diff options
author | Linus Torvalds <torvalds@linux-foundation.org> | 2020-11-06 12:58:11 -0800 |
---|---|---|
committer | Linus Torvalds <torvalds@linux-foundation.org> | 2020-11-06 12:58:11 -0800 |
commit | bb72bbe8f6c70e67c85d773e5c9b04c7fe36a0ab (patch) | |
tree | 72522131eb6a2d5913aee8c5d30d5f936b15cef5 | |
parent | fc7b66ef076644dd646eb9f11563684edc479649 (diff) | |
parent | a6c96672a64f4f0e1bac9f37b5bb57d8ab551b4b (diff) |
Merge tag 'sound-5.10-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Quite a bunch of small fixes that have been gathered since the last
pull, including changes like below:
- HD-audio runtime PM fixes and refactoring
- HD-audio and USB-audio quirks
- SOF warning fix
- Various ASoC device-specific fixes for Intel, Qualcomm, etc"
* tag 'sound-5.10-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (26 commits)
ALSA: usb-audio: Add implicit feedback quirk for Qu-16
ASoC: mchp-spdiftx: Do not set Validity bit(s)
ALSA: usb-audio: Add implicit feedback quirk for MODX
ALSA: usb-audio: add usb vendor id as DSD-capable for Khadas devices
ALSA: hda/realtek - Enable headphone for ASUS TM420
ALSA: hda: prevent undefined shift in snd_hdac_ext_bus_get_link()
ASoC: qcom: lpass-cpu: Fix clock disable failure
ASoC: qcom: lpass-sc7180: Fix MI2S bitwidth field bit positions
ASoC: codecs: wcd9335: Set digital gain range correctly
ASoC: codecs: wcd934x: Set digital gain range correctly
ALSA: hda: Reinstate runtime_allow() for all hda controllers
ALSA: hda: Separate runtime and system suspend
ALSA: hda: Refactor codec PM to use direct-complete optimization
ALSA: hda/realtek - Fixed HP headset Mic can't be detected
ALSA: usb-audio: Add implicit feedback quirk for Zoom UAC-2
ALSA: make snd_kcontrol_new name a normal string
ALSA: fix kernel-doc markups
ASoC: SOF: loader: handle all SOF_IPC_EXT types
ASoC: cs42l51: manage mclk shutdown delay
ASoC: qcom: sdm845: set driver name correctly
...
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") |