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

[PATCH v1 00/10] Fix sound on ASUS Transformers
by Svyatoslav Ryhel 06 Mar '23

06 Mar '23

- add quirk for headset detection used by some T30 devices (ASUS Transformers, LG Optimus 4X HD and Vu); - add RT5631 and MAX9808x machine drivers - add Fortemedia FM34NE DSP driver used by ASUS Transformers and mandatory for correct sound work - bind everything into working configuration David Heidelberg (1): dt-bindings: sound: nvidia,tegra-audio: add RT5631 CODEC Svyatoslav Ryhel (9): dt-bindings: sound: nvidia,tegra-audio-common: add new property sound: soc: jack: allow multiple interrupt per gpio ASoC: tegra: Support RT5631 by machine driver dt-bindings: sound: nvidia,tegra-audio: add MAX9808x CODEC ASoC: tegra: Support MAX9808x by machine driver ARM: tegra: transformers: update bindings of sound graph dt-bindings: dsp: add Fortemedia FM34 DSP staging: dsp: add support for Fortemedia FM34NE DSP ARM: tegra: transformers: bind FM34NE DSP on supported devices .../bindings/dsp/fortemedia,dsp.yaml | 95 ++ .../sound/nvidia,tegra-audio-common.yaml | 4 + .../sound/nvidia,tegra-audio-max9808x.yaml | 93 ++ .../sound/nvidia,tegra-audio-rt5631.yaml | 88 ++ arch/arm/boot/dts/tegra20-asus-tf101.dts | 22 +- arch/arm/boot/dts/tegra30-asus-tf201.dts | 46 + arch/arm/boot/dts/tegra30-asus-tf300t.dts | 34 +- arch/arm/boot/dts/tegra30-asus-tf300tg.dts | 36 + arch/arm/boot/dts/tegra30-asus-tf700t.dts | 36 + .../dts/tegra30-asus-transformer-common.dtsi | 9 +- .../arm/boot/dts/tegra30-pegatron-chagall.dts | 29 +- drivers/staging/Kconfig | 2 + drivers/staging/Makefile | 1 + drivers/staging/dsp/Kconfig | 7 + drivers/staging/dsp/Makefile | 2 + drivers/staging/dsp/dsp-fm34ne.c | 364 ++++++++ drivers/staging/dsp/dsp-fm34ne.h | 845 ++++++++++++++++++ sound/soc/soc-jack.c | 1 + sound/soc/tegra/Kconfig | 18 + sound/soc/tegra/tegra_asoc_machine.c | 120 ++- 20 files changed, 1837 insertions(+), 15 deletions(-) create mode 100644 Documentation/devicetree/bindings/dsp/fortemedia,dsp.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max9808x.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5631.yaml create mode 100644 drivers/staging/dsp/Kconfig create mode 100644 drivers/staging/dsp/Makefile create mode 100644 drivers/staging/dsp/dsp-fm34ne.c create mode 100644 drivers/staging/dsp/dsp-fm34ne.h -- 2.37.2

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[PATCH][next] ASoC: SOF: ipc4-topology: Replace fake flexible arrays with flexible-array member
by Gustavo A. R. Silva 06 Mar '23

