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

[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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[PATCH] ASoC: rt5514: make array div static const, makes object smaller
by Colin King 04 Aug '21

04 Aug '21

From: Colin Ian King <colin.king(a)canonical.com> Don't populate the array div on the stack but instead it static const. Makes the object code smaller by 48 bytes. Before: text data bss dec hex filename 53894 16368 128 70390 112f6 ./sound/soc/codecs/rt5514.o After: text data bss dec hex filename 53750 16464 128 70342 112c6 ./sound/soc/codecs/rt5514.o (gcc version 10.2.0) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt5514.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 7081142a355e..4b1ad5054e8d 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -494,7 +494,7 @@ static const struct snd_kcontrol_new rt5514_sto2_dmic_mux = */ static int rt5514_calc_dmic_clk(struct snd_soc_component *component, int rate) { - int div[] = {2, 3, 4, 8, 12, 16, 24, 32}; + static const int div[] = {2, 3, 4, 8, 12, 16, 24, 32}; int i; if (rate < 1000000 * div[0]) { -- 2.31.1

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[PATCH 0/4] ASoC: SOF/Intel: machine driver updates
by Pierre-Louis Bossart 04 Aug '21

04 Aug '21

Two quirks added for Dell XPS 9710 and Intel reference boards, and two minor cleanups. Pierre-Louis Bossart (3): ASoC: Intel: sof_sdw: add quirk for Dell XPS 9710 ASoC: Intel: update sof_pcm512x quirks ASoC: Intel: sof_sdw_max98373: remove useless inits jairaj arava (1): ASoC: SOF: Intel: Use DMI string to search for adl_mx98373_rt5682 variant sound/soc/intel/boards/sof_pcm512x.c | 13 +++++++++++-- sound/soc/intel/boards/sof_rt5682.c | 14 ++++++++++++++ sound/soc/intel/boards/sof_sdw.c | 12 ++++++++++++ sound/soc/intel/boards/sof_sdw_max98373.c | 4 ++-- sound/soc/sof/sof-pci-dev.c | 9 +++++++++ 5 files changed, 48 insertions(+), 4 deletions(-) -- 2.25.1

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