August 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH v5 1/2] ASoC: dt-bindings: Add Everest ES8326 audio CODEC
by Zhu Ning 16 Aug '22

16 Aug '22

Add device tree binding documentation for Everest ES8326 Signed-off-by: David Yang <yangxiaohua(a)everest-semi.com> Signed-off-by: Zhu Ning <zhuning(a)everest-semi.com> ---- v5 tested by dtschema --- .../bindings/sound/everest,es8326.yaml | 116 ++++++++++++++++++ 1 file changed, 116 insertions(+) create mode 100755 Documentation/devicetree/bindings/sound/everest,es8326.yaml diff --git a/Documentation/devicetree/bindings/sound/everest,es8326.yaml b/Documentation/devicetree/bindings/sound/everest,es8326.yaml new file mode 100755 index 000000000000..07781408e788 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es8326.yaml @@ -0,0 +1,116 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/everest,es8326.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Everest ES8326 audio CODEC + +maintainers: + - David Yang <yangxiaohua(a)everest-semi.com> + +properties: + compatible: + const: everest,es8326 + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for master clock (MCLK) + + clock-names: + items: + - const: mclk + + "#sound-dai-cells": + const: 0 + + everest,jack-pol: + $ref: /schemas/types.yaml#/definitions/uint8 + description: | + just the value of reg 57. Bit(3) decides whether the jack polarity is inverted. + Bit(2) decides whether the button on the headset is inverted. + Bit(1)/(0) decides the mic properity to be OMTP/CTIA or auto. + minimum: 0x00 + maximum: 0x0f + default: 0x0f + + everest,mic1-src: + $ref: /schemas/types.yaml#/definitions/uint8 + description: + the value of reg 2A when headset plugged. + minimum: 0x00 + maximum: 0x77 + default: 0x22 + + everest,mic2-src: + $ref: /schemas/types.yaml#/definitions/uint8 + description: + the value of reg 2A when headset unplugged. + minimum: 0x00 + maximum: 0x77 + default: 0x44 + + everest,jack-detect-inverted: + $ref: /schemas/types.yaml#/definitions/flag + description: + Defined to invert the jack detection. + + everest,interrupt-src: + $ref: /schemas/types.yaml#/definitions/uint8 + description: | + value of reg 0x58, Defines the interrupt source. + Bit(2) 1 means button press triggers irq, 0 means not. + Bit(3) 1 means PIN9 is the irq source for jack detection. When set to 0, + bias change on PIN9 do not triggers irq. + Bit(4) 1 means PIN27 is the irq source for jack detection. + Bit(5) 1 means PIN9 is the irq source after MIC detect. + Bit(6) 1 means PIN27 is the irq source after MIC detect. + minimum: 0 + maximum: 0x3c + default: 0x08 + + everest,interrupt-clk: + $ref: /schemas/types.yaml#/definitions/uint8 + description: | + value of reg 0x59, Defines the interrupt output behavior. + Bit(0-3) 0 means irq pulse equals 512*internal clock + 1 means irq pulse equals 1024*internal clock + 2 means ... + 7 means irq pulse equals 65536*internal clock + 8 means irq mutes PA + 9 means irq mutes PA and DAC output + Bit(4) 1 means we invert the interrupt output. + Bit(6) 1 means the chip do not detect jack type after button released. + 0 means the chip detect jack type again after button released. + minimum: 0 + maximum: 0x7f + default: 0x45 + +required: + - compatible + - reg + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + es8326: codec@19 { + compatible = "everest,es8326"; + reg = <0x19>; + clocks = <&clks 10>; + clock-names = "mclk"; + #sound-dai-cells = <0>; + everest,mic1-src = [22]; + everest,mic2-src = [44]; + everest,jack-pol = [0e]; + everest,interrupt-src = [08]; + everest,interrupt-clk = [45]; + }; + }; -- 2.36.1

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[PATCH v2] ASoC: fsl_sai: fix incorrect mclk number in error message
by Pieterjan Camerlynck 16 Aug '22