06 Mar '23

Zero-length arrays as fake flexible arrays are deprecated and we are moving towards adopting C99 flexible-array members, instead. Use the DECLARE_FLEX_ARRAY() helper macro to transform zero-length arrays in unions with flexible-array members. Address the following warnings found with GCC-13 and -fstrict-flex-arrays=3 enabled: sound/soc/sof/ipc4-control.c:176:77: warning: array subscript i is outside array bounds of ‘struct sof_ipc4_ctrl_value_chan[0]’ [-Warray-bounds=] sound/soc/sof/ipc4-control.c:78:29: warning: array subscript 0 is outside array bounds of ‘struct sof_ipc4_ctrl_value_chan[0]’ [-Warray-bounds=] sound/soc/sof/ipc4-control.c:80:33: warning: array subscript i is outside array bounds of ‘struct sof_ipc4_ctrl_value_chan[0]’ [-Warray-bounds=] sound/soc/sof/ipc4-control.c:95:53: warning: array subscript i is outside array bounds of ‘struct sof_ipc4_ctrl_value_chan[0]’ [-Warray-bounds=] sound/soc/sof/ipc4-control.c:96:53: warning: array subscript i is outside array bounds of ‘struct sof_ipc4_ctrl_value_chan[0]’ [-Warray-bounds=] sound/soc/sof/ipc4-control.c:93:53: warning: array subscript 0 is outside array bounds of ‘struct sof_ipc4_ctrl_value_chan[0]’ [-Warray-bounds=] sound/soc/sof/ipc4-control.c:140:58: warning: array subscript i is outside array bounds of ‘struct sof_ipc4_ctrl_value_chan[0]’ [-Warray-bounds=] sound/soc/sof/ipc4-control.c:141:29: warning: array subscript i is outside array bounds of ‘struct sof_ipc4_ctrl_value_chan[0]’ [-Warray-bounds=] sound/soc/sof/ipc4-control.c:142:29: warning: array subscript i is outside array bounds of ‘struct sof_ipc4_ctrl_value_chan[0]’ [-Warray-bounds=] sound/soc/sof/ipc4-topology.c:1475:36: warning: array subscript i is outside array bounds of ‘struct sof_ipc4_ctrl_value_chan[0]’ [-Warray-bounds=] sound/soc/sof/ipc4-topology.c:1476:36: warning: array subscript i is outside array bounds of ‘struct sof_ipc4_ctrl_value_chan[0]’ [-Warray-bounds=] This helps with the ongoing efforts to tighten the FORTIFY_SOURCE routines on memcpy() and help us make progress towards globally enabling -fstrict-flex-arrays=3 [1]. Link: https://github.com/KSPP/linux/issues/21 Link: https://github.com/KSPP/linux/issues/193 Link: https://github.com/KSPP/linux/issues/258 Link: https://gcc.gnu.org/pipermail/gcc-patches/2022-October/602902.html [1] Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- sound/soc/sof/ipc4-topology.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h index 72529179ac22..8507171a6e4d 100644 --- a/sound/soc/sof/ipc4-topology.h +++ b/sound/soc/sof/ipc4-topology.h @@ -266,8 +266,8 @@ struct sof_ipc4_control_data { int index; union { - struct sof_ipc4_ctrl_value_chan chanv[0]; - struct sof_abi_hdr data[0]; + DECLARE_FLEX_ARRAY(struct sof_ipc4_ctrl_value_chan, chanv); + DECLARE_FLEX_ARRAY(struct sof_abi_hdr, data); }; }; -- 2.34.1

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[PATCH 0/2] Add support for the TAS5733
by Kamel Bouhara 06 Mar '23

06 Mar '23

Hello, This small series extends the tas571x driver to support the TAS5733 audio power amplifier. Kamel Bouhara (2): ASoC: tas571x: add tas5733 compatible ASoC: tas571x: add support for TAS5733 .../devicetree/bindings/sound/tas571x.txt | 1 + sound/soc/codecs/tas571x.c | 59 +++++++++++++++++++ 2 files changed, 60 insertions(+) -- 2.35.1

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[PATCH 1/2] ALSA: ice1712: Do not left ice->gpio_mutex locked in aureon_add_controls()
by Dmitry Fomin 03 Mar '23

