diff options
Diffstat (limited to 'sound')
39 files changed, 306 insertions, 153 deletions
diff --git a/sound/core/control.c b/sound/core/control.c index 35324a8e83c8..eeb691d1911f 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1170,6 +1170,10 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, if (info->count < 1) return -EINVAL; + if (!*info->id.name) + return -EINVAL; + if (strnlen(info->id.name, sizeof(info->id.name)) >= sizeof(info->id.name)) + return -EINVAL; access = info->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE : (info->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE| SNDRV_CTL_ELEM_ACCESS_INACTIVE| diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index b03a638b420c..279e24f61305 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1552,6 +1552,8 @@ static int snd_pcm_do_drain_init(struct snd_pcm_substream *substream, int state) if (! snd_pcm_playback_empty(substream)) { snd_pcm_do_start(substream, SNDRV_PCM_STATE_DRAINING); snd_pcm_post_start(substream, SNDRV_PCM_STATE_DRAINING); + } else { + runtime->status->state = SNDRV_PCM_STATE_SETUP; } break; case SNDRV_PCM_STATE_RUNNING: diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index f62780ed64ad..7821b07415a7 100644 --- a/sound/drivers/opl3/opl3_midi.c +++ b/sound/drivers/opl3/opl3_midi.c @@ -105,6 +105,8 @@ static void snd_opl3_calc_pitch(unsigned char *fnum, unsigned char *blocknum, int pitchbend = chan->midi_pitchbend; int segment; + if (pitchbend < -0x2000) + pitchbend = -0x2000; if (pitchbend > 0x1FFF) pitchbend = 0x1FFF; diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index 0d580186ef1a..5cc356db5351 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -33,7 +33,7 @@ */ #define MAX_MIDI_RX_BLOCKS 8 -#define TRANSFER_DELAY_TICKS 0x2e00 /* 479.17 µs */ +#define TRANSFER_DELAY_TICKS 0x2e00 /* 479.17 microseconds */ /* isochronous header parameters */ #define ISO_DATA_LENGTH_SHIFT 16 @@ -78,7 +78,7 @@ static void pcm_period_tasklet(unsigned long data); int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit, enum amdtp_stream_direction dir, enum cip_flags flags) { - s->unit = fw_unit_get(unit); + s->unit = unit; s->direction = dir; s->flags = flags; s->context = ERR_PTR(-1); @@ -102,7 +102,6 @@ void amdtp_stream_destroy(struct amdtp_stream *s) { WARN_ON(amdtp_stream_running(s)); mutex_destroy(&s->mutex); - fw_unit_put(s->unit); } EXPORT_SYMBOL(amdtp_stream_destroy); diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index fc19c99654aa..611b7dae7ee5 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -116,11 +116,22 @@ end: return err; } +/* + * This module releases the FireWire unit data after all ALSA character devices + * are released by applications. This is for releasing stream data or finishing + * transactions safely. Thus at returning from .remove(), this module still keep + * references for the unit. + */ static void bebob_card_free(struct snd_card *card) { struct snd_bebob *bebob = card->private_data; + snd_bebob_stream_destroy_duplex(bebob); + fw_unit_put(bebob->unit); + + kfree(bebob->maudio_special_quirk); + if (bebob->card_index >= 0) { mutex_lock(&devices_mutex); clear_bit(bebob->card_index, devices_used); @@ -205,7 +216,7 @@ bebob_probe(struct fw_unit *unit, card->private_free = bebob_card_free; bebob->card = card; - bebob->unit = unit; + bebob->unit = fw_unit_get(unit); bebob->spec = spec; mutex_init(&bebob->mutex); spin_lock_init(&bebob->lock); @@ -306,10 +317,11 @@ static void bebob_remove(struct fw_unit *unit) if (bebob == NULL) return; - kfree(bebob->maudio_special_quirk); + /* Awake bus-reset waiters. */ + if (!completion_done(&bebob->bus_reset)) + complete_all(&bebob->bus_reset); - snd_bebob_stream_destroy_duplex(bebob); - snd_card_disconnect(bebob->card); + /* No need to wait for releasing card object in this context. */ snd_card_free_when_closed(bebob->card); } diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 0ebcabfdc7ce..98e4fc8121a1 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -410,8 +410,6 @@ break_both_connections(struct snd_bebob *bebob) static void destroy_both_connections(struct snd_bebob *bebob) { - break_both_connections(bebob); - cmp_connection_destroy(&bebob->in_conn); cmp_connection_destroy(&bebob->out_conn); } @@ -712,22 +710,16 @@ void snd_bebob_stream_update_duplex(struct snd_bebob *bebob) mutex_unlock(&bebob->mutex); } +/* + * This function should be called before starting streams or after stopping + * streams. + */ void snd_bebob_stream_destroy_duplex(struct snd_bebob *bebob) { - mutex_lock(&bebob->mutex); - - amdtp_stream_pcm_abort(&bebob->rx_stream); - amdtp_stream_pcm_abort(&bebob->tx_stream); - - amdtp_stream_stop(&bebob->rx_stream); - amdtp_stream_stop(&bebob->tx_stream); - amdtp_stream_destroy(&bebob->rx_stream); amdtp_stream_destroy(&bebob->tx_stream); destroy_both_connections(bebob); - - mutex_unlock(&bebob->mutex); } /* diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index