May 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH 00/14] ALSA: emu10k1: various improvements to the DSP-based mixer code
by Oswald Buddenhagen 12 May '23

12 May '23

Oswald Buddenhagen (14): ALSA: emu10k1: make tone control switch mono ALSA: emu10k1: roll up loops in DSP setup code for Audigy ALSA: emu10k1: fix+optimize E-MU stereo capture DSP code ALSA: emu10k1: simplify snd_emu10k1_audigy_dsp_convert_32_to_2x16() ALSA: emu10k1: apply channel delay hack to all E-MU cards ALSA: emu10k1: simplify tone control switch DSP code ALSA: emu10k1: properly assert DSP init constraints ALSA: emu10k1: polish audigy GPR allocation ALSA: emu10k1: fix non-zero mixer control defaults in highres mode ALSA: emu10k1: validate min/max values of translated controls ALSA: emu10k1: omit non-applicable mixer controls for E-MU cards ALSA: emu10k1: skip mic capture PCM for cards without AC97 codec ALSA: emu10k1: enable bit-exact playback, part 1: DSP attenuation ALSA: emu10k1: enable bit-exact playback, part 2: voice attenuation Documentation/sound/cards/audigy-mixer.rst | 2 +- Documentation/sound/cards/sb-live-mixer.rst | 2 +- include/sound/emu10k1.h | 10 +- include/uapi/sound/emu10k1.h | 8 +- sound/pci/emu10k1/emu10k1.c | 8 +- sound/pci/emu10k1/emufx.c | 763 +++++++++----------- sound/pci/emu10k1/emumixer.c | 15 +- sound/pci/emu10k1/emupcm.c | 4 +- 8 files changed, 369 insertions(+), 443 deletions(-) -- 2.40.0.152.g15d061e6df

2 18

[PATCH] ALSA: hda/realtek: Fix mute and micmute LEDs for yet another HP laptop
by Kai-Heng Feng 12 May '23

12 May '23

There's yet another laptop that needs the fixup to enable mute and micmute LEDs. So do it accordingly. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 172ffc2c332b..2e3f87e9b4ee 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9480,6 +9480,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8b8f, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b92, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b96, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8b97, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8bf0, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), -- 2.34.1

2 1

[PATCH] ALSA: usb-audio: Add a sample rate workaround for Line6 Pod Go
by Takashi Iwai 12 May '23

12 May '23

Line6 Pod Go (0e41:424b) requires the similar workaround for the fixed 48k sample rate like other Line6 models. This patch adds the corresponding entry to line6_parse_audio_format_rate_quirk(). Reported-by: John Humlick <john(a)humlick.org> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/format.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/format.c b/sound/usb/format.c index 4b1c5ba121f3..ab5fed9f55b6 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -423,6 +423,7 @@ static int line6_parse_audio_format_rates_quirk(struct snd_usb_audio *chip, case USB_ID(0x0e41, 0x4248): /* Line6 Helix >= fw 2.82 */ case USB_ID(0x0e41, 0x4249): /* Line6 Helix Rack >= fw 2.82 */ case USB_ID(0x0e41, 0x424a): /* Line6 Helix LT >= fw 2.82 */ + case USB_ID(0x0e41, 0x424b): /* Line6 Pod Go */ case USB_ID(0x19f7, 0x0011): /* Rode Rodecaster Pro */ return set_fixed_rate(fp, 48000, SNDRV_PCM_RATE_48000); } -- 2.35.3

1 0

[PATCH] ALSA: emu10k1: don't create regular S/PDIF controls for E-MU cards
by Oswald Buddenhagen 12 May '23

12 May '23

These ports are unused on these cards. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- sound/pci/emu10k1/emumixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 610700be1e37..48f0d3f8b8e7 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -2055,7 +2055,7 @@ int snd_emu10k1_mixer(struct snd_emu10k1 *emu, mix->attn[0] = 0xffff; } - if (! emu->card_capabilities->ecard) { /* FIXME: APS has these controls? */ + if (!emu->card_capabilities->ecard && !emu->card_capabilities->emu_model) { /* sb live! and audigy */ kctl = snd_ctl_new1(&snd_emu10k1_spdif_mask_control, emu); if (!kctl) -- 2.40.0.152.g15d061e6df

2 1

[PATCH 1/4] ALSA: emu10k1: automate encoding of sub-register definitions
by Oswald Buddenhagen 12 May '23

