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

[PATCH] soc: meson: fix a missing check on list iterator
by Xiaomeng Tong 27 Mar '22

27 Mar '22

The bug is here: *dai_name = dai->driver->name; For for_each_component_dais(), just like list_for_each_entry, the list iterator 'runtime' will point to a bogus position containing HEAD if the list is empty or no element is found. This case must be checked before any use of the iterator, otherwise it will lead to a invalid memory access. To fix the bug, just move the assignment into loop and return 0 when element is found, otherwise return -EINVAL; Cc: stable(a)vger.kernel.org Fixes: 6ae9ca9ce986b ("ASoC: meson: aiu: add i2s and spdif support") Signed-off-by: Xiaomeng Tong <xiam0nd.tong(a)gmail.com> --- sound/soc/meson/aiu.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c index d299a70db7e5..b52915c6f53b 100644 --- a/sound/soc/meson/aiu.c +++ b/sound/soc/meson/aiu.c @@ -61,14 +61,14 @@ int aiu_of_xlate_dai_name(struct snd_soc_component *component, return -EINVAL; for_each_component_dais(component, dai) { - if (id == 0) - break; + if (id == 0) { + *dai_name = dai->driver->name; + return 0; + } id--; } - *dai_name = dai->driver->name; - - return 0; + return -EINVAL; } static int aiu_cpu_of_xlate_dai_name(struct snd_soc_component *component, -- 2.17.1

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[PATCH] cs423x: cs4236: fix an incorrect NULL check on list iterator
by Xiaomeng Tong 27 Mar '22

27 Mar '22

The bug is here: err = snd_card_cs423x_pnp(dev, card->private_data, pdev, cdev); The list iterator value 'cdev' will *always* be set and non-NULL by list_for_each_entry(), so it is incorrect to assume that the iterator value will be NULL if the list is empty or no element is found. To fix the bug, use a new variable 'iter' as the list iterator, while use the original variable 'cdev' as a dedicated pointer to point to the found element. And snd_card_cs423x_pnp() itself has NULL check for cdev. Cc: stable(a)vger.kernel.org Fixes: c2b73d1458014 ("ALSA: cs4236: cs4232 and cs4236 driver merge to solve PnP BIOS detection") Signed-off-by: Xiaomeng Tong <xiam0nd.tong(a)gmail.com> --- sound/isa/cs423x/cs4236.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index b6bdebd9ef27..10112e1bb25d 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -494,7 +494,7 @@ static int snd_cs423x_pnpbios_detect(struct pnp_dev *pdev, static int dev; int err; struct snd_card *card; - struct pnp_dev *cdev; + struct pnp_dev *cdev, *iter; char cid[PNP_ID_LEN]; if (pnp_device_is_isapnp(pdev)) @@ -510,9 +510,11 @@ static int snd_cs423x_pnpbios_detect(struct pnp_dev *pdev, strcpy(cid, pdev->id[0].id); cid[5] = '1'; cdev = NULL; - list_for_each_entry(cdev, &(pdev->protocol->devices), protocol_list) { - if (!strcmp(cdev->id[0].id, cid)) + list_for_each_entry(iter, &(pdev->protocol->devices), protocol_list) { + if (!strcmp(iter->id[0].id, cid)) { + cdev = iter; break; + } } err = snd_cs423x_card_new(&pdev->dev, dev, &card); if (err < 0) -- 2.17.1

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[PATCH] ALSA: hda/realtek: Enable headset mic on Lenovo P360
by Kai-Heng Feng 27 Mar '22

27 Mar '22

Lenovo P360 is another platform equipped with ALC897, and it needs ALC897_FIXUP_HEADSET_MIC_PIN quirk to make its headset mic work. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e88fbef57c40c..4c33cb57963db 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11152,6 +11152,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x14cd, 0x5003, "USI", ALC662_FIXUP_USI_HEADSET_MODE), SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC662_FIXUP_LENOVO_MULTI_CODECS), + SND_PCI_QUIRK(0x17aa, 0x1057, "Lenovo P360", ALC897_FIXUP_HEADSET_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x32ca, "Lenovo ThinkCentre M80", ALC897_FIXUP_HEADSET_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x32cb, "Lenovo ThinkCentre M70", ALC897_FIXUP_HEADSET_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x32cf, "Lenovo ThinkCentre M950", ALC897_FIXUP_HEADSET_MIC_PIN), -- 2.34.1