fa9cf761b610..07dbd01d7a6b 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -311,14 +311,21 @@ end: return err; } +/* + * This function should be called before starting streams or after stopping + * streams. + */ static void destroy_stream(struct snd_dice *dice, struct amdtp_stream *stream) { - amdtp_stream_destroy(stream); + struct fw_iso_resources *resources; if (stream == &dice->tx_stream) - fw_iso_resources_destroy(&dice->tx_resources); + resources = &dice->tx_resources; else - fw_iso_resources_destroy(&dice->rx_resources); + resources = &dice->rx_resources; + + amdtp_stream_destroy(stream); + fw_iso_resources_destroy(resources); } int snd_dice_stream_init_duplex(struct snd_dice *dice) @@ -332,6 +339,8 @@ int snd_dice_stream_init_duplex(struct snd_dice *dice) goto end; err = init_stream(dice, &dice->rx_stream); + if (err < 0) + destroy_stream(dice, &dice->tx_stream); end: return err; } @@ -340,10 +349,7 @@ void snd_dice_stream_destroy_duplex(struct snd_dice *dice) { snd_dice_transaction_clear_enable(dice); - stop_stream(dice, &dice->tx_stream); destroy_stream(dice, &dice->tx_stream); - - stop_stream(dice, &dice->rx_stream); destroy_stream(dice, &dice->rx_stream); dice->substreams_counter = 0; diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 90d8f40ff727..70a111d7f428 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -226,11 +226,20 @@ static void dice_card_strings(struct snd_dice *dice) strcpy(card->mixername, "DICE"); } +/* + * This module releases the FireWire unit data after all ALSA character devices + * are released by applications. This is for releasing stream data or finishing + * transactions safely. Thus at returning from .remove(), this module still keep + * references for the unit. + */ static void dice_card_free(struct snd_card *card) { struct snd_dice *dice = card->private_data; + snd_dice_stream_destroy_duplex(dice); snd_dice_transaction_destroy(dice); + fw_unit_put(dice->unit); + mutex_destroy(&dice->mutex); } @@ -251,7 +260,7 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) dice = card->private_data; dice->card = card; - dice->unit = unit; + dice->unit = fw_unit_get(unit); card->private_free = dice_card_free; spin_lock_init(&dice->lock); @@ -305,10 +314,7 @@ static void dice_remove(struct fw_unit *unit) { struct snd_dice *dice = dev_get_drvdata(&unit->device); - snd_card_disconnect(dice->card); - - snd_dice_stream_destroy_duplex(dice); - + /* No need to wait for releasing card object in this context. */ snd_card_free_when_closed(dice->card); } diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index 3e2ed8e82cbc..2682e7e3e5c9 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -173,11 +173,23 @@ end: return err; } +/* + * This module releases the FireWire unit data after all ALSA character devices + * are released by applications. This is for releasing stream data or finishing + * transactions safely. Thus at returning from .remove(), this module still keep + * references for the unit. + */ static void efw_card_free(struct snd_card *card) { struct snd_efw *efw = card->private_data; + snd_efw_stream_destroy_duplex(efw); + snd_efw_transaction_remove_instance(efw); + fw_unit_put(efw->unit); + + kfree(efw->resp_buf); + if (efw->card_index >= 0) { mutex_lock(&devices_mutex); clear_bit(efw->card_index, devices_used); @@ -185,7 +197,6 @@ efw_card_free(struct snd_card *card) } mutex_destroy(&efw->mutex); - kfree(efw->resp_buf); } static int @@ -218,7 +229,7 @@ efw_probe(struct fw_unit *unit, card->private_free = efw_card_free; efw->card = card; - efw->unit = unit; + efw->unit = fw_unit_get(unit); mutex_init(&efw->mutex); spin_lock_init(&efw->lock); init_waitqueue_head(&efw->hwdep_wait); @@ -289,10 +300,7 @@ static void efw_remove(struct fw_unit *unit) { struct snd_efw *efw = dev_get_drvdata(&unit->device); - snd_efw_stream_destroy_duplex(efw); - snd_efw_transaction_remove_instance(efw); - - snd_card_disconnect(efw->card); + /* No need to wait for releasing card object in this context. */ snd_card_free_when_closed(efw->card); } diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c index 4f440e163667..c55db1bddc80 100644 --- a/sound/firewire/fireworks/fireworks_stream.c +++ b/sound/firewire/fireworks/fireworks_stream.c @@ -100,17 +100,22 @@ end: return err; } +/* + * This function should be called before starting the stream or after stopping + * the streams. + */ static void destroy_stream(struct snd_efw *efw, struct amdtp_stream *stream) { - stop_stream(efw, stream); - - amdtp_stream_destroy(stream); + struct cmp_connection *conn; if (stream == &efw->tx_stream) - cmp_connection_destroy(&efw->out_conn); + conn = &efw->out_conn; else - cmp_connection_destroy(&efw->in_conn); + conn = &efw->in_conn; + + amdtp_stream_destroy(stream); + cmp_connection_destroy(&efw->out_conn); } static int @@ -319,12 +324,8 @@ void snd_efw_stream_update_duplex(struct snd_efw *efw) void snd_efw_stream_destroy_duplex(struct snd_efw *efw) { - mutex_lock(&efw->mutex); - destroy_stream(efw, &efw->rx_stream); destroy_stream(efw, &efw->tx_stream); - - mutex_unlock(&efw->mutex); } void snd_efw_stream_lock_changed(struct snd_efw *efw) diff --git a/sound/firewire/iso-resources.c b/sound/firewire/iso-resources.c index 5f17b77ee152..f0e4d502d604 100644 --- a/sound/firewire/iso-resources.c +++ b/sound/firewire/iso-resources.c @@ -26,7 +26,7 @@ int fw_iso_resources_init(struct fw_iso_resources *r, struct fw_unit *unit) { r->channels_mask = ~0uLL; - r->unit = fw_unit_get(unit); + r->unit = unit; mutex_init(&r->mutex); r->allocated = false; @@ -42,7 +42,6 @@ void fw_iso_resources_destroy(struct fw_iso_resources *r) { WARN_ON(r->allocated); mutex_destroy(&r->mutex); - fw_unit_put(r->unit); } EXPORT_SYMBOL(fw_iso_resources_destroy); diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index bda845afb470..e6757cd85724 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -171,9 +171,10 @@ static int start_stream(struct snd_oxfw *oxfw, struct amdtp_stream *stream, } /* Wait first packet */ - err = amdtp_stream_wait_callback(stream, CALLBACK_TIMEOUT); - if (err < 0) + if (!amdtp_stream_wait_callback(stream, CALLBACK_TIMEOUT)) { stop_stream(oxfw, stream); + err = -ETIMEDOUT; + } end: return err; } @@ -337,6 +338,10 @@ void snd_oxfw_stream_stop_simplex(struct snd_oxfw *oxfw, stop_stream(oxfw, stream); } +/* + * This function should be called before starting the stream or after stopping + * the streams. + */ void snd_oxfw_stream_destroy_simplex(struct snd_oxfw *oxfw, struct amdtp_stream *stream) { @@ -347,8 +352,6 @@ void snd_oxfw_stream_destroy_simplex(struct snd_oxfw *oxfw, else conn = &oxfw->in_conn; - stop_stream(oxfw, stream); - amdtp_stream_destroy(stream); cmp_connection_destroy(conn); } diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 60e5cad0531a..8c6ce019f437 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -104,11 +104,23 @@ end: return err; } +/* + * This module releases the FireWire unit data after all ALSA character devices + * are released by applications. This is for releasing stream data or finishing + * transactions safely. Thus at returning from .remove(), this module still keep + * references for the unit. + */ static void oxfw_card_free(struct snd_card *card) { struct snd_oxfw *oxfw = card->private_data; unsigned int i; + snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream); + if (oxfw->has_output) + snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->tx_stream); + + fw_unit_put(oxfw->unit); + for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) { kfree(oxfw->tx_stream_formats[i]); kfree(oxfw->rx_stream_formats[i]); @@ -136,7 +148,7 @@ static int oxfw_probe(struct fw_unit *unit, oxfw = card->private_data; oxfw->card = card; mutex_init(&oxfw->mutex); - oxfw->unit = unit; + oxfw->unit = fw_unit_get(unit); oxfw->device_info = (const struct device_info *)id->driver_data; spin_lock_init(&oxfw->lock); init_waitqueue_head(&oxfw->hwdep_wait); @@ -212,12 +224,7 @@ static void oxfw_remove(struct fw_unit *unit) { struct snd_oxfw *oxfw = dev_get_drvdata(&unit->device); - snd_card_disconnect(oxfw->card); - - snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream); - if (oxfw->has_output) - snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->tx_stream); - + /* No need to wait for releasing card object in this context. */ snd_card_free_when_closed(oxfw->card); } diff --git a/sound/isa/msnd/msnd_pinnacle_mixer.c b/sound/isa/msnd/msnd_pinnacle_mixer.c index 17e49a071af4..b408540798c1 100644 --- a/sound/isa/msnd/msnd_pinnacle_mixer.c +++ b/sound/isa/msnd/msnd_pinnacle_mixer.c @@ -306,11 +306,12 @@ int snd_msndmix_new(struct snd_card *card) spin_lock_init(&chip->mixer_lock); strcpy(card->mixername, "MSND Pinnacle Mixer"); - for (idx = 0; idx < ARRAY_SIZE(snd_msnd_controls); idx++) + for (idx = 0; idx < ARRAY_SIZE(snd_msnd_controls); idx++) { err = snd_ctl_add(card, snd_ctl_new1(snd_msnd_controls + idx, chip)); if (err < 0) return err; + } return 0; } diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index dfcb5e929f9f..17c2637d842c 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -961,7 +961,6 @@ static int azx_alloc_cmd_io(struct azx *chip) dev_err(chip->card->dev, "cannot allocate CORB/RIRB\n"); return err; } -EXPORT_SYMBOL_GPL(azx_alloc_cmd_io); static void