16 Aug '22

In commit c3ecef21c3f26 ("ASoC: fsl_sai: add sai master mode support") the loop was changed to start iterating from 1 instead of 0. The error message however was not updated, reporting the wrong clock to the user. Signed-off-by: Pieterjan Camerlynck <pieterjan.camerlynck(a)gmail.com> --- V2: rebase against latest version --- sound/soc/fsl/fsl_sai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 7523bb944b21..d430eece1d6b 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1306,7 +1306,7 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->mclk_clk[i] = devm_clk_get(dev, tmp); if (IS_ERR(sai->mclk_clk[i])) { dev_err(dev, "failed to get mclk%d clock: %ld\n", - i + 1, PTR_ERR(sai->mclk_clk[i])); + i, PTR_ERR(sai->mclk_clk[i])); sai->mclk_clk[i] = NULL; } } -- 2.25.1

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[PATCH] ALSA: hda/realtek: Add quirk for Lenovo Yoga7 14IAL7
by Takashi Iwai 16 Aug '22

16 Aug '22

Lenovo Yoga7 14IAL7 requires the same quirk as Lenovo Yoga9 14IAP7 for fixing the bass speaker problems. Reported-by: Pascal Gross <baipush(a)tutanota.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/N9_CjBz--3-2@tutanota.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- 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 f544761eb11b..b42496c01c43 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9492,6 +9492,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3852, "Lenovo Yoga 7 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3855, "Legion 7 16ITHG6", ALC287_FIXUP_LEGION_16ITHG6), + SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), -- 2.35.3

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[GIT PULL] ASoC fixes for v6.0-rc1
by Mark Brown 16 Aug '22

16 Aug '22

The following changes since commit 40d060b8158e26438398bf1132925f666e3b6480: ASoC: q6asm: use kcalloc() instead of kzalloc() (2022-07-28 11:59:10 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.0-rc1 for you to fetch changes up to b4b5f29a076e52181f63e45a2ad1bc88593072e3: ASoC: codec: tlv320aic32x4: fix mono playback via I2S (2022-08-10 14:52:05 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.0 A relatively large batch of fixes that came in since my pull request, none of them too major and mostly device specific apart from a series of security/robustness improvements from Takashi. ---------------------------------------------------------------- Biju Das (1): ASoC: sh: rz-ssi: Improve error handling in rz_ssi_probe() error path Justin Stitt (1): ASoC: SOF: ipc3-topology: Fix clang -Wformat warning Mark Brown (2): ASoC: Fix theoretical buffer overflow by snprintf() TAS2770 fixes Martin Povišer (4): ASoC: tas2770: Set correct FSYNC polarity ASoC: tas2770: Allow mono streams ASoC: tas2770: Drop conflicting set_bias_level power setting ASoC: tas2770: Fix handling of mute/unmute Oder Chiou (1): ASoC: rt5640: Fix the JD voltage dropping issue Philipp Zabel (1): ASoC: codec: tlv320aic32x4: fix mono playback via I2S Pierre-Louis Bossart (1): ASoC: Intel: fix sof_es8336 probe Takashi Iwai (4): ASoC: Intel: avs: Fix potential buffer overflow by snprintf() ASoC: SOF: debug: Fix potential buffer overflow by snprintf() ASoC: SOF: Intel: hda: Fix potential buffer overflow by snprintf() ASoC: DPCM: Don't pick up BE without substream syed sabakareem (1): ASoC: amd: yc: Update DMI table entries for AMD platforms sound/soc/amd/yc/acp6x-mach.c | 28 +++++++++++ sound/soc/codecs/rt5640.c | 5 +- sound/soc/codecs/tas2770.c | 98 ++++++++++++++++--------------------- sound/soc/codecs/tas2770.h | 5 ++ sound/soc/codecs/tlv320aic32x4.c | 9 ++++ sound/soc/intel/avs/pcm.c | 4 +- sound/soc/intel/boards/sof_es8336.c | 4 +- sound/soc/sh/rz-ssi.c | 26 +++++----- sound/soc/soc-pcm.c | 3 ++ sound/soc/sof/debug.c | 6 +-- sound/soc/sof/intel/hda.c | 2 +- sound/soc/sof/ipc3-topology.c | 2 +- 12 files changed, 116 insertions(+), 76 deletions(-)

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[PATCH v2] ALSA: hda: cs35l41: Clarify support for CSC3551 without _DSD Properties
by Stefan Binding 16 Aug '22

