diff options
author | Takashi Iwai <tiwai@suse.de> | 2009-01-17 12:01:12 +0100 |
---|---|---|
committer | Takashi Iwai <tiwai@suse.de> | 2009-01-17 12:01:12 +0100 |
commit | d1a020050c6ce1a0794ff73582ccf47e4db536f7 (patch) | |
tree | 1b7250410f24703cd77c76156e758db9887137aa /sound | |
parent | ALSA: rename "Device" to "Toshiba SB-0500" via quirks (diff) | |
parent | ALSA: usb-audio - Cache mixer values (diff) | |
download | linux-d1a020050c6ce1a0794ff73582ccf47e4db536f7.tar.xz linux-d1a020050c6ce1a0794ff73582ccf47e4db536f7.zip |
Merge branch 'topic/usb-mixer-cache' into next/usb-audio
Diffstat (limited to 'sound')
34 files changed, 790 insertions, 380 deletions
diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index ef6539eea579..35afd0c33be5 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -321,10 +321,6 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) { int ret; - ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, 0, "AC97", NULL); - if (ret < 0) - goto err; - if (cpu_is_pxa25x() || cpu_is_pxa27x()) { pxa_gpio_mode(GPIO31_SYNC_AC97_MD); pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD); @@ -339,7 +335,7 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) if (IS_ERR(ac97conf_clk)) { ret = PTR_ERR(ac97conf_clk); ac97conf_clk = NULL; - goto err_irq; + goto err_conf; } } @@ -347,19 +343,30 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) if (IS_ERR(ac97_clk)) { ret = PTR_ERR(ac97_clk); ac97_clk = NULL; - goto err_irq; + goto err_clk; } - return clk_enable(ac97_clk); + ret = clk_enable(ac97_clk); + if (ret) + goto err_clk2; + + ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, IRQF_DISABLED, "AC97", NULL); + if (ret < 0) + goto err_irq; + + return 0; err_irq: GCR |= GCR_ACLINK_OFF; +err_clk2: + clk_put(ac97_clk); + ac97_clk = NULL; +err_clk: if (ac97conf_clk) { clk_put(ac97conf_clk); ac97conf_clk = NULL; } - free_irq(IRQ_AC97, NULL); -err: +err_conf: return ret; } EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_probe); diff --git a/sound/core/sound.c b/sound/core/sound.c index 44a69bb8d4f0..7872a02f6ca9 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -152,6 +152,10 @@ static int __snd_open(struct inode *inode, struct file *file) } old_fops = file->f_op; file->f_op = fops_get(mptr->f_ops); + if (file->f_op == NULL) { + file->f_op = old_fops; + return -ENODEV; + } if (file->f_op->open) err = file->f_op->open(inode, file); if (err) { diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c index 549b4eba1496..9d98a6658ac9 100644 --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -84,7 +84,7 @@ static void snd_tea575x_set_freq(struct snd_tea575x *tea) * Linux Video interface */ -static int snd_tea575x_ioctl(struct inode *inode, struct file *file, +static long snd_tea575x_ioctl(struct file *file, unsigned int cmd, unsigned long data) { struct snd_tea575x *tea = video_drvdata(file); @@ -174,14 +174,14 @@ static void snd_tea575x_release(struct video_device *vfd) { } -static int snd_tea575x_exclusive_open(struct inode *inode, struct file *file) +static int snd_tea575x_exclusive_open(struct file *file) { struct snd_tea575x *tea = video_drvdata(file); return test_and_set_bit(0, &tea->in_use) ? -EBUSY : 0; } -static int snd_tea575x_exclusive_release(struct inode *inode, struct file *file) +static int snd_tea575x_exclusive_release(struct file *file) { struct snd_tea575x *tea = video_drvdata(file); diff --git a/sound/oss/aedsp16.c b/sound/oss/aedsp16.c index a0274f3dac08..3ee9900ffd7b 100644 --- a/sound/oss/aedsp16.c +++ b/sound/oss/aedsp16.c @@ -157,7 +157,7 @@ Started Fri Mar 17 16:13:18 MET 1995 - v0.1 (ALPHA, was an user-level program called AudioExcelDSP16.c) + v0.1 (ALPHA, was a user-level program called AudioExcelDSP16.c) - Initial code. v0.2 (ALPHA) - Cleanups. diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c index 4d45bd63718b..57d9f154c88b 100644 --- a/sound/oss/dmasound/dmasound_atari.c +++ b/sound/oss/dmasound/dmasound_atari.c @@ -851,8 +851,9 @@ static int __init AtaIrqInit(void) mfp.tim_dt_a = 1; /* Cause interrupt after first event. */ mfp.tim_ct_a = 8; /* Turn on event counting. */ /* Register interrupt handler. */ - request_irq(IRQ_MFP_TIMA, AtaInterrupt, IRQ_TYPE_SLOW, "DMA sound", - AtaInterrupt); + if (request_irq(IRQ_MFP_TIMA, AtaInterrupt, IRQ_TYPE_SLOW, "DMA sound", + AtaInterrupt)) + return 0; mfp.int_en_a |= 0x20; /* Turn interrupt on. */ mfp.int_mk_a |= 0x20; return 1; diff --git a/sound/oss/dmasound/dmasound_q40.c b/sound/oss/dmasound/dmasound_q40.c index 1855b14d90c3..99bcb21c2281 100644 --- a/sound/oss/dmasound/dmasound_q40.c +++ b/sound/oss/dmasound/dmasound_q40.c @@ -371,8 +371,9 @@ static void Q40Free(void *ptr, unsigned int size) static int __init Q40IrqInit(void) { /* Register interrupt handler. */ - request_irq(Q40_IRQ_SAMPLE, Q40StereoInterrupt, 0, - "DMA sound", Q40Interrupt); + if (request_irq(Q40_IRQ_SAMPLE, Q40StereoInterrupt, 0, + "DMA sound", Q40Interrupt)) + return 0; return(1); } @@ -401,6 +402,7 @@ static void Q40PlayNextFrame(int index) u_char *start; u_long size; u_char speed; + int error; /* used by Q40Play() if all doubts whether there really is something * to be played are already wiped out. @@ -419,11 +421,13 @@ static void Q40PlayNextFrame(int index) master_outb( 0,SAMPLE_ENABLE_REG); free_irq(Q40_IRQ_SAMPLE, Q40Interrupt); if (dmasound.soft.stereo) - request_irq(Q40_IRQ_SAMPLE, Q40StereoInterrupt, 0, - "Q40 sound", Q40Interrupt); + error = request_irq(Q40_IRQ_SAMPLE, Q40StereoInterrupt, 0, + "Q40 sound", Q40Interrupt); else - request_irq(Q40_IRQ_SAMPLE, Q40MonoInterrupt, 0, - "Q40 sound", Q40Interrupt); + error = request_irq(Q40_IRQ_SAMPLE, Q40MonoInterrupt, 0, + "Q40 sound", Q40Interrupt); + if (error && printk_ratelimit()) + pr_err("Couldn't register sound interrupt\n"); master_outb( speed, SAMPLE_RATE_REG); master_outb( 1,SAMPLE_CLEAR_REG); diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index e00421c0d8ba..960fd7970384 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -135,7 +135,6 @@ void snd_hda_detach_beep_device(struct hda_codec *codec) struct hda_beep *beep = codec->beep; if (beep) { cancel_work_sync(&beep->beep_work); - flush_scheduled_work(); input_unregister_device(beep->dev); kfree(beep); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index e16cf63821ae..3c596da2b9b5 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -373,7 +373,7 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex) unsol->queue[wp] = res; unsol->queue[wp + 1] = res_ex; - schedule_work(&unsol->work); + queue_work(bus->workq, &unsol->work); return 0; } @@ -437,15 +437,17 @@ static int snd_hda_bus_free(struct hda_bus *bus) if (!bus) return 0; - if (bus->unsol) { - flush_scheduled_work(); + if (bus->workq) + flush_workqueue(bus->workq); + if (bus->unsol) kfree(bus->unsol); - } list_for_each_entry_safe(codec, n, &bus->codec_list, list) { snd_hda_codec_free(codec); } if (bus->ops.private_free) bus->ops.private_free(bus); + if (bus->workq) + destroy_workqueue(bus->workq); kfree(bus); return 0; } @@ -485,6 +487,7 @@ int /*__devinit*/ snd_hda_bus_new(struct snd_card *card, { struct hda_bus *bus; int err; + char qname[8]; static struct snd_device_ops dev_ops = { .dev_register = snd_hda_bus_dev_register, .dev_free = snd_hda_bus_dev_free, @@ -514,6 +517,14 @@ int /*__devinit*/ snd_hda_bus_new(struct snd_card *card, mutex_init(&bus->cmd_mutex); INIT_LIST_HEAD(&bus->codec_list); + snprintf(qname, sizeof(qname), "hda%d", card->number); + bus->workq = create_workqueue(qname); + if (!bus->workq) { + snd_printk(KERN_ERR "cannot create workqueue %s\n", qname); + kfree(bus); + return -ENOMEM; + } + err = snd_device_new(card, SNDRV_DEV_BUS, bus, &dev_ops); if (err < 0) { snd_hda_bus_free(bus); @@ -684,7 +695,7 @@ static void snd_hda_codec_free(struct hda_codec *codec) return; #ifdef CONFIG_SND_HDA_POWER_SAVE cancel_delayed_work(&codec->power_work); - flush_scheduled_work(); + flush_workqueue(codec->bus->workq); #endif list_del(&codec->list); snd_array_free(&codec->mixers); @@ -735,6 +746,7 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr codec->bus = bus; codec->addr = codec_addr; mutex_init(&codec->spdif_mutex); + mutex_init(&codec->control_mutex); init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); snd_array_init(&codec->mixers, sizeof(struct snd_kcontrol *), 32); @@ -1272,7 +1284,7 @@ void snd_hda_codec_reset(struct hda_codec *codec) #ifdef CONFIG_SND_HDA_POWER_SAVE cancel_delayed_work(&codec->power_work); - flush_scheduled_work(); + flush_workqueue(codec->bus->workq); #endif snd_hda_ctls_clear(codec); /* relase PCMs */ @@ -1418,12 +1430,12 @@ int snd_hda_mixer_bind_switch_get(struct snd_kcontrol *kcontrol, unsigned long pval; int err; - mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->control_mutex); pval = kcontrol->private_value; kcontrol->private_value = pval & ~AMP_VAL_IDX_MASK; /* index 0 */ err = snd_hda_mixer_amp_switch_get(kcontrol, ucontrol); kcontrol->private_value = pval; - mutex_unlock(&codec->spdif_mutex); + mutex_unlock(&codec->control_mutex); return err; } EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_switch_get); @@ -1435,7 +1447,7 @@ int snd_hda_mixer_bind_switch_put(struct snd_kcontrol *kcontrol, unsigned long pval; int i, indices, err = 0, change = 0; - mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->control_mutex); pval = kcontrol->private_value; indices = (pval & AMP_VAL_IDX_MASK) >> AMP_VAL_IDX_SHIFT; for (i = 0; i < indices; i++) { @@ -1447,7 +1459,7 @@ int snd_hda_mixer_bind_switch_put(struct snd_kcontrol *kcontrol, change |= err; } kcontrol->private_value = pval; - mutex_unlock(&codec->spdif_mutex); + mutex_unlock(&codec->control_mutex); return err < 0 ? err : change; } EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_switch_put); @@ -1462,12 +1474,12 @@ int snd_hda_mixer_bind_ctls_info(struct snd_kcontrol *kcontrol, struct hda_bind_ctls *c; int err; - mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->control_mutex); c = (struct hda_bind_ctls *)kcontrol->private_value; kcontrol->private_value = *c->values; err = c->ops->info(kcontrol, uinfo); kcontrol->private_value = (long)c; - mutex_unlock(&codec->spdif_mutex); + mutex_unlock(&codec->control_mutex); return err; } EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_ctls_info); @@ -1479,12 +1491,12 @@ int snd_hda_mixer_bind_ctls_get(struct snd_kcontrol *kcontrol, struct hda_bind_ctls *c; int err; - mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->control_mutex); c = (struct hda_bind_ctls *)kcontrol->private_value; kcontrol->private_value = *c->values; err = c->ops->get(kcontrol, ucontrol); kcontrol->private_value = (long)c; - mutex_unlock(&codec->spdif_mutex); + mutex_unlock(&codec->control_mutex); return err; } EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_ctls_get); @@ -1497,7 +1509,7 @@ int snd_hda_mixer_bind_ctls_put(struct snd_kcontrol *kcontrol, unsigned long *vals; int err = 0, change = 0; - mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->control_mutex); c = (struct hda_bind_ctls *)kcontrol->private_value; for (vals = c->values; *vals; vals++) { kcontrol->private_value = *vals; @@ -1507,7 +1519,7 @@ int snd_hda_mixer_bind_ctls_put(struct snd_kcontrol *kcontrol, change |= err; } kcontrol->private_value = (long)c; - mutex_unlock(&codec->spdif_mutex); + mutex_unlock(&codec->control_mutex); return err < 0 ? err : change; } EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_ctls_put); @@ -1519,12 +1531,12 @@ int snd_hda_mixer_bind_tlv(struct snd_kcontrol *kcontrol, int op_flag, struct hda_bind_ctls *c; int err; - mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->control_mutex); c = (struct hda_bind_ctls *)kcontrol->private_value; kcontrol->private_value = *c->values; err = c->ops->tlv(kcontrol, op_flag, size, tlv); kcontrol->private_value = (long)c; - mutex_unlock(&codec->spdif_mutex); + mutex_unlock(&codec->control_mutex); return err; } EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_tlv); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 729fc7642d7f..5810ef588402 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -614,6 +614,7 @@ struct hda_bus { /* unsolicited event queue */ struct hda_bus_unsolicited *unsol; + struct workqueue_struct *workq; /* common workqueue for codecs */ /* assigned PCMs */ DECLARE_BITMAP(pcm_dev_bits, SNDRV_PCM_DEVICES); @@ -771,6 +772,7 @@ struct hda_codec { struct hda_cache_rec cmd_cache; /* cache for other commands */ struct mutex spdif_mutex; + struct mutex control_mutex; unsigned int spdif_status; /* IEC958 status bits */ unsigned short spdif_ctls; /* SPDIF control bits */ unsigned int spdif_in_enable; /* SPDIF input enable? */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index f04de115ee11..11e791b965f6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -996,10 +996,11 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) spin_unlock(&chip->reg_lock); snd_pcm_period_elapsed(azx_dev->substream); spin_lock(&chip->reg_lock); - } else { + } else if (chip->bus && chip->bus->workq) { /* bogus IRQ, process it later */ azx_dev->irq_pending = 1; - schedule_work(&chip->irq_pending_work); + queue_work(chip->bus->workq, + &chip->irq_pending_work); } } } @@ -1741,7 +1742,6 @@ static void azx_clear_irq_pending(struct azx *chip) for (i = 0; i < chip->num_streams; i++) chip->azx_dev[i].irq_pending = 0; spin_unlock_irq(&chip->reg_lock); - flush_scheduled_work(); } static struct snd_pcm_ops azx_pcm_ops = { diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 26247cfe749d..2e7371ec2e23 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3900,6 +3900,7 @@ static const char *ad1884a_models[AD1884A_MODELS] = { static struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE), + SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE), SND_PCI_QUIRK(0x103c, 0x30e6, "HP 6730b", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30e7, "HP EliteBook 8530p", AD1884A_LAPTOP), @@ -4262,13 +4263,13 @@ static int patch_ad1882(struct hda_codec *codec) spec->num_adc_nids = ARRAY_SIZE(ad1882_adc_nids); spec->adc_nids = ad1882_adc_nids; spec->capsrc_nids = ad1882_capsrc_nids; - if (codec->vendor_id == 0x11d1882) + if (codec->vendor_id == 0x11d41882) spec->input_mux = &ad1882_capture_source; else spec->input_mux = &ad1882a_capture_source; spec->num_mixers = 2; spec->mixers[0] = ad1882_base_mixers; - if (codec->vendor_id == 0x11d1882) + if (codec->vendor_id == 0x11d41882) spec->mixers[1] = ad1882_loopback_mixers; else spec->mixers[1] = ad1882a_loopback_mixers; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index b20e1cede00b..75de40aaab0a 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -25,6 +25,8 @@ #include <linux/slab.h> #include <linux/pci.h> #include <sound/core.h> +#include <sound/jack.h> + #include "hda_codec.h" #include "hda_local.h" @@ -37,8 +39,21 @@ #define CONEXANT_HP_EVENT 0x37 #define CONEXANT_MIC_EVENT 0x38 +/* Conexant 5051 specific */ + +#define CXT5051_SPDIF_OUT 0x1C +#define CXT5051_PORTB_EVENT 0x38 +#define CXT5051_PORTC_EVENT 0x39 +struct conexant_jack { + + hda_nid_t nid; + int type; + struct snd_jack *jack; + +}; + struct conexant_spec { struct snd_kcontrol_new *mixers[5]; @@ -83,6 +98,9 @@ struct conexant_spec { unsigned int spdif_route; + /* jack detection */ + struct snd_array jacks; + /* dynamic controls, init_verbs and input_mux */ struct auto_pin_cfg autocfg; struct hda_input_mux private_imux; @@ -329,6 +347,86 @@ static int conexant_mux_enum_put(struct snd_kcontrol *kcontrol, &spec->cur_mux[adc_idx]); } +static int conexant_add_jack(struct hda_codec *codec, + hda_nid_t nid, int type) +{ + struct conexant_spec *spec; + struct conexant_jack *jack; + const char *name; + + spec = codec->spec; + snd_array_init(&spec->jacks, sizeof(*jack), 32); + jack = snd_array_new(&spec->jacks); + name = (type == SND_JACK_HEADPHONE) ? "Headphone" : "Mic" ; + + if (!jack) + return -ENOMEM; + + jack->nid = nid; + jack->type = type; + + return snd_jack_new(codec->bus->card, name, type, &jack->jack); +} + +static void conexant_report_jack(struct hda_codec *codec, hda_nid_t nid) +{ + struct conexant_spec *spec = codec->spec; + struct conexant_jack *jacks = spec->jacks.list; + + if (jacks) { + int i; + for (i = 0; i < spec->jacks.used; i++) { + if (jacks->nid == nid) { + unsigned int present; + present = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_SENSE, 0) & + AC_PINSENSE_PRESENCE; + + present = (present) ? jacks->type : 0 ; + + snd_jack_report(jacks->jack, + present); + } + jacks++; + } + } +} + +static int conexant_init_jacks(struct hda_codec *codec) +{ +#ifdef CONFIG_SND_JACK + struct conexant_spec *spec = codec->spec; + int i; + + for (i = 0; i < spec->num_init_verbs; i++) { + const struct hda_verb *hv; + + hv = spec->init_verbs[i]; + while (hv->nid) { + int err = 0; + switch (hv->param ^ AC_USRSP_EN) { + case CONEXANT_HP_EVENT: + err = conexant_add_jack(codec, hv->nid, + SND_JACK_HEADPHONE); + conexant_report_jack(codec, hv->nid); + break; + case CXT5051_PORTC_EVENT: + case CONEXANT_MIC_EVENT: + err = conexant_add_jack(codec, hv->nid, + SND_JACK_MICROPHONE); + conexant_report_jack(codec, hv->nid); + break; + } + if (err < 0) + return err; + ++hv; + } + } +#endif + return 0; + +} + static int conexant_init(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; @@ -341,6 +439,16 @@ static int conexant_init(struct hda_codec *codec) static void conexant_free(struct hda_codec *codec) { +#ifdef CONFIG_SND_JACK + struct conexant_spec *spec = codec->spec; + if (spec->jacks.list) { + struct conexant_jack *jacks = spec->jacks.list; + int i; + for (i = 0; i < spec->jacks.used; i++) + snd_device_free(codec->bus->card, &jacks[i].jack); + snd_array_free(&spec->jacks); + } +#endif kfree(codec->spec); } @@ -1526,9 +1634,6 @@ static int patch_cxt5047(struct hda_codec *codec) /* Conexant 5051 specific */ static hda_nid_t cxt5051_dac_nids[1] = { 0x10 }; static hda_nid_t cxt5051_adc_nids[2] = { 0x14, 0x15 }; -#define CXT5051_SPDIF_OUT 0x1C -#define CXT5051_PORTB_EVENT 0x38 -#define CXT5051_PORTC_EVENT 0x39 static struct hda_channel_mode cxt5051_modes[1] = { { 2, NULL }, @@ -1608,6 +1713,7 @@ static void