diff options
Diffstat (limited to 'sound')
30 files changed, 211 insertions, 178 deletions
diff --git a/sound/Kconfig b/sound/Kconfig index 1140e9988..76febc378 100644 --- a/sound/Kconfig +++ b/sound/Kconfig @@ -1,6 +1,6 @@ menuconfig SOUND tristate "Sound card support" - depends on HAS_IOMEM + depends on HAS_IOMEM || UML help If you have a sound card in your computer, i.e. if it can say more than an occasional beep, say Y. diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index 00d826b04..eb6735f16 100644 --- a/sound/core/control_compat.c +++ b/sound/core/control_compat.c @@ -236,7 +236,7 @@ static int copy_ctl_value_from_user(struct snd_card *card, { struct snd_ctl_elem_value32 __user *data32 = userdata; int i, type, size; - int uninitialized_var(count); + int count; unsigned int indirect; if (copy_from_user(&data->id, &data32->id, sizeof(data->id))) diff --git a/sound/core/info.c b/sound/core/info.c index 2ac656db0..b2c459ca5 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -72,7 +72,7 @@ struct snd_info_private_data { }; static int snd_info_version_init(void); -static void snd_info_disconnect(struct snd_info_entry *entry); +static void snd_info_clear_entries(struct snd_info_entry *entry); /* @@ -598,11 +598,16 @@ void snd_info_card_disconnect(struct snd_card *card) { if (!card) return; - mutex_lock(&info_mutex); + proc_remove(card->proc_root_link); - card->proc_root_link = NULL; if (card->proc_root) - snd_info_disconnect(card->proc_root); + proc_remove(card->proc_root->p); + + mutex_lock(&info_mutex); + if (card->proc_root) + snd_info_clear_entries(card->proc_root); + card->proc_root_link = NULL; + card->proc_root = NULL; mutex_unlock(&info_mutex); } @@ -776,15 +781,14 @@ struct snd_info_entry *snd_info_create_card_entry(struct snd_card *card, } EXPORT_SYMBOL(snd_info_create_card_entry); -static void snd_info_disconnect(struct snd_info_entry *entry) +static void snd_info_clear_entries(struct snd_info_entry *entry) { struct snd_info_entry *p; if (!entry->p) return; list_for_each_entry(p, &entry->children, list) - snd_info_disconnect(p); - proc_remove(entry->p); + snd_info_clear_entries(p); entry->p = NULL; } @@ -801,8 +805,9 @@ void snd_info_free_entry(struct snd_info_entry * entry) if (!entry) return; if (entry->p) { + proc_remove(entry->p); mutex_lock(&info_mutex); - snd_info_disconnect(entry); + snd_info_clear_entries(entry); mutex_unlock(&info_mutex); } diff --git a/sound/core/jack.c b/sound/core/jack.c index 074b15fcb..06e0fc7b6 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -378,6 +378,7 @@ void snd_jack_report(struct snd_jack *jack, int status) { struct snd_jack_kctl *jack_kctl; #ifdef CONFIG_SND_JACK_INPUT_DEV + struct input_dev *idev; int i; #endif @@ -389,30 +390,28 @@ void snd_jack_report(struct snd_jack *jack, int status) status & jack_kctl->mask_bits); #ifdef CONFIG_SND_JACK_INPUT_DEV - mutex_lock(&jack->input_dev_lock); - if (!jack->input_dev) { - mutex_unlock(&jack->input_dev_lock); + idev = input_get_device(jack->input_dev); + if (!idev) return; - } for (i = 0; i < ARRAY_SIZE(jack->key); i++) { int testbit = SND_JACK_BTN_0 >> i; if (jack->type & testbit) - input_report_key(jack->input_dev, jack->key[i], + input_report_key(idev, jack->key[i], status & testbit); } for (i = 0; i < ARRAY_SIZE(jack_switch_types); i++) { int testbit = 1 << i; if (jack->type & testbit) - input_report_switch(jack->input_dev, + input_report_switch(idev, jack_switch_types[i], status & testbit); } - input_sync(jack->input_dev); - mutex_unlock(&jack->input_dev_lock); + input_sync(idev); + input_put_device(idev); #endif /* CONFIG_SND_JACK_INPUT_DEV */ } EXPORT_SYMBOL(snd_jack_report); diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 8eed6244b..601f60bb2 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -266,6 +266,7 @@ static char *snd_pcm_state_names[] = { STATE(DRAINING), STATE(PAUSED), STATE(SUSPENDED), + STATE(DISCONNECTED), }; static char *snd_pcm_access_names[] = { diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index 946ab080a..7c5799fec 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -329,10 +329,14 @@ static int