August 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH 00/15] ALSA: Improved WC memory handling
by Takashi Iwai 04 Aug '21

04 Aug '21

Hi, this is a patchset to enhance the ALSA memory allocation helper to support the allocation of WC pages more generically as well as supporting the fixed size buffer allocation. It allows us to reduce lots of redundant code in various ASoC drivers that still haven't used the standard preallocation. Takashi === Takashi Iwai (15): ALSA: memalloc: Minor refactoring ALSA: memalloc: Correctly name as WC ALSA: pcm: Allow exact buffer preallocation ALSA: memalloc: Support WC allocation on all architectures ALSA: pxa2xx: Use managed PCM buffer allocation ASoC: bcm: Use managed PCM buffer allocation ASoC: fsl: imx-pcm-fiq: Use managed buffer allocation ASoC: fsl: imx-pcm-rpmsg: Use managed buffer allocation ASoC: tegra: Use managed buffer allocation ASoC: fsl_asrc_dma: Use managed buffer allocation ASoC: fsl_dma: Use managed buffer allocation ASoC: mpc5200: Use managed buffer allocation ASoC: qcom: lpass: Use managed buffer allocation ASoC: qcom: qdsp6: Use managed buffer allocation ASoC: sprd: Use managed buffer allocation include/sound/memalloc.h | 6 +- include/sound/pcm.h | 47 ++++++++++++-- include/sound/pxa2xx-lib.h | 13 +--- sound/arm/pxa2xx-ac97.c | 18 ++---- sound/arm/pxa2xx-pcm-lib.c | 90 ++------------------------ sound/core/memalloc.c | 94 +++++++++++++++++----------- sound/core/memalloc_local.h | 2 +- sound/core/pcm_memory.c | 67 +++++++++++++++----- sound/core/sgbuf.c | 21 ++++--- sound/pci/hda/hda_controller.c | 2 +- sound/pci/hda/hda_intel.c | 2 +- sound/pci/intel8x0.c | 2 +- sound/soc/bcm/bcm63xx-pcm-whistler.c | 82 ++---------------------- sound/soc/fsl/fsl_asrc_dma.c | 47 +------------- sound/soc/fsl/fsl_dma.c | 50 +-------------- sound/soc/fsl/imx-pcm-fiq.c | 75 +--------------------- sound/soc/fsl/imx-pcm-rpmsg.c | 89 +------------------------- sound/soc/fsl/mpc5200_dma.c | 63 +------------------ sound/soc/pxa/pxa-ssp.c | 3 - sound/soc/pxa/pxa2xx-ac97.c | 3 - sound/soc/pxa/pxa2xx-i2s.c | 3 - sound/soc/pxa/pxa2xx-pcm.c | 3 - sound/soc/qcom/lpass-platform.c | 60 +----------------- sound/soc/qcom/qdsp6/q6asm-dai.c | 63 +------------------ sound/soc/sprd/sprd-pcm-dma.c | 67 +------------------- sound/soc/tegra/tegra210_admaif.c | 6 -- sound/soc/tegra/tegra_pcm.c | 93 +-------------------------- sound/soc/tegra/tegra_pcm.h | 7 --- sound/x86/intel_hdmi_audio.c | 2 +- 29 files changed, 209 insertions(+), 871 deletions(-) === Cc: Banajit Goswami <bgoswami(a)codeaurora.org> Cc: Daniel Mack <daniel(a)zonque.org> Cc: Fabio Estevam <festevam(a)gmail.com> Cc: Haojian Zhuang <haojian.zhuang(a)gmail.com> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Robert Jarzmik <robert.jarzmik(a)free.fr> Cc: Shengjiu Wang <shengjiu.wang(a)gmail.com> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> -- 2.26.2

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[PATCH v2] ALSA: memalloc: Store snd_dma_buffer.addr for continuous pages, too
by Takashi Iwai 04 Aug '21

