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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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[GIT PULL] ASoC fixes for v5.9-rc1
by Mark Brown 19 Aug '20

19 Aug '20

The following changes since commit 8e34f1e867b572f1e20b5250c2897fe5f041c99f: Merge series "ASoC: core: Two step component registration" from Cezary Rojewski <cezary.rojewski(a)intel.com>: (2020-07-31 19:36:00 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.9-rc1 for you to fetch changes up to 062fa09f44f4fb3776a23184d5d296b0c8872eb9: ASoC: intel: Fix memleak in sst_media_open (2020-08-17 12:28:57 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.9 A bunch of fixes that came in during the merge window, mostly for issues that were uncovered by the changes to report errors on invalid register access plus one important fix in that code itself. ---------------------------------------------------------------- Dinghao Liu (1): ASoC: intel: Fix memleak in sst_media_open Fabio Estevam (1): ASoC: wm8962: Do not remove ADDITIONAL_CONTROL_4 from readable register list Hui Wang (1): ASoC: amd: renoir: restore two more registers during resume Mark Brown (1): Merge series "ASoC: tegra: Fix compile warning with CONFIG_PM=n" from Takashi Iwai <tiwai(a)suse.de>: Ravulapati Vishnu vardhan rao (1): ASoC: amd: Replacing component->name with codec_dai->name. Shengjiu Wang (1): ASoC: fsl-asoc-card: Remove fsl_asoc_card_set_bias_level function Srinivas Kandagatla (3): ASoC: q6afe-dai: mark all widgets registers as SND_SOC_NOPM ASoC: q6routing: add dummy register read/write function ASoC: msm8916-wcd-analog: fix register Interrupt offset Sylwester Nawrocki (2): ASoC: wm8994: Prevent access to invalid VU register bits on WM1811 ASoC: wm8994: Avoid attempts to read unreadable registers Takashi Iwai (7): ASoC: tegra: tegra186_dspk: Fix compile warning with CONFIG_PM=n ASoC: tegra: tegra210_admaif: Fix compile warning with CONFIG_PM=n ASoC: tegra: tegra210_ahub: Fix compile warning with CONFIG_PM=n ASoC: tegra: tegra210_dmic: Fix compile warning with CONFIG_PM=n ASoC: tegra: tegra210_i2s: Fix compile warning with CONFIG_PM=n ASoC: fsl: Fix unused variable warning ASoC: Make soc_component_read() returning an error code again sound/soc/amd/acp3x-rt5682-max9836.c | 2 +- sound/soc/amd/renoir/acp3x-pdm-dma.c | 29 ++-- sound/soc/codecs/msm8916-wcd-analog.c | 4 +- sound/soc/codecs/wm8958-dsp2.c | 4 + sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8994.c | 60 ++++++-- sound/soc/fsl/fsl-asoc-card.c | 154 +++++++++----------- sound/soc/fsl/mpc5200_dma.c | 1 - sound/soc/intel/atom/sst-mfld-platform-pcm.c | 5 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 210 +++++++++++++-------------- sound/soc/qcom/qdsp6/q6routing.c | 16 ++ sound/soc/soc-component.c | 2 +- sound/soc/tegra/tegra186_dspk.c | 4 +- sound/soc/tegra/tegra210_admaif.c | 4 +- sound/soc/tegra/tegra210_ahub.c | 4 +- sound/soc/tegra/tegra210_dmic.c | 4 +- sound/soc/tegra/tegra210_i2s.c | 4 +- 17 files changed, 266 insertions(+), 243 deletions(-)

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ucm: let rval_sysfs check the device/driver/module first
by GitHub pull_request - opened 19 Aug '20

19 Aug '20

alsa-project/alsa-lib pull request #79 was opened from jason77-wang: We plan to use the sound driver module's name for the top ucm's name, I checked it on 3 different ASoC machines, the device/driver links to machine driver while deice/driver/module links to the sound driver module, so change the code to readlink the device/driver/module first, if it fails then readlink the device/driver. This is the output from those 3 machines: $readlink /sys/class/sound/card0/device/driver ../../../../bus/platform/drivers/skl_hda_dsp_generic $readlink /sys/class/sound/card0/device/driver/module ../../../../module/snd_soc_skl_hda_dsp $readlink /sys/class/sound/card2/device/driver ../../../../../bus/platform/drivers/acp_pdm_mach $readlink /sys/class/sound/card2/device/driver/module ../../../../module/snd_acp3x_rn $readlink /sys/class/sound/card0/device/driver ../../../../bus/platform/drivers/sof_sdw $readlink /sys/class/sound/card0/device/driver/module ../../../../module/snd_soc_sof_sdw Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Request URL : https://github.com/alsa-project/alsa-lib/pull/79 Patch URL : https://github.com/alsa-project/alsa-lib/pull/79.patch Repository URL: https://github.com/alsa-project/alsa-lib

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