diff options
Diffstat (limited to 'sound')
76 files changed, 795 insertions, 330 deletions
diff --git a/sound/Kconfig b/sound/Kconfig index 1140e9988fc5..76febc37862d 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 3fc216644e0e..eb6735f16b93 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))) @@ -319,7 +319,9 @@ static int ctl_elem_read_user(struct snd_card *card, err = snd_power_wait(card, SNDRV_CTL_POWER_D0); if (err < 0) goto error; + down_read(&card->controls_rwsem); err = snd_ctl_elem_read(card, data); + up_read(&card->controls_rwsem); if (err < 0) goto error; err = copy_ctl_value_to_user(userdata, valuep, data, type, count); @@ -347,7 +349,9 @@ static int ctl_elem_write_user(struct snd_ctl_file *file, err = snd_power_wait(card, SNDRV_CTL_POWER_D0); if (err < 0) goto error; + down_write(&card->controls_rwsem); err = snd_ctl_elem_write(card, file, data); + up_write(&card->controls_rwsem); if (err < 0) goto error; err = copy_ctl_value_to_user(userdata, valuep, data, type, count); diff --git a/sound/core/info.c b/sound/core/info.c index 2ac656db0b1c..b2c459ca56d0 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 074b15fcb0ac..06e0fc7b6417 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/oss/pcm_plugin.h b/sound/core/oss/pcm_plugin.h index c9cd29d86efd..64a2057aa061 100644 --- a/sound/core/oss/pcm_plugin.h +++ b/sound/core/oss/pcm_plugin.h @@ -156,6 +156,14 @@ int snd_pcm_area_copy(const struct snd_pcm_channel_area *src_channel, void *snd_pcm_plug_buf_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t size); void snd_pcm_plug_buf_unlock(struct snd_pcm_substream *plug, void *ptr); +#else + +static inline snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t drv_size) { return drv_size; } +static inline snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t clt_size) { return clt_size; } +static inline int snd_pcm_plug_slave_format(int format, const struct snd_mask *format_mask) { return format; } + +#endif + snd_pcm_sframes_t snd_pcm_oss_write3(struct snd_pcm_substream *substream, const char *ptr, snd_pcm_uframes_t size, int in_kernel); @@ -166,14 +174,6 @@ snd_pcm_sframes_t snd_pcm_oss_writev3(struct snd_pcm_substream *substream, snd_pcm_sframes_t snd_pcm_oss_readv3(struct snd_pcm_substream *substream, void **bufs, snd_pcm_uframes_t frames); -#else - -static inline snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t drv_size) { return drv_size; } -static inline snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t clt_size) { return clt_size; } -static inline int snd_pcm_plug_slave_format(int format, const struct snd_mask *format_mask) { return format; } - -#endif - #ifdef PLUGIN_DEBUG #define pdprintf(fmt, args...) printk(KERN_DEBUG "plugin: " fmt, ##args) #else diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 8eed6244b832..601f60bb2e8a 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 946ab080ac00..7c5799fecfa1 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 838c3c8b403c..2ddfd6fed122 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/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c index 5b0388202bac..ac854beb8347 100644 --- a/sound/core/seq/seq_memory.c +++ b/sound/core/seq/seq_memory.c @@ -126,15 +126,19 @@ EXPORT_SYMBOL(snd_seq_dump_var_event); * expand the variable length event to linear buffer space. */ -static int seq_copy_in_kernel(char **bufptr, const void *src, int size) +static int seq_copy_in_kernel(void *ptr, void *src, int size) { + char **bufptr = ptr; + memcpy(*bufptr, src, size); *bufptr += size; return 0; } -static int seq_copy_in_user(char __user **bufptr, const void *src, int size) +static int seq_copy_in_user(void *ptr, void *src, int size) { + char __user **bufptr = ptr; + if (copy_to_user(*bufptr, src, size)) return -EFAULT; *bufptr += size; @@ -163,8 +167,7 @@ int snd_seq_expand_var_event(const struct snd_seq_event *event, int count, char return newlen; } err = snd_seq_dump_var_event(event, - in_kernel ? (snd_seq_dump_func_t)seq_copy_in_kernel : - (snd_seq_dump_func_t)seq_copy_in_user, + in_kernel ? seq_copy_in_kernel : seq_copy_in_user, &buf); return err < 0 ? err : newlen; } diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c index b68e71ca7abd..7dceb1e1c3b4 100644 --- a/sound/drivers/mts64.c +++ b/sound/drivers/mts64.c @@ -830,6 +830,9 @@ static void snd_mts64_interrupt(void *private) u8 status, data; struct snd_rawmidi_substream *substream; + if (!mts) + return; + spin_lock(&mts->lock); ret = mts64_read(mts->pardev->port); data = ret & 0x00ff; diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index eee422390d8e..2569f82b6fa0 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/i2c/cs8427.c b/sound/i2c/cs8427.c index 8afa2f888466..ef40501cf898 100644 --- a/sound/i2c/cs8427.c +++ b/sound/i2c/cs8427.c @@ -568,10 +568,13 @@ int snd_cs8427_iec958_active(struct snd_i2c_device *cs8427, int active) if (snd_BUG_ON(!cs8427)) return -ENXIO; chip = cs8427->private_data; - if (active) + if (active) { memcpy(chip->playback.pcm_status, chip->playback.def_status, 24); - chip->playback.pcm_ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; + chip->playback.pcm_ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; + } else { + chip->playback.pcm_ctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; + } snd_ctl_notify(cs8427->bus->card, SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO, &chip->playback.pcm_ctl->id); diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c index c16c8151160c..970aef2cf513 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 a276c4283c7b..64a1bd420637 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/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c index 2864698436a5..6a49f897c4d9 100644 --- a/sound/pci/asihpi/hpi6205.c +++ b/sound/pci/asihpi/hpi6205.c @@ -441,7 +441,7 @@ void HPI_6205(struct hpi_message *phm, struct hpi_response *phr) pao = hpi_find_adapter(phm->adapter_index); } else { /* subsys messages don't address an adapter */ - _HPI_6205(NULL, phm, phr); + phr->error = HPI_ERROR_INVALID_OBJ_INDEX; return; } diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index 3f06986fbecf..d8c244a5dce0 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -359,7 +359,7 @@ int asihpi_adapter_probe(struct pci_dev *pci_dev, pci_dev->device, pci_dev->subsystem_vendor, pci_dev->subsystem_device, pci_dev->devfn); - if (pci_enable_device(pci_dev) < 0) { + if (pcim_enable_device(pci_dev) < 0) { dev_err(&pci_dev->dev, "pci_enable_device failed, disabling device\n"); return -EIO; diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 1f25e6d029d8..84d98c098b74 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/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 623776b13f8d..54f09fbd786f 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -1258,7 +1258,7 @@ static int snd_emu10k1_capture_mic_close(struct snd_pcm_substream *substream) { struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream); - emu->capture_interrupt = NULL; + emu->capture_mic_interrupt = NULL; emu->pcm_capture_mic_substream = NULL; return 0; } @@ -1366,7 +1366,7 @@ static int snd_emu10k1_capture_efx_close(struct snd_pcm_substream *substream) { struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream); - emu->capture_interrupt = NULL; + emu->capture_efx_interrupt = NULL; emu->pcm_capture_efx_substream = NULL; return 0; } diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index ff263ad19230..f4b07dc6f1cc 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -1159,8 +1159,8 @@ static bool path_has_mixer(struct hda_codec *codec, int path_idx, int ctl_type) return path && path->ctls[ctl_type]; } -static const char * const channel_name[4] = { - "Front", "Surround", "CLFE", "Side" +static const char * const channel_name[] = { + "Front", "Surround", "CLFE", "Side", "Back", }; /* give some appropriate ctl name prefix for the given line out channel */ @@ -1186,7 +1186,7 @@ static const char *get_line_out_pfx(struct hda_codec *codec, int ch, /* multi-io channels */ if (ch >= cfg->line_outs) - return channel_name[ch]; + goto fixed_name; switch (cfg->line_out_type) { case AUTO_PIN_SPEAKER_OUT: @@ -1238,6 +1238,7 @@ static const char *get_line_out_pfx(struct hda_codec *codec, int ch, if (cfg->line_outs == 1 && !spec->multi_ios) return "Line Out"; + fixed_name: if (ch >= ARRAY_SIZE(channel_name)) { snd_BUG(); return "PCM"; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 