May 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: qdsp6: q6core: Use struct_size() in kmemdup()
by Gustavo A. R. Silva 23 May '19

23 May '19

Make use of the struct_size() helper instead of an open-coded version in order to avoid any potential type mistakes, in particular in the context in which this code is being used. So, replace code of the following form: sizeof(*fwk) + fwk->num_services * sizeof(fwk->svc_api_info[0]); with: struct_size(fwk, svc_api_info, fwk->num_services) and so on... Notice that variables bytes and len are unnecessary, hence they are removed. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/qcom/qdsp6/q6core.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6core.c b/sound/soc/qcom/qdsp6/q6core.c index cdfc8ab6cfc0..ae314a652efe 100644 --- a/sound/soc/qcom/qdsp6/q6core.c +++ b/sound/soc/qcom/qdsp6/q6core.c @@ -98,13 +98,13 @@ static int q6core_callback(struct apr_device *adev, struct apr_resp_pkt *data) } case AVCS_CMDRSP_GET_FWK_VERSION: { struct avcs_cmdrsp_get_fwk_version *fwk; - int bytes; fwk = data->payload; - bytes = sizeof(*fwk) + fwk->num_services * - sizeof(fwk->svc_api_info[0]); - core->fwk_version = kmemdup(data->payload, bytes, GFP_ATOMIC); + core->fwk_version = kmemdup(data->payload, + struct_size(fwk, svc_api_info, + fwk->num_services), + GFP_ATOMIC); if (!core->fwk_version) return -ENOMEM; @@ -115,13 +115,13 @@ static int q6core_callback(struct apr_device *adev, struct apr_resp_pkt *data) } case AVCS_GET_VERSIONS_RSP: { struct avcs_cmdrsp_get_version *v; - int len; v = data->payload; - len = sizeof(*v) + v->num_services * sizeof(v->svc_api_info[0]); - - core->svc_version = kmemdup(data->payload, len, GFP_ATOMIC); + core->svc_version = kmemdup(data->payload, + struct_size(v, svc_api_info, + v->num_services), + GFP_ATOMIC); if (!core->svc_version) return -ENOMEM; -- 2.21.0

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[alsa-devel] [PATCH 1/5] ASoC: madera: Add DT bindings for Cirrus Logic Madera codecs
by Charles Keepax 23 May '19

