March 2024 - Alsa-devel - mailman.alsa-project.org

Strange trigger_htstamp after snd_pcm_drop
by GitHub issues - opened 07 Mar '24

07 Mar '24

alsa-project/alsa-lib issue #387 was opened from z-s-e: I have conceptually the following situation (actual code can be found [here](https://gitlab.com/z-s-e/alpe/-/blob/main/src/alpe_alsa.cpp?ref_type=…): ``` // sound is currently playing snd_pcm_status(pcm, pcm_status); snd_pcm_status_get_htstamp(pcm_status, &ts1); snd_pcm_drop(pcm); snd_pcm_status(pcm, pcm_status); snd_pcm_status_get_trigger_htstamp(status, &ts2); ``` On my hardware I get a `ts2` timestamp that is earlier than `ts1`. From my understanding that should be impossible, as `ts2` should contain the timestamp of when the drop happened, which should obviously be after `ts1`. Is that a bug or do I misunderstand the meaning of those timestamps? Issue URL : https://github.com/alsa-project/alsa-lib/issues/387 Repository URL: https://github.com/alsa-project/alsa-lib

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RE: [EXTERNAL] Re: [PATCH v10] The tas2783 is a smart audio amplifier with integrated MIPI SoundWire interface (Version 1.2.1 compliant), I2C, and I2S/TDM interfaces designed for portable applications. An on-chip DSP supports Texas Instruments SmartAmp sp
by Ding, Shenghao 07 Mar '24

07 Mar '24

> -----Original Message----- > From: Mark Brown <broonie(a)kernel.org> > Sent: Wednesday, March 6, 2024 3:19 AM > To: Ding, Shenghao <shenghao-ding(a)ti.com> > Cc: andriy.shevchenko(a)linux.intel.com; lgirdwood(a)gmail.com; > perex(a)perex.cz; pierre-louis.bossart(a)linux.intel.com; > 13916275206(a)139.com; alsa-devel(a)alsa-project.org; linux- > kernel(a)vger.kernel.org; liam.r.girdwood(a)intel.com; bard.liao(a)intel.com; > mengdong.lin(a)intel.com; yung-chuan.liao(a)linux.intel.com; Xu, Baojun > <baojun.xu(a)ti.com>; Lu, Kevin <kevin-lu(a)ti.com>; tiwai(a)suse.de; > soyer(a)irl.hu; Baojun.Xu(a)fpt.com; Navada Kanyana, Mukund > <navada(a)ti.com> > Subject: [EXTERNAL] Re: [PATCH v10] The tas2783 is a smart audio amplifier > with integrated MIPI SoundWire interface (Version 1.2.1 compliant), I2C, and > I2S/TDM interfaces designed for portable applications. An on-chip DSP > supports Texas Instruments SmartAmp spe... > > On Tue, Mar 05, 2024 at 04:43:35PM +0800, Shenghao Ding wrote: > > The ASoC component provides the majority of the functionality of the > > device, all the audio functions. > > > +static const struct reg_default tas2783_reg_defaults[] = { > > + /* Default values for ROM mode. Activated. */ > > + { 0x8002, 0x1a }, /* AMP inactive. */ > > + { 0x8097, 0xc8 }, > > + { 0x80b5, 0x74 }, > > + { 0x8099, 0x20 }, > > + { 0xfe8d, 0x0d }, > > + { 0xfebe, 0x4a }, > > + { 0x8230, 0x00 }, > > + { 0x8231, 0x00 }, > > + { 0x8232, 0x00 }, > > + { 0x8233, 0x01 }, > > + { 0x8418, 0x00 }, /* Set volume to 0 dB. */ > > + { 0x8419, 0x00 }, > > + { 0x841a, 0x00 }, > > + { 0x841b, 0x00 }, > > + { 0x8428, 0x40 }, /* Unmute channel */ > > + { 0x8429, 0x00 }, > > + { 0x842a, 0x00 }, > > + { 0x842b, 0x00 }, > > + { 0x8548, 0x00 }, /* Set volume to 0 dB. */ > > + { 0x8549, 0x00 }, > > + { 0x854a, 0x00 }, > > + { 0x854b, 0x00 }, > > + { 0x8558, 0x40 }, /* Unmute channel */ > > + { 0x8559, 0x00 }, > > + { 0x855a, 0x00 }, > > + { 0x855b, 0x00 }, > > + { 0x800a, 0x3a }, /* Enable both channel */ > > These comments sound like this register default table is not actually the > physical default register values that the chip has... > > > +static const struct regmap_config tasdevice_regmap = { > > + .reg_bits = 32, > > + .val_bits = 8, > > + .readable_reg = tas2783_readable_register, > > + .volatile_reg = tas2783_volatile_register, > > + .max_register = 0x44ffffff, > > + .reg_defaults = tas2783_reg_defaults, > > + .num_reg_defaults = ARRAY_SIZE(tas2783_reg_defaults), > > ...but this is set as the register defaults. This will cause problems with things > like cache sync where we don't write values out if they're not the defaults. > We also try to keep default settings from the silicon except in the most > obvious cases, it avoids issues with trying to work out what to set and > accomodate different use cases for different systems. > > If you do need to set non-default values then either just regular writes during > probe or a regmap patch would do it. So, can I remove ".reg_defaults = tas2783_reg_defaults," and tas2783_reg_defaults from the code? > > > + .cache_type = REGCACHE_RBTREE, > > + .use_single_read = true, > > + .use_single_write = true, > > REGCACHE_MAPLE is generally the most sensible choice for modern devices > - it's a newer and fancier data structure underlying it and there's only a few > cases with low end devices, mostly doing block I/O which this doesn't > support, where the RBTREE cache is still better. Accept > > > + u16 dev_info; > > + int ret; > > + > > + if (!tas_dev->sdw_peripheral) { > > + dev_err(tas_dev->dev, "%s: peripheral doesn't exist.\n", > > + __func__); > > + return; > > + } > > + > > + dev_info = tas_dev->sdw_peripheral->bus->link_id | > > + tas_dev->sdw_peripheral->id.unique_id << 16; > > I'm kind of surprised dev_info works as a variable name without something > getting upset that it aliases the function of the same name. Accept

