May 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH AUTOSEL 6.3 15/37] ASoC: lpass: Fix for KASAN use_after_free out of bounds
by Sasha Levin 31 May '23

31 May '23

From: Ravulapati Vishnu Vardhan Rao <quic_visr(a)quicinc.com> [ Upstream commit 75e5fab7db0cecb6e16b22c34608f0b40a4c7cd1 ] When we run syzkaller we get below Out of Bounds error. "KASAN: slab-out-of-bounds Read in regcache_flat_read" Below is the backtrace of the issue: BUG: KASAN: slab-out-of-bounds in regcache_flat_read+0x10c/0x110 Read of size 4 at addr ffffff8088fbf714 by task syz-executor.4/14144 CPU: 6 PID: 14144 Comm: syz-executor.4 Tainted: G W Hardware name: Qualcomm Technologies, Inc. sc7280 CRD platform (rev5+) (DT) Call trace: dump_backtrace+0x0/0x4ec show_stack+0x34/0x50 dump_stack_lvl+0xdc/0x11c print_address_description+0x30/0x2d8 kasan_report+0x178/0x1e4 __asan_report_load4_noabort+0x44/0x50 regcache_flat_read+0x10c/0x110 regcache_read+0xf8/0x5a0 _regmap_read+0x45c/0x86c _regmap_update_bits+0x128/0x290 regmap_update_bits_base+0xc0/0x15c snd_soc_component_update_bits+0xa8/0x22c snd_soc_component_write_field+0x68/0xd4 tx_macro_put_dec_enum+0x1d0/0x268 snd_ctl_elem_write+0x288/0x474 By Error checking and checking valid values issue gets rectifies. Signed-off-by: Ravulapati Vishnu Vardhan Rao <quic_visr(a)quicinc.com Link: https://lore.kernel.org/r/20230511112532.16106-1-quic_visr@quicinc.com Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/lpass-tx-macro.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index 589c490a8c487..88d8fa58ec1eb 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -746,6 +746,8 @@ static int tx_macro_put_dec_enum(struct snd_kcontrol *kcontrol, struct tx_macro *tx = snd_soc_component_get_drvdata(component); val = ucontrol->value.enumerated.item[0]; + if (val >= e->items) + return -EINVAL; switch (e->reg) { case CDC_TX_INP_MUX_ADC_MUX0_CFG0: @@ -772,6 +774,9 @@ static int tx_macro_put_dec_enum(struct snd_kcontrol *kcontrol, case CDC_TX_INP_MUX_ADC_MUX7_CFG0: mic_sel_reg = CDC_TX7_TX_PATH_CFG0; break; + default: + dev_err(component->dev, "Error in configuration!!\n"); + return -EINVAL; } if (val != 0) { -- 2.39.2

1 0

[PATCH] ASoC: nau8825: Add the management of headset detection for power saving
by David Lin 31 May '23

31 May '23

The patch is to manage HSD feature for power saving. The detail is to disable HSD feature after the headset detection is done. When the jack is inserted, the HSD feature will be enabled again. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index cc3e18207c42..f6dd84b32e0b 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1704,6 +1704,10 @@ static void nau8825_setup_auto_irq(struct nau8825 *nau8825) { struct regmap *regmap = nau8825->regmap; + /* Enable HSD function */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, + NAU8825_HSD_AUTO_MODE, NAU8825_HSD_AUTO_MODE); + /* Enable headset jack type detection complete interruption and * jack ejection interruption. */ @@ -1955,6 +1959,9 @@ static int nau8825_jack_insert(struct nau8825 *nau8825) regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_MCLK_SRC_MASK, 0xf); + /* Disable HSD function */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, NAU8825_HSD_AUTO_MODE, 0); + /* Leaving HPOL/R grounded after jack insert by default. They will be * ungrounded as part of the widget power up sequence at the beginning * of playback to reduce pop. -- 2.25.1

