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snd_hda_intel initialization failure with Xen PCI passthrough
by Jason Andryuk 24 Mar '22

24 Mar '22

Hi, I'm running Xen hypervisor and using PCI passthrough to assign an Intel HDA audio device (00:1f.3 Audio device: Intel Corporation Cannon Point-LP High Definition Audio Controller (rev 30)) to a Xen HVM virtual machine. I do this for both Linux 5.4.185 and a different Windows 10 VM (only one at a time). The Windows VM seems to work every time. The Linux VM has issues after the first VM boot. This is one boot of the physical hardware and multiple boots of the virtual machines. For Linux, on first boot, the sound card is detected and works properly. After that, things usually don't work. I just ran a reboot loop and it was: 1st boot - audio detected and working 2 & 3 - no audio 4th - audio detected and working 5 - 20 - no audio For boots 2, 3, 5-7, dmesg shows: [ 0.760401] hdaudio hdaudioC0D0: no AFG or MFG node found [ 0.760415] snd_hda_intel 0000:00:06.0: no codecs initialized For boots 8+, the errors changed to: [ 0.783397] hdaudio hdaudioC0D0: cannot read sub nodes for FG 0x10 [ 0.783413] snd_hda_intel 0000:00:06.0: no codecs initialized At this point, I booted a Windows 10 VM and audio works Trying to boot Linux again gives a new error message [ 0.789041] snd_hda_intel 0000:00:06.0: Unknown capability 0 [ 1.811205] snd_hda_intel 0000:00:06.0: No response from codec, resetting bus: last cmd=0x0eef0004 [ 1.811246] hdaudio hdaudioC0D0: cannot read sub nodes for FG 0x10ee [ 1.811263] snd_hda_intel 0000:00:06.0: no codecs initialized Reboot VM and it's back to: [ 0.775917] hdaudio hdaudioC0D0: no AFG or MFG node found [ 0.775932] snd_hda_intel 0000:00:06.0: no codecs initialized Reboot VM and again: [ 0.789069] hdaudio hdaudioC0D0: cannot read sub nodes for FG 0x10 [ 0.789084] snd_hda_intel 0000:00:06.0: no codecs initialized Reboot physical laptop: 1. boot Windows 10 - audio works 2. boot Linux - audio works 3. reboot Linux - no audio [ 0.773111] hdaudio hdaudioC0D0: no AFG or MFG node found [ 0.773151] snd_hda_intel 0000:00:06.0: no codecs initialized This seems to me like Windows does a better job resetting the card to get the audio hardware working. Any suggestions on what to investigate? Thanks, Jason

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[PATCH 1/1] ALSA: hda/realtek: Add mute and micmut LED support for Zbook Fury 17 G9
by Kai-Heng Feng 24 Mar '22

24 Mar '22

Zbook Fury 17 G9 requires the same ALC285_FIXUP_HP_GPIO_LED quirk to make its audio LEDs work. So apply the quirk, and make it the last one since it's an LED quirk. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f6ee67f41c45a..e88fbef57c40c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7011,6 +7011,7 @@ enum { ALC287_FIXUP_LEGION_16ACHG6, ALC287_FIXUP_CS35L41_I2C_2, ALC245_FIXUP_CS35L41_SPI_2, + ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED, ALC245_FIXUP_CS35L41_SPI_4, ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED, ALC285_FIXUP_HP_SPEAKERS_MICMUTE_LED, @@ -8776,6 +8777,12 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cs35l41_fixup_spi_two, }, + [ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs35l41_fixup_spi_two, + .chained = true, + .chain_id = ALC285_FIXUP_HP_GPIO_LED, + }, [ALC245_FIXUP_CS35L41_SPI_4] = { .type = HDA_FIXUP_FUNC, .v.func = cs35l41_fixup_spi_four, @@ -9031,7 +9038,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x89ac, "HP EliteBook 640 G9", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89ae, "HP EliteBook 650 G9", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89c3, "Zbook Studio G9", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), - SND_PCI_QUIRK(0x103c, 0x89c6, "Zbook Fury 17 G9", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x103c, 0x89c6, "Zbook Fury 17 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89ca, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 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

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RE: [PATCH 00/19] ASoC: fsl_micfil: Driver updates
by S.J. Wang 24 Mar '22