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[PATCH 0/4] ASoC: fsl: Support register and unregister rpmsg sound card through remoteproc
by Chancel Liu 07 Mar '24

07 Mar '24

echo /lib/firmware/fw.elf > /sys/class/remoteproc/remoteproc0/firmware (A) echo start > /sys/class/remoteproc/remoteproc0/state (B) echo stop > /sys/class/remoteproc/remoteproc0/state The rpmsg sound card is registered in (A) and unregistered in (B). After "start", imx-audio-rpmsg registers devices for ASoC platform driver and machine driver. Then sound card is registered. After "stop", imx-audio-rpmsg unregisters devices for ASoC platform driver and machine driver. Then sound card is unregistered. Chancel Liu (4): ASoC: fsl: imx_pcm_rpmsg: Register component with rpmsg channel name ASoC: fsl: imx-audio-rpmsg: Register device with rpmsg channel name ASoC: fsl: Let imx-audio-rpmsg register platform device for card ASoC: fsl: imx-rpmsg: Update to correct DT node sound/soc/fsl/fsl_rpmsg.c | 11 ----------- sound/soc/fsl/imx-audio-rpmsg.c | 22 +++++++++++++++++++--- sound/soc/fsl/imx-pcm-rpmsg.c | 11 ++++++++--- sound/soc/fsl/imx-rpmsg.c | 21 ++++++++++++++++++--- 4 files changed, 45 insertions(+), 20 deletions(-) -- 2.43.0

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[PATCH v1 2/2] ASoC: codecs: ES8326: change support for ES8326
by Zhang Yi 07 Mar '24

07 Mar '24

> > On Wed, Mar 06, 2024 at 09:34:14AM +0800, Zhang Yi wrote: > > > Removed mic1-src and mic2-src. and changed default value > > > of interrupt-clk > > > > We could do with a better changelog here. > > > > > - everest,mic1-src: > > > - $ref: /schemas/types.yaml#/definitions/uint8 > > > - description: > > > - the value of reg 2A when headset plugged. > > > - minimum: 0x00 > > > - maximum: 0x77 > > > - default: 0x22 > > > - > > > - everest,mic2-src: > > > - $ref: /schemas/types.yaml#/definitions/uint8 > > > - description: > > > - the value of reg 2A when headset unplugged. > > > - minimum: 0x00 > > > - maximum: 0x77 > > > - default: 0x44 > > > > This will make any DTs using the properties instantly buggy. I believe > > there's a way of marking properties as deprecated, you should use that > > instead. Or we could just leave the properties there and ignore them at > > runtime. > 'deprecated: true' is the way. OK.I will modify the properties, thank you for your support

