August 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH v6 0/2] Add mediatek codec mt6359 driver
by Jiaxin Yu 20 Aug '20

20 Aug '20

Add mediatek codec (MT6359) driver MT6359 support playback and capture feature. On downlink path, it includes three DACs for handset, headset, and lineout path. On unlink path, it includeds three ADCs for main mic, second mic, 3rd mic, and headset mic. By scenario, select *_MUX widget to create damp path. And by select mic_type_mux to choose DMIC/AMIC/.... For example, select these MUX widget to create headset path (1) DAC In Mux --> "Normal Path" (2) HP Mux --> "Audio Playback" v6 changes: 1. Remove the compatible string in mt6359codec because MFD should be registering the platform device. v5 changes: 1. Don't need to unregister the component whic is already relegated to devm. 2. patchwork link: https://patchwork.kernel.org/cover/11716387/ https://patchwork.kernel.org/patch/11717757/ https://patchwork.kernel.org/patch/11716491/ v4 changes: 1. Add a remove() function to undo regulator_enable(). 2. Removed unnecessary logs. 3. patchwork link: https://patchwork.kernel.org/cover/11715553/ https://patchwork.kernel.org/patch/11716015/ https://patchwork.kernel.org/patch/11715557/ v3 changes: 1. patchwork link: https://patchwork.kernel.org/cover/11714291/ https://patchwork.kernel.org/patch/11714295/ https://patchwork.kernel.org/patch/11714339/ v2 changes: 1. patchwork link: https://patchwork.kernel.org/cover/11706935/ https://patchwork.kernel.org/patch/11708865/ https://patchwork.kernel.org/patch/11706937/ v1 changes: 1.lkml link: https://lkml.org/lkml/2020/3/5/1257 Jiaxin Yu (2): ASoC: mediatek: mt6359: add codec driver dt-bindings: mediatek: mt6359: add codec document .../devicetree/bindings/sound/mt6359.yaml | 61 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mt6359.c | 2753 +++++++++++++++++ sound/soc/codecs/mt6359.h | 2640 ++++++++++++++++ 5 files changed, 5464 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt6359.yaml create mode 100644 sound/soc/codecs/mt6359.c create mode 100644 sound/soc/codecs/mt6359.h -- 2.18.0

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Request to pick up couple NVIDIA Tegra ASoC patches into 5.7 kernel
by Dmitry Osipenko 20 Aug '20

20 Aug '20

Hello, stable-kernel maintainers! Could you please cherry-pick these commits into the v5.7.x kernel? commit 0de6db30ef79b391cedd749801a49c485d2daf4b Author: Sowjanya Komatineni <skomatineni(a)nvidia.com> Date: Mon Jan 13 23:24:17 2020 -0800 ASoC: tegra: Use device managed resource APIs to get the clock commit 1e4e0bf136aa4b4aa59c1e6af19844bd6d807794 Author: Sowjanya Komatineni <skomatineni(a)nvidia.com> Date: Mon Jan 13 23:24:23 2020 -0800 ASoC: tegra: Add audio mclk parent configuration commit ff5d18cb04f4ecccbcf05b7f83ab6df2a0d95c16 Author: Sowjanya Komatineni <skomatineni(a)nvidia.com> Date: Mon Jan 13 23:24:24 2020 -0800 ASoC: tegra: Enable audio mclk during tegra_asoc_utils_init() It will fix a huge warnings splat during of kernel boot on NVIDIA Tegra SoCs. For some reason these patches haven't made into 5.7 when it was released and several people complained about the warnings. Thanks in advance!