24 Mar '22

Hi > > This series has a bunch of cleanups for the FSL MICFIL driver. There is not > much chance for regressions in this series as the driver currently can't work > at all. The MICFIL needs multififo support in the i.MX SDMA engine which is > added with this series, see 10/19. > > The multififo support is selected in the dma phandle arguments in the device > tree, the transfer type must be '25' aka IMX_DMATYPE_MULTI_SAI. > This is set already to 25 in the upstream i.MX8M[NM] dtsi files, but the SDMA > driver silently ignores unsupported values instead of throwing an error. This > is fixed in this series and multififo support is added. > > I think the series should go via the ASoC tree, so I'll need an ack from the > DMA guys for 9/19 and 10/10. Thanks for this series, basically I am ok with the fsl_micfil part, just a comments for sdma change as we have discussed. Best regards Wang Shengjiu > > Sascha > > Sascha Hauer (19): > ASoC: fsl_micfil: Drop unnecessary register read > ASoC: fsl_micfil: Drop unused register read > ASoC: fsl_micfil: drop fsl_micfil_set_mclk_rate() > ASoC: fsl_micfil: do not define SHIFT/MASK for single bits > ASoC: fsl_micfil: use GENMASK to define register bit fields > ASoC: fsl_micfil: use clear/set bits > ASoC: fsl_micfil: drop error messages from failed register accesses > ASoC: fsl_micfil: drop unused variables > dma: imx-sdma: error out on unsupported transfer types > dma: imx-sdma: Add multi fifo support > ASoC: fsl_micfil: add multi fifo support > ASoC: fsl_micfil: use define for OSR default value > ASoC: fsl_micfil: Drop get_pdm_clk() > ASoC: fsl_micfil: simplify clock setting > ASoC: fsl_micfil: rework quality setting > ASoC: fsl_micfil: drop unused include > ASoC: fsl_micfil: drop only once used defines > ASoC: fsl_micfil: drop support for undocumented property > ASoC: fsl_micfil: fold fsl_set_clock_params() into its only user > > drivers/dma/imx-sdma.c | 71 ++++- > include/linux/platform_data/dma-imx.h | 7 + > sound/soc/fsl/fsl_micfil.c | 368 +++++++++----------------- > sound/soc/fsl/fsl_micfil.h | 269 +++++-------------- > 4 files changed, 265 insertions(+), 450 deletions(-) > > -- > 2.30.2

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[v6 0/4] ASoC: mediatek: mt8192: support rt1015p_rt5682s
by Jiaxin Yu 24 Mar '22

24 Mar '22

The series reuses mt8192-mt6359-rt10150rt5682.c for supporting machine driver with rt1015p speaker amplifier and rt5682s headset codec. Changes from v5: - "mediatek,headset-codec" and "mediatek,speaker-codec" drop prefix and move to properties from patternProperties. Changes form v4: - split a large patch into three small patches for easy reviewing - correct coding style Changes from v3: - fix build error: too many arguments for format [-Werror-format-extra-args] Changes from v2: - fix build warnings such as "data argument not used by format string" Changes from v1: - uses the snd_soc_of_get_dai_link_codecs to complete the configuration of dai_link's codecs - uses definitions to simplifies card name and compatible name Jiaxin Yu (4): ASoC: dt-bindings: mt8192-mt6359: add new compatible and new properties ASoC: mediatek: mt8192: refactor for I2S3 DAI link of speaker ASoC: mediatek: mt8192: refactor for I2S8/I2S9 DAI links of headset ASoC: mediatek: mt8192: support rt1015p_rt5682s .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 32 +++ sound/soc/mediatek/Kconfig | 1 + .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 201 +++++++++++------- 3 files changed, 156 insertions(+), 78 deletions(-) -- 2.18.0

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[PATCH v4 1/3] rename tas2764 to tas27xx-Makefile and Kconfig
by Raphael-Xu 23 Mar '22

