From 292709b9cf3ba470af94b62c9bb60284cc581b79 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Sat, 7 May 2022 20:14:13 +0800 Subject: ASoC: fsl_micfil: explicitly clear software reset bit SRES is self-cleared bit, but REG_MICFIL_CTRL1 is defined as non volatile register, it still remain in regmap cache after set, then every update of REG_MICFIL_CTRL1, software reset happens. to avoid this, clear it explicitly. Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1651925654-32060-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_micfil.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 9f90989ac59a..cb84d95c3aac 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -191,6 +191,17 @@ static int fsl_micfil_reset(struct device *dev) return ret; } + /* + * SRES is self-cleared bit, but REG_MICFIL_CTRL1 is defined + * as non-volatile register, so SRES still remain in regmap + * cache after set, that every update of REG_MICFIL_CTRL1, + * software reset happens. so clear it explicitly. + */ + ret = regmap_clear_bits(micfil->regmap, REG_MICFIL_CTRL1, + MICFIL_CTRL1_SRES); + if (ret) + return ret; + return 0; } -- cgit v1.2.1 From b776c4a4618ec1b5219d494c423dc142f23c4e8f Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Sat, 7 May 2022 20:14:14 +0800 Subject: ASoC: fsl_micfil: explicitly clear CHnF flags There may be failure when start 1 channel recording after 8 channels recording. The reason is that the CHnF flags are not cleared successfully by software reset. This issue is triggerred by the change of clearing software reset bit. CHnF flags are write 1 clear bits. Clear them by force write. Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1651925654-32060-2-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_micfil.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index cb84d95c3aac..d1cd104f8584 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -202,6 +202,14 @@ static int fsl_micfil_reset(struct device *dev) if (ret) return ret; + /* + * Set SRES should clear CHnF flags, But even add delay here + * the CHnF may not be cleared sometimes, so clear CHnF explicitly. + */ + ret = regmap_write_bits(micfil->regmap, REG_MICFIL_STAT, 0xFF, 0xFF); + if (ret) + return ret; + return 0; } -- cgit v1.2.1 From 698813ba8c580efb356ace8dbf55f61dac6063a8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 11 May 2022 14:41:36 +0100 Subject: ASoC: ops: Fix bounds check for _sx controls For _sx controls the semantics of the max field is not the usual one, max is the number of steps rather than the maximum value. This means that our check in snd_soc_put_volsw_sx() needs to just check against the maximum value. Fixes: 4f1e50d6a9cf9c1b ("ASoC: ops: Reject out of bounds values in snd_soc_put_volsw_sx()") Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220511134137.169575-1-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/soc-ops.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index e693070f51fe..1ac7e2ce31a1 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -435,7 +435,7 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, val = ucontrol->value.integer.value[0]; if (mc->platform_max && val > mc->platform_max) return -EINVAL; - if (val > max - min) + if (val > max) return -EINVAL; val_mask = mask << shift; val = (val + min) & mask; -- cgit v1.2.1 From 97eea946b93961fffd29448dcda7398d0d51c4b2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 11 May 2022 14:41:37 +0100 Subject: ASoC: ops: Check bounds for second channel in snd_soc_put_volsw_sx() The bounds checks in snd_soc_put_volsw_sx() are only being applied to the first channel, meaning it is possible to write out of bounds values to the second channel in stereo controls. Add appropriate checks. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220511134137.169575-2-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/soc-ops.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 1ac7e2ce31a1..7cac26a64e0c 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -451,6 +451,12 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, val_mask = mask << rshift; val2 = (ucontrol->value.integer.value[1] + min) & mask; + + if (mc->platform_max && val2 > mc->platform_max) + return -EINVAL; + if (val2 > max) + return -EINVAL; + val2 = val2 << rshift; err = snd_soc_component_update_bits(component, reg2, val_mask, -- cgit v1.2.1 From 19c5bda74dc45fee598a57600b550c9ea7662f10 Mon Sep 17 00:00:00 2001 From: Hui Tang Date: Thu, 12 May 2022 15:46:40 +0800 Subject: ASoC: tlv320adc3xxx: Fix build error for implicit function declaration MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit sound/soc/codecs/tlv320adc3xxx.c: In function ‘adc3xxx_i2c_probe’: sound/soc/codecs/tlv320adc3xxx.c:1359:21: error: implicit declaration of function ‘devm_gpiod_get’; did you mean ‘devm_gpio_free’? [-Werror=implicit-function-declaration] adc3xxx->rst_pin = devm_gpiod_get(dev, "reset", GPIOD_OUT_LOW); ^~~~~~~~~~~~~~ devm_gpio_free CC [M] drivers/gpu/drm/nouveau/nvkm/engine/disp/sorgt215.o LD [M] sound/soc/codecs/snd-soc-ak4671.o LD [M] sound/soc/codecs/snd-soc-arizona.o LD [M] sound/soc/codecs/snd-soc-cros-ec-codec.o LD [M] sound/soc/codecs/snd-soc-ak4641.o LD [M] sound/soc/codecs/snd-soc-alc5632.o sound/soc/codecs/tlv320adc3xxx.c:1359:50: error: ‘GPIOD_OUT_LOW’ undeclared (first use in this function); did you mean ‘GPIOF_INIT_LOW’? adc3xxx->rst_pin = devm_gpiod_get(dev, "reset", GPIOD_OUT_LOW); ^~~~~~~~~~~~~ GPIOF_INIT_LOW sound/soc/codecs/tlv320adc3xxx.c:1359:50: note: each undeclared identifier is reported only once for each function it appears in LD [M] sound/soc/codecs/snd-soc-cs35l32.o sound/soc/codecs/tlv320adc3xxx.c:1408:2: error: implicit declaration of function ‘gpiod_set_value_cansleep’; did you mean ‘gpio_set_value_cansleep’? [-Werror=implicit-function-declaration] gpiod_set_value_cansleep(adc3xxx->rst_pin, 1); ^~~~~~~~~~~~~~~~~~~~~~~~ gpio_set_value_cansleep LD [M] sound/soc/codecs/snd-soc-cs35l41-lib.o LD [M] sound/soc/codecs/snd-soc-cs35l36.o LD [M] sound/soc/codecs/snd-soc-cs35l34.o LD [M] sound/soc/codecs/snd-soc-cs35l41.o CC [M] drivers/gpu/drm/nouveau/nvkm/engine/disp/sormcp89.o cc1: all warnings being treated as errors Fixes: e9a3b57efd28 ("ASoC: codec: tlv320adc3xxx: New codec driver") Signed-off-by: Hui Tang Link: https://lore.kernel.org/r/20220512074640.75550-3-tanghui20@huawei.com Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320adc3xxx.