May 2020 - Alsa-devel - mailman.alsa-project.org

snd_dice: Clicking artifacts with TC StudioKonnekt 48
by Mathias Buhr 10 May '20

10 May '20

Hi, thanks to this group, both of my firewire interfaces are (almost) working! Big thank you! While the TC Konnekt 24D works fine (playback and recording), the StudioKonnekt 48 produces clicking artifacts during playback when using snd_dice. This interface is working flawless on Windows and on a Jack/FFADO combination. This artifact occurs in all use cases (alsa via aplay, pulseaudio and jack) and does not seem to be in recorded streams. After recording the playback with another device, it looks like the level of the artifact is scaling with the signal and its interval is interestingly 256ms. Is there anything I can do to further debug this issue? Capture firewire packets? I would love to get this device fully working. I'm using kernel version 5.5.4 but this issue has been there for a long time. Thanks, Mathias

3 9

[PATCH] ALSA: hda/realtek: Add quirk for Samsung Notebook
by Mike Pozulp 10 May '20

10 May '20

Some models of the Samsung Notebook 9 have very quiet and distorted headphone output. This quirk changes the VREF value of the ALC298 codec NID 0x1a from default HIZ to new 100. Signed-off-by: Mike Pozulp <pozulp.kernel(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 63e1a56f705b..9b14ed5c75d7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5923,6 +5923,7 @@ enum { ALC294_FIXUP_ASUS_DUAL_SPK, ALC285_FIXUP_THINKPAD_HEADSET_JACK, ALC294_FIXUP_ASUS_HPE, + ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, }; static const struct hda_fixup alc269_fixups[] = { @@ -7061,6 +7062,13 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC }, + [ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc5 }, + { } + }, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7336,6 +7344,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE), + SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), + SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), #if 0 /* Below is a quirk table taken from the old code. -- 2.24.1

2 1

crackling microphone
by GitHub issues - reopened 09 May '20

09 May '20

alsa-project/alsa-plugins issue #10 was reopened from riddicc: Hi! When I send a 800Hz sine wave with rate 12345 to pcm.outmic and when I arecord from pcm.micend at rate 8000 with 1 channel ([/etc/asound.conf](https://github.com/alsa-project/alsa-plugins/files/4597703/etc-asound-conf.txt)), then I get a lot of tiny little gaps (see attachment) and a lot of messages like "overrun!!! (at least 0.014 ms long)" (the "ms" vary from 0.008 to 0.043; and arecord continues to record) and sometimes even this error: "arecord: xrun:1672: read/write error, state = RUNNING" (it causes arecord to terminate). 1. > speaker-test -D outmic -t sine -f 800 -c2 -r 12345 2. > arecord -D micend -r 8000 -c 1 -f S16_LE /tmp/bla.wav twinkle makes even bigger gaps, when I let it use pcm.micend as microphone... But with hw:Generic,0,0 as mic device twinkle works fine... https://github.com/LubosD/twinkle/issues/207 "samplerate_best" and "samplerate_order" seems to make it worse... How can I fix that? Thx. Bye. ![audacity screenshot](https://user-images.githubusercontent.com/33246439/81384628-7ad… Issue URL : https://github.com/alsa-project/alsa-plugins/issues/10 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[alsa-devel] [alsa-lib][RFC PATCH] ucm: reset config id of condition items
by libin.yang＠linux.intel.com 09 May '20