2 1

[PATCH] dirvers/gpio: remove redundant reset gpio suffix
by Lizhe 31 May '23

31 May '23

no need to add the suffix "gpio" or "gpios" after "reset", as they will be automatically added, see of_find_gpio(). Signed-off-by: Lizhe <sensor1010(a)163.com> --- sound/soc/codecs/cs35l34.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l34.c b/sound/soc/codecs/cs35l34.c index b3f98023e6a7..cbf0f2f4f9df 100644 --- a/sound/soc/codecs/cs35l34.c +++ b/sound/soc/codecs/cs35l34.c @@ -1061,7 +1061,7 @@ static int cs35l34_i2c_probe(struct i2c_client *i2c_client) dev_err(&i2c_client->dev, "Failed to request IRQ: %d\n", ret); cs35l34->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, - "reset-gpios", GPIOD_OUT_LOW); + "reset", GPIOD_OUT_LOW); if (IS_ERR(cs35l34->reset_gpio)) { ret = PTR_ERR(cs35l34->reset_gpio); goto err_regulator; -- 2.34.1

1 0

[PATCH v2 0/6] Add cs42l43 PC focused SoundWire CODEC
by Charles Keepax 31 May '23

31 May '23

This patch chain adds support for the Cirrus Logic cs42l43 PC focused SoundWire CODEC. The chain is currently based of Lee's for-mfd-next branch. Change notes are included with each patch, most of the changes are trivial, the notable ones are moving the IRQs out of irqchip and into the MFD, and moving the DT binding to sound. Thanks, Charles Charles Keepax (4): dt-bindings: mfd: cirrus,cs42l43: Add initial DT binding mfd: cs42l43: Add support for cs42l43 core driver pinctrl: cs42l43: Add support for the cs42l43 ASoC: cs42l43: Add support for the cs42l43 Lucas Tanure (2): soundwire: bus: Allow SoundWire peripherals to register IRQ handlers spi: cs42l43: Add SPI controller support .../bindings/sound/cirrus,cs42l43.yaml | 320 +++ MAINTAINERS | 5 + drivers/mfd/Kconfig | 23 + drivers/mfd/Makefile | 3 + drivers/mfd/cs42l43-i2c.c | 86 + drivers/mfd/cs42l43-sdw.c | 213 ++ drivers/mfd/cs42l43.c | 1141 +++++++++ drivers/mfd/cs42l43.h | 23 + drivers/pinctrl/cirrus/Kconfig | 11 + drivers/pinctrl/cirrus/Makefile | 2 + drivers/pinctrl/cirrus/pinctrl-cs42l43.c | 609 +++++ drivers/soundwire/bus.c | 31 + drivers/soundwire/bus_type.c | 12 + drivers/spi/Kconfig | 7 + drivers/spi/Makefile | 1 + drivers/spi/spi-cs42l43.c | 279 ++ include/linux/mfd/cs42l43-regs.h | 1172 +++++++++ include/linux/mfd/cs42l43.h | 102 + include/linux/soundwire/sdw.h | 9 + include/sound/cs42l43.h | 17 + sound/soc/codecs/Kconfig | 16 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs42l43-jack.c | 951 +++++++ sound/soc/codecs/cs42l43-sdw.c | 75 + sound/soc/codecs/cs42l43.c | 2275 +++++++++++++++++ sound/soc/codecs/cs42l43.h | 126 + 26 files changed, 7513 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs42l43.yaml create mode 100644 drivers/mfd/cs42l43-i2c.c create mode 100644 drivers/mfd/cs42l43-sdw.c create mode 100644 drivers/mfd/cs42l43.c create mode 100644 drivers/mfd/cs42l43.h create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs42l43.c create mode 100644 drivers/spi/spi-cs42l43.c create mode 100644 include/linux/mfd/cs42l43-regs.h create mode 100644 include/linux/mfd/cs42l43.h create mode 100644 include/sound/cs42l43.h create mode 100644 sound/soc/codecs/cs42l43-jack.c create mode 100644 sound/soc/codecs/cs42l43-sdw.c create mode 100644 sound/soc/codecs/cs42l43.c create mode 100644 sound/soc/codecs/cs42l43.h -- 2.30.2