23 Mar '22

Signed-off-by: Raphael-Xu <13691752556(a)139.com> --- sound/soc/codecs/Kconfig | 13 ++++++++----- sound/soc/codecs/Makefile | 4 ++-- 2 files changed, 10 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d3e5ae8310ef..95f45f3b6b17 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -212,8 +212,8 @@ config SND_SOC_ALL_CODECS imply SND_SOC_STI_SAS imply SND_SOC_TAS2552 imply SND_SOC_TAS2562 - imply SND_SOC_TAS2764 imply SND_SOC_TAS2770 + imply SND_SOC_TAS27XX imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 @@ -1459,14 +1459,17 @@ config SND_SOC_TAS2562 tristate "Texas Instruments TAS2562 Mono Audio amplifier" depends on I2C -config SND_SOC_TAS2764 - tristate "Texas Instruments TAS2764 Mono Audio amplifier" - depends on I2C - config SND_SOC_TAS2770 tristate "Texas Instruments TAS2770 speaker amplifier" depends on I2C +config SND_SOC_TAS27XX + tristate "Texas Instruments TAS27xx speaker amplifier" + depends on I2C + help + Enable support for Texas Instruments TAS2764 and TAS2780 + power amplifiers + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ac7f20972470..2cd2b2352ee7 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -334,7 +334,7 @@ snd-soc-simple-amplifier-objs := simple-amplifier.o snd-soc-tpa6130a2-objs := tpa6130a2.o snd-soc-tas2552-objs := tas2552.o snd-soc-tas2562-objs := tas2562.o -snd-soc-tas2764-objs := tas2764.o +snd-soc-tas27xx-objs := tas27xx.o # Mux snd-soc-simple-mux-objs := simple-mux.o @@ -570,7 +570,7 @@ obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_STI_SAS) += snd-soc-sti-sas.o obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o obj-$(CONFIG_SND_SOC_TAS2562) += snd-soc-tas2562.o -obj-$(CONFIG_SND_SOC_TAS2764) += snd-soc-tas2764.o +obj-$(CONFIG_SND_SOC_TAS27XX) += snd-soc-tas27xx.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o -- 2.35.1

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[PATCH -next] ASoC: tlv320adc3xxx: Fix build error when GPIOLIB is not set
by Baisong Zhong 23 Mar '22

23 Mar '22

sound/soc/codecs/tlv320adc3xxx.c:1352:21: error: implicit declaration of function ‘devm_gpiod_get’; did you mean ‘devm_gpio_free’? sound/soc/codecs/tlv320adc3xxx.c:1352:50: error: ‘GPIOD_OUT_LOW’ undeclared (first use in this function); did you mean ‘GPIOF_INIT_LOW’? sound/soc/codecs/tlv320adc3xxx.c:1400:2: error: implicit declaration of function ‘gpiod_set_value_cansleep’; did you mean ‘gpio_set_value_cansleep’? Fixes: e9a3b57efd28 ("ASoC: codec: tlv320adc3xxx: New codec driver") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Baisong Zhong <zhongbaisong(a)huawei.com> --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f46a22660103..1400fc75fd65 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1535,7 +1535,7 @@ config SND_SOC_TFA989X config SND_SOC_TLV320ADC3XXX tristate "Texas Instruments TLV320ADC3001/3101 audio ADC" - depends on I2C + depends on I2C && GPIOLIB help Enable support for Texas Instruments TLV320ADC3001 and TLV320ADC3101 ADCs. -- 2.25.1

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Re: realtek ALC1220 : recording samplerate 10x too slow
by fenugrec＠mail.com 23 Mar '22

23 Mar '22

Hi, > Does it communicate with alsa-lib pulse plugin? Or is it the direct > ALSA device access? Ah, good question. I'm not sure how to determine that but looking into it. Tried strace but I wasn't sure what to look for in the deluge of info. > Do you mean that the sound from arecord gets garbled? Correct : playing back the file just produced (arecord -f S16_LE -c 2 -r 44100 arec_44k.wav), sound is normal until I started audacity, then the rest of the audio is 10x accelerated. > arecord, is it over pulseaudio (with alsa-lib pulse plugin)? Probably ? alsa-lib is installed and pulseaudio running normally, and I just gave default options to arecord. > No, I mean the way to trigger without audacity I understand. I haven't found a simpler way (yet). Thanks, Chris

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[PATCH v3] dt-bindings: mfd: Fix pinctrl node name warnings
by Rob Herring 23 Mar '22

23 Mar '22

The recent addition pinctrl.yaml in commit c09acbc499e8 ("dt-bindings: pinctrl: use pinctrl.yaml") resulted in some node name warnings: Documentation/devicetree/bindings/mfd/cirrus,lochnagar.example.dt.yaml: \ lochnagar-pinctrl: $nodename:0: 'lochnagar-pinctrl' does not match '^(pinctrl|pinmux)(@[0-9a-f]+)?$' Documentation/devicetree/bindings/mfd/cirrus,madera.example.dt.yaml: \ codec@1a: $nodename:0: 'codec@1a' does not match '^(pinctrl|pinmux)(@[0-9a-f]+)?$' Documentation/devicetree/bindings/mfd/brcm,cru.example.dt.yaml: \ pin-controller@1c0: $nodename:0: 'pin-controller@1c0' does not match '^(pinctrl|pinmux)(@[0-9a-f]+)?$' Fix the node names to the preferred 'pinctrl'. For cirrus,madera, nothing from pinctrl.yaml schema is used, so just drop the reference. Fixes: c09acbc499e8 ("dt-bindings: pinctrl: use pinctrl.yaml") Cc: Rafał Miłecki <rafal(a)milecki.pl> Signed-off-by: Rob Herring <robh(a)kernel.org> --- v2: - Fix lochnagar-pinctrl nodename in example v3: - And fix lochnagar-pinctrl nodename in 'required'. Sigh... --- Documentation/devicetree/bindings/mfd/brcm,cru.yaml | 4 ++-- Documentation/devicetree/bindings/mfd/cirrus,lochnagar.yaml | 6 +++--- .../devicetree/bindings/pinctrl/cirrus,madera.yaml | 3 --- 3 files changed, 5 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/mfd/brcm,cru.yaml b/Documentation/devicetree/bindings/mfd/brcm,cru.yaml index be4a2df71c25..b85819fbb07c 100644 --- a/Documentation/devicetree/bindings/mfd/brcm,cru.yaml +++ b/Documentation/devicetree/bindings/mfd/brcm,cru.yaml @@ -39,7 +39,7 @@ patternProperties: '^phy@[a-f0-9]+$': $ref: ../phy/bcm-ns-usb2-phy.yaml - '^pin-controller@[a-f0-9]+$': + '^pinctrl@[a-f0-9]+$': $ref: ../pinctrl/brcm,ns-pinmux.yaml '^syscon@[a-f0-9]+$': @@ -94,7 +94,7 @@ examples: reg = <0x180 0x4>; }; - pin-controller@1c0 { + pinctrl@1c0 { compatible = "brcm,bcm4708-pinmux"; reg = <0x1c0 0x24>; reg-names = "cru_gpio_control"; diff --git a/Documentation/devicetree/bindings/mfd/cirrus,lochnagar.yaml b/Documentation/devicetree/bindings/mfd/cirrus,lochnagar.yaml index c00ad3e21c21..ad285cb480c9 100644 --- a/Documentation/devicetree/bindings/mfd/cirrus,lochnagar.yaml +++ b/Documentation/devicetree/bindings/mfd/cirrus,lochnagar.yaml @@ -126,7 +126,7 @@ properties: clock-frequency: const: 12288000 - lochnagar-pinctrl: + pinctrl: type: object $ref: /schemas/pinctrl/cirrus,lochnagar.yaml# @@ -255,7 +255,7 @@ required: - reg - reset-gpios - lochnagar-clk - - lochnagar-pinctrl + - pinctrl additionalProperties: false @@ -293,7 +293,7 @@ examples: clock-frequency = <32768>; }; - lochnagar-pinctrl { + pinctrl { compatible = "cirrus,lochnagar-pinctrl"; gpio-controller; diff --git a/Documentation/devicetree/bindings/pinctrl/cirrus,madera.yaml b/Documentation/devicetree/bindings/pinctrl/cirrus,madera.yaml index c85f759ae5a3..8a90d8273767 100644 --- a/Documentation/devicetree/bindings/pinctrl/cirrus,madera.yaml +++ b/Documentation/devicetree/bindings/pinctrl/cirrus,madera.yaml @@ -107,9 +107,6 @@ properties: additionalProperties: false -allOf: - - $ref: "pinctrl.yaml#" - required: - pinctrl-0 - pinctrl-names -- 2.32.0

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realtek ALC1220 : recording samplerate 10x too slow
by cblondin＠mail.com 23 Mar '22

