channel */ + struct via_spec *spec = codec->spec; + unsigned int parm = spec->hp_independent_mode + ? AMP_OUT_MUTE : AMP_OUT_UNMUTE; + hda_nid_t nids_sw3[CODEC_TYPES] = { + [VT1708] = 0x1b, + [VT1709_10CH] = 0x29, + [VT1709_6CH] = 0, + [VT1708B_8CH] = 0x27, + [VT1708B_4CH] = 0, + [VT1708S] = 0x27, + [VT1702] = 0, + }; + + if (!nids_sw3[spec->codec_type]) + return 0; + + snd_hda_codec_write(codec, nids_sw3[spec->codec_type], 0, + AC_VERB_SET_AMP_GAIN_MUTE, parm); + return 0; +} + static int via_independent_hp_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -596,47 +621,19 @@ struct via_spec *spec = codec->spec; hda_nid_t nid = spec->autocfg.hp_pins[0]; unsigned int pinsel = ucontrol->value.enumerated.item[0]; - unsigned int con_nid = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONNECT_LIST, 0) & 0xff; - if (con_nid == spec->multiout.hp_nid) { - if (pinsel == 0) { - if (!spec->hp_independent_mode) { - if (spec->multiout.num_dacs > 1) - spec->multiout.num_dacs -= 1; - spec->hp_independent_mode = 1; - } - } else if (pinsel == 1) { - if (spec->hp_independent_mode) { - if (spec->multiout.num_dacs > 1) - spec->multiout.num_dacs += 1; - spec->hp_independent_mode = 0; - } - } - } else { - if (pinsel == 0) { - if (spec->hp_independent_mode) { - if (spec->multiout.num_dacs > 1) - spec->multiout.num_dacs += 1; - spec->hp_independent_mode = 0; - } - } else if (pinsel == 1) { - if (!spec->hp_independent_mode) { - if (spec->multiout.num_dacs > 1) - spec->multiout.num_dacs -= 1; - spec->hp_independent_mode = 1; - } - } - } - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, - pinsel); - - if (spec->multiout.hp_nid && - spec->multiout.hp_nid != spec->multiout.dac_nids[HDA_FRONT]) - snd_hda_codec_setup_stream(codec, - spec->multiout.hp_nid, - 0, 0, 0); + /* Get Independent Mode index of headphone pin widget */ + spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel + ? 1 : 0; + + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, pinsel); + + if (spec->multiout.hp_nid && spec->multiout.hp_nid + != spec->multiout.dac_nids[HDA_FRONT]) + snd_hda_codec_setup_stream(codec, spec->multiout.hp_nid, + 0, 0, 0); + update_side_mute_status(codec); return 0; } @@ -1444,6 +1441,7 @@ return 0; spec->multiout.hp_nid = VT1708_HP_NID; /* AOW3 */ + spec->hp_independent_mode_index = 1; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Headphone Playback Volume", @@ -1979,6 +1977,7 @@ spec->multiout.hp_nid = VT1709_HP_DAC_NID; else if (spec->multiout.num_dacs == 3) /* 6 channels */ spec->multiout.hp_nid = 0; + spec->hp_independent_mode_index = 1; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Headphone Playback Volume", @@ -2538,6 +2537,7 @@ return 0; spec->multiout.hp_nid = VT1708B_HP_NID; /* AOW3 */ + spec->hp_independent_mode_index = 1; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Headphone Playback Volume", @@ -3008,6 +3008,7 @@ return 0; spec->multiout.hp_nid = VT1708S_HP_NID; /* AOW3 */ + spec->hp_independent_mode_index = 1; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Headphone Playback Volume", @@ -3367,6 +3368,7 @@ if (!pin) return 0; spec->multiout.hp_nid = 0x1D; + spec->hp_independent_mode_index = 0; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Headphone Playback Volume",

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[alsa-devel] [PATCH 7/31] HDA patch_via.c: Remove unused argument of via_new_analog_input
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Remove unused argument of via_new_analog_input Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:09:07.