diff options
Diffstat (limited to 'sound')
68 files changed, 966 insertions, 248 deletions
diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 76e0d569507..823359ed95e 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -166,7 +166,9 @@ void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id) } else { printk(KERN_ERR "%s: DMA error on channel %d (DCSR=%#x)\n", rtd->params->name, dma_ch, dcsr); + snd_pcm_stream_lock(substream); snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock(substream); } } EXPORT_SYMBOL(pxa2xx_pcm_dma_irq); diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 99db892d729..19799931c51 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -668,14 +668,48 @@ static int snd_compr_stop(struct snd_compr_stream *stream) return -EPERM; retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_STOP); if (!retval) { - stream->runtime->state = SNDRV_PCM_STATE_SETUP; - wake_up(&stream->runtime->sleep); + snd_compr_drain_notify(stream); stream->runtime->total_bytes_available = 0; stream->runtime->total_bytes_transferred = 0; } return retval; } +static int snd_compress_wait_for_drain(struct snd_compr_stream *stream) +{ + int ret; + + /* + * We are called with lock held. So drop the lock while we wait for + * drain complete notfication from the driver + * + * It is expected that driver will notify the drain completion and then + * stream will be moved to SETUP state, even if draining resulted in an + * error. We can trigger next track after this. + */ + stream->runtime->state = SNDRV_PCM_STATE_DRAINING; + mutex_unlock(&stream->device->lock); + + /* we wait for drain to complete here, drain can return when + * interruption occurred, wait returned error or success. + * For the first two cases we don't do anything different here and + * return after waking up + */ + + ret = wait_event_interruptible(stream->runtime->sleep, + (stream->runtime->state != SNDRV_PCM_STATE_DRAINING)); + if (ret == -ERESTARTSYS) + pr_debug("wait aborted by a signal"); + else if (ret) + pr_debug("wait for drain failed with %d\n", ret); + + + wake_up(&stream->runtime->sleep); + mutex_lock(&stream->device->lock); + + return ret; +} + static int snd_compr_drain(struct snd_compr_stream *stream) { int retval; @@ -683,12 +717,15 @@ static int snd_compr_drain(struct snd_compr_stream *stream) if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || stream->runtime->state == SNDRV_PCM_STATE_SETUP) return -EPERM; + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_DRAIN); - if (!retval) { - stream->runtime->state = SNDRV_PCM_STATE_DRAINING; + if (retval) { + pr_debug("SND_COMPR_TRIGGER_DRAIN failed %d\n", retval); wake_up(&stream->runtime->sleep); + return retval; } - return retval; + + return snd_compress_wait_for_drain(stream); } static int snd_compr_next_track(struct snd_compr_stream *stream) @@ -724,9 +761,14 @@ static int snd_compr_partial_drain(struct snd_compr_stream *stream) return -EPERM; retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_PARTIAL_DRAIN); + if (retval) { + pr_debug("Partial drain returned failure\n"); + wake_up(&stream->runtime->sleep); + return retval; + } stream->next_track = false; - return retval; + return snd_compress_wait_for_drain(stream); } static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) @@ -743,7 +785,7 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) mutex_lock(&stream->device->lock); switch (_IOC_NR(cmd)) { case _IOC_NR(SNDRV_COMPRESS_IOCTL_VERSION): - put_user(SNDRV_COMPRESS_VERSION, + retval = put_user(SNDRV_COMPRESS_VERSION, (int __user *)arg) ? -EFAULT : 0; break; case _IOC_NR(SNDRV_COMPRESS_GET_CAPS): @@ -837,7 +879,8 @@ static int snd_compress_dev_disconnect(struct snd_device *device) struct snd_compr *compr; compr = device->device_data; - snd_unregister_device(compr->direction, compr->card, compr->device); + snd_unregister_device(SNDRV_DEVICE_TYPE_COMPRESS, compr->card, + compr->device); return 0; } diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 17f45e8aa89..e1e9e0c999f 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -49,6 +49,8 @@ static struct snd_pcm *snd_pcm_get(struct snd_card *card, int device) struct snd_pcm *pcm; list_for_each_entry(pcm, &snd_pcm_devices, list) { + if (pcm->internal) + continue; if (pcm->card == card && pcm->device == device) return pcm; } @@ -60,6 +62,8 @@ static int snd_pcm_next(struct snd_card *card, int device) struct snd_pcm *pcm; list_for_each_entry(pcm, &snd_pcm_devices, list) { + if (pcm->internal) + continue; if (pcm->card == card && pcm->device > device) return pcm->device; else if (pcm->card->number > card->number) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 41b3dfe6869..3284940a4af 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1936,6 +1936,8 @@ static int wait_for_avail(struct snd_pcm_substream *substream, case SNDRV_PCM_STATE_DISCONNECTED: err = -EBADFD; goto _endloop; + case SNDRV_PCM_STATE_PAUSED: + continue; } if (!tout) { snd_printd("%s write error (DMA or IRQ trouble?)\n", diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c index e3cb46fef2c..b3f39b5ed74 100644 --- a/sound/core/seq/oss/seq_oss_init.c +++ b/sound/core/seq/oss/seq_oss_init.c @@ -31,6 +31,7 @@ #include <linux/export.h> #include <linux/moduleparam.h> #include <linux/slab.h> +#include <linux/workqueue.h> /* * common variables @@ -60,6 +61,14 @@ static void free_devinfo(void *private); #define call_ctl(type,rec) snd_seq_kernel_client_ctl(system_client, type, rec) +/* call snd_seq_oss_midi_lookup_ports() asynchronously */ +static void async_call_lookup_ports(struct work_struct *work) +{ + snd_seq_oss_midi_lookup_ports(system_client); +} + +static DECLARE_WORK(async_lookup_work, async_call_lookup_ports); + /* * create sequencer client for OSS sequencer */ @@ -85,9 +94,6 @@ snd_seq_oss_create_client(void) system_client = rc; debug_printk(("new client = %d\n", rc)); - /* look up midi devices */ - snd_seq_oss_midi_lookup_ports(system_client); - /* create annoucement receiver port */ memset(port, 0, sizeof(*port)); strcpy(port->name, "Receiver"); @@ -115,6 +121,9 @@ snd_seq_oss_create_client(void) } rc = 0; + /* look up midi devices */ + schedule_work(&async_lookup_work); + __error: kfree(port); return rc; @@ -160,6 +169,7 @@ receive_announce(struct snd_seq_event *ev, int direct, void *private, int atomic int snd_seq_oss_delete_client(void) { + cancel_work_sync(&async_lookup_work); if (system_client >= 0) snd_seq_delete_kernel_client(system_client); diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index 677dc84590c..862d84893ee 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -72,7 +72,7 @@ static int send_midi_event(struct seq_oss_devinfo *dp, struct snd_seq_event *ev, * look up the existing ports * this looks a very exhausting job. */ -int __init +int snd_seq_oss_midi_lookup_ports(int client) { struct snd_seq_client_info *clinfo; diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c index ddabb406b14..3a7946ebbe2 100644 --- a/sound/isa/msnd/msnd_pinnacle.c +++ b/sound/isa/msnd/msnd_pinnacle.c @@ -73,9 +73,11 @@ #ifdef MSND_CLASSIC # include "msnd_classic.h" # define LOGNAME "msnd_classic" +# define DEV_NAME "msnd-classic" #else # include "msnd_pinnacle.h" # define LOGNAME "snd_msnd_pinnacle" +# define DEV_NAME "msnd-pinnacle" #endif static void set_default_audio_parameters(struct snd_msnd *chip) @@ -1068,8 +1070,6 @@ static int snd_msnd_isa_remove(struct device *pdev, unsigned int dev) return 0; } -#define DEV_NAME "msnd-pinnacle" - static struct isa_driver snd_msnd_driver = { .match = snd_msnd_isa_match, .probe = snd_msnd_isa_probe, diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index b41ed8661b2..e427dbf7636 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -173,11 +173,7 @@ MODULE_DEVICE_TABLE(pnp_card, snd_opti9xx_pnpids); #endif /* CONFIG_PNP */ -#ifdef OPTi93X -#define DEV_NAME "opti93x" -#else -#define DEV_NAME "opti92x" -#endif +#define DEV_NAME KBUILD_MODNAME static char * snd_opti9xx_names[] = { "unknown", @@ -1168,7 +1164,7 @@ static int snd_opti9xx_pnp_resume(struct pnp_card_link *pcard) static struct pnp_card_driver opti9xx_pnpc_driver = { .flags = PNP_DRIVER_RES_DISABLE, - .name = "opti9xx", + .name = DEV_NAME, .id_table = snd_opti9xx_pnpids, .probe = snd_opti9xx_pnp_probe, .remove = snd_opti9xx_pnp_remove, diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index fe6fa93a626..daac7c7ebe9 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -30,6 +30,7 @@ config SND_ALS300 select SND_PCM select SND_AC97_CODEC select SND_OPL3_LIB + select ZONE_DMA help Say 'Y' or 'M' to include support for Avance Logic ALS300/ALS300+ @@ -54,6 +55,7 @@ config SND_ALI5451 tristate "ALi M5451 PCI Audio Controller" select SND_MPU401_UART select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for the integrated AC97 sound device on motherboards using the ALi M5451 Audio Controller @@ -158,6 +160,7 @@ config SND_AZT3328 select SND_PCM select SND_RAWMIDI select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for Aztech AZF3328 (PCI168) soundcards. @@ -463,6 +466,7 @@ config SND_EMU10K1 select SND_HWDEP select SND_RAWMIDI select SND_AC97_CODEC + select ZONE_DMA help Say Y to include support for Sound Blaster PCI 512, Live!, Audigy and E-mu APS (partially supported) soundcards. @@ -478,6 +482,7 @@ config SND_EMU10K1X tristate "Emu10k1X (Dell OEM Version)" select SND_AC97_CODEC select SND_RAWMIDI + select ZONE_DMA help Say Y here to include support for the Dell OEM version of the Sound Blaster Live!. @@ -511,6 +516,7 @@ config SND_ES1938 select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for soundcards based on ESS Solo-1 (ES1938, ES1946, ES1969) chips. @@ -522,6 +528,7 @@ config SND_ES1968 tristate "ESS ES1968/1978 (Maestro-1/2/2E)" select SND_MPU401_UART select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for soundcards based on ESS Maestro 1/2/2E chips. @@ -603,6 +610,7 @@ config SND_ICE1712 select SND_MPU401_UART select SND_AC97_CODEC select BITREVERSE + select ZONE_DMA help Say Y here to include support for soundcards based on the ICE1712 (Envy24) chip. @@ -690,6 +698,7 @@ config SND_LX6464ES config SND_MAESTRO3 tristate "ESS Allegro/Maestro3" select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for soundcards based on ESS Maestro 3 (Allegro) chips. @@ -786,6 +795,7 @@ config SND_SIS7019 tristate "SiS 7019 Audio Accelerator" depends on X86 && !X86_64 select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for the SiS 7019 Audio Accelerator. @@ -797,6 +807,7 @@ config SND_SONICVIBES select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for soundcards based on the S3 SonicVibes chip. @@ -808,6 +819,7 @@ config SND_TRIDENT tristate "Trident 4D-Wave DX/NX; SiS 7018" select SND_MPU401_UART select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for soundcards based on Trident 4D-Wave DX/NX or SiS 7018 chips. diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index fbc17203613..a471d821c60 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -769,7 +769,10 @@ static void snd_card_asihpi_timer_function(unsigned long data) s->number); ds->drained_count++; if (ds->drained_count > 20) { + unsigned long flags; + snd_pcm_stream_lock_irqsave(s, flags); snd_pcm_stop(s, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irqrestore(s, flags); continue; } } else { diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 6e78c678985..819430ac6b3 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -689,7 +689,9 @@ static void snd_atiixp_xrun_dma(struct atiixp *chip, struct atiixp_dma *dma) if (! dma->substream || ! dma->running) return; snd_printdd("atiixp: XRUN detected (DMA %d)\n", dma->ops->type); + snd_pcm_stream_lock(dma->substream); snd_pcm_stop(dma->substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock(dma->substream); } /* diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index d0bec7ba3b0..57f41820263 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -638,7 +638,9 @@ static void snd_atiixp_xrun_dma(struct atiixp_modem *chip, if (! dma->substream || ! dma->running) return; snd_printdd("atiixp-modem: XRUN detected (DMA %d)\n", dma->ops->type); + snd_pcm_stream_lock(dma->substream); snd_pcm_stop(dma->substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock(dma->substream); } /* diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 