diff options
Diffstat (limited to 'sound')
28 files changed, 226 insertions, 93 deletions
diff --git a/sound/isa/sb/emu8000_patch.c b/sound/isa/sb/emu8000_patch.c index e09f144177f..c99c6078be3 100644 --- a/sound/isa/sb/emu8000_patch.c +++ b/sound/isa/sb/emu8000_patch.c @@ -22,7 +22,6 @@ #include "emu8000_local.h" #include <asm/uaccess.h> #include <linux/moduleparam.h> -#include <linux/moduleparam.h> static int emu8000_reset_addr; module_param(emu8000_reset_addr, int, 0444); diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 95ffa6a9db6..496f14c1a73 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -2684,10 +2684,9 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); if (err < 0) goto out_err; + opl3->private_data = chip; } - opl3->private_data = chip; - sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, chip->ctrl_io, chip->irq); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c2c65f63bf0..684307372d7 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1759,7 +1759,11 @@ static void put_vol_mute(struct hda_codec *codec, struct hda_amp_info *info, parm = ch ? AC_AMP_SET_RIGHT : AC_AMP_SET_LEFT; parm |= direction == HDA_OUTPUT ? AC_AMP_SET_OUTPUT : AC_AMP_SET_INPUT; parm |= index << AC_AMP_SET_INDEX_SHIFT; - parm |= val; + if ((val & HDA_AMP_MUTE) && !(info->amp_caps & AC_AMPCAP_MUTE) && + (info->amp_caps & AC_AMPCAP_MIN_MUTE)) + ; /* set the zero value as a fake mute */ + else + parm |= val; snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, parm); info->vol[ch] = val; } @@ -2026,7 +2030,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, val1 = -((caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT); val1 += ofs; val1 = ((int)val1) * ((int)val2); - if (min_mute) + if (min_mute || (caps & AC_AMPCAP_MIN_MUTE)) val2 |= TLV_DB_SCALE_MUTE; if (put_user(SNDRV_CTL_TLVT_DB_SCALE, _tlv)) return -EFAULT; @@ -5114,7 +5118,7 @@ static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid, const char *pfx = "", *sfx = ""; /* handle as a speaker if it's a fixed line-out */ - if (!strcmp(name, "Line-Out") && attr == INPUT_PIN_ATTR_INT) + if (!strcmp(name, "Line Out") && attr == INPUT_PIN_ATTR_INT) name = "Speaker"; /* check the location */ switch (attr) { @@ -5173,7 +5177,7 @@ int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, switch (get_defcfg_device(def_conf)) { case AC_JACK_LINE_OUT: - return fill_audio_out_name(codec, nid, cfg, "Line-Out", + return fill_audio_out_name(codec, nid, cfg, "Line Out", label, maxlen, indexp); case AC_JACK_SPEAKER: return fill_audio_out_name(codec, nid, cfg, "Speaker", diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index e9f71dc0d46..f0f1943a4b2 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -298,6 +298,9 @@ enum { #define AC_AMPCAP_MUTE (1<<31) /* mute capable */ #define AC_AMPCAP_MUTE_SHIFT 31 +/* driver-specific amp-caps: using bits 24-30 */ +#define AC_AMPCAP_MIN_MUTE (1 << 30) /* min-volume = mute */ + /* Connection list */ #define AC_CLIST_LENGTH (0x7f<<0) #define AC_CLIST_LONG (1<<7) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 35abe3c6290..21d91d580da 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -728,18 +728,19 @@ static int ca0132_hp_switch_put(struct snd_kcontrol *kcontrol, err = chipio_read(codec, REG_CODEC_MUTE, &data); if (err < 0) - return err; + goto exit; /* *valp 0 is mute, 1 is unmute */ data = (data & 0x7f) | (*valp ? 