diff options
Diffstat (limited to 'sound')
49 files changed, 663 insertions, 192 deletions
diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index a59de24695ec..dfe5a64e19d2 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -468,6 +468,76 @@ static int snd_pcm_ioctl_sync_ptr_x32(struct snd_pcm_substream *substream, } #endif /* CONFIG_X86_X32 */ +#ifdef __BIG_ENDIAN +typedef char __pad_before_u32[4]; +typedef char __pad_after_u32[0]; +#else +typedef char __pad_before_u32[0]; +typedef char __pad_after_u32[4]; +#endif + +/* PCM 2.0.15 API definition had a bug in mmap control; it puts the avail_min + * at the wrong offset due to a typo in padding type. + * The bug hits only 32bit. + * A workaround for incorrect read/write is needed only in 32bit compat mode. + */ +struct __snd_pcm_mmap_control64_buggy { + __pad_before_u32 __pad1; + __u32 appl_ptr; + __pad_before_u32 __pad2; /* SiC! here is the bug */ + __pad_before_u32 __pad3; + __u32 avail_min; + __pad_after_uframe __pad4; +}; + +static int snd_pcm_ioctl_sync_ptr_buggy(struct snd_pcm_substream *substream, + struct snd_pcm_sync_ptr __user *_sync_ptr) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_pcm_sync_ptr sync_ptr; + struct __snd_pcm_mmap_control64_buggy *sync_cp; + volatile struct snd_pcm_mmap_status *status; + volatile struct snd_pcm_mmap_control *control; + int err; + + memset(&sync_ptr, 0, sizeof(sync_ptr)); + sync_cp = (struct __snd_pcm_mmap_control64_buggy *)&sync_ptr.c.control; + if (get_user(sync_ptr.flags, (unsigned __user *)&(_sync_ptr->flags))) + return -EFAULT; + if (copy_from_user(sync_cp, &(_sync_ptr->c.control), sizeof(*sync_cp))) + return -EFAULT; + status = runtime->status; + control = runtime->control; + if (sync_ptr.flags & SNDRV_PCM_SYNC_PTR_HWSYNC) { + err = snd_pcm_hwsync(substream); + if (err < 0) + return err; + } + snd_pcm_stream_lock_irq(substream); + if (!(sync_ptr.flags & SNDRV_PCM_SYNC_PTR_APPL)) { + err = pcm_lib_apply_appl_ptr(substream, sync_cp->appl_ptr); + if (err < 0) { + snd_pcm_stream_unlock_irq(substream); + return err; + } + } else { + sync_cp->appl_ptr = control->appl_ptr; + } + if (!(sync_ptr.flags & SNDRV_PCM_SYNC_PTR_AVAIL_MIN)) + control->avail_min = sync_cp->avail_min; + else + sync_cp->avail_min = control->avail_min; + sync_ptr.s.status.state = status->state; + sync_ptr.s.status.hw_ptr = status->hw_ptr; + sync_ptr.s.status.tstamp = status->tstamp; + sync_ptr.s.status.suspended_state = status->suspended_state; + sync_ptr.s.status.audio_tstamp = status->audio_tstamp; + snd_pcm_stream_unlock_irq(substream); + if (copy_to_user(_sync_ptr, &sync_ptr, sizeof(sync_ptr))) + return -EFAULT; + return 0; +} + /* */ enum { @@ -537,7 +607,7 @@ static long snd_pcm_ioctl_compat(struct file *file, unsigned int cmd, unsigned l if (in_x32_syscall()) return snd_pcm_ioctl_sync_ptr_x32(substream, argp); #endif /* CONFIG_X86_X32 */ - return snd_pcm_common_ioctl(file, substream, cmd, argp); + return snd_pcm_ioctl_sync_ptr_buggy(substream, argp); case SNDRV_PCM_IOCTL_HW_REFINE32: return snd_pcm_ioctl_hw_params_compat(substream, 1, argp); case SNDRV_PCM_IOCTL_HW_PARAMS32: diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 6c0a4a67ad2e..6f30231bdb88 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -873,12 +873,21 @@ static long snd_rawmidi_ioctl(struct file *file, unsigned int cmd, unsigned long return -EINVAL; } } + case SNDRV_RAWMIDI_IOCTL_USER_PVERSION: + if (get_user(rfile->user_pversion, (unsigned int __user *)arg)) + return -EFAULT; + return 0; + case SNDRV_RAWMIDI_IOCTL_PARAMS: { struct snd_rawmidi_params params; if (copy_from_user(¶ms, argp, sizeof(struct snd_rawmidi_params))) return -EFAULT; + if (rfile->user_pversion < SNDRV_PROTOCOL_VERSION(2, 0, 2)) { + params.mode = 0; + memset(params.reserved, 0, sizeof(params.reserved)); + } switch (params.stream) { case SNDRV_RAWMIDI_STREAM_OUTPUT: if (rfile->output == NULL) diff --git a/sound/core/seq_device.c b/sound/core/seq_device.c index 382275c5b193..7f3fd8eb016f 100644 --- a/sound/core/seq_device.c +++ b/sound/core/seq_device.c @@ -156,6 +156,8 @@ static int snd_seq_device_dev_free(struct snd_device *device) struct snd_seq_device *dev = device->device_data; cancel_autoload_drivers(); + if (dev->private_free) + dev->private_free(dev); put_device(&dev->dev); return 0; } @@ -183,11 +185,7 @@ static int snd_seq_device_dev_disconnect(struct snd_device *device) static void snd_seq_dev_release(struct device *dev) { - struct snd_seq_device *sdev = to_seq_dev(dev); - - if (sdev->private_free) - sdev->private_free(sdev); - kfree(sdev); + kfree(to_seq_dev(dev)); } /* diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c index ed40d0f7432c..773db4bf0876 100644 --- a/sound/drivers/pcsp/pcsp_lib.c +++ b/sound/drivers/pcsp/pcsp_lib.c @@ -143,7 +143,7 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) if (pointer_update) pcsp_pointer_update(chip); - hrtimer_forward(handle, hrtimer_get_expires(handle), ns_to_ktime(ns)); + hrtimer_forward_now(handle, ns_to_ktime(ns)); return HRTIMER_RESTART; } diff --git a/sound/firewire/motu/amdtp-motu.c b/sound/firewire/motu/amdtp-motu.c index 5388b85fb60e..a18c2c033e83 100644 --- a/sound/firewire/motu/amdtp-motu.c +++ b/sound/firewire/motu/amdtp-motu.c @@ -276,10 +276,11 @@ static void __maybe_unused copy_message(u64 *frames, __be32 *buffer, /* This is just for v2/v3 protocol. */ for (i = 0; i < data_blocks; ++i) { - *frames = (be32_to_cpu(buffer[1]) << 16) | - (be32_to_cpu(buffer[2]) >> 16); + *frames = be32_to_cpu(buffer[1]); + *frames <<= 16; + *frames |= be32_to_cpu(buffer[2]) >> 16; + ++frames; buffer += data_block_quadlets; - frames++; } } diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index cb5b5e3a481b..daf731364695 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -184,13 +184,16 @@ static int detect_quirks(struct snd_oxfw *oxfw, const struct ieee1394_device_id model = val; } - /* - * Mackie Onyx Satellite with base station has a quirk to report a wrong - * value in 'dbs' field of CIP header against its format information. - */ - if (vendor == VENDOR_LOUD && model == MODEL_SATELLITE) + if (vendor == VENDOR_LOUD) { + // Mackie Onyx Satellite with base station has a quirk to report a wrong + // value in 'dbs' field of CIP header against its format information. oxfw->quirks |= SND_OXFW_QUIRK_WRONG_DBS; + // OXFW971-based models may transfer events by blocking method. + if (!