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PulseAudio only working after setting the UCM - kblda7219max on the ChromeOS Kernel, 5.10.70
by GitHub issues - opened 26 Mar '22

26 Mar '22

alsa-project/alsa-ucm-conf issue #142 was opened from MilkyDeveloper: On my `kblda7219max` Chromebook (baseboard: NAMI), internal audio works properly when the following conditions are met: * Kernel 5.10.70 (ChromeOS fork) is used (the latest ChromeOS kernel has some kind of topology-related kernel bug) * FW is copied from the ChromeOS rootfs to Linux * UCM is converted to UCM2 * Only ALSA is used However, whenever I start PulseAudio (`pulseaudio -vvvv`), audio is disabled, going to a dummy output. PulseAudio passes through probing for the UCM files but errors out in this particular spot: ``` I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_PREPARE failed (-22) I: [pulseaudio] alsa-util.c: snd_pcm_hw_params failed: Invalid argument D: [pulseaudio] alsa-util.c: Trying hw:kblda7219max,0 without SND_PCM_NO_AUTO_FORMAT ... D: [pulseaudio] alsa-util.c: Managed to open hw:kblda7219max,0 D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 21845 ms I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_PREPARE failed (-22) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_PREPARE failed (-22) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_PREPARE failed (-22) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_PREPARE failed (-22) ``` which causes it to say: ``` E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="0" name="platform-kbl_da7219_mx98357a" card_name="alsa_card.platform-kbl_da7219_mx98357a" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed. ``` My `dmesg` gets filled with: ``` Kbl Audio Port: ASoC: no backend DAIs enabled for Kbl Audio Port Kbl Audio Port: ASoC: dpcm_fe_dai_prepare() failed (-22) ``` However, once I kill PulseAudio and run `alsaucm -c kblda7219max set _verb HiFi set _enadev Speaker`, audio begins working again (but this is in pure ALSA and PA apps are not compatible) and the `no backend DAIs enabled for Kbl Audio Port` is no longer logged to `dmesg`. I think this is a UCM issue because the audio is working fine kernel-level and with ALSA, but PulseAudio seems to mess something up with the UCM and the UCM needs to be set again. Huge thanks for any help! [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8355612/… (My PulseAudio log is slightly different from the time I ran PulseAudio and got the PCM_IOCTL_PREPARE error) [pulseaudio-log.txt](https://github.com/alsa-project/alsa-ucm-conf/files/835… [HiFi.conf.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8355631/… [kblda7219max.conf.txt](https://github.com/alsa-project/alsa-ucm-conf/files/… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/142 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Conceptual bug on SoundWire probe/remove?
by Pierre-Louis Bossart 26 Mar '22

26 Mar '22

Hi, I could use feedback/guidance on a possible conceptual bug in the SoundWire probe and bus handling. When we probe a driver, the code does this: static int sdw_drv_probe(struct device *dev) { struct sdw_slave *slave = dev_to_sdw_dev(dev); struct sdw_driver *drv = drv_to_sdw_driver(dev->driver); const struct sdw_device_id *id; const char *name; int ret; /* * fw description is mandatory to bind */ if (!dev->fwnode) return -ENODEV; if (!IS_ENABLED(CONFIG_ACPI) && !dev->of_node) return -ENODEV; id = sdw_get_device_id(slave, drv); if (!id) return -ENODEV; slave->ops = drv->ops; The last line is the problematic one. If at some point, the user does an rmmod and unbinds the SoundWire codec driver, the .remove will be called and the 'drv' will no longer be valid, but we will still have a reference to drv->ops and use that pointer in the bus code, e.g. /* Update the Slave driver */ if (slave_notify && slave->ops && slave->ops->interrupt_callback) { slave_intr.sdca_cascade = sdca_cascade; slave_intr.control_port = clear; memcpy(slave_intr.port, &port_status, sizeof(slave_intr.port)); slave->ops->interrupt_callback(slave, &slave_intr); } I noodled with a potential fix in https://github.com/thesofproject/linux/pull/3534/commits/82d64fb0fd39b53226… where I force-reset this slave->ops pointer, but it is likely to be very racy. We probably need to avoid such references, or have a clean mechanism to unbind, e.g. with all commands and interrupts stopped while the codec driver .remove routine is handled. Initial error reports at https://github.com/thesofproject/linux/issues/3531 Suggestions and comments welcome, thanks! -Pierre

