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

[PATCH v2 0/2] Fix unchecked return value for clk_prepare_enable
by Shengjiu Wang 23 Jun '20

23 Jun '20

First patch is to remove the check of clock pointer before calling clk API. Second patch is to fix the issue that the return value of clk_prepare_enable is not checked. changes in v2: - split the patch to separate patches Shengjiu Wang (2): ASoC: fsl_mqs: Don't check clock is NULL before calling clk API ASoC: fsl_mqs: Fix unchecked return value for clk_prepare_enable sound/soc/fsl/fsl_mqs.c | 23 ++++++++++++++--------- 1 file changed, 14 insertions(+), 9 deletions(-) -- 2.21.0

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[PATCH] ASoC: rt1011: fix KASAN out-of-bounds bug in find_next_bit()
by Pierre-Louis Bossart 23 Jun '20

23 Jun '20

From: Fred Oh <fred.oh(a)linux.intel.com> KASAN throws the following warning in rt1011.c: [ 170.777603] BUG: KASAN: stack-out-of-bounds in _find_next_bit.constprop.0+0x3e/0xf0 find_next_bit() relies on unsigned long pointer arguments, but this driver uses a type cast that generates the KASAN warning. Replace find_next_bit() and find_last_bit() with __ffs() and __fls() to pass the value and avoid casting pointers to make the warning go away. Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Fred Oh <fred.oh(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/rt1011.c | 20 +++++++++++++------- 1 file changed, 13 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index dec5638060c3..098ecf13814d 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1849,13 +1849,13 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, /* Rx slot configuration */ rx_slotnum = hweight_long(rx_mask); - first_bit = find_next_bit((unsigned long *)&rx_mask, 32, 0); - if (rx_slotnum > 1 || rx_slotnum == 0) { + if (rx_slotnum > 1 || !rx_slotnum) { ret = -EINVAL; - dev_dbg(component->dev, "too many rx slots or zero slot\n"); + dev_err(component->dev, "too many rx slots or zero slot\n"); goto _set_tdm_err_; } + first_bit = __ffs(rx_mask); switch (first_bit) { case 0: case 2: @@ -1892,11 +1892,17 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, /* Tx slot configuration */ tx_slotnum = hweight_long(tx_mask); - first_bit = find_next_bit((unsigned long *)&tx_mask, 32, 0); - last_bit = find_last_bit((unsigned long *)&tx_mask, 32); - if (tx_slotnum > 2 || (last_bit-first_bit) > 1) { + if (tx_slotnum > 2 || !tx_slotnum) { ret = -EINVAL; - dev_dbg(component->dev, "too many tx slots or tx slot location error\n"); + dev_err(component->dev, "too many tx slots or zero slot\n"); + goto _set_tdm_err_; + } + + first_bit = __ffs(tx_mask); + last_bit = __fls(tx_mask); + if (last_bit - first_bit > 1) { + ret = -EINVAL; + dev_err(component->dev, "tx slot location error\n"); goto _set_tdm_err_; } base-commit: 39853b1438bf9b07349c8c44b48f6c2eda6f8840 -- 2.20.1

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[PATCH] ASoC: mediatek: mt8183-da7219: set playback and capture constraints
by Tzung-Bi Shih 23 Jun '20

23 Jun '20

Sets playback and capture constraints to S16_LE, stereo, 48kHz. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- .../mediatek/mt8183/mt8183-da7219-max98357.c | 42 +++++++++++++++++++ 1 file changed, 42 insertions(+) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index ffd7c931e7bb..b61bb2de4ec3 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -119,6 +119,46 @@ static int mt8183_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } +static int +mt8183_da7219_max98357_startup( + struct snd_pcm_substream *substream) +{ + static const unsigned int rates[] = { + 48000, + }; + static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, + }; + static const unsigned int channels[] = { + 2, + }; + static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, + }; + + struct snd_pcm_runtime *runtime = substream->runtime; + + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); + runtime->hw.channels_max = 2; + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + + return 0; +} + +static const struct snd_soc_ops mt8183_da7219_max98357_ops = { + .startup = mt8183_da7219_max98357_startup, +}; + static int mt8183_da7219_max98357_bt_sco_startup( struct snd_pcm_substream *substream) @@ -256,6 +296,7 @@ static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = { SND_SOC_DPCM_TRIGGER_PRE}, .dynamic = 1, .dpcm_playback = 1, + .ops = &mt8183_da7219_max98357_ops, SND_SOC_DAILINK_REG(playback1), }, { @@ -303,6 +344,7 @@ static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = { SND_SOC_DPCM_TRIGGER_PRE}, .dynamic = 1, .dpcm_capture = 1, + .ops = &mt8183_da7219_max98357_ops, SND_SOC_DAILINK_REG(capture3), }, { -- 2.27.0.111.gc72c7da667-goog

