diff options
Diffstat (limited to 'sound')
39 files changed, 588 insertions, 320 deletions
diff --git a/sound/core/control.c b/sound/core/control.c index 9aa15bfc7936..649d3217590e 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -348,6 +348,40 @@ static int snd_ctl_find_hole(struct snd_card *card, unsigned int count) return 0; } +/* add a new kcontrol object; call with card->controls_rwsem locked */ +static int __snd_ctl_add(struct snd_card *card, struct snd_kcontrol *kcontrol) +{ + struct snd_ctl_elem_id id; + unsigned int idx; + unsigned int count; + + id = kcontrol->id; + if (id.index > UINT_MAX - kcontrol->count) + return -EINVAL; + + if (snd_ctl_find_id(card, &id)) { + dev_err(card->dev, + "control %i:%i:%i:%s:%i is already present\n", + id.iface, id.device, id.subdevice, id.name, id.index); + return -EBUSY; + } + + if (snd_ctl_find_hole(card, kcontrol->count) < 0) + return -ENOMEM; + + list_add_tail(&kcontrol->list, &card->controls); + card->controls_count += kcontrol->count; + kcontrol->id.numid = card->last_numid + 1; + card->last_numid += kcontrol->count; + + id = kcontrol->id; + count = kcontrol->count; + for (idx = 0; idx < count; idx++, id.index++, id.numid++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_ADD, &id); + + return 0; +} + /** * snd_ctl_add - add the control instance to the card * @card: the card instance @@ -364,45 +398,18 @@ static int snd_ctl_find_hole(struct snd_card *card, unsigned int count) */ int snd_ctl_add(struct snd_card *card, struct snd_kcontrol *kcontrol) { - struct snd_ctl_elem_id id; - unsigned int idx; - unsigned int count; int err = -EINVAL; if (! kcontrol) return err; if (snd_BUG_ON(!card || !kcontrol->info)) goto error; - id = kcontrol->id; - if (id.index > UINT_MAX - kcontrol->count) - goto error; down_write(&card->controls_rwsem); - if (snd_ctl_find_id(card, &id)) { - up_write(&card->controls_rwsem); - dev_err(card->dev, "control %i:%i:%i:%s:%i is already present\n", - id.iface, - id.device, - id.subdevice, - id.name, - id.index); - err = -EBUSY; - goto error; - } - if (snd_ctl_find_hole(card, kcontrol->count) < 0) { - up_write(&card->controls_rwsem); - err = -ENOMEM; - goto error; - } - list_add_tail(&kcontrol->list, &card->controls); - card->controls_count += kcontrol->count; - kcontrol->id.numid = card->last_numid + 1; - card->last_numid += kcontrol->count; - id = kcontrol->id; - count = kcontrol->count; + err = __snd_ctl_add(card, kcontrol); up_write(&card->controls_rwsem); - for (idx = 0; idx < count; idx++, id.index++, id.numid++) - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_ADD, &id); + if (err < 0) + goto error; return 0; error: @@ -1361,9 +1368,12 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, kctl->tlv.c = snd_ctl_elem_user_tlv; /* This function manage to free the instance on failure. */ - err = snd_ctl_add(card, kctl); - if (err < 0) - return err; + down_write(&card->controls_rwsem); + err = __snd_ctl_add(card, kctl); + if (err < 0) { + snd_ctl_free_one(kctl); + goto unlock; + } offset = snd_ctl_get_ioff(kctl, &info->id); snd_ctl_build_ioff(&info->id, kctl, offset); /* @@ -1374,10 +1384,10 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, * which locks the element. */ - down_write(&card->controls_rwsem); card->user_ctl_count++; - up_write(&card->controls_rwsem); + unlock: + up_write(&card->controls_rwsem); return 0; } diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index f8d4a419f3af..467039b342b5 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1062,8 +1062,8 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream) runtime->oss.channels = params_channels(params); runtime->oss.rate = params_rate(params); - vfree(runtime->oss.buffer); - runtime->oss.buffer = vmalloc(runtime->oss.period_bytes); + kvfree(runtime->oss.buffer); + runtime->oss.buffer = kvzalloc(runtime->oss.period_bytes, GFP_KERNEL); if (!runtime->oss.buffer) { err = -ENOMEM; goto failure; @@ -2328,7 +2328,7 @@ static void snd_pcm_oss_release_substream(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime; runtime = substream->runtime; - vfree(runtime->oss.buffer); + kvfree(runtime->oss.buffer); runtime->oss.buffer = NULL; #ifdef CONFIG_SND_PCM_OSS_PLUGINS snd_pcm_oss_plugin_clear(substream); diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 141c5f3a9575..31cb2acf8afc 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -66,8 +66,8 @@ static int snd_pcm_plugin_alloc(struct snd_pcm_plugin *plugin, snd_pcm_uframes_t return -ENXIO; size /= 8; if (plugin->buf_frames < frames) { - vfree(plugin->buf); - plugin->buf = vmalloc(size); + kvfree(plugin->buf); + plugin->buf = kvzalloc(size, GFP_KERNEL); plugin->buf_frames = frames; } if (!plugin->buf) { @@ -191,7 +191,7 @@ int snd_pcm_plugin_free(struct snd_pcm_plugin *plugin) if (plugin->private_free) plugin->private_free(plugin); kfree(plugin->buf_channels); - vfree(plugin->buf); + kvfree(plugin->buf); kfree(plugin); return 0; } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 66c90f486af9..818dff1de545 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -36,6 +36,7 @@ #include <sound/timer.h> #include <sound/minors.h> #include <linux/uio.h> +#include <linux/delay.h> #include "pcm_local.h" @@ -91,12 +92,12 @@ static DECLARE_RWSEM(snd_pcm_link_rwsem); * and this may lead to a deadlock when the code path takes read sem * twice (e.g. one in snd_pcm_action_nonatomic() and another in * snd_pcm_stream_lock()). As a (suboptimal) workaround, let writer to - * spin until it gets the lock. + * sleep until all the readers are completed without blocking by writer. */ -static inline void down_write_nonblock(struct rw_semaphore *lock) +static inline void down_write_nonfifo(struct rw_semaphore *lock) { while (!down_write_trylock(lock)) - cond_resched(); + msleep(1); } #define PCM_LOCK_DEFAULT 0 @@ -1967,7 +1968,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) res = -ENOMEM; goto _nolock; } - down_write_nonblock(&snd_pcm_link_rwsem); + down_write_nonfifo(&snd_pcm_link_rwsem); write_lock_irq(&snd_pcm_link_rwlock); if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN || substream->runtime->status->state != substream1->runtime->status->state || @@ -2014,7 +2015,7 @@ static int snd_pcm_unlink(struct snd_pcm_substream *substream) struct snd_pcm_substream *s; int res = 0; - down_write_nonblock(&snd_pcm_link_rwsem); + down_write_nonfifo(&snd_pcm_link_rwsem); write_lock_irq(&snd_pcm_link_rwlock); if (!snd_pcm_stream_linked(substream)) { res = -EALREADY; @@ -2369,7 +2370,8 @@ int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) static void pcm_release_private(struct snd_pcm_substream *substream) { - snd_pcm_unlink(substream); + if (snd_pcm_stream_linked(substream)) + snd_pcm_unlink(substream); } void snd_pcm_release_substream(struct snd_pcm_substream *substream) diff --git a/sound/firewire/fireface/ff-protocol-ff400.c b/sound/firewire/fireface/ff-protocol-ff400.c index 64c3cb0fb926..654a50319198 100644 --- a/sound/firewire/fireface/ff-protocol-ff400.c +++ b/sound/firewire/fireface/ff-protocol-ff400.c @@ -30,7 +30,7 @@ static int ff400_get_clock(struct snd_ff *ff, unsigned int *rate, int err; err = snd_fw_transaction(ff->unit, TCODE_READ_QUADLET_REQUEST, - FF400_SYNC_STATUS, ®, sizeof(reg), 0); + FF400_CLOCK_CONFIG, ®, sizeof(reg), 0); if (err < 0) return err; data = le32_to_cpu(reg); diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 32453f81b95a..3a5008837576 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -1531,7 +1531,6 @@ static int snd_wss_playback_open(struct snd_pcm_substream *substream) if (err < 0) { if (chip->release_dma) chip->release_dma(chip, chip->dma_private_data, chip->dma1); - snd_free_pages(runtime->dma_area, runtime->dma_bytes); return err; } chip->playback_substream = substream; @@ -1572,7 +1571,6 @@ static int snd_wss_capture_open(struct snd_pcm_substream *substream) if (err < 0) { if (chip->release_dma) chip->release_dma(chip, chip->dma_private_data, chip->dma2); - snd_free_pages(runtime->dma_area, runtime->dma_bytes); return err; } chip->capture_substream = substream; diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index