diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index a9a2e63c0222..e8a1d18774b2 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -76,6 +76,7 @@ static int snd_mixer_oss_open(struct inode *inode, struct file *file) snd_card_unref(card); return -EFAULT; } + snd_card_unref(card); return 0; } diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index f337b66a020b..4c1cc51772e6 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2454,6 +2454,7 @@ static int snd_pcm_oss_open(struct inode *inode, struct file *file) mutex_unlock(&pcm->open_mutex); if (err < 0) goto __error; + snd_card_unref(pcm->card); return err; __error: diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index b329ca58e63c..c76f6375a96b 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2124,7 +2124,8 @@ static int snd_pcm_playback_open(struct inode *inode, struct file *file) pcm = snd_lookup_minor_data(iminor(inode), SNDRV_DEVICE_TYPE_PCM_PLAYBACK); err = snd_pcm_open(file, pcm, SNDRV_PCM_STREAM_PLAYBACK); - snd_card_unref(pcm->card); + if (pcm) + snd_card_unref(pcm->card); return err; } @@ -2137,7 +2138,8 @@ static int snd_pcm_capture_open(struct inode *inode, struct file *file) pcm = snd_lookup_minor_data(iminor(inode), SNDRV_DEVICE_TYPE_PCM_CAPTURE); err = snd_pcm_open(file, pcm, SNDRV_PCM_STREAM_CAPTURE); - snd_card_unref(pcm->card); + if (pcm) + snd_card_unref(pcm->card); return err; } diff --git a/sound/core/sound.c b/sound/core/sound.c index 89780c323f19..70ccdab74153 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -114,7 +114,7 @@ void *snd_lookup_minor_data(unsigned int minor, int type) mreg = snd_minors[minor]; if (mreg && mreg->type == type) { private_data = mreg->private_data; - if (mreg->card_ptr) + if (private_data && mreg->card_ptr) atomic_inc(&mreg->card_ptr->refcount); } else private_data = NULL; diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c index e1d79ee35906..726a49ac9725 100644 --- a/sound/core/sound_oss.c +++ b/sound/core/sound_oss.c @@ -54,7 +54,7 @@ void *snd_lookup_oss_minor_data(unsigned int minor, int type) mreg = snd_oss_minors[minor]; if (mreg && mreg->type == type) { private_data = mreg->private_data; - if (mreg->card_ptr) + if (private_data && mreg->card_ptr) atomic_inc(&mreg->card_ptr->refcount); } else private_data = NULL; diff --git a/sound/i2c/other/ak4113.c b/sound/i2c/other/ak4113.c index ef68d710d08c..e04e750a77ed 100644 --- a/sound/i2c/other/ak4113.c +++ b/sound/i2c/other/ak4113.c @@ -426,7 +426,7 @@ static struct snd_kcontrol_new snd_ak4113_iec958_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = "IEC958 Preample Capture Default", + .name = "IEC958 Preamble Capture Default", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = snd_ak4113_spdif_pinfo, diff --git a/sound/i2c/other/ak4114.c b/sound/i2c/other/ak4114.c index 816e7d225fb0..5bf4fca19e48 100644 --- a/sound/i2c/other/ak4114.c +++ b/sound/i2c/other/ak4114.c @@ -401,7 +401,7 @@ static struct snd_kcontrol_new snd_ak4114_iec958_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = "IEC958 Preample Capture Default", + .name = "IEC958 Preamble Capture Default", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = snd_ak4114_spdif_pinfo, .get = snd_ak4114_spdif_pget, diff --git a/sound/i2c/other/ak4117.c b/sound/i2c/other/ak4117.c index b4b2a51fc117..40e33c9f2b09 100644 --- a/sound/i2c/other/ak4117.c +++ b/sound/i2c/other/ak4117.c @@ -380,7 +380,7 @@ static struct snd_kcontrol_new snd_ak4117_iec958_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = "IEC958 Preample Capture Default", + .name = "IEC958 Preamble Capture Default", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = snd_ak4117_spdif_pinfo, .get = snd_ak4117_spdif_pget, diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 5d0e568fdea1..50169bcfd903 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -2655,6 +2655,8 @@ static struct ess_device_list pm_whitelist[] __devinitdata = { { TYPE_MAESTRO2E, 0x1179 }, { TYPE_MAESTRO2E, 0x14c0 }, /* HP omnibook 4150 */ { TYPE_MAESTRO2E, 0x1558 }, + { TYPE_MAESTRO2E, 0x125d }, /* a PCI card, e.g. Terratec DMX */ + { TYPE_MAESTRO2, 0x125d }, /* a PCI card, e.g. SF64-PCE2 */ }; static struct ess_device_list mpu_blacklist[] __devinitdata = { diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index bffaaeb0b4ba..2cc07c184d31 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3683,6 +3683,8 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* Teradici */ { PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA | AZX_DCAPS_NO_64BIT }, + { PCI_DEVICE(0x6549, 0x2200), + .driver_data = AZX_DRIVER_TERA | AZX_DCAPS_NO_64BIT }, /* Creative X-Fi (CA0110-IBG) */ /* CTHDA chips */ { PCI_DEVICE(0x1102, 0x0010), diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index cdd43eadbc67..1eeba7386666 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -545,6 +545,7 @@ static int ad198x_build_pcms(struct hda_codec *codec) if (spec->multiout.dig_out_nid) { info++; codec->num_pcms++; + codec->spdif_status_reset = 1; info->name = "AD198x Digital"; info->pcm_type = HDA_PCM_TYPE_SPDIF; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad198x_pcm_digital_playback; diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index a7f8790ae885..b9039dbd704d 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -101,8 +101,8 @@ enum { #define CS420X_VENDOR_NID 0x11 #define CS_DIG_OUT1_PIN_NID 0x10 #define CS_DIG_OUT2_PIN_NID 0x15 -#define CS_DMIC1_PIN_NID 0x12 -#define CS_DMIC2_PIN_NID 0x0e +#define CS_DMIC1_PIN_NID 0x0e +#define CS_DMIC2_PIN_NID 0x12 /* coef indices */ #define IDX_SPDIF_STAT 0x0000 @@ -1080,14 +1080,18 @@ static void init_input(struct hda_codec *codec) cs_automic(codec, NULL); coef = 0x000a; /* ADC1/2 - Digital and Analog Soft Ramp */ + cs_vendor_coef_set(codec, IDX_ADC_CFG, coef); + + coef = cs_vendor_coef_get(codec, IDX_BEEP_CFG); if (is_active_pin(codec, CS_DMIC2_PIN_NID)) - coef |= 0x0500; /* DMIC2 2 chan on, GPIO1 off */ + coef |= 1 << 4; /* DMIC2 2 chan on, GPIO1 off */ if (is_active_pin(codec, CS_DMIC1_PIN_NID)) - coef |= 0x1800; /* DMIC1 2 chan on, GPIO0 off + coef |= 1 << 3; /* DMIC1 2 chan on, GPIO0 off * No effect if SPDIF_OUT2 is * selected in IDX_SPDIF_CTL. */ - cs_vendor_coef_set(codec, IDX_ADC_CFG, coef); + + cs_vendor_coef_set(codec, IDX_BEEP_CFG, coef); } else { if (spec->mic_detect) cs_automic(codec, NULL); @@ -1108,7 +1112,7 @@ static const struct hda_verb cs_coef_init_verbs[] = { | 0x0400 /* Disable Coefficient Auto increment */ )}, /* Beep */ - {0x11, AC_VERB_SET_COEF_INDEX, IDX_DAC_CFG}, + {0x11, AC_VERB_SET_COEF_INDEX, IDX_BEEP_CFG}, {0x11, AC_VERB_SET_PROC_COEF, 0x0007}, /* Enable Beep thru DAC1/2/3 */ {} /* terminator */ @@ -1729,8 +1733,7 @@ static int cs421x_mux_enum_put(struct snd_kcontrol *kcontrol, } -static struct snd_kcontrol_new cs421x_capture_source = { - +static const struct snd_kcontrol_new cs421x_capture_source = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Source", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -1947,7 +1950,7 @@ static int cs421x_suspend(struct hda_codec *codec) } #endif -static struct hda_codec_ops cs421x_patch_ops = { +static const struct hda_codec_ops cs421x_patch_ops = { .build_controls = cs421x_build_controls, .build_pcms = cs_build_pcms, .init = cs421x_init, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 96c08792d210..bc71be32232e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5841,7 +5841,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) return alc_parse_auto_config(codec, alc269_ignore, ssids); } -static void alc269_toggle_power_output(struct hda_codec *codec, int power_up) +static void alc269vb_toggle_power_output(struct hda_codec *codec, int power_up) { int val = alc_read_coef_idx(codec, 0x04); if (power_up) @@ -5858,10 +5858,10 @@ static void alc269_shutup(struct hda_codec *codec) if (spec->codec_variant != ALC269_TYPE_ALC269VB) return; - if ((alc_get_coef0(codec) & 0x00ff) == 0x017) - alc269_toggle_power_output(codec, 0); - if ((alc_get_coef0(codec) & 0x00ff) == 0x018) { - alc269_toggle_power_output(codec, 0); + if (spec->codec_variant == ALC269_TYPE_ALC269VB) + alc269vb_toggle_power_output(codec, 0); + if (spec->codec_variant == ALC269_TYPE_ALC269VB && + (alc_get_coef0(codec) & 0x00ff) == 0x018) { msleep(150); } } @@ -5871,24 +5871,22 @@ static int alc269_resume(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - if (spec->codec_variant == ALC269_TYPE_ALC269VB || + if (spec->codec_variant == ALC269_TYPE_ALC269VB) + alc269vb_toggle_power_output(codec, 0); + if (spec->codec_variant == ALC269_TYPE_ALC269VB && (alc_get_coef0(codec) & 0x00ff) == 0x018) { - alc269_toggle_power_output(codec, 0); msleep(150); } codec->patch_ops.init(codec); - if (spec->codec_variant == ALC269_TYPE_ALC269VB || + if (spec->codec_variant == ALC269_TYPE_ALC269VB) + alc269vb_toggle_power_output(codec, 1); + if (spec->codec_variant == ALC269_TYPE_ALC269VB && (alc_get_coef0(codec) & 0x00ff) == 0x017) { - alc269_toggle_power_output(codec, 1); msleep(200); } - if (spec->codec_variant == ALC269_TYPE_ALC269VB || - (alc_get_coef0(codec) & 0x00ff) == 0x018) - alc269_toggle_power_output(codec, 1); - snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); hda_call_check_power_status(codec, 0x01); @@ -7080,6 +7078,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { .patch = patch_alc662 }, { .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 }, { .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 }, + { .id = 0x10ec0668, .name = "ALC668", .patch = patch_alc662 }, { .id = 0x10ec0670, .name = "ALC670", .patch = patch_alc662 }, { .id = 0x10ec0680, .name = "ALC680", .patch = patch_alc680 }, { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 }, @@ -7097,6 +7096,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc882 }, { .id = 0x10ec0892, .name = "ALC892", .patch = patch_alc662 }, { .id = 0x10ec0899, .name = "ALC898", .patch = patch_alc882 }, + { .id = 0x10ec0900, .name = "ALC1150", .patch = patch_alc882 }, {} /* terminator */ }; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 72a2f60b087c..019e1a00414a 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1809,11 +1809,11 @@ static int via_auto_fill_dac_nids(struct hda_codec *codec) { struct via_spec *spec = codec->spec; const struct auto_pin_cfg *cfg = &spec->autocfg; - int i, dac_num; + int i; hda_nid_t nid; + spec->multiout.num_dacs = 0; spec->multiout.dac_nids = spec->private_dac_nids; - dac_num = 0; for (i = 0; i < cfg->line_outs; i++) { hda_nid_t dac = 0; nid = cfg->line_out_pins[i]; @@ -1824,16 +1824,13 @@ static int via_auto_fill_dac_nids(struct hda_codec *codec) if (!i && parse_output_path(codec, nid, dac, 1, &spec->out_mix_path)) dac = spec->out_mix_path.path[0]; - if (dac) { - spec->private_dac_nids[i] = dac; - dac_num++; - } + if (dac) + spec->private_dac_nids[spec->multiout.num_dacs++] = dac; } if (!spec->out_path[0].depth && spec->out_mix_path.depth) { spec->out_path[0] = spec->out_mix_path; spec->out_mix_path.depth = 0; } - spec->multiout.num_dacs = dac_num; return 0; } @@ -3628,6 +3625,7 @@ static void set_widgets_power_state_vt2002P(struct hda_codec *codec) */ enum { VIA_FIXUP_INTMIC_BOOST, + VIA_FIXUP_ASUS_G75, }; static void via_fixup_intmic_boost(struct hda_codec *codec, @@ -3642,13 +3640,35 @@ static const struct hda_fixup via_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = via_fixup_intmic_boost, }, + [VIA_FIXUP_ASUS_G75] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + /* set 0x24 and 0x33 as speakers */ + { 0x24, 0x991301f0 }, + { 0x33, 0x991301f1 }, /* subwoofer */ + { } + } + }, }; static const struct snd_pci_quirk vt2002p_fixups[] = { + SND_PCI_QUIRK(0x1043, 0x1487, "Asus G75", VIA_FIXUP_ASUS_G75), SND_PCI_QUIRK(0x1043, 0x8532, "Asus X202E", VIA_FIXUP_INTMIC_BOOST), {} }; +/* NIDs 0x24 and 0x33 on VT1802 have connections to non-existing NID 0x3e + * Replace this with mixer NID 0x1c + */ +static void fix_vt1802_connections(struct hda_codec *codec) +{ + static hda_nid_t conn_24[] = { 0x14, 0x1c }; + static hda_nid_t conn_33[] = { 0x1c }; + + snd_hda_override_conn_list(codec, 0x24, ARRAY_SIZE(conn_24), conn_24); + snd_hda_override_conn_list(codec, 0x33, ARRAY_SIZE(conn_33), conn_33); +} + /* patch for vt2002P */ static int patch_vt2002P(struct hda_codec *codec) { @@ -3663,6 +3683,8 @@ static int patch_vt2002P(struct hda_codec *codec) spec->aa_mix_nid = 0x21; override_mic_boost(codec, 0x2b, 0, 3, 40); override_mic_boost(codec, 