0 : 0x80); - chipio_write(codec, REG_CODEC_MUTE, data); + err = chipio_write(codec, REG_CODEC_MUTE, data); if (err < 0) - return err; + goto exit; spec->curr_hp_switch = *valp; + exit: snd_hda_power_down(codec); - return 1; + return err < 0 ? err : 1; } static int ca0132_speaker_switch_get(struct snd_kcontrol *kcontrol, @@ -770,18 +771,19 @@ static int ca0132_speaker_switch_put(struct snd_kcontrol *kcontrol, err = chipio_read(codec, REG_CODEC_MUTE, &data); if (err < 0) - return err; + goto exit; /* *valp 0 is mute, 1 is unmute */ data = (data & 0xef) | (*valp ? 0 : 0x10); - chipio_write(codec, REG_CODEC_MUTE, data); + err = chipio_write(codec, REG_CODEC_MUTE, data); if (err < 0) - return err; + goto exit; spec->curr_speaker_switch = *valp; + exit: snd_hda_power_down(codec); - return 1; + return err < 0 ? err : 1; } static int ca0132_hp_volume_get(struct snd_kcontrol *kcontrol, @@ -819,25 +821,26 @@ static int ca0132_hp_volume_put(struct snd_kcontrol *kcontrol, err = chipio_read(codec, REG_CODEC_HP_VOL_L, &data); if (err < 0) - return err; + goto exit; val = 31 - left_vol; data = (data & 0xe0) | val; - chipio_write(codec, REG_CODEC_HP_VOL_L, data); + err = chipio_write(codec, REG_CODEC_HP_VOL_L, data); if (err < 0) - return err; + goto exit; val = 31 - right_vol; data = (data & 0xe0) | val; - chipio_write(codec, REG_CODEC_HP_VOL_R, data); + err = chipio_write(codec, REG_CODEC_HP_VOL_R, data); if (err < 0) - return err; + goto exit; spec->curr_hp_volume[0] = left_vol; spec->curr_hp_volume[1] = right_vol; + exit: snd_hda_power_down(codec); - return 1; + return err < 0 ? err : 1; } static int add_hp_switch(struct hda_codec *codec, hda_nid_t nid) @@ -936,6 +939,8 @@ static int ca0132_build_controls(struct hda_codec *codec) if (err < 0) return err; err = add_in_volume(codec, spec->dig_in, "IEC958"); + if (err < 0) + return err; } return 0; } diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index bc5a993d114..c83ccdba1e5 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -609,7 +609,7 @@ static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx, "Front Speaker", "Surround Speaker", "Bass Speaker" }; static const char * const line_outs[] = { - "Front Line-Out", "Surround Line-Out", "Bass Line-Out" + "Front Line Out", "Surround Line Out", "Bass Line Out" }; fix_volume_caps(codec, dac); @@ -635,7 +635,7 @@ static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx, if (num_ctls > 1) name = line_outs[idx]; else - name = "Line-Out"; + name = "Line Out"; break; } diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 51e3ed4527c..f584f6d8ffc 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3482,7 +3482,7 @@ static int cx_automute_mode_info(struct snd_kcontrol *kcontrol, "Disabled", "Enabled" }; static const char * const texts3[] = { - "Disabled", "Speaker Only", "Line-Out+Speaker" + "Disabled", "Speaker Only", "Line Out+Speaker" }; const char * const *texts; @@ -4079,7 +4079,8 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, err = snd_hda_ctl_add(codec, nid, kctl); if (err < 0) return err; - if (!(query_amp_caps(codec, nid, hda_dir) & AC_AMPCAP_MUTE)) + if (!(query_amp_caps(codec, nid, hda_dir) & + (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE))) break; } return 0; @@ -4379,6 +4380,22 @@ static const struct snd_pci_quirk cxt_fixups[] = { {} }; +/* add "fake" mute amp-caps to DACs on cx5051 so that mixer mute switches + * can be created (bko#42825) + */ +static void add_cx5051_fake_mutes(struct hda_codec *codec) +{ + static hda_nid_t out_nids[] = { + 0x10, 0x11, 0 + }; + hda_nid_t *p; + + for (p = out_nids; *p; p++) + snd_hda_override_amp_caps(codec, *p, HDA_OUTPUT, + AC_AMPCAP_MIN_MUTE | + query_amp_caps(codec, *p, HDA_OUTPUT)); +} + static int patch_conexant_auto(struct hda_codec *codec) { struct conexant_spec *spec; @@ -4397,6 +4414,9 @@ static int patch_conexant_auto(struct hda_codec *codec) case 0x14f15045: spec->single_adc_amp = 1; break; + case 