June 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH -next 1/5] ASoC: img-i2s-in: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 15 Jun '21

15 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/img/img-i2s-in.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c index fd3432a1d6ab..1bf5d6edbd32 100644 --- a/sound/soc/img/img-i2s-in.c +++ b/sound/soc/img/img-i2s-in.c @@ -434,8 +434,7 @@ static int img_i2s_in_probe(struct platform_device *pdev) i2s->dev = dev; - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_ioremap_resource(dev, res); + base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(base)) return PTR_ERR(base); -- 2.25.1

2 5

[PATCH v2] ASoC: codecs: wcd938x: remove incorrect module interdependency
by Srinivas Kandagatla 15 Jun '21

15 Jun '21

For some reason we ended up with cyclic dependency between snd_soc_wcd938x and snd_soc_wcd938x_sdw modules. Remove this cyclic dependency by handling them in respective modules. Without this below error is reported during make modules_install depmod: ERROR: Cycle detected: snd_soc_wcd938x -> snd_soc_wcd938x_sdw -> snd_soc_wcd938x depmod: ERROR: Found 2 modules in dependency cycles! Fixes: 045442228868 ("ASoC: codecs: wcd938x: add audio routing and Kconfig") Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes since v1: - included irq.h as x86 allmodconfig build reported implicit declaration of function handle_nested_irq sound/soc/codecs/wcd938x-sdw.c | 20 +++++++++++++------- sound/soc/codecs/wcd938x.c | 27 +++++---------------------- sound/soc/codecs/wcd938x.h | 4 +--- 3 files changed, 19 insertions(+), 32 deletions(-) diff --git a/sound/soc/codecs/wcd938x-sdw.c b/sound/soc/codecs/wcd938x-sdw.c index d82c40ec6898..d8d4caf48dc7 100644 --- a/sound/soc/codecs/wcd938x-sdw.c +++ b/sound/soc/codecs/wcd938x-sdw.c @@ -9,6 +9,7 @@ #include <linux/component.h> #include <sound/soc.h> #include <linux/pm_runtime.h> +#include <linux/irq.h> #include <linux/irqdomain.h> #include <linux/of.h> #include <linux/soundwire/sdw.h> @@ -176,8 +177,19 @@ static int wcd9380_interrupt_callback(struct sdw_slave *slave, struct sdw_slave_intr_status *status) { struct wcd938x_sdw_priv *wcd = dev_get_drvdata(&slave->dev); + struct irq_domain *slave_irq = wcd->slave_irq; + struct regmap *regmap = dev_get_regmap(&slave->dev, NULL); + u32 sts1, sts2, sts3; - return wcd938x_handle_sdw_irq(wcd); + do { + handle_nested_irq(irq_find_mapping(slave_irq, 0)); + regmap_read(regmap, WCD938X_DIGITAL_INTR_STATUS_0, &sts1); + regmap_read(regmap, WCD938X_DIGITAL_INTR_STATUS_1, &sts2); + regmap_read(regmap, WCD938X_DIGITAL_INTR_STATUS_2, &sts3); + + } while (sts1 || sts2 || sts3); + + return IRQ_HANDLED; } static struct sdw_slave_ops wcd9380_slave_ops = { @@ -239,16 +251,10 @@ static int wcd9380_probe(struct sdw_slave *pdev, SDW_SCP_INT1_PARITY; pdev->prop.lane_control_support = true; if (wcd->is_tx) { - struct regmap *rm; - pdev->prop.source_ports = GENMASK(WCD938X_MAX_SWR_PORTS, 0); pdev->prop.src_dpn_prop = wcd938x_dpn_prop; wcd->ch_info = &wcd938x_sdw_tx_ch_info[0]; pdev->prop.wake_capable = true; - - rm = devm_regmap_init_sdw(pdev, &wcd938x_regmap_config); - if (IS_ERR(rm)) - return PTR_ERR(rm); } else { pdev->prop.sink_ports = GENMASK(WCD938X_MAX_SWR_PORTS, 0); pdev->prop.sink_dpn_prop = wcd938x_dpn_prop; diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index a2c76dc8fd89..24afe93830ce 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -1198,7 +1198,7 @@ static bool wcd938x_volatile_register(struct device *dev, unsigned int reg) return false; } -struct regmap_config wcd938x_regmap_config = { +static struct regmap_config wcd938x_regmap_config = { .name = "wcd938x_csr", .reg_bits = 32, .val_bits = 8, @@ -1211,7 +1211,6 @@ struct regmap_config wcd938x_regmap_config = { .volatile_reg = wcd938x_volatile_register, .can_multi_write = true, }; -EXPORT_SYMBOL_GPL(wcd938x_regmap_config); static const struct regmap_irq wcd938x_irqs[WCD938X_NUM_IRQS] = { REGMAP_IRQ_REG(WCD938X_IRQ_MBHC_BUTTON_PRESS_DET, 0, 0x01), @@ -3472,24 +3471,6 @@ static int wcd938x_reset(struct wcd938x_priv *wcd938x) return 0; } -int wcd938x_handle_sdw_irq(struct wcd938x_sdw_priv *wcd) -{ - struct wcd938x_priv *wcd938x = wcd->wcd938x; - struct irq_domain *slave_irq = wcd938x->virq; - u32 sts1, sts2, sts3; - - do { - handle_nested_irq(irq_find_mapping(slave_irq, 0)); - regmap_read(wcd938x->regmap, WCD938X_DIGITAL_INTR_STATUS_0, &sts1); - regmap_read(wcd938x->regmap, WCD938X_DIGITAL_INTR_STATUS_1, &sts2); - regmap_read(wcd938x->regmap, WCD938X_DIGITAL_INTR_STATUS_2, &sts3); - - } while (sts1 || sts2 || sts3); - - return IRQ_HANDLED; -} -EXPORT_SYMBOL_GPL(wcd938x_handle_sdw_irq); - static int wcd938x_codec_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -3573,6 +3554,7 @@ static int wcd938x_bind(struct device *dev) } wcd938x->sdw_priv[AIF1_PB] = dev_get_drvdata(wcd938x->rxdev); wcd938x->sdw_priv[AIF1_PB]->wcd938x = wcd938x; + wcd938x->sdw_priv[AIF1_PB]->slave_irq = wcd938x->virq; wcd938x->txdev = wcd938x_sdw_device_get(wcd938x->txnode); if (!wcd938x->txdev) { @@ -3581,6 +3563,7 @@ static int wcd938x_bind(struct device *dev) } wcd938x->sdw_priv[AIF1_CAP] = dev_get_drvdata(wcd938x->txdev); wcd938x->sdw_priv[AIF1_CAP]->wcd938x = wcd938x; + wcd938x->sdw_priv[AIF1_CAP]->slave_irq = wcd938x->virq; wcd938x->tx_sdw_dev = dev_to_sdw_dev(wcd938x->txdev); if (!wcd938x->tx_sdw_dev) { dev_err(dev, "could not get txslave with matching of dev\n"); @@ -3607,8 +3590,8 @@ static int wcd938x_bind(struct device *dev) return -EINVAL; } - wcd938x->regmap = dev_get_regmap(wcd938x->txdev, NULL); - if (!wcd938x->regmap) { + wcd938x->regmap = devm_regmap_init_sdw(wcd938x->tx_sdw_dev, &wcd938x_regmap_config); + if (IS_ERR(wcd938x->regmap)) { dev_err(dev, "%s: tx csr regmap not found\n", __func__); return PTR_ERR(wcd938x->regmap); } diff --git a/sound/soc/codecs/wcd938x.h b/sound/soc/codecs/wcd938x.h index 9db3ab6e47a6..07b08de4cebf 100644 --- a/sound/soc/codecs/wcd938x.h +++ b/sound/soc/codecs/wcd938x.h @@ -663,11 +663,9 @@ struct wcd938x_sdw_priv { int num_ports; bool is_tx; struct wcd938x_priv *wcd938x; + struct irq_domain *slave_irq; }; -extern struct regmap_config wcd938x_regmap_config; -int wcd938x_handle_sdw_irq(struct wcd938x_sdw_priv *priv); - #if IS_ENABLED(CONFIG_SND_SOC_WCD938X_SDW) int wcd938x_sdw_free(struct wcd938x_sdw_priv *wcd, struct snd_pcm_substream *substream, -- 2.21.0

2 1

[PATCH -next] ASoC: hisilicon: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 15 Jun '21

15 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/hisilicon/hi6210-i2s.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c index ff05b9779e4b..a297d4af5099 100644 --- a/sound/soc/hisilicon/hi6210-i2s.c +++ b/sound/soc/hisilicon/hi6210-i2s.c @@ -556,8 +556,7 @@ static int hi6210_i2s_probe(struct platform_device *pdev) i2s->dev = dev; spin_lock_init(&i2s->lock); - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - i2s->base = devm_ioremap_resource(dev, res); + i2s->base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(i2s->base)) return PTR_ERR(i2s->base); -- 2.25.1

2 1

[PATCH AUTOSEL 4.19 01/12] ASoC: rt5659: Fix the lost powers for the HDA header
by Sasha Levin 15 Jun '21