7c11d46b84d..48a9d004d6d 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -860,7 +860,7 @@ void snd_hda_pick_fixup(struct hda_codec *codec, } } if (id < 0 && quirk) { - for (q = quirk; q->subvendor; q++) { + for (q = quirk; q->subvendor || q->subdevice; q++) { unsigned int vendorid = q->subdevice | (q->subvendor << 16); unsigned int mask = 0xffff0000 | q->subdevice_mask; diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 55108b5fb29..aeefec74a06 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2517,9 +2517,6 @@ int snd_hda_codec_reset(struct hda_codec *codec) cancel_delayed_work_sync(&codec->jackpoll_work); #ifdef CONFIG_PM cancel_delayed_work_sync(&codec->power_work); - codec->power_on = 0; - codec->power_transition = 0; - codec->power_jiffies = jiffies; flush_workqueue(bus->workq); #endif snd_hda_ctls_clear(codec); @@ -3927,6 +3924,10 @@ static void hda_call_codec_resume(struct hda_codec *codec) * in the resume / power-save sequence */ hda_keep_power_on(codec); + if (codec->pm_down_notified) { + codec->pm_down_notified = 0; + hda_call_pm_notify(codec->bus, true); + } hda_set_power_state(codec, AC_PWRST_D0); restore_shutup_pins(codec); hda_exec_init_verbs(codec); @@ -4789,8 +4790,8 @@ static void hda_power_work(struct work_struct *work) spin_unlock(&codec->power_lock); state = hda_call_codec_suspend(codec, true); - codec->pm_down_notified = 0; - if (!bus->power_keep_link_on && (state & AC_PWRST_CLK_STOP_OK)) { + if (!codec->pm_down_notified && + !bus->power_keep_link_on && (state & AC_PWRST_CLK_STOP_OK)) { codec->pm_down_notified = 1; hda_call_pm_notify(bus, false); } diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 4b1524a861f..cb4d3700f33 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -468,6 +468,20 @@ static void invalidate_nid_path(struct hda_codec *codec, int idx) memset(path, 0, sizeof(*path)); } +/* return a DAC if paired to the given pin by codec driver */ +static hda_nid_t get_preferred_dac(struct hda_codec *codec, hda_nid_t pin) +{ + struct hda_gen_spec *spec = codec->spec; + const hda_nid_t *list = spec->preferred_dacs; + + if (!list) + return 0; + for (; *list; list += 2) + if (*list == pin) + return list[1]; + return 0; +} + /* look for an empty DAC slot */ static hda_nid_t look_for_dac(struct hda_codec *codec, hda_nid_t pin, bool is_digital) @@ -519,7 +533,7 @@ static bool same_amp_caps(struct hda_codec *codec, hda_nid_t nid1, } #define nid_has_mute(codec, nid, dir) \ - check_amp_caps(codec, nid, dir, AC_AMPCAP_MUTE) + check_amp_caps(codec, nid, dir, (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) #define nid_has_volume(codec, nid, dir) \ check_amp_caps(codec, nid, dir, AC_AMPCAP_NUM_STEPS) @@ -621,7 +635,7 @@ static int get_amp_val_to_activate(struct hda_codec *codec, hda_nid_t nid, if (enable) val = (caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT; } - if (caps & AC_AMPCAP_MUTE) { + if (caps & (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) { if (!enable) val |= HDA_AMP_MUTE; } @@ -645,7 +659,7 @@ static unsigned int get_amp_mask_to_modify(struct hda_codec *codec, { unsigned int mask = 0xff; - if (caps & AC_AMPCAP_MUTE) { + if (caps & (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) { if (is_ctl_associated(codec, nid, dir, idx, NID_PATH_MUTE_CTL)) mask &= ~0x80; } @@ -786,10 +800,10 @@ static void set_pin_eapd(struct hda_codec *codec, hda_nid_t pin, bool enable) if (spec->own_eapd_ctl || !(snd_hda_query_pin_caps(codec, pin) & AC_PINCAP_EAPD)) return; - if (codec->inv_eapd) - enable = !enable; if (spec->keep_eapd_on && !enable) return; + if (codec->inv_eapd) + enable = !enable; snd_hda_codec_update_cache(codec, pin, 0, AC_VERB_SET_EAPD_BTLENABLE, enable ? 0x02 : 0x00); @@ -840,7 +854,7 @@ static int add_control_with_pfx(struct hda_gen_spec *spec, int type, const char *pfx, const char *dir, const char *sfx, int cidx, unsigned long val) { - char name[32]; + char name[44]; snprintf(name, sizeof(name), "%s %s %s", pfx, dir, sfx); if (!add_control(spec, type, name, cidx, val)) return -ENOMEM; @@ -1134,7 +1148,14 @@ static int try_assign_dacs(struct hda_codec *codec, int num_outs, continue; } - dacs[i] = look_for_dac(codec, pin, false); + dacs[i] = get_preferred_dac(codec, pin); + if (dacs[i]) { + if (is_dac_already_used(codec, dacs[i])) + badness += bad->shared_primary; + } + + if (!dacs[i]) + dacs[i] = look_for_dac(codec, pin, false); if (!dacs[i] && !i) { /* try to steal the DAC of surrounds for the front */ for (j = 1; j < num_outs; j++) { @@ -2445,12 +2466,8 @@ static int create_out_jack_modes(struct hda_codec *codec, int num_pins, for (i = 0; i < num_pins; i++) { hda_nid_t pin = pins[i]; - if (pin == spec->hp_mic_pin) { - int ret = create_hp_mic_jack_mode(codec, pin); - if (ret < 0) - return ret; + if (pin == spec->hp_mic_pin) continue; - } if (get_out_jack_num_items(codec, pin) > 1) { struct snd_kcontrol_new *knew; char name[44]; @@ -2703,7 +2720,7 @@ static int hp_mic_jack_mode_put(struct snd_kcontrol *kcontrol, val &= ~(AC_PINCTL_VREFEN | PIN_HP); val |= get_vref_idx(vref_caps, idx) | PIN_IN; } else - val = snd_hda_get_default_vref(codec, nid); + val = snd_hda_get_default_vref(codec, nid) | PIN_IN; } snd_hda_set_pin_ctl_cache(codec, nid, val); call_hp_automute(codec, NULL); @@ -2723,9 +2740,6 @@ static int create_hp_mic_jack_mode(struct hda_codec *codec, hda_nid_t pin) struct hda_gen_spec *spec = codec->spec; struct snd_kcontrol_new *knew; - if (get_out_jack_num_items(codec, pin) <= 1 && - get_in_jack_num_items(codec, pin) <= 1) - return 0; /* no need */ knew = snd_hda_gen_add_kctl(spec, "Headphone Mic Jack Mode", &hp_mic_jack_mode_enum); if (!knew) @@ -2754,6 +2768,44 @@ static int add_loopback_list(struct hda_gen_spec *spec, hda_nid_t mix, int idx) return 0; } +/* return true if either a volume or a mute amp is found for the given + * aamix path; the amp has to be either in the mixer node or its direct leaf + */ +static bool look_for_mix_leaf_ctls(struct hda_codec *codec, hda_nid_t mix_nid, + hda_nid_t pin, unsigned int *mix_val, + unsigned int *mute_val) +{ + int idx, num_conns; + const hda_nid_t *list; + hda_nid_t nid; + + idx = snd_hda_get_conn_index(codec, mix_nid, pin, true); + if (idx < 0) + return false; + + *mix_val = *mute_val = 0; + if (nid_has_volume(codec, mix_nid, HDA_INPUT)) + *mix_val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT); + if (nid_has_mute(codec, mix_nid, HDA_INPUT)) + *mute_val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT); + if (*mix_val && *mute_val) + return true; + + /* check leaf node */ + num_conns = snd_hda_get_conn_list(codec, mix_nid, &list); + if (num_conns < idx) + return false; + nid = list[idx]; + if (!*mix_val && nid_has_volume(codec, nid, HDA_OUTPUT) && + !is_ctl_associated(codec, nid, HDA_OUTPUT, 0, NID_PATH_VOL_CTL)) + *mix_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT); + if (!*mute_val && nid_has_mute(codec, nid, HDA_OUTPUT) && + !is_ctl_associated(codec, nid, HDA_OUTPUT, 0, NID_PATH_MUTE_CTL)) + *mute_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT); + + return *mix_val || *mute_val; +} + /* create input playback/capture controls for the given pin */ static int new_analog_input(struct hda_codec *codec, int input_idx, hda_nid_t pin, const char *ctlname, int ctlidx, @@ -2761,12 +2813,11 @@ static int new_analog_input(struct hda_codec *codec, int input_idx, { struct hda_gen_spec *spec = codec->spec; struct nid_path *path; - unsigned int val; + unsigned int mix_val, mute_val; int err, idx; - if (!nid_has_volume(codec, mix_nid, HDA_INPUT) && - !nid_has_mute(codec, mix_nid, HDA_INPUT)) - return 0; /* no need for analog loopback */ + if (!look_for_mix_leaf_ctls(codec, mix_nid, pin, &mix_val, &mute_val)) + return 0; path = snd_hda_add_new_path(codec, pin, mix_nid, 0); if (!path) @@ -2775,20 +2826,18 @@ static int new_analog_input(struct hda_codec *codec, int input_idx, spec->loopback_paths[input_idx] = snd_hda_get_path_idx(codec, path); idx = path->idx[path->depth - 1]; - if (nid_has_volume(codec, mix_nid, HDA_INPUT)) { - val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT); - err = __add_pb_vol_ctrl(spec, HDA_CTL_WIDGET_VOL, ctlname, ctlidx, val); + if (mix_val) { + err = __add_pb_vol_ctrl(spec, HDA_CTL_WIDGET_VOL, ctlname, ctlidx, mix_val); if (err < 0) return err; - path->ctls[NID_PATH_VOL_CTL] = val; + path->ctls[NID_PATH_VOL_CTL] = mix_val; } - if (nid_has_mute(codec, mix_nid, HDA_INPUT)) { - val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT); - err = __add_pb_sw_ctrl(spec, HDA_CTL_WIDGET_MUTE, ctlname, ctlidx, val); + if (mute_val) { + err = __add_pb_sw_ctrl(spec, HDA_CTL_WIDGET_MUTE, ctlname, ctlidx, mute_val); if (err < 0) return err; - path->ctls[NID_PATH_MUTE_CTL] = val; + path->ctls[NID_PATH_MUTE_CTL] = mute_val; } path->active = true; @@ -3474,7 +3523,7 @@ static int create_capture_mixers(struct hda_codec *codec) if (!multi) err = create_single_cap_vol_ctl(codec, n, vol, sw, inv_dmic); - else if (!multi_cap_vol) + else if (!multi_cap_vol && !inv_dmic) err = create_bind_cap_vol_ctl(codec, n, vol, sw); else err = create_multi_cap_vol_ctl(codec); @@ -4175,6 +4224,26 @@ static unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, return AC_PWRST_D3; } +/* mute all aamix inputs initially; parse up to the first leaves */ +static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix) +{ + int i, nums; + const hda_nid_t *conn; + bool has_amp; + + nums = snd_hda_get_conn_list(codec, mix, &conn); + has_amp = nid_has_mute(codec, mix, HDA_INPUT); + for (i = 0; i < nums; i++) { + if (has_amp) + snd_hda_codec_amp_stereo(codec, mix, + HDA_INPUT, i, + 0xff, HDA_AMP_MUTE); + else if (nid_has_volume(codec, conn[i], HDA_OUTPUT)) + snd_hda_codec_amp_stereo(codec, conn[i], + HDA_OUTPUT, 0, + 0xff, HDA_AMP_MUTE); + } +} /* * Parse the given BIOS configuration and set up the hda_gen_spec @@ -4287,6 +4356,17 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, if (err < 0) return err; + /* create "Headphone Mic Jack Mode" if no input selection is + * available (or user specifies add_jack_modes hint) + */ + if (spec->hp_mic_pin && + (spec->auto_mic || spec->input_mux.num_items == 1 || + spec->add_jack_modes)) { + err = create_hp_mic_jack_mode(codec, spec->hp_mic_pin); + if (err < 0) + return err; + } + if (spec->add_jack_modes) { if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { err = create_out_jack_modes(codec, cfg->line_outs, @@ -4302,6 +4382,10 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, } } + /* mute all aamix input initially */ + if (spec->mixer_nid) + mute_all_mixer_nid(codec, spec->mixer_nid); + dig_only: parse_digital(codec); @@ -4383,9 +4467,11 @@ int snd_hda_gen_build_controls(struct hda_codec *codec) true, &spec->vmaster_mute.sw_kctl); if (err < 0) return err; - if (spec->vmaster_mute.hook) + if (spec->vmaster_mute.hook) { snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute, spec->vmaster_mute_enum); + snd_hda_sync_vmaster_hook(&spec->vmaster_mute); + } } free_kctls(spec); /* no longer needed */ diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 76200314ee9..a18a1005002 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -241,6 +241,9 @@ struct hda_gen_spec { const struct badness_table *main_out_badness; const struct badness_table *extra_out_badness; + /* preferred pin/DAC pairs; an array of paired NIDs */ + const hda_nid_t *preferred_dacs; + /* loopback mixing mode */ bool aamix_mode; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index de18722c487..5f055d7ee85 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3332,9 +3332,14 @@ static void check_probe_mask(struct azx *chip, int dev) * white/black-list for enable_msi */ static struct snd_pci_quirk msi_black_list[] = { + SND_PCI_QUIRK(0x103c, 0x2191, "HP", 0), /* AMD Hudson */ + SND_PCI_QUIRK(0x103c, 0x2192, "HP", 0), /* AMD Hudson */ + SND_PCI_QUIRK(0x103c, 0x21f7, "HP", 0), /* AMD Hudson */ + SND_PCI_QUIRK(0x103c, 0x21fa, "HP", 0), /* AMD Hudson */ SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */ SND_PCI_QUIRK(0x1043, 0x81f6, "ASUS", 0), /* nvidia */ SND_PCI_QUIRK(0x1043, 0x822d, "ASUS", 0), /* Athlon64 X2 + nvidia MCP55 */ + SND_PCI_QUIRK(0x1179, 