12 May '23

The idea to encode the bitfield manipulation in the register address is quite clever, but doing that by hand is ugly and error-prone. So derive it automatically from the mask instead. Macros cannot #define other macros, so we now declare enums instead. This also adds macros for decoding the register definitions. These will be used by later commits. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- include/sound/emu10k1.h | 113 ++++++++++++++++++---------------------- 1 file changed, 50 insertions(+), 63 deletions(-) diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 8fe80dcee71b..f4e05a2897e8 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -38,6 +38,22 @@ #define IP_TO_CP(ip) ((ip == 0) ? 0 : (((0x00001000uL | (ip & 0x00000FFFL)) << (((ip >> 12) & 0x000FL) + 4)) & 0xFFFF0000uL)) +#define SUB_REG_NC(reg, field, mask) \ + enum { \ + field ## _MASK = mask, \ + field = reg | \ + (__builtin_ctz(mask) << 16) | \ + (__builtin_popcount(mask) << 24), \ + }; +#define SUB_REG(reg, field, mask) SUB_REG_NC(reg, reg ## _ ## field, mask) + +#define REG_SHIFT(r) (((r) >> 16) & 0x1f) +#define REG_SIZE(r) (((r) >> 24) & 0x1f) +#define REG_MASK0(r) ((1U << REG_SIZE(r)) - 1U) +#define REG_MASK(r) (REG_MASK0(r) << REG_SHIFT(r)) +#define REG_VAL_GET(r, v) ((v & REG_MASK(r)) >> REG_SHIFT(r)) +#define REG_VAL_PUT(r, v) ((v) << REG_SHIFT(r)) + // Audigy specify registers are prefixed with 'A_' /************************************************************************************************/ @@ -148,12 +164,10 @@ #define INTE_MIDIRXENABLE 0x00000001 /* Enable MIDI receive-buffer-empty interrupts */ #define WC 0x10 /* Wall Clock register */ -#define WC_SAMPLECOUNTER_MASK 0x03FFFFC0 /* Sample periods elapsed since reset */ -#define WC_SAMPLECOUNTER 0x14060010 -#define WC_CURRENTCHANNEL_MASK 0x0000003F /* Channel [0..63] currently being serviced */ +SUB_REG(WC, SAMPLECOUNTER, 0x03FFFFC0) /* Sample periods elapsed since reset */ +SUB_REG(WC, CURRENTCHANNEL, 0x0000003F) /* Channel [0..63] currently being serviced */ /* NOTE: Each channel takes 1/64th of a sample */ /* period to be serviced. */ -#define WC_CURRENTCHANNEL 0x06000010 #define HCFG 0x14 /* Hardware config register */ /* NOTE: There is no reason to use the legacy */ @@ -225,9 +239,8 @@ /* async audio source */ #define HCFG_LOCKSOUNDCACHE 0x00000008 /* 1 = Cancel bustmaster accesses to soundcache */ /* NOTE: This should generally never be used. */ -#define HCFG_LOCKTANKCACHE_MASK 0x00000004 /* 1 = Cancel bustmaster accesses to tankcache */ +SUB_REG(HCFG, LOCKTANKCACHE, 0x00000004) /* 1 = Cancel bustmaster accesses to tankcache */ /* NOTE: This should generally never be used. */ -#define HCFG_LOCKTANKCACHE 0x01020014 #define HCFG_MUTEBUTTONENABLE 0x00000002 /* 1 = Master mute button sets AUDIOENABLE = 0. */ /* NOTE: This is a 'cheap' way to implement a */ /* master mute function on the mute button, and */ @@ -382,55 +395,38 @@ // which the current registers "swerve" gradually. #define CPF 0x00 /* Current pitch and fraction register */ -#define CPF_CURRENTPITCH_MASK 0xffff0000 /* Current pitch (linear, 0x4000 == unity pitch shift) */ -#define CPF_CURRENTPITCH 0x10100000 +SUB_REG(CPF, CURRENTPITCH, 0xffff0000) /* Current pitch (linear, 0x4000 == unity pitch shift) */ #define CPF_STEREO_MASK 0x00008000 /* 1 = Even channel interleave, odd channel locked */ #define CPF_STOP_MASK 0x00004000 /* 1 = Current pitch forced to 0 */ #define CPF_FRACADDRESS_MASK 0x00003fff /* Linear fractional address of the current channel */ #define PTRX 0x01 /* Pitch target and send A/B amounts register */ -#define PTRX_PITCHTARGET_MASK 0xffff0000 /* Pitch target of specified channel */ -#define PTRX_PITCHTARGET 0x10100001 -#define PTRX_FXSENDAMOUNT_A_MASK 0x0000ff00 /* Linear level of channel output sent to FX send bus A */ -#define PTRX_FXSENDAMOUNT_A 0x08080001 -#define PTRX_FXSENDAMOUNT_B_MASK 0x000000ff /* Linear level of channel output sent to FX send bus B */ -#define PTRX_FXSENDAMOUNT_B 