cxt5051_hp_automute(struct hda_codec *codec) static void cxt5051_hp_unsol_event(struct hda_codec *codec, unsigned int res) { + int nid = (res & AC_UNSOL_RES_SUBTAG) >> 20; switch (res >> 26) { case CONEXANT_HP_EVENT: cxt5051_hp_automute(codec); @@ -1619,6 +1725,7 @@ static void cxt5051_hp_unsol_event(struct hda_codec *codec, cxt5051_portc_automic(codec); break; } + conexant_report_jack(codec, nid); } static struct snd_kcontrol_new cxt5051_mixers[] = { @@ -1693,6 +1800,7 @@ static struct hda_verb cxt5051_init_verbs[] = { static int cxt5051_init(struct hda_codec *codec) { conexant_init(codec); + conexant_init_jacks(codec); if (codec->patch_ops.unsol_event) { cxt5051_hp_automute(codec); cxt5051_portb_automic(codec); diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index 0270fda0bda5..96952a37d884 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -162,12 +162,14 @@ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = { { .id = 0x10de0002, .name = "MCP78 HDMI", .patch = patch_nvhdmi }, { .id = 0x10de0007, .name = "MCP7A HDMI", .patch = patch_nvhdmi }, { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi }, + { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi }, {} /* terminator */ }; MODULE_ALIAS("snd-hda-codec-id:10de0002"); MODULE_ALIAS("snd-hda-codec-id:10de0007"); MODULE_ALIAS("snd-hda-codec-id:10de0067"); +MODULE_ALIAS("snd-hda-codec-id:10de8001"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Nvidia HDMI HD-audio codec"); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0bd4e6bf354d..ea4c88fe05c4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1502,11 +1502,11 @@ static int alc_cap_vol_info(struct snd_kcontrol *kcontrol, struct alc_spec *spec = codec->spec; int err; - mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->control_mutex); kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[0], 3, 0, HDA_INPUT); err = snd_hda_mixer_amp_volume_info(kcontrol, uinfo); - mutex_unlock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_unlock(&codec->control_mutex); return err; } @@ -1517,11 +1517,11 @@ static int alc_cap_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, struct alc_spec *spec = codec->spec; int err; - mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->control_mutex); kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[0], 3, 0, HDA_INPUT); err = snd_hda_mixer_amp_tlv(kcontrol, op_flag, size, tlv); - mutex_unlock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_unlock(&codec->control_mutex); return err; } @@ -1537,11 +1537,11 @@ static int alc_cap_getput_caller(struct snd_kcontrol *kcontrol, unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); int err; - mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->control_mutex); kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[adc_idx], 3, 0, HDA_INPUT); err = func(kcontrol, ucontrol); - mutex_unlock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_unlock(&codec->control_mutex); return err; } @@ -8461,12 +8461,17 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x013e, "Acer Aspire 4930G", ALC888_ACER_ASPIRE_4930G), + SND_PCI_QUIRK(0x1025, 0x013f, "Acer Aspire 5930G", + ALC888_ACER_ASPIRE_4930G), + SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G", + ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), /* default Acer */ SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL), SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG), SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x103c, 0x2a66, "HP Acacia", ALC888_3ST_HP), SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V), SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x1043, 0x82fe, "Asus P5Q-EM HDMI", ALC1200_ASUS_P5Q), @@ -8521,6 +8526,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66), SND_PCI_QUIRK(0x8086, 0x0001, "DG33BUC", ALC883_3ST_6ch_INTEL), SND_PCI_QUIRK(0x8086, 0x0002, "DG33FBC", ALC883_3ST_6ch_INTEL), + SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC883_3ST_6ch_INTEL), SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch), {} }; @@ -11688,6 +11694,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x015b, "Acer Aspire One", ALC268_ACER_ASPIRE_ONE), SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL), + SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron Mini9", ALC268_DELL), SND_PCI_QUIRK(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA), SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA), @@ -16638,9 +16645,9 @@ static struct hda_codec_preset snd_hda_preset_realtek[] = { .patch = patch_alc882 }, /* should be patch_alc883() in future */ { .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 }, { .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc883 }, - { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc883 }, { .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200", .patch = patch_alc883 }, + { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc883 }, { .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc883 }, {} /* terminator */ }; diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 0dfa0540ce2c..bb8d8c766b9d 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -1239,7 +1239,7 @@ static int __devinit snd_vt1724_pcm_spdif(struct snd_ice1712 *ice, int device) if (ice->force_pdma4 || ice->force_rdma1) name = "ICE1724 Secondary"; else - name = "IEC1724 IEC958"; + name = "ICE1724 IEC958"; err = snd_pcm_new(ice->card, name, device, play, capt, &pcm); if (err < 0) return err; diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 1fb59a9d3719..6ea04be911d0 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -221,8 +221,8 @@ static int at91sam9g20ek_wm8731_init(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); /* not connected */ - snd_soc_dapm_disable_pin(codec, "RLINEIN"); - snd_soc_dapm_disable_pin(codec, "LLINEIN"); + snd_soc_dapm_nc_pin(codec, "RLINEIN"); + snd_soc_dapm_nc_pin(codec, "LLINEIN"); /* always connected */ snd_soc_dapm_enable_pin(codec, "Int Mic"); diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 74c823d60f91..bc8d654576c0 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -187,7 +187,7 @@ static int au1x_pcm_dbdma_realloc(struct au1xpsc_audio_dmadata *pcd, au1x_pcm_dmatx_cb, (void *)pcd); if (!pcd->ddma_chan) - return -ENOMEM;; + return -ENOMEM; au1xxx_dbdma_set_devwidth(pcd->ddma_chan, msbits); au1xxx_dbdma_ring_alloc(pcd->ddma_chan, 2); diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c41289b5f586..d0e0d691ae51 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1,3 +1,13 @@ +# Helper to resolve issues with configs that have SPI enabled but I2C +# modular, meaning we can't build the codec driver in with I2C support. +# We use an ordered list of conditional defaults to pick the appropriate +# setting - SPI can't be modular so that case doesn't need to be covered. +config SND_SOC_I2C_AND_SPI + tristate + default m if I2C=m + default y if I2C=y + default y if SPI_MASTER=y + config SND_SOC_ALL_CODECS tristate "Build all ASoC CODEC drivers" select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS @@ -14,12 +24,12 @@ config SND_SOC_ALL_CODECS select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C select SND_SOC_WM8350 if MFD_WM8350 - select SND_SOC_WM8510 if (I2C || SPI_MASTER) + select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8580 if I2C - select SND_SOC_WM8728 if (I2C || SPI_MASTER) - select SND_SOC_WM8731 if (I2C || SPI_MASTER) - select SND_SOC_WM8750 if (I2C || SPI_MASTER) - select SND_SOC_WM8753 if (I2C || SPI_MASTER) + select SND_SOC_WM8728 if SND_SOC_I2C_AND_SPI + select SND_SOC_WM8731 if SND_SOC_I2C_AND_SPI + select SND_SOC_WM8750 if SND_SOC_I2C_AND_SPI + select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8900 if I2C select SND_SOC_WM8903 if I2C select SND_SOC_WM8971 if I2C diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 51848880504a..ea370a4f86d5 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -192,39 +192,51 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) /* Earpiece */ static const char *twl4030_earpiece_texts[] = - {"Off", "DACL1", "DACL2", "Invalid", "DACR1"}; + {"Off", "DACL1", "DACL2", "DACR1"}; + +static const unsigned int twl4030_earpiece_values[] = + {0x0, 0x1, 0x2, 0x4}; static const struct soc_enum twl4030_earpiece_enum = - SOC_ENUM_SINGLE(TWL4030_REG_EAR_CTL, 1, + SOC_VALUE_ENUM_SINGLE(TWL4030_REG_EAR_CTL, 1, 0x7, ARRAY_SIZE(twl4030_earpiece_texts), - twl4030_earpiece_texts); + twl4030_earpiece_texts, + twl4030_earpiece_values); static const struct snd_kcontrol_new twl4030_dapm_earpiece_control = -SOC_DAPM_ENUM("Route", twl4030_earpiece_enum); +SOC_DAPM_VALUE_ENUM("Route", twl4030_earpiece_enum); /* PreDrive Left */ static const char *twl4030_predrivel_texts[] = - {"Off", "DACL1", "DACL2", "Invalid", "DACR2"}; + {"Off", "DACL1", "DACL2", "DACR2"}; + +static const unsigned int twl4030_predrivel_values[] = + {0x0, 0x1, 0x2, 0x4}; static const struct soc_enum twl4030_predrivel_enum = - SOC_ENUM_SINGLE(TWL4030_REG_PREDL_CTL, 1, + SOC_VALUE_ENUM_SINGLE(TWL4030_REG_PREDL_CTL, 1, 