0xfb44, "Toshiba Satellite C870", 0), /* AMD Hudson */ SND_PCI_QUIRK(0x1849, 0x0888, "ASRock", 0), /* Athlon64 X2 + nvidia */ SND_PCI_QUIRK(0xa0a0, 0x0575, "Aopen MZ915-M", 0), /* ICH6 */ {} diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index e0bf7534fa1..2e7493ef8ee 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -562,6 +562,14 @@ static inline unsigned int get_wcaps_channels(u32 wcaps) return chans; } +static inline void snd_hda_override_wcaps(struct hda_codec *codec, + hda_nid_t nid, u32 val) +{ + if (nid >= codec->start_nid && + nid < codec->start_nid + codec->num_nodes) + codec->wcaps[nid - codec->start_nid] = val; +} + u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction); int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps); @@ -667,7 +675,7 @@ snd_hda_check_power_state(struct hda_codec *codec, hda_nid_t nid, if (state & AC_PWRST_ERROR) return true; state = (state >> 4) & 0x0f; - return (state != target_state); + return (state == target_state); } unsigned int snd_hda_codec_eapd_power_filter(struct hda_codec *codec, diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 977b0d878da..5a6527668c0 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1197,8 +1197,12 @@ static int alloc_ad_spec(struct hda_codec *codec) static void ad_fixup_inv_jack_detect(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - if (action == HDA_FIXUP_ACT_PRE_PROBE) + struct ad198x_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { codec->inv_jack_detect = 1; + spec->gen.keep_eapd_on = 1; + } } enum { @@ -1223,6 +1227,14 @@ static int ad1986a_parse_auto_config(struct hda_codec *codec) { int err; struct ad198x_spec *spec; + static hda_nid_t preferred_pairs[] = { + 0x1a, 0x03, + 0x1b, 0x03, + 0x1c, 0x04, + 0x1d, 0x05, + 0x1e, 0x03, + 0 + }; err = alloc_ad_spec(codec); if (err < 0) @@ -1243,6 +1255,8 @@ static int ad1986a_parse_auto_config(struct hda_codec *codec) * So, let's disable the shared stream. */ spec->gen.multiout.no_share_stream = 1; + /* give fixed DAC/pin pairs */ + spec->gen.preferred_dacs = preferred_pairs; snd_hda_pick_fixup(codec, NULL, ad1986a_fixup_tbl, ad1986a_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -1666,6 +1680,7 @@ static int ad1983_parse_auto_config(struct hda_codec *codec) return err; spec = codec->spec; + spec->gen.mixer_nid = 0x0e; spec->gen.beep_nid = 0x10; set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); err = ad198x_parse_auto_config(codec); @@ -2112,6 +2127,9 @@ static void ad_vmaster_eapd_hook(void *private_data, int enabled) { struct hda_codec *codec = private_data; struct ad198x_spec *spec = codec->spec; + + if (!spec->eapd_nid) + return; snd_hda_codec_update_cache(codec, spec->eapd_nid, 0, AC_VERB_SET_EAPD_BTLENABLE, enabled ? 0x02 : 0x00); @@ -3601,13 +3619,16 @@ static void ad1884_fixup_hp_eapd(struct hda_codec *codec, { struct ad198x_spec *spec = codec->spec; - if (action == HDA_FIXUP_ACT_PRE_PROBE) { + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->gen.vmaster_mute.hook = ad_vmaster_eapd_hook; + break; + case HDA_FIXUP_ACT_PROBE: if (spec->gen.autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT) spec->eapd_nid = spec->gen.autocfg.line_out_pins[0]; else spec->eapd_nid = spec->gen.autocfg.speaker_pins[0]; - if (spec->eapd_nid) - spec->gen.vmaster_mute.hook = ad_vmaster_eapd_hook; + break; } } diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index b314d3e6d7f..1868d3a6e31 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3225,6 +3225,7 @@ enum { CXT_PINCFG_LEMOTE_A1205, CXT_FIXUP_STEREO_DMIC, CXT_FIXUP_INC_MIC_BOOST, + CXT_FIXUP_GPIO1, }; static void cxt_fixup_stereo_dmic(struct hda_codec *codec, @@ -3303,6 +3304,15 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt5066_increase_mic_boost, }, + [CXT_FIXUP_GPIO1] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x01, AC_VERB_SET_GPIO_MASK, 0x01 }, + { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01 }, + { 0x01, AC_VERB_SET_GPIO_DATA, 0x01 }, + { } + }, + }, }; static const struct snd_pci_quirk cxt5051_fixups[] = { @@ -3312,6 +3322,7 @@ static const struct snd_pci_quirk cxt5051_fixups[] = { 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_GPIO1), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410), @@ -3480,6 +3491,8 @@ static const struct hda_codec_preset snd_hda_preset_conexant[] = { .patch = patch_conexant_auto }, { .id = 0x14f15115, .name = "CX20757", .patch = patch_conexant_auto }, + { .id = 0x14f151d7, .name = "CX20952", + .patch = patch_conexant_auto }, {} /* terminator */ }; @@ -3506,6 +3519,7 @@ MODULE_ALIAS("snd-hda-codec-id:14f15111"); MODULE_ALIAS("snd-hda-codec-id:14f15113"); MODULE_ALIAS("snd-hda-codec-id:14f15114"); MODULE_ALIAS("snd-hda-codec-id:14f15115"); +MODULE_ALIAS("snd-hda-codec-id:14f151d7"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Conexant HD-audio codec"); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index e12f7a030c5..ba442d24257 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -67,6 +67,8 @@ struct hdmi_spec_per_pin { struct delayed_work work; struct snd_kcontrol *eld_ctl; int repoll_count; + bool setup; /* the stream has been set up by prepare callback */ + int channels; /* current number of channels */ bool non_pcm; bool chmap_set; /* channel-map override by ALSA API? */ unsigned char chmap[8]; /* ALSA API channel-map */ @@ -84,6 +86,9 @@ struct hdmi_spec { unsigned int channels_max; /* max over all cvts */ struct hdmi_eld temp_eld; + + bool dyn_pin_out; + /* * Non-generic ATI/NVIDIA specific */ @@ -448,15 +453,25 @@ static void hdmi_write_dip_byte(struct hda_codec *codec, hda_nid_t pin_nid, static void hdmi_init_pin(struct hda_codec *codec, hda_nid_t pin_nid) { + struct hdmi_spec *spec = codec->spec; + int pin_out; + /* Unmute */ if (get_wcaps(codec, pin_nid) & AC_WCAP_OUT_AMP) snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); - /* Enable pin out: some machines with GM965 gets broken output when - * the pin is disabled or changed while using with HDMI - */ + + if (spec->dyn_pin_out) + /* Disable pin out until stream is active */ + pin_out = 0; + else + /* Enable pin out: some machines with GM965 gets broken output + * when the pin is disabled or changed while using with HDMI + */ + pin_out = PIN_OUT; + snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + AC_VERB_SET_PIN_WIDGET_CONTROL, pin_out); } static int hdmi_get_channel_count(struct hda_codec *codec, hda_nid_t cvt_nid) @@ -551,6 +566,17 @@ static int hdmi_channel_allocation(struct hdmi_eld *eld, int channels) } } + if (!ca) { + /* if there was no match, select the regular ALSA channel + * allocation with the matching number of channels */ + for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) { + if (channels == channel_allocations[i].channels) { + ca = channel_allocations[i].ca_index; + break; + } + } + } + snd_print_channel_allocation(eld->info.spk_alloc, buf, sizeof(buf)); snd_printdd("HDMI: select CA 0x%x for %d-channel allocation: %s\n", ca, channels, buf); @@ -725,9 +751,10 @@ static int hdmi_manual_setup_channel_mapping(struct hda_codec *codec, static void hdmi_setup_fake_chmap(unsigned char *map, int ca) { int i; + int ordered_ca = get_channel_allocation_order(ca); for (i = 0; i < 8; i++) { - if (i < channel_allocations[ca].channels) - map[i] = from_cea_slot((hdmi_channel_mapping[ca][i] >> 4) & 0x0f); + if (i < channel_allocations[ordered_ca].channels) + map[i] = from_cea_slot(hdmi_channel_mapping[ca][i] & 0x0f); else map[i] = 0; } @@ -868,18 +895,19 @@ static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, return true; } -static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, - bool non_pcm, - struct snd_pcm_substream *substream) +static void hdmi_setup_audio_infoframe(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin, + bool non_pcm) { - struct hdmi_spec *spec = codec->spec; - struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); hda_nid_t pin_nid = per_pin->pin_nid; - int channels = substream->runtime->channels; + int channels = per_pin->channels; struct hdmi_eld *eld; int ca; union audio_infoframe ai; + if (!channels) + return; + eld = &per_pin->sink_eld; if (!eld->monitor_present) return; @@ -916,6 +944,14 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, } /* + * always configure channel mapping, it may have been changed by the + * user in the meantime + */ + hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca, + channels, per_pin->chmap, + per_pin->chmap_set); + + /* * sizeof(ai) is used instead of sizeof(*hdmi_ai) or * sizeof(*dp_ai) to avoid partial match/update problems when * the user switches between HDMI/DP monitors. @@ -926,20 +962,10 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, "pin=%d channels=%d\n", pin_nid, channels); - hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca, - channels, per_pin->chmap, - per_pin->chmap_set); hdmi_stop_infoframe_trans(codec, pin_nid); hdmi_fill_audio_infoframe(codec, pin_nid, ai.bytes, sizeof(ai)); hdmi_start_infoframe_trans(codec, pin_nid); - } else { - /* For non-pcm audio switch, setup new channel mapping - * accordingly */ - if (per_pin->non_pcm != non_pcm) - hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca, - channels, per_pin->chmap, - per_pin->chmap_set); } per_pin->non_pcm = non_pcm; @@ -1146,7 +1172,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, per_cvt->assigned = 1; hinfo->nid = per_cvt->cvt_nid; - snd_hda_codec_write(codec, per_pin->pin_nid, 0, + snd_hda_codec_write_cache(codec, per_pin->pin_nid, 0, AC_VERB_SET_CONNECT_SEL, mux_idx); snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); @@ -1263,6 +1289,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) eld_changed = true; } if (update_eld) { + bool old_eld_valid = pin_eld->eld_valid; pin_eld->eld_valid = eld->eld_valid; eld_changed = pin_eld->eld_size != eld->eld_size || memcmp(pin_eld->eld_buffer, eld->eld_buffer, @@ -1272,6 +1299,18 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) eld->eld_size); pin_eld->eld_size = eld->eld_size; pin_eld->info = eld->info; + + /* Haswell-specific workaround: re-setup when the transcoder is + * changed during the stream playback + */ + if (codec->vendor_id == 0x80862807 && + eld->eld_valid && !old_eld_valid && per_pin->setup) { + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_UNMUTE); + hdmi_setup_audio_infoframe(codec, per_pin, + per_pin->non_pcm); + } } mutex_unlock(&pin_eld->lock); @@ -1444,14 +1483,26 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, hda_nid_t cvt_nid = hinfo->nid; struct hdmi_spec *spec = codec->spec; int pin_idx = hinfo_to_pin_index(spec, hinfo); - hda_nid_t pin_nid = get_pin(spec, pin_idx)->pin_nid; + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); + hda_nid_t pin_nid = per_pin->pin_nid; bool non_pcm; + int pinctl; non_pcm = check_non_pcm_per_cvt(codec, cvt_nid); + per_pin->channels = substream->runtime->channels; + per_pin->setup = true; hdmi_set_channel_count(codec, cvt_nid, substream->runtime->channels); - hdmi_setup_audio_infoframe(codec, pin_idx, non_pcm, substream); + hdmi_setup_audio_infoframe(codec, per_pin, non_pcm); + + if (spec->dyn_pin_out) { + pinctl = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + pinctl | PIN_OUT); + } return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); } @@ -1472,6 +1523,7 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, int cvt_idx, pin_idx; struct hdmi_spec_per_cvt *per_cvt; struct hdmi_spec_per_pin *per_pin; + int pinctl; if (hinfo->nid) { cvt_idx = cvt_nid_to_cvt_index(spec, hinfo->nid); @@ -1488,9 +1540,20 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, return -EINVAL; per_pin = get_pin(spec, pin_idx); + if (spec->dyn_pin_out) { + pinctl = snd_hda_codec_read(codec, per_pin->pin_nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_codec_write(codec, per_pin->pin_nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + pinctl & ~PIN_OUT); + } + snd_hda_spdif_ctls_unassign(codec, pin_idx); per_pin->chmap_set = false; memset(per_pin->chmap, 0, sizeof(per_pin->chmap)); + + per_pin->setup = false; + per_pin->channels = 0; } return 0; @@ -1626,8 +1689,7 @@ static int hdmi_chmap_ctl_put(struct snd_kcontrol *kcontrol, per_pin->chmap_set = true; memcpy(per_pin->chmap, chmap, sizeof(chmap)); if (prepared) - hdmi_setup_audio_infoframe(codec, pin_idx, per_pin->non_pcm, - substream); + hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); return 0; } @@ -1715,6 +1777,9 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) struct snd_pcm_chmap *chmap; struct snd_kcontrol *kctl; int i; + + if (!codec->pcm_info[pin_idx].pcm) + break; err = snd_pcm_add_chmap_ctls(codec->pcm_info[pin_idx].pcm, SNDRV_PCM_STREAM_PLAYBACK, NULL, 0, pin_idx, &chmap); @@ -1967,8 +2032,9 @@ static int simple_playback_build_controls(struct hda_codec *codec) int err; per_cvt = get_cvt(spec, 0); - err = snd_hda_create_spdif_out_ctls(codec, per_cvt->cvt_nid, - per_cvt->cvt_nid); + err = snd_hda_create_dig_out_ctls(codec, per_cvt->cvt_nid, + per_cvt->cvt_nid, + HDA_PCM_TYPE_HDMI); if (err < 0) return err; return simple_hdmi_build_jack(codec, 0); @@ -2441,6 +2507,21 @@ static int patch_nvhdmi_8ch_7x(struct hda_codec *codec) return 0; } +static int patch_nvhdmi(struct hda_codec *codec) +{ + struct hdmi_spec *spec; + int err; + + err = patch_generic_hdmi(codec); + if (err) + return err; + + spec = codec->spec; + spec->dyn_pin_out = true; + + return 0; +} + /* * ATI-specific implementations * @@ -2513,29 +2594,30 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x10de0005, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x }, { .id = 0x10de0006, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x }, { .id = 0x10de0007, .name = "MCP79/7A HDMI", .patch = patch_nvhdmi_8ch_7x }, -{ .id = 0x10de000a, .name = "GPU 0a HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de000b, .name = "GPU 0b HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de000c, .name = "MCP89 HDMI", .patch = patch_generic_hdmi }, -{ .id = 0x10de000d, .name = "GPU 0d HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0010, .name = "GPU 10 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0011, .name = "GPU 11 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0012, .name = "GPU 12 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0013, .name = "GPU 13 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0014, .name = "GPU 14 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0015, .name = "GPU 15 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0016, .name = "GPU 16 HDMI/DP", .patch = patch_generic_hdmi }, +{ .id = 0x10de000a, .name = "GPU 0a HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de000b, .name = "GPU 0b HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de000c, .name = "MCP89 HDMI", .patch = patch_nvhdmi }, +{ .id = 0x10de000d, .name = "GPU 0d HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0010, .name = "GPU 10 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0011, .name = "GPU 11 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0012, .name = "GPU 12 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0013, .name = "GPU 13 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0014, .name = "GPU 14 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0015, .name = "GPU 15 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0016, .name = "GPU 16 HDMI/DP", .patch = patch_nvhdmi }, /* 17 is known to be absent */ -{ .id = 0x10de0018, .name = "GPU 18 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0019, .name = "GPU 19 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de001a, .name = "GPU 1a HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de001b, .name = "GPU 1b HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de001c, .name = "GPU 1c HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0040, .name = "GPU 40 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0041, .name = "GPU 41 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0042, .name = "GPU 42 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0043, .name = "GPU 43 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0044, .name = "GPU 44 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0051, .name = "GPU 51 HDMI/DP", .patch = patch_generic_hdmi }, +{ .id = 0x10de0018, .name = "GPU 18 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0019, .name = "GPU 19 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de001a, .name = "GPU 1a HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de001b, .name = "GPU 1b HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de001c, .name = "GPU 1c HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0040, .name = "GPU 40 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0041, .name = "GPU 41 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0042, .name = "GPU 42 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0043, .name = "GPU 43 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0044, .name = "GPU 44 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0051, .name = "GPU 51 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0060, .name = "GPU 60 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x11069f80, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi }, @@ -2588,6 +2670,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0042"); MODULE_ALIAS("snd-hda-codec-id:10de0043"); MODULE_ALIAS("snd-hda-codec-id:10de0044"); MODULE_ALIAS("snd-hda-codec-id:10de0051"); +MODULE_ALIAS("snd-hda-codec-id:10de0060"); MODULE_ALIAS("snd-hda-codec-id:10de0067"); MODULE_ALIAS("snd-hda-codec-id:10de8001"); MODULE_ALIAS("snd-hda-codec-id:11069f80"); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 403010c9e82..e0bdcb3ecf0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1027,6 +1027,7 @@ enum { ALC880_FIXUP_GPIO2, ALC880_FIXUP_MEDION_RIM, ALC880_FIXUP_LG, + ALC880_FIXUP_LG_LW25, ALC880_FIXUP_W810, ALC880_FIXUP_EAPD_COEF, ALC880_FIXUP_TCL_S700, @@ -1036,6 +1037,7 @@ enum { ALC880_FIXUP_UNIWILL, ALC880_FIXUP_UNIWILL_DIG, ALC880_FIXUP_Z71V, + ALC880_FIXUP_ASUS_W5A, ALC880_FIXUP_3ST_BASE, ALC880_FIXUP_3ST, ALC880_FIXUP_3ST_DIG, @@ -1085,6 +1087,14 @@ static const struct hda_fixup alc880_fixups[] = { { } } }, + [ALC880_FIXUP_LG_LW25] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x0181344f }, /* line-in */ + { 0x1b, 0x0321403f }, /* headphone */ + { } + } + }, [ALC880_FIXUP_W810] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -1198,6 +1208,26 @@ static const struct hda_fixup alc880_fixups[] = { { } } }, + [ALC880_FIXUP_ASUS_W5A] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + /* set up the whole pins as BIOS is utterly broken */ + { 0x14, 0x0121411f }, /* HP */ + { 0x15, 0x411111f0 }, /* N/A */ + { 0x16, 0x411111f0 }, /* N/A */ + { 0x17, 0x411111f0 }, /* N/A */ + { 0x18, 0x90a60160 }, /* mic */ + { 0x19, 0x411111f0 }, /* N/A */ + { 0x1a, 0x411111f0 }, /* N/A */ + { 0x1b, 0x411111f0 }, /* N/A */ + { 0x1c, 0x411111f0 }, /* N/A */ + { 0x1d, 0x411111f0 }, /* N/A */ + { 0x1e, 0xb743111e }, /* SPDIF out */ + { } + }, + .chained = true, + .chain_id = ALC880_FIXUP_GPIO1, + }, [ALC880_FIXUP_3ST_BASE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -1319,6 +1349,7 @@ static const struct hda_fixup alc880_fixups[] = { static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_FIXUP_W810), + SND_PCI_QUIRK(0x1043, 0x10c3, "ASUS W5A", ALC880_FIXUP_ASUS_W5A), SND_PCI_QUIRK(0x1043, 0x1964, "ASUS Z71V", ALC880_FIXUP_Z71V), SND_PCI_QUIRK_VENDOR(0x1043, "ASUS", ALC880_FIXUP_GPIO1), SND_PCI_QUIRK(0x1558, 0x5401, "Clevo GPIO2", ALC880_FIXUP_GPIO2), @@ -1337,6 +1368,7 @@ static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_FIXUP_LG), SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_FIXUP_LG), SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_FIXUP_LG), + SND_PCI_QUIRK(0x1854, 0x0077, "LG LW25", ALC880_FIXUP_LG_LW25), SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_FIXUP_TCL_S700), /* Below is the copied entries from alc880_quirks.c. @@ -1463,6 +1495,7 @@ enum { ALC260_FIXUP_KN1, ALC260_FIXUP_FSC_S7020, ALC260_FIXUP_FSC_S7020_JWSE, + ALC260_FIXUP_VAIO_PINS, }; static void alc260_gpio1_automute(struct hda_codec *codec) @@ -1603,6 +1636,24 @@ static const struct hda_fixup alc260_fixups[] = { .chained = true, .chain_id = ALC260_FIXUP_FSC_S7020, }, + [ALC260_FIXUP_VAIO_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + /* Pin configs are missing completely on some VAIOs */ + { 0x0f, 0x01211020 }, + { 0x10, 0x0001003f }, + { 0x11, 0x411111f0 }, + { 0x12, 0x01a15930 }, + { 0x13, 0x411111f0 }, + { 0x14, 0x411111f0 }, + { 0x15, 0x411111f0 }, + { 0x16, 0x411111f0 }, + { 0x17, 0x411111f0 }, + { 0x18, 0x411111f0 }, + { 0x19, 0x411111f0 }, + { } + } + }, }; static const struct snd_pci_quirk alc260_fixup_tbl[] = { @@ -1611,6 +1662,8 @@ static const struct snd_pci_quirk alc260_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_FIXUP_GPIO1), SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", ALC260_FIXUP_HP_DC5750), SND_PCI_QUIRK(0x103c, 0x30ba, "HP Presario B1900", ALC260_FIXUP_HP_B1900), + SND_PCI_QUIRK(0x104d, 0x81bb, "Sony VAIO", ALC260_FIXUP_VAIO_PINS), + SND_PCI_QUIRK(0x104d, 0x81e2, "Sony VAIO TX", ALC260_FIXUP_HP_PIN_0F), SND_PCI_QUIRK(0x10cf, 0x1326, "FSC LifeBook S7020", ALC260_FIXUP_FSC_S7020), SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FIXUP_GPIO1), SND_PCI_QUIRK(0x152d, 0x0729, "Quanta KN1", ALC260_FIXUP_KN1), @@ -1710,8 +1763,12 @@ enum { ALC889_FIXUP_DAC_ROUTE, ALC889_FIXUP_MBP_VREF, ALC889_FIXUP_IMAC91_VREF, + ALC889_FIXUP_MBA11_VREF, + ALC889_FIXUP_MBA21_VREF, + ALC889_FIXUP_MP11_VREF, ALC882_FIXUP_INV_DMIC, ALC882_FIXUP_NO_PRIMARY_HP, + ALC887_FIXUP_ASUS_BASS, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -1812,17 +1869,13 @@ static void alc889_fixup_mbp_vref(struct hda_codec *codec, } } -/* Set VREF on speaker pins on imac91 */ -static void alc889_fixup_imac91_vref(struct hda_codec *codec, - const struct hda_fixup *fix, int action) +static void alc889_fixup_mac_pins(struct hda_codec *codec, + const hda_nid_t *nids, int num_nids) { struct alc_spec *spec = codec->spec; - static hda_nid_t nids[2] = { 0x18, 0x1a }; int i; - if (action != HDA_FIXUP_ACT_INIT) - return; - for (i = 0; i < ARRAY_SIZE(nids); i++) { + for (i = 0; i < num_nids; i++) { unsigned int val; val = snd_hda_codec_get_pin_target(codec, nids[i]); val |= AC_PINCTL_VREF_50; @@ -1831,6 +1884,36 @@ static void alc889_fixup_imac91_vref(struct hda_codec *codec, spec->gen.keep_vref_in_automute = 1; } +/* Set VREF on speaker pins on imac91 */ +static void alc889_fixup_imac91_vref(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static hda_nid_t nids[2] = { 0x18, 0x1a }; + + if (action == HDA_FIXUP_ACT_INIT) + alc889_fixup_mac_pins(codec, nids, ARRAY_SIZE(nids)); +} + +/* Set VREF on speaker pins on mba11 */ +static void alc889_fixup_mba11_vref(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static hda_nid_t nids[1] = { 0x18 }; + + if (action == HDA_FIXUP_ACT_INIT) + alc889_fixup_mac_pins(codec, nids, ARRAY_SIZE(nids)); +} + +/* Set VREF on speaker pins on mba21 */ +static void alc889_fixup_mba21_vref(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static hda_nid_t nids[2] = { 0x18, 0x19 }; + + if (action == HDA_FIXUP_ACT_INIT) + alc889_fixup_mac_pins(codec, nids, ARRAY_SIZE(nids)); +} + /* Don't take HP output as primary * Strangely, the speaker output doesn't work on Vaio Z and some Vaio * all-in-one desktop PCs (for example VGC-LN51JGB) through DAC 0x05 @@ -2025,6 +2108,24 @@ static const struct hda_fixup alc882_fixups[] = { .chained = true, .chain_id = ALC882_FIXUP_GPIO1, }, + [ALC889_FIXUP_MBA11_VREF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc889_fixup_mba11_vref, + .chained = true, + .chain_id = ALC889_FIXUP_MBP_VREF, + }, + [ALC889_FIXUP_MBA21_VREF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc889_fixup_mba21_vref, + .chained = true, + .chain_id = ALC889_FIXUP_MBP_VREF, + }, + [ALC889_FIXUP_MP11_VREF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc889_fixup_mba11_vref, + .chained = true, + .chain_id = ALC885_FIXUP_MACPRO_GPIO, + }, [ALC882_FIXUP_INV_DMIC] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic_0x12, @@ -2033,6 +2134,13 @@ static const struct hda_fixup alc882_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc882_fixup_no_primary_hp, }, + [ALC887_FIXUP_ASUS_BASS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + {0x16, 0x99130130}, /* bass speaker */ + {} + }, + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -2066,6 +2174,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V), SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_FIXUP_ASUS_W2JC), SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601), + SND_PCI_QUIRK(0x1043, 0x84bc, "ASUS ET2700", ALC887_FIXUP_ASUS_BASS), SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP), SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP), @@ -2074,14 +2183,14 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x00a1, "Macbook", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x00a4, "MacbookPro 4,1", ALC889_FIXUP_MBP_VREF), - SND_PCI_QUIRK(0x106b, 0x0c00, "Mac Pro", ALC885_FIXUP_MACPRO_GPIO), + SND_PCI_QUIRK(0x106b, 0x0c00, "Mac Pro", ALC889_FIXUP_MP11_VREF), SND_PCI_QUIRK(0x106b, 0x1000, "iMac 24", ALC885_FIXUP_MACPRO_GPIO), SND_PCI_QUIRK(0x106b, 0x2800, "AppleTV", ALC885_FIXUP_MACPRO_GPIO), SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x3000, "iMac", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x3200, "iMac 7,1 Aluminum", ALC882_FIXUP_EAPD), - SND_PCI_QUIRK(0x106b, 0x3400, "MacBookAir 1,1", ALC889_FIXUP_MBP_VREF), - SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC889_FIXUP_MBP_VREF), + SND_PCI_QUIRK(0x106b, 0x3400, "MacBookAir 1,1", ALC889_FIXUP_MBA11_VREF), + SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC889_FIXUP_MBA21_VREF), SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_FIXUP_MACPRO_GPIO), @@ -2370,6 +2479,7 @@ static const struct hda_verb alc268_beep_init_verbs[] = { enum { ALC268_FIXUP_INV_DMIC, ALC268_FIXUP_HP_EAPD, + ALC268_FIXUP_SPDIF, }; static const struct hda_fixup alc268_fixups[] = { @@ -2384,6 +2494,13 @@ static const struct hda_fixup alc268_fixups[] = { {} } }, + [ALC268_FIXUP_SPDIF] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1e, 0x014b1180 }, /* enable SPDIF out */ + {} + } + }, }; static const struct hda_model_fixup alc268_fixup_models[] = { @@ -2393,6 +2510,7 @@ static const struct hda_model_fixup alc268_fixup_models[] = { }; static const struct snd_pci_quirk alc268_fixup_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x0139, "Acer TravelMate 6293", ALC268_FIXUP_SPDIF), SND_PCI_QUIRK(0x1025, 0x015b, "Acer AOA 150 (ZG5)", ALC268_FIXUP_INV_DMIC), /* below is codec SSID since multiple Toshiba laptops have the * same PCI SSID 1179:ff00 @@ -2521,6 +2639,7 @@ enum { ALC269_TYPE_ALC282, ALC269_TYPE_ALC284, ALC269_TYPE_ALC286, + ALC269_TYPE_ALC255, }; /* @@ -2545,6 +2664,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) case ALC269_TYPE_ALC269VD: case ALC269_TYPE_ALC282: case ALC269_TYPE_ALC286: + case ALC269_TYPE_ALC255: ssids = alc269_ssids; break; default: @@ -2744,6 +2864,23 @@ static void alc269_fixup_mic_mute_hook(void *private_data, int enabled) snd_hda_set_pin_ctl_cache(codec, spec->mute_led_nid, pinval); } +/* Make sure the led works even in runtime suspend */ +static unsigned int led_power_filter(struct hda_codec *codec, + hda_nid_t nid, + unsigned int power_state) +{ + struct alc_spec *spec = codec->spec; + + if (power_state != AC_PWRST_D3 || nid != spec->mute_led_nid) + return power_state; + + /* Set pin ctl again, it might have just been set to 0 */ + snd_hda_set_pin_ctl(codec, nid, + snd_hda_codec_get_pin_target(codec, nid)); + + return AC_PWRST_D0; +} + static void alc269_fixup_hp_mute_led(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -2763,6 +2900,7 @@ static void alc269_fixup_hp_mute_led(struct hda_codec *codec, spec->mute_led_nid = pin - 0x0a + 0x18; spec->gen.vmaster_mute.hook = alc269_fixup_mic_mute_hook; spec->gen.vmaster_mute_enum = 1; + codec->power_filter = led_power_filter; snd_printd("Detected mute LED for %x:%d\n", spec->mute_led_nid, spec->mute_led_polarity); break; @@ -2778,6 +2916,7 @@ static void alc269_fixup_hp_mute_led_mic1(struct hda_codec *codec, spec->mute_led_nid = 0x18; spec->gen.vmaster_mute.hook = alc269_fixup_mic_mute_hook; spec->gen.vmaster_mute_enum = 1; + codec->power_filter = led_power_filter; } } @@ -2790,6 +2929,7 @@ static void alc269_fixup_hp_mute_led_mic2(struct hda_codec *codec, spec->mute_led_nid = 0x19; spec->gen.vmaster_mute.hook = alc269_fixup_mic_mute_hook; spec->gen.vmaster_mute_enum = 1; + codec->power_filter = led_power_filter; } } @@ -2948,6 +3088,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) alc_write_coef_idx(codec, 0x18, 0x7388); break; case 0x10ec0668: + alc_write_coef_idx(codec, 0x11, 0x0001); alc_write_coef_idx(codec, 0x15, 0x0d60); alc_write_coef_idx(codec, 0xc3, 0x0000); break; @@ -2970,6 +3111,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) alc_write_coef_idx(codec, 0x18, 0x7388); break; case 0x10ec0668: + alc_write_coef_idx(codec, 0x11, 0x0001); alc_write_coef_idx(codec, 0x15, 0x0d50); alc_write_coef_idx(codec, 0xc3, 0x0000); break; @@ -3030,8 +3172,10 @@ static void alc_update_headset_mode(struct hda_codec *codec) else new_headset_mode = ALC_HEADSET_MODE_HEADPHONE; - if (new_headset_mode == spec->current_headset_mode) + if (new_headset_mode == spec->current_headset_mode) { + snd_hda_gen_update_outputs(codec); return; + } switch (new_headset_mode) { case ALC_HEADSET_MODE_UNPLUGGED: @@ -3190,6 +3334,15 @@ static void alc269_fixup_limit_int_mic_boost(struct hda_codec *codec, } } +static void alc290_fixup_mono_speakers(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) + /* Remove DAC node 0x03, as it seems to be + giving mono output */ + snd_hda_override_wcaps(codec, 0x03, 0); +} + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -3213,9 +3366,12 @@ enum { ALC269_FIXUP_HP_GPIO_LED, ALC269_FIXUP_INV_DMIC, ALC269_FIXUP_LENOVO_DOCK, + ALC286_FIXUP_SONY_MIC_NO_PRESENCE, ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, + ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, + ALC290_FIXUP_MONO_SPEAKERS, ALC269_FIXUP_HEADSET_MODE, ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, ALC269_FIXUP_ASUS_X101_FUNC, @@ -3402,6 +3558,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC269_FIXUP_DELL3_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, [ALC269_FIXUP_HEADSET_MODE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode, @@ -3410,6 +3575,13 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode_no_hp_mic, }, + [ALC286_FIXUP_SONY_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_x101_headset_mic, @@ -3467,6 +3639,12 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_limit_int_mic_boost, }, + [ALC290_FIXUP_MONO_SPEAKERS] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc290_fixup_mono_speakers, + .chained = true, + .chain_id = ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -3495,9 +3673,15 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f8, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05f9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05fb, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0614, "Dell Inspiron 3135", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_MONO_SPEAKERS), + SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), @@ -3516,6 +3700,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x8516, "ASUS X101CH", ALC269_FIXUP_ASUS_X101), + SND_PCI_QUIRK(0x104d, 0x90b5, "Sony VAIO Pro 11", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x104d, 0x90b6, "Sony VAIO Pro 13", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2), SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), @@ -3716,6 +3902,9 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0286: spec->codec_variant = ALC269_TYPE_ALC286; break; + case 0x10ec0255: + spec->codec_variant = ALC269_TYPE_ALC255; + break; } /* automatic parse from the BIOS config */ @@ -3758,6 +3947,7 @@ enum { ALC861_FIXUP_AMP_VREF_0F, ALC861_FIXUP_NO_JACK_DETECT, ALC861_FIXUP_ASUS_A6RP, + ALC660_FIXUP_ASUS_W7J, }; /* On some laptops, VREF of pin 0x0f is abused for controlling the main amp */ @@ -3807,10 +3997,22 @@ static const struct hda_fixup alc861_fixups[] = { .v.func = alc861_fixup_asus_amp_vref_0f, .chained = true, .chain_id = ALC861_FIXUP_NO_JACK_DETECT, + }, + [ALC660_FIXUP_ASUS_W7J] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* ASUS W7J needs a magic pin setup on unused NID 0x10 + * for enabling outputs + */ + {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + { } + }, } }; static const struct snd_pci_quirk alc861_fixup_tbl[] = { + SND_PCI_QUIRK(0x1043, 0x1253, "ASUS W7J", ALC660_FIXUP_ASUS_W7J), + SND_PCI_QUIRK(0x1043, 0x1263, "ASUS Z35HL", ALC660_FIXUP_ASUS_W7J), SND_PCI_QUIRK(0x1043, 0x1393, "ASUS A6Rp", ALC861_FIXUP_ASUS_A6RP), SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", ALC861_FIXUP_AMP_VREF_0F), SND_PCI_QUIRK(0x1462, 0x7254, "HP DX2200", ALC861_FIXUP_NO_JACK_DETECT), @@ -4194,13 +4396,17 @@ static const struct hda_fixup alc662_fixups[] = { static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_FIXUP_ASUS_MODE2), + SND_PCI_QUIRK(0x1025, 0x022f, "Acer Aspire One", ALC662_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x1025, 0x0349, "eMachines eM250", ALC662_FIXUP_INV_DMIC), + SND_PCI_QUIRK(0x1025, 0x034a, "Gateway LT27", ALC662_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), + SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_ASUS_MODE4), + SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_ASUS_MODE4), SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_FIXUP_ASUS_MODE2), SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), @@ -4361,6 +4567,7 @@ static int patch_alc662(struct hda_codec *codec) case 0x10ec0272: case 0x10ec0663: case 0x10ec0665: + case 0x10ec0668: set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); break; case 0x10ec0273: @@ -4418,7 +4625,9 @@ static int patch_alc680(struct hda_codec *codec) */ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0221, .name = "ALC221", .patch = patch_alc269 }, + { .id = 0x10ec0231, .name = "ALC231", .patch = patch_alc269 }, { .id = 0x10ec0233, .name = "ALC233", .patch = patch_alc269 }, + { .id = 0x10ec0255, .name = "ALC255", .patch = patch_alc269 }, { .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 }, { .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 }, { .id = 0x10ec0267, .name = "ALC267", .patch = patch_alc268 }, diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 1d9d6427e0b..dc4833f47a2 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -417,9 +417,11 @@ static void stac_update_outputs(struct hda_codec *codec) val &= ~spec->eapd_mask; else val |= spec->eapd_mask; - if (spec->gpio_data != val) + if (spec->gpio_data != val) { + spec->gpio_data = val; stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, val); + } } } @@ -2813,6 +2815,7 @@ static const struct hda_pintbl ecs202_pin_configs[] = { /* codec SSIDs for Intel Mac sharing the same PCI SSID 8384:7680 */ static const struct snd_pci_quirk stac922x_intel_mac_fixup_tbl[] = { + SND_PCI_QUIRK(0x0000, 0x0100, "Mac Mini", STAC_INTEL_MAC_V3), SND_PCI_QUIRK(0x106b, 0x0800, "Mac", STAC_INTEL_MAC_V1), SND_PCI_QUIRK(0x106b, 0x0600, "Mac", STAC_INTEL_MAC_V2), SND_PCI_QUIRK(0x106b, 0x0700, "Mac", STAC_INTEL_MAC_V2), @@ -3227,7 +3230,7 @@ static const struct hda_fixup stac927x_fixups[] = { /* configure the analog microphone on some laptops */ { 0x0c, 0x90a79130 }, /* correct the front output jack as a hp out */ - { 0x0f, 0x0227011f }, + { 0x0f, 0x0221101f }, /* correct the front input jack as a mic */ { 0x0e, 0x02a79130 }, {} @@ -3608,20 +3611,18 @@ static int stac_parse_auto_config(struct hda_codec *codec) static int stac_init(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - unsigned int gpio; int i; /* override some hints */ stac_store_hints(codec); /* set up GPIO */ - gpio = spec->gpio_data; /* turn on EAPD statically when spec->eapd_switch isn't set. * otherwise, unsol event will turn it on/off dynamically */ if (!spec->eapd_switch) - gpio |= spec->eapd_mask; - stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, gpio); + spec->gpio_data |= spec->eapd_mask; + stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data); snd_hda_gen_init(codec); @@ -3921,6 +3922,7 @@ static void stac_setup_gpio(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; + spec->gpio_mask |= spec->eapd_mask; if (spec->gpio_led) { if (!spec->vref_mute_led_nid) { spec->gpio_mask |= spec->gpio_led; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index e5245544eb5..aed19c3f846 