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:09:18.000000000 +0800 @@ -317,8 +317,8 @@ } /* create input playback/capture controls for the given pin */ -static int via_new_analog_input(struct via_spec *spec, hda_nid_t pin, - const char *ctlname, int idx, int mix_nid) +static int via_new_analog_input(struct via_spec *spec, const char *ctlname, + int idx, int mix_nid) { char name[32]; int err; @@ -1480,8 +1480,7 @@ idx = 1; break; } - err = via_new_analog_input(spec, cfg->input_pins[i], labels[i], - idx, 0x17); + err = via_new_analog_input(spec, labels[i], idx, 0x17); if (err < 0) return err; imux->items[imux->num_items].label = labels[i]; @@ -2014,8 +2013,7 @@ idx = 1; break; } - err = via_new_analog_input(spec, cfg->input_pins[i], labels[i], - idx, 0x18); + err = via_new_analog_input(spec, labels[i], idx, 0x18); if (err < 0) return err; imux->items[imux->num_items].label = labels[i]; @@ -2576,8 +2574,7 @@ idx = 1; break; } - err = via_new_analog_input(spec, cfg->input_pins[i], labels[i], - idx, 0x16); + err = via_new_analog_input(spec, labels[i], idx, 0x16); if (err < 0) return err; imux->items[imux->num_items].label = labels[i]; @@ -3048,8 +3045,7 @@ idx = 1; break; } - err = via_new_analog_input(spec, cfg->input_pins[i], labels[i], - idx, 0x16); + err = via_new_analog_input(spec, labels[i], idx, 0x16); if (err < 0) return err; imux->items[imux->num_items].label = labels[i]; @@ -3402,8 +3398,7 @@ idx = 3; break; } - err = via_new_analog_input(spec, cfg->input_pins[i], - labels[i], idx, 0x1A); + err = via_new_analog_input(spec, labels[i], idx, 0x1A); if (err < 0) return err; imux->items[imux->num_items].label = labels[i];

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[alsa-devel] [PATCH 6/31] HDA patch_via.c: Add low current mode for power saving.
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Add low current mode for power saving. For VT1708B, VT1708S and VT1702, enter low current mode if no analog stream is opened and all aa path mute. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:08:44.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:08:59.000000000 +0800 @@ -783,6 +783,10 @@ struct snd_pcm_substream *substream) { struct via_spec *spec = codec->spec; + int idle = substream->pstr->substream_opened == 1 + && substream->ref_count == 0; + + analog_low_current_mode(codec, idle); return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, hinfo); } @@ -1089,6 +1093,11 @@ if (err < 0) return err; } + + /* init power states */ + set_jack_power_state(codec); + analog_low_current_mode(codec, 1); + via_free_kctls(codec); /* no longer needed */ return 0; } @@ -2312,6 +2321,17 @@ { } }; +static int via_pcm_open_close(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + int idle = substream->pstr->substream_opened == 1 + && substream->ref_count == 0; + + analog_low_current_mode(codec, idle); + return 0; +} + static struct hda_pcm_stream vt1708B_8ch_pcm_analog_playback = { .substreams = 2, .channels_min = 2, @@ -2320,7 +2340,8 @@ .ops = { .open = via_playback_pcm_open, .prepare = via_playback_multi_pcm_prepare, - .cleanup = via_playback_multi_pcm_cleanup + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -2342,8 +2363,10 @@ .channels_max = 2, .nid = 0x13, /* NID to query formats and rates */ .ops = { + .open = via_pcm_open_close, .prepare = via_capture_pcm_prepare, - .cleanup = via_capture_pcm_cleanup + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -2800,7 +2823,8 @@ .ops = { .open = via_playback_pcm_open, .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup + .cleanup = via_playback_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -2810,8 +2834,10 @@ .channels_max = 2, .nid = 0x13, /* NID to query formats and rates */ .ops = { + .open = via_pcm_open_close, .prepare = via_capture_pcm_prepare, - .cleanup = via_capture_pcm_cleanup + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -3236,7 +3262,8 @@ .ops = { .open = via_playback_pcm_open, .prepare = via_playback_multi_pcm_prepare, - .cleanup = via_playback_multi_pcm_cleanup + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -3246,8 +3273,10 @@ .channels_max = 2, .nid = 0x12, /* NID to query formats and rates */ .ops = { + .open = via_pcm_open_close, .prepare = via_capture_pcm_prepare, - .cleanup = via_capture_pcm_cleanup + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close }, };

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[alsa-devel] [PATCH 5/31] HDA patch_via.c: Add VIA_CTL_WIDGET_ANALOG_MUTE control type
