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[PATCH] sound: arm: aaci: fixed formatting
by Paul Lemmermann 30 Jun '22

30 Jun '22

added proper spacing before parenthesis and around 'or' operator. Signed-off-by: Paul Lemmermann <thepaulodoom(a)thepaulodoom.com> --- sound/arm/aaci.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 0817ad21a..724a1d50b 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -239,7 +239,7 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask) aacirun->bytes -= len; /* reading 16 bytes at a time */ - for( ; len > 0; len -= 16) { + for ( ; len > 0; len -= 16) { asm( "ldmia %1, {r0, r1, r2, r3}\n\t" "stmia %0!, {r0, r1, r2, r3}" @@ -250,7 +250,7 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask) if (ptr >= aacirun->end) ptr = aacirun->start; } - } while(1); + } while (1); aacirun->ptr = ptr; @@ -662,7 +662,7 @@ static void aaci_pcm_capture_start(struct aaci_runtime *aacirun) writel(aacirun->cr, aacirun->base + AACI_RXCR); ie = readl(aacirun->base + AACI_IE); - ie |= IE_ORIE |IE_RXIE; // overrun and rx interrupt -- half full + ie |= IE_ORIE | IE_RXIE; // overrun and rx interrupt -- half full writel(ie, aacirun->base + AACI_IE); } -- 2.36.1

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Re: [PATCH v1] Fix: SYNCOBJ TIMELINE Test failed.
by Christian König 30 Jun '22

30 Jun '22

Am 29.06.22 um 08:02 schrieb jie1zhan: > The issue cause by the commit : > > 721255b527(drm/syncobj: flatten dma_fence_chains on transfer). > > Because it use the point of dma_fence incorrectly > > Correct the point of dma_fence by fence array Well that patch is just utterly nonsense as far as I can see. > > Signed-off-by: jie1zhan <jesse.zhang(a)amd.com> > > Reviewed-by: Christian König <christian.koenig(a)amd.com> > > Reviewed-by: Nirmoy Das <nirmoy.das(a)linux.intel.com> I have strong doubts that Nirmoy has reviewed this and I certainly haven't reviewed it. Christian. > --- > drivers/gpu/drm/drm_syncobj.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > diff --git a/drivers/gpu/drm/drm_syncobj.c b/drivers/gpu/drm/drm_syncobj.c > index 7e48dcd1bee4..d5db818f1c76 100644 > --- a/drivers/gpu/drm/drm_syncobj.c > +++ b/drivers/gpu/drm/drm_syncobj.c > @@ -887,7 +887,7 @@ static int drm_syncobj_flatten_chain(struct dma_fence **f) > goto free_fences; > > dma_fence_put(*f); > - *f = &array->base; > + *f = array->fences[0]; > return 0; > > free_fences:

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[PATCH v3] ASoC: codecs: wsa883x: add control, dapm widgets and map
by Srinivas Kandagatla 30 Jun '22