09 May '20

From: Libin Yang <libin.yang(a)intel.com> UCMv2 supports "If" statement and will merge the same items with compound_merge(). If the items have the same id, it will fail to add the config items. And the id of the item in an array is automatically generated with the increased number. It is probably that some items to be merged have the same id. Let's add prefix in the id to avoid such situation. For example: If.seq1 { Condition { Type ControlExists Control "name='PGA1.0 1 Master Playback Volume'" } True { EnableSequence [ cset "name='PGA1.0 1 Master Playback Volume' 50" ] } } If.seq2 { Condition { Type ControlExists Control "name='PGA2.0 2 Master Playback Volume'" } True { EnableSequence [ cset "name='PGA2.0 2 Master Playback Volume' 50" ] } } If.seq3 { Condition { Type ControlExists Control "name='PGA3.0 3 Master Playback Volume'" } True { EnableSequence [ cset "name='PGA3.0 3 Master Playback Volume' 50" ] } } If seq1, seq2 and seq3 conditions are true, UCM will fail to initialize. This patch rename the config id to avoid the same id conflict. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- include/conf.h | 2 +- src/conf.c | 21 +++++++++++++++++++++ src/ucm/ucm_cond.c | 28 ++++++++++++++++++++++++---- 3 files changed, 46 insertions(+), 5 deletions(-) diff --git a/include/conf.h b/include/conf.h index 456b272..adb3d84 100644 --- a/include/conf.h +++ b/include/conf.h @@ -139,7 +139,7 @@ int snd_config_imake_safe_string(snd_config_t **config, const char *key, const c int snd_config_imake_pointer(snd_config_t **config, const char *key, const void *ptr); snd_config_type_t snd_config_get_type(const snd_config_t *config); - +int snd_config_of_array(const snd_config_t *config); int snd_config_set_id(snd_config_t *config, const char *id); int snd_config_set_integer(snd_config_t *config, long value); int snd_config_set_integer64(snd_config_t *config, long long value); diff --git a/src/conf.c b/src/conf.c index 50d0403..43d565b 100644 --- a/src/conf.c +++ b/src/conf.c @@ -435,6 +435,8 @@ struct _snd_config { char *id; snd_config_type_t type; int refcount; /* default = 0 */ + /* member of an array */ + int of_array; union { long integer; long long integer64; @@ -1123,6 +1125,7 @@ static int _snd_config_make(snd_config_t **config, char **id, snd_config_type_t *id = NULL; } n->type = type; + n->of_array = 0; if (type == SND_CONFIG_TYPE_COMPOUND) INIT_LIST_HEAD(&n->u.compound.fields); *config = n; @@ -1316,6 +1319,8 @@ static int parse_array_def(snd_config_t *parent, input_t *input, int *idx, int s default: unget_char(c, input); err = parse_value(&n, parent, input, &id, skip); + /* this is a member of an array */ + n->of_array = 1; if (err < 0) goto __end; break; @@ -1784,6 +1789,22 @@ snd_config_type_t snd_config_get_type(const snd_config_t *config) } /** + * \brief Returns the of_array of a configuration node. + * \param config Handle to the configuration node. + * \return of_array of the node + * + * The returned value indicates whether the node is a member of an array. + * + * \par Conforming to: + * LSB 3.2 + */ +int snd_config_of_array(const snd_config_t *config) +{ + assert(config); + return config->of_array; +} + +/** * \brief Returns the id of a configuration node. * \param[in] config Handle to the configuration node. * \param[out] id The function puts the pointer to the id string at the diff --git a/src/ucm/ucm_cond.c b/src/ucm/ucm_cond.c index 22b418d..725a69e 100644 --- a/src/ucm/ucm_cond.c +++ b/src/ucm/ucm_cond.c @@ -347,14 +347,15 @@ static void config_dump(snd_config_t *cfg) } #endif -static int compound_merge(const char *id, +static int compound_merge(const char *id, const char *cname, snd_config_t *dst, snd_config_t *src, snd_config_t *before, snd_config_t *after) { snd_config_iterator_t i, next; snd_config_t *n, *_before = NULL, *_after = NULL; const char *s; - int err; + char s1[32]; + int err, cnt; if (snd_config_get_type(src) != SND_CONFIG_TYPE_COMPOUND) { uc_error("compound type expected for If True/False block"); @@ -387,8 +388,22 @@ static int compound_merge(const char *id, return -EINVAL; } + cnt = 0; snd_config_for_each(i, next, src) { n = snd_config_iterator_entry(i); + /* + * If n is an array member, n->id is automatically generated. + * It is prossible that n->id is used by other array member, + * which will be merged with this one. So let's add prefix + * to the id to avoid the conflict. + */ + if (snd_config_of_array(n)) { + err = snd_config_get_id(n, &s); + if (err < 0) + return err; /* FIXME: this will never happen */ + snprintf(s1, sizeof(s1), "%s-%d-%s", cname, cnt++, s); + snd_config_set_id(n, (const char *)s1); + } err = snd_config_remove(n); if (err < 0) return err; @@ -422,7 +437,7 @@ int uc_mgr_evaluate_condition(snd_use_case_mgr_t *uc_mgr, { snd_config_iterator_t i, i2, next, next2; snd_config_t *a, *n, *n2, *parent2, *before, *after; - const char *id; + const char *id, *cname; int err; if (uc_mgr->conf_format < 2) { @@ -437,6 +452,10 @@ int uc_mgr_evaluate_condition(snd_use_case_mgr_t *uc_mgr, snd_config_for_each(i, next, cond) { n = snd_config_iterator_entry(i); + err = snd_config_get_id(n, &cname); + if (err < 0) + return err; /* FIXME: this will never happen */ + before = after = NULL; err = if_eval_one(uc_mgr, n, &a, &before, &after); if (err < 0) @@ -469,7 +488,8 @@ __add: err = snd_config_search(parent, id, &parent2); if (err == -ENOENT) goto __add; - err = compound_merge(id, parent2, n2, before, after); + err = compound_merge(id, cname, parent2, n2, + before, after); if (err < 0) return err; } -- 2.7.4