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[PATCH] ASoC: codecs: add support for ES8326 dts
by zhuning＠everest-semi.com 26 Mar '22

26 Mar '22

From: yangxiaohua <yangxiaohua(a)everest-semi.com> Subject: [PATCH 1/1] ASoC: codecs: add support for ES8326 dts The ES8326 codec is not compatible with ES8316 and requires a dedicated driver. This is a separate patch for ES8326 dts Signed-off-by: David Yang <yangxiaohua(a)everest-semi.com> --- .../bindings/sound/everest,es8326.yaml | 109 ++++++++++++++++++ 1 file changed, 109 insertions(+) create mode 100755 Documentation/devicetree/bindings/sound/everest,es8326.yaml diff --git a/Documentation/devicetree/bindings/sound/everest,es8326.yaml b/Documentation/devicetree/bindings/sound/everest,es8326.yaml new file mode 100755 index 000000000000..8cae9c328dbc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es8326.yaml @@ -0,0 +1,109 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/everest,es8326.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Everest ES8326 audio CODEC + +maintainers: + - David Yang <yangxiaohua(a)everest-semi.com> + +properties: + compatible: + const: everest,es8326 + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for master clock (MCLK) + + clock-names: + items: + - const: mclk + + "#sound-dai-cells": + const: 0 + + mclk-rate: + description: + clock rate of mclk + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + + everest,jack-pol: + description: + just the value of reg 57. Bit(3) decides whether the jack polarity is inverted. + Bit(2) decides whether the bottom on the headset is inverted. + Bit(1)/(0) decides the mic properity to be OMTP/CTIA or auto. + $ref: /schemas/types.yaml#/definitions/uint8-array + minimum: 0 + maximum: 0x0f + default: 0x0f + + everest,mic1-src: + description: + the value of reg 2A when headset plugged. + $ref: /schemas/types.yaml#/definitions/uint8-array + minimum: 0x00 + maximum: 0x77 + default: 0x22 + + everest,mic2-src: + description: + the value of reg 2A when headset unplugged. + $ref: /schemas/types.yaml#/definitions/uint8-array + minimum: 0x00 + maximum: 0x77 + default: 0x44 + + everest,amic1-src: + description: + the value of reg 23 when headset plugged. + $ref: /schemas/types.yaml#/definitions/uint8-array + minimum: 0x19 + maximum: 0x29 + default: 0x19 + + everest,amic2-src: + description: + the value of reg 23 when headset unplugged. + Set to 0x19 if dmic used. Set to 0x29 if amic used. + $ref: /schemas/types.yaml#/definitions/uint8-array + minimum: 0x19 + maximum: 0x29 + default: 0x19 + + everest,jack-detect-inverted: + description: + Defined to invert the jack detection. + $ref: /schemas/types.yaml#/definitions/flag + +required: + - compatible + - reg + - "#sound-dai-cells" + + +additionalProperties: false + +examples: + - | + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + es8326: codec@19 { + compatible = "everest,es8326"; + reg = <0x19>; + clocks = <&clks 10>; + clock-names = "mclk"; + #sound-dai-cells = <0>; + mic1-src = [22]; + mic2-src = [44]; + amic1-src = [19]; + amic2-src = [19]; + jack-pol = [0e]; + }; + }; -- 2.17.1

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Add UCM for PinePhone
by GitHub pull_request - edited 26 Mar '22