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[PATCH] ALSA: usb-audio: Add prevent wakeup from s3 state trig by Lenovo ThinkCentre TI024Gen3 USB-audio
by penghao 20 Aug '20

20 Aug '20

TI024Gen3 USB-audio is controlled by TI024Gen3,when TI024Gens enter sleep mode, USB-audio will disconnect from USB bus port, wakup form s3 state Signed-off-by: penghao <penghao(a)uniontech.com> --- sound/usb/card.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index 696e788c5d31..6bdbb34009b3 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -658,6 +658,12 @@ static int usb_audio_probe(struct usb_interface *intf, } dev_set_drvdata(&dev->dev, chip); + /* + *ALSA: usb-audio: Add prevent wakeup from s3 state trig by Lenovo + *ThinkCentre TI024Gen3 usb-audio + */ + if ((usb_id->idVendor == 0x17ef) && (usb_id->idProduct == 0xa012)) + device_set_wakeup_enable(＆dev->dev, 0); /* * For devices with more than one control interface, we assume the -- 2.11.0

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[PATCH v2] ALSA: hda: avoid reset of sdo_limit
by Sameer Pujar 19 Aug '20

19 Aug '20

By default 'sdo_limit' is initialized with a default value of '8' as per spec. This is overridden in cases where a different value is required. However this is getting reset when snd_hdac_bus_init_chip() is called again, which happens during runtime PM cycle. Avoid this reset by moving 'sdo_limit' setup to 'snd_hdac_bus_init()' function which would be called only once. Fixes: 67ae482a59e9 ("ALSA: hda: add member to store ratio for stripe control") Cc: <stable(a)vger.kernel.org> Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> --- sound/hda/hdac_bus.c | 12 ++++++++++++ sound/hda/hdac_controller.c | 11 ----------- 2 files changed, 12 insertions(+), 11 deletions(-) diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c index 09ddab5..9766f6a 100644 --- a/sound/hda/hdac_bus.c +++ b/sound/hda/hdac_bus.c @@ -46,6 +46,18 @@ int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev, INIT_LIST_HEAD(&bus->hlink_list); init_waitqueue_head(&bus->rirb_wq); bus->irq = -1; + + /* + * Default value of '8' is as per the HD audio specification (Rev 1.0a). + * Following relation is used to derive STRIPE control value. + * For sample rate <= 48K: + * { ((num_channels * bits_per_sample) / number of SDOs) >= 8 } + * For sample rate > 48K: + * { ((num_channels * bits_per_sample * rate/48000) / + * number of SDOs) >= 8 } + */ + bus->sdo_limit = 8; + return 0; } EXPORT_SYMBOL_GPL(snd_hdac_bus_init); diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 011b17c..b98449f 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -529,17 +529,6 @@ bool snd_hdac_bus_init_chip(struct hdac_bus *bus, bool full_reset) bus->chip_init = true; - /* - * Default value of '8' is as per the HD audio specification (Rev 1.0a). - * Following relation is used to derive STRIPE control value. - * For sample rate <= 48K: - * { ((num_channels * bits_per_sample) / number of SDOs) >= 8 } - * For sample rate > 48K: - * { ((num_channels * bits_per_sample * rate/48000) / - * number of SDOs) >= 8 } - */ - bus->sdo_limit = 8; - return true; } EXPORT_SYMBOL_GPL(snd_hdac_bus_init_chip); -- 2.7.4

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[PATCH] ALSA: hda: avoid reset of sdo_limit
by Sameer Pujar 19 Aug '20

19 Aug '20

By default 'sdo_limit' is initialized with a default value of '8' as per spec. This is overridden in cases where a different value is required. However this is getting reset when snd_hdac_bus_init_chip() is called again, which happens during runtime PM cycle. Avoid reset by not initializing to default value everytime. Fixes: 67ae482a59e9 ("ALSA: hda: add member to store ratio for stripe control") Cc: <stable(a)vger.kernel.org> Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> --- sound/hda/hdac_controller.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 011b17c..0e26e96 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -538,7 +538,8 @@ bool snd_hdac_bus_init_chip(struct hdac_bus *bus, bool full_reset) * { ((num_channels * bits_per_sample * rate/48000) / * number of SDOs) >= 8 } */ - bus->sdo_limit = 8; + if (!bus->sdo_limit) + bus->sdo_limit = 8; return true; } -- 2.7.4

