From 07c774dd64ba0c605dbf844132122e3edbdbea93 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 19 Feb 2026 04:53:52 +0000 Subject: ASoC: soc-compress: use function to clear symmetric params Current soc-compress.c clears symmetric_rate, but it clears rate only, not clear other symmetric_channels/sample_bits. static int soc_compr_clean(...) { ... if (!snd_soc_dai_active(cpu_dai)) => cpu_dai->symmetric_rate = 0; if (!snd_soc_dai_active(codec_dai)) => codec_dai->symmetric_rate = 0; ... }; This feature was added when v3.7 kernel [1], and there was only symmetric_rate, no symmetric_channels/sample_bits in that timing. symmetric_channels/sample_bits were added in v3.14 [2], but I guess it didn't notice that soc-compress.c is updating symmetric_xxx. We are clearing symmetry_xxx by soc_pcm_set_dai_params(), but is soc-pcm.c local function. Makes it global function and clear symmetry_xxx by it. [1] commit 1245b7005de02 ("ASoC: add compress stream support") [2] commit 3635bf09a89cf ("ASoC: soc-pcm: add symmetry for channels and sample bits") Fixes: 3635bf09a89c ("ASoC: soc-pcm: add symmetry for channels and sample bits") Cc: Nicolin Chen Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87ms15e3kv.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc.h | 3 +++ 1 file changed, 3 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 7d8376c8e1be..1e0b7cd8d956 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1412,6 +1412,9 @@ struct snd_soc_dai *snd_soc_find_dai( struct snd_soc_dai *snd_soc_find_dai_with_mutex( const struct snd_soc_dai_link_component *dlc); +void soc_pcm_set_dai_params(struct snd_soc_dai *dai, + struct snd_pcm_hw_params *params); + #include static inline -- cgit v1.2.3 From d075cef4af6327a5de4bee7bf77591e3201e54f4 Mon Sep 17 00:00:00 2001 From: Stefano Radaelli Date: Fri, 13 Feb 2026 16:03:55 +0100 Subject: ASoC: simple-card-utils: add sysclk ordering support When simple-audio-card programs sysclk for CPU and codec DAIs during hw_params, the ordering of these calls may matter on some platforms. Some CPU DAIs finalize or adjust the MCLK rate as part of their set_sysclk() callback (for example by calling clk_set_rate()). If the codec sysclk is configured before the CPU DAI applies the final MCLK rate, the codec may configure its internal clocking based on a non-final MCLK value. Such situations can arise depending on the clock provider/consumer relationship between the CPU DAI and the codec. Introduce an explicit sysclk ordering enum in simple-card-utils and use it to control the order of snd_soc_dai_set_sysclk() calls in the mclk-fs handling path. The default behaviour remains unchanged (codec-first) to avoid regressions. Signed-off-by: Stefano Radaelli Acked-by: Kuninori Morimoto Link: https://patch.msgid.link/20260213150355.442609-1-stefano.r@variscite.com Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'include') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 69a9c9c4d0e9..915e6ae5f68d 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -54,6 +54,11 @@ struct prop_nums { int platforms; }; +enum simple_util_sysclk_order { + SIMPLE_SYSCLK_ORDER_CODEC_FIRST = 0, + SIMPLE_SYSCLK_ORDER_CPU_FIRST, +}; + struct simple_util_priv { struct snd_soc_card snd_card; struct simple_dai_props { @@ -63,6 +68,7 @@ struct simple_util_priv { struct snd_soc_codec_conf *codec_conf; struct prop_nums num; unsigned int mclk_fs; + enum simple_util_sysclk_order sysclk_order; } *dai_props; struct simple_util_jack hp_jack; struct simple_util_jack mic_jack; -- cgit v1.2.3 From 80930d81c4b0753ba2ca750708e4d2fcc0627dc8 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Mon, 23 Feb 2026 15:02:55 +0000 Subject: ASoC: soc_sdw_utils: Add device info for CS47L47 Add a device info entry for the Cirrus Logic CS47L47. CS47L47 has UAJ (headset speaker + mic + jack detect) and DMICs. The audio ports are similar to the CS42L45 so can be based on the CS42L45 code. Signed-off-by: Richard Fitzgerald Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20260223150256.326143-3-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- include/sound/soc_sdw_utils.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include') diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h index 227347c8f0b3..98531e500cbb 100644 --- a/include/sound/soc_sdw_utils.h +++ b/include/sound/soc_sdw_utils.h @@ -259,6 +259,8 @@ int asoc_sdw_cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_so int asoc_sdw_cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_cs42l45_hs_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_cs42l45_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int asoc_sdw_cs47l47_hs_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int asoc_sdw_cs47l47_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); /* TI */ -- cgit v1.2.3 From 501efdcb3b3ab099fc0ce2f6e668b1c4095dd476 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 25 Feb 2026 14:01:18 +0000 Subject: ASoC: SDCA: Pull the Q7.8 volume helpers out of soc-ops It is cleaner to keep the SDCA code contained and not update the core code for things that are unlikely to see reuse outside of SDCA. Move the Q7.8 volume helpers back into the SDCA core code. Reviewed-by: Pierre-Louis Bossart Signed-off-by: Charles Keepax Link: https://patch.msgid.link/20260225140118.402695-5-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- include/sound/soc.h | 1 - 1 file changed, 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 7d8376c8e1be..172bd68e1315 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1239,7 +1239,6 @@ struct soc_mixer_control { unsigned int sign_bit; unsigned int invert:1; unsigned int autodisable:1; - unsigned int sdca_q78:1; #ifdef CONFIG_SND_SOC_TOPOLOGY struct snd_soc_dobj dobj; #endif -- cgit v1.2.3 From 2974aa42e6696a1d95b727d677dc01a71af5b998 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 27 Feb 2026 01:17:19 +0000 Subject: ASoC: remove snd_soc_pcm_subclass enum snd_soc_pcm_subclass has added at v3.1 commit b8c0dab9bf337 ("ASoC: core - PCM mutex per rtd"), but has never been used during this 15 years. Let's remove it. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/878qcfyogw.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc.h | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 7bf7ce085516..b1c5dad26edb 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -422,11 +422,6 @@ struct snd_soc_jack_pin; #include #include -enum snd_soc_pcm_subclass { - SND_SOC_PCM_CLASS_PCM = 0, - SND_SOC_PCM_CLASS_BE = 1, -}; - int snd_soc_register_card(struct snd_soc_card *card); void snd_soc_unregister_card(struct snd_soc_card *card); int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card); @@ -999,7 +994,6 @@ struct snd_soc_card { /* Mutex for PCM operations */ struct mutex pcm_mutex; - enum snd_soc_pcm_subclass pcm_subclass; int (*probe)(struct snd_soc_card *card); int (*late_probe)(struct snd_soc_card *card); @@ -1519,7 +1513,7 @@ static inline void _snd_soc_dapm_mutex_assert_held_d(struct snd_soc_dapm_context */ static inline void _snd_soc_dpcm_mutex_lock_c(struct snd_soc_card *card) { - mutex_lock_nested(&card->pcm_mutex, card->pcm_subclass); + mutex_lock(&card->pcm_mutex); } static inline void _snd_soc_dpcm_mutex_unlock_c(struct snd_soc_card *card) -- cgit v1.2.3 From 84446536f63d471ab16b2faa25eeab1df21ace0a Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Tue, 24 Feb 2026 21:56:19 +0100 Subject: ALSA: control: Verify put() result when in debug mode The put() operation is expected to return: 1) 0 on success if no changes were made 2) 1 on success if changes were made 3) error code otherwise Currently 2) is usually ignored when writing control-operations. While forcing compliance is not an option right now, make it easier for developers to adhere to the expectations and notice problems by logging them when CONFIG_SND_CTL_DEBUG is enabled. Due to large size of struct snd_ctl_elem_value, 'value_buf' is provided as a reusable buffer for kctl->put() verification. This prevents exhausting the stack when verifying the operation. >From user perspective, patch introduces a new trace/events category 'snd_ctl' containing a single 'snd_ctl_put' event type. Log sample: amixer-1086 [003] ..... 