15 Jun '21

From: Jack Yu <jack.yu(a)realtek.com> [ Upstream commit 6308c44ed6eeadf65c0a7ba68d609773ed860fbb ] The power of "LDO2", "MICBIAS1" and "Mic Det Power" were powered off after the DAPM widgets were added, and these powers were set by the JD settings "RT5659_JD_HDA_HEADER" in the probe function. In the codec probe function, these powers were ignored to prevent them controlled by DAPM. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Signed-off-by: Jack Yu <jack.yu(a)realtek.com> Message-Id: <15fced51977b458798ca4eebf03dafb9(a)realtek.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5659.c | 26 +++++++++++++++++++++----- 1 file changed, 21 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index b331b3ba61a9..ab73f84b5970 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -2473,13 +2473,18 @@ static int set_dmic_power(struct snd_soc_dapm_widget *w, return 0; } -static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { +static const struct snd_soc_dapm_widget rt5659_particular_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("LDO2", RT5659_PWR_ANLG_3, RT5659_PWR_LDO2_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0, - NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT, + 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5659_PWR_VOL, RT5659_PWR_MIC_DET_BIT, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0, + NULL, 0), SND_SOC_DAPM_SUPPLY("Mono Vref", RT5659_PWR_ANLG_1, RT5659_PWR_VREF3_BIT, 0, NULL, 0), @@ -2504,8 +2509,6 @@ static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { RT5659_ADC_MONO_R_ASRC_SFT, 0, NULL, 0), /* Input Side */ - SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT, - 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5659_PWR_ANLG_2, RT5659_PWR_MB2_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS3", RT5659_PWR_ANLG_2, RT5659_PWR_MB3_BIT, @@ -3700,10 +3703,23 @@ static int rt5659_set_bias_level(struct snd_soc_component *component, static int rt5659_probe(struct snd_soc_component *component) { + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component); rt5659->component = component; + switch (rt5659->pdata.jd_src) { + case RT5659_JD_HDA_HEADER: + break; + + default: + snd_soc_dapm_new_controls(dapm, + rt5659_particular_dapm_widgets, + ARRAY_SIZE(rt5659_particular_dapm_widgets)); + break; + } + return 0; } -- 2.30.2

1 0

[PATCH AUTOSEL 5.4 02/15] ASoC: rt5659: Fix the lost powers for the HDA header
by Sasha Levin 15 Jun '21

15 Jun '21

From: Jack Yu <jack.yu(a)realtek.com> [ Upstream commit 6308c44ed6eeadf65c0a7ba68d609773ed860fbb ] The power of "LDO2", "MICBIAS1" and "Mic Det Power" were powered off after the DAPM widgets were added, and these powers were set by the JD settings "RT5659_JD_HDA_HEADER" in the probe function. In the codec probe function, these powers were ignored to prevent them controlled by DAPM. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Signed-off-by: Jack Yu <jack.yu(a)realtek.com> Message-Id: <15fced51977b458798ca4eebf03dafb9(a)realtek.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5659.c | 26 +++++++++++++++++++++----- 1 file changed, 21 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index afd61599d94c..a28afb480060 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -2470,13 +2470,18 @@ static int set_dmic_power(struct snd_soc_dapm_widget *w, return 0; } -static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { +static const struct snd_soc_dapm_widget rt5659_particular_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("LDO2", RT5659_PWR_ANLG_3, RT5659_PWR_LDO2_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0, - NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT, + 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5659_PWR_VOL, RT5659_PWR_MIC_DET_BIT, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0, + NULL, 0), SND_SOC_DAPM_SUPPLY("Mono Vref", RT5659_PWR_ANLG_1, RT5659_PWR_VREF3_BIT, 0, NULL, 0), @@ -2501,8 +2506,6 @@ static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { RT5659_ADC_MONO_R_ASRC_SFT, 0, NULL, 0), /* Input Side */ - SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT, - 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5659_PWR_ANLG_2, RT5659_PWR_MB2_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS3", RT5659_PWR_ANLG_2, RT5659_PWR_MB3_BIT, @@ -3697,10 +3700,23 @@ static int rt5659_set_bias_level(struct snd_soc_component *component, static int rt5659_probe(struct snd_soc_component *component) { + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component); rt5659->component = component; + switch (rt5659->pdata.jd_src) { + case RT5659_JD_HDA_HEADER: + break; + + default: + snd_soc_dapm_new_controls(dapm, + rt5659_particular_dapm_widgets, + ARRAY_SIZE(rt5659_particular_dapm_widgets)); + break; + } + return 0; } -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 21/30] ASoC: qcom: lpass-cpu: Fix pop noise during audio capture begin
by Sasha Levin 15 Jun '21

15 Jun '21

From: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> [ Upstream commit c8a4556d98510ca05bad8d02265a4918b03a8c0b ] This patch fixes PoP noise of around 15ms observed during audio capture begin. Enables BCLK and LRCLK in snd_soc_dai_ops prepare call for introducing some delay before capture start. (am from https://patchwork.kernel.org/patch/12276369/) (also found at https://lore.kernel.org/r/20210524142114.18676-1-srivasam@codeaurora.org) Co-developed-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20210604154545.1198337-1-judyhsiao@chromium.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/qcom/lpass-cpu.c | 79 ++++++++++++++++++++++++++++++++++++++ sound/soc/qcom/lpass.h | 4 ++ 2 files changed, 83 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 7a30a12519a7..e620a62ef534 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -93,8 +93,30 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); + /* + * Ensure LRCLK is disabled even in device node validation. + * Will not impact if disabled in lpass_cpu_daiops_trigger() + * suspend. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); + else + regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_DISABLE); + + /* + * BCLK may not be enabled if lpass_cpu_daiops_prepare is called before + * lpass_cpu_daiops_shutdown. It's paired with the clk_enable in + * lpass_cpu_daiops_prepare. + */ + if (drvdata->mi2s_was_prepared[dai->driver->id]) { + drvdata->mi2s_was_prepared[dai->driver->id] = false; + clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]); + } + clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); } @@ -275,6 +297,18 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* + * Ensure lpass BCLK/LRCLK is enabled during + * device resume as lpass_cpu_daiops_prepare() is not called + * after the device resumes. We don't check mi2s_was_prepared before + * enable/disable BCLK in trigger events because: + * 1. These trigger events are paired, so the BCLK + * enable_count is balanced. + * 2. the BCLK can be shared (ex: headset and headset mic), + * we need to increase the enable_count so that we don't + * turn off the shared BCLK while other devices are using + * it. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); @@ -296,6 +330,10 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* + * To ensure lpass BCLK/LRCLK is disabled during + * device suspend. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); @@ -315,12 +353,53 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, return ret; } +static int lpass_cpu_daiops_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; + int ret; + + /* + * Ensure lpass BCLK/LRCLK is enabled bit before playback/capture + * data flow starts. This allows other codec to have some delay before + * the data flow. + * (ex: to drop start up pop noise before capture starts). + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); + else + ret = regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_ENABLE); + + if (ret) { + dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); + return ret; + } + + /* + * Check mi2s_was_prepared before enabling BCLK as lpass_cpu_daiops_prepare can + * be called multiple times. It's paired with the clk_disable in + * lpass_cpu_daiops_shutdown. + */ + if (!drvdata->mi2s_was_prepared[dai->driver->id]) { + ret = clk_enable(drvdata->mi2s_bit_clk[id]); + if (ret) { + dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); + return ret; + } + drvdata->mi2s_was_prepared[dai->driver->id] = true; + } + return 0; +} + const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { .set_sysclk = lpass_cpu_daiops_set_sysclk, .startup = lpass_cpu_daiops_startup, .shutdown = lpass_cpu_daiops_shutdown, .hw_params = lpass_cpu_daiops_hw_params, .trigger = lpass_cpu_daiops_trigger, + .prepare = lpass_cpu_daiops_prepare, }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 1d926dd5f590..0484ad39b3dc 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -67,6 +67,10 @@ struct lpass_data { /* MI2S SD lines to use for playback/capture */ unsigned int mi2s_playback_sd_mode[LPASS_MAX_MI2S_PORTS]; unsigned int mi2s_capture_sd_mode[LPASS_MAX_MI2S_PORTS]; + + /* The state of MI2S prepare dai_ops was called */ + bool mi2s_was_prepared[LPASS_MAX_MI2S_PORTS]; + int hdmi_port_enable; /* low-power audio interface (LPAIF) registers */ -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 18/30] ASoC: rt5682: Fix the fast discharge for headset unplugging in soundwire mode
by Sasha Levin 15 Jun '21

15 Jun '21

From: Oder Chiou <oder_chiou(a)realtek.com> [ Upstream commit 49783c6f4a4f49836b5a109ae0daf2f90b0d7713 ] Based on ("5a15cd7fce20b1fd4aece6a0240e2b58cd6a225d"), the setting also should be set in soundwire mode. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Link: https://lore.kernel.org/r/20210604063150.29925-1-oder_chiou@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5682-sdw.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index 58fb13132602..aa6c325faeab 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -455,7 +455,8 @@ static int rt5682_io_init(struct device *dev, struct sdw_slave *slave) regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_2, RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL); - regmap_write(rt5682->regmap, RT5682_CBJ_CTRL_1, 0xd042); + regmap_write(rt5682->regmap, RT5682_CBJ_CTRL_1, 0xd142); + regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_5, 0x0700, 0x0600); regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_3, RT5682_CBJ_IN_BUF_EN, RT5682_CBJ_IN_BUF_EN); regmap_update_bits(rt5682->regmap, RT5682_SAR_IL_CMD_1, -- 2.30.2

