diff options
Diffstat (limited to 'sound/pci')
| -rw-r--r-- | sound/pci/ac97/ac97_patch.c | 48 | ||||
| -rw-r--r-- | sound/pci/aw2/aw2-alsa.c | 4 | ||||
| -rw-r--r-- | sound/pci/emu10k1/emu10k1_main.c | 15 | ||||
| -rw-r--r-- | sound/pci/hda/patch_analog.c | 51 | ||||
| -rw-r--r-- | sound/pci/hda/patch_cmedia.c | 1 | ||||
| -rw-r--r-- | sound/pci/hda/patch_realtek.c | 6 | ||||
| -rw-r--r-- | sound/pci/hda/patch_sigmatel.c | 2 | ||||
| -rw-r--r-- | sound/pci/hda/patch_via.c | 20 | ||||
| -rw-r--r-- | sound/pci/oxygen/oxygen_mixer.c | 12 |
9 files changed, 107 insertions, 52 deletions
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 2da89810ca10..1292dcee072d 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -1971,6 +1971,9 @@ static int snd_ac97_ad1888_lohpsel_get(struct snd_kcontrol *kcontrol, struct snd val = ac97->regs[AC97_AD_MISC]; ucontrol->value.integer.value[0] = !(val & AC97_AD198X_LOSEL); + if (ac97->spec.ad18xx.lo_as_master) + ucontrol->value.integer.value[0] = + !ucontrol->value.integer.value[0]; return 0; } @@ -1979,8 +1982,10 @@ static int snd_ac97_ad1888_lohpsel_put(struct snd_kcontrol *kcontrol, struct snd struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); unsigned short val; - val = !ucontrol->value.integer.value[0] - ? (AC97_AD198X_LOSEL | AC97_AD198X_HPSEL) : 0; + val = !ucontrol->value.integer.value[0]; + if (ac97->spec.ad18xx.lo_as_master) + val = !val; + val = val ? (AC97_AD198X_LOSEL | AC97_AD198X_HPSEL) : 0; return snd_ac97_update_bits(ac97, AC97_AD_MISC, AC97_AD198X_LOSEL | AC97_AD198X_HPSEL, val); } @@ -2031,7 +2036,7 @@ static void ad1888_update_jacks(struct snd_ac97 *ac97) { unsigned short val = 0; /* clear LODIS if shared jack is to be used for Surround out */ - if (is_shared_linein(ac97)) + if (!ac97->spec.ad18xx.lo_as_master && is_shared_linein(ac97)) val |= (1 << 12); /* clear CLDIS if shared jack is to be used for C/LFE out */ if (is_shared_micin(ac97)) @@ -2067,9 +2072,13 @@ static const struct snd_kcontrol_new snd_ac97_ad1888_controls[] = { static int patch_ad1888_specific(struct snd_ac97 *ac97) { - /* rename 0x04 as "Master" and 0x02 as "Master Surround" */ - snd_ac97_rename_vol_ctl(ac97, "Master Playback", "Master Surround Playback"); - snd_ac97_rename_vol_ctl(ac97, "Headphone Playback", "Master Playback"); + if (!ac97->spec.ad18xx.lo_as_master) { + /* rename 0x04 as "Master" and 0x02 as "Master Surround" */ + snd_ac97_rename_vol_ctl(ac97, "Master Playback", + "Master Surround Playback"); + snd_ac97_rename_vol_ctl(ac97, "Headphone Playback", + "Master Playback"); + } return patch_build_controls(ac97, snd_ac97_ad1888_controls, ARRAY_SIZE(snd_ac97_ad1888_controls)); } @@ -2088,16 +2097,27 @@ static int patch_ad1888(struct snd_ac97 * ac97) patch_ad1881(ac97); ac97->build_ops = &patch_ad1888_build_ops; - /* Switch FRONT/SURROUND LINE-OUT/HP-OUT default connection */ - /* it seems that most vendors connect line-out connector to headphone out of AC'97 */ + + /* + * LO can be used as a real line-out on some devices, + * and we need to revert the front/surround mixer switches + */ + if (ac97->subsystem_vendor == 0x1043 && + ac97->subsystem_device == 0x1193) /* ASUS A9T laptop */ + ac97->spec.ad18xx.lo_as_master = 1; + + misc = snd_ac97_read(ac97, AC97_AD_MISC); /* AD-compatible mode */ /* Stereo mutes enabled */ - misc = snd_ac97_read(ac97, AC97_AD_MISC); - snd_ac97_write_cache(ac97, AC97_AD_MISC, misc | - AC97_AD198X_LOSEL | - AC97_AD198X_HPSEL | - AC97_AD198X_MSPLT | - AC97_AD198X_AC97NC); + misc |= AC97_AD198X_MSPLT | AC97_AD198X_AC97NC; + if (!ac97->spec.ad18xx.lo_as_master) + /* Switch FRONT/SURROUND LINE-OUT/HP-OUT default connection */ + /* it seems that most vendors connect line-out connector to + * headphone out of AC'97 + */ + misc |= AC97_AD198X_LOSEL | AC97_AD198X_HPSEL; + + snd_ac97_write_cache(ac97, AC97_AD_MISC, misc); ac97->flags |= AC97_STEREO_MUTES; return 0; } diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c index 56f87cd33c19..3f00ddf450f8 100644 --- a/sound/pci/aw2/aw2-alsa.c +++ b/sound/pci/aw2/aw2-alsa.c @@ -316,6 +316,8 @@ static int __devinit snd_aw2_create(struct snd_card *card, return -ENOMEM; } + /* (2) initialization of the chip hardware */ + snd_aw2_saa7146_setup(&chip->saa7146, chip->iobase_virt); if (request_irq(pci->irq, snd_aw2_saa7146_interrupt, IRQF_SHARED, "Audiowerk2", chip)) { @@ -329,8 +331,6 @@ static int __devinit snd_aw2_create(struct snd_card *card, } chip->irq = pci->irq; - /* (2) initialization of the chip hardware */ - snd_aw2_saa7146_setup(&chip->saa7146, chip->iobase_virt); err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); if (err < 0) { free_irq(chip->irq, (void *)chip); diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index abde5b901884..548c9cc81af5 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1818,13 +1818,6 @@ int __devinit snd_emu10k1_create(struct snd_card *card, } emu->port = pci_resource_start(pci, 0); - if (request_irq(pci->irq, snd_emu10k1_interrupt, IRQF_SHARED, - "EMU10K1", emu)) { - err = -EBUSY; - goto error; - } - emu->irq = pci->irq; - emu->max_cache_pages = max_cache_bytes >> PAGE_SHIFT; if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), 32 * 1024, &emu->ptb_pages) < 0) { @@ -1887,6 +1880,14 @@ int __devinit snd_emu10k1_create(struct snd_card *card, emu->fx8010.etram_pages.area = NULL; emu->fx8010.etram_pages.bytes = 0; + /* irq handler must be registered after I/O ports are activated */ + if (request_irq(pci->irq, snd_emu10k1_interrupt, IRQF_SHARED, + "EMU10K1", emu)) { + err = -EBUSY; + goto error; + } + emu->irq = pci->irq; + /* * Init to 0x02109204 : * Clock accuracy = 0 (1000ppm) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index e0a605adde42..a99e86d74278 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2858,6 +2858,7 @@ static const char *ad1988_models[AD1988_MODEL_LAST] = { static struct snd_pci_quirk ad1988_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x81ec, "Asus P5B-DLX", AD1988_6STACK_DIG), SND_PCI_QUIRK(0x1043, 0x81f6, "Asus M2N-SLI", AD1988_6STACK_DIG), + SND_PCI_QUIRK(0x1043, 0x8277, "Asus P5K-E/WIFI-AP", AD1988_6STACK_DIG), {} }; @@ -3643,33 +3644,17 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = { { } /* end */ }; -static struct hda_input_mux ad1884a_mobile_capture_source = { - .num_items = 2, - .items = { - { "Mic", 0x1 }, /* port-C */ - { "Mix", 0x3 }, - }, -}; - static struct snd_kcontrol_new ad1884a_mobile_mixers[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), 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("Mic Playback Volume", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, 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", 0x15, 0x0, 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), HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, { } /* end */ }; @@ -3686,14 +3671,31 @@ static void ad1884a_hp_automute(struct hda_codec *codec) present ? 0x00 : 0x02); } +/* switch to external mic if plugged */ +static void ad1884a_hp_automic(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x14, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL, + present ? 0 : 1); +} + #define AD1884A_HP_EVENT 0x37 +#define AD1884A_MIC_EVENT 0x36 /* unsolicited event for HP jack sensing */ static void ad1884a_hp_unsol_event(struct hda_codec *codec, unsigned int res) { - if ((res >> 26) != AD1884A_HP_EVENT) - return; - ad1884a_hp_automute(codec); + switch (res >> 26) { + case AD1884A_HP_EVENT: + ad1884a_hp_automute(codec); + break; + case AD1884A_MIC_EVENT: + ad1884a_hp_automic(codec); + break; + } } /* initialize jack-sensing, too */ @@ -3701,6 +3703,7 @@ static int ad1884a_hp_init(struct hda_codec *codec) { ad198x_init(codec); ad1884a_hp_automute(codec); + ad1884a_hp_automic(codec); return 0; } @@ -3714,10 +3717,15 @@ static struct hda_verb ad1884a_laptop_verbs[] = { /* Port-F pin */ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Port-C pin - internal mic-in */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */ + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */ /* analog mix */ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* unsolicited event for pin-sense */ {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT}, { } /* end */ }; @@ -3877,7 +3885,6 @@ static int patch_ad1884a(struct hda_codec *codec) spec->mixers[0] = ad1884a_mobile_mixers; spec->init_verbs[spec->num_init_verbs++] = ad1884a_laptop_verbs; spec->multiout.dig_out_nid = 0; - spec->input_mux = &ad1884a_mobile_capture_source; codec->patch_ops.unsol_event = ad1884a_hp_unsol_event; codec->patch_ops.init = ad1884a_hp_init; break; diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index c73ce074a6ea..6ef57fbfb6eb 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -611,6 +611,7 @@ static const char *cmi9880_models[CMI_MODELS] = { static struct snd_pci_quirk cmi9880_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", CMI_FULL_DIG), + SND_PCI_QUIRK(0x1854, 0x002b, "LG LS75", CMI_MINIMAL), SND_PCI_QUIRK(0x1854, 0x0032, "LG", CMI_FULL_DIG), {} /* terminator */ }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 864b2f598c38..b0a2a262ece2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -853,6 +853,7 @@ do_sku: case 0x10ec0269: case 0x10ec0862: case 0x10ec0662: + case 0x10ec0889: snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_EAPD_BTLENABLE, 2); snd_hda_codec_write(codec, 0x15, 0, @@ -877,6 +878,7 @@ do_sku: case 0x10ec0883: case 0x10ec0885: case 0x10ec0888: + case 0x10ec0889: snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 7); tmp = snd_hda_codec_read(codec, 0x20, 0, @@ -940,7 +942,6 @@ do_sku: AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT); spec->unsol_event = alc_sku_unsol_event; - spec->init_hook = alc_sku_automute; } /* @@ -7743,6 +7744,7 @@ 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(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG), + SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG), SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD), @@ -8640,6 +8642,7 @@ static struct hda_verb alc262_sony_unsol_verbs[] = { {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {} }; /* mute/unmute internal speaker according to the hp jack and mute state */ @@ -10510,6 +10513,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA), SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA), + SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA), SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER), SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1), SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO), diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 393f7fd2b1be..a4f44a00bae8 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -840,7 +840,7 @@ static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { static struct snd_kcontrol_new stac925x_mixer[] = { STAC_INPUT_SOURCE(1), HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x14, 0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Mux Volume", 0x0f, 0, HDA_OUTPUT), { } /* end */ }; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 52b1d81a26f7..e7e43524f8c7 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -447,6 +447,23 @@ static struct hda_pcm_stream vt1708_pcm_analog_playback = { }, }; +static struct hda_pcm_stream vt1708_pcm_analog_s16_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 8, + .nid = 0x10, /* NID to query formats and rates */ + /* We got noisy outputs on the right channel on VT1708 when + * 24bit samples are used. Until any workaround is found, + * disable the 24bit format, so far. + */ + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .ops = { + .open = via_playback_pcm_open, + .prepare = via_playback_pcm_prepare, + .cleanup = via_playback_pcm_cleanup + }, +}; + static struct hda_pcm_stream vt1708_pcm_analog_capture = { .substreams = 2, .channels_min = 2, @@ -899,6 +916,9 @@ static int patch_vt1708(struct hda_codec *codec) spec->stream_name_analog = "VT1708 Analog"; spec->stream_analog_playback = &vt1708_pcm_analog_playback; + /* disable 32bit format on VT1708 */ + if (codec->vendor_id == 0x11061708) + spec->stream_analog_playback = &vt1708_pcm_analog_s16_playback; spec->stream_analog_capture = &vt1708_pcm_analog_capture; spec->stream_name_digital = "VT1708 Digital"; diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index cc0cddadd589..6facac5aed90 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -936,11 +936,13 @@ static int add_controls(struct oxygen *chip, for (i = 0; i < count; ++i) { template = controls[i]; - err = chip->model->control_filter(&template); - if (err < 0) - return err; - if (err == 1) - continue; + if (chip->model->control_filter) { + err = chip->model->control_filter(&template); + if (err < 0) + return err; + if (err == 1) + continue; + } if (!strcmp(template.name, "Master Playback Volume") && chip->model->dac_tlv) { template.tlv.p = chip->model->dac_tlv; |