30 Jun '22

Add controls, dapm widgets along with route. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes since v2: - updated dev mode text entries to start with Capital letter sound/soc/codecs/wsa883x.c | 200 +++++++++++++++++++++++++++++++++++++ 1 file changed, 200 insertions(+) diff --git a/sound/soc/codecs/wsa883x.c b/sound/soc/codecs/wsa883x.c index 856709ec017e..e8f519e89213 100644 --- a/sound/soc/codecs/wsa883x.c +++ b/sound/soc/codecs/wsa883x.c @@ -473,6 +473,19 @@ enum wsa_port_ids { WSA883X_PORT_VISENSE, }; +static const char * const wsa_dev_mode_text[] = { + "Speaker", "Receiver", "Ultrasound" +}; + +enum { + SPEAKER, + RECEIVER, + ULTRASOUND, +}; + +static const struct soc_enum wsa_dev_mode_enum = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(wsa_dev_mode_text), wsa_dev_mode_text); + /* 4 ports */ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA883X_MAX_SWR_PORTS] = { { @@ -1066,6 +1079,94 @@ static struct sdw_slave_ops wsa883x_slave_ops = { .port_prep = wsa883x_port_prep, }; +static int wsa_dev_mode_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct wsa883x_priv *wsa883x = snd_soc_component_get_drvdata(component); + + ucontrol->value.enumerated.item[0] = wsa883x->dev_mode; + + return 0; +} + +static int wsa_dev_mode_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct wsa883x_priv *wsa883x = snd_soc_component_get_drvdata(component); + + if (wsa883x->dev_mode == ucontrol->value.enumerated.item[0]) + return 0; + + wsa883x->dev_mode = ucontrol->value.enumerated.item[0]; + + return 1; +} + +static const DECLARE_TLV_DB_SCALE(pa_gain, -300, 150, -300); + +static int wsa883x_get_swr_port(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct wsa883x_priv *data = snd_soc_component_get_drvdata(comp); + struct soc_mixer_control *mixer = (struct soc_mixer_control *)kcontrol->private_value; + int portidx = mixer->reg; + + ucontrol->value.integer.value[0] = data->port_enable[portidx]; + + return 0; +} + +static int wsa883x_set_swr_port(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct wsa883x_priv *data = snd_soc_component_get_drvdata(comp); + struct soc_mixer_control *mixer = (struct soc_mixer_control *)kcontrol->private_value; + int portidx = mixer->reg; + + if (ucontrol->value.integer.value[0]) { + if (data->port_enable[portidx]) + return 0; + + data->port_enable[portidx] = true; + } else { + if (!data->port_enable[portidx]) + return 0; + + data->port_enable[portidx] = false; + } + + return 1; +} + +static int wsa883x_get_comp_offset(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct wsa883x_priv *wsa883x = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = wsa883x->comp_offset; + + return 0; +} + +static int wsa883x_set_comp_offset(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct wsa883x_priv *wsa883x = snd_soc_component_get_drvdata(component); + + if (wsa883x->comp_offset == ucontrol->value.integer.value[0]) + return 0; + + wsa883x->comp_offset = ucontrol->value.integer.value[0]; + + return 1; +} + static int wsa883x_codec_probe(struct snd_soc_component *comp) { struct wsa883x_priv *wsa883x = snd_soc_component_get_drvdata(comp); @@ -1075,9 +1176,108 @@ static int wsa883x_codec_probe(struct snd_soc_component *comp) return 0; } +static int wsa883x_spkr_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct wsa883x_priv *wsa883x = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + switch (wsa883x->dev_mode) { + case RECEIVER: + snd_soc_component_write_field(component, WSA883X_CDC_PATH_MODE, + WSA883X_RXD_MODE_MASK, + WSA883X_RXD_MODE_HIFI); + snd_soc_component_write_field(component, WSA883X_SPKR_PWM_CLK_CTL, + WSA883X_SPKR_PWM_FREQ_SEL_MASK, + WSA883X_SPKR_PWM_FREQ_F600KHZ); + snd_soc_component_write_field(component, WSA883X_DRE_CTL_0, + WSA883X_DRE_PROG_DELAY_MASK, 0x0); + break; + case SPEAKER: + snd_soc_component_write_field(component, WSA883X_CDC_PATH_MODE, + WSA883X_RXD_MODE_MASK, + WSA883X_RXD_MODE_NORMAL); + snd_soc_component_write_field(component, WSA883X_SPKR_PWM_CLK_CTL, + WSA883X_SPKR_PWM_FREQ_SEL_MASK, + WSA883X_SPKR_PWM_FREQ_F300KHZ); + snd_soc_component_write_field(component, WSA883X_DRE_CTL_0, + WSA883X_DRE_PROG_DELAY_MASK, 0x9); + default: + break; + } + + snd_soc_component_write_field(component, WSA883X_DRE_CTL_1, + WSA883X_DRE_GAIN_EN_MASK, + WSA883X_DRE_GAIN_FROM_CSR); + if (wsa883x->port_enable[WSA883X_PORT_COMP]) + snd_soc_component_write_field(component, WSA883X_DRE_CTL_0, + WSA883X_DRE_OFFSET_MASK, + wsa883x->comp_offset); + snd_soc_component_write_field(component, WSA883X_VBAT_ADC_FLT_CTL, + WSA883X_VBAT_ADC_COEF_SEL_MASK, + WSA883X_VBAT_ADC_COEF_F_1DIV16); + snd_soc_component_write_field(component, WSA883X_VBAT_ADC_FLT_CTL, + WSA883X_VBAT_ADC_FLT_EN_MASK, 0x1); + snd_soc_component_write_field(component, WSA883X_PDM_WD_CTL, + WSA883X_PDM_EN_MASK, + WSA883X_PDM_ENABLE); + snd_soc_component_write_field(component, WSA883X_PA_FSM_CTL, + WSA883X_GLOBAL_PA_EN_MASK, + WSA883X_GLOBAL_PA_ENABLE); + + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_component_write_field(component, WSA883X_VBAT_ADC_FLT_CTL, + WSA883X_VBAT_ADC_FLT_EN_MASK, 0x0); + snd_soc_component_write_field(component, WSA883X_VBAT_ADC_FLT_CTL, + WSA883X_VBAT_ADC_COEF_SEL_MASK, + WSA883X_VBAT_ADC_COEF_F_1DIV2); + snd_soc_component_write_field(component, WSA883X_PA_FSM_CTL, + WSA883X_GLOBAL_PA_EN_MASK, 0); + snd_soc_component_write_field(component, WSA883X_PDM_WD_CTL, + WSA883X_PDM_EN_MASK, 0); + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget wsa883x_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("IN"), + SND_SOC_DAPM_SPK("SPKR", wsa883x_spkr_event), +}; + +static const struct snd_kcontrol_new wsa883x_snd_controls[] = { + SOC_SINGLE_RANGE_TLV("PA Volume", WSA883X_DRE_CTL_1, 1, + 0x0, 0x1f, 1, pa_gain), + SOC_ENUM_EXT("WSA MODE", wsa_dev_mode_enum, + wsa_dev_mode_get, wsa_dev_mode_put), + SOC_SINGLE_EXT("COMP Offset", SND_SOC_NOPM, 0, 4, 0, + wsa883x_get_comp_offset, wsa883x_set_comp_offset), + SOC_SINGLE_EXT("DAC Switch", WSA883X_PORT_DAC, 0, 1, 0, + wsa883x_get_swr_port, wsa883x_set_swr_port), + SOC_SINGLE_EXT("COMP Switch", WSA883X_PORT_COMP, 0, 1, 0, + wsa883x_get_swr_port, wsa883x_set_swr_port), + SOC_SINGLE_EXT("BOOST Switch", WSA883X_PORT_BOOST, 0, 1, 0, + wsa883x_get_swr_port, wsa883x_set_swr_port), + SOC_SINGLE_EXT("VISENSE Switch", WSA883X_PORT_VISENSE, 0, 1, 0, + wsa883x_get_swr_port, wsa883x_set_swr_port), +}; + +static const struct snd_soc_dapm_route wsa883x_audio_map[] = { + {"SPKR", NULL, "IN"}, +}; + static const struct snd_soc_component_driver wsa883x_component_drv = { .name = "WSA883x", .probe = wsa883x_codec_probe, + .controls = wsa883x_snd_controls, + .num_controls = ARRAY_SIZE(wsa883x_snd_controls), + .dapm_widgets = wsa883x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wsa883x_dapm_widgets), + .dapm_routes = wsa883x_audio_map, + .num_dapm_routes = ARRAY_SIZE(wsa883x_audio_map), }; static int wsa883x_hw_params(struct snd_pcm_substream *substream, -- 2.25.1