0x14f15051: + add_cx5051_fake_mutes(codec); + break; } apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a8e82be3d2f..22c73b78ac6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -80,6 +80,8 @@ enum { ALC_AUTOMUTE_MIXER, /* mute/unmute mixer widget AMP */ }; +#define MAX_VOL_NIDS 0x40 + struct alc_spec { /* codec parameterization */ const struct snd_kcontrol_new *mixers[5]; /* mixer arrays */ @@ -118,8 +120,8 @@ struct alc_spec { const hda_nid_t *capsrc_nids; hda_nid_t dig_in_nid; /* digital-in NID; optional */ hda_nid_t mixer_nid; /* analog-mixer NID */ - DECLARE_BITMAP(vol_ctls, 0x20 << 1); - DECLARE_BITMAP(sw_ctls, 0x20 << 1); + DECLARE_BITMAP(vol_ctls, MAX_VOL_NIDS << 1); + DECLARE_BITMAP(sw_ctls, MAX_VOL_NIDS << 1); /* capture setup for dynamic dual-adc switch */ hda_nid_t cur_adc; @@ -800,7 +802,7 @@ static int alc_automute_mode_info(struct snd_kcontrol *kcontrol, "Disabled", "Enabled" }; static const char * const texts3[] = { - "Disabled", "Speaker Only", "Line-Out+Speaker" + "Disabled", "Speaker Only", "Line Out+Speaker" }; const char * const *texts; @@ -1854,7 +1856,9 @@ static const char * const alc_slave_vols[] = { "Headphone Playback Volume", "Speaker Playback Volume", "Mono Playback Volume", - "Line-Out Playback Volume", + "Line Out Playback Volume", + "CLFE Playback Volume", + "Bass Speaker Playback Volume", "PCM Playback Volume", NULL, }; @@ -1869,7 +1873,9 @@ static const char * const alc_slave_sws[] = { "Speaker Playback Switch", "Mono Playback Switch", "IEC958 Playback Switch", - "Line-Out Playback Switch", + "Line Out Playback Switch", + "CLFE Playback Switch", + "Bass Speaker Playback Switch", "PCM Playback Switch", NULL, }; @@ -2062,12 +2068,16 @@ static int alc_build_controls(struct hda_codec *codec) */ static void alc_init_special_input_src(struct hda_codec *codec); +static int alc269_fill_coef(struct hda_codec *codec); static int alc_init(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; unsigned int i; + if (codec->vendor_id == 0x10ec0269) + alc269_fill_coef(codec); + alc_fix_pll(codec); alc_auto_init_amp(codec, spec->init_amp); @@ -2318,7 +2328,7 @@ static int alc_build_pcms(struct hda_codec *codec) "%s Analog", codec->chip_name); info->name = spec->stream_name_analog; - if (spec->multiout.dac_nids > 0) { + if (spec->multiout.num_dacs > 0) { p = spec->stream_analog_playback; if (!p) p = &alc_pcm_analog_playback; @@ -3145,7 +3155,10 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) static inline unsigned int get_ctl_pos(unsigned int data) { hda_nid_t nid = get_amp_nid_(data); - unsigned int dir = get_amp_direction_(data); + unsigned int dir; + if (snd_BUG_ON(nid >= MAX_VOL_NIDS)) + return 0; + dir = get_amp_direction_(data); return (nid << 1) | dir; } @@ -3788,7 +3801,7 @@ static void alc_auto_init_input_src(struct hda_codec *codec) else nums = spec->num_adc_nids; for (c = 0; c < nums; c++) - alc_mux_select(codec, 0, spec->cur_mux[c], true); + alc_mux_select(codec, c, spec->cur_mux[c], true); } /* add mic boosts if needed */ @@ -4358,6 +4371,7 @@ enum { ALC882_FIXUP_PB_M5210, ALC882_FIXUP_ACER_ASPIRE_7736, ALC882_FIXUP_ASUS_W90V, + ALC889_FIXUP_CD, ALC889_FIXUP_VAIO_TT, ALC888_FIXUP_EEE1601, ALC882_FIXUP_EAPD, @@ -4370,6 +4384,7 @@ enum { ALC882_FIXUP_ACER_ASPIRE_8930G, ALC882_FIXUP_ASPIRE_8930G_VERBS, ALC885_FIXUP_MACPRO_GPIO, + ALC889_FIXUP_DAC_ROUTE, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -4423,6 +4438,31 @@ static void alc885_fixup_macpro_gpio(struct hda_codec *codec, alc882_gpio_mute(codec, 1, 0); } +/* Fix the connection of some pins for ALC889: + * At least, Acer Aspire 5935 shows the connections to DAC3/4 don't + * work correctly (bko#42740) + */ +static void alc889_fixup_dac_route(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + if (action == ALC_FIXUP_ACT_PRE_PROBE) { + /* fake the connections during parsing the tree */ + hda_nid_t conn1[2] = { 0x0c, 0x0d }; + hda_nid_t conn2[2] = { 0x0e, 0x0f }; + snd_hda_override_conn_list(codec, 0x14, 2, conn1); + snd_hda_override_conn_list(codec, 0x15, 2, conn1); + snd_hda_override_conn_list(codec, 0x18, 2, conn2); + snd_hda_override_conn_list(codec, 0x1a, 2, conn2); + } else if (action == ALC_FIXUP_ACT_PROBE) { + /* restore the connections */ + hda_nid_t conn[5] = { 0x0c, 0x0d, 0x0e, 0x0f, 0x26 }; + snd_hda_override_conn_list(codec, 0x14, 5, conn); + snd_hda_override_conn_list(codec, 0x15, 5, conn); + snd_hda_override_conn_list(codec, 0x18, 5, conn); + snd_hda_override_conn_list(codec, 0x1a, 5, conn); + } +} + static const struct alc_fixup alc882_fixups[] = { [ALC882_FIXUP_ABIT_AW9D_MAX] = { .type = ALC_FIXUP_PINS, @@ -4459,6 +4499,13 @@ static const struct alc_fixup alc882_fixups[] = { { } } }, + [ALC889_FIXUP_CD] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x1c, 0x993301f0 }, /* CD */ + { } + } + }, [ALC889_FIXUP_VAIO_TT] = { .type = ALC_FIXUP_PINS, .v.pins = (const struct alc_pincfg[]) { @@ -4570,6 +4617,10 @@ static const struct alc_fixup alc882_fixups[] = { .type = ALC_FIXUP_FUNC, .v.func = alc885_fixup_macpro_gpio, }, + [ALC889_FIXUP_DAC_ROUTE] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc889_fixup_dac_route, + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -4594,6 +4645,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G", ALC882_FIXUP_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210), + SND_PCI_QUIRK(0x1025, 0x0259, "Acer Aspire 5935", ALC889_FIXUP_DAC_ROUTE), SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", ALC882_FIXUP_ACER_ASPIRE_7736), SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V), @@ -4610,6 +4662,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), + SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3", ALC889_FIXUP_CD), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), @@ -5427,8 +5480,12 @@ static const struct alc_model_fixup alc269_fixup_models[] = { static int alc269_fill_coef(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; int val; + if (spec->codec_variant != ALC269_TYPE_ALC269VB) + return 0; + if ((alc_get_coef0(codec) & 0x00ff) < 0x015) { alc_write_coef_idx(codec, 0xf, 0x960b); alc_write_coef_idx(codec, 0xe, 0x8817); @@ -5623,6 +5680,7 @@ static const struct alc_fixup alc861_fixups[] = { static const struct snd_pci_quirk alc861_fixup_tbl[] = { SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", PINFIX_ASUS_A6RP), + SND_PCI_QUIRK(0x1584, 0x0000, "Uniwill ECS M31EI", PINFIX_ASUS_A6RP), SND_PCI_QUIRK(0x1584, 0x2b01, "Haier W18", PINFIX_ASUS_A6RP), SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", PINFIX_FSC_AMILO_PI1505), {} diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 948f0be2f4f..9dbb5735d77 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4629,7 +4629,7 @@ static void stac92xx_hp_detect(struct hda_codec *codec) unsigned int val = AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN; if (no_hp_sensing(spec, i)) continue; - if (presence) + if (1 /*presence*/) stac92xx_set_pinctl(codec, cfg->hp_pins[i], val); #if 0 /* FIXME */ /* Resetting the pinctl like below may lead to (a sort of) regressions @@ -5078,9 +5078,9 @@ static int stac92xx_update_led_status(struct hda_codec *codec) spec->gpio_dir, spec->gpio_data); } else { notmtd_lvl = spec->gpio_led_polarity ? - AC_PINCTL_VREF_HIZ : AC_PINCTL_VREF_GRD; + AC_PINCTL_VREF_50 : AC_PINCTL_VREF_GRD; muted_lvl = spec->gpio_led_polarity ? - AC_PINCTL_VREF_GRD : AC_PINCTL_VREF_HIZ; + AC_PINCTL_VREF_GRD : AC_PINCTL_VREF_50; spec->vref_led = muted ? muted_lvl : notmtd_lvl; stac_vrefout_set(codec, spec->vref_mute_led_nid, spec->vref_led); diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 284e311040f..dff9a00ee8f 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -666,6 +666,9 @@ static void via_auto_init_analog_input(struct hda_codec *codec) /* init input-src */ for (i = 0; i < spec->num_adc_nids; i++) { int adc_idx = spec->inputs[spec->cur_mux[i]].adc_idx; + /* secondary ADCs must have the unique MUX */ + if (i > 0 && !spec->mux_nids[i]) + break; if (spec->mux_nids[adc_idx]) { int mux_idx = spec->inputs[spec->cur_mux[i]].mux_idx; snd_hda_codec_write(codec, spec->mux_nids[adc_idx], 0, diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 9f3b01bb72c..e0a4263baa2 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -2102,6 +2102,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { }, { .subvendor = 0x161f, + .subdevice = 0x202f, + .name = "Gateway M520", + .type = AC97_TUNE_INV_EAPD + }, + { + .subvendor = 0x161f, .subdevice = 0x203a, .name = "Gateway 4525GZ", /* AD1981B */ .type = AC97_TUNE_INV_EAPD diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 26c7e8bcb22..c0dbb52d45b 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -618,9 +618,12 @@ static int ac97_volume_get(struct snd_kcontrol *ctl, mutex_lock(&chip->mutex); reg = oxygen_read_ac97(chip, codec, index); mutex_unlock(&chip->mutex); - value->value.integer.value[0] = 31 - (reg & 0x1f); - if (stereo) - value->value.integer.value[1] = 31 - ((reg >> 8) & 0x1f); + if (!stereo) { + value->value.integer.value[0] = 31 - (reg & 0x1f); + } else { + value->value.integer.value[0] = 31 - ((reg >> 8) & 0x1f); + value->value.integer.value[1] = 31 - (reg & 0x1f); + } return 0; } @@ -636,14 +639,14 @@ static int ac97_volume_put(struct snd_kcontrol *ctl, mutex_lock(&chip->mutex); oldreg = oxygen_read_ac97(chip, codec, index); - newreg = oldreg; - newreg = (newreg & ~0x1f) | - (31 - (value->value.integer.value[0] & 0x1f)); - if (stereo) - newreg = (newreg & ~0x1f00) | - ((31 - (value->value.integer.value[1] & 0x1f)) << 8); - else - newreg = (newreg & ~0x1f00) | ((newreg & 0x1f) << 8); + if (!stereo) { + newreg = oldreg & ~0x1f; + newreg |= 31 - (value->value.integer.value[0] & 0x1f); + } else { + newreg = oldreg & ~0x1f1f; + newreg |= (31 - (value->value.integer.value[0] & 0x1f)) << 8; + newreg |= 31 - (value->value.integer.value[1] & 0x1f); + } change = newreg != oldreg; if (change) oxygen_write_ac97(chip, codec, index, newreg); diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index cc9f6c83d66..bc030a2088d 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6333,6 +6333,7 @@ static int __devinit snd_hdspm_create_hwdep(struct snd_card *card, hw->ops.open = snd_hdspm_hwdep_dummy_op; hw->ops.ioctl = snd_hdspm_hwdep_ioctl; + hw->ops.ioctl_compat = snd_hdspm_hwdep_ioctl; hw->ops.release = snd_hdspm_hwdep_dummy_op; return 0; diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 5ef70b5d27e..278c0a0575f 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -146,13 +146,10 @@ static const struct snd_kcontrol_new ak4642_snd_controls[] = { SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC, 0, 0xFF, 1, out_tlv), - - SOC_SINGLE("Headphone Switch", PW_MGMT2, 6, 1, 0), }; -static const struct snd_kcontrol_new ak4642_hpout_mixer_controls[] = { - SOC_DAPM_SINGLE("DACH", MD_CTL4, 0, 1, 0), -}; +static const struct snd_kcontrol_new ak4642_headphone_control = + SOC_DAPM_SINGLE("Switch", PW_MGMT2, 6, 1, 