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[PATCH AUTOSEL 5.10 14/30] ASoC: tas2562: Fix TDM_CFG0_SAMPRATE values
by Sasha Levin 15 Jun '21

15 Jun '21

From: Richard Weinberger <richard(a)nod.at> [ Upstream commit 8bef925e37bdc9b6554b85eda16ced9a8e3c135f ] TAS2562_TDM_CFG0_SAMPRATE_MASK starts at bit 1, not 0. So all values need to be left shifted by 1. Signed-off-by: Richard Weinberger <richard(a)nod.at> Link: https://lore.kernel.org/r/20210530203446.19022-1-richard@nod.at Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tas2562.h | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h index 81866aeb3fbf..55b2a1f52ca3 100644 --- a/sound/soc/codecs/tas2562.h +++ b/sound/soc/codecs/tas2562.h @@ -57,13 +57,13 @@ #define TAS2562_TDM_CFG0_RAMPRATE_MASK BIT(5) #define TAS2562_TDM_CFG0_RAMPRATE_44_1 BIT(5) #define TAS2562_TDM_CFG0_SAMPRATE_MASK GENMASK(3, 1) -#define TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ 0x0 -#define TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ 0x1 -#define TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ 0x2 -#define TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ 0x3 -#define TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ 0x4 -#define TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ 0x5 -#define TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ 0x6 +#define TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ (0x0 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ (0x1 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ (0x2 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ (0x3 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ (0x4 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ (0x5 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ (0x6 << 1) #define TAS2562_TDM_CFG2_RIGHT_JUSTIFY BIT(6) -- 2.30.2

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[PATCH AUTOSEL 5.10 13/30] ASoC: AMD Renoir: Remove fix for DMI entry on Lenovo 2020 platforms
by Sasha Levin 15 Jun '21

15 Jun '21

From: Mark Pearson <markpearson(a)lenovo.com> [ Upstream commit 320232caf1d8febea17312dab4b2dfe02e033520 ] Unfortunately the previous patch to fix issues using the AMD ACP bridge has the side effect of breaking the dmic in other cases and needs to be reverted. Removing the changes while we revisit the fix and find something better. Apologies for the churn. Suggested-by: Gabriel Craciunescu <unix.or.die(a)gmail.com> Signed-off-by: Mark Pearson <markpearson(a)lenovo.com> Link: https://lore.kernel.org/r/20210602171251.3243-1-markpearson@lenovo.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/amd/renoir/rn-pci-acp3x.c | 35 ----------------------------- 1 file changed, 35 deletions(-) diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index 6de536b02b57..917536def5f2 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -192,41 +192,6 @@ static const struct dmi_system_id rn_acp_quirk_table[] = { DMI_EXACT_MATCH(DMI_BOARD_NAME, "20NLCTO1WW"), } }, - { - /* Lenovo ThinkPad P14s Gen 1 (20Y1) */ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20Y1"), - } - }, - { - /* Lenovo ThinkPad T14s Gen1 */ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20UH"), - } - }, - { - /* Lenovo ThinkPad T14s Gen1 Campus */ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20UJ"), - } - }, - { - /* Lenovo ThinkPad T14 Gen 1*/ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20UD"), - } - }, - { - /* Lenovo ThinkPad X13 Gen 1*/ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20UF"), - } - }, {} }; -- 2.30.2

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[PATCH AUTOSEL 5.10 10/30] ASoC: AMD Renoir - add DMI entry for Lenovo 2020 AMD platforms
by Sasha Levin 15 Jun '21

15 Jun '21

From: Mark Pearson <markpearson(a)lenovo.com> [ Upstream commit 19a0aa9b04c5ab9a063b6ceaf7211ee7d9a9d24d ] More laptops identified where the AMD ACP bridge needs to be blocked or the microphone will not work when connected to HDMI. Use DMI to block the microphone PCM device for these platforms. Suggested-by: Gabriel Craciunescu <nix.or.die(a)gmail.com> Signed-off-by: Mark Pearson <markpearson(a)lenovo.com> Link: https://lore.kernel.org/r/20210531145502.6079-1-markpearson@lenovo.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/amd/renoir/rn-pci-acp3x.c | 35 +++++++++++++++++++++++++++++ 1 file changed, 35 insertions(+) diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index 917536def5f2..6de536b02b57 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -192,6 +192,41 @@ static const struct dmi_system_id rn_acp_quirk_table[] = { DMI_EXACT_MATCH(DMI_BOARD_NAME, "20NLCTO1WW"), } }, + { + /* Lenovo ThinkPad P14s Gen 1 (20Y1) */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20Y1"), + } + }, + { + /* Lenovo ThinkPad T14s Gen1 */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UH"), + } + }, + { + /* Lenovo ThinkPad T14s Gen1 Campus */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UJ"), + } + }, + { + /* Lenovo ThinkPad T14 Gen 1*/ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UD"), + } + }, + { + /* Lenovo ThinkPad X13 Gen 1*/ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UF"), + } + }, {} }; -- 2.30.2

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[PATCH AUTOSEL 5.10 07/30] ASoC: fsl-asoc-card: Set .owner attribute when registering card.
by Sasha Levin 15 Jun '21

15 Jun '21

From: Nicolas Cavallari <nicolas.cavallari(a)green-communications.fr> [ Upstream commit a8437f05384cb472518ec21bf4fffbe8f0a47378 ] Otherwise, when compiled as module, a WARN_ON is triggered: WARNING: CPU: 0 PID: 5 at sound/core/init.c:208 snd_card_new+0x310/0x39c [snd] [...] CPU: 0 PID: 5 Comm: kworker/0:0 Not tainted 5.10.39 #1 Hardware name: Freescale i.MX6 Quad/DualLite (Device Tree) Workqueue: events deferred_probe_work_func [<c0111988>] (unwind_backtrace) from [<c010c8ac>] (show_stack+0x10/0x14) [<c010c8ac>] (show_stack) from [<c092784c>] (dump_stack+0xdc/0x104) [<c092784c>] (dump_stack) from [<c0129710>] (__warn+0xd8/0x114) [<c0129710>] (__warn) from [<c0922a48>] (warn_slowpath_fmt+0x5c/0xc4) [<c0922a48>] (warn_slowpath_fmt) from [<bf0496f8>] (snd_card_new+0x310/0x39c [snd]) [<bf0496f8>] (snd_card_new [snd]) from [<bf1d7df8>] (snd_soc_bind_card+0x334/0x9c4 [snd_soc_core]) [<bf1d7df8>] (snd_soc_bind_card [snd_soc_core]) from [<bf1e9cd8>] (devm_snd_soc_register_card+0x30/0x6c [snd_soc_core]) [<bf1e9cd8>] (devm_snd_soc_register_card [snd_soc_core]) from [<bf22d964>] (fsl_asoc_card_probe+0x550/0xcc8 [snd_soc_fsl_asoc_card]) [<bf22d964>] (fsl_asoc_card_probe [snd_soc_fsl_asoc_card]) from [<c060c930>] (platform_drv_probe+0x48/0x98) [...] Signed-off-by: Nicolas Cavallari <nicolas.cavallari(a)green-communications.fr> Acked-by: Shengjiu Wang <shengjiu.wang(a)gmail.com> Link: https://lore.kernel.org/r/20210527163409.22049-1-nicolas.cavallari@green-co… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl-asoc-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index a2dd3b6b7fec..7cd14d6b9436 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -720,6 +720,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Initialize sound card */ priv->pdev = pdev; priv->card.dev = &pdev->dev; + priv->card.owner = THIS_MODULE; ret = snd_soc_of_parse_card_name(&priv->card, "model"); if (ret) { snprintf(priv->name, sizeof(priv->name), "%s-audio", -- 2.30.2

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[PATCH AUTOSEL 5.10 05/30] ASoC: rt5659: Fix the lost powers for the HDA header
by Sasha Levin 15 Jun '21

15 Jun '21

From: Jack Yu <jack.yu(a)realtek.com> [ Upstream commit 6308c44ed6eeadf65c0a7ba68d609773ed860fbb ] The power of "LDO2", "MICBIAS1" and "Mic Det Power" were powered off after the DAPM widgets were added, and these powers were set by the JD settings "RT5659_JD_HDA_HEADER" in the probe function. In the codec probe function, these powers were ignored to prevent them controlled by DAPM. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Signed-off-by: Jack Yu <jack.yu(a)realtek.com> Message-Id: <15fced51977b458798ca4eebf03dafb9(a)realtek.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5659.c | 26 +++++++++++++++++++++----- 1 file changed, 21 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 91a4ef7f620c..a9b079d56fd6 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -2433,13 +2433,18 @@ static int set_dmic_power(struct snd_soc_dapm_widget *w, return 0; } -static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { +static const struct snd_soc_dapm_widget rt5659_particular_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("LDO2", RT5659_PWR_ANLG_3, RT5659_PWR_LDO2_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0, - NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT, + 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5659_PWR_VOL, RT5659_PWR_MIC_DET_BIT, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0, + NULL, 0), SND_SOC_DAPM_SUPPLY("Mono Vref", RT5659_PWR_ANLG_1, RT5659_PWR_VREF3_BIT, 0, NULL, 0), @@ -2464,8 +2469,6 @@ static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { RT5659_ADC_MONO_R_ASRC_SFT, 0, NULL, 0), /* Input Side */ - SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT, - 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5659_PWR_ANLG_2, RT5659_PWR_MB2_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS3", RT5659_PWR_ANLG_2, RT5659_PWR_MB3_BIT, @@ -3660,10 +3663,23 @@ static int rt5659_set_bias_level(struct snd_soc_component *component, static int rt5659_probe(struct snd_soc_component *component) { + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component); rt5659->component = component; + switch (rt5659->pdata.jd_src) { + case RT5659_JD_HDA_HEADER: + break; + + default: + snd_soc_dapm_new_controls(dapm, + rt5659_particular_dapm_widgets, + ARRAY_SIZE(rt5659_particular_dapm_widgets)); + break; + } + return 0; } -- 2.30.2