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[PATCH v5 0/3] Add TDM audio on StarFive JH7110
by Walker Chen 31 May '23

31 May '23

This patchset adds TDM audio driver for the StarFive JH7110 SoC. The first patch adds device tree binding for TDM module. The second patch adds tdm driver support for JH7110 SoC. The last patch adds device tree node and pins configuration of tdm to JH7110 dts. The series has been tested on the VisionFive 2 board by plugging an audio expansion board. For more information of audio expansion board, you can take a look at the following webpage: https://wiki.seeedstudio.com/ReSpeaker_2_Mics_Pi_HAT/ The last patch should be applied after the following patch: https://lore.kernel.org/all/20230322094820.24738-5-walker.chen@starfivetech… Changes since v4: - Rebased on Linux 6.4-rc2. - Adjusted code indentation in jh7110_tdm_syncdiv(). - Removed '__maybe_unused' for jh7110_tdm_system_suspend() and jh7110_tdm_system_resume(). - Removed initialization for variable 'ret' in jh7110_tdm_hw_params(). - Return immedially when entering unsurpported case in jh7110_tdm_set_dai_fmt(). - Check the return value with IS_ERR_OR_NULL() for the call to devm_reset_control_array_get_exclusive(). - Dropped the jh7110_tdm_clk_enable() in probe and rely on runtime PM check if CONFIG_PM is enabled. - Used RUNTIME_PM_OPS() instead of SET_RUNTIME_PM_OPS() and SYSTEM_SLEEP_PM_OPS() instead of SET_SYSTEM_SLEEP_PM_OPS() in structure jh7110_tdm_pm_ops. Changes since v3: - Modified the subject of patch [1/3] to reflect which subsystem it is. - Use the set of functions with 'clk_bulk_' to get/enable/disable clocks. - Dropped the overlay from patch [3/3]. - Dropped the redundant macro 'CONFIG_PM' and 'CONFIG_PM_SLEEP' around suspend() and resume(). Changes since v2: - Use dt-overlay to describe sound card because need to plug the audio expansion board into the VisionFive2 board. - Modified the coding style for driver. - Moved assignment of stop_dma_first to startup function of dai_driver. - Dropped some useless macro definition. - Use loops to get/enable/disable clocks. Changes since v1: - Rebased on Linux 6.3-rc4. - Added the dts file dedicated to describe audio card. - Added the item for JH7110 audio board to the dt-binding of StarFive SoC-based boards. --- v4: https://lore.kernel.org/all/20230511091549.28003-1-walker.chen@starfivetech… v3: https://lore.kernel.org/all/20230506090116.9206-1-walker.chen@starfivetech.… v2: https://lore.kernel.org/all/20230420024118.22677-1-walker.chen@starfivetech… v1: https://lore.kernel.org/all/20230329153320.31390-1-walker.chen@starfivetech… Walker Chen (3): ASoC: dt-bindings: Add TDM controller bindings for StarFive JH7110 ASoC: starfive: Add JH7110 TDM driver riscv: dts: starfive: add the node and pins configuration for tdm .../bindings/sound/starfive,jh7110-tdm.yaml | 98 +++ MAINTAINERS | 6 + .../jh7110-starfive-visionfive-2.dtsi | 40 ++ arch/riscv/boot/dts/starfive/jh7110.dtsi | 21 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/starfive/Kconfig | 15 + sound/soc/starfive/Makefile | 2 + sound/soc/starfive/jh7110_tdm.c | 679 ++++++++++++++++++ 9 files changed, 863 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/starfive,jh7110-tdm.yaml create mode 100644 sound/soc/starfive/Kconfig create mode 100644 sound/soc/starfive/Makefile create mode 100644 sound/soc/starfive/jh7110_tdm.c base-commit: f1fcbaa18b28dec10281551dfe6ed3a3ed80e3d6 prerequisite-patch-id: 398744c61913c76a35754de867c4f820ca7a8d99 prerequisite-patch-id: 07986c6e98ce85d3ee84f38c780e72145f098fc8 prerequisite-patch-id: 1babe83d6bf999bad17584dc595480f9070a5369 prerequisite-patch-id: 63c1f70ca2bcf827429542e60d74411b7666ceff prerequisite-patch-id: 9fbb7ad1dd258bb8ff5946c4a0e59de4bfd82a04 prerequisite-patch-id: a7f61973ccb6bc425daa8a86c048fd0f5a915b76 prerequisite-patch-id: 39e1be2a3d1593577ab997f55f59367cba665aa7 prerequisite-patch-id: 0159f09bb0a1ff711a00ae17ef5b12662c9c7d3d prerequisite-patch-id: 2ddada18ab6ea5cd1da14212aaf59632f5203d40 prerequisite-patch-id: 584c256c9acb52ee2773d0c81c3f4977fc18155a prerequisite-patch-id: b37ac15032973e1fcd918f157c82a0606775c9e9 prerequisite-patch-id: 6abf359fa445f4104432ddee27044dfbfb128417 -- 2.17.1