0x7, ARRAY_SIZE(twl4030_predrivel_texts), - twl4030_predrivel_texts); + twl4030_predrivel_texts, + twl4030_predrivel_values); static const struct snd_kcontrol_new twl4030_dapm_predrivel_control = -SOC_DAPM_ENUM("Route", twl4030_predrivel_enum); +SOC_DAPM_VALUE_ENUM("Route", twl4030_predrivel_enum); /* PreDrive Right */ static const char *twl4030_predriver_texts[] = - {"Off", "DACR1", "DACR2", "Invalid", "DACL2"}; + {"Off", "DACR1", "DACR2", "DACL2"}; + +static const unsigned int twl4030_predriver_values[] = + {0x0, 0x1, 0x2, 0x4}; static const struct soc_enum twl4030_predriver_enum = - SOC_ENUM_SINGLE(TWL4030_REG_PREDR_CTL, 1, + SOC_VALUE_ENUM_SINGLE(TWL4030_REG_PREDR_CTL, 1, 0x7, ARRAY_SIZE(twl4030_predriver_texts), - twl4030_predriver_texts); + twl4030_predriver_texts, + twl4030_predriver_values); static const struct snd_kcontrol_new twl4030_dapm_predriver_control = -SOC_DAPM_ENUM("Route", twl4030_predriver_enum); +SOC_DAPM_VALUE_ENUM("Route", twl4030_predriver_enum); /* Headset Left */ static const char *twl4030_hsol_texts[] = @@ -298,28 +310,90 @@ static const struct soc_enum twl4030_handsfreer_enum = static const struct snd_kcontrol_new twl4030_dapm_handsfreer_control = SOC_DAPM_ENUM("Route", twl4030_handsfreer_enum); -static int outmixer_event(struct snd_soc_dapm_widget *w, +/* Left analog microphone selection */ +static const char *twl4030_analoglmic_texts[] = + {"Off", "Main mic", "Headset mic", "AUXL", "Carkit mic"}; + +static const unsigned int twl4030_analoglmic_values[] = + {0x0, 0x1, 0x2, 0x4, 0x8}; + +static const struct soc_enum twl4030_analoglmic_enum = + SOC_VALUE_ENUM_SINGLE(TWL4030_REG_ANAMICL, 0, 0xf, + ARRAY_SIZE(twl4030_analoglmic_texts), + twl4030_analoglmic_texts, + twl4030_analoglmic_values); + +static const struct snd_kcontrol_new twl4030_dapm_analoglmic_control = +SOC_DAPM_VALUE_ENUM("Route", twl4030_analoglmic_enum); + +/* Right analog microphone selection */ +static const char *twl4030_analogrmic_texts[] = + {"Off", "Sub mic", "AUXR"}; + +static const unsigned int twl4030_analogrmic_values[] = + {0x0, 0x1, 0x4}; + +static const struct soc_enum twl4030_analogrmic_enum = + SOC_VALUE_ENUM_SINGLE(TWL4030_REG_ANAMICR, 0, 0x5, + ARRAY_SIZE(twl4030_analogrmic_texts), + twl4030_analogrmic_texts, + twl4030_analogrmic_values); + +static const struct snd_kcontrol_new twl4030_dapm_analogrmic_control = +SOC_DAPM_VALUE_ENUM("Route", twl4030_analogrmic_enum); + +/* TX1 L/R Analog/Digital microphone selection */ +static const char *twl4030_micpathtx1_texts[] = + {"Analog", "Digimic0"}; + +static const struct soc_enum twl4030_micpathtx1_enum = + SOC_ENUM_SINGLE(TWL4030_REG_ADCMICSEL, 0, + ARRAY_SIZE(twl4030_micpathtx1_texts), + twl4030_micpathtx1_texts); + +static const struct snd_kcontrol_new twl4030_dapm_micpathtx1_control = +SOC_DAPM_ENUM("Route", twl4030_micpathtx1_enum); + +/* TX2 L/R Analog/Digital microphone selection */ +static const char *twl4030_micpathtx2_texts[] = + {"Analog", "Digimic1"}; + +static const struct soc_enum twl4030_micpathtx2_enum = + SOC_ENUM_SINGLE(TWL4030_REG_ADCMICSEL, 2, + ARRAY_SIZE(twl4030_micpathtx2_texts), + twl4030_micpathtx2_texts); + +static const struct snd_kcontrol_new twl4030_dapm_micpathtx2_control = +SOC_DAPM_ENUM("Route", twl4030_micpathtx2_enum); + +static int micpath_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - int ret = 0; - int val; - - switch (e->reg) { - case TWL4030_REG_PREDL_CTL: - case TWL4030_REG_PREDR_CTL: - case TWL4030_REG_EAR_CTL: - val = w->value >> e->shift_l; - if (val == 3) { - printk(KERN_WARNING - "Invalid MUX setting for register 0x%02x (%d)\n", - e->reg, val); - ret = -1; - } - break; + struct soc_enum *e = (struct soc_enum *)w->kcontrols->private_value; + unsigned char adcmicsel, micbias_ctl; + + adcmicsel = twl4030_read_reg_cache(w->codec, TWL4030_REG_ADCMICSEL); + micbias_ctl = twl4030_read_reg_cache(w->codec, TWL4030_REG_MICBIAS_CTL); + /* Prepare the bits for the given TX path: + * shift_l == 0: TX1 microphone path + * shift_l == 2: TX2 microphone path */ + if (e->shift_l) { + /* TX2 microphone path */ + if (adcmicsel & TWL4030_TX2IN_SEL) + micbias_ctl |= TWL4030_MICBIAS2_CTL; /* digimic */ + else + micbias_ctl &= ~TWL4030_MICBIAS2_CTL; + } else { + /* TX1 microphone path */ + if (adcmicsel & TWL4030_TX1IN_SEL) + micbias_ctl |= TWL4030_MICBIAS1_CTL; /* digimic */ + else + micbias_ctl &= ~TWL4030_MICBIAS1_CTL; } - return ret; + twl4030_write(w->codec, TWL4030_REG_MICBIAS_CTL, micbias_ctl); + + return 0; } static int handsfree_event(struct snd_soc_dapm_widget *w, @@ -503,162 +577,6 @@ static int snd_soc_put_volsw_r2_twl4030(struct snd_kcontrol *kcontrol, return err; } -static int twl4030_get_left_input(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = kcontrol->private_data; - u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL); - int result = 0; - - /* one bit must be set a time */ - reg &= TWL4030_CKMIC_EN | TWL4030_AUXL_EN | TWL4030_HSMIC_EN - | TWL4030_MAINMIC_EN; - if (reg != 0) { - result++; - while ((reg & 1) == 0) { - result++; - reg >>= 1; - } - } - - ucontrol->value.integer.value[0] = result; - return 0; -} - -static int twl4030_put_left_input(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = kcontrol->private_data; - int value = ucontrol->value.integer.value[0]; - u8 anamicl, micbias, avadc_ctl; - - anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL); - anamicl &= ~(TWL4030_CKMIC_EN | TWL4030_AUXL_EN | TWL4030_HSMIC_EN - | TWL4030_MAINMIC_EN); - micbias = twl4030_read_reg_cache(codec, TWL4030_REG_MICBIAS_CTL); - micbias &= ~(TWL4030_HSMICBIAS_EN | TWL4030_MICBIAS1_EN); - avadc_ctl = twl4030_read_reg_cache(codec, TWL4030_REG_AVADC_CTL); - - switch (value) { - case 1: - anamicl |= TWL4030_MAINMIC_EN; - micbias |= TWL4030_MICBIAS1_EN; - break; - case 2: - anamicl |= TWL4030_HSMIC_EN; - micbias |= TWL4030_HSMICBIAS_EN; - break; - case 3: - anamicl |= TWL4030_AUXL_EN; - break; - case 4: - anamicl |= TWL4030_CKMIC_EN; - break; - default: - break; - } - - /* If some input is selected, enable amp and ADC */ - if (value != 0) { - anamicl |= TWL4030_MICAMPL_EN; - avadc_ctl |= TWL4030_ADCL_EN; - } else { - anamicl &= ~TWL4030_MICAMPL_EN; - avadc_ctl &= ~TWL4030_ADCL_EN; - } - - twl4030_write(codec, TWL4030_REG_ANAMICL, anamicl); - twl4030_write(codec, TWL4030_REG_MICBIAS_CTL, micbias); - twl4030_write(codec, TWL4030_REG_AVADC_CTL, avadc_ctl); - - return 1; -} - -static int twl4030_get_right_input(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = kcontrol->private_data; - u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICR); - int value = 0; - - reg &= TWL4030_SUBMIC_EN|TWL4030_AUXR_EN; - switch (reg) { - case TWL4030_SUBMIC_EN: - value = 1; - break; - case TWL4030_AUXR_EN: - value = 2; - break; - default: - break; - } - - ucontrol->value.integer.value[0] = value; - return 0; -} - -static int twl4030_put_right_input(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = kcontrol->private_data; - int value = ucontrol->value.integer.value[0]; - u8 anamicr, micbias, avadc_ctl; - - anamicr = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICR); - anamicr &= ~(TWL4030_SUBMIC_EN|TWL4030_AUXR_EN); - micbias = twl4030_read_reg_cache(codec, TWL4030_REG_MICBIAS_CTL); - micbias &= ~TWL4030_MICBIAS2_EN; - avadc_ctl = twl4030_read_reg_cache(codec, TWL4030_REG_AVADC_CTL); - - switch (value) { - case 1: - anamicr |= TWL4030_SUBMIC_EN; - micbias |= TWL4030_MICBIAS2_EN; - break; - case 2: - anamicr |= TWL4030_AUXR_EN; - break; - default: - break; - } - - if (value != 0) { - anamicr |= TWL4030_MICAMPR_EN; - avadc_ctl |= TWL4030_ADCR_EN; - } else { - anamicr &= ~TWL4030_MICAMPR_EN; - avadc_ctl &= ~TWL4030_ADCR_EN; - } - - twl4030_write(codec, TWL4030_REG_ANAMICR, anamicr); - twl4030_write(codec, TWL4030_REG_MICBIAS_CTL, micbias); - twl4030_write(codec, TWL4030_REG_AVADC_CTL, avadc_ctl); - - return 1; -} - -static const char *twl4030_left_in_sel[] = { - "None", - "Main Mic", - "Headset Mic", - "Line In", - "Carkit Mic", -}; - -static const char *twl4030_right_in_sel[] = { - "None", - "Sub Mic", - "Line In", -}; - -static const struct soc_enum twl4030_left_input_mux = - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(twl4030_left_in_sel), - twl4030_left_in_sel); - -static const struct soc_enum twl4030_right_input_mux = - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(twl4030_right_in_sel), - twl4030_right_in_sel); - /* * FGAIN volume control: * from -62 to 0 dB in 1 dB steps (mute instead of -63 dB) @@ -741,18 +659,15 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { TWL4030_REG_EAR_CTL, 4, 3, 0, output_tvl), /* Common capture gain controls */ - SOC_DOUBLE_R_TLV("Capture Volume", + SOC_DOUBLE_R_TLV("TX1 Digital Capture Volume", TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA, 0, 0x1f, 0, digital_capture_tlv), + SOC_DOUBLE_R_TLV("TX2 Digital Capture Volume", + TWL4030_REG_AVTXL2PGA, TWL4030_REG_AVTXR2PGA, + 0, 0x1f, 0, digital_capture_tlv), - SOC_DOUBLE_TLV("Input Boost Volume", TWL4030_REG_ANAMIC_GAIN, + SOC_DOUBLE_TLV("Analog Capture Volume", TWL4030_REG_ANAMIC_GAIN, 0, 3, 5, 0, input_gain_tlv), - - /* Input