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[PATCH AUTOSEL 5.12 23/33] ASoC: qcom: lpass-cpu: Fix pop noise during audio capture begin
by Sasha Levin 15 Jun '21

15 Jun '21

From: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> [ Upstream commit c8a4556d98510ca05bad8d02265a4918b03a8c0b ] This patch fixes PoP noise of around 15ms observed during audio capture begin. Enables BCLK and LRCLK in snd_soc_dai_ops prepare call for introducing some delay before capture start. (am from https://patchwork.kernel.org/patch/12276369/) (also found at https://lore.kernel.org/r/20210524142114.18676-1-srivasam@codeaurora.org) Co-developed-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20210604154545.1198337-1-judyhsiao@chromium.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/qcom/lpass-cpu.c | 79 ++++++++++++++++++++++++++++++++++++++ sound/soc/qcom/lpass.h | 4 ++ 2 files changed, 83 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 936384a94f25..74d3d8c58608 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -93,8 +93,30 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); + /* + * Ensure LRCLK is disabled even in device node validation. + * Will not impact if disabled in lpass_cpu_daiops_trigger() + * suspend. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); + else + regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_DISABLE); + + /* + * BCLK may not be enabled if lpass_cpu_daiops_prepare is called before + * lpass_cpu_daiops_shutdown. It's paired with the clk_enable in + * lpass_cpu_daiops_prepare. + */ + if (drvdata->mi2s_was_prepared[dai->driver->id]) { + drvdata->mi2s_was_prepared[dai->driver->id] = false; + clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]); + } + clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); } @@ -275,6 +297,18 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* + * Ensure lpass BCLK/LRCLK is enabled during + * device resume as lpass_cpu_daiops_prepare() is not called + * after the device resumes. We don't check mi2s_was_prepared before + * enable/disable BCLK in trigger events because: + * 1. These trigger events are paired, so the BCLK + * enable_count is balanced. + * 2. the BCLK can be shared (ex: headset and headset mic), + * we need to increase the enable_count so that we don't + * turn off the shared BCLK while other devices are using + * it. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); @@ -296,6 +330,10 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* + * To ensure lpass BCLK/LRCLK is disabled during + * device suspend. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); @@ -315,12 +353,53 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, return ret; } +static int lpass_cpu_daiops_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; + int ret; + + /* + * Ensure lpass BCLK/LRCLK is enabled bit before playback/capture + * data flow starts. This allows other codec to have some delay before + * the data flow. + * (ex: to drop start up pop noise before capture starts). + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); + else + ret = regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_ENABLE); + + if (ret) { + dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); + return ret; + } + + /* + * Check mi2s_was_prepared before enabling BCLK as lpass_cpu_daiops_prepare can + * be called multiple times. It's paired with the clk_disable in + * lpass_cpu_daiops_shutdown. + */ + if (!drvdata->mi2s_was_prepared[dai->driver->id]) { + ret = clk_enable(drvdata->mi2s_bit_clk[id]); + if (ret) { + dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); + return ret; + } + drvdata->mi2s_was_prepared[dai->driver->id] = true; + } + return 0; +} + const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { .set_sysclk = lpass_cpu_daiops_set_sysclk, .startup = lpass_cpu_daiops_startup, .shutdown = lpass_cpu_daiops_shutdown, .hw_params = lpass_cpu_daiops_hw_params, .trigger = lpass_cpu_daiops_trigger, + .prepare = lpass_cpu_daiops_prepare, }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 83b2e08ade06..7f72214404ba 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -67,6 +67,10 @@ struct lpass_data { /* MI2S SD lines to use for playback/capture */ unsigned int mi2s_playback_sd_mode[LPASS_MAX_MI2S_PORTS]; unsigned int mi2s_capture_sd_mode[LPASS_MAX_MI2S_PORTS]; + + /* The state of MI2S prepare dai_ops was called */ + bool mi2s_was_prepared[LPASS_MAX_MI2S_PORTS]; + int hdmi_port_enable; /* low-power audio interface (LPAIF) registers */ -- 2.30.2

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[PATCH AUTOSEL 5.12 20/33] ASoC: rt5682: Fix the fast discharge for headset unplugging in soundwire mode
by Sasha Levin 15 Jun '21

15 Jun '21

From: Oder Chiou <oder_chiou(a)realtek.com> [ Upstream commit 49783c6f4a4f49836b5a109ae0daf2f90b0d7713 ] Based on ("5a15cd7fce20b1fd4aece6a0240e2b58cd6a225d"), the setting also should be set in soundwire mode. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Link: https://lore.kernel.org/r/20210604063150.29925-1-oder_chiou@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5682-sdw.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index b49f1e16125d..d1dd7f720ba4 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -462,7 +462,8 @@ static int rt5682_io_init(struct device *dev, struct sdw_slave *slave) regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_2, RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL); - regmap_write(rt5682->regmap, RT5682_CBJ_CTRL_1, 0xd042); + regmap_write(rt5682->regmap, RT5682_CBJ_CTRL_1, 0xd142); + regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_5, 0x0700, 0x0600); regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_3, RT5682_CBJ_IN_BUF_EN, RT5682_CBJ_IN_BUF_EN); regmap_update_bits(rt5682->regmap, RT5682_SAR_IL_CMD_1, -- 2.30.2

1 0

[PATCH AUTOSEL 5.12 15/33] ASoC: tas2562: Fix TDM_CFG0_SAMPRATE values
by Sasha Levin 15 Jun '21

15 Jun '21

From: Richard Weinberger <richard(a)nod.at> [ Upstream commit 8bef925e37bdc9b6554b85eda16ced9a8e3c135f ] TAS2562_TDM_CFG0_SAMPRATE_MASK starts at bit 1, not 0. So all values need to be left shifted by 1. Signed-off-by: Richard Weinberger <richard(a)nod.at> Link: https://lore.kernel.org/r/20210530203446.19022-1-richard@nod.at Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tas2562.h | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h index 81866aeb3fbf..55b2a1f52ca3 100644 --- a/sound/soc/codecs/tas2562.h +++ b/sound/soc/codecs/tas2562.h @@ -57,13 +57,13 @@ #define TAS2562_TDM_CFG0_RAMPRATE_MASK BIT(5) #define TAS2562_TDM_CFG0_RAMPRATE_44_1 BIT(5) #define TAS2562_TDM_CFG0_SAMPRATE_MASK GENMASK(3, 1) -#define TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ 0x0 -#define TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ 0x1 -#define TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ 0x2 -#define TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ 0x3 -#define TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ 0x4 -#define TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ 0x5 -#define TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ 0x6 +#define TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ (0x0 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ (0x1 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ (0x2 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ (0x3 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ (0x4 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ (0x5 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ (0x6 << 1) #define TAS2562_TDM_CFG2_RIGHT_JUSTIFY BIT(6) -- 2.30.2

1 0

[PATCH AUTOSEL 5.12 14/33] ASoC: AMD Renoir: Remove fix for DMI entry on Lenovo 2020 platforms
by Sasha Levin 15 Jun '21

15 Jun '21

From: Mark Pearson <markpearson(a)lenovo.com> [ Upstream commit 320232caf1d8febea17312dab4b2dfe02e033520 ] Unfortunately the previous patch to fix issues using the AMD ACP bridge has the side effect of breaking the dmic in other cases and needs to be reverted. Removing the changes while we revisit the fix and find something better. Apologies for the churn. Suggested-by: Gabriel Craciunescu <unix.or.die(a)gmail.com> Signed-off-by: Mark Pearson <markpearson(a)lenovo.com> Link: https://lore.kernel.org/r/20210602171251.3243-1-markpearson@lenovo.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/amd/renoir/rn-pci-acp3x.c | 35 ----------------------------- 1 file changed, 35 deletions(-) diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index 47a4dfd81a46..050a61fe9693 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -199,41 +199,6 @@ static const struct dmi_system_id rn_acp_quirk_table[] = { DMI_EXACT_MATCH(DMI_BOARD_NAME, "20NLCTO1WW"), } }, - { - /* Lenovo ThinkPad P14s Gen 1 (20Y1) */ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20Y1"), - } - }, - { - /* Lenovo ThinkPad T14s Gen1 */ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20UH"), - } - }, - { - /* Lenovo ThinkPad T14s Gen1 Campus */ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20UJ"), - } - }, - { - /* Lenovo ThinkPad T14 Gen 1*/ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20UD"), - } - }, - { - /* Lenovo ThinkPad X13 Gen 1*/ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20UF"), - } - }, {} }; -- 2.30.2