snd_pcm_ioctl_hw_params_compat(struct snd_pcm_substream *substream, goto error; } - if (refine) + if (refine) { err = snd_pcm_hw_refine(substream, data); - else + if (err < 0) + goto error; + err = fixup_unreferenced_params(substream, data); + } else { err = snd_pcm_hw_params(substream, data); + } if (err < 0) goto error; if (copy_to_user(data32, data, sizeof(*data32)) || diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index 838c3c8b4..2ddfd6fed 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -50,6 +50,7 @@ struct seq_oss_midi { struct snd_midi_event *coder; /* MIDI event coder */ struct seq_oss_devinfo *devinfo; /* assigned OSSseq device */ snd_use_lock_t use_lock; + struct mutex open_mutex; }; @@ -184,6 +185,7 @@ snd_seq_oss_midi_check_new_port(struct snd_seq_port_info *pinfo) mdev->flags = pinfo->capability; mdev->opened = 0; snd_use_lock_init(&mdev->use_lock); + mutex_init(&mdev->open_mutex); /* copy and truncate the name of synth device */ strlcpy(mdev->name, pinfo->name, sizeof(mdev->name)); @@ -332,14 +334,16 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) int perm; struct seq_oss_midi *mdev; struct snd_seq_port_subscribe subs; + int err; if ((mdev = get_mididev(dp, dev)) == NULL) return -ENODEV; + mutex_lock(&mdev->open_mutex); /* already used? */ if (mdev->opened && mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return -EBUSY; + err = -EBUSY; + goto unlock; } perm = 0; @@ -349,14 +353,14 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) perm |= PERM_READ; perm &= mdev->flags; if (perm == 0) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } /* already opened? */ if ((mdev->opened & perm) == perm) { - snd_use_lock_free(&mdev->use_lock); - return 0; + err = 0; + goto unlock; } perm &= ~mdev->opened; @@ -381,13 +385,17 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) } if (! mdev->opened) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } mdev->devinfo = dp; + err = 0; + + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); - return 0; + return err; } /* @@ -401,10 +409,9 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) if ((mdev = get_mididev(dp, dev)) == NULL) return -ENODEV; - if (! mdev->opened || mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return 0; - } + mutex_lock(&mdev->open_mutex); + if (!mdev->opened || mdev->devinfo != dp) + goto unlock; memset(&subs, 0, sizeof(subs)); if (mdev->opened & PERM_WRITE) { @@ -423,6 +430,8 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) mdev->opened = 0; mdev->devinfo = NULL; + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); return 0; } diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index eee422390..2569f82b6 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -241,8 +241,10 @@ struct hdac_stream *snd_hdac_stream_assign(struct hdac_bus *bus, struct hdac_stream *res = NULL; /* make a non-zero unique key for the substream */ - int key = (substream->pcm->device << 16) | (substream->number << 2) | - (substream->stream + 1); + int key = (substream->number << 2) | (substream->stream + 1); + + if (substream->pcm) + key |= (substream->pcm->device << 16); list_for_each_entry(azx_dev, &bus->stream_list, list) { if (azx_dev->direction != substream->stream) diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c index c16c81511..970aef2cf 100644 --- a/sound/isa/sb/sb16_csp.c +++ b/sound/isa/sb/sb16_csp.c @@ -116,7 +116,7 @@ static void info_read(struct snd_info_entry *entry, struct snd_info_buffer *buff int snd_sb_csp_new(struct snd_sb *chip, int device, struct snd_hwdep ** rhwdep) { struct snd_sb_csp *p; - int uninitialized_var(version); + int version; int err; struct snd_hwdep *hw; diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index a276c4283..64a1bd420 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -2026,10 +2026,9 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, .dev_disconnect = snd_ac97_dev_disconnect, }; - if (rac97) - *rac97 = NULL; - if (snd_BUG_ON(!bus || !template)) + if (snd_BUG_ON(!bus || !template || !rac97)) return -EINVAL; + *rac97 = NULL; if (snd_BUG_ON(template->num >= 4)) return -EINVAL; if (bus->codec[template->num]) diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 1f25e6d02..84d98c098 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -1550,14 +1550,8 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) gpr += 2; /* Master volume (will be renamed later) */ - A_OP(icode, &ptr, iMAC0, A_GPR(playback+0+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+0+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+1+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+1+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+2+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+2+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+3+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+3+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+4+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+4+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+5+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+5+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+6+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+6+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+7+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+7+SND_EMU10K1_PLAYBACK_CHANNELS)); + for (z = 0; z < 8; z++) + A_OP(icode, &ptr, iMAC0, A_GPR(playback+z+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+z+SND_EMU10K1_PLAYBACK_CHANNELS)); snd_emu10k1_init_mono_control(&controls[nctl++], "Wave Master Playback Volume", gpr, 0); gpr += 2; @@ -1641,102 +1635,14 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) dev_dbg(emu->card->dev, "emufx.c: gpr=0x%x, tmp=0x%x\n", gpr, tmp); */ - /* For the EMU1010: How to get 32bit values from the DSP. High 16bits into L, low 16bits into R. */ - /* A_P16VIN(0) is delayed by one sample, - * so all other A_P16VIN channels will need to also be delayed - */ - /* Left ADC in. 1 of 2 */ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_P16VIN(0x0), A_FXBUS2(0) ); - /* Right ADC in 1 of 2 */ - gpr_map[gpr++] = 0x00000000; - /* Delaying by one sample: instead of copying the input - * value A_P16VIN to output A_FXBUS2 as in the first channel, - * we use an auxiliary register, delaying the value by one - * sample - */ - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(2) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x1), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(4) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x2), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(6) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x3), A_C_00000000, A_C_00000000); - /* For 96kHz mode */ - /* Left ADC in. 2 of 2 */ - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0x8) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x4), A_C_00000000, A_C_00000000); - /* Right ADC in 2 of 2 */ - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xa) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x5), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xc) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x6), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xe) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x7), A_C_00000000, A_C_00000000); - /* Pavel Hofman - we still have voices, A_FXBUS2s, and - * A_P16VINs available - - * let's add 8 more capture channels - total of 16 - */ - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x10)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x8), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x12)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x9), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x14)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xa), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x16)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xb), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x18)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xc), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x1a)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xd), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x1c)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xe), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x1e)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xf), - A_C_00000000, A_C_00000000); + /* A_P16VIN(0) is