source controls */ - SOC_ENUM_EXT("Left Input Source", twl4030_left_input_mux, - twl4030_get_left_input, twl4030_put_left_input), - SOC_ENUM_EXT("Right Input Source", twl4030_right_input_mux, - twl4030_get_right_input, twl4030_put_right_input), }; /* add non dapm controls */ @@ -772,9 +687,19 @@ static int twl4030_add_controls(struct snd_soc_codec *codec) } static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { - SND_SOC_DAPM_INPUT("INL"), - SND_SOC_DAPM_INPUT("INR"), - + /* Left channel inputs */ + SND_SOC_DAPM_INPUT("MAINMIC"), + SND_SOC_DAPM_INPUT("HSMIC"), + SND_SOC_DAPM_INPUT("AUXL"), + SND_SOC_DAPM_INPUT("CARKITMIC"), + /* Right channel inputs */ + SND_SOC_DAPM_INPUT("SUBMIC"), + SND_SOC_DAPM_INPUT("AUXR"), + /* Digital microphones (Stereo) */ + SND_SOC_DAPM_INPUT("DIGIMIC0"), + SND_SOC_DAPM_INPUT("DIGIMIC1"), + + /* Outputs */ SND_SOC_DAPM_OUTPUT("OUTL"), SND_SOC_DAPM_OUTPUT("OUTR"), SND_SOC_DAPM_OUTPUT("EARPIECE"), @@ -809,16 +734,13 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { /* Output MUX controls */ /* Earpiece */ - SND_SOC_DAPM_MUX_E("Earpiece Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_earpiece_control, outmixer_event, - SND_SOC_DAPM_PRE_REG), + SND_SOC_DAPM_VALUE_MUX("Earpiece Mux", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_earpiece_control), /* PreDrivL/R */ - SND_SOC_DAPM_MUX_E("PredriveL Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_predrivel_control, outmixer_event, - SND_SOC_DAPM_PRE_REG), - SND_SOC_DAPM_MUX_E("PredriveR Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_predriver_control, outmixer_event, - SND_SOC_DAPM_PRE_REG), + SND_SOC_DAPM_VALUE_MUX("PredriveL Mux", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_predrivel_control), + SND_SOC_DAPM_VALUE_MUX("PredriveR Mux", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_predriver_control), /* HeadsetL/R */ SND_SOC_DAPM_MUX("HeadsetL Mux", SND_SOC_NOPM, 0, 0, &twl4030_dapm_hsol_control), @@ -837,8 +759,48 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { &twl4030_dapm_handsfreer_control, handsfree_event, SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_ADC("ADCL", "Left Capture", SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_ADC("ADCR", "Right Capture", SND_SOC_NOPM, 0, 0), + /* Introducing four virtual ADC, since TWL4030 have four channel for + capture */ + SND_SOC_DAPM_ADC("ADC Virtual Left1", "Left Front Capture", + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC Virtual Right1", "Right Front Capture", + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC Virtual Left2", "Left Rear Capture", + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC Virtual Right2", "Right Rear Capture", + SND_SOC_NOPM, 0, 0), + + /* Analog/Digital mic path selection. + TX1 Left/Right: either analog Left/Right or Digimic0 + TX2 Left/Right: either analog Left/Right or Digimic1 */ + SND_SOC_DAPM_MUX_E("TX1 Capture Route", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_micpathtx1_control, micpath_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD| + SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_MUX_E("TX2 Capture Route", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_micpathtx2_control, micpath_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD| + SND_SOC_DAPM_POST_REG), + + /* Analog input muxes with power switch for the physical ADCL/R */ + SND_SOC_DAPM_VALUE_MUX("Analog Left Capture Route", + TWL4030_REG_AVADC_CTL, 3, 0, &twl4030_dapm_analoglmic_control), + SND_SOC_DAPM_VALUE_MUX("Analog Right Capture Route", + TWL4030_REG_AVADC_CTL, 1, 0, &twl4030_dapm_analogrmic_control), + + SND_SOC_DAPM_PGA("Analog Left Amplifier", + TWL4030_REG_ANAMICL, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Analog Right Amplifier", + TWL4030_REG_ANAMICR, 4, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Digimic0 Enable", + TWL4030_REG_ADCMICSEL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("Digimic1 Enable", + TWL4030_REG_ADCMICSEL, 3, 0, NULL, 0), + + SND_SOC_DAPM_MICBIAS("Mic Bias 1", TWL4030_REG_MICBIAS_CTL, 0, 0), + SND_SOC_DAPM_MICBIAS("Mic Bias 2", TWL4030_REG_MICBIAS_CTL, 1, 0), + SND_SOC_DAPM_MICBIAS("Headset Mic Bias", TWL4030_REG_MICBIAS_CTL, 2, 0), }; static const struct snd_soc_dapm_route intercon[] = { @@ -894,9 +856,39 @@ static const struct snd_soc_dapm_route intercon[] = { {"HFL", NULL, "HandsfreeL Mux"}, {"HFR", NULL, "HandsfreeR Mux"}, - /* inputs */ - {"ADCL", NULL, "INL"}, - {"ADCR", NULL, "INR"}, + /* Capture path */ + {"Analog Left Capture Route", "Main mic", "MAINMIC"}, + {"Analog Left Capture Route", "Headset mic", "HSMIC"}, + {"Analog Left Capture Route", "AUXL", "AUXL"}, + {"Analog Left Capture Route", "Carkit mic", "CARKITMIC"}, + + {"Analog Right Capture Route", "Sub mic", "SUBMIC"}, + {"Analog Right Capture Route", "AUXR", "AUXR"}, + + {"Analog Left Amplifier", NULL, "Analog Left Capture Route"}, + {"Analog Right Amplifier", NULL, "Analog Right Capture Route"}, + + {"Digimic0 Enable", NULL, "DIGIMIC0"}, + {"Digimic1 Enable", NULL, "DIGIMIC1"}, + + /* TX1 Left capture path */ + {"TX1 Capture Route", "Analog", "Analog Left Amplifier"}, + {"TX1 Capture Route", "Digimic0", "Digimic0 Enable"}, + /* TX1 Right capture path */ + {"TX1 Capture Route", "Analog", "Analog Right Amplifier"}, + {"TX1 Capture Route", "Digimic0", "Digimic0 Enable"}, + /* TX2 Left capture path */ + {"TX2 Capture Route", "Analog", "Analog Left Amplifier"}, + {"TX2 Capture Route", "Digimic1", "Digimic1 Enable"}, + /* TX2 Right capture path */ + {"TX2 Capture Route", "Analog", "Analog Right Amplifier"}, + {"TX2 Capture Route", "Digimic1", "Digimic1 Enable"}, + + {"ADC Virtual Left1", NULL, "TX1 Capture Route"}, + {"ADC Virtual Right1", NULL, "TX1 Capture Route"}, + {"ADC Virtual Left2", NULL, "TX2 Capture Route"}, + {"ADC Virtual Right2", NULL, "TX2 Capture Route"}, + }; static int twl4030_add_widgets(struct snd_soc_codec *codec) @@ -923,6 +915,7 @@ static void twl4030_power_up(struct snd_soc_codec *codec) twl4030_write(codec, TWL4030_REG_ANAMICL, anamicl | TWL4030_CNCL_OFFSET_START); + /* wait for offset cancellation to complete */ do { /* this takes a little while, so don't slam i2c */ @@ -1287,6 +1280,8 @@ static int twl4030_remove(struct platform_device *pdev) struct snd_soc_codec *codec = socdev->codec; printk(KERN_INFO "TWL4030 Audio Codec remove\n"); + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); kfree(codec); return 0; diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 54615c76802b..442e5a828617 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -147,6 +147,13 @@ #define TWL4030_AVADC_CLK_PRIORITY 0x04 #define TWL4030_ADCR_EN 0x02 +/* TWL4030_REG_ADCMICSEL (0x08) Fields */ + +#define TWL4030_DIGMIC1_EN 0x08 +#define TWL4030_TX2IN_SEL 0x04 +#define TWL4030_DIGMIC0_EN 0x02 +#define TWL4030_TX1IN_SEL 0x01 + /* AUDIO_IF (0x0E) Fields */ #define TWL4030_AIF_SLAVE_EN 0x80 diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 01b948bb55a1..54851f318568 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -26,7 +26,6 @@ #include "davinci-pcm.h" #include "davinci-i2s.h" -#define EVM_CODEC_CLOCK 22579200 #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF) @@ -37,6 +36,21 @@ static int evm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret = 0; + unsigned sysclk; + + /* ASP1 on DM355 EVM is clocked by an external oscillator */ + if (machine_is_davinci_dm355_evm()) + sysclk = 27000000; + + /* ASP0 in DM6446 EVM is clocked by U55, as configured by + * board-dm644x-evm.c using GPIOs from U18. There are six + * options; here we "know" we use a 48 KHz sample rate. + */ + else if (machine_is_davinci_evm()) + sysclk = 12288000; + + else + return -EINVAL; /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, AUDIO_FORMAT); @@ -49,8 +63,7 @@ static int evm_hw_params(struct snd_pcm_substream *substream, return ret; /* set the codec system clock */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, EVM_CODEC_CLOCK, - SND_SOC_CLOCK_OUT); + ret = snd_soc_dai_set_sysclk(codec_dai, 0, sysclk, SND_SOC_CLOCK_OUT); if (ret < 0) return ret; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 74abc9b4f1cc..366049d8578c 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -212,7 +212,7 @@ davinci_pcm_pointer(struct snd_pcm_substream *substream) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) count = src - runtime->dma_addr; else - count = dst - runtime->dma_addr;; + count = dst - runtime->dma_addr; spin_unlock(&prtd->lock); diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c index f67579d52765..4935d1bcbd8d 100644 --- a/sound/soc/davinci/davinci-sffsdr.c +++ b/sound/soc/davinci/davinci-sffsdr.c @@ -24,6 +24,7 @@ #include <sound/soc-dapm.h> #include <asm/dma.h> +#include <asm/mach-types.h> #include <asm/plat-sffsdr/sffsdr-fpga.h> #include <mach/mcbsp.h> @@ -115,6 +116,9 @@ static int __init sffsdr_init(void) { int ret; + if (!machine_is_sffsdr()) + return -EINVAL; + sffsdr_snd_device = platform_device_alloc("soc-audio", 0); if (!sffsdr_snd_device) { printk(KERN_ERR "platform device allocation