1 0

[PATCH AUTOSEL 5.12 07/33] ASoC: fsl-asoc-card: Set .owner attribute when registering card.
by Sasha Levin 15 Jun '21

15 Jun '21

From: Nicolas Cavallari <nicolas.cavallari(a)green-communications.fr> [ Upstream commit a8437f05384cb472518ec21bf4fffbe8f0a47378 ] Otherwise, when compiled as module, a WARN_ON is triggered: WARNING: CPU: 0 PID: 5 at sound/core/init.c:208 snd_card_new+0x310/0x39c [snd] [...] CPU: 0 PID: 5 Comm: kworker/0:0 Not tainted 5.10.39 #1 Hardware name: Freescale i.MX6 Quad/DualLite (Device Tree) Workqueue: events deferred_probe_work_func [<c0111988>] (unwind_backtrace) from [<c010c8ac>] (show_stack+0x10/0x14) [<c010c8ac>] (show_stack) from [<c092784c>] (dump_stack+0xdc/0x104) [<c092784c>] (dump_stack) from [<c0129710>] (__warn+0xd8/0x114) [<c0129710>] (__warn) from [<c0922a48>] (warn_slowpath_fmt+0x5c/0xc4) [<c0922a48>] (warn_slowpath_fmt) from [<bf0496f8>] (snd_card_new+0x310/0x39c [snd]) [<bf0496f8>] (snd_card_new [snd]) from [<bf1d7df8>] (snd_soc_bind_card+0x334/0x9c4 [snd_soc_core]) [<bf1d7df8>] (snd_soc_bind_card [snd_soc_core]) from [<bf1e9cd8>] (devm_snd_soc_register_card+0x30/0x6c [snd_soc_core]) [<bf1e9cd8>] (devm_snd_soc_register_card [snd_soc_core]) from [<bf22d964>] (fsl_asoc_card_probe+0x550/0xcc8 [snd_soc_fsl_asoc_card]) [<bf22d964>] (fsl_asoc_card_probe [snd_soc_fsl_asoc_card]) from [<c060c930>] (platform_drv_probe+0x48/0x98) [...] Signed-off-by: Nicolas Cavallari <nicolas.cavallari(a)green-communications.fr> Acked-by: Shengjiu Wang <shengjiu.wang(a)gmail.com> Link: https://lore.kernel.org/r/20210527163409.22049-1-nicolas.cavallari@green-co… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl-asoc-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index f62f81ceab0d..9dcbe5d5a428 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -732,6 +732,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Initialize sound card */ priv->pdev = pdev; priv->card.dev = &pdev->dev; + priv->card.owner = THIS_MODULE; ret = snd_soc_of_parse_card_name(&priv->card, "model"); if (ret) { snprintf(priv->name, sizeof(priv->name), "%s-audio", -- 2.30.2

1 0

[PATCH AUTOSEL 5.12 05/33] ASoC: rt5659: Fix the lost powers for the HDA header
by Sasha Levin 15 Jun '21

15 Jun '21

From: Jack Yu <jack.yu(a)realtek.com> [ Upstream commit 6308c44ed6eeadf65c0a7ba68d609773ed860fbb ] The power of "LDO2", "MICBIAS1" and "Mic Det Power" were powered off after the DAPM widgets were added, and these powers were set by the JD settings "RT5659_JD_HDA_HEADER" in the probe function. In the codec probe function, these powers were ignored to prevent them controlled by DAPM. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Signed-off-by: Jack Yu <jack.yu(a)realtek.com> Message-Id: <15fced51977b458798ca4eebf03dafb9(a)realtek.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5659.c | 26 +++++++++++++++++++++----- 1 file changed, 21 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 91a4ef7f620c..a9b079d56fd6 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -2433,13 +2433,18 @@ static int set_dmic_power(struct snd_soc_dapm_widget *w, return 0; } -static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { +static const struct snd_soc_dapm_widget rt5659_particular_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("LDO2", RT5659_PWR_ANLG_3, RT5659_PWR_LDO2_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0, - NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT, + 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5659_PWR_VOL, RT5659_PWR_MIC_DET_BIT, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0, + NULL, 0), SND_SOC_DAPM_SUPPLY("Mono Vref", RT5659_PWR_ANLG_1, RT5659_PWR_VREF3_BIT, 0, NULL, 0), @@ -2464,8 +2469,6 @@ static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { RT5659_ADC_MONO_R_ASRC_SFT, 0, NULL, 0), /* Input Side */ - SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT, - 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5659_PWR_ANLG_2, RT5659_PWR_MB2_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS3", RT5659_PWR_ANLG_2, RT5659_PWR_MB3_BIT, @@ -3660,10 +3663,23 @@ static int rt5659_set_bias_level(struct snd_soc_component *component, static int rt5659_probe(struct snd_soc_component *component) { + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component); rt5659->component = component; + switch (rt5659->pdata.jd_src) { + case RT5659_JD_HDA_HEADER: + break; + + default: + snd_soc_dapm_new_controls(dapm, + rt5659_particular_dapm_widgets, + ARRAY_SIZE(rt5659_particular_dapm_widgets)); + break; + } + return 0; } -- 2.30.2

1 0

Undefined symbol in static library
by GitHub issues - opened 15 Jun '21

15 Jun '21

alsa-project/alsa-lib issue #157 was opened from jpalus: There is a mismatch between declaration: https://github.com/alsa-project/alsa-lib/blob/f4c061f349188c548497607efd462… and actual definition: https://github.com/alsa-project/alsa-lib/blob/f4c061f349188c548497607efd462… resulting in undefined symbol in static library. Judging by convention used in other files I suppose the latter should be renamed to match declaration. Issue URL : https://github.com/alsa-project/alsa-lib/issues/157 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH -next] ASoC: codecs: wcd938x: constify static struct snd_soc_dai_ops
by Pu Lehui 15 Jun '21

15 Jun '21

The snd_soc_dai_ops structures is only stored in the ops field of a snd_soc_dai_driver structure, so make the snd_soc_dai_ops structure const to allow the compiler to put it in read-only memory. Signed-off-by: Pu Lehui <pulehui(a)huawei.com> --- sound/soc/codecs/wcd938x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index 2cf6145b7219..8ec9357fd855 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -3425,7 +3425,7 @@ static int wcd938x_codec_set_sdw_stream(struct snd_soc_dai *dai, } -static struct snd_soc_dai_ops wcd938x_sdw_dai_ops = { +static const struct snd_soc_dai_ops wcd938x_sdw_dai_ops = { .hw_params = wcd938x_codec_hw_params, .hw_free = wcd938x_codec_free, .set_sdw_stream = wcd938x_codec_set_sdw_stream, -- 2.17.1

2 1

[PATCH 0/3] ASoC: tlv320aic32x4: Add support for TAS2505
by Claudius Heine 15 Jun '21

15 Jun '21

Hi, this is an early patchset that provides support for the TAS2505 by the tlv320aic32x4 driver. Playback with aplay and the right mixer settings seems to be working, but some mixer settings either cause distortions or popping. Any input on how to prevent that is welcome. Also with this change TAS2505 is compiled in unconditionally whenever SND_SOC_TLV320AIC32X4_I2C is enabled. Maybe an additional option for the TAS2505 should be added, but then we probably also need a setting to provide only support for the TAS2505 and not the TLV320AIC32X4. Not sure how to set that up in the best way. This is my first time working with the sound subsystem, so expect stupidity and obvious mistakes. regards, Claudius Claudius Heine (3): ASoC: tlv320aic32x4: prepare driver for different device variants ASoC: tlv320aic32x4: add support for TAS2505 ASoC: tlv320aic32x4: dt-bindings: add TAS2505 to compatible .../bindings/sound/tlv320aic32x4.txt | 1 + sound/soc/codecs/tlv320aic32x4-i2c.c | 22 ++- sound/soc/codecs/tlv320aic32x4-spi.c | 23 ++- sound/soc/codecs/tlv320aic32x4.c | 173 +++++++++++++++++- sound/soc/codecs/tlv320aic32x4.h | 10 + 5 files changed, 219 insertions(+), 10 deletions(-) base-commit: 009c9aa5be652675a06d5211e1640e02bbb1c33d -- 2.32.0

2 9

[PATCH] ASoC: dt-bindings: fsl-sai: Add compatible string for imx8mm/8mn/8mp/8ulp
by Shengjiu Wang 15 Jun '21

15 Jun '21

Add compatible string for imx8mm/8mn/8mp/8ulp, these platforms all support SAI IP. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl-sai.txt | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 0dc83cc4a236..c71c5861d787 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -9,8 +9,10 @@ Required properties: - compatible : Compatible list, contains "fsl,vf610-sai", "fsl,imx6sx-sai", "fsl,imx6ul-sai", - "fsl,imx7ulp-sai", "fsl,imx8mq-sai" or - "fsl,imx8qm-sai". + "fsl,imx7ulp-sai", "fsl,imx8mq-sai", + "fsl,imx8qm-sai", "fsl,imx8mm-sai", + "fsl,imx8mn-sai", "fsl,imx8mp-sai", or + "fsl,imx8ulp-sai". - reg : Offset and length of the register set for the device. -- 2.27.0

3 2

[PATCH v6 0/4] Unify NVIDIA Tegra ASoC machine drivers
by Dmitry Osipenko 15 Jun '21