f4459d1a9d67..27b468f057dd 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -824,7 +824,7 @@ static int snd_ac97_put_spsa(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ { struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0xff; + int shift = (kcontrol->private_value >> 8) & 0x0f; int mask = (kcontrol->private_value >> 16) & 0xff; // int invert = (kcontrol->private_value >> 24) & 0xff; unsigned short value, old, new; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d8eb2b5f51ae..76f03abd15ab 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2169,6 +2169,8 @@ static struct snd_pci_quirk power_save_blacklist[] = { /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1849, 0xc892, "Asrock B85M-ITX", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ + SND_PCI_QUIRK(0x1849, 0x0397, "Asrock N68C-S UCC", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1849, 0x7662, "Asrock H81M-HDS", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1043, 0x8733, "Asus Prime X370-Pro", 0), @@ -2496,6 +2498,10 @@ static const struct pci_device_id azx_ids[] = { /* AMD Hudson */ { PCI_DEVICE(0x1022, 0x780d), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB }, + /* AMD Stoney */ + { PCI_DEVICE(0x1022, 0x157a), + .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB | + AZX_DCAPS_PM_RUNTIME }, /* AMD Raven */ { PCI_DEVICE(0x1022, 0x15e3), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB | diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 0a24037184c3..0a567634e5fa 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1177,6 +1177,7 @@ static const struct snd_pci_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1028, 0x0708, "Alienware 15 R2 2016", QUIRK_ALIENWARE), SND_PCI_QUIRK(0x1102, 0x0010, "Sound Blaster Z", QUIRK_SBZ), SND_PCI_QUIRK(0x1102, 0x0023, "Sound Blaster Z", QUIRK_SBZ), + SND_PCI_QUIRK(0x1102, 0x0033, "Sound Blaster ZxR", QUIRK_SBZ), SND_PCI_QUIRK(0x1458, 0xA016, "Recon3Di", QUIRK_R3DI), SND_PCI_QUIRK(0x1458, 0xA026, "Gigabyte G1.Sniper Z97", QUIRK_R3DI), SND_PCI_QUIRK(0x1458, 0xA036, "Gigabyte GA-Z170X-Gaming 7", QUIRK_R3DI), @@ -8413,7 +8414,7 @@ static void ca0132_free(struct hda_codec *codec) snd_hda_power_down(codec); if (spec->mem_base) - iounmap(spec->mem_base); + pci_iounmap(codec->bus->pci, spec->mem_base); kfree(spec->spec_init_verbs); kfree(codec->spec); } @@ -8488,7 +8489,7 @@ static void ca0132_config(struct hda_codec *codec) break; case QUIRK_AE5: codec_dbg(codec, "%s: QUIRK_AE5 applied.\n", __func__); - snd_hda_apply_pincfgs(codec, r3di_pincfgs); + snd_hda_apply_pincfgs(codec, ae5_pincfgs); break; } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fa61674a5605..15021c839372 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -388,6 +388,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0285: case 0x10ec0298: case 0x10ec0289: + case 0x10ec0300: alc_update_coef_idx(codec, 0x10, 1<<9, 0); break; case 0x10ec0275: @@ -2830,6 +2831,7 @@ enum { ALC269_TYPE_ALC215, ALC269_TYPE_ALC225, ALC269_TYPE_ALC294, + ALC269_TYPE_ALC300, ALC269_TYPE_ALC700, }; @@ -2864,6 +2866,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) case ALC269_TYPE_ALC215: case ALC269_TYPE_ALC225: case ALC269_TYPE_ALC294: + case ALC269_TYPE_ALC300: case ALC269_TYPE_ALC700: ssids = alc269_ssids; break; @@ -4985,9 +4988,18 @@ static void alc_fixup_tpt470_dock(struct hda_codec *codec, { 0x19, 0x21a11010 }, /* dock mic */ { } }; + /* Assure the speaker pin to be coupled with DAC NID 0x03; otherwise + * the speaker output becomes too low by some reason on Thinkpads with + * ALC298 codec + */ + static hda_nid_t preferred_pairs[] = { + 0x14, 0x03, 0x17, 0x02, 0x21, 0x02, + 0 + }; struct alc_spec *spec = codec->spec; if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->gen.preferred_dacs = preferred_pairs; spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; snd_hda_apply_pincfgs(codec, pincfgs); } else if (action == HDA_FIXUP_ACT_INIT) { @@ -5358,6 +5370,16 @@ static void alc274_fixup_bind_dacs(struct hda_codec *codec, spec->gen.preferred_dacs = preferred_pairs; } +/* The DAC of NID 0x3 will introduce click/pop noise on headphones, so invalidate it */ +static void alc285_fixup_invalidate_dacs(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + snd_hda_override_wcaps(codec, 0x03, 0); +} + /* for hda_fixup_thinkpad_acpi() */ #include "thinkpad_helper.c" @@ -5495,6 +5517,12 @@ enum { ALC255_FIXUP_DELL_HEADSET_MIC, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, + ALC285_FIXUP_LENOVO_HEADPHONE_NOISE, + ALC295_FIXUP_HP_AUTO_MUTE, + ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE, + ALC294_FIXUP_ASUS_MIC, + ALC294_FIXUP_ASUS_HEADSET_MIC, + ALC294_FIXUP_ASUS_SPK, }; static const struct hda_fixup alc269_fixups[] = { @@ -5659,6 +5687,8 @@ static const struct hda_fixup alc269_fixups[] = { [ALC269_FIXUP_HP_MUTE_LED_MIC3] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_hp_mute_led_mic3, + .chained = true, + .chain_id = ALC295_FIXUP_HP_AUTO_MUTE }, [ALC269_FIXUP_HP_GPIO_LED] = { .type = HDA_FIXUP_FUNC, @@ -6362,6 +6392,55 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC285_FIXUP_LENOVO_HEADPHONE_NOISE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_invalidate_dacs, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI + }, + [ALC295_FIXUP_HP_AUTO_MUTE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_auto_mute_via_amp, + }, + [ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MIC + }, + [ALC294_FIXUP_ASUS_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x13, 0x90a60160 }, /* use as internal mic */ + { 0x19, 0x04a11120 }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, + [ALC294_FIXUP_ASUS_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1113c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, + [ALC294_FIXUP_ASUS_SPK] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* Set EAPD high */ + { 0x20, AC_VERB_SET_COEF_INDEX, 0x40 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x8800 }, + { } + }, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -6376,7 +6455,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0762, "Acer Aspire E1-472", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS), + SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK), + SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS), SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X), @@ -6481,6 +6564,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2336, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x221c, "HP EliteBook 755 G2", ALC280_FIXUP_HP_HEADSET_MIC), + SND_PCI_QUIRK(0x103c, 0x820d, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8256, "HP", ALC221_FIXUP_HP_FRONT_MIC), SND_PCI_QUIRK(0x103c, 0x827e, "HP x360", ALC295_FIXUP_HP_X360), SND_PCI_QUIRK(0x103c, 0x82bf, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE), @@ -6499,6 +6583,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x12e0, "ASUS X541SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), + SND_PCI_QUIRK(0x1043, 0x14a1, "ASUS UX533FD", ALC294_FIXUP_ASUS_SPK), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), @@ -6531,6 +6616,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC233_FIXUP_LENOVO_MULTI_CODECS), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), @@ -7033,6 +7119,15 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x12, 0x90a60130}, {0x19, 0x03a11020}, {0x21, 0x0321101f}), + SND_HDA_PIN_QUIRK(0x10ec0285, 0x17aa, "Lenovo", ALC285_FIXUP_LENOVO_HEADPHONE_NOISE, + {0x12, 0x90a60130}, + {0x14, 0x90170110}, + {0x19, 0x04a11040}, + {0x21, 0x04211020}), + SND_HDA_PIN_QUIRK(0x10ec0286, 0x1025, "Acer", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE, + {0x12, 0x90a60130}, + {0x17, 0x90170110}, + {0x21, 0x02211020}), SND_HDA_PIN_QUIRK(0x10ec0288, 0x1028, "Dell", ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60120}, {0x14, 0x90170110}, @@ -7096,6 +7191,14 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0293, 0x1028, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_STANDARD_PINS, {0x13, 0x90a60140}), + SND_HDA_PIN_QUIRK(0x10ec0294, 