16 Aug '22

For devices which use HID CSC3551, correct ACPI _DSD properties are required to be able support those systems. Add error message to clarify this. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/pci/hda/cs35l41_hda.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 129bffb431c2..15e2a0009080 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -1163,6 +1163,11 @@ static int cs35l41_no_acpi_dsd(struct cs35l41_hda *cs35l41, struct device *physd hw_cfg->gpio1.func = CS35l41_VSPK_SWITCH; hw_cfg->gpio1.valid = true; } else { + /* + * Note: CLSA010(0/1) are special cases which use a slightly different design. + * All other HIDs e.g. CSC3551 require valid ACPI _DSD properties to be supported. + */ + dev_err(cs35l41->dev, "Error: ACPI _DSD Properties are missing for HID %s.\n", hid); hw_cfg->valid = false; hw_cfg->gpio1.valid = false; hw_cfg->gpio2.valid = false; -- 2.34.1

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[PATCH v2 0/6] ALSA: hda: Unify codec construction
by Cezary Rojewski 16 Aug '22

16 Aug '22

A follow up to the recent HDAudio fixes series [1]. Given the recently reported regression [2], before the page fault occurring on codec shutdown can be fixed, codec construction procedure needs to be updated for skylake and sof-intel drivers. Drivers: pci-hda and avs need no changes - already making use of snd_hda_codec_device_init(). As subject touches code used by the sof-driver, additional review has been conducted on thesofproject/linux [3]. Changes in v2: - dropped snd_hda_ext_core <-> snd_hda_codec dependency by calling snd_hda_codec_device_init() directly in skylake and sof drivers probe enumeration routines, as suggested by Takashi - skylake/sof portion of the change has been split into two separate patches - new functions that aim to replace hdac_ext codec init & exit functionality are added first - for skylake and sof drivers both - third patch in the series now combines the "field -> pointer" change for hdac_hda_priv->codec plus the codec-enumeration adjustments for skylake and sof drivers Both above are here to keep git bisect happy, as suggested by Pierre [1]: https://lore.kernel.org/alsa-devel/20220706120230.427296-7-cezary.rojewski@… [2]: https://lore.kernel.org/alsa-devel/3c40df55-3aee-1e08-493b-7b30cd84dc00@lin… [3]: https://github.com/thesofproject/linux/pull/3775 Cezary Rojewski (6): ASoC: Intel: Skylake: Introduce HDA codec init and exit routines ASoC: SOF: Intel: Introduce HDA codec init and exit routines ASoC: Intel: Drop hdac_ext usage for codec device creation ALSA: hda: Always free codec on the device release ALSA: hda: Remove codec init and exit routines ALSA: hda: Fix page fault in snd_hda_codec_shutdown() include/sound/hda_codec.h | 2 - include/sound/hdaudio_ext.h | 3 -- sound/hda/ext/hdac_ext_bus.c | 53 ------------------- sound/pci/hda/hda_codec.c | 49 ++++++++--------- sound/soc/codecs/hdac_hda.c | 26 ++++----- sound/soc/codecs/hdac_hda.h | 2 +- sound/soc/intel/boards/hda_dsp_common.c | 2 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- sound/soc/intel/skylake/skl.c | 53 ++++++++++++++----- sound/soc/sof/intel/hda-codec.c | 55 ++++++++++++++------ 10 files changed, 113 insertions(+), 134 deletions(-) -- 2.25.1

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[PATCH] ASoC: Fix the include guard used for include/sound/wm8904.h
by Christophe JAILLET 16 Aug '22

16 Aug '22

__MFD_WM8994_PDATA_H__ is already used for: include/linux/mfd/wm8994/pdata.h Based on file names, use __MFD_WM8904_PDATA_H__ instead here. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- Not sure of if a Fixes: tag is needed. If yes, I would use: 9e50108668a7 ("mfd: Add initial WM8994 support") which introduced the conflict, even if: a91eb199e4dc8 ("ASoC: Initial WM8904 CODEC driver") which introduced the typo is more logical. No strong opinion on it. --- include/sound/wm8904.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/wm8904.h b/include/sound/wm8904.h index 88ac1870510e..f74293a83a17 100644 --- a/include/sound/wm8904.h +++ b/include/sound/wm8904.h @@ -7,8 +7,8 @@ * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.com> */ -#ifndef __MFD_WM8994_PDATA_H__ -#define __MFD_WM8994_PDATA_H__ +#ifndef __MFD_WM8904_PDATA_H__ +#define __MFD_WM8904_PDATA_H__ /* Used to enable configuration of a GPIO to all zeros */ #define WM8904_GPIO_NO_CONFIG 0x8000 -- 2.34.1