15 Jun '21

This series squashes all the ASoC machine drivers into a single one, this change was suggested by Jon Hunter. It also sets driver_name and components string of each card, allowing userspace alsa-lib to find UCMs at predictable path. Changelog: v6: - Fixed missed configuration of AC97 clock rate for the WM9712 codec in the unified driver. - Added new patch that removes now obsolete "utils" helpers and moves code into the unified driver. ASoC: tegra: Squash utils into common machine driver v5: - The v4 removed the customization of components string for Nexus 7, but I missed to remove the "components" hook which is unused now, it's removed in v5 for consistency. - Slightly improved naming of the common 12MHz MCLK rate function to make it more consistent with the rest of the driver functions. v4: - Moved out mclk_rate callback that is currently used only by WM8903 machine driver from the common driver. This was suggested by Jon Hunter. - Dropped patch which was setting custom components string for Nexus 7. Jaroslav Kysela wants it to be specified in a device-tree, but the components string doesn't have a firm specification for today. It's better to drop this change for now since it's optional anyways. - Fixed compilation error that was reported by kernel robot for v3. - Jaroslav Kysela merged alsa-ucm-conf PR [1] which added UCMs for Nexus 7 and Acer A500. The UCMs are fully working using a combination of updated kernel + alsa-ucm-conf master + alsa-lib master, meaning that they will work with the next releases of kernel and ALSA userspace upstream packages. - Added ack from Jaroslav Kysela to the "Specify components string for each card" patch that he gave to v3. v3: - Added components string as was suggested by Jaroslav Kysela to v2. - Renamed MCLK rate function that is used by max98090 and other codecs to make it look more generic. Added option for specifying CLK ID per device. This all was suggested by Jon Hunter to v2. v2: - Dropped use of of_device_compatible_match(), like it was suggested by Rob Herring in a review comment to v1. - Added patch that sets card's driver_name of as Tegra ASoC drivers. In a comment to v1 Jaroslav Kysela suggested that the Tegra drivers don't set the card name properly and he was right. I opened pull request with the new Tegra UCMs and updated lookup paths for older UCMs [1]. [1] https://github.com/alsa-project/alsa-ucm-conf/pull/92 Dmitry Osipenko (4): ASoC: tegra: Set driver_name=tegra for all machine drivers ASoC: tegra: Unify ASoC machine drivers ASoC: tegra: Specify components string for each card ASoC: tegra: Squash utils into common machine driver sound/soc/tegra/Kconfig | 12 + sound/soc/tegra/Makefile | 19 +- sound/soc/tegra/tegra_alc5632.c | 259 -------- sound/soc/tegra/tegra_asoc_machine.c | 854 +++++++++++++++++++++++++++ sound/soc/tegra/tegra_asoc_machine.h | 49 ++ sound/soc/tegra/tegra_asoc_utils.c | 225 ------- sound/soc/tegra/tegra_asoc_utils.h | 38 -- sound/soc/tegra/tegra_max98090.c | 276 --------- sound/soc/tegra/tegra_rt5640.c | 222 ------- sound/soc/tegra/tegra_rt5677.c | 324 ---------- sound/soc/tegra/tegra_sgtl5000.c | 211 ------- sound/soc/tegra/tegra_wm8753.c | 185 ------ sound/soc/tegra/tegra_wm8903.c | 351 +++-------- sound/soc/tegra/tegra_wm9712.c | 166 ------ sound/soc/tegra/trimslice.c | 172 ------ 15 files changed, 996 insertions(+), 2367 deletions(-) delete mode 100644 sound/soc/tegra/tegra_alc5632.c create mode 100644 sound/soc/tegra/tegra_asoc_machine.c create mode 100644 sound/soc/tegra/tegra_asoc_machine.h delete mode 100644 sound/soc/tegra/tegra_asoc_utils.c delete mode 100644 sound/soc/tegra/tegra_asoc_utils.h delete mode 100644 sound/soc/tegra/tegra_max98090.c delete mode 100644 sound/soc/tegra/tegra_rt5640.c delete mode 100644 sound/soc/tegra/tegra_rt5677.c delete mode 100644 sound/soc/tegra/tegra_sgtl5000.c delete mode 100644 sound/soc/tegra/tegra_wm8753.c delete mode 100644 sound/soc/tegra/tegra_wm9712.c delete mode 100644 sound/soc/tegra/trimslice.c -- 2.30.2

2 6

[PATCH] ASoC: dt-bindings: fsl, spdif: Add compatible string for imx8ulp
by Shengjiu Wang 15 Jun '21

15 Jun '21

Add compatible string for imx8ulp, which supports spdif module Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl,spdif.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml index 4454aca34d56..f226ec13167a 100644 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml @@ -25,6 +25,7 @@ properties: - fsl,imx8mq-spdif - fsl,imx8mm-spdif - fsl,imx8mn-spdif + - fsl,imx8ulp-spdif reg: maxItems: 1 -- 2.27.0

3 2

Xone43C: USB communication
by Roman Muller 15 Jun '21

15 Jun '21

Hello Gerardo, hello all of you, > Is your capture complete from a software standpoint? i.e. are you > starting Serato or Traktor on the VM while doing the usbmon capture? Not tested yet, I need to get a few items for that and investigate the matter! It is complete up to loading the Win10 driver (which has been supplied by the manufacturer). Maybe I am lucky and this is already sufficient. Have a nice day, Roman

2 1

[PATCH -next] ASoC: axi-spdif: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 14 Jun '21

14 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/adi/axi-spdif.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/adi/axi-spdif.c b/sound/soc/adi/axi-spdif.c index 9b3d81c41c8c..8d4a6cb4e5c5 100644 --- a/sound/soc/adi/axi-spdif.c +++ b/sound/soc/adi/axi-spdif.c @@ -189,8 +189,7 @@ static int axi_spdif_probe(struct platform_device *pdev) platform_set_drvdata(pdev, spdif); - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_ioremap_resource(&pdev->dev, res); + base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(base)) return PTR_ERR(base); -- 2.25.1

2 1

[PATCH -next] ASoC: codecs: wcd: Remove unneeded semicolon
by Yang Li 14 Jun '21

14 Jun '21

Eliminate the following coccicheck warning: ./sound/soc/codecs/wcd-mbhc-v2.c:990:2-3: Unneeded semicolon. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- sound/soc/codecs/wcd-mbhc-v2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd-mbhc-v2.c b/sound/soc/codecs/wcd-mbhc-v2.c index dee9410..405128c 100644 --- a/sound/soc/codecs/wcd-mbhc-v2.c +++ b/sound/soc/codecs/wcd-mbhc-v2.c @@ -987,7 +987,7 @@ static void wcd_mbhc_adc_update_fsm_source(struct wcd_mbhc *mbhc, wcd_mbhc_write_field(mbhc, WCD_MBHC_BTN_ISRC_CTL, 0); break; - }; + } } static void wcd_mbhc_bcs_enable(struct wcd_mbhc *mbhc, int plug_type, bool enable) -- 1.8.3.1

3 2

[PATCH -next] ASoC: mchp-i2s-mcc: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 14 Jun '21

14 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/atmel/mchp-i2s-mcc.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c index 673bc16cb46a..8988f024a732 100644 --- a/sound/soc/atmel/mchp-i2s-mcc.c +++ b/sound/soc/atmel/mchp-i2s-mcc.c @@ -1008,8 +1008,7 @@ static int mchp_i2s_mcc_probe(struct platform_device *pdev) if (!dev) return -ENOMEM; - mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_ioremap_resource(&pdev->dev, mem); + base = devm_platform_get_and_ioremap_resource(pdev, 0, &mem); if (IS_ERR(base)) return PTR_ERR(base); -- 2.25.1

2 1

[RESEND PATCH] ASoC: rt5640: Make codec selectable
by Martin Blumenstingl 14 Jun '21

14 Jun '21

The Realtek rt5640 codec driver can be used with the generic sound card drivers, so it should be selectable. For example, with the addition of #sound-dai-cells = <0> property in DT, it can be used with simple and graph card drivers. Signed-off-by: Martin Blumenstingl <martin.blumenstingl(a)googlemail.com> --- Re-sent because my last mail did not make it to the list This will be used in arch/arm/boot/dts/meson8b-ec100.dts sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2a7b3e363069..96fc0144f9fc 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1180,7 +1180,7 @@ config SND_SOC_RT5631 depends on I2C config SND_SOC_RT5640 - tristate + tristate "Realtek RT5640/RT5639 Codec" depends on I2C config SND_SOC_RT5645 -- 2.31.1

2 1

[PATCH] ASoC: soc-pcm: fix the return value in dpcm_apply_symmetry()
by Jaroslav Kysela 14 Jun '21

14 Jun '21

In case, where the loops are not executed for a reason, the uninitialized variable 'err' is returned to the caller. Make code fully predictible and assign zero in the declaration. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Mark Brown <broonie(a)kernel.org> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8659089a87a0..46513bb97904 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1700,7 +1700,7 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream, struct snd_soc_dpcm *dpcm; struct snd_soc_pcm_runtime *fe = asoc_substream_to_rtd(fe_substream); struct snd_soc_dai *fe_cpu_dai; - int err; + int err = 0; int i; /* apply symmetry for FE */ -- 2.30.2