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[PATCH] ASoC: codecs: rt*-sdw: use SDW_SLAVE_ENTRY_EXT
by Bard Liao 19 Aug '20

19 Aug '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Add version and class information explicitly to prepare for support for new devices. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/codecs/rt1308-sdw.c | 2 +- sound/soc/codecs/rt5682-sdw.c | 2 +- sound/soc/codecs/rt700-sdw.c | 2 +- sound/soc/codecs/rt711-sdw.c | 2 +- sound/soc/codecs/rt715-sdw.c | 2 +- 5 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c index 81e3e2caeb2f..ecb03ad8102e 100644 --- a/sound/soc/codecs/rt1308-sdw.c +++ b/sound/soc/codecs/rt1308-sdw.c @@ -696,7 +696,7 @@ static int rt1308_sdw_probe(struct sdw_slave *slave, } static const struct sdw_device_id rt1308_id[] = { - SDW_SLAVE_ENTRY(0x025d, 0x1308, 0), + SDW_SLAVE_ENTRY_EXT(0x025d, 0x1308, 0x2, 0, 0), {}, }; MODULE_DEVICE_TABLE(sdw, rt1308_id); diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index 148d4cff267b..4d5bc8ac38df 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -722,7 +722,7 @@ static int rt5682_sdw_remove(struct sdw_slave *slave) } static const struct sdw_device_id rt5682_id[] = { - SDW_SLAVE_ENTRY(0x025d, 0x5682, 0), + SDW_SLAVE_ENTRY_EXT(0x025d, 0x5682, 0x2, 0, 0), {}, }; MODULE_DEVICE_TABLE(sdw, rt5682_id); diff --git a/sound/soc/codecs/rt700-sdw.c b/sound/soc/codecs/rt700-sdw.c index 2d475405b20d..b08ccc3da1dd 100644 --- a/sound/soc/codecs/rt700-sdw.c +++ b/sound/soc/codecs/rt700-sdw.c @@ -483,7 +483,7 @@ static int rt700_sdw_remove(struct sdw_slave *slave) } static const struct sdw_device_id rt700_id[] = { - SDW_SLAVE_ENTRY(0x025d, 0x700, 0), + SDW_SLAVE_ENTRY_EXT(0x025d, 0x700, 0x1, 0, 0), {}, }; MODULE_DEVICE_TABLE(sdw, rt700_id); diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c index 7a1ae7442e75..c72cd4fb64fd 100644 --- a/sound/soc/codecs/rt711-sdw.c +++ b/sound/soc/codecs/rt711-sdw.c @@ -484,7 +484,7 @@ static int rt711_sdw_remove(struct sdw_slave *slave) } static const struct sdw_device_id rt711_id[] = { - SDW_SLAVE_ENTRY(0x025d, 0x711, 0), + SDW_SLAVE_ENTRY_EXT(0x025d, 0x711, 0x2, 0, 0), {}, }; MODULE_DEVICE_TABLE(sdw, rt711_id); diff --git a/sound/soc/codecs/rt715-sdw.c b/sound/soc/codecs/rt715-sdw.c index 761d4663e813..c87a3fc0678b 100644 --- a/sound/soc/codecs/rt715-sdw.c +++ b/sound/soc/codecs/rt715-sdw.c @@ -546,7 +546,7 @@ static int rt715_sdw_probe(struct sdw_slave *slave, } static const struct sdw_device_id rt715_id[] = { - SDW_SLAVE_ENTRY(0x025d, 0x715, 0), + SDW_SLAVE_ENTRY_EXT(0x025d, 0x715, 0x2, 0, 0), {}, }; MODULE_DEVICE_TABLE(sdw, rt715_id); -- 2.17.1

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[PATCH] ASoC: Intel: skl_hda_dsp_generic: Fix NULLptr dereference in autosuspend delay
by Mateusz Gorski 19 Aug '20