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -910,6 +910,8 @@ static const struct hda_verb vt1708S_init_verbs[] = { static void override_mic_boost(struct hda_codec *codec, hda_nid_t pin, int offset, int num_steps, int step_size) { + snd_hda_override_wcaps(codec, pin, + get_wcaps(codec, pin) | AC_WCAP_IN_AMP); snd_hda_override_amp_caps(codec, pin, HDA_INPUT, (offset << AC_AMPCAP_OFFSET_SHIFT) | (num_steps << AC_AMPCAP_NUM_STEPS_SHIFT) | diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index 773a67fff4c..431bf6897dd 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -285,7 +285,7 @@ static char channel_map_9636_ds[26] = { /* ADAT channels are remapped */ 1, 3, 5, 7, 9, 11, 13, 15, /* channels 8 and 9 are S/PDIF */ - 24, 25 + 24, 25, /* others don't exist */ -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1 }; diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c index 1d38fd0bc4e..d1282652679 100644 --- a/sound/soc/atmel/atmel-pcm-dma.c +++ b/sound/soc/atmel/atmel-pcm-dma.c @@ -81,7 +81,9 @@ static void atmel_pcm_dma_irq(u32 ssc_sr, /* stop RX and capture: will be enabled again at restart */ ssc_writex(prtd->ssc->regs, SSC_CR, prtd->mask->ssc_disable); + snd_pcm_stream_lock(substream); snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock(substream); /* now drain RHR and read status to remove xrun condition */ ssc_readx(prtd->ssc->regs, SSC_RHR); diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index dd0c2a4f83a..e0869aaa1e9 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -111,6 +111,7 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, bf5xx_i2s->tcr2 |= 7; bf5xx_i2s->rcr2 |= 7; sport_handle->wdsize = 1; + break; case SNDRV_PCM_FORMAT_S16_LE: bf5xx_i2s->tcr2 |= 15; bf5xx_i2s->rcr2 |= 15; diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 60159c07448..6fd174be3bd 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -351,6 +351,9 @@ static int snd_soc_put_volsw_2r_st(struct snd_kcontrol *kcontrol, val = ucontrol->value.integer.value[0]; val2 = ucontrol->value.integer.value[1]; + if (val >= ARRAY_SIZE(st_table) || val2 >= ARRAY_SIZE(st_table)) + return -EINVAL; + err = snd_soc_update_bits(codec, reg, 0x3f, st_table[val].m); if (err < 0) return err; diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index a153b168129..bce45c197e1 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -1225,13 +1225,18 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol, struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev); struct device *dev = codec->dev; bool apply_fir, apply_iir; - int req, status; + unsigned int req; + int status; dev_dbg(dev, "%s: Enter.\n", __func__); mutex_lock(&drvdata->anc_lock); req = ucontrol->value.integer.value[0]; + if (req >= ARRAY_SIZE(enum_anc_state)) { + status = -EINVAL; + goto cleanup; + } if (req != ANC_APPLY_FIR_IIR && req != ANC_APPLY_FIR && req != ANC_APPLY_IIR) { dev_err(dev, "%s: ERROR: Unsupported status to set '%s'!\n", diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index dafdbe87ede..0c499c63869 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -64,7 +64,7 @@ #define ADAU1701_SEROCTL_WORD_LEN_24 0x0000 #define ADAU1701_SEROCTL_WORD_LEN_20 0x0001 -#define ADAU1701_SEROCTL_WORD_LEN_16 0x0010 +#define ADAU1701_SEROCTL_WORD_LEN_16 0x0002 #define ADAU1701_SEROCTL_WORD_LEN_MASK 0x0003 #define ADAU1701_AUXNPOW_VBPD 0x40 diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 2d037870970..687565d08d9 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -257,7 +257,7 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * This operation came from example code of * "ASAHI KASEI AK4642" (japanese) manual p94. */ - snd_soc_write(codec, SG_SL1, PMMP | MGAIN0); + snd_soc_update_bits(codec, SG_SL1, PMMP | MGAIN0, PMMP | MGAIN0); snd_soc_write(codec, TIMER, ZTM(0x3) | WTM(0x3)); snd_soc_write(codec, ALC_CTL1, ALC | LMTH0); snd_soc_update_bits(codec, PW_MGMT1, PMADL, PMADL); diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 389f2325383..663a2a74862 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1455,6 +1455,8 @@ static void arizona_enable_fll(struct arizona_fll *fll, try_wait_for_completion(&fll->ok); regmap_update_bits(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_FREERUN, 0); + regmap_update_bits(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); if (fll->ref_src >= 0 && fll->sync_src >= 0 && fll->ref_src != fll->sync_src) @@ -1473,6 +1475,8 @@ static void arizona_disable_fll(struct arizona_fll *fll) struct arizona *arizona = fll->arizona; bool change; + regmap_update_bits(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_FREERUN, ARIZONA_FLL1_FREERUN); regmap_update_bits_check(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, 0, &change); regmap_update_bits(arizona->regmap, fll->base + 0x11, diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 987f728718c..ee25f325d65 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -451,7 +451,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { SOC_ENUM("Beep Pitch", beep_pitch_enum), SOC_ENUM("Beep on Time", beep_ontime_enum), SOC_ENUM("Beep off Time", beep_offtime_enum), - SOC_SINGLE_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x1f, 0x07, hl_tlv), + SOC_SINGLE_SX_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x07, 0x1f, hl_tlv), SOC_SINGLE("Beep Mixer Switch", CS42L52_BEEP_TONE_CTL, 5, 1, 1), SOC_ENUM("Beep Treble Corner Freq", beep_treble_enum), SOC_ENUM("Beep Bass Corner Freq", beep_bass_enum), diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h index 4277012c471..a935d7381af 100644 --- a/sound/soc/codecs/cs42l52.h +++ b/sound/soc/codecs/cs42l52.h @@ -179,7 +179,7 @@ #define CS42L52_MICB_CTL 0x11 #define CS42L52_MIC_CTL_MIC_SEL_MASK 0xBF #define CS42L52_MIC_CTL_MIC_SEL_SHIFT 6 -#define CS42L52_MIC_CTL_TYPE_MASK 0xDF +#define CS42L52_MIC_CTL_TYPE_MASK 0x20 #define CS42L52_MIC_CTL_TYPE_SHIFT 5 diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 3eeada57e87..566a367c94f 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1612,7 +1612,7 @@ static int max98088_dai2_digital_mute(struct snd_soc_dai *codec_dai, int mute) static void max98088_sync_cache(struct snd_soc_codec *codec) { - u16 *reg_cache = codec->reg_cache; + u8 *reg_cache = codec->reg_cache; int i; if (!codec->cache_sync) diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 41cdd164297..8dbcacd44e6 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -1863,7 +1863,7 @@ static int max98095_put_eq_enum(struct snd_kcontrol *kcontrol, struct max98095_pdata *pdata = max98095->pdata; int channel = max98095_get_eq_channel(kcontrol->id.name); struct max98095_cdata *cdata; - int sel = ucontrol->value.integer.value[0]; + unsigned int sel = ucontrol->value.integer.value[0]; struct max98095_eq_cfg *coef_set; int fs, best, best_val, i; int regmask, regsave; @@ -2016,7 +2016,7 @@ static int max98095_put_bq_enum(struct snd_kcontrol *kcontrol, struct max98095_pdata *pdata = max98095->pdata; int channel = max98095_get_bq_channel(codec, kcontrol->id.name); struct max98095_cdata *cdata; - int sel = ucontrol->value.integer.value[0]; + unsigned int sel = ucontrol->value.integer.value[0]; struct max98095_biquad_cfg *coef_set; int fs, best, best_val, i; int regmask, regsave; diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 5402dfbbb71..8a8d9364e87 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -126,6 +126,10 @@ static int mc13783_write(struct snd_soc_codec *codec, ret = mc13xxx_reg_write(priv->mc13xxx, reg, value); + /* include errata fix for spi audio problems */ + if (reg == MC13783_AUDIO_CODEC || reg == MC13783_AUDIO_DAC) + ret = mc13xxx_reg_write(priv->mc13xxx, reg, value); + mc13xxx_unlock(priv->mc13xxx); return ret; diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 92bbfec9b10..ea479388fb5 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -37,7 +37,7 @@ static const u16 sgtl5000_regs[SGTL5000_MAX_REG_OFFSET] = { [SGTL5000_CHIP_CLK_CTRL] = 0x0008, [SGTL5000_CHIP_I2S_CTRL] = 0x0010, - [SGTL5000_CHIP_SSS_CTRL] = 0x0008, + [SGTL5000_CHIP_SSS_CTRL] = 0x0010, [SGTL5000_CHIP_DAC_VOL] = 0x3c3c, [SGTL5000_CHIP_PAD_STRENGTH] = 0x015f, [SGTL5000_CHIP_ANA_HP_CTRL] = 0x1818, diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index 8a9f43534b7..d3a68bbfea0 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -347,7 +347,7 @@ #define SGTL5000_PLL_INT_DIV_MASK 0xf800 #define SGTL5000_PLL_INT_DIV_SHIFT 11 #define SGTL5000_PLL_INT_DIV_WIDTH 5 -#define SGTL5000_PLL_FRAC_DIV_MASK 0x0700 +#define SGTL5000_PLL_FRAC_DIV_MASK 0x07ff #define SGTL5000_PLL_FRAC_DIV_SHIFT 0 #define SGTL5000_PLL_FRAC_DIV_WIDTH 11 diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 88ad7db52dd..3775394c9c8 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -37,6 +37,95 @@ struct wm5110_priv { struct arizona_fll fll[2]; }; +static const struct reg_default wm5110_sysclk_revd_patch[] = { + { 0x3093, 0x1001 }, + { 0x30E3, 0x1301 }, + { 0x3133, 0x1201 }, + { 0x3183, 0x1501 }, + { 0x31D3, 0x1401 }, + { 0x0049, 0x01ea }, + { 0x004a, 0x01f2 }, + { 0x0057, 0x01e7 }, + { 0x0058, 0x01fb }, + { 0x33ce, 0xc4f5 }, + { 0x33cf, 0x1361 }, + { 0x33d0, 0x0402 }, + { 0x33d1, 0x4700 }, + { 0x33d2, 0x026d }, + { 0x33d3, 0xff00 }, + { 0x33d4, 0x026d }, + { 0x33d5, 0x0101 }, + { 0x33d6, 0xc4f5 }, + { 0x33d7, 0x0361 }, + { 0x33d8, 0x0402 }, + { 0x33d9, 0x6701 }, + { 0x33da, 0xc4f5 }, + { 0x33db, 0x136f }, + { 0x33dc, 0xc4f5 }, + { 0x33dd, 0x134f }, + { 0x33de, 0xc4f5 }, + { 0x33df, 0x131f }, + { 0x33e0, 0x026d }, + { 0x33e1, 0x4f01 }, + { 0x33e2, 0x026d }, + { 0x33e3, 0xf100 }, + { 0x33e4, 0x026d }, + { 0x33e5, 0x0001 }, + { 0x33e6, 0xc4f5 }, + { 0x33e7, 0x0361 }, + { 0x33e8, 0x0402 }, + { 0x33e9, 0x6601 }, + { 0x33ea, 0xc4f5 }, + { 0x33eb, 0x136f }, + { 0x33ec, 0xc4f5 }, + { 0x33ed, 0x134f }, + { 0x33ee, 0xc4f5 }, + { 0x33ef, 0x131f }, + { 0x33f0, 0x026d }, + { 0x33f1, 0x4e01 }, + { 0x33f2, 0x026d }, + { 0x33f3, 0xf000 }, + { 0x33f6, 0xc4f5 }, + { 0x33f7, 0x1361 }, + { 0x33f8, 0x0402 }, + { 0x33f9, 0x4600 }, + { 0x33fa, 0x026d }, + { 0x33fb, 0xfe00 }, +}; + +static int wm5110_sysclk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + struct regmap *regmap = codec->control_data; + const struct reg_default *patch = NULL; + int i, patch_size; + + switch (arizona->rev) { + case 3: + patch = wm5110_sysclk_revd_patch; + patch_size = ARRAY_SIZE(wm5110_sysclk_revd_patch); + break; + default: + return 0; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (patch) + for (i = 0; i < patch_size; i++) + regmap_write(regmap, patch[i].reg, + patch[i].def); + break; + + default: + break; + } + + return 0; +} + static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); @@ -386,7 +475,7 @@ static const struct snd_kcontrol_new wm5110_aec_loopback_mux = static const struct snd_soc_dapm_widget wm5110_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, - 0, NULL, 0), + 0, wm5110_sysclk_ev, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1, ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK, @@ -856,7 +945,7 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "HPOUT2R", NULL, "OUT2R" }, { "HPOUT3L", NULL, "OUT3L" }, - { "HPOUT3R", NULL, "OUT3L" }, + { "HPOUT3R", NULL, "OUT3R" }, { "SPKOUTLN", NULL, "OUT4L" }, { "SPKOUTLP", NULL, "OUT4L" }, diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 5276062d6c7..10d492b6a5b 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -407,10 +407,10 @@ static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai, iface |= 0x0001; break; case SND_SOC_DAIFMT_DSP_A: - iface |= 0x0003; + iface |= 0x0013; break; case SND_SOC_DAIFMT_DSP_B: - iface |= 0x0013; + iface |= 0x0003; break; default: return -EINVAL; diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 3ff195c541d..af62f843a69 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1449,7 +1449,7 @@ static int wm8904_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: - aif1 |= WM8904_AIF_LRCLK_INV; + aif1 |= 0x3 | WM8904_AIF_LRCLK_INV; case SND_SOC_DAIFMT_DSP_A: aif1 |= 0x3; break; diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 0a4ffdd1d2a..5e5af898f7f 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -857,9 +857,9 @@ static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, if (pll_div.k) { reg |= 