2 1

Re: [PATCH] ASoC:fsl_easrc:Remove superfluous error message around platform_get_irq()
by Mark Brown 14 Jun '21

14 Jun '21

On Thu, 10 Jun 2021 20:50:52 +0800, Zhongjun Tan wrote: > Clean up the check for irq.dev_err is superfluous as platform_get_irq() > already prints an error.Remove curly braces to confirm to styling > requirements. Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/1] ASoC:fsl_easrc:Remove superfluous error message around platform_get_irq() commit: 4d5f3a096f3d9e7067c7c2e730d989668e06d552 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 v8 0/9] ASoC: codecs: add wcd938x support
by Srinivas Kandagatla 14 Jun '21

14 Jun '21

This patchset adds support for Qualcomm WCD938X codec. Qualcomm WCD9380/WCD9385 Codec is a standalone Hi-Fi audio codec IC connected over SoundWire. This device has two SoundWire devices, RX and TX respectively supporting 4 x ADCs, ClassH, Ear, Aux PA, 2xHPH, 7 x TX diff inputs, 8 DMICs and MBHC. Eventhough this device has two SoundWire devices, only tx device has access to main codec Control/Status Registers! For codec driver to be functional it would need both tx and rx Soundwire devices to be up and this is taken care by using device component framework and device-links are used to ensure proper pm dependencies. Ex tx does not enter suspend before rx or codec is suspended. This patchset along with other SoundWire patches on the list have been tested on SM8250 MTP device. Thanks, srini Changes since v7: - Fixed errors reported by kernel test robot by moving SoundWire specific bits to sdw driver. Srinivas Kandagatla (9): ASoC: dt-bindings: wcd938x: add bindings for wcd938x ASoC: codecs: wcd-clsh: add new version support ASoC: codecs: wcd938x: add basic driver ASoC: dt-bindings: wcd938x-sdw: add bindings for wcd938x-sdw ASoC: codecs: wcd938x-sdw: add SoundWire driver ASoC: codecs: wcd938x: add basic controls ASoC: codecs: wcd938x: add playback dapm widgets ASoC: codecs: wcd938x: add capture dapm widgets ASoC: codecs: wcd938x: add audio routing .../bindings/sound/qcom,wcd938x-sdw.yaml | 70 + .../bindings/sound/qcom,wcd938x.yaml | 146 + sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/wcd-clsh-v2.c | 348 +- sound/soc/codecs/wcd-clsh-v2.h | 16 + sound/soc/codecs/wcd938x-sdw.c | 315 ++ sound/soc/codecs/wcd938x.c | 3743 +++++++++++++++++ sound/soc/codecs/wcd938x.h | 720 ++++ 9 files changed, 5365 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd938x-sdw.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml create mode 100644 sound/soc/codecs/wcd938x-sdw.c create mode 100644 sound/soc/codecs/wcd938x.c create mode 100644 sound/soc/codecs/wcd938x.h -- 2.21.0

2 14

[PATCH -next] ASoC: atmel-i2s: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 14 Jun '21

14 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/atmel/atmel-i2s.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index 584656cc7d3c..e8fe1a7a4d83 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -610,8 +610,7 @@ static int atmel_i2s_probe(struct platform_device *pdev) dev->caps = match->data; /* Map I/O registers. */ - mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_ioremap_resource(&pdev->dev, mem); + base = devm_platform_get_and_ioremap_resource(pdev, 0, &mem); if (IS_ERR(base)) return PTR_ERR(base); -- 2.25.1

2 1

[PATCH] dt-bindings: ASoC: sgtl5000: Add audio-graph-card port
by Marek Vasut 14 Jun '21

14 Jun '21

The SGTL5000 codec can be connected via audio-graph-card, add the missing port: entry into the bindings. Signed-off-by: Marek Vasut <marex(a)denx.de> Cc: Fabio Estevam <festevam(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Rob Herring <robh+dt(a)kernel.org> Cc: kernel(a)dh-electronics.com Cc: alsa-devel(a)alsa-project.org To: devicetree(a)vger.kernel.org --- Documentation/devicetree/bindings/sound/sgtl5000.yaml | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.yaml b/Documentation/devicetree/bindings/sound/sgtl5000.yaml index 70b4a8831073..e762c320b574 100644 --- a/Documentation/devicetree/bindings/sound/sgtl5000.yaml +++ b/Documentation/devicetree/bindings/sound/sgtl5000.yaml @@ -75,6 +75,10 @@ properties: $ref: "/schemas/types.yaml#/definitions/uint32" enum: [ 0, 1, 2, 3 ] + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + required: - compatible - reg -- 2.30.2

2 1

[PATCH v9 0/9] ASoC: codecs: add wcd938x support
by Srinivas Kandagatla 14 Jun '21

14 Jun '21

This patchset adds support for Qualcomm WCD938X codec. Qualcomm WCD9380/WCD9385 Codec is a standalone Hi-Fi audio codec IC connected over SoundWire. This device has two SoundWire devices, RX and TX respectively supporting 4 x ADCs, ClassH, Ear, Aux PA, 2xHPH, 7 x TX diff inputs, 8 DMICs and MBHC. Eventhough this device has two SoundWire devices, only tx device has access to main codec Control/Status Registers! For codec driver to be functional it would need both tx and rx Soundwire devices to be up and this is taken care by using device component framework and device-links are used to ensure proper pm dependencies. Ex tx does not enter suspend before rx or codec is suspended. This patchset along with other SoundWire patches on the list have been tested on SM8250 MTP device. Thanks, srini Changes since v8: - moved Kconfig and Makefile changes to last patch as suggested by Mark - removed array of enums and used static entries instead. Suggested by Mark - return true if put succeeds, Suggested by Mark - removed some unneeded semi-colons in switch Srinivas Kandagatla (9): ASoC: dt-bindings: wcd938x: add bindings for wcd938x ASoC: codecs: wcd-clsh: add new version support ASoC: codecs: wcd938x: add basic driver ASoC: dt-bindings: wcd938x-sdw: add bindings for wcd938x-sdw ASoC: codecs: wcd938x-sdw: add SoundWire driver ASoC: codecs: wcd938x: add basic controls ASoC: codecs: wcd938x: add playback dapm widgets ASoC: codecs: wcd938x: add capture dapm widgets ASoC: codecs: wcd938x: add audio routing and Kconfig .../bindings/sound/qcom,wcd938x-sdw.yaml | 70 + .../bindings/sound/qcom,wcd938x.yaml | 146 + sound/soc/codecs/Kconfig | 14 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/wcd-clsh-v2.c | 348 +- sound/soc/codecs/wcd-clsh-v2.h | 16 + sound/soc/codecs/wcd938x-sdw.c | 315 ++ sound/soc/codecs/wcd938x.c | 3753 +++++++++++++++++ sound/soc/codecs/wcd938x.h | 720 ++++ 9 files changed, 5376 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd938x-sdw.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml create mode 100644 sound/soc/codecs/wcd938x-sdw.c create mode 100644 sound/soc/codecs/wcd938x.c create mode 100644 sound/soc/codecs/wcd938x.h -- 2.21.0

2 10

[PATCH] ASoC: rt5645: Avoid upgrading static warnings to errors
by Mark Brown 14 Jun '21

14 Jun '21

One of the fixes reverted as part of the UMN fallout was actually fine, however rather than undoing the revert the process that handled all this stuff resulted in a patch which attempted to add extra error checks instead. Unfortunately this new change wasn't really based on a good understanding of the subsystem APIs and bypassed the usual patch flow without ensuring it was reviewed by people with subsystem knowledge and was merged as a fix rather than during the merge window. The effect of the new fix is to upgrade what were previously warnings on static data in the code to hard errors on that data. If this actually happens then it would break existing systems, if it doesn't happen then the change has no effect so this was not a safe change to apply as a fix to the release candidates. Since the new code has not been tested and doesn't in practice improve error handling revert it instead, and also drop the original revert since the original fix was fine. This takes the driver back to what it was in -rc1. Fixes: 5e70b8e22b64e ("ASoC: rt5645: add error checking to rt5645_probe function") Fixes: 1e0ce84215dbf ("Revert "ASoC: rt5645: fix a NULL pointer dereference") Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Phillip Potter <phil(a)philpotter.co.uk> --- sound/soc/codecs/rt5645.c | 49 +++++++++------------------------------ 1 file changed, 11 insertions(+), 38 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 438fa18bcb55..9408ee63cb26 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3388,44 +3388,30 @@ static int rt5645_probe(struct snd_soc_component *component) { struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component); - int ret = 0; rt5645->component = component; switch (rt5645->codec_type) { case CODEC_TYPE_RT5645: - ret = snd_soc_dapm_new_controls(dapm, + snd_soc_dapm_new_controls(dapm, rt5645_specific_dapm_widgets, ARRAY_SIZE(rt5645_specific_dapm_widgets)); - if (ret < 0) - goto exit; - - ret = snd_soc_dapm_add_routes(dapm, + snd_soc_dapm_add_routes(dapm, rt5645_specific_dapm_routes, ARRAY_SIZE(rt5645_specific_dapm_routes)); - if (ret < 0) - goto exit; - if (rt5645->v_id < 3) { - ret = snd_soc_dapm_add_routes(dapm, + snd_soc_dapm_add_routes(dapm, rt5645_old_dapm_routes, ARRAY_SIZE(rt5645_old_dapm_routes)); - if (ret < 0) - goto exit; } break; case CODEC_TYPE_RT5650: - ret = snd_soc_dapm_new_controls(dapm, + snd_soc_dapm_new_controls(dapm, rt5650_specific_dapm_widgets, ARRAY_SIZE(rt5650_specific_dapm_widgets)); - if (ret < 0) - goto exit; - - ret = snd_soc_dapm_add_routes(dapm, + snd_soc_dapm_add_routes(dapm, rt5650_specific_dapm_routes, ARRAY_SIZE(rt5650_specific_dapm_routes)); - if (ret < 0) - goto exit; break; } @@ -3433,17 +3419,9 @@ static int rt5645_probe(struct snd_soc_component *component) /* for JD function */ if (rt5645->pdata.jd_mode) { - ret = snd_soc_dapm_force_enable_pin(dapm, "JD Power"); - if (ret < 0) - goto exit; - - ret = snd_soc_dapm_force_enable_pin(dapm, "LDO2"); - if (ret < 0) - goto exit; - - ret = snd_soc_dapm_sync(dapm); - if (ret < 0) - goto exit; + snd_soc_dapm_force_enable_pin(dapm, "JD Power"); + snd_soc_dapm_force_enable_pin(dapm, "LDO2"); + snd_soc_dapm_sync(dapm); } if (rt5645->pdata.long_name) @@ -3454,14 +3432,9 @@ static int rt5645_probe(struct snd_soc_component *component) GFP_KERNEL); if (!rt5645->eq_param) - ret = -ENOMEM; -exit: - /* - * If there was an error above, everything will be cleaned up by the - * caller if we return an error here. This will be done with a later - * call to rt5645_remove(). - */ - return ret; + return -ENOMEM; + + return 0; } static void rt5645_remove(struct snd_soc_component *component) -- 2.20.1