0x08000001 +SUB_REG(PTRX, PITCHTARGET, 0xffff0000) /* Pitch target of specified channel */ +SUB_REG(PTRX, FXSENDAMOUNT_A, 0x0000ff00) /* Linear level of channel output sent to FX send bus A */ +SUB_REG(PTRX, FXSENDAMOUNT_B, 0x000000ff) /* Linear level of channel output sent to FX send bus B */ #define CVCF 0x02 /* Current volume and filter cutoff register */ -#define CVCF_CURRENTVOL_MASK 0xffff0000 /* Current linear volume of specified channel */ -#define CVCF_CURRENTVOL 0x10100002 -#define CVCF_CURRENTFILTER_MASK 0x0000ffff /* Current filter cutoff frequency of specified channel */ -#define CVCF_CURRENTFILTER 0x10000002 +SUB_REG(CVCF, CURRENTVOL, 0xffff0000) /* Current linear volume of specified channel */ +SUB_REG(CVCF, CURRENTFILTER, 0x0000ffff) /* Current filter cutoff frequency of specified channel */ #define VTFT 0x03 /* Volume target and filter cutoff target register */ -#define VTFT_VOLUMETARGET_MASK 0xffff0000 /* Volume target of specified channel */ -#define VTFT_VOLUMETARGET 0x10100003 -#define VTFT_FILTERTARGET_MASK 0x0000ffff /* Filter cutoff target of specified channel */ -#define VTFT_FILTERTARGET 0x10000003 +SUB_REG(VTFT, VOLUMETARGET, 0xffff0000) /* Volume target of specified channel */ +SUB_REG(VTFT, FILTERTARGET, 0x0000ffff) /* Filter cutoff target of specified channel */ #define Z1 0x05 /* Filter delay memory 1 register */ #define Z2 0x04 /* Filter delay memory 2 register */ #define PSST 0x06 /* Send C amount and loop start address register */ -#define PSST_FXSENDAMOUNT_C_MASK 0xff000000 /* Linear level of channel output sent to FX send bus C */ - -#define PSST_FXSENDAMOUNT_C 0x08180006 - -#define PSST_LOOPSTARTADDR_MASK 0x00ffffff /* Loop start address of the specified channel */ -#define PSST_LOOPSTARTADDR 0x18000006 +SUB_REG(PSST, FXSENDAMOUNT_C, 0xff000000) /* Linear level of channel output sent to FX send bus C */ +SUB_REG(PSST, LOOPSTARTADDR, 0x00ffffff) /* Loop start address of the specified channel */ #define DSL 0x07 /* Send D amount and loop end address register */ -#define DSL_FXSENDAMOUNT_D_MASK 0xff000000 /* Linear level of channel output sent to FX send bus D */ - -#define DSL_FXSENDAMOUNT_D 0x08180007 - -#define DSL_LOOPENDADDR_MASK 0x00ffffff /* Loop end address of the specified channel */ -#define DSL_LOOPENDADDR 0x18000007 +SUB_REG(DSL, FXSENDAMOUNT_D, 0xff000000) /* Linear level of channel output sent to FX send bus D */ +SUB_REG(DSL, LOOPENDADDR, 0x00ffffff) /* Loop end address of the specified channel */ #define CCCA 0x08 /* Filter Q, interp. ROM, byte size, cur. addr register */ -#define CCCA_RESONANCE_MASK 0xf0000000 /* Lowpass filter resonance (Q) height */ -#define CCCA_RESONANCE 0x041c0008 +SUB_REG(CCCA, RESONANCE, 0xf0000000) /* Lowpass filter resonance (Q) height */ #define CCCA_INTERPROM_MASK 0x0e000000 /* Selects passband of interpolation ROM */ /* 1 == full band, 7 == lowpass */ /* ROM 0 is used when pitch shifting downward or less */ @@ -447,27 +443,24 @@ #define CCCA_INTERPROM_7 0x0e000000 /* Select interpolation ROM 7 */ #define CCCA_8BITSELECT 0x01000000 /* 1 = Sound memory for this channel uses 8-bit samples */ /* 8-bit samples are unsigned, 16-bit ones signed */ -#define CCCA_CURRADDR_MASK 0x00ffffff /* Current address of the selected channel */ -#define CCCA_CURRADDR 0x18000008 +SUB_REG(CCCA, CURRADDR, 0x00ffffff) /* Current address of the selected channel */ #define CCR 0x09 /* Cache control register */ -#define CCR_CACHEINVALIDSIZE 0x07190009 -#define CCR_CACHEINVALIDSIZE_MASK 0xfe000000 /* Number of invalid samples before the read address */ +SUB_REG(CCR, CACHEINVALIDSIZE, 0xfe000000) /* Number of invalid samples before the read address */ #define CCR_CACHELOOPFLAG 0x01000000 /* 1 = Cache has a loop service pending */ #define CCR_INTERLEAVEDSAMPLES 0x00800000 /* 1 = A cache service will fetch interleaved samples */ /* Auto-set from CPF_STEREO_MASK */ #define CCR_WORDSIZEDSAMPLES 0x00400000 /* 1 = A cache service will fetch word sized samples */ /* Auto-set from CCCA_8BITSELECT */ -#define CCR_READADDRESS 0x06100009 -#define CCR_READADDRESS_MASK 0x003f0000 /* Next cached sample to play */ -#define CCR_LOOPINVALSIZE 0x0000fe00 /* Number of invalid samples in cache prior to loop */ +SUB_REG(CCR, READADDRESS, 0x003f0000) /* Next cached sample to play */ +SUB_REG(CCR, LOOPINVALSIZE, 0x0000fe00) /* Number of invalid samples in cache prior to loop */ /* NOTE: This is valid only if CACHELOOPFLAG is set */ #define CCR_LOOPFLAG 0x00000100 /* Set for a single sample period when a loop occurs */ -#define CCR_CACHELOOPADDRHI 0x000000ff /* CLP_LOOPSTARTADDR's hi byte if CACHELOOPFLAG is set */ +SUB_REG(CCR, CACHELOOPADDRHI, 0x000000ff) /* CLP_LOOPSTARTADDR's hi byte if CACHELOOPFLAG is set */ #define CLP 0x0a /* Cache loop register (valid if CCR_CACHELOOPFLAG = 1) */ /* NOTE: This register is normally not used */ -#define CLP_CACHELOOPADDR 0x0000ffff /* Cache loop address low word */ +SUB_REG(CLP, CACHELOOPADDR, 0x0000ffff) /* Cache loop address low word */ #define FXRT 0x0b /* Effects send routing register */ /* NOTE: It is illegal to assign the same routing to */ @@ -537,20 +530,17 @@ #define IP_UNITY 0x0000e000 /* Unity pitch shift */ #define IFATN 0x19 /* Initial filter cutoff and attenuation register */ -#define IFATN_FILTERCUTOFF_MASK 0x0000ff00 /* Initial filter cutoff frequency in exponential units */ +SUB_REG(IFATN, FILTERCUTOFF, 0x0000ff00) /* Initial filter cutoff frequency in exponential units */ /* 6 most significant bits are semitones */ /* 2 least significant bits are fractions */ -#define IFATN_FILTERCUTOFF 0x08080019 -#define IFATN_ATTENUATION_MASK 0x000000ff /* Initial attenuation in 0.375dB steps */ -#define IFATN_ATTENUATION 0x08000019 +SUB_REG(IFATN, ATTENUATION, 0x000000ff) /* Initial attenuation in 0.375dB steps */ #define PEFE 0x1a /* Pitch envelope and filter envelope amount register */ -#define PEFE_PITCHAMOUNT_MASK 0x0000ff00 /* Pitch envlope amount */ +SUB_REG(PEFE, PITCHAMOUNT, 0x0000ff00) /* Pitch envlope amount */ /* Signed 2's complement, +/- one octave peak extremes */ -#define PEFE_PITCHAMOUNT 0x0808001a -#define PEFE_FILTERAMOUNT_MASK 0x000000ff /* Filter envlope amount */ +SUB_REG(PEFE, FILTERAMOUNT, 0x000000ff) /* Filter envlope amount */ /* Signed 2's complement, +/- six octaves peak extremes */ -#define PEFE_FILTERAMOUNT 0x0800001a + #define FMMOD 0x1b /* Vibrato/filter modulation from LFO register */ #define FMMOD_MODVIBRATO 0x0000ff00 /* Vibrato LFO modulation depth */ @@ -793,22 +783,19 @@ #define SRCS_SPDIFRATE_96 0x00080000 #define MICIDX 0x63 /* Microphone recording buffer index register */ -#define MICIDX_MASK 0x0000ffff /* 16-bit value */ -#define MICIDX_IDX 0x10000063 +SUB_REG(MICIDX, IDX, 0x0000ffff) #define ADCIDX 0x64 /* ADC recording buffer index register */ -#define ADCIDX_MASK 0x0000ffff /* 16 bit index field */ -#define ADCIDX_IDX 0x10000064 +SUB_REG(ADCIDX, IDX, 0x0000ffff) #define A_ADCIDX 0x63 -#define A_ADCIDX_IDX 0x10000063 +SUB_REG(A_ADCIDX, IDX, 0x0000ffff) #define A_MICIDX 0x64 -#define A_MICIDX_IDX 0x10000064 +SUB_REG(A_MICIDX, IDX, 0x0000ffff) #define FXIDX 0x65 /* FX recording buffer index register */ -#define FXIDX_MASK 0x0000ffff /* 16-bit value */ -#define FXIDX_IDX 0x10000065 +SUB_REG(FXIDX, IDX, 0x0000ffff) /* The 32-bit HLIEx and HLIPx registers all have one bit per channel control/status */ #define HLIEL 0x66 /* Channel half loop interrupt enable low register */ @@ -852,8 +839,8 @@ #define A_SPDIF_44100 0x00000080 #define A_SPDIF_MUTED 0x000000c0 -#define A_I2S_CAPTURE_RATE_MASK 0x00000e00 /* This sets the capture PCM rate, but it is */ -#define A_I2S_CAPTURE_RATE 0x03090076 /* unclear if this sets the ADC rate as well. */ +SUB_REG_NC(A_EHC, A_I2S_CAPTURE_RATE, 0x00000e00) /* This sets the capture PCM rate, but it is */ + /* unclear if this sets the ADC rate as well. */ #define A_I2S_CAPTURE_48000 0x0 #define A_I2S_CAPTURE_192000 0x1 #define A_I2S_CAPTURE_96000 0x2 -- 2.40.0.152.g15d061e6df

