diff options
Diffstat (limited to 'sound')
46 files changed, 723 insertions, 426 deletions
diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 59d62f05658f..1545f8fdb4db 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -205,13 +205,14 @@ static snd_pcm_sframes_t calc_dst_frames(struct snd_pcm_substream *plug, plugin = snd_pcm_plug_first(plug); while (plugin && frames > 0) { plugin_next = plugin->next; + if (check_size && plugin->buf_frames && + frames > plugin->buf_frames) + frames = plugin->buf_frames; if (plugin->dst_frames) { frames = plugin->dst_frames(plugin, frames); if (frames < 0) return frames; } - if (check_size && frames > plugin->buf_frames) - frames = plugin->buf_frames; plugin = plugin_next; } return frames; @@ -225,14 +226,15 @@ static snd_pcm_sframes_t calc_src_frames(struct snd_pcm_substream *plug, plugin = snd_pcm_plug_last(plug); while (plugin && frames > 0) { - if (check_size && frames > plugin->buf_frames) - frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) { frames = plugin->src_frames(plugin, frames); if (frames < 0) return frames; } + if (check_size && plugin->buf_frames && + frames > plugin->buf_frames) + frames = plugin->buf_frames; plugin = plugin_prev; } return frames; diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index 4ca6b09056f3..3bc9224d5e4f 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -21,16 +21,17 @@ config SND_HDA_EXT_CORE select SND_HDA_CORE config SND_HDA_PREALLOC_SIZE - int "Pre-allocated buffer size for HD-audio driver" if !SND_DMA_SGBUF + int "Pre-allocated buffer size for HD-audio driver" range 0 32768 - default 0 if SND_DMA_SGBUF + default 2048 if SND_DMA_SGBUF default 64 if !SND_DMA_SGBUF help Specifies the default pre-allocated buffer-size in kB for the HD-audio driver. A larger buffer (e.g. 2048) is preferred for systems using PulseAudio. The default 64 is chosen just for compatibility reasons. - On x86 systems, the default is zero as we need no preallocation. + On x86 systems, the default is 2048 as a reasonable value for + most of modern systems. Note that the pre-allocation size can be changed dynamically via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too. diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index e764816a8f7a..b039429e6871 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -867,10 +867,13 @@ static void snd_miro_write(struct snd_miro *chip, unsigned char reg, spin_unlock_irqrestore(&chip->lock, flags); } +static inline void snd_miro_write_mask(struct snd_miro *chip, + unsigned char reg, unsigned char value, unsigned char mask) +{ + unsigned char oldval = snd_miro_read(chip, reg); -#define snd_miro_write_mask(chip, reg, value, mask) \ - snd_miro_write(chip, reg, \ - (snd_miro_read(chip, reg) & ~(mask)) | ((value) & (mask))) + snd_miro_write(chip, reg, (oldval & ~mask) | (value & mask)); +} /* * Proc Interface diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index d06b29693c85..0e6d20e49158 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -317,10 +317,13 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, } -#define snd_opti9xx_write_mask(chip, reg, value, mask) \ - snd_opti9xx_write(chip, reg, \ - (snd_opti9xx_read(chip, reg) & ~(mask)) | ((value) & (mask))) +static inline void snd_opti9xx_write_mask(struct snd_opti9xx *chip, + unsigned char reg, unsigned char value, unsigned char mask) +{ + unsigned char oldval = snd_opti9xx_read(chip, reg); + snd_opti9xx_write(chip, reg, (oldval & ~mask) | (value & mask)); +} static int snd_opti9xx_configure(struct snd_opti9xx *chip, long port, diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index 6e3177bcc709..015c0d676897 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -168,7 +168,7 @@ static int src_get_rsc_ctrl_blk(void **rblk) static int src_put_rsc_ctrl_blk(void *blk) { - kfree((struct src_rsc_ctrl_blk *)blk); + kfree(blk); return 0; } @@ -494,7 +494,7 @@ static int src_mgr_get_ctrl_blk(void **rblk) static int src_mgr_put_ctrl_blk(void *blk) { - kfree((struct src_mgr_ctrl_blk *)blk); + kfree(blk); return 0; } @@ -515,7 +515,7 @@ static int srcimp_mgr_get_ctrl_blk(void **rblk) static int srcimp_mgr_put_ctrl_blk(void *blk) { - kfree((struct srcimp_mgr_ctrl_blk *)blk); + kfree(blk); return 0; } @@ -702,7 +702,7 @@ static int amixer_rsc_get_ctrl_blk(void **rblk) static int amixer_rsc_put_ctrl_blk(void *blk) { - kfree((struct amixer_rsc_ctrl_blk *)blk); + kfree(blk); return 0; } @@ -909,7 +909,7 @@ static int dai_get_ctrl_blk(void **rblk) static int dai_put_ctrl_blk(void *blk) { - kfree((struct dai_ctrl_blk *)blk); + kfree(blk); return 0; } @@ -958,7 +958,7 @@ static int dao_get_ctrl_blk(void **rblk) static int dao_put_ctrl_blk(void *blk) { - kfree((struct dao_ctrl_blk *)blk); + kfree(blk); return 0; } @@ -1156,7 +1156,7 @@ static int daio_mgr_get_ctrl_blk(struct hw *hw, void **rblk) static int daio_mgr_put_ctrl_blk(void *blk) { - kfree((struct daio_mgr_ctrl_blk *)blk); + kfree(blk); return 0; } diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index a34a2c9f4bcf..7e3ae4534df9 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -641,8 +641,18 @@ static void hda_jackpoll_work(struct work_struct *work) struct hda_codec *codec = container_of(work, struct hda_codec, jackpoll_work.work); - snd_hda_jack_set_dirty_all(codec); - snd_hda_jack_poll_all(codec); + /* for non-polling trigger: we need nothing if already powered on */ + if (!codec->jackpoll_interval && snd_hdac_is_power_on(&codec->core)) + return; + + /* the power-up/down sequence triggers the runtime resume */ + snd_hda_power_up_pm(codec); + /* update jacks manually if polling is required, too */ + if (codec->jackpoll_interval) { + snd_hda_jack_set_dirty_all(codec); + snd_hda_jack_poll_all(codec); + } + snd_hda_power_down_pm(codec); if (!codec->jackpoll_interval) return; @@ -2951,18 +2961,14 @@ static int hda_codec_runtime_resume(struct device *dev) static int hda_codec_force_resume(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); - bool forced_resume = !codec->relaxed_resume && codec->jacktbl.used; int ret; - /* The get/put pair below enforces the runtime resume even if the - * device hasn't been used at suspend time. This trick is needed to - * update the jack state change during the sleep. - */ - if (forced_resume) - pm_runtime_get_noresume(dev); ret = pm_runtime_force_resume(dev); - if (forced_resume) - pm_runtime_put(dev); + /* schedule jackpoll work for jack detection update */ + if (codec->jackpoll_interval || + (pm_runtime_suspended(dev) && hda_codec_need_resume(codec))) + schedule_delayed_work(&codec->jackpoll_work, + codec->jackpoll_interval); return ret; } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index bd093593f8fb..0310193ea1bd 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1004,7 +1004,8 @@ static void __azx_runtime_resume(struct azx *chip, bool from_rt) if (status && from_rt) { list_for_each_codec(codec, &chip->bus) - if (status & (1 << codec->addr)) + if (!codec->relaxed_resume && + (status & (1 << codec->addr))) schedule_delayed_work(&codec->jackpoll_work, codec->jackpoll_interval); } @@ -1027,7 +1028,7 @@ static int azx_suspend(struct device *dev) chip = card->private_data; bus = azx_bus(chip); snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - __azx_runtime_suspend(chip); + pm_runtime_force_suspend(dev); if (bus->irq >= 0) { free_irq(bus->irq, chip); bus->irq = -1; @@ -1055,7 +1056,8 @@ static int azx_resume(struct device *dev) chip->msi = 0; if (azx_acquire_irq(chip, 1) < 0) return -EIO; - __azx_runtime_resume(chip, false); + + pm_runtime_force_resume(dev); snd_power_change_state(card, SNDRV_CTL_POWER_D0); trace_azx_resume(chip); @@ -1071,6 +1073,8 @@ static int azx_freeze_noirq(struct device *dev) struct azx *chip = card->private_data; struct pci_dev *pci = to_pci_dev(dev); + if (!azx_is_pm_ready(card)) + return 0; if (chip->driver_type == AZX_DRIVER_SKL) pci_set_power_state(pci, PCI_D3hot); @@ -1083,6 +1087,8 @@ static int azx_thaw_noirq(struct device *dev) struct azx *chip = card->private_data; struct pci_dev *pci = to_pci_dev(dev); + if (!azx_is_pm_ready(card)) + return 0; if (chip->driver_type == AZX_DRIVER_SKL) pci_set_power_state(pci, PCI_D0); @@ -1098,12 +1104,12 @@ static int azx_runtime_suspend(struct device *dev) if (!azx_is_pm_ready(card)) return 0; chip = card->private_data; - if (!azx_has_pm_runtime(chip)) - return 0; /* enable controller wake up event */ - azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | - STATESTS_INT_MASK); + if (snd_power_get_state(card) == SNDRV_CTL_POWER_D0) { + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | + STATESTS_INT_MASK); + } __azx_runtime_suspend(chip); trace_azx_runtime_suspend(chip); @@ -1114,17 +1120,18 @@ static int azx_runtime_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; + bool from_rt = snd_power_get_state(card) == SNDRV_CTL_POWER_D0; if (!azx_is_pm_ready(card)) return 0; chip = card->private_data; - if (!azx_has_pm_runtime(chip)) - return 0; - __azx_runtime_resume(chip, true); + __azx_runtime_resume(chip, from_rt); /* disable controller Wake Up event*/ - azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & - ~STATESTS_INT_MASK); + if (from_rt) { + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & + ~STATESTS_INT_MASK); + } trace_azx_runtime_resume(chip); return 0; @@ -1199,10 +1206,8 @@ static void azx_vs_set_state(struct pci_dev *pci, if (!disabled) { dev_info(chip->card->dev, "Start delayed initialization\n"); - if (azx_probe_continue(chip) < 0) { + if (azx_probe_continue(chip) < 0) dev_err(chip->card->dev, "initialization error\n"); - hda->init_failed = true; - } } } else { dev_info(chip->card->dev, "%s via vga_switcheroo\n", @@ -1335,12 +1340,15 @@ static int register_vga_switcheroo(struct azx *chip) /* * destructor */ -static int azx_free(struct azx *chip) +static void azx_free(struct azx *chip) { struct pci_dev *pci = chip->pci; struct hda_intel *hda = container_of(chip, struct hda_intel, chip); struct hdac_bus *bus = azx_bus(chip); + if (hda->freed) + return; + if (azx_has_pm_runtime(chip) && chip->running) pm_runtime_get_noresume(&pci->dev); chip->running = 0; @@ -1384,9 +1392,8 @@ static int azx_free(struct azx *chip) if (chip->driver_caps & AZX_DCAPS_I915_COMPONENT) snd_hdac_i915_exit(bus); - kfree(hda); - return 0; + hda->freed = 1; } static int azx_dev_disconnect(struct snd_device *device) @@ -1402,7 +1409,8 @@ static int azx_dev_disconnect(struct snd_device *device) static int azx_dev_free(struct snd_device *device) { - return azx_free(device->device_data); + azx_free(device->device_data); + return 0; } #ifdef SUPPORT_VGA_SWITCHEROO @@ -1769,7 +1777,7 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, if (err < 0) return err; - hda = kzalloc(sizeof(*hda), GFP_KERNEL); + hda = devm_kzalloc(&pci->dev, sizeof(*hda), GFP_KERNEL); if (!hda) { pci_disable_device(pci); return -ENOMEM; @@ -1810,7 +1818,6 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, err = azx_bus_init(chip, model[dev]); if (err < 0) { - kfree(hda); pci_disable_device(pci); return err; } @@ -2005,7 +2012,7 @@ static int azx_first_init(struct azx *chip) /* codec detection */ if (!azx_bus(chip)->codec_mask) { dev_err(card->dev, "no codecs found!\n"); - return -ENODEV; + /* keep running the rest for the runtime PM */ } if (azx_acquire_irq(chip, 0) < 0) @@ -2027,24 +2034,15 @@ static void azx_firmware_cb(const struct firmware *fw, void *context) { struct snd_card *card = context; struct azx *chip = card->private_data; - struct pci_dev *pci = chip->pci; - - if (!fw) { - dev_err(card->dev, "Cannot load firmware, aborting\n"); - goto error; - } - chip->fw = fw; + if (fw) + chip->fw = fw; + else + dev_err(card->dev, "Cannot load firmware, continue without patching\n"); if (!chip->disabled) { /* continue probing */ - if (azx_probe_continue(chip)) - goto error; + azx_probe_continue(chip); } - return; /* OK */ - - error: - snd_card_free(card); - pci_set_drvdata(pci, NULL); } #endif @@ -2080,10 +2078,10 @@ static void pcm_mmap_prepare(struct snd_pcm_substream *substream, * some HD-audio PCI entries are exposed without any codecs, and such devices * should be ignored from the beginning. */ -static const struct snd_pci_quirk driver_blacklist[] = { - SND_PCI_QUIRK(0x1043, 0x874f, "ASUS ROG Zenith II / Strix", 0), - SND_PCI_QUIRK(0x1462, 0xcb59, "MSI TRX40 Creator", 0), - SND_PCI_QUIRK(0x1462, 0xcb60, "MSI TRX40", 0), +static const struct pci_device_id driver_blacklist[] = { + { PCI_DEVICE_SUB(0x1022, 0x1487, 0x1043, 0x874f) }, /* ASUS ROG Zenith II / Strix */ + { PCI_DEVICE_SUB(0x1022, 0x1487, 0x1462, 0xcb59) }, /* MSI TRX40 Creator */ + { PCI_DEVICE_SUB(0x1022, 0x1487, 0x1462, 0xcb60) }, /* MSI TRX40 */ {} }; @@ -2103,7 +2101,7 @@ static int azx_probe(struct pci_dev *pci, bool schedule_probe; int err; - if (snd_pci_quirk_lookup(pci, driver_blacklist)) { + if (pci_match_id(driver_blacklist, pci)) { dev_info(&pci->dev, "Skipping the blacklisted device\n"); return -ENODEV; } @@ -2308,9 +2306,11 @@ static int azx_probe_continue(struct azx *chip) #endif /* create codec instances */ - err = azx_probe_codecs(chip, azx_max_codecs[chip->driver_type]); - if (err < 0) - goto out_free; + if (bus->codec_mask) { + err = azx_probe_codecs(chip, azx_max_codecs[chip->driver_type]); + if (err < 0) + goto out_free; + } #ifdef CONFIG_SND_HDA_PATCH_LOADER if (chip->fw) { @@ -2324,7 +2324,7 @@ static int azx_probe_continue(struct azx *chip) #endif } #endif - if ((probe_only[dev] & 1) == 0) { + if (bus->codec_mask && !(probe_only[dev] & 1)) { err = azx_codec_configure(chip); if (err < 0) goto out_free; @@ -2341,17 +2341,23 @@ static int azx_probe_continue(struct azx *chip) set_default_power_save(chip); - if (azx_has_pm_runtime(chip)) + if (azx_has_pm_runtime(chip)) { + pm_runtime_use_autosuspend(&pci->dev); + pm_runtime_allow(&pci->dev); pm_runtime_put_autosuspend(&pci->dev); + } out_free: - if (err < 0 || !hda->need_i915_power) + if (err < 0) { + azx_free(chip); + return err; + } + + if (!hda->need_i915_power) display_power(chip, false); - if (err < 0) - hda->init_failed = 1; complete_all(&hda->probe_wait); to_hda_bus(bus)->bus_probing = 0; - return err; + return 0; } static void azx_remove(struct pci_dev *pci) diff --git a/sound/pci/hda/hda_intel.h b/sound/pci/hda/hda_intel.h index 2acfff3da1a0..3fb119f09040 100644 --- a/sound/pci/hda/hda_intel.h +++ b/sound/pci/hda/hda_intel.h @@ -27,6 +27,7 @@ struct hda_intel { unsigned int use_vga_switcheroo:1; unsigned int vga_switcheroo_registered:1; unsigned int init_failed:1; /* delayed init failed */ + unsigned int freed:1; /* resources already released */ bool need_i915_power:1; /* the hda controller needs i915 power */ }; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index bb287a916dae..93760a3564cf 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -38,6 +38,10 @@ static bool static_hdmi_pcm; module_param(static_hdmi_pcm, bool, 0644); MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); +static bool enable_acomp = true; +module_param(enable_acomp, bool, 0444); +MODULE_PARM_DESC(enable_acomp, "Enable audio component binding (default=yes)"); + struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; int assigned; @@ -1844,8 +1848,10 @@ static bool check_non_pcm_per_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) /* Add sanity check to pass klockwork check. * This should never happen. */ - if (WARN_ON(spdif == NULL)) + if (WARN_ON(spdif == NULL)) { + mutex_unlock(&codec->spdif_mutex); return true; + } non_pcm = !!(spdif->status & IEC958_AES0_NONAUDIO); mutex_unlock(&codec->spdif_mutex); return non_pcm; @@ -2194,7 +2200,9 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); + struct hdmi_eld *pin_eld = &per_pin->sink_eld; + pin_eld->eld_valid = false; hdmi_present_sense(per_pin, 0); } @@ -2505,6 +2513,11 @@ static void generic_acomp_init(struct hda_codec *codec, { struct hdmi_spec *spec = codec->spec; + if (!enable_acomp) { + codec_info(codec, "audio component disabled by module option\n"); + return; + } + spec->port2pin = port2pin; setup_drm_audio_ops(codec, ops); if (!snd_hdac_acomp_init(&codec->bus->core, &spec->drm_audio_ops, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index de2826f90d34..c16f63957c5a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -377,6 +377,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0233: case 0x10ec0235: case 0x10ec0236: + case 0x10ec0245: case 0x10ec0255: case 0x10ec0256: case 0x10ec0257: @@ -797,9 +798,11 @@ static void alc_ssid_check(struct hda_codec *codec, const hda_nid_t *ports) { if (!alc_subsystem_id(codec, ports)) { struct alc_spec *spec = codec->spec; - codec_dbg(codec, - "realtek: Enable default setup for auto mode as fallback\n"); - spec->init_amp = ALC_INIT_DEFAULT; + if (spec->init_amp == ALC_INIT_UNDEFINED) { + codec_dbg(codec, + "realtek: Enable default setup for auto mode as fallback\n"); + spec->init_amp = ALC_INIT_DEFAULT; + } } } @@ -7378,6 +7381,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x19ce, "ASUS