From 0eba2a7e858907a746ba69cd002eb9eb4dbd7bf3 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 28 Feb 2025 15:14:56 +0000 Subject: [PATCH 1/7] ASoC: ops: Consistently treat platform_max as control value This reverts commit 9bdd10d57a88 ("ASoC: ops: Shift tested values in snd_soc_put_volsw() by +min"), and makes some additional related updates. There are two ways the platform_max could be interpreted; the maximum register value, or the maximum value the control can be set to. The patch moved from treating the value as a control value to a register one. When the patch was applied it was technically correct as snd_soc_limit_volume() also used the register interpretation. However, even then most of the other usages treated platform_max as a control value, and snd_soc_limit_volume() has since been updated to also do so in commit fb9ad24485087 ("ASoC: ops: add correct range check for limiting volume"). That patch however, missed updating snd_soc_put_volsw() back to the control interpretation, and fixing snd_soc_info_volsw_range(). The control interpretation makes more sense as limiting is typically done from the machine driver, so it is appropriate to use the customer facing representation rather than the internal codec representation. Update all the code to consistently use this interpretation of platform_max. Finally, also add some comments to the soc_mixer_control struct to hopefully avoid further patches switching between the two approaches. Fixes: fb9ad24485087 ("ASoC: ops: add correct range check for limiting volume") Signed-off-by: Charles Keepax Link: https://patch.msgid.link/20250228151456.3703342-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- include/sound/soc.h | 5 ++++- sound/soc/soc-ops.c | 15 +++++++-------- 2 files changed, 11 insertions(+), 9 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index fcdb5adfcd5ec..b3e84bc47c6fd 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1261,7 +1261,10 @@ void snd_soc_close_delayed_work(struct snd_soc_pcm_runtime *rtd); /* mixer control */ struct soc_mixer_control { - int min, max, platform_max; + /* Minimum and maximum specified as written to the hardware */ + int min, max; + /* Limited maximum value specified as presented through the control */ + int platform_max; int reg, rreg; unsigned int shift, rshift; unsigned int sign_bit; diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 19928f098d8dc..b0e4e4168f38d 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -337,7 +337,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, if (ucontrol->value.integer.value[0] < 0) return -EINVAL; val = ucontrol->value.integer.value[0]; - if (mc->platform_max && ((int)val + min) > mc->platform_max) + if (mc->platform_max && val > mc->platform_max) return -EINVAL; if (val > max - min) return -EINVAL; @@ -350,7 +350,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, if (ucontrol->value.integer.value[1] < 0) return -EINVAL; val2 = ucontrol->value.integer.value[1]; - if (mc->platform_max && ((int)val2 + min) > mc->platform_max) + if (mc->platform_max && val2 > mc->platform_max) return -EINVAL; if (val2 > max - min) return -EINVAL; @@ -503,17 +503,16 @@ int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol, { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - int platform_max; - int min = mc->min; + int max; - if (!mc->platform_max) - mc->platform_max = mc->max; - platform_max = mc->platform_max; + max = mc->max - mc->min; + if (mc->platform_max && mc->platform_max < max) + max = mc->platform_max; uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; uinfo->value.integer.min = 0; - uinfo->value.integer.max = platform_max - min; + uinfo->value.integer.max = max; return 0; } From e26f1cfeac6712516bfeed80890da664f4f2e88a Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 6 Mar 2025 13:32:54 +0000 Subject: [PATCH 2/7] ASoC: cs42l43: Fix maximum ADC Volume The range of ADC volume is -1 -> 3 (-6 to 18dB) so the number of levels should actually be 4. Fixes: fc918cbe874e ("ASoC: cs42l43: Add support for the cs42l43") Signed-off-by: Charles Keepax Link: https://patch.msgid.link/20250306133254.1861046-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l43.