0x1043, "ASUS", ALC294_FIXUP_ASUS_MIC, + {0x14, 0x90170110}, + {0x1b, 0x90a70130}, + {0x21, 0x04211020}), + SND_HDA_PIN_QUIRK(0x10ec0294, 0x1043, "ASUS", ALC294_FIXUP_ASUS_SPK, + {0x12, 0x90a60130}, + {0x17, 0x90170110}, + {0x21, 0x04211020}), SND_HDA_PIN_QUIRK(0x10ec0295, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, ALC295_STANDARD_PINS, {0x17, 0x21014020}, @@ -7168,6 +7271,37 @@ static void alc269_fill_coef(struct hda_codec *codec) alc_update_coef_idx(codec, 0x4, 0, 1<<11); } +static void alc294_hp_init(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + int i, val; + + if (!hp_pin) + return; + + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + + msleep(100); + + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + + alc_update_coef_idx(codec, 0x6f, 0x000f, 0);/* Set HP depop to manual mode */ + alc_update_coefex_idx(codec, 0x58, 0x00, 0x8000, 0x8000); /* HP depop procedure start */ + + /* Wait for depop procedure finish */ + val = alc_read_coefex_idx(codec, 0x58, 0x01); + for (i = 0; i < 20 && val & 0x0080; i++) { + msleep(50); + val = alc_read_coefex_idx(codec, 0x58, 0x01); + } + /* Set HP depop to auto mode */ + alc_update_coef_idx(codec, 0x6f, 0x000f, 0x000b); + msleep(50); +} + /* */ static int patch_alc269(struct hda_codec *codec) @@ -7293,6 +7427,11 @@ static int patch_alc269(struct hda_codec *codec) spec->codec_variant = ALC269_TYPE_ALC294; spec->gen.mixer_nid = 0; /* ALC2x4 does not have any loopback mixer path */ alc_update_coef_idx(codec, 0x6b, 0x0018, (1<<4) | (1<<3)); /* UAJ MIC Vref control by verb */ + alc294_hp_init(codec); + break; + case 0x10ec0300: + spec->codec_variant = ALC269_TYPE_ALC300; + spec->gen.mixer_nid = 0; /* no loopback on ALC300 */ break; case 0x10ec0700: case 0x10ec0701: @@ -7300,6 +7439,7 @@ static int patch_alc269(struct hda_codec *codec) spec->codec_variant = ALC269_TYPE_ALC700; spec->gen.mixer_nid = 0; /* ALC700 does not have any loopback mixer path */ alc_update_coef_idx(codec, 0x4a, 1 << 15, 0); /* Combo jack auto trigger control */ + alc294_hp_init(codec); break; } @@ -8404,6 +8544,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0295, "ALC295", patch_alc269), HDA_CODEC_ENTRY(0x10ec0298, "ALC298", patch_alc269), HDA_CODEC_ENTRY(0x10ec0299, "ALC299", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0300, "ALC300", patch_alc269), HDA_CODEC_REV_ENTRY(0x10ec0861, 0x100340, "ALC660", patch_alc861), HDA_CODEC_ENTRY(0x10ec0660, "ALC660-VD", patch_alc861vd), HDA_CODEC_ENTRY(0x10ec0861, "ALC861", patch_alc861), diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c index 97f49b751e6e..568575b72f2f 100644 --- a/sound/pci/hda/thinkpad_helper.c +++ b/sound/pci/hda/thinkpad_helper.c @@ -58,8 +58,8 @@ static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, removefunc = false; } if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0 && - snd_hda_gen_add_micmute_led(codec, - update_tpacpi_micmute) > 0) + !snd_hda_gen_add_micmute_led(codec, + update_tpacpi_micmute)) removefunc = false; } diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 4e9854889a95..e63d6e33df48 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -2187,11 +2187,6 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) */ snd_hdac_codec_read(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - err = snd_hdac_display_power(bus, false); - if (err < 0) { - dev_err(dev, "Cannot turn on display power on i915\n"); - return err; - } hlink = snd_hdac_ext_bus_get_link(bus, dev_name(dev)); if (!hlink) { @@ -2201,7 +2196,11 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) snd_hdac_ext_bus_link_put(bus, hlink); - return 0; + err = snd_hdac_display_power(bus, false); + if (err < 0) + dev_err(dev, "Cannot turn off display power on i915\n"); + + return err; } static int hdac_hdmi_runtime_resume(struct device *dev) diff --git a/sound/soc/codecs/pcm186x.h b/sound/soc/codecs/pcm186x.h index 2c6ba55bf394..bb3f0c42a1cd 100644 --- a/sound/soc/codecs/pcm186x.h +++ b/sound/soc/codecs/pcm186x.h @@ -139,7 +139,7 @@ enum pcm186x_type { #define PCM186X_MAX_REGISTER PCM186X_CURR_TRIM_CTRL /* PCM186X_PAGE */ -#define PCM186X_RESET 0xff +#define PCM186X_RESET 0xfe /* PCM186X_ADCX_INPUT_SEL_X */ #define PCM186X_ADC_INPUT_SEL_POL BIT(7) diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c index 494d9d662be8..771b46e1974b 100644 --- a/sound/soc/codecs/pcm3060.c +++ b/sound/soc/codecs/pcm3060.c @@ -198,20 +198,16 @@ static const struct snd_kcontrol_new pcm3060_dapm_controls[] = { }; static const struct snd_soc_dapm_widget pcm3060_dapm_widgets[] = { - SND_SOC_DAPM_OUTPUT("OUTL+"), - SND_SOC_DAPM_OUTPUT("OUTR+"), - SND_SOC_DAPM_OUTPUT("OUTL-"), - SND_SOC_DAPM_OUTPUT("OUTR-"), + SND_SOC_DAPM_OUTPUT("OUTL"), + SND_SOC_DAPM_OUTPUT("OUTR"), SND_SOC_DAPM_INPUT("INL"), SND_SOC_DAPM_INPUT("INR"), }; static const struct snd_soc_dapm_route pcm3060_dapm_map[] = { - { "OUTL+", NULL, "Playback" }, - { "OUTR+", NULL, "Playback" }, - { "OUTL-", NULL, "Playback" }, - { "OUTR-", NULL, "Playback" }, + { "OUTL", NULL, "Playback" }, + { "OUTR", NULL, "Playback" }, { "Capture", NULL, "INL" }, { "Capture", NULL, "INR" }, diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index a53dc174bbf0..66501b8dc46f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -765,38 +765,41 @@ static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *mem, static void wm_adsp2_show_fw_status(struct wm_adsp *dsp) { - u16 scratch[4]; + unsigned int scratch[4]; + unsigned int addr = dsp->base + ADSP2_SCRATCH0; + unsigned int i; int ret; - ret = regmap_raw_read(dsp->regmap, dsp->base + ADSP2_SCRATCH0, - scratch, sizeof(scratch)); - if (ret) { - adsp_err(dsp, "Failed to read SCRATCH regs: %d\n", ret); - return; + for (i = 0; i < ARRAY_SIZE(scratch); ++i) { + ret = regmap_read(dsp->regmap, addr + i, &scratch[i]); + if (ret) { + adsp_err(dsp, "Failed to read SCRATCH%u: %d\n", i, ret); + return; + } } adsp_dbg(dsp, "FW SCRATCH 0:0x%x 1:0x%x 2:0x%x 3:0x%x\n", - be16_to_cpu(scratch[0]), - be16_to_cpu(scratch[1]), - be16_to_cpu(scratch[2]), - be16_to_cpu(scratch[3])); + scratch[0], scratch[1], scratch[2], scratch[3]); } static void wm_adsp2v2_show_fw_status(struct wm_adsp *dsp) { - u32 scratch[2]; + unsigned int scratch[2]; int ret; - ret = regmap_raw_read(dsp->regmap, dsp->base + ADSP2V2_SCRATCH0_1, - scratch, sizeof(scratch)); - + ret = regmap_read(dsp->regmap, dsp->base + ADSP2V2_SCRATCH0_1, + &scratch[0]); if (ret) { - adsp_err(dsp, "Failed to read SCRATCH regs: %d\n", ret); + adsp_err(dsp, "Failed to read SCRATCH0_1: %d\n", ret); return; } - scratch[0] = be32_to_cpu(scratch[0]); - scratch[1] = be32_to_cpu(scratch[1]); + ret = regmap_read(dsp->regmap, dsp->base + ADSP2V2_SCRATCH2_3, + &scratch[1]); + if (ret) { + adsp_err(dsp, "Failed to read SCRATCH2_3: %d\n", ret); + return; + } adsp_dbg(dsp, "FW SCRATCH 0:0x%x 1:0x%x 2:0x%x 3:0x%x\n", scratch[0] & 0xFFFF, diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 0caa1f4eb94d..18e717703685 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -101,22 +101,42 @@ config SND_SST_ATOM_HIFI2_PLATFORM_ACPI codec, then enable this option by saying Y or m. This is a recommended option -config SND_SOC_INTEL_SKYLAKE_SSP_CLK - tristate - config SND_SOC_INTEL_SKYLAKE tristate "SKL/BXT/KBL/GLK/CNL... Platforms" depends on PCI && ACPI + select SND_SOC_INTEL_SKYLAKE_COMMON + help + If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ + GeminiLake or CannonLake platform with the DSP enabled in the BIOS + then enable this option by saying Y or m. + +if SND_SOC_INTEL_SKYLAKE + +config SND_SOC_INTEL_SKYLAKE_SSP_CLK + tristate + +config SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC + bool "HDAudio codec support" + help + If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ + GeminiLake or CannonLake platform with an HDaudio codec + then enable this option by saying Y + +config SND_SOC_INTEL_SKYLAKE_COMMON + tristate select SND_HDA_EXT_CORE select SND_HDA_DSP_LOADER select SND_SOC_TOPOLOGY select SND_SOC_INTEL_SST + select SND_SOC_HDAC_HDA if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC select SND_SOC_ACPI_INTEL_MATCH help