03 Mar '23

If snd_ctl_add() fails in aureon_add_controls(), it immediately returns and leaves ice->gpio_mutex locked. ice->gpio_mutex locks in snd_ice1712_save_gpio_status and unlocks in snd_ice1712_restore_gpio_status(ice). It seems that the mutex is required only for aureon_cs8415_get(), so snd_ice1712_restore_gpio_status(ice) can be placed just after that. Compile tested only. Found by Linux Verification Center (linuxtesting.org) with SVACE. Fixes: 1da177e4c3f4 ("Linux-2.6.12-rc2") Signed-off-by: Dmitry Fomin <fomindmitriyfoma(a)mail.ru> --- sound/pci/ice1712/aureon.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c index 9a30f6d35d13..40a0e0095030 100644 --- a/sound/pci/ice1712/aureon.c +++ b/sound/pci/ice1712/aureon.c @@ -1892,6 +1892,7 @@ static int aureon_add_controls(struct snd_ice1712 *ice) unsigned char id; snd_ice1712_save_gpio_status(ice); id = aureon_cs8415_get(ice, CS8415_ID); + snd_ice1712_restore_gpio_status(ice); if (id != 0x41) dev_info(ice->card->dev, "No CS8415 chip. Skipping CS8415 controls.\n"); @@ -1909,7 +1910,6 @@ static int aureon_add_controls(struct snd_ice1712 *ice) kctl->id.device = ice->pcm->device; } } - snd_ice1712_restore_gpio_status(ice); } return 0; -- 2.37.2

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[PATCH] ALSA: hda/realtek: Add quirk for HP EliteDesk 800 G6 Tower PC
by Łukasz Stelmach 03 Mar '23

03 Mar '23

HP EliteDesk 800 G6 Tower PC (103c:870c) requires a quirk for enabling headset-mic. Signed-off-by: Łukasz Stelmach <l.stelmach(a)samsung.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) Verified by echo 0x19 0x01a1913c > /sys/class/sound/card0/device/sound/card0/hwC0D0/user_pin_configs echo 1 > /sys/class/sound/card0/device/sound/card0/hwC0D0/reconfig diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6fab7c8fc19a..ac196db573e4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11599,6 +11599,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0698, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x069f, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), + SND_PCI_QUIRK(0x103c, 0x870c, "HP", ALC897_FIXUP_HP_HSMIC_VERB), SND_PCI_QUIRK(0x103c, 0x8719, "HP", ALC897_FIXUP_HP_HSMIC_VERB), SND_PCI_QUIRK(0x103c, 0x873e, "HP", ALC671_FIXUP_HP_HEADSET_MIC2), SND_PCI_QUIRK(0x103c, 0x877e, "HP 288 Pro G6", ALC671_FIXUP_HP_HEADSET_MIC2), -- 2.30.2

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[PATCH] ALSA: hda/realtek: Improve support for Dell Precision 3260
by Jaroslav Kysela 03 Mar '23

03 Mar '23

The headset jack works better with model=alc283-dac-wcaps. Without this option, the headset insertion (separate physical jack) may not be handled correctly (re-insertion is required). It seems that it follows the "Intel Reference Board" defaults. Reported-by: steven_wu2(a)dell.com Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- 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 e103bb3693c0..7afbb708b705 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9260,6 +9260,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a62, "Dell Precision 5560", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0a9d, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0a9e, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0ac9, "Dell Precision 3260", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x1028, 0x0b19, "Dell XPS 15 9520", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0b1a, "Dell Precision 5570", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0b37, "Dell Inspiron 16 Plus 7620 2-in-1", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), -- 2.39.0

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[PATCH V2] ucm2: Rockchip: rk817: Add ALSA UCM support
by Chris Morgan 02 Mar '23