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[PATCH v2 0/2] ASoC SSM3515 codec driver
by Martin Povišer 12 May '23

12 May '23

Hi, Analog Devices SSM3515 is a simple speaker amp that Apple is using in their 2021 iMacs, possibly elsewhere. This is a v2 of an ASoC driver for those, to be used with upcoming ASoC support (in upstream kernels anyway) of those platforms. Martin Changes since v1: https://lore.kernel.org/asahi/0fcc4617-e706-8743-e65d-1ff99f38ddac@linaro.o… - drop 'adi,ana-gain' from the binding - move device reset to bus probe - fix schema copyright, other schema tweaks Martin Povišer (2): ASoC: dt-bindings: Add adi,ssm3515 amp schema ASoC: ssm3515: Add new amp driver .../bindings/sound/adi,ssm3515.yaml | 49 ++ MAINTAINERS | 2 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ssm3515.c | 448 ++++++++++++++++++ 5 files changed, 507 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/adi,ssm3515.yaml create mode 100644 sound/soc/codecs/ssm3515.c -- 2.38.3

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[PATCH v4] ASoC: dt-bindings: nau8825: Convert to dtschema
by David Lin 12 May '23

12 May '23

Convert the NAU8825 audio CODEC bindings to DT schema. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> Changes: V3 -> V4: - remove the description from interrupts, clocks and clock-names properites - add sound-dai-cells property and update the dts example V2 -> V3: - refine node name from "nau8825" to "codec" for generic purpose V1 -> V2: - add interrupts properties - add maximum to nuvoton,jack-insert-debounce and nuvoton,jack-eject-debounce properites - add a enum item for nuvoton,short-key-debounce properites - add default value for most properites - add maxItems to clocks properites and mclk entries to clock-names properites - refine wrong indentation from clocks and clock-names - refine dts example for interrupts and clocks - remove headset label from dts example --- .../devicetree/bindings/sound/nau8825.txt | 111 -------- .../bindings/sound/nuvoton,nau8825.yaml | 239 ++++++++++++++++++ 2 files changed, 239 insertions(+), 111 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nau8825.txt create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8825.yaml diff --git a/Documentation/devicetree/bindings/sound/nau8825.txt b/Documentation/devicetree/bindings/sound/nau8825.txt deleted file mode 100644 index a9c34526f4cb..000000000000 --- a/Documentation/devicetree/bindings/sound/nau8825.txt +++ /dev/null @@ -1,111 +0,0 @@ -Nuvoton NAU8825 audio codec - -This device supports I2C only. - -Required properties: - - compatible : Must be "nuvoton,nau8825" - - - reg : the I2C address of the device. This is either 0x1a (CSB=0) or 0x1b (CSB=1). - -Optional properties: - - nuvoton,jkdet-enable: Enable jack detection via JKDET pin. - - nuvoton,jkdet-pull-enable: Enable JKDET pin pull. If set - pin pull enabled, - otherwise pin in high impedance state. - - nuvoton,jkdet-pull-up: Pull-up JKDET pin. If set then JKDET pin is pull up, otherwise pull down. - - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low. - - - nuvoton,vref-impedance: VREF Impedance selection - 0 - Open - 1 - 25 kOhm - 2 - 125 kOhm - 3 - 2.5 kOhm - - - nuvoton,micbias-voltage: Micbias voltage level. - 0 - VDDA - 1 - VDDA - 2 - VDDA * 1.1 - 3 - VDDA * 1.2 - 4 - VDDA * 1.3 - 5 - VDDA * 1.4 - 6 - VDDA * 1.53 - 7 - VDDA * 1.53 - - - nuvoton,sar-threshold-num: Number of buttons supported - - nuvoton,sar-threshold: Impedance threshold for each button. Array that contains up to 8 buttons configuration. SAR value is calculated as - SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) - where MICBIAS is configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by 'nuvoton,sar-voltage', R - button impedance. - Refer datasheet section 10.2 for more information about threshold calculation. - - - nuvoton,sar-hysteresis: Button impedance measurement hysteresis. - - - nuvoton,sar-voltage: Reference voltage for button impedance measurement. - 0 - VDDA - 1 - VDDA - 2 - VDDA * 1.1 - 3 - VDDA * 1.2 - 4 - VDDA * 1.3 - 5 - VDDA * 1.4 - 6 - VDDA * 1.53 - 7 - VDDA * 1.53 - - - nuvoton,sar-compare-time: SAR compare time - 0 - 500 ns - 1 - 1 us - 2 - 2 us - 3 - 4 us - - - nuvoton,sar-sampling-time: SAR sampling time - 0 - 2 us - 1 - 4 us - 2 - 8 us - 3 - 16 us - - - nuvoton,short-key-debounce: Button short key press debounce time. - 0 - 30 ms - 1 - 50 ms - 2 - 100 ms - 3 - 30 ms - - - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - - - nuvoton,crosstalk-enable: make crosstalk function enable if set. - - - nuvoton,adcout-drive-strong: make the drive strength of ADCOUT IO PIN strong if set. - Otherwise, the drive keeps normal strength. - - - nuvoton,adc-delay-ms: Delay (in ms) to make input path stable and avoid pop noise. The - default value is 125 and range between 125 to 500 ms. - - - clocks: list of phandle and clock specifier pairs according to common clock bindings for the - clocks described in clock-names - - clock-names: should include "mclk" for the MCLK master clock - -Example: - - headset: nau8825@1a { - compatible = "nuvoton,nau8825"; - reg = <0x1a>; - interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(E, 6) IRQ_TYPE_LEVEL_LOW>; - nuvoton,jkdet-enable; - nuvoton,jkdet-pull-enable; - nuvoton,jkdet-pull-up; - nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; - nuvoton,vref-impedance = <2>; - nuvoton,micbias-voltage = <6>; - // Setup 4 buttons impedance according to Android specification - nuvoton,sar-threshold-num = <4>; - nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>; - nuvoton,sar-hysteresis = <1>; - nuvoton,sar-voltage = <0>; - nuvoton,sar-compare-time = <0>; - nuvoton,sar-sampling-time = <0>; - nuvoton,short-key-debounce = <2>; - nuvoton,jack-insert-debounce = <7>; - nuvoton,jack-eject-debounce = <7>; - nuvoton,crosstalk-enable; - - clock-names = "mclk"; - clocks = <&tegra_pmc TEGRA_PMC_CLK_OUT_2>; - }; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8825.