diff options
33 files changed, 265 insertions, 117 deletions
diff --git a/include/sound/control.h b/include/sound/control.h index e128cff10dfa..77d9fa10812d 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -42,7 +42,7 @@ struct snd_kcontrol_new { snd_ctl_elem_iface_t iface; /* interface identifier */ unsigned int device; /* device/client number */ unsigned int subdevice; /* subdevice (substream) number */ - const unsigned char *name; /* ASCII name of item */ + const char *name; /* ASCII name of item */ unsigned int index; /* index of item */ unsigned int access; /* access rights */ unsigned int count; /* count of same elements */ diff --git a/include/sound/core.h b/include/sound/core.h index 381a010a1bd4..0462c577d7a3 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -332,7 +332,8 @@ void __snd_printk(unsigned int level, const char *file, int line, #define snd_BUG() WARN(1, "BUG?\n") /** - * Suppress high rates of output when CONFIG_SND_DEBUG is enabled. + * snd_printd_ratelimit - Suppress high rates of output when + * CONFIG_SND_DEBUG is enabled. */ #define snd_printd_ratelimit() printk_ratelimit() diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 2ba5df2c9e23..2336bf9243e1 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1284,8 +1284,8 @@ snd_pcm_sgbuf_get_ptr(struct snd_pcm_substream *substream, unsigned int ofs) } /** - * snd_pcm_sgbuf_chunk_size - Compute the max size that fits within the contig. - * page from the given size + * snd_pcm_sgbuf_get_chunk_size - Compute the max size that fits within the + * contig. page from the given size * @substream: PCM substream * @ofs: byte offset * @size: byte size to examine diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 7184265c0b0d..9555f31c8425 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -144,7 +144,7 @@ struct snd_compr_metadata { __u32 value[8]; } __attribute__((packed, aligned(4))); -/** +/* * compress path ioctl definitions * SNDRV_COMPRESS_GET_CAPS: Query capability of DSP * SNDRV_COMPRESS_GET_CODEC_CAPS: Query capability of a codec diff --git a/sound/core/control.c b/sound/core/control.c index 421ddc76f264..4373de42a5a0 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1925,8 +1925,8 @@ EXPORT_SYMBOL(snd_ctl_unregister_ioctl); #ifdef CONFIG_COMPAT /** - * snd_ctl_unregister_ioctl - de-register the device-specific compat 32bit - * control-ioctls + * snd_ctl_unregister_ioctl_compat - de-register the device-specific compat + * 32bit control-ioctls * @fcn: ioctl callback function to unregister */ int snd_ctl_unregister_ioctl_compat(snd_kctl_ioctl_func_t fcn) diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 4d059ff2b2e4..4d0e8fe535a1 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -356,7 +356,8 @@ int snd_dmaengine_pcm_close(struct snd_pcm_substream *substream) EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_close); /** - * snd_dmaengine_pcm_release_chan_close - Close a dmaengine based PCM substream and release channel + * snd_dmaengine_pcm_close_release_chan - Close a dmaengine based PCM + * substream and release channel * @substream: PCM substream * * Releases the DMA channel associated with the PCM substream. diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index d531e1bc2b81..bda3514c7b2d 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -490,7 +490,7 @@ void snd_pcm_set_ops(struct snd_pcm *pcm, int direction, EXPORT_SYMBOL(snd_pcm_set_ops); /** - * snd_pcm_sync - set the PCM sync id + * snd_pcm_set_sync - set the PCM sync id * @substream: the pcm substream * * Sets the PCM sync identifier for the card. diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 9e0b2d73faf6..47b155a49226 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -112,7 +112,7 @@ void snd_pcm_stream_lock(struct snd_pcm_substream *substream) EXPORT_SYMBOL_GPL(snd_pcm_stream_lock); /** - * snd_pcm_stream_lock - Unlock the PCM stream + * snd_pcm_stream_unlock - Unlock the PCM stream * @substream: PCM substream * * This unlocks the PCM stream that has been locked via snd_pcm_stream_lock(). @@ -595,7 +595,7 @@ static void snd_pcm_sync_stop(struct snd_pcm_substream *substream) } /** - * snd_pcm_hw_param_choose - choose a configuration defined by @params + * snd_pcm_hw_params_choose - choose a configuration defined by @params * @pcm: PCM instance * @params: the hw_params instance * diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index 4d060d5b1db6..b0c0ef824d7d 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -148,6 +148,8 @@ struct hdac_ext_link *snd_hdac_ext_bus_get_link(struct hdac_bus *bus, return NULL; if (bus->idx != bus_idx) return NULL; + if (addr < 0 || addr > 31) + return NULL; list_for_each_entry(hlink, &bus->hlink_list, list) { for (i = 0; i < HDA_MAX_CODECS; i++) { diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index a356c21edb90..4bb58e8b08a8 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2934,7 +2934,7 @@ static void hda_call_codec_resume(struct hda_codec *codec) snd_hdac_leave_pm(&codec->core); } -static int hda_codec_runtime_suspend(struct device *dev) +static int hda_codec_suspend(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); unsigned int state; @@ -2953,7 +2953,7 @@ static int hda_codec_runtime_suspend(struct device *dev) return 0; } -static int hda_codec_runtime_resume(struct device *dev) +static int hda_codec_resume(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); @@ -2967,57 +2967,70 @@ static int hda_codec_runtime_resume(struct device *dev) pm_runtime_mark_last_busy(dev); return 0; } + +static int hda_codec_runtime_suspend(struct device *dev) +{ + return hda_codec_suspend(dev); +} + +static int hda_codec_runtime_resume(struct device *dev) +{ + return hda_codec_resume(dev); +} + #endif /* CONFIG_PM */ #ifdef CONFIG_PM_SLEEP -static int hda_codec_force_resume(struct device *dev) +static int hda_codec_pm_prepare(struct device *dev) +{ + return pm_runtime_suspended(dev); +} + +static void hda_codec_pm_complete(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); - int ret; - ret = pm_runtime_force_resume(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; + if (pm_runtime_suspended(dev) && (codec->jackpoll_interval || + hda_codec_need_resume(codec) || codec->forced_resume)) + pm_request_resume(dev); } static int hda_codec_pm_suspend(struct device *dev) { dev->power.power_state = PMSG_SUSPEND; - return pm_runtime_force_suspend(dev); + return hda_codec_suspend(dev); } static int hda_codec_pm_resume(struct device *dev) { dev->power.power_state = PMSG_RESUME; - return hda_codec_force_resume(dev); + return hda_codec_resume(dev); } static int hda_codec_pm_freeze(struct device *dev) { dev->power.power_state = PMSG_FREEZE; - return pm_runtime_force_suspend(dev); + return hda_codec_suspend(dev); } static int hda_codec_pm_thaw(struct device *dev) { dev->power.power_state = PMSG_THAW; - return hda_codec_force_resume(dev); + return hda_codec_resume(dev); } static int hda_codec_pm_restore(struct device *dev) { dev->power.power_state = PMSG_RESTORE; - return hda_codec_force_resume(dev); + return hda_codec_resume(dev); } #endif /* CONFIG_PM_SLEEP */ /* referred in hda_bind.c */ const struct dev_pm_ops hda_codec_driver_pm = { #ifdef CONFIG_PM_SLEEP + .prepare = hda_codec_pm_prepare, + .complete = hda_codec_pm_complete, .suspend = hda_codec_pm_suspend, .resume = hda_codec_pm_resume, .freeze = hda_codec_pm_freeze, diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index be63ead8161f..68f9668788ea 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 */ -#define AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP (1 << 27) /* Workaround for spurious wakeups after suspend */ +/* 27 unused */ #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 */ @@ -143,6 +143,7 @@ struct azx { unsigned int align_buffer_size:1; unsigned