23 May '19

From: Richard Fitzgerald <rf(a)opensource.cirrus.com> The Cirrus Logic Madera codecs are a family of related codecs with extensive digital and analogue I/O, digital mixing and routing, signal processing and programmable DSPs. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- Documentation/devicetree/bindings/sound/madera.txt | 67 ++++++++++++++++++++++ MAINTAINERS | 1 + include/dt-bindings/sound/madera.h | 29 ++++++++++ 3 files changed, 97 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/madera.txt create mode 100644 include/dt-bindings/sound/madera.h diff --git a/Documentation/devicetree/bindings/sound/madera.txt b/Documentation/devicetree/bindings/sound/madera.txt new file mode 100644 index 0000000000000..1114fcf1aa4c2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/madera.txt @@ -0,0 +1,67 @@ +Cirrus Logic Madera class audio codecs + +This describes audio configuration bindings for these codecs. + +See also the core bindings for the parent MFD driver: +See Documentation/devicetree/bindings/mfd/madera.txt + +and defines for values used in these bindings: +include/dt-bindings/sound/madera.h + +These properties are all contained in the parent MFD node. + +Optional properties: + - cirrus,dmic-ref : Indicates how the MICBIAS pins have been externally + connected to DMICs on each input, one cell per input. + <IN1 IN2 IN3 ...> + A value of 0 indicates MICVDD and is the default, other values depend on the + codec: + For CS47L35 one of the CS47L35_DMIC_REF_xxx values + For all other codecs one of the MADERA_DMIC_REF_xxx values + Also see the datasheet for a description of the INn_DMIC_SUP field. + + - cirrus,inmode : A list of input mode settings for each input. A maximum of + 16 cells, with four cells per input in the order INnAL, INnAR INnBL INnBR. + For non-muxed inputs the first two cells for that input set the mode for + the left and right channel and the second two cells must be 0. + For muxed inputs the first two cells for that input set the mode of the + left and right A inputs and the second two cells set the mode of the left + and right B inputs. + Valid mode values are one of the MADERA_INMODE_xxx. If the array is shorter + than the number of inputs the unspecified inputs default to + MADERA_INMODE_DIFF. + + - cirrus,out-mono : Mono bit for each output, must contain six cells if + specified. A non-zero value indicates the corresponding output is mono. + + - cirrus,max-channels-clocked : Maximum number of channels that I2S clocks + will be generated for. Useful when clock master for systems where the I2S + bus has multiple data lines. + One cell for each AIF, use a value of zero for AIFs that should be handled + normally. + + - cirrus,pdm-fmt : PDM speaker data format, must contain 2 cells + (OUT5 and OUT6). See the PDM_SPKn_FMT field in the datasheet for a + description of this value. + The second cell is ignored for codecs that do not have OUT6. + + - cirrus,pdm-mute : PDM mute format, must contain 2 cells + (OUT5 and OUT6). See the PDM_SPKn_CTRL_1 register in the datasheet for a + description of this value. + The second cell is ignored for codecs that do not have OUT6. + +Example: + +cs47l35@0 { + compatible = "cirrus,cs47l35"; + + cirrus,dmic-ref = <0 0 CS47L35_DMIC_REF_MICBIAS1B 0>; + cirrus,inmode = < + MADERA_INMODE_DMIC MADERA_INMODE_DMIC /* IN1A digital */ + MADERA_INMODE_SE MADERA_INMODE_SE /* IN1B single-ended */ + MADERA_INMODE_DIFF MADERA_INMODE_DIFF /* IN2 differential */ + 0 0 /* not used on this codec */ + >; + cirrus,out-mono = <0 0 0 0 0 0>; + cirrus,max-channels-clocked = <2 0 0>; +}; diff --git a/MAINTAINERS b/MAINTAINERS index 5cfbea4ce5750..642cb5610dd50 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -3915,6 +3915,7 @@ W: https://github.com/CirrusLogic/linux-drivers/wiki S: Supported F: Documentation/devicetree/bindings/mfd/madera.txt F: Documentation/devicetree/bindings/pinctrl/cirrus,madera-pinctrl.txt +F: include/dt-bindings/sound/madera* F: include/linux/irqchip/irq-madera* F: include/linux/mfd/madera/* F: drivers/gpio/gpio-madera* diff --git a/include/dt-bindings/sound/madera.h b/include/dt-bindings/sound/madera.h new file mode 100644 index 0000000000000..9ff4eae5259b0 --- /dev/null +++ b/include/dt-bindings/sound/madera.h @@ -0,0 +1,29 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Device Tree defines for Madera codecs + * + * Copyright (C) 2016-2017 Cirrus Logic, Inc. and + * Cirrus Logic International Semiconductor Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef DT_BINDINGS_SOUND_MADERA_H +#define DT_BINDINGS_SOUND_MADERA_H + +#define MADERA_INMODE_DIFF 0 +#define MADERA_INMODE_SE 1 +#define MADERA_INMODE_DMIC 2 + +#define MADERA_DMIC_REF_MICVDD 0 +#define MADERA_DMIC_REF_MICBIAS1 1 +#define MADERA_DMIC_REF_MICBIAS2 2 +#define MADERA_DMIC_REF_MICBIAS3 3 + +#define CS47L35_DMIC_REF_MICBIAS1B 1 +#define CS47L35_DMIC_REF_MICBIAS2A 2 +#define CS47L35_DMIC_REF_MICBIAS2B 3 + +#endif -- 2.11.0

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[alsa-devel] [PATCH] ASoC: Mediatek: MT8183: enable dev runtime suspend and resume
by Jiaxin Yu 23 May '19