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[PATCH] ASoC: wm_adsp: Minor clean and redundant code removal
by Charles Keepax 30 Jun '22

30 Jun '22

The cs_dsp core will return an error if passed a NULL cs_dsp struct so there is no need for the wm_adsp_write|read_ctl functions to manually check that. The cs_dsp core will also check the data is within bounds of the control so the additional bounds check is redundant too. Simplify things a bit by removing said code. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 25 +++++-------------------- 1 file changed, 5 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index a7784ac15dde6..cfaa45ede916a 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -675,21 +675,12 @@ static void wm_adsp_control_remove(struct cs_dsp_coeff_ctl *cs_ctl) int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, int type, unsigned int alg, void *buf, size_t len) { - struct cs_dsp_coeff_ctl *cs_ctl; + struct cs_dsp_coeff_ctl *cs_ctl = cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg); struct wm_coeff_ctl *ctl; struct snd_kcontrol *kcontrol; char ctl_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; int ret; - cs_ctl = cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg); - if (!cs_ctl) - return -EINVAL; - - ctl = cs_ctl->priv; - - if (len > cs_ctl->len) - return -EINVAL; - ret = cs_dsp_coeff_write_ctrl(cs_ctl, 0, buf, len); if (ret) return ret; @@ -697,6 +688,8 @@ int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, int type, if (cs_ctl->flags & WMFW_CTL_FLAG_SYS) return 0; + ctl = cs_ctl->priv; + if (dsp->component->name_prefix) snprintf(ctl_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s %s", dsp->component->name_prefix, ctl->name); @@ -720,16 +713,8 @@ EXPORT_SYMBOL_GPL(wm_adsp_write_ctl); int wm_adsp_read_ctl(struct wm_adsp *dsp, const char *name, int type, unsigned int alg, void *buf, size_t len) { - struct cs_dsp_coeff_ctl *cs_ctl; - - cs_ctl = cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg); - if (!cs_ctl) - return -EINVAL; - - if (len > cs_ctl->len) - return -EINVAL; - - return cs_dsp_coeff_read_ctrl(cs_ctl, 0, buf, len); + return cs_dsp_coeff_read_ctrl(cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg), + 0, buf, len); } EXPORT_SYMBOL_GPL(wm_adsp_read_ctl); -- 2.30.2

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Re: [PATCH] sound: delete a semicolon
by Mark Brown 30 Jun '22

30 Jun '22

On Thu, 30 Jun 2022 10:03:47 +0800, Li kunyu wrote: > extra semicolons could be deleted. > > Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/1] sound: delete a semicolon commit: d8d6253b36f55d199590ef908712fe52bb39ee97 All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark

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[PATCH] ASoC: samsung: change neo1973_audio from a global to static
by Tom Rix 30 Jun '22

30 Jun '22

sparse reports sound/soc/samsung/neo1973_wm8753.c:347:24: warning: symbol 'neo1973_audio' was not declared. Should it be static? neo1973_audio is only used in neo1973_wm8753.c, so it's storage class specifier should be static. Fixes: e26a2abcc246 ("ASoC: samsung: neo1973: turn into platform driver") Signed-off-by: Tom Rix <trix(a)redhat.com> --- sound/soc/samsung/neo1973_wm8753.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index c98b68567a89..e9f2334028bf 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -344,7 +344,7 @@ static int neo1973_probe(struct platform_device *pdev) return devm_snd_soc_register_card(dev, &neo1973); } -struct platform_driver neo1973_audio = { +static struct platform_driver neo1973_audio = { .driver = { .name = "neo1973-audio", .pm = &snd_soc_pm_ops, -- 2.27.0

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[PATCH 0/2] ASoC: soc-core.c: fixup snd_soc_of_get_dai_link_cpus()
by Kuninori Morimoto 30 Jun '22

30 Jun '22

Hi Mark Current ASoC has fixup both snd_soc_of_get_dai_link_cpus/codecs(). I guess cpu was copied from codec, but it is using "codec" naming everwhere in "cpu" function. It is strange, and thus, error case will be issue (It should call cpu function instead of codec). This patch tidyup it, and try to cleanup. [1/2] is for bug-fix, [2/2] is for new feature. It needs Tested-by/Reviewed-by/Acked-by from someone. Kuninori Morimoto (2): ASoC: soc-core.c: fixup snd_soc_of_get_dai_link_cpus() ASoC: soc-core.c: share code for snd_soc_of_get_dai_link_cpus/codecs() sound/soc/soc-core.c | 146 ++++++++++++++++++++++--------------------- 1 file changed, 75 insertions(+), 71 deletions(-) -- 2.25.1