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/tlv320adc3xxx.c b/sound/soc/codecs/tlv320adc3xxx.c index ae18982ac310..104167c6fd93 100644 --- a/sound/soc/codecs/tlv320adc3xxx.c +++ b/sound/soc/codecs/tlv320adc3xxx.c @@ -14,6 +14,7 @@ #include #include +#include #include #include #include @@ -1025,7 +1026,9 @@ static const struct gpio_chip adc3xxx_gpio_chip = { static void adc3xxx_free_gpio(struct adc3xxx *adc3xxx) { +#ifdef CONFIG_GPIOLIB gpiochip_remove(&adc3xxx->gpio_chip); +#endif } static void adc3xxx_init_gpio(struct adc3xxx *adc3xxx) -- cgit v1.2.1 From 1683d3282f240336a2b4b6b541d435facfe8bbb6 Mon Sep 17 00:00:00 2001 From: Paul Cercueil Date: Tue, 25 Oct 2022 16:01:49 +0100 Subject: ASoC: dapm: Don't use prefix for regulator name When a component has a prefix, and uses a SND_SOC_DAPM_REGULATOR_SUPPLY, the name of the regulator should not use the prefix, otherwise it won't be properly matched in the DT/ACPI. Fixes: 3caac759681e ("ASoC: soc-dapm.c: fixup snd_soc_dapm_new_control_unlocked() error handling") Signed-off-by: Paul Cercueil Link: https://lore.kernel.org/r/20221025150149.113129-1-paul@crapouillou.net Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d515e7a78ea8..879cf1be67a9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3645,7 +3645,7 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, switch (w->id) { case snd_soc_dapm_regulator_supply: - w->regulator = devm_regulator_get(dapm->dev, w->name); + w->regulator = devm_regulator_get(dapm->dev, widget->name); if (IS_ERR(w->regulator)) { ret = PTR_ERR(w->regulator); goto request_failed; -- cgit v1.2.1 From d40b6529c6269cd5afddb1116a383cab9f126694 Mon Sep 17 00:00:00 2001 From: Brent Mendelsohn Date: Mon, 24 Oct 2022 18:42:27 +0100 Subject: ASoC: amd: yc: Add Alienware m17 R5 AMD into DMI table This model requires an additional detection quirk to enable the internal microphone - BIOS doesn't seem to support AcpDmicConnected (nothing in acpidump output). Link: https://bugzilla.kernel.org/show_bug.cgi?id=216590 Signed-off-by: Brent Mendelsohn Reviewed-by: Mario Limonciello Link: https://lore.kernel.org/r/20221024174227.4160-1-mendiebm@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 6c0f1de10429..d9715bea965e 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -206,6 +206,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "UM5302TA"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Alienware"), + DMI_MATCH(DMI_PRODUCT_NAME, "Alienware m17 R5 AMD"), + } + }, {} }; -- cgit v1.2.1 From 8bb0ac0e6f64ebdf15d963c26b028de391c9bcf9 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Tue, 25 Oct 2022 16:09:42 +0200 Subject: ASoC: Intel: bytcht_es8316: Add quirk for the Nanote UMPC-01 The Nanote UMPC-01 mini laptop has stereo speakers, while the default bytcht_es8316 settings assume a mono speaker setup. Add a quirk for this. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20221025140942.509066-1-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcht_es8316.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 6432b83f616f..a935c5fd9edb 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -443,6 +443,13 @@ static const struct dmi_system_id byt_cht_es8316_quirk_table[] = { | BYT_CHT_ES8316_INTMIC_IN2_MAP | BYT_CHT_ES8316_JD_INVERTED), }, + { /* Nanote UMPC-01 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "RWC CO.,LTD"), + DMI_MATCH(DMI_PRODUCT_NAME, "UMPC-01"), + }, + .driver_data = (void *)BYT_CHT_ES8316_INTMIC_IN1_MAP, + }, { /* Teclast X98 Plus II */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "TECLAST"), -- cgit v1.2.1 From 69d1abc0214e944dff1d30e201f8fc332a1adf1a Mon Sep 17 00:00:00 2001 From: Tzung-Bi Shih Date: Tue, 25 Oct 2022 10:49:29 +0800 Subject: MAINTAINERS: update Tzung-Bi's email address Use kernel.org account instead. Signed-off-by: Tzung-Bi Shih Link: https://lore.kernel.org/r/20221025024929.2652134-1-tzungbi@kernel.org Signed-off-by: Mark Brown --- .mailmap | 1 + Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml | 2 +- Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml | 2 +- MAINTAINERS | 2 +- 4 files changed, 4 insertions(+), 3 deletions(-) diff --git a/.mailmap b/.mailmap index 380378e2db36..84342d781407 100644 --- a/.mailmap +++ b/.mailmap @@ -414,6 +414,7 @@ TripleX Chung TripleX Chung Tsuneo Yoshioka Tycho Andersen +Tzung-Bi Shih Uwe Kleine-König Uwe Kleine-König Uwe Kleine-König diff --git a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml index c3e9f3485449..dea293f403d9 100644 --- a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml +++ b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml @@ -8,7 +8,7 @@ title: Audio codec controlled by ChromeOS EC maintainers: - Cheng-Yi Chiang - - Tzung-Bi Shih + - Tzung-Bi Shih description: | Google's ChromeOS EC codec is a digital mic codec provided by the diff --git a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml index 1d73204451b1..ea7d4900ee4a 100644 --- a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml +++ b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: Realtek rt1015p codec devicetree bindings maintainers: - - Tzung-Bi Shih + - Tzung-Bi Shih description: | Rt1015p is a rt1015 variant which does not support I2C and diff --git a/MAINTAINERS b/MAINTAINERS index cf0f18502372..f9749afc0b9d 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -4906,7 +4906,7 @@ F: drivers/platform/chrome/ CHROMEOS EC CODEC DRIVER M: Cheng-Yi Chiang -M: Tzung-Bi Shih +M: Tzung-Bi Shih R: Guenter Roeck L: chrome-platform@lists.linux.dev S: Maintained -- cgit v1.2.1 From 6ec27c53886c8963729885bcf2dd996eba2767a7 Mon Sep 17 00:00:00 2001 From: Chen Zhongjin Date: Fri, 28 Oct 2022 11:16:03 +0800 Subject: ASoC: core: Fix use-after-free in snd_soc_exit() KASAN reports a use-after-free: BUG: KASAN: use-after-free in device_del+0xb5b/0xc60 Read of size 8 at addr ffff888008655050 by task rmmod/387 CPU: 2 PID: 387 Comm: rmmod Hardware name: QEMU Standard PC (i440FX + PIIX, 1996) Call Trace: dump_stack_lvl+0x79/0x9a print_report+0x17f/0x47b kasan_report+0xbb/0xf0 device_del+0xb5b/0xc60 platform_device_del.part.0+0x24/0x200 platform_device_unregister+0x2e/0x40 snd_soc_exit+0xa/0x22 [snd_soc_core] __do_sys_delete_module.constprop.0+0x34f/0x5b0 do_syscall_64+0x3a/0x90 entry_SYSCALL_64_after_hwframe+0x63/0xcd ... It's bacause in snd_soc_init(), snd_soc_util_init() is possble to fail, but its ret is ignored, which makes soc_dummy_dev unregistered twice. snd_soc_init() snd_soc_util_init() platform_device_register_simple(soc_dummy_dev) platform_driver_register() # fail platform_device_unregister(soc_dummy_dev) platform_driver_register() # success ... snd_soc_exit() snd_soc_util_exit() # soc_dummy_dev will be unregistered for second time To fix it, handle error and stop snd_soc_init() when util_init() fail. Also clean debugfs when util_init() or driver_register() fail. Fixes: fb257897bf20 ("ASoC: Work around allmodconfig failure") Signed-off-by: Chen Zhongjin Link: https://lore.kernel.org/r/20221028031603.59416-1-chenzhongjin@huawei.