23 May '19

System suspend will power off audio, so we need regcache sync when system resume. Remove pm_runtime_get_sync in dev probe and pm_runtime_put_sync in dev remove.So that audio dev can triggle runitme suspend and resume. Signed-off-by: Jiaxin Yu <jiaxin.yu(a)mediatek.com> --- Hi, This patch is based on for-next and test passed. --- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c index 56c3732..4a31106 100644 --- a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c +++ b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c @@ -1142,8 +1142,6 @@ static int mt8183_afe_pcm_dev_probe(struct platform_device *pdev) regcache_cache_only(afe->regmap, true); regcache_mark_dirty(afe->regmap); - pm_runtime_get_sync(&pdev->dev); - /* init memif */ afe->memif_size = MT8183_MEMIF_NUM; afe->memif = devm_kcalloc(dev, afe->memif_size, sizeof(*afe->memif), @@ -1235,11 +1233,10 @@ static int mt8183_afe_pcm_dev_probe(struct platform_device *pdev) static int mt8183_afe_pcm_dev_remove(struct platform_device *pdev) { - pm_runtime_put_sync(&pdev->dev); - pm_runtime_disable(&pdev->dev); if (!pm_runtime_status_suspended(&pdev->dev)) mt8183_afe_runtime_suspend(&pdev->dev); + return 0; } -- 1.8.1.1.dirty

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[alsa-devel] [PATCH v6 0/2] ASoC: CX2072X codec support (revised, v6)
by Takashi Iwai 23 May '19

23 May '19

Hi, the merge window was closed, and now it's time for fun again. This is a v6 revision of ASoC CX2072X codec support patchset for Intel platforms. Takashi === v5->v6: Drop buggy&untested EQ and DRC codes Lots of code reduction/cleanup in codec driver Add more comments about platform-specific stuff v4->v5: Move clk management to runtime PM Sparse warning fixes Some more code simplification Drop tricky regcache fiddling Apply mutex locks around possible racy sequences Move exported jack detection stuff into set_jack callback v3->v4: Fix the wrong endianess conversion in reg write Minor code cleanups v2->v3: CX2072X codec driver: * Move register tables to appropriate place * Remove some confusing codes * Set snd_ctl_boolean_* helpers directly * Fix EQ put callback * Rename to "DAC1 Switch" from "DAC1 Mute Switch" * Drop superfluous regmap calls at shutdown * Avoid regmap_register_patch() * Add missing register definitions * Fix register access on big-endian machine * Remove regcache messes CX2072X machine driver: * Add Pierre's ack * Drop superfluous ssp0 routes v1->v2: Uncomment SOF entries in ACPI binding Move snd_soc_dai_set_bclk_ratio() call into init callback The original cover letter is below: === It's been long time ago the first version of CX2072X codec patch was submitted, and I hoped that it'd be resubmitted, but never happened, as the original author seems to have left, unfortunately. Since I have such a laptop (ASUS E200HA), I took some time to cook and brush up the patches. The patches are for 5.2, should be applicable to ASoC tree for-next branch. Will post the corresponding UCM profile patch once when this gets accepted. thanks, Takashi === Simon Ho (1): ASoC: Add support for Conexant CX2072X CODEC Takashi Iwai (1): ASoC: Intel: Add machine driver for CX2072X on BYT/CHT platforms sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cx2072x.c | 1727 +++++++++++++++++++++ sound/soc/codecs/cx2072x.h | 314 ++++ sound/soc/intel/boards/Kconfig | 11 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bytcht_cx2072x.c | 262 ++++ sound/soc/intel/common/soc-acpi-intel-byt-match.c | 8 + sound/soc/intel/common/soc-acpi-intel-cht-match.c | 8 + 9 files changed, 2341 insertions(+) create mode 100644 sound/soc/codecs/cx2072x.c create mode 100644 sound/soc/codecs/cx2072x.h create mode 100644 sound/soc/intel/boards/bytcht_cx2072x.c -- 2.16.4

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[alsa-devel] [PATCH] ASoC: mediatek: mt8183: remove unused DAPM pins
by Tzung-Bi Shih 23 May '19