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 4257dbefe9dd1..d307b56a7f38e 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -1146,7 +1146,7 @@ static const struct snd_kcontrol_new cs42l43_controls[] = { SOC_DOUBLE_R_SX_TLV("ADC Volume", CS42L43_ADC_B_CTRL1, CS42L43_ADC_B_CTRL2, CS42L43_ADC_PGA_GAIN_SHIFT, - 0xF, 5, cs42l43_adc_tlv), + 0xF, 4, cs42l43_adc_tlv), SOC_DOUBLE("PDM1 Invert Switch", CS42L43_DMIC_PDM_CTRL, CS42L43_PDM1L_INV_SHIFT, CS42L43_PDM1R_INV_SHIFT, 1, 0), From 0704a15b930cf97073ce091a0cd7ad32f2304329 Mon Sep 17 00:00:00 2001 From: Thomas Mizrahi Date: Sat, 8 Mar 2025 01:06:28 -0300 Subject: [PATCH 3/7] ASoC: amd: yc: Support mic on another Lenovo ThinkPad E16 Gen 2 model The internal microphone on the Lenovo ThinkPad E16 model requires a quirk entry to work properly. This was fixed in a previous patch (linked below), but depending on the specific variant of the model, the product name may be "21M5" or "21M6". The following patch fixed this issue for the 21M5 variant: https://lore.kernel.org/all/20240725065442.9293-1-tiwai@suse.de/ This patch adds support for the microphone on the 21M6 variant. Link: https://github.com/ramaureirac/thinkpad-e14-linux/issues/31 Cc: stable@vger.kernel.org Signed-off-by: Thomas Mizrahi Link: https://patch.msgid.link/20250308041303.198765-1-thomasmizra@gmail.com Signed-off-by: Mark Brown --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index b16587d8f97a8..a7637056972aa 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -248,6 +248,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "21M5"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "21M6"), + } + }, { .driver_data = &acp6x_card, .matches = { From 247fba13416af65b155949bae582d55c310f58b6 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 10 Mar 2025 16:04:40 +0800 Subject: [PATCH 4/7] ASoC: rt722-sdca: add missing readable registers SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT722_SDCA_ENT_FU15, RT722_SDCA_CTL_FU_CH_GAIN, CH_01) ... SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT722_SDCA_ENT_FU15, RT722_SDCA_CTL_FU_CH_GAIN, CH_04) are used by the "FU15 Boost Volume" control, but not marked as readable. And the mbq size are 2 for those registers. Fixes: 7f5d6036ca005 ("ASoC: rt722-sdca: Add RT722 SDCA driver") Signed-off-by: Bard Liao Reviewed-by: Ranjani Sridharan Reviewed-by: Shuming Fan Link: https://patch.msgid.link/20250310080440.58797-1-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt722-sdca-sdw.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/rt722-sdca-sdw.c b/sound/soc/codecs/rt722-sdca-sdw.c index 25fc13687bc83..4d3043627bd04 100644 --- a/sound/soc/codecs/rt722-sdca-sdw.c +++ b/sound/soc/codecs/rt722-sdca-sdw.c @@ -86,6 +86,10 @@ static bool rt722_sdca_mbq_readable_register(struct device *dev, unsigned int re case 0x6100067: case 0x6100070 ... 0x610007c: case 0x6100080: + case SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT722_SDCA_ENT_FU15, RT722_SDCA_CTL_FU_CH_GAIN, + CH_01) ... + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT722_SDCA_ENT_FU15, RT722_SDCA_CTL_FU_CH_GAIN, + CH_04): case SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT722_SDCA_ENT_USER_FU1E, RT722_SDCA_CTL_FU_VOLUME, CH_01): case SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT722_SDCA_ENT_USER_FU1E, RT722_SDCA_CTL_FU_VOLUME, From ed92bc5264c4357d4fca292c769ea9967cd3d3b6 Mon Sep 17 00:00:00 2001 From: Christophe JAILLET Date: Mon, 10 Mar 2025 18:45:36 +0100 Subject: [PATCH 5/7] ASoC: codecs: wm0010: Fix error handling path in wm0010_spi_probe() Free some resources in the error handling path of the probe, as already done in the remove function. Fixes: e3523e01869d ("ASoC: wm0010: Add initial wm0010 DSP driver") Fixes: fd8b96574456 ("ASoC: wm0010: Clear IRQ as wake source and include missing header") Signed-off-by: Christophe JAILLET Reviewed-by: Charles Keepax Link: https://patch.msgid.link/5139ba1ab8c4c157ce04e56096a0f54a1683195c.1741549792.git.christophe.jaillet@wanadoo.fr Signed-off-by: Mark Brown --- sound/soc/codecs/wm0010.