diff options
Diffstat (limited to 'sound')
27 files changed, 366 insertions, 120 deletions
diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 33e72c809e50..494b7b533366 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -465,7 +465,6 @@ static int snd_pcm_hw_param_near(struct snd_pcm_substream *pcm, v = snd_pcm_hw_param_last(pcm, params, var, dir); else v = snd_pcm_hw_param_first(pcm, params, var, dir); - snd_BUG_ON(v < 0); return v; } @@ -1370,8 +1369,11 @@ static ssize_t snd_pcm_oss_write1(struct snd_pcm_substream *substream, const cha if ((tmp = snd_pcm_oss_make_ready(substream)) < 0) return tmp; - mutex_lock(&runtime->oss.params_lock); while (bytes > 0) { + if (mutex_lock_interruptible(&runtime->oss.params_lock)) { + tmp = -ERESTARTSYS; + break; + } if (bytes < runtime->oss.period_bytes || runtime->oss.buffer_used > 0) { tmp = bytes; if (tmp + runtime->oss.buffer_used > runtime->oss.period_bytes) @@ -1415,14 +1417,18 @@ static ssize_t snd_pcm_oss_write1(struct snd_pcm_substream *substream, const cha xfer += tmp; if ((substream->f_flags & O_NONBLOCK) != 0 && tmp != runtime->oss.period_bytes) - break; + tmp = -EAGAIN; } - } - mutex_unlock(&runtime->oss.params_lock); - return xfer; - err: - mutex_unlock(&runtime->oss.params_lock); + mutex_unlock(&runtime->oss.params_lock); + if (tmp < 0) + break; + if (signal_pending(current)) { + tmp = -ERESTARTSYS; + break; + } + tmp = 0; + } return xfer > 0 ? (snd_pcm_sframes_t)xfer : tmp; } @@ -1470,8 +1476,11 @@ static ssize_t snd_pcm_oss_read1(struct snd_pcm_substream *substream, char __use if ((tmp = snd_pcm_oss_make_ready(substream)) < 0) return tmp; - mutex_lock(&runtime->oss.params_lock); while (bytes > 0) { + if (mutex_lock_interruptible(&runtime->oss.params_lock)) { + tmp = -ERESTARTSYS; + break; + } if (bytes < runtime->oss.period_bytes || runtime->oss.buffer_used > 0) { if (runtime->oss.buffer_used == 0) { tmp = snd_pcm_oss_read2(substream, runtime->oss.buffer, runtime->oss.period_bytes, 1); @@ -1502,12 +1511,16 @@ static ssize_t snd_pcm_oss_read1(struct snd_pcm_substream *substream, char __use bytes -= tmp; xfer += tmp; } - } - mutex_unlock(&runtime->oss.params_lock); - return xfer; - err: - mutex_unlock(&runtime->oss.params_lock); + mutex_unlock(&runtime->oss.params_lock); + if (tmp < 0) + break; + if (signal_pending(current)) { + tmp = -ERESTARTSYS; + break; + } + tmp = 0; + } return xfer > 0 ? (snd_pcm_sframes_t)xfer : tmp; } diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 727ac44d39f4..a84a1d3d23e5 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -591,18 +591,26 @@ snd_pcm_sframes_t snd_pcm_plug_write_transfer(struct snd_pcm_substream *plug, st snd_pcm_sframes_t frames = size; plugin = snd_pcm_plug_first(plug); - while (plugin && frames > 0) { + while (plugin) { + if (frames <= 0) + return frames; if ((next = plugin->next) != NULL) { snd_pcm_sframes_t frames1 = frames; - if (plugin->dst_frames) + if (plugin->dst_frames) { frames1 = plugin->dst_frames(plugin, frames); + if (frames1 <= 0) + return frames1; + } if ((err = next->client_channels(next, frames1, &dst_channels)) < 0) { return err; } if (err != frames1) { frames = err; - if (plugin->src_frames) + if (plugin->src_frames) { frames = plugin->src_frames(plugin, frames1); + if (frames <= 0) + return frames; + } } } else dst_channels = NULL; diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index cd20f91326fe..4c145d6bccd4 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -578,7 +578,6 @@ static inline unsigned int muldiv32(unsigned int a, unsigned int b, { u_int64_t n = (u_int64_t) a * b; if (c == 0) { - snd_BUG_ON(!n); *r = 0; return UINT_MAX; } @@ -1664,7 +1663,7 @@ int snd_pcm_hw_param_first(struct snd_pcm_substream *pcm, return changed; if (params->rmask) { int err = snd_pcm_hw_refine(pcm, params); - if (snd_BUG_ON(err < 0)) + if (err < 0) return err; } return snd_pcm_hw_param_value(params, var, dir); @@ -1711,7 +1710,7 @@ int snd_pcm_hw_param_last(struct snd_pcm_substream *pcm, return changed; if (params->rmask) { int err = snd_pcm_hw_refine(pcm, params); - if (snd_BUG_ON(err < 0)) + if (err < 0) return err; } return snd_pcm_hw_param_value(params, var, dir); diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index b450a27588c8..16f8124b1150 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -579,15 +579,14 @@ static int snd_rawmidi_info_user(struct snd_rawmidi_substream *substream, return 0; } -int snd_rawmidi_info_select(struct snd_card *card, struct snd_rawmidi_info *info) +static int __snd_rawmidi_info_select(struct snd_card *card, + struct snd_rawmidi_info *info) { struct snd_rawmidi *rmidi; struct snd_rawmidi_str *pstr; struct snd_rawmidi_substream *substream; - mutex_lock(®ister_mutex); rmidi = snd_rawmidi_search(card, info->device); - mutex_unlock(®ister_mutex); if (!rmidi) return -ENXIO; if (info->stream < 0 || info->stream > 1) @@ -603,6 +602,16 @@ int snd_rawmidi_info_select(struct snd_card *card, struct snd_rawmidi_info *info } return -ENXIO; } + +int snd_rawmidi_info_select(struct snd_card *card, struct snd_rawmidi_info *info) +{ + int ret; + + mutex_lock(®ister_mutex); + ret = __snd_rawmidi_info_select(card, info); + mutex_unlock(®ister_mutex); + return ret; +} EXPORT_SYMBOL(snd_rawmidi_info_select); static int snd_rawmidi_info_select_user(struct snd_card *card, diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index b36de76f24e2..167b943469ab 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -236,6 +236,7 @@ static struct snd_seq_client *seq_create_client1(int client_index, int poolsize) rwlock_init(&client->ports_lock); mutex_init(&client->ports_mutex); INIT_LIST_HEAD(&client->ports_list_head); + mutex_init(&client->ioctl_mutex); /* find free slot in the client table */ spin_lock_irqsave(&clients_lock, flags); @@ -1011,7 +1012,7 @@ static ssize_t snd_seq_write(struct file *file, const char __user *buf, { struct snd_seq_client *client = file->private_data; int written = 0, len; - int err = -EINVAL; + int err; struct snd_seq_event event; if (!(snd_seq_file_flags(file) & SNDRV_SEQ_LFLG_OUTPUT)) @@ -1026,11 +1027,15 @@ static ssize_t snd_seq_write(struct file *file, const char __user *buf, /* allocate the pool now if the pool is not allocated yet */ if (client->pool->size > 0 && !snd_seq_write_pool_allocated(client)) { - if (snd_seq_pool_init(client->pool) < 0) + mutex_lock(&client->ioctl_mutex); + err = snd_seq_pool_init(client->pool); + mutex_unlock(&client->ioctl_mutex); + if (err < 0) return -ENOMEM; } /* only process whole events */ + err = -EINVAL; while (count >= sizeof(struct snd_seq_event)) { /* Read in the event header from the user */ len = sizeof(event); @@ -2220,11 +2225,15 @@ static int snd_seq_do_ioctl(struct snd_seq_client *client, unsigned int cmd, static long snd_seq_ioctl(struct file *file, unsigned int cmd, unsigned long arg) { struct snd_seq_client *client = file->private_data; + long ret; if (snd_BUG_ON(!client)) return -ENXIO; - return snd_seq_do_ioctl(client, cmd, (void __user *) arg); + mutex_lock(&client->ioctl_mutex); + ret = snd_seq_do_ioctl(client, cmd, (void __user *) arg); + mutex_unlock(&client->ioctl_mutex); + return ret; } #ifdef CONFIG_COMPAT diff --git a/sound/core/seq/seq_clientmgr.h b/sound/core/seq/seq_clientmgr.h index 20f0a725ec7d..91f8f165bfdc 100644 --- a/sound/core/seq/seq_clientmgr.h +++ b/sound/core/seq/seq_clientmgr.h @@ -59,6 +59,7 @@ struct snd_seq_client { struct list_head ports_list_head; rwlock_t ports_lock; struct mutex ports_mutex; + struct mutex ioctl_mutex; int convert32; /* convert 32->64bit */ /* output pool */ diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 54f348a4fb78..cbd20cb8ca11 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -39,6 +39,7 @@ #include <sound/core.h> #include <sound/control.h> #include <sound/pcm.h> +#include <sound/pcm_params.h> #include <sound/info.h> #include <sound/initval.h> @@ -305,19 +306,6 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) return 0; } -static void params_change_substream(struct loopback_pcm *dpcm, - struct snd_pcm_runtime *runtime) -{ - struct snd_pcm_runtime *dst_runtime; - - if (dpcm == NULL || dpcm->substream == NULL) - return; - dst_runtime = dpcm->substream->runtime; - if (dst_runtime == NULL) - return; - dst_runtime->hw = dpcm->cable->hw; -} - static void params_change(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -329,10 +317,6 @@ static void params_change(struct snd_pcm_substream *substream) cable->hw.rate_max = runtime->rate; cable->hw.channels_min = runtime->channels; cable->hw.channels_max = runtime->channels; - params_change_substream(cable->streams[SNDRV_PCM_STREAM_PLAYBACK], - runtime); - params_change_substream(cable->streams[SNDRV_PCM_STREAM_CAPTURE], - runtime); } static int loopback_prepare(struct snd_pcm_substream *substream) @@ -620,26 +604,29 @@ static unsigned int get_cable_index(struct snd_pcm_substream *substream) static int rule_format(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { + struct loopback_pcm *dpcm = rule->private; + struct loopback_cable *cable = dpcm->cable; + struct snd_mask m; - struct snd_pcm_hardware *hw = rule->private; - struct snd_mask *maskp = hw_param_mask(params, rule->var); - - maskp->bits[0] &= (u_int32_t)hw->formats; - maskp->bits[1] &= (u_int32_t)(hw->formats >> 32); - memset(maskp->bits + 2, 0, (SNDRV_MASK_MAX-64) / 8); /* clear rest */ - if (! maskp->bits[0] && ! maskp->bits[1]) - return -EINVAL; - return 0; + snd_mask_none(&m); + mutex_lock(&dpcm->loopback->cable_lock); + m.bits[0] = (u_int32_t)cable->hw.formats; + m.bits[1] = (u_int32_t)(cable->hw.formats >> 32); + mutex_unlock(&dpcm->loopback->cable_lock); + return snd_mask_refine(hw_param_mask(params, rule->var), &m); } static int rule_rate(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - struct snd_pcm_hardware *hw = rule->private; + struct loopback_pcm *dpcm = rule->private; + struct loopback_cable *cable = dpcm->cable; struct snd_interval t; - t.min = hw->rate_min; - t.max = hw->rate_max; + mutex_lock(&dpcm->loopback->cable_lock); + t.min = cable->hw.rate_min; + t.max = cable->hw.rate_max; + mutex_unlock(&dpcm->loopback->cable_lock); t.openmin = t.openmax = 0; t.integer = 0; return snd_interval_refine(hw_param_interval(params, rule->var), &t); @@ -648,22 +635,44 @@ static int rule_rate(struct snd_pcm_hw_params *params, static int rule_channels(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - struct snd_pcm_hardware *hw = rule->private; + struct loopback_pcm *dpcm = rule->private; + struct loopback_cable *cable = dpcm->cable; struct snd_interval t; - t.min = hw->channels_min; - t.max = hw->channels_max; + mutex_lock(&dpcm->loopback->cable_lock); + t.min = cable->hw.channels_min; + t.max = cable->hw.channels_max; + mutex_unlock(&dpcm->loopback->cable_lock); t.openmin = t.openmax = 0; t.integer = 0; return snd_interval_refine(hw_param_interval(params, rule->var), &t); } +static void free_cable(struct snd_pcm_substream *substream) +{ + struct loopback *loopback = substream->private_data; + int dev = get_cable_index(substream); + struct loopback_cable *cable; + + cable = loopback->cables[substream->number][dev]; + if (!cable) + return; + if (cable->streams[!substream->stream]) { + /* other stream is still alive */ + cable->streams[substream->stream] = NULL; + } else { + /* free the cable */ + loopback->cables[substream->number][dev] = NULL; + kfree(cable); + } +} + static int loopback_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct loopback *loopback = substream->private_data; struct loopback_pcm *dpcm; - struct loopback_cable *cable; + struct loopback_cable *cable = NULL; int err = 0; int dev = get_cable_index(substream); @@ -682,7 +691,6 @@ static int loopback_open(struct snd_pcm_substream *substream) if (!cable) { cable = kzalloc(sizeof(*cable), GFP_KERNEL); if (!cable) { - kfree(dpcm); err = -ENOMEM; goto unlock; } @@ -700,19 +708,19 @@ static int loopback_open(struct snd_pcm_substream *substream) /* are cached -> they do not reflect the actual state */ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, - rule_format, &runtime->hw, + rule_format, dpcm, SNDRV_PCM_HW_PARAM_FORMAT, -1); if (err < 0) goto unlock; err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - rule_rate, &runtime->hw, + rule_rate, dpcm, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) goto unlock; err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - rule_channels, &runtime->hw, + rule_channels, dpcm, SNDRV_PCM_HW_PARAM_CHANNELS, -1); if (err < 0) goto unlock; @@ -724,6 +732,10 @@ static int loopback_open(struct snd_pcm_substream *substream) else runtime->hw = cable->hw; unlock: + if (err < 0) { + free_cable(substream); + kfree(dpcm); + } mutex_unlock(&loopback->cable_lock); return err; } @@ -732,20 +744,10 @@ static int loopback_close(struct snd_pcm_substream *substream) { struct loopback *loopback = substream->private_data; struct loopback_pcm *dpcm = substream->runtime->private_data; - struct loopback_cable *cable; - int dev = get_cable_index(substream); loopback_timer_stop(dpcm); mutex_lock(&loopback->cable_lock); - cable = loopback->cables[substream->number][dev]; - if (cable->streams[!substream->stream]) { - /* other stream is still alive */ - cable->streams[substream->stream] = NULL; - } else { - /* free the cable */ - loopback->cables[substream->number][dev] = NULL; - kfree(cable); - } + free_cable(substream); mutex_unlock(&loopback->cable_lock); return 0; } diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 8fef1b8d1fd8..bd7bcf428bcf 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -183,7 +183,7 @@ static int hdac_component_master_match(struct device *dev, void *data) */ int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops *aops) { - if (WARN_ON(!hdac_acomp)) + if (!hdac_acomp) return -ENODEV; hdac_acomp->audio_ops = aops; @@ -240,7 +240,8 @@ out_master_del: out_err: kfree(acomp); bus->audio_component = NULL; - dev_err(dev, "failed to add i915 component master (%d)\n", ret); + hdac_acomp = NULL; + dev_info(dev, "failed to add i915 component master (%d)\n", ret); return ret; } @@ -273,6 +274,7 @@ int snd_hdac_i915_exit(struct hdac_bus *bus) kfree(acomp); bus->audio_component = NULL; + hdac_acomp = NULL; return 0; } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e2e08fc73b50..e2212830df0c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -179,7 +179,7 @@ static const struct kernel_param_ops param_ops_xint = { }; #define param_check_xint param_check_int -static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; +static int power_save = -1; module_param(power_save, xint, 0644); MODULE_PARM_DESC(power_save, "Automatic power-saving timeout " "(in second, 0 = disable)."); @@ -2055,6 +2055,24 @@ out_free: return err; } +#ifdef CONFIG_PM +/* On some boards setting power_save to a non 0 value leads to clicking / + * popping sounds when ever we enter/leave powersaving mode. Ideally we would + * figure out how to avoid these sounds, but that is not always feasible. + * So we keep a list of devices where we disable powersaving as its known + * to causes problems on these devices. + */ +static struct snd_pci_quirk power_save_blacklist[] = { + /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ + SND_PCI_QUIRK(0x1849, 0x0c0c, "Asrock B85M-ITX", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ + SND_PCI_QUIRK(0x1043, 0x8733, "Asus Prime X370-Pro", 0), + /* https://bugzilla.kernel.org/show_bug.cgi?id=198611 */ + SND_PCI_QUIRK(0x17aa, 0x2227, "Lenovo X1 Carbon 3rd Gen", 0), + {} +}; +#endif /* CONFIG_PM */ + /* number of codec slots for each chipset: 0 = default slots (i.e. 4) */ static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] = { [AZX_DRIVER_NVIDIA] = 8, @@ -2067,6 +2085,7 @@ static int azx_probe_continue(struct azx *chip) struct hdac_bus *bus = azx_bus(chip); struct pci_dev *pci = chip->pci; int dev = chip->dev_index; + int val; int err; hda->probe_continued = 1; @@ -2088,9 +2107,11 @@ static int azx_probe_continue(struct azx *chip) * for other chips, still continue probing as other * codecs can be on the same link. */ - if (CONTROLLER_IN_GPU(pci)) + if (CONTROLLER_IN_GPU(pci)) { + dev_err(chip->card->dev, + "HSW/BDW HD-audio HDMI/DP requires binding with gfx driver\n"); goto out_free; - else + } else goto skip_i915; } @@ -2140,7 +2161,22 @@ static int azx_probe_continue(struct azx *chip) chip->running = 1; azx_add_card_list(chip); - snd_hda_set_power_save(&chip->bus, power_save * 1000); + + val = power_save; +#ifdef CONFIG_PM + if (val == -1) { + const struct snd_pci_quirk *q; + + val = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; + q = snd_pci_quirk_lookup(chip->pci, power_save_blacklist); + if (q && val) { + dev_info(chip->card->dev, "device %04x:%04x is on the power_save blacklist, forcing power_save to 0\n", + q->subvendor, q->subdevice); + val = 0; + } + } +#endif /* CONFIG_PM */ + snd_hda_set_power_save(&chip->bus, val * 1000); if (azx_has_pm_runtime(chip) || hda->use_vga_switcheroo) pm_runtime_put_noidle(&pci->dev); diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index c146d0de53d8..29e1ce2263bc 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1482,6 +1482,9 @@ static int dspio_scp(struct hda_codec *codec, } else if (ret_size != reply_data_size) { codec_dbg(codec, "RetLen and HdrLen .NE.\n"); return -EINVAL; + } else if (!reply) { + codec_dbg(codec, "NULL reply\n"); + return -EINVAL; } else { *reply_len = ret_size*sizeof(unsigned int); memcpy(reply, scp_reply.data, *reply_len); diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 80bbadc83721..d6e079f4ec09 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -408,6 +408,7 @@ static const struct snd_pci_quirk cs420x_fixup_tbl[] = { /*SND_PCI_QUIRK(0x8086, 0x7270, "IMac 27 Inch", CS420X_IMAC27),*/ /* codec SSID */ + SND_PCI_QUIRK(0x106b, 0x0600, "iMac 14,1", CS420X_IMAC27_122), SND_PCI_QUIRK(0x106b, 0x1c00, "MacBookPro 8,1", CS420X_MBP81), SND_PCI_QUIRK(0x106b, 0x2000, "iMac 12,2", CS420X_IMAC27_122), SND_PCI_QUIRK(0x106b, 0x2800, "MacBookPro 10,1", CS420X_MBP101), diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index ac5de4365e15..c92b7ba344ef 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -261,6 +261,7 @@ enum { CXT_FIXUP_HP_530, CXT_FIXUP_CAP_MIX_AMP_5047, CXT_FIXUP_MUTE_LED_EAPD, + CXT_FIXUP_HP_DOCK, CXT_FIXUP_HP_SPECTRE, CXT_FIXUP_HP_GATE_MIC, }; @@ -778,6 +779,14 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_mute_led_eapd, }, + [CXT_FIXUP_HP_DOCK] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x16, 0x21011020 }, /* line-out */ + { 0x18, 0x2181103f }, /* line-in */ + { } + } + }, [CXT_FIXUP_HP_SPECTRE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -839,6 +848,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_ASPIRE_DMIC), SND_PCI_QUIRK(0x1025, 0x054f, "Acer Aspire 4830T", CXT_FIXUP_ASPIRE_DMIC), + SND_PCI_QUIRK(0x103c, 0x8079, "HP EliteBook 840 G3", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), SND_PCI_QUIRK(0x103c, 0x8115, "HP Z1 Gen3", CXT_FIXUP_HP_GATE_MIC), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), @@ -872,6 +882,7 @@ static const struct hda_model_fixup cxt5066_fixup_models[] = { { .id = CXT_PINCFG_LEMOTE_A1205, .name = "lemote-a1205" }, { .id = CXT_FIXUP_OLPC_XO, .name = "olpc-xo" }, { .id = CXT_FIXUP_MUTE_LED_EAPD, .name = "mute-led-eapd" }, + { .id = CXT_FIXUP_HP_DOCK, .name = "hp-dock" }, {} }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e5730a7d0480..b302d056e5d3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3130,6 +3130,19 @@ static void alc269_fixup_pincfg_no_hp_to_lineout(struct hda_codec *codec, spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; } +static void alc269_fixup_pincfg_U7x7_headset_mic(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + unsigned int cfg_headphone = snd_hda_codec_get_pincfg(codec, 0x21); + unsigned int cfg_headset_mic = snd_hda_codec_get_pincfg(codec, 0x19); + + if (cfg_headphone && cfg_headset_mic == 0x411111f0) + snd_hda_codec_set_pincfg(codec, 0x19, + (cfg_headphone & ~AC_DEFCFG_DEVICE) | + (AC_JACK_MIC_IN << AC_DEFCFG_DEVICE_SHIFT)); +} + static void alc269_fixup_hweq(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -4782,6 +4795,7 @@ enum { ALC269_FIXUP_LIFEBOOK_EXTMIC, ALC269_FIXUP_LIFEBOOK_HP_PIN, ALC269_FIXUP_LIFEBOOK_NO_HP_TO_LINEOUT, + ALC255_FIXUP_LIFEBOOK_U7x7_HEADSET_MIC, ALC269_FIXUP_AMIC, ALC269_FIXUP_DMIC, ALC269VB_FIXUP_AMIC, @@ -4839,6 +4853,7 @@ enum { ALC286_FIXUP_HP_GPIO_LED, ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY, ALC280_FIXUP_HP_DOCK_PINS, + ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED, ALC280_FIXUP_HP_9480M, ALC288_FIXUP_DELL_HEADSET_MODE, ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, @@ -4971,6 +4986,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_pincfg_no_hp_to_lineout, }, + [ALC255_FIXUP_LIFEBOOK_U7x7_HEADSET_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_pincfg_U7x7_headset_mic, + }, [ALC269_FIXUP_AMIC] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -5377,6 +5396,16 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC280_FIXUP_HP_GPIO4 }, + [ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1b, 0x21011020 }, /* line-out */ + { 0x18, 0x2181103f }, /* line-in */ + { }, + }, + .chained = true, + .chain_id = ALC269_FIXUP_HP_GPIO_MIC1_LED + }, [ALC280_FIXUP_HP_9480M] = { .type = HDA_FIXUP_FUNC, .v.func = alc280_fixup_hp_9480m, @@ -5589,6 +5618,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x1028, 0x0798, "Dell Inspiron 17 7000 Gaming", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), + SND_PCI_QUIRK(0x1028, 0x082a, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), @@ -5629,7 +5659,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2256, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2257, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2259, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), - SND_PCI_QUIRK(0x103c, 0x225a, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x225a, "HP", ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2260, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2263, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2264, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), @@ -5675,6 +5705,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x159f, "Lifebook E780", ALC269_FIXUP_LIFEBOOK_NO_HP_TO_LINEOUT), SND_PCI_QUIRK(0x10cf, 0x15dc, "Lifebook T731", ALC269_FIXUP_LIFEBOOK_HP_PIN), SND_PCI_QUIRK(0x10cf, 0x1757, "Lifebook E752", ALC269_FIXUP_LIFEBOOK_HP_PIN), + SND_PCI_QUIRK(0x10cf, 0x1629, "Lifebook U7x7", ALC255_FIXUP_LIFEBOOK_U7x7_HEADSET_MIC), SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC), SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_BXBT2807_MIC), @@ -5794,6 +5825,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, .name = "headset-mode-no-hp-mic"}, {.id = ALC269_FIXUP_LENOVO_DOCK, .name = "lenovo-dock"}, {.id = ALC269_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"}, + {.id = ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED, .name = "hp-dock-gpio-mic1-led"}, {.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, {.id = ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, .name = "dell-headset-dock"}, {.id = ALC283_FIXUP_CHROME_BOOK, .name = "alc283-dac-wcaps"}, @@ -5942,6 +5974,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1b, 0x01011020}, {0x21, 0x02211010}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x90a60130}, + {0x14, 0x90170110}, + {0x1b, 0x01011020}, + {0x21, 0x0221101f}), + SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60160}, {0x14, 0x90170120}, {0x21, 0x02211030}), @@ -5958,6 +5995,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x14, 0x90170110}, {0x21, 0x02211020}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x90a60130}, + {0x14, 0x90170110}, + {0x14, 0x01011020}, + {0x21, 0x0221101f}), + SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, ALC256_STANDARD_PINS), SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4, {0x12, 0x90a60130}, @@ -6013,6 +6055,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x12, 0x90a60120}, {0x14, 0x90170110}, {0x21, 0x0321101f}), + SND_HDA_PIN_QUIRK(0x10ec0289, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0xb7a60130}, + {0x14, 0x90170110}, + {0x21, 0x04211020}), SND_HDA_PIN_QUIRK(0x10ec0290, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC290_STANDARD_PINS, {0x15, 0x04211040}, diff --git a/sound/soc/codecs/pcm512x-spi.c b/sound/soc/codecs/pcm512x-spi.c index 712ed6598c48..ebdf9bd5a64c 100644 --- a/sound/soc/codecs/pcm512x-spi.c +++ b/sound/soc/codecs/pcm512x-spi.c @@ -70,3 +70,7 @@ static struct spi_driver pcm512x_spi_driver = { }; module_spi_driver(pcm512x_spi_driver); + +MODULE_DESCRIPTION("ASoC PCM512x codec driver - SPI"); +MODULE_AUTHOR("Mark Brown <broonie@kernel.org>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index a5a4e9f75c57..a06395507225 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -232,7 +232,7 @@ static struct twl4030_codec_data *twl4030_get_pdata(struct snd_soc_codec *codec) struct twl4030_codec_data *pdata = dev_get_platdata(codec->dev); struct device_node *twl4030_codec_node = NULL; - twl4030_codec_node = of_find_node_by_name(codec->dev->parent->of_node, + twl4030_codec_node = of_get_child_by_name(codec->dev->parent->of_node, "codec"); if (!pdata && twl4030_codec_node) { @@ -241,9 +241,11 @@ static struct twl4030_codec_data *twl4030_get_pdata(struct snd_soc_codec *codec) GFP_KERNEL); if (!pdata) { dev_err(codec->dev, "Can not allocate memory\n"); + of_node_put(twl4030_codec_node); return NULL; } twl4030_setup_pdata_of(pdata, twl4030_codec_node); + of_node_put(twl4030_codec_node); } return pdata; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 95d2392303eb..7ca67613e0d4 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1408,12 +1408,6 @@ static int fsl_ssi_probe(struct platform_device *pdev) sizeof(fsl_ssi_ac97_dai)); fsl_ac97_data = ssi_private; - - ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); - if (ret) { - dev_err(&pdev->dev, "could not set AC'97 ops\n"); - return ret; - } } else { /* Initialize this copy of the CPU DAI driver structure */ memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template, @@ -1473,6 +1467,14 @@ static int fsl_ssi_probe(struct platform_device *pdev) return ret; } + if (fsl_ssi_is_ac97(ssi_private)) { + ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + if (ret) { + dev_err(&pdev->dev, "could not set AC'97 ops\n"); + goto error_ac97_ops; + } + } + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_ssi_component, &ssi_private->cpu_dai_drv, 1); if (ret) { @@ -1556,6 +1558,10 @@ error_sound_card: fsl_ssi_debugfs_remove(&ssi_private->dbg_stats); error_asoc_register: + if (fsl_ssi_is_ac97(ssi_private)) + snd_soc_set_ac97_ops(NULL); + +error_ac97_ops: if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index ff6fcd9f92f7..0b1b6fcb7500 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -343,13 +343,19 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, snprintf(prop, sizeof(prop), "%scpu", prefix); cpu = of_get_child_by_name(node, prop); + if (!cpu) { + ret = -EINVAL; + dev_err(dev, "%s: Can't find %s DT node\n", __func__, prop); + goto dai_link_of_err; + } + snprintf(prop, sizeof(prop), "%splat", prefix); plat = of_get_child_by_name(node, prop); snprintf(prop, sizeof(prop), "%scodec", prefix); codec = of_get_child_by_name(node, prop); - if (!cpu || !codec) { + if (!codec) { ret = -EINVAL; dev_err(dev, "%s: Can't find %s DT node\n", __func__, prop); goto dai_link_of_err; diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index d430ef5a4f38..79c29330c56a 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -24,7 +24,6 @@ config SND_SST_IPC_PCI config SND_SST_IPC_ACPI tristate select SND_SST_IPC - depends on ACPI config SND_SOC_INTEL_SST tristate @@ -91,7 +90,7 @@ config SND_SOC_INTEL_BROADWELL_MACH config SND_SOC_INTEL_BYTCR_RT5640_MACH tristate "ASoC Audio DSP Support for MID BYT Platform" - depends on X86 && I2C + depends on X86 && I2C && ACPI select SND_SOC_RT5640 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI @@ -103,7 +102,7 @@ config SND_SOC_INTEL_BYTCR_RT5640_MACH config