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 17 +++++++++++++++-- 1 file changed, 15 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 12a82f5a3ff6..a409fbed8f34 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3477,10 +3477,23 @@ EXPORT_SYMBOL_GPL(snd_soc_of_get_dai_link_cpus); static int __init snd_soc_init(void) { + int ret; + snd_soc_debugfs_init(); - snd_soc_util_init(); + ret = snd_soc_util_init(); + if (ret) + goto err_util_init; - return platform_driver_register(&soc_driver); + ret = platform_driver_register(&soc_driver); + if (ret) + goto err_register; + return 0; + +err_register: + snd_soc_util_exit(); +err_util_init: + snd_soc_debugfs_exit(); + return ret; } module_init(snd_soc_init); -- cgit v1.2.1 From 6a564338a23cefcfc29c4a535b98402d13efdda6 Mon Sep 17 00:00:00 2001 From: Maarten Zanders Date: Fri, 28 Oct 2022 16:11:28 +0200 Subject: ASoC: fsl_asrc fsl_esai fsl_sai: allow CONFIG_PM=N When CONFIG_PM=N, pm_runtime_put_sync() returns -ENOSYS which breaks the probe function of these drivers. Other users of pm_runtime_put_sync() typically don't check the return value. In order to keep the program flow as intended, check for -ENOSYS. This commit is similar to commit 0434d3f (omap-mailbox.c). Fixes: cab04ab5900f ("ASoC: fsl_asrc: Don't use devm_regmap_init_mmio_clk") Fixes: 203773e39347 ("ASoC: fsl_esai: Don't use devm_regmap_init_mmio_clk") Fixes: 2277e7e36b4b ("ASoC: fsl_sai: Don't use devm_regmap_init_mmio_clk") Signed-off-by: Maarten Zanders Reviewed-by: Daniel Baluta Link: https://lore.kernel.org/r/20221028141129.100702-1-maarten.zanders@mind.be Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_asrc.c | 2 +- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_sai.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 936aef5d2767..e16e7b3fa96c 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -1232,7 +1232,7 @@ static int fsl_asrc_probe(struct platform_device *pdev) } ret = pm_runtime_put_sync(&pdev->dev); - if (ret < 0) + if (ret < 0 && ret != -ENOSYS) goto err_pm_get_sync; ret = devm_snd_soc_register_component(&pdev->dev, &fsl_asrc_component, diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 5c21fc490fce..17fefd27ec90 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -1069,7 +1069,7 @@ static int fsl_esai_probe(struct platform_device *pdev) regmap_write(esai_priv->regmap, REG_ESAI_RSMB, 0); ret = pm_runtime_put_sync(&pdev->dev); - if (ret < 0) + if (ret < 0 && ret != -ENOSYS) goto err_pm_get_sync; /* diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 81f89f6767a2..e60c7b344562 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1446,7 +1446,7 @@ static int fsl_sai_probe(struct platform_device *pdev) } ret = pm_runtime_put_sync(dev); - if (ret < 0) + if (ret < 0 && ret != -ENOSYS) goto err_pm_get_sync; /* -- cgit v1.2.1 From e59bf547a7dd366f93bfebb7487959580ca6c0ec Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Martin=20Povi=C5=A1er?= Date: Thu, 27 Oct 2022 11:57:58 +0200 Subject: ASoC: tas2770: Fix set_tdm_slot in case of single slot MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit There's a special branch in the set_tdm_slot op for the case of nslots being 1, but: (1) That branch can never work (there's a check for tx_mask being non-zero, later there's another check for it *being* zero; one or the other always throws -EINVAL). (2) The intention of the branch seems to be what the general other branch reduces to in case of nslots being 1. For those reasons remove the 'nslots being 1' special case. Fixes: 1a476abc723e ("tas2770: add tas2770 smart PA kernel driver") Suggested-by: Jos Dehaes Signed-off-by: Martin Povišer Link: https://lore.kernel.org/r/20221027095800.16094-1-povik+lin@cutebit.org Signed-off-by: Mark Brown --- sound/soc/codecs/tas2770.c | 20 ++++++-------------- 1 file changed, 6 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index b6765235a4b3..8557759acb1f 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -395,21 +395,13 @@ static int tas2770_set_dai_tdm_slot(struct snd_soc_dai *dai, if (tx_mask == 0 || rx_mask != 0) return -EINVAL; - if (slots == 1) { - if (tx_mask != 1) - return -EINVAL; - - left_slot = 0; - right_slot = 0; + left_slot = __ffs(tx_mask); + tx_mask &= ~(1 << left_slot); + if (tx_mask == 0) { + right_slot = left_slot; } else { - left_slot = __ffs(tx_mask); - tx_mask &= ~(1 << left_slot); - if (tx_mask == 0) { - right_slot = left_slot; - } else { - right_slot = __ffs(tx_mask); - tx_mask &= ~(1 << right_slot); - } + right_slot = __ffs(tx_mask); + tx_mask &= ~(1 << right_slot); } if (tx_mask != 0 || left_slot >= slots || right_slot >= slots) -- cgit v1.2.1 From faac764ea1ea6898d93e46c403271fb105c0906e Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Martin=20Povi=C5=A1er?= Date: Thu, 27 Oct 2022 11:57:59 +0200 Subject: ASoC: tas2764: Fix set_tdm_slot in case of single slot MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit There's a special branch in the set_tdm_slot op for the case of nslots being 1, but: (1) That branch can never work (there's a check for tx_mask being non-zero, later there's another check for it *being* zero; one or the other always throws -EINVAL). (2) The intention of the branch seems to be what the general other branch reduces to in case of nslots being 1. For those reasons remove the 'nslots being 1' special case. Fixes: 827ed8a0fa50 ("ASoC: tas2764: Add the driver for the TAS2764") Suggested-by: Jos Dehaes Signed-off-by: Martin Povišer Link: https://lore.kernel.org/r/20221027095800.16094-2-povik+lin@cutebit.org Signed-off-by: Mark Brown --- sound/soc/codecs/tas2764.c | 19 ++++++------------- 1 file changed, 6 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/tas2764.c b/sound/soc/codecs/tas2764.c index 51b87a936179..2e0ed3e68fa5 100644 --- a/sound/soc/codecs/tas2764.c +++ b/sound/soc/codecs/tas2764.c @@ -438,20 +438,13 @@ static int tas2764_set_dai_tdm_slot(struct snd_soc_dai *dai, if (tx_mask == 0 || rx_mask != 0) return -EINVAL; - if (slots == 1) { - if (tx_mask != 1) - return -EINVAL; - left_slot = 0; - right_slot = 0; + left_slot = __ffs(tx_mask); + tx_mask &= ~(1 << left_slot); + if (tx_mask == 0) { + right_slot = left_slot; } else { - left_slot = __ffs(tx_mask); - tx_mask &= ~(1 << left_slot); - if (tx_mask == 0) { - right_slot = left_slot; - } else { - right_slot = __ffs(tx_mask); - tx_mask &= ~(1 << right_slot); - } + right_slot = __ffs(tx_mask); + tx_mask &= ~(1 << right_slot); } if (tx_mask != 0 || left_slot >= slots || right_slot >= slots) -- cgit v1.2.1 From 6f934afa6a980bb8d3ce73836b1a9922685e50d7 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Martin=20Povi=C5=A1er?