If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ GeminiLake or CannonLake platform with the DSP enabled in the BIOS then enable this option by saying Y or m. +endif ## SND_SOC_INTEL_SKYLAKE + config SND_SOC_ACPI_INTEL_MATCH tristate select SND_SOC_ACPI if ACPI diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 73ca1350aa31..b177db2a0dbb 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -293,16 +293,6 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH Say Y if you have such a device. If unsure select "N". -config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH - tristate "SKL/KBL/BXT/APL with HDA Codecs" - select SND_SOC_HDAC_HDMI - select SND_SOC_HDAC_HDA - help - This adds support for ASoC machine driver for Intel platforms - SKL/KBL/BXT/APL with iDisp, HDA audio codecs. - Say Y or m if you have such a device. This is a recommended option. - If unsure select "N". - config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH tristate "GLK with RT5682 and MAX98357A in I2S Mode" depends on MFD_INTEL_LPSS && I2C && ACPI @@ -319,4 +309,18 @@ config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH endif ## SND_SOC_INTEL_SKYLAKE +if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC + +config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH + tristate "SKL/KBL/BXT/APL with HDA Codecs" + select SND_SOC_HDAC_HDMI + # SND_SOC_HDAC_HDA is already selected + help + This adds support for ASoC machine driver for Intel platforms + SKL/KBL/BXT/APL with iDisp, HDA audio codecs. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + +endif ## SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC + endif ## SND_SOC_INTEL_MACH diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index db6976f4ddaa..9d9f6e41d81c 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -19,6 +19,7 @@ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ */ +#include <linux/dmi.h> #include <linux/module.h> #include <linux/platform_device.h> #include <linux/slab.h> @@ -35,6 +36,8 @@ #define CHT_PLAT_CLK_3_HZ 19200000 #define CHT_CODEC_DAI "HiFi" +#define QUIRK_PMC_PLT_CLK_0 0x01 + struct cht_mc_private { struct clk *mclk; struct snd_soc_jack jack; @@ -385,11 +388,29 @@ static struct snd_soc_card snd_soc_card_cht = { .num_controls = ARRAY_SIZE(cht_mc_controls), }; +static const struct dmi_system_id cht_max98090_quirk_table[] = { + { + /* Swanky model Chromebook (Toshiba Chromebook 2) */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Swanky"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + {} +}; + static int snd_cht_mc_probe(struct platform_device *pdev) { + const struct dmi_system_id *dmi_id; struct device *dev = &pdev->dev; int ret_val = 0; struct cht_mc_private *drv; + const char *mclk_name; + int quirks = 0; + + dmi_id = dmi_first_match(cht_max98090_quirk_table); + if (dmi_id) + quirks = (unsigned long)dmi_id->driver_data; drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); if (!drv) @@ -411,11 +432,16 @@ static int snd_cht_mc_probe(struct platform_device *pdev) snd_soc_card_cht.dev = &pdev->dev; snd_soc_card_set_drvdata(&snd_soc_card_cht, drv); - drv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); + if (quirks & QUIRK_PMC_PLT_CLK_0) + mclk_name = "pmc_plt_clk_0"; + else + mclk_name = "pmc_plt_clk_3"; + + drv->mclk = devm_clk_get(&pdev->dev, mclk_name); if (IS_ERR(drv->mclk)) { dev_err(&pdev->dev, - "Failed to get MCLK from pmc_plt_clk_3: %ld\n", - PTR_ERR(drv->mclk)); + "Failed to get MCLK from %s: %ld\n", + mclk_name, PTR_ERR(drv->mclk)); return PTR_ERR(drv->mclk); } diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 29225623b4b4..7487f388e65d 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -37,7 +37,9 @@ #include "skl.h" #include "skl-sst-dsp.h" #include "skl-sst-ipc.h" +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) #include "../../../soc/codecs/hdac_hda.h" +#endif /* * initialize the PCI registers @@ -658,6 +660,8 @@ static void skl_clock_device_unregister(struct skl *skl) platform_device_unregister(skl->clk_dev); } +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) + #define IDISP_INTEL_VENDOR_ID 0x80860000 /* @@ -676,6 +680,8 @@ static void load_codec_module(struct hda_codec *codec) #endif } +#endif /* CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC */ + /* * Probe the given codec address */ @@ -685,9 +691,11 @@ static int probe_codec(struct hdac_bus *bus, int addr) (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID; unsigned int res = -1; struct skl *skl = bus_to_skl(bus); +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) struct hdac_hda_priv *hda_codec; - struct hdac_device *hdev; int err; +#endif + struct hdac_device *hdev; mutex_lock(&bus->cmd_mutex); snd_hdac_bus_send_cmd(bus, cmd); @@ -697,6 +705,7 @@ static int probe_codec(struct hdac_bus *bus, int addr) return -EIO; dev_dbg(bus->dev, "codec #%d probed OK: %x\n", addr, res); +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) hda_codec = devm_kzalloc(&skl->pci->dev, sizeof(*hda_codec), GFP_KERNEL); if (!hda_codec) @@ -715,6 +724,13 @@ static int probe_codec(struct hdac_bus *bus, int addr) load_codec_module(&hda_codec->codec); } return 0; +#else + hdev = devm_kzalloc(&skl->pci->dev, sizeof(*hdev), GFP_KERNEL); + if (!hdev) + return -ENOMEM; + + return snd_hdac_ext_bus_device_init(bus, addr, hdev); +#endif /* CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC */ } /* Codec initialization */ @@ -815,6 +831,12 @@ static void skl_probe_work(struct work_struct *work) } } + /* + * we are done probing so decrement link counts + */ + list_for_each_entry(hlink, &bus->hlink_list, list) + snd_hdac_ext_bus_link_put(bus, hlink); + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { err = snd_hdac_display_power(bus, false); if (err < 0) { @@ -824,12 +846,6 @@ static void skl_probe_work(struct work_struct *work) } } - /* - * we are done probing so decrement link counts - */ - list_for_each_entry(hlink, &bus->hlink_list, list) - snd_hdac_ext_bus_link_put(bus, hlink); - /* configure PM */ pm_runtime_put_noidle(bus->dev); pm_runtime_allow(bus->dev); @@ -870,7 +886,7 @@ static int skl_create(struct pci_dev *pci, hbus = skl_to_hbus(skl); bus = skl_to_bus(skl); -#if IS_ENABLED(CONFIG_SND_SOC_HDAC_HDA) +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) ext_ops = snd_soc_hdac_hda_get_ops(); #endif snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, io_ops, ext_ops); diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index d5ae9eb8c756..fed45b41f9d3 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -36,6 +36,8 @@ #include "../codecs/twl6040.h" struct abe_twl6040 { + struct snd_soc_card card; + struct snd_soc_dai_link dai_links[2]; int jack_detection; /* board can detect jack events */ int mclk_freq; /* MCLK frequency speed for twl6040 */ }; @@ -208,40 +210,10 @@ static int omap_abe_dmic_init(struct snd_soc_pcm_runtime *rtd) ARRAY_SIZE(dmic_audio_map)); } -/* Digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link abe_twl6040_dai_links[] = { - { - .name = "TWL6040", - .stream_name = "TWL6040", - .codec_dai_name = "twl6040-legacy", - .codec_name = "twl6040-codec", - .init = omap_abe_twl6040_init, - .ops = &omap_abe_ops, - }, - { - .name = "DMIC", - .stream_name = "DMIC Capture", - .codec_dai_name = "dmic-hifi", - .codec_name = "dmic-codec", - .init = omap_abe_dmic_init, - .ops = &omap_abe_dmic_ops, - }, -}; - -/* Audio machine driver */ -static struct snd_soc_card omap_abe_card = { - .owner = THIS_MODULE, - - .dapm_widgets = twl6040_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets), - .dapm_routes = audio_map, - .num_dapm_routes = ARRAY_SIZE(audio_map), -}; - static int omap_abe_probe(struct platform_device *pdev) { struct device_node *node = pdev->dev.of_node; - struct snd_soc_card *card = &omap_abe_card; + struct snd_soc_card *card; struct device_node *dai_node; struct abe_twl6040 *priv; int num_links = 0; @@ -252,12 +224,18 @@ static int omap_abe_probe(struct platform_device *pdev) return -ENODEV; } - card->dev = &pdev->dev; - priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL); if (priv == NULL) return -ENOMEM; + card = &priv->card; + card->dev = &pdev->dev; + card->owner = THIS_MODULE; + card->dapm_widgets = twl6040_dapm_widgets; + card->num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets); + card->dapm_routes = audio_map; + card->num_dapm_routes = ARRAY_SIZE(audio_map); + if (snd_soc_of_parse_card_name(card, "ti,model")) { dev_err(&pdev->dev, "Card name is not provided\n"); return -ENODEV; @@ -274,14 +252,27 @@ static int omap_abe_probe(struct platform_device *pdev) dev_err(&pdev->dev, "McPDM node is not provided\n"); return -EINVAL; } - abe_twl6040_dai_links[0].cpu_of_node = dai_node; - abe_twl6040_dai_links[0].platform_of_node = dai_node; + + priv->dai_links[0].name = "DMIC"; + priv->dai_links[0].stream_name = "TWL6040"; + priv->dai_links[0].cpu_of_node = dai_node; + priv->dai_links[0].platform_of_node = dai_node; + priv->dai_links[0].codec_dai_name = "twl6040-legacy"; + priv->dai_links[0].codec_name = "twl6040-codec"; + priv->dai_links[0].init = omap_abe_twl6040_init; + priv->dai_links[0].ops = &omap_abe_ops; dai_node = of_parse_phandle(node, "ti,dmic", 0); if (dai_node) { num_links = 2; - abe_twl6040_dai_links[1].cpu_of_node = dai_node; - abe_twl6040_dai_links[1].platform_of_node = dai_node; + priv->dai_links[1].name = "TWL6040"; + priv->dai_links[1].stream_name = "DMIC Capture"; + priv->dai_links[1].cpu_of_node = dai_node; + priv->dai_links[1].platform_of_node = dai_node; + priv->dai_links[1].codec_dai_name = "dmic-hifi"; + priv->dai_links[1].codec_name = "dmic-codec"; + priv->dai_links[1].init = omap_abe_dmic_init; + priv->dai_links[1].ops = &omap_abe_dmic_ops; } else { num_links = 1; } @@ -300,7 +291,7 @@ static int omap_abe_probe(struct platform_device *pdev) return -ENODEV; } - card->dai_link = abe_twl6040_dai_links; + card->dai_link = priv->dai_links; card->num_links = num_links; snd_soc_card_set_drvdata(card, priv); diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index fe966272bd0c..cba9645b6487 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -48,6 +48,8 @@ struct omap_dmic { struct device *dev; void __iomem *io_base; struct clk *fclk; + struct pm_qos_request pm_qos_req; + int latency; int fclk_freq; int out_freq; int clk_div; @@ -124,6 +126,8 @@ static void omap_dmic_dai_shutdown(struct snd_pcm_substream *substream, mutex_lock(&dmic->mutex); + pm_qos_remove_request(&dmic->pm_qos_req); + if (!dai->active) dmic->active = 0; @@ -228,6 +232,8 @@ static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream, /* packet size is threshold * channels */ dma_data = snd_soc_dai_get_dma_data(dai, substream); dma_data->maxburst = dmic->threshold * channels; + dmic->latency = (OMAP_DMIC_THRES_MAX - dmic->threshold) * USEC_PER_SEC / + params_rate(params); return 0; } @@ -238,6 +244,9 @@ static int omap_dmic_dai_prepare(struct snd_pcm_substream *substream, struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai); u32 ctrl; + if (pm_qos_request_active(&dmic->pm_qos_req)) + pm_qos_update_request(&dmic->pm_qos_req, dmic->latency); + /* Configure uplink threshold */ omap_dmic_write(dmic, OMAP_DMIC_FIFO_CTRL_REG, dmic->threshold); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index d0ebb6b9bfac..2d6decbfc99e 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -308,9 +308,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, pkt_size = channels; } - latency = ((((buffer_size - pkt_size) / channels) * 1000) - / (params->rate_num / params->rate_den)); - + latency = (buffer_size - pkt_size) / channels; + latency = latency * USEC_PER_SEC / + (params->rate_num / params->rate_den); mcbsp->latency[substream->stream] = latency; omap_mcbsp_set_threshold(substream, pkt_size); diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 4c1be36c2207..7d5bdc5a2890 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -54,6 +54,8 @@ struct omap_mcpdm { unsigned long phys_base; void __iomem *io_base; int irq; + struct pm_qos_request pm_qos_req; + int latency[2]; struct mutex mutex; @@ -277,6 +279,9 @@ static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); + int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE; + int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; mutex_lock(&mcpdm->mutex); @@ -289,6 +294,14 @@ static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream, } } + if (mcpdm->latency[stream2]) + pm_qos_update_request(&mcpdm->pm_qos_req, + mcpdm->latency[stream2]); + else if (mcpdm->latency[stream1]) + pm_qos_remove_request(&mcpdm->pm_qos_req); + + mcpdm->latency[stream1] = 0; + mutex_unlock(&mcpdm->mutex); } @@ -300,7 +313,7 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, int stream = substream->stream; struct snd_dmaengine_dai_dma_data *dma_data; u32 threshold; - int channels; + int channels, latency; int link_mask = 0; channels = params_channels(params); @@ -344,14 +357,25 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, dma_data->maxburst = (MCPDM_DN_THRES_MAX - threshold) * channels; + latency = threshold; } else { /* If playback is not running assume a stereo stream to come */ if (!mcpdm->config[!stream].link_mask) mcpdm->config[!stream].link_mask = (0x3 << 3); dma_data->maxburst = threshold * channels; + latency = (MCPDM_DN_THRES_MAX - threshold); } + /* + * The DMA must act to a DMA request within latency time (usec) to avoid + * under/overflow + */ + mcpdm->latency[stream] = latency * USEC_PER_SEC / params_rate(params); + + if (!mcpdm->latency[stream]) + mcpdm->latency[stream] = 10; + /* Check if we need to restart McPDM with this stream */ if (mcpdm->config[stream].link_mask && mcpdm->config[stream].link_mask != link_mask) @@ -366,6 +390,20 @@ static int omap_mcpdm_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); + struct pm_qos_request *pm_qos_req = &mcpdm->pm_qos_req; + int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE; + int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; + int latency = mcpdm->latency[stream2]; + + /* Prevent omap hardware from hitting off between FIFO fills */ + if (!latency || mcpdm->latency[stream1] < latency) + latency = mcpdm->latency[stream1]; + + if (pm_qos_request_active(pm_qos_req)) + pm_qos_update_request(pm_qos_req, latency); + else if (latency) + pm_qos_add_request(pm_qos_req, PM_QOS_CPU_DMA_LATENCY, latency); if (!omap_mcpdm_active(mcpdm)) { omap_mcpdm_start(mcpdm); @@ -427,6 +465,9 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai) free_irq(mcpdm->irq, (void *)mcpdm); pm_runtime_disable(mcpdm->dev); + if (pm_qos_request_active(&mcpdm->pm_qos_req)) + pm_qos_remove_request(&mcpdm->pm_qos_req); + return 0; } diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c index eb1b9da05dd4..4715527054e5 100644 --- a/sound/soc/qcom/common.c +++ b/sound/soc/qcom/common.c @@ -13,6 +13,7 @@ int qcom_snd_parse_of(struct snd_soc_card *card) struct device_node *cpu = NULL; struct device *dev = card->dev; struct snd_soc_dai_link *link; + struct of_phandle_args args; int ret, num_links; ret = snd_soc_of_parse_card_name(card, "model"); @@ -47,12 +48,14 @@ int qcom_snd_parse_of(struct snd_soc_card *card) goto err; } - link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0); - if (!link->cpu_of_node) { + ret = of_parse_phandle_with_args(cpu, "sound-dai", + "#sound-dai-cells", 0, &args); + if (ret) { dev_err(card->dev, "error getting cpu phandle\n"); - ret = -EINVAL; goto err; } + link->cpu_of_node = args.np; + link->id = args.args[0]; ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name); if (ret) { diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index 60ff4a2d3577..8f6c8fc073a9 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -1112,204 +1112,204 @@ static int q6afe_of_xlate_dai_name(struct snd_soc_component *component, } static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = { - SND_SOC_DAPM_AIF_OUT("HDMI_RX", "HDMI Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SLIMBUS_0_RX", "Slimbus Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SLIMBUS_1_RX", "Slimbus1 Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SLIMBUS_2_RX", "Slimbus2 Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SLIMBUS_3_RX", "Slimbus3 Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SLIMBUS_4_RX", "Slimbus4 Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SLIMBUS_5_RX", "Slimbus5 Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SLIMBUS_6_RX", "Slimbus6 Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SLIMBUS_0_TX", "Slimbus Capture", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SLIMBUS_1_TX", "Slimbus1 Capture", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SLIMBUS_2_TX", "Slimbus2 Capture", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SLIMBUS_3_TX", "Slimbus3 Capture", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SLIMBUS_4_TX", "Slimbus4 Capture", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SLIMBUS_5_TX", "Slimbus5 Capture", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SLIMBUS_6_TX", "Slimbus6 Capture", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUAT_MI2S_RX", "Quaternary MI2S Playback", + SND_SOC_DAPM_AIF_IN("HDMI_RX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_0_RX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_1_RX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_2_RX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_3_RX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_4_RX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_5_RX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_6_RX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("SLIMBUS_0_TX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("SLIMBUS_1_TX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("SLIMBUS_2_TX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("SLIMBUS_3_TX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("SLIMBUS_4_TX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("SLIMBUS_5_TX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("SLIMBUS_6_TX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("QUAT_MI2S_RX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUAT_MI2S_TX", "Quaternary MI2S Capture", + SND_SOC_DAPM_AIF_OUT("QUAT_MI2S_TX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("TERT_MI2S_RX", "Tertiary MI2S Playback", + SND_SOC_DAPM_AIF_IN("TERT_MI2S_RX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("TERT_MI2S_TX", "Tertiary MI2S Capture", + SND_SOC_DAPM_AIF_OUT("TERT_MI2S_TX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SEC_MI2S_RX", "Secondary MI2S Playback", + SND_SOC_DAPM_AIF_IN("SEC_MI2S_RX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SEC_MI2S_TX", "Secondary MI2S Capture", + SND_SOC_DAPM_AIF_OUT("SEC_MI2S_TX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SEC_MI2S_RX_SD1", + SND_SOC_DAPM_AIF_IN("SEC_MI2S_RX_SD1", "Secondary MI2S Playback SD1", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("PRI_MI2S_RX", "Primary MI2S Playback", + SND_SOC_DAPM_AIF_IN("PRI_MI2S_RX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("PRI_MI2S_TX", "Primary MI2S Capture", + SND_SOC_DAPM_AIF_OUT("PRI_MI2S_TX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_0", "Primary TDM0 Playback", + SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_0", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_1", "Primary TDM1 Playback", + SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_1", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_2", "Primary TDM2 Playback", + SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_2", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_3", "Primary TDM3 Playback", + SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_3", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_4", "Primary TDM4 Playback", + SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_4", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_5", "Primary TDM5 Playback", + SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_5", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_6", "Primary TDM6 Playback", + SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_6", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_7", "Primary TDM7 Playback", + SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_7", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_0", "Primary TDM0 Capture", + SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_0", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_1", "Primary TDM1 Capture", + SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_1", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_2", "Primary TDM2 Capture", + SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_2", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_3", "Primary TDM3 Capture", + SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_3", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_4", "Primary TDM4 Capture", + SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_4", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_5", "Primary TDM5 Capture", + SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_5", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_6", "Primary TDM6 Capture", + SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_6", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_7", "Primary TDM7 Capture", + SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_7", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_0", "Secondary TDM0 Playback", + SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_0", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_1", "Secondary TDM1 Playback", + SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_1", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_2", "Secondary TDM2 Playback", + SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_2", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_3", "Secondary TDM3 Playback", + SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_3", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_4", "Secondary TDM4 Playback", + SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_4", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_5", "Secondary TDM5 Playback", + SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_5", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_6", "Secondary TDM6 Playback", + SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_6", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_7", "Secondary TDM7 Playback", + SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_7", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_0", "Secondary TDM0 Capture", + SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_0", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_1", "Secondary TDM1 Capture", + SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_1", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_2", "Secondary TDM2 Capture", + SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_2", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_3", "Secondary TDM3 Capture", + SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_3", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_4", "Secondary TDM4 Capture", + SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_4", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_5", "Secondary TDM5 Capture", + SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_5", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_6", "Secondary TDM6 Capture", + SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_6", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_7", "Secondary TDM7 Capture", + SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_7", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_0", "Tertiary TDM0 Playback", + SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_0", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_1", "Tertiary TDM1 Playback", + SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_1", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_2", "Tertiary TDM2 Playback", + SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_2", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_3", "Tertiary TDM3 