failed\n"); diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index a7b1d77b2105..4f7f04014585 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -10,6 +10,7 @@ config SND_OMAP_SOC_N810 tristate "SoC Audio support for Nokia N810" depends on SND_OMAP_SOC && MACH_NOKIA_N810 select SND_OMAP_SOC_MCBSP + select OMAP_MUX select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on Nokia N810. diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index bd91594496b1..fcc2f5d9a878 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -180,6 +180,19 @@ static int omap3pandora_in_init(struct snd_soc_codec *codec) { int ret; + /* All TWL4030 output pins are floating */ + snd_soc_dapm_nc_pin(codec, "OUTL"), + snd_soc_dapm_nc_pin(codec, "OUTR"), + snd_soc_dapm_nc_pin(codec, "EARPIECE"), + snd_soc_dapm_nc_pin(codec, "PREDRIVEL"), + snd_soc_dapm_nc_pin(codec, "PREDRIVER"), + snd_soc_dapm_nc_pin(codec, "HSOL"), + snd_soc_dapm_nc_pin(codec, "HSOR"), + snd_soc_dapm_nc_pin(codec, "CARKITL"), + snd_soc_dapm_nc_pin(codec, "CARKITR"), + snd_soc_dapm_nc_pin(codec, "HFL"), + snd_soc_dapm_nc_pin(codec, "HFR"), + ret = snd_soc_dapm_new_controls(codec, omap3pandora_in_dapm_widgets, ARRAY_SIZE(omap3pandora_in_dapm_widgets)); if (ret < 0) diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index c670d08e7c9e..53b9fb127a6d 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -61,9 +61,9 @@ static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) __pxa2xx_pcm_hw_free(substream); - if (prtd->dma_ch) { + if (prtd->dma_ch >= 0) { pxa_free_dma(prtd->dma_ch); - prtd->dma_ch = 0; + prtd->dma_ch = -1; } return 0; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b098c0b4c584..55fdb4abb179 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1300,6 +1300,8 @@ EXPORT_SYMBOL_GPL(snd_soc_test_bits); /** * snd_soc_new_pcms - create new sound card and pcms * @socdev: the SoC audio device + * @idx: ALSA card index + * @xid: card identification * * Create a new sound card based upon the codec and interface pcms. * @@ -1472,7 +1474,7 @@ EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams); * snd_soc_cnew - create new control * @_template: control template * @data: control private data - * @lnng_name: control long name + * @long_name: control long name * * Create a new mixer control from a template control. * @@ -1522,7 +1524,7 @@ EXPORT_SYMBOL_GPL(snd_soc_info_enum_double); /** * snd_soc_get_enum_double - enumerated double mixer get callback * @kcontrol: mixer control - * @uinfo: control element information + * @ucontrol: control element information * * Callback to get the value of a double enumerated mixer. * @@ -1551,7 +1553,7 @@ EXPORT_SYMBOL_GPL(snd_soc_get_enum_double); /** * snd_soc_put_enum_double - enumerated double mixer put callback * @kcontrol: mixer control - * @uinfo: control element information + * @ucontrol: control element information * * Callback to set the value of a double enumerated mixer. * @@ -1583,6 +1585,80 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); /** + * snd_soc_get_value_enum_double - semi enumerated double mixer get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value of a double semi enumerated mixer. + * + * Semi enumerated mixer: the enumerated items are referred as values. Can be + * used for handling bitfield coded enumeration for example. + * + * Returns 0 for success. + */ +int snd_soc_get_value_enum_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned short reg_val, val, mux; + + reg_val = snd_soc_read(codec, e->reg); + val = (reg_val >> e->shift_l) & e->mask; + for (mux = 0; mux < e->max; mux++) { + if (val == e->values[mux]) + break; + } + ucontrol->value.enumerated.item[0] = mux; + if (e->shift_l != e->shift_r) { + val = (reg_val >> e->shift_r) & e->mask; + for (mux = 0; mux < e->max; mux++) { + if (val == e->values[mux]) + break; + } + ucontrol->value.enumerated.item[1] = mux; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_value_enum_double); + +/** + * snd_soc_put_value_enum_double - semi enumerated double mixer put callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to set the value of a double semi enumerated mixer. + * + * Semi enumerated mixer: the enumerated items are referred as values. Can be + * used for handling bitfield coded enumeration for example. + * + * Returns 0 for success. + */ +int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned short val; + unsigned short mask; + + if (ucontrol->value.enumerated.item[0] > e->max - 1) + return -EINVAL; + val = e->values[ucontrol->value.enumerated.item[0]] << e->shift_l; + mask = e->mask << e->shift_l; + if (e->shift_l != e->shift_r) { + if (ucontrol->value.enumerated.item[1] > e->max - 1) + return -EINVAL; + val |= e->values[ucontrol->value.enumerated.item[1]] << e->shift_r; + mask |= e->mask << e->shift_r; + } + + return snd_soc_update_bits(codec, e->reg, mask, val); +} +EXPORT_SYMBOL_GPL(snd_soc_put_value_enum_double); + +/** * snd_soc_info_enum_ext - external enumerated single mixer info callback * @kcontrol: mixer control * @uinfo: control element information @@ -1668,7 +1744,7 @@ EXPORT_SYMBOL_GPL(snd_soc_info_volsw); /** * snd_soc_get_volsw - single mixer get callback * @kcontrol: mixer control - * @uinfo: control element information + * @ucontrol: control element information * * Callback to get the value of a single mixer control. * @@ -1707,7 +1783,7 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw); /** * snd_soc_put_volsw - single mixer put callback * @kcontrol: mixer control - * @uinfo: control element information + * @ucontrol: control element information * * Callback to set the value of a single mixer control. * @@ -1775,7 +1851,7 @@ EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r); /** * snd_soc_get_volsw_2r - double mixer get callback * @kcontrol: mixer control - * @uinfo: control element information + * @ucontrol: control element information * * Callback to get the value of a double mixer control that spans 2 registers. * @@ -1812,7 +1888,7 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r); /** * snd_soc_put_volsw_2r - double mixer set callback * @kcontrol: mixer control - * @uinfo: control element information + * @ucontrol: control element information * * Callback to set the value of a double mixer control that spans 2 registers. * @@ -1882,7 +1958,7 @@ EXPORT_SYMBOL_GPL(snd_soc_info_volsw_s8); /** * snd_soc_get_volsw_s8 - signed mixer get callback * @kcontrol: mixer control - * @uinfo: control element information + * @ucontrol: control element information * * Callback to get the value of a signed mixer control. * @@ -1909,7 +1985,7 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_s8); /** * snd_soc_put_volsw_sgn - signed mixer put callback * @kcontrol: mixer control - * @uinfo: control element information + * @ucontrol: control element information * * Callback to set the value of a signed mixer control. * @@ -1954,7 +2030,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); /** * snd_soc_dai_set_clkdiv - configure DAI clock dividers. * @dai: DAI - * @clk_id: DAI specific clock divider ID + * @div_id: DAI specific clock divider ID * @div: new clock divisor. * * Configures the clock dividers. This is used to derive the best DAI bit and @@ -2060,7 +2136,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); /** * snd_soc_register_card - Register a card with the ASoC core * - * @param card Card to register + * @card: Card to register * * Note that currently this is an internal only function: it will be * exposed to machine drivers after further backporting of ASoC v2 @@ -2087,7 +2163,7 @@ static int snd_soc_register_card(struct snd_soc_card *card) /** * snd_soc_unregister_card - Unregister a card with the ASoC core * - * @param card Card to unregister + * @card: Card to unregister * * Note that currently this is an internal only function: it will be * exposed to machine drivers after further backporting of ASoC v2 @@ -2107,7 +2183,7 @@ static int snd_soc_unregister_card(struct snd_soc_card *card) /** * snd_soc_register_dai - Register a DAI with the ASoC core * - * @param dai DAI to register + * @dai: DAI to register */ int snd_soc_register_dai(struct snd_soc_dai *dai) { @@ -2134,7 +2210,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_dai); /** * snd_soc_unregister_dai - Unregister a DAI from the ASoC core * - * @param dai DAI to unregister + * @dai: DAI to unregister */ void snd_soc_unregister_dai(struct snd_soc_dai *dai) { @@ -2149,8 +2225,8 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_dai); /** * snd_soc_register_dais - Register multiple DAIs with the ASoC core * - * @param dai Array of DAIs to register - * @param count Number of DAIs + * @dai: Array of DAIs to register + * @count: Number of DAIs */ int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count) { @@ -2175,8 +2251,8 @@ EXPORT_SYMBOL_GPL(snd_soc_register_dais); /** * snd_soc_unregister_dais - Unregister multiple DAIs from the ASoC core * - * @param dai Array of DAIs to unregister - * @param count Number of DAIs + * @dai: Array of DAIs to unregister + * @count: Number of DAIs */ void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count) { @@ -2190,7 +2266,7 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_dais); /** * snd_soc_register_platform - Register a platform with the ASoC core * - * @param platform platform to register + * @platform: platform to register */ int snd_soc_register_platform(struct snd_soc_platform *platform) { @@ -2213,7 +2289,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_platform); /** * snd_soc_unregister_platform - Unregister a platform from the ASoC core * - * @param platform platform to unregister + * @platform: platform to unregister */ void snd_soc_unregister_platform(struct snd_soc_platform *platform) { @@ -2228,7 +2304,7 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_platform); /** * snd_soc_register_codec - Register a codec with the ASoC core * - * @param codec codec to register + * @codec: codec to register */ int snd_soc_register_codec(struct snd_soc_codec *codec) { @@ -2255,7 +2331,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_codec); /** * snd_soc_unregister_codec - Unregister a codec from the ASoC core * - * @param codec codec to unregister + * @codec: codec to unregister */ void snd_soc_unregister_codec(struct snd_soc_codec *codec) { diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8863eddbac02..493a4e8aa273 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -53,13 +53,15 @@ /* dapm power sequences - make this per codec in the future */ static int dapm_up_seq[] = { snd_soc_dapm_pre, snd_soc_dapm_micbias, snd_soc_dapm_mic, - snd_soc_dapm_mux, snd_soc_dapm_dac, snd_soc_dapm_mixer, snd_soc_dapm_pga, - snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk, snd_soc_dapm_post + snd_soc_dapm_mux, snd_soc_dapm_value_mux, snd_soc_dapm_dac, + snd_soc_dapm_mixer, snd_soc_dapm_pga, snd_soc_dapm_adc, snd_soc_dapm_hp, + snd_soc_dapm_spk, snd_soc_dapm_post }; static int dapm_down_seq[] = { snd_soc_dapm_pre, snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk, snd_soc_dapm_pga, snd_soc_dapm_mixer, snd_soc_dapm_dac, snd_soc_dapm_mic, - snd_soc_dapm_micbias, snd_soc_dapm_mux, snd_soc_dapm_post + snd_soc_dapm_micbias, snd_soc_dapm_mux, snd_soc_dapm_value_mux, + snd_soc_dapm_post }; static int dapm_status = 1; @@ -134,6 +136,25 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, } } break; + case snd_soc_dapm_value_mux: { + struct soc_enum *e = (struct soc_enum *) + w->kcontrols[i].private_value; + int val, item; + + val = snd_soc_read(w->codec, e->reg); + val = (val >> e->shift_l) & e->mask; + for (item = 0; item < e->max; item++) { + if (val == e->values[item]) + break; + } + + p->connect = 0; + for (i = 0; i < e->max; i++) { + if (!(strcmp(p->name, e->texts[i])) && item == i) + p->connect = 1; + } + } + break; /* does not effect routing - always connected */ case snd_soc_dapm_pga: case snd_soc_dapm_output: @@ -653,6 +674,7 @@ static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action) case snd_soc_dapm_vmid: continue; case snd_soc_dapm_mux: + case snd_soc_dapm_value_mux: case snd_soc_dapm_output: case snd_soc_dapm_input: case snd_soc_dapm_switch: @@ -960,6 +982,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, path->connect = 1; return 0; case snd_soc_dapm_mux: + case snd_soc_dapm_value_mux: ret = dapm_connect_mux(codec, wsource, wsink, path, control, &wsink->kcontrols[0]); if (ret != 0) @@ -1047,6 +1070,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec) dapm_new_mixer(codec, w); break; case snd_soc_dapm_mux: + case snd_soc_dapm_value_mux: dapm_new_mux(codec, w); break; case snd_soc_dapm_adc: @@ -1077,7 +1101,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets); /** * snd_soc_dapm_get_volsw - dapm mixer get callback * @kcontrol: mixer control - * @uinfo: control element information + * @ucontrol: control element information * * Callback to get the value of a dapm mixer control. * @@ -1122,7 +1146,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_volsw); /** * snd_soc_dapm_put_volsw - dapm mixer set callback * @kcontrol: mixer control - * @uinfo: control element information + * @ucontrol: control element information * * Callback to set the value of a dapm mixer control. * @@ -1193,7 +1217,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw); /** * snd_soc_dapm_get_enum_double - dapm enumerated double mixer get callback * @kcontrol: mixer control - * @uinfo: control element information + * @ucontrol: control element information * * Callback to get the value of a dapm enumerated double mixer control. * @@ -1221,7 +1245,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double); /** * snd_soc_dapm_put_enum_double - dapm enumerated double mixer set callback * @kcontrol: mixer control - * @uinfo: control element information + * @ucontrol: control element information * * Callback to set the value of a dapm enumerated double mixer control. * @@ -1274,6 +1298,103 @@ out: EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double); /** + * snd_soc_dapm_get_value_enum_double - dapm semi enumerated double mixer get + * callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value of a dapm semi enumerated double mixer control. + * + * Semi enumerated mixer: the enumerated items are referred as values. Can be + * used for handling bitfield coded enumeration for example. + * + * Returns 0 for success. + */ +int snd_soc_dapm_get_value_enum_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned short reg_val, val, mux; + + reg_val = snd_soc_read(widget->codec, e->reg); + val = (reg_val >> e->shift_l) & e->mask; + for (mux = 0; mux < e->max; mux++) { + if (val == e->values[mux]) + break; + } + ucontrol->value.enumerated.item[0] = mux; + if (e->shift_l != e->shift_r) { + val = (reg_val >> e->shift_r) & e->mask; + for (mux = 0; mux < e->max; mux++) { + if (val == e->values[mux]) + break; + } + ucontrol->value.enumerated.item[1] = mux; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_get_value_enum_double); + +/** + * snd_soc_dapm_put_value_enum_double - dapm semi enumerated double mixer set + * callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to set the value of a dapm semi enumerated double mixer control. + * + * Semi enumerated mixer: the enumerated items are referred as values. Can be + * used for handling bitfield coded enumeration for example. + * + * Returns 0 for success. + */ +int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned short val, mux; + unsigned short mask; + int ret = 0; + + if (ucontrol->value.enumerated.item[0] > e->max - 1) + return -EINVAL; + mux = ucontrol->value.enumerated.item[0]; + val = e->values[ucontrol->value.enumerated.item[0]] << e->shift_l; + mask = e->mask << e->shift_l; + if (e->shift_l != e->shift_r) { + if (ucontrol->value.enumerated.item[1] > e->max - 1) + return -EINVAL; + val |= e->values[ucontrol->value.enumerated.item[1]] << e->shift_r; + mask |= e->mask << e->shift_r; + } + + mutex_lock(&widget->codec->mutex); + widget->value = val; + dapm_mux_update_power(widget, kcontrol, mask, mux, val, e); + if (widget->event) { + if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { + ret = widget->event(widget, + kcontrol, SND_SOC_DAPM_PRE_REG); + if (ret < 0) + goto out; + } + ret = snd_soc_update_bits(widget->codec, e->reg, mask, val); + if (widget->event_flags & SND_SOC_DAPM_POST_REG) + ret = widget->event(widget, + kcontrol, SND_SOC_DAPM_POST_REG); + } else + ret = snd_soc_update_bits(widget->codec, e->reg, mask, val); + +out: + mutex_unlock(&widget->codec->mutex); + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_put_value_enum_double); + +/** * snd_soc_dapm_new_control - create new dapm control * @codec: audio codec * @widget: widget template @@ -1419,7 +1540,7 @@ int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, /** * snd_soc_dapm_enable_pin - enable pin. - * @snd_soc_codec: SoC codec + * @codec: SoC codec * @pin: pin name * * Enables input/output pin and it's parents or children widgets iff there is diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index d44bf98e965e..41c387587474 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -2057,7 +2057,7 @@ static int __devinit cs4231_ebus_probe(struct of_device *op, const struct of_dev if (err) return err; - sprintf(card->longname, "%s at 0x%lx, irq %d", + sprintf(card->longname, "%s at 0x%llx, irq %d", card->shortname, op->resource[0].start, op->irqs[0]); diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c index a62500e387a6..41c36b055f6b 100644 --- a/sound/usb/caiaq/caiaq-device.c +++ b/sound/usb/caiaq/caiaq-device.c @@ -42,7 +42,7 @@ #endif MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>"); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.9"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.10"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," diff --git a/sound/usb/caiaq/caiaq-device.h b/sound/usb/caiaq/caiaq-device.h index f9fbdbae269d..ab56e738c5fc 100644 --- a/sound/usb/caiaq/caiaq-device.h +++ b/sound/usb/caiaq/caiaq-device.h @@ -75,6 +75,7 @@ struct snd_usb_caiaqdev { wait_queue_head_t ep1_wait_queue; wait_queue_head_t prepare_wait_queue; int