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alsa-lib: Bug fixes to namehint, dsnoop and minor changes
by Mark Hills 23 Jun '20

23 Jun '20

I've been running these patches in some form for a while. Now tidied up for RFC. It's quite possible I misinterpreted something; the behaviours seem to be quite long-standing, hence a first pass for comments. This has taken me to the _avail() functions, which do not seem to be prepared for signed output, yet callers interpret as signed and ultimately end up in external API. Their handling of wraparound conditions etc. may not be as expected. I've already started to look at these. -- Mark

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[PATCH] ALSA: hda/conexant: Fix missing headphone node after reboot from Windows
by Takashi Iwai 23 Jun '20

23 Jun '20

When Lenovo Thinkpad E495 is rebooted from Windows, the headphone control is lost. This seems because BIOS doesn't initialize the headphone node (NID 0x16) properly after the reboot from Windows, as Windows likely turns off the power at its exit. The node is left as a vendor-specific unknown node and ignored by the HD-audio parser. This patch forcibly initializes the node at the probe time, by overriding the widget caps and the connection list, so that the headphone control is detected always at probe time. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207741 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_conexant.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 396b5503038a..37c8add48791 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -215,6 +215,7 @@ enum { CXT_FIXUP_MUTE_LED_GPIO, CXT_FIXUP_HEADSET_MIC, CXT_FIXUP_HP_MIC_NO_PRESENCE, + CXT_FIXUP_THINKPAD_E495, }; /* for hda_fixup_thinkpad_acpi() */ @@ -673,6 +674,22 @@ static void cxt_fixup_mute_led_gpio(struct hda_codec *codec, spec->gpio_led); } +static void cxt_fixup_thinkpad_e495(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + static const hda_nid_t conns[] = { 0x10, 0x11 }; + /* BIOS invalidates the headphone pin NID 0x16 when rebooted + * after Windows by some reason; + * forcibly restoring the whole setup + */ + snd_hda_override_wcaps(codec, 0x16, 0x400581); + snd_hda_codec_set_pincfg(codec, 0x16, 0x03211040); + snd_hda_override_conn_list(codec, 0x16, ARRAY_SIZE(conns), + conns); + } +} /* ThinkPad X200 & co with cxt5051 */ static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = { @@ -846,6 +863,12 @@ static const struct hda_fixup cxt_fixups[] = { .chained = true, .chain_id = CXT_FIXUP_HEADSET_MIC, }, + [CXT_FIXUP_THINKPAD_E495] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_thinkpad_e495, + .chained = true, + .chain_id = CXT_FIXUP_THINKPAD_ACPI, + }, }; static const struct snd_pci_quirk cxt5045_fixups[] = { @@ -932,6 +955,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo G50-70", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x17aa, 0x5124, "Lenovo Thinkpad E495", CXT_FIXUP_THINKPAD_E495), SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", CXT_FIXUP_THINKPAD_ACPI), SND_PCI_QUIRK(0x1c06, 0x2011, "Lemote A1004", CXT_PINCFG_LEMOTE_A1004), SND_PCI_QUIRK(0x1c06, 0x2012, "Lemote A1205", CXT_PINCFG_LEMOTE_A1205), -- 2.16.4

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Announcement for alsa-gobject v0.1.0 release
by Takashi Sakamoto 23 Jun '20