(oxfw->quirks & SND_OXFW_QUIRK_JUMBO_PAYLOAD)) + oxfw->quirks |= SND_OXFW_QUIRK_BLOCKING_TRANSMISSION; + } + return 0; } diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 062da7a7a586..f7bd6e2db085 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -421,8 +421,9 @@ int snd_hdac_bus_reset_link(struct hdac_bus *bus, bool full_reset) if (!full_reset) goto skip_reset; - /* clear STATESTS */ - snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); + /* clear STATESTS if not in reset */ + if (snd_hdac_chip_readb(bus, GCTL) & AZX_GCTL_RESET) + snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); /* reset controller */ snd_hdac_bus_enter_link_reset(bus); diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 2523b23389e9..1c8bffc3eec6 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -298,29 +298,31 @@ int snd_hda_codec_configure(struct hda_codec *codec) { int err; + if (codec->configured) + return 0; + if (is_generic_config(codec)) codec->probe_id = HDA_CODEC_ID_GENERIC; else codec->probe_id = 0; - err = snd_hdac_device_register(&codec->core); - if (err < 0) - return err; + if (!device_is_registered(&codec->core.dev)) { + err = snd_hdac_device_register(&codec->core); + if (err < 0) + return err; + } if (!codec->preset) codec_bind_module(codec); if (!codec->preset) { err = codec_bind_generic(codec); if (err < 0) { - codec_err(codec, "Unable to bind the codec\n"); - goto error; + codec_dbg(codec, "Unable to bind the codec\n"); + return err; } } + codec->configured = 1; return 0; - - error: - snd_hdac_device_unregister(&codec->core); - return err; } EXPORT_SYMBOL_GPL(snd_hda_codec_configure); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index a9ebefd60cf6..0c4a337c9fc0 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -791,6 +791,7 @@ void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec) snd_array_free(&codec->nids); remove_conn_list(codec); snd_hdac_regmap_exit(&codec->core); + codec->configured = 0; } EXPORT_SYMBOL_GPL(snd_hda_codec_cleanup_for_unbind); diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 7cd452831fd3..930ae4002a81 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -25,6 +25,7 @@ #include <sound/core.h> #include <sound/initval.h> #include "hda_controller.h" +#include "hda_local.h" #define CREATE_TRACE_POINTS #include "hda_controller_trace.h" @@ -1248,17 +1249,24 @@ EXPORT_SYMBOL_GPL(azx_probe_codecs); int azx_codec_configure(struct azx *chip) { struct hda_codec *codec, *next; + int success = 0; - /* use _safe version here since snd_hda_codec_configure() deregisters - * the device upon error and deletes itself from the bus list. - */ - list_for_each_codec_safe(codec, next, &chip->bus) { - snd_hda_codec_configure(codec); + list_for_each_codec(codec, &chip->bus) { + if (!snd_hda_codec_configure(codec)) + success++; } - if (!azx_bus(chip)->num_codecs) - return -ENODEV; - return 0; + if (success) { + /* unregister failed codecs if any codec has been probed */ + list_for_each_codec_safe(codec, next, &chip->bus) { + if (!codec->configured) { + codec_err(codec, "Unable to configure, disabling\n"); + snd_hdac_device_unregister(&codec->core); + } + } + } + + return success ? 0 : -ENODEV; } EXPORT_SYMBOL_GPL(azx_codec_configure); diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index 3062f87380b1..f5bf295eb830 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -41,7 +41,7 @@ /* 24 unused */ #define AZX_DCAPS_COUNT_LPIB_DELAY (1 << 25) /* Take LPIB as delay */ #define AZX_DCAPS_PM_RUNTIME (1 << 26) /* runtime PM support */ -/* 27 unused */ +#define AZX_DCAPS_RETRY_PROBE (1 << 27) /* retry probe if no codec is configured */ #define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */ #define AZX_DCAPS_NO_MSI64 (1 << 29) /* Stick to 32-bit MSIs */ #define AZX_DCAPS_SEPARATE_STREAM_TAG (1 << 30) /* capture and playback use separate stream tag */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 3aa432d814a2..4d22e7adeee8 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -307,7 +307,8 @@ enum { /* quirks for AMD SB */ #define AZX_DCAPS_PRESET_AMD_SB \ (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_AMD_WORKAROUND |\ - AZX_DCAPS_SNOOP_TYPE(ATI) | AZX_DCAPS_PM_RUNTIME) + AZX_DCAPS_SNOOP_TYPE(ATI) | AZX_DCAPS_PM_RUNTIME |\ + AZX_DCAPS_RETRY_PROBE) /* quirks for Nvidia */ #define AZX_DCAPS_PRESET_NVIDIA \ @@ -883,10 +884,11 @@ static unsigned int azx_get_pos_skl(struct azx *chip, struct azx_dev *azx_dev) return azx_get_pos_posbuf(chip, azx_dev); } -static void azx_shutdown_chip(struct azx *chip) +static void __azx_shutdown_chip(struct azx *chip, bool skip_link_reset) { azx_stop_chip(chip); - azx_enter_link_reset(chip); + if (!skip_link_reset) + azx_enter_link_reset(chip); azx_clear_irq_pending(chip); display_power(chip, false); } @@ -895,6 +897,11 @@ static void azx_shutdown_chip(struct azx *chip) static DEFINE_MUTEX(card_list_lock); static LIST_HEAD(card_list); +static void azx_shutdown_chip(struct azx *chip) +{ + __azx_shutdown_chip(chip, false); +} + static void azx_add_card_list(struct azx *chip) { struct hda_intel *hda = container_of(chip, struct hda_intel, chip); @@ -1717,7 +1724,7 @@ static void azx_check_snoop_available(struct azx *chip) static void azx_probe_work(struct work_struct *work) { - struct hda_intel *hda = container_of(work, struct hda_intel, probe_work); + struct hda_intel *hda = container_of(work, struct hda_intel, probe_work.work); azx_probe_continue(&hda->chip); } @@ -1822,7 +1829,7 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, } /* continue probing in work context as may trigger request module */ - INIT_WORK(&hda->probe_work, azx_probe_work); + INIT_DELAYED_WORK(&hda->probe_work, azx_probe_work); *rchip = chip; @@ -2136,7 +2143,7 @@ static int azx_probe(struct pci_dev *pci, #endif if (schedule_probe) - schedule_work(&hda->probe_work); + schedule_delayed_work(&hda->probe_work, 0); dev++; if (chip->disabled) @@ -2222,6 +2229,11 @@ static int azx_probe_continue(struct azx *chip) int dev = chip->dev_index; int err; + if (chip->disabled || hda->init_failed) + return -EIO; + if (hda->probe_retry) + goto probe_retry; + to_hda_bus(bus)->bus_probing = 1; hda->probe_continued = 1; @@ -2283,10 +2295,20 @@ static int azx_probe_continue(struct azx *chip) #endif } #endif + + probe_retry: if (bus->codec_mask && !(probe_only[dev] & 1)) { err = azx_codec_configure(chip); - if (err < 0) + if (err) { + if ((chip->driver_caps & AZX_DCAPS_RETRY_PROBE) && + ++hda->probe_retry < 60) { + schedule_delayed_work(&hda->probe_work, + msecs_to_jiffies(1000)); + return 0; /* keep things up */ + } + dev_err(chip->card->dev, "Cannot probe codecs, giving up\n"); goto out_free; + } } err = snd_card_register(chip->card); @@ -2316,6 +2338,7 @@ out_free: display_power(chip, false); complete_all(&hda->probe_wait); to_hda_bus(bus)->bus_probing = 0; + hda->probe_retry = 0; return 0; } @@ -2341,7 +2364,7 @@ static void azx_remove(struct pci_dev *pci) * device during cancel_work_sync() call. */ device_unlock(&pci->dev); - cancel_work_sync(&hda->probe_work); + cancel_delayed_work_sync(&hda->probe_work); device_lock(&pci->dev); snd_card_free(card); @@ -2357,7 +2380,7 @@ static void azx_shutdown(struct pci_dev *pci) return; chip = card->private_data; if (chip && chip->running) - azx_shutdown_chip(chip); + __azx_shutdown_chip(chip, true); } /* PCI IDs */ diff --git a/sound/pci/hda/hda_intel.h b/sound/pci/hda/hda_intel.h index 3fb119f09040..0f39418f9328 100644 --- a/sound/pci/hda/hda_intel.h +++ b/sound/pci/hda/hda_intel.h @@ -14,7 +14,7 @@ struct hda_intel { /* sync probing */ struct completion probe_wait; - struct work_struct probe_work; + struct delayed_work probe_work; /* card list (for power_save trigger) */ struct list_head list; @@ -30,6 +30,8 @@ struct hda_intel { unsigned int freed:1; /* resources already released */ bool need_i915_power:1; /* the hda controller needs i915 power */ + + int