= Date: Thu, 27 Oct 2022 11:58:00 +0200 Subject: ASoC: tas2780: Fix set_tdm_slot in case of single slot MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit There's a special branch in the set_tdm_slot op for the case of nslots being 1, but: (1) That branch can never work (there's a check for tx_mask being non-zero, later there's another check for it *being* zero; one or the other always throws -EINVAL). (2) The intention of the branch seems to be what the general other branch reduces to in case of nslots being 1. For those reasons remove the 'nslots being 1' special case. Fixes: eae9f9ce181b ("ASoC: add tas2780 driver") Suggested-by: Jos Dehaes Signed-off-by: Martin Povišer Link: https://lore.kernel.org/r/20221027095800.16094-3-povik+lin@cutebit.org Signed-off-by: Mark Brown --- sound/soc/codecs/tas2780.c | 19 ++++++------------- 1 file changed, 6 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/tas2780.c b/sound/soc/codecs/tas2780.c index a6db6f0e5431..afdf0c863aa1 100644 --- a/sound/soc/codecs/tas2780.c +++ b/sound/soc/codecs/tas2780.c @@ -380,20 +380,13 @@ static int tas2780_set_dai_tdm_slot(struct snd_soc_dai *dai, if (tx_mask == 0 || rx_mask != 0) return -EINVAL; - if (slots == 1) { - if (tx_mask != 1) - return -EINVAL; - left_slot = 0; - right_slot = 0; + left_slot = __ffs(tx_mask); + tx_mask &= ~(1 << left_slot); + if (tx_mask == 0) { + right_slot = left_slot; } else { - left_slot = __ffs(tx_mask); - tx_mask &= ~(1 << left_slot); - if (tx_mask == 0) { - right_slot = left_slot; - } else { - right_slot = __ffs(tx_mask); - tx_mask &= ~(1 << right_slot); - } + right_slot = __ffs(tx_mask); + tx_mask &= ~(1 << right_slot); } if (tx_mask != 0 || left_slot >= slots || right_slot >= slots) -- cgit v1.2.1 From 9a1d248bb4beaf1b43d17ba12481ee0629fa29b9 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 31 Oct 2022 15:58:36 -0400 Subject: ASoC: Intel: soc-acpi: add ES83x6 support to IceLake Missing entry to find a machine driver for ES83x6-based platforms. Link: https://github.com/thesofproject/linux/issues/3873 Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20221031195836.250193-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-icl-match.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-icl-match.c b/sound/soc/intel/common/soc-acpi-intel-icl-match.c index b032bc07de8b..d0062f2cd256 100644 --- a/sound/soc/intel/common/soc-acpi-intel-icl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-icl-match.c @@ -10,6 +10,11 @@ #include #include "../skylake/skl.h" +static const struct snd_soc_acpi_codecs essx_83x6 = { + .num_codecs = 3, + .codecs = { "ESSX8316", "ESSX8326", "ESSX8336"}, +}; + static struct skl_machine_pdata icl_pdata = { .use_tplg_pcm = true, }; @@ -27,6 +32,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_icl_machines[] = { .drv_name = "sof_rt5682", .sof_tplg_filename = "sof-icl-rt5682.tplg", }, + { + .comp_ids = &essx_83x6, + .drv_name = "sof-essx8336", + .sof_tplg_filename = "sof-icl-es8336", /* the tplg suffix is added at run time */ + .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_SSP_NUMBER | + SND_SOC_ACPI_TPLG_INTEL_SSP_MSB | + SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER, + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_icl_machines); -- cgit v1.2.1 From 5d73263f9e7c54ccb20814dc50809b9deb9e2bc7 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 31 Oct 2022 15:56:39 -0400 Subject: ASoC: hda: intel-dsp-config: add ES83x6 quirk for IceLake Yet another hardware variant we need to handle. Link: https://github.com/thesofproject/linux/issues/3873 Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Takashi Iwai Link: https://lore.kernel.org/r/20221031195639.250062-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/hda/intel-dsp-config.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index b9eb3208f288..ae31bb127594 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -320,6 +320,11 @@ static const struct config_entry config_table[] = { {} } }, + { + .flags = FLAG_SOF, + .device = 0x34c8, + .codec_hid = &essx_83x6, + }, { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0x34c8, -- cgit v1.2.1 From 003b786b678919e072c2b12ffa73901ef840963e Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Tue, 1 Nov 2022 13:49:13 +0200 Subject: ASoC: SOF: ipc3-topology: use old pipeline teardown flow with SOF2.1 and older MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Originally in commit b2ebcf42a48f ("ASoC: SOF: free widgets in sof_tear_down_pipelines() for static pipelines"), freeing of pipeline components at suspend was only done with recent FW as there were known limitations in older firmware versions. Tests show that if static pipelines are used, i.e. all pipelines are setup whenever firmware is powered up, the reverse action of freeing all components at power down, leads to firmware failures with also SOF2.0 and SOF2.1 based firmware. The problems can be specific to certain topologies with e.g. components not prepared to be freed at suspend (as this did not happen with older SOF kernels). To avoid hitting these problems when kernel is upgraded and used with an older firmware, bump the firmware requirement to SOF2.2 or newer. If an older firmware is used, and pipeline is a static one, do not free the components at suspend. This ensures the suspend flow remains backwards compatible with older firmware versions. This limitation does not apply if the product configuration is updated to dynamic pipelines. The limitation is not linked to firmware ABI, as the interface to free pipeline components has been available already before ABI3.19. The problem is in the implementation, so firmware version should be used to decide whether it is safe to use the newer flow or not. This patch adds a new SOF_FW_VER() macro to compare SOF firmware release versions. Link: https://github.com/thesofproject/sof/issues/6475 Signed-off-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20221101114913.1292671-1-kai.vehmanen@linux.intel.com Signed-off-by: Mark Brown --- include/sound/sof/info.h | 4 ++++ sound/soc/sof/ipc3-topology.c | 15 ++++++++++----- 2 files changed, 14 insertions(+), 5 deletions(-) diff --git a/include/sound/sof/info.h b/include/sound/sof/info.h index 65e86e4e9fd8..75193850ead0 100644 --- a/include/sound/sof/info.h +++ b/include/sound/sof/info.h @@ -36,6 +36,10 @@ enum sof_ipc_ext_data { SOF_IPC_EXT_USER_ABI_INFO = 4, }; +/* Build u32 number in format MMmmmppp */ +#define SOF_FW_VER(MAJOR, MINOR, PATCH) ((uint32_t)( \ + ((MAJOR) << 24) | ((MINOR) << 12) | (PATCH))) + /* FW version - SOF_IPC_GLB_VERSION */ struct sof_ipc_fw_version { struct sof_ipc_hdr hdr; diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index c148715aa0f9..0720e1eae084 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -2275,6 +2275,7 @@ static int sof_ipc3_tear_down_all_pipelines(struct snd_sof_dev *sdev, bool verif struct sof_ipc_fw_version *v = &sdev->fw_ready.version; struct snd_sof_widget *swidget; struct snd_sof_route *sroute; + bool dyn_widgets = false; int ret; /* @@ -2284,12 +2285,14 @@ static int sof_ipc3_tear_down_all_pipelines(struct snd_sof_dev *sdev, bool verif * topology loading