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Add VIA_CTL_WIDGET_ANALOG_MUTE control type Enter low power state if AA-Path volume is muted. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:08:20.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:09:02.000000000 +0800 @@ -128,6 +128,7 @@ enum { VIA_CTL_WIDGET_VOL, VIA_CTL_WIDGET_MUTE, + VIA_CTL_WIDGET_ANALOG_MUTE, }; enum { @@ -177,9 +178,34 @@ return 0; } +static void analog_low_current_mode(struct hda_codec *codec, int stream_idle); +static void set_jack_power_state(struct hda_codec *codec); + +static int analog_input_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int change = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + + set_jack_power_state(codec); + analog_low_current_mode(snd_kcontrol_chip(kcontrol), -1); + return change; +} + +/* modify .put = snd_hda_mixer_amp_switch_put */ +#define ANALOG_INPUT_MUTE \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = NULL, \ + .index = 0, \ + .info = snd_hda_mixer_amp_switch_info, \ + .get = snd_hda_mixer_amp_switch_get, \ + .put = analog_input_switch_put, \ + .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0) } + static struct snd_kcontrol_new vt1708_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), + ANALOG_INPUT_MUTE, }; @@ -303,7 +329,7 @@ if (err < 0) return err; sprintf(name, "%s Playback Switch", ctlname); - err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name, + err = via_add_control(spec, VIA_CTL_WIDGET_ANALOG_MUTE, name, HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT)); if (err < 0) return err; @@ -362,6 +388,131 @@ } } + +static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid, + unsigned int *affected_parm) +{ + unsigned parm; + unsigned def_conf = snd_hda_codec_get_pincfg(codec, nid); + unsigned no_presence = (def_conf & AC_DEFCFG_MISC) + >> AC_DEFCFG_MISC_SHIFT + & AC_DEFCFG_MISC_NO_PRESENCE; /* do not support pin sense */ + unsigned present = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_SENSE, 0) >> 31; + + if ((no_presence || present) && get_defcfg_connect(def_conf) + != AC_JACK_PORT_NONE) { + *affected_parm = AC_PWRST_D0; /* if it's connected */ + parm = AC_PWRST_D0; + } else + parm = AC_PWRST_D3; + + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm); +} + +static void set_jack_power_state(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer; + unsigned int parm; + + if (spec->codec_type == VT1702) { + imux_is_smixer = snd_hda_codec_read( + codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; + /* inputs */ + /* PW 1/2/5 (14h/15h/18h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x14, &parm); + set_pin_power_state(codec, 0x15, &parm); + set_pin_power_state(codec, 0x18, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; /* SW0 = stereo mixer (idx 3) */ + /* SW0 (13h), AIW 0/1/2 (12h/1fh/20h) */ + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* outputs */ + /* PW 3/4 (16h/17h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x16, &parm); + set_pin_power_state(codec, 0x17, &parm); + /* MW0 (1ah), AOW 0/1 (10h/1dh) */ + snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, + parm); + } else if (spec->codec_type == VT1708B_8CH + || spec->codec_type == VT1708B_4CH + || spec->codec_type == VT1708S) { + /* SW0 (17h) = stereo mixer */ + int is_8ch = spec->codec_type != VT1708B_4CH; + imux_is_smixer = snd_hda_codec_read( + codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) + == ((spec->codec_type == VT1708S) ? 