0); static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = { SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0), @@ -165,13 +162,12 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HPOUTR"), SND_SOC_DAPM_OUTPUT("LINEOUT"), - SND_SOC_DAPM_MIXER("HPOUTL Mixer", PW_MGMT2, 5, 0, - &ak4642_hpout_mixer_controls[0], - ARRAY_SIZE(ak4642_hpout_mixer_controls)), + SND_SOC_DAPM_PGA("HPL Out", PW_MGMT2, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPR Out", PW_MGMT2, 4, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("Headphone Enable", SND_SOC_NOPM, 0, 0, + &ak4642_headphone_control), - SND_SOC_DAPM_MIXER("HPOUTR Mixer", PW_MGMT2, 4, 0, - &ak4642_hpout_mixer_controls[0], - ARRAY_SIZE(ak4642_hpout_mixer_controls)), + SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0, &ak4642_lout_mixer_controls[0], @@ -184,12 +180,17 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { static const struct snd_soc_dapm_route ak4642_intercon[] = { /* Outputs */ - {"HPOUTL", NULL, "HPOUTL Mixer"}, - {"HPOUTR", NULL, "HPOUTR Mixer"}, + {"HPOUTL", NULL, "HPL Out"}, + {"HPOUTR", NULL, "HPR Out"}, {"LINEOUT", NULL, "LINEOUT Mixer"}, - {"HPOUTL Mixer", "DACH", "DAC"}, - {"HPOUTR Mixer", "DACH", "DAC"}, + {"HPL Out", NULL, "Headphone Enable"}, + {"HPR Out", NULL, "Headphone Enable"}, + + {"Headphone Enable", "Switch", "DACH"}, + + {"DACH", NULL, "DAC"}, + {"LINEOUT Mixer", "DACL", "DAC"}, }; diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 9d38db8f191..78979b3e0e9 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1113,7 +1113,7 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, priv->config[id].mmcc &= 0xC0; priv->config[id].mmcc |= cs42l73_mclk_coeffs[mclk_coeff].mmcc; priv->config[id].spc &= 0xFC; - priv->config[id].spc &= MCK_SCLK_64FS; + priv->config[id].spc |= MCK_SCLK_MCLK; } else { /* CS42L73 Slave */ priv->config[id].spc &= 0xFC; diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index bda3da887d7..0ac228b7dc0 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2564,7 +2564,7 @@ static int dsp2_event(struct snd_soc_dapm_widget *w, return 0; } -static const char *st_text[] = { "None", "Right", "Left" }; +static const char *st_text[] = { "None", "Left", "Right" }; static const struct soc_enum str_enum = SOC_ENUM_SINGLE(WM8962_DAC_DSP_MIXING_1, 2, 3, st_text); @@ -3159,13 +3159,13 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: break; case SNDRV_PCM_FORMAT_S20_3LE: - aif0 |= 0x40; + aif0 |= 0x4; break; case SNDRV_PCM_FORMAT_S24_LE: - aif0 |= 0x80; + aif0 |= 0x8; break; case SNDRV_PCM_FORMAT_S32_LE: - aif0 |= 0xc0; + aif0 |= 0xc; break; default: return -EINVAL; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 93d27b66025..ec69a6c152f 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -770,6 +770,8 @@ static void vmid_reference(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + pm_runtime_get_sync(codec->dev); + wm8994->vmid_refcount++; dev_dbg(codec->dev, "Referencing VMID, refcount is now %d\n", @@ -783,7 +785,12 @@ static void vmid_reference(struct snd_soc_codec *codec) WM8994_VMID_RAMP_MASK, WM8994_STARTUP_BIAS_ENA | WM8994_VMID_BUF_ENA | - (0x11 << WM8994_VMID_RAMP_SHIFT)); + (0x3 << WM8994_VMID_RAMP_SHIFT)); + + /* Remove discharge for line out */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_1, + WM8994_LINEOUT1_DISCH | + WM8994_LINEOUT2_DISCH, 0); /* Main bias enable, VMID=2x40k */ snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, @@ -837,6 +844,8 @@ static void vmid_dereference(struct snd_soc_codec *codec) WM8994_VMID_BUF_ENA | WM8994_VMID_RAMP_MASK, 0); } + + pm_runtime_put(codec->dev); } static int vmid_event(struct snd_soc_dapm_widget *w, @@ -2753,11 +2762,6 @@ static int wm8994_resume(struct