delayed by one sample, so all other A_P16VIN channels + * will need to also be delayed; we use an auxiliary register for that. */ + for (z = 1; z < 0x10; z++) { + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr), A_FXBUS2(z * 2) ); + A_OP(icode, &ptr, iACC3, A_GPR(gpr), A_P16VIN(z), A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + } } #if 0 diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 40d596248..e66d8729c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2364,12 +2364,15 @@ static struct snd_pci_quirk power_save_blacklist[] = { SND_PCI_QUIRK(0x8086, 0x2068, "Intel NUC7i3BNB", 0), /* https://bugzilla.kernel.org/show_bug.cgi?id=198611 */ SND_PCI_QUIRK(0x17aa, 0x2227, "Lenovo X1 Carbon 3rd Gen", 0), + SND_PCI_QUIRK(0x17aa, 0x316e, "Lenovo ThinkCentre M70q", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1689623 */ SND_PCI_QUIRK(0x17aa, 0x367b, "Lenovo IdeaCentre B550", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1572975 */ SND_PCI_QUIRK(0x17aa, 0x36a7, "Lenovo C50 All in one", 0), /* https://bugs.launchpad.net/bugs/1821663 */ SND_PCI_QUIRK(0x1631, 0xe017, "Packard Bell NEC IMEDIA 5204", 0), + /* KONTRON SinglePC may cause a stall at runtime resume */ + SND_PCI_QUIRK(0x1734, 0x1232, "KONTRON SinglePC", 0), {} }; #endif /* CONFIG_PM */ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e5d858877..2b345ba08 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1917,6 +1917,7 @@ enum { ALC887_FIXUP_ASUS_AUDIO, ALC887_FIXUP_ASUS_HMIC, ALCS1200A_FIXUP_MIC_VREF, + ALC888VD_FIXUP_MIC_100VREF, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -2470,6 +2471,13 @@ static const struct hda_fixup alc882_fixups[] = { {} } }, + [ALC888VD_FIXUP_MIC_100VREF] = { + .type = HDA_FIXUP_PINCTLS, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, PIN_VREF100 }, /* headset mic */ + {} + } + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -2539,6 +2547,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_MBA11_VREF), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD), + SND_PCI_QUIRK(0x10ec, 0x12d8, "iBase Elo Touch", ALC888VD_FIXUP_MIC_100VREF), SND_PCI_QUIRK(0x13fe, 0x1009, "Advantech MIT-W101", ALC886_FIXUP_EAPD), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE), SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), @@ -7168,6 +7177,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x10a1, "ASUS UX391UA", ALC294_FIXUP_ASUS_SPK), SND_PCI_QUIRK(0x1043, 0x10c0, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x10d0, "ASUS X540LA/X540LJ", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1043, 0x10d3, "ASUS K6500ZC", ALC294_FIXUP_ASUS_SPK), SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x11c0, "ASUS X556UR", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1271, "ASUS X430UN", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), @@ -8596,6 +8606,17 @@ static void alc897_fixup_lenovo_headset_mic(struct hda_codec *codec, } } +static void alc897_fixup_lenovo_headset_mode(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; + spec->gen.hp_automute_hook = alc897_hp_automute_hook; + } +} + static const struct coef_fw alc668_coefs[] = { WRITE_COEF(0x01, 0xbebe), WRITE_COEF(0x02, 0xaaaa), WRITE_COEF(0x03, 0x0), WRITE_COEF(0x04, 0x0180), WRITE_COEF(0x06, 0x0), WRITE_COEF(0x07, 0x0f80), @@ -8678,6 +8699,8 @@ enum { ALC897_FIXUP_LENOVO_HEADSET_MIC, ALC897_FIXUP_HEADSET_MIC_PIN, ALC897_FIXUP_HP_HSMIC_VERB, + ALC897_FIXUP_LENOVO_HEADSET_MODE, + ALC897_FIXUP_HEADSET_MIC_PIN2, }; static const struct hda_fixup alc662_fixups[] = { @@ -9085,6 +9108,19 @@ static const struct hda_fixup alc662_fixups[] = { { } }, }, + [ALC897_FIXUP_LENOVO_HEADSET_MODE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc897_fixup_lenovo_headset_mode, + }, + [ALC897_FIXUP_HEADSET_MIC_PIN2] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x01a11140 }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC897_FIXUP_LENOVO_HEADSET_MODE + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -9134,6 +9170,10 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x32cb, "Lenovo ThinkCentre M70", ALC897_FIXUP_HEADSET_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x32cf, "Lenovo ThinkCentre M950", ALC897_FIXUP_HEADSET_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x32f7, "Lenovo ThinkCentre M90", ALC897_FIXUP_HEADSET_MIC_PIN), + SND_PCI_QUIRK(0x17aa, 0x3321, "Lenovo ThinkCentre M70 Gen4", ALC897_FIXUP_HEADSET_MIC_PIN), + SND_PCI_QUIRK(0x17aa, 0x331b, "Lenovo ThinkCentre M90 Gen4", ALC897_FIXUP_HEADSET_MIC_PIN), + SND_PCI_QUIRK(0x17aa, 0x3364, "Lenovo ThinkCentre M90 Gen5", ALC897_FIXUP_HEADSET_MIC_PIN), + SND_PCI_QUIRK(0x17aa, 0x3742, "Lenovo TianYi510Pro-14IOB", ALC897_FIXUP_HEADSET_MIC_PIN2), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x1849, 0x5892, "ASRock B150M", ALC892_FIXUP_ASROCK_MOBO), diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index 99cc73150..ab7f76117 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -174,11 +174,14 @@ struct atmel_i2s_gck_param { #define I2S_MCK_12M288 12288000UL #define I2S_MCK_11M2896 11289600UL +#define I2S_MCK_6M144 6144000UL /* mck = (32 * (imckfs+1) / (imckdiv+1)) * fs */ static const struct atmel_i2s_gck_param gck_params[] = { + /* mck = 6.144Mhz */ + { 8000, I2S_MCK_6M144, 1, 47}, /* mck = 768 fs */ + /* mck = 12.288MHz */ - { 8000, I2S_MCK_12M288, 0, 47}, /* mck = 1536 fs */ { 16000, I2S_MCK_12M288, 1, 47}, /* mck = 768 fs */ { 24000, I2S_MCK_12M288, 3, 63}, /* mck = 512 fs */ { 32000, I2S_MCK_12M288, 3, 47}, /* mck = 384 fs */ diff --git a/sound/soc/codecs/cs42l51-i2c.c b/sound/soc/codecs/cs42l51-i2c.c index 4b5731a41..cd93e93a5 100644 --- a/sound/soc/codecs/cs42l51-i2c.c +++ b/sound/soc/codecs/cs42l51-i2c.c @@ -23,6 +23,12 @@ static struct i2c_device_id cs42l51_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, cs42l51_i2c_id); +const struct of_device_id cs42l51_of_match[] = { + { .compatible = "cirrus,cs42l51", }, + { } +}; +MODULE_DEVICE_TABLE(of, cs42l51_of_match); + static int cs42l51_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 5080d7a3c..662f1f85b 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -563,13 +563,6 @@ error: } EXPORT_SYMBOL_GPL(cs42l51_probe); -const struct of_device_id cs42l51_of_match[] = { - { .compatible = "cirrus,cs42l51", }, - { } -}; -MODULE_DEVICE_TABLE(of, cs42l51_of_match); -EXPORT_SYMBOL_GPL(cs42l51_of_match); - MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>"); MODULE_DESCRIPTION("Cirrus Logic CS42L51 ALSA SoC Codec Driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42l51.h b/sound/soc/codecs/cs42l51.h index 0ca805492..8c55bf384 100644 --- a/sound/soc/codecs/cs42l51.h +++ b/sound/soc/codecs/cs42l51.h @@ -22,7 +22,6 @@ struct device; extern const struct regmap_config cs42l51_regmap; int cs42l51_probe(struct device *dev, struct regmap *regmap); -extern const struct of_device_id cs42l51_of_match[]; #define CS42L51_CHIP_ID 0x1B #define CS42L51_CHIP_REV_A 0x00 diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 2c7d5088e..7e18e007a 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -351,11 +351,15 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data) struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); u8 events[DA7219_AAD_IRQ_REG_MAX]; u8 statusa; - int i, report = 0, mask = 0; + int i, ret, report = 0, mask = 0; /* Read current IRQ events */ - regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, - events, DA7219_AAD_IRQ_REG_MAX); + ret = regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, + events, DA7219_AAD_IRQ_REG_MAX); + if (ret) { + dev_warn_ratelimited(component->dev, "Failed to read IRQ events: %d\n", ret); + return IRQ_NONE; + } if (!events[DA7219_AAD_IRQ_REG_A] && !events[DA7219_AAD_IRQ_REG_B]) return IRQ_NONE; @@ -859,6 +863,8 @@ void da7219_aad_suspend(struct snd_soc_component *component) } } } + + synchronize_irq(da7219_aad->irq); } void da7219_aad_resume(struct snd_soc_component *component) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 57130edaf..0fc4755fd 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -45,7 +45,12 @@ static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(dac_vol_tlv, -9600, 50, 1); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_vol_tlv, -9600, 50, 1); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_max_gain_tlv, -650, 150, 0); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_min_gain_tlv, -1200, 150, 0); -static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_target_tlv, -1650, 150, 0); + +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(alc_target_tlv, + 0, 10, TLV_DB_SCALE_ITEM(-1650, 150, 0), + 11, 11, TLV_DB_SCALE_ITEM(-150, 0, 0), +); + static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpmixer_gain_tlv, 0, 4, TLV_DB_SCALE_ITEM(-1200, 150, 0), 8, 11, TLV_DB_SCALE_ITEM(-450, 150, 0), @@ -107,7 +112,7 @@ static const struct snd_kcontrol_new es8316_snd_controls[] = { alc_max_gain_tlv), SOC_SINGLE_TLV("ALC Capture Min Volume", ES8316_ADC_ALC2, 0, 28, 0, alc_min_gain_tlv), - SOC_SINGLE_TLV("ALC Capture Target Volume", ES8316_ADC_ALC3, 4, 10, 0, + SOC_SINGLE_TLV("ALC Capture Target Volume", ES8316_ADC_ALC3, 4, 11, 0, alc_target_tlv), SOC_SINGLE("ALC Capture Hold Time", ES8316_ADC_ALC3, 0, 10, 0), SOC_SINGLE("ALC Capture Decay Time", ES8316_ADC_ALC4, 4, 10, 0), @@ -140,7 +145,7 @@ static const char * const es8316_dmic_txt[] = { "dmic data at high level", "dmic data at low level", }; -static const unsigned int es8316_dmic_values[] = { 0, 1, 2 }; +static const unsigned int es8316_dmic_values[] = { 0, 2, 3 }; static const struct soc_enum es8316_dmic_src_enum = SOC_VALUE_ENUM_SINGLE(ES8316_ADC_DMIC, 0, 3, ARRAY_SIZE(es8316_dmic_txt), diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index d34000182..a713e9649 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3278,6 +3278,8 @@ int rt5645_set_jack_detect(struct snd_soc_component *component, RT5645_GP1_PIN_IRQ, RT5645_GP1_PIN_IRQ); regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL1, RT5645_DIG_GATE_CTRL, RT5645_DIG_GATE_CTRL); + regmap_update_bits(rt5645->regmap, RT5645_DEPOP_M1, + RT5645_HP_CB_MASK, RT5645_HP_CB_PU); } rt5645_irq(0, rt5645); diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 6ba99f5ed..a7ed2a19c 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4475,6 +4475,8 @@ static void rt5665_remove(struct snd_soc_component *component) struct rt5665_priv *rt5665 = snd_soc_component_get_drvdata(component); regmap_write(rt5665->regmap, RT5665_RESET, 0); + + regulator_bulk_disable(ARRAY_SIZE(rt5665->supplies), rt5665->supplies); } #ifdef CONFIG_PM diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index d14e851b9..03d3b0f17 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2264,6 +2264,9 @@ static int wm8904_i2c_probe(struct i2c_client *i2c, regmap_update_bits(wm8904->regmap, WM8904_BIAS_CONTROL_0, WM8904_POBCTRL, 0); + /* Fill the cache for the ADC test register */ + regmap_read(wm8904->regmap, WM8904_ADC_TEST_0, &val); + /* Can leave the device powered off until we need it */ regcache_cache_only(wm8904->regmap, true); regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 740b90df4..0a1ba64ed 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -614,6 +614,8 @@ static int fsl_spdif_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_PAUSE_PUSH: regmap_update_bits(regmap, REG_SPDIF_SCR, dmaen, 0); regmap_update_bits(regmap, REG_SPDIF_SIE, intr, 0); + regmap_write(regmap, REG_SPDIF_STL, 0x0); + regmap_write(regmap, REG_SPDIF_STR, 0x0); break; default: return -EINVAL; diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 64bf3560c..7567ee380 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -404,10 +404,12 @@ static int asoc_simple_card_probe(struct platform_device *pdev) } else { struct asoc_simple_card_info *cinfo; + ret = -EINVAL; + cinfo = dev->platform_data; if (!cinfo) { dev_err(dev, "no info for asoc-simple-card\n"); - return -EINVAL; + goto err; } if (!cinfo->name || @@ -416,7 +418,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) !cinfo->platform || !cinfo->cpu_dai.name) { dev_err(dev, "insufficient asoc_simple_card_info settings\n"); - return -EINVAL; + goto err; } card->name = (cinfo->card) ? cinfo->card : cinfo->name; diff --git a/sound/soc/intel/skylake/skl-sst-utils.c b/sound/soc/intel/skylake/skl-sst-utils.c index 2ae405617..9e1e9bac1 100644 --- a/sound/soc/intel/skylake/skl-sst-utils.c +++ b/sound/soc/intel/skylake/skl-sst-utils.c @@ -317,6 +317,7 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, module->instance_id = devm_kzalloc(ctx->dev, size, GFP_KERNEL); if (!module->instance_id) { ret = -ENOMEM; + kfree(module); goto free_uuid_list; } diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c index 43e390f93..a195160b6 100644 --- a/sound/soc/meson/axg-tdm-formatter.c +++ b/sound/soc/meson/axg-tdm-formatter.c @@ -28,27 +28,32 @@ int axg_tdm_formatter_set_channel_masks(struct regmap *map, struct axg_tdm_stream *ts, unsigned int offset) { - unsigned int val, ch = ts->channels; - unsigned long mask; - int i, j; + unsigned int ch = ts->channels; + u32 val[AXG_TDM_NUM_LANES]; + int i, j, k; + + /* + * We need to mimick the slot distribution used by the HW to keep the + * channel placement consistent regardless of the number of channel + * in the stream. This is why the odd algorithm below is used. + */ + memset(val, 0, sizeof(*val) * AXG_TDM_NUM_LANES); /* * Distribute the channels of the stream over the available slots - * of each TDM lane + * of each TDM lane. We need to go over the 32 slots ... */ - for (i = 0; i < AXG_TDM_NUM_LANES; i++) { - val = 0; - mask = ts->mask[i]; - - for (j = find_first_bit(&mask, 32); - (j < 32) && ch; - j = find_next_bit(&mask, 32, j + 1)) { - val |= 1 << j; - ch -= 1; + for (i = 0; (i < 32) && ch; i += 2) { + /* ... of all the lanes ... */ + for (j = 0; j < AXG_TDM_NUM_LANES; j++) { + /* ... then distribute the channels in pairs */ + for (k = 0; k < 2; k++) { + if ((BIT(i + k) & ts->mask[j]) && ch) { + val[j] |= BIT(i + k); + ch -= 1; + } + } } - - regmap_write(map, offset, val); - offset += regmap_get_reg_stride(map); } /* @@ -61,6 +66,11 @@ int axg_tdm_formatter_set_channel_masks(struct regmap *map, return -EINVAL; } + for (i = 0; i < AXG_TDM_NUM_LANES; i++) { + regmap_write(map, offset, val[i]); + offset += regmap_get_reg_stride(map); + } + return 0; } EXPORT_SYMBOL_GPL(axg_tdm_formatter_set_channel_masks); diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 4dce494df..ef9fda16c 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -300,7 +300,7 @@ static int cx81801_open(struct tty_struct *tty) static void cx81801_close(struct tty_struct *tty) { struct snd_soc_component *component = tty->disc_data; - struct snd_soc_dapm_context *dapm = &component->card->dapm; + struct snd_soc_dapm_context *dapm; del_timer_sync(&cx81801_timer); @@ -312,6 +312,8 @@ static void cx81801_close(struct tty_struct *tty) v253_ops.close(tty); + dapm = &component->card->dapm; + /* Revert back to default audio input/output constellation */ snd_soc_dapm_mutex_lock(dapm); diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 69033e1a8..49481dadb 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -795,7 +795,7 @@ static int pxa_ssp_probe(struct snd_soc_dai *dai) if (IS_ERR(priv->extclk)) { ret = PTR_ERR(priv->extclk); if (ret == -EPROBE_DEFER) - return ret; + goto err_priv; priv->extclk = NULL; } diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index e428d8b36..56119a96d 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -324,7 +324,7 @@ static void queue_pending_output_urbs(struct snd_usb_endpoint *ep) while (test_bit(EP_FLAG_RUNNING, &ep->flags)) { unsigned long flags; - struct snd_usb_packet_info *uninitialized_var(packet); + struct snd_usb_packet_info *packet; struct snd_urb_ctx *ctx = NULL; int err, i; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index e72f744bc..6c546f520 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3677,5 +3677,34 @@ ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */ } } }, +{ + /* Advanced modes of the Mythware XA001AU. + * For the standard mode, Mythware XA001AU has ID ffad:a001 + */ + USB_DEVICE_VENDOR_SPEC(0xffad, 0xa001), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Mythware", + .product_name = "XA001AU", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE, + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE, + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE, + }, + { + .ifnum = -1 + } + } + } +}, #undef USB_DEVICE_VENDOR_SPEC |