azx_init_cmd_io(struct azx *chip) { @@ -1026,7 +1025,6 @@ static void azx_init_cmd_io(struct azx *chip) azx_writeb(chip, RIRBCTL, AZX_RBCTL_DMA_EN | AZX_RBCTL_IRQ_EN); spin_unlock_irq(&chip->reg_lock); } -EXPORT_SYMBOL_GPL(azx_init_cmd_io); static void azx_free_cmd_io(struct azx *chip) { @@ -1036,7 +1034,6 @@ static void azx_free_cmd_io(struct azx *chip) azx_writeb(chip, CORBCTL, 0); spin_unlock_irq(&chip->reg_lock); } -EXPORT_SYMBOL_GPL(azx_free_cmd_io); static unsigned int azx_command_addr(u32 cmd) { @@ -1167,7 +1164,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, } } - if (!bus->no_response_fallback) + if (bus->no_response_fallback) return -1; if (!chip->polling_mode && chip->poll_count < 2) { @@ -1316,7 +1313,6 @@ static int azx_send_cmd(struct hda_bus *bus, unsigned int val) else return azx_corb_send_cmd(bus, val); } -EXPORT_SYMBOL_GPL(azx_send_cmd); /* get a response */ static unsigned int azx_get_response(struct hda_bus *bus, @@ -1330,7 +1326,6 @@ static unsigned int azx_get_response(struct hda_bus *bus, else return azx_rirb_get_response(bus, addr); } -EXPORT_SYMBOL_GPL(azx_get_response); #ifdef CONFIG_SND_HDA_DSP_LOADER /* diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index b680b4ec6331..fe18071bf93a 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -692,7 +692,23 @@ static void init_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx) { unsigned int caps = query_amp_caps(codec, nid, dir); int val = get_amp_val_to_activate(codec, nid, dir, caps, false); - snd_hda_codec_amp_init_stereo(codec, nid, dir, idx, 0xff, val); + + if (get_wcaps(codec, nid) & AC_WCAP_STEREO) + snd_hda_codec_amp_init_stereo(codec, nid, dir, idx, 0xff, val); + else + snd_hda_codec_amp_init(codec, nid, 0, dir, idx, 0xff, val); +} + +/* update the amp, doing in stereo or mono depending on NID */ +static int update_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx, + unsigned int mask, unsigned int val) +{ + if (get_wcaps(codec, nid) & AC_WCAP_STEREO) + return snd_hda_codec_amp_stereo(codec, nid, dir, idx, + mask, val); + else + return snd_hda_codec_amp_update(codec, nid, 0, dir, idx, + mask, val); } /* calculate amp value mask we can modify; @@ -732,7 +748,7 @@ static void activate_amp(struct hda_codec *codec, hda_nid_t nid, int dir, return; val &= mask; - snd_hda_codec_amp_stereo(codec, nid, dir, idx, mask, val); + update_amp(codec, nid, dir, idx, mask, val); } static void activate_amp_out(struct hda_codec *codec, struct nid_path *path, @@ -4424,13 +4440,11 @@ static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix) 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); + update_amp(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); + update_amp(codec, conn[i], HDA_OUTPUT, 0, + 0xff, HDA_AMP_MUTE); } } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 36d2f20db7a4..4ca3d5d02436 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1966,7 +1966,7 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* Panther Point */ { PCI_DEVICE(0x8086, 0x1e20), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* Lynx Point */ { PCI_DEVICE(0x8086, 0x8c20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 1589c9bcce3e..dd2b3d92071f 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -393,6 +393,7 @@ static const struct snd_pci_quirk cs420x_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x1c00, "MacBookPro 8,1", CS420X_MBP81), SND_PCI_QUIRK(0x106b, 0x2000, "iMac 12,2", CS420X_IMAC27_122), SND_PCI_QUIRK(0x106b, 0x2800, "MacBookPro 10,1", CS420X_MBP101), + SND_PCI_QUIRK(0x106b, 0x5600, "MacBookAir 5,2", CS420X_MBP81), SND_PCI_QUIRK(0x106b, 0x5b00, "MacBookAir 4,2", CS420X_MBA42), SND_PCI_QUIRK_VENDOR(0x106b, "Apple", CS420X_APPLE), {} /* terminator */ @@ -584,6 +585,7 @@ static int patch_cs420x(struct hda_codec *codec) return -ENOMEM; spec->gen.automute_hook = cs_automute; + codec->single_adc_amp = 1; snd_hda_pick_fixup(codec, cs420x_models, cs420x_fixup_tbl, cs420x_fixups); diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index fd3ed18670e9..da67ea8645a6 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -223,6 +223,7 @@ enum { CXT_PINCFG_LENOVO_TP410, CXT_PINCFG_LEMOTE_A1004, CXT_PINCFG_LEMOTE_A1205, + CXT_PINCFG_COMPAQ_CQ60, CXT_FIXUP_STEREO_DMIC, CXT_FIXUP_INC_MIC_BOOST, CXT_FIXUP_HEADPHONE_MIC_PIN, @@ -660,6 +661,15 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_PINS, .v.pins = cxt_pincfg_lemote, }, + [CXT_PINCFG_COMPAQ_CQ60] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + /* 0x17 was falsely set up as a mic, it should 0x1d */ + { 0x17, 0x400001f0 }, + { 0x1d, 0x97a70120 }, + { } + } + }, [CXT_FIXUP_STEREO_DMIC] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_stereo_dmic, @@ -769,6 +779,7 @@ static const struct hda_model_fixup