23 Jun '20

Hi, This is an announcement for the first official release of alsa-gobject project, addressed in Audio Mini Conference 2018[1]. The release is available in below URL. * https://github.com/alsa-project/alsa-gobject/releases/tag/v0.1.0 The release includes below libraries to operate corresponding character devices of Linux Sound subsystem (=Advanced Linux Sound Architecture, a.k.a ALSA)[2]: * libalsactl0 * libalsatimer0 * libalsaseq0 * libalsahwdep0 * libalsarawmidi0 The libraries are also compatible with gobject-introspection (g-i)[3] to produce gir files: * ALSACtl-0.0 * ALSATimer-0.0 * ALSASeq-0.0 * ALSAHwdep-0.0 * ALSARawmidi-0.0 Any language binding for g-i can handle the gir files to use the the libraries. For example, PyGObject[4] is available as dynamic g-i loader in Python 3 runtime, and some Python 3 sample scripts are available in `samples` directory. * https://github.com/alsa-project/alsa-gobject/tree/master/samples Rust FFI/Safe API crates are provided in alsa-gobject-rs, automatically generated by gtk-rs/gir[5]. Its README includes code snippet. * https://github.com/alsa-project/alsa-gobject-rs Documentation is available in below URLs. They're auto-generated by gtk-doc[6] with function annotations: * https://alsa-project.github.io/alsa-gobject-docs/ As a design note, the libraries provide API to execute system calls directly to ALSA core implementation in kernel land. The libraries produce minimal set of API, and delegate enhanced features to application side. They perform like 'glue' code between application and in-kernel functionalities. This makes it easy to write test code for the in-kernel functionalities as well as to write ehhanced features by each language. The libraries have no dependency on alsa-lib[7]. The alsa-lib includes own software features such as configuration space with plug-in system. Although the plug-in system is useful to add sample operating feature into PCM interface, it's not necessarily useful for the other interfaces. Furthermore it's better design to implement enhanced software features in user library and user application side, instead of common library side. For the above points, alsa-gobject produces languag bindings different from the existent ones[8][9]. In the release, PCM interface is not supported yet because the 'glue' is already provided by the other projects; GStreamer[10], PortAudio[11], SDL[12], and so on. They mostly satisfy the demand, therefore the support is judged not to be so urgent. It's planned to support PCM interfaces in future release as 'ALSAPcm-0.0'. Compress-Offload interface is not supported as well because it's difficult to buy free test device in market as of 2020. [1] https://www.alsa-project.org/wiki/Miniconf_2018#GObject_introspection [2] http://www.alsa-project.org/ [3] https://gi.readthedocs.io/ [4] https://pygobject.readthedocs.io/ [5] https://github.com/gtk-rs/gir [6] https://developer.gnome.org/gtk-doc-manual/ [7] https://git.alsa-project.org/?p=alsa-lib.git;a=summary [8] https://git.alsa-project.org/?p=alsa-python.git;a=summary [9] https://github.com/diwic/alsa-rs/ [10] https://gstreamer.freedesktop.org/bindings/ [11] http://www.portaudio.com/ [12] https://www.libsdl.org/languages.php Regards Takashi Sakamoto

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[PATCH v3 0/3] ASoC: ti: Add support for audio on J721e EVM
by Peter Ujfalusi 23 Jun '20

23 Jun '20

Hi, Changes since v2: - DT binding: - use proper (?) patch subject for the binding docuemtn patch - drop pll4 and pll15 from DT - driver should check the rate via clk_get_parent. If it is not available (as it is not currently) then use the match_data provided rates. - add simple explanation for the clocking setup - Use descriptive names for clocks: cpb/ivi-mcasp-auxclk and cpb/ivi-codec-scki - dt_binding_check shows no errors/warnings - ASoC machine driver: - Try to read the PLL4/15 rate with clk API (parent of the two clock divider) if it is not available then use the match_data provided numbers. - Support for single PLL setup Changes since v1: - Fixed DT binding documentation errors - Rebased on ASoC head and updated the driver to compile and work This series adds support for the analog audio setup on the j721e EVM. The audio setup of the EVM is: Common Processor Board (CPB): McASP10 <-> pcm3168a Infotainment Expansion Board (IVI): McASP0 <-> 2x pcm3168a Both CPB and IVI wired in parallel serializer setup. The first patch adds the stream_name for McASP driver as it is needed in multicodec (and would be needed in DPCM) setup for proper DAPM handling. The second patch adds two DT schema, one for the cpb and one for the cpb+ivi card. Regards, Peter --- Peter Ujfalusi (3): ASoC: ti: davinci-mcasp: Specify stream_name for playback/capture ASoC: dt-bindings: Add documentation for TI j721e EVM (CPB and IVI) ASoC: ti: Add custom machine driver for j721e EVM (CPB and IVI) .../bindings/sound/ti,j721e-cpb-audio.yaml | 95 ++ .../sound/ti,j721e-cpb-ivi-audio.yaml | 150 +++ sound/soc/ti/Kconfig | 8 + sound/soc/ti/Makefile | 2 + sound/soc/ti/davinci-mcasp.c | 3 + sound/soc/ti/j721e-evm.c | 887 ++++++++++++++++++ 6 files changed, 1145 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml create mode 100644 Documentation/devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml create mode 100644 sound/soc/ti/j721e-evm.c -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[PATCH] dt-bindings: ASoC: Convert UniPhier EVEA codec to json-schema
by Masahiro Yamada 23 Jun '20