snd_soc_codec *codec) codec->cache_only = 0; } - /* Restore the registers */ - ret = snd_soc_cache_sync(codec); - if (ret != 0) - dev_err(codec->dev, "Failed to sync cache: %d\n", ret); - wm8994_set_bias_level(codec, SND_SOC_BIAS_STANDBY); for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) { diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index ea2672455d0..8a68cea4a3e 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -586,8 +586,8 @@ SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER1, 0, 1, 0), }; static const struct snd_kcontrol_new line2_mix[] = { -SOC_DAPM_SINGLE("IN2R Switch", WM8993_LINE_MIXER2, 2, 1, 0), -SOC_DAPM_SINGLE("IN2L Switch", WM8993_LINE_MIXER2, 1, 1, 0), +SOC_DAPM_SINGLE("IN1L Switch", WM8993_LINE_MIXER2, 2, 1, 0), +SOC_DAPM_SINGLE("IN1R Switch", WM8993_LINE_MIXER2, 1, 1, 0), SOC_DAPM_SINGLE("Output Switch", WM8993_LINE_MIXER2, 0, 1, 0), }; @@ -848,8 +848,8 @@ static const struct snd_soc_dapm_route lineout1_se_routes[] = { }; static const struct snd_soc_dapm_route lineout2_diff_routes[] = { - { "LINEOUT2 Mixer", "IN2L Switch", "IN2L PGA" }, - { "LINEOUT2 Mixer", "IN2R Switch", "IN2R PGA" }, + { "LINEOUT2 Mixer", "IN1L Switch", "IN1L PGA" }, + { "LINEOUT2 Mixer", "IN1R Switch", "IN1R PGA" }, { "LINEOUT2 Mixer", "Output Switch", "Right Output PGA" }, { "LINEOUT2N Driver", NULL, "LINEOUT2 Mixer" }, diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 01d1f749cf0..b6adbed6e50 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -112,7 +112,7 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) break; case SND_SOC_DAIFMT_DSP_A: /* data on rising edge of bclk, frame high 1clk before data */ - strcr |= SSI_STCR_TFSL | SSI_STCR_TEFS; + strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0 | SSI_STCR_TEFS; break; } diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index c6012ff5bd3..d23b19a59d8 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -367,7 +367,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { .platform_name = "samsung-audio", .cpu_dai_name = "s3c24xx-iis", .codec_dai_name = "wm8753-hifi", - .codec_name = "wm8753-codec.0-001a", + .codec_name = "wm8753.0-001a", .init = neo1973_wm8753_init, .ops = &neo1973_hifi_ops, }, @@ -376,7 +376,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { .stream_name = "Voice", .cpu_dai_name = "dfbmcs320-pcm", .codec_dai_name = "wm8753-voice", - .codec_name = "wm8753-codec.0-001a", + .codec_name = "wm8753.0-001a", .ops = &neo1973_voice_ops, }, }; diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index db6c89a28bd..ea4a82d0116 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1152,12 +1152,8 @@ static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream) { struct fsi_priv *fsi = fsi_get_priv(substream); struct fsi_stream *io = fsi_get_stream(fsi, fsi_is_play(substream)); - int samples_pos = io->buff_sample_pos - 1; - if (samples_pos < 0) - samples_pos = 0; - - return fsi_sample2frame(fsi, samples_pos); + return fsi_sample2frame(fsi, io->buff_sample_pos); } static struct snd_pcm_ops fsi_pcm_ops = { diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b5ecf6d2321..92cee24ed2d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -567,6 +567,17 @@ int snd_soc_suspend(struct device *dev) if (!codec->suspended && codec->driver->suspend) { switch (codec->dapm.bias_level) { case SND_SOC_BIAS_STANDBY: + /* + * If the CODEC is capable of idle + * bias off then being in STANDBY + * means it's doing something, + * otherwise fall through. + */ + if (codec->dapm.idle_bias_off) { + dev_dbg(codec->dev, + "idle_bias_off CODEC on over suspend\n"); + break; + } case SND_SOC_BIAS_OFF: codec->driver->suspend(codec); codec->suspended = 1; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1f55ded4047..1315663c1c0 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3068,9 +3068,13 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) * standby. */ if (powerdown) { - snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_PREPARE); + if (dapm->bias_level == SND_SOC_BIAS_ON) + snd_soc_dapm_set_bias_level(dapm, + SND_SOC_BIAS_PREPARE); dapm_seq_run(dapm, &down_list, 0, false); - snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_STANDBY); + if (dapm->bias_level == SND_SOC_BIAS_PREPARE) + snd_soc_dapm_set_bias_level(dapm, + SND_SOC_BIAS_STANDBY); } } @@ -3083,7 +3087,9 @@ void snd_soc_dapm_shutdown(struct snd_soc_card *card) list_for_each_entry(codec, &card->codec_dev_list, list) { soc_dapm_shutdown_codec(&codec->dapm); - snd_soc_dapm_set_bias_level(&codec->dapm, SND_SOC_BIAS_OFF); + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) + snd_soc_dapm_set_bias_level(&codec->dapm, + SND_SOC_BIAS_OFF); } } diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 2cf87f5afed..fde9a7a29cb 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -311,8 +311,10 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub) spin_lock(&dev->spinlock); - if (dev->input_panic || dev->output_panic) + if (dev->input_panic || dev->output_panic) { ptr = SNDRV_PCM_POS_XRUN; + goto unlock; + } if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) ptr = bytes_to_frames(sub->runtime, @@ -321,6 +323,7 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub) ptr = bytes_to_frames(sub->runtime, dev->audio_in_buf_pos[index]); +unlock: spin_unlock(&dev->spinlock); return ptr; } diff --git a/sound/usb/card.h b/sound/usb/card.h index a39edcc32a9..da5fa1ac4ed 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -1,6 +1,7 @@ #ifndef __USBAUDIO_CARD_H #define __USBAUDIO_CARD_H +#define MAX_NR_RATES 1024 #define MAX_PACKS 20 #define MAX_PACKS_HS (MAX_PACKS * 8) /* in high speed mode */ #define MAX_URBS 8 diff --git a/sound/usb/format.c b/sound/usb/format.c index e09aba19375..ddfef57c4c9 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -209,8 +209,6 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof return 0; } -#define MAX_UAC2_NR_RATES 1024 - /* * Helper function to walk the array of sample rate triplets reported by * the device. The problem is that we need to parse whole array first to @@ -255,7 +253,7 @@ static int parse_uac2_sample_rate_range(struct audioformat *fp, int nr_triplets, fp->rates |= snd_pcm_rate_to_rate_bit(rate); nr_rates++; - if (nr_rates >= MAX_UAC2_NR_RATES) { + if (nr_rates >= MAX_NR_RATES) { snd_printk(KERN_ERR "invalid uac2 rates\n"); break; } diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 8edc5035fc8..d89ab4c7d44 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1618,6 +1618,14 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + /* Edirol UM-3G */ + USB_DEVICE_VENDOR_SPEC(0x0582, 0x0108), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .ifnum = 0, + .type = QUIRK_MIDI_STANDARD_INTERFACE + } +}, +{ /* Boss JS-8 Jam Station */ USB_DEVICE(0x0582, 0x0109), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index a3ddac0deff..27817266867 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -132,10 +132,14 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, unsigned *rate_table = NULL; fp = kmemdup(quirk->data, sizeof(*fp), GFP_KERNEL); - if (! fp) { + if (!fp) { snd_printk(KERN_ERR "cannot memdup\n"); return -ENOMEM; } + if (fp->nr_rates > MAX_NR_RATES) { + kfree(fp); + return -EINVAL; + } if (fp->nr_rates > 0) { rate_table = kmemdup(fp->rate_table, sizeof(int) * fp->nr_rates, GFP_KERNEL); |