8.035939: snd_ctl_put: success: expected=0, actual=0 for ctl numid=1, iface=MIXER, name='Master Playback Volume', index=0, device=0, subdevice=0, card=0 amixer-1087 [003] ..... 8.938721: snd_ctl_put: success: expected=1, actual=1 for ctl numid=1, iface=MIXER, name='Master Playback Volume', index=0, device=0, subdevice=0, card=0 amixer-1088 [003] ..... 9.631470: snd_ctl_put: success: expected=1, actual=1 for ctl numid=1, iface=MIXER, name='Master Playback Volume', index=0, device=0, subdevice=0, card=0 amixer-1089 [000] ..... 9.636786: snd_ctl_put: fail: expected=1, actual=0 for ctl numid=5, iface=MIXER, name='Loopback Mute', index=0, device=0, subdevice=0, card=0 Signed-off-by: Cezary Rojewski Reviewed-by: Mark Brown Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20260224205619.584795-1-cezary.rojewski@intel.com --- include/sound/core.h | 3 +++ 1 file changed, 3 insertions(+) (limited to 'include') diff --git a/include/sound/core.h b/include/sound/core.h index 64327e971122..4093ec82a0a1 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -133,6 +133,9 @@ struct snd_card { #ifdef CONFIG_SND_DEBUG struct dentry *debugfs_root; /* debugfs root for card */ #endif +#ifdef CONFIG_SND_CTL_DEBUG + struct snd_ctl_elem_value *value_buf; /* buffer for kctl->put() verification */ +#endif #ifdef CONFIG_PM unsigned int power_state; /* power state */ -- cgit v1.2.3 From 73942a6ea26bd7e02b7c260b8b7aa942397be894 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Tue, 24 Feb 2026 16:18:05 +0000 Subject: firmware: cs_dsp: Add API to hibernate the DSP For some parts, the DSP is kept running when in low power mode (hibernation), leaving the firmware ALSA controls enabled, but the registers are inaccessible. Attempts to access volatile firmware controls whilst in this state would produce errors in the kernel log due to a regmap_raw_read() into DSP registers whilst the regmap is in cache_only. To remove this error log, add a hibernating flag to indicate that the controls are inaccessible, so we no longer try to read or write to the registers whilst the regmap is in cache_only. This would still produce an error when trying to read or write to these controls, but this would be a different error (-EPERM instead of -EBUSY), and would not produce a spurious error log in the kernel. Upon wake from hibernation, the control caches are re-synced to the hardware, if the DSP is running. Signed-off-by: Stefan Binding Link: https://patch.msgid.link/20260224161821.93365-2-sbinding@opensource.cirrus.com Reviewed-by: Richard Fitzgerald Signed-off-by: Mark Brown --- include/linux/firmware/cirrus/cs_dsp.h | 3 +++ 1 file changed, 3 insertions(+) (limited to 'include') diff --git a/include/linux/firmware/cirrus/cs_dsp.h b/include/linux/firmware/cirrus/cs_dsp.h index 0ec1cdc5585d..4e3baa557068 100644 --- a/include/linux/firmware/cirrus/cs_dsp.h +++ b/include/linux/firmware/cirrus/cs_dsp.h @@ -179,6 +179,7 @@ struct cs_dsp { bool booted; bool running; + bool hibernating; struct list_head ctl_list; @@ -354,4 +355,6 @@ int cs_dsp_chunk_write(struct cs_dsp_chunk *ch, int nbits, u32 val); int cs_dsp_chunk_flush(struct cs_dsp_chunk *ch); int cs_dsp_chunk_read(struct cs_dsp_chunk *ch, int nbits); +void cs_dsp_hibernate(struct cs_dsp *dsp, bool hibernating); + #endif -- cgit v1.2.3 From 98eb42c7de6b0185c914df4cca61b49ff76821ee Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 25 Feb 2026 21:50:02 +0800 Subject: ASoC: add snd_soc_lookup_component_by_name helper MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add a helper to help user to get the component by name. Signed-off-by: Bard Liao Reviewed-by: Charles Keepax Reviewed-by: Péter Ujfalusi Link: https://patch.msgid.link/20260225135004.2322987-2-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- include/sound/soc.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 7d8376c8e1be..1a486153dc76 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -465,6 +465,7 @@ struct snd_soc_component *snd_soc_lookup_component_nolocked(struct device *dev, const char *driver_name); struct snd_soc_component *snd_soc_lookup_component(struct device *dev, const char *driver_name); +struct snd_soc_component *snd_soc_lookup_component_by_name(const char *component_name); int soc_new_pcm(struct snd_soc_pcm_runtime *rtd); #ifdef CONFIG_SND_SOC_COMPRESS -- cgit v1.2.3 From a6fe20d67dc7d512f9b5dc11c5777fb1e1ff70ce Mon Sep 17 00:00:00 2001 From: Maciej Strozek Date: Fri, 6 Mar 2026 15:28:10 +0000 Subject: mfd: cs42l43: Add support for the B variant Introducing CS42L43B codec, a variant of CS42L43 which can be driven by the same driver. Changes in CS42L43 driver specific for CS42L43B: - Decimator 1 and 2 are dedicated to ADC, can't be selected for PDM - Decimators 3 and 4 are connected to PDM1 - Added Decimator 5 and 6 for PDM2 - Supports SoundWire Clock Gearing - Updated ROM requiring no patching - Reduced RAM space - Each ISRC has 4 decimators now Signed-off-by: Maciej Strozek Acked-by: Lee Jones Reviewed-by: Charles Keepax Link: https://patch.msgid.link/20260306152829.3130530-4-mstrozek@opensource.cirrus.com Signed-off-by: Mark Brown --- include/linux/mfd/cs42l43-regs.h | 76 ++++++++++++++++++++++++++++++++++++++++ include/linux/mfd/cs42l43.h | 1 + 2 files changed, 77 insertions(+) (limited to 'include') diff --git a/include/linux/mfd/cs42l43-regs.h b/include/linux/mfd/cs42l43-regs.h index c39a49269cb7..68831f113589 100644 --- a/include/linux/mfd/cs42l43-regs.h +++ b/include/linux/mfd/cs42l43-regs.h @@ -1181,4 +1181,80 @@ /* CS42L43_FW_MISSION_CTRL_MM_MCU_CFG_REG */ #define CS42L43_FW_MISSION_CTRL_MM_MCU_CFG_DISABLE_VAL 0xF05AA50F +/* CS42L43B VARIANT REGISTERS */ +#define CS42L43B_DEVID_VAL 0x0042A43B + +#define CS42L43B_DECIM_VOL_CTRL_CH1_CH2 0x00008280 +#define CS42L43B_DECIM_VOL_CTRL_CH3_CH4 0x00008284 + +#define CS42L43B_DECIM_VOL_CTRL_CH5_CH6 0x00008290 +#define CS42L43B_DECIM_VOL_CTRL_UPDATE 0x0000829C + +#define CS42L43B_DECIM_HPF_WNF_CTRL5 0x000082A0 +#define CS42L43B_DECIM_HPF_WNF_CTRL6 0x000082A4 + +#define CS42L43B_SWIRE_DP3_CH3_INPUT 0x0000C320 +#define CS42L43B_SWIRE_DP3_CH4_INPUT 0x0000C330 +#define CS42L43B_SWIRE_DP4_CH3_INPUT 0x0000C340 +#define CS42L43B_SWIRE_DP4_CH4_INPUT 0x0000C350 + +#define CS42L43B_ISRC1DEC3_INPUT1 0x0000C780 +#define CS42L43B_ISRC1DEC4_INPUT1 0x0000C790 +#define CS42L43B_ISRC2DEC3_INPUT1 0x0000C7A0 +#define CS42L43B_ISRC2DEC4_INPUT1 0x0000C7B0 + +#define CS42L43B_FW_MISSION_CTRL_NEED_CONFIGS 0x00117E00 +#define CS42L43B_FW_MISSION_CTRL_HAVE_CONFIGS 0x00117E04 +#define CS42L43B_FW_MISSION_CTRL_PATCH_START_ADDR_REG 0x00117E08 +#define