23 May '19

Kernel emits the following messages when booting. ASoC: DAPM unknown pin Headset Mic ASoC: DAPM unknown pin Headphone Remove the unused DAPM pins. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- .../soc/mediatek/mt8183/mt8183-da7219-max98357.c | 15 +-------------- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 16 +--------------- 2 files changed, 2 insertions(+), 29 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index 31ea8632c397..231fb49d79b9 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -18,18 +18,6 @@ static struct snd_soc_jack headset_jack; -/* Headset jack detection DAPM pins */ -static struct snd_soc_jack_pin headset_jack_pins[] = { - { - .pin = "Headphone", - .mask = SND_JACK_HEADPHONE, - }, - { - .pin = "Headset Mic", - .mask = SND_JACK_MICROPHONE, - }, -}; - static struct snd_soc_dai_link_component mt8183_da7219_max98357_external_codecs[] = { { @@ -375,8 +363,7 @@ mt8183_da7219_max98357_headset_init(struct snd_soc_component *component) SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, &headset_jack, - headset_jack_pins, - ARRAY_SIZE(headset_jack_pins)); + NULL, 0); if (ret) return ret; diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index 4e44e5689d6f..1ba7aa58e4b6 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -17,19 +17,6 @@ static struct snd_soc_jack headset_jack; -/* Headset jack detection DAPM pins */ -static struct snd_soc_jack_pin headset_jack_pins[] = { - { - .pin = "Headphone", - .mask = SND_JACK_HEADPHONE, - }, - { - .pin = "Headset Mic", - .mask = SND_JACK_MICROPHONE, - }, - -}; - static int mt8183_mt6358_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -328,8 +315,7 @@ mt8183_mt6358_ts3a227_max98357_headset_init(struct snd_soc_component *component) SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, &headset_jack, - headset_jack_pins, - ARRAY_SIZE(headset_jack_pins)); + NULL, 0); if (ret) return ret; -- 2.21.0.1020.gf2820cf01a-goog

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[alsa-devel] [PATCH] ASoC: mediatek: mt8183: move card registration to the end of probe
by Tzung-Bi Shih 23 May '19

23 May '19

mt8183_mt6358_ts3a227_max98357_dev_probe() does not return immediately if devm_snd_soc_register_card() returns fail. As a result, the rest of code still get executed. Two options to fix: 1. add an extra return statement if devm_snd_soc_register_card() returns fail 2. move devm_snd_soc_register_card() to the end of probe Adopt option 2 which is more clear and common. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- .../soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index 4e44e5689d6f..1efc7f5d04e4 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -345,7 +345,7 @@ mt8183_mt6358_ts3a227_max98357_dev_probe(struct platform_device *pdev) struct device_node *platform_node; struct snd_soc_dai_link *dai_link; struct pinctrl *default_pins; - int ret, i; + int i; card->dev = &pdev->dev; @@ -380,11 +380,6 @@ mt8183_mt6358_ts3a227_max98357_dev_probe(struct platform_device *pdev) return -EINVAL; } - ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) - dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", - __func__, ret); - default_pins = devm_pinctrl_get_select(&pdev->dev, PINCTRL_STATE_DEFAULT); if (IS_ERR(default_pins)) { @@ -393,7 +388,7 @@ mt8183_mt6358_ts3a227_max98357_dev_probe(struct platform_device *pdev) return PTR_ERR(default_pins); } - return ret; + return devm_snd_soc_register_card(&pdev->dev, card); } #ifdef CONFIG_OF -- 2.21.0.1020.gf2820cf01a-goog

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[alsa-devel] [RFC PATCH] ASoC: simple-card: Fix configuration of DAI format
by Jon Hunter 23 May '19