26 Mar '22

alsa-project/alsa-ucm-conf pull request #134 was edited from tomfitzhenry: Fixes https://github.com/alsa-project/alsa-ucm-conf/issues/124. This PR adds support for the https://wiki.pine64.org/index.php/PinePhone This PR contains: * This UCM config copied from https://gitlab.com/pine64-org/pine64-alsa-ucm/, which was developed by many developers ( https://gitlab.com/pine64-org/pine64-alsa-ucm/-/merge_requests/1 ) and is BSD-3-Clause licensed. * Fix paths to be relative to alsa-ucm-conf root. * Some refactoring of the UCM config, incl. removing duplicate directives, and moving volume directives to BootSequence. * Updating the routing/playback/boost volumes, in response to comments on this PR. I have tested: * HiFi * Record audio via the headset. * Record audio via the internal mic. * Play audio on the headphones. * Play audio on the speaker. * Play audio on the internal speaker. * Voice Call * Speak on the headset. * Speak on the internal mic. * Listen on the headphones. * Listen on the speaker. This induces an echo, and given the audio routing is entirely on chip, and the chip doesn't advertise echo cancellation, I don't see how this is avoidable. * Listen on the internal speaker. See also: * `alsa-info`: https://alsa-project.org/db/?f=5e331f5ef10bd167b3bd7640bfcf0e879100f80d * `pa-info`: [pa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8015083/pa… * https://gitlab.com/pine64-org/pine64-alsa-ucm/-/issues/3 * Schematics/code * The device-tree configuration has had a unique audio card name since 5.14: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/arc… * http://files.pine64.org/doc/PinePhone/PinePhone%20v1.2%20Released%20Schemat… * https://xnux.eu/devices/feature/audio-pp.html * https://github.com/torvalds/linux/blob/master/arch/arm64/boot/dts/allwinner… * https://github.com/torvalds/linux/blob/master/arch/arm64/boot/dts/allwinner… * https://github.com/torvalds/linux/blob/master/sound/soc/sunxi/sun50i-codec-… * https://files.pine64.org/doc/datasheet/pine64/Allwinner_A64_User_Manual_V1.… * https://static.chipdip.ru/lib/126/DOC001126567.pdf Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/134 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/134.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Add UCM for PinePhone
by GitHub pull_request - edited 26 Mar '22

26 Mar '22

alsa-project/alsa-ucm-conf pull request #134 was edited from tomfitzhenry: Fixes https://github.com/alsa-project/alsa-ucm-conf/issues/124. This PR adds support for the https://wiki.pine64.org/index.php/PinePhone This PR contains: * This UCM config copied from https://gitlab.com/pine64-org/pine64-alsa-ucm/, which was developed by many developers ( https://gitlab.com/pine64-org/pine64-alsa-ucm/-/merge_requests/1 ) and is BSD-3-Clause licensed. * Fix paths to be relative to alsa-ucm-conf root. * Some refactoring of the UCM config, incl. removing duplicate directives, and moving volume directives to BootSequence. * Updating the playback/boost volumes, in response to comments on this PR. I have tested: * HiFi * Record audio via the headset. * Record audio via the internal mic. * Play audio on the headphones. * Play audio on the speaker. * Play audio on the internal speaker. * Voice Call * Speak on the headset. * Speak on the internal mic. * Listen on the headphones. * Listen on the speaker. This induces an echo, and given the audio routing is entirely on chip, and the chip doesn't advertise echo cancellation, I don't see how this is avoidable. * Listen on the internal speaker. See also: * `alsa-info`: https://alsa-project.org/db/?f=5e331f5ef10bd167b3bd7640bfcf0e879100f80d * `pa-info`: [pa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8015083/pa… * https://gitlab.com/pine64-org/pine64-alsa-ucm/-/issues/3 * Schematics/code * The device-tree configuration has had a unique audio card name since 5.14: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/arc… * http://files.pine64.org/doc/PinePhone/PinePhone%20v1.2%20Released%20Schemat… * https://xnux.eu/devices/feature/audio-pp.html * https://github.com/torvalds/linux/blob/master/arch/arm64/boot/dts/allwinner… * https://github.com/torvalds/linux/blob/master/arch/arm64/boot/dts/allwinner… * https://github.com/torvalds/linux/blob/master/sound/soc/sunxi/sun50i-codec-… * https://files.pine64.org/doc/datasheet/pine64/Allwinner_A64_User_Manual_V1.… * https://static.chipdip.ru/lib/126/DOC001126567.pdf Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/134 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/134.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH -next] ASoC: SOF: Intel: Fix build error without SND_SOC_SOF_PCI_DEV
by Zheng Bin 25 Mar '22