CS42L43B_FW_MISSION_CTRL_MM_CTRL_SELECTION 0x00117E0C +#define CS42L43B_FW_MISSION_CTRL_MM_MCU_CFG_REG 0x00117E10 + +#define CS42L43B_MCU_SW_REV 0x00117314 +#define CS42L43B_PATCH_START_ADDR 0x00117318 +#define CS42L43B_CONFIG_SELECTION 0x0011731C +#define CS42L43B_NEED_CONFIGS 0x00117320 +#define CS42L43B_BOOT_STATUS 0x00117330 + +#define CS42L43B_FW_MISSION_CTRL_NEED_CONFIGS 0x00117E00 +#define CS42L43B_FW_MISSION_CTRL_HAVE_CONFIGS 0x00117E04 +#define CS42L43B_FW_MISSION_CTRL_PATCH_START_ADDR_REG 0x00117E08 +#define CS42L43B_FW_MISSION_CTRL_MM_CTRL_SELECTION 0x00117E0C +#define CS42L43B_FW_MISSION_CTRL_MM_MCU_CFG_REG 0x00117E10 + +#define CS42L43B_MCU_RAM_MAX 0x00117FFF + +/* CS42L43B_DECIM_DECIM_VOL_CTRL_CH5_CH6 */ +#define CS42L43B_DECIM6_MUTE_MASK 0x80000000 +#define CS42L43B_DECIM6_MUTE_SHIFT 31 +#define CS42L43B_DECIM6_VOL_MASK 0x3FC00000 +#define CS42L43B_DECIM6_VOL_SHIFT 22 +#define CS42L43B_DECIM6_PATH1_VOL_FALL_RATE_MASK 0x00380000 +#define CS42L43B_DECIM6_PATH1_VOL_FALL_RATE_SHIFT 19 +#define CS42L43B_DECIM6_PATH1_VOL_RISE_RATE_MASK 0x00070000 +#define CS42L43B_DECIM6_PATH1_VOL_RISE_RATE_SHIFT 16 +#define CS42L43B_DECIM5_MUTE_MASK 0x00008000 +#define CS42L43B_DECIM5_MUTE_SHIFT 15 +#define CS42L43B_DECIM5_VOL_MASK 0x00003FC0 +#define CS42L43B_DECIM5_VOL_SHIFT 6 +#define CS42L43B_DECIM5_PATH1_VOL_FALL_RATE_MASK 0x00000038 +#define CS42L43B_DECIM5_PATH1_VOL_FALL_RATE_SHIFT 3 +#define CS42L43B_DECIM5_PATH1_VOL_RISE_RATE_MASK 0x00000007 +#define CS42L43B_DECIM5_PATH1_VOL_RISE_RATE_SHIFT 0 + +/* CS42L43B_DECIM_VOL_CTRL_UPDATE */ +#define CS42L43B_DECIM6_PATH1_VOL_TRIG_MASK 0x00000800 +#define CS42L43B_DECIM6_PATH1_VOL_TRIG_SHIFT 11 +#define CS42L43B_DECIM5_PATH1_VOL_TRIG_MASK 0x00000100 +#define CS42L43B_DECIM5_PATH1_VOL_TRIG_SHIFT 8 +#define CS42L43B_DECIM4_VOL_UPDATE_MASK 0x00000020 +#define CS42L43B_DECIM4_VOL_UPDATE_SHIFT 5 + +/* CS42L43_ISRC1_CTRL..CS42L43_ISRC2_CTRL */ +#define CS42L43B_ISRC_DEC4_EN_MASK 0x00000008 +#define CS42L43B_ISRC_DEC4_EN_SHIFT 3 +#define CS42L43B_ISRC_DEC4_EN_WIDTH 1 +#define CS42L43B_ISRC_DEC3_EN_MASK 0x00000004 +#define CS42L43B_ISRC_DEC3_EN_SHIFT 2 +#define CS42L43B_ISRC_DEC3_EN_WIDTH 1 + #endif /* CS42L43_CORE_REGS_H */ diff --git a/include/linux/mfd/cs42l43.h b/include/linux/mfd/cs42l43.h index 2239d8585e78..ff0f7e365a19 100644 --- a/include/linux/mfd/cs42l43.h +++ b/include/linux/mfd/cs42l43.h @@ -98,6 +98,7 @@ struct cs42l43 { bool sdw_pll_active; bool attached; bool hw_lock; + long variant_id; }; #endif /* CS42L43_CORE_EXT_H */ -- cgit v1.2.3 From 97af961568c8682c44506c9ad4b26c8a5455ec1d Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Mon, 9 Mar 2026 12:45:43 +0000 Subject: ASoC: cs35l56: Put OTP register defines in correct address order Move the defines for the OTP registers to keep the register defines in order of increasing address. Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20260309124543.1135247-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- include/sound/cs35l56.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'include') diff --git a/include/sound/cs35l56.h b/include/sound/cs35l56.h index ae1e1489b671..51b153bb8598 100644 --- a/include/sound/cs35l56.h +++ b/include/sound/cs35l56.h @@ -32,9 +32,6 @@ struct snd_ctl_elem_value; #define CS35L56_UPDATE_REGS 0x0002A0C #define CS35L56_REFCLK_INPUT 0x0002C04 #define CS35L56_GLOBAL_SAMPLE_RATE 0x0002C0C -#define CS35L56_OTP_MEM_53 0x00300D4 -#define CS35L56_OTP_MEM_54 0x00300D8 -#define CS35L56_OTP_MEM_55 0x00300DC #define CS35L56_ASP1_ENABLES1 0x0004800 #define CS35L56_ASP1_CONTROL1 0x0004804 #define CS35L56_ASP1_CONTROL2 0x0004808 @@ -86,6 +83,9 @@ struct snd_ctl_elem_value; #define CS35L56_DIE_STS1 0x0017040 #define CS35L56_DIE_STS2 0x0017044 #define CS35L56_DSP_RESTRICT_STS1 0x00190F0 +#define CS35L56_OTP_MEM_53 0x00300D4 +#define CS35L56_OTP_MEM_54 0x00300D8 +#define CS35L56_OTP_MEM_55 0x00300DC #define CS35L56_DSP1_XMEM_PACKED_0 0x2000000 #define CS35L56_DSP1_XMEM_PACKED_6143 0x2005FFC #define CS35L56_DSP1_XMEM_UNPACKED32_0 0x2400000 -- cgit v1.2.3 From 7d8632f1ef6c8ed0b53771c16f130f18d636931e Mon Sep 17 00:00:00 2001 From: James Calligeros Date: Sun, 1 Mar 2026 18:05:23 +1000 Subject: ASoC: soc-dai: define possible idle TDM slot modes Some audio devices, such as certain Texas Instruments codecs, include configurable bus keepers. We currently don't have a standardised way to configure such hardware, and instead rely on the hardware initialising setting itself up into a sane state. There are situations where this is insufficient, however, and some platforms require more concrete guarantees as to the state of the bus, and being able to explicitly configure bus keepers enables this. For example, some Apple Silicon machines have an odd bus topology where the SDOUT pins of all codecs are split across two data lines, which are summed via an OR gate in front of the receiving port on the SoC's I2S peripheral. Each line must transmit 0 while a codec on the other line is actively transmitting data, or the SoC will receive garbage data. To do this, one codec on each line must be configured to transmit zeroes during the other line's active TDM slots. Thus, we define seven possible bus-keeping modes that a device can be in: NONE (UB/as initialised), OFF (explicitly disabled), ZERO (actively transmit a 0), PULLDOWN, HIZ (floating), PULLUP, and DRIVE_HIGH. These will be consumed by CODEC/CPU drivers via a common DAI op, enabling the explicit configuration of bus keepers where required. Signed-off-by: James Calligeros Link: https://patch.msgid.link/20260301-tdm-idle-slots-v3-4-c6ac5351489a@gmail.com Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 15 +++++++++++++++ 1 file changed, 15 insertions(+) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 224396927aef..44dd06add52e 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -52,6 +52,21 @@ struct snd_compr_stream; #define SND_SOC_POSSIBLE_DAIFMT_AC97 (1 << SND_SOC_DAI_FORMAT_AC97) #define SND_SOC_POSSIBLE_DAIFMT_PDM (1 << SND_SOC_DAI_FORMAT_PDM) +/* + * DAI TDM slot idle modes + * + * Describes a CODEC/CPU's behaviour when not actively receiving or + * transmitting on a given TDM slot. NONE is undefined behaviour. + * Add new modes to the end. + */ +#define SND_SOC_DAI_TDM_IDLE_NONE 0 +#define SND_SOC_DAI_TDM_IDLE_OFF 1 +#define SND_SOC_DAI_TDM_IDLE_ZERO 2 +#define SND_SOC_DAI_TDM_IDLE_PULLDOWN 3 +#define SND_SOC_DAI_TDM_IDLE_HIZ 4 +#define SND_SOC_DAI_TDM_IDLE_PULLUP 5 +#define SND_SOC_DAI_TDM_IDLE_DRIVE_HIGH 6 + /* * DAI Clock gating. * -- cgit v1.2.3 From b758d3574e88537f9089bd757a51b35cf9675179 Mon Sep 17 00:00:00 2001 From: James Calligeros Date: Sun, 1 Mar 2026 18:05:24 +1000 Subject: ASoC: soc-dai: add common operation to set TDM idle mode Some audio devices, like certain Texas Instruments codecs, integrate configurable bus keepers that dictate the codec's behaviour during idle TDM slots. Now that we have definitions for various idle modes, add a snd_soc_dai_set_tdm_idle() operation to control this in a standardised way. This is useful on Apple Silicon laptops, where a single I2S bus is comprised of two physical lines which are ORed just before the receiving port. When a codec on one line is transmitting, we must guarantee that the other line is low. We can achieve this by configuring one codec on each line to use its bus keeper to fill its line with zeroes during the active slots of the other line. Signed-off-by: James Calligeros Link: https://patch.msgid.link/20260301-tdm-idle-slots-v3-5-c6ac5351489a@gmail.com Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 44dd06add52e..6a42812bba8c 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -196,6 +196,10 @@ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); +int snd_soc_dai_set_tdm_idle(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int tx_mode, int rx_mode); + int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, unsigned int tx_num, const unsigned int *tx_slot, unsigned int rx_num, const unsigned int *rx_slot); @@ -312,6 +316,9 @@ struct snd_soc_dai_ops { int (*set_tdm_slot)(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); + int (*set_tdm_idle)(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int tx_mode, int rx_mode); int (*set_channel_map)(struct snd_soc_dai *dai, unsigned int tx_num, const unsigned int *tx_slot, unsigned int rx_num, const unsigned int *rx_slot); -- cgit v1.2.3 From cb15d8e6cbe8d085ac585016deb2e1e0107b99e5 Mon Sep 17 00:00:00 2001 From: Linus Walleij Date: Sat, 14 Mar 2026 23:56:49 +0100 Subject: ASoC: codec: arizona: Convert to use GPIO descriptors This converts the Arizona driver to use GPIO descriptors exclusively, deletes the legacy code path an updates the in-tree user of legacy GPIO. The GPIO lines for mic detect polarity and headphone ID detection are made exclusively descriptor-oriented. The headphone ID detection could actually only be used by the legacy GPIO code, but I converted it to use a descriptor if someone would actually need it so we don't just drop useful code. The compatible "wlf,hpdet-id-gpio" is not in the device tree bindings and only intended to be used by software nodes if any. If someone insists I can try to add a binding for it, but I doubt there is any real user so it seems pointless. Signed-off-by: Linus Walleij Reviewed-by: Charles Keepax Reviewed-by: Bartosz Golaszewski Acked-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20260314-asoc-arizona-v1-1-ecc9a165307c@kernel.org Signed-off-by: Mark Brown --- include/linux/mfd/arizona/pdata.h | 10 ---------- 1 file changed, 10 deletions(-) (limited to 'include') diff --git a/include/linux/mfd/arizona/pdata.h b/include/linux/mfd/arizona/pdata.h index f72e6d4b14a7..d465dcd8c90a 100644 --- a/include/linux/mfd/arizona/pdata.h +++ b/include/linux/mfd/arizona/pdata.h @@ -117,11 +117,6 @@ struct arizona_pdata { /** Check for line output with HPDET method */ bool hpdet_acc_id_line; -#ifdef CONFIG_GPIOLIB_LEGACY - /** GPIO used for mic isolation with HPDET */ - int hpdet_id_gpio; -#endif - /** Channel to use for headphone detection */ unsigned int hpdet_channel; @@ -131,11 +126,6 @@ struct arizona_pdata { /** Extra debounce timeout used during initial mic detection (ms) */ unsigned int micd_detect_debounce; -#ifdef CONFIG_GPIOLIB_LEGACY - /** GPIO for mic detection polarity */ - int micd_pol_gpio; -#endif - /** Mic detect ramp rate */ unsigned int micd_bias_start_time; -- cgit v1.2.3 From 68130eef1e0d3c1770952e738f7f8d9f340bd42d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 16 Mar 2026 02:24:43 +0000 Subject: ASoC: soc-component: re-add pcm_new()/pcm_free() Because old pcm_new()/pcm_free() didn't care about parameter component, to avoid name collisions, we have added pcm_construct()/pcm_destruct() by commit c64bfc9066007 ("ASoC: soc-core: add new pcm_construct/pcm_destruct") Because all driver switch to new pcm_construct()/pcm_destruct(), old pcm_new()/pcm_free() were remoted by commit e9067bb502787 ("ASoC: soc-component: remove snd_pcm_ops from component driver") But naming of pcm_construct()/pcm_destruct() are not goot. re-add pcm_new()/pcm_free(), and switch to use it, again. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87a4w8lde4.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc-component.h | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'include') diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 2a2b74b24a60..0435ba376369 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -90,6 +90,10 @@ struct snd_soc_component_driver { struct snd_soc_pcm_runtime *rtd); void (*pcm_destruct)(struct snd_soc_component *component, struct snd_pcm *pcm); + int (*pcm_new)(struct snd_soc_component *component, + struct snd_soc_pcm_runtime *rtd); + void (*pcm_free)(struct snd_soc_component *component, + struct snd_pcm *pcm); /* component wide operations */ int (*set_sysclk)(struct snd_soc_component *component, -- cgit v1.2.3 From 175f733325ac2ce875cafd051980be2d2c06dec9 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 16 Mar 2026 02:27:57 +0000 Subject: ASoC: soc-component: remove pcm_construct()/pcm_destruct() All driver have switched to use pcm_new()/pcm_free(), let's remove pcm_construct()/pcm_destruct(). Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/875x6wjyoa.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc-component.h | 4 ---- 1 file changed, 4 deletions(-) (limited to 'include') diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 0435ba376369..60f73c4b0bbb 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -86,10 +86,6 @@ struct snd_soc_component_driver { unsigned int reg, unsigned int val); /* pcm creation and destruction */ - int (*pcm_construct)(struct snd_soc_component *component, - struct snd_soc_pcm_runtime *rtd); - void (*pcm_destruct)(struct snd_soc_component *component, - struct snd_pcm *pcm); int (*pcm_new)(struct snd_soc_component *component, struct snd_soc_pcm_runtime *rtd); void (*pcm_free)(struct snd_soc_component *component, -- cgit v1.2.3 From 032322b44c02f5e8a127d1dca6798f91cc72eb1d Mon Sep 17 00:00:00 2001 From: Cen Zhang Date: Mon, 16 Mar 2026 16:50:47 +0800 Subject: ALSA: pcm: oss: use proper stream lock for runtime->state access __snd_pcm_set_state() writes runtime->state under the PCM stream lock. However, the OSS I/O functions snd_pcm_oss_write3(), snd_pcm_oss_read3(), snd_pcm_oss_writev3() and snd_pcm_oss_readv3() read runtime->state without holding the stream lock, only holding oss.params_lock (a different mutex that does not synchronize with the stream lock). Since __snd_pcm_set_state() is called from IRQ context (e.g., snd_pcm_period_elapsed -> snd_pcm_update_state -> __snd_pcm_xrun -> snd_pcm_stop -> snd_pcm_post_stop) while the OSS read/write paths run in process context, these are concurrent accesses that constitute a data race. Rather than using READ_ONCE()/WRITE_ONCE() barriers, introduce a snd_pcm_get_state() helper that reads runtime->state under the stream lock, matching the locking discipline used elsewhere in the PCM layer. Also export snd_pcm_set_state() for completeness. Use snd_pcm_get_state() in all four OSS I/O functions, caching the result in a local variable where the same snapshot is used for multiple comparisons to avoid taking the lock repeatedly. Signed-off-by: Cen Zhang Link: https://patch.msgid.link/20260316085047.2876451-1-zzzccc427@gmail.com Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index a7860c047503..76fc33dce537 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -729,6 +729,10 @@ static inline void __snd_pcm_set_state(struct snd_pcm_runtime *runtime, runtime->status->state = state; /* copy for mmap */ } +void snd_pcm_set_state(struct snd_pcm_substream *substream, + snd_pcm_state_t state); +snd_pcm_state_t snd_pcm_get_state(struct snd_pcm_substream *substream); + /** * bytes_to_samples - Unit