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[PATCH] ASoC: intel: Replace scnprintf() calls with sysfs_emit_at()
by zhaoxiao 16 Aug '22

16 Aug '22

For sysfs outputs, it's safer to use a new helper, sysfs_emit(), instead of the raw sprintf() & co. This patch replaces the open-code with a new helper, sysfs_emit_at(), by passing the string offset. Signed-off-by: zhaoxiao <zhaoxiao(a)uniontech.com> --- sound/soc/sof/intel/hda.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 6d4ecbe14adf..917d639671a7 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -574,7 +574,7 @@ static void hda_dsp_dump_ext_rom_status(struct snd_sof_dev *sdev, const char *le chip = get_chip_info(sdev->pdata); for (i = 0; i < HDA_EXT_ROM_STATUS_SIZE; i++) { value = snd_sof_dsp_read(sdev, HDA_DSP_BAR, chip->rom_status_reg + i * 0x4); - len += scnprintf(msg + len, sizeof(msg) - len, " 0x%x", value); + len += sysfs_emit_at(msg, len, " 0x%x", value); } dev_printk(level, sdev->dev, "extended rom status: %s", msg); -- 2.20.1

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[PATCH] ASoC: Intel: sof_rt5682: Add support for jsl_rt5682_rt1019
by Sean Hong 16 Aug '22

16 Aug '22

This patch adds the driver data for rt5682 support jsl_rt5682_rt1019. Signed-off-by: Sean Hong <sean.hong(a)quanta.corp-partner.google.com> --- sound/soc/intel/boards/sof_rt5682.c | 9 +++++++++ sound/soc/intel/common/soc-acpi-intel-jsl-match.c | 12 ++++++++++++ 2 files changed, 21 insertions(+) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 045965312..3a840f3a9 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -1100,6 +1100,15 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(1) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .name = "jsl_rt5682_rt1019", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT1019_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1)), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c index b95c4b2cd..e374bf163 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -29,6 +29,11 @@ static const struct snd_soc_acpi_codecs rt1015p_spk = { .codecs = {"RTL1015"} }; +static struct snd_soc_acpi_codecs rt1019p_spk = { + .num_codecs = 1, + .codecs = {"RTL1019"} +}; + static const struct snd_soc_acpi_codecs mx98360a_spk = { .num_codecs = 1, .codecs = {"MX98360A"} @@ -78,6 +83,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { .quirk_data = &mx98360a_spk, .sof_tplg_filename = "sof-jsl-rt5682-mx98360a.tplg", }, + { + .comp_ids = &rt5682_rt5682s_hp, + .drv_name = "jsl_rt5682_rt1019", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &rt1019p_spk, + .sof_tplg_filename = "sof-jsl-rt5682-rt1019.tplg", + }, { .id = "10134242", .drv_name = "jsl_cs4242_mx98360a", -- 2.25.1

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Re: [PATCH 07/13] ASoC: amd: add acp6.2 pdm driver dma ops
by Syed Saba Kareem 16 Aug '22