Playback", + SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_3", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_4", "Tertiary TDM4 Playback", + SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_4", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_5", "Tertiary TDM5 Playback", + SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_5", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_6", "Tertiary TDM6 Playback", + SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_6", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_7", "Tertiary TDM7 Playback", + SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_7", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_0", "Tertiary TDM0 Capture", + SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_0", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_1", "Tertiary TDM1 Capture", + SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_1", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_2", "Tertiary TDM2 Capture", + SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_2", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_3", "Tertiary TDM3 Capture", + SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_3", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_4", "Tertiary TDM4 Capture", + SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_4", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_5", "Tertiary TDM5 Capture", + SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_5", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_6", "Tertiary TDM6 Capture", + SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_6", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_7", "Tertiary TDM7 Capture", + SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_7", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_0", "Quaternary TDM0 Playback", + SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_0", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_1", "Quaternary TDM1 Playback", + SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_1", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_2", "Quaternary TDM2 Playback", + SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_2", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_3", "Quaternary TDM3 Playback", + SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_3", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_4", "Quaternary TDM4 Playback", + SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_4", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_5", "Quaternary TDM5 Playback", + SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_5", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_6", "Quaternary TDM6 Playback", + SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_6", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_7", "Quaternary TDM7 Playback", + SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_7", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_0", "Quaternary TDM0 Capture", + SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_0", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_1", "Quaternary TDM1 Capture", + SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_1", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_2", "Quaternary TDM2 Capture", + SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_2", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_3", "Quaternary TDM3 Capture", + SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_3", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_4", "Quaternary TDM4 Capture", + SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_4", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_5", "Quaternary TDM5 Capture", + SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_5", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_6", "Quaternary TDM6 Capture", + SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_6", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_7", "Quaternary TDM7 Capture", + SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_7", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_0", "Quinary TDM0 Playback", + SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_0", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_1", "Quinary TDM1 Playback", + SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_1", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_2", "Quinary TDM2 Playback", + SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_2", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_3", "Quinary TDM3 Playback", + SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_3", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_4", "Quinary TDM4 Playback", + SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_4", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_5", "Quinary TDM5 Playback", + SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_5", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_6", "Quinary TDM6 Playback", + SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_6", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_7", "Quinary TDM7 Playback", + SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_7", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_0", "Quinary TDM0 Capture", + SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_0", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_1", "Quinary TDM1 Capture", + SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_1", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_2", "Quinary TDM2 Capture", + SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_2", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_3", "Quinary TDM3 Capture", + SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_3", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_4", "Quinary TDM4 Capture", + SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_4", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_5", "Quinary TDM5 Capture", + SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_5", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_6", "Quinary TDM6 Capture", + SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_6", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_7", "Quinary TDM7 Capture", + SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_7", NULL, 0, 0, 0, 0), }; diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index 000775b4bba8..829b5e987b2a 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -49,14 +49,14 @@ #define AFE_PORT_I2S_SD1 0x2 #define AFE_PORT_I2S_SD2 0x3 #define AFE_PORT_I2S_SD3 0x4 -#define AFE_PORT_I2S_SD0_MASK BIT(0x1) -#define AFE_PORT_I2S_SD1_MASK BIT(0x2) -#define AFE_PORT_I2S_SD2_MASK BIT(0x3) -#define AFE_PORT_I2S_SD3_MASK BIT(0x4) -#define AFE_PORT_I2S_SD0_1_MASK GENMASK(2, 1) -#define AFE_PORT_I2S_SD2_3_MASK GENMASK(4, 3) -#define AFE_PORT_I2S_SD0_1_2_MASK GENMASK(3, 1) -#define AFE_PORT_I2S_SD0_1_2_3_MASK GENMASK(4, 1) +#define AFE_PORT_I2S_SD0_MASK BIT(0x0) +#define AFE_PORT_I2S_SD1_MASK BIT(0x1) +#define AFE_PORT_I2S_SD2_MASK BIT(0x2) +#define AFE_PORT_I2S_SD3_MASK BIT(0x3) +#define AFE_PORT_I2S_SD0_1_MASK GENMASK(1, 0) +#define AFE_PORT_I2S_SD2_3_MASK GENMASK(3, 2) +#define AFE_PORT_I2S_SD0_1_2_MASK GENMASK(2, 0) +#define AFE_PORT_I2S_SD0_1_2_3_MASK GENMASK(3, 0) #define AFE_PORT_I2S_QUAD01 0x5 #define AFE_PORT_I2S_QUAD23 0x6 #define AFE_PORT_I2S_6CHS 0x7 diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index a16c71c03058..86115de5c1b2 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -122,7 +122,6 @@ static struct snd_pcm_hardware q6asm_dai_hardware_playback = { .rate_max = 48000, \ }, \ .name = "MultiMedia"#num, \ - .probe = fe_dai_probe, \ .id = MSM_FRONTEND_DAI_MULTIMEDIA##num, \ } @@ -511,38 +510,6 @@ static void q6asm_dai_pcm_free(struct snd_pcm *pcm) } } -static const struct snd_soc_dapm_route afe_pcm_routes[] = { - {"MM_DL1", NULL, "MultiMedia1 Playback" }, - {"MM_DL2", NULL, "MultiMedia2 Playback" }, - {"MM_DL3", NULL, "MultiMedia3 Playback" }, - {"MM_DL4", NULL, "MultiMedia4 Playback" }, - {"MM_DL5", NULL, "MultiMedia5 