23 Mar '22

Hi, I was redirected here by pulseaudio maintainers for an issue I thought was at that layer : https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1344 After a fresh reboot, arecord / parecord work fine, and the level meters in pavucontrol react normally. If I start audacity, the card gets in a state where the level meters become sluggish, and recording either drops 90% of samples, or records approximately 10x too fast. Playback is always ok, unaffected. I cannot fix that state other than by a reboot. The obvious "closing audacity", restarting the pulseaudio daemon, have no effect. Tried kernels 5.15.24 and a recent 5.17; same problem. I posted some detailed debugging info on that pulseaudio issue tracker, as well as pastebinned alsa-info.sh output here: http://alsa-project.org/db/?f=7ee784003379252acfef25de5fecbdfd559fe342 The problematic hardware is Card 1, Audio device [0403]: Advanced Micro Devices, Inc. [AMD] Family 17h (Models 00h-0fh) HD Audio Controller [1022:1457]. Test command : pasuspender -- arecord -D hw:1,0 -f S16_LE -c 2 -r 44100 arec_pasusp.wav Here's an example recording of me saying 'test' : https://gitlab.freedesktop.org/pulseaudio/pulseaudio/uploads/5a51081fcd55ec… I have no ~/.asoundrc or related files. Anything I can try to narrow down the cause of this ? Thanks! Chris

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[PATCH v5 00/16] Support external boost at CS35l41 ASoC driver
by Lucas Tanure 23 Mar '22

23 Mar '22

Move the support for CS35L41 external boost to its shared library for ASoC use. This move resulted in cs35l41_hda_reg_sequence being removed, and its steps were broken down into regmap writes or functions from the library. And hardware configuration struct was unified for its use in the shared lib. While at it, some minor bugs were found and fixed it. v5 changelog: - Fixed wrong indentation at Documentation patch - Use of consistent prefix v4 changelog: - Separated GPIO 1 and 2 function enums v3 changelog: - Remove patches already accepted - Improved logic in documentation patch - Documentation patch goes before its code - Fixed missing Signed-off-by - Fixed subject for HDA patches v2 changelog: - Instead of removing the log, playback actions will log the last regmap access. - Documentation patch with the correct subject line and fixed bug reported by Rob Herring on the provided example. Previous versions: v1: https://lkml.org/lkml/2022/3/3/759 v2: https://lkml.org/lkml/2022/3/4/743 v3: https://lkml.org/lkml/2022/3/8/975 v4: https://lkml.org/lkml/2022/3/17/267 David Rhodes (1): ASoC: cs35l41: Document CS35l41 External Boost Lucas Tanure (15): ALSA: cs35l41: Unify hardware configuration ALSA: cs35l41: Check hw_config before using it ALSA: cs35l41: Move cs35l41_gpio_config to shared lib ALSA: hda: cs35l41: Fix I2S params comments ALSA: hda: cs35l41: Always configure the DAI ALSA: hda: cs35l41: Add Boost type flag ALSA: hda: cs35l41: Put the device into safe mode for external boost ALSA: hda: cs35l41: Mute the device before shutdown ALSA: cs35l41: Enable Internal Boost in shared lib ALSA: hda: cs35l41: Move boost config to initialization code ALSA: hda: cs35l41: Remove cs35l41_hda_reg_sequence struct ALSA: hda: cs35l41: Reorganize log for playback actions ALSA: hda: cs35l41: Handle all external boost setups the same way ALSA: hda: cs35l41: Move external boost handling to lib for ASoC use ASoC: cs35l41: Support external boost .../bindings/sound/cirrus,cs35l41.yaml | 44 ++- include/sound/cs35l41.h | 59 +++- sound/pci/hda/cs35l41_hda.c | 295 ++++++------------ sound/pci/hda/cs35l41_hda.h | 27 +- sound/soc/codecs/cs35l41-i2c.c | 4 +- sound/soc/codecs/cs35l41-lib.c | 190 ++++++++++- sound/soc/codecs/cs35l41-spi.c | 4 +- sound/soc/codecs/cs35l41.c | 166 +++++----- sound/soc/codecs/cs35l41.h | 5 +- 9 files changed, 443 insertions(+), 351 deletions(-) -- 2.35.1

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