40d596248fab..e66d8729c72f 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_ca0132.c b/sound/pci/hda/patch_ca0132.c index 23f00ba993cb..ca3c9f161829 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1070,6 +1070,7 @@ static const struct snd_pci_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1458, 0xA026, "Gigabyte G1.Sniper Z97", QUIRK_R3DI), SND_PCI_QUIRK(0x1458, 0xA036, "Gigabyte GA-Z170X-Gaming 7", QUIRK_R3DI), SND_PCI_QUIRK(0x3842, 0x1038, "EVGA X99 Classified", QUIRK_R3DI), + SND_PCI_QUIRK(0x3842, 0x104b, "EVGA X299 Dark", QUIRK_R3DI), SND_PCI_QUIRK(0x3842, 0x1055, "EVGA Z390 DARK", QUIRK_R3DI), SND_PCI_QUIRK(0x1102, 0x0013, "Recon3D", QUIRK_R3D), {} @@ -1917,7 +1918,7 @@ static int dspio_set_uint_param_no_source(struct hda_codec *codec, int mod_id, static int dspio_alloc_dma_chan(struct hda_codec *codec, unsigned int *dma_chan) { int status = 0; - unsigned int size = sizeof(dma_chan); + unsigned int size = sizeof(*dma_chan); codec_dbg(codec, " dspio_alloc_dma_chan() -- begin\n"); status = dspio_scp(codec, MASTERCONTROL, 0x20, @@ -3620,8 +3621,10 @@ static int tuning_ctl_set(struct hda_codec *codec, hda_nid_t nid, for (i = 0; i < TUNING_CTLS_COUNT; i++) if (nid == ca0132_tuning_ctls[i].nid) - break; + goto found; + return -EINVAL; +found: snd_hda_power_up(codec); dspio_set_param(codec, ca0132_tuning_ctls[i].mid, 0x20, ca0132_tuning_ctls[i].req, diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index b1a8ee8cf17e..cfa958dc2dd5 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -952,7 +952,10 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x3905, "Lenovo G50-30", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x390b, "Lenovo G50-80", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), - SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_PINCFG_LENOVO_NOTEBOOK), + /* NOTE: we'd need to extend the quirk for 17aa:3977 as the same + * PCI SSID is used on multiple Lenovo models + */ + SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo G50-70", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", CXT_FIXUP_THINKPAD_ACPI), @@ -974,6 +977,7 @@ static const struct hda_model_fixup cxt5066_fixup_models[] = { { .id = CXT_FIXUP_HP_DOCK, .name = "hp-dock" }, { .id = CXT_FIXUP_MUTE_LED_GPIO, .name = "mute-led-gpio" }, { .id = CXT_FIXUP_HP_MIC_NO_PRESENCE, .name = "hp-mic-fix" }, + { .id = CXT_PINCFG_LENOVO_NOTEBOOK, .name = "lenovo-20149" }, {} }; @@ -1106,6 +1110,7 @@ static const struct hda_device_id snd_hda_id_conexant[] = { HDA_CODEC_ENTRY(0x14f11f86, "CX8070", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f12008, "CX8200", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f120d0, "CX11970", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f120d1, "SN6180", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15045, "CX20549 (Venice)", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15047, "CX20551 (Waikiki)", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15051, "CX20561 (Hermosa)", patch_conexant_auto), diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index cbd5118570fd..e3f0326d81c2 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1804,33 +1804,43 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) static int hdmi_parse_codec(struct hda_codec *codec) { - hda_nid_t nid; + hda_nid_t start_nid; + unsigned int caps; int i, nodes; - nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &nid); - if (!nid || nodes < 0) { + nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &start_nid); + if (!start_nid || nodes < 0) { codec_warn(codec, "HDMI: failed to get afg sub nodes\n"); return -EINVAL; } - for (i = 0; i < nodes; i++, nid++) { - unsigned int caps; - unsigned int type; + /* + * hdmi_add_pin() assumes total amount of converters to + * be known, so first discover all converters + */ + for (i = 0; i < nodes; i++) { + hda_nid_t nid = start_nid + i; caps = get_wcaps(codec, nid); - type = get_wcaps_type(caps); if (!(caps & AC_WCAP_DIGITAL)) continue; - switch (type) { - case AC_WID_AUD_OUT: + if (get_wcaps_type(caps) == AC_WID_AUD_OUT) hdmi_add_cvt(codec, nid); - break; - case AC_WID_PIN: + } + + /* discover audio pins */ + for (i = 0; i < nodes; i++) { + hda_nid_t nid = start_nid + i; + + caps = get_wcaps(codec, nid); + + if (!(caps & AC_WCAP_DIGITAL)) + continue; + + if (get_wcaps_type(caps) == AC_WID_PIN) hdmi_add_pin(codec, nid); - break; - } } return 0; @@ -3927,6 +3937,11 @@ HDA_CODEC_ENTRY(0x10de009d, "GPU 9d HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de009e, "GPU 9e HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de009f, "GPU 9f HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de00a0, "GPU a0 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de00a3, "GPU a3 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de00a4, "GPU a4 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de00a5, "GPU a5 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de00a6, "GPU a6 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de00a7, "GPU a7 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de8001, "MCP73 HDMI", patch_nvhdmi_2ch), HDA_CODEC_ENTRY(0x10de8067, "MCP67/68 HDMI", patch_nvhdmi_2ch), HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP", patch_via_hdmi), diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9670db6ad1e1..2b345ba083d8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -771,7 +771,7 @@ do_sku: alc_setup_gpio(codec, 0x02); break; case 7: - alc_setup_gpio(codec, 0x03); + alc_setup_gpio(codec, 0x04); break; case 5: default: @@ -956,7 +956,7 @@ struct alc_codec_rename_pci_table { const char *name; }; -static struct alc_codec_rename_table rename_tbl[] = { +static const struct alc_codec_rename_table rename_tbl[] = { { 0x10ec0221, 0xf00f, 0x1003, "ALC231" }, { 0x10ec0269, 0xfff0, 0x3010, "ALC277" }, { 0x10ec0269, 0xf0f0, 0x2010, "ALC259" }, @@ -977,7 +977,7 @@ static struct alc_codec_rename_table rename_tbl[] = { { } /* terminator */ }; -static struct alc_codec_rename_pci_table rename_pci_tbl[] = { +static const struct alc_codec_rename_pci_table rename_pci_tbl[] = { { 0x10ec0280, 0x1028, 0, "ALC3220" }, { 0x10ec0282, 0x1028, 0, "ALC3221" }, { 0x10ec0283, 0x1028, 0, "ALC3223" }, @@ -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), @@ -2556,6 +2565,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1462, 0xda57, "MSI Z270-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), + SND_PCI_QUIRK(0x1558, 0x3702, "Clevo X370SN[VW]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x50d3, "Clevo PC50[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x65d1, "Clevo PB51[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x65d2, "Clevo PB51R[CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), @@ -3115,7 +3125,7 @@ static void alc269_shutup(struct hda_codec *codec) alc_shutup_pins(codec); } -static struct coef_fw alc282_coefs[] = { +static const struct coef_fw alc282_coefs[] = { WRITE_COEF(0x03, 0x0002), /* Power Down Control */ UPDATE_COEF(0x05, 0xff3f, 0x0700), /* FIFO and filter clock */ WRITE_COEF(0x07, 0x0200), /* DMIC control */ @@ -3227,7 +3237,7 @@ static void alc282_shutup(struct hda_codec *codec) alc_write_coef_idx(codec, 0x78, coef78); } -static struct coef_fw alc283_coefs[] = { +static const struct coef_fw alc283_coefs[] = { WRITE_COEF(0x03, 0x0002), /* Power Down Control */ UPDATE_COEF(0x05, 0xff3f, 0x0700), /* FIFO and filter clock */ WRITE_COEF(0x07, 0x0200), /* DMIC control */ @@ -4234,7 +4244,7 @@ static void alc269_fixup_hp_line1_mic1_led(struct hda_codec *codec, } } -static struct coef_fw alc225_pre_hsmode[] = { +static const struct coef_fw alc225_pre_hsmode[] = { UPDATE_COEF(0x4a, 1<<8, 0), UPDATE_COEFEX(0x57, 0x05, 1<<14, 0), UPDATE_COEF(0x63, 3<<14, 3<<14), @@ -4247,7 +4257,7 @@ static struct coef_fw alc225_pre_hsmode[] = { static void alc_headset_mode_unplugged(struct hda_codec *codec) { - static struct coef_fw coef0255[] = { + static const struct coef_fw coef0255[] = { WRITE_COEF(0x1b, 0x0c0b), /* LDO and MISC control */ WRITE_COEF(0x45, 0xd089), /* UAJ function set to menual mode */ UPDATE_COEFEX(0x57, 0x05, 1<<14, 0), /* Direct Drive HP Amp control(Set to verb control)*/ @@ -4255,7 +4265,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) WRITE_COEFEX(0x57, 0x03, 0x8aa6), /* Direct Drive HP Amp control */ {} }; - static struct coef_fw coef0256[] = { + static const struct coef_fw