02 Mar '23

From: Chris Morgan <macromorgan(a)hotmail.com> Add ALSA-UCM support for the Rockchip RK817 audio codec. This codec is typically configured either with an internal or external amplifier as reflected by the longname. This configuration has been tested on the Anbernic RG353P (rk817_ext) and the Odroid Go Advance (rk817_int). Changes from V1: - Use a constant to control if the output mux is set to HP or SPK. Signed-off-by: Chris Morgan <macromorgan(a)hotmail.com> --- ucm2/Rockchip/rk817-sound/HiFi.conf | 68 ++++++++++++++++++++++ ucm2/Rockchip/rk817-sound/rk817-sound.conf | 6 ++ ucm2/conf.d/simple-card/rk817_ext.conf | 1 + ucm2/conf.d/simple-card/rk817_int.conf | 1 + 4 files changed, 76 insertions(+) create mode 100644 ucm2/Rockchip/rk817-sound/HiFi.conf create mode 100644 ucm2/Rockchip/rk817-sound/rk817-sound.conf create mode 120000 ucm2/conf.d/simple-card/rk817_ext.conf create mode 120000 ucm2/conf.d/simple-card/rk817_int.conf diff --git a/ucm2/Rockchip/rk817-sound/HiFi.conf b/ucm2/Rockchip/rk817-sound/HiFi.conf new file mode 100644 index 0000000..8d90567 --- /dev/null +++ b/ucm2/Rockchip/rk817-sound/HiFi.conf @@ -0,0 +1,68 @@ +Define.pbk_mux "SPK" + +If.1 { + Condition { + Type ControlExists + Control "name='Internal Speakers Switch'" + } + + True { + Define.pbk_mux "HP" + SectionDevice."Speaker".EnableSequence [ + cset "name='Internal Speakers Switch' on" + ] + + SectionDevice."Speaker".DisableSequence [ + cset "name='Internal Speakers Switch' off" + ] + } +} + +SectionDevice."Speaker" { + Comment "Internal Speaker" + + EnableSequence [ + cset "name='Playback Mux' ${var:pbk_mux}" + ] + + Value { + PlaybackMixerElem "Master Playback Volume" + PlaybackPriority 100 + PlaybackPCM "hw:${CardId}" + } + + ConflictingDevice [ + "Headphones" + ] +} + +SectionDevice."Mic" { + Comment "Microphone" + + Value { + CapturePriority 100 + CapturePCM "hw:${CardId}" + CaptureMixerElem "Mic Capture Gain" + CaptureMasterElem "Master Capture Volume" + } +} + +SectionDevice."Headphones" { + Comment "Headphones" + + EnableSequence [ + cset "name='Playback Mux' HP" + ] + + Value { + PlaybackMixerElem "Master Playback Volume" + PlaybackPriority 200 + PlaybackPCM "hw:${CardId}" + JackControl "Headphones Jack" + JackHWMute "Speaker" + } + + ConflictingDevice [ + "Speaker" + ] +} diff --git a/ucm2/Rockchip/rk817-sound/rk817-sound.conf b/ucm2/Rockchip/rk817-sound/rk817-sound.conf new file mode 100644 index 0000000..90ec794 --- /dev/null +++ b/ucm2/Rockchip/rk817-sound/rk817-sound.conf @@ -0,0 +1,6 @@ +Syntax 4 + +SectionUseCase."HiFi" { + File "/Rockchip/rk817-sound/HiFi.conf" + Comment "Play HiFi quality music" +} diff --git a/ucm2/conf.d/simple-card/rk817_ext.conf b/ucm2/conf.d/simple-card/rk817_ext.conf new file mode 120000 index 0000000..6fa8db7 --- /dev/null +++ b/ucm2/conf.d/simple-card/rk817_ext.conf @@ -0,0 +1 @@ +../../Rockchip/rk817-sound/rk817-sound.conf \ No newline at end of file diff --git a/ucm2/conf.d/simple-card/rk817_int.conf b/ucm2/conf.d/simple-card/rk817_int.conf new file mode 120000 index 0000000..6fa8db7 --- /dev/null +++ b/ucm2/conf.d/simple-card/rk817_int.conf @@ -0,0 +1 @@ +../../Rockchip/rk817-sound/rk817-sound.conf \ No newline at end of file -- 2.34.1

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[PATCH v3 0/3] ASoC: mchp-pdmc: fix poc noises when starting capture
by Claudiu Beznea 28 Feb '23