4 10

speaker-test: allow controlling how long the noise is heard by the user
by GitHub issues - opened 30 May '23

30 May '23

alsa-project/alsa-utils issue #219 was opened from jidanni: $ speaker-test --nloops 1 makes a pink noise for about one second. On the man page mention how to make this into say a half second, as some users would like to listen to shorter test sounds, and no matter if the user does ``` $ speaker-test --device hdmi:HDMI -c 2 --nloops 1 --period 5 $ speaker-test --device hdmi:HDMI -c 2 --nloops 1 --period 50 $ speaker-test --device hdmi:HDMI -c 2 --nloops 1 --period 500 $ speaker-test --device hdmi:HDMI -c 2 --nloops 1 --period 5000 ``` what the user hears is exactly the same, therefore --period is not what controls this item. I am saying that some users can confirm their speakers work without listing to a full second of pink noise, and thus would like a way to listen to a shorter length. As far as ``` -b | --buffer TIME Use buffer size of TIME microseconds. When 0 is given, use the maximal buffer size. The default value is 0. -p | --period TIME Use period size of TIME microseconds. When 0 is given, the pe‐ riods given by -P option is used. The default value is 0. ``` are concerned, they must be documenting something completely different. Debian Package: alsa-utils Version: 1.2.8-1 Issue URL : https://github.com/alsa-project/alsa-utils/issues/219 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH 0/3] ASoC: topology: allow for partial dailink stream_name match
by Pierre-Louis Bossart 30 May '23

30 May '23

The topology file and the machine driver rely on common definitions for the dailink stream_name. To avoid any backwards-compatibility problems, the machine driver stream names are set in stone and cannot be modified. This is problematic when we try to name some of the topology widgets after the stream_name, since the widget name is limited to 44 characters tools/include/uapi/sound/asound.h:#define SNDRV_CTL_ELEM_ID_NAME_MAXLEN 44 Existing examples include "Analog Playback and Capture" for HDaudio dailinks, which leaves less than 20 chars to identify widgets/controls with a meaningful name. Since the 44-char limit is part of the UAPI definitions, we assumed there is no way to increase it. This patchset suggests instead a partial match which allows topology files to use a shorter stream_name, which in turn allows for self-explanatory widget names that comply with the 44-char limit. This should not break any existing setup but with the introduction of a partial match new dailinks should be named carefully to avoid confusions between e.g. 'link1' and 'link10'. The last patch fixes such an issue in the 'nocodec' test topology used by Intel. Ranjani Sridharan (3): ASoC: topology: Allow partial matching when finding DAI link ASoC: SOF: topology: Use partial match for connecting DAI link and DAI widget ASoC: SOF: Intel: HDA: Limit the number of dai drivers for nocodec mode sound/soc/soc-topology.c | 6 +++--- sound/soc/sof/intel/hda.c | 6 +++++- sound/soc/sof/intel/hda.h | 4 +++- sound/soc/sof/topology.c | 2 +- 4 files changed, 12 insertions(+), 6 deletions(-) -- 2.37.2