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8825.yaml new file mode 100644 index 000000000000..a54f194a0b49 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8825.yaml @@ -0,0 +1,239 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nuvoton,nau8825.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NAU8825 audio CODEC + +maintainers: + - John Hsu <KCHSU0(a)nuvoton.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - nuvoton,nau8825 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + nuvoton,jkdet-enable: + description: + Enable jack detection via JKDET pin. + type: boolean + + nuvoton,jkdet-pull-enable: + description: + Enable JKDET pin pull. + If set - pin pull enabled, otherwise pin in high impedance state. + type: boolean + + nuvoton,jkdet-pull-up: + description: + Pull-up JKDET pin. + If set then JKDET pin is pull up, otherwise pull down. + type: boolean + + nuvoton,jkdet-polarity: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + JKDET pin polarity. + enum: + - 0 # active high + - 1 # active low + default: 1 + + nuvoton,vref-impedance: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + VREF Impedance selection. + enum: + - 0 # Open + - 1 # 25 kOhm + - 2 # 125 kOhm + - 3 # 2.5 kOhm + default: 2 + + nuvoton,micbias-voltage: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Micbias voltage level. + enum: + - 0 # VDDA + - 1 # VDDA + - 2 # VDDA * 1.1 + - 3 # VDDA * 1.2 + - 4 # VDDA * 1.3 + - 5 # VDDA * 1.4 + - 6 # VDDA * 1.53 + - 7 # VDDA * 1.53 + default: 6 + + nuvoton,sar-threshold-num: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Number of buttons supported. + minimum: 1 + maximum: 4 + default: 4 + + nuvoton,sar-threshold: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: + Impedance threshold for each button. Array that contains up to 8 buttons + configuration. SAR value is calculated as + SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) where MICBIAS is + configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by + 'nuvoton,sar-voltage', R - button impedance. + Refer datasheet section 10.2 for more information about threshold + calculation. + minItems: 1 + maxItems: 4 + items: + minimum: 0 + maximum: 255 + + nuvoton,sar-hysteresis: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Button impedance measurement hysteresis. + default: 0 + + nuvoton,sar-voltage: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Reference voltage for button impedance measurement. + enum: + - 0 # VDDA + - 1 # VDDA + - 2 # VDDA * 1.1 + - 3 # VDDA * 1.2 + - 4 # VDDA * 1.3 + - 5 # VDDA * 1.4 + - 6 # VDDA * 1.53 + - 7 # VDDA * 1.53 + default: 6 + + nuvoton,sar-compare-time: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + SAR compare time. + enum: + - 0 # 500 ns + - 1 # 1 us + - 2 # 2 us + - 3 # 4 us + default: 1 + + nuvoton,sar-sampling-time: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + SAR sampling time. + enum: + - 0 # 2 us + - 1 # 4 us + - 2 # 8 us + - 3 # 16 us + default: 1 + + nuvoton,short-key-debounce: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Button short key press debounce time. + enum: + - 0 # 30 ms + - 1 # 50 ms + - 2 # 100 ms + - 3 # 30 ms + default: 3 + + nuvoton,jack-insert-debounce: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + number from 0 to 7 that sets debounce time to 2^(n+2) ms. + maximum: 7 + default: 7 + + nuvoton,jack-eject-debounce: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + number from 0 to 7 that sets debounce time to 2^(n+2) ms + maximum: 7 + default: 0 + + nuvoton,crosstalk-enable: + description: + make crosstalk function enable if set. + type: boolean + + nuvoton,adcout-drive-strong: + description: + make the drive strength of ADCOUT IO PIN strong if set. + Otherwise, the drive keeps normal strength. + type: boolean + + nuvoton,adc-delay-ms: + description: + Delay (in ms) to make input path stable and avoid pop noise. + The default value is 125 and range between 125 to 500 ms. + minimum: 125 + maximum: 500 + default: 125 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: mclk + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1a { + #sound-dai-cells = <0>; + compatible = "nuvoton,nau8825"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = <38 IRQ_TYPE_LEVEL_LOW>; + nuvoton,jkdet-enable; + nuvoton,jkdet-pull-enable; + nuvoton,jkdet-pull-up; + nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; + nuvoton,vref-impedance = <2>; + nuvoton,micbias-voltage = <6>; + // Setup 4 buttons impedance according to Android specification + nuvoton,sar-threshold-num = <4>; + nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>; + nuvoton,sar-hysteresis = <1>; + nuvoton,sar-voltage = <0>; + nuvoton,sar-compare-time = <0>; + nuvoton,sar-sampling-time = <0>; + nuvoton,short-key-debounce = <2>; + nuvoton,jack-insert-debounce = <7>; + nuvoton,jack-eject-debounce = <7>; + nuvoton,crosstalk-enable; + + clock-names = "mclk"; + clocks = <&tegra_pmc 1>; + }; + }; -- 2.25.1