23 May '19

When configuring a codec to be both bit-clock and frame-master, it was found that the codec was always configured as bit-clock and frame-slave. Looking at the simple_dai_link_of() function there appears to be two problems with the configuration of the DAI format, which are ... 1. The function asoc_simple_parse_daifmt() is called before the function asoc_simple_parse_codec() and this means that the device-tree node for the codec has not been parsed yet, which is needed by the function asoc_simple_parse_daifmt() to determine who is the codec. 2. The phandle passed to asoc_simple_parse_daifmt() is the phandle to the 'codec' node and not the phandle of the actual codec defined by the 'sound-dai' property under the 'codec' node. Fix the above by moving the call to asoc_simple_parse_daifmt() after the the call to asoc_simple_parse_codec() and pass the phandle for the codec to asoc_simple_parse_daifmt(). Signed-off-by: Jon Hunter <jonathanh(a)nvidia.com> --- sound/soc/generic/simple-card.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 9b568f578bcd..c2c8dcbcf795 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -283,11 +283,6 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, codec_dai = dai_props->codec_dai = &priv->dais[li->dais++]; - ret = asoc_simple_parse_daifmt(dev, node, codec, - prefix, &dai_link->dai_fmt); - if (ret < 0) - goto dai_link_of_err; - simple_parse_mclk_fs(top, cpu, codec, dai_props, prefix); ret = asoc_simple_parse_cpu(cpu, dai_link, &single_cpu); @@ -298,6 +293,11 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, if (ret < 0) goto dai_link_of_err; + ret = asoc_simple_parse_daifmt(dev, node, dai_link->codecs->of_node, + prefix, &dai_link->dai_fmt); + if (ret < 0) + goto dai_link_of_err; + ret = asoc_simple_parse_platform(plat, dai_link); if (ret < 0) goto dai_link_of_err; -- 2.7.4

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[alsa-devel] [PATCH] ASoC: sound/soc/intel/boards: limit some drivers to X86 since headers are only in arch/x86/
by Randy Dunlap 23 May '19

23 May '19

From: Randy Dunlap <rdunlap(a)infradead.org> Several drivers in sound/soc/intel/boards/ #include header files that only exist in arch/x86/include/asm. This causes build errors, so make these drivers depend on X86. Fixes these build errors (on ia64): ../sound/soc/intel/boards/bxt_da7219_max98357a.c:19:10: fatal error: asm/cpu_device_id.h: No such file or directory #include <asm/cpu_device_id.h> ../sound/soc/intel/boards/bytcr_rt5640.c:31:10: fatal error: asm/cpu_device_id.h: No such file or directory #include <asm/cpu_device_id.h> ../sound/soc/intel/boards/bytcr_rt5651.c:33:10: fatal error: asm/cpu_device_id.h: No such file or directory #include <asm/cpu_device_id.h> ../sound/soc/intel/boards/cht_bsw_rt5645.c:29:10: fatal error: asm/cpu_device_id.h: No such file or directory #include <asm/cpu_device_id.h> ../sound/soc/intel/boards/bytcht_es8316.c:33:10: fatal error: asm/cpu_device_id.h: No such file or directory #include <asm/cpu_device_id.h> ../sound/soc/intel/boards/bytcht_da7213.c:26:10: fatal error: asm/platform_sst_audio.h: No such file or directory #include <asm/platform_sst_audio.h> And more drivers determined by: > grep "include.*asm.cpu_device_id.h" *.c bxt_da7219_max98357a.c:#include <asm/cpu_device_id.h> bytcht_es8316.c:#include <asm/cpu_device_id.h> bytcr_rt5640.c:#include <asm/cpu_device_id.h> bytcr_rt5651.c:#include <asm/cpu_device_id.h> cht_bsw_rt5645.c:#include <asm/cpu_device_id.h> sof_rt5682.c:#include <asm/cpu_device_id.h> and > grep "include.*asm.platform_sst_audio.h" *.c bytcht_da7213.c:#include <asm/platform_sst_audio.h> bytcht_es8316.c:#include <asm/platform_sst_audio.h> Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: Jie Yang <yang.jie(a)linux.intel.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/intel/boards/Kconfig | 6 ++++++ 1 file changed, 6 insertions(+) --- lnx-52-rc1.orig/sound/soc/intel/boards/Kconfig +++ lnx-52-rc1/sound/soc/intel/boards/Kconfig @@ -87,6 +87,7 @@ config SND_SOC_INTEL_BYTCR_RT5640_MACH tristate "Baytrail and Baytrail-CR with RT5640 codec" depends on I2C && ACPI depends on X86_INTEL_LPSS || COMPILE_TEST + depends on X86 select SND_SOC_ACPI select SND_SOC_RT5640 help @@ -99,6 +100,7 @@ config SND_SOC_INTEL_BYTCR_RT5651_MACH tristate "Baytrail and Baytrail-CR with RT5651 codec" depends on I2C && ACPI depends on X86_INTEL_LPSS || COMPILE_TEST + depends on X86 select SND_SOC_ACPI select SND_SOC_RT5651 help @@ -123,6 +125,7 @@ config SND_SOC_INTEL_CHT_BSW_RT5645_MACH tristate "Cherrytrail & Braswell with RT5645/5650 codec" depends on I2C && ACPI depends on X86_INTEL_LPSS || COMPILE_TEST + depends on X86 select SND_SOC_ACPI select SND_SOC_RT5645 help @@ -159,6 +162,7 @@ config SND_SOC_INTEL_BYT_CHT_DA7213_MACH tristate "Baytrail & Cherrytrail with DA7212/7213 codec" depends on I2C && ACPI depends on X86_INTEL_LPSS || COMPILE_TEST + depends on X86 select SND_SOC_ACPI select SND_SOC_DA7213 help @@ -171,6 +175,7 @@ config SND_SOC_INTEL_BYT_CHT_ES8316_MACH tristate "Baytrail & Cherrytrail with ES8316 codec" depends on I2C && ACPI depends on X86_INTEL_LPSS || COMPILE_TEST + depends on X86 select SND_SOC_ACPI select SND_SOC_ES8316 help @@ -249,6 +254,7 @@ config SND_SOC_INTEL_BXT_DA7219_MAX98357 tristate "Broxton with DA7219 and MAX98357A in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST + depends on X86 select SND_SOC_DA7219 select SND_SOC_MAX98357A select SND_SOC_DMIC