0x20; - snd_soc_write(codec, WM8960_PLL2, (pll_div.k >> 18) & 0x3f); - snd_soc_write(codec, WM8960_PLL3, (pll_div.k >> 9) & 0x1ff); - snd_soc_write(codec, WM8960_PLL4, pll_div.k & 0x1ff); + snd_soc_write(codec, WM8960_PLL2, (pll_div.k >> 16) & 0xff); + snd_soc_write(codec, WM8960_PLL3, (pll_div.k >> 8) & 0xff); + snd_soc_write(codec, WM8960_PLL4, pll_div.k & 0xff); } snd_soc_write(codec, WM8960_PLL1, reg); diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index e9710280e5e..e3cd86514ce 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1600,7 +1600,6 @@ static int wm8962_put_hp_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - u16 *reg_cache = codec->reg_cache; int ret; /* Apply the update (if any) */ @@ -1609,16 +1608,19 @@ static int wm8962_put_hp_sw(struct snd_kcontrol *kcontrol, return 0; /* If the left PGA is enabled hit that VU bit... */ - if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTL_PGA_ENA) - return snd_soc_write(codec, WM8962_HPOUTL_VOLUME, - reg_cache[WM8962_HPOUTL_VOLUME]); + ret = snd_soc_read(codec, WM8962_PWR_MGMT_2); + if (ret & WM8962_HPOUTL_PGA_ENA) { + snd_soc_write(codec, WM8962_HPOUTL_VOLUME, + snd_soc_read(codec, WM8962_HPOUTL_VOLUME)); + return 1; + } /* ...otherwise the right. The VU is stereo. */ - if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTR_PGA_ENA) - return snd_soc_write(codec, WM8962_HPOUTR_VOLUME, - reg_cache[WM8962_HPOUTR_VOLUME]); + if (ret & WM8962_HPOUTR_PGA_ENA) + snd_soc_write(codec, WM8962_HPOUTR_VOLUME, + snd_soc_read(codec, WM8962_HPOUTR_VOLUME)); - return 0; + return 1; } /* The VU bits for the speakers are in a different register to the mute @@ -3374,7 +3376,6 @@ static int wm8962_probe(struct snd_soc_codec *codec) int ret; struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); struct wm8962_pdata *pdata = dev_get_platdata(codec->dev); - u16 *reg_cache = codec->reg_cache; int i, trigger, irq_pol; bool dmicclk, dmicdat; @@ -3432,8 +3433,9 @@ static int wm8962_probe(struct snd_soc_codec *codec) /* Put the speakers into mono mode? */ if (pdata->spk_mono) - reg_cache[WM8962_CLASS_D_CONTROL_2] - |= WM8962_SPK_MONO; + snd_soc_update_bits(codec, WM8962_CLASS_D_CONTROL_2, + WM8962_SPK_MONO_MASK, WM8962_SPK_MONO); + /* Micbias setup, detection enable and detection * threasholds. */ @@ -3684,6 +3686,8 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, if (ret < 0) goto err_enable; + regcache_cache_only(wm8962->regmap, true); + /* The drivers should power up as needed */ regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 837978e16e9..ded9ed854a1 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1264,6 +1264,8 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ snd_soc_write(codec, WM8990_ANTIPOP2, 0x0); + + codec->cache_sync = 1; break; } diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 3470b649c0b..6dbb17d050c 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1073,13 +1073,17 @@ static int wm_adsp2_ena(struct wm_adsp *dsp) return ret; /* Wait for the RAM to start, should be near instantaneous */ - count = 0; - do { + for (count = 0; count < 10; ++count) { ret = regmap_read(dsp->regmap, dsp->base + ADSP2_STATUS1, &val); if (ret != 0) return ret; - } while (!(val & ADSP2_RAM_RDY) && ++count < 10); + + if (val & ADSP2_RAM_RDY) + break; + + msleep(1); + } if (!(val & ADSP2_RAM_RDY)) { adsp_err(dsp, "Failed to start DSP RAM\n"); diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index f5d81b94875..7a0466eb7ed 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -530,6 +530,7 @@ static int hp_supply_event(struct snd_soc_dapm_widget *w, hubs->hp_startup_mode); break; } + break; case SND_SOC_DAPM_PRE_PMD: snd_soc_update_bits(codec, WM8993_CHARGE_PUMP_1, diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 670b96b0ce2..dcfd0fae0b3 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -42,7 +42,8 @@ struct imx_pcm_runtime_data { struct hrtimer hrt; int poll_time_ns; struct snd_pcm_substream *substream; - atomic_t running; + atomic_t playing; + atomic_t capturing; }; static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) @@ -54,7 +55,7 @@ static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) struct pt_regs regs; unsigned long delta; - if (!atomic_read(&iprtd->running)) + if (!atomic_read(&iprtd->playing) && !atomic_read(&iprtd->capturing)) return HRTIMER_NORESTART; get_fiq_regs(®s); @@ -122,7 +123,6 @@ static int snd_imx_pcm_prepare(struct snd_pcm_substream *substream) return 0; } -static int fiq_enable; static int imx_pcm_fiq; static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) @@ -134,23 +134,27 @@ static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - atomic_set(&iprtd->running, 1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + atomic_set(&iprtd->playing, 1); + else + atomic_set(&iprtd->capturing, 1); hrtimer_start(&iprtd->hrt, ns_to_ktime(iprtd->poll_time_ns), HRTIMER_MODE_REL); - if (++fiq_enable == 1) - enable_fiq(imx_pcm_fiq); - + enable_fiq(imx_pcm_fiq); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - atomic_set(&iprtd->running, 0); - - if (--fiq_enable == 0) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + atomic_set(&iprtd->playing, 0); + else + atomic_set(&iprtd->capturing, 0); + if (!atomic_read(&iprtd->playing) && + !atomic_read(&iprtd->capturing)) disable_fiq(imx_pcm_fiq); - break; + default: return -EINVAL; } @@ -198,7 +202,8 @@ static int snd_imx_open(struct snd_pcm_substream *substream) iprtd->substream = substream; - atomic_set(&iprtd->running, 0); + atomic_set(&iprtd->playing, 0); + atomic_set(&iprtd->capturing, 0); hrtimer_init(&iprtd->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL); iprtd->hrt.function = snd_hrtimer_callback; diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c index 1358c7de252..d0740a76296 100644 --- a/sound/soc/s6000/s6000-pcm.c +++ b/sound/soc/s6000/s6000-pcm.c @@ -128,7 +128,9 @@ static irqreturn_t s6000_pcm_irq(int irq, void *data) substream->runtime && snd_pcm_running(substream)) { dev_dbg(pcm->dev, "xrun\n"); + snd_pcm_stream_lock(substream); snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock(substream); ret = IRQ_HANDLED; } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c7051c457b7..c2ecb4e0159 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -682,13 +682,14 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w) return -EINVAL; } - path = list_first_entry(&w->sources, struct snd_soc_dapm_path, - list_sink); - if (!path) { + if (list_empty(&w->sources)) { dev_err(dapm->dev, "ASoC: mux %s has no paths\n", w->name); return -EINVAL; } + path = list_first_entry(&w->sources, struct snd_soc_dapm_path, + list_sink); + ret = dapm_create_or_share_mixmux_kcontrol(w, 0, path); if (ret < 0) return ret; @@ -1796,7 +1797,7 @@ static ssize_t dapm_widget_power_read_file(struct file *file, w->active ? "active" : "inactive"); list_for_each_entry(p, &w->sources, list_sink) { - if (p->connected && !p->connected(w, p->sink)) + if (p->connected && !p->connected(w, p->source)) continue; if (p->connect) diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index 2f70ea7f661..05676c022a1 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -399,9 +399,9 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) ac97->capture_dma_data.slave_id = of_dma[1]; ac97->playback_dma_data.addr = mem->start + TEGRA20_AC97_FIFO_TX1; - ac97->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; - ac97->capture_dma_data.maxburst = 4; - ac97->capture_dma_data.slave_id = of_dma[0]; + ac97->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + ac97->playback_dma_data.maxburst = 4; + ac97->playback_dma_data.slave_id = of_dma[1]; ret = snd_soc_register_component(&pdev->dev, &tegra20_ac97_component, &tegra20_ac97_dai, 1); diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index 52af7f6fb37..540832e9e68 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -74,7 +74,7 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai); - unsigned int mask, val; + unsigned int mask = 0, val = 0; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: @@ -83,10 +83,10 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } - mask = TEGRA20_I2S_CTRL_MASTER_ENABLE; + mask |= TEGRA20_I2S_CTRL_MASTER_ENABLE; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - val = TEGRA20_I2S_CTRL_MASTER_ENABLE; + val |= TEGRA20_I2S_CTRL_MASTER_ENABLE; break; case SND_SOC_DAIFMT_CBM_CFM: break; diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 5eaa12cdc6e..2e7d4aca3d7 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -67,15 +67,15 @@ static int tegra20_spdif_hw_params(struct snd_pcm_substream *substream, { struct device *dev = dai->dev; struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai); - unsigned int mask, val; + unsigned int mask = 0, val = 0; int ret, spdifclock; - mask = TEGRA20_SPDIF_CTRL_PACK | - TEGRA20_SPDIF_CTRL_BIT_MODE_MASK; + mask |= TEGRA20_SPDIF_CTRL_PACK | + TEGRA20_SPDIF_CTRL_BIT_MODE_MASK; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: - val = TEGRA20_SPDIF_CTRL_PACK | - TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT; + val |= TEGRA20_SPDIF_CTRL_PACK | + TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT; break; default: return -EINVAL; @@ -323,8 +323,8 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) } spdif->playback_dma_data.addr = mem->start + TEGRA20_SPDIF_DATA_OUT; - spdif->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; - spdif->capture_dma_data.maxburst = 4; + spdif->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + spdif->playback_dma_data.maxburst = 4; spdif->playback_dma_data.slave_id = dmareq->start; pm_runtime_enable(&pdev->dev); diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 31d092d83c7..5c6520b8ec0 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -117,7 +117,7 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); - unsigned int mask, val; + unsigned int mask = 0, val = 0; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: @@ -126,10 +126,10 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } - mask = TEGRA30_I2S_CTRL_MASTER_ENABLE; + mask |= TEGRA30_I2S_CTRL_MASTER_ENABLE; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - val = TEGRA30_I2S_CTRL_MASTER_ENABLE; + val |= TEGRA30_I2S_CTRL_MASTER_ENABLE; break; case SND_SOC_DAIFMT_CBM_CFM: break; @@ -228,7 +228,7 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream, reg = TEGRA30_I2S_CIF_RX_CTRL; } else { val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX; - reg = TEGRA30_I2S_CIF_RX_CTRL; + reg = TEGRA30_I2S_CIF_TX_CTRL; } regmap_write(i2s->regmap, reg, val); diff --git a/sound/usb/6fire/chip.c b/sound/usb/6fire/chip.c index 4394ae79635..0716ba69139 100644 --- a/sound/usb/6fire/chip.c +++ b/sound/usb/6fire/chip.c @@ -101,7 +101,7 @@ static int usb6fire_chip_probe(struct usb_interface *intf, usb_set_intfdata(intf, chips[i]); mutex_unlock(®ister_mutex); return 0; - } else if (regidx < 0) + } else if (!devices[i] && regidx < 0) regidx = i; } if (regidx < 0) { diff --git a/sound/usb/6fire/comm.c b/sound/usb/6fire/comm.c index 9e6e3ffd86b..23452ee617e 100644 --- a/sound/usb/6fire/comm.c +++ b/sound/usb/6fire/comm.c @@ -110,19 +110,37 @@ static int usb6fire_comm_send_buffer(u8 *buffer, struct usb_device *dev) static int usb6fire_comm_write8(struct comm_runtime *rt, u8 request, u8 reg, u8 value) { - u8 buffer[13]; /* 13: maximum length of message */ + u8 *buffer; + int ret; + + /* 13: maximum length of message */ + buffer = kmalloc(13, GFP_KERNEL); + if (!buffer) + return -ENOMEM; usb6fire_comm_init_buffer(buffer, 0x00, request, reg, value, 0x00); - return usb6fire_comm_send_buffer(buffer, rt->chip->dev); + ret = usb6fire_comm_send_buffer(buffer, rt->chip->dev); + + kfree(buffer); + return ret; } static int usb6fire_comm_write16(struct comm_runtime *rt, u8 request, u8 reg, u8 vl, u8 vh) { - u8 buffer[13]; /* 13: maximum length of message */ + u8 *buffer; + int ret; + + /* 13: maximum length of message */ + buffer = kmalloc(13, GFP_KERNEL); + if (!buffer) + return -ENOMEM; usb6fire_comm_init_buffer(buffer, 0x00, request, reg, vl, vh); - return usb6fire_comm_send_buffer(buffer, rt->chip->dev); + ret = usb6fire_comm_send_buffer(buffer, rt->chip->dev); + + kfree(buffer); + return ret; } int usb6fire_comm_init(struct