3 2

[PATCH 0/6] Fix soc compress playback deadlock in 5.15
by yixuanjiang 12 May '23

12 May '23

The patch in 5.15 aa9ff6a4955f ("ASoC: soc-compress: Reposition and add pcm_mutex") will cause the deadlock when close compress playback. It needs backport patch b7898396f4 ("ASoC: soc-pcm: Fix and cleanup DPCM locking commit") to prevent the deadlock issue. Rootcause is patch b7898396f4 will remove pcm_mutex in soc_pcm_hw_clean then patch aa9ff6a4955f add pcm_mutex acqurie back in snd_compr_free to fix regression cause from b7898396f4. If only apply aa9ff6a4955f, it will acqurie pcm_mutex twice cause the deadlock. There are total 6 patches on these related backport [ 212.063445] sysrq: Show Blocked State [ 212.064964] Call trace: [ 212.064974] __switch_to+0x244/0x454 [ 212.065075] __schedule+0x5fc/0xbdc [ 212.065092] schedule+0x154/0x258 [ 212.065103] schedule_preempt_disabled+0x30/0x50 [ 212.065115] __mutex_lock+0x36c/0x794 [ 212.065135] __mutex_lock_slowpath+0x1c/0x2c [ 212.065145] soc_pcm_hw_clean+0x6c/0x364 [ 212.065177] dpcm_be_dai_hw_free+0x20c/0x304 [ 212.065188] soc_compr_free_fe+0x130/0x448 [ 212.065230] snd_compr_free+0xac/0x148 [ 212.065316] __fput+0x104/0x438 [ 212.065347] ____fput+0x18/0x28 [ 212.065409] task_work_run+0x144/0x1f8 [ 212.065474] do_notify_resume+0x1d24/0x27b0 [ 212.065518] el0_svc+0x68/0xc4 [ 212.065601] el0t_64_sync_handler+0x8c/0xfc [ 212.065610] el0t_64_sync+0x1b0/0x1b4 *** BLURB HERE *** Pierre-Louis Bossart (4): ASoC: soc-pcm: use GFP_ATOMIC for dpcm structure ASoC: soc-pcm: align BE 'atomicity' with that of the FE ASoC: soc-pcm: test refcount before triggering ASoC: soc-pcm: fix BE handling of PAUSE_RELEASE Takashi Iwai (2): ASoC: soc-pcm: Fix and cleanup DPCM locking ASoC: soc-pcm: serialize BE triggers include/sound/soc-dpcm.h | 2 + include/sound/soc.h | 2 - sound/soc/soc-core.c | 1 - sound/soc/soc-pcm.c | 351 +++++++++++++++++++++++++++------------ 4 files changed, 246 insertions(+), 110 deletions(-) -- 2.40.1.521.gf1e218fcd8-goog

2 7

Looking for alsaunmute script
by GitHub issues - opened 12 May '23

12 May '23

alsa-project/alsa-utils issue #211 was opened from DenverEllis: I have a friend that is having trouble with their legion 7 laptop. Perusing a very long kernel thread I saw mention of an alsaunmute script and I am having a hard time tracking it down. I believ perex authored it and I saw it mentioned that it was a part of alsa-utils, but alas I have had no luck in finding this script. Issue URL : https://github.com/alsa-project/alsa-utils/issues/211 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH AUTOSEL 5.10 2/3] ALSA: hda: Glenfly: add HD Audio PCI IDs and HDMI Codec Vendor IDs.
by Sasha Levin 11 May '23