Playback" }, - {"MM_DL6", NULL, "MultiMedia6 Playback" }, - {"MM_DL7", NULL, "MultiMedia7 Playback" }, - {"MM_DL7", NULL, "MultiMedia8 Playback" }, - {"MultiMedia1 Capture", NULL, "MM_UL1"}, - {"MultiMedia2 Capture", NULL, "MM_UL2"}, - {"MultiMedia3 Capture", NULL, "MM_UL3"}, - {"MultiMedia4 Capture", NULL, "MM_UL4"}, - {"MultiMedia5 Capture", NULL, "MM_UL5"}, - {"MultiMedia6 Capture", NULL, "MM_UL6"}, - {"MultiMedia7 Capture", NULL, "MM_UL7"}, - {"MultiMedia8 Capture", NULL, "MM_UL8"}, - -}; - -static int fe_dai_probe(struct snd_soc_dai *dai) -{ - struct snd_soc_dapm_context *dapm; - - dapm = snd_soc_component_get_dapm(dai->component); - snd_soc_dapm_add_routes(dapm, afe_pcm_routes, - ARRAY_SIZE(afe_pcm_routes)); - - return 0; -} - - static const struct snd_soc_component_driver q6asm_fe_dai_component = { .name = DRV_NAME, .ops = &q6asm_dai_ops, diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index c6b51571be94..d61b8404f7da 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -909,6 +909,25 @@ static const struct snd_soc_dapm_route intercon[] = { {"MM_UL6", NULL, "MultiMedia6 Mixer"}, {"MM_UL7", NULL, "MultiMedia7 Mixer"}, {"MM_UL8", NULL, "MultiMedia8 Mixer"}, + + {"MM_DL1", NULL, "MultiMedia1 Playback" }, + {"MM_DL2", NULL, "MultiMedia2 Playback" }, + {"MM_DL3", NULL, "MultiMedia3 Playback" }, + {"MM_DL4", NULL, "MultiMedia4 Playback" }, + {"MM_DL5", NULL, "MultiMedia5 Playback" }, + {"MM_DL6", NULL, "MultiMedia6 Playback" }, + {"MM_DL7", NULL, "MultiMedia7 Playback" }, + {"MM_DL8", NULL, "MultiMedia8 Playback" }, + + {"MultiMedia1 Capture", NULL, "MM_UL1"}, + {"MultiMedia2 Capture", NULL, "MM_UL2"}, + {"MultiMedia3 Capture", NULL, "MM_UL3"}, + {"MultiMedia4 Capture", NULL, "MM_UL4"}, + {"MultiMedia5 Capture", NULL, "MM_UL5"}, + {"MultiMedia6 Capture", NULL, "MM_UL6"}, + {"MultiMedia7 Capture", NULL, "MM_UL7"}, + {"MultiMedia8 Capture", NULL, "MM_UL8"}, + }; static int routing_hw_params(struct snd_pcm_substream *substream, diff --git a/sound/soc/rockchip/rockchip_pcm.c b/sound/soc/rockchip/rockchip_pcm.c index 9e7b5fa4cf59..4ac78d7a4b2d 100644 --- a/sound/soc/rockchip/rockchip_pcm.c +++ b/sound/soc/rockchip/rockchip_pcm.c @@ -33,6 +33,7 @@ static const struct snd_pcm_hardware snd_rockchip_hardware = { static const struct snd_dmaengine_pcm_config rk_dmaengine_pcm_config = { .pcm_hardware = &snd_rockchip_hardware, + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, .prealloc_buffer_size = 32 * 1024, }; diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index fcb4df23248c..6ec78f3096dd 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -306,7 +306,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, if (rsnd_ssi_is_multi_slave(mod, io)) return 0; - if (ssi->rate) { + if (ssi->usrcnt > 1) { if (ssi->rate != rate) { dev_err(dev, "SSI parent/child should use same rate\n"); return -EINVAL; diff --git a/sound/soc/soc-acpi.c b/sound/soc/soc-acpi.c index b8e72b52db30..4fb29f0e561e 100644 --- a/sound/soc/soc-acpi.c +++ b/sound/soc/soc-acpi.c @@ -10,11 +10,17 @@ struct snd_soc_acpi_mach * snd_soc_acpi_find_machine(struct snd_soc_acpi_mach *machines) { struct snd_soc_acpi_mach *mach; + struct snd_soc_acpi_mach *mach_alt; for (mach = machines; mach->id[0]; mach++) { if (acpi_dev_present(mach->id, NULL, -1)) { - if (mach->machine_quirk) - mach = mach->machine_quirk(mach); + if (mach->machine_quirk) { + mach_alt = mach->machine_quirk(mach); + if (!mach_alt) + continue; /* not full match, ignore */ + mach = mach_alt; + } + return mach; } } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6ddcf12bc030..b29d0f65611e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2131,6 +2131,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) } card->instantiated = 1; + dapm_mark_endpoints_dirty(card); snd_soc_dapm_sync(&card->dapm); mutex_unlock(&card->mutex); mutex_unlock(&client_mutex); diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index ea05cc91aa05..211589b0b2ef 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -390,7 +390,7 @@ static int stm32_sai_add_mclk_provider(struct stm32_sai_sub_data *sai) char *mclk_name, *p, *s = (char *)pname; int ret, i = 0; - mclk = devm_kzalloc(dev, sizeof(mclk), GFP_KERNEL); + mclk = devm_kzalloc(dev, sizeof(*mclk), GFP_KERNEL); if (!mclk) return -ENOMEM; diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index 66aad0d3f9c7..8134c3c94229 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -31,7 +31,7 @@ config SND_SUN8I_CODEC_ANALOG config SND_SUN50I_CODEC_ANALOG tristate "Allwinner sun50i Codec Analog Controls Support" depends on (ARM64 && ARCH_SUNXI) || COMPILE_TEST - select SND_SUNXI_ADDA_PR_REGMAP + select SND_SUN8I_ADDA_PR_REGMAP help Say Y or M if you want to add support for the analog controls for the codec embedded in Allwinner A64 SoC. diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index 522a72fde78d..92c5de026c43 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -481,7 +481,11 @@ static const struct snd_soc_dapm_route sun8i_codec_dapm_routes[] = { { "Right Digital DAC Mixer", "AIF1 Slot 0 Digital DAC Playback Switch", "AIF1 Slot 0 Right"}, - /* ADC routes */ + /* ADC Routes */ + { "AIF1 Slot 0 Right ADC", NULL, "ADC" }, + { "AIF1 Slot 0 Left ADC", NULL, "ADC" }, + + /* ADC Mixer Routes */ { "Left Digital ADC Mixer", "AIF1 Data Digital ADC Capture Switch", "AIF1 Slot 0 Left ADC" }, { "Right Digital ADC Mixer", "AIF1 Data Digital ADC Capture Switch", @@ -605,16 +609,10 @@ err_pm_disable: static int sun8i_codec_remove(struct platform_device *pdev) { - struct snd_soc_card *card = platform_get_drvdata(pdev); - struct sun8i_codec *scodec = snd_soc_card_get_drvdata(card); - pm_runtime_disable(&pdev->dev); if (!pm_runtime_status_suspended(&pdev->dev)) sun8i_codec_runtime_suspend(&pdev->dev); - clk_disable_unprepare(scodec->clk_module); - clk_disable_unprepare(scodec->clk_bus); - return 0; } diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index e73c962590eb..079063d8038d 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -1146,10 +1146,8 @@ static int snd_cs4231_playback_open(struct snd_pcm_substream *substream) runtime->hw = snd_cs4231_playback; err = snd_cs4231_open(chip, CS4231_MODE_PLAY); - if (err < 0) { - snd_free_pages(runtime->dma_area, runtime->dma_bytes); + if (err < 0) return err; - } chip->playback_substream = substream; chip->p_periods_sent = 0; snd_pcm_set_sync(substream); @@ -1167,10 +1165,8 @@ static int snd_cs4231_capture_open(struct snd_pcm_substream *substream) runtime->hw = snd_cs4231_capture; err = snd_cs4231_open(chip, CS4231_MODE_RECORD); - if (err < 0) { - snd_free_pages(runtime->dma_area, runtime->dma_bytes); + if (err < 0) return err; - } chip->capture_substream = substream; chip->c_periods_sent = 0; snd_pcm_set_sync(substream); diff --git a/sound/usb/card.c b/sound/usb/card.c index 2bfe4e80a6b9..a105947eaf55 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -682,9 +682,12 @@ static int usb_audio_probe(struct usb_interface *intf, __error: if (chip) { + /* chip->active is inside the chip->card object, + * decrement before memory is possibly returned. + */ + atomic_dec(&chip->active); if (!chip->num_interfaces) snd_card_free(chip->card); - atomic_dec(&chip->active); } mutex_unlock(®ister_mutex); return err; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 849953e5775c..37fc0447c071 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3382,5 +3382,15 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .ifnum = QUIRK_NO_INTERFACE } }, +/* Dell WD19 Dock */ +{ + USB_DEVICE(0x0bda, 0x402e), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Dell", + .product_name = "WD19 Dock", + .profile_name = "Dell-WD15-Dock", + .ifnum = QUIRK_NO_INTERFACE + } +}, #undef USB_DEVICE_VENDOR_SPEC diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 8a945ece9869..6623cafc94f2 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1373,6 +1373,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; + case USB_ID(0x152a, 0x85de): /* SMSL D1 DAC */ case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */ case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */ |