diff options
Diffstat (limited to 'sound')
34 files changed, 260 insertions, 99 deletions
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 86ed2fbee0c8..840bb9cfe789 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -1025,7 +1025,7 @@ static u64 snd_compr_seqno_next(struct snd_compr_stream *stream) static int snd_compr_task_new(struct snd_compr_stream *stream, struct snd_compr_task *utask) { struct snd_compr_task_runtime *task; - int retval; + int retval, fd_i, fd_o; if (stream->runtime->total_tasks >= stream->runtime->fragments) return -EBUSY; @@ -1039,19 +1039,27 @@ static int snd_compr_task_new(struct snd_compr_stream *stream, struct snd_compr_ retval = stream->ops->task_create(stream, task); if (retval < 0) goto cleanup; - utask->input_fd = dma_buf_fd(task->input, O_WRONLY|O_CLOEXEC); - if (utask->input_fd < 0) { - retval = utask->input_fd; + /* similar functionality as in dma_buf_fd(), but ensure that both + file descriptors are allocated before fd_install() */ + if (!task->input || !task->input->file || !task->output || !task->output->file) { + retval = -EINVAL; goto cleanup; } - utask->output_fd = dma_buf_fd(task->output, O_RDONLY|O_CLOEXEC); - if (utask->output_fd < 0) { - retval = utask->output_fd; + fd_i = get_unused_fd_flags(O_WRONLY|O_CLOEXEC); + if (fd_i < 0) + goto cleanup; + fd_o = get_unused_fd_flags(O_RDONLY|O_CLOEXEC); + if (fd_o < 0) { + put_unused_fd(fd_i); goto cleanup; } /* keep dmabuf reference until freed with task free ioctl */ - dma_buf_get(utask->input_fd); - dma_buf_get(utask->output_fd); + get_dma_buf(task->input); + get_dma_buf(task->output); + fd_install(fd_i, task->input->file); + fd_install(fd_o, task->output->file); + utask->input_fd = fd_i; + utask->output_fd = fd_o; list_add_tail(&task->list, &stream->runtime->tasks); stream->runtime->total_tasks++; return 0; @@ -1069,7 +1077,7 @@ static int snd_compr_task_create(struct snd_compr_stream *stream, unsigned long return -EPERM; task = memdup_user((void __user *)arg, sizeof(*task)); if (IS_ERR(task)) - return PTR_ERR(no_free_ptr(task)); + return PTR_ERR(task); retval = snd_compr_task_new(stream, task); if (retval >= 0) if (copy_to_user((void __user *)arg, task, sizeof(*task))) @@ -1130,7 +1138,7 @@ static int snd_compr_task_start_ioctl(struct snd_compr_stream *stream, unsigned return -EPERM; task = memdup_user((void __user *)arg, sizeof(*task)); if (IS_ERR(task)) - return PTR_ERR(no_free_ptr(task)); + return PTR_ERR(task); retval = snd_compr_task_start(stream, task); if (retval >= 0) if (copy_to_user((void __user *)arg, task, sizeof(*task))) @@ -1174,18 +1182,18 @@ typedef void (*snd_compr_seq_func_t)(struct snd_compr_stream *stream, static int snd_compr_task_seq(struct snd_compr_stream *stream, unsigned long arg, snd_compr_seq_func_t fcn) { - struct snd_compr_task_runtime *task; + struct snd_compr_task_runtime *task, *temp; __u64 seqno; int retval; if (stream->runtime->state != SNDRV_PCM_STATE_SETUP) return -EPERM; - retval = get_user(seqno, (__u64 __user *)arg); - if (retval < 0) - return retval; + retval = copy_from_user(&seqno, (__u64 __user *)arg, sizeof(seqno)); + if (retval) + return -EFAULT; retval = 0; if (seqno == 0) { - list_for_each_entry_reverse(task, &stream->runtime->tasks, list) + list_for_each_entry_safe_reverse(task, temp, &stream->runtime->tasks, list) fcn(stream, task); } else { task = snd_compr_find_task(stream, seqno); @@ -1221,7 +1229,7 @@ static int snd_compr_task_status_ioctl(struct snd_compr_stream *stream, unsigned return -EPERM; status = memdup_user((void __user *)arg, sizeof(*status)); if (IS_ERR(status)) - return PTR_ERR(no_free_ptr(status)); + return PTR_ERR(status); retval = snd_compr_task_status(stream, status); if (retval >= 0) if (copy_to_user((void __user *)arg, status, sizeof(*status))) @@ -1247,6 +1255,7 @@ void snd_compr_task_finished(struct snd_compr_stream *stream, } EXPORT_SYMBOL_GPL(snd_compr_task_finished); +MODULE_IMPORT_NS("DMA_BUF"); #endif /* CONFIG_SND_COMPRESS_ACCEL */ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) diff --git a/sound/core/control_led.c b/sound/core/control_led.c index 65a1ebe87776..e33dfcf863cf 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -668,10 +668,16 @@ static void snd_ctl_led_sysfs_add(struct snd_card *card) goto cerr; led->cards[card->number] = led_card; snprintf(link_name, sizeof(link_name), "led-%s", led->name); - WARN(sysfs_create_link(&card->ctl_dev->kobj, &led_card->dev.kobj, link_name), - "can't create symlink to controlC%i device\n", card->number); - WARN(sysfs_create_link(&led_card->dev.kobj, &card->card_dev.kobj, "card"), - "can't create symlink to card%i\n", card->number); + if (sysfs_create_link(&card->ctl_dev->kobj, &led_card->dev.kobj, + link_name)) + dev_err(card->dev, + "%s: can't create symlink to controlC%i device\n", + __func__, card->number); + if (sysfs_create_link(&led_card->dev.kobj, &card->card_dev.kobj, + "card")) + dev_err(card->dev, + "%s: can't create symlink to card%i\n", + __func__, card->number); continue; cerr: diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 13b71069ae18..b3853583d2ae 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -505,7 +505,7 @@ static void *snd_dma_wc_alloc(struct snd_dma_buffer *dmab, size_t size) if (!p) return NULL; dmab->addr = dma_map_single(dmab->dev.dev, p, size, DMA_BIDIRECTIONAL); - if (dmab->addr == DMA_MAPPING_ERROR) { + if (dma_mapping_error(dmab->dev.dev, dmab->addr)) { do_free_pages(dmab->area, size, true); return NULL; } diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c index e3394919daa0..51ee4c00a843 100644 --- a/sound/core/seq/oss/seq_oss_synth.c +++ b/sound/core/seq/oss/seq_oss_synth.c @@ -66,6 +66,7 @@ static struct seq_oss_synth midi_synth_dev = { }; static DEFINE_SPINLOCK(register_lock); +static DEFINE_MUTEX(sysex_mutex); /* * prototypes @@ -497,6 +498,7 @@ snd_seq_oss_synth_sysex(struct seq_oss_devinfo *dp, int dev, unsigned char *buf, if (!info) return -ENXIO; + guard(mutex)(&sysex_mutex); sysex = info->sysex; if (sysex == NULL) { sysex = kzalloc(sizeof(*sysex), GFP_KERNEL); diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 3930e2f9082f..77b6ac9b5c11 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -1275,10 +1275,16 @@ static int snd_seq_ioctl_set_client_info(struct snd_seq_client *client, if (client->type != client_info->type) return -EINVAL; - /* check validity of midi_version field */ - if (client->user_pversion >= SNDRV_PROTOCOL_VERSION(1, 0, 3) && - client_info->midi_version > SNDRV_SEQ_CLIENT_UMP_MIDI_2_0) - return -EINVAL; + if (client->user_pversion >= SNDRV_PROTOCOL_VERSION(1, 0, 3)) { + /* check validity of midi_version field */ + if (client_info->midi_version > SNDRV_SEQ_CLIENT_UMP_MIDI_2_0) + return -EINVAL; + + /* check if UMP is supported in kernel */ + if (!IS_ENABLED(CONFIG_SND_SEQ_UMP) && + client_info->midi_version > 0) + return -EINVAL; + } /* fill the info fields */ if (client_info->name[0]) diff --git a/sound/core/ump.c b/sound/core/ump.c index fe4d39ae1159..9198bff4768c 100644 --- a/sound/core/ump.c +++ b/sound/core/ump.c @@ -1244,7 +1244,7 @@ static int fill_legacy_mapping(struct snd_ump_endpoint *ump) num = 0; for (i = 0; i < SNDRV_UMP_MAX_GROUPS; i++) - if ((group_maps & (1U << i)) && ump->groups[i].valid) + if (group_maps & (1U << i)) ump->legacy_mapping[num++] = i; return num; diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index d96266c8cb38..4ef7878e8fd4 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -151,10 +151,6 @@ static int cs35l56_hda_runtime_resume(struct device *dev) } } - ret = cs35l56_force_sync_asp1_registers_from_cache(&cs35l56->base); - if (ret) - goto err; - return 0; err: @@ -1059,9 +1055,6 @@ int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int hid, int id) regmap_multi_reg_write(cs35l56->base.regmap, cs35l56_hda_dai_config, ARRAY_SIZE(cs35l56_hda_dai_config)); - ret = cs35l56_force_sync_asp1_registers_from_cache(&cs35l56->base); - if (ret) - goto dsp_err; /* * By default only enable one ASP1TXn, where n=amplifier index, @@ -1087,7 +1080,6 @@ int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int hid, int id) pm_err: pm_runtime_disable(cs35l56->base.dev); -dsp_err: cs_dsp_remove(&cs35l56->cs_dsp); err: gpiod_set_value_cansleep(cs35l56->base.reset_gpio, 0); diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index e4673a71551a..d40197fb5fbd 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1134,7 +1134,6 @@ struct ca0132_spec { struct hda_codec *codec; struct delayed_work unsol_hp_work; - int quirk; #ifdef ENABLE_TUNING_CONTROLS long cur_ctl_vals[TUNING_CTLS_COUNT]; @@ -1166,7 +1165,6 @@ struct ca0132_spec { * CA0132 quirks table */ enum { - QUIRK_NONE, QUIRK_ALIENWARE, QUIRK_ALIENWARE_M17XR4, QUIRK_SBZ, @@ -1176,10 +1174,11 @@ enum { QUIRK_R3D, QUIRK_AE5, QUIRK_AE7, + QUIRK_NONE = HDA_FIXUP_ID_NOT_SET, }; #ifdef CONFIG_PCI -#define ca0132_quirk(spec) ((spec)->quirk) +#define ca0132_quirk(spec) ((spec)->codec->fixup_id) #define ca0132_use_pci_mmio(spec) ((spec)->use_pci_mmio) #define ca0132_use_alt_functions(spec) ((spec)->use_alt_functions) #define ca0132_use_alt_controls(spec) ((spec)->use_alt_controls) @@ -1293,7 +1292,7 @@ static const struct hda_pintbl ae7_pincfgs[] = { {} }; -static const struct snd_pci_quirk ca0132_quirks[] = { +static const struct hda_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1028, 0x057b, "Alienware M17x R4", QUIRK_ALIENWARE_M17XR4), SND_PCI_QUIRK(0x1028, 0x0685, "Alienware 15 2015", QUIRK_ALIENWARE), SND_PCI_QUIRK(0x1028, 0x0688, "Alienware 17 2015", QUIRK_ALIENWARE), @@ -1316,6 +1315,19 @@ static const struct snd_pci_quirk ca0132_quirks[] = { {} }; +static const struct hda_model_fixup ca0132_quirk_models[] = { + { .id = QUIRK_ALIENWARE, .name = "alienware" }, + { .id = QUIRK_ALIENWARE_M17XR4, .name = "alienware-m17xr4" }, + { .id = QUIRK_SBZ, .name = "sbz" }, + { .id = QUIRK_ZXR, .name = "zxr" }, + { .id = QUIRK_ZXR_DBPRO, .name = "zxr-dbpro" }, + { .id = QUIRK_R3DI, .name = "r3di" }, + { .id = QUIRK_R3D, .name = "r3d" }, + { .id = QUIRK_AE5, .name = "ae5" }, + { .id = QUIRK_AE7, .name = "ae7" }, + {} +}; + /* Output selection quirk info structures. */ #define MAX_QUIRK_MMIO_GPIO_SET_VALS 3 #define MAX_QUIRK_SCP_SET_VALS 2 @@ -9957,17 +9969,15 @@ static int ca0132_prepare_verbs(struct hda_codec *codec) */ static void sbz_detect_quirk(struct hda_codec *codec) { - struct ca0132_spec *spec = codec->spec; - switch (codec->core.subsystem_id) { case 0x11020033: - spec->quirk = QUIRK_ZXR; + codec->fixup_id = QUIRK_ZXR; break; case 0x1102003f: - spec->quirk = QUIRK_ZXR_DBPRO; + codec->fixup_id = QUIRK_ZXR_DBPRO; break; default: - spec->quirk = QUIRK_SBZ; + codec->fixup_id = QUIRK_SBZ; break; } } @@ -9976,7 +9986,6 @@ static int patch_ca0132(struct hda_codec *codec) { struct ca0132_spec *spec; int err; - const struct snd_pci_quirk *quirk; codec_dbg(codec, "patch_ca0132\n"); @@ -9987,11 +9996,7 @@ static int patch_ca0132(struct hda_codec *codec) spec->codec = codec; /* Detect codec quirk */ - quirk = snd_pci_quirk_lookup(codec->bus->pci, ca0132_quirks); - if (quirk) - spec->quirk = quirk->value; - else - spec->quirk = QUIRK_NONE; + snd_hda_pick_fixup(codec, ca0132_quirk_models, ca0132_quirks, NULL); if (ca0132_quirk(spec) == QUIRK_SBZ) sbz_detect_quirk(codec); @@ -10068,7 +10073,7 @@ static int patch_ca0132(struct hda_codec *codec) spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); if (spec->mem_base == NULL) { codec_warn(codec, "pci_iomap failed! Setting quirk to QUIRK_NONE."); - spec->quirk = QUIRK_NONE; + codec->fixup_id = QUIRK_NONE; } } #endif diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 17392b21d5bf..ad66378d7321 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7714,6 +7714,7 @@ enum { ALC274_FIXUP_HP_MIC, ALC274_FIXUP_HP_HEADSET_MIC, ALC274_FIXUP_HP_ENVY_GPIO, + ALC274_FIXUP_ASUS_ZEN_AIO_27, ALC256_FIXUP_ASUS_HPE, ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK, ALC287_FIXUP_HP_GPIO_LED, @@ -9516,6 +9517,26 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc274_fixup_hp_envy_gpio, }, + [ALC274_FIXUP_ASUS_ZEN_AIO_27] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0x10 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xc420 }, + { 0x20, AC_VERB_SET_COEF_INDEX, 0x40 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x8800 }, + { 0x20, AC_VERB_SET_COEF_INDEX, 0x49 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0249 }, + { 0x20, AC_VERB_SET_COEF_INDEX, 0x4a }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x202b }, + { 0x20, AC_VERB_SET_COEF_INDEX, 0x62 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xa007 }, + { 0x20, AC_VERB_SET_COEF_INDEX, 0x6b }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x5060 }, + {} + }, + .chained = true, + .chain_id = ALC2XX_FIXUP_HEADSET_MIC, + }, [ALC256_FIXUP_ASUS_HPE] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -10142,6 +10163,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x1430, "Acer TravelMate B311R-31", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1466, "Acer Aspire A515-56", ALC255_FIXUP_ACER_HEADPHONE_AND_MIC), SND_PCI_QUIRK(0x1025, 0x1534, "Acer Predator PH315-54", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1025, 0x159c, "Acer Nitro 5 AN515-58", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x169a, "Acer Swift SFG16", ALC256_FIXUP_ACER_SFG16_MICMUTE_LED), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x053c, "Dell Latitude E5430", ALC292_FIXUP_DELL_E7X), @@ -10619,6 +10641,8 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1e1f, "ASUS Vivobook 15 X1504VAP", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1e51, "ASUS Zephyrus M15", ALC294_FIXUP_ASUS_GU502_PINS), SND_PCI_QUIRK(0x1043, 0x1e5e, "ASUS ROG Strix G513", ALC294_FIXUP_ASUS_G513_PINS), + SND_PCI_QUIRK(0x1043, 0x1e63, "ASUS H7606W", ALC285_FIXUP_ASUS_GU605_SPI_SPEAKER2_TO_DAC1), + SND_PCI_QUIRK(0x1043, 0x1e83, "ASUS GA605W", ALC285_FIXUP_ASUS_GU605_SPI_SPEAKER2_TO_DAC1), SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1eb3, "ASUS Ally RCLA72", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x1043, 0x1ed3, "ASUS HN7306W", ALC287_FIXUP_CS35L41_I2C_2), @@ -10630,6 +10654,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1f62, "ASUS UX7602ZM", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1f92, "ASUS ROG Flow X16", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), + SND_PCI_QUIRK(0x1043, 0x31d0, "ASUS Zen AIO 27 Z272SD_A272SD", ALC274_FIXUP_ASUS_ZEN_AIO_27), SND_PCI_QUIRK(0x1043, 0x3a20, "ASUS G614JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), SND_PCI_QUIRK(0x1043, 0x3a30, "ASUS G814JVR/JIR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), SND_PCI_QUIRK(0x1043, 0x3a40, "ASUS G814JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), @@ -10907,8 +10932,8 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x38e0, "Yoga Y990 Intel VECO Dual", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38f8, "Yoga Book 9i", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38df, "Y990 YG DUAL", ALC287_FIXUP_TAS2781_I2C), - SND_PCI_QUIRK(0x17aa, 0x38f9, "Thinkbook 16P Gen5", ALC287_FIXUP_CS35L41_I2C_2), - SND_PCI_QUIRK(0x17aa, 0x38fa, "Thinkbook 16P Gen5", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x38f9, "Thinkbook 16P Gen5", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x38fa, "Thinkbook 16P Gen5", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x38fd, "ThinkBook plus Gen5 Hybrid", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3913, "Lenovo 145", ALC236_FIXUP_LENOVO_INV_DMIC), @@ -10972,6 +10997,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1945, "Redmi G", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1947, "RedmiBook Air", ALC255_FIXUP_XIAOMI_HEADSET_MIC), + SND_PCI_QUIRK(0x1f66, 0x0105, "Ayaneo Portable Game Player", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x2782, 0x0214, "VAIO VJFE-CL", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x2782, 0x0228, "Infinix ZERO BOOK 13", ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13), SND_PCI_QUIRK(0x2782, 0x0232, "CHUWI CoreBook XPro", ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO), @@ -10986,6 +11012,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0xf111, 0x0001, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0xf111, 0x0006, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0xf111, 0x0009, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0xf111, 0x000c, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), #if 0 /* Below is a quirk table taken from the old code. @@ -11177,6 +11204,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, .name = "alc255-acer-headphone-and-mic"}, {.id = ALC285_FIXUP_HP_GPIO_AMP_INIT, .name = "alc285-hp-amp-init"}, {.id = ALC236_FIXUP_LENOVO_INV_DMIC, .name = "alc236-fixup-lenovo-inv-mic"}, + {.id = ALC2XX_FIXUP_HEADSET_MIC, .name = "alc2xx-fixup-headset-mic"}, {} }; #define ALC225_STANDARD_PINS \ diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 0af015806aba..0e42b87dadb8 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -142,6 +142,9 @@ static int tas2781_read_acpi(struct tasdevice_priv *p, const char *hid) } sub = acpi_get_subsystem_id(ACPI_HANDLE(physdev)); if (IS_ERR(sub)) { + /* No subsys id in older tas2563 projects. */ + if (!strncmp(hid, "INT8866", sizeof("INT8866"))) + goto end_2563; dev_err(p->dev, "Failed to get SUBSYS ID.\n"); ret = PTR_ERR(sub); goto err; @@ -164,6 +167,7 @@ static int tas2781_read_acpi(struct tasdevice_priv *p, const char *hid) p->speaker_id = NULL; } +end_2563: acpi_dev_free_resource_list(&resources); strscpy(p->dev_name, hid, sizeof(p->dev_name)); put_device(physdev); diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c index a4d07438ad64..3f5422145c5e 100644 --- a/sound/sh/sh_dac_audio.c +++ b/sound/sh/sh_dac_audio.c @@ -163,7 +163,7 @@ static int snd_sh_dac_pcm_copy(struct snd_pcm_substream *substream, /* channel is not used (interleaved data) */ struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); - if (copy_from_iter(chip->data_buffer + pos, src, count) != count) + if (copy_from_iter(chip->data_buffer + pos, count, src) != count) return -EFAULT; chip->buffer_end = chip->data_buffer + pos + count; diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index 823a69bf778b..4575326d0635 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -375,11 +375,18 @@ static int get_acp63_device_config(struct pci_dev *pci, struct acp63_dev_data *a { struct acpi_device *pdm_dev; const union acpi_object *obj; + acpi_handle handle; + acpi_integer dmic_status; u32 config; bool is_dmic_dev = false; bool is_sdw_dev = false; + bool wov_en, dmic_en; int ret; + /* IF WOV entry not found, enable dmic based on acp-audio-device-type entry*/ + wov_en = true; + dmic_en = false; + config = readl(acp_data->acp63_base + ACP_PIN_CONFIG); switch (config) { case ACP_CONFIG_4: @@ -412,10 +419,18 @@ static int get_acp63_device_config(struct pci_dev *pci, struct acp63_dev_data *a if (!acpi_dev_get_property(pdm_dev, "acp-audio-device-type", ACPI_TYPE_INTEGER, &obj) && obj->integer.value == ACP_DMIC_DEV) - is_dmic_dev = true; + dmic_en = true; } + + handle = ACPI_HANDLE(&pci->dev); + ret = acpi_evaluate_integer(handle, "_WOV", NULL, &dmic_status); + if (!ACPI_FAILURE(ret)) + wov_en = dmic_status; } + if (dmic_en && wov_en) + is_dmic_dev = true; + if (acp_data->is_sdw_config) { ret = acp_scan_sdw_devices(&pci->dev, ACP63_SDW_ADDR); if (!ret && acp_data->info.link_mask) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index e38c5885dadf..ecf57a6cb7c3 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -578,14 +578,19 @@ static int acp6x_probe(struct platform_device *pdev) handle = ACPI_HANDLE(pdev->dev.parent); ret = acpi_evaluate_integer(handle, "_WOV", NULL, &dmic_status); - if (!ACPI_FAILURE(ret)) + if (!ACPI_FAILURE(ret)) { wov_en = dmic_status; + if (!wov_en) + return -ENODEV; + } else { + /* Incase of ACPI method read failure then jump to check_dmi_entry */ + goto check_dmi_entry; + } - if (is_dmic_enable && wov_en) + if (is_dmic_enable) platform_set_drvdata(pdev, &acp6x_card); - else - return 0; +check_dmi_entry: /* check for any DMI overrides */ dmi_id = dmi_first_match(yc_acp_quirk_table); if (dmi_id) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0f2df7c91e18..0b9e87dc2b6c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -2451,6 +2451,7 @@ config SND_SOC_WM8993 config SND_SOC_WM8994 tristate + depends on MFD_WM8994 config SND_SOC_WM8995 tristate diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 4236f78beec0..83c21c17fb80 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -2404,6 +2404,7 @@ static int cs42l43_codec_runtime_resume(struct device *dev) static const struct dev_pm_ops cs42l43_codec_pm_ops = { RUNTIME_PM_OPS(NULL, cs42l43_codec_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) }; static const struct platform_device_id cs42l43_codec_id_table[] = { diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 61729e5b50a8..f508df01145b 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -39,7 +39,9 @@ struct es8316_priv { struct snd_soc_jack *jack; int irq; unsigned int sysclk; - unsigned int allowed_rates[ARRAY_SIZE(supported_mclk_lrck_ratios)]; + /* ES83xx supports halving the MCLK so it supports twice as many rates + */ + unsigned int allowed_rates[ARRAY_SIZE(supported_mclk_lrck_ratios) * 2]; struct snd_pcm_hw_constraint_list sysclk_constraints; bool jd_inverted; }; @@ -386,6 +388,12 @@ static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai, if (freq % ratio == 0) es8316->allowed_rates[count++] = freq / ratio; + + /* We also check if the halved MCLK produces a valid rate + * since the codec supports halving the MCLK. + */ + if ((freq / ratio) % 2 == 0) + es8316->allowed_rates[count++] = freq / ratio / 2; } if (count) { diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index a5603b617688..b06eead7e0f6 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -616,7 +616,7 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) 0x0F, 0x0F); if (es8326->version > ES8326_VERSION_B) { regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x10); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x30); } } } else { @@ -631,6 +631,8 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) regmap_write(es8326->regmap, ES8326_HPR_OFFSET_INI, offset_r); es8326->calibrated = true; } + regmap_update_bits(es8326->regmap, ES8326_CLK_INV, 0xc0, 0x00); + regmap_update_bits(es8326->regmap, ES8326_CLK_MUX, 0x80, 0x00); if (direction == SNDRV_PCM_STREAM_PLAYBACK) { regmap_update_bits(es8326->regmap, ES8326_DAC_DSM, 0x01, 0x01); usleep_range(1000, 5000); @@ -645,7 +647,7 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) } else { msleep(300); if (es8326->version > ES8326_VERSION_B) { - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x50); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x70); regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x00); } regmap_update_bits(es8326->regmap, ES8326_ADC_MUTE, @@ -676,6 +678,10 @@ static int es8326_set_bias_level(struct snd_soc_component *codec, regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x00); regmap_update_bits(es8326->regmap, ES8326_CLK_CTL, 0x20, 0x20); regmap_update_bits(es8326->regmap, ES8326_RESET, 0x02, 0x00); + if (es8326->version > ES8326_VERSION_B) { + regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x30); + } break; case SND_SOC_BIAS_PREPARE: break; @@ -683,6 +689,12 @@ static int es8326_set_bias_level(struct snd_soc_component *codec, regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x3b); regmap_update_bits(es8326->regmap, ES8326_CLK_CTL, 0x20, 0x00); regmap_write(es8326->regmap, ES8326_SDINOUT1_IO, ES8326_IO_INPUT); + if (es8326->version > ES8326_VERSION_B) { + regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x10); + } + regmap_update_bits(es8326->regmap, ES8326_CLK_INV, 0xc0, 0xc0); + regmap_update_bits(es8326->regmap, ES8326_CLK_MUX, 0x80, 0x80); break; case SND_SOC_BIAS_OFF: clk_disable_unprepare(es8326->mclk); @@ -773,7 +785,10 @@ static void es8326_jack_button_handler(struct work_struct *work) case 0x6f: case 0x4b: /* button volume up */ - cur_button = SND_JACK_BTN_1; + if ((iface == 0x6f) && (es8326->version > ES8326_VERSION_B)) + cur_button = SND_JACK_BTN_0; + else + cur_button = SND_JACK_BTN_1; break; case 0x27: /* button volume down */ @@ -1082,7 +1097,7 @@ static void es8326_init(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_ADC2_SRC, 0x66); es8326_disable_micbias(es8326->component); if (es8326->version > ES8326_VERSION_B) { - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x13); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x10); regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); } diff --git a/sound/soc/codecs/rt722-sdca.c b/sound/soc/codecs/rt722-sdca.c index 908846e994df..e17a142d03b9 100644 --- a/sound/soc/codecs/rt722-sdca.c +++ b/sound/soc/codecs/rt722-sdca.c @@ -1468,13 +1468,18 @@ static void rt722_sdca_jack_preset(struct rt722_sdca_priv *rt722) 0x008d); /* check HP calibration FSM status */ for (loop_check = 0; loop_check < chk_cnt; loop_check++) { + usleep_range(10000, 11000); ret = rt722_sdca_index_read(rt722, RT722_VENDOR_CALI, RT722_DAC_DC_CALI_CTL3, &calib_status); - if (ret < 0 || loop_check == chk_cnt) + if (ret < 0) dev_dbg(&rt722->slave->dev, "calibration failed!, ret=%d\n", ret); if ((calib_status & 0x0040) == 0x0) break; } + + if (loop_check == chk_cnt) + dev_dbg(&rt722->slave->dev, "%s, calibration time-out!\n", __func__); + /* Set ADC09 power entity floating control */ rt722_sdca_index_write(rt722, RT722_VENDOR_HDA_CTL, RT722_ADC0A_08_PDE_FLOAT_CTL, 0x2a12); diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index be2ca5eb6c93..728bf78ae71f 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -78,7 +78,7 @@ static const struct bulk_reg_val tas2781_cali_start_reg[] = { X2781_CL_STT_VAL(TAS2781_PRM_INT_MASK_REG, 0xfe, false), X2781_CL_STT_VAL(TAS2781_PRM_CLK_CFG_REG, 0xdd, false), X2781_CL_STT_VAL(TAS2781_PRM_RSVD_REG, 0x20, false), - X2781_CL_STT_VAL(TAS2781_PRM_TEST_57_REG, 0x14, false), + X2781_CL_STT_VAL(TAS2781_PRM_TEST_57_REG, 0x14, true), X2781_CL_STT_VAL(TAS2781_PRM_TEST_62_REG, 0x45, true), X2781_CL_STT_VAL(TAS2781_PRM_PVDD_UVLO_REG, 0x03, false), X2781_CL_STT_VAL(TAS2781_PRM_CHNL_0_REG, 0xa8, false), @@ -370,7 +370,7 @@ static void sngl_calib_start(struct tasdevice_priv *tas_priv, int i, tasdevice_dev_read(tas_priv, i, p[j].reg, (int *)&p[j].val[0]); } else { - switch (p[j].reg) { + switch (tas2781_cali_start_reg[j].reg) { case 0: { if (!reg[0]) continue; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 8e88830e8e57..678540b78280 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -29,8 +29,8 @@ config SND_SOC_FSL_SAI config SND_SOC_FSL_MQS tristate "Medium Quality Sound (MQS) module support" depends on SND_SOC_FSL_SAI + depends on IMX_SCMI_MISC_DRV || !IMX_SCMI_MISC_DRV select REGMAP_MMIO - select IMX_SCMI_MISC_DRV if IMX_SCMI_MISC_EXT !=n help Say Y if you want to add Medium Quality Sound (MQS) support for the Freescale CPUs. diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index b6ff04f7138a..ee946e0d3f49 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1204,7 +1204,7 @@ static struct snd_kcontrol_new fsl_spdif_ctrls[] = { }, /* DPLL lock info get controller */ { - .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = RX_SAMPLE_RATE_KCONTROL, .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index 1e0bfd59d511..9c184ab73468 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -171,7 +171,7 @@ static int fsl_xcvr_capds_put(struct snd_kcontrol *kcontrol, } static struct snd_kcontrol_new fsl_xcvr_earc_capds_kctl = { - .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capabilities Data Structure", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, .info = fsl_xcvr_type_capds_bytes_info, diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index 5280c1b20d85..1f5c4e8ff1b9 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -771,7 +771,7 @@ static void graph_link_init(struct simple_util_priv *priv, of_node_get(port_codec); if (graph_lnk_is_multi(port_codec)) { ep_codec = graph_get_next_multi_ep(&port_codec); - of_node_put(port_cpu); + of_node_put(port_codec); port_codec = ep_to_port(ep_codec); } else { ep_codec = of_graph_get_next_port_endpoint(port_codec, NULL); diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 810be7c949a5..c9f9c9b0de9b 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -632,7 +632,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { .callback = sof_sdw_quirk_cb, .matches = { DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"), - DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "233C") + DMI_MATCH(DMI_PRODUCT_NAME, "21QB") }, /* Note this quirk excludes the CODEC mic */ .driver_data = (void *)(SOC_SDW_CODEC_MIC), @@ -641,9 +641,26 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { .callback = sof_sdw_quirk_cb, .matches = { DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"), - DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "233B") + DMI_MATCH(DMI_PRODUCT_NAME, "21QA") }, - .driver_data = (void *)(SOC_SDW_SIDECAR_AMPS), + /* Note this quirk excludes the CODEC mic */ + .driver_data = (void *)(SOC_SDW_CODEC_MIC), + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "21Q6") + }, + .driver_data = (void *)(SOC_SDW_SIDECAR_AMPS | SOC_SDW_CODEC_MIC), + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "21Q7") + }, + .driver_data = (void *)(SOC_SDW_SIDECAR_AMPS | SOC_SDW_CODEC_MIC), }, /* ArrowLake devices */ @@ -1067,8 +1084,12 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) return ret; } - /* One per DAI link, worst case is a DAI link for every endpoint */ - sof_dais = kcalloc(num_ends, sizeof(*sof_dais), GFP_KERNEL); + /* + * One per DAI link, worst case is a DAI link for every endpoint, also + * add one additional to act as a terminator such that code can iterate + * until it hits an uninitialised DAI. + */ + sof_dais = kcalloc(num_ends + 1, sizeof(*sof_dais), GFP_KERNEL); if (!sof_dais) return -ENOMEM; diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.c b/sound/soc/mediatek/common/mtk-afe-platform-driver.c index 9b72b2a7ae91..6b6330583941 100644 --- a/sound/soc/mediatek/common/mtk-afe-platform-driver.c +++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.c @@ -120,8 +120,8 @@ int mtk_afe_pcm_new(struct snd_soc_component *component, struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); size = afe->mtk_afe_hardware->buffer_bytes_max; - snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV, - afe->dev, size, size); + snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV, afe->dev, 0, size); + return 0; } EXPORT_SYMBOL_GPL(mtk_afe_pcm_new); diff --git a/sound/soc/renesas/rcar/adg.c b/sound/soc/renesas/rcar/adg.c index 0f190abf00e7..191f212d338c 100644 --- a/sound/soc/renesas/rcar/adg.c +++ b/sound/soc/renesas/rcar/adg.c @@ -374,12 +374,12 @@ int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *ssi_mod, unsigned int rate) return 0; } -void rsnd_adg_clk_control(struct rsnd_priv *priv, int enable) +int rsnd_adg_clk_control(struct rsnd_priv *priv, int enable) { struct rsnd_adg *adg = rsnd_priv_to_adg(priv); struct rsnd_mod *adg_mod = rsnd_mod_get(adg); struct clk *clk; - int i; + int ret = 0, i; if (enable) { rsnd_mod_bset(adg_mod, BRGCKR, 0x80770000, adg->ckr); @@ -389,18 +389,33 @@ void rsnd_adg_clk_control(struct rsnd_priv *priv, int enable) for_each_rsnd_clkin(clk, adg, i) { if (enable) { - clk_prepare_enable(clk); + ret = clk_prepare_enable(clk); /* * We shouldn't use clk_get_rate() under * atomic context. Let's keep it when * rsnd_adg_clk_enable() was called */ + if (ret < 0) + break; + adg->clkin_rate[i] = clk_get_rate(clk); } else { - clk_disable_unprepare(clk); + if (adg->clkin_rate[i]) + clk_disable_unprepare(clk); + + adg->clkin_rate[i] = 0; } } + + /* + * rsnd_adg_clk_enable() might return error (_disable() will not). + * We need to rollback in such case + */ + if (ret < 0) + rsnd_adg_clk_disable(priv); + + return ret; } static struct clk *rsnd_adg_create_null_clk(struct rsnd_priv *priv, @@ -753,7 +768,10 @@ int rsnd_adg_probe(struct rsnd_priv *priv) if (ret) return ret; - rsnd_adg_clk_enable(priv); + ret = rsnd_adg_clk_enable(priv); + if (ret) + return ret; + rsnd_adg_clk_dbg_info(priv, NULL); return 0; diff --git a/sound/soc/renesas/rcar/core.c b/sound/soc/renesas/rcar/core.c index e2234928c9e8..d3709fd0409e 100644 --- a/sound/soc/renesas/rcar/core.c +++ b/sound/soc/renesas/rcar/core.c @@ -2086,9 +2086,7 @@ static int __maybe_unused rsnd_resume(struct device *dev) { struct rsnd_priv *priv = dev_get_drvdata(dev); - rsnd_adg_clk_enable(priv); - - return 0; + return rsnd_adg_clk_enable(priv); } static const struct dev_pm_ops rsnd_pm_ops = { diff --git a/sound/soc/renesas/rcar/rsnd.h b/sound/soc/renesas/rcar/rsnd.h index 3c164d8e3b16..a5f54b65313c 100644 --- a/sound/soc/renesas/rcar/rsnd.h +++ b/sound/soc/renesas/rcar/rsnd.h @@ -608,7 +608,7 @@ int rsnd_adg_set_cmd_timsel_gen2(struct rsnd_mod *cmd_mod, struct rsnd_dai_stream *io); #define rsnd_adg_clk_enable(priv) rsnd_adg_clk_control(priv, 1) #define rsnd_adg_clk_disable(priv) rsnd_adg_clk_control(priv, 0) -void rsnd_adg_clk_control(struct rsnd_priv *priv, int enable); +int rsnd_adg_clk_control(struct rsnd_priv *priv, int enable); void rsnd_adg_clk_dbg_info(struct rsnd_priv *priv, struct seq_file *m); /* diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 4b1ea7b2c796..60b4b7b75215 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -127,8 +127,9 @@ config SND_SOC_SAMSUNG_TM2_WM5110 config SND_SOC_SAMSUNG_ARIES_WM8994 tristate "SoC I2S Audio support for WM8994 on Aries" - depends on SND_SOC_SAMSUNG && MFD_WM8994 && IIO && EXTCON + depends on SND_SOC_SAMSUNG && I2C && IIO && EXTCON select SND_SOC_BT_SCO + select MFD_WM8994 select SND_SOC_WM8994 select SND_SAMSUNG_I2S help @@ -140,8 +141,9 @@ config SND_SOC_SAMSUNG_ARIES_WM8994 config SND_SOC_SAMSUNG_MIDAS_WM1811 tristate "SoC I2S Audio support for Midas boards" - depends on SND_SOC_SAMSUNG && IIO + depends on SND_SOC_SAMSUNG && I2C && IIO select SND_SAMSUNG_I2S + select MFD_WM8994 select SND_SOC_WM8994 help Say Y if you want to add support for SoC audio on the Midas boards. diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index c13f89b7065e..0db2a3e554fb 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -103,8 +103,10 @@ hda_dai_get_ops(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai return sdai->platform_private; } -int hda_link_dma_cleanup(struct snd_pcm_substream *substream, struct hdac_ext_stream *hext_stream, - struct snd_soc_dai *cpu_dai) +static int +hda_link_dma_cleanup(struct snd_pcm_substream *substream, + struct hdac_ext_stream *hext_stream, + struct snd_soc_dai *cpu_dai, bool release) { const struct hda_dai_widget_dma_ops *ops = hda_dai_get_ops(substream, cpu_dai); struct sof_intel_hda_stream *hda_stream; @@ -128,6 +130,17 @@ int hda_link_dma_cleanup(struct snd_pcm_substream *substream, struct hdac_ext_st snd_hdac_ext_bus_link_clear_stream_id(hlink, stream_tag); } + if (!release) { + /* + * Force stream reconfiguration without releasing the channel on + * subsequent stream restart (without free), including LinkDMA + * reset. + * The stream is released via hda_dai_hw_free() + */ + hext_stream->link_prepared = 0; + return 0; + } + if (ops->release_hext_stream) ops->release_hext_stream(sdev, cpu_dai, substream); @@ -211,7 +224,7 @@ static int __maybe_unused hda_dai_hw_free(struct snd_pcm_substream *substream, if (!hext_stream) return 0; - return hda_link_dma_cleanup(substream, hext_stream, cpu_dai); + return hda_link_dma_cleanup(substream, hext_stream, cpu_dai, true); } static int __maybe_unused hda_dai_hw_params_data(struct snd_pcm_substream *substream, @@ -304,7 +317,8 @@ static int __maybe_unused hda_dai_trigger(struct snd_pcm_substream *substream, i switch (cmd) { case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: - ret = hda_link_dma_cleanup(substream, hext_stream, dai); + ret = hda_link_dma_cleanup(substream, hext_stream, dai, + cmd == SNDRV_PCM_TRIGGER_STOP ? false : true); if (ret < 0) { dev_err(sdev->dev, "%s: failed to clean up link DMA\n", __func__); return ret; @@ -660,8 +674,7 @@ static int hda_dai_suspend(struct hdac_bus *bus) } ret = hda_link_dma_cleanup(hext_stream->link_substream, - hext_stream, - cpu_dai); + hext_stream, cpu_dai, true); if (ret < 0) return ret; } diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 22bd9c3c8216..ee4ccc1a5490 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -1038,8 +1038,6 @@ const struct hda_dai_widget_dma_ops * hda_select_dai_widget_ops(struct snd_sof_dev *sdev, struct snd_sof_widget *swidget); int hda_dai_config(struct snd_soc_dapm_widget *w, unsigned int flags, struct snd_sof_dai_config_data *data); -int hda_link_dma_cleanup(struct snd_pcm_substream *substream, struct hdac_ext_stream *hext_stream, - struct snd_soc_dai *cpu_dai); static inline struct snd_sof_dev *widget_to_sdev(struct snd_soc_dapm_widget *w) { diff --git a/sound/usb/format.c b/sound/usb/format.c index 0cbf1d4fbe6e..6049d957694c 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -60,6 +60,8 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, pcm_formats |= SNDRV_PCM_FMTBIT_SPECIAL; /* flag potentially raw DSD capable altsettings */ fp->dsd_raw = true; + /* clear special format bit to avoid "unsupported format" msg below */ + format &= ~UAC2_FORMAT_TYPE_I_RAW_DATA; } format <<= 1; @@ -71,8 +73,11 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, sample_width = as->bBitResolution; sample_bytes = as->bSubslotSize; - if (format & UAC3_FORMAT_TYPE_I_RAW_DATA) + if (format & UAC3_FORMAT_TYPE_I_RAW_DATA) { pcm_formats |= SNDRV_PCM_FMTBIT_SPECIAL; + /* clear special format bit to avoid "unsupported format" msg below */ + format &= ~UAC3_FORMAT_TYPE_I_RAW_DATA; + } format <<= 1; break; diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c index 6eb7d93b358d..20ac32635f1f 100644 --- a/sound/usb/mixer_us16x08.c +++ b/sound/usb/mixer_us16x08.c @@ -687,7 +687,7 @@ static int snd_us16x08_meter_get(struct snd_kcontrol *kcontrol, struct usb_mixer_elem_info *elem = kcontrol->private_data; struct snd_usb_audio *chip = elem->head.mixer->chip; struct snd_us16x08_meter_store *store = elem->private_data; - u8 meter_urb[64]; + u8 meter_urb[64] = {0}; switch (kcontrol->private_value) { case 0: { diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 00101875d9a8..8ba0aff8be2e 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2179,6 +2179,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_MIC_RES_384), DEVICE_FLG(0x046d, 0x09a4, /* Logitech QuickCam E 3500 */ QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_IGNORE_CTL_ERROR), + DEVICE_FLG(0x0499, 0x1506, /* Yamaha THR5 */ + QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x0499, 0x1509, /* Steinberg UR22 */ QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x0499, 0x3108, /* Yamaha YIT-W12TX */ @@ -2323,6 +2325,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_DSD_RAW), DEVICE_FLG(0x2522, 0x0007, /* LH Labs Geek Out HD Audio 1V5 */ QUIRK_FLAG_SET_IFACE_FIRST), + DEVICE_FLG(0x262a, 0x9302, /* ddHiFi TC44C */ + QUIRK_FLAG_DSD_RAW), DEVICE_FLG(0x2708, 0x0002, /* Audient iD14 */ QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x2912, 0x30c8, /* Audioengine D1 */ |