16 Aug '22

On 8/12/22 19:50, Amadeusz Sławiński wrote: > [CAUTION: External Email] > > On 8/12/2022 2:07 PM, Syed Saba kareem wrote: >> This patch adds PDM driver DMA operations for Pink Sardine Platform. >> >> Signed-off-by: Syed Saba Kareem <Syed.SabaKareem(a)amd.com> >> --- >> sound/soc/amd/ps/acp62.h | 41 +++++ >> sound/soc/amd/ps/ps-pdm-dma.c | 310 ++++++++++++++++++++++++++++++++++ >> 2 files changed, 351 insertions(+) >> >> diff --git a/sound/soc/amd/ps/acp62.h b/sound/soc/amd/ps/acp62.h >> index 563252834b09..525e8bd225a2 100644 >> --- a/sound/soc/amd/ps/acp62.h >> +++ b/sound/soc/amd/ps/acp62.h >> @@ -27,6 +27,31 @@ >> #define ACP_EXT_INTR_STAT_CLEAR_MASK 0xFFFFFFFF >> #define PDM_DMA_STAT 0x10 >> >> +#define PDM_DMA_INTR_MASK 0x10000 >> +#define ACP_ERROR_STAT 29 >> +#define PDM_DECIMATION_FACTOR 2 >> +#define ACP_PDM_CLK_FREQ_MASK 7 >> +#define ACP_WOV_MISC_CTRL_MASK 0x10 >> +#define ACP_PDM_ENABLE 1 >> +#define ACP_PDM_DISABLE 0 >> +#define ACP_PDM_DMA_EN_STATUS 2 >> +#define TWO_CH 2 >> +#define DELAY_US 5 >> +#define ACP_COUNTER 20000 >> + >> +#define ACP_SRAM_PTE_OFFSET 0x03800000 >> +#define PAGE_SIZE_4K_ENABLE 2 >> +#define PDM_PTE_OFFSET 0 >> +#define PDM_MEM_WINDOW_START 0x4000000 >> + >> +#define CAPTURE_MIN_NUM_PERIODS 4 >> +#define CAPTURE_MAX_NUM_PERIODS 4 >> +#define CAPTURE_MAX_PERIOD_SIZE 8192 >> +#define CAPTURE_MIN_PERIOD_SIZE 4096 >> + >> +#define MAX_BUFFER (CAPTURE_MAX_PERIOD_SIZE * CAPTURE_MAX_NUM_PERIODS) >> +#define MIN_BUFFER MAX_BUFFER >> + >> enum acp_config { >> ACP_CONFIG_0 = 0, >> ACP_CONFIG_1, >> @@ -46,6 +71,22 @@ enum acp_config { >> ACP_CONFIG_15, >> }; >> >> +struct pdm_stream_instance { >> + u16 num_pages; >> + u16 channels; >> + dma_addr_t dma_addr; >> + u64 bytescount; >> + void __iomem *acp62_base; >> +}; >> + >> +union acp_pdm_dma_count { >> + struct { >> + u32 low; >> + u32 high; >> + } bcount; > > Fix indentation. > Also you can remove struct name, and access fields directly, so instead > of accessing somevariable.bcount.low, you would use somevariable.low, it > would probably be more readable. > Okay Will fix it. >> + u64 bytescount; >> +}; >> + >> struct pdm_dev_data { >> u32 pdm_irq; >> void __iomem *acp62_base; >> diff --git a/sound/soc/amd/ps/ps-pdm-dma.c >> b/sound/soc/amd/ps/ps-pdm-dma.c >> index bca2ac3e81f5..a98a2015015d 100644 > > ... > >> + >> +static int acp62_pdm_dma_open(struct snd_soc_component *component, >> + struct snd_pcm_substream *substream) >> +{ >> + struct snd_pcm_runtime *runtime; >> + struct pdm_dev_data *adata; >> + struct pdm_stream_instance *pdm_data; >> + int ret; >> + >> + runtime = substream->runtime; >> + adata = dev_get_drvdata(component->dev); >> + pdm_data = kzalloc(sizeof(*pdm_data), GFP_KERNEL); >> + if (!pdm_data) >> + return -EINVAL; >> + >> + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) >> + runtime->hw = acp62_pdm_hardware_capture; >> + >> + ret = snd_pcm_hw_constraint_integer(runtime, >> + SNDRV_PCM_HW_PARAM_PERIODS); >> + if (ret < 0) { >> + dev_err(component->dev, "set integer constraint >> failed\n"); >> + kfree(pdm_data); >> + return ret; >> + } >> + >> + acp62_enable_pdm_interrupts(adata->acp62_base); >> + >> + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) >> + adata->capture_stream = substream; >> + >> + pdm_data->acp62_base = adata->acp62_base; >> + runtime->private_data = pdm_data; > > Should probably kfree(runtime->private_data) in acp62_pdm_dma_close(), > otherwise won't there be a memory leak? > Will fix it. >> + return ret; >> +} >> + > > ... >

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