0x29, 0, 3, 40); + if (spec->codec_type == VT1802) + fix_vt1802_connections(codec); add_secret_dac_path(codec); snd_hda_pick_fixup(codec, NULL, vt2002p_fixups, via_fixups); diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 58b4b6f85db3..26feeb81a709 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2866,7 +2866,12 @@ static int snd_vt1724_resume(struct device *dev) ice->set_spdif_clock(ice, 0); } else { /* internal on-card clock */ - snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 1); + int rate; + if (ice->cur_rate) + rate = ice->cur_rate; + else + rate = ice->pro_rate_default; + snd_vt1724_set_pro_rate(ice, rate, 1); } update_spdif_bits(ice, ice->pm_saved_spdif_ctrl); diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index f1cd1e387801..748e36c66603 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -3979,7 +3979,8 @@ static int snd_hdspm_get_sync_check(struct snd_kcontrol *kcontrol, case 8: /* SYNC IN */ val = hdspm_sync_in_sync_check(hdspm); break; default: - val = hdspm_s1_sync_check(hdspm, ucontrol->id.index-1); + val = hdspm_s1_sync_check(hdspm, + kcontrol->private_value-1); } break; @@ -4899,7 +4900,7 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry, insel = "Coaxial"; break; default: - insel = "Unkown"; + insel = "Unknown"; } snd_iprintf(buffer, diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 61599298fb26..4d8db3685e96 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -763,7 +763,7 @@ static int cs42l52_set_sysclk(struct snd_soc_dai *codec_dai, if ((freq >= CS42L52_MIN_CLK) && (freq <= CS42L52_MAX_CLK)) { cs42l52->sysclk = freq; } else { - dev_err(codec->dev, "Invalid freq paramter\n"); + dev_err(codec->dev, "Invalid freq parameter\n"); return -EINVAL; } return 0; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 3fddc7ad1127..b2b2b37131bd 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3722,7 +3722,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) } while (count--); if (count == 0) - dev_warn(codec->dev, "No impedence range reported for jack\n"); + dev_warn(codec->dev, "No impedance range reported for jack\n"); #ifndef CONFIG_SND_SOC_WM8994_MODULE trace_snd_soc_jack_irq(dev_name(codec->dev)); diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 7f78c6d782b0..34de6f2faf61 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -35,6 +35,7 @@ #define EP_FLAG_ACTIVATED 0 #define EP_FLAG_RUNNING 1 +#define EP_FLAG_STOPPING 2 /* * snd_usb_endpoint is a model that abstracts everything related to an @@ -502,10 +503,20 @@ static int wait_clear_urbs(struct snd_usb_endpoint *ep) if (alive) snd_printk(KERN_ERR "timeout: still %d active urbs on EP #%x\n", alive, ep->ep_num); + clear_bit(EP_FLAG_STOPPING, &ep->flags); return 0; } +/* sync the pending stop operation; + * this function itself doesn't trigger the stop operation + */ +void snd_usb_endpoint_sync_pending_stop(struct snd_usb_endpoint *ep) +{ + if (ep && test_bit(EP_FLAG_STOPPING, &ep->flags)) + wait_clear_urbs(ep); +} + /* * unlink active urbs. */ @@ -918,6 +929,8 @@ void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, if (wait) wait_clear_urbs(ep); + else + set_bit(EP_FLAG_STOPPING, &ep->flags); } } diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 6376ccf10fd4..3d4c9705041f 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -19,6 +19,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, int can_sleep); void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, int force, int can_sleep, int wait); +void snd_usb_endpoint_sync_pending_stop(struct snd_usb_endpoint *ep); int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep); int snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep); void snd_usb_endpoint_free(struct list_head *head); diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 8ca6edf29fee..e7329d0449f2 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -565,6 +565,9 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) goto unlock; } + snd_usb_endpoint_sync_pending_stop(subs->sync_endpoint); + snd_usb_endpoint_sync_pending_stop(subs->data_endpoint); + ret = set_format(subs, subs->cur_audiofmt); if (ret < 0) goto unlock;