probe_retry; /* being probe-retry */ }; #endif diff --git a/sound/pci/hda/patch_cs8409.c b/sound/pci/hda/patch_cs8409.c index 3c7ef55d016e..31ff11ab868e 100644 --- a/sound/pci/hda/patch_cs8409.c +++ b/sound/pci/hda/patch_cs8409.c @@ -1207,6 +1207,9 @@ void dolphin_fixups(struct hda_codec *codec, const struct hda_fixup *fix, int ac snd_hda_jack_add_kctl(codec, DOLPHIN_LO_PIN_NID, "Line Out", true, SND_JACK_HEADPHONE, NULL); + snd_hda_jack_add_kctl(codec, DOLPHIN_AMIC_PIN_NID, "Microphone", true, + SND_JACK_MICROPHONE, NULL); + cs8409_fix_caps(codec, DOLPHIN_HP_PIN_NID); cs8409_fix_caps(codec, DOLPHIN_LO_PIN_NID); cs8409_fix_caps(codec, DOLPHIN_AMIC_PIN_NID); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8b7a389b6aed..965b096f416f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -526,6 +526,8 @@ static void alc_shutup_pins(struct hda_codec *codec) struct alc_spec *spec = codec->spec; switch (codec->core.vendor_id) { + case 0x10ec0236: + case 0x10ec0256: case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: @@ -2533,11 +2535,13 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x65d2, "Clevo PB51R[CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x65e1, "Clevo PB51[ED][DF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x65e5, "Clevo PC50D[PRS](?:-D|-G)?", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x65f1, "Clevo PC50HS", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67d1, "Clevo PB71[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67e1, "Clevo PB71[DE][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67e5, "Clevo PC70D[PRS](?:-D|-G)?", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x70d1, "Clevo PC70[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), - SND_PCI_QUIRK(0x1558, 0x7714, "Clevo X170", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x7714, "Clevo X170SM", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x7715, "Clevo X170KM-G", ALC1220_FIXUP_CLEVO_PB51ED), SND_PCI_QUIRK(0x1558, 0x9501, "Clevo P950HR", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1558, 0x9506, "Clevo P955HQ", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1558, 0x950a, "Clevo P955H[PR]", ALC1220_FIXUP_CLEVO_P950), @@ -3528,7 +3532,8 @@ static void alc256_shutup(struct hda_codec *codec) /* If disable 3k pulldown control for alc257, the Mic detection will not work correctly * when booting with headset plugged. So skip setting it for the codec alc257 */ - if (codec->core.vendor_id != 0x10ec0257) + if (spec->codec_variant != ALC269_TYPE_ALC257 && + spec->codec_variant != ALC269_TYPE_ALC256) alc_update_coef_idx(codec, 0x46, 0, 3 << 12); if (!spec->no_shutup_pins) @@ -6401,6 +6406,44 @@ static void alc_fixup_no_int_mic(struct hda_codec *codec, } } +/* GPIO1 = amplifier on/off + * GPIO3 = mic mute LED + */ +static void alc285_fixup_hp_spectre_x360_eb1(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static const hda_nid_t conn[] = { 0x02 }; + + struct alc_spec *spec = codec->spec; + static const struct hda_pintbl pincfgs[] = { + { 0x14, 0x90170110 }, /* front/high speakers */ + { 0x17, 0x90170130 }, /* back/bass speakers */ + { } + }; + + //enable micmute led + alc_fixup_hp_gpio_led(codec, action, 0x00, 0x04); + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->micmute_led_polarity = 1; + /* needed for amp of back speakers */ + spec->gpio_mask |= 0x01; + spec->gpio_dir |= 0x01; + snd_hda_apply_pincfgs(codec, pincfgs); + /* share DAC to have unified volume control */ + snd_hda_override_conn_list(codec, 0x14, ARRAY_SIZE(conn), conn); + snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn); + break; + case HDA_FIXUP_ACT_INIT: + /* need to toggle GPIO to enable the amp of back speakers */ + alc_update_gpio_data(codec, 0x01, true); + msleep(100); + alc_update_gpio_data(codec, 0x01, false); + break; + } +} + static void alc285_fixup_hp_spectre_x360(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -6429,12 +6472,44 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, hda_fixup_thinkpad_acpi(codec, fix, action); } +/* Fixup for Lenovo Legion 15IMHg05 speaker output on headset removal. */ +static void alc287_fixup_legion_15imhg05_speakers(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->gen.suppress_auto_mute = 1; + break; + } +} + /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" /* for alc285_fixup_ideapad_s740_coef() */ #include "ideapad_s740_helper.c" +static void alc256_fixup_tongfang_reset_persistent_settings(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + /* + * A certain other OS sets these coeffs to different values. On at least one TongFang + * barebone these settings might survive even a cold reboot. So to restore a clean slate the + * values are explicitly reset to default here. Without this, the external microphone is + * always in a plugged-in state, while the internal microphone is always in an unplugged + * state, breaking the ability to use the internal microphone. + */ + alc_write_coef_idx(codec, 0x24, 0x0000); + alc_write_coef_idx(codec, 0x26, 0x0000); + alc_write_coef_idx(codec, 0x29, 0x3000); + alc_write_coef_idx(codec, 0x37, 0xfe05); + alc_write_coef_idx(codec, 0x45, 0x5089); +} + enum { ALC269_FIXUP_GPIO2, ALC269_FIXUP_SONY_VAIO, @@ -6521,6 +6596,7 @@ enum { ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED, ALC280_FIXUP_HP_9480M, ALC245_FIXUP_HP_X360_AMP, + ALC285_FIXUP_HP_SPECTRE_X360_EB1, ALC288_FIXUP_DELL_HEADSET_MODE, ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, ALC288_FIXUP_DELL_XPS_13, @@ -6646,6 +6722,11 @@ enum { ALC623_FIXUP_LENOVO_THINKSTATION_P340, ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST, + ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS, + ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE, + ALC287_FIXUP_YOGA7_14ITL_SPEAKERS, + ALC287_FIXUP_13S_GEN2_SPEAKERS, + ALC256_FIXUP_TONGFANG_RESET_PERSISTENT_SETTINGS, }; static const struct hda_fixup alc269_fixups[] = { @@ -8209,6 +8290,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_hp_spectre_x360, }, + [ALC285_FIXUP_HP_SPECTRE_X360_EB1] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_hp_spectre_x360_eb1 + }, [ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_ideapad_s740_coef, @@ -8236,6 +8321,117 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF, }, + [ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS] = { + .type = HDA_FIXUP_VERBS, + //.v.verbs = legion_15imhg05_coefs, + .v.verbs = (const struct hda_verb[]) { + // set left speaker Legion 7i. + { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x41 }, + + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xc }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x1a }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + // set right speaker Legion 7i. + { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x42 }, + + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xc }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x2a }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {} + }, + .chained = true, + .chain_id = ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE, + }, + [ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc287_fixup_legion_15imhg05_speakers, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE, + }, + [ALC287_FIXUP_YOGA7_14ITL_SPEAKERS] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + // set left speaker Yoga 7i. + { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x41 }, + + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xc }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x1a }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + // set right speaker Yoga 7i. + { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x46 }, + + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xc }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x2a }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {} + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE, + }, + [ALC287_FIXUP_13S_GEN2_SPEAKERS] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x41 }, + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x42 }, + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {} + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE, + }, + [ALC256_FIXUP_TONGFANG_RESET_PERSISTENT_SETTINGS] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc256_fixup_tongfang_reset_persistent_settings, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8327,6 +8523,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a30, "Dell", ALC236_FIXUP_DELL_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0a58, "Dell", ALC255_FIXUP_DELL_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0a61, "Dell XPS 15 9510", ALC289_FIXUP_DUAL_SPK), + SND_PCI_QUIRK(0x1028, 0x0a62, "Dell Precision 5560", ALC289_FIXUP_DUAL_SPK), + SND_PCI_QUIRK(0x1028, 0x0a9d, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0a9e, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), @@ -8429,6 +8628,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), SND_PCI_QUIRK(0x103c, 0x8805, "HP ProBook 650 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x880d, "HP EliteBook 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8811, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), + SND_PCI_QUIRK(0x103c, 0x8812, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8847, "HP EliteBook x360 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884b, "HP EliteBook 840 Aero G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), @@ -8630,6 +8831,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME), SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF), SND_PCI_QUIRK(0x17aa, 0x3843, "Yoga 9i", ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP), + SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3852, "Lenovo Yoga 7 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3819, "Lenovo 13s Gen2 ITL", ALC287_FIXUP_13S_GEN2_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), @@ -8660,6 +8865,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802), SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X), + SND_PCI_QUIRK(0x1d05, 0x1132, "TongFang PHxTxX1", ALC256_FIXUP_TONGFANG_RESET_PERSISTENT_SETTINGS), SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), @@ -8845,6 +9051,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC245_FIXUP_HP_X360_AMP, .name = "alc245-hp-x360-amp"}, {.id = ALC295_FIXUP_HP_OMEN, .name = "alc295-hp-omen"}, {.id = ALC285_FIXUP_HP_SPECTRE_X360, .name = "alc285-hp-spectre-x360"}, + {.id = ALC285_FIXUP_HP_SPECTRE_X360_EB1, .name = "alc285-hp-spectre-x360-eb1"}, {.id = ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, .name = "alc287-ideapad-bass-spk-amp"}, {.id = ALC623_FIXUP_LENOVO_THINKSTATION_P340, .name = "alc623-lenovo-thinkstation-p340"}, {.id = ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, .name = "alc255-acer-headphone-and-mic"}, @@ -10037,6 +10244,9 @@ enum { ALC671_FIXUP_HP_HEADSET_MIC2, ALC662_FIXUP_ACER_X2660G_HEADSET_MODE, ALC662_FIXUP_ACER_NITRO_HEADSET_MODE, + ALC668_FIXUP_ASUS_NO_HEADSET_MIC, + ALC668_FIXUP_HEADSET_MIC, + ALC668_FIXUP_MIC_DET_COEF, }; static const struct hda_fixup alc662_fixups[] = { @@ -10420,6 +10630,29 @@ static const struct hda_fixup alc662_fixups[] = { .chained = true, .chain_id = ALC662_FIXUP_USI_FUNC }, + [ALC668_FIXUP_ASUS_NO_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1b, 0x04a1112c }, + { } + }, + .chained = true, + .chain_id = ALC668_FIXUP_HEADSET_MIC + }, + [ALC668_FIXUP_HEADSET_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_headset_mic, + .chained = true, + .chain_id = ALC668_FIXUP_MIC_DET_COEF + }, + [ALC668_FIXUP_MIC_DET_COEF] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0x15 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0d60 }, + {} + }, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -10455,6 +10688,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x15a7, "ASUS UX51VZH", ALC662_FIXUP_BASS_16), SND_PCI_QUIRK(0x1043, 0x177d, "ASUS N551", ALC668_FIXUP_ASUS_Nx51), SND_PCI_QUIRK(0x1043, 0x17bd, "ASUS N751", ALC668_FIXUP_ASUS_Nx51), + SND_PCI_QUIRK(0x1043, 0x185d, "ASUS G551JW", ALC668_FIXUP_ASUS_NO_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71SL", ALC662_FIXUP_ASUS_MODE8), SND_PCI_QUIRK(0x1043, 0x1b73, "ASUS N55SF", ALC662_FIXUP_BASS_16), SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c index 87d24224c042..23f253effb4f 100644 --- a/sound/pci/pcxhr/pcxhr_core.c +++ b/sound/pci/pcxhr/pcxhr_core.c @@ -52,7 +52,7 @@ #define PCXHR_DSP 2 #if (PCXHR_DSP_OFFSET_MAX > PCXHR_PLX_OFFSET_MIN) -#undef PCXHR_REG_TO_PORT(x) +#error PCXHR_REG_TO_PORT(x) #else #define PCXHR_REG_TO_PORT(x) ((x)>PCXHR_DSP_OFFSET_MAX ? PCXHR_PLX : PCXHR_DSP) #endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 82ee233a269d..216cea04ad70 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1583,6 +1583,7 @@ config SND_SOC_WCD938X_SDW tristate "WCD9380/WCD9385 Codec - SDW" select SND_SOC_WCD938X select SND_SOC_WCD_MBHC + select REGMAP_IRQ depends on SOUNDWIRE select REGMAP_SOUNDWIRE help diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index fb1e4c33e27d..9a463ab54bdd 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -922,7 +922,6 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) struct snd_soc_component *component = dai->component; struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); unsigned int regval; - u8 fullScaleVol; int ret; if (mute) { @@ -993,20 +992,11 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) cs42l42->stream_use |= 1 << stream; if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* Read the headphone load */ - regval = snd_soc_component_read(component, CS42L42_LOAD_DET_RCSTAT); - if (((regval & CS42L42_RLA_STAT_MASK) >> CS42L42_RLA_STAT_SHIFT) == - CS42L42_RLA_STAT_15_OHM) { - fullScaleVol = CS42L42_HP_FULL_SCALE_VOL_MASK; - } else { - fullScaleVol = 0; - } - - /* Un-mute the headphone, set the full scale volume flag */ + /* Un-mute the headphone */ snd_soc_component_update_bits(component, CS42L42_HP_CTL, CS42L42_HP_ANA_AMUTE_MASK | - CS42L42_HP_ANA_BMUTE_MASK | - CS42L42_HP_FULL_SCALE_VOL_MASK, fullScaleVol); + CS42L42_HP_ANA_BMUTE_MASK, + 0); } } diff --git a/sound/soc/codecs/cs4341.c b/sound/soc/codecs/cs4341.c index 7d3e54d8eef3..29d05e32d341 100644 --- a/sound/soc/codecs/cs4341.c +++ b/sound/soc/codecs/cs4341.c @@ -305,12 +305,19 @@ static int cs4341_spi_probe(struct spi_device *spi) return cs4341_probe(&spi->dev); } +static const struct spi_device_id cs4341_spi_ids[] = { + { "cs4341a" }, + { } +}; +MODULE_DEVICE_TABLE(spi, cs4341_spi_ids); + static struct spi_driver cs4341_spi_driver = { .driver = { .name = "cs4341-spi", .of_match_table = of_match_ptr(cs4341_dt_ids), }, .probe = cs4341_spi_probe, + .id_table = cs4341_spi_ids, }; #endif diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index db88be48c998..f946ef65a4c1 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -867,8 +867,8 @@ static void nau8824_jdet_work(struct work_struct *work) struct regmap *regmap = nau8824->regmap; int adc_value, event = 0, event_mask = 0; - snd_soc_dapm_enable_pin(dapm, "MICBIAS"); - snd_soc_dapm_enable_pin(dapm, "SAR"); + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin(dapm, "SAR"); snd_soc_dapm_sync(dapm); msleep(100); diff --git a/sound/soc/codecs/pcm179x-spi.c b/sound/soc/codecs/pcm179x-spi.c index 0a542924ec5f..ebf63ea90a1c 100644 --- a/sound/soc/codecs/pcm179x-spi.c +++ b/sound/soc/codecs/pcm179x-spi.c @@ -36,6 +36,7 @@ static const struct of_device_id pcm179x_of_match[] = { MODULE_DEVICE_TABLE(of, pcm179x_of_match); static const struct spi_device_id pcm179x_spi_ids[] = { + { "pcm1792a", 0 }, { "pcm179x", 0 }, { }, }; diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 4dc844f3c1fc..60dee41816dc 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -116,6 +116,8 @@ static const struct reg_default pcm512x_reg_defaults[] = { { PCM512x_FS_SPEED_MODE, 0x00 }, { PCM512x_IDAC_1, 0x01 }, { PCM512x_IDAC_2, 0x00 }, + { PCM512x_I2S_1, 0x02 }, + { PCM512x_I2S_2, 0x00 }, }; static bool pcm512x_readable(struct device *dev, unsigned int reg) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index f0daf8defcf1..52de7d14b139 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -4144,10 +4144,10 @@ static int wcd938x_codec_set_jack(struct snd_soc_component *comp, { struct wcd938x_priv *wcd = dev_get_drvdata(comp->dev); - if (!jack) + if (jack) return wcd_mbhc_start(wcd->wcd_mbhc, &wcd->mbhc_cfg, jack); - - wcd_mbhc_stop(wcd->wcd_mbhc); + else + wcd_mbhc_stop(wcd->wcd_mbhc); return 0; } diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 9e621a254392..499604f1e178 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -742,9 +742,16 @@ static int wm8960_configure_clocking(struct snd_soc_component *component) int i, j, k; int ret; - if (!(iface1 & (1<<6))) { - dev_dbg(component->dev, - "Codec is slave mode, no need to configure clock\n"); + /* + * For Slave mode clocking should still be configured, + * so this if statement should be removed, but some platform + * may not work if the sysclk is not configured, to avoid such + * compatible issue, just add '!wm8960->sysclk' condition in + * this if statement. + */ + if (!(iface1 & (1 << 6)) && !wm8960->sysclk) { + dev_warn(component->dev, + "slave mode, but proceeding with no clock configuration\n"); return 0; } diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index a961f837cd09..bda66b30e063 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -1073,6 +1073,16 @@ static int fsl_esai_probe(struct platform_device *pdev) if (ret < 0) goto err_pm_get_sync; + /* + * Register platform component before registering cpu dai for there + * is not defer probe for platform component in snd_soc_add_pcm_runtime(). + */ + ret = imx_pcm_dma_init(pdev, IMX_ESAI_DMABUF_SIZE); + if (ret) { + dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret); + goto err_pm_get_sync; + } + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_esai_component, &fsl_esai_dai, 1); if (ret) { @@ -1082,12 +1092,6 @@ static int fsl_esai_probe(struct platform_device *pdev) INIT_WORK(&esai_priv->work, fsl_esai_hw_reset); - ret = imx_pcm_dma_init(pdev, IMX_ESAI_DMABUF_SIZE); - if (ret) { - dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret); - goto err_pm_get_sync; - } - return ret; err_pm_get_sync: diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 8c0c75ce9490..9f90989ac59a 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -737,18 +737,23 @@ static int fsl_micfil_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); regcache_cache_only(micfil->regmap, true); + /* + * Register platform component before registering cpu dai for there + * is not defer probe for platform component in snd_soc_add_pcm_runtime(). + */ + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) { + dev_err(&pdev->dev, "failed to pcm register\n"); + return ret; + } + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_micfil_component, &fsl_micfil_dai, 1); if (ret) { dev_err(&pdev->dev, "failed to register component %s\n", fsl_micfil_component.name); - return ret; } - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); - if (ret) - dev_err(&pdev->dev, "failed to pcm register\n"); - return ret; } diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 223fcd15bfcc..38f6362099d5 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1152,11 +1152,10 @@ static int fsl_sai_probe(struct platform_device *pdev) if (ret < 0) goto err_pm_get_sync; - ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component, - &sai->cpu_dai_drv, 1); - if (ret) - goto err_pm_get_sync; - + /* + * Register platform component before registering cpu dai for there + * is not defer probe for platform component in snd_soc_add_pcm_runtime(). + */ if (sai->soc_data->use_imx_pcm) { ret = imx_pcm_dma_init(pdev, IMX_SAI_DMABUF_SIZE); if (ret) @@ -1167,6 +1166,11 @@ static int fsl_sai_probe(struct platform_device *pdev) goto err_pm_get_sync; } + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component, + &sai->cpu_dai_drv, 1); + if (ret) + goto err_pm_get_sync; + return ret; err_pm_get_sync: diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 8ffb1a6048d6..1c53719bb61e 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1434,16 +1434,20 @@ static int fsl_spdif_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); regcache_cache_only(spdif_priv->regmap, true); - ret = devm_snd_soc_register_component(&pdev->dev, &fsl_spdif_component, - &spdif_priv->cpu_dai_drv, 1); + /* + * Register platform component before registering cpu dai for there + * is not defer probe for platform component in snd_soc_add_pcm_runtime(). + */ + ret = imx_pcm_dma_init(pdev, IMX_SPDIF_DMABUF_SIZE); if (ret) { - dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); + dev_err_probe(&pdev->dev, ret, "imx_pcm_dma_init failed\n"); goto err_pm_disable; } - ret = imx_pcm_dma_init(pdev, IMX_SPDIF_DMABUF_SIZE); + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_spdif_component, + &spdif_priv->cpu_dai_drv, 1); if (ret) { - dev_err_probe(&pdev->dev, ret, "imx_pcm_dma_init failed\n"); + dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); goto err_pm_disable; } diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index 31c5ee641fe7..d0556c79fdb1 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -487,8 +487,9 @@ static int fsl_xcvr_prepare(struct snd_pcm_substream *substream, return ret; } - /* clear DPATH RESET */ + /* set DPATH RESET */ m_ctl |= FSL_XCVR_EXT_CTRL_DPTH_RESET(tx); + v_ctl |= FSL_XCVR_EXT_CTRL_DPTH_RESET(tx); ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_CTRL, m_ctl, v_ctl); if (ret < 0) { dev_err(dai->dev, "Error while setting EXT_CTRL: %d\n", ret); @@ -590,10 +591,6 @@ static void fsl_xcvr_shutdown(struct snd_pcm_substream *substream, val |= FSL_XCVR_EXT_CTRL_CMDC_RESET(tx); } - /* set DPATH RESET */ - mask |= FSL_XCVR_EXT_CTRL_DPTH_RESET(tx); - val |= FSL_XCVR_EXT_CTRL_DPTH_RESET(tx); - ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_CTRL, mask, val); if (ret < 0) { dev_err(dai->dev, "Err setting DPATH RESET: %d\n", ret); @@ -643,6 +640,16 @@ static int fsl_xcvr_trigger(struct snd_pcm_substream *substream, int cmd, dev_err(dai->dev, "Failed to enable DMA: %d\n", ret); return ret; } + + /* clear DPATH RESET */ + ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_CTRL, + FSL_XCVR_EXT_CTRL_DPTH_RESET(tx), + 0); + if (ret < 0) { + dev_err(dai->dev, "Failed to clear DPATH RESET: %d\n", ret); + return ret; + } + break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: @@ -1215,18 +1222,23 @@ static int fsl_xcvr_probe(struct platform_device *pdev) pm_runtime_enable(dev); regcache_cache_only(xcvr->regmap, true); + /* + * Register platform component before registering cpu dai for there + * is not defer probe for platform component in snd_soc_add_pcm_runtime(). + */ + ret = devm_snd_dmaengine_pcm_register(dev, NULL, 0); + if (ret) { + dev_err(dev, "failed to pcm register\n"); + return ret; + } + ret = devm_snd_soc_register_component(dev, &fsl_xcvr_comp, &fsl_xcvr_dai, 1); if (ret) { dev_err(dev, "failed to register component %s\n", fsl_xcvr_comp.name); - return ret; } - ret = devm_snd_dmaengine_pcm_register(dev, NULL, 0); - if (ret) - dev_err(dev, "failed to pcm register\n"); - return ret; } diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 055248f104b2..4d313d0d0f23 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -456,12 +456,12 @@ static const struct dmi_system_id byt_cht_es8316_quirk_table[] = { static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) { + struct device *dev = &pdev->dev; static const char * const mic_name[] = { "in1", "in2" }; + struct snd_soc_acpi_mach *mach = dev_get_platdata(dev); struct property_entry props[MAX_NO_PROPS] = {}; struct byt_cht_es8316_private *priv; const struct dmi_system_id *dmi_id; - struct device *dev = &pdev->dev; - struct snd_soc_acpi_mach *mach; struct fwnode_handle *fwnode; const char *platform_name; struct acpi_device *adev; @@ -476,7 +476,6 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) if (!priv) return -ENOMEM; - mach = dev->platform_data; /* fix index of codec dai */ for (i = 0; i < ARRAY_SIZE(byt_cht_es8316_dais); i++) { if (!strcmp(byt_cht_es8316_dais[i].codecs->name, @@ -494,7 +493,7 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) put_device(&adev->dev); byt_cht_es8316_dais[dai_index].codecs->name = codec_name; } else { - dev_err(&pdev->dev, "Error cannot find '%s' dev\n", mach->id); + dev_err(dev, "Error cannot find '%s' dev\n", mach->id); return -ENXIO; } @@ -533,11 +532,8 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) /* get the clock */ priv->mclk = devm_clk_get(dev, "pmc_plt_clk_3"); - if (IS_ERR(priv->mclk)) { - ret = PTR_ERR(priv->mclk); - dev_err(dev, "clk_get pmc_plt_clk_3 failed: %d\n", ret); - return ret; - } + if (IS_ERR(priv->mclk)) + return dev_err_probe(dev, PTR_ERR(priv->mclk), "clk_get pmc_plt_clk_3 failed\n"); /* get speaker enable GPIO */ codec_dev = acpi_get_first_physical_node(adev); @@ -567,22 +563,13 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) devm_acpi_dev_add_driver_gpios(codec_dev, byt_cht_es8316_gpios); priv->speaker_en_gpio = - gpiod_get_index(codec_dev, "speaker-enable", 0, - /* see comment in byt_cht_es8316_resume */ - GPIOD_OUT_LOW | GPIOD_FLAGS_BIT_NONEXCLUSIVE); - + gpiod_get_optional(codec_dev, "speaker-enable", + /* see comment in byt_cht_es8316_resume() */ + GPIOD_OUT_LOW | GPIOD_FLAGS_BIT_NONEXCLUSIVE); if (IS_ERR(priv->speaker_en_gpio)) { - ret = PTR_ERR(priv->speaker_en_gpio); - switch (ret) { - case -ENOENT: - priv->speaker_en_gpio = NULL; - break; - default: - dev_err(dev, "get speaker GPIO failed: %d\n", ret); - fallthrough; - case -EPROBE_DEFER: - goto err_put_codec; - } + ret = dev_err_probe(dev, PTR_ERR(priv->speaker_en_gpio), + "get speaker GPIO failed\n"); + goto err_put_codec; } snprintf(components_string, sizeof(components_string), @@ -597,7 +584,7 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) byt_cht_es8316_card.long_name = long_name; #endif - sof_parent = snd_soc_acpi_sof_parent(&pdev->dev); + sof_parent = snd_soc_acpi_sof_parent(dev); /* set card and driver name */ if (sof_parent) { diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 6602eda89e8e..6b06248a9327 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -929,6 +929,11 @@ static int create_sdw_dailink(struct snd_soc_card *card, cpus + *cpu_id, cpu_dai_num, codecs, codec_num, NULL, &sdw_ops); + /* + * SoundWire DAILINKs use 'stream' functions and Bank Switch operations + * based on wait_for_completion(), tag them as 'nonatomic'. + */ + dai_links[*be_index].nonatomic = true; ret = set_codec_init_func(card, link, dai_links + (*be_index)++, playback, group_id); diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 5a2f4667d50b..81ad2dcee9eb 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -1,6 +1,7 @@ # SPDX-License-Identifier: GPL-2.0-only config SND_SOC_MEDIATEK tristate + select REGMAP_MMIO config SND_SOC_MT2701 tristate "ASoC support for Mediatek MT2701 chip" @@ -188,7 +189,9 @@ config SND_SOC_MT8192_MT6359_RT1015_RT5682 config SND_SOC_MT8195 tristate "ASoC support for Mediatek MT8195 chip" depends on ARCH_MEDIATEK || COMPILE_TEST + depends on COMMON_CLK select SND_SOC_MEDIATEK + select MFD_SYSCON if SND_SOC_MT6359 help This adds ASoC platform driver support for Mediatek MT8195 chip that can be used with other codecs. diff --git a/sound/soc/mediatek/common/mtk-afe-fe-dai.c b/sound/soc/mediatek/common/mtk-afe-fe-dai.c index baaa5881b1d4..e95c7c018e7d 100644 --- a/sound/soc/mediatek/common/mtk-afe-fe-dai.c +++ b/sound/soc/mediatek/common/mtk-afe-fe-dai.c @@ -334,9 +334,11 @@ int mtk_afe_suspend(struct snd_soc_component *component) devm_kcalloc(dev, afe->reg_back_up_list_num, sizeof(unsigned int), GFP_KERNEL); - for (i = 0; i < afe->reg_back_up_list_num; i++) - regmap_read(regmap, afe->reg_back_up_list[i], - &afe->reg_back_up[i]); + if (afe->reg_back_up) { + for (i = 0; i < afe->reg_back_up_list_num; i++) + regmap_read(regmap, afe->reg_back_up_list[i], + &afe->reg_back_up[i]); + } afe->suspended = true; afe->runtime_suspend(dev); @@ -356,12 +358,13 @@ int mtk_afe_resume(struct snd_soc_component *component) afe->runtime_resume(dev); - if (!afe->reg_back_up) + if (!afe->reg_back_up) { dev_dbg(dev, "%s no reg_backup\n", __func__); - - for (i = 0; i < afe->reg_back_up_list_num; i++) - mtk_regmap_write(regmap, afe->reg_back_up_list[i], - afe->reg_back_up[i]); + } else { + for (i = 0; i < afe->reg_back_up_list_num; i++) + mtk_regmap_write(regmap, afe->reg_back_up_list[i], + afe->reg_back_up[i]); + } afe->suspended = false; return 0; diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c index c97ace7387b4..de09f67c0450 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c @@ -424,8 +424,8 @@ static int mt8195_hdmi_codec_init(struct snd_soc_pcm_runtime *rtd) return snd_soc_component_set_jack(cmpnt_codec, &priv->hdmi_jack, NULL); } -static int mt8195_hdmitx_dptx_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) +static int mt8195_dptx_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) { /* fix BE i2s format to 32bit, clean param mask first */ @@ -902,7 +902,7 @@ static struct snd_soc_dai_link mt8195_mt6359_rt1019_rt5682_dai_links[] = { .no_pcm = 1, .dpcm_playback = 1, .ops = &mt8195_dptx_ops, - .be_hw_params_fixup = mt8195_hdmitx_dptx_hw_params_fixup, + .be_hw_params_fixup = mt8195_dptx_hw_params_fixup, SND_SOC_DAILINK_REG(DPTX_BE), }, [DAI_LINK_ETDM1_IN_BE] = { @@ -953,7 +953,6 @@ static struct snd_soc_dai_link mt8195_mt6359_rt1019_rt5682_dai_links[] = { SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .dpcm_playback = 1, - .be_hw_params_fixup = mt8195_hdmitx_dptx_hw_params_fixup, SND_SOC_DAILINK_REG(ETDM3_OUT_BE), }, [DAI_LINK_PCM1_BE] = { diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c830e96afba2..80ca260595fd 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2599,6 +2599,7 @@ int snd_soc_component_initialize(struct snd_soc_component *component, INIT_LIST_HEAD(&component->dai_list); INIT_LIST_HEAD(&component->dobj_list); INIT_LIST_HEAD(&component->card_list); + INIT_LIST_HEAD(&component->list); mutex_init(&component->io_mutex); component->name = fmt_single_name(dev, &component->id); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7b67f1e19ae9..59d07648a7e7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2561,6 +2561,7 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, const char *pin, int status) { struct snd_soc_dapm_widget *w = dapm_find_widget(dapm, pin, true); + int ret = 0; dapm_assert_locked(dapm); @@ -2573,13 +2574,14 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, dapm_mark_dirty(w, "pin configuration"); dapm_widget_invalidate_input_paths(w); dapm_widget_invalidate_output_paths(w); + ret = 1; } w->connected = status; if (status == 0) w->force = 0; - return 0; + return ret; } /** @@ -3583,14 +3585,15 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, { struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); const char *pin = (const char *)kcontrol->private_value; + int ret; if (ucontrol->value.integer.value[0]) - snd_soc_dapm_enable_pin(&card->dapm, pin); + ret = snd_soc_dapm_enable_pin(&card->dapm, pin); else - snd_soc_dapm_disable_pin(&card->dapm, pin); + ret = snd_soc_dapm_disable_pin(&card->dapm, pin); snd_soc_dapm_sync(&card->dapm); - return 0; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch); @@ -4023,7 +4026,7 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, rtd->params_select = ucontrol->value.enumerated.item[0]; - return 0; + return 1; } static void diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 3e4dd4a86363..59d0d7b2b55c 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -371,7 +371,6 @@ int snd_sof_device_remove(struct device *dev) dev_warn(dev, "error: %d failed to prepare DSP for device removal", ret); - snd_sof_fw_unload(sdev); snd_sof_ipc_free(sdev); snd_sof_free_debug(sdev); snd_sof_free_trace(sdev); @@ -394,8 +393,7 @@ int snd_sof_device_remove(struct device *dev) snd_sof_remove(sdev); /* release firmware */ - release_firmware(pdata->fw); - pdata->fw = NULL; + snd_sof_fw_unload(sdev); return 0; } diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index 12fedf0984bd..7e9723a10d02 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -365,7 +365,14 @@ static int imx8_remove(struct snd_sof_dev *sdev) /* on i.MX8 there is 1 to 1 match between type and BAR idx */ static int imx8_get_bar_index(struct snd_sof_dev *sdev, u32 type) { - return type; + /* Only IRAM and SRAM bars are valid */ + switch (type) { + case SOF_FW_BLK_TYPE_IRAM: + case SOF_FW_BLK_TYPE_SRAM: + return type; + default: + return -EINVAL; + } } static void imx8_ipc_msg_data(struct snd_sof_dev *sdev, diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c index cb822d953767..892e1482f97f 100644 --- a/sound/soc/sof/imx/imx8m.c +++ b/sound/soc/sof/imx/imx8m.c @@ -228,7 +228,14 @@ static int imx8m_remove(struct snd_sof_dev *sdev) /* on i.MX8 there is 1 to 1 match between type and BAR idx */ static int imx8m_get_bar_index(struct snd_sof_dev *sdev, u32 type) { - return type; + /* Only IRAM and SRAM bars are valid */ + switch (type) { + case SOF_FW_BLK_TYPE_IRAM: + case SOF_FW_BLK_TYPE_SRAM: + return type; + default: + return -EINVAL; + } } static void imx8m_ipc_msg_data(struct snd_sof_dev *sdev, diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c index 2b38a77cd594..bb79c77775b3 100644 --- a/sound/soc/sof/loader.c +++ b/sound/soc/sof/loader.c @@ -729,10 +729,10 @@ int snd_sof_load_firmware_raw(struct snd_sof_dev *sdev) ret = request_firmware(&plat_data->fw, fw_filename, sdev->dev); if (ret < 0) { - dev_err(sdev->dev, "error: request firmware %s failed err: %d\n", - fw_filename, ret); dev_err(sdev->dev, - "you may need to download the firmware from https://github.com/thesofproject/sof-bin/\n"); + "error: sof firmware file is missing, you might need to\n"); + dev_err(sdev->dev, + " download it from https://github.com/thesofproject/sof-bin/\n"); goto err; } else { dev_dbg(sdev->dev, "request_firmware %s successful\n", @@ -880,5 +880,7 @@ EXPORT_SYMBOL(snd_sof_run_firmware); void snd_sof_fw_unload(struct snd_sof_dev *sdev) { /* TODO: support module unloading at runtime */ + release_firmware(sdev->pdata->fw); + sdev->pdata->fw = NULL; } EXPORT_SYMBOL(snd_sof_fw_unload); diff --git a/sound/soc/sof/trace.c b/sound/soc/sof/trace.c index f72a6e83e6af..58f6ca5cf491 100644 --- a/sound/soc/sof/trace.c +++ b/sound/soc/sof/trace.c @@ -530,7 +530,6 @@ void snd_sof_trace_notify_for_error(struct snd_sof_dev *sdev) return; if (sdev->dtrace_is_enabled) { - dev_err(sdev->dev, "error: waking up any trace sleepers\n"); sdev->dtrace_error = true; wake_up(&sdev->trace_sleep); } diff --git a/sound/soc/sof/xtensa/core.c b/sound/soc/sof/xtensa/core.c index bbb9a2282ed9..f6e3411b33cf 100644 --- a/sound/soc/sof/xtensa/core.c +++ b/sound/soc/sof/xtensa/core.c @@ -122,9 +122,9 @@ static void xtensa_stack(struct snd_sof_dev *sdev, void *oops, u32 *stack, * 0x0049fbb0: 8000f2d0 0049fc00 6f6c6c61 00632e63 */ for (i = 0; i < stack_words; i += 4) { - hex_dump_to_buffer(stack + i * 4, 16, 16, 4, + hex_dump_to_buffer(stack + i, 16, 16, 4, buf, sizeof(buf), false); - dev_err(sdev->dev, "0x%08x: %s\n", stack_ptr + i, buf); + dev_err(sdev->dev, "0x%08x: %s\n", stack_ptr + i * 4, buf); } } diff --git a/sound/usb/card.c b/sound/usb/card.c index fd570a42f043..1764b9302d46 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -1054,7 +1054,7 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) return 0; } -static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) +static int usb_audio_resume(struct usb_interface *intf) { struct snd_usb_audio *chip = usb_get_intfdata(intf); struct snd_usb_stream *as; @@ -1080,7 +1080,7 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) * we just notify and restart the mixers */ list_for_each_entry(mixer, &chip->mixer_list, list) { - err = snd_usb_mixer_resume(mixer, reset_resume); + err = snd_usb_mixer_resume(mixer); if (err < 0) goto err_out; } @@ -1100,20 +1100,10 @@ err_out: atomic_dec(&chip->active); /* allow autopm after this point */ return err; } - -static int usb_audio_resume(struct usb_interface *intf) -{ - return __usb_audio_resume(intf, false); -} - -static int usb_audio_reset_resume(struct usb_interface *intf) -{ - return __usb_audio_resume(intf, true); -} #else #define usb_audio_suspend NULL #define usb_audio_resume NULL -#define usb_audio_reset_resume NULL +#define usb_audio_resume NULL #endif /* CONFIG_PM */ static const struct usb_device_id usb_audio_ids [] = { @@ -1135,7 +1125,7 @@ static struct usb_driver usb_audio_driver = { .disconnect = usb_audio_disconnect, .suspend = usb_audio_suspend, .resume = usb_audio_resume, - .reset_resume = usb_audio_reset_resume, + .reset_resume = usb_audio_resume, .id_table = usb_audio_ids, .supports_autosuspend = 1, }; diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 43bc59575a6e..8e030b1c061a 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1198,6 +1198,13 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, cval->res = 1; } break; + case USB_ID(0x1224, 0x2a25): /* Jieli Technology USB PHY 2.0 */ + if (!strcmp(kctl->id.name, "Mic Capture Volume")) { + usb_audio_info(chip, + "set resolution quirk: cval->res = 16\n"); + cval->res = 16; + } + break; } } @@ -3653,33 +3660,16 @@ static int restore_mixer_value(struct usb_mixer_elem_list *list) return 0; } -static int default_mixer_reset_resume(struct usb_mixer_elem_list *list) -{ - int err; - - if (list->resume) { - err = list->resume(list); - if (err < 0) - return err; - } - return restore_mixer_value(list); -} - -int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume) +int snd_usb_mixer_resume(struct usb_mixer_interface *mixer) { struct usb_mixer_elem_list *list; - usb_mixer_elem_resume_func_t f; int id, err; /* restore cached mixer values */ for (id = 0; id < MAX_ID_ELEMS; id++) { for_each_mixer_elem(list, mixer, id) { - if (reset_resume) - f = list->reset_resume; - else - f = list->resume; - if (f) { - err = f(list); + if (list->resume) { + err = list->resume(list); if (err < 0) return err; } @@ -3700,7 +3690,6 @@ void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list, list->id = unitid; list->dump = snd_usb_mixer_dump_cval; #ifdef CONFIG_PM - list->resume = NULL; - list->reset_resume = default_mixer_reset_resume; + list->resume = restore_mixer_value; #endif } diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 876bbc9a71ad..98ea24d91d80 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -70,7 +70,6 @@ struct usb_mixer_elem_list { bool is_std_info; usb_mixer_elem_dump_func_t dump; usb_mixer_elem_resume_func_t resume; - usb_mixer_elem_resume_func_t reset_resume; }; /* iterate over mixer element list of the given unit id */ @@ -121,7 +120,7 @@ int snd_usb_mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, #ifdef CONFIG_PM int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer); -int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume); +int snd_usb_mixer_resume(struct usb_mixer_interface *mixer); #endif int snd_usb_set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index a66ce0375fd9..46082dc57be0 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -151,7 +151,7 @@ static int add_single_ctl_with_resume(struct usb_mixer_interface *mixer, *listp = list; list->mixer = mixer; list->id = id; - list->reset_resume = resume; + list->resume = resume; kctl = snd_ctl_new1(knew, list); if (!kctl) { kfree(list); diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 3d5848d5481b..53ebabf42472 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -2450,6 +2450,8 @@ static int scarlett2_update_monitor_other(struct usb_mixer_interface *mixer) err = scarlett2_usb_get_config(mixer, SCARLETT2_CONFIG_TALKBACK_MAP, 1, &bitmap); + if (err < 0) + return err; for (i = 0; i < num_mixes; i++, bitmap >>= 1) private->talkback_map[i] = bitmap & 1; } diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index e03043f7dad3..2af8c68fac27 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -78,6 +78,48 @@ { USB_DEVICE_VENDOR_SPEC(0x041e, 0x3f19) }, /* + * Creative Technology, Ltd Live! Cam Sync HD [VF0770] + * The device advertises 8 formats, but only a rate of 48kHz is honored by the + * hardware and 24 bits give chopped audio, so only report the one working + * combination. + */ +{ + USB_DEVICE(0x041e, 0x4095), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, + { + .ifnum = 3, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels = 2, + .fmt_bits = 16, + .iface = 3, + .altsetting = 4, + .altset_idx = 4, + .endpoint = 0x82, + .ep_attr = 0x05, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 48000 }, + }, + }, + { + .ifnum = -1 + }, + }, + }, +}, + +/* * HP Wireless Audio * When not ignored, causes instability issues for some users, forcing them to * skip the entire module. @@ -3970,6 +4012,38 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* + * Sennheiser GSP670 + * Change order of interfaces loaded + */ + USB_DEVICE(0x1395, 0x0300), + .bInterfaceClass = USB_CLASS_PER_INTERFACE, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + // Communication + { + .ifnum = 3, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + // Recording + { + .ifnum = 4, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + // Main + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, #undef USB_DEVICE_VENDOR_SPEC #undef USB_AUDIO_DEVICE diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 6ee6d24c847f..8929d9abe8aa 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1719,6 +1719,11 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, */ fp->attributes &= ~UAC_EP_CS_ATTR_FILL_MAX; break; + case USB_ID(0x1224, 0x2a25): /* Jieli Technology USB PHY 2.0 */ + /* mic works only when ep packet size is set to wMaxPacketSize */ + fp->attributes |= UAC_EP_CS_ATTR_FILL_MAX; + break; + } } @@ -1884,10 +1889,14 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x2912, 0x30c8, /* Audioengine D1 */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x30be, 0x0101, /* Schiit Hel */ + QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x413c, 0xa506, /* Dell AE515 sound bar */ QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x534d, 0x2109, /* MacroSilicon MS2109 */ QUIRK_FLAG_ALIGN_TRANSFER), + DEVICE_FLG(0x1224, 0x2a25, /* Jieli Technology USB PHY 2.0 */ + QUIRK_FLAG_GET_SAMPLE_RATE), /* Vendor matches */ VENDOR_FLG(0x045e, /* MS Lifecam */ @@ -1900,6 +1909,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_CTL_MSG_DELAY | QUIRK_FLAG_IFACE_DELAY), VENDOR_FLG(0x07fd, /* MOTU */ QUIRK_FLAG_VALIDATE_RATES), + VENDOR_FLG(0x1235, /* Focusrite Novation */ + QUIRK_FLAG_VALIDATE_RATES), VENDOR_FLG(0x152a, /* Thesycon devices */ QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x1de7, /* Phoenix Audio */ |