19 Aug '20

Different modules for HDMI codec are used depending on the "hda_codec_use_common_hdmi" option being enabled or not. Driver private context for both of them is different. This leads to null-pointer dereference error when driver tries to set autosuspend delay for HDMI codec while the option is off (hdac_hdmi module is used for HDMI). Change the string in conditional statement to "ehdaudio0D0" to ensure that only the HDAudio codec is handled by this function. Fixes: 5bf73b1b1dec ("ASoC: intel/skl/hda - fix oops on systems without i915 audio codec") Signed-off-by: Mateusz Gorski <mateusz.gorski(a)linux.intel.com> --- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index ca4900036ead..bc50eda297ab 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -181,7 +181,7 @@ static void skl_set_hda_codec_autosuspend_delay(struct snd_soc_card *card) struct snd_soc_dai *dai; for_each_card_rtds(card, rtd) { - if (!strstr(rtd->dai_link->codecs->name, "ehdaudio")) + if (!strstr(rtd->dai_link->codecs->name, "ehdaudio0D0")) continue; dai = asoc_rtd_to_codec(rtd, 0); hda_pvt = snd_soc_component_get_drvdata(dai->component); -- 2.17.1

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[PATCH V2] dt-bindings: sound: Convert NXP spdif to json-schema
by Anson Huang 19 Aug '20

19 Aug '20

Convert the NXP SPDIF binding to DT schema format using json-schema. Signed-off-by: Anson Huang <Anson.Huang(a)nxp.com> Reviewed-by: Rob Herring <robh(a)kernel.org> --- changes since V1: - add "additionalProperties: false". --- .../devicetree/bindings/sound/fsl,spdif.txt | 68 ------------- .../devicetree/bindings/sound/fsl,spdif.yaml | 110 +++++++++++++++++++++ 2 files changed, 110 insertions(+), 68 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.yaml diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.txt b/Documentation/devicetree/bindings/sound/fsl,spdif.txt deleted file mode 100644 index e1365b0..0000000 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.txt +++ /dev/null @@ -1,68 +0,0 @@ -Freescale Sony/Philips Digital Interface Format (S/PDIF) Controller - -The Freescale S/PDIF audio block is a stereo transceiver that allows the -processor to receive and transmit digital audio via an coaxial cable or -a fibre cable. - -Required properties: - - - compatible : Compatible list, should contain one of the following - compatibles: - "fsl,imx35-spdif", - "fsl,vf610-spdif", - "fsl,imx6sx-spdif", - - - reg : Offset and length of the register set for the device. - - - interrupts : Contains the spdif interrupt. - - - dmas : Generic dma devicetree binding as described in - Documentation/devicetree/bindings/dma/dma.txt. - - - dma-names : Two dmas have to be defined, "tx" and "rx". - - - clocks : Contains an entry for each entry in clock-names. - - - clock-names : Includes the following entries: - "core" The core clock of spdif controller. - "rxtx<0-7>" Clock source list for tx and rx clock. - This clock list should be identical to the source - list connecting to the spdif clock mux in "SPDIF - Transceiver Clock Diagram" of SoC reference manual. - It can also be referred to TxClk_Source bit of - register SPDIF_STC. - "spba" The spba clock is required when SPDIF is placed as a - bus slave of the Shared Peripheral Bus and when two - or more bus masters (CPU, DMA or DSP) try to access - it. This property is optional depending on the SoC - design. - -Optional properties: - - - big-endian : If this property is absent, the native endian mode - will be in use as default, or the big endian mode - will be in use for all the device registers. - -Example: - -spdif: spdif@2004000 { - compatible = "fsl,imx35-spdif"; - reg = <0x02004000 0x4000>; - interrupts = <0 52 0x04>; - dmas = <&sdma 14 18 0>, - <&sdma 15 18 0>; - dma-names = "rx", "tx"; - - clocks = <&clks 197>, <&clks 3>, - <&clks 197>, <&clks 107>, - <&clks 0>, <&clks 118>, - <&clks 62>, <&clks 139>, - <&clks 0>; - clock-names = "core", "rxtx0", - "rxtx1", "rxtx2", - "rxtx3", "rxtx4", - "rxtx5", "rxtx6", - "rxtx7"; - - big-endian; -}; diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml new file mode 100644 index 0000000..2ac671f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml @@ -0,0 +1,110 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,spdif.