B9450FA", ALC294_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC), @@ -7416,6 +7420,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x8560, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1558, 0x8561, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC233_FIXUP_LENOVO_MULTI_CODECS), + SND_PCI_QUIRK(0x17aa, 0x1048, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), @@ -8195,6 +8200,7 @@ static int patch_alc269(struct hda_codec *codec) spec->gen.mixer_nid = 0; break; case 0x10ec0215: + case 0x10ec0245: case 0x10ec0285: case 0x10ec0289: spec->codec_variant = ALC269_TYPE_ALC215; @@ -9456,6 +9462,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0234, "ALC234", patch_alc269), HDA_CODEC_ENTRY(0x10ec0235, "ALC233", patch_alc269), HDA_CODEC_ENTRY(0x10ec0236, "ALC236", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0245, "ALC245", patch_alc269), HDA_CODEC_ENTRY(0x10ec0255, "ALC255", patch_alc269), HDA_CODEC_ENTRY(0x10ec0256, "ALC256", patch_alc269), HDA_CODEC_ENTRY(0x10ec0257, "ALC257", patch_alc269), diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 024a7ee54cd5..e499c00e0c66 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -89,9 +89,9 @@ static int acp3x_5682_init(struct snd_soc_pcm_runtime *rtd) } snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); - snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_1, KEY_VOLUMEUP); - snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); - snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); ret = snd_soc_component_set_jack(component, &pco_jack, NULL); if (ret) { diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e6a0c5d05fa5..e60e0b6a689c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1525,6 +1525,7 @@ config SND_SOC_WM8804_SPI config SND_SOC_WM8900 tristate + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8903 tristate "Wolfson Microelectronics WM8903 CODEC" @@ -1576,6 +1577,7 @@ config SND_SOC_WM8985 config SND_SOC_WM8988 tristate + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8990 tristate @@ -1594,6 +1596,7 @@ config SND_SOC_WM8994 config SND_SOC_WM8995 tristate + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8996 tristate diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index fba9b749839d..f26b77faed59 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -142,14 +142,14 @@ static struct hdac_hdmi_pcm * hdac_hdmi_get_pcm_from_cvt(struct hdac_hdmi_priv *hdmi, struct hdac_hdmi_cvt *cvt) { - struct hdac_hdmi_pcm *pcm = NULL; + struct hdac_hdmi_pcm *pcm; list_for_each_entry(pcm, &hdmi->pcm_list, head) { if (pcm->cvt == cvt) - break; + return pcm; } - return pcm; + return NULL; } static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm, diff --git a/sound/soc/codecs/madera.c b/sound/soc/codecs/madera.c index 40de9d7811d1..a448d2a2918a 100644 --- a/sound/soc/codecs/madera.c +++ b/sound/soc/codecs/madera.c @@ -1903,7 +1903,6 @@ const struct soc_enum madera_isrc_fsh[] = { MADERA_ISRC4_FSH_SHIFT, 0xf, MADERA_RATE_ENUM_SIZE, madera_rate_text, madera_rate_val), - }; EXPORT_SYMBOL_GPL(madera_isrc_fsh); @@ -1924,7 +1923,6 @@ const struct soc_enum madera_isrc_fsl[] = { MADERA_ISRC4_FSL_SHIFT, 0xf, MADERA_RATE_ENUM_SIZE, madera_rate_text, madera_rate_val), - }; EXPORT_SYMBOL_GPL(madera_isrc_fsl); @@ -1938,7 +1936,6 @@ const struct soc_enum madera_asrc1_rate[] = { MADERA_ASYNC_RATE_ENUM_SIZE, madera_rate_text + MADERA_SYNC_RATE_ENUM_SIZE, madera_rate_val + MADERA_SYNC_RATE_ENUM_SIZE), - }; EXPORT_SYMBOL_GPL(madera_asrc1_rate); @@ -1964,7 +1961,6 @@ const struct soc_enum madera_asrc2_rate[] = { MADERA_ASYNC_RATE_ENUM_SIZE, madera_rate_text + MADERA_SYNC_RATE_ENUM_SIZE, madera_rate_val + MADERA_SYNC_RATE_ENUM_SIZE), - }; EXPORT_SYMBOL_GPL(madera_asrc2_rate); diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d5130193b4a2..e8a8bf7b4ffe 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1653,6 +1653,40 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, dev_err(&client->dev, "Error %d initializing CHIP_CLK_CTRL\n", ret); + /* Mute everything to avoid pop from the following power-up */ + ret = regmap_write(sgtl5000->regmap, SGTL5000_CHIP_ANA_CTRL, + SGTL5000_CHIP_ANA_CTRL_DEFAULT); + if (ret) { + dev_err(&client->dev, + "Error %d muting outputs via CHIP_ANA_CTRL\n", ret); + goto disable_clk; + } + + /* + * If VAG is powered-on (e.g. from previous boot), it would be disabled + * by the write to ANA_POWER in later steps of the probe code. This + * may create a loud pop even with all outputs muted. The proper way + * to circumvent this is disabling the bit first and waiting the proper + * cool-down time. + */ + ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ANA_POWER, &value); + if (ret) { + dev_err(&client->dev, "Failed to read ANA_POWER: %d\n", ret); + goto disable_clk; + } + if (value & SGTL5000_VAG_POWERUP) { + ret = regmap_update_bits(sgtl5000->regmap, + SGTL5000_CHIP_ANA_POWER, + SGTL5000_VAG_POWERUP, + 0); + if (ret) { + dev_err(&client->dev, "Error %d disabling VAG\n", ret); + goto disable_clk; + } + + msleep(SGTL5000_VAG_POWERDOWN_DELAY); + } + /* Follow section 2.2.1.1 of AN3663 */ ana_pwr = SGTL5000_ANA_POWER_DEFAULT; if (sgtl5000->num_supplies <= VDDD) { diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index a4bf4bca95bf..56ec5863f250 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -233,6 +233,7 @@ /* * SGTL5000_CHIP_ANA_CTRL */ +#define SGTL5000_CHIP_ANA_CTRL_DEFAULT 0x0133 #define SGTL5000_LINE_OUT_MUTE 0x0100 #define SGTL5000_HP_SEL_MASK 0x0040 #define SGTL5000_HP_SEL_SHIFT 6 diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 1554631cb397..5b7f9fcf6cbf 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -820,8 +820,10 @@ static int tas571x_i2c_probe(struct i2c_client *client, priv->regmap = devm_regmap_init(dev, NULL, client, priv->chip->regmap_config); - if (IS_ERR(priv->regmap)) - return PTR_ERR(priv->regmap); + if (IS_ERR(priv->regmap)) { + ret = PTR_ERR(priv->regmap); + goto disable_regs; + } priv->pdn_gpio = devm_gpiod_get_optional(dev, "pdn", GPIOD_OUT_LOW); if (IS_ERR(priv->pdn_gpio)) { @@ -845,7 +847,7 @@ static int tas571x_i2c_probe(struct i2c_client *client, ret = regmap_write(priv->regmap, TAS571X_OSC_TRIM_REG, 0); if (ret) - return ret; + goto disable_regs; usleep_range(50000, 60000); @@ -861,12 +863,20 @@ static int tas571x_i2c_probe(struct i2c_client *client, */ ret = regmap_update_bits(priv->regmap, TAS571X_MVOL_REG, 1, 0); if (ret) - return ret; + goto disable_regs; } - return devm_snd_soc_register_component(&client->dev, + ret = devm_snd_soc_register_component(&client->dev, &priv->component_driver, &tas571x_dai, 1); + if (ret) + goto disable_regs; + + return ret; + +disable_regs: + regulator_bulk_disable(priv->chip->num_supply_names, priv->supplies); + return ret; } static int tas571x_i2c_remove(struct i2c_client *client) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 55112c1bba5e..6cf0f6612bda 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -860,8 +860,7 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, wm8960->is_stream_in_use[tx] = true; - if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_ON && - !wm8960->is_stream_in_use[!tx]) + if (!wm8960->is_stream_in_use[!tx]) return wm8960_configure_clocking(component); return 0; diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index f2d6f2f81f14..d39d479e2378 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -394,6 +394,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = { .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, + .read_only_wordlength = true, }, { /* COMP */ .num = 2, @@ -401,6 +402,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = { .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, + .read_only_wordlength = true, }, { /* BOOST */ .num = 3, @@ -408,6 +410,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = { .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, + .read_only_wordlength = true, }, { /* VISENSE */ .num = 4, @@ -415,6 +418,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = { .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, + .read_only_wordlength = true, } }; diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c index bcedec6c6117..7d85bd5aff9f 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c @@ -113,14 +113,6 @@ static const struct snd_soc_acpi_adr_device rt1308_1_adr[] = { } }; -static const struct snd_soc_acpi_adr_device rt1308_2_adr[] = { - { - .adr = 0x000210025D130800, - .num_endpoints = 1, - .endpoints = &single_endpoint, - } -}; - static const struct snd_soc_acpi_adr_device rt1308_1_group1_adr[] = { { .adr = 0x000110025D130800, diff --git a/sound/soc/intel/common/soc-acpi-intel-icl-match.c b/sound/soc/intel/common/soc-acpi-intel-icl-match.c index ef8500349f2f..16ec9f382b0f 100644 --- a/sound/soc/intel/common/soc-acpi-intel-icl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-icl-match.c @@ -87,14 +87,6 @@ static const struct snd_soc_acpi_adr_device rt1308_1_adr[] = { } }; -static const struct snd_soc_acpi_adr_device rt1308_2_adr[] = { - { - .adr = 0x000210025D130800, - .num_endpoints = 1, - .endpoints = &single_endpoint, - } -}; - static const struct snd_soc_acpi_adr_device rt1308_1_group1_adr[] = { { .adr = 0x000110025D130800, diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index af46845f4ef2..89f7f64747cd 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -338,8 +338,10 @@ static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np, if (axg_card_cpu_is_tdm_iface(dai_link->cpus->of_node)) ret = axg_card_parse_tdm(card, np, index); - else if (axg_card_cpu_is_codec(dai_link->cpus->of_node)) + else if (axg_card_cpu_is_codec(dai_link->cpus->of_node)) { dai_link->params = &codec_params; + dai_link->no_pcm = 0; /* link is not a DPCM BE */ + } return ret; } diff --git a/sound/soc/meson/gx-card.c b/sound/soc/meson/gx-card.c index 7b01dcb73e5e..4abf7efb7eac 100644 --- a/sound/soc/meson/gx-card.c +++ b/sound/soc/meson/gx-card.c @@ -108,8 +108,10 @@ static int gx_card_add_link(struct snd_soc_card *card, struct device_node *np, ret = gx_card_parse_i2s(card, np, index); /* Or apply codec to codec params if necessary */ - else if (gx_card_cpu_identify(dai_link->cpus, "CODEC CTRL")) + else if (gx_card_cpu_identify(dai_link->cpus, "CODEC CTRL")) { dai_link->params = &codec_params; + dai_link->no_pcm = 0; /* link is not a DPCM BE */ + } return ret; } diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index d55e3ad96716..287ad2aa27f3 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -116,10 +116,8 @@ static int apq8096_platform_probe(struct platform_device *pdev) card->dev = dev; dev_set_drvdata(dev, card); ret = qcom_snd_parse_of(card); - if (ret) { - dev_err(dev, "Error parsing OF data\n"); + if (ret) goto err; - } apq8096_add_be_ops(card); ret = snd_soc_register_card(card); diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index c1a7624eaf17..2a5302f1db98 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -902,6 +902,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, @@ -917,6 +919,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, @@ -931,6 +935,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, @@ -946,6 +952,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, @@ -960,6 +968,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, @@ -975,6 +985,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, @@ -989,6 +1001,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, @@ -1004,6 +1018,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = { SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, .rate_min = 8000, .rate_max = 48000, }, diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index b2de65c7f95c..68e9388ff46f 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -559,10 +559,8 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) card->dev = dev; dev_set_drvdata(dev, card); ret = qcom_snd_parse_of(card); - if (ret) { - dev_err(dev, "Error parsing OF data\n"); + if (ret) goto parse_dt_fail; - } data->card = card; snd_soc_card_set_drvdata(card, data); diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 358887848293..5e95c30fb2ba 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -656,60 +656,6 @@ void s3c_i2sv2_cleanup(struct snd_soc_dai *dai, } EXPORT_SYMBOL_GPL(s3c_i2sv2_cleanup); -#ifdef CONFIG_PM -static int s3c2412_i2s_suspend(struct snd_soc_dai *dai) -{ - struct s3c_i2sv2_info *i2s = to_info(dai); - u32 iismod; - - if (dai->active) { - i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD); - i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON); - i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR); - - /* some basic suspend checks */ - - iismod = readl(i2s->regs + S3C2412_IISMOD); - - if (iismod & S3C2412_IISCON_RXDMA_ACTIVE) - pr_warn("%s: RXDMA active?\n", __func__); - - if (iismod & S3C2412_IISCON_TXDMA_ACTIVE) - pr_warn("%s: TXDMA active?\n", __func__); - - if (iismod & S3C2412_IISCON_IIS_ACTIVE) - pr_warn("%s: IIS active\n", __func__); - } - - return 0; -} - -static int s3c2412_i2s_resume(struct snd_soc_dai *dai) -{ - struct s3c_i2sv2_info *i2s = to_info(dai); - - pr_info("dai_active %d, IISMOD %08x, IISCON %08x\n", - dai->active, i2s->suspend_iismod, i2s->suspend_iiscon); - - if (dai->active) { - writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON); - writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD); - writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR); - - writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH, - i2s->regs + S3C2412_IISFIC); - - ndelay(250); - writel(0x0, i2s->regs + S3C2412_IISFIC); - } - - return 0; -} -#else -#define s3c2412_i2s_suspend NULL -#define s3c2412_i2s_resume NULL -#endif - int s3c_i2sv2_register_component(struct device *dev, int id, const struct snd_soc_component_driver *cmp_drv, struct snd_soc_dai_driver *dai_drv) @@ -727,9 +673,6 @@ int s3c_i2sv2_register_component(struct device *dev, int id, if (!ops->delay) ops->delay = s3c2412_i2s_delay; - dai_drv->suspend = s3c2412_i2s_suspend; - dai_drv->resume = s3c2412_i2s_resume; - return devm_snd_soc_register_component(dev, cmp_drv, dai_drv, 1); } EXPORT_SYMBOL_GPL(s3c_i2sv2_register_component); diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 787a3f6e9f24..b35d828c1cfe 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -117,6 +117,60 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } +#ifdef CONFIG_PM +static int s3c2412_i2s_suspend(struct snd_soc_component *component) +{ + struct s3c_i2sv2_info *i2s = snd_soc_component_get_drvdata(component); + u32 iismod; + + if (component->active) { + i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD); + i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON); + i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR); + + /* some basic suspend checks */ + + iismod = readl(i2s->regs + S3C2412_IISMOD); + + if (iismod & S3C2412_IISCON_RXDMA_ACTIVE) + pr_warn("%s: RXDMA active?\n", __func__); + + if (iismod & S3C2412_IISCON_TXDMA_ACTIVE) + pr_warn("%s: TXDMA active?\n", __func__); + + if (iismod & S3C2412_IISCON_IIS_ACTIVE) + pr_warn("%s: IIS active\n", __func__); + } + + return 0; +} + +static int s3c2412_i2s_resume(struct snd_soc_component *component) +{ + struct s3c_i2sv2_info *i2s = snd_soc_component_get_drvdata(component); + + pr_info("component_active %d, IISMOD %08x, IISCON %08x\n", + component->active, i2s->suspend_iismod, i2s->suspend_iiscon); + + if (component->active) { + writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON); + writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD); + writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR); + + writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH, + i2s->regs + S3C2412_IISFIC); + + ndelay(250); + writel(0x0, i2s->regs + S3C2412_IISFIC); + } + + return 0; +} +#else +#define s3c2412_i2s_suspend NULL +#define s3c2412_i2s_resume NULL +#endif + #define S3C2412_I2S_RATES \ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ @@ -146,6 +200,8 @@ static struct snd_soc_dai_driver s3c2412_i2s_dai = { static const struct snd_soc_component_driver s3c2412_i2s_component = { .name = "s3c2412-i2s", + .suspend = s3c2412_i2s_suspend, + .resume = s3c2412_i2s_resume, }; static int s3c2412_iis_dev_probe(struct platform_device *pdev) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index fc5d089868df..4a7d3413917f 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -594,10 +594,16 @@ static int rsnd_ssi_stop(struct rsnd_mod *mod, * Capture: It might not receave data. Do nothing */ if (rsnd_io_is_play(io)) { - rsnd_mod_write(mod, SSICR, cr | EN); + rsnd_mod_write(mod, SSICR, cr | ssi->cr_en); rsnd_ssi_status_check(mod, DIRQ); } + /* In multi-SSI mode, stop is performed by setting ssi0129 in + * SSI_CONTROL to 0 (in rsnd_ssio_stop_gen2). Do nothing here. + */ + if (rsnd_ssi_multi_slaves_runtime(io)) + return 0; + /* * disable SSI, * and, wait idle state @@ -737,6 +743,9 @@ static void rsnd_ssi_parent_attach(struct rsnd_mod *mod, if (!rsnd_rdai_is_clk_master(rdai)) return; + if (rsnd_ssi_is_multi_slave(mod, io)) + return; + switch (rsnd_mod_id(mod)) { case 1: case 2: diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index f35d88211887..9c7c3e7539c9 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -221,7 +221,7 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, i; for_each_rsnd_mod_array(i, pos, io, rsnd_ssi_array) { - shift = (i * 4) + 16; + shift = (i * 4) + 20; val = (val & ~(0xF << shift)) | rsnd_mod_id(pos) << shift; } diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 8f3cad8db89a..31c41559034b 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -295,24 +295,17 @@ int snd_soc_dai_startup(struct snd_soc_dai *dai, { int ret = 0; - if (!dai->started[substream->stream] && - dai->driver->ops->startup) + if (dai->driver->ops->startup) ret = dai->driver->ops->startup(substream, dai); - if (ret == 0) - dai->started[substream->stream] = 1; - return ret; } void snd_soc_dai_shutdown(struct snd_soc_dai *dai, struct snd_pcm_substream *substream) { - if (dai->started[substream->stream] && - dai->driver->ops->shutdown) + if (dai->driver->ops->shutdown) dai->driver->ops->shutdown(substream, dai); - - dai->started[substream->stream] = 0; } int snd_soc_dai_prepare(struct snd_soc_dai *dai, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 679ed60d850e..e2632841b321 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -423,7 +423,7 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, memset(&template, 0, sizeof(template)); template.reg = e->reg; - template.mask = e->mask << e->shift_l; + template.mask = e->mask; template.shift = e->shift_l; template.off_val = snd_soc_enum_item_to_val(e, 0); template.on_val = template.off_val; @@ -546,8 +546,22 @@ static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, if (data->value == value) return false; - if (data->widget) - data->widget->on_val = value; + if (data->widget) { + switch (dapm_kcontrol_get_wlist(kcontrol)->widgets[0]->id) { + case snd_soc_dapm_switch: + case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: + data->widget->on_val = value & data->widget->mask; + break; + case snd_soc_dapm_demux: + case snd_soc_dapm_mux: + data->widget->on_val = value >> data->widget->shift; + break; + default: + data->widget->on_val = value; + break; + } + } data->value = value; @@ -4165,6 +4179,8 @@ snd_soc_dapm_new_dai(struct snd_soc_card *card, w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); if (IS_ERR(w)) { ret = PTR_ERR(w); + dev_err(rtd->dev, "ASoC: Failed to create %s widget: %d\n", + link_name, ret); goto outfree_kcontrol_news; } @@ -4283,52 +4299,58 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) return 0; } -static void dapm_add_valid_dai_widget(struct snd_soc_card *card, - struct snd_soc_pcm_runtime *rtd, - struct snd_soc_dai *codec_dai, - struct snd_soc_dai *cpu_dai) +static void dapm_connect_dai_routes(struct snd_soc_dapm_context *dapm, + struct snd_soc_dai *src_dai, + struct snd_soc_dapm_widget *src, + struct snd_soc_dapm_widget *dai, + struct snd_soc_dai *sink_dai, + struct snd_soc_dapm_widget *sink) { - struct snd_soc_dapm_widget *playback = NULL, *capture = NULL; - struct snd_soc_dapm_widget *codec, *playback_cpu, *capture_cpu; + dev_dbg(dapm->dev, "connected DAI link %s:%s -> %s:%s\n", + src_dai->component->name, src->name, + sink_dai->component->name, sink->name); + + if (dai) { + snd_soc_dapm_add_path(dapm, src, dai, NULL, NULL); + src = dai; + } + + snd_soc_dapm_add_path(dapm, src, sink, NULL, NULL); +} + +static void dapm_connect_dai_pair(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *codec_dai, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_dai_link *dai_link = rtd->dai_link; + struct snd_soc_dapm_widget *dai, *codec, *playback_cpu, *capture_cpu; struct snd_pcm_substream *substream; struct snd_pcm_str *streams = rtd->pcm->streams; - if (rtd->dai_link->params) { + if (dai_link->params) { playback_cpu = cpu_dai->capture_widget; capture_cpu = cpu_dai->playback_widget; } else { - playback = cpu_dai->playback_widget; - capture = cpu_dai->capture_widget; - playback_cpu = playback; - capture_cpu = capture; + playback_cpu = cpu_dai->playback_widget; + capture_cpu = cpu_dai->capture_widget; } /* connect BE DAI playback if widgets are valid */ codec = codec_dai->playback_widget; if (playback_cpu && codec) { - if (!playback) { + if (dai_link->params && !dai_link->playback_widget) { substream = streams[SNDRV_PCM_STREAM_PLAYBACK].substream; - playback = snd_soc_dapm_new_dai(card, substream, - "playback"); - if (IS_ERR(playback)) { - dev_err(rtd->dev, - "ASoC: Failed to create DAI %s: %ld\n", - codec_dai->name, - PTR_ERR(playback)); + dai = snd_soc_dapm_new_dai(card, substream, "playback"); + if (IS_ERR(dai)) goto capture; - } - - snd_soc_dapm_add_path(&card->dapm, playback_cpu, - playback, NULL, NULL); + dai_link->playback_widget = dai; } - dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", - cpu_dai->component->name, playback_cpu->name, - codec_dai->component->name, codec->name); - - snd_soc_dapm_add_path(&card->dapm, playback, codec, - NULL, NULL); + dapm_connect_dai_routes(&card->dapm, cpu_dai, playback_cpu, + dai_link->playback_widget, + codec_dai, codec); } capture: @@ -4336,50 +4358,18 @@ capture: codec = codec_dai->capture_widget; if (codec && capture_cpu) { - if (!capture) { + if (dai_link->params && !dai_link->capture_widget) { substream = streams[SNDRV_PCM_STREAM_CAPTURE].substream; - capture = snd_soc_dapm_new_dai(card, substream, - "capture"); - if (IS_ERR(capture)) { - dev_err(rtd->dev, - "ASoC: Failed to create DAI %s: %ld\n", - codec_dai->name, - PTR_ERR(capture)); + dai = snd_soc_dapm_new_dai(card, substream, "capture"); + if (IS_ERR(dai)) return; - } - - snd_soc_dapm_add_path(&card->dapm, capture, - capture_cpu, NULL, NULL); + dai_link->capture_widget = dai; } - dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", - codec_dai->component->name, codec->name, - cpu_dai->component->name, capture_cpu->name); - - snd_soc_dapm_add_path(&card->dapm, codec, capture, - NULL, NULL); - } -} - -static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, - struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_dai *codec_dai; - int i; - - if (rtd->num_cpus == 1) { - for_each_rtd_codec_dais(rtd, i, codec_dai) - dapm_add_valid_dai_widget(card, rtd, codec_dai, - rtd->cpu_dais[0]); - } else if (rtd->num_codecs == rtd->num_cpus) { - for_each_rtd_codec_dais(rtd, i, codec_dai) - dapm_add_valid_dai_widget(card, rtd, codec_dai, - rtd->cpu_dais[i]); - } else { - dev_err(card->dev, - "N cpus to M codecs link is not supported yet\n"); + dapm_connect_dai_routes(&card->dapm, codec_dai, codec, + dai_link->capture_widget, + cpu_dai, capture_cpu); } - } static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream, @@ -4422,6 +4412,8 @@ static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream, void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; + int i; /* for each BE DAI link... */ for_each_card_rtds(card, rtd) { @@ -4432,7 +4424,18 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) if (rtd->dai_link->dynamic) continue; - dapm_connect_dai_link_widgets(card, rtd); + if (rtd->num_cpus == 1) { + for_each_rtd_codec_dais(rtd, i, codec_dai) + dapm_connect_dai_pair(card, rtd, codec_dai, + rtd->cpu_dais[0]); + } else if (rtd->num_codecs == rtd->num_cpus) { + for_each_rtd_codec_dais(rtd, i, codec_dai) + dapm_connect_dai_pair(card, rtd, codec_dai, + rtd->cpu_dais[i]); + } else { + dev_err(card->dev, + "N cpus to M codecs link is not supported yet\n"); + } } } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 289aebc15529..1f302de44052 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2911,8 +2911,17 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) int i; if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) { - playback = rtd->dai_link->dpcm_playback; - capture = rtd->dai_link->dpcm_capture; + cpu_dai = asoc_rtd_to_cpu(rtd, 0); + if (rtd->num_cpus > 1) { + dev_err(rtd->dev, + "DPCM doesn't support Multi CPU yet\n"); + return -EINVAL; + } + + playback = rtd->dai_link->dpcm_playback && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK); + capture = rtd->dai_link->dpcm_capture && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE); } else { /* Adapt stream for codec2codec links */ int cpu_capture = rtd->dai_link->params ? diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 87f75edba3dc..6df3b0d12d87 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -894,7 +894,13 @@ static int