4 8

crackling microphone
by GitHub issues - reopened 09 May '20

09 May '20

alsa-project/alsa-plugins issue #10 was reopened from riddicc: Hi! When I send a 800Hz sine wave with rate 12345 to pcm.outmic and when I arecord from pcm.micend at rate 8000 with 1 channel ([/etc/asound.conf](https://github.com/alsa-project/alsa-plugins/files/4597703/etc-asound-conf.txt)), then I get a lot of tiny little gaps (see attachment) and a lot of messages like "overrun!!! (at least 0.014 ms long)" (the "ms" vary from 0.008 to 0.043; and arecord continues to record) and sometimes even this error: "arecord: xrun:1672: read/write error, state = RUNNING" (it causes arecord to terminate). 1. > speaker-test -D outmic -t sine -f 800 -c2 -r 12345 2. > arecord -D micend -r 8000 -c 1 -f S16_LE /tmp/bla.wav twinkle makes even bigger gaps, when I let it use pcm.micend as microphone... But with hw:Generic,0,0 as mic device twinkle works fine... https://github.com/LubosD/twinkle/issues/207 "samplerate_best" and "samplerate_order" seems to make it worse... How can I fix that? Thx. Bye. ![audacity screenshot](https://user-images.githubusercontent.com/33246439/81384628-7ad… Issue URL : https://github.com/alsa-project/alsa-plugins/issues/10 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH 1/2] dt-bindings: Added device tree binding for max98390
by Steve Lee 09 May '20

09 May '20

Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- .../devicetree/bindings/sound/max98390.txt | 26 +++++++++++++++++++ 1 file changed, 26 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98390.txt diff --git a/Documentation/devicetree/bindings/sound/max98390.txt b/Documentation/devicetree/bindings/sound/max98390.txt new file mode 100644 index 000000000000..147dfd88cd3b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98390.txt @@ -0,0 +1,26 @@ +Maxim Integrated MAX98390 Speaker Amplifier + +This device supports I2C. + +Required properties: + + - compatible : "maxim,max98390" + + - reg : the I2C address of the device. + +Optional properties: + + - maxim,temperature_calib + u32. The calculated temperature data was measured while doing the calibration. Data calibration : Temp / 100 * 2^12 + + - maxim,r0_calib + u32. This is r0 calibration data which was measured in factory mode. + +Example: + +codec: max98390@38 { + compatible = "maxim,max98390"; + reg = <0x38>; + maxim,temperature_calib = <1024>; + maxim,r0_calib = <0x224050>; +}; -- 2.17.1

1 0

[PATCH] ASoC: cros_ec_codec: allocate shash_desc dynamically
by Arnd Bergmann 08 May '20

08 May '20

The wov_hotword_model_put() function has multiple large variables on its stack, the largest of which is the result of SHASH_DESC_ON_STACK(). In total, this exceeds the warning limit for 32-bit architectures: sound/soc/codecs/cros_ec_codec.c:776:12: error: stack frame size of 1152 bytes in function 'wov_hotword_model_put' [-Werror,-Wframe-larger-than=] The function already has a dynamic crypto_alloc_shash() allocation, so using kmalloc() for the descriptor is correct as well and does not introduce any additional failure scenarios. With this, the stack usage of wov_hotword_model_put() gets reduced to 480 bytes in my test configuration. Fixes: b6bc07d4360d ("ASoC: cros_ec_codec: support WoV") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/cros_ec_codec.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index d3dc42aa6825..1948bc6971f6 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -108,22 +108,23 @@ static int calculate_sha256(struct cros_ec_codec_priv *priv, uint8_t *buf, uint32_t size, uint8_t *digest) { struct crypto_shash *tfm; + struct shash_desc *desc; tfm = crypto_alloc_shash("sha256", CRYPTO_ALG_TYPE_SHASH, 0); if (IS_ERR(tfm)) { dev_err(priv->dev, "can't alloc shash\n"); return PTR_ERR(tfm); } - - { - SHASH_DESC_ON_STACK(desc, tfm); - - desc->tfm = tfm; - - crypto_shash_digest(desc, buf, size, digest); - shash_desc_zero(desc); + desc = kmalloc(sizeof(*desc) + crypto_shash_descsize(tfm), GFP_KERNEL); + if (!desc) { + crypto_free_shash(tfm); + return -ENOMEM; } + desc->tfm = tfm; + crypto_shash_digest(desc, buf, size, digest); + shash_desc_zero(desc); + kfree(desc); crypto_free_shash(tfm); #ifdef DEBUG -- 2.26.0