the sound card unavailable to open PCMs. */ list_for_each_entry(swidget, &sdev->widget_list, list) { - if (swidget->dynamic_pipeline_widget) + if (swidget->dynamic_pipeline_widget) { + dyn_widgets = true; continue; + } - /* Do not free widgets for static pipelines with FW ABI older than 3.19 */ + /* Do not free widgets for static pipelines with FW older than SOF2.2 */ if (!verify && !swidget->dynamic_pipeline_widget && - v->abi_version < SOF_ABI_VER(3, 19, 0)) { + SOF_FW_VER(v->major, v->minor, v->micro) < SOF_FW_VER(2, 2, 0)) { swidget->use_count = 0; swidget->complete = 0; continue; @@ -2303,9 +2306,11 @@ static int sof_ipc3_tear_down_all_pipelines(struct snd_sof_dev *sdev, bool verif /* * Tear down all pipelines associated with PCMs that did not get suspended * and unset the prepare flag so that they can be set up again during resume. - * Skip this step for older firmware. + * Skip this step for older firmware unless topology has any + * dynamic pipeline (in which case the step is mandatory). */ - if (!verify && v->abi_version >= SOF_ABI_VER(3, 19, 0)) { + if (!verify && (dyn_widgets || SOF_FW_VER(v->major, v->minor, v->micro) >= + SOF_FW_VER(2, 2, 0))) { ret = sof_tear_down_left_over_pipelines(sdev); if (ret < 0) { dev_err(sdev->dev, "failed to tear down paused pipelines\n"); -- cgit v1.2.1 From 392cc13c5ec72ccd6bbfb1bc2339502cc59dd285 Mon Sep 17 00:00:00 2001 From: Jason Montleon Date: Thu, 3 Nov 2022 10:46:11 -0400 Subject: ASoC: rt5514: fix legacy dai naming Starting with 6.0-rc1 these messages are logged and the sound card is unavailable. Adding legacy_dai_naming to the rt5514-spi causes it to function properly again. [ 16.928454] kbl_r5514_5663_max kbl_r5514_5663_max: ASoC: CPU DAI spi-PRP0001:00 not registered [ 16.928561] platform kbl_r5514_5663_max: deferred probe pending Fixes: fc34ece41f71 ("ASoC: Refactor non_legacy_dai_naming flag") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=216641 Signed-off-by: Jason Montleon Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20221103144612.4431-1-jmontleo@redhat.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5514-spi.c | 15 ++++++++------- 1 file changed, 8 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 1a25a3787935..362663abcb89 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -298,13 +298,14 @@ static int rt5514_spi_pcm_new(struct snd_soc_component *component, } static const struct snd_soc_component_driver rt5514_spi_component = { - .name = DRV_NAME, - .probe = rt5514_spi_pcm_probe, - .open = rt5514_spi_pcm_open, - .hw_params = rt5514_spi_hw_params, - .hw_free = rt5514_spi_hw_free, - .pointer = rt5514_spi_pcm_pointer, - .pcm_construct = rt5514_spi_pcm_new, + .name = DRV_NAME, + .probe = rt5514_spi_pcm_probe, + .open = rt5514_spi_pcm_open, + .hw_params = rt5514_spi_hw_params, + .hw_free = rt5514_spi_hw_free, + .pointer = rt5514_spi_pcm_pointer, + .pcm_construct = rt5514_spi_pcm_new, + .legacy_dai_naming = 1, }; /** -- cgit v1.2.1 From a1dca8774faf3f77eb34fa0ac6f3e2b82290b1e4 Mon Sep 17 00:00:00 2001 From: Jason Montleon Date: Thu, 3 Nov 2022 10:46:12 -0400 Subject: ASoC: rt5677: fix legacy dai naming Starting with 6.0-rc1 the CPU DAI is not registered and the sound card is unavailable. Adding legacy_dai_naming causes it to function properly again. Fixes: fc34ece41f71 ("ASoC: Refactor non_legacy_dai_naming flag") Signed-off-by: Jason Montleon Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20221103144612.4431-2-jmontleo@redhat.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677-spi.c | 19 ++++++++++--------- 1 file changed, 10 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index 8f3993a4c1cc..d25703dd7499 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -396,15 +396,16 @@ static int rt5677_spi_pcm_probe(struct snd_soc_component *component) } static const struct snd_soc_component_driver rt5677_spi_dai_component = { - .name = DRV_NAME, - .probe = rt5677_spi_pcm_probe, - .open = rt5677_spi_pcm_open, - .close = rt5677_spi_pcm_close, - .hw_params = rt5677_spi_hw_params, - .hw_free = rt5677_spi_hw_free, - .prepare = rt5677_spi_prepare, - .pointer = rt5677_spi_pcm_pointer, - .pcm_construct = rt5677_spi_pcm_new, + .name = DRV_NAME, + .probe = rt5677_spi_pcm_probe, + .open = rt5677_spi_pcm_open, + .close = rt5677_spi_pcm_close, + .hw_params = rt5677_spi_hw_params, + .hw_free = rt5677_spi_hw_free, + .prepare = rt5677_spi_prepare, + .pointer = rt5677_spi_pcm_pointer, + .pcm_construct = rt5677_spi_pcm_new, + .legacy_dai_naming = 1, }; /* Select a suitable transfer command for the next transfer to ensure -- cgit v1.2.1 From 314d34fe7f0a5836cb0472950c1f17744b4efde8 Mon Sep 17 00:00:00 2001 From: Chen Zhongjin Date: Mon, 31 Oct 2022 21:40:31 +0800 Subject: ASoC: soc-utils: Remove __exit for snd_soc_util_exit() snd_soc_util_exit() is called in __init snd_soc_init() for cleanup. Remove the __exit annotation for it to fix the build warning: WARNING: modpost: sound/soc/snd-soc-core.o: section mismatch in reference: init_module (section: .init.text) -> snd_soc_util_exit (section: .exit.text) Fixes: 6ec27c53886c ("ASoC: core: Fix use-after-free in snd_soc_exit()") Signed-off-by: Chen Zhongjin Link: https://lore.kernel.org/r/20221031134031.256511-1-chenzhongjin@huawei.com Signed-off-by: Mark Brown --- sound/soc/soc-utils.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index a3b6df2378b4..a4dba0b751e7 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -264,7 +264,7 @@ int __init snd_soc_util_init(void) return ret; } -void __exit snd_soc_util_exit(void) +void snd_soc_util_exit(void) { platform_driver_unregister(&soc_dummy_driver); platform_device_unregister(soc_dummy_dev); -- cgit v1.2.1 From 3d59eaef49ca2db581156a7b77c9afc0546eefc0 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 7 Nov 2022 11:04:33 +0200 Subject: ASoC: SOF: topology: No need to assign core ID if token parsing failed Move the return value check before attempting to assign the core ID to the swidget since we are going to fail the sof_widget_ready() and free up swidget anyways. Fixes: 909dadf21aae ("ASoC: SOF: topology: Make DAI widget parsing IPC agnostic") Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20221107090433.5146-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/topology.