diff options
| author | Thierry Reding <treding@nvidia.com> | 2026-04-30 09:31:28 +0200 |
|---|---|---|
| committer | Thierry Reding <treding@nvidia.com> | 2026-04-30 09:31:28 +0200 |
| commit | d87dbdefa6d2160a29b734cfb69de75ddcb46c25 (patch) | |
| tree | c8f5915ec14dd4dfeae7985e2bc90db5b551967c | |
| parent | 0cb0f612490e40d60cbdb3c05ba462ca292e86f7 (diff) | |
| parent | e052a1f7199260eda4d6ca08a59c3b98738f8491 (diff) | |
| download | linux-next-d87dbdefa6d2160a29b734cfb69de75ddcb46c25.tar.gz | |
Merge branch 'for-linus' of https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git
32 files changed, 266 insertions, 122 deletions
diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index a140a0d9abb80..33fd34f0d615d 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2155,10 +2155,16 @@ static int snd_pcm_oss_get_trigger(struct snd_pcm_oss_file *pcm_oss_file) psubstream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK]; csubstream = pcm_oss_file->streams[SNDRV_PCM_STREAM_CAPTURE]; - if (psubstream && psubstream->runtime && psubstream->runtime->oss.trigger) - result |= PCM_ENABLE_OUTPUT; - if (csubstream && csubstream->runtime && csubstream->runtime->oss.trigger) - result |= PCM_ENABLE_INPUT; + if (psubstream && psubstream->runtime) { + guard(mutex)(&psubstream->runtime->oss.params_lock); + if (psubstream->runtime->oss.trigger) + result |= PCM_ENABLE_OUTPUT; + } + if (csubstream && csubstream->runtime) { + guard(mutex)(&csubstream->runtime->oss.params_lock); + if (csubstream->runtime->oss.trigger) + result |= PCM_ENABLE_INPUT; + } return result; } @@ -2832,6 +2838,17 @@ static int snd_pcm_oss_capture_ready(struct snd_pcm_substream *substream) runtime->oss.period_frames; } +static bool need_input_retrigger(struct snd_pcm_runtime *runtime) +{ + bool ret; + + guard(mutex)(&runtime->oss.params_lock); + ret = runtime->oss.trigger; + if (ret) + runtime->oss.trigger = 0; + return ret; +} + static __poll_t snd_pcm_oss_poll(struct file *file, poll_table * wait) { struct snd_pcm_oss_file *pcm_oss_file; @@ -2864,11 +2881,11 @@ static __poll_t snd_pcm_oss_poll(struct file *file, poll_table * wait) snd_pcm_oss_capture_ready(csubstream)) mask |= EPOLLIN | EPOLLRDNORM; } - if (ostate != SNDRV_PCM_STATE_RUNNING && runtime->oss.trigger) { + if (ostate != SNDRV_PCM_STATE_RUNNING && + need_input_retrigger(runtime)) { struct snd_pcm_oss_file ofile; memset(&ofile, 0, sizeof(ofile)); ofile.streams[SNDRV_PCM_STREAM_CAPTURE] = pcm_oss_file->streams[SNDRV_PCM_STREAM_CAPTURE]; - runtime->oss.trigger = 0; snd_pcm_oss_set_trigger(&ofile, PCM_ENABLE_INPUT); } } diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index aa0d2fcb1a180..a37a1695f51c7 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -99,6 +99,9 @@ struct loopback_ops { struct loopback_cable { spinlock_t lock; struct loopback_pcm *streams[2]; + /* in-flight peer stops running outside cable->lock */ + atomic_t stop_count; + wait_queue_head_t stop_wait; struct snd_pcm_hardware hw; /* flags */ unsigned int valid; @@ -366,8 +369,11 @@ static int loopback_check_format(struct loopback_cable *cable, int stream) return 0; if (stream == SNDRV_PCM_STREAM_CAPTURE) return -EIO; - else if (cruntime->state == SNDRV_PCM_STATE_RUNNING) + else if (cruntime->state == SNDRV_PCM_STATE_RUNNING) { + /* close must not free the peer runtime below */ + atomic_inc(&cable->stop_count); stop_capture = true; + } } setup = get_setup(dpcm_play); @@ -396,8 +402,11 @@ static int loopback_check_format(struct loopback_cable *cable, int stream) } } - if (stop_capture) + if (stop_capture) { snd_pcm_stop(dpcm_capt->substream, SNDRV_PCM_STATE_DRAINING); + if (atomic_dec_and_test(&cable->stop_count)) + wake_up(&cable->stop_wait); + } return 0; } @@ -1049,23 +1058,29 @@ static void free_cable(struct snd_pcm_substream *substream) struct loopback *loopback = substream->private_data; int dev = get_cable_index(substream); struct loopback_cable *cable; + struct loopback_pcm *dpcm; + bool other_alive; cable = loopback->cables[substream->number][dev]; if (!cable) return; - if (cable->streams[!substream->stream]) { - /* other stream is still alive */ - guard(spinlock_irq)(&cable->lock); - cable->streams[substream->stream] = NULL; - } else { - struct loopback_pcm *dpcm = substream->runtime->private_data; - if (cable->ops && cable->ops->close_cable && dpcm) - cable->ops->close_cable(dpcm); - /* free the cable */ - loopback->cables[substream->number][dev] = NULL; - kfree(cable); + scoped_guard(spinlock_irq, &cable->lock) { + cable->streams[substream->stream] = NULL; + other_alive = cable->streams[!substream->stream]; } + + /* Pair with the stop_count increment in loopback_check_format(). */ + wait_event(cable->stop_wait, !atomic_read(&cable->stop_count)); + if (other_alive) + return; + + dpcm = substream->runtime->private_data; + if (cable->ops && cable->ops->close_cable && dpcm) + cable->ops->close_cable(dpcm); + /* free the cable */ + loopback->cables[substream->number][dev] = NULL; + kfree(cable); } static int loopback_jiffies_timer_open(struct loopback_pcm *dpcm) @@ -1260,6 +1275,8 @@ static int loopback_open(struct snd_pcm_substream *substream) goto unlock; } spin_lock_init(&cable->lock); + atomic_set(&cable->stop_count, 0); + init_waitqueue_head(&cable->stop_wait); cable->hw = loopback_pcm_hardware; if (loopback->timer_source) cable->ops = &loopback_snd_timer_ops; diff --git a/sound/hda/codecs/Makefile b/sound/hda/codecs/Makefile index e7f03e281999f..88d2f8a79467b 100644 --- a/sound/hda/codecs/Makefile +++ b/sound/hda/codecs/Makefile @@ -7,7 +7,6 @@ snd-hda-codec-cm9825-y := cm9825.o snd-hda-codec-analog-y := analog.o snd-hda-codec-ca0110-y := ca0110.o snd-hda-codec-ca0132-y := ca0132.o -snd-hda-codec-cmedia-y := cmedia.o snd-hda-codec-conexant-y := conexant.o snd-hda-codec-idt-y := sigmatel.o snd-hda-codec-senarytech-y := senarytech.o @@ -26,7 +25,6 @@ obj-$(CONFIG_SND_HDA_CODEC_CM9825) += snd-hda-codec-cm9825.o obj-$(CONFIG_SND_HDA_CODEC_ANALOG) += snd-hda-codec-analog.o obj-$(CONFIG_SND_HDA_CODEC_CA0110) += snd-hda-codec-ca0110.o obj-$(CONFIG_SND_HDA_CODEC_CA0132) += snd-hda-codec-ca0132.o -obj-$(CONFIG_SND_HDA_CODEC_CMEDIA) += snd-hda-codec-cmedia.o obj-$(CONFIG_SND_HDA_CODEC_CONEXANT) += snd-hda-codec-conexant.o obj-$(CONFIG_SND_HDA_CODEC_SIGMATEL) += snd-hda-codec-idt.o obj-$(CONFIG_SND_HDA_CODEC_SENARYTECH) += snd-hda-codec-senarytech.o diff --git a/sound/hda/codecs/conexant.c b/sound/hda/codecs/conexant.c index 3a9717df39b4b..e3b6aaabe3a9c 100644 --- a/sound/hda/codecs/conexant.c +++ b/sound/hda/codecs/conexant.c @@ -1175,6 +1175,7 @@ static void add_cx5051_fake_mutes(struct hda_codec *codec) static int cx_probe(struct hda_codec *codec, const struct hda_device_id *id) { struct conexant_spec *spec; + struct hda_jack_callback *callback; int err; codec_info(codec, "%s: BIOS auto-probing.\n", codec->core.chip_name); @@ -1190,7 +1191,12 @@ static int cx_probe(struct hda_codec *codec, const struct hda_device_id *id) case 0x14f11f86: case 0x14f11f87: spec->is_cx11880_sn6140 = true; - snd_hda_jack_detect_enable_callback(codec, 0x19, cx_update_headset_mic_vref); + callback = snd_hda_jack_detect_enable_callback(codec, 0x19, + cx_update_headset_mic_vref); + if (IS_ERR(callback)) { + err = PTR_ERR(callback); + goto error; + } break; } diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index d720565db4aaf..a9cd03bb73c41 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -3694,6 +3694,17 @@ static void alc287_fixup_lenovo_thinkpad_with_alc1318(struct hda_codec *codec, spec->power_hook = alc287_s4_power_gpio3_default; spec->gen.pcm_playback_hook = alc287_alc1318_playback_pcm_hook; } + +static void alc287_fixup_tb_vmaster_led(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) + spec->gen.vmaster_mute_led = 1; + + alc287_fixup_bind_dacs(codec, fix, action); +} /* GPIO2: mute led GPIO3: micmute led */ static void alc245_tas2781_spi_hp_fixup_muteled(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -6448,7 +6459,7 @@ static const struct hda_fixup alc269_fixups[] = { }, [ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD] = { .type = HDA_FIXUP_FUNC, - .v.func = alc287_fixup_bind_dacs, + .v.func = alc287_fixup_tb_vmaster_led, .chained = true, .chain_id = ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI, }, @@ -6717,6 +6728,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x159c, "Acer Nitro 5 AN515-58", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1597, "Acer Nitro 5 AN517-55", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x160e, "Acer PT316-51S", ALC2XX_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x1640, "Acer Aspire A315-44P", ALC256_FIXUP_ACER_SFG16_MICMUTE_LED), SND_PCI_QUIRK(0x1025, 0x1679, "Acer Nitro 16 AN16-41", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x169a, "Acer Swift SFG16", ALC256_FIXUP_ACER_SFG16_MICMUTE_LED), SND_PCI_QUIRK(0x1025, 0x171e, "Acer Nitro ANV15-51", ALC245_FIXUP_ACER_MICMUTE_LED), diff --git a/sound/hda/codecs/side-codecs/cs35l56_hda.c b/sound/hda/codecs/side-codecs/cs35l56_hda.c index 1ace4beef5085..4c8d01799931c 100644 --- a/sound/hda/codecs/side-codecs/cs35l56_hda.c +++ b/sound/hda/codecs/side-codecs/cs35l56_hda.c @@ -180,11 +180,15 @@ static int cs35l56_hda_mixer_get(struct snd_kcontrol *kcontrol, { struct cs35l56_hda *cs35l56 = snd_kcontrol_chip(kcontrol); unsigned int reg_val; - int i; + int