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index edd2cb185c42c..9e67fbfc2ccaf 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -920,7 +920,7 @@ static int wm0010_spi_probe(struct spi_device *spi) if (ret) { dev_err(wm0010->dev, "Failed to set IRQ %d as wake source: %d\n", irq, ret); - return ret; + goto free_irq; } if (spi->max_speed_hz) @@ -932,9 +932,18 @@ static int wm0010_spi_probe(struct spi_device *spi) &soc_component_dev_wm0010, wm0010_dai, ARRAY_SIZE(wm0010_dai)); if (ret < 0) - return ret; + goto disable_irq_wake; return 0; + +disable_irq_wake: + irq_set_irq_wake(wm0010->irq, 0); + +free_irq: + if (wm0010->irq) + free_irq(wm0010->irq, wm0010); + + return ret; } static void wm0010_spi_remove(struct spi_device *spi) From 658fb7fe8e7f4014ea17a4da0e0c1d9bc319fa35 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Wed, 5 Mar 2025 18:27:32 +0100 Subject: [PATCH 6/7] ASoC: cs42l43: convert to SYSTEM_SLEEP_PM_OPS The custom suspend function causes a build warning when CONFIG_PM_SLEEP is disabled: sound/soc/codecs/cs42l43.c:2405:12: error: unused function 'cs42l43_codec_runtime_force_suspend' [-Werror,-Wunused-function] Change SET_SYSTEM_SLEEP_PM_OPS() to the newer SYSTEM_SLEEP_PM_OPS(), to avoid this. Fixes: 164b7dd4546b ("ASoC: cs42l43: Add jack delay debounce after suspend") Signed-off-by: Arnd Bergmann Reviewed-by: Maciej Strozek Reviewed-by: Charles Keepax Link: https://patch.msgid.link/20250305172738.3437513-1-arnd@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l43.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index d307b56a7f38e..ea84ac64c775e 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -2417,7 +2417,7 @@ static int cs42l43_codec_runtime_force_suspend(struct device *dev) static const struct dev_pm_ops cs42l43_codec_pm_ops = { RUNTIME_PM_OPS(NULL, cs42l43_codec_runtime_resume, NULL) - SET_SYSTEM_SLEEP_PM_OPS(cs42l43_codec_runtime_force_suspend, pm_runtime_force_resume) + SYSTEM_SLEEP_PM_OPS(cs42l43_codec_runtime_force_suspend, pm_runtime_force_resume) }; static const struct platform_device_id cs42l43_codec_id_table[] = { From de74ec718e0788e1998eb7289ad07970e27cae27 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 28 Feb 2025 00:29:30 +0000 Subject: [PATCH 7/7] ASoC: simple-card-utils: Don't use __free(device_node) at graph_util_parse_dai() commit 419d1918105e ("ASoC: simple-card-utils: use __free(device_node) for device node") uses __free(device_node) for dlc->of_node, but we need to keep it while driver is in use. Don't use __free(device_node) in graph_util_parse_dai(). Fixes: 419d1918105e ("ASoC: simple-card-utils: use __free(device_node) for device node") Reported-by: Thuan Nguyen Reported-by: Detlev Casanova Signed-off-by: Kuninori Morimoto Tested-by: Thuan Nguyen Tested-by: Detlev Casanova Link: https://patch.msgid.link/87eczisyhh.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/generic/simple-card-utils.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index c2445c5ccd84c..32efb30c55d69 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -1077,6 +1077,7 @@ static int graph_get_dai_id(struct device_node *ep) int graph_util_parse_dai(struct device *dev, struct device_node *ep, struct snd_soc_dai_link_component *dlc, int *is_single_link) { + struct device_node *node; struct of_phandle_args args = {}; struct snd_soc_dai *dai; int ret; @@ -1084,7 +1085,7 @@ int graph_util_parse_dai(struct device *dev, struct device_node *ep, if (!ep) return 0; - struct device_node *node __free(device_node) = of_graph_get_port_parent(ep); + node = of_graph_get_port_parent(ep); /* * Try to find from DAI node @@ -1126,8 +1127,10 @@ int graph_util_parse_dai(struct device *dev, struct device_node *ep, * if he unbinded CPU or Codec. */ ret = snd_soc_get_dlc(&args, dlc); - if (ret < 0) + if (ret < 0) { + of_node_put(node); return ret; + } parse_dai_end: if (is_single_link)