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Sony/Philips Digital Interface Format (S/PDIF) Controller + +maintainers: + - Shengjiu Wang <shengjiu.wang(a)nxp.com> + +description: | + The Freescale S/PDIF audio block is a stereo transceiver that allows the + processor to receive and transmit digital audio via an coaxial cable or + a fibre cable. + +properties: + compatible: + enum: + - fsl,imx35-spdif + - fsl,vf610-spdif + - fsl,imx6sx-spdif + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + dmas: + items: + - description: DMA controller phandle and request line for RX + - description: DMA controller phandle and request line for TX + + dma-names: + items: + - const: rx + - const: tx + + clocks: + items: + - description: The core clock of spdif controller. + - description: Clock for tx0 and rx0. + - description: Clock for tx1 and rx1. + - description: Clock for tx2 and rx2. + - description: Clock for tx3 and rx3. + - description: Clock for tx4 and rx4. + - description: Clock for tx5 and rx5. + - description: Clock for tx6 and rx6. + - description: Clock for tx7 and rx7. + - description: The spba clock is required when SPDIF is placed as a bus + slave of the Shared Peripheral Bus and when two or more bus masters + (CPU, DMA or DSP) try to access it. This property is optional depending + on the SoC design. + minItems: 9 + + clock-names: + items: + - const: core + - const: rxtx0 + - const: rxtx1 + - const: rxtx2 + - const: rxtx3 + - const: rxtx4 + - const: rxtx5 + - const: rxtx6 + - const: rxtx7 + - const: spba + minItems: 9 + + big-endian: + $ref: /schemas/types.yaml#/definitions/flag + description: | + If this property is absent, the native endian mode will be in use + as default, or the big endian mode will be in use for all the device + registers. Set this flag for HCDs with big endian descriptors and big + endian registers. + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + spdif@2004000 { + compatible = "fsl,imx35-spdif"; + reg = <0x02004000 0x4000>; + interrupts = <0 52 0x04>; + dmas = <&sdma 14 18 0>, + <&sdma 15 18 0>; + dma-names = "rx", "tx"; + clocks = <&clks 197>, <&clks 3>, + <&clks 197>, <&clks 107>, + <&clks 0>, <&clks 118>, + <&clks 62>, <&clks 139>, + <&clks 0>; + clock-names = "core", "rxtx0", + "rxtx1", "rxtx2", + "rxtx3", "rxtx4", + "rxtx5", "rxtx6", + "rxtx7"; + big-endian; + }; -- 2.7.4

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[PATCH] soundwire: bus: fix typo in comment on INTSTAT registers
by Bard Liao 19 Aug '20

19 Aug '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> s/Instat/Intstat/ Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/bus.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index e6e0fb9a81b4..da0201693c24 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -1372,7 +1372,7 @@ static int sdw_handle_slave_alerts(struct sdw_slave *slave) return ret; } - /* Read Instat 1, Instat 2 and Instat 3 registers */ + /* Read Intstat 1, Intstat 2 and Intstat 3 registers */ ret = sdw_read(slave, SDW_SCP_INT1); if (ret < 0) { dev_err(slave->bus->dev, -- 2.17.1

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[PATCH] ALSA: hda/realtek: Add quirk for Samsung Galaxy Book Ion
by Mike Pozulp 19 Aug '20

19 Aug '20

The Galaxy Book Ion uses the same ALC298 codec as other Samsung laptops which have the no headphone sound bug, like my Samsung Notebook. The Galaxy Book owner confirmed that this patch fixes the bug. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207423 Signed-off-by: Mike Pozulp <pozulp.kernel(a)gmail.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 29f5878f0c50..72aa3da479c9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7612,6 +7612,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), + SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8), SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), -- 2.28.0

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