cxt5047_fixup_models[] = { }; static const struct snd_pci_quirk cxt5051_fixups[] = { + SND_PCI_QUIRK(0x103c, 0x360b, "Compaq CQ60", CXT_PINCFG_COMPAQ_CQ60), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT_PINCFG_LENOVO_X200), {} }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b2b24a8b3dac..526398a4a442 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5209,6 +5209,13 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x17, 0x40000000}, {0x1d, 0x40700001}, {0x21, 0x02211040}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC255_STANDARD_PINS, + {0x12, 0x90a60170}, + {0x14, 0x90170140}, + {0x17, 0x40000000}, + {0x1d, 0x40700001}, + {0x21, 0x02211050}), SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4, {0x12, 0x90a60130}, {0x13, 0x40000000}, diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 6d36c5b78805..87eff3173ce9 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -79,6 +79,7 @@ enum { STAC_ALIENWARE_M17X, STAC_92HD89XX_HP_FRONT_JACK, STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK, + STAC_92HD73XX_ASUS_MOBO, STAC_92HD73XX_MODELS }; @@ -1911,7 +1912,18 @@ static const struct hda_fixup stac92hd73xx_fixups[] = { [STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK] = { .type = HDA_FIXUP_PINS, .v.pins = stac92hd89xx_hp_z1_g2_right_mic_jack_pin_configs, - } + }, + [STAC_92HD73XX_ASUS_MOBO] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + /* enable 5.1 and SPDIF out */ + { 0x0c, 0x01014411 }, + { 0x0d, 0x01014410 }, + { 0x0e, 0x01014412 }, + { 0x22, 0x014b1180 }, + { } + } + }, }; static const struct hda_model_fixup stac92hd73xx_models[] = { @@ -1923,6 +1935,7 @@ static const struct hda_model_fixup stac92hd73xx_models[] = { { .id = STAC_DELL_M6_BOTH, .name = "dell-m6" }, { .id = STAC_DELL_EQ, .name = "dell-eq" }, { .id = STAC_ALIENWARE_M17X, .name = "alienware" }, + { .id = STAC_92HD73XX_ASUS_MOBO, .name = "asus-mobo" }, {} }; @@ -1975,6 +1988,8 @@ static const struct snd_pci_quirk stac92hd73xx_fixup_tbl[] = { "HP Z1 G2", STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x2b17, "unknown HP", STAC_92HD89XX_HP_FRONT_JACK), + SND_PCI_QUIRK(PCI_VENDOR_ID_ASUSTEK, 0x83f8, "ASUS AT4NM10", + STAC_92HD73XX_ASUS_MOBO), {} /* terminator */ }; diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index f5ad214663f9..8de836165cf2 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -46,8 +46,6 @@ #include <sound/pcm_params.h> #include <sound/soc.h> -#include <asm/mach-types.h> - #include "../codecs/wm8731.h" #include "atmel-pcm.h" #include "atmel_ssc_dai.h" @@ -171,9 +169,7 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) int ret; if (!np) { - if (!(machine_is_at91sam9g20ek() || - machine_is_at91sam9g20ek_2mmc())) - return -ENODEV; + return -ENODEV; } ret = atmel_ssc_set_audio(0); @@ -210,39 +206,37 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) card->dev = &pdev->dev; /* Parse device node info */ - if (np) { - ret = snd_soc_of_parse_card_name(card, "atmel,model"); - if (ret) - goto err; - - ret = snd_soc_of_parse_audio_routing(card, - "atmel,audio-routing"); - if (ret) - goto err; - - /* Parse codec info */ - at91sam9g20ek_dai.codec_name = NULL; - codec_np = of_parse_phandle(np, "atmel,audio-codec", 0); - if (!codec_np) { - dev_err(&pdev->dev, "codec info missing\n"); - return -EINVAL; - } - at91sam9g20ek_dai.codec_of_node = codec_np; - - /* Parse dai and platform info */ - at91sam9g20ek_dai.cpu_dai_name = NULL; - at91sam9g20ek_dai.platform_name = NULL; - cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0); - if (!cpu_np) { - dev_err(&pdev->dev, "dai and pcm info missing\n"); - return -EINVAL; - } - at91sam9g20ek_dai.cpu_of_node = cpu_np; - at91sam9g20ek_dai.platform_of_node = cpu_np; - - of_node_put(codec_np); - of_node_put(cpu_np); + ret = snd_soc_of_parse_card_name(card, "atmel,model"); + if (ret) + goto err; + + ret = snd_soc_of_parse_audio_routing(card, + "atmel,audio-routing"); + if (ret) + goto err; + + /* Parse codec info */ + at91sam9g20ek_dai.codec_name = NULL; + codec_np = of_parse_phandle(np, "atmel,audio-codec", 0); + if (!codec_np) { + dev_err(&pdev->dev, "codec info missing\n"); + return -EINVAL; + } + at91sam9g20ek_dai.codec_of_node = codec_np; + + /* Parse dai and platform info */ + at91sam9g20ek_dai.cpu_dai_name = NULL; + at91sam9g20ek_dai.platform_name = NULL; + cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0); + if (!cpu_np) { + dev_err(&pdev->dev, "dai and pcm info missing\n"); + return -EINVAL; } + at91sam9g20ek_dai.cpu_of_node = cpu_np; + at91sam9g20ek_dai.platform_of_node = cpu_np; + + of_node_put(codec_np); + of_node_put(cpu_np); ret = snd_soc_register_card(card); if (ret) { diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig index 7b7fbcd49e5e..c7cd60f009e9 100644 --- a/sound/soc/cirrus/Kconfig +++ b/sound/soc/cirrus/Kconfig @@ -16,7 +16,7 @@ config SND_EP93XX_SOC_AC97 config SND_EP93XX_SOC_SNAPPERCL15 tristate "SoC Audio support for Bluewater Systems Snapper CL15 module" - depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 + depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 && I2C select SND_EP93XX_SOC_I2S select SND_SOC_TLV320AIC23_I2C help diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 064e6c18e109..ea9f0e31f9d4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -69,7 +69,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98088 if I2C select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C - select SND_SOC_MAX98357A + select SND_SOC_MAX98357A if GPIOLIB select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C select SND_SOC_MAX9877 if I2C diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 1806333ea29e..e9e6efbc21dd 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -12,9 +12,19 @@ * max98357a.c -- MAX98357A ALSA SoC Codec driver */ -#include <linux/module.h> +#include <linux/device.h> +#include <linux/err.h> #include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <linux/kernel.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <sound/pcm.h> #include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> #define DRV_NAME "max98357a" diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index e1a4a45c57e2..fd102613d20d 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -225,7 +225,6 @@ static bool rt5670_volatile_register(struct device *dev, unsigned int reg) case RT5670_ADC_EQ_CTRL1: case RT5670_EQ_CTRL1: case RT5670_ALC_CTRL_1: - case RT5670_IRQ_CTRL1: case RT5670_IRQ_CTRL2: case RT5670_INT_IRQ_ST: case RT5670_IL_CMD: @@ -2703,6 +2702,12 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, regmap_write(rt5670->regmap, RT5670_RESET, 0); + regmap_read(rt5670->regmap, RT5670_VENDOR_ID, &val); + if (val >= 4) + regmap_write(rt5670->regmap, RT5670_GPIO_CTRL3, 0x0980); + else + regmap_write(rt5670->regmap, RT5670_GPIO_CTRL3, 0x0d00); + ret = regmap_register_patch(rt5670->regmap, init_list, ARRAY_SIZE(init_list)); if (ret != 0) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 5d0bb8748dd1..fb9c20eace3f 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -3284,8 +3284,8 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "IB45 Bypass Mux", "Bypass", "IB45 Mux" }, { "IB45 Bypass Mux", "Pass SRC", "IB45 Mux" }, - { "IB6 Mux", "IF1 DAC 6", "IF1 DAC6" }, - { "IB6 Mux", "IF2 DAC 6", "IF2 DAC6" }, + { "IB6 Mux", "IF1 DAC 6", "IF1 DAC6 Mux" }, + { "IB6 Mux", "IF2 DAC 6", "IF2 DAC6 Mux" }, { "IB6 Mux", "SLB DAC 6", "SLB DAC6" }, { "IB6 Mux", "STO4 ADC MIX L", "Stereo4 ADC MIXL" }, { "IB6 Mux", "IF4 DAC L", "IF4 DAC L" }, @@ -3293,8 +3293,8 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "IB6 Mux", "STO2 ADC MIX L", "Stereo2 ADC MIXL" }, { "IB6 Mux", "STO3 ADC MIX L", "Stereo3 ADC MIXL" }, - { "IB7 Mux", "IF1 DAC 7", "IF1 DAC7" }, - { "IB7 Mux", "IF2 DAC 7", "IF2 DAC7" }, + { "IB7 Mux", "IF1 DAC 7", "IF1 DAC7 Mux" }, + { "IB7 Mux", "IF2 DAC 7", "IF2 DAC7 Mux" }, { "IB7 Mux", "SLB DAC 7", "SLB DAC7" }, { "IB7 Mux", "STO4 ADC MIX R", "Stereo4 ADC MIXR" }, { "IB7 Mux", "IF4 DAC R", "IF4 DAC R" }, @@ -3635,15 +3635,15 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "DAC1 FS", NULL, "DAC1 MIXL" }, { "DAC1 FS", NULL, "DAC1 MIXR" }, - { "DAC2 L Mux", "IF1 DAC 2", "IF1 DAC2" }, - { "DAC2 L Mux", "IF2 DAC 2", "IF2 DAC2" }, + { "DAC2 L Mux", "IF1 DAC 2", "IF1 DAC2 Mux" }, + { "DAC2 L Mux", "IF2 DAC 2", "IF2 DAC2 Mux" }, { "DAC2 L Mux", "IF3 DAC L", "IF3 DAC L" }, { "DAC2 L Mux", "IF4 DAC L", "IF4 DAC L" }, { "DAC2 L Mux", "SLB DAC 2", "SLB DAC2" }, { "DAC2 L Mux", "OB 2", "OutBound2" }, - { "DAC2 R Mux", "IF1 DAC 3", "IF1 DAC3" }, - { "DAC2 R Mux", "IF2 DAC 3", "IF2 DAC3" }, + { "DAC2 R Mux", "IF1 DAC 3", "IF1 DAC3 Mux" }, + { "DAC2 R Mux", "IF2 DAC 3", "IF2 DAC3 Mux" }, { "DAC2 R Mux", "IF3 DAC R", "IF3 DAC R" }, { "DAC2 R Mux", "IF4 DAC R", "IF4 DAC R" }, { "DAC2 R Mux", "SLB DAC 3", "SLB DAC3" }, @@ -3651,29 +3651,29 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "DAC2 R Mux", "Haptic Generator", "Haptic Generator" }, { "DAC2 R Mux", "VAD ADC", "VAD ADC Mux" }, - { "DAC3 L Mux", "IF1 DAC 4", "IF1 DAC4" }, - { "DAC3 L Mux", "IF2 DAC 4", "IF2 DAC4" }, + { "DAC3 L Mux", "IF1 DAC 4", "IF1 DAC4 Mux" }, + { "DAC3 L Mux", "IF2 DAC 4", "IF2 DAC4 Mux" }, { "DAC3 L Mux", "IF3 DAC L", "IF3 DAC L" }, { "DAC3 L Mux", "IF4 DAC L", "IF4 DAC L" }, { "DAC3 L Mux", "SLB DAC 4", "SLB DAC4" }, { "DAC3 L Mux", "OB 4", "OutBound4" }, - { "DAC3 R Mux", "IF1 DAC 5", "IF1 DAC4" }, - { "DAC3 R Mux", "IF2 DAC 5", "IF2 DAC4" }, + { "DAC3 R Mux", "IF1 DAC 5", "IF1 DAC5 Mux" }, + { "DAC3 R