5 : 0); + /* inputs */ + /* PW 1/2/5 (1ah/1bh/1eh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1a, &parm); + set_pin_power_state(codec, 0x1b, &parm); + set_pin_power_state(codec, 0x1e, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* SW0 (17h), AIW 0/1 (13h/14h) */ + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* outputs */ + /* PW0 (19h), SW1 (18h), AOW1 (11h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x19, &parm); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* PW6 (22h), SW2 (26h), AOW2 (24h) */ + if (is_8ch) { + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x22, &parm); + snd_hda_codec_write(codec, 0x26, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x24, 0, + AC_VERB_SET_POWER_STATE, parm); + } + + /* PW 3/4/7 (1ch/1dh/23h) */ + parm = AC_PWRST_D3; + /* force to D0 for internal Speaker */ + set_pin_power_state(codec, 0x1c, &parm); + set_pin_power_state(codec, 0x1d, &parm); + if (is_8ch) + set_pin_power_state(codec, 0x23, &parm); + /* MW0 (16h), Sw3 (27h), AOW 0/3 (10h/25h) */ + snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, + parm); + if (is_8ch) { + snd_hda_codec_write(codec, 0x25, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x27, 0, + AC_VERB_SET_POWER_STATE, parm); + } + } +} + /* * input MUX handling */ @@ -504,6 +655,93 @@ }, { } /* end */ }; + +/* check AA path's mute statue */ +static int is_aa_path_mute(struct hda_codec *codec) +{ + int mute = 1; + hda_nid_t nid_mixer; + int start_idx; + int end_idx; + int i; + struct via_spec *spec = codec->spec; + /* get nid of MW0 and start & end index */ + switch (spec->codec_type) { + case VT1708B_8CH: + case VT1708B_4CH: + case VT1708S: + nid_mixer = 0x16; + start_idx = 2; + end_idx = 4; + break; + case VT1702: + nid_mixer = 0x1a; + start_idx = 1; + end_idx = 3; + break; + default: + return 0; + } + /* check AA path's mute status */ + for (i = start_idx; i <= end_idx; i++) { + unsigned int con_list = snd_hda_codec_read( + codec, nid_mixer, 0, AC_VERB_GET_CONNECT_LIST, i/4*4); + int shift = 8 * (i % 4); + hda_nid_t nid_pin = (con_list & (0xff << shift)) >> shift; + unsigned int defconf = snd_hda_codec_get_pincfg(codec, nid_pin); + if (get_defcfg_connect(defconf) == AC_JACK_PORT_COMPLEX) { + /* check mute status while the pin is connected */ + int mute_l = snd_hda_codec_amp_read(codec, nid_mixer, 0, + HDA_INPUT, i) >> 7; + int mute_r = snd_hda_codec_amp_read(codec, nid_mixer, 1, + HDA_INPUT, i) >> 7; + if (!mute_l || !mute_r) { + mute = 0; + break; + } + } + } + return mute; +} + +/* enter/exit analog low-current mode */ +static void analog_low_current_mode(struct hda_codec *codec, int stream_idle) +{ + struct via_spec *spec = codec->spec; + static int saved_stream_idle = 1; /* saved stream idle status */ + int enable = is_aa_path_mute(codec); + unsigned int verb = 0; + unsigned int parm = 0; + + if (stream_idle == -1) /* stream status did not change */ + enable = enable && saved_stream_idle; + else { + enable = enable && stream_idle; + saved_stream_idle = stream_idle; + } + + /* decide low current mode's verb & parameter */ + switch (spec->codec_type) { + case VT1708B_8CH: + case VT1708B_4CH: + verb = 0xf70; + parm = enable ? 0x02 : 0x00; /* 0x02: 2/3x, 0x00: 1x */ + break; + case VT1708S: + verb = 0xf73; + parm = enable ? 0x51 : 0xe1; /* 0x51: 4/28x, 0xe1: 1x */ + break; + case VT1702: + verb = 0xf73; + parm = enable ? 0x01 : 0x1d; /* 0x01: 4/40x, 0x1d: 1x */ + break; + default: + return; /* other codecs are not supported */ + } + /* send verb */ + snd_hda_codec_write(codec, codec->afg, 0, verb, parm); +} + /* * generic initialization of ADC, input mixers and output mixers */

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[alsa-devel] [PATCH 4/31] HDA patch_via.c: Modify VT1702 AA-Path max volume
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Modify VT1702 AA-Path max volume according to customer request, VT1702 AA-Path max volume (12 dB) is too high, so limit to 0 dB. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-04 23:05:19.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 00:24:34.000000000 +0800 @@ -3166,6 +3166,12 @@ err = vt1702_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); if (err < 0) return err; + /* limit AA path volume to 0 dB */ + snd_hda_override_amp_caps(codec, 0x1A, HDA_INPUT, + (0x17 << AC_AMPCAP_OFFSET_SHIFT) | + (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x5 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (1 << AC_AMPCAP_MUTE_SHIFT)); err = vt1702_auto_create_analog_input_ctls(spec, &spec->autocfg); if (err < 0) return err;

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