conversion of the size from bytes to samples * @runtime: PCM runtime instance -- cgit v1.2.3 From e29d097ead33d0172f028b5b23f10812fe8e8335 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 3 Mar 2026 15:53:07 +0000 Subject: ASoC: dapm: Add a named controls variant of a mux widget There is already a version of the mixer widget that forces use of the specified control name, rather than factoring in the widget name. Add the same feature for mux widgets. Signed-off-by: Charles Keepax Link: https://patch.msgid.link/20260303155308.138989-2-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 49f0fe05db01..4f8fb7622a13 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -424,6 +424,7 @@ enum snd_soc_dapm_type { snd_soc_dapm_input = 0, /* input pin */ snd_soc_dapm_output, /* output pin */ snd_soc_dapm_mux, /* selects 1 analog signal from many inputs */ + snd_soc_dapm_mux_named_ctl, /* mux with named controls */ snd_soc_dapm_demux, /* connects the input to one of multiple outputs */ snd_soc_dapm_mixer, /* mixes several analog signals together */ snd_soc_dapm_mixer_named_ctl, /* mixer with named controls */ -- cgit v1.2.3 From 785639b5bf2a87eaf0cda14baaa068b3728c7be2 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?C=C3=A1ssio=20Gabriel?= Date: Mon, 16 Mar 2026 10:39:38 -0300 Subject: ALSA: timer: keep a list of open masters for slave lookup MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit snd_timer_check_slave() still walks all registered timers and all open timer instances to find a matching master for a newly opened slave. Maintain a global list of open master instances that can accept slave links and use it for the slave lookup path instead. This keeps the existing matching semantics while avoiding the nested walk over snd_timer_list and each timer open_list_head. The reverse path in snd_timer_check_master() already scans only the pending slave list, so this makes both lookup paths closer in shape without changing the master/slave linking logic. Signed-off-by: Cássio Gabriel Link: https://patch.msgid.link/20260316-alsa-timer-master-list-v1-1-fb95e547110a@gmail.com Signed-off-by: Takashi Iwai --- include/sound/timer.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/timer.h b/include/sound/timer.h index 760e132cc0cd..83bafe70cf33 100644 --- a/include/sound/timer.h +++ b/include/sound/timer.h @@ -102,6 +102,7 @@ struct snd_timer_instance { unsigned int slave_id; struct list_head open_list; struct list_head active_list; + struct list_head master_list; struct list_head ack_list; struct list_head slave_list_head; struct list_head slave_active_head; -- cgit v1.2.3 From c2da4813882b8037198cd8e67182293e17b44573 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 12 Mar 2026 00:15:03 +0000 Subject: ASoC: soc-component: add snd_soc_component_regmap_val_bytes() component has component->val_bytes which is set via snd_soc_component_setup_regmap(). But it can be calculated via component->regmap. No need to keep it as component->val_bytes. Add snd_soc_component_regmap_val_bytes() for it. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/878qbxzywo.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc-component.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 2a2b74b24a60..c08922290cf1 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -327,6 +327,7 @@ int snd_soc_component_stream_event(struct snd_soc_component *component, int snd_soc_component_set_bias_level(struct snd_soc_component *component, enum snd_soc_bias_level level); +int snd_soc_component_regmap_val_bytes(struct snd_soc_component *component); void snd_soc_component_setup_regmap(struct snd_soc_component *component); #ifdef CONFIG_REGMAP void snd_soc_component_init_regmap(struct snd_soc_component *component, -- cgit v1.2.3 From b84d27531744e046a72120882f513f42e361269d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 12 Mar 2026 00:15:55 +0000 Subject: ASoC: soc-component: remove component->val_bytes component has component->val_bytes which is set via snd_soc_component_setup_regmap(). But it can be calculated via component->regmap. No need to keep it as component->val_bytes. No one is using component->val_bytes. Remove it. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/874imlzyv8.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc-component.h | 2 -- 1 file changed, 2 deletions(-) (limited to 'include') diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index c08922290cf1..65dc540246aa 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -224,7 +224,6 @@ struct snd_soc_component { int num_dai; struct regmap *regmap; - int val_bytes; struct mutex io_mutex; @@ -328,7 +327,6 @@ int snd_soc_component_set_bias_level(struct snd_soc_component *component, enum snd_soc_bias_level level); int snd_soc_component_regmap_val_bytes(struct snd_soc_component *component); -void snd_soc_component_setup_regmap(struct snd_soc_component *component); #ifdef CONFIG_REGMAP void snd_soc_component_init_regmap(struct snd_soc_component *component, struct regmap *regmap); -- cgit v1.2.3 From d12dc8c558b955e74bf3dc9c01926c3c109f2d69 Mon Sep 17 00:00:00 2001 From: Niranjan H Y Date: Tue, 24 Mar 2026 09:42:59 +0530 Subject: ASoC: sdw_utils: add vendor_id to asoc_sdw_codec_info struct asoc_sdw_codec_info has part_id which is not sufficient to uniquely identify devices. This change adds the vendor_id field and updates the codec_info list with the corresponding vendor id as per the Manufacturer's id in https://mid.mipi.org/ Signed-off-by: Niranjan H Y Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Reviewed-by: Charles Keepax Link: https://patch.msgid.link/20260324041300.784-1-niranjan.hy@ti.com Signed-off-by: Mark Brown --- include/sound/soc_sdw_utils.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h index 98531e500cbb..48f516ba682f 100644 --- a/include/sound/soc_sdw_utils.h +++ b/include/sound/soc_sdw_utils.h @@ -71,6 +71,7 @@ struct asoc_sdw_aux_info { }; struct asoc_sdw_codec_info { + const int vendor_id; const int part_id; const int version_id; const char *name_prefix; -- cgit v1.2.3 From 37c277f050e8d24cb3db6d090d4e9cdd263ba1a3 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 25 Mar 2026 17:08:40 +0000 Subject: ASoC: soc.h: Add SOC_SINGLE_BOOL_EXT_ACC() to allow setting access flags Add a macro SOC_SINGLE_BOOL_EXT_ACC() to allow the access permission flags to be set. This is the same as SOC_SINGLE_BOOL_EXT() but with an extra argument for the access flags. Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20260325170841.1405368-2-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- include/sound/soc.h | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 7d8376c8e1be..8055def75195 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -311,6 +311,12 @@ struct platform_device; .info = snd_soc_info_bool_ext, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = xdata } +#define SOC_SINGLE_BOOL_EXT_ACC(xname, xdata, xhandler_get, xhandler_put, xaccess) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = xaccess, \ + .info = snd_soc_info_bool_ext, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = xdata } #define SOC_ENUM_EXT(xname, xenum, xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_enum_double, \ -- cgit v1.2.3 From ee7d655dbaf5e57145c73fd3925b5f44f7a1a5cc Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 25 Mar 2026 17:08:41 +0000 Subject: ASoC: cs35l56: Allow factory calibration through ALSA controls Add support for using ALSA controls to trigger a factory calibration. This is protected by a new Kconfig option so that it is only available if explicitly enabled in the kernel. By default it is not enabled. Factory calibration is normally done through debugfs files. Google have requested that factory calibration can be performed by repair shops. These repair shops only have access to the standard "user" kernel, which does not include debugfs. Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20260325170841.1405368-3-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- include/sound/cs35l56.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/cs35l56.h b/include/sound/cs35l56.h index 28f9f5940ab6..4c1969cd84d8 100644 --- a/include/sound/cs35l56.h +++ b/include/sound/cs35l56.h @@ -435,6 +435,7 @@ ssize_t cs35l56_cal_data_debugfs_read(struct cs35l56_base *cs35l56_base, ssize_t cs35l56_cal_data_debugfs_write(struct cs35l56_base *cs35l56_base, const char __user *from, size_t count, loff_t *ppos); +int cs35l56_factory_calibrate(struct cs35l56_base *cs35l56_base); void cs35l56_create_cal_debugfs(struct cs35l56_base *cs35l56_base, const struct cs35l56_cal_debugfs_fops *fops); void cs35l56_remove_cal_debugfs(struct cs35l56_base *cs35l56_base); -- cgit v1.2.3 From dd4a1963ddf0d0f5e129efec03f34ea37109b4b7 Mon Sep 17 00:00:00 2001 From: Derek Fang Date: Thu, 26 Mar 2026 15:53:01 +0800 Subject: ASoC: SOF: Intel: Add a is_amp flag to fix the wrong name prefix According to the Intel sof design, it will create the name prefix appended with amp index for the amp codec only, such as: rt1318-1, rt1318-2, etc... But the rt1320 is a codec with amp and mic codec functions, it doesn't have the amp index in its name prefix as above. And then it will be hard to identify the codec if in multi-rt1320 case. So we add a flag to force the amp index to be appended. Signed-off-by: Derek Fang Signed-off-by: Bard Liao Link: https://patch.msgid.link/20260326075303.1083567-3-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- include/sound/soc_sdw_utils.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include') diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h index 48f516ba682f..489083183673 100644 --- a/include/sound/soc_sdw_utils.h +++ b/include/sound/soc_sdw_utils.h @@ -83,6 +83,8 @@ struct asoc_sdw_codec_info { const int dai_num; struct asoc_sdw_aux_info auxs[SOC_SDW_MAX_AUX_NUM]; const int aux_num; + /* Force AMP-style name_prefix handling (append AMP index) even if MIC/Jack DAIs exist */ + const bool is_amp; int (*codec_card_late_probe)(struct snd_soc_card *card); -- cgit v1.2.3 From 7caae0aed04137545e9f8c146d8d1dbb7a8e9865 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 27 Mar 2026 02:43:31 +0000 Subject: ASoC: soc-core: remove unused dobj_list commit 8a9782346dccd ("ASoC: topology: Add topology core") added dobj_list to Component and Card, but Card side has never been used. Remove it. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/874im2xa98.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc.h | 3 --- 1 file changed, 3 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index fd6c1c8055d2..d66164fd83e5 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1073,9 +1073,6 @@ struct snd_soc_card { struct list_head dapm_list; struct list_head dapm_dirty; - /* attached dynamic objects */ - struct list_head dobj_list; - /* Generic DAPM context for the card */ struct snd_soc_dapm_context *dapm; struct snd_soc_dapm_stats dapm_stats; -- cgit v1.2.3 From 3bd246d1cf609a80cae19e4aefb599256a72b1a6 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?C=C3=A1ssio=20Gabriel?= Date: Sat, 28 Mar 2026 01:53:35 -0300 Subject: ALSA: hda/proc: show GPI and GPO state in codec proc output MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit print_gpio() prints the GPIO capability header and the bidirectional GPIO state, but it never reports the separate GPI and GPO pins even though AC_PAR_GPIO_CAP exposes their counts. The HD-audio specification defines dedicated GPI and GPO verbs alongside the GPIO ones, so codecs with input-only or output-only general-purpose pins currently lose that state from /proc/asound/card*/codec#* altogether. Add the missing read verb definitions and extend print_gpio() to dump the GPI and GPO pins, too, while leaving the existing IO[] output unchanged. Signed-off-by: Cássio Gabriel Link: https://patch.msgid.link/20260328-hda-proc-gpi-gpo-v1-1-fabb36564bee@gmail.com Signed-off-by: Takashi Iwai --- include/sound/hda_verbs.h | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/hda_verbs.h b/include/sound/hda_verbs.h index 006d358acce2..127e7016e4fe 100644 --- a/include/sound/hda_verbs.h +++ b/include/sound/hda_verbs.h @@ -56,7 +56,12 @@ enum { #define AC_VERB_GET_DIGI_CONVERT_1 0x0f0d #define AC_VERB_GET_DIGI_CONVERT_2 0x0f0e /* unused */ #define AC_VERB_GET_VOLUME_KNOB_CONTROL 0x0f0f -/* f10-f1a: GPIO */ +/* f10-f1a: GPI/GPO/GPIO */ +#define AC_VERB_GET_GPI_DATA 0x0f10 +#define AC_VERB_GET_GPI_WAKE_MASK 0x0f11 +#define AC_VERB_GET_GPI_UNSOLICITED_RSP_MASK 0x0f12 +#define AC_VERB_GET_GPI_STICKY_MASK 0x0f13 +#define AC_VERB_GET_GPO_DATA 0x0f14 #define AC_VERB_GET_GPIO_DATA 0x0f15 #define AC_VERB_GET_GPIO_MASK 0x0f16 #define AC_VERB_GET_GPIO_DIRECTION 0x0f17 -- cgit v1.2.3 From 38f6e93dedbc1b6c2a6e97110ab7e872c257a5e3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 28 Mar 2026 14:43:05 +0100 Subject: ALSA: hda: Add missing SET_GPI_* and SET_GPO_* verb definitions We've added the definitions of the missing GPI and GPO verbs for reading in the previous commit, but the counter-part for setting values is missing. Add the definitions of missing verbs for comprehensiveness. Link: https://patch.msgid.link/20260328134319.207482-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/hda_verbs.h | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'include') diff --git a/include/sound/hda_verbs.h b/include/sound/hda_verbs.h index 127e7016e4fe..6066954409aa 100644 --- a/include/sound/hda_verbs.h +++ b/include/sound/hda_verbs.h @@ -104,6 +104,11 @@ enum { #define AC_VERB_SET_DIGI_CONVERT_2 0x70e #define AC_VERB_SET_DIGI_CONVERT_3 0x73e #define AC_VERB_SET_VOLUME_KNOB_CONTROL 0x70f +#define AC_VERB_SET_GPI_DATA 0x710 +#define AC_VERB_SET_GPI_WAKE_MASK 0x711 +#define AC_VERB_SET_SPI_UNSOLICITED_RSP_MASK 0x712 +#define AC_VERB_SET_GPI_STICKY_MASK 0x713 +#define AC_VERB_SET_GPO_DATA 0x714 #define AC_VERB_SET_GPIO_DATA 0x715 #define AC_VERB_SET_GPIO_MASK 0x716 #define AC_VERB_SET_GPIO_DIRECTION 0x717 -- cgit v1.2.3 From 5a77906982df26975aa26caefb81b7d6f53d9b3f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 1 Apr 2026 00:19:06 +0000 Subject: ASoC: soc.h: remove snd_soc_of_parse_audio_prefix() No one is using snd_soc_of_parse_audio_prefix(). Remove it. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/877bqrttvp.