Mux", "IF2 DAC 5", "IF2 DAC5 Mux" }, { "DAC3 R Mux", "IF3 DAC R", "IF3 DAC R" }, { "DAC3 R Mux", "IF4 DAC R", "IF4 DAC R" }, { "DAC3 R Mux", "SLB DAC 5", "SLB DAC5" }, { "DAC3 R Mux", "OB 5", "OutBound5" }, - { "DAC4 L Mux", "IF1 DAC 6", "IF1 DAC6" }, - { "DAC4 L Mux", "IF2 DAC 6", "IF2 DAC6" }, + { "DAC4 L Mux", "IF1 DAC 6", "IF1 DAC6 Mux" }, + { "DAC4 L Mux", "IF2 DAC 6", "IF2 DAC6 Mux" }, { "DAC4 L Mux", "IF3 DAC L", "IF3 DAC L" }, { "DAC4 L Mux", "IF4 DAC L", "IF4 DAC L" }, { "DAC4 L Mux", "SLB DAC 6", "SLB DAC6" }, { "DAC4 L Mux", "OB 6", "OutBound6" }, - { "DAC4 R Mux", "IF1 DAC 7", "IF1 DAC7" }, - { "DAC4 R Mux", "IF2 DAC 7", "IF2 DAC7" }, + { "DAC4 R Mux", "IF1 DAC 7", "IF1 DAC7 Mux" }, + { "DAC4 R Mux", "IF2 DAC 7", "IF2 DAC7 Mux" }, { "DAC4 R Mux", "IF3 DAC R", "IF3 DAC R" }, { "DAC4 R Mux", "IF4 DAC R", "IF4 DAC R" }, { "DAC4 R Mux", "SLB DAC 7", "SLB DAC7" }, diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 3a1343fa109b..007a0e3bc273 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -106,13 +106,11 @@ static const struct reg_default sta32x_regs[] = { }; static const struct regmap_range sta32x_write_regs_range[] = { - regmap_reg_range(STA32X_CONFA, STA32X_AUTO2), - regmap_reg_range(STA32X_C1CFG, STA32X_FDRC2), + regmap_reg_range(STA32X_CONFA, STA32X_FDRC2), }; static const struct regmap_range sta32x_read_regs_range[] = { - regmap_reg_range(STA32X_CONFA, STA32X_AUTO2), - regmap_reg_range(STA32X_C1CFG, STA32X_FDRC2), + regmap_reg_range(STA32X_CONFA, STA32X_FDRC2), }; static const struct regmap_range sta32x_volatile_regs_range[] = { diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2595611e8a6d..b9fabbf69db6 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -603,10 +603,6 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, factor = (div2 + 1) * (7 * psr + 1) * 2; for (i = 0; i < 255; i++) { - /* The bclk rate must be smaller than 1/5 sysclk rate */ - if (factor * (i + 1) < 5) - continue; - tmprate = freq * factor * (i + 2); if (baudclk_is_used) @@ -614,6 +610,13 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, else clkrate = clk_round_rate(ssi_private->baudclk, tmprate); + /* + * Hardware limitation: The bclk rate must be + * never greater than 1/5 IPG clock rate + */ + if (clkrate * 5 > clk_get_rate(ssi_private->clk)) + continue; + clkrate /= factor; afreq = clkrate / (i + 1); diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index f7c6734bd5da..fb550b5869d2 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -372,6 +372,11 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, strlen(dai_link->cpu_dai_name) + strlen(dai_link->codec_dai_name) + 2, GFP_KERNEL); + if (!name) { + ret = -ENOMEM; + goto dai_link_of_err; + } + sprintf(name, "%s-%s", dai_link->cpu_dai_name, dai_link->codec_dai_name); dai_link->name = dai_link->stream_name = name; diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h index dfebfdd5eb2a..daecc58f28af 100644 --- a/sound/soc/intel/sst-atom-controls.h +++ b/sound/soc/intel/sst-atom-controls.h @@ -150,7 +150,7 @@ enum sst_cmd_type { enum sst_task { SST_TASK_SBA = 1, - SST_TASK_MMX, + SST_TASK_MMX = 3, }; enum sst_type { diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index 8a8d56a146e7..11c578651c1c 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -350,7 +350,9 @@ static inline void sst_save_shim64(struct intel_sst_drv *ctx, spin_lock_irqsave(&ctx->ipc_spin_lock, irq_flags); - shim_regs->imrx = sst_shim_read64(shim, SST_IMRX), + shim_regs->imrx = sst_shim_read64(shim, SST_IMRX); + shim_regs->csr = sst_shim_read64(shim, SST_CSR); + spin_unlock_irqrestore(&ctx->ipc_spin_lock, irq_flags); } @@ -367,6 +369,7 @@ static inline void sst_restore_shim64(struct intel_sst_drv *ctx, */ spin_lock_irqsave(&ctx->ipc_spin_lock, irq_flags); sst_shim_write64(shim, SST_IMRX, shim_regs->imrx), + sst_shim_write64(shim, SST_CSR, shim_regs->csr), spin_unlock_irqrestore(&ctx->ipc_spin_lock, irq_flags); } @@ -379,6 +382,10 @@ void sst_configure_runtime_pm(struct intel_sst_drv *ctx) * initially active. So change the state to active before * enabling the pm */ + + if (!acpi_disabled) + pm_runtime_set_active(ctx->dev); + pm_runtime_enable(ctx->dev); if (acpi_disabled) @@ -409,6 +416,7 @@ static int intel_sst_runtime_suspend(struct device *dev) synchronize_irq(ctx->irq_num); flush_workqueue(ctx->post_msg_wq); + ctx->ops->reset(ctx); /* save the shim registers because PMC doesn't save state */ sst_save_shim64(ctx, ctx->shim, ctx->shim_regs64); diff --git a/sound/soc/omap/omap-hdmi-audio.c b/sound/soc/omap/omap-hdmi-audio.c index ccfb41c22e53..f7eb42aa3f38 100644 --- a/sound/soc/omap/omap-hdmi-audio.c +++ b/sound/soc/omap/omap-hdmi-audio.c @@ -352,6 +352,9 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) return