soc_tplg_dmixer_create(struct soc_tplg *tplg, unsigned int count, } /* create any TLV data */ - soc_tplg_create_tlv(tplg, &kc, &mc->hdr); + err = soc_tplg_create_tlv(tplg, &kc, &mc->hdr); + if (err < 0) { + dev_err(tplg->dev, "ASoC: failed to create TLV %s\n", + mc->hdr.name); + kfree(sm); + continue; + } /* pass control to driver for optional further init */ err = soc_tplg_init_kcontrol(tplg, &kc, @@ -1118,6 +1124,7 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg, struct snd_soc_tplg_hdr *hdr) { struct snd_soc_tplg_ctl_hdr *control_hdr; + int ret; int i; if (tplg->pass != SOC_TPLG_PASS_MIXER) { @@ -1146,25 +1153,30 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg, case SND_SOC_TPLG_CTL_RANGE: case SND_SOC_TPLG_DAPM_CTL_VOLSW: case SND_SOC_TPLG_DAPM_CTL_PIN: - soc_tplg_dmixer_create(tplg, 1, - le32_to_cpu(hdr->payload_size)); + ret = soc_tplg_dmixer_create(tplg, 1, + le32_to_cpu(hdr->payload_size)); break; case SND_SOC_TPLG_CTL_ENUM: case SND_SOC_TPLG_CTL_ENUM_VALUE: case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: - soc_tplg_denum_create(tplg, 1, - le32_to_cpu(hdr->payload_size)); + ret = soc_tplg_denum_create(tplg, 1, + le32_to_cpu(hdr->payload_size)); break; case SND_SOC_TPLG_CTL_BYTES: - soc_tplg_dbytes_create(tplg, 1, - le32_to_cpu(hdr->payload_size)); + ret = soc_tplg_dbytes_create(tplg, 1, + le32_to_cpu(hdr->payload_size)); break; default: soc_bind_err(tplg, control_hdr, i); return -EINVAL; } + if (ret < 0) { + dev_err(tplg->dev, "ASoC: invalid control\n"); + return ret; + } + } return 0; @@ -1272,7 +1284,9 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg, routes[i]->dobj.index = tplg->index; list_add(&routes[i]->dobj.list, &tplg->comp->dobj_list); - soc_tplg_add_route(tplg, routes[i]); + ret = soc_tplg_add_route(tplg, routes[i]); + if (ret < 0) + break; /* add route, but keep going if some fail */ snd_soc_dapm_add_routes(dapm, routes[i], 1); @@ -1355,7 +1369,13 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( } /* create any TLV data */ - soc_tplg_create_tlv(tplg, &kc[i], &mc->hdr); + err = soc_tplg_create_tlv(tplg, &kc[i], &mc->hdr); + if (err < 0) { + dev_err(tplg->dev, "ASoC: failed to create TLV %s\n", + mc->hdr.name); + kfree(sm); + continue; + } /* pass control to driver for optional further init */ err = soc_tplg_init_kcontrol(tplg, &kc[i], @@ -1766,10 +1786,13 @@ static int soc_tplg_dapm_complete(struct soc_tplg *tplg) return 0; } -static void set_stream_info(struct snd_soc_pcm_stream *stream, +static int set_stream_info(struct snd_soc_pcm_stream *stream, struct snd_soc_tplg_stream_caps *caps) { stream->stream_name = kstrdup(caps->name, GFP_KERNEL); + if (!stream->stream_name) + return -ENOMEM; + stream->channels_min = le32_to_cpu(caps->channels_min); stream->channels_max = le32_to_cpu(caps->channels_max); stream->rates = le32_to_cpu(caps->rates); @@ -1777,6 +1800,8 @@ static void set_stream_info(struct snd_soc_pcm_stream *stream, stream->rate_max = le32_to_cpu(caps->rate_max); stream->formats = le64_to_cpu(caps->formats); stream->sig_bits = le32_to_cpu(caps->sig_bits); + + return 0; } static void set_dai_flags(struct snd_soc_dai_driver *dai_drv, @@ -1812,20 +1837,29 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, if (dai_drv == NULL) return -ENOMEM; - if (strlen(pcm->dai_name)) + if (strlen(pcm->dai_name)) { dai_drv->name = kstrdup(pcm->dai_name, GFP_KERNEL); + if (!dai_drv->name) { + ret = -ENOMEM; + goto err; + } + } dai_drv->id = le32_to_cpu(pcm->dai_id); if (pcm->playback) { stream = &dai_drv->playback; caps = &pcm->caps[SND_SOC_TPLG_STREAM_PLAYBACK]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (pcm->capture) { stream = &dai_drv->capture; caps = &pcm->caps[SND_SOC_TPLG_STREAM_CAPTURE]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (pcm->compress) @@ -1835,11 +1869,7 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, ret = soc_tplg_dai_load(tplg, dai_drv, pcm, NULL); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); - kfree(dai_drv->playback.stream_name); - kfree(dai_drv->capture.stream_name); - kfree(dai_drv->name); - kfree(dai_drv); - return ret; + goto err; } dai_drv->dobj.index = tplg->index; @@ -1860,6 +1890,14 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, return ret; } + return 0; + +err: + kfree(dai_drv->playback.stream_name); + kfree(dai_drv->capture.stream_name); + kfree(dai_drv->name); + kfree(dai_drv); + return ret; } @@ -1916,11 +1954,20 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg, if (strlen(pcm->pcm_name)) { link->name = kstrdup(pcm->pcm_name, GFP_KERNEL); link->stream_name = kstrdup(pcm->pcm_name, GFP_KERNEL); + if (!link->name || !link->stream_name) { + ret = -ENOMEM; + goto err; + } } link->id = le32_to_cpu(pcm->pcm_id); - if (strlen(pcm->dai_name)) + if (strlen(pcm->dai_name)) { link->cpus->dai_name = kstrdup(pcm->dai_name, GFP_KERNEL); + if (!link->cpus->dai_name) { + ret = -ENOMEM; + goto err; + } + } link->codecs->name = "snd-soc-dummy"; link->codecs->dai_name = "snd-soc-dummy-dai"; @@ -2088,7 +2135,9 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, _pcm = pcm; } else { abi_match = false; - pcm_new_ver(tplg, pcm, &_pcm); + ret = pcm_new_ver(tplg, pcm, &_pcm); + if (ret < 0) + return ret; } /* create the FE DAIs and DAI links */ @@ -2436,13 +2485,17 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg, if (d->playback) { stream = &dai_drv->playback; caps = &d->caps[SND_SOC_TPLG_STREAM_PLAYBACK]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (d->capture) { stream = &dai_drv->capture; caps = &d->caps[SND_SOC_TPLG_STREAM_CAPTURE]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (d->flag_mask) @@ -2454,10 +2507,15 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg, ret = soc_tplg_dai_load(tplg, dai_drv, NULL, dai); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); - return ret; + goto err; } return 0; + +err: + kfree(dai_drv->playback.stream_name); + kfree(dai_drv->capture.stream_name); + return ret; } /* load physical DAI elements */ @@ -2466,7 +2524,7 @@ static int soc_tplg_dai_elems_load(struct soc_tplg *tplg, { struct snd_soc_tplg_dai *dai; int count; - int i; + int i, ret; count = le32_to_cpu(hdr->count); @@ -2481,7 +2539,12 @@ static int soc_tplg_dai_elems_load(struct soc_tplg *tplg, return -EINVAL; } - soc_tplg_dai_config(tplg, dai); + ret = soc_tplg_dai_config(tplg, dai); + if (ret < 0) { + dev_err(tplg->dev, "ASoC: failed to configure DAI\n"); + return ret; + } + tplg->pos += (sizeof(*dai) + le32_to_cpu(dai->priv.size)); } @@ -2589,7 +2652,7 @@ static int soc_valid_header(struct soc_tplg *tplg, } /* big endian firmware objects not supported atm */ - if (hdr->magic == SOC_TPLG_MAGIC_BIG_ENDIAN) { + if (le32_to_cpu(hdr->magic) == SOC_TPLG_MAGIC_BIG_ENDIAN) { dev_err(tplg->dev, "ASoC: pass %d big endian not supported header got %x at offset 0x%lx size 0x%zx.\n", tplg->pass, hdr->magic, diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c index 6c23c5769330..a32a3ef78ec5 100644 --- a/sound/soc/sof/intel/bdw.c +++ b/sound/soc/sof/intel/bdw.c @@ -567,9 +567,25 @@ static void bdw_set_mach_params(const struct snd_soc_acpi_mach *mach, static struct snd_soc_dai_driver bdw_dai[] = { { .name = "ssp0-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "ssp1-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, }; diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c index f84391294f12..29fd1d86156c 100644 --- a/sound/soc/sof/intel/byt.c +++ b/sound/soc/sof/intel/byt.c @@ -459,21 +459,69 @@ static void byt_set_mach_params(const struct snd_soc_acpi_mach *mach, static struct snd_soc_dai_driver byt_dai[] = { { .name = "ssp0-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "ssp1-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "ssp2-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + } }, { .name = "ssp3-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "ssp4-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "ssp5-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, }; diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 0d0c9afd8791..41f01c3e639e 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -837,7 +837,7 @@ static int stm32_sai_set_config(struct snd_soc_dai *cpu_dai, cr1 = SAI_XCR1_DS_SET(SAI_DATASIZE_32); break; default: - dev_err(cpu_dai->dev, "Data format not supported"); + dev_err(cpu_dai->dev, "Data format not supported\n"); return -EINVAL; } @@ -1547,6 +1547,9 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) return ret; } + if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) + conf = &stm32_sai_pcm_config_spdif; + ret = snd_dmaengine_pcm_register(&pdev->dev, conf, 0); if (ret) { if (ret != -EPROBE_DEFER) @@ -1556,15 +1559,10 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) ret = snd_soc_register_component(&pdev->dev, &stm32_component, &sai->cpu_dai_drv, 1); - if (ret) { + if (ret) snd_dmaengine_pcm_unregister(&pdev->dev); - return ret; - } - - if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) - conf = &stm32_sai_pcm_config_spdif; - return 0; + return ret; } static int stm32_sai_sub_remove(struct platform_device *pdev) diff --git a/sound/usb/format.c b/sound/usb/format.c index 50e1874c847c..5ffb457cc88c 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -278,6 +278,52 @@ static bool s1810c_valid_sample_rate(struct audioformat *fp, } /* + * Many Focusrite devices supports a limited set of sampling rates per + * altsetting. Maximum rate is exposed in the last 4 bytes of Format Type + * descriptor which has a non-standard bLength = 10. + */ +static bool focusrite_valid_sample_rate(struct snd_usb_audio *chip, + struct audioformat *fp, + unsigned int rate) +{ + struct usb_interface *iface; + struct usb_host_interface *alts; + unsigned char *fmt; + unsigned int max_rate; + + iface = usb_ifnum_to_if(chip->dev, fp->iface); + if (!iface) + return true; + + alts = &iface->altsetting[fp->altset_idx]; + fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, + NULL, UAC_FORMAT_TYPE); + if (!fmt) + return true; + + if (fmt[0] == 10) { /* bLength */ + max_rate = combine_quad(&fmt[6]); + + /* Validate max rate */ + if (max_rate != 48000 && + max_rate != 96000 && + max_rate != 192000 && + max_rate != 384000) { + + usb_audio_info(chip, + "%u:%d : unexpected max rate: %u\n", + fp->iface, fp->altsetting, max_rate); + + return true; + } + + return rate <= max_rate; + } + + return true; +} + +/* * Helper function to walk the array of sample rate triplets reported by * the device. The problem is that we need to parse whole array first to * get to know how many sample rates we have to expect. @@ -319,6 +365,11 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip, !s1810c_valid_sample_rate(fp, rate)) goto skip_rate; + /* Filter out invalid rates on Focusrite devices */ + if (USB_ID_VENDOR(chip->usb_id) == 0x1235 && + !focusrite_valid_sample_rate(chip, fp, rate)) + goto skip_rate; + if (fp->rate_table) fp->rate_table[nr_rates] = rate; if (!fp->rate_min || rate < fp->rate_min) diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c index d37db32ecd3b..e39dc85c355a 100644 --- a/sound/usb/line6/podhd.c +++ b/sound/usb/line6/podhd.c @@ -21,8 +21,7 @@ enum { LINE6_PODHD300, LINE6_PODHD400, - LINE6_PODHD500_0, - LINE6_PODHD500_1, + LINE6_PODHD500, LINE6_PODX3, LINE6_PODX3LIVE, LINE6_PODHD500X, @@ -318,8 +317,7 @@ static const struct usb_device_id podhd_id_table[] = { /* TODO: no need to alloc data interfaces when only audio is used */ { LINE6_DEVICE(0x5057), .driver_info = LINE6_PODHD300 }, { LINE6_DEVICE(0x5058), .driver_info = LINE6_PODHD400 }, - { LINE6_IF_NUM(0x414D, 0), .driver_info = LINE6_PODHD500_0 }, - { LINE6_IF_NUM(0x414D, 1), .driver_info = LINE6_PODHD500_1 }, + { LINE6_IF_NUM(0x414D, 0), .driver_info = LINE6_PODHD500 }, { LINE6_IF_NUM(0x414A, 0), .driver_info = LINE6_PODX3 }, { LINE6_IF_NUM(0x414B, 0), .driver_info = LINE6_PODX3LIVE }, { LINE6_IF_NUM(0x4159, 0), .driver_info = LINE6_PODHD500X }, @@ -352,23 +350,13 @@ static const struct line6_properties podhd_properties_table[] = { .ep_audio_r = 0x82, .ep_audio_w = 0x01, }, - [LINE6_PODHD500_0] = { + [LINE6_PODHD500] = { .id = "PODHD500", .name = "POD HD500", - .capabilities = LINE6_CAP_PCM + .capabilities = LINE6_CAP_PCM | LINE6_CAP_CONTROL | LINE6_CAP_HWMON, .altsetting = 1, - .ep_ctrl_r = 0x81, - .ep_ctrl_w = 0x01, - .ep_audio_r = 0x86, - .ep_audio_w = 0x02, - }, - [LINE6_PODHD500_1] = { - .id = "PODHD500", - .name = "POD HD500", - .capabilities = LINE6_CAP_PCM - | LINE6_CAP_HWMON, - .altsetting = 0, + .ctrl_if = 1, .ep_ctrl_r = 0x81, .ep_ctrl_w = 0x01, .ep_audio_r = 0x86, diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 721d12130d0c..a88d7854513b 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1457,7 +1457,7 @@ error: usb_audio_err(chip, "cannot get connectors status: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n", UAC_GET_CUR, validx, idx, cval->val_type); - return ret; + return filter_error(cval, ret); } ucontrol->value.integer.value[0] = val; @@ -1771,10 +1771,16 @@ static void get_connector_control_name(struct usb_mixer_interface *mixer, /* Build a mixer control for a UAC connector control (jack-detect) */ static void build_connector_control(struct usb_mixer_interface *mixer, + const struct usbmix_name_map *imap, struct usb_audio_term *term, bool is_input) { struct snd_kcontrol *kctl; struct usb_mixer_elem_info *cval; + const struct usbmix_name_map *map; + + map = find_map(imap, term->id, 0); + if (check_ignored_ctl(map)) + return; cval = kzalloc(sizeof(*cval), GFP_KERNEL); if (!cval) @@ -1805,8 +1811,12 @@ static void build_connector_control(struct usb_mixer_interface *mixer, usb_mixer_elem_info_free(cval); return; } - get_connector_control_name(mixer, term, is_input, kctl->id.name, - sizeof(kctl->id.name)); + + if (check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name))) + strlcat(kctl->id.name, " Jack", sizeof(kctl->id.name)); + else + get_connector_control_name(mixer, term, is_input, kctl->id.name, + sizeof(kctl->id.name)); kctl->private_free = snd_usb_mixer_elem_free; snd_usb_mixer_add_control(&cval->head, kctl); } @@ -2109,8 +2119,9 @@ static int parse_audio_input_terminal(struct mixer_build *state, int unitid, check_input_term(state, term_id, &iterm); /* Check for jack detection. */ - if (uac_v2v3_control_is_readable(bmctls, control)) - build_connector_control(state->mixer, &iterm, true); + if ((iterm.type & 0xff00) != 0x0100 && + uac_v2v3_control_is_readable(bmctls, control)) + build_connector_control(state->mixer, state->map, &iterm, true); return 0; } @@ -3071,13 +3082,13 @@ static int snd_usb_mixer_controls_badd(struct usb_mixer_interface *mixer, memset(&iterm, 0, sizeof(iterm)); iterm.id = UAC3_BADD_IT_ID4; iterm.type = UAC_BIDIR_TERMINAL_HEADSET; - build_connector_control(mixer, &iterm, true); + build_connector_control(mixer, map->map, &iterm, true); /* Output Term - Insertion control */ memset(&oterm, 0, sizeof(oterm)); oterm.id = UAC3_BADD_OT_ID3; oterm.type = UAC_BIDIR_TERMINAL_HEADSET; - build_connector_control(mixer, &oterm, false); + build_connector_control(mixer, map->map, &oterm, false); } return 0; @@ -3106,7 +3117,8 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) if (map->id == state.chip->usb_id) { state.map = map->map; state.selector_map = map->selector_map; - mixer->ignore_ctl_error = map->ignore_ctl_error; + mixer->connector_map = map->connector_map; + mixer->ignore_ctl_error |= map->ignore_ctl_error; break; } } @@ -3149,10 +3161,11 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) if (err < 0 && err != -EINVAL) return err; - if (uac_v2v3_control_is_readable(le16_to_cpu(desc->bmControls), + if ((state.oterm.type & 0xff00) != 0x0100 && + uac_v2v3_control_is_readable(le16_to_cpu(desc->bmControls), UAC2_TE_CONNECTOR)) { - build_connector_control(state.mixer, &state.oterm, - false); + build_connector_control(state.mixer, state.map, + &state.oterm, false); } } else { /* UAC_VERSION_3 */ struct uac3_output_terminal_descriptor *desc = p; @@ -3174,10 +3187,11 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) if (err < 0 && err != -EINVAL) return err; - if (uac_v2v3_control_is_readable(le32_to_cpu(desc->bmControls), + if ((state.oterm.type & 0xff00) != 0x0100 && + uac_v2v3_control_is_readable(le32_to_cpu(desc->bmControls), UAC3_TE_INSERTION)) { - build_connector_control(state.mixer, &state.oterm, - false); + build_connector_control(state.mixer, state.map, + &state.oterm, false); } } } @@ -3185,10 +3199,32 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) return 0; } +static int delegate_notify(struct usb_mixer_interface *mixer, int unitid, + u8 *control, u8 *channel) +{ + const struct usbmix_connector_map *map = mixer->connector_map; + + if (!map) + return unitid; + + for (; map->id; map++) { + if (map->id == unitid) { + if (control && map->control) + *control = map->control; + if (channel && map->channel) + *channel = map->channel; + return map->delegated_id; + } + } + return unitid; +} + void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid) { struct