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[alsa-devel] [PATCH] ALSA: Enable micmute LED for Huawei laptops
by Ayman Bagabas 23 May '19

23 May '19

Since this LED is found on all Huawei laptops, we can hook it to huawei-wmi platform driver to control it. Also, some renames have been made to use product name instead of common name to avoid confusions. Signed-off-by: Ayman Bagabas <ayman.bagabas(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f83f21d64dd4..ab6a856ec614 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5747,7 +5747,7 @@ enum { ALC298_FIXUP_TPT470_DOCK, ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, - ALC256_FIXUP_HUAWEI_MBXP_PINS, + ALC256_FIXUP_HUAWEI_MACH_WX9_PINS, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, ALC285_FIXUP_LENOVO_HEADPHONE_NOISE, @@ -6038,7 +6038,7 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, - [ALC256_FIXUP_HUAWEI_MBXP_PINS] = { + [ALC256_FIXUP_HUAWEI_MACH_WX9_PINS] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { {0x12, 0x90a60130}, @@ -7063,9 +7063,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), - SND_PCI_QUIRK(0x19e5, 0x3200, "Huawei MBX", ALC255_FIXUP_MIC_MUTE_LED), - SND_PCI_QUIRK(0x19e5, 0x3201, "Huawei MBX", ALC255_FIXUP_MIC_MUTE_LED), - SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MBXP", ALC256_FIXUP_HUAWEI_MBXP_PINS), + SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ #if 0 @@ -7124,6 +7122,7 @@ static const struct snd_pci_quirk alc269_fixup_vendor_tbl[] = { SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", ALC269_FIXUP_THINKPAD_ACPI), + SND_PCI_QUIRK_VENDOR(0x19e5, "Huawei Matebook", ALC255_FIXUP_MIC_MUTE_LED), {} }; -- 2.20.1

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Re: [alsa-devel] [PATCH] ASoC: fsl_esai: fix the channel swap issue after xrun
by S.j. Wang 23 May '19

23 May '19

Hi > > + /* > > + * Add fifo reset here, because the regcache_sync will > > + * write one more data to ETDR. > > + * Which will cause channel shift. > > Sounds like a bug to me...should fix it first by marking the data registers as > volatile. > The ETDR is a writable register, it is not volatile. Even we change it to Volatile, I don't think we can't avoid this issue. for the regcache_sync Just to write this register, it is correct behavior. Best regards Wang shengjiu

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