04 Aug '21

In the recent fix commit eda80d7c9c4d ("ALSA: memalloc: Fix regression with SNDRV_DMA_TYPE_CONTINUOUS"), we replaced the pfn argument of the remap_page_pfn() call from the uninitialized dmab->addr. It was the right fix, but it'd be more generic if we actually initialize dmab->area for SNDRV_DMA_TYPE_CONTINOUS, too. e.g. the field is used in the common snd_sgbuf_get_addr(), too. This patch adds the initialization of addr field and does revert of the previous change to refer to it again in the mmap call. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: Fix a wrong buffer pointer to convert sound/core/memalloc.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 084b2600a423..1cea8cb9668f 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -242,8 +242,11 @@ EXPORT_SYMBOL(snd_sgbuf_get_chunk_size); static void *snd_dma_continuous_alloc(struct snd_dma_buffer *dmab, size_t size) { gfp_t gfp = snd_mem_get_gfp_flags(dmab, GFP_KERNEL); + void *p = alloc_pages_exact(size, gfp); - return alloc_pages_exact(size, gfp); + if (p) + dmab->addr = page_to_phys(virt_to_page(p)); + return p; } static void snd_dma_continuous_free(struct snd_dma_buffer *dmab) @@ -255,7 +258,7 @@ static int snd_dma_continuous_mmap(struct snd_dma_buffer *dmab, struct vm_area_struct *area) { return remap_pfn_range(area, area->vm_start, - page_to_pfn(virt_to_page(dmab->area)), + dmab->addr >> PAGE_SHIFT, area->vm_end - area->vm_start, area->vm_page_prot); } -- 2.26.2

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[PATCH] ALSA: memalloc: Store snd_dma_buffer.addr for continuous pages, too
by Takashi Iwai 04 Aug '21

04 Aug '21

In the recent fix commit eda80d7c9c4d ("ALSA: memalloc: Fix regression with SNDRV_DMA_TYPE_CONTINUOUS"), we replaced the pfn argument of the remap_page_pfn() call from the uninitialized dmab->addr. It was the right fix, but it'd be more generic if we actually initialize dmab->area for SNDRV_DMA_TYPE_CONTINOUS, too. e.g. the field is used in the common snd_sgbuf_get_addr(), too. This patch adds the initialization of addr field and does revert of the previous change to refer to it again in the mmap call. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/memalloc.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 084b2600a423..21f57f1e81a4 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -242,8 +242,11 @@ EXPORT_SYMBOL(snd_sgbuf_get_chunk_size); static void *snd_dma_continuous_alloc(struct snd_dma_buffer *dmab, size_t size) { gfp_t gfp = snd_mem_get_gfp_flags(dmab, GFP_KERNEL); + void *p = alloc_pages_exact(size, gfp); - return alloc_pages_exact(size, gfp); + if (p) + dmab->addr = page_to_phys(virt_to_page(dmab->area)); + return p; } static void snd_dma_continuous_free(struct snd_dma_buffer *dmab) @@ -255,7 +258,7 @@ static int snd_dma_continuous_mmap(struct snd_dma_buffer *dmab, struct vm_area_struct *area) { return remap_pfn_range(area, area->vm_start, - page_to_pfn(virt_to_page(dmab->area)), + dmab->addr >> PAGE_SHIFT, area->vm_end - area->vm_start, area->vm_page_prot); } -- 2.26.2

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[PATCH] ALSA: memalloc: Fix pgprot for WC mmap on x86
by Takashi Iwai 04 Aug '21