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 38855dd60617..6a0e7f3b5023 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1344,16 +1344,6 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, break; } - if (sof_debug_check_flag(SOF_DBG_DISABLE_MULTICORE)) { - swidget->core = SOF_DSP_PRIMARY_CORE; - } else { - int core = sof_get_token_value(SOF_TKN_COMP_CORE_ID, swidget->tuples, - swidget->num_tuples); - - if (core >= 0) - swidget->core = core; - } - /* check token parsing reply */ if (ret < 0) { dev_err(scomp->dev, @@ -1365,6 +1355,16 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, return ret; } + if (sof_debug_check_flag(SOF_DBG_DISABLE_MULTICORE)) { + swidget->core = SOF_DSP_PRIMARY_CORE; + } else { + int core = sof_get_token_value(SOF_TKN_COMP_CORE_ID, swidget->tuples, + swidget->num_tuples); + + if (core >= 0) + swidget->core = core; + } + /* bind widget to external event */ if (tw->event_type) { if (widget_ops[w->id].bind_event) { -- cgit v1.2.1 From 89cdb224f2abe37ec4ac21ba0d9ddeb5a6a9cf68 Mon Sep 17 00:00:00 2001 From: Zhu Ning Date: Fri, 28 Oct 2022 10:04:56 +0800 Subject: ASoC: sof_es8336: reduce pop noise on speaker The Speaker GPIO needs to be turned on slightly behind the codec turned on. It also need to be turned off slightly before the codec turned down. Current code uses delay in DAPM_EVENT to do it but the mdelay delays the DAPM itself and thus has no effect. A delayed_work is added to turn on the speaker. The Speaker is turned off in .trigger since trigger is called slightly before the DAPM events. Signed-off-by: Zhu Ning ------------ v1: cancel delayed work while disabling speaker. Link: https://lore.kernel.org/r/20221028020456.90286-1-zhuning0077@gmail.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_es8336.c | 60 ++++++++++++++++++++++++++++--------- 1 file changed, 46 insertions(+), 14 deletions(-) diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index fbb42e54947a..70713e4b07dc 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -63,6 +63,7 @@ struct sof_es8336_private { struct snd_soc_jack jack; struct list_head hdmi_pcm_list; bool speaker_en; + struct delayed_work pcm_pop_work; }; struct sof_hdmi_pcm { @@ -111,6 +112,46 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk headset at mic1 port enabled\n"); } +static void pcm_pop_work_events(struct work_struct *work) +{ + struct sof_es8336_private *priv = + container_of(work, struct sof_es8336_private, pcm_pop_work.work); + + gpiod_set_value_cansleep(priv->gpio_speakers, priv->speaker_en); + + if (quirk & SOF_ES8336_HEADPHONE_GPIO) + gpiod_set_value_cansleep(priv->gpio_headphone, priv->speaker_en); + +} + +static int sof_8336_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct sof_es8336_private *priv = snd_soc_card_get_drvdata(card); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + break; + + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + if (priv->speaker_en == false) + if (substream->stream == 0) { + cancel_delayed_work(&priv->pcm_pop_work); + gpiod_set_value_cansleep(priv->gpio_speakers, true); + } + break; + default: + return -EINVAL; + } + + return 0; +} + static int sof_es8316_speaker_power_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -122,19 +163,7 @@ static int sof_es8316_speaker_power_event(struct snd_soc_dapm_widget *w, priv->speaker_en = !SND_SOC_DAPM_EVENT_ON(event); - if (SND_SOC_DAPM_EVENT_ON(event)) - msleep(70); - - gpiod_set_value_cansleep(priv->gpio_speakers, priv->speaker_en); - - if (!(quirk & SOF_ES8336_HEADPHONE_GPIO)) - return 0; - - if (SND_SOC_DAPM_EVENT_ON(event)) - msleep(70); - - gpiod_set_value_cansleep(priv->gpio_headphone, priv->speaker_en); - + queue_delayed_work(system_wq, &priv->pcm_pop_work, msecs_to_jiffies(70)); return 0; } @@ -344,6 +373,7 @@ static int sof_es8336_hw_params(struct snd_pcm_substream *substream, /* machine stream operations */ static struct snd_soc_ops sof_es8336_ops = { .hw_params = sof_es8336_hw_params, + .trigger = sof_8336_trigger, }; static struct snd_soc_dai_link_component platform_component[] = { @@ -723,7 +753,8 @@ static int sof_es8336_probe(struct platform_device *pdev) } INIT_LIST_HEAD(&priv->hdmi_pcm_list); - + INIT_DELAYED_WORK(&priv->pcm_pop_work, + pcm_pop_work_events); snd_soc_card_set_drvdata(card, priv); if (mach->mach_params.dmic_num > 0) { @@ -752,6 +783,7 @@ static int sof_es8336_remove(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); struct sof_es8336_private *priv = snd_soc_card_get_drvdata(card); + cancel_delayed_work(&priv->pcm_pop_work); gpiod_put(priv->gpio_speakers); device_remove_software_node(priv->codec_dev); put_device(priv->codec_dev); -- cgit v1.2.1 From 7d945b046be3d2605dbb1806e73095aadd7ae129 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Wed, 9 Nov 2022 18:08:49 +0100 Subject: ASoC: stm32: dfsdm: manage cb buffers cleanup Ensure that resources allocated by iio_channel_get_all_cb() are released on driver unbind. Signed-off-by: Olivier Moysan Link: https://lore.kernel.org/r/20221109170849.273719-1-olivier.moysan@foss.st.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_adfsdm.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c index 643fc8a17018..837c1848d9bf 100644 --- a/sound/soc/stm/stm32_adfsdm.c +++ b/sound/soc/stm/stm32_adfsdm.c @@ -304,6 +304,11 @@ static int stm32_adfsdm_dummy_cb(const void *data, void *private) return 0; } +static void stm32_adfsdm_cleanup(void *data) +{ + iio_channel_release_all_cb(data); +} + static struct snd_soc_component_driver stm32_adfsdm_soc_platform = { .open = stm32_adfsdm_pcm_open, .close = stm32_adfsdm_pcm_close, @@ -350,6 +355,12 @@ static int stm32_adfsdm_probe(struct platform_device *pdev) if (IS_ERR(priv->iio_cb)) return PTR_ERR(priv->iio_cb); + ret = devm_add_action_or_reset(&pdev->dev, stm32_adfsdm_cleanup, priv->iio_cb); + if (ret < 0) { + dev_err(&pdev->dev, "Unable to add action\n"); + return ret; + } + component = devm_kzalloc(&pdev->dev, sizeof(*component), GFP_KERNEL); if (!component) return -ENOMEM; -- cgit v1.2.1 From 3ca507bf99611c82dafced73e921c1b10ee12869 Mon Sep 17 00:00:00 2001 From: Chancel Liu Date: Wed, 9 Nov 2022 20:13:54 +0800 Subject: ASoC: wm8962: Wait for updated value of WM8962_CLOCKING1 register DSPCLK_DIV field in WM8962_CLOCKING1 register is used to generate correct frequency of LRCLK and BCLK. Sometimes the read-only value can't be updated timely after enabling SYSCLK. This results in wrong calculation values. Delay is introduced here to wait for newest value from register. The time of the delay should be at least 500~1000us according to test. Signed-off-by: Chancel Liu Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20221109121354.123958-1-chancel.liu@nxp.