ret; card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + card->name = devm_kasprintf(dev, GFP_KERNEL, "HDMI %s", dev_name(ad->dssdev)); card->owner = THIS_MODULE; diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index c7eb9dd67f60..fd99d89de6a8 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -530,8 +530,19 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, case OMAP_MCBSP_SYSCLK_CLKX_EXT: regs->srgr2 |= CLKSM; + regs->pcr0 |= SCLKME; + /* + * If McBSP is master but yet the CLKX/CLKR pin drives the SRG, + * disable output on those pins. This enables to inject the + * reference clock through CLKX/CLKR. For this to work + * set_dai_sysclk() _needs_ to be called after set_dai_fmt(). + */ + regs->pcr0 &= ~CLKXM; + break; case OMAP_MCBSP_SYSCLK_CLKR_EXT: regs->pcr0 |= SCLKME; + /* Disable ouput on CLKR pin in master mode */ + regs->pcr0 &= ~CLKRM; break; default: err = -ENODEV; diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index f4b05bc23e4b..1343ecbf0bd5 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -201,7 +201,7 @@ static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd) struct snd_pcm *pcm = rtd->pcm; int ret; - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(64)); + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); if (ret) return ret; diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 3cebf6ca03df..0632a36852c8 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -174,7 +174,7 @@ config SND_SOC_SMDK_WM8994_PCM config SND_SOC_SPEYSIDE tristate "Audio support for Wolfson Speyside" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C && SPI_MASTER select SND_SAMSUNG_I2S select SND_SOC_WM8996 select SND_SOC_WM9081 @@ -189,7 +189,7 @@ config SND_SOC_TOBERMORY config SND_SOC_BELLS tristate "Audio support for Wolfson Bells" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && MFD_ARIZONA + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && MFD_ARIZONA && I2C && SPI_MASTER select SND_SAMSUNG_I2S select SND_SOC_WM5102 select SND_SOC_WM5110 @@ -206,7 +206,7 @@ config SND_SOC_LOWLAND config SND_SOC_LITTLEMILL tristate "Audio support for Wolfson Littlemill" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C select SND_SAMSUNG_I2S select MFD_WM8994 select SND_SOC_WM8994 @@ -223,7 +223,7 @@ config SND_SOC_SNOW config SND_SOC_ODROIDX2 tristate "Audio support for Odroid-X2 and Odroid-U3" - depends on SND_SOC_SAMSUNG + depends on SND_SOC_SAMSUNG && I2C select SND_SOC_MAX98090 select SND_SAMSUNG_I2S help @@ -231,6 +231,6 @@ config SND_SOC_ODROIDX2 config SND_SOC_ARNDALE_RT5631_ALC5631 tristate "Audio support for RT5631(ALC5631) on Arndale Board" - depends on SND_SOC_SAMSUNG + depends on SND_SOC_SAMSUNG && I2C select SND_SAMSUNG_I2S select SND_SOC_RT5631 diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 1b53605f7154..110577c52317 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1252,6 +1252,8 @@ static int rsnd_probe(struct platform_device *pdev) goto exit_snd_probe; } + dev_set_drvdata(dev, priv); + /* * asoc register */ @@ -1268,8 +1270,6 @@ static int rsnd_probe(struct platform_device *pdev) goto exit_snd_soc; } - dev_set_drvdata(dev, priv); - pm_runtime_enable(dev); dev_info(dev, "probed\n"); diff --git a/sound/usb/line6/playback.c b/sound/usb/line6/playback.c index 05dee690f487..97ed593f6010 100644 --- a/sound/usb/line6/playback.c +++ b/sound/usb/line6/playback.c @@ -39,7 +39,7 @@ static void change_volume(struct urb *urb_out, int volume[], for (; p < buf_end; ++p) { short pv = le16_to_cpu(*p); int val = (pv * volume[chn & 1]) >> 8; - pv = clamp(val, 0x7fff, -0x8000); + pv = clamp(val, -0x8000, 0x7fff); *p = cpu_to_le16(pv); ++chn; } @@ -54,7 +54,7 @@ static void change_volume(struct urb *urb_out, int volume[], val = p[0] + (p[1] << 8) + ((signed char)p[2] << 16); val = (val * volume[chn & 1]) >> 8; - val = clamp(val, 0x7fffff, -0x800000); + val = clamp(val, -0x800000, 0x7fffff); p[0] = val; p[1] = val >> 8; p[2] = val >> 16; @@ -126,7 +126,7 @@ static void add_monitor_signal(struct urb *urb_out, unsigned char *signal, short pov = le16_to_cpu(*po); short piv = le16_to_cpu(*pi); int val = pov + ((piv * volume) >> 8); - pov = clamp(val, 0x7fff, -0x8000); + pov = clamp(val, -0x8000, 0x7fff); *po = cpu_to_le16(pov); } } diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 67d476548dcf..07f984d5f516 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1773,6 +1773,36 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + USB_DEVICE(0x0582, 0x0159), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Roland", */ + /* .product_name = "UA-22", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, /* this catches most recent vendor-specific Roland devices */ { .match_flags = USB_DEVICE_ID_MATCH_VENDOR | |