23 Jun '20

Convert the UniPhier EVEA sound codec binding to DT schema format. Signed-off-by: Masahiro Yamada <yamada.masahiro(a)socionext.com> --- .../sound/socionext,uniphier-evea.yaml | 62 +++++++++++++++++++ .../bindings/sound/uniphier,evea.txt | 26 -------- 2 files changed, 62 insertions(+), 26 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml delete mode 100644 Documentation/devicetree/bindings/sound/uniphier,evea.txt diff --git a/Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml b/Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml new file mode 100644 index 000000000000..7ac1c0140d5d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml @@ -0,0 +1,62 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/socionext,uniphier-evea.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: UniPhier EVEA SoC-internal sound codec + +maintainers: + - <alsa-devel(a)alsa-project.org> + +properties: + compatible: + const: socionext,uniphier-evea + + reg: + maxItems: 1 + + clock-names: + items: + - const: evea + - const: exiv + + clocks: + minItems: 2 + maxItems: 2 + + reset-names: + items: + - const: evea + - const: exiv + - const: adamv + + resets: + minItems: 3 + maxItems: 3 + + "#sound-dai-cells": + const: 1 + +additionalProperties: false + +required: + - compatible + - reg + - clock-names + - clocks + - reset-names + - resets + - "#sound-dai-cells" + +examples: + - | + codec@57900000 { + compatible = "socionext,uniphier-evea"; + reg = <0x57900000 0x1000>; + clock-names = "evea", "exiv"; + clocks = <&sys_clk 41>, <&sys_clk 42>; + reset-names = "evea", "exiv", "adamv"; + resets = <&sys_rst 41>, <&sys_rst 42>, <&adamv_rst 0>; + #sound-dai-cells = <1>; + }; diff --git a/Documentation/devicetree/bindings/sound/uniphier,evea.txt b/Documentation/devicetree/bindings/sound/uniphier,evea.txt deleted file mode 100644 index 3f31b235f18b..000000000000 --- a/Documentation/devicetree/bindings/sound/uniphier,evea.txt +++ /dev/null @@ -1,26 +0,0 @@ -Socionext EVEA - UniPhier SoC internal codec driver - -Required properties: -- compatible : should be "socionext,uniphier-evea". -- reg : offset and length of the register set for the device. -- clock-names : should include following entries: - "evea", "exiv" -- clocks : a list of phandle, should contain an entry for each - entries in clock-names. -- reset-names : should include following entries: - "evea", "exiv", "adamv" -- resets : a list of phandle, should contain reset entries of - reset-names. -- #sound-dai-cells: should be 1. - -Example: - - codec { - compatible = "socionext,uniphier-evea"; - reg = <0x57900000 0x1000>; - clock-names = "evea", "exiv"; - clocks = <&sys_clk 41>, <&sys_clk 42>; - reset-names = "evea", "exiv", "adamv"; - resets = <&sys_rst 41>, <&sys_rst 42>, <&adamv_rst 0>; - #sound-dai-cells = <1>; - }; -- 2.25.1

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[PATCH] dt-bindings: ASoC: Convert UniPhier AIO audio system to json-schema
by Masahiro Yamada 23 Jun '20