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index b4b4355c6728..b901e4c65e8a 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2503,6 +2503,14 @@ static void wm8962_configure_bclk(struct snd_soc_component *component) snd_soc_component_update_bits(component, WM8962_CLOCKING2, WM8962_SYSCLK_ENA_MASK, WM8962_SYSCLK_ENA); + /* DSPCLK_DIV field in WM8962_CLOCKING1 register is used to generate + * correct frequency of LRCLK and BCLK. Sometimes the read-only value + * can't be updated timely after enabling SYSCLK. This results in wrong + * calculation values. Delay is introduced here to wait for newest + * value from register. The time of the delay should be at least + * 500~1000us according to test. + */ + usleep_range(500, 1000); dspclk = snd_soc_component_read(component, WM8962_CLOCKING1); if (snd_soc_component_get_bias_level(component) != SND_SOC_BIAS_ON) -- cgit v1.2.1 From 7c0f8f1462c9edeaa202a2cbea1bde0960434b09 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Thu, 10 Nov 2022 09:44:06 +0100 Subject: ASoC: stm32: i2s: remove irqf_oneshot flag The IRQF_ONESHOT flag allows to ensure that the interrupt is not unmasked after the hard interrupt context handler has been executed and the thread has been woken. The interrupt line is unmasked after the thread handler function has been executed. The STM32 I2S driver does not implement a threaded IRQ handler. So, the IRQF_ONESHOT flag is not useful in I2S driver. Remove this flag to allow the interrupt routine to be managed as a thread in RT mode. Signed-off-by: Olivier Moysan Link: https://lore.kernel.org/r/20221110084406.287117-1-olivier.moysan@foss.st.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c index ce7f6942308f..f3dd9f8e621c 100644 --- a/sound/soc/stm/stm32_i2s.c +++ b/sound/soc/stm/stm32_i2s.c @@ -1077,7 +1077,7 @@ static int stm32_i2s_parse_dt(struct platform_device *pdev, if (irq < 0) return irq; - ret = devm_request_irq(&pdev->dev, irq, stm32_i2s_isr, IRQF_ONESHOT, + ret = devm_request_irq(&pdev->dev, irq, stm32_i2s_isr, 0, dev_name(&pdev->dev), i2s); if (ret) { dev_err(&pdev->dev, "irq request returned %d\n", ret); -- cgit v1.2.1 From 37882100cd0629d830db430a8cee0b724fe1fea3 Mon Sep 17 00:00:00 2001 From: Junxiao Chang Date: Thu, 10 Nov 2022 07:40:23 +0800 Subject: ASoC: hdac_hda: fix hda pcm buffer overflow issue When KASAN is enabled, below log might be dumped with Intel EHL hardware: [ 48.583597] ================================================================== [ 48.585921] BUG: KASAN: slab-out-of-bounds in hdac_hda_dai_hw_params+0x20a/0x22b [snd_soc_hdac_hda] [ 48.587995] Write of size 4 at addr ffff888103489708 by task pulseaudio/759 [ 48.589237] CPU: 2 PID: 759 Comm: pulseaudio Tainted: G U E 5.15.71-intel-ese-standard-lts #9 [ 48.591272] Hardware name: Intel Corporation Elkhart Lake Embedded Platform/ElkhartLake LPDDR4x T3 CRB, BIOS EHLSFWI1.R00.4251.A01.2206130432 06/13/2022 [ 48.593010] Call Trace: [ 48.593648] [ 48.593852] dump_stack_lvl+0x34/0x48 [ 48.594404] print_address_description.constprop.0+0x1f/0x140 [ 48.595174] ? hdac_hda_dai_hw_params+0x20a/0x22b [snd_soc_hdac_hda] [ 48.595868] ? hdac_hda_dai_hw_params+0x20a/0x22b [snd_soc_hdac_hda] [ 48.596519] kasan_report.cold+0x7f/0x11b [ 48.597003] ? hdac_hda_dai_hw_params+0x20a/0x22b [snd_soc_hdac_hda] [ 48.597885] hdac_hda_dai_hw_params+0x20a/0x22b [snd_soc_hdac_hda] HDAC_LAST_DAI_ID is last index id, pcm buffer array size should be +1 to avoid out of bound access. Fixes: 608b8c36c371 ("ASoC: hdac_hda: add support for HDMI/DP as a HDA codec") Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Signed-off-by: Junxiao Chang Signed-off-by: Furong Zhou Link: https://lore.kernel.org/r/20221109234023.3111035-1-junxiao.chang@intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/hdac_hda.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/hdac_hda.h b/sound/soc/codecs/hdac_hda.h index fc19c34ca00e..b65560981abb 100644 --- a/sound/soc/codecs/hdac_hda.h +++ b/sound/soc/codecs/hdac_hda.h @@ -14,7 +14,7 @@ enum { HDAC_HDMI_1_DAI_ID, HDAC_HDMI_2_DAI_ID, HDAC_HDMI_3_DAI_ID, - HDAC_LAST_DAI_ID = HDAC_HDMI_3_DAI_ID, + HDAC_DAI_ID_NUM }; struct hdac_hda_pcm { @@ -24,7 +24,7 @@ struct hdac_hda_pcm { struct hdac_hda_priv { struct hda_codec *codec; - struct hdac_hda_pcm pcm[HDAC_LAST_DAI_ID]; + struct hdac_hda_pcm pcm[HDAC_DAI_ID_NUM]; bool need_display_power; }; -- cgit v1.2.1 From 0bb8e9b36b5b7f2e77892981ff6c27ee831d8026 Mon Sep 17 00:00:00 2001 From: Detlev Casanova Date: Thu, 10 Nov 2022 14:06:12 -0500 Subject: ASoC: sgtl5000: Reset the CHIP_CLK_CTRL reg on remove Since commit bf2aebccddef ("ASoC: sgtl5000: Fix noise on shutdown/remove"), the device power control registers are reset when the driver is removed/shutdown. This is an issue when the device is configured to use the PLL clock. The device will stop responding if it is still configured to use the PLL clock but the PLL clock is powered down. When rebooting linux, the probe function will show: sgtl5000 0-000a: Error reading chip id -11 Make sure that the CHIP_CLK_CTRL is reset to its default value before powering down the device. Fixes: bf2aebccddef ("ASoC: sgtl5000: Fix noise on shutdown/remove") Signed-off-by: Detlev Casanova Reviewed-by: Fabio Estevam Link: https://lore.kernel.org/r/20221110190612.1341469-1-detlev.casanova@collabora.com Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 4b2135eba74d..a916f4619ea3 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1794,6 +1794,7 @@ static void sgtl5000_i2c_remove(struct i2c_client *client) { struct sgtl5000_priv *sgtl5000 = i2c_get_clientdata(client); + regmap_write(sgtl5000->regmap, SGTL5000_CHIP_CLK_CTRL, SGTL5000_CHIP_CLK_CTRL_DEFAULT); regmap_write(sgtl5000->regmap, SGTL5000_CHIP_DIG_POWER, SGTL5000_DIG_POWER_DEFAULT); regmap_write(sgtl5000->regmap, SGTL5000_CHIP_ANA_POWER, SGTL5000_ANA_POWER_DEFAULT); -- cgit v1.2.1 From 39bd801d6908900e9ab0cdc2655150f95ddd4f1a Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Fri, 4 Nov 2022 13:22:13 +0000 Subject: ASoC: soc-pcm: Don't zero TDM masks in __soc_pcm_open() The DAI tx_mask and rx_mask are set by snd_soc_dai_set_tdm_slot() and used by later code that depends on the TDM settings. So __soc_pcm_open() should not be obliterating those mask values. The code in __soc_pcm_hw_params() uses these masks to calculate the active channels so that only the AIF_IN/AIF_OUT widgets for the active TDM slots are enabled. The zeroing of the masks in __soc_pcm_open() disables this functionality so all AIF widgets were enabled even for channels that are not assigned to a TDM slot. Signed-off-by: Richard Fitzgerald Fixes: 2e5894d73789 ("ASoC: pcm: Add support for DAI multicodec") Link: https://lore.kernel.org/r/20221104132213.121847-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index fb87d6d23408..8ceded22a3c4 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -822,11 +822,6 @@ static int __soc_pcm_open(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_startup(dai, substream); if (ret < 0) goto err; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dai->tx_mask = 0; - else - dai->rx_mask = 0; } /* Dynamic PCM DAI links compat checks use dynamic capabilities */ -- cgit v1.2.1 From c7d7d4e7bb1290cc473610b0bb96d9fa606d00e7 Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Wed, 16 Nov 2022 17:03:18 +0800 Subject: ASoC: rt711-sdca: fix the latency time of clock stop prepare state machine transitions Due to the hardware behavior, it takes some time for CBJ detection/impedance