3 2

[PATCH] ASoC: qcom: Use devm_platform_ioremap_resource_byname() to simplify code
by Tang Bin 08 May '20

08 May '20

Use devm_platform_ioremap_resource_byname() instead of platform_get_resource_byname() + devm_ioremap_resource(). Signed-off-by: Zhang Shengju <zhangshengju(a)cmss.chinamobile.com> Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/qcom/lpass-cpu.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index dbce7e92b..885c1f2e7 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -417,7 +417,6 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) { struct lpass_data *drvdata; struct device_node *dsp_of_node; - struct resource *res; struct lpass_variant *variant; struct device *dev = &pdev->dev; const struct of_device_id *match; @@ -442,9 +441,8 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) drvdata->variant = (struct lpass_variant *)match->data; variant = drvdata->variant; - res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "lpass-lpaif"); - - drvdata->lpaif = devm_ioremap_resource(&pdev->dev, res); + drvdata->lpaif = devm_platform_ioremap_resource_byname(pdev, + "lpass-lpaif") if (IS_ERR((void const __force *)drvdata->lpaif)) { dev_err(&pdev->dev, "error mapping reg resource: %ld\n", PTR_ERR((void const __force *)drvdata->lpaif)); -- 2.20.1.windows.1

3 3

[PATCH] ASoC: mxs-saif: Add variable dev to simplify code
by Tang Bin 08 May '20

08 May '20

Add variable 'dev' to make the code cleaner in the function mxs_saif_probe(). And now that the function mxs_saif_mclk_init() have defined the variables 'ret' as the error returned value, then it should be used instead in this place. Signed-off-by: Zhang Shengju <zhangshengju(a)cmss.chinamobile.com> Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/mxs/mxs-saif.c | 32 ++++++++++++++++---------------- 1 file changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index dc197883e..f4e441183 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -719,7 +719,7 @@ static int mxs_saif_mclk_init(struct platform_device *pdev) if (ret == -EEXIST) return 0; dev_err(&pdev->dev, "failed to register mclk: %d\n", ret); - return PTR_ERR(clk); + return ret; } ret = of_clk_add_provider(np, of_clk_src_simple_get, clk); @@ -732,6 +732,7 @@ static int mxs_saif_mclk_init(struct platform_device *pdev) static int mxs_saif_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; + struct device *dev = &pdev->dev; struct mxs_saif *saif; int irq, ret; struct device_node *master; @@ -739,7 +740,7 @@ static int mxs_saif_probe(struct platform_device *pdev) if (!np) return -EINVAL; - saif = devm_kzalloc(&pdev->dev, sizeof(*saif), GFP_KERNEL); + saif = devm_kzalloc(dev, sizeof(*saif), GFP_KERNEL); if (!saif) return -ENOMEM; @@ -750,7 +751,7 @@ static int mxs_saif_probe(struct platform_device *pdev) saif->id = ret; if (saif->id >= ARRAY_SIZE(mxs_saif)) { - dev_err(&pdev->dev, "get wrong saif id\n"); + dev_err(dev, "get wrong saif id\n"); return -EINVAL; } @@ -770,18 +771,17 @@ static int mxs_saif_probe(struct platform_device *pdev) saif->master_id = ret; if (saif->master_id >= ARRAY_SIZE(mxs_saif)) { - dev_err(&pdev->dev, "get wrong master id\n"); + dev_err(dev, "get wrong master id\n"); return -EINVAL; } } mxs_saif[saif->id] = saif; - saif->clk = devm_clk_get(&pdev->dev, NULL); + saif->clk = devm_clk_get(dev, NULL); if (IS_ERR(saif->clk)) { ret = PTR_ERR(saif->clk); - dev_err(&pdev->dev, "Cannot get the clock: %d\n", - ret); + dev_err(dev, "Cannot get the clock: %d\n", ret); return ret; } @@ -793,11 +793,11 @@ static int mxs_saif_probe(struct platform_device *pdev) if (irq < 0) return irq; - saif->dev = &pdev->dev; - ret = devm_request_irq(&pdev->dev, irq, mxs_saif_irq, 0, - dev_name(&pdev->dev), saif); + saif->dev = dev; + ret = devm_request_irq(dev, irq, mxs_saif_irq, 0, + dev_name(dev), saif); if (ret) { - dev_err(&pdev->dev, "failed to request irq\n"); + dev_err(dev, "failed to request irq\n"); return ret; } @@ -807,19 +807,19 @@ static int mxs_saif_probe(struct platform_device *pdev) if (saif->id == 0) { ret = mxs_saif_mclk_init(pdev); if (ret) - dev_warn(&pdev->dev, "failed to init clocks\n"); + dev_warn(dev, "failed to init clocks\n"); } - ret = devm_snd_soc_register_component(&pdev->dev, &mxs_saif_component, + ret = devm_snd_soc_register_component(dev, &mxs_saif_component, &mxs_saif_dai, 1); if (ret) { - dev_err(&pdev->dev, "register DAI failed\n"); + dev_err(dev, "register DAI failed\n"); return ret; } - ret = mxs_pcm_platform_register(&pdev->dev); + ret = mxs_pcm_platform_register(dev); if (ret) { - dev_err(&pdev->dev, "register PCM failed: %d\n", ret); + dev_err(dev, "register PCM failed: %d\n", ret); return ret; } -- 2.20.1.windows.1