i, ret; cs35l56_hda_wait_dsp_ready(cs35l56); - regmap_read(cs35l56->base.regmap, kcontrol->private_value, ®_val); + ret = regmap_read(cs35l56->base.regmap, kcontrol->private_value, + ®_val); + if (ret) + return ret; + reg_val &= CS35L56_ASP_TXn_SRC_MASK; for (i = 0; i < CS35L56_NUM_INPUT_SRC; ++i) { @@ -203,15 +207,20 @@ static int cs35l56_hda_mixer_put(struct snd_kcontrol *kcontrol, struct cs35l56_hda *cs35l56 = snd_kcontrol_chip(kcontrol); unsigned int item = ucontrol->value.enumerated.item[0]; bool changed; + int ret; if (item >= CS35L56_NUM_INPUT_SRC) return -EINVAL; cs35l56_hda_wait_dsp_ready(cs35l56); - regmap_update_bits_check(cs35l56->base.regmap, kcontrol->private_value, - CS35L56_INPUT_MASK, cs35l56_tx_input_values[item], - &changed); + ret = regmap_update_bits_check(cs35l56->base.regmap, + kcontrol->private_value, + CS35L56_INPUT_MASK, + cs35l56_tx_input_values[item], + &changed); + if (ret) + return ret; return changed; } @@ -967,6 +976,7 @@ static int cs35l56_hda_system_resume(struct device *dev) static int cs35l56_hda_fixup_yoga9(struct cs35l56_hda *cs35l56, int *bus_addr) { /* The cirrus,dev-index property has the wrong values */ + cs35l56->num_amps = 2; switch (*bus_addr) { case 0x30: cs35l56->index = 1; @@ -1016,7 +1026,6 @@ static int cs35l56_hda_read_acpi(struct cs35l56_hda *cs35l56, int hid, int id) char hid_string[8]; struct acpi_device *adev; const char *property, *sub; - size_t nval; int i, ret; /* @@ -1052,13 +1061,14 @@ static int cs35l56_hda_read_acpi(struct cs35l56_hda *cs35l56, int hid, int id) ret = -EINVAL; goto err; } - nval = ret; + cs35l56->num_amps = ret; - ret = device_property_read_u32_array(cs35l56->base.dev, property, values, nval); + ret = device_property_read_u32_array(cs35l56->base.dev, property, values, + cs35l56->num_amps); if (ret) goto err; - for (i = 0; i < nval; i++) { + for (i = 0; i < cs35l56->num_amps; i++) { if (values[i] == id) { cs35l56->index = i; break; @@ -1081,7 +1091,8 @@ static int cs35l56_hda_read_acpi(struct cs35l56_hda *cs35l56, int hid, int id) "Read ACPI _SUB failed(%ld): fallback to generic firmware\n", PTR_ERR(sub)); } else { - ret = cirrus_scodec_get_speaker_id(cs35l56->base.dev, cs35l56->index, nval, -1); + ret = cirrus_scodec_get_speaker_id(cs35l56->base.dev, cs35l56->index, + cs35l56->num_amps, -1); if (ret == -ENOENT) { cs35l56->system_name = sub; } else if (ret >= 0) { diff --git a/sound/hda/codecs/side-codecs/cs35l56_hda.h b/sound/hda/codecs/side-codecs/cs35l56_hda.h index cb4b5e7356a35..3705af7c186b3 100644 --- a/sound/hda/codecs/side-codecs/cs35l56_hda.h +++ b/sound/hda/codecs/side-codecs/cs35l56_hda.h @@ -26,6 +26,7 @@ struct cs35l56_hda { struct work_struct dsp_work; int index; + int num_amps; const char *system_name; const char *amp_name; diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_spi.c b/sound/hda/codecs/side-codecs/tas2781_hda_spi.c index 560f2385212dd..0e4f3553f2738 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_spi.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_spi.c @@ -132,10 +132,18 @@ static int tasdevice_spi_dev_update_bits(struct tasdevice_priv *tas_priv, int ret, val; /* - * In our TAS2781 SPI mode, read/write was masked in last bit of - * address, it cause regmap_update_bits() not work as expected. + * In TAS2781 SPI mode, when accessing non-book-zero or page numbers + * greater than 1 in book 0, an additional byte must be read. The + * first byte in such cases is a dummy byte and should be ignored. */ - ret = tasdevice_dev_read(tas_priv, chn, reg, &val); + if ((TASDEVICE_BOOK_ID(reg) > 0) || (TASDEVICE_PAGE_ID(reg) > 1)) { + unsigned char buf[2]; + + ret = tasdevice_dev_bulk_read(tas_priv, chn, reg, buf, 2); + val = buf[1]; + } else { + ret = tasdevice_dev_read(tas_priv, chn, reg, &val); + } if (ret < 0) { dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); return ret; diff --git a/sound/hda/core/hdmi_chmap.c b/sound/hda/core/hdmi_chmap.c index 7b276047f85a7..c897fc443467c 100644 --- a/sound/hda/core/hdmi_chmap.c +++ b/sound/hda/core/hdmi_chmap.c @@ -353,13 +353,16 @@ static void hdmi_std_setup_channel_mapping(struct hdac_chmap *chmap, if (hdmi_channel_mapping[ca][1] == 0) { int hdmi_slot = 0; /* fill actual channel mappings in ALSA channel (i) order */ - for (i = 0; i < ch_alloc->channels; i++) { - while (!WARN_ON(hdmi_slot >= 8) && - !ch_alloc->speakers[7 - hdmi_slot]) - hdmi_slot++; /* skip zero slots */ + for (i = 0; i < ch_alloc->channels && hdmi_slot < 8; i++) { + while (!ch_alloc->speakers[7 - hdmi_slot]) { + /* skip zero slots */ + if (++hdmi_slot >= 8) + goto out; + } hdmi_channel_mapping[ca][i] = (i << 4) | hdmi_slot++; } + out: /* fill the rest of the slots with ALSA channel 0xf */ for (hdmi_slot = 0; hdmi_slot < 