spec_received, audio_parm_answer; + int midi_out_active; char vendor_name[CAIAQ_USB_STR_LEN]; char product_name[CAIAQ_USB_STR_LEN]; diff --git a/sound/usb/caiaq/caiaq-midi.c b/sound/usb/caiaq/caiaq-midi.c index 30b57f97c6e4..f19fd360c936 100644 --- a/sound/usb/caiaq/caiaq-midi.c +++ b/sound/usb/caiaq/caiaq-midi.c @@ -59,6 +59,11 @@ static int snd_usb_caiaq_midi_output_open(struct snd_rawmidi_substream *substrea static int snd_usb_caiaq_midi_output_close(struct snd_rawmidi_substream *substream) { + struct snd_usb_caiaqdev *dev = substream->rmidi->private_data; + if (dev->midi_out_active) { + usb_kill_urb(&dev->midi_out_urb); + dev->midi_out_active = 0; + } return 0; } @@ -69,7 +74,8 @@ static void snd_usb_caiaq_midi_send(struct snd_usb_caiaqdev *dev, dev->midi_out_buf[0] = EP1_CMD_MIDI_WRITE; dev->midi_out_buf[1] = 0; /* port */ - len = snd_rawmidi_transmit_peek(substream, dev->midi_out_buf+3, EP1_BUFSIZE-3); + len = snd_rawmidi_transmit(substream, dev->midi_out_buf + 3, + EP1_BUFSIZE - 3); if (len <= 0) return; @@ -79,24 +85,24 @@ static void snd_usb_caiaq_midi_send(struct snd_usb_caiaqdev *dev, ret = usb_submit_urb(&dev->midi_out_urb, GFP_ATOMIC); if (ret < 0) - log("snd_usb_caiaq_midi_send(%p): usb_submit_urb() failed, %d\n", - substream, ret); + log("snd_usb_caiaq_midi_send(%p): usb_submit_urb() failed," + "ret=%d, len=%d\n", + substream, ret, len); + else + dev->midi_out_active = 1; } static void snd_usb_caiaq_midi_output_trigger(struct snd_rawmidi_substream *substream, int up) { struct snd_usb_caiaqdev *dev = substream->rmidi->private_data; - if (dev->midi_out_substream != NULL) - return; - - if (!up) { + if (up) { + dev->midi_out_substream = substream; + if (!dev->midi_out_active) + snd_usb_caiaq_midi_send(dev, substream); + } else { dev->midi_out_substream = NULL; - return; } - - dev->midi_out_substream = substream; - snd_usb_caiaq_midi_send(dev, substream); } @@ -161,16 +167,14 @@ int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device) void snd_usb_caiaq_midi_output_done(struct urb* urb) { struct snd_usb_caiaqdev *dev = urb->context; - char *buf = urb->transfer_buffer; + dev->midi_out_active = 0; if (urb->status != 0) return; if (!dev->midi_out_substream) return; - snd_rawmidi_transmit_ack(dev->midi_out_substream, buf[2]); - dev->midi_out_substream = NULL; snd_usb_caiaq_midi_send(dev, dev->midi_out_substream); } diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index bc8bd00047ad..330f2fbff2d1 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -111,6 +111,8 @@ struct mixer_build { const struct usbmix_selector_map *selector_map; }; +#define MAX_CHANNELS 10 /* max logical channels */ + struct usb_mixer_elem_info { struct usb_mixer_interface *mixer; struct usb_mixer_elem_info *next_id_elem; /* list of controls with same id */ @@ -121,6 +123,8 @@ struct usb_mixer_elem_info { int channels; int val_type; int min, max, res; + int cached; + int cache_val[MAX_CHANNELS]; u8 initialized; }; @@ -182,8 +186,6 @@ enum { USB_PROC_DCR_RELEASE = 6, }; -#define MAX_CHANNELS 10 /* max logical channels */ - /* * manual mapping of mixer names @@ -377,11 +379,35 @@ static int get_cur_ctl_value(struct usb_mixer_elem_info *cval, int validx, int * } /* channel = 0: master, 1 = first channel */ -static inline int get_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, int *value) +static inline int get_cur_mix_raw(struct usb_mixer_elem_info *cval, + int channel, int *value) { return get_ctl_value(cval, GET_CUR, (cval->control << 8) | channel, value); } +static int get_cur_mix_value(struct usb_mixer_elem_info *cval, + int channel, int index, int *value) +{ + int err; + + if (cval->cached & (1 << channel)) { + *value = cval->cache_val[index]; + return 0; + } + err = get_cur_mix_raw(cval, channel, value); + if (err < 0) { + if (!cval->mixer->ignore_ctl_error) + snd_printd(KERN_ERR "cannot get current value for " + "control %d ch %d: err = %d\n", + cval->control, channel, err); + return err; + } + cval->cached |= 1 << channel; + cval->cache_val[index] = *value; + return 0; +} + + /* * set a mixer value */ @@ -413,9 +439,17 @@ static int set_cur_ctl_value(struct usb_mixer_elem_info *cval, int validx, int v return set_ctl_value(cval, SET_CUR, validx, value); } -static inline int set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, int value) +static int set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, + int index, int value) { - return set_ctl_value(cval, SET_CUR, (cval->control << 8) | channel, value); + int err; + err = set_ctl_value(cval, SET_CUR, (cval->control << 8) | channel, + value); + if (err < 0) + return err; + cval->cached |= 1 << channel; + cval->cache_val[index] = value; + return 0; } /* @@ -719,7 +753,7 @@ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min) if (cval->min + cval->res < cval->max) { int last_valid_res = cval->res; int saved, test, check; - get_cur_mix_value(cval, minchn, &saved); + get_cur_mix_raw(cval, minchn, &saved); for (;;) { test = saved; if (test < cval->max) @@ -727,8 +761,8 @@ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min) else test -= cval->res; if (test < cval->min || test > cval->max || - set_cur_mix_value(cval, minchn, test) || - get_cur_mix_value(cval, minchn, &check)) { + set_cur_mix_value(cval, minchn, 0, test) || + get_cur_mix_raw(cval, minchn, &check)) { cval->res = last_valid_res; break; } @@ -736,7 +770,7 @@ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min) break; cval->res *= 2; } - set_cur_mix_value(cval, minchn, saved); + set_cur_mix_value(cval, minchn, 0, saved); } cval->initialized = 1; @@ -776,35 +810,25 @@ static int mixer_ctl_feature_get(struct snd_kcontrol *kcontrol, struct snd_ctl_e struct usb_mixer_elem_info *cval = kcontrol->private_data; int c, cnt, val, err; + ucontrol->value.integer.value[0] = cval->min; if (cval->cmask) { cnt = 0; for (c = 0; c < MAX_CHANNELS; c++) { - if (cval->cmask & (1 << c)) { - err = get_cur_mix_value(cval, c + 1, &val); - if (err < 0) { - if (cval->mixer->ignore_ctl_error) { - ucontrol->value.integer.value[0] = cval->min; - return 0; - } - snd_printd(KERN_ERR "cannot get current value for control %d ch %d: err = %d\n", cval->control, c + 1, err); - return err; - } - val = get_relative_value(cval, val); - ucontrol->value.integer.value[cnt] = val; - cnt++; - } + if (!(cval->cmask & (1 << c))) + continue; + err = get_cur_mix_value(cval, c + 1, cnt, &val); + if (err < 0) + return cval->mixer->ignore_ctl_error ? 0 : err; + val = get_relative_value(cval, val); + ucontrol->value.integer.value[cnt] = val; + cnt++; } + return 0; } else { /* master channel */ - err = get_cur_mix_value(cval, 0, &val); - if (err < 0) { - if (cval->mixer->ignore_ctl_error) { - ucontrol->value.integer.value[0] = cval->min; - return 0; - } - snd_printd(KERN_ERR "cannot get current value for control %d master ch: err = %d\n", cval->control, err); - return err; - } + err = get_cur_mix_value(cval, 0, 0, &val); + if (err < 0) + return cval->mixer->ignore_ctl_error ? 0 : err; val = get_relative_value(cval, val); ucontrol->value.integer.value[0] = val; } @@ -821,34 +845,28 @@ static int mixer_ctl_feature_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e if (cval->cmask) { cnt = 0; for (c = 0; c < MAX_CHANNELS; c++) { - if (cval->cmask & (1 << c)) { - err = get_cur_mix_value(cval, c + 1, &oval); - if (err < 0) { - if (cval->mixer->ignore_ctl_error) - return 0; - return err; - } - val = ucontrol->value.integer.value[cnt]; - val = get_abs_value(cval, val); - if (oval != val) { - set_cur_mix_value(cval, c + 1, val); - changed = 1; - } - get_cur_mix_value(cval, c + 1, &val); - cnt++; + if (!(cval->cmask & (1 << c))) + continue; + err = get_cur_mix_value(cval, c + 1, cnt, &oval); + if (err < 0) + return cval->mixer->ignore_ctl_error ? 0 : err; + val = ucontrol->value.integer.value[cnt]; + val = get_abs_value(cval, val); + if (oval != val) { + set_cur_mix_value(cval, c + 1, cnt, val); + changed = 1; } + cnt++; } } else { /* master channel */ - err = get_cur_mix_value(cval, 0, &oval); - if (err < 0 && cval->mixer->ignore_ctl_error) - return 0; + err = get_cur_mix_value(cval, 0, 0, &oval); if (err < 0) - return err; + return cval->mixer->ignore_ctl_error ? 0 : err; val = ucontrol->value.integer.value[0]; val = get_abs_value(cval, val); if (val != oval) { - set_cur_mix_value(cval, 0, val); + set_cur_mix_value(cval, 0, 0, val); changed = 1; } } diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index ca26c532e77e..11639bd72a51 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -238,7 +238,7 @@ static void i_usX2Y_In04Int(struct urb *urb) send = 0; for (j = 0; j < URBS_AsyncSeq && !err; ++j) if (0 == usX2Y->AS04.urb[j]->status) { - struct us428_p4out *p4out = us428ctls->p4out + send; // FIXME if more then 1 p4out is new, 1 gets lost. + struct us428_p4out *p4out = us428ctls->p4out + send; // FIXME if more than 1 p4out is new, 1 gets lost. usb_fill_bulk_urb(usX2Y->AS04.urb[j], usX2Y->chip.dev, usb_sndbulkpipe(usX2Y->chip.dev, 0x04), &p4out->val.vol, p4out->type == eLT_Light ? sizeof(struct us428_lights) : 5, |