23 Jun '20

Convert the UniPhier AIO audio system binding to DT schema format. Signed-off-by: Masahiro Yamada <yamada.masahiro(a)socionext.com> --- .../sound/socionext,uniphier-aio.yaml | 73 +++++++++++++++++++ .../bindings/sound/uniphier,aio.txt | 45 ------------ 2 files changed, 73 insertions(+), 45 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml delete mode 100644 Documentation/devicetree/bindings/sound/uniphier,aio.txt diff --git a/Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml b/Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml new file mode 100644 index 000000000000..bea8b06ff1b9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml @@ -0,0 +1,73 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/socionext,uniphier-aio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: UniPhier AIO audio system + +maintainers: + - <alsa-devel(a)alsa-project.org> + +properties: + compatible: + enum: + - socionext,uniphier-ld11-aio + - socionext,uniphier-ld20-aio + - socionext,uniphier-pxs2-aio + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clock-names: + const: aio + + clocks: + maxItems: 1 + + reset-names: + const: aio + + resets: + maxItems: 1 + + socionext,syscon: + description: | + Specifies a phandle to soc-glue, which is used for changing mode of S/PDIF + signal pin to output from Hi-Z. This property is optional if you use I2S + signal pins only. + $ref: "/schemas/types.yaml#/definitions/phandle" + + "#sound-dai-cells": + const: 1 + +additionalProperties: false + +required: + - compatible + - reg + - interrupts + - clock-names + - clocks + - reset-names + - resets + - "#sound-dai-cells" + +examples: + - | + audio@56000000 { + compatible = "socionext,uniphier-ld20-aio"; + reg = <0x56000000 0x80000>; + interrupts = <0 144 4>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_aout>; + clock-names = "aio"; + clocks = <&sys_clk 40>; + reset-names = "aio"; + resets = <&sys_rst 40>; + #sound-dai-cells = <1>; + socionext,syscon = <&soc_glue>; + }; diff --git a/Documentation/devicetree/bindings/sound/uniphier,aio.txt b/Documentation/devicetree/bindings/sound/uniphier,aio.txt deleted file mode 100644 index 4ce68ed6f2f2..000000000000 --- a/Documentation/devicetree/bindings/sound/uniphier,aio.txt +++ /dev/null @@ -1,45 +0,0 @@ -Socionext UniPhier SoC audio driver - -The Socionext UniPhier audio subsystem consists of I2S and S/PDIF blocks in -the same register space. - -Required properties: -- compatible : should be one of the following: - "socionext,uniphier-ld11-aio" - "socionext,uniphier-ld20-aio" - "socionext,uniphier-pxs2-aio" -- reg : offset and length of the register set for the device. -- interrupts : should contain I2S or S/PDIF interrupt. -- pinctrl-names : should be "default". -- pinctrl-0 : defined I2S signal pins for an external codec chip. -- clock-names : should include following entries: - "aio" -- clocks : a list of phandle, should contain an entry for each - entry in clock-names. -- reset-names : should include following entries: - "aio" -- resets : a list of phandle, should contain an entry for each - entry in reset-names. -- #sound-dai-cells: should be 1. - -Optional properties: -- socionext,syscon: a phandle, should contain soc-glue. - The soc-glue is used for changing mode of S/PDIF signal pin - to Output from Hi-Z. This property is optional if you use - I2S signal pins only. - -Example: - audio { - compatible = "socionext,uniphier-ld20-aio"; - reg = <0x56000000 0x80000>; - interrupts = <0 144 4>; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_aout>; - clock-names = "aio"; - clocks = <&sys_clk 40>; - reset-names = "aio"; - resets = <&sys_rst 40>; - #sound-dai-cells = <1>; - - socionext,syscon = <&sg>; - }; -- 2.25.1

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Re: [PATCH] ALSA: usb-audio: Add registration quirk for Kingston HyperX Cloud Flight S
by Takashi Iwai 23 Jun '20

23 Jun '20

On Fri, 19 Jun 2020 13:48:22 +0200, Christoffer Nielsen wrote: > > Similar to the Kingston HyperX AMP, the Kingston HyperX Cloud > Alpha S (0951:0x16ea) uses two interfaces, but only the second > interface contains the capture stream. This patch delays the > registration until the second interface appears. > > > Signed-off-by: Christoffer Nielsen <segadora93(a)gmail.com> Applied now (with some correction and Cc to stable). thanks, Takashi

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