8; hdmi_slot++) if (!ch_alloc->speakers[7 - hdmi_slot]) diff --git a/sound/soc/amd/acp/acp-legacy-mach.c b/sound/soc/amd/acp/acp-legacy-mach.c index a7a551366a409..235d6cc83fa98 100644 --- a/sound/soc/amd/acp/acp-legacy-mach.c +++ b/sound/soc/amd/acp/acp-legacy-mach.c @@ -174,7 +174,7 @@ static int acp_asoc_probe(struct platform_device *pdev) acp_card_drvdata->acp_rev = mach->mach_params.subsystem_rev; dmi_id = dmi_first_match(acp_quirk_table); - if (dmi_id && dmi_id->driver_data) + if (dmi_id && dmi_id->driver_data == (void *)QUIRK_TDM_MODE_ENABLE) acp_card_drvdata->tdm_mode = dmi_id->driver_data; ret = acp_legacy_dai_links_create(card); diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index 09f6c9a2c0410..ef784cca13f2b 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -20,6 +20,7 @@ #include <sound/soc.h> #include <linux/input.h> #include <linux/module.h> +#include <linux/dmi.h> #include "../../codecs/rt5682.h" #include "../../codecs/rt1019.h" @@ -37,15 +38,21 @@ #define NAU8821_FREQ_OUT 12288000 #define MAX98388_CODEC_DAI "max98388-aif1" -#define TDM_MODE_ENABLE 1 - const struct dmi_system_id acp_quirk_table[] = { { /* Google skyrim proto-0 */ .matches = { DMI_EXACT_MATCH(DMI_PRODUCT_FAMILY, "Google_Skyrim"), }, - .driver_data = (void *)TDM_MODE_ENABLE, + .driver_data = (void *)QUIRK_TDM_MODE_ENABLE, + }, + { + /* Valve Steam Deck OLED */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Valve"), + DMI_MATCH(DMI_PRODUCT_NAME, "Galileo"), + }, + .driver_data = (void *)QUIRK_REMAP_DMIC_BT, }, {} }; @@ -1401,6 +1408,7 @@ int acp_sofdsp_dai_links_create(struct snd_soc_card *card) struct snd_soc_dai_link *links; struct device *dev = card->dev; struct acp_card_drvdata *drv_data = card->drvdata; + const struct dmi_system_id *dmi_id = dmi_first_match(acp_quirk_table); int i = 0, num_links = 0; if (drv_data->hs_cpu_id) @@ -1572,6 +1580,9 @@ int acp_sofdsp_dai_links_create(struct snd_soc_card *card) links[i].codecs = &snd_soc_dummy_dlc; links[i].num_codecs = 1; } + + if (dmi_id && dmi_id->driver_data == (void *)QUIRK_REMAP_DMIC_BT) + links[i].id = DMIC_BE_ID; i++; } @@ -1587,6 +1598,11 @@ int acp_sofdsp_dai_links_create(struct snd_soc_card *card) links[i].capture_only = 1; links[i].nonatomic = true; links[i].no_pcm = 1; + + if (dmi_id && dmi_id->driver_data == (void *)QUIRK_REMAP_DMIC_BT) { + links[i].id = BT_BE_ID; + dev_dbg(dev, "quirk REMAP_DMIC_BT enabled\n"); + } } card->dai_link = links; diff --git a/sound/soc/amd/acp/acp-mach.h b/sound/soc/amd/acp/acp-mach.h index f94c30c20f20b..7177d3fd96192 100644 --- a/sound/soc/amd/acp/acp-mach.h +++ b/sound/soc/amd/acp/acp-mach.h @@ -26,6 +26,10 @@ #define acp_get_drvdata(card) ((struct acp_card_drvdata *)(card)->drvdata) +/* List of DMI quirks - check acp-mach-common.c for usage. */ +#define QUIRK_TDM_MODE_ENABLE 1 +#define QUIRK_REMAP_DMIC_BT 2 + enum be_id { HEADSET_BE_ID = 0, AMP_BE_ID, diff --git a/sound/soc/amd/acp/acp-sof-mach.c b/sound/soc/amd/acp/acp-sof-mach.c index 6215e31eceddf..36ecef7013b9c 100644 --- a/sound/soc/amd/acp/acp-sof-mach.c +++ b/sound/soc/amd/acp/acp-sof-mach.c @@ -110,7 +110,7 @@ static int acp_sof_probe(struct platform_device *pdev) acp_card_drvdata = card->drvdata; dmi_id = dmi_first_match(acp_quirk_table); - if (dmi_id && dmi_id->driver_data) + if (dmi_id && dmi_id->driver_data == (void *)QUIRK_TDM_MODE_ENABLE) acp_card_drvdata->tdm_mode = dmi_id->driver_data; acp_card_drvdata->acp_rev = mach->mach_params.subsystem_rev; diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index fdda1b747bf7e..8ab2e60f80b4f 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2496,13 +2496,13 @@ static int ab8500_codec_probe(struct snd_soc_component *component) return status; } fc = (struct filter_control *) - &ab8500_filter_controls[AB8500_FILTER_ANC_FIR].private_value; + ab8500_filter_controls[AB8500_FILTER_ANC_FIR].private_value; drvdata->anc_fir_values = (long *)fc->value; fc = (struct filter_control *) - &ab8500_filter_controls[AB8500_FILTER_ANC_IIR].private_value; + ab8500_filter_controls[AB8500_FILTER_ANC_IIR].private_value; drvdata->anc_iir_values = (long *)fc->value; fc = (struct filter_control *) - &ab8500_filter_controls[AB8500_FILTER_SID_FIR].private_value; + ab8500_filter_controls[AB8500_FILTER_SID_FIR].private_value; drvdata->sid_fir_values = (long *)fc->value; snd_soc_dapm_disable_pin(dapm, "ANC Configure Input"); diff --git a/sound/soc/codecs/aw88395/aw88395.c b/sound/soc/codecs/aw88395/aw88395.c index 3602b5b9f7d77..dd09bac652f7f 100644 --- a/sound/soc/codecs/aw88395/aw88395.c +++ b/sound/soc/codecs/aw88395/aw88395.c @@ -456,8 +456,6 @@ static void aw88395_hw_reset(struct aw88395 *aw88395) usleep_range(AW88395_1000_US, AW88395_1000_US + 10); gpiod_set_value_cansleep(aw88395->reset_gpio, 1); usleep_range(AW88395_1000_US, AW88395_1000_US + 10); - } else { - dev_err(aw88395->aw_pa->dev, "%s failed", __func__); } } @@ -522,9 +520,10 @@ static int aw88395_i2c_probe(struct i2c_client *i2c) i2c_set_clientdata(i2c, aw88395); aw88395->reset_gpio = devm_gpiod_get_optional(&i2c->dev, "reset", GPIOD_OUT_LOW); - if (IS_ERR(aw88395->reset_gpio)) - dev_info(&i2c->dev, "reset gpio not defined\n"); - + if (IS_ERR(aw88395->reset_gpio)) { + return dev_err_probe(&i2c->dev, PTR_ERR(aw88395->reset_gpio), + "failed to get reset gpio\n"); + } /* hardware reset */ aw88395_hw_reset(aw88395); diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index e05d975ba7945..033e56d5e9db4 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -108,8 +108,6 @@ int cs35l56_set_patch(struct cs35l56_base *cs35l56_base) EXPORT_SYMBOL_NS_GPL(cs35l56_set_patch, "SND_SOC_CS35L56_SHARED"); static const struct reg_default cs35l56_reg_defaults[] = { - /* no defaults for OTP_MEM - first read populates cache */ - { CS35L56_ASP1_ENABLES1, 0x00000000 }, { CS35L56_ASP1_CONTROL1, 0x00000028 }, { CS35L56_ASP1_CONTROL2, 0x18180200 }, @@ -138,8 +136,6 @@ static const struct reg_default cs35l56_reg_defaults[] = { }; static const struct reg_default cs35l63_reg_defaults[] = { - /* no defaults for OTP_MEM - first read populates cache */ - { CS35L56_ASP1_ENABLES1, 0x00000000 }, { CS35L56_ASP1_CONTROL1, 0x00000028 }, { CS35L56_ASP1_CONTROL2, 0x18180200 }, @@ -282,6 +278,9 @@ static bool cs35l56_common_volatile_reg(unsigned int reg) case CS35L56_GLOBAL_ENABLES: /* owned by firmware */ case CS35L56_BLOCK_ENABLES: /* owned by firmware */ case CS35L56_BLOCK_ENABLES2: /* owned by firmware */ + case CS35L56_OTP_MEM_53: + case CS35L56_OTP_MEM_54: + case CS35L56_OTP_MEM_55: case CS35L56_SYNC_GPIO1_CFG ... CS35L56_ASP2_DIO_GPIO13_CFG: case CS35L56_UPDATE_REGS: case CS35L56_REFCLK_INPUT: /* owned by firmware */ diff --git a/sound/soc/codecs/es8389.c b/sound/soc/codecs/es8389.c index 8d418cae371a8..449d9574b03a8 100644 --- a/sound/soc/codecs/es8389.c +++ b/sound/soc/codecs/es8389.c @@ -892,7 +892,7 @@ static int es8389_probe(struct snd_soc_component *component) return ret; } - es8389->mclk = devm_clk_get(component->dev, "mclk"); + es8389->mclk = devm_clk_get_optional(component->dev, "mclk"); if (IS_ERR(es8389->mclk)) return dev_err_probe(component->dev, PTR_ERR(es8389->mclk), "ES8389 is unable to get mclk\n"); diff --git a/sound/soc/codecs/tas2764.c b/sound/soc/codecs/tas2764.c index 423b7073b3022..6aab6d2b74191 100644 --- a/sound/soc/codecs/tas2764.c +++ b/sound/soc/codecs/tas2764.c @@ -904,6 +904,7 @@ static bool tas2764_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case TAS2764_SW_RST: + case TAS2764_TEMP: case TAS2764_INT_LTCH0 ... TAS2764_INT_LTCH4: case TAS2764_INT_CLK_CFG: return true; diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index d4d7d056141bd..50501bcbe9167 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -624,7 +624,7 @@ static int tas2770_read_die_temp(struct tas2770_priv *tas2770, long *result) /* * As per datasheet: divide register by 16 and subtract 93 to get * degrees Celsius. hwmon requires millidegrees. Let's avoid rounding - * errors by subtracting 93 * 16 then multiplying by 1000 / 16. + * errors by subtracting 93 * 16 and scaling before dividing. * * NOTE: The ADC registers are initialised to 0 on reset. This means * that the temperature will read -93 *C until the chip is brought out @@ -633,7 +633,7 @@ static int tas2770_read_die_temp(struct tas2770_priv *tas2770, long *result) * value read back from its registers will be the last value sampled * before entering software shutdown. */ - *result = (reading - (93 * 16)) * (1000 / 16); + *result = (reading - (93 * 16)) * 1000 / 16; return 0; } diff --git a/sound/soc/codecs/wcd937x.c b/sound/soc/codecs/wcd937x.c index 10a2d598caa71..72a53f95d6887 100644 --- a/sound/soc/codecs/wcd937x.c +++ b/sound/soc/codecs/wcd937x.c @@ -547,6 +547,9 @@ static int wcd937x_codec_aux_dac_event(struct snd_soc_dapm_widget *w, WCD937X_DIGITAL_CDC_ANA_CLK_CTL, BIT(2), BIT(2)); snd_soc_component_update_bits(component, + WCD937X_AUX_AUXPA, + BIT(4), BIT(4)); + snd_soc_component_update_bits(component, WCD937X_DIGITAL_CDC_DIG_CLK_CTL, BIT(2), BIT(2)); snd_soc_component_update_bits(component, @@ -562,6 +565,9 @@ static int wcd937x_codec_aux_dac_event(struct snd_soc_dapm_widget *w, snd_soc_component_update_bits(component, WCD937X_DIGITAL_CDC_ANA_CLK_CTL, BIT(2), 0x00); + snd_soc_component_update_bits(component, + WCD937X_AUX_AUXPA, + BIT(4), 0x00); break; } @@ -730,10 +736,23 @@ static int wcd937x_codec_enable_aux_pa(struct snd_soc_dapm_widget *w, snd_soc_component_update_bits(component, WCD937X_ANA_RX_SUPPLIES, BIT(1), BIT(1)); + /* Enable AUX PA related RX supplies */ + snd_soc_component_update_bits(component, + WCD937X_ANA_RX_SUPPLIES, + BIT(6), BIT(6)); + snd_soc_component_update_bits(component, + WCD937X_ANA_RX_SUPPLIES, + BIT(7), BIT(7)); enable_irq(wcd937x->aux_pdm_wd_int); break; case SND_SOC_DAPM_PRE_PMD: disable_irq_nosync(wcd937x->aux_pdm_wd_int); + snd_soc_component_update_bits(component, + WCD937X_ANA_RX_SUPPLIES, + BIT(6), 0x00); + snd_soc_component_update_bits(component, + WCD937X_ANA_RX_SUPPLIES, + BIT(7), 0x00); break; case SND_SOC_DAPM_POST_PMD: usleep_range(2000, 2010); @@ -2051,7 +2070,12 @@ static const struct snd_kcontrol_new wcd937x_snd_controls[] = { wcd937x_get_swr_port, wcd937x_set_swr_port), SOC_SINGLE_EXT("LO Switch", WCD937X_LO, 0, 1, 0, wcd937x_get_swr_port, wcd937x_set_swr_port), - + SOC_SINGLE_EXT("CLSH PA Switch", WCD937X_CLSH, 0, 1, 0, + wcd937x_get_swr_port, wcd937x_set_swr_port), + SOC_SINGLE_EXT("DSD_L Switch", WCD937X_DSD_L, 0, 1, 0, + wcd937x_get_swr_port, wcd937x_set_swr_port), + SOC_SINGLE_EXT("DSD_R Switch", WCD937X_DSD_R, 0, 1, 0, + wcd937x_get_swr_port, wcd937x_set_swr_port), SOC_SINGLE_EXT("ADC1 Switch", WCD937X_ADC1, 1, 1, 0, wcd937x_get_swr_port, wcd937x_set_swr_port), SOC_SINGLE_EXT("ADC2 Switch", WCD937X_ADC2, 1, 1, 0, diff --git a/sound/soc/intel/boards/bytcr_wm5102.c b/sound/soc/intel/boards/bytcr_wm5102.c index 4879f79aef292..4aa0cf49b0335 100644 --- a/sound/soc/intel/boards/bytcr_wm5102.c +++ b/sound/soc/intel/boards/bytcr_wm5102.c @@ -170,6 +170,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, ret = byt_wm5102_prepare_and_enable_pll1(codec_dai, 48000); if (ret) { dev_err(card->dev, "Error setting codec sysclk: %d\n", ret); + clk_disable_unprepare(priv->mclk); return ret; } } else { diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index b3d61d973ce40..8662b422eb807 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -1412,7 +1412,8 @@ static struct snd_soc_acpi_mach *hda_sdw_machine_select(struct snd_sof_dev *sdev link_mask |= BIT(peripherals->array[i]->bus->link_id); link_num = hweight32(link_mask); - links = devm_kcalloc(sdev->dev, link_num, sizeof(*links), GFP_KERNEL); + /* An empty adr_link is needed to terminate the adr_link loop */ + links = devm_kcalloc(sdev->dev, link_num + 1, sizeof(*links), GFP_KERNEL); if (!links) return NULL; diff --git a/sound/soc/spacemit/k1_i2s.c b/sound/soc/spacemit/k1_i2s.c index 1cb99f1abc7cd..43481f387c446 100644 --- a/sound/soc/spacemit/k1_i2s.c +++ b/sound/soc/spacemit/k1_i2s.c @@ -93,8 +93,8 @@ static void spacemit_i2s_init(struct spacemit_i2s_dev *i2s) u32 sscr_val, sspsp_val, ssfcr_val, ssrwt_val; sscr_val = SSCR_TRAIL | SSCR_FRF_PSP; - ssfcr_val = FIELD_PREP(SSFCR_FIELD_TFT, 5) | - FIELD_PREP(SSFCR_FIELD_RFT, 5) | + ssfcr_val = FIELD_PREP(SSFCR_FIELD_TFT, 0xF) | + FIELD_PREP(SSFCR_FIELD_RFT, 0xF) | SSFCR_RSRE | SSFCR_TSRE; ssrwt_val = SSRWT_RWOT; sspsp_val = SSPSP_SFRMP; @@ -106,6 +106,37 @@ static void spacemit_i2s_init(struct spacemit_i2s_dev *i2s) writel(0, i2s->base + SSINTEN); } +static int spacemit_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct spacemit_i2s_dev *i2s = snd_soc_dai_get_drvdata(dai); + + switch (i2s->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, + 2, 2); + snd_pcm_hw_constraint_mask64(substream->runtime, + SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_FMTBIT_S16_LE); + break; + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, + 1, 1); + snd_pcm_hw_constraint_mask64(substream->runtime, + SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_FMTBIT_S32_LE); + break; + default: + dev_dbg(i2s->dev, "unexpected format type"); + return -EINVAL; + } + + return 0; +} + static int spacemit_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -157,22 +188,9 @@ static int spacemit_i2s_hw_params(struct snd_pcm_substream *substream, dma_data->maxburst = 32; dma_data->addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; } - - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_CHANNELS, - 1, 2); - snd_pcm_hw_constraint_mask64(substream->runtime, - SNDRV_PCM_HW_PARAM_FORMAT, - SNDRV_PCM_FMTBIT_S16_LE); break; case SND_SOC_DAIFMT_DSP_A: case SND_SOC_DAIFMT_DSP_B: - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_CHANNELS, - 1, 1); - snd_pcm_hw_constraint_mask64(substream->runtime, - SNDRV_PCM_HW_PARAM_FORMAT, - SNDRV_PCM_FMTBIT_S32_LE); break; default: dev_dbg(i2s->dev, "unexpected format type"); @@ -303,6 +321,7 @@ static int spacemit_i2s_dai_remove(struct snd_soc_dai *dai) static const struct snd_soc_dai_ops spacemit_i2s_dai_ops = { .probe = spacemit_i2s_dai_probe, .remove = spacemit_i2s_dai_remove, + .startup = spacemit_i2s_startup, .hw_params = spacemit_i2s_hw_params, .set_sysclk = spacemit_i2s_set_sysclk, .set_fmt = spacemit_i2s_set_fmt, diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 3f396c87802ed..1c18ef6971c09 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -1,7 +1,6 @@ # SPDX-License-Identifier: GPL-2.0 # Tegra platform Support snd-soc-tegra-pcm-y := tegra_pcm.o -snd-soc-tegra-utils-y += tegra_asoc_utils.o snd-soc-tegra20-ac97-y := tegra20_ac97.o snd-soc-tegra20-das-y := tegra20_das.o snd-soc-tegra20-i2s-y := tegra20_i2s.o diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 8af0c04041ee3..b20aae0caf60a 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -366,7 +366,7 @@ static int setup_card(struct snd_usb_caiaqdev *cdev) #ifdef CONFIG_SND_USB_CAIAQ_INPUT ret = snd_usb_caiaq_input_init(cdev); - if (ret < 0) { + if (ret < 0 && ret != -ENODEV) { dev_err(dev, "Unable to set up input system (ret=%d)\n", ret); return ret; } @@ -423,6 +423,7 @@ static int create_card(struct usb_device *usb_dev, cdev = caiaqdev(card); cdev->chip.dev = usb_get_dev(usb_dev); + card->private_free = card_free; cdev->chip.card = card; cdev->chip.usb_id = USB_ID(le16_to_cpu(usb_dev->descriptor.idVendor), le16_to_cpu(usb_dev->descriptor.idProduct)); @@ -511,10 +512,9 @@ static int init_card(struct snd_usb_caiaqdev *cdev) scnprintf(card->longname, sizeof(card->longname), "%s %s (%s)", cdev->vendor_name, cdev->product_name, usbpath); - card->private_free = card_free; err = setup_card(cdev); if (err < 0) - return err; + goto err_kill_urb; return 0; diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c index a9130891bb696..5c70fdf61cc13 100644 --- a/sound/usb/caiaq/input.c +++ b/sound/usb/caiaq/input.c @@ -804,7 +804,7 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *cdev) default: /* no input methods supported on this device */ - ret = -EINVAL; + ret = -ENODEV; goto exit_free_idev; } diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c index 49b3dd8d827da..d129b42eb979d 100644 --- a/sound/usb/misc/ua101.c +++ b/sound/usb/misc/ua101.c @@ -974,6 +974,13 @@ static int detect_usb_format(struct ua101 *ua) ua->capture.channels = fmt_capture->bNrChannels; ua->playback.channels = fmt_playback->bNrChannels; + if (!ua->capture.channels || !ua->playback.channels) { + dev_err(&ua->dev->dev, + "invalid channel count: capture %u, playback %u\n", + ua->capture.channels, ua->playback.channels); + return -EINVAL; + } + ua->capture.frame_bytes = fmt_capture->bSubframeSize * ua->capture.channels; ua->playback.frame_bytes = diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 85653112e7f3b..5fba456eb4a96 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1190,6 +1190,16 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, cval->res = 1; } break; + + case USB_ID(0x0e6f, 0x024a): /* PDP Riffmaster for PS4 */ + case USB_ID(0x0e6f, 0x0249): /* PDP Riffmaster for PS5 */ + if (!strcmp(kctl->id.name, "PCM Playback Volume")) { + usb_audio_info(chip, + "set volume quirk for PDP Riffmaster for PS4/PS5\n"); + cval->min = -2560; /* Mute under it */ + } + break; + case USB_ID(0x3302, 0x12db): /* MOONDROP Quark2 */ if (!strcmp(kctl->id.name, "PCM Playback Volume")) { usb_audio_info(chip, diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 803e03d4d77b8..4e9cfff4047fc 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2652,6 +2652,9 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, + +#endif /* disabled */ + { /* * The AudioBox USB advertises S24_3LE as the only supported format @@ -2700,7 +2703,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, -#endif /* disabled */ { /* diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 7b803ad58487f..0b4ecc2c6bcc4 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -125,7 +125,7 @@ static int add_audio_stream_from_fixed_fmt(struct snd_usb_audio *chip, snd_usb_audioformat_set_sync_ep(chip, fp); - err = snd_usb_add_audio_stream(chip, stream, fp); + err = snd_usb_add_audio_stream(chip, stream, fp, NULL); if (err < 0) return err; diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 2532bf97e05e0..b2c5c8198281a 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -79,7 +79,7 @@ static void snd_usb_audio_pcm_free(struct snd_pcm *pcm) static void snd_usb_init_substream(struct snd_usb_stream *as, int stream, struct audioformat *fp, - struct snd_usb_power_domain *pd) + struct snd_usb_power_domain **pdptr) { struct snd_usb_substream *subs = &as->substream[stream]; @@ -105,10 +105,11 @@ static void snd_usb_init_substream(struct snd_usb_stream *as, if (fp->channels > subs->channels_max) subs->channels_max = fp->channels; - if (pd) { - subs->str_pd = pd; + if (pdptr && *pdptr) { + subs->str_pd = *pdptr; + *pdptr = NULL; /* assigned */ /* Initialize Power Domain to idle status D1 */ - snd_usb_power_domain_set(subs->stream->chip, pd, + snd_usb_power_domain_set(subs->stream->chip, subs->str_pd, UAC3_PD_STATE_D1); } @@ -352,6 +353,8 @@ snd_pcm_chmap_elem *convert_chmap_v3(struct uac3_cluster_header_descriptor if (len < sizeof(*cs_desc)) break; cs_len = le16_to_cpu(cs_desc->wLength); + if (cs_len < sizeof(*cs_desc)) + break; if (len < cs_len) break; cs_type = cs_desc->bSegmentType; @@ -492,11 +495,14 @@ snd_pcm_chmap_elem *convert_chmap_v3(struct uac3_cluster_header_descriptor * if not, create a new pcm stream. note, fp is added to the substream * fmt_list and will be freed on the chip instance release. do not free * fp or do remove it from the substream fmt_list to avoid double-free. + * + * pdptr is optional and can be NULL. When it's non-NULL and the PD gets + * assigned to the stream, *pdptr is cleared to NULL upon return. */ -static int __snd_usb_add_audio_stream(struct snd_usb_audio *chip, - int stream, - struct audioformat *fp, - struct snd_usb_power_domain *pd) +int snd_usb_add_audio_stream(struct snd_usb_audio *chip, + int stream, + struct audioformat *fp, + struct snd_usb_power_domain **pdptr) { struct snd_usb_stream *as; @@ -529,7 +535,7 @@ static int __snd_usb_add_audio_stream(struct snd_usb_audio *chip, err = snd_pcm_new_stream(as->pcm, stream, 1); if (err < 0) return err; - snd_usb_init_substream(as, stream, fp, pd); + snd_usb_init_substream(as, stream, fp, pdptr); return add_chmap(as->pcm, stream, subs); } @@ -558,7 +564,7 @@ static int __snd_usb_add_audio_stream(struct snd_usb_audio *chip, else strscpy(pcm->name, "USB Audio"); - snd_usb_init_substream(as, stream, fp, pd); + snd_usb_init_substream(as, stream, fp, pdptr); /* * Keep using head insertion for M-Audio Audiophile USB (tm) which has a @@ -576,21 +582,6 @@ static int __snd_usb_add_audio_stream(struct snd_usb_audio *chip, return add_chmap(pcm, stream, &as->substream[stream]); } -int snd_usb_add_audio_stream(struct snd_usb_audio *chip, - int stream, - struct audioformat *fp) -{ - return __snd_usb_add_audio_stream(chip, stream, fp, NULL); -} - -static int snd_usb_add_audio_stream_v3(struct snd_usb_audio *chip, - int stream, - struct audioformat *fp, - struct snd_usb_power_domain *pd) -{ - return __snd_usb_add_audio_stream(chip, stream, fp, pd); -} - static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip, struct usb_host_interface *alts, int protocol, int iface_no) @@ -1003,7 +994,7 @@ snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip, * and request Cluster Descriptor */ wLength = le16_to_cpu(hc_header.wLength); - if (wLength < sizeof(cluster)) + if (wLength < sizeof(*cluster)) return NULL; cluster = kzalloc(wLength, GFP_KERNEL); if (!cluster) @@ -1113,8 +1104,7 @@ found_clock: } } - if (pd) - *pd_out = pd; + *pd_out = pd; return fp; } @@ -1129,7 +1119,6 @@ static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip, struct usb_interface_descriptor *altsd; int i, altno, err, stream; struct audioformat *fp = NULL; - struct snd_usb_power_domain *pd = NULL; bool set_iface_first; int num, protocol; @@ -1171,6 +1160,12 @@ static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip, if (snd_usb_apply_interface_quirk(chip, iface_no, altno)) continue; + /* pd may be allocated at snd_usb_get_audioformat_uac3() and + * assigned at snd_usb_add_audio_stream(); otherwise it'll be + * freed automatically by cleanup at each loop. + */ + struct snd_usb_power_domain *pd __free(kfree) = NULL; + /* * Roland audio streaming interfaces are marked with protocols * 0/1/2, but are UAC 1 compatible. @@ -1226,23 +1221,16 @@ static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip, *has_non_pcm = true; if ((fp->fmt_type == UAC_FORMAT_TYPE_I) == non_pcm) { audioformat_free(fp); - kfree(pd); fp = NULL; - pd = NULL; continue; } snd_usb_audioformat_set_sync_ep(chip, fp); dev_dbg(&dev->dev, "%u:%d: add audio endpoint %#x\n", iface_no, altno, fp->endpoint); - if (protocol == UAC_VERSION_3) - err = snd_usb_add_audio_stream_v3(chip, stream, fp, pd); - else - err = snd_usb_add_audio_stream(chip, stream, fp); - + err = snd_usb_add_audio_stream(chip, stream, fp, &pd); if (err < 0) { audioformat_free(fp); - kfree(pd); return err; } diff --git a/sound/usb/stream.h b/sound/usb/stream.h index d92e18d5818fe..61b9a133da018 100644 --- a/sound/usb/stream.h +++ b/sound/usb/stream.h @@ -7,7 +7,8 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int snd_usb_add_audio_stream(struct snd_usb_audio *chip, int stream, - struct audioformat *fp); + struct audioformat *fp, + struct snd_usb_power_domain **pdptr); #endif /* __USBAUDIO_STREAM_H */ |