coef0256[] = { WRITE_COEF(0x1b, 0x0c4b), /* LDO and MISC control */ WRITE_COEF(0x45, 0xd089), /* UAJ function set to menual mode */ WRITE_COEF(0x06, 0x6104), /* Set MIC2 Vref gate with HP */ @@ -4263,7 +4273,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) UPDATE_COEFEX(0x57, 0x05, 1<<14, 0), /* Direct Drive HP Amp control(Set to verb control)*/ {} }; - static struct coef_fw coef0233[] = { + static const struct coef_fw coef0233[] = { WRITE_COEF(0x1b, 0x0c0b), WRITE_COEF(0x45, 0xc429), UPDATE_COEF(0x35, 0x4000, 0), @@ -4273,7 +4283,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) WRITE_COEF(0x32, 0x42a3), {} }; - static struct coef_fw coef0288[] = { + static const struct coef_fw coef0288[] = { UPDATE_COEF(0x4f, 0xfcc0, 0xc400), UPDATE_COEF(0x50, 0x2000, 0x2000), UPDATE_COEF(0x56, 0x0006, 0x0006), @@ -4281,18 +4291,18 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) UPDATE_COEF(0x67, 0x2000, 0), {} }; - static struct coef_fw coef0298[] = { + static const struct coef_fw coef0298[] = { UPDATE_COEF(0x19, 0x1300, 0x0300), {} }; - static struct coef_fw coef0292[] = { + static const struct coef_fw coef0292[] = { WRITE_COEF(0x76, 0x000e), WRITE_COEF(0x6c, 0x2400), WRITE_COEF(0x18, 0x7308), WRITE_COEF(0x6b, 0xc429), {} }; - static struct coef_fw coef0293[] = { + static const struct coef_fw coef0293[] = { UPDATE_COEF(0x10, 7<<8, 6<<8), /* SET Line1 JD to 0 */ UPDATE_COEFEX(0x57, 0x05, 1<<15|1<<13, 0x0), /* SET charge pump by verb */ UPDATE_COEFEX(0x57, 0x03, 1<<10, 1<<10), /* SET EN_OSW to 1 */ @@ -4301,16 +4311,16 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) UPDATE_COEF(0x4a, 0x000f, 0x000e), /* Combo Jack auto detect */ {} }; - static struct coef_fw coef0668[] = { + static const struct coef_fw coef0668[] = { WRITE_COEF(0x15, 0x0d40), WRITE_COEF(0xb7, 0x802b), {} }; - static struct coef_fw coef0225[] = { + static const struct coef_fw coef0225[] = { UPDATE_COEF(0x63, 3<<14, 0), {} }; - static struct coef_fw coef0274[] = { + static const struct coef_fw coef0274[] = { UPDATE_COEF(0x4a, 0x0100, 0), UPDATE_COEFEX(0x57, 0x05, 0x4000, 0), UPDATE_COEF(0x6b, 0xf000, 0x5000), @@ -4375,25 +4385,25 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, hda_nid_t mic_pin) { - static struct coef_fw coef0255[] = { + static const struct coef_fw coef0255[] = { WRITE_COEFEX(0x57, 0x03, 0x8aa6), WRITE_COEF(0x06, 0x6100), /* Set MIC2 Vref gate to normal */ {} }; - static struct coef_fw coef0256[] = { + static const struct coef_fw coef0256[] = { UPDATE_COEFEX(0x57, 0x05, 1<<14, 1<<14), /* Direct Drive HP Amp control(Set to verb control)*/ WRITE_COEFEX(0x57, 0x03, 0x09a3), WRITE_COEF(0x06, 0x6100), /* Set MIC2 Vref gate to normal */ {} }; - static struct coef_fw coef0233[] = { + static const struct coef_fw coef0233[] = { UPDATE_COEF(0x35, 0, 1<<14), WRITE_COEF(0x06, 0x2100), WRITE_COEF(0x1a, 0x0021), WRITE_COEF(0x26, 0x008c), {} }; - static struct coef_fw coef0288[] = { + static const struct coef_fw coef0288[] = { UPDATE_COEF(0x4f, 0x00c0, 0), UPDATE_COEF(0x50, 0x2000, 0), UPDATE_COEF(0x56, 0x0006, 0), @@ -4402,30 +4412,30 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, UPDATE_COEF(0x67, 0x2000, 0x2000), {} }; - static struct coef_fw coef0292[] = { + static const struct coef_fw coef0292[] = { WRITE_COEF(0x19, 0xa208), WRITE_COEF(0x2e, 0xacf0), {} }; - static struct coef_fw coef0293[] = { + static const struct coef_fw coef0293[] = { UPDATE_COEFEX(0x57, 0x05, 0, 1<<15|1<<13), /* SET charge pump by verb */ UPDATE_COEFEX(0x57, 0x03, 1<<10, 0), /* SET EN_OSW to 0 */ UPDATE_COEF(0x1a, 1<<3, 0), /* Combo JD gating without LINE1-VREFO */ {} }; - static struct coef_fw coef0688[] = { + static const struct coef_fw coef0688[] = { WRITE_COEF(0xb7, 0x802b), WRITE_COEF(0xb5, 0x1040), UPDATE_COEF(0xc3, 0, 1<<12), {} }; - static struct coef_fw coef0225[] = { + static const struct coef_fw coef0225[] = { UPDATE_COEFEX(0x57, 0x05, 1<<14, 1<<14), UPDATE_COEF(0x4a, 3<<4, 2<<4), UPDATE_COEF(0x63, 3<<14, 0), {} }; - static struct coef_fw coef0274[] = { + static const struct coef_fw coef0274[] = { UPDATE_COEFEX(0x57, 0x05, 0x4000, 0x4000), UPDATE_COEF(0x4a, 0x0010, 0), UPDATE_COEF(0x6b, 0xf000, 0), @@ -4511,7 +4521,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, static void alc_headset_mode_default(struct hda_codec *codec) { - static struct coef_fw coef0225[] = { + static const struct coef_fw coef0225[] = { UPDATE_COEF(0x45, 0x3f<<10, 0x30<<10), UPDATE_COEF(0x45, 0x3f<<10, 0x31<<10), UPDATE_COEF(0x49, 3<<8, 0<<8), @@ -4520,14 +4530,14 @@ static void alc_headset_mode_default(struct hda_codec *codec) UPDATE_COEF(0x67, 0xf000, 0x3000), {} }; - static struct coef_fw coef0255[] = { + static const struct coef_fw coef0255[] = { WRITE_COEF(0x45, 0xc089), WRITE_COEF(0x45, 0xc489), WRITE_COEFEX(0x57, 0x03, 0x8ea6), WRITE_COEF(0x49, 0x0049), {} }; - static struct coef_fw coef0256[] = { + static const struct coef_fw coef0256[] = { WRITE_COEF(0x45, 0xc489), WRITE_COEFEX(0x57, 0x03, 0x0da3), WRITE_COEF(0x49, 0x0049), @@ -4535,12 +4545,12 @@ static void alc_headset_mode_default(struct hda_codec *codec) WRITE_COEF(0x06, 0x6100), {} }; - static struct coef_fw coef0233[] = { + static const struct coef_fw coef0233[] = { WRITE_COEF(0x06, 0x2100), WRITE_COEF(0x32, 0x4ea3), {} }; - static struct coef_fw coef0288[] = { + static const struct coef_fw coef0288[] = { UPDATE_COEF(0x4f, 0xfcc0, 0xc400), /* Set to TRS type */ UPDATE_COEF(0x50, 0x2000, 0x2000), UPDATE_COEF(0x56, 0x0006, 0x0006), @@ -4548,26 +4558,26 @@ static void alc_headset_mode_default(struct hda_codec *codec) UPDATE_COEF(0x67, 0x2000, 0), {} }; - static struct coef_fw coef0292[] = { + static const struct coef_fw coef0292[] = { WRITE_COEF(0x76, 0x000e), WRITE_COEF(0x6c, 0x2400), WRITE_COEF(0x6b, 0xc429), WRITE_COEF(0x18, 0x7308), {} }; - static struct coef_fw coef0293[] = { + static const struct coef_fw coef0293[] = { UPDATE_COEF(0x4a, 0x000f, 0x000e), /* Combo Jack auto detect */ WRITE_COEF(0x45, 0xC429), /* Set to TRS type */ UPDATE_COEF(0x1a, 1<<3, 0), /* Combo JD gating without LINE1-VREFO */ {} }; - static struct coef_fw coef0688[] = { + static const struct coef_fw coef0688[] = { WRITE_COEF(0x11, 0x0041), WRITE_COEF(0x15, 0x0d40), WRITE_COEF(0xb7, 0x802b), {} }; - static struct coef_fw coef0274[] = { + static const struct coef_fw coef0274[] = { WRITE_COEF(0x45, 0x4289), UPDATE_COEF(0x4a, 0x0010, 0x0010), UPDATE_COEF(0x6b, 0x0f00, 0), @@ -4630,53 +4640,53 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) { int val; - static struct coef_fw coef0255[] = { + static const struct coef_fw coef0255[] = { WRITE_COEF(0x45, 0xd489), /* Set to CTIA type */ WRITE_COEF(0x1b, 0x0c2b), WRITE_COEFEX(0x57, 0x03, 0x8ea6), {} }; - static struct coef_fw coef0256[] = { + static const struct coef_fw coef0256[] = { WRITE_COEF(0x45, 0xd489), /* Set to CTIA type */ WRITE_COEF(0x1b, 0x0e6b), {} }; - static struct coef_fw coef0233[] = { + static const struct coef_fw coef0233[] = { WRITE_COEF(0x45, 0xd429), WRITE_COEF(0x1b, 0x0c2b), WRITE_COEF(0x32, 0x4ea3), {} }; - static struct coef_fw coef0288[] = { + static const struct coef_fw coef0288[] = { UPDATE_COEF(0x50, 0x2000, 0x2000), UPDATE_COEF(0x56, 0x0006, 0x0006), UPDATE_COEF(0x66, 0x0008, 0), UPDATE_COEF(0x67, 0x2000, 0), {} }; - static struct coef_fw coef0292[] = { + static const struct coef_fw coef0292[] = { WRITE_COEF(0x6b, 0xd429), WRITE_COEF(0x76, 0x0008), WRITE_COEF(0x18, 0x7388), {} }; - static struct coef_fw coef0293[] = { + static const struct coef_fw coef0293[] = { WRITE_COEF(0x45, 0xd429), /* Set to ctia type */ UPDATE_COEF(0x10, 7<<8, 7<<8), /* SET Line1 JD to 1 */ {} }; - static struct coef_fw coef0688[] = { + static const struct coef_fw coef0688[] = { WRITE_COEF(0x11, 0x0001), WRITE_COEF(0x15, 0x0d60), WRITE_COEF(0xc3, 0x0000), {} }; - static struct coef_fw coef0225_1[] = { + static const struct coef_fw coef0225_1[] = { UPDATE_COEF(0x45, 0x3f<<10, 0x35<<10), UPDATE_COEF(0x63, 3<<14, 2<<14), {} }; - static struct coef_fw coef0225_2[] = { + static const struct coef_fw coef0225_2[] = { UPDATE_COEF(0x45, 0x3f<<10, 0x35<<10), UPDATE_COEF(0x63, 3<<14, 1<<14), {} @@ -4748,48 +4758,48 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) /* Nokia type */ static void alc_headset_mode_omtp(struct hda_codec *codec) { - static struct coef_fw coef0255[] = { + static const struct coef_fw coef0255[] = { WRITE_COEF(0x45, 0xe489), /* Set to OMTP Type */ WRITE_COEF(0x1b, 0x0c2b), WRITE_COEFEX(0x57, 0x03, 0x8ea6), {} }; - static struct coef_fw coef0256[] = { + static const struct coef_fw coef0256[] = { WRITE_COEF(0x45, 0xe489), /* Set to OMTP Type */ WRITE_COEF(0x1b, 0x0e6b), {} }; - static struct coef_fw coef0233[] = { + static const struct coef_fw coef0233[] = { WRITE_COEF(0x45, 0xe429), WRITE_COEF(0x1b, 0x0c2b), WRITE_COEF(0x32, 0x4ea3), {} }; - static struct coef_fw coef0288[] = { + static const struct coef_fw coef0288[] = { UPDATE_COEF(0x50, 0x2000, 0x2000), UPDATE_COEF(0x56, 0x0006, 0x0006), UPDATE_COEF(0x66, 0x0008, 0), UPDATE_COEF(0x67, 0x2000, 0), {} }; - static struct coef_fw coef0292[] = { + static const struct coef_fw coef0292[] = { WRITE_COEF(0x6b, 0xe429), WRITE_COEF(0x76, 0x0008), WRITE_COEF(0x18, 0x7388), {} }; - static struct coef_fw coef0293[] = { + static const struct coef_fw coef0293[] = { WRITE_COEF(0x45, 0xe429), /* Set to omtp type */ UPDATE_COEF(0x10, 7<<8, 7<<8), /* SET Line1 JD to 1 */ {} }; - static struct coef_fw coef0688[] = { + static const struct coef_fw coef0688[] = { WRITE_COEF(0x11, 0x0001), WRITE_COEF(0x15, 0x0d50), WRITE_COEF(0xc3, 0x0000), {} }; - static struct coef_fw coef0225[] = { + static const struct coef_fw coef0225[] = { UPDATE_COEF(0x45, 0x3f<<10, 0x39<<10), UPDATE_COEF(0x63, 3<<14, 2<<14), {} @@ -4849,17 +4859,17 @@ static void alc_determine_headset_type(struct hda_codec *codec) int val; bool is_ctia = false; struct alc_spec *spec = codec->spec; - static struct coef_fw coef0255[] = { + static const struct coef_fw coef0255[] = { WRITE_COEF(0x45, 0xd089), /* combo jack auto switch control(Check type)*/ WRITE_COEF(0x49, 0x0149), /* combo jack auto switch control(Vref conteol) */ {} }; - static struct coef_fw coef0288[] = { + static const struct coef_fw coef0288[] = { UPDATE_COEF(0x4f, 0xfcc0, 0xd400), /* Check Type */ {} }; - static struct coef_fw coef0298[] = { + static const struct coef_fw coef0298[] = { UPDATE_COEF(0x50, 0x2000, 0x2000), UPDATE_COEF(0x56, 0x0006, 0x0006), UPDATE_COEF(0x66, 0x0008, 0), @@ -4867,19 +4877,19 @@ static void alc_determine_headset_type(struct hda_codec *codec) UPDATE_COEF(0x19, 0x1300, 0x1300), {} }; - static struct coef_fw coef0293[] = { + static const struct coef_fw coef0293[] = { UPDATE_COEF(0x4a, 0x000f, 0x0008), /* Combo Jack auto detect */ WRITE_COEF(0x45, 0xD429), /* Set to ctia type */ {} }; - static struct coef_fw coef0688[] = { + static const struct coef_fw coef0688[] = { WRITE_COEF(0x11, 0x0001), WRITE_COEF(0xb7, 0x802b), WRITE_COEF(0x15, 0x0d60), WRITE_COEF(0xc3, 0x0c00), {} }; - static struct coef_fw coef0274[] = { + static const struct coef_fw coef0274[] = { UPDATE_COEF(0x4a, 0x0010, 0), UPDATE_COEF(0x4a, 0x8000, 0), WRITE_COEF(0x45, 0xd289), @@ -5164,7 +5174,7 @@ static void alc_fixup_headset_mode_no_hp_mic(struct hda_codec *codec, static void alc255_set_default_jack_type(struct hda_codec *codec) { /* Set to iphone type */ - static struct coef_fw alc255fw[] = { + static const struct coef_fw alc255fw[] = { WRITE_COEF(0x1b, 0x880b), WRITE_COEF(0x45, 0xd089), WRITE_COEF(0x1b, 0x080b), @@ -5172,7 +5182,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec) WRITE_COEF(0x1b, 0x0c0b), {} }; - static struct coef_fw alc256fw[] = { + static const struct coef_fw alc256fw[] = { WRITE_COEF(0x1b, 0x884b), WRITE_COEF(0x45, 0xd089), WRITE_COEF(0x1b, 0x084b), @@ -7167,6 +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), @@ -8510,7 +8521,103 @@ static void alc662_fixup_usi_headset_mic(struct hda_codec *codec, } } -static struct coef_fw alc668_coefs[] = { +static void alc662_aspire_ethos_mute_speakers(struct hda_codec *codec, + struct hda_jack_callback *cb) +{ + /* surround speakers at 0x1b already get muted automatically when + * headphones are plugged in, but we have to mute/unmute the remaining + * channels manually: + * 0x15 - front left/front right + * 0x18 - front center/ LFE + */ + if (snd_hda_jack_detect_state(codec, 0x1b) == HDA_JACK_PRESENT) { + snd_hda_set_pin_ctl_cache(codec, 0x15, 0); + snd_hda_set_pin_ctl_cache(codec, 0x18, 0); + } else { + snd_hda_set_pin_ctl_cache(codec, 0x15, PIN_OUT); + snd_hda_set_pin_ctl_cache(codec, 0x18, PIN_OUT); + } +} + +static void alc662_fixup_aspire_ethos_hp(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + /* Pin 0x1b: shared headphones jack and surround speakers */ + if (!is_jack_detectable(codec, 0x1b)) + return; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_jack_detect_enable_callback(codec, 0x1b, + alc662_aspire_ethos_mute_speakers); + /* subwoofer needs an extra GPIO setting to become audible */ + alc_setup_gpio(codec, 0x02); + break; + case HDA_FIXUP_ACT_INIT: + /* Make sure to start in a correct state, i.e. if + * headphones have been plugged in before powering up the system + */ + alc662_aspire_ethos_mute_speakers(codec, NULL); + break; + } +} + +static void alc671_fixup_hp_headset_mic2(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + static const struct hda_pintbl pincfgs[] = { + { 0x19, 0x02a11040 }, /* use as headset mic, with its own jack detect */ + { 0x1b, 0x0181304f }, + { } + }; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->gen.mixer_nid = 0; + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; + snd_hda_apply_pincfgs(codec, pincfgs); + break; + case HDA_FIXUP_ACT_INIT: + alc_write_coef_idx(codec, 0x19, 0xa054); + break; + } +} + +static void alc897_hp_automute_hook(struct hda_codec *codec, + struct hda_jack_callback *jack) +{ + struct alc_spec *spec = codec->spec; + int vref; + + snd_hda_gen_hp_automute(codec, jack); + vref = spec->gen.hp_jack_present ? (PIN_HP | AC_PINCTL_VREF_100) : PIN_HP; + snd_hda_codec_write(codec, 0x1b, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + vref); +} + +static void alc897_fixup_lenovo_headset_mic(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->gen.hp_automute_hook = alc897_hp_automute_hook; + } +} + +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), WRITE_COEF(0x08, 0x0031), WRITE_COEF(0x0a, 0x0060), WRITE_COEF(0x0b, 0x0), @@ -8581,6 +8688,19 @@ enum { ALC662_FIXUP_USI_FUNC, ALC662_FIXUP_USI_HEADSET_MODE, ALC662_FIXUP_LENOVO_MULTI_CODECS, + ALC669_FIXUP_ACER_ASPIRE_ETHOS, + ALC669_FIXUP_ACER_ASPIRE_ETHOS_HEADSET, + ALC671_FIXUP_HP_HEADSET_MIC2, + ALC662_FIXUP_ACER_X2660G_HEADSET_MODE, + ALC662_FIXUP_ACER_NITRO_HEADSET_MODE, + ALC668_FIXUP_ASUS_NO_HEADSET_MIC, + ALC668_FIXUP_HEADSET_MIC, + ALC668_FIXUP_MIC_DET_COEF, + 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[] = { @@ -8907,6 +9027,100 @@ static const struct hda_fixup alc662_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc233_alc662_fixup_lenovo_dual_codecs, }, + [ALC669_FIXUP_ACER_ASPIRE_ETHOS_HEADSET] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc662_fixup_aspire_ethos_hp, + }, + [ALC669_FIXUP_ACER_ASPIRE_ETHOS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x15, 0x92130110 }, /* front speakers */ + { 0x18, 0x99130111 }, /* center/subwoofer */ + { 0x1b, 0x11130012 }, /* surround plus jack for HP */ + { } + }, + .chained = true, + .chain_id = ALC669_FIXUP_ACER_ASPIRE_ETHOS_HEADSET + }, + [ALC671_FIXUP_HP_HEADSET_MIC2] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc671_fixup_hp_headset_mic2, + }, + [ALC662_FIXUP_ACER_X2660G_HEADSET_MODE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x02a1113c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC662_FIXUP_USI_FUNC + }, + [ALC662_FIXUP_ACER_NITRO_HEADSET_MODE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x01a11140 }, /* use as headset mic, without its own jack detect */ + { 0x1b, 0x0221144f }, + { } + }, + .chained = true, + .chain_id = ALC662_FIXUP_USI_FUNC + }, + [ALC668_FIXUP_ASUS_NO_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1b, 0x04a1112c }, + { } + }, + .chained = true, + .chain_id = ALC668_FIXUP_HEADSET_MIC + }, + [ALC668_FIXUP_HEADSET_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_headset_mic, + .chained = true, + .chain_id = ALC668_FIXUP_MIC_DET_COEF + }, + [ALC668_FIXUP_MIC_DET_COEF] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0x15 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0d60 }, + {} + }, + }, + [ALC897_FIXUP_LENOVO_HEADSET_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc897_fixup_lenovo_headset_mic, + }, + [ALC897_FIXUP_HEADSET_MIC_PIN] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x03a11050 }, + { } + }, + .chained = true, + .chain_id = ALC897_FIXUP_LENOVO_HEADSET_MIC + }, + [ALC897_FIXUP_HP_HSMIC_VERB] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + }, + [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[] = { @@ -8918,6 +9132,8 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0349, "eMachines eM250", ALC662_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x034a, "Gateway LT27", ALC662_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), + SND_PCI_QUIRK(0x1025, 0x123c, "Acer Nitro N50-600", ALC662_FIXUP_ACER_NITRO_HEADSET_MODE), + SND_PCI_QUIRK(0x1025, 0x124e, "Acer 2660G", ALC662_FIXUP_ACER_X2660G_HEADSET_MODE), SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05fe, "Dell XPS 15", ALC668_FIXUP_DELL_XPS13), @@ -8929,6 +9145,9 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0698, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x069f, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), + SND_PCI_QUIRK(0x103c, 0x8719, "HP", ALC897_FIXUP_HP_HSMIC_VERB), + SND_PCI_QUIRK(0x103c, 0x872b, "HP", ALC897_FIXUP_HP_HSMIC_VERB), + SND_PCI_QUIRK(0x103c, 0x873e, "HP", ALC671_FIXUP_HP_HEADSET_MIC2), SND_PCI_QUIRK(0x1043, 0x1080, "Asus UX501VW", ALC668_FIXUP_HEADSET_MODE), SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_ASUS_Nx50), SND_PCI_QUIRK(0x1043, 0x13df, "Asus N550JX", ALC662_FIXUP_BASS_1A), @@ -8938,6 +9157,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x15a7, "ASUS UX51VZH", ALC662_FIXUP_BASS_16), SND_PCI_QUIRK(0x1043, 0x177d, "ASUS N551", ALC668_FIXUP_ASUS_Nx51), SND_PCI_QUIRK(0x1043, 0x17bd, "ASUS N751", ALC668_FIXUP_ASUS_Nx51), + SND_PCI_QUIRK(0x1043, 0x185d, "ASUS G551JW", ALC668_FIXUP_ASUS_NO_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71SL", ALC662_FIXUP_ASUS_MODE8), SND_PCI_QUIRK(0x1043, 0x1b73, "ASUS N55SF", ALC662_FIXUP_BASS_16), SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), @@ -8946,12 +9166,21 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x14cd, 0x5003, "USI", ALC662_FIXUP_USI_HEADSET_MODE), SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC662_FIXUP_LENOVO_MULTI_CODECS), + SND_PCI_QUIRK(0x17aa, 0x32ca, "Lenovo ThinkCentre M80", ALC897_FIXUP_HEADSET_MIC_PIN), + 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), SND_PCI_QUIRK(0x19da, 0xa130, "Zotac Z68", ALC662_FIXUP_ZOTAC_Z68), SND_PCI_QUIRK(0x1b0a, 0x01b8, "ACER Veriton", ALC662_FIXUP_ACER_VERITON), SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T), + SND_PCI_QUIRK(0x1025, 0x0566, "Acer Aspire Ethos 8951G", ALC669_FIXUP_ACER_ASPIRE_ETHOS), #if 0 /* Below is a quirk table taken from the old code. @@ -9044,6 +9273,7 @@ static const struct hda_model_fixup alc662_fixup_models[] = { {.id = ALC892_FIXUP_ASROCK_MOBO, .name = "asrock-mobo"}, {.id = ALC662_FIXUP_USI_HEADSET_MODE, .name = "usi-headset"}, {.id = ALC662_FIXUP_LENOVO_MULTI_CODECS, .name = "dual-codecs"}, + {.id = ALC669_FIXUP_ACER_ASPIRE_ETHOS, .name = "aspire-ethos"}, {} }; @@ -9086,6 +9316,23 @@ static const struct snd_hda_pin_quirk alc662_pin_fixup_tbl[] = { {0x12, 0x90a60130}, {0x14, 0x90170110}, {0x15, 0x0321101f}), + SND_HDA_PIN_QUIRK(0x10ec0671, 0x103c, "HP cPC", ALC671_FIXUP_HP_HEADSET_MIC2, + {0x14, 0x01014010}, + {0x17, 0x90170150}, + {0x19, 0x02a11060}, + {0x1b, 0x01813030}, + {0x21, 0x02211020}), + SND_HDA_PIN_QUIRK(0x10ec0671, 0x103c, "HP cPC", ALC671_FIXUP_HP_HEADSET_MIC2, + {0x14, 0x01014010}, + {0x18, 0x01a19040}, + {0x1b, 0x01813030}, + {0x21, 0x02211020}), + SND_HDA_PIN_QUIRK(0x10ec0671, 0x103c, "HP cPC", ALC671_FIXUP_HP_HEADSET_MIC2, + {0x14, 0x01014020}, + {0x17, 0x90170110}, + {0x18, 0x01a19050}, + {0x1b, 0x01813040}, + {0x21, 0x02211030}), {} }; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 8d09312b2e42..e91df1152612 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1723,6 +1723,7 @@ static const struct snd_pci_quirk stac925x_fixup_tbl[] = { }; static const struct hda_pintbl ref92hd73xx_pin_configs[] = { + // Port A-H { 0x0a, 0x02214030 }, { 0x0b, 0x02a19040 }, { 0x0c, 0x01a19020 }, @@ -1731,9 +1732,12 @@ static const struct hda_pintbl ref92hd73xx_pin_configs[] = { { 0x0f, 0x01014010 }, { 0x10, 0x01014020 }, { 0x11, 0x01014030 }, + // CD in { 0x12, 0x02319040 }, + // Digial Mic ins { 0x13, 0x90a000f0 }, { 0x14, 0x90a000f0 }, + // Digital outs { 0x22, 0x01452050 }, { 0x23, 0x01452050 }, {} @@ -1774,6 +1778,7 @@ static const struct hda_pintbl alienware_m17x_pin_configs[] = { }; static const struct hda_pintbl intel_dg45id_pin_configs[] = { + // Analog outputs { 0x0a, 0x02214230 }, { 0x0b, 0x02A19240 }, { 0x0c, 0x01013214 }, @@ -1781,6 +1786,9 @@ static const struct hda_pintbl intel_dg45id_pin_configs[] = { { 0x0e, 0x01A19250 }, { 0x0f, 0x01011212 }, { 0x10, 0x01016211 }, + // Digital output + { 0x22, 0x01451380 }, + { 0x23, 0x40f000f0 }, {} }; @@ -1971,6 +1979,8 @@ static const struct snd_pci_quirk stac92hd73xx_fixup_tbl[] = { "DFI LanParty", STAC_92HD73XX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_92HD73XX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x5001, + "Intel DP45SG", STAC_92HD73XX_INTEL), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x5002, "Intel DG45ID", STAC_92HD73XX_INTEL), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x5003, diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 937155b1fae0..9e2252eee626 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -834,6 +834,9 @@ static int add_secret_dac_path(struct hda_codec *codec) return 0; nums = snd_hda_get_connections(codec, spec->gen.mixer_nid, conn, ARRAY_SIZE(conn) - 1); + if (nums < 0) + return nums; + for (i = 0; i < nums; i++) { if (get_wcaps_type(get_wcaps(codec, conn[i])) == AC_WID_AUD_OUT) return 0; diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c index c9411dfff5a4..3473f1040d92 100644 --- a/sound/pci/ice1712/aureon.c +++ b/sound/pci/ice1712/aureon.c @@ -1906,6 +1906,7 @@ static int aureon_add_controls(struct snd_ice1712 *ice) unsigned char id; snd_ice1712_save_gpio_status(ice); id = aureon_cs8415_get(ice, CS8415_ID); + snd_ice1712_restore_gpio_status(ice); if (id != 0x41) dev_info(ice->card->dev, "No CS8415 chip. Skipping CS8415 controls.\n"); @@ -1923,7 +1924,6 @@ static int aureon_add_controls(struct snd_ice1712 *ice) kctl->id.device = ice->pcm->device; } } - snd_ice1712_restore_gpio_status(ice); } return 0; diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index a80684bdc30d..46f536209671 100644 --- a/sound/pci/lx6464es/lx_core.c +++ b/sound/pci/lx6464es/lx_core.c @@ -508,12 +508,11 @@ int lx_buffer_ask(struct lx6464es *chip, u32 pipe, int is_capture, dev_dbg(chip->card->dev, "CMD_08_ASK_BUFFERS: needed %d, freed %d\n", *r_needed, *r_freed); - for (i = 0; i < MAX_STREAM_BUFFER; ++i) { - for (i = 0; i != chip->rmh.stat_len; ++i) - dev_dbg(chip->card->dev, - " stat[%d]: %x, %x\n", i, - chip->rmh.stat[i], - chip->rmh.stat[i] & MASK_DATA_SIZE); + for (i = 0; i < MAX_STREAM_BUFFER && i < chip->rmh.stat_len; + ++i) { + dev_dbg(chip->card->dev, " stat[%d]: %x, %x\n", i, + chip->rmh.stat[i], + chip->rmh.stat[i] & MASK_DATA_SIZE); } } diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 81af21ac1439..ba8721337d5a 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -730,7 +730,7 @@ static int ac97_fp_rec_volume_put(struct snd_kcontrol *ctl, oldreg = oxygen_read_ac97(chip, 1, AC97_REC_GAIN); newreg = oldreg & ~0x0707; newreg = newreg | (value->value.integer.value[0] & 7); - newreg = newreg | ((value->value.integer.value[0] & 7) << 8); + newreg = newreg | ((value->value.integer.value[1] & 7) << 8); change = newreg != oldreg; if (change) oxygen_write_ac97(chip, 1, AC97_REC_GAIN, newreg); diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index 99cc73150576..ab7f76117474 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/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 06d32257ddb6..5041f43ee5f7 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -59,6 +59,35 @@ */ #undef ENABLE_MIC_INPUT +static struct clk *mclk; + +static int at91sam9g20ek_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + static int mclk_on; + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + if (!mclk_on) + ret = clk_enable(mclk); + if (ret == 0) + mclk_on = 1; + break; + + case SND_SOC_BIAS_OFF: + case SND_SOC_BIAS_STANDBY: + if (mclk_on) + clk_disable(mclk); + mclk_on = 0; + break; + } + + return ret; +} + static const struct snd_soc_dapm_widget at91sam9g20ek_dapm_widgets[] = { SND_SOC_DAPM_MIC("Int Mic", NULL), SND_SOC_DAPM_SPK("Ext Spk", NULL), @@ -117,6 +146,7 @@ static struct snd_soc_card snd_soc_at91sam9g20ek = { .owner = THIS_MODULE, .dai_link = &at91sam9g20ek_dai, .num_links = 1, + .set_bias_level = at91sam9g20ek_set_bias_level, .dapm_widgets = at91sam9g20ek_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(at91sam9g20ek_dapm_widgets), @@ -129,6 +159,7 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; struct device_node *codec_np, *cpu_np; + struct clk *pllb; struct snd_soc_card *card = &snd_soc_at91sam9g20ek; int ret; @@ -142,6 +173,31 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) return -EINVAL; } + /* + * Codec MCLK is supplied by PCK0 - set it up. + */ + mclk = clk_get(NULL, "pck0"); + if (IS_ERR(mclk)) { + dev_err(&pdev->dev, "Failed to get MCLK\n"); + ret = PTR_ERR(mclk); + goto err; + } + + pllb = clk_get(NULL, "pllb"); + if (IS_ERR(pllb)) { + dev_err(&pdev->dev, "Failed to get PLLB\n"); + ret = PTR_ERR(pllb); + goto err_mclk; + } + ret = clk_set_parent(mclk, pllb); + clk_put(pllb); + if (ret != 0) { + dev_err(&pdev->dev, "Failed to set MCLK parent\n"); + goto err_mclk; + } + + clk_set_rate(mclk, MCLK_RATE); + card->dev = &pdev->dev; /* Parse device node info */ @@ -185,6 +241,9 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) return ret; +err_mclk: + clk_put(mclk); + mclk = NULL; err: atmel_ssc_put_audio(0); return ret; @@ -194,6 +253,8 @@ static int at91sam9g20ek_audio_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); + clk_disable(mclk); + mclk = NULL; snd_soc_unregister_card(card); atmel_ssc_put_audio(0); diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c index 73fa784646e5..8436df40bbda 100644 --- a/sound/soc/codecs/cs35l33.c +++ b/sound/soc/codecs/cs35l33.c @@ -26,13 +26,11 @@ #include <sound/soc-dapm.h> #include <sound/initval.h> #include <sound/tlv.h> -#include <linux/gpio.h> #include <linux/gpio/consumer.h> #include <sound/cs35l33.h> #include <linux/pm_runtime.h> #include <linux/regulator/consumer.h> #include <linux/regulator/machine.h> -#include <linux/of_gpio.h> #include <linux/of.h> #include <linux/of_device.h> #include <linux/of_irq.h> @@ -1171,7 +1169,7 @@ static int cs35l33_i2c_probe(struct i2c_client *i2c_client, /* We could issue !RST or skip it based on AMP topology */ cs35l33->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, - "reset-gpios", GPIOD_OUT_HIGH); + "reset", GPIOD_OUT_HIGH); if (IS_ERR(cs35l33->reset_gpio)) { dev_err(&i2c_client->dev, "%s ERROR: Can't get reset GPIO\n", __func__); diff --git a/sound/soc/codecs/cs35l34.c b/sound/soc/codecs/cs35l34.c index 5063c05afa27..72c7c8426f3f 100644 --- a/sound/soc/codecs/cs35l34.c +++ b/sound/soc/codecs/cs35l34.c @@ -24,14 +24,12 @@ #include <linux/regulator/machine.h> #include <linux/pm_runtime.h> #include <linux/of_device.h> -#include <linux/of_gpio.h> #include <linux/of_irq.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-dapm.h> -#include <linux/gpio.h> #include <linux/gpio/consumer.h> #include <sound/initval.h> #include <sound/tlv.h> @@ -1062,7 +1060,7 @@ static int cs35l34_i2c_probe(struct i2c_client *i2c_client, dev_err(&i2c_client->dev, "Failed to request IRQ: %d\n", ret); cs35l34->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, - "reset-gpios", GPIOD_OUT_LOW); + "reset", GPIOD_OUT_LOW); if (IS_ERR(cs35l34->reset_gpio)) return PTR_ERR(cs35l34->reset_gpio); diff --git a/sound/soc/codecs/cs42l51-i2c.c b/sound/soc/codecs/cs42l51-i2c.c index 4b5731a41876..cd93e93a5983 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 5080d7a3c279..662f1f85ba36 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 0ca805492ac4..8c55bf384bc6 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/cs42l56.c b/sound/soc/codecs/cs42l56.c index deaad703a7db..a4826a7d0a98 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -1204,18 +1204,12 @@ static int cs42l56_i2c_probe(struct i2c_client *i2c_client, if (pdata) { cs42l56->pdata = *pdata; } else { - pdata = devm_kzalloc(&i2c_client->dev, sizeof(*pdata), - GFP_KERNEL); - if (!pdata) - return -ENOMEM; - if (i2c_client->dev.of_node) { ret = cs42l56_handle_of_data(i2c_client, &cs42l56->pdata); if (ret != 0) return ret; } - cs42l56->pdata = *pdata; } if (cs42l56->pdata.gpio_nreset) { diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index cf29dec28b5e..0ffd93564555 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -581,7 +581,7 @@ static int cs43130_set_sp_fmt(int dai_id, unsigned int bitwidth_sclk, break; case SND_SOC_DAIFMT_LEFT_J: hi_size = bitwidth_sclk; - frm_delay = 2; + frm_delay = 0; frm_phase = 1; break; case SND_SOC_DAIFMT_DSP_A: @@ -1686,7 +1686,7 @@ static ssize_t cs43130_show_dc_r(struct device *dev, return cs43130_show_dc(dev, buf, HP_RIGHT); } -static u16 const cs43130_ac_freq[CS43130_AC_FREQ] = { +static const u16 cs43130_ac_freq[CS43130_AC_FREQ] = { 24, 43, 93, @@ -2365,7 +2365,7 @@ static const struct regmap_config cs43130_regmap = { .use_single_rw = true, /* needed for regcache_sync */ }; -static u16 const cs43130_dc_threshold[CS43130_DC_THRESHOLD] = { +static const u16 cs43130_dc_threshold[CS43130_DC_THRESHOLD] = { 50, 120, }; diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 2c7d5088e6f2..e3515ac8b223 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; @@ -655,7 +659,7 @@ static struct da7219_aad_pdata *da7219_aad_fw_to_pdata(struct snd_soc_component aad_pdata->mic_det_thr = da7219_aad_fw_mic_det_thr(component, fw_val32); else - aad_pdata->mic_det_thr = DA7219_AAD_MIC_DET_THR_500_OHMS; + aad_pdata->mic_det_thr = DA7219_AAD_MIC_DET_THR_200_OHMS; if (fwnode_property_read_u32(aad_np, "dlg,jack-ins-deb", &fw_val32) >= 0) aad_pdata->jack_ins_deb = @@ -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 57130edaf3ab..0fc4755fd0d9 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/nau8824.c b/sound/soc/codecs/nau8824.c index 4f18bb272e92..0ecea65a80b4 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1899,6 +1899,30 @@ static const struct dmi_system_id nau8824_quirk_table[] = { }, .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), }, + { + /* Positivo CW14Q01P */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "CW14Q01P"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, + { + /* Positivo K1424G */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "K1424G"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, + { + /* Positivo N14ZP74G */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "N14ZP74G"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, {} }; diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 5272c81641c1..310cfceab41f 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -1471,7 +1471,7 @@ int pcm512x_probe(struct device *dev, struct regmap *regmap) if (val > 6) { dev_err(dev, "Invalid pll-in\n"); ret = -EINVAL; - goto err_clk; + goto err_pm; } pcm512x->pll_in = val; } @@ -1480,7 +1480,7 @@ int pcm512x_probe(struct device *dev, struct regmap *regmap) if (val > 6) { dev_err(dev, "Invalid pll-out\n"); ret = -EINVAL; - goto err_clk; + goto err_pm; } pcm512x->pll_out = val; } @@ -1489,12 +1489,12 @@ int pcm512x_probe(struct device *dev, struct regmap *regmap) dev_err(dev, "Error: both pll-in and pll-out, or none\n"); ret = -EINVAL; - goto err_clk; + goto err_pm; } if (pcm512x->pll_in && pcm512x->pll_in == pcm512x->pll_out) { dev_err(dev, "Error: pll-in == pll-out\n"); ret = -EINVAL; - goto err_clk; + goto err_pm; } } #endif diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 06cdba4edfe2..3181b91a025b 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -1169,6 +1169,13 @@ static const struct dmi_system_id force_combo_jack_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Geminilake") } }, + { + .ident = "Intel Kabylake R RVP", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_NAME, "Kabylake Client platform") + } + }, { } }; diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index d34000182f67..37ad3bee66a4 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -419,6 +419,7 @@ struct rt5645_priv { struct regulator_bulk_data supplies[ARRAY_SIZE(rt5645_supply_names)]; struct rt5645_eq_param_s *eq_param; struct timer_list btn_check_timer; + struct mutex jd_mutex; int codec_type; int sysclk; @@ -3216,6 +3217,8 @@ static int rt5645_jack_detect(struct snd_soc_component *component, int jack_inse rt5645_enable_push_button_irq(component, true); } } else { + if (rt5645->en_button_func) + rt5645_enable_push_button_irq(component, false); snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); snd_soc_dapm_sync(dapm); rt5645->jack_type = SND_JACK_HEADPHONE; @@ -3278,6 +3281,8 @@ 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); @@ -3294,6 +3299,8 @@ static void rt5645_jack_detect_work(struct work_struct *work) if (!rt5645->component) return; + mutex_lock(&rt5645->jd_mutex); + switch (rt5645->pdata.jd_mode) { case 0: /* Not using rt5645 JD */ if (rt5645->gpiod_hp_det) { @@ -3318,7 +3325,7 @@ static void rt5645_jack_detect_work(struct work_struct *work) if (!val && (rt5645->jack_type == 0)) { /* jack in */ report = rt5645_jack_detect(rt5645->component, 1); - } else if (!val && rt5645->jack_type != 0) { + } else if (!val && rt5645->jack_type == SND_JACK_HEADSET) { /* for push button and jack out */ btn_type = 0; if (snd_soc_component_read32(rt5645->component, RT5645_INT_IRQ_ST) & 0x4) { @@ -3374,6 +3381,8 @@ static void rt5645_jack_detect_work(struct work_struct *work) rt5645_jack_detect(rt5645->component, 0); } + mutex_unlock(&rt5645->jd_mutex); + snd_soc_jack_report(rt5645->hp_jack, report, SND_JACK_HEADPHONE); snd_soc_jack_report(rt5645->mic_jack, report, SND_JACK_MICROPHONE); if (rt5645->en_button_func) @@ -4070,6 +4079,7 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, } timer_setup(&rt5645->btn_check_timer, rt5645_btn_check_callback, 0); + mutex_init(&rt5645->jd_mutex); INIT_DELAYED_WORK(&rt5645->jack_detect_work, rt5645_jack_detect_work); INIT_DELAYED_WORK(&rt5645->rcclock_work, rt5645_rcclock_work); diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 6ba99f5ed3f4..a7ed2a19c3ec 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/rt5670.c b/sound/soc/codecs/rt5670.c index 6a2a58e107e3..9dd99d123e44 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -3217,8 +3217,6 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, if (ret < 0) goto err; - pm_runtime_put(&i2c->dev); - return 0; err: pm_runtime_disable(&i2c->dev); diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 501a4e73b185..06f382c794b2 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -67,6 +67,18 @@ static const struct reg_default ssm2602_reg[SSM2602_CACHEREGNUM] = { { .reg = 0x09, .def = 0x0000 } }; +/* + * ssm2602 register patch + * Workaround for playback distortions after power up: activates digital + * core, and then powers on output, DAC, and whole chip at the same time + */ + +static const struct reg_sequence ssm2602_patch[] = { + { SSM2602_ACTIVE, 0x01 }, + { SSM2602_PWR, 0x07 }, + { SSM2602_RESET, 0x00 }, +}; + /*Appending several "None"s just for OSS mixer use*/ static const char *ssm2602_input_select[] = { @@ -577,6 +589,9 @@ static int ssm260x_component_probe(struct snd_soc_component *component) return ret; } + regmap_register_patch(ssm2602->regmap, ssm2602_patch, + ARRAY_SIZE(ssm2602_patch)); + /* set the update bits */ regmap_update_bits(ssm2602->regmap, SSM2602_LINVOL, LINVOL_LRIN_BOTH, LINVOL_LRIN_BOTH); diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index d14e851b9160..03d3b0f17f87 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/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index e3e069277a3f..13ef2bebf6da 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3715,7 +3715,12 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) } else { dev_dbg(component->dev, "Jack not detected\n"); + /* Release wm8994->accdet_lock to avoid deadlock: + * cancel_delayed_work_sync() takes wm8994->mic_work internal + * lock and wm1811_mic_work takes wm8994->accdet_lock */ + mutex_unlock(&wm8994->accdet_lock); cancel_delayed_work_sync(&wm8994->mic_work); + mutex_lock(&wm8994->accdet_lock); snd_soc_component_update_bits(component, WM8958_MICBIAS2, WM8958_MICB2_DISCH, WM8958_MICB2_DISCH); diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 65112b9d8588..90b8814d7506 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -132,13 +132,13 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id) /* Error Handling: TX */ if (isr[i] & ISR_TXFO) { - dev_err(dev->dev, "TX overrun (ch_id=%d)\n", i); + dev_err_ratelimited(dev->dev, "TX overrun (ch_id=%d)\n", i); irq_valid = true; } /* Error Handling: TX */ if (isr[i] & ISR_RXFO) { - dev_err(dev->dev, "RX overrun (ch_id=%d)\n", i); + dev_err_ratelimited(dev->dev, "RX overrun (ch_id=%d)\n", i); irq_valid = true; } } diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 740b90df44bb..0a1ba64ed63c 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 64bf3560c1d1..7567ee380283 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/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index c4d19b88d17d..d27dd170beda 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -400,6 +400,18 @@ static int byt_rt5640_aif1_hw_params(struct snd_pcm_substream *substream, /* Please keep this list alphabetically sorted */ static const struct dmi_system_id byt_rt5640_quirk_table[] = { + { /* Acer Iconia One 7 B1-750 */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Insyde"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "VESPA2"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD1_IN4P | + BYT_RT5640_OVCD_TH_1500UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* Acer Iconia Tab 8 W1-810 */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Acer"), @@ -438,6 +450,21 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_MCLK_EN), }, { + /* Advantech MICA-071 */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Advantech"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "MICA-071"), + }, + /* OVCD Th = 1500uA to reliable detect head-phones vs -set */ + .driver_data = (void *)(BYT_RT5640_IN3_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_1500UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_MCLK_EN), + }, + { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ARCHOS 80 Cesium"), diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 6b2c8c6e7a00..5195e012dc6d 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1450,6 +1450,7 @@ int skl_platform_register(struct device *dev) dais = krealloc(skl->dais, sizeof(skl_fe_dai) + sizeof(skl_platform_dai), GFP_KERNEL); if (!dais) { + kfree(skl->dais); ret = -ENOMEM; goto err; } @@ -1462,8 +1463,10 @@ int skl_platform_register(struct device *dev) ret = devm_snd_soc_register_component(dev, &skl_component, skl->dais, num_dais); - if (ret) + if (ret) { + kfree(skl->dais); dev_err(dev, "soc component registration failed %d\n", ret); + } err: return ret; } diff --git a/sound/soc/intel/skylake/skl-sst-utils.c b/sound/soc/intel/skylake/skl-sst-utils.c index 2ae405617876..9e1e9bac1790 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/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index 255cc45905b8..51f75523b691 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -90,7 +90,7 @@ kirkwood_dma_conf_mbus_windows(void __iomem *base, int win, /* try to find matching cs for current dma address */ for (i = 0; i < dram->num_cs; i++) { - const struct mbus_dram_window *cs = dram->cs + i; + const struct mbus_dram_window *cs = &dram->cs[i]; if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) { writel(cs->base & 0xffff0000, base + KIRKWOOD_AUDIO_WIN_BASE_REG(win)); diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c index cdb394071037..9f8d2a00a1cd 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c @@ -199,14 +199,16 @@ static int mt8173_rt5650_rt5514_dev_probe(struct platform_device *pdev) if (!mt8173_rt5650_rt5514_codecs[0].of_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto out; } mt8173_rt5650_rt5514_codecs[1].of_node = of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 1); if (!mt8173_rt5650_rt5514_codecs[1].of_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto out; } mt8173_rt5650_rt5514_codec_conf[0].of_node = mt8173_rt5650_rt5514_codecs[1].of_node; @@ -218,6 +220,7 @@ static int mt8173_rt5650_rt5514_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", __func__, ret); +out: of_node_put(platform_node); return ret; } diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c index 43e390f9358a..a195160b6820 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 4dce494dfbd3..ef9fda16ce13 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/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c index d2d4652de32c..5969aa66410d 100644 --- a/sound/soc/pxa/mmp-pcm.c +++ b/sound/soc/pxa/mmp-pcm.c @@ -90,7 +90,7 @@ static bool filter(struct dma_chan *chan, void *param) devname = kasprintf(GFP_KERNEL, "%s.%d", dma_data->dma_res->name, dma_data->ssp_id); - if ((strcmp(dev_name(chan->device->dev), devname) == 0) && + if (devname && (strcmp(dev_name(chan->device->dev), devname) == 0) && (chan->chan_id == dma_data->dma_res->start)) { found = true; } diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 69033e1a84e6..49481dadb923 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/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c index ad16c8310dd3..7dfd1e6b2c25 100644 --- a/sound/soc/rockchip/rockchip_pdm.c +++ b/sound/soc/rockchip/rockchip_pdm.c @@ -303,6 +303,7 @@ static int rockchip_pdm_runtime_resume(struct device *dev) ret = clk_prepare_enable(pdm->hclk); if (ret) { + clk_disable_unprepare(pdm->clk); dev_err(pdm->dev, "hclock enable failed %d\n", ret); return ret; } diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index a89fe9b6463b..5ac726da6015 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -89,6 +89,7 @@ static int __maybe_unused rk_spdif_runtime_resume(struct device *dev) ret = clk_prepare_enable(spdif->hclk); if (ret) { + clk_disable_unprepare(spdif->mclk); dev_err(spdif->dev, "hclk clock enable failed %d\n", ret); return ret; } diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 409d082e80d1..7745a3e9044f 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -944,7 +944,7 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) rtd->fe_compr = 1; if (rtd->dai_link->dpcm_playback) be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; - else if (rtd->dai_link->dpcm_capture) + if (rtd->dai_link->dpcm_capture) be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops)); } else { diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 2faf95d4bb75..e01f3bf3ef17 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -458,8 +458,15 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, return err; if (snd_soc_volsw_is_stereo(mc)) { + val2 = ucontrol->value.integer.value[1]; + + if (mc->platform_max && val2 > mc->platform_max) + return -EINVAL; + if (val2 > max) + return -EINVAL; + val_mask = mask << rshift; - val2 = (ucontrol->value.integer.value[1] + min) & mask; + val2 = (val2 + min) & mask; val2 = val2 << rshift; err = snd_soc_component_update_bits(component, reg2, val_mask, diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index c03b653bf6ff..1fabb285b016 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1266,6 +1266,8 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, return; be_substream = snd_soc_dpcm_get_substream(be, stream); + if (!be_substream) + return; list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) { if (dpcm->fe == fe) diff --git a/sound/synth/emux/emux_nrpn.c b/sound/synth/emux/emux_nrpn.c index 9729a15b6ae6..f4aa2706aeb6 100644 --- a/sound/synth/emux/emux_nrpn.c +++ b/sound/synth/emux/emux_nrpn.c @@ -363,6 +363,9 @@ int snd_emux_xg_control(struct snd_emux_port *port, struct snd_midi_channel *chan, int param) { + if (param >= ARRAY_SIZE(chan->control)) + return -EINVAL; + return send_converted_effect(xg_effects, ARRAY_SIZE(xg_effects), port, chan, param, chan->control[param], diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c index e883659ea6e7..19951e1dbbb0 100644 --- a/sound/usb/caiaq/input.c +++ b/sound/usb/caiaq/input.c @@ -817,6 +817,7 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *cdev) default: /* no input methods supported on this device */ + ret = -EINVAL; goto exit_free_idev; } diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index e428d8b36c00..56119a96d350 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/format.c b/sound/usb/format.c index 01ba7a939ac4..342d6edb06ad 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -53,8 +53,12 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, case UAC_VERSION_1: default: { struct uac_format_type_i_discrete_descriptor *fmt = _fmt; - if (format >= 64) - return 0; /* invalid format */ + if (format >= 64) { + usb_audio_info(chip, + "%u:%d: invalid format type 0x%llx is detected, processed as PCM\n", + fp->iface, fp->altsetting, format); + format = UAC_FORMAT_TYPE_I_PCM; + } sample_width = fmt->bBitResolution; sample_bytes = fmt->bSubframeSize; format = 1ULL << format; diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index 67d74218d861..2399d500b881 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -318,7 +318,8 @@ static void line6_data_received(struct urb *urb) for (;;) { done = line6_midibuf_read(mb, line6->buffer_message, - LINE6_MIDI_MESSAGE_MAXLEN); + LINE6_MIDI_MESSAGE_MAXLEN, + LINE6_MIDIBUF_READ_RX); if (done <= 0) break; diff --git a/sound/usb/line6/midi.c b/sound/usb/line6/midi.c index e2cf55c53ea8..6df1cf26e440 100644 --- a/sound/usb/line6/midi.c +++ b/sound/usb/line6/midi.c @@ -48,7 +48,8 @@ static void line6_midi_transmit(struct snd_rawmidi_substream *substream) int req, done; for (;;) { - req = min(line6_midibuf_bytes_free(mb), line6->max_packet_size); + req = min3(line6_midibuf_bytes_free(mb), line6->max_packet_size, + LINE6_FALLBACK_MAXPACKETSIZE); done = snd_rawmidi_transmit_peek(substream, chunk, req); if (done == 0) @@ -60,7 +61,8 @@ static void line6_midi_transmit(struct snd_rawmidi_substream *substream) for (;;) { done = line6_midibuf_read(mb, chunk, - LINE6_FALLBACK_MAXPACKETSIZE); + LINE6_FALLBACK_MAXPACKETSIZE, + LINE6_MIDIBUF_READ_TX); if (done == 0) break; diff --git a/sound/usb/line6/midibuf.c b/sound/usb/line6/midibuf.c index c931d48801eb..4622234723a6 100644 --- a/sound/usb/line6/midibuf.c +++ b/sound/usb/line6/midibuf.c @@ -13,6 +13,7 @@ #include "midibuf.h" + static int midibuf_message_length(unsigned char code) { int message_length; @@ -24,12 +25,7 @@ static int midibuf_message_length(unsigned char code) message_length = length[(code >> 4) - 8]; } else { - /* - Note that according to the MIDI specification 0xf2 is - the "Song Position Pointer", but this is used by Line 6 - to send sysex messages to the host. - */ - static const int length[] = { -1, 2, -1, 2, -1, -1, 1, 1, 1, 1, + static const int length[] = { -1, 2, 2, 2, -1, -1, 1, 1, 1, -1, 1, 1, 1, -1, 1, 1 }; message_length = length[code & 0x0f]; @@ -129,7 +125,7 @@ int line6_midibuf_write(struct midi_buffer *this, unsigned char *data, } int line6_midibuf_read(struct midi_buffer *this, unsigned char *data, - int length) + int length, int read_type) { int bytes_used; int length1, length2; @@ -152,9 +148,22 @@ int line6_midibuf_read(struct midi_buffer *this, unsigned char *data, length1 = this->size - this->pos_read; - /* check MIDI command length */ command = this->buf[this->pos_read]; + /* + PODxt always has status byte lower nibble set to 0010, + when it means to send 0000, so we correct if here so + that control/program changes come on channel 1 and + sysex message status byte is correct + */ + if (read_type == LINE6_MIDIBUF_READ_RX) { + if (command == 0xb2 || command == 0xc2 || command == 0xf2) { + unsigned char fixed = command & 0xf0; + this->buf[this->pos_read] = fixed; + command = fixed; + } + } + /* check MIDI command length */ if (command & 0x80) { midi_length = midibuf_message_length(command); this->command_prev = command; diff --git a/sound/usb/line6/midibuf.h b/sound/usb/line6/midibuf.h index 6ea21ffb6627..187f49c975c2 100644 --- a/sound/usb/line6/midibuf.h +++ b/sound/usb/line6/midibuf.h @@ -12,6 +12,9 @@ #ifndef MIDIBUF_H #define MIDIBUF_H +#define LINE6_MIDIBUF_READ_TX 0 +#define LINE6_MIDIBUF_READ_RX 1 + struct midi_buffer { unsigned char *buf; int size; @@ -27,7 +30,7 @@ extern void line6_midibuf_destroy(struct midi_buffer *mb); extern int line6_midibuf_ignore(struct midi_buffer *mb, int length); extern int line6_midibuf_init(struct midi_buffer *mb, int size, int split); extern int line6_midibuf_read(struct midi_buffer *mb, unsigned char *data, - int length); + int length, int read_type); extern void line6_midibuf_reset(struct midi_buffer *mb); extern int line6_midibuf_write(struct midi_buffer *mb, unsigned char *data, int length); diff --git a/sound/usb/line6/pod.c b/sound/usb/line6/pod.c index dff8e7d5f305..41cb655eb4a6 100644 --- a/sound/usb/line6/pod.c +++ b/sound/usb/line6/pod.c @@ -169,8 +169,9 @@ static struct line6_pcm_properties pod_pcm_properties = { .bytes_per_channel = 3 /* SNDRV_PCM_FMTBIT_S24_3LE */ }; + static const char pod_version_header[] = { - 0xf2, 0x7e, 0x7f, 0x06, 0x02 + 0xf0, 0x7e, 0x7f, 0x06, 0x02 }; /* forward declarations: */ diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index e72f744bc305..6c546f520f99 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 |