2 4

[PATCH v2] SoC: ti: davinci-mcasp: Use pcm_for_each_format() macro
by Min-Hua Chen 30 May '23

30 May '23

Use pcm_for_each_format for the PCM format iteration and fix the following sparse warnings. sound/soc/ti/davinci-mcasp.c:1336:26: sparse: warning: restricted snd_pcm_format_t degrades to integer sound/soc/ti/davinci-mcasp.c:1358:26: sparse: warning: restricted snd_pcm_format_t degrades to integer sound/soc/ti/davinci-mcasp.c:1438:26: sparse: warning: restricted snd_pcm_format_t degrades to integer No functional changes. Signed-off-by: Min-Hua Chen <minhuadotchen(a)gmail.com> --- Change since v1: use clear subject format --- sound/soc/ti/davinci-mcasp.c | 27 +++++++++++++++------------ 1 file changed, 15 insertions(+), 12 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index c0892be2992b..172fea764a31 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -1328,15 +1328,16 @@ static int davinci_mcasp_hw_rule_slot_width(struct snd_pcm_hw_params *params, struct davinci_mcasp_ruledata *rd = rule->private; struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); struct snd_mask nfmt; - int i, slot_width; + int slot_width; + snd_pcm_format_t i; snd_mask_none(&nfmt); slot_width = rd->mcasp->slot_width; - for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { - if (snd_mask_test(fmt, i)) { + pcm_for_each_format(i) { + if (snd_mask_test_format(fmt, i)) { if (snd_pcm_format_width(i) <= slot_width) { - snd_mask_set(&nfmt, i); + snd_mask_set_format(&nfmt, i); } } } @@ -1350,15 +1351,16 @@ static int davinci_mcasp_hw_rule_format_width(struct snd_pcm_hw_params *params, struct davinci_mcasp_ruledata *rd = rule->private; struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); struct snd_mask nfmt; - int i, format_width; + int format_width; + snd_pcm_format_t i; snd_mask_none(&nfmt); format_width = rd->mcasp->max_format_width; - for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { - if (snd_mask_test(fmt, i)) { + pcm_for_each_format(i) { + if (snd_mask_test_format(fmt, i)) { if (snd_pcm_format_width(i) == format_width) { - snd_mask_set(&nfmt, i); + snd_mask_set_format(&nfmt, i); } } } @@ -1431,12 +1433,13 @@ static int davinci_mcasp_hw_rule_format(struct snd_pcm_hw_params *params, struct snd_mask nfmt; int rate = params_rate(params); int slots = rd->mcasp->tdm_slots; - int i, count = 0; + int count = 0; + snd_pcm_format_t i; snd_mask_none(&nfmt); - for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { - if (snd_mask_test(fmt, i)) { + pcm_for_each_format(i) { + if (snd_mask_test_format(fmt, i)) { uint sbits = snd_pcm_format_width(i); unsigned int sysclk_freq; int ppm; @@ -1454,7 +1457,7 @@ static int davinci_mcasp_hw_rule_format(struct snd_pcm_hw_params *params, sbits * slots * rate, false); if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) { - snd_mask_set(&nfmt, i); + snd_mask_set_format(&nfmt, i); count++; } } -- 2.34.1

2 1

[PATCH] ASoC: amd: vangogh: Add check for acp config flags in vangogh platform
by Venkata Prasad Potturu 30 May '23