int region_requested:1; unsigned int disabled:1; /* disabled by vga_switcheroo */ + unsigned int pm_prepared:1; /* GTS present */ unsigned int gts_present:1; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 749b88090970..d539f52009a1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -297,8 +297,7 @@ enum { /* PCH for HSW/BDW; with runtime PM */ /* no i915 binding for this as HSW/BDW has another controller for HDMI */ #define AZX_DCAPS_INTEL_PCH \ - (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\ - AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP) + (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME) /* HSW HDMI */ #define AZX_DCAPS_INTEL_HASWELL \ @@ -985,7 +984,7 @@ static void __azx_runtime_suspend(struct azx *chip) display_power(chip, false); } -static void __azx_runtime_resume(struct azx *chip, bool from_rt) +static void __azx_runtime_resume(struct azx *chip) { struct hda_intel *hda = container_of(chip, struct hda_intel, chip); struct hdac_bus *bus = azx_bus(chip); @@ -1002,7 +1001,8 @@ static void __azx_runtime_resume(struct azx *chip, bool from_rt) azx_init_pci(chip); hda_intel_init_chip(chip, true); - if (from_rt) { + /* Avoid codec resume if runtime resume is for system suspend */ + if (!chip->pm_prepared) { list_for_each_codec(codec, &chip->bus) { if (codec->relaxed_resume) continue; @@ -1018,6 +1018,29 @@ static void __azx_runtime_resume(struct azx *chip, bool from_rt) } #ifdef CONFIG_PM_SLEEP +static int azx_prepare(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip; + + chip = card->private_data; + chip->pm_prepared = 1; + + /* HDA controller always requires different WAKEEN for runtime suspend + * and system suspend, so don't use direct-complete here. + */ + return 0; +} + +static void azx_complete(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip; + + chip = card->private_data; + chip->pm_prepared = 0; +} + static int azx_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); @@ -1029,15 +1052,7 @@ static int azx_suspend(struct device *dev) chip = card->private_data; bus = azx_bus(chip); - snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - /* An ugly workaround: direct call of __azx_runtime_suspend() and - * __azx_runtime_resume() for old Intel platforms that suffer from - * spurious wakeups after S3 suspend - */ - if (chip->driver_caps & AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP) - __azx_runtime_suspend(chip); - else - pm_runtime_force_suspend(dev); + __azx_runtime_suspend(chip); if (bus->irq >= 0) { free_irq(bus->irq, chip); bus->irq = -1; @@ -1066,11 +1081,7 @@ static int azx_resume(struct device *dev) if (azx_acquire_irq(chip, 1) < 0) return -EIO; - if (chip->driver_caps & AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP) - __azx_runtime_resume(chip, false); - else - pm_runtime_force_resume(dev); - snd_power_change_state(card, SNDRV_CTL_POWER_D0); + __azx_runtime_resume(chip); trace_azx_resume(chip); return 0; @@ -1118,10 +1129,7 @@ static int azx_runtime_suspend(struct device *dev) chip = card->private_data; /* enable controller wake up event */ - if (snd_power_get_state(card) == SNDRV_CTL_POWER_D0) { - azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | - STATESTS_INT_MASK); - } + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | STATESTS_INT_MASK); __azx_runtime_suspend(chip); trace_azx_runtime_suspend(chip); @@ -1132,18 +1140,14 @@ 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; - __azx_runtime_resume(chip, from_rt); + __azx_runtime_resume(chip); /* disable controller Wake Up event*/ - if (from_rt) { - azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & - ~STATESTS_INT_MASK); - } + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & ~STATESTS_INT_MASK); trace_azx_runtime_resume(chip); return 0; @@ -1177,6 +1181,8 @@ static int azx_runtime_idle(struct device *dev) static const struct dev_pm_ops azx_pm = { SET_SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume) #ifdef CONFIG_PM_SLEEP + .prepare = azx_prepare, + .complete = azx_complete, .freeze_noirq = azx_freeze_noirq, .thaw_noirq = azx_thaw_noirq, #endif @@ -2356,6 +2362,7 @@ static int azx_probe_continue(struct azx *chip) if (azx_has_pm_runtime(chip)) { pm_runtime_use_autosuspend(&pci->dev); + pm_runtime_allow(&pci->dev); pm_runtime_put_autosuspend(&pci->dev); } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f2398721ac1e..6899089d132e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6008,6 +6008,27 @@ static void alc285_fixup_invalidate_dacs(struct hda_codec *codec, snd_hda_override_wcaps(codec, 0x03, 0); } +static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) +{ + switch (codec->core.vendor_id) { + case 0x10ec0274: + case 0x10ec0294: + case 0x10ec0225: + case 0x10ec0295: + case 0x10ec0299: + alc_update_coef_idx(codec, 0x4a, 0x8000, 1 << 15); /* Reset HP JD */ + alc_update_coef_idx(codec, 0x4a, 0x8000, 0 << 15); + break; + case 0x10ec0235: + case 0x10ec0236: + case 0x10ec0255: + case 0x10ec0256: + alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ + alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); + break; + } +} + static void alc295_fixup_chromebook(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -6018,16 +6039,7 @@ static void alc295_fixup_chromebook(struct hda_codec *codec, spec->ultra_low_power = true; break; case HDA_FIXUP_ACT_INIT: - switch (codec->core.vendor_id) { - case 0x10ec0295: - alc_update_coef_idx(codec, 0x4a, 0x8000, 1 << 15); /* Reset HP JD */ - alc_update_coef_idx(codec, 0x4a, 0x8000, 0 << 15); - break; - case 0x10ec0236: - alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ - alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); - break; - } + alc_combo_jack_hp_jd_restart(codec); break; } } @@ -6083,6 +6095,16 @@ static void alc285_fixup_hp_gpio_amp_init(struct hda_codec *codec, alc_write_coef_idx(codec, 0x65, 0x0); } +static void alc274_fixup_hp_headset_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + switch (action) { + case HDA_FIXUP_ACT_INIT: + alc_combo_jack_hp_jd_restart(codec); + break; + } +} + /* for hda_fixup_thinkpad_acpi() */ #include "thinkpad_helper.c" @@ -6277,6 +6299,8 @@ enum { ALC256_FIXUP_INTEL_NUC8_RUGGED, ALC255_FIXUP_XIAOMI_HEADSET_MIC, ALC274_FIXUP_HP_MIC, + ALC274_FIXUP_HP_HEADSET_MIC, + ALC256_FIXUP_ASUS_HPE, }; static const struct hda_fixup alc269_fixups[] = { @@ -7664,6 +7688,23 @@ static const struct hda_fixup alc269_fixups[] = { { } }, }, + [ALC274_FIXUP_HP_HEADSET_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc274_fixup_hp_headset_mic, + .chained = true, + .chain_id = ALC274_FIXUP_HP_MIC + }, + [ALC256_FIXUP_ASUS_HPE] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* Set EAPD high */ + { 0x20, AC_VERB_SET_COEF_INDEX, 0x0f }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x7778 }, + { } + }, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7815,7 +7856,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8729, "HP", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_AMP_INIT), - SND_PCI_QUIRK(0x103c, 0x874e, "HP", ALC274_FIXUP_HP_MIC), SND_PCI_QUIRK(0x103c, 0x8760, "HP", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x877a, "HP", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x877d, "HP", ALC236_FIXUP_HP_MUTE_LED), @@ -7848,6 +7888,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), + SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS), @@ -8339,6 +8380,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1a, 0x90a70130}, {0x1b, 0x90170110}, {0x21, 0x03211020}), + SND_HDA_PIN_QUIRK(0x10ec0274, 0x103c, "HP", ALC274_FIXUP_HP_HEADSET_MIC, + {0x17, 0x90170110}, + {0x19, 0x03a11030}, + {0x21, 0x03211020}), SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4, {0x12, 0x90a60130}, {0x14, 0x90170110}, diff --git a/sound/soc/atmel/mchp-spdiftx.c