sensing/de-bounce. The ClockStop_NotFinished flag will be raised until these functions are completed. In ClockStopMode0 mode case, the SdW controller might check this flag from D3 to D0 when the jack detection interrupt happened. Signed-off-by: Shuming Fan Link: https://lore.kernel.org/r/20221116090318.5017-1-shumingf@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt711-sdca-sdw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt711-sdca-sdw.c b/sound/soc/codecs/rt711-sdca-sdw.c index 4120842fe699..88a8392a58ed 100644 --- a/sound/soc/codecs/rt711-sdca-sdw.c +++ b/sound/soc/codecs/rt711-sdca-sdw.c @@ -230,7 +230,7 @@ static int rt711_sdca_read_prop(struct sdw_slave *slave) } /* set the timeout values */ - prop->clk_stop_timeout = 20; + prop->clk_stop_timeout = 700; /* wake-up event */ prop->wake_capable = 1; -- cgit v1.2.1 From 60591bbf6d5eb44f275eb733943b7757325c1b60 Mon Sep 17 00:00:00 2001 From: Jiasheng Jiang Date: Wed, 16 Nov 2022 16:25:08 +0800 Subject: ASoC: max98373: Add checks for devm_kcalloc As the devm_kcalloc may return NULL pointer, it should be better to check the return value in order to avoid NULL poineter dereference. Fixes: 349dd23931d1 ("ASoC: max98373: don't access volatile registers in bias level off") Signed-off-by: Jiasheng Jiang Link: https://lore.kernel.org/r/20221116082508.17418-1-jiasheng@iscas.ac.cn Signed-off-by: Mark Brown --- sound/soc/codecs/max98373-i2c.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/max98373-i2c.c b/sound/soc/codecs/max98373-i2c.c index 3e04c7f0cce4..ec0905df65d1 100644 --- a/sound/soc/codecs/max98373-i2c.c +++ b/sound/soc/codecs/max98373-i2c.c @@ -549,6 +549,10 @@ static int max98373_i2c_probe(struct i2c_client *i2c) max98373->cache = devm_kcalloc(&i2c->dev, max98373->cache_num, sizeof(*max98373->cache), GFP_KERNEL); + if (!max98373->cache) { + ret = -ENOMEM; + return ret; + } for (i = 0; i < max98373->cache_num; i++) max98373->cache[i].reg = max98373_i2c_cache_reg[i]; -- cgit v1.2.1 From f5f8ad3fcdc49e4d794973007525ed864f93f3fb Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 17 Nov 2022 17:21:20 -0600 Subject: ASoC: SOF: dai: move AMD_HS to end of list to restore backwards-compatibility The addition of AMD_HS breaks Mediatek platforms by using an index previously allocated to Mediatek. This is a backwards-compatibility issue and needs to be fixed. All firmware released by AMD needs to be re-generated and re-distributed. Fixes: ed2562c64b4f ("ASoC: SOF: Adding amd HS functionality to the sof core") Link: https://github.com/thesofproject/sof/issues/6615 Link: https://lore.kernel.org/alsa-devel/36a45c7a-820a-7675-d740-c0e83ae2c417@collabora.com/ Reported-by: AngeloGioacchino Del Regno Reviewed-by: AngeloGioacchino Del Regno Reviewed-by: Daniel Baluta Reviewed-by: Basavaraj Hiregoudar Reviewed-by: V sujith kumar Reddy Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20221117232120.112639-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/sound/sof/dai.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/sof/dai.h b/include/sound/sof/dai.h index 83fd81c82e4c..9fbd3832bcdc 100644 --- a/include/sound/sof/dai.h +++ b/include/sound/sof/dai.h @@ -84,8 +84,8 @@ enum sof_ipc_dai_type { SOF_DAI_AMD_BT, /**< AMD ACP BT*/ SOF_DAI_AMD_SP, /**< AMD ACP SP */ SOF_DAI_AMD_DMIC, /**< AMD ACP DMIC */ - SOF_DAI_AMD_HS, /**< Amd HS */ SOF_DAI_MEDIATEK_AFE, /**< Mediatek AFE */ + SOF_DAI_AMD_HS, /**< Amd HS */ }; /* general purpose DAI configuration */ -- cgit v1.2.1 From db8f91d424fe0ea6db337aca8bc05908bbce1498 Mon Sep 17 00:00:00 2001 From: Srinivasa Rao Mandadapu Date: Tue, 22 Nov 2022 12:01:13 +0530 Subject: ASoC: soc-pcm: Add NULL check in BE reparenting Add NULL check in dpcm_be_reparent API, to handle kernel NULL pointer dereference error. The issue occurred in fuzzing test. Signed-off-by: Srinivasa Rao Mandadapu Link: https://lore.kernel.org/r/1669098673-29703-1-git-send-email-quic_srivasam@quicinc.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8ceded22a3c4..35a16c3f9591 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1247,6 +1247,8 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, return; be_substream = snd_soc_dpcm_get_substream(be, stream); + if (!be_substream) + return; for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) -- cgit v1.2.1 From f33bcc506050f89433a52a3052054d4ebd37b1c1 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 25 Nov 2022 16:23:47 +0000 Subject: ASoC: ops: Correct bounds check for second channel on SX controls Currently the check against the max value for the control is being applied after the value has had the minimum applied and been masked. But the max value simply indicates the number of volume levels on an SX control, and as such should just be applied on the raw value. Fixes: 97eea946b939 ("ASoC: ops: Check bounds for second channel in snd_soc_put_volsw_sx()") Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20221125162348.1288005-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/soc-ops.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 1970bda074d8..55b009d3c681 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -464,16 +464,15 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, ret = err; if (snd_soc_volsw_is_stereo(mc)) { - unsigned int val2; - - val_mask = mask << rshift; - val2 = (ucontrol->value.integer.value[1] + min) & mask; + unsigned int val2 = ucontrol->value.integer.value[1]; if (mc->platform_max && val2 > mc->platform_max) return -EINVAL; if (val2 > max) return -EINVAL; + val_mask = mask << rshift; + val2 = (val2 + min) & mask; val2 = val2 << rshift; err = snd_soc_component_update_bits(component, reg2, val_mask, -- cgit v1.2.1 From 3d1bb6cc1a654c8693a85b1d262e610196edec8b Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 25 Nov 2022 16:23:48 +0000 Subject: ASoC: cs42l51: Correct PGA Volume minimum value The table in the datasheet actually shows the volume values in the wrong order, with the two -3dB values being reversed. This appears to have caused the lower of the two values to be used in the driver when the higher should have been, correct this mixup. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20221125162348.1288005-2-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l51.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 51721edd8f53..e88d9ff95cdf 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -143,7 +143,7 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = { 0, 0xA0, 96, adc_att_tlv), SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L51_ALC_PGA_CTL, CS42L51_ALC_PGB_CTL, - 0, 0x19, 30, pga_tlv), + 0, 0x1A, 30, pga_tlv), SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0), SOC_SINGLE("Auto-Mute Switch", CS42L51_DAC_CTL, 2, 1, 0), SOC_SINGLE("Soft Ramp Switch", CS42L51_DAC_CTL, 1, 1, 0), -- cgit v1.2.1