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[PATCH 1/5] ASoC: wm8962: Enable oscillator if selecting WM8962_FLL_OSC
by Stuart Henderson 06 Mar '24

06 Mar '24

Signed-off-by: Stuart Henderson <stuarth(a)opensource.cirrus.com> --- sound/soc/codecs/wm8962.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index fb90ae6a8a34..6d7bb696b135 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2914,8 +2914,12 @@ static int wm8962_set_fll(struct snd_soc_component *component, int fll_id, int s switch (fll_id) { case WM8962_FLL_MCLK: case WM8962_FLL_BCLK: + fll1 |= (fll_id - 1) << WM8962_FLL_REFCLK_SRC_SHIFT; + break; case WM8962_FLL_OSC: fll1 |= (fll_id - 1) << WM8962_FLL_REFCLK_SRC_SHIFT; + snd_soc_component_update_bits(component, WM8962_PLL2, + WM8962_OSC_ENA, WM8962_OSC_ENA); break; case WM8962_FLL_INT: snd_soc_component_update_bits(component, WM8962_FLL_CONTROL_1, -- 2.39.2

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[PATCH v1 0/2] ASoC: codecs: ES8326: change members of private structure
by Zhang Yi 06 Mar '24

06 Mar '24

We found that using 0x45 as the default value for interrupt-clk would cause a headset detection error.So we took 0x00 as the default value for interrupt-clk and passed the test. We removed mic1-src and mic2-src, which were not used. *** BLURB HERE *** Zhang Yi (2): ASoC: codecs: ES8326: Changing members of private structure ASoC: codecs: ES8326: change support for ES8326 .../bindings/sound/everest,es8326.yaml | 22 ++----------------- sound/soc/codecs/es8326.c | 18 +-------------- 2 files changed, 3 insertions(+), 37 deletions(-) -- 2.17.1

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[PATCH 0/5] ASoC: Harden DAPM route checks and Intel fixes
by Cezary Rojewski 06 Mar '24

06 Mar '24

Set of loosely connected patches. Most impactful change is dropping any permisiveness when snd_soc_dapm_add_routes() fails in soc-topology.c To do it safely, disable route checks for all skylake-driver boards. Relevant background: Since commit daa480bde6b3 ("ASoC: soc-core: tidyup for snd_soc_dapm_add_routes()") route checks are no longer permissive. Probe failures for Intel boards have been partially addressed by commit a22ae72b86a4 ("ASoC: soc-core: isable route checks for legacy devices") and its follow up but only skl_nau88l25_ssm4567.c is patched. The rest of the boards still need fixing. After that, removal of copy-pastas found in ssm4567.c and redundant code in i2s_test.c for avs-boards. Amadeusz Sławiński (1): ASoC: Intel: avs: i2s_test: Remove redundant dapm routes Cezary Rojewski (4): ASoC: Intel: Disable route checks for Skylake boards ASoC: topology: Do not ignore route checks when parsing graphs ASoC: Intel: avs: ssm4567: Do not ignore route checks ASoC: Intel: avs: ssm4567: Board cleanup sound/soc/intel/avs/boards/i2s_test.c | 79 ------------------- sound/soc/intel/avs/boards/ssm4567.c | 5 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 1 + sound/soc/intel/boards/bxt_rt298.c | 1 + sound/soc/intel/boards/glk_rt5682_max98357a.c | 1 + sound/soc/intel/boards/kbl_da7219_max98357a.c | 1 + sound/soc/intel/boards/kbl_da7219_max98927.c | 4 + sound/soc/intel/boards/kbl_rt5660.c | 1 + sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 + .../intel/boards/kbl_rt5663_rt5514_max98927.c | 1 + sound/soc/intel/boards/skl_hda_dsp_generic.c | 1 + .../soc/intel/boards/skl_nau88l25_max98357a.c | 1 + sound/soc/intel/boards/skl_rt286.c | 1 + sound/soc/soc-topology.c | 5 +- 14 files changed, 19 insertions(+), 85 deletions(-) -- 2.25.1