SND_SOC_INTEL_CHT_BSW_RT5672_MACH tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5672 codec" - depends on X86_INTEL_LPSS && I2C + depends on X86_INTEL_LPSS && I2C && ACPI select SND_SOC_RT5670 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI @@ -115,7 +114,7 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH config SND_SOC_INTEL_CHT_BSW_RT5645_MACH tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5645/5650 codec" - depends on X86_INTEL_LPSS && I2C + depends on X86_INTEL_LPSS && I2C && ACPI select SND_SOC_RT5645 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 15c04e2eae34..976967675387 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -9,7 +9,7 @@ config SND_SOC_MEDIATEK config SND_SOC_MT8173_MAX98090 tristate "ASoC Audio driver for MT8173 with MAX98090 codec" - depends on SND_SOC_MEDIATEK + depends on SND_SOC_MEDIATEK && I2C select SND_SOC_MAX98090 help This adds ASoC driver for Mediatek MT8173 boards @@ -19,7 +19,7 @@ config SND_SOC_MT8173_MAX98090 config SND_SOC_MT8173_RT5650_RT5676 tristate "ASoC Audio driver for MT8173 with RT5650 RT5676 codecs" - depends on SND_SOC_MEDIATEK + depends on SND_SOC_MEDIATEK && I2C select SND_SOC_RT5645 select SND_SOC_RT5677 help diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index 5a806da89f42..5e2eb4cc5cf1 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -54,7 +54,7 @@ static const struct of_device_id rk_spdif_match[] = { }; MODULE_DEVICE_TABLE(of, rk_spdif_match); -static int rk_spdif_runtime_suspend(struct device *dev) +static int __maybe_unused rk_spdif_runtime_suspend(struct device *dev) { struct rk_spdif_dev *spdif = dev_get_drvdata(dev); @@ -64,7 +64,7 @@ static int rk_spdif_runtime_suspend(struct device *dev) return 0; } -static int rk_spdif_runtime_resume(struct device *dev) +static int __maybe_unused rk_spdif_runtime_resume(struct device *dev) { struct rk_spdif_dev *spdif = dev_get_drvdata(dev); int ret; @@ -316,26 +316,30 @@ static int rk_spdif_probe(struct platform_device *pdev) spdif->mclk = devm_clk_get(&pdev->dev, "mclk"); if (IS_ERR(spdif->mclk)) { dev_err(&pdev->dev, "Can't retrieve rk_spdif master clock\n"); - return PTR_ERR(spdif->mclk); + ret = PTR_ERR(spdif->mclk); + goto err_disable_hclk; } ret = clk_prepare_enable(spdif->mclk); if (ret) { dev_err(spdif->dev, "clock enable failed %d\n", ret); - return ret; + goto err_disable_clocks; } res = platform_get_resource(pdev, IORESOURCE_MEM, 0); regs = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(regs)) - return PTR_ERR(regs); + if (IS_ERR(regs)) { + ret = PTR_ERR(regs); + goto err_disable_clocks; + } spdif->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "hclk", regs, &rk_spdif_regmap_config); if (IS_ERR(spdif->regmap)) { dev_err(&pdev->dev, "Failed to initialise managed register map\n"); - return PTR_ERR(spdif->regmap); + ret = PTR_ERR(spdif->regmap); + goto err_disable_clocks; } spdif->playback_dma_data.addr = res->start + SPDIF_SMPDR; @@ -367,6 +371,10 @@ static int rk_spdif_probe(struct platform_device *pdev) err_pm_runtime: pm_runtime_disable(&pdev->dev); +err_disable_clocks: + clk_disable_unprepare(spdif->mclk); +err_disable_hclk: + clk_disable_unprepare(spdif->hclk); return ret; } diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 085329878525..5976e3992dd1 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -235,6 +235,7 @@ enum rsnd_mod_type { RSND_MOD_MIX, RSND_MOD_CTU, RSND_MOD_SRC, + RSND_MOD_SSIP, /* SSI parent */ RSND_MOD_SSI, RSND_MOD_MAX, }; @@ -365,6 +366,7 @@ struct rsnd_dai_stream { }; #define rsnd_io_to_mod(io, i) ((i) < RSND_MOD_MAX ? (io)->mod[(i)] : NULL) #define rsnd_io_to_mod_ssi(io) rsnd_io_to_mod((io), RSND_MOD_SSI) +#define rsnd_io_to_mod_ssip(io) rsnd_io_to_mod((io), RSND_MOD_SSIP) #define rsnd_io_to_mod_src(io) rsnd_io_to_mod((io), RSND_MOD_SRC) #define rsnd_io_to_mod_ctu(io) rsnd_io_to_mod((io), RSND_MOD_CTU) #define rsnd_io_to_mod_mix(io) rsnd_io_to_mod((io), RSND_MOD_MIX) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index c62a2947ac14..38aae96267c9 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -550,11 +550,16 @@ static int rsnd_ssi_dma_remove(struct rsnd_mod *mod, struct rsnd_priv *priv) { struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + struct rsnd_mod *pure_ssi_mod = rsnd_io_to_mod_ssi(io); struct device *dev = rsnd_priv_to_dev(priv); int irq = ssi->info->irq; rsnd_dma_quit(io, rsnd_mod_to_dma(mod)); + /* Do nothing if non SSI (= SSI parent, multi SSI) mod */ + if (pure_ssi_mod != mod) + return 0; + /* PIO will request IRQ again */ devm_free_irq(dev, irq, mod); diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index ba9fc099cf67..503aef8fcde2 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -164,3 +164,7 @@ static struct platform_driver snd_soc_mop500_driver = { }; module_platform_driver(snd_soc_mop500_driver); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("ASoC MOP500 board driver"); +MODULE_AUTHOR("Ola Lilja"); diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index f12c01dddc8d..d35ba7700f46 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -165,3 +165,8 @@ int ux500_pcm_unregister_platform(struct platform_device *pdev) return 0; } EXPORT_SYMBOL_GPL(ux500_pcm_unregister_platform); + +MODULE_AUTHOR("Ola Lilja"); +MODULE_AUTHOR("Roger Nilsson"); +MODULE_DESCRIPTION("ASoC UX500 driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 0ed9ae030ce1..c5447ff078b3 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -343,17 +343,20 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret) { struct snd_usb_audio *chip = cval->head.mixer->chip; - unsigned char buf[4 + 3 * sizeof(__u32)]; /* enough space for one range */ + /* enough space for one range */ + unsigned char buf[sizeof(__u16) + 3 * sizeof(__u32)]; unsigned char *val; - int idx = 0, ret, size; + int idx = 0, ret, val_size, size; __u8 bRequest; + val_size = uac2_ctl_value_size(cval->val_type); + if (request == UAC_GET_CUR) { bRequest = UAC2_CS_CUR; - size = uac2_ctl_value_size(cval->val_type); + size = val_size; } else { bRequest = UAC2_CS_RANGE; - size = sizeof(buf); + size = sizeof(__u16) + 3 * val_size; } memset(buf, 0, sizeof(buf)); @@ -386,16 +389,17 @@ error: val = buf + sizeof(__u16); break; case UAC_GET_MAX: - val = buf + sizeof(__u16) * 2; + val = buf + sizeof(__u16) + val_size; break; case UAC_GET_RES: - val = buf + sizeof(__u16) * 3; + val = buf + sizeof(__u16) + val_size * 2; break; default: return -EINVAL; } - *value_ret = convert_signed_value(cval, snd_usb_combine_bytes(val, sizeof(__u16))); + *value_ret = convert_signed_value(cval, + snd_usb_combine_bytes(val, val_size)); return 0; } @@ -2101,20 +2105,25 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, kctl->private_value = (unsigned long)namelist; kctl->private_free = usb_mixer_selector_elem_free; - nameid = uac_selector_unit_iSelector(desc); + /* check the static mapping table at first */ len = check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name)); - if (len) - ; - else if (nameid) - len = snd_usb_copy_string_desc(state, nameid, kctl->id.name, - sizeof(kctl->id.name)); - else - len = get_term_name(state, &state->oterm, - kctl->id.name, sizeof(kctl->id.name), 0); - if (!len) { - strlcpy(kctl->id.name, "USB", sizeof(kctl->id.name)); + /* no mapping ? */ + /* if iSelector is given, use it */ + nameid = uac_selector_unit_iSelector(desc); + if (nameid) + len = snd_usb_copy_string_desc(state, nameid, + kctl->id.name, + sizeof(kctl->id.name)); + /* ... or pick up the terminal name at next */ + if (!len) + len = get_term_name(state, &state->oterm, + kctl->id.name, sizeof(kctl->id.name), 0); + /* ... or use the fixed string "USB" as the last resort */ + if (!len) + strlcpy(kctl->id.name, "USB", sizeof(kctl->id.name)); + /* and add the proper suffix */ if (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR) append_ctl_name(kctl, " Clock Source"); else if ((state->oterm.type & 0xff00) == 0x0100) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 48afae053c56..8e8db4ddf365 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -348,6 +348,15 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, alts = &iface->altsetting[1]; goto add_sync_ep; + case USB_ID(0x1397, 0x0002): + ep = 0x81; + iface = usb_ifnum_to_if(dev, 1); + + if (!iface || iface->num_altsetting == 0) + return -EINVAL; + + alts = &iface->altsetting[1]; + goto add_sync_ep; } if (attr == USB_ENDPOINT_SYNC_ASYNC && altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC && diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 8a59d4782a0f..69bf5cf1e91e 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3277,4 +3277,51 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } }, +{ + /* + * Bower's & Wilkins PX headphones only support the 48 kHz sample rate + * even though it advertises more. The capture interface doesn't work + * even on windows. + */ + USB_DEVICE(0x19b5, 0x0021), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, + /* Capture */ + { + .ifnum = 1, + .type = QUIRK_IGNORE_INTERFACE, + }, + /* Playback */ + { + .ifnum = 2, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels = 2, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_FILL_MAX | + UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x03, + .ep_attr = USB_ENDPOINT_XFER_ISOC, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { + 48000 + } + } + }, + } + } +}, + #undef USB_DEVICE_VENDOR_SPEC |