28 Feb '23

To start capture on Microchip PDMC the enable bits for each supported microphone need to be set. After this bit is set the PDMC starts to receive data from microphones and it considers this data as valid data. Thus if microphones are not ready the PDMC captures anyway data from its lines. This data is interpreted by the human ear as poc noises. To avoid this the following software workaround need to be applied when starting capture: 1/ enable PDMC channel 2/ wait 150ms 3/ execute 16 dummy reads from RHR 4/ clear interrupts 5/ enable interrupts 6/ enable DMA channel For this workaround to work step 6 need to be executed at the end. For step 6 was added patch 1/3 from this series. With this, component DAI driver sets its struct snd_soc_component_driver::start_dma_last = 1 and proper action is taken based on this flag when starting DAI trigger vs DMA. Thank you, Claudiu Beznea Changes in v3: - update the commit message of patch 2/3 to be more descriptive - in patch 3/3 initialize dd->startup_delay_us before calling of_property_read_u32() and don't check its return value; property is optional and the the default value is 150ms Changes in v2: - patch 1/3 from v1 is now "ASoC: soc-pcm: add option to start DMA after DAI" - pass start_dma_last from component DAI driver object (struct snd_soc_component_driver::start_dma_last); adapt patch 3/3 after this; - in patch 1/3 s/Do we need to start dma first/Do we need to start dma last in comment from soc_pcm_trigger() - collect review tag from Krzysztof Claudiu Beznea (3): ASoC: soc-pcm: add option to start DMA after DAI ASoC: dt-bindings: sama7g5-pdmc: add microchip,startup-delay-us binding ASoC: mchp-pdmc: fix poc noise at capture startup Claudiu Beznea (3): ASoC: soc-pcm: add option to start DMA after DAI ASoC: dt-bindings: sama7g5-pdmc: add microchip,startup-delay-us binding ASoC: mchp-pdmc: fix poc noise at capture startup .../sound/microchip,sama7g5-pdmc.yaml | 6 +++ include/sound/soc-component.h | 2 + sound/soc/atmel/mchp-pdmc.c | 53 +++++++++++++++++-- sound/soc/soc-pcm.c | 27 ++++++++-- 4 files changed, 78 insertions(+), 10 deletions(-) -- 2.34.1

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[PATCH] ASoC: zl38060 add gpiolib dependency
by Arnd Bergmann 28 Feb '23

28 Feb '23

From: Arnd Bergmann <arnd(a)arndb.de> Without gpiolib, this driver fails to link: arm-linux-gnueabi-ld: sound/soc/codecs/zl38060.o: in function `chip_gpio_get': zl38060.c:(.text+0x30): undefined reference to `gpiochip_get_data' arm-linux-gnueabi-ld: sound/soc/codecs/zl38060.o: in function `zl38_spi_probe': zl38060.c:(.text+0xa18): undefined reference to `devm_gpiochip_add_data_with_key' This appears to have been in the driver since the start, but is hard to hit in randconfig testing since gpiolib is almost always selected by something else. Fixes: 52e8a94baf90 ("ASoC: Add initial ZL38060 driver") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bd72c426a93d..07747565c3b5 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -2103,6 +2103,7 @@ config SND_SOC_WSA883X config SND_SOC_ZL38060 tristate "Microsemi ZL38060 Connected Home Audio Processor" depends on SPI_MASTER + depends on GPIOLIB select REGMAP help Support for ZL38060 Connected Home Audio Processor from Microsemi, -- 2.39.2

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[PATCH] ASoC: sam9g20ek: Disable capture unless building with microphone input
by Mark Brown 28 Feb '23

28 Feb '23

Without modification the AT91SAM9G20-EK has no capture support, none of the inputs of the CODEC are wired to anything to useful and there are no paths supporting loopback. Since the audio is clocked from the CODEC and the DAPM inputs are marked as unusable this means that capture will fail to transfer any data as the ADC path can't be powered up. Flag this in the device description so apps don't see unusable capture support, guarded with the existing optional define for mic input. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/atmel/sam9g20_wm8731.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 1430642c8433..785b9d01d8af 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -98,6 +98,9 @@ static struct snd_soc_dai_link at91sam9g20ek_dai = { .init = at91sam9g20ek_wm8731_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBP_CFP, +#ifndef ENABLE_MIC_INPUT + .playback_only = true, +#endif SND_SOC_DAILINK_REG(pcm), }; --- base-commit: cac85e4616b1cf4a90844b952b49b9cbc4562530 change-id: 20230225-asoc-sam9g20ek-f4e775205d8c Best regards, -- Mark Brown <broonie(a)kernel.org>

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