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[PATCH] ASoC: soc-card: Fix missing locking in snd_soc_card_get_kcontrol()
by Richard Fitzgerald 06 Mar '24

06 Mar '24

snd_soc_card_get_kcontrol() must be holding a read lock on card->controls_rwsem while walking the controls list. Compare with snd_ctl_find_numid(). The existing function is renamed snd_soc_card_get_kcontrol_locked() so that it can be called from contexts that are already holding card->controls_rwsem (for example, control get/put functions). There are few direct or indirect callers of snd_soc_card_get_kcontrol(), and most are safe. Three require changes, which have been included in this patch: codecs/cs35l45.c: cs35l45_activate_ctl() is called from a control put() function so is changed to call snd_soc_card_get_kcontrol_locked(). codecs/cs35l56.c: cs35l56_sync_asp1_mixer_widgets_with_firmware() is called from control get()/put() functions so is changed to call snd_soc_card_get_kcontrol_locked(). fsl/fsl_xcvr.c: fsl_xcvr_activate_ctl() is called from three places, one of which already holds card->controls_rwsem: 1. fsl_xcvr_mode_put(), a control put function, which will already be holding card->controls_rwsem. 2. fsl_xcvr_startup(), a DAI startup function. 3. fsl_xcvr_shutdown(), a DAI shutdown function. To fix this, fsl_xcvr_activate_ctl() has been changed to call snd_soc_card_get_kcontrol_locked() so that it is safe to call directly from fsl_xcvr_mode_put(). The fsl_xcvr_startup() and fsl_xcvr_shutdown() functions have been changed to take a read lock on card->controls_rsem() around calls to fsl_xcvr_activate_ctl(). While this is not very elegant, it keeps the change small, to avoid this patch creating a large collateral churn in fsl/fsl_xcvr.c. Analysis of other callers of snd_soc_card_get_kcontrol() is that they do not need any changes, they are not holding card->controls_rwsem when they call snd_soc_card_get_kcontrol(). Direct callers of snd_soc_card_get_kcontrol(): fsl/fsl_spdif.c: fsl_spdif_dai_probe() - DAI probe function fsl/fsl_micfil.c: voice_detected_fn() - IRQ handler Indirect callers via soc_component_notify_control(): codecs/cs42l43: cs42l43_mic_shutter() - IRQ handler codecs/cs42l43: cs42l43_spk_shutter() - IRQ handler codecs/ak4118.c: ak4118_irq_handler() - IRQ handler codecs/wm_adsp.c: wm_adsp_write_ctl() - not currently used Indirect callers via snd_soc_limit_volume(): qcom/sc8280xp.c: sc8280xp_snd_init() - DAIlink init function ti/rx51.c: rx51_aic34_init() - DAI init function I don't have hardware to test the fsl/*, qcom/sc828xp.c, ti/rx51.c and ak4118.c changes. Backport note: The fsl/, qcom/, cs35l45, cs35l56 and cs42l43 callers were added since the Fixes commit so won't all be present on older kernels. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Fixes: 209c6cdfd283 ("ASoC: soc-card: move snd_soc_card_get_kcontrol() to soc-card") --- It would be great if people could test the fsl/, qcom/, ti/rx51 and ak4418 drivers. --- include/sound/soc-card.h | 2 ++ sound/soc/codecs/cs35l45.c | 2 +- sound/soc/codecs/cs35l56.c | 2 +- sound/soc/fsl/fsl_xcvr.c | 12 +++++++++++- sound/soc/soc-card.c | 24 ++++++++++++++++++++++-- 5 files changed, 37 insertions(+), 5 deletions(-) diff --git a/include/sound/soc-card.h b/include/sound/soc-card.h index ecc02e955279..1f4c39922d82 100644 --- a/include/sound/soc-card.h +++ b/include/sound/soc-card.h @@ -30,6 +30,8 @@ static inline void snd_soc_card_mutex_unlock(struct snd_soc_card *card) struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, const char *name); +struct snd_kcontrol *snd_soc_card_get_kcontrol_locked(struct snd_soc_card *soc_card, + const char *name); int snd_soc_card_jack_new(struct snd_soc_card *card, const char *id, int type, struct snd_soc_jack *jack); int snd_soc_card_jack_new_pins(struct snd_soc_card *card, const char *id, diff --git a/sound/soc/codecs/cs35l45.c b/sound/soc/codecs/cs35l45.c index 44c221745c3b..2392c6effed8 100644 --- a/sound/soc/codecs/cs35l45.c +++ b/sound/soc/codecs/cs35l45.c @@ -184,7 +184,7 @@ static int cs35l45_activate_ctl(struct snd_soc_component *component, else snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s", ctl_name); - kcontrol = snd_soc_card_get_kcontrol(component->card, name); + kcontrol = snd_soc_card_get_kcontrol_locked(component->card, name); if (!kcontrol) { dev_err(component->dev, "Can't find kcontrol %s\n", name); return -EINVAL; diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index 2c1313e34cce..6dd0319bc843 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -114,7 +114,7 @@ static int cs35l56_sync_asp1_mixer_widgets_with_firmware(struct cs35l56_private name = full_name; } - kcontrol = snd_soc_card_get_kcontrol(dapm->card, name); + kcontrol = snd_soc_card_get_kcontrol_locked(dapm->card, name); if (!kcontrol) { dev_warn(cs35l56->base.dev, "Could not find control %s\n", name); continue; diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index f0fb33d719c2..c46f64557a7f 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -174,7 +174,9 @@ static int fsl_xcvr_activate_ctl(struct snd_soc_dai *dai, const char *name, struct snd_kcontrol *kctl; bool enabled; - kctl = snd_soc_card_get_kcontrol(card, name); + lockdep_assert_held(&card->snd_card->controls_rwsem); + + kctl = snd_soc_card_get_kcontrol_locked(card, name); if (kctl == NULL) return -ENOENT; @@ -576,10 +578,14 @@ static int fsl_xcvr_startup(struct snd_pcm_substream *substream, xcvr->streams |= BIT(substream->stream); if (!xcvr->soc_data->spdif_only) { + struct snd_soc_card *card = dai->component->card; + /* Disable XCVR controls if there is stream started */ + down_read(&card->snd_card->controls_rwsem); fsl_xcvr_activate_ctl(dai, fsl_xcvr_mode_kctl.name, false); fsl_xcvr_activate_ctl(dai, fsl_xcvr_arc_mode_kctl.name, false); fsl_xcvr_activate_ctl(dai, fsl_xcvr_earc_capds_kctl.name, false); + up_read(&card->snd_card->controls_rwsem); } return 0; @@ -598,11 +604,15 @@ static void fsl_xcvr_shutdown(struct snd_pcm_substream *substream, /* Enable XCVR controls if there is no stream started */ if (!xcvr->streams) { if (!xcvr->soc_data->spdif_only) { + struct snd_soc_card *card = dai->component->card; + + down_read(&card->snd_card->controls_rwsem); fsl_xcvr_activate_ctl(dai, fsl_xcvr_mode_kctl.name, true); fsl_xcvr_activate_ctl(dai, fsl_xcvr_arc_mode_kctl.name, (xcvr->mode == FSL_XCVR_MODE_ARC)); fsl_xcvr_activate_ctl(dai, fsl_xcvr_earc_capds_kctl.name, (xcvr->mode == FSL_XCVR_MODE_EARC)); + up_read(&card->snd_card->controls_rwsem); } ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_IER0, FSL_XCVR_IRQ_EARC_ALL, 0); diff --git a/sound/soc/soc-card.c b/sound/soc/soc-card.c index 285ab4c9c716..8a2f163da6bc 100644 --- a/sound/soc/soc-card.c +++ b/sound/soc/soc-card.c @@ -5,6 +5,9 @@ // Copyright (C) 2019 Renesas Electronics Corp. // Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> // + +#include <linux/lockdep.h> +#include <linux/rwsem.h> #include <sound/soc.h> #include <sound/jack.h> @@ -26,12 +29,15 @@ static inline int _soc_card_ret(struct snd_soc_card *card, return ret; } -struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, - const char *name) +struct snd_kcontrol *snd_soc_card_get_kcontrol_locked(struct snd_soc_card *soc_card, + const char *name) { struct snd_card *card = soc_card->snd_card; struct snd_kcontrol *kctl; + /* must be held read or write */ + lockdep_assert_held(&card->controls_rwsem); + if (unlikely(!name)) return NULL; @@ -40,6 +46,20 @@ struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, return kctl; return NULL; } +EXPORT_SYMBOL_GPL(snd_soc_card_get_kcontrol_locked); + +struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, + const char *name) +{ + struct snd_card *card = soc_card->snd_card; + struct snd_kcontrol *kctl; + + down_read(&card->controls_rwsem); + kctl = snd_soc_card_get_kcontrol_locked(soc_card, name); + up_read(&card->controls_rwsem); + + return kctl; +} EXPORT_SYMBOL_GPL(snd_soc_card_get_kcontrol); static int jack_new(struct snd_soc_card *card, const char *id, int type, -- 2.30.2