04 Aug '21

We have a special handling of WC pages on x86, and it's currently specific to HD-audio. The last forgotten piece was the pgprot setup for the mmap with WC pages. This patch moves the pgprot setup for WC pages from HD-audio-specific mmap callback to the common helper code. It allows us to remove the superfluous mmap callback in HD-audio and its prepare_mmap redirection. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/memalloc.c | 4 ++++ sound/pci/hda/hda_controller.c | 11 ----------- sound/pci/hda/hda_controller.h | 2 -- sound/pci/hda/hda_intel.c | 12 ------------ 4 files changed, 4 insertions(+), 25 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 58bab2cfdb87..084b2600a423 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -407,6 +407,10 @@ static void snd_dma_dev_free(struct snd_dma_buffer *dmab) static int snd_dma_dev_mmap(struct snd_dma_buffer *dmab, struct vm_area_struct *area) { +#ifdef CONFIG_X86 + if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC) + area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); +#endif return dma_mmap_coherent(dmab->dev.dev, area, dmab->area, dmab->addr, dmab->bytes); } diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index ea85c714d1cf..7cd452831fd3 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -669,16 +669,6 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) return err; } -static int azx_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *area) -{ - struct azx_pcm *apcm = snd_pcm_substream_chip(substream); - struct azx *chip = apcm->chip; - if (chip->ops->pcm_mmap_prepare) - chip->ops->pcm_mmap_prepare(substream, area); - return snd_pcm_lib_default_mmap(substream, area); -} - static const struct snd_pcm_ops azx_pcm_ops = { .open = azx_pcm_open, .close = azx_pcm_close, @@ -688,7 +678,6 @@ static const struct snd_pcm_ops azx_pcm_ops = { .trigger = azx_pcm_trigger, .pointer = azx_pcm_pointer, .get_time_info = azx_get_time_info, - .mmap = azx_pcm_mmap, }; static void azx_pcm_free(struct snd_pcm *pcm) diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index e4b290de81de..3062f87380b1 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -74,8 +74,6 @@ struct azx; struct hda_controller_ops { /* Disable msi if supported, PCI only */ int (*disable_msi_reset_irq)(struct azx *); - void (*pcm_mmap_prepare)(struct snd_pcm_substream *substream, - struct vm_area_struct *area); /* Check if current position is acceptable */ int (*position_check)(struct azx *chip, struct azx_dev *azx_dev); /* enable/disable the link power */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9771996e2ef5..3aa432d814a2 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2037,17 +2037,6 @@ static int disable_msi_reset_irq(struct azx *chip) return 0; } -static void pcm_mmap_prepare(struct snd_pcm_substream *substream, - struct vm_area_struct *area) -{ -#ifdef CONFIG_X86 - struct azx_pcm *apcm = snd_pcm_substream_chip(substream); - struct azx *chip = apcm->chip; - if (chip->uc_buffer) - area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); -#endif -} - /* Denylist for skipping the whole probe: * some HD-audio PCI entries are exposed without any codecs, and such devices * should be ignored from the beginning. @@ -2061,7 +2050,6 @@ static const struct pci_device_id driver_denylist[] = { static const struct hda_controller_ops pci_hda_ops = { .disable_msi_reset_irq = disable_msi_reset_irq, - .pcm_mmap_prepare = pcm_mmap_prepare, .position_check = azx_position_check, }; -- 2.26.2

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[PATCH] ASoC: mediatek: mt6359: convert to use module_platform_driver
by Nobuhiro Iwamatsu 04 Aug '21

04 Aug '21

Simplify the code by using module_platform_driver macro for mt6359-accdet. Signed-off-by: Nobuhiro Iwamatsu <nobuhiro1.iwamatsu(a)toshiba.co.jp> --- sound/soc/codecs/mt6359-accdet.c | 17 +---------------- 1 file changed, 1 insertion(+), 16 deletions(-) diff --git a/sound/soc/codecs/mt6359-accdet.c b/sound/soc/codecs/mt6359-accdet.c index 4222aed013f106..b2354a613b1587 100644 --- a/sound/soc/codecs/mt6359-accdet.c +++ b/sound/soc/codecs/mt6359-accdet.c @@ -1057,22 +1057,7 @@ static struct platform_driver mt6359_accdet_driver = { .probe = mt6359_accdet_probe, }; -static int __init mt6359_accdet_driver_init(void) -{ - int ret = 0; - - ret = platform_driver_register(&mt6359_accdet_driver); - if (ret) - return -ENODEV; - return 0; -} - -static void __exit mt6359_accdet_driver_exit(void) -{ - platform_driver_unregister(&mt6359_accdet_driver); -} -module_init(mt6359_accdet_driver_init); -module_exit(mt6359_accdet_driver_exit); +module_platform_driver(mt6359_accdet_driver) /* Module information */ MODULE_DESCRIPTION("MT6359 ALSA SoC codec jack driver"); -- 2.30.0