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc.h | 9 --------- 1 file changed, 9 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index d66164fd83e5..f70edd9c23b2 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1337,15 +1337,6 @@ void snd_soc_of_parse_node_prefix(struct device_node *np, struct snd_soc_codec_conf *codec_conf, struct device_node *of_node, const char *propname); -static inline -void snd_soc_of_parse_audio_prefix(struct snd_soc_card *card, - struct snd_soc_codec_conf *codec_conf, - struct device_node *of_node, - const char *propname) -{ - snd_soc_of_parse_node_prefix(card->dev->of_node, - codec_conf, of_node, propname); -} int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname); -- cgit v1.2.3 From e46957f27c6004f3ab5ec456f4e848950364ebc1 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Thu, 2 Apr 2026 08:11:11 +0000 Subject: ASoC: dt-bindings: qcom: add LPASS LPI MI2S dai ids Add new dai ids entries for LPASS LPI MI2S and SENARY MI2S audio lines. Co-developed-by: Mohammad Rafi Shaik Signed-off-by: Mohammad Rafi Shaik Signed-off-by: Srinivas Kandagatla Reviewed-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20260402081118.348071-7-srinivas.kandagatla@oss.qualcomm.com Signed-off-by: Mark Brown --- include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h | 12 ++++++++++++ 1 file changed, 12 insertions(+) (limited to 'include') diff --git a/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h b/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h index 6d1ce7f5da51..45850f2d4342 100644 --- a/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h +++ b/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h @@ -140,6 +140,18 @@ #define DISPLAY_PORT_RX_6 134 #define DISPLAY_PORT_RX_7 135 #define USB_RX 136 +#define LPI_MI2S_RX_0 137 +#define LPI_MI2S_TX_0 138 +#define LPI_MI2S_RX_1 139 +#define LPI_MI2S_TX_1 140 +#define LPI_MI2S_RX_2 141 +#define LPI_MI2S_TX_2 142 +#define LPI_MI2S_RX_3 143 +#define LPI_MI2S_TX_3 144 +#define LPI_MI2S_RX_4 145 +#define LPI_MI2S_TX_4 146 +#define SENARY_MI2S_RX 147 +#define SENARY_MI2S_TX 148 #define LPASS_CLK_ID_PRI_MI2S_IBIT 1 #define LPASS_CLK_ID_PRI_MI2S_EBIT 2 -- cgit v1.2.3 From ba2a0e81d4d71c3bbc61c420b6fac5abaeddd77d Mon Sep 17 00:00:00 2001 From: Niranjan H Y Date: Wed, 1 Apr 2026 18:51:45 +0530 Subject: ASoC: SDCA: Export Q7.8 volume control helpers Export the Q7.8 volume control helpers to allow reuse by other ASoC drivers. These functions handle 16-bit signed Q7.8 fixed-point format values for volume controls. Changes include: - Rename q78_get_volsw to sdca_asoc_q78_get_volsw - Rename q78_put_volsw to sdca_asoc_q78_put_volsw - Add a convenience macro SDCA_SINGLE_Q78_TLV and SDCA_DOUBLE_Q78_TLV for creating mixer controls This allows other ASoC drivers to easily implement controls using the Q7.8 fixed-point format without duplicating code. Signed-off-by: Niranjan H Y Reviewed-by: Charles Keepax Link: https://patch.msgid.link/20260401132148.2367-1-niranjan.hy@ti.com Signed-off-by: Mark Brown --- include/sound/sdca_asoc.h | 43 ++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 42 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/sdca_asoc.h b/include/sound/sdca_asoc.h index aa9124f93218..46a61a52decc 100644 --- a/include/sound/sdca_asoc.h +++ b/include/sound/sdca_asoc.h @@ -13,6 +13,8 @@ struct device; struct regmap; struct sdca_function_data; +struct snd_ctl_elem_value; +struct snd_kcontrol; struct snd_kcontrol_new; struct snd_pcm_hw_params; struct snd_pcm_substream; @@ -23,6 +25,42 @@ struct snd_soc_dai_ops; struct snd_soc_dapm_route; struct snd_soc_dapm_widget; +/* convenient macro to handle the mono volume in 7.8 fixed format representation */ +#define SDCA_SINGLE_Q78_TLV(xname, xreg, xmin, xmax, xstep, tlv_array) \ +{ \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, \ + .get = sdca_asoc_q78_get_volsw, \ + .put = sdca_asoc_q78_put_volsw, \ + .private_value = (unsigned long)&(struct soc_mixer_control) { \ + .reg = (xreg), .rreg = (xreg), \ + .min = (xmin), .max = (xmax), \ + .shift = (xstep), .rshift = (xstep), \ + .sign_bit = 15 \ + } \ +} + +/* convenient macro for stereo volume in 7.8 fixed format with separate registers for L/R */ +#define SDCA_DOUBLE_Q78_TLV(xname, xreg_l, xreg_r, xmin, xmax, xstep, tlv_array) \ +{ \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, \ + .get = sdca_asoc_q78_get_volsw, \ + .put = sdca_asoc_q78_put_volsw, \ + .private_value = (unsigned long)&(struct soc_mixer_control) { \ + .reg = (xreg_l), .rreg = (xreg_r), \ + .min = (xmin), .max = (xmax), \ + .shift = (xstep), .rshift = (xstep), \ + .sign_bit = 15 \ + } \ +} + int sdca_asoc_count_component(struct device *dev, struct sdca_function_data *function, int *num_widgets, int *num_routes, int *num_controls, int *num_dais); @@ -57,5 +95,8 @@ int sdca_asoc_hw_params(struct device *dev, struct regmap *regmap, struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai); - +int sdca_asoc_q78_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int sdca_asoc_q78_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); #endif // __SDCA_ASOC_H__ -- cgit v1.2.3 From d38e9457ddf0780dd55c953886ae48abbe4d33b8 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?C=C3=A1ssio=20Gabriel?= Date: Mon, 6 Apr 2026 00:20:03 -0300 Subject: ALSA: gus: add shared GF1 suspend and resume helpers MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit gusclassic and gusextreme still leave their ISA PM callbacks disabled because the shared GF1 core only provides probe-time startup and full shutdown paths. Those helpers are not suitable for suspend and resume. They reset software handlers and tear down runtime state such as the DRAM allocator, timer state, DMA queues, PCM state and UART setup. Resume instead needs a narrower recovery path that rebuilds the GF1 hardware state without rerunning probe-only detection or discarding the bookkeeping kept by the card instance. Add shared GF1 suspend and resume helpers for that recovery path. Suspend now quiesces GF1 PCM, aborts queued GF1 DMA work, resets the UART and powers the chip down without tearing down allocator, timer or rawmidi bookkeeping. Resume rebuilds the GF1 hardware state, restores timer and UART handlers, and brings the chip back to a usable post-resume state for the ISA front-ends. The scope is limited to restoring post-resume usability. It does not attempt transparent continuation of active GF1 PCM or synth state across suspend, and userspace may still need to reprepare streams or reload onboard sample data after resume. Open rawmidi substreams are restored only to a usable post-resume state. Signed-off-by: Cássio Gabriel Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20260406-b4-alsa-gus-isa-pm-v1-1-b6829a7457cd@gmail.com --- include/sound/gus.h | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'include') diff --git a/include/sound/gus.h b/include/sound/gus.h index 321ae93625eb..3feb42627de1 100644 --- a/include/sound/gus.h +++ b/include/sound/gus.h @@ -536,6 +536,7 @@ int snd_gf1_dma_transfer_block(struct snd_gus_card * gus, struct snd_gf1_dma_block * block, int atomic, int synth); +void snd_gf1_dma_suspend(struct snd_gus_card *gus); /* gus_volume.c */ @@ -552,6 +553,8 @@ struct snd_gus_voice *snd_gf1_alloc_voice(struct snd_gus_card * gus, int type, i void snd_gf1_free_voice(struct snd_gus_card * gus, struct snd_gus_voice *voice); int snd_gf1_start(struct snd_gus_card * gus); int snd_gf1_stop(struct snd_gus_card * gus); +int snd_gf1_suspend(struct snd_gus_card *gus); +int snd_gf1_resume(struct snd_gus_card *gus); /* gus_mixer.c */ @@ -572,6 +575,8 @@ int snd_gus_create(struct snd_card *card, int effect, struct snd_gus_card ** rgus); int snd_gus_initialize(struct snd_gus_card * gus); +int snd_gus_suspend(struct snd_gus_card *gus); +int snd_gus_resume(struct snd_gus_card *gus); /* gus_irq.c */ @@ -583,6 +588,8 @@ void snd_gus_irq_profile_init(struct snd_gus_card *gus); /* gus_uart.c */ int snd_gf1_rawmidi_new(struct snd_gus_card *gus, int device); +void snd_gf1_uart_suspend(struct snd_gus_card *gus); +void snd_gf1_uart_resume(struct snd_gus_card *gus); /* gus_dram.c */ int snd_gus_dram_write(struct snd_gus_card *gus, char __user *ptr, @@ -593,5 +600,6 @@ int snd_gus_dram_read(struct snd_gus_card *gus, char __user *ptr, /* gus_timer.c */ void snd_gf1_timers_init(struct snd_gus_card *gus); void snd_gf1_timers_done(struct snd_gus_card *gus); +void snd_gf1_timers_resume(struct snd_gus_card *gus); #endif /* __SOUND_GUS_H */ -- cgit v1.2.3 From d78ddeb8938a366aabfabf60255c1a94de8d8ea1 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 6 Apr 2026 05:51:51 +0000 Subject: ASoC: soc.h: remove unused card->pmdown_time commit f0fba2ad1b6b ("ASoC: multi-component - ASoC Multi-Component Support") has replaced "card->pmdown_time" to "rtd->pmdown_time". card->pmdown_time has been not used this 15 years. Let's remove it. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87eckstz49.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc.h | 2 -- 1 file changed, 2 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index f70edd9c23b2..5e3eb617d832 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1027,8 +1027,6 @@ struct snd_soc_card { void (*remove_dai_link)(struct snd_soc_card *, struct snd_soc_dai_link *link); - long pmdown_time; - /* CPU <--> Codec DAI links */ struct snd_soc_dai_link *dai_link; /* predefined links only */ int num_links; /* predefined links only */ -- cgit v1.2.3 From 19cbb3e0c27f28feb7781641994226cb2ee206a2 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?C=C3=A1ssio=20Gabriel?= Date: Tue, 7 Apr 2026 12:35:42 -0300 Subject: ALSA: tea6330t: add mixer state restore helper MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The InterWave STB variant uses a TEA6330T mixer on its private I2C bus. The mixer state is cached in software, but there is no helper to push that register image back to hardware after system resume. Add a small restore helper that reapplies the cached TEA6330T register image to the device so board drivers can restore the external mixer state as part of their PM resume path. Take snd_i2c_lock() around the full device lookup and restore sequence so the bus device list traversal is also protected. Signed-off-by: Cássio Gabriel Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20260407-alsa-interwave-pm-v2-2-8dd96c6129e9@gmail.com --- include/sound/tea6330t.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/tea6330t.h b/include/sound/tea6330t.h index 1c77b78f6533..3a34033d2aa3 100644 --- a/include/sound/tea6330t.h +++ b/include/sound/tea6330t.h @@ -12,5 +12,6 @@ int snd_tea6330t_detect(struct snd_i2c_bus *bus, int equalizer); int snd_tea6330t_update_mixer(struct snd_card *card, struct snd_i2c_bus *bus, int equalizer, int fader); +int snd_tea6330t_restore_mixer(struct snd_i2c_bus *bus); #endif /* __SOUND_TEA6330T_H */ -- cgit v1.2.3 From b0762dd2fcab5b8b4b953314f3f6eb9d92bc16bc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Apr 2026 11:38:14 +0200 Subject: ALSA: hda: Add sync version of snd_hda_codec_write() We used snd_hda_codec_read() for the verb write when a synchronization is needed after the write, e.g. for the power state toggle or such cases. It works in principle, but it looks rather confusing and too hackish. For improving the code readability, introduce a new helper function, snd_hda_codec_write_sync(), which is another variant of snd_hda_codec_write(), and replace the existing snd_hda_codec_read() calls with this one. No behavior change but just the code refactoring. Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20260409093826.1317626-2-tiwai@suse.de --- include/sound/hda_codec.h | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'include') diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index 5d9f0ef228af..292d6024388b 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -336,6 +336,17 @@ snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags, return snd_hdac_codec_write(&codec->core, nid, flags, verb, parm); } +/* sync after write */ +static inline int +snd_hda_codec_write_sync(struct hda_codec *codec, hda_nid_t nid, int flags, + unsigned int verb, unsigned int parm) +{ + /* use snd_hda_codec_read() for writing; + * the returned value is usually discarded + */ + return snd_hdac_codec_read(&codec->core, nid, flags, verb, parm); +} + #define snd_hda_param_read(codec, nid, param) \ snd_hdac_read_parm(&(codec)->core, nid, param) #define snd_hda_get_sub_nodes(codec, nid, start_nid) \ -- cgit v1.2.3 From cd8fd5a0566e0d93fbd408e6b06ca484a78b5ccd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Apr 2026 11:38:15 +0200 Subject: ALSA: hda: Add a simple GPIO setup helper function Introduce a common GPIO setup helper function, so that we can clean up the open code found in many codec drivers later. Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20260409093826.1317626-3-tiwai@suse.de --- include/sound/hda_codec.h | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'include') diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index 292d6024388b..24581080e26a 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -481,6 +481,10 @@ void snd_hda_unlock_devices(struct hda_bus *bus); void snd_hda_bus_reset(struct hda_bus *bus); void snd_hda_bus_reset_codecs(struct hda_bus *bus); +void snd_hda_codec_set_gpio(struct hda_codec *codec, unsigned int mask, + unsigned int dir, unsigned int data, + unsigned int delay); + int snd_hda_codec_set_name(struct hda_codec *codec, const char *name); /* -- cgit v1.2.3 From 0cb7aa965ad02e90ba7d6bf847f3de07e8d0c05e Mon Sep 17 00:00:00 2001 From: Linus Walleij Date: Thu, 9 Apr 2026 23:39:31 +0200 Subject: ASoC: uda1380: Modernize the driver This codec driver depended on the legacy GPIO API, and nothing in the kernel is defining the platform data, so get rid of this. Two in-kernel device trees are defining this codec using undocumented device tree properties, so support these for now. The same properties can be defined using software nodes if board files are desired. The device tree use the "-gpio" rather than "-gpios" suffix but the GPIO DT parser will deal with that. Since there may be out of tree users, migrate to GPIO descriptors, drop the platform data that is unused, and assign the dac_clk the value that was used in all platforms found in a historical dig, and support setting the clock to the PLL using the undocumented device tree property. Add some menuconfig so the codec can be selected and tested. Signed-off-by: Linus Walleij Link: https://patch.msgid.link/20260409-asoc-uda1380-v3-1-b3d5a53f31be@kernel.org Signed-off-by: Mark Brown --- include/sound/uda1380.h | 19 ------------------- 1 file changed, 19 deletions(-) delete mode 100644 include/sound/uda1380.h (limited to 'include') diff --git a/include/sound/uda1380.h b/include/sound/uda1380.h deleted file mode 100644 index 2e42ea2d0cfd..000000000000 --- a/include/sound/uda1380.h +++ /dev/null @@ -1,19 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ -/* - * UDA1380 ALSA SoC Codec driver - * - * Copyright 2009 Philipp Zabel - */ - -#ifndef __UDA1380_H -#define __UDA1380_H - -struct uda1380_platform_data { - int gpio_power; - int gpio_reset; - int dac_clk; -#define UDA1380_DAC_CLK_SYSCLK 0 -#define UDA1380_DAC_CLK_WSPLL 1 -}; - -#endif /* __UDA1380_H */ -- cgit v1.2.3