usb_mixer_elem_list *list; + unitid = delegate_notify(mixer, unitid, NULL, NULL); + for_each_mixer_elem(list, mixer, unitid) { struct usb_mixer_elem_info *info = mixer_elem_list_to_info(list); @@ -3258,6 +3294,8 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer, return; } + unitid = delegate_notify(mixer, unitid, &control, &channel); + for_each_mixer_elem(list, mixer, unitid) count++; diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 65d6d08c96f5..41ec9dc4139b 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -6,6 +6,13 @@ struct media_mixer_ctl; +struct usbmix_connector_map { + u8 id; + u8 delegated_id; + u8 control; + u8 channel; +}; + struct usb_mixer_interface { struct snd_usb_audio *chip; struct usb_host_interface *hostif; @@ -18,6 +25,9 @@ struct usb_mixer_interface { /* the usb audio specification version this interface complies to */ int protocol; + /* optional connector delegation map */ + const struct usbmix_connector_map *connector_map; + /* Sound Blaster remote control stuff */ const struct rc_config *rc_cfg; u32 rc_code; diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 72b575c34860..0260c750e156 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -27,6 +27,7 @@ struct usbmix_ctl_map { u32 id; const struct usbmix_name_map *map; const struct usbmix_selector_map *selector_map; + const struct usbmix_connector_map *connector_map; int ignore_ctl_error; }; @@ -360,13 +361,42 @@ static const struct usbmix_name_map corsair_virtuoso_map[] = { }; /* Some mobos shipped with a dummy HD-audio show the invalid GET_MIN/GET_MAX - * response for Input Gain Pad (id=19, control=12). Skip it. + * response for Input Gain Pad (id=19, control=12) and the connector status + * for SPDIF terminal (id=18). Skip them. */ static const struct usbmix_name_map asus_rog_map[] = { + { 18, NULL }, /* OT, connector control */ { 19, NULL, 12 }, /* FU, Input Gain Pad */ {} }; +/* TRX40 mobos with Realtek ALC1220-VB */ +static const struct usbmix_name_map trx40_mobo_map[] = { + { 18, NULL }, /* OT, IEC958 - broken response, disabled */ + { 19, NULL, 12 }, /* FU, Input Gain Pad - broken response, disabled */ + { 16, "Speaker" }, /* OT */ + { 22, "Speaker Playback" }, /* FU */ + { 7, "Line" }, /* IT */ + { 19, "Line Capture" }, /* FU */ + { 17, "Front Headphone" }, /* OT */ + { 23, "Front Headphone Playback" }, /* FU */ + { 8, "Mic" }, /* IT */ + { 20, "Mic Capture" }, /* FU */ + { 9, "Front Mic" }, /* IT */ + { 21, "Front Mic Capture" }, /* FU */ + { 24, "IEC958 Playback" }, /* FU */ + {} +}; + +static const struct usbmix_connector_map trx40_mobo_connector_map[] = { + { 10, 16 }, /* (Back) Speaker */ + { 11, 17 }, /* Front Headphone */ + { 13, 7 }, /* Line */ + { 14, 8 }, /* Mic */ + { 15, 9 }, /* Front Mic */ + {} +}; + /* * Control map entries */ @@ -498,7 +528,8 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { }, { /* Gigabyte TRX40 Aorus Pro WiFi */ .id = USB_ID(0x0414, 0xa002), - .map = asus_rog_map, + .map = trx40_mobo_map, + .connector_map = trx40_mobo_connector_map, }, { /* ASUS ROG Zenith II */ .id = USB_ID(0x0b05, 0x1916), @@ -510,11 +541,13 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { }, { /* MSI TRX40 Creator */ .id = USB_ID(0x0db0, 0x0d64), - .map = asus_rog_map, + .map = trx40_mobo_map, + .connector_map = trx40_mobo_connector_map, }, { /* MSI TRX40 */ .id = USB_ID(0x0db0, 0x543d), - .map = asus_rog_map, + .map = trx40_mobo_map, + .connector_map = trx40_mobo_connector_map, }, { 0 } /* terminator */ }; diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 02b036b2aefb..a5f65a9a0254 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1509,11 +1509,15 @@ static int snd_microii_spdif_default_get(struct snd_kcontrol *kcontrol, /* use known values for that card: interface#1 altsetting#1 */ iface = usb_ifnum_to_if(chip->dev, 1); - if (!iface || iface->num_altsetting < 2) - return -EINVAL; + if (!iface || iface->num_altsetting < 2) { + err = -EINVAL; + goto end; + } alts = &iface->altsetting[1]; - if (get_iface_desc(alts)->bNumEndpoints < 1) - return -EINVAL; + if (get_iface_desc(alts)->bNumEndpoints < 1) { + err = -EINVAL; + goto end; + } ep = get_endpoint(alts, 0)->bEndpointAddress; err = snd_usb_ctl_msg(chip->dev, diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index e009d584e7d0..a1df4c5b4f8c 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2756,90 +2756,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_MIDI_NOVATION } }, -{ - /* - * Focusrite Scarlett Solo 2nd generation - * Reports that playback should use Synch: Synchronous - * while still providing a feedback endpoint. Synchronous causes - * snapping on some sample rates. - * Force it to use Synch: Asynchronous. - */ - USB_DEVICE(0x1235, 0x8205), - .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 = 1, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = & (const struct audioformat) { - .formats = SNDRV_PCM_FMTBIT_S32_LE, - .channels = 2, - .iface = 1, - .altsetting = 1, - .altset_idx = 1, - .attributes = 0, - .endpoint = 0x01, - .ep_attr = USB_ENDPOINT_XFER_ISOC | - USB_ENDPOINT_SYNC_ASYNC, - .protocol = UAC_VERSION_2, - .rates = SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_88200 | - SNDRV_PCM_RATE_96000 | - SNDRV_PCM_RATE_176400 | - SNDRV_PCM_RATE_192000, - .rate_min = 44100, - .rate_max = 192000, - .nr_rates = 6, - .rate_table = (unsigned int[]) { - 44100, 48000, 88200, - 96000, 176400, 192000 - }, - .clock = 41 - } - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = & (const struct audioformat) { - .formats = SNDRV_PCM_FMTBIT_S32_LE, - .channels = 2, - .iface = 2, - .altsetting = 1, - .altset_idx = 1, - .attributes = 0, - .endpoint = 0x82, - .ep_attr = USB_ENDPOINT_XFER_ISOC | - USB_ENDPOINT_SYNC_ASYNC | - USB_ENDPOINT_USAGE_IMPLICIT_FB, - .protocol = UAC_VERSION_2, - .rates = SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_88200 | - SNDRV_PCM_RATE_96000 | - SNDRV_PCM_RATE_176400 | - SNDRV_PCM_RATE_192000, - .rate_min = 44100, - .rate_max = 192000, - .nr_rates = 6, - .rate_table = (unsigned int[]) { - 44100, 48000, 88200, - 96000, 176400, 192000 - }, - .clock = 41 - } - }, - { - .ifnum = 3, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = -1 - } - } - } -}, /* Access Music devices */ { @@ -3635,4 +3551,18 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } }, +#define ALC1220_VB_DESKTOP(vend, prod) { \ + USB_DEVICE(vend, prod), \ + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { \ + .vendor_name = "Realtek", \ + .product_name = "ALC1220-VB-DT", \ + .profile_name = "Realtek-ALC1220-VB-Desktop", \ + .ifnum = QUIRK_NO_INTERFACE \ + } \ +} +ALC1220_VB_DESKTOP(0x0414, 0xa002), /* Gigabyte TRX40 Aorus Pro WiFi */ +ALC1220_VB_DESKTOP(0x0db0, 0x0d64), /* MSI TRX40 Creator */ +ALC1220_VB_DESKTOP(0x0db0, 0x543d), /* MSI TRX40 */ +#undef ALC1220_VB_DESKTOP + #undef USB_DEVICE_VENDOR_SPEC diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index a8ece1701068..848a4cc25bed 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1687,7 +1687,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ case USB_ID(0x10cb, 0x0103): /* The Bit Opus #3; with fp->dsd_raw */ - case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */ + case USB_ID(0x16d0, 0x06b2): /* NuPrime DAC-10 */ case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */ case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */ case USB_ID(0x16d0, 0x09db): /* NuPrime Audio DAC-9 */ @@ -1806,6 +1806,20 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, */ fp->attributes &= ~UAC_EP_CS_ATTR_FILL_MAX; break; + case USB_ID(0x1235, 0x8200): /* Focusrite Scarlett 2i4 2nd gen */ + case USB_ID(0x1235, 0x8202): /* Focusrite Scarlett 2i2 2nd gen */ + case USB_ID(0x1235, 0x8205): /* Focusrite Scarlett Solo 2nd gen */ + /* + * Reports that playback should use Synch: Synchronous + * while still providing a feedback endpoint. + * Synchronous causes snapping on some sample rates. + * Force it to use Synch: Asynchronous. + */ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + fp->ep_attr &= ~USB_ENDPOINT_SYNCTYPE; + fp->ep_attr |= USB_ENDPOINT_SYNC_ASYNC; + } + break; } } diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 37d290fe9d43..ecaf41265dcd 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -681,6 +681,8 @@ static int usX2Y_rate_set(struct usX2Ydev *usX2Y, int rate) us->submitted = 2*NOOF_SETRATE_URBS; for (i = 0; i < NOOF_SETRATE_URBS; ++i) { struct urb *urb = us->urb[i]; + if (!urb) + continue; if (urb->status) { if (!err) err = -ENODEV; |