sfire_chip *chip) @@ -135,6 +153,12 @@ int usb6fire_comm_init(struct sfire_chip *chip) if (!rt) return -ENOMEM; + rt->receiver_buffer = kzalloc(COMM_RECEIVER_BUFSIZE, GFP_KERNEL); + if (!rt->receiver_buffer) { + kfree(rt); + return -ENOMEM; + } + urb = &rt->receiver; rt->serial = 1; rt->chip = chip; @@ -153,6 +177,7 @@ int usb6fire_comm_init(struct sfire_chip *chip) urb->interval = 1; ret = usb_submit_urb(urb, GFP_KERNEL); if (ret < 0) { + kfree(rt->receiver_buffer); kfree(rt); snd_printk(KERN_ERR PREFIX "cannot create comm data receiver."); return ret; @@ -171,6 +196,9 @@ void usb6fire_comm_abort(struct sfire_chip *chip) void usb6fire_comm_destroy(struct sfire_chip *chip) { - kfree(chip->comm); + struct comm_runtime *rt = chip->comm; + + kfree(rt->receiver_buffer); + kfree(rt); chip->comm = NULL; } diff --git a/sound/usb/6fire/comm.h b/sound/usb/6fire/comm.h index 6a0840b0dcf..780d5ed8e5d 100644 --- a/sound/usb/6fire/comm.h +++ b/sound/usb/6fire/comm.h @@ -24,7 +24,7 @@ struct comm_runtime { struct sfire_chip *chip; struct urb receiver; - u8 receiver_buffer[COMM_RECEIVER_BUFSIZE]; + u8 *receiver_buffer; u8 serial; /* urb serial */ diff --git a/sound/usb/6fire/midi.c b/sound/usb/6fire/midi.c index 26722423330..f3dd7266c39 100644 --- a/sound/usb/6fire/midi.c +++ b/sound/usb/6fire/midi.c @@ -19,6 +19,10 @@ #include "chip.h" #include "comm.h" +enum { + MIDI_BUFSIZE = 64 +}; + static void usb6fire_midi_out_handler(struct urb *urb) { struct midi_runtime *rt = urb->context; @@ -156,6 +160,12 @@ int usb6fire_midi_init(struct sfire_chip *chip) if (!rt) return -ENOMEM; + rt->out_buffer = kzalloc(MIDI_BUFSIZE, GFP_KERNEL); + if (!rt->out_buffer) { + kfree(rt); + return -ENOMEM; + } + rt->chip = chip; rt->in_received = usb6fire_midi_in_received; rt->out_buffer[0] = 0x80; /* 'send midi' command */ @@ -169,6 +179,7 @@ int usb6fire_midi_init(struct sfire_chip *chip) ret = snd_rawmidi_new(chip->card, "6FireUSB", 0, 1, 1, &rt->instance); if (ret < 0) { + kfree(rt->out_buffer); kfree(rt); snd_printk(KERN_ERR PREFIX "unable to create midi.\n"); return ret; @@ -197,6 +208,9 @@ void usb6fire_midi_abort(struct sfire_chip *chip) void usb6fire_midi_destroy(struct sfire_chip *chip) { - kfree(chip->midi); + struct midi_runtime *rt = chip->midi; + + kfree(rt->out_buffer); + kfree(rt); chip->midi = NULL; } diff --git a/sound/usb/6fire/midi.h b/sound/usb/6fire/midi.h index c321006e543..84851b9f555 100644 --- a/sound/usb/6fire/midi.h +++ b/sound/usb/6fire/midi.h @@ -16,10 +16,6 @@ #include "common.h" -enum { - MIDI_BUFSIZE = 64 -}; - struct midi_runtime { struct sfire_chip *chip; struct snd_rawmidi *instance; @@ -32,7 +28,7 @@ struct midi_runtime { struct snd_rawmidi_substream *out; struct urb out_urb; u8 out_serial; /* serial number of out packet */ - u8 out_buffer[MIDI_BUFSIZE]; + u8 *out_buffer; int buffer_offset; void (*in_received)(struct midi_runtime *rt, u8 *data, int length); diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index 40dd50a80f5..25f9e61ad88 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -543,7 +543,7 @@ static snd_pcm_uframes_t usb6fire_pcm_pointer( snd_pcm_uframes_t ret; if (rt->panic || !sub) - return SNDRV_PCM_STATE_XRUN; + return SNDRV_PCM_POS_XRUN; spin_lock_irqsave(&sub->lock, flags); ret = sub->dma_off; @@ -580,6 +580,33 @@ static void usb6fire_pcm_init_urb(struct pcm_urb *urb, urb->instance.number_of_packets = PCM_N_PACKETS_PER_URB; } +static int usb6fire_pcm_buffers_init(struct pcm_runtime *rt) +{ + int i; + + for (i = 0; i < PCM_N_URBS; i++) { + rt->out_urbs[i].buffer = kzalloc(PCM_N_PACKETS_PER_URB + * PCM_MAX_PACKET_SIZE, GFP_KERNEL); + if (!rt->out_urbs[i].buffer) + return -ENOMEM; + rt->in_urbs[i].buffer = kzalloc(PCM_N_PACKETS_PER_URB + * PCM_MAX_PACKET_SIZE, GFP_KERNEL); + if (!rt->in_urbs[i].buffer) + return -ENOMEM; + } + return 0; +} + +static void usb6fire_pcm_buffers_destroy(struct pcm_runtime *rt) +{ + int i; + + for (i = 0; i < PCM_N_URBS; i++) { + kfree(rt->out_urbs[i].buffer); + kfree(rt->in_urbs[i].buffer); + } +} + int usb6fire_pcm_init(struct sfire_chip *chip) { int i; @@ -591,6 +618,13 @@ int usb6fire_pcm_init(struct sfire_chip *chip) if (!rt) return -ENOMEM; + ret = usb6fire_pcm_buffers_init(rt); + if (ret) { + usb6fire_pcm_buffers_destroy(rt); + kfree(rt); + return ret; + } + rt->chip = chip; rt->stream_state = STREAM_DISABLED; rt->rate = ARRAY_SIZE(rates); @@ -612,6 +646,7 @@ int usb6fire_pcm_init(struct sfire_chip *chip) ret = snd_pcm_new(chip->card, "DMX6FireUSB", 0, 1, 1, &pcm); if (ret < 0) { + usb6fire_pcm_buffers_destroy(rt); kfree(rt); snd_printk(KERN_ERR PREFIX "cannot create pcm instance.\n"); return ret; @@ -627,6 +662,7 @@ int usb6fire_pcm_init(struct sfire_chip *chip) snd_dma_continuous_data(GFP_KERNEL), MAX_BUFSIZE, MAX_BUFSIZE); if (ret) { + usb6fire_pcm_buffers_destroy(rt); kfree(rt); snd_printk(KERN_ERR PREFIX "error preallocating pcm buffers.\n"); @@ -641,17 +677,25 @@ int usb6fire_pcm_init(struct sfire_chip *chip) void usb6fire_pcm_abort(struct sfire_chip *chip) { struct pcm_runtime *rt = chip->pcm; + unsigned long flags; int i; if (rt) { rt->panic = true; - if (rt->playback.instance) + if (rt->playback.instance) { + snd_pcm_stream_lock_irqsave(rt->playback.instance, flags); snd_pcm_stop(rt->playback.instance, SNDRV_PCM_STATE_XRUN); - if (rt->capture.instance) + snd_pcm_stream_unlock_irqrestore(rt->playback.instance, flags); + } + + if (rt->capture.instance) { + snd_pcm_stream_lock_irqsave(rt->capture.instance, flags); snd_pcm_stop(rt->capture.instance, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irqrestore(rt->capture.instance, flags); + } for (i = 0; i < PCM_N_URBS; i++) { usb_poison_urb(&rt->in_urbs[i].instance); @@ -663,6 +707,9 @@ void usb6fire_pcm_abort(struct sfire_chip *chip) void usb6fire_pcm_destroy(struct sfire_chip *chip) { - kfree(chip->pcm); + struct pcm_runtime *rt = chip->pcm; + + usb6fire_pcm_buffers_destroy(rt); + kfree(rt); chip->pcm = NULL; } diff --git a/sound/usb/6fire/pcm.h b/sound/usb/6fire/pcm.h index 9b01133ee3f..f5779d6182c 100644 --- a/sound/usb/6fire/pcm.h +++ b/sound/usb/6fire/pcm.h @@ -32,7 +32,7 @@ struct pcm_urb { struct urb instance; struct usb_iso_packet_descriptor packets[PCM_N_PACKETS_PER_URB]; /* END DO NOT SEPARATE */ - u8 buffer[PCM_N_PACKETS_PER_URB * PCM_MAX_PACKET_SIZE]; + u8 *buffer; struct pcm_urb *peer; }; diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 225dfd73726..ba2664200d1 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -14,6 +14,7 @@ config SND_USB_AUDIO select SND_HWDEP select SND_RAWMIDI select SND_PCM + select BITREVERSE help Say Y here to include support for USB audio and USB MIDI devices. diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 7a444b5501d..659950e5b94 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -591,17 +591,16 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, ep->stride = frame_bits >> 3; ep->silence_value = pcm_format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0; - /* calculate max. frequency */ - if (ep->maxpacksize) { + /* assume max. frequency is 25% higher than nominal */ + ep->freqmax = ep->freqn + (ep->freqn >> 2); + maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) + >> (16 - ep->datainterval); + /* but wMaxPacketSize might reduce this */ + if (ep->maxpacksize && ep->maxpacksize < maxsize) { /* whatever fits into a max. size packet */ maxsize = ep->maxpacksize; ep->freqmax = (maxsize / (frame_bits >> 3)) << (16 - ep->datainterval); - } else { - /* no max. packet size: just take 25% higher than nominal */ - ep->freqmax = ep->freqn + (ep->freqn >> 2); - maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) - >> (16 - ep->datainterval); } if (ep->fill_max) diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c index 6ad617b9473..76d832908fe 100644 --- a/sound/usb/misc/ua101.c +++ b/sound/usb/misc/ua101.c @@ -613,14 +613,24 @@ static int start_usb_playback(struct ua101 *ua) static void abort_alsa_capture(struct ua101 *ua) { - if (test_bit(ALSA_CAPTURE_RUNNING, &ua->states)) + unsigned long flags; + + if (test_bit(ALSA_CAPTURE_RUNNING, &ua->states)) { + snd_pcm_stream_lock_irqsave(ua->capture.substream, flags); snd_pcm_stop(ua->capture.substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irqrestore(ua->capture.substream, flags); + } } static void abort_alsa_playback(struct ua101 *ua) { - if (test_bit(ALSA_PLAYBACK_RUNNING, &ua->states)) + unsigned long flags; + + if (test_bit(ALSA_PLAYBACK_RUNNING, &ua->states)) { + snd_pcm_stream_lock_irqsave(ua->playback.substream, flags); snd_pcm_stop(ua->playback.substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irqrestore(ua->playback.substream, flags); + } } static int set_stream_hw(struct ua101 *ua, struct snd_pcm_substream *substream, diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index d5438083fd6..95558ef4a7a 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -888,6 +888,7 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, case USB_ID(0x046d, 0x081b): /* HD Webcam c310 */ case USB_ID(0x046d, 0x081d): /* HD Webcam c510 */ case USB_ID(0x046d, 0x0825): /* HD Webcam c270 */ + case USB_ID(0x046d, 0x0826): /* HD Webcam c525 */ case USB_ID(0x046d, 0x0991): /* Most audio usb devices lie about volume resolution. * Most Logitech webcams have res = 384. diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index d0323a693ba..999550bbad4 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -262,7 +262,9 @@ static int usb_stream_hwdep_mmap(struct snd_hwdep *hw, } area->vm_ops = &usb_stream_hwdep_vm_ops; - area->vm_flags |= VM_DONTEXPAND | VM_DONTDUMP; + area->vm_flags |= VM_DONTDUMP; + if (!read) + area->vm_flags |= VM_DONTEXPAND; area->vm_private_data = us122l; atomic_inc(&us122l->mmap_count); out: diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index b37653247ef..cd69a80b5ca 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -273,7 +273,11 @@ static void usX2Y_clients_stop(struct usX2Ydev *usX2Y) struct snd_usX2Y_substream *subs = usX2Y->subs[s]; if (subs) { if (atomic_read(&subs->state) >= state_PRERUNNING) { + unsigned long flags; + + snd_pcm_stream_lock_irqsave(subs->pcm_substream, flags); snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irqrestore(subs->pcm_substream, flags); } for (u = 0; u < NRURBS; u++) { struct urb *urb = subs->urb[u]; @@ -295,19 +299,6 @@ static void usX2Y_error_urb_status(struct usX2Ydev *usX2Y, usX2Y_clients_stop(usX2Y); } -static void usX2Y_error_sequence(struct usX2Ydev *usX2Y, - struct snd_usX2Y_substream *subs, struct urb *urb) -{ - snd_printk(KERN_ERR -"Sequence Error!(hcd_frame=%i ep=%i%s;wait=%i,frame=%i).\n" -"Most probably some urb of usb-frame %i is still missing.\n" -"Cause could be too long delays in usb-hcd interrupt handling.\n", - usb_get_current_frame_number(usX2Y->dev), - subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out", - usX2Y->wait_iso_frame, urb->start_frame, usX2Y->wait_iso_frame); - usX2Y_clients_stop(usX2Y); -} - static void i_usX2Y_urb_complete(struct urb *urb) { struct snd_usX2Y_substream *subs = urb->context; @@ -324,12 +315,9 @@ static void i_usX2Y_urb_complete(struct urb *urb) usX2Y_error_urb_status(usX2Y, subs, urb); return; } - if (likely((urb->start_frame & 0xFFFF) == (usX2Y->wait_iso_frame & 0xFFFF))) - subs->completed_urb = urb; - else { - usX2Y_error_sequence(usX2Y, subs, urb); - return; - } + + subs->completed_urb = urb; + { struct snd_usX2Y_substream *capsubs = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE], *playbacksubs = usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK]; diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index f2a1acdc4d8..814d0e887c6 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -244,13 +244,8 @@ static void i_usX2Y_usbpcm_urb_complete(struct urb *urb) usX2Y_error_urb_status(usX2Y, subs, urb); return; } - if (likely((urb->start_frame & 0xFFFF) == (usX2Y->wait_iso_frame & 0xFFFF))) - subs->completed_urb = urb; - else { - usX2Y_error_sequence(usX2Y, subs, urb); - return; - } + subs->completed_urb = urb; capsubs = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE]; capsubs2 = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE + 2]; playbacksubs = usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK]; |