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[PATCH 0/4] ASoC: soc-ops: cleanup cppcheck warning
by Kuninori Morimoto 04 Aug '21

04 Aug '21

Hi Mark Now I'm posting audio-graph-card2 patch-set, but it seems it needs longer discussion. Thus I want to post more easy patch first, and reduce my local patches. These are cppcheck warning cleanup patches for soc-ops. Kuninori Morimoto (4): ASoC: soc-ops: cleanup cppcheck warning at snd_soc_put_volsw_sx() ASoC: soc-ops: cleanup cppcheck warning at snd_soc_limit_volume() ASoC: soc-ops: cleanup cppcheck warning at snd_soc_get_xr_sx() ASoC: soc-ops: cleanup cppcheck warning at snd_soc_put_xr_sx() sound/soc/soc-ops.c | 21 ++++++++++----------- 1 file changed, 10 insertions(+), 11 deletions(-) -- 2.25.1

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[PATCH] ASoC: codecs: ad193x: add support for 96kHz and 192kHz playback rates
by Codrin Ciubotariu 04 Aug '21

04 Aug '21

ad193x devices support 96KHz and 192KHz sampling rates, when PLL/MCLK is referenced to 48kHz. Tested on ad1934. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- sound/soc/codecs/ad193x.c | 30 ++++++++++++++++++++++++++++-- sound/soc/codecs/ad193x.h | 4 ++++ 2 files changed, 32 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index f37ab7eda615..278a55af158b 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -316,6 +316,13 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, int word_len = 0, master_rate = 0; struct snd_soc_component *component = dai->component; struct ad193x_priv *ad193x = snd_soc_component_get_drvdata(component); + bool is_playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + u8 dacc0; + + dev_dbg(dai->dev, "%s() rate=%u format=%#x width=%u channels=%u\n", + __func__, params_rate(params), params_format(params), + params_width(params), params_channels(params)); + /* bit size */ switch (params_width(params)) { @@ -346,6 +353,25 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, break; } + if (is_playback) { + switch (params_rate(params)) { + case 48000: + dacc0 = AD193X_DAC_SR_48; + break; + case 96000: + dacc0 = AD193X_DAC_SR_96; + break; + case 192000: + dacc0 = AD193X_DAC_SR_192; + break; + default: + dev_err(dai->dev, "invalid sampling rate: %d\n", params_rate(params)); + return -EINVAL; + } + + regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL0, AD193X_DAC_SR_MASK, dacc0); + } + regmap_update_bits(ad193x->regmap, AD193X_PLL_CLK_CTRL0, AD193X_PLL_INPUT_MASK, master_rate); @@ -385,7 +411,7 @@ static struct snd_soc_dai_driver ad193x_dai = { .stream_name = "Playback", .channels_min = 2, .channels_max = 8, - .rates = SNDRV_PCM_RATE_48000, + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000, .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, }, @@ -407,7 +433,7 @@ static struct snd_soc_dai_driver ad193x_no_adc_dai = { .stream_name = "Playback", .channels_min = 2, .channels_max = 8, - .rates = SNDRV_PCM_RATE_48000, + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000, .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, }, diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index 377854712c20..61f4648861d5 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -37,6 +37,10 @@ int ad193x_probe(struct device *dev, struct regmap *regmap, #define AD193X_PLL_CLK_SRC_MCLK (1 << 1) #define AD193X_DAC_CTRL0 0x02 #define AD193X_DAC_POWERDOWN 0x01 +#define AD193X_DAC_SR_MASK 0x06 +#define AD193X_DAC_SR_48 (0 << 1) +#define AD193X_DAC_SR_96 (1 << 1) +#define AD193X_DAC_SR_192 (2 << 1) #define AD193X_DAC_SERFMT_MASK 0xC0 #define AD193X_DAC_SERFMT_STEREO (0 << 6) #define AD193X_DAC_SERFMT_TDM (1 << 6) -- 2.30.2