11 May '23

From: jasontao <jasontao(a)glenfly.com> [ Upstream commit c51e431052e2eacfb23fbf6b39bc6c8770d9827a ] Add a set of HD Audio PCI IDS, and the HDMI codec vendor IDs for Glenfly Gpus. - In default_bdl_pos_adj, set bdl to 128 as Glenfly Gpus have hardware limitation, need to increase hdac interrupt interval. - In azx_first_init, enable polling mode for Glenfly Gpu. When the codec complete the command, it sends interrupt and writes response entries to memory, howerver, the write requests sometimes are not actually synchronized to memory when driver handle hdac interrupt on Glenfly Gpus. If the RIRB status is not updated in the interrupt handler, azx_rirb_get_response keeps trying to recevie a response from rirb until 1s timeout. Enabling polling mode for Glenfly Gpu can fix the issue. - In patch_gf_hdmi, set Glenlfy Gpu Codec's no_sticky_stream as it need driver to do actual clean-ups for the linked codec when switch from one codec to another. Signed-off-by: jasontao <jasontao(a)glenfly.com> Signed-off-by: Reaper Li <reaperlioc(a)glenfly.com> Link: https://lore.kernel.org/r/20230426013059.4329-1-reaperlioc@glenfly.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 21 +++++++++++++++++++++ sound/pci/hda/patch_hdmi.c | 22 ++++++++++++++++++++++ 2 files changed, 43 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index de1fe604905f3..1f641712233ef 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -264,6 +264,7 @@ enum { AZX_DRIVER_ATI, AZX_DRIVER_ATIHDMI, AZX_DRIVER_ATIHDMI_NS, + AZX_DRIVER_GFHDMI, AZX_DRIVER_VIA, AZX_DRIVER_SIS, AZX_DRIVER_ULI, @@ -386,6 +387,7 @@ static const char * const driver_short_names[] = { [AZX_DRIVER_ATI] = "HDA ATI SB", [AZX_DRIVER_ATIHDMI] = "HDA ATI HDMI", [AZX_DRIVER_ATIHDMI_NS] = "HDA ATI HDMI", + [AZX_DRIVER_GFHDMI] = "HDA GF HDMI", [AZX_DRIVER_VIA] = "HDA VIA VT82xx", [AZX_DRIVER_SIS] = "HDA SIS966", [AZX_DRIVER_ULI] = "HDA ULI M5461", @@ -1783,6 +1785,12 @@ static int default_bdl_pos_adj(struct azx *chip) } switch (chip->driver_type) { + /* + * increase the bdl size for Glenfly Gpus for hardware + * limitation on hdac interrupt interval + */ + case AZX_DRIVER_GFHDMI: + return 128; case AZX_DRIVER_ICH: case AZX_DRIVER_PCH: return 1; @@ -1902,6 +1910,12 @@ static int azx_first_init(struct azx *chip) pci_write_config_dword(pci, PCI_BASE_ADDRESS_1, 0); } #endif + /* + * Fix response write request not synced to memory when handle + * hdac interrupt on Glenfly Gpus + */ + if (chip->driver_type == AZX_DRIVER_GFHDMI) + bus->polling_mode = 1; err = pci_request_regions(pci, "ICH HD audio"); if (err < 0) @@ -2011,6 +2025,7 @@ static int azx_first_init(struct azx *chip) chip->playback_streams = ATIHDMI_NUM_PLAYBACK; chip->capture_streams = ATIHDMI_NUM_CAPTURE; break; + case AZX_DRIVER_GFHDMI: case AZX_DRIVER_GENERIC: default: chip->playback_streams = ICH6_NUM_PLAYBACK; @@ -2756,6 +2771,12 @@ static const struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x1002, 0xab38), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | AZX_DCAPS_PM_RUNTIME }, + /* GLENFLY */ + { PCI_DEVICE(0x6766, PCI_ANY_ID), + .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, + .class_mask = 0xffffff, + .driver_data = AZX_DRIVER_GFHDMI | AZX_DCAPS_POSFIX_LPIB | + AZX_DCAPS_NO_MSI | AZX_DCAPS_NO_64BIT }, /* VIA VT8251/VT8237A */ { PCI_DEVICE(0x1106, 0x3288), .driver_data = AZX_DRIVER_VIA }, /* VIA GFX VT7122/VX900 */ diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 1afe9cddb69eb..bd151465f2089 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4287,6 +4287,22 @@ static int patch_via_hdmi(struct hda_codec *codec) return patch_simple_hdmi(codec, VIAHDMI_CVT_NID, VIAHDMI_PIN_NID); } +static int patch_gf_hdmi(struct hda_codec *codec) +{ + int err; + + err = patch_generic_hdmi(codec); + if (err) + return err; + + /* + * Glenfly GPUs have two codecs, stream switches from one codec to + * another, need to do actual clean-ups in codec_cleanup_stream + */ + codec->no_sticky_stream = 1; + return 0; +} + /* * patch entries */ @@ -4376,6 +4392,12 @@ HDA_CODEC_ENTRY(0x10de009f, "GPU 9f HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de00a0, "GPU a0 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de8001, "MCP73 HDMI", patch_nvhdmi_2ch), HDA_CODEC_ENTRY(0x10de8067, "MCP67/68 HDMI", patch_nvhdmi_2ch), +HDA_CODEC_ENTRY(0x67663d82, "Arise 82 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d83, "Arise 83 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d84, "Arise 84 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d85, "Arise 85 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d86, "Arise 86 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d87, "Arise 87 HDMI/DP", patch_gf_hdmi), HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP", patch_via_hdmi), HDA_CODEC_ENTRY(0x11069f81, "VX900 HDMI/DP", patch_via_hdmi), HDA_CODEC_ENTRY(0x11069f84, "VX11 HDMI/DP", patch_generic_hdmi), -- 2.39.2

1 0