2 1

[alsa-devel] [PATCH v4 00/12] ASoC: Add support to WCD9340/WCD9341 codec
by Srinivas Kandagatla 08 May '20

08 May '20

This patchset adds support to Qualcomm WCD9340/WCD9341 Codec which is a standalone Hi-Fi audio codec IC. This codec supports both I2S/I2C and SLIMbus audio interfaces. On slimbus interface it supports two data lanes; 16 Tx ports and 8 Rx ports. It has Five DACs and seven dedicated interpolators, Multibutton headset control (MBHC), Active noise cancellation, Sidetone paths, MAD (mic activity detection) and codec processing engine. It supports Class-H differential earpiece out and stereo single ended headphones out. This codec also has integrated SoundWire controller. Patchset for this is already sent for review at https://patchwork.kernel.org/cover/11185769/ This patchset has been tested on SDM845 based DragonBoard DB845c and Lenovo Yoga C630 laptop with WSA881x smart speaker amplifiers via soundwire and 4 DMICs. gpio controller patch does not have any link dependency, it can go by its own. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. Thanks, srini Changes since v3: - Added new SLIMBus common bindings suggested by Rob. - renamed microvolts to microvolt in bindings as suggested by Rob - dumped pinctrl driver and added new simple gpio driver - few cosmetic cleanups in mfd driver. Srinivas Kandagatla (12): dt-bindings: SLIMBus: add slim devices optional properties ASoC: dt-bindings: add dt bindings for WCD9340/WCD9341 audio codec mfd: wcd934x: add support to wcd9340/wcd9341 codec ASoC: wcd934x: add support to wcd9340/wcd9341 codec ASoC: wcd934x: add basic controls ASoC: wcd934x: add playback dapm widgets ASoC: wcd934x: add capture dapm widgets ASoC: wcd934x: add audio routings dt-bindings: gpio: wcd934x: Add bindings for gpio gpio: wcd934x: Add support to wcd934x gpio controller ASoC: qcom: dt-bindings: Add compatible for DB845c and Lenovo Yoga ASoC: qcom: sdm845: add support to DB845c and Lenovo Yoga .../bindings/gpio/qcom,wcd934x-gpio.yaml | 47 + .../devicetree/bindings/slimbus/bus.txt | 10 + .../devicetree/bindings/sound/qcom,sdm845.txt | 5 +- .../bindings/sound/qcom,wcd934x.yaml | 163 + drivers/gpio/Kconfig | 8 + drivers/gpio/Makefile | 1 + drivers/gpio/gpio-wcd934x.c | 104 + drivers/mfd/Kconfig | 12 + drivers/mfd/Makefile | 1 + drivers/mfd/wcd934x.c | 306 + include/linux/mfd/wcd934x/registers.h | 531 ++ include/linux/mfd/wcd934x/wcd934x.h | 31 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd934x.c | 5084 +++++++++++++++++ sound/soc/qcom/sdm845.c | 86 +- 16 files changed, 6399 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/gpio/qcom,wcd934x-gpio.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml create mode 100644 drivers/gpio/gpio-wcd934x.c create mode 100644 drivers/mfd/wcd934x.c create mode 100644 include/linux/mfd/wcd934x/registers.h create mode 100644 include/linux/mfd/wcd934x/wcd934x.h create mode 100644 sound/soc/codecs/wcd934x.c -- 2.21.0

4 17