25 Mar '22

If SND_SOC_SOF_PCI_DEV is n, bulding fails: sound/soc/sof/intel/pci-tng.o:(.data+0x1c0): undefined reference to `sof_pci_probe' sound/soc/sof/intel/pci-tng.o:(.data+0x1c8): undefined reference to `sof_pci_remove' sound/soc/sof/intel/pci-tng.o:(.data+0x1e0): undefined reference to `sof_pci_shutdown' sound/soc/sof/intel/pci-tng.o:(.data+0x290): undefined reference to `sof_pci_pm' Make SND_SOC_SOF_MERRIFIELD select SND_SOC_SOF_PCI_DEV to fix this. Fixes: 8d4ba1be3d22 ("ASoC: SOF: pci: split PCI into different drivers") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Zheng Bin <zhengbin13(a)huawei.com> --- sound/soc/sof/intel/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index b53f216d4ecc..172419392b33 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -84,6 +84,7 @@ if SND_SOC_SOF_PCI config SND_SOC_SOF_MERRIFIELD tristate "SOF support for Tangier/Merrifield" default SND_SOC_SOF_PCI + select SND_SOC_SOF_PCI_DEV select SND_SOC_SOF_INTEL_ATOM_HIFI_EP help This adds support for Sound Open Firmware for Intel(R) platforms -- 2.31.1

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[PATCH] ASoC: rt5682s: Separate the regulator consumer controls
by derek.fang＠realtek.com 25 Mar '22

25 Mar '22

From: Derek Fang <derek.fang(a)realtek.com> Control the regulators separately instead of using regulator_bulk to accord to the timing request in the datasheet. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt5682s.c | 26 +++++++++++++++++++++----- sound/soc/codecs/rt5682s.h | 6 +++++- 2 files changed, 26 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index 1cba8ec7cedb..b8bf9aede29c 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -42,8 +42,8 @@ static const struct rt5682s_platform_data i2s_default_platform_data = { }; static const char *rt5682s_supply_names[RT5682S_NUM_SUPPLIES] = { - "AVDD", - "MICVDD", + [RT5682S_SUPPLY_AVDD] = "AVDD", + [RT5682S_SUPPLY_MICVDD] = "MICVDD", }; static const struct reg_sequence patch_list[] = { @@ -3025,8 +3025,17 @@ static struct snd_soc_dai_driver rt5682s_dai[] = { static void rt5682s_i2c_disable_regulators(void *data) { struct rt5682s_priv *rt5682s = data; + int ret; + + ret = regulator_disable(rt5682s->supplies[RT5682S_SUPPLY_AVDD].consumer); + if (ret) + pr_err("%s: Failed to disable supply AVDD: %d\n", __func__, ret); - regulator_bulk_disable(ARRAY_SIZE(rt5682s->supplies), rt5682s->supplies); + usleep_range(1000, 1500); + + ret = regulator_disable(rt5682s->supplies[RT5682S_SUPPLY_MICVDD].consumer); + if (ret) + pr_err("%s: Failed to disable supply MICVDD: %d\n", __func__, ret); } static int rt5682s_i2c_probe(struct i2c_client *i2c, @@ -3071,9 +3080,16 @@ static int rt5682s_i2c_probe(struct i2c_client *i2c, if (ret) return ret; - ret = regulator_bulk_enable(ARRAY_SIZE(rt5682s->supplies), rt5682s->supplies); + ret = regulator_enable(rt5682s->supplies[RT5682S_SUPPLY_MICVDD].consumer); + if (ret) { + dev_err(&i2c->dev, "Failed to enable supply MICVDD: %d\n", ret); + return ret; + } + usleep_range(1000, 1500); + + ret = regulator_enable(rt5682s->supplies[RT5682S_SUPPLY_AVDD].consumer); if (ret) { - dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret); + dev_err(&i2c->dev, "Failed to enable supply AVDD: %d\n", ret); return ret; } diff --git a/sound/soc/codecs/rt5682s.h b/sound/soc/codecs/rt5682s.h index 397a2531b6f6..7353831c73dd 100644 --- a/sound/soc/codecs/rt5682s.h +++ b/sound/soc/codecs/rt5682s.h @@ -1434,7 +1434,11 @@ struct pll_calc_map { bool sel_ps; }; -#define RT5682S_NUM_SUPPLIES 2 +enum { + RT5682S_SUPPLY_AVDD, + RT5682S_SUPPLY_MICVDD, + RT5682S_NUM_SUPPLIES, +}; struct rt5682s_priv { struct snd_soc_component *component; -- 2.17.1

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