30 May '23

We have SOF and generic ACP support enabled for Vangogh platform on some machines. Since we have same PCI id used for probing, add check for machine configuration flag to avoid conflict with newer pci drivers. Such machine flag has been initialized via dmi match on few Vangogh based machines. If no flag is specified probe and register older platform device. Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/amd/vangogh/acp5x.h | 2 ++ sound/soc/amd/vangogh/pci-acp5x.c | 7 ++++++- 2 files changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/vangogh/acp5x.h b/sound/soc/amd/vangogh/acp5x.h index bd9f1c5684d1..ac1936a8c43f 100644 --- a/sound/soc/amd/vangogh/acp5x.h +++ b/sound/soc/amd/vangogh/acp5x.h @@ -147,6 +147,8 @@ static inline void acp_writel(u32 val, void __iomem *base_addr) writel(val, base_addr - ACP5x_PHY_BASE_ADDRESS); } +int snd_amd_acp_find_config(struct pci_dev *pci); + static inline u64 acp_get_byte_count(struct i2s_stream_instance *rtd, int direction) { diff --git a/sound/soc/amd/vangogh/pci-acp5x.c b/sound/soc/amd/vangogh/pci-acp5x.c index e0df17c88e8e..c4634a8a17cd 100644 --- a/sound/soc/amd/vangogh/pci-acp5x.c +++ b/sound/soc/amd/vangogh/pci-acp5x.c @@ -125,10 +125,15 @@ static int snd_acp5x_probe(struct pci_dev *pci, { struct acp5x_dev_data *adata; struct platform_device_info pdevinfo[ACP5x_DEVS]; - unsigned int irqflags; + unsigned int irqflags, flag; int ret, i; u32 addr, val; + /* Return if acp config flag is defined */ + flag = snd_amd_acp_find_config(pci); + if (flag) + return -ENODEV; + irqflags = IRQF_SHARED; if (pci->revision != 0x50) return -ENODEV; -- 2.25.1

2 1

[PATCH] ASoC: amd: ps: fix for acp_lock access in pdm driver
by Vijendar Mukunda 30 May '23

30 May '23

Sending the mutex address(acp_lock) as platform data during ACP PDM platform driver register sequence, its creating copy of the platform data. Referencing this platform data in ACP PDM driver results incorrect reference to the common lock usage. Instead of directly passing the lock address as platform data, retrieve it from parent driver data structure and use the same lock reference in ACP PDM driver. Fixes: 45aa83cb9388 ("ASoC: amd: ps: use acp_lock to protect common registers in pdm driver") Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/ps/pci-ps.c | 3 +-- sound/soc/amd/ps/ps-pdm-dma.c | 10 +++++----- 2 files changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index c957718abefc..54752d6040d6 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -198,8 +198,7 @@ static int create_acp63_platform_devs(struct pci_dev *pci, struct acp63_dev_data case ACP63_PDM_DEV_MASK: adata->pdm_dev_index = 0; acp63_fill_platform_dev_info(&pdevinfo[0], parent, NULL, "acp_ps_pdm_dma", - 0, adata->res, 1, &adata->acp_lock, - sizeof(adata->acp_lock)); + 0, adata->res, 1, NULL, 0); acp63_fill_platform_dev_info(&pdevinfo[1], parent, NULL, "dmic-codec", 0, NULL, 0, NULL, 0); acp63_fill_platform_dev_info(&pdevinfo[2], parent, NULL, "acp_ps_mach", diff --git a/sound/soc/amd/ps/ps-pdm-dma.c b/sound/soc/amd/ps/ps-pdm-dma.c index 3ecc6cf3fd34..bdbbb797c74d 100644 --- a/sound/soc/amd/ps/ps-pdm-dma.c +++ b/sound/soc/amd/ps/ps-pdm-dma.c @@ -358,12 +358,12 @@ static int acp63_pdm_audio_probe(struct platform_device *pdev) { struct resource *res; struct pdm_dev_data *adata; + struct acp63_dev_data *acp_data; + struct device *parent; int status; - if (!pdev->dev.platform_data) { - dev_err(&pdev->dev, "platform_data not retrieved\n"); - return -ENODEV; - } + parent = pdev->dev.parent; + acp_data = dev_get_drvdata(parent); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!res) { dev_err(&pdev->dev, "IORESOURCE_MEM FAILED\n"); @@ -379,7 +379,7 @@ static int acp63_pdm_audio_probe(struct platform_device *pdev) return -ENOMEM; adata->capture_stream = NULL; - adata->acp_lock = pdev->dev.platform_data; + adata->acp_lock = &acp_data->acp_lock; dev_set_drvdata(&pdev->dev, adata); status = devm_snd_soc_register_component(&pdev->dev, &acp63_pdm_component, -- 2.34.1

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