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Re: [PATCH] ASoC: qdsp6: fix potential memory leak in q6apm_get_audioreach_graph()
by Mark Brown 30 Jun '22

30 Jun '22

On Thu, 30 Jun 2022 02:25:20 +0800, Jianglei Nie wrote: > q6apm_get_audioreach_graph() allocates a memory chunk for graph->graph > with audioreach_alloc_graph_pkt(). When idr_alloc() fails, graph->graph > is not released, which will lead to a memory leak. > > We can release the graph->graph with kfree() when idr_alloc() fails to > fix the memory leak. > > [...] Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/1] ASoC: qdsp6: fix potential memory leak in q6apm_get_audioreach_graph() commit: 0063ae55ccd2ade509be828762addcfccee52c4c All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark

1 0

[PATCH 1/4] ASoC: tas2764: Add post reset delays
by Martin Povišer 30 Jun '22

30 Jun '22

Make sure there is at least 1 ms delay from reset to first command as is specified in the datasheet. This is a fix similar to commit 307f31452078 ("ASoC: tas2770: Insert post reset delay"). Fixes: 827ed8a0fa50 ("ASoC: tas2764: Add the driver for the TAS2764") Signed-off-by: Martin Povišer <povik+lin(a)cutebit.org> --- sound/soc/codecs/tas2764.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas2764.c b/sound/soc/codecs/tas2764.c index 42f0c1e449ba..039bf1900880 100644 --- a/sound/soc/codecs/tas2764.c +++ b/sound/soc/codecs/tas2764.c @@ -42,10 +42,12 @@ static void tas2764_reset(struct tas2764_priv *tas2764) gpiod_set_value_cansleep(tas2764->reset_gpio, 0); msleep(20); gpiod_set_value_cansleep(tas2764->reset_gpio, 1); + usleep_range(1000, 2000); } snd_soc_component_write(tas2764->component, TAS2764_SW_RST, TAS2764_RST); + usleep_range(1000, 2000); } static int tas2764_set_bias_level(struct snd_soc_component *component, @@ -107,8 +109,10 @@ static int tas2764_codec_resume(struct snd_soc_component *component) struct tas2764_priv *tas2764 = snd_soc_component_get_drvdata(component); int ret; - if (tas2764->sdz_gpio) + if (tas2764->sdz_gpio) { gpiod_set_value_cansleep(tas2764->sdz_gpio, 1); + usleep_range(1000, 2000); + } ret = snd_soc_component_update_bits(component, TAS2764_PWR_CTRL, TAS2764_PWR_CTRL_MASK, @@ -501,8 +505,10 @@ static int tas2764_codec_probe(struct snd_soc_component *component) tas2764->component = component; - if (tas2764->sdz_gpio) + if (tas2764->sdz_gpio) { gpiod_set_value_cansleep(tas2764->sdz_gpio, 1); + usleep_range(1000, 2000); + } tas2764_reset(tas2764); -- 2.33.0

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[PATCH] ASoC: dt-bindings: Fix description for msm8916
by Bryan O'Donoghue 30 Jun '22

30 Jun '22

For the existing msm8916 bindings the minimum reg/reg-names is 1 not 2. Similarly the minimum interrupt/interrupt-names is 1 not 2. Fixes: f3fc4fbfa2d2 ("ASoC: dt-bindings: Add SC7280 lpass cpu bindings") Signed-off-by: Bryan O'Donoghue <bryan.odonoghue(a)linaro.org> --- .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml index e9a533080b32d..ef18a572a1ff3 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml @@ -25,12 +25,12 @@ properties: - qcom,sc7280-lpass-cpu reg: - minItems: 2 + minItems: 1 maxItems: 6 description: LPAIF core registers reg-names: - minItems: 2 + minItems: 1 maxItems: 6 clocks: @@ -42,12 +42,12 @@ properties: maxItems: 10 interrupts: - minItems: 2 + minItems: 1 maxItems: 4 description: LPAIF DMA buffer interrupt interrupt-names: - minItems: 2 + minItems: 1 maxItems: 4 qcom,adsp: -- 2.36.1

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