b/sound/soc/atmel/mchp-spdiftx.c index 82c1eecd2528..3bd350afb743 100644 --- a/sound/soc/atmel/mchp-spdiftx.c +++ b/sound/soc/atmel/mchp-spdiftx.c @@ -487,7 +487,6 @@ static int mchp_spdiftx_hw_params(struct snd_pcm_substream *substream, } mchp_spdiftx_channel_status_write(dev); spin_unlock_irqrestore(&ctrl->lock, flags); - mr |= SPDIFTX_MR_VALID1 | SPDIFTX_MR_VALID2; if (dev->gclk_enabled) { clk_disable_unprepare(dev->gclk); diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 097c4e8d9950..c61b17dc2af8 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -254,8 +254,28 @@ static const struct snd_soc_dapm_widget cs42l51_dapm_widgets[] = { &cs42l51_adcr_mux_controls), }; +static int mclk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); + struct cs42l51_private *cs42l51 = snd_soc_component_get_drvdata(comp); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + return clk_prepare_enable(cs42l51->mclk_handle); + case SND_SOC_DAPM_POST_PMD: + /* Delay mclk shutdown to fulfill power-down sequence requirements */ + msleep(20); + clk_disable_unprepare(cs42l51->mclk_handle); + break; + } + + return 0; +} + static const struct snd_soc_dapm_widget cs42l51_dapm_mclk_widgets[] = { - SND_SOC_DAPM_CLOCK_SUPPLY("MCLK") + SND_SOC_DAPM_SUPPLY("MCLK", SND_SOC_NOPM, 0, 0, mclk_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route cs42l51_routes[] = { diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index f2d9d52ee171..4d2b1ec7c03b 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -618,7 +618,7 @@ static const char * const sb_tx8_mux_text[] = { "ZERO", "RX_MIX_TX8", "DEC8", "DEC8_192" }; -static const DECLARE_TLV_DB_SCALE(digital_gain, 0, 1, 0); +static const DECLARE_TLV_DB_SCALE(digital_gain, -8400, 100, -8400); static const DECLARE_TLV_DB_SCALE(line_gain, 0, 7, 1); static const DECLARE_TLV_DB_SCALE(analog_gain, 0, 25, 1); static const DECLARE_TLV_DB_SCALE(ear_pa_gain, 0, 150, 0); diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 35697b072367..40f682f5dab8 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -551,7 +551,7 @@ struct wcd_iir_filter_ctl { struct soc_bytes_ext bytes_ext; }; -static const DECLARE_TLV_DB_SCALE(digital_gain, 0, 1, 0); +static const DECLARE_TLV_DB_SCALE(digital_gain, -8400, 100, -8400); static const DECLARE_TLV_DB_SCALE(line_gain, 0, 7, 1); static const DECLARE_TLV_DB_SCALE(analog_gain, 0, 25, 1); static const DECLARE_TLV_DB_SCALE(ear_pa_gain, 0, 150, 0); diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index 68e774e69c85..4530b74f5921 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -1026,6 +1026,8 @@ static struct snd_soc_dai_driver wsa881x_dais[] = { .id = 0, .playback = { .stream_name = "SPKR Playback", + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, .rate_max = 48000, .rate_min = 48000, .channels_min = 1, diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index d5bae5d1ab6f..a5b446d5af19 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -15,22 +15,6 @@ config SND_SOC_INTEL_SST_TOPLEVEL if SND_SOC_INTEL_SST_TOPLEVEL -config SND_SST_IPC - tristate - # This option controls the IPC core for HiFi2 platforms - -config SND_SST_IPC_PCI - tristate - select SND_SST_IPC - # This option controls the PCI-based IPC for HiFi2 platforms - # (Medfield, Merrifield). - -config SND_SST_IPC_ACPI - tristate - select SND_SST_IPC - # This option controls the ACPI-based IPC for HiFi2 platforms - # (Baytrail, Cherrytrail) - config SND_SOC_INTEL_SST tristate @@ -57,7 +41,6 @@ config SND_SST_ATOM_HIFI2_PLATFORM config SND_SST_ATOM_HIFI2_PLATFORM_PCI tristate "PCI HiFi2 (Merrifield) Platforms" depends on X86 && PCI - select SND_SST_IPC_PCI select SND_SST_ATOM_HIFI2_PLATFORM help If you have a Intel Merrifield/Edison platform, then @@ -70,7 +53,6 @@ config SND_SST_ATOM_HIFI2_PLATFORM_ACPI tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms" default ACPI depends on X86 && ACPI && PCI - select SND_SST_IPC_ACPI select SND_SST_ATOM_HIFI2_PLATFORM select SND_SOC_ACPI_INTEL_MATCH select IOSF_MBI diff --git a/sound/soc/intel/atom/Makefile b/sound/soc/intel/atom/Makefile index a9326d5ec44c..c66f03f5d8d6 100644 --- a/sound/soc/intel/atom/Makefile +++ b/sound/soc/intel/atom/Makefile @@ -6,4 +6,4 @@ snd-soc-sst-atom-hifi2-platform-objs := sst-mfld-platform-pcm.o \ obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM) += snd-soc-sst-atom-hifi2-platform.o # DSP driver -obj-$(CONFIG_SND_SST_IPC) += sst/ +obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM) += sst/ diff --git a/sound/soc/intel/atom/sst/Makefile b/sound/soc/intel/atom/sst/Makefile index f17c905df3e2..5761d30a5f9d 100644 --- a/sound/soc/intel/atom/sst/Makefile +++ b/sound/soc/intel/atom/sst/Makefile @@ -3,6 +3,6 @@ snd-intel-sst-core-objs := sst.o sst_ipc.o sst_stream.o sst_drv_interface.o sst_ snd-intel-sst-pci-objs += sst_pci.o snd-intel-sst-acpi-objs += sst_acpi.o -obj-$(CONFIG_SND_SST_IPC) += snd-intel-sst-core.o -obj-$(CONFIG_SND_SST_IPC_PCI) += snd-intel-sst-pci.o -obj-$(CONFIG_SND_SST_IPC_ACPI) += snd-intel-sst-acpi.o +obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM) += snd-intel-sst-core.o +obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM_PCI) += snd-intel-sst-pci.o +obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM_ACPI) += snd-intel-sst-acpi.o diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 3ea4602dfb3e..9a4b3d0973f6 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -401,17 +401,40 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_soc_dpcm *dpcm = container_of( - params, struct snd_soc_dpcm, hw_params); - struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; - struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; + struct snd_soc_dpcm *dpcm, *rtd_dpcm = NULL; + + /* + * The following loop will be called only for playback stream + * In this platform, there is only one playback device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) { + rtd_dpcm = dpcm; + break; + } + + /* + * This following loop will be called only for capture stream + * In this platform, there is only one capture device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) { + rtd_dpcm = dpcm; + break; + } + + if (!rtd_dpcm) + return -EINVAL; + + /* + * The above 2 loops are mutually exclusive based on the stream direction, + * thus rtd_dpcm variable will never be overwritten + */ /* * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ - if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || - !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || - !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { + if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) { rate->min = rate->max = 48000; chan->min = chan->max = 2; snd_mask_none(fmt); @@ -421,7 +444,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, * The speaker on the SSP0 supports S16_LE and not S24_LE. * thus changing the mask here */ - if (!strcmp(be_dai_link->name, "SSP0-Codec")) + if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec")) snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); return 0; diff --git a/sound/soc/intel/catpt/dsp.c b/sound/soc/intel/catpt/dsp.c index 7d2968571951..9e807b941732 100644 --- a/sound/soc/intel/catpt/dsp.c +++ b/sound/soc/intel/catpt/dsp.c @@ -267,9 +267,12 @@ static int catpt_dsp_select_lpclock(struct catpt_dev *cdev, bool lp, bool waiti) reg, (reg & CATPT_ISD_DCPWM), 500, 10000); if (ret) { - dev_err(cdev->dev, "await WAITI timeout\n"); - mutex_unlock(&cdev->clk_mutex); - return ret; + dev_warn(cdev->dev, "await WAITI timeout\n"); + /* no signal - only high clock selection allowed */ + if (lp) { + mutex_unlock(&cdev->clk_mutex); + return 0; + } } } diff --git a/sound/soc/intel/catpt/pcm.c b/sound/soc/intel/catpt/pcm.c index f78018c857b8..ba653ebea7d1 100644 --- a/sound/soc/intel/catpt/pcm.c +++ b/sound/soc/intel/catpt/pcm.c @@ -667,7 +667,17 @@ static int catpt_dai_pcm_new(struct snd_soc_pcm_runtime *rtm, break; } + /* see if this is a new configuration */ + if (!memcmp(&cdev->devfmt[devfmt.iface], &devfmt, sizeof(devfmt))) + return 0; + + pm_runtime_get_sync(cdev->dev); + ret = catpt_ipc_set_device_format(cdev, &devfmt); + + pm_runtime_mark_last_busy(cdev->dev); + pm_runtime_put_autosuspend(cdev->dev); + if (ret) return CATPT_IPC_ERROR(ret); diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index c2c1eb16fcc0..26e7d9a7198f 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -630,15 +630,34 @@ static struct snd_soc_codec_conf mt8183_da7219_rt1015_codec_conf[] = { }, }; +static const struct snd_kcontrol_new mt8183_da7219_rt1015_snd_controls[] = { + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + +static const +struct snd_soc_dapm_widget mt8183_da7219_rt1015_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), + SND_SOC_DAPM_PINCTRL("TDM_OUT_PINCTRL", + "aud_tdm_out_on", "aud_tdm_out_off"), +}; + +static const struct snd_soc_dapm_route mt8183_da7219_rt1015_dapm_routes[] = { + {"Left Spk", NULL, "Left SPO"}, + {"Right Spk", NULL, "Right SPO"}, + {"I2S Playback", NULL, "TDM_OUT_PINCTRL"}, +}; + static struct snd_soc_card mt8183_da7219_rt1015_card = { .name = "mt8183_da7219_rt1015", .owner = THIS_MODULE, - .controls = mt8183_da7219_max98357_snd_controls, - .num_controls = ARRAY_SIZE(mt8183_da7219_max98357_snd_controls), - .dapm_widgets = mt8183_da7219_max98357_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(mt8183_da7219_max98357_dapm_widgets), - .dapm_routes = mt8183_da7219_max98357_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(mt8183_da7219_max98357_dapm_routes), + .controls = mt8183_da7219_rt1015_snd_controls, + .num_controls = ARRAY_SIZE(mt8183_da7219_rt1015_snd_controls), + .dapm_widgets = mt8183_da7219_rt1015_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(mt8183_da7219_rt1015_dapm_widgets), + .dapm_routes = mt8183_da7219_rt1015_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(mt8183_da7219_rt1015_dapm_routes), .dai_link = mt8183_da7219_dai_links, .num_links = ARRAY_SIZE(mt8183_da7219_dai_links), .aux_dev = &mt8183_da7219_max98357_headset_dev, diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index ba2aca301a9b..9d17c87445a9 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -80,6 +80,12 @@ static int lpass_cpu_daiops_startup(struct snd_pcm_substream *substream, dev_err(dai->dev, "error in enabling mi2s osr clk: %d\n", ret); return ret; } + ret = clk_prepare(drvdata->mi2s_bit_clk[dai->driver->id]); + if (ret) { + dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); + clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); + return ret; + } return 0; } @@ -88,9 +94,8 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream, { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); - clk_disable_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); - clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); + clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); } static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, @@ -303,10 +308,10 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); - ret = clk_prepare_enable(drvdata->mi2s_bit_clk[id]); + ret = clk_enable(drvdata->mi2s_bit_clk[id]); if (ret) { dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); - clk_disable_unprepare(drvdata->mi2s_osr_clk[id]); + clk_disable(drvdata->mi2s_osr_clk[id]); return ret; } @@ -324,6 +329,7 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, if (ret) dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); + clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]); break; } diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index c6292f9e613f..bc998d501600 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -188,7 +188,7 @@ static struct lpass_variant sc7180_data = { .micmode = REG_FIELD_ID(0x1000, 4, 8, 3, 0x1000), .micmono = REG_FIELD_ID(0x1000, 3, 3, 3, 0x1000), .wssrc = REG_FIELD_ID(0x1000, 2, 2, 3, 0x1000), - .bitwidth = REG_FIELD_ID(0x1000, 0, 0, 3, 0x1000), + .bitwidth = REG_FIELD_ID(0x1000, 0, 1, 3, 0x1000), .rdma_dyncclk = REG_FIELD_ID(0xC000, 21, 21, 5, 0x1000), .rdma_bursten = REG_FIELD_ID(0xC000, 20, 20, 5, 0x1000), diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index ab1bf23c21a6..6c2760e27ea6 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -17,6 +17,7 @@ #include "qdsp6/q6afe.h" #include "../codecs/rt5663.h" +#define DRIVER_NAME "sdm845" #define DEFAULT_SAMPLE_RATE_48K 48000 #define DEFAULT_MCLK_RATE 24576000 #define TDM_BCLK_RATE 6144000 @@ -552,6 +553,7 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) if (!data) return -ENOMEM; + card->driver_name = DRIVER_NAME; card->dapm_widgets = sdm845_snd_widgets; card->num_dapm_widgets = ARRAY_SIZE(sdm845_snd_widgets); card->dev = dev; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ea3986a46c12..05a085f6dc7c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2341,7 +2341,7 @@ struct snd_soc_dai *snd_soc_register_dai(struct snd_soc_component *component, } /** - * snd_soc_unregister_dai - Unregister DAIs from the ASoC core + * snd_soc_unregister_dais - Unregister DAIs from the ASoC core * * @component: The component for which the DAIs should be unregistered */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 980f2c330b87..7f87b449f950 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1276,7 +1276,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, } /** - * snd_soc_dapm_get_connected_widgets - query audio path and it's widgets. + * snd_soc_dapm_dai_get_connected_widgets - query audio path and it's widgets. * @dai: the soc DAI. * @stream: stream direction. * @list: list of active widgets for this stream. diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c index 68ed454f7ddf..ba9ed66f98bc 100644 --- a/sound/soc/sof/loader.c +++ b/sound/soc/sof/loader.c @@ -118,6 +118,11 @@ int snd_sof_fw_parse_ext_data(struct snd_sof_dev *sdev, u32 bar, u32 offset) case SOF_IPC_EXT_CC_INFO: ret = get_cc_info(sdev, ext_hdr); break; + case SOF_IPC_EXT_UNUSED: + case SOF_IPC_EXT_PROBE_INFO: + case SOF_IPC_EXT_USER_ABI_INFO: + /* They are supported but we don't do anything here */ + break; default: dev_warn(sdev->dev, "warning: unknown ext header type %d size 0x%x\n", ext_hdr->type, ext_hdr->hdr.size); diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b401ee894e1b..a860303cc522 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -336,6 +336,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, switch (subs->stream->chip->usb_id) { case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */ + case USB_ID(0x22f0, 0x0006): /* Allen&Heath Qu-16 */ ep = 0x81; ifnum = 3; goto add_sync_ep_from_ifnum; @@ -345,6 +346,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ifnum = 2; goto add_sync_ep_from_ifnum; case USB_ID(0x2466, 0x8003): /* Fractal Audio Axe-Fx II */ + case USB_ID(0x0499, 0x172a): /* Yamaha MODX */ ep = 0x86; ifnum = 2; goto add_sync_ep_from_ifnum; @@ -352,6 +354,10 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ep = 0x81; ifnum = 2; goto add_sync_ep_from_ifnum; + case USB_ID(0x1686, 0xf029): /* Zoom UAC-2 */ + ep = 0x82; + ifnum = 2; + goto add_sync_ep_from_ifnum; case USB_ID(0x1397, 0x0001): /* Behringer UFX1604 */ case USB_ID(0x1397, 0x0002): /* Behringer UFX1204 */ ep = 0x81; diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index b4fa80ef730d..c989ad8052ae 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1800,6 +1800,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case 0x278b: /* Rotel? */ case 0x292b: /* Gustard/Ess based devices */ case 0x2ab6: /* T+A devices */ + case 0x3353: /* Khadas devices */ case 0x3842: /* EVGA */ case 0xc502: /* HiBy devices */ if (fp->dsd_raw) |