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[PATCH v1 1/1] ASoC: Intel: atom: sst_ipc: Remove unused intel-mid.h
by Andy Shevchenko 06 Mar '24

06 Mar '24

intel-mid.h is providing some core parts of the South Complex PM, which are usually are not used by individual drivers. In particular, this driver doesn't use it, so simply remove the unused header. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/atom/sst/sst_ipc.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst/sst_ipc.c b/sound/soc/intel/atom/sst/sst_ipc.c index 3fc2c9a6c44d..0630e58b9d6b 100644 --- a/sound/soc/intel/atom/sst/sst_ipc.c +++ b/sound/soc/intel/atom/sst/sst_ipc.c @@ -19,8 +19,9 @@ #include <sound/pcm.h> #include <sound/soc.h> #include <sound/compress_driver.h> -#include <asm/intel-mid.h> + #include <asm/platform_sst_audio.h> + #include "../sst-mfld-platform.h" #include "sst.h" -- 2.43.0.rc1.1.gbec44491f096

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[PATCH] ASoC: dt-bindings: rt1015: Convert to dtschema
by Javier García 06 Mar '24

06 Mar '24

Convert the RT1015 Mono Class D Audio Amplifier to DT schema. Signed-off-by: Javier García <javier.gar.tab(a)gmail.com> --- .../bindings/sound/realtek,rt1015.yaml | 41 +++++++++++++++++++ .../devicetree/bindings/sound/rt1015.txt | 23 ----------- 2 files changed, 41 insertions(+), 23 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt1015.yaml delete mode 100644 Documentation/devicetree/bindings/sound/rt1015.txt diff --git a/Documentation/devicetree/bindings/sound/realtek,rt1015.yaml b/Documentation/devicetree/bindings/sound/realtek,rt1015.yaml new file mode 100644 index 000000000000..880196081a60 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt1015.yaml @@ -0,0 +1,41 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt1015.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: RT1015 Mono Class D Audio Amplifier + +maintainers: + - Jack Yu <jack.yu(a)realtek.com> + +properties: + compatible: + enum: + - realtek,rt1015 + + reg: + maxItems: 1 + + realtek,power-up-delay-ms: + description: Set a delay time for flush work to be completed, + this vlaue is adjustable depending on platform. + maxItems: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@28 { + compatible = "realtek,rt1015"; + reg = <0x28>; + realtek,power-up-delay-ms = <50>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/rt1015.txt b/Documentation/devicetree/bindings/sound/rt1015.txt deleted file mode 100644 index e498966d436f..000000000000 --- a/Documentation/devicetree/bindings/sound/rt1015.txt +++ /dev/null @@ -1,23 +0,0 @@ -RT1015 Mono Class D Audio Amplifier - -This device supports I2C only. - -Required properties: - -- compatible : "realtek,rt1015". - -- reg : The I2C address of the device. - -Optional properties: - -- realtek,power-up-delay-ms - Set a delay time for flush work to be completed, - this value is adjustable depending on platform. - -Example: - -rt1015: codec@28 { - compatible = "realtek,rt1015"; - reg = <0x28>; - realtek,power-up-delay-ms = <50>; -}; -- 2.41.0

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