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[PATCH 1/2] ASoC: cs42l42: Fix inversion of ADC Notch Switch control
by Richard Fitzgerald 04 Aug '21

04 Aug '21

The underlying register field has inverted sense (0 = enabled) so the control definition must be marked as inverted. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Fixes: 2c394ca79604 ("ASoC: Add support for CS42L42 codec") --- sound/soc/codecs/cs42l42.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 08ca05bfbeb3..2db87e3a4dc3 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -437,7 +437,7 @@ static SOC_ENUM_SINGLE_DECL(cs42l42_wnf05_freq_enum, CS42L42_ADC_WNF_HPF_CTL, static const struct snd_kcontrol_new cs42l42_snd_controls[] = { /* ADC Volume and Filter Controls */ SOC_SINGLE("ADC Notch Switch", CS42L42_ADC_CTL, - CS42L42_ADC_NOTCH_DIS_SHIFT, true, false), + CS42L42_ADC_NOTCH_DIS_SHIFT, true, true), SOC_SINGLE("ADC Weak Force Switch", CS42L42_ADC_CTL, CS42L42_ADC_FORCE_WEAK_VCM_SHIFT, true, false), SOC_SINGLE("ADC Invert Switch", CS42L42_ADC_CTL, -- 2.11.0

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[PATCH] ASoC: SOF: Intel: hda-ipc: fix reply size checking
by Pierre-Louis Bossart 04 Aug '21

04 Aug '21

From: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Checking that two values don't have common bits makes no sense, strict equality is meant. Fixes: f3b433e4699f ("ASoC: SOF: Implement Probe IPC API") Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/hda-ipc.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c index c91aa951df22..acfeca42604c 100644 --- a/sound/soc/sof/intel/hda-ipc.c +++ b/sound/soc/sof/intel/hda-ipc.c @@ -107,8 +107,8 @@ void hda_dsp_ipc_get_reply(struct snd_sof_dev *sdev) } else { /* reply correct size ? */ if (reply.hdr.size != msg->reply_size && - /* getter payload is never known upfront */ - !(reply.hdr.cmd & SOF_IPC_GLB_PROBE)) { + /* getter payload is never known upfront */ + ((reply.hdr.cmd & SOF_GLB_TYPE_MASK) != SOF_IPC_GLB_PROBE)) { dev_err(sdev->dev, "error: reply expected %zu got %u bytes\n", msg->reply_size, reply.hdr.size); ret = -EINVAL; -- 2.25.1

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[PATCH] ASoC: SOF: Intel: Kconfig: fix SoundWire dependencies
by Pierre-Louis Bossart 04 Aug '21

04 Aug '21

The previous Kconfig cleanup added simplifications but also introduced a new one by moving a boolean to a tristate. This leads to randconfig problems. This patch moves the select operations in the SOUNDWIRE_LINK_BASELINE option. The INTEL_SOUNDWIRE config remains a tristate for backwards compatibility with older configurations but is essentially an on/off switch. Fixes: cf5807f5f814f ('ASoC: SOF: Intel: SoundWire: simplify Kconfig') Reported-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> --- sound/soc/sof/intel/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 4bce89b5ea40..4447f515e8b1 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -278,6 +278,8 @@ config SND_SOC_SOF_HDA config SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE tristate + select SOUNDWIRE_INTEL if SND_SOC_SOF_INTEL_SOUNDWIRE + select SND_INTEL_SOUNDWIRE_ACPI if SND_SOC_SOF_INTEL_SOUNDWIRE config SND_SOC_SOF_INTEL_SOUNDWIRE tristate "SOF support for SoundWire" @@ -285,8 +287,6 @@ config SND_SOC_SOF_INTEL_SOUNDWIRE depends on SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE depends on ACPI && SOUNDWIRE depends on !(SOUNDWIRE=m && SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE=y) - select SOUNDWIRE_INTEL - select SND_INTEL_SOUNDWIRE_ACPI help This adds support for SoundWire with Sound Open Firmware for Intel(R) platforms. -- 2.25.1

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