2 2

[PATCH -next] ASoC: bcm: cygnus_ssp: Use devm_platform_ioremap_resource_byname()
by Yang Yingliang 14 Jun '21

14 Jun '21

Use the devm_platform_ioremap_resource_byname() helper instead of calling platform_get_resource_byname() and devm_ioremap_resource() separately. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/bcm/cygnus-ssp.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c index ba03bb62ba96..fca5a3f2eec5 100644 --- a/sound/soc/bcm/cygnus-ssp.c +++ b/sound/soc/bcm/cygnus-ssp.c @@ -1308,7 +1308,6 @@ static int cygnus_ssp_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; struct device_node *child_node; - struct resource *res; struct cygnus_audio *cygaud; int err; int node_count; @@ -1320,13 +1319,11 @@ static int cygnus_ssp_probe(struct platform_device *pdev) dev_set_drvdata(dev, cygaud); - res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "aud"); - cygaud->audio = devm_ioremap_resource(dev, res); + cygaud->audio = devm_platform_ioremap_resource_byname(pdev, "aud"); if (IS_ERR(cygaud->audio)) return PTR_ERR(cygaud->audio); - res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "i2s_in"); - cygaud->i2s_in = devm_ioremap_resource(dev, res); + cygaud->i2s_in = devm_platform_ioremap_resource_byname(pdev, "i2s_in"); if (IS_ERR(cygaud->i2s_in)) return PTR_ERR(cygaud->i2s_in); -- 2.25.1

2 1

[PATCH] ASoC: fsl_spdif: Fix error handler with pm_runtime_enable
by Shengjiu Wang 14 Jun '21

14 Jun '21

There is error message when defer probe happens: fsl-spdif-dai 2dab0000.spdif: Unbalanced pm_runtime_enable! Fix the error handler with pm_runtime_enable and add fsl_spdif_remove() for pm_runtime_disable. Fixes: 9cb2b3796e08 ("ASoC: fsl_spdif: Add pm runtime function") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_spdif.c | 20 +++++++++++++++++--- 1 file changed, 17 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 2a76714eb8e6..76a7c9581333 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1441,16 +1441,29 @@ static int fsl_spdif_probe(struct platform_device *pdev) &spdif_priv->cpu_dai_drv, 1); if (ret) { dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); - return ret; + goto err_pm_disable; } ret = imx_pcm_dma_init(pdev, IMX_SPDIF_DMABUF_SIZE); - if (ret && ret != -EPROBE_DEFER) - dev_err(&pdev->dev, "imx_pcm_dma_init failed: %d\n", ret); + if (ret) { + dev_err_probe(&pdev->dev, ret, "imx_pcm_dma_init failed\n"); + goto err_pm_disable; + } + + return ret; +err_pm_disable: + pm_runtime_disable(&pdev->dev); return ret; } +static int fsl_spdif_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + + return 0; +} + #ifdef CONFIG_PM static int fsl_spdif_runtime_suspend(struct device *dev) { @@ -1554,6 +1567,7 @@ static struct platform_driver fsl_spdif_driver = { .pm = &fsl_spdif_pm, }, .probe = fsl_spdif_probe, + .remove = fsl_spdif_remove, }; module_platform_driver(fsl_spdif_driver); -- 2.27.0

2 1

[PATCH -next] ASoC: atmel-pdmic: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 14 Jun '21

14 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/atmel/atmel-pdmic.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/atmel/atmel-pdmic.c b/sound/soc/atmel/atmel-pdmic.c index 8e1d8230b180..42117de299e7 100644 --- a/sound/soc/atmel/atmel-pdmic.c +++ b/sound/soc/atmel/atmel-pdmic.c @@ -620,8 +620,7 @@ static int atmel_pdmic_probe(struct platform_device *pdev) return ret; } - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - io_base = devm_ioremap_resource(dev, res); + io_base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(io_base)) return PTR_ERR(io_base); -- 2.25.1

2 1

[PATCH v3] ASoC: qcom: Add four speaker support on MI2S secondary
by Srinivasa Rao Mandadapu 14 Jun '21

14 Jun '21

Add four speaker support on MI2S secondary block by using I2S SD1 line on gpio52 pin, and add channel map control support in the lpass-cpu audio driver. Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- Changes Since V2: -- Added empty entry at the end of channel maps array. Changes Since V1: -- removed set_channel_map/get_channel_map implementation as default kcontrols added in pcm_new API. sound/soc/qcom/lpass-cpu.c | 33 +++++++++++++++++++++++++++++++++ sound/soc/qcom/lpass-sc7180.c | 1 + sound/soc/qcom/lpass.h | 2 ++ 3 files changed, 36 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index c62d2612e8f5..aff39c9f6326 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -29,6 +29,15 @@ #define LPASS_CPU_I2S_SD0_1_2_MASK GENMASK(2, 0) #define LPASS_CPU_I2S_SD0_1_2_3_MASK GENMASK(3, 0) +/* + * Channel maps for Quad channel playbacks on MI2S Secondary + */ +static struct snd_pcm_chmap_elem lpass_quad_chmaps[] = { + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_RL, + SNDRV_CHMAP_FR, SNDRV_CHMAP_RR } }, + { } +}; static int lpass_cpu_init_i2sctl_bitfields(struct device *dev, struct lpaif_i2sctl *i2sctl, struct regmap *map) { @@ -324,6 +333,25 @@ const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); +int lpass_cpu_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + int ret; + struct snd_soc_dai_driver *drv = dai->driver; + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + + if (drvdata->mi2s_playback_sd_mode[dai->id] == LPAIF_I2SCTL_MODE_QUAD01) { + ret = snd_pcm_add_chmap_ctls(rtd->pcm, SNDRV_PCM_STREAM_PLAYBACK, + lpass_quad_chmaps, drv->playback.channels_max, 0, + NULL); + if (ret < 0) + return ret; + } + + return 0; +} +EXPORT_SYMBOL_GPL(lpass_cpu_pcm_new); + int asoc_qcom_lpass_cpu_dai_probe(struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); @@ -856,6 +884,11 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) PTR_ERR(drvdata->mi2s_bit_clk[dai_id])); return PTR_ERR(drvdata->mi2s_bit_clk[dai_id]); } + if (drvdata->mi2s_playback_sd_mode[dai_id] == + LPAIF_I2SCTL_MODE_QUAD01) { + variant->dai_driver[dai_id].playback.channels_min = 4; + variant->dai_driver[dai_id].playback.channels_max = 4; + } } /* Allocation for i2sctl regmap fields */ diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index 8c168d3c589e..77a556b27cf0 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -58,6 +58,7 @@ static struct snd_soc_dai_driver sc7180_lpass_cpu_dai_driver[] = { }, .probe = &asoc_qcom_lpass_cpu_dai_probe, .ops = &asoc_qcom_lpass_cpu_dai_ops, + .pcm_new = lpass_cpu_pcm_new, }, { .id = LPASS_DP_RX, .name = "Hdmi", diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 83b2e08ade06..623ddccdafff 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -259,5 +259,7 @@ void asoc_qcom_lpass_cpu_platform_shutdown(struct platform_device *pdev); int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev); int asoc_qcom_lpass_cpu_dai_probe(struct snd_soc_dai *dai); extern const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops; +int lpass_cpu_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai); #endif /* __LPASS_H__ */ -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[PATCH -next] ASoC: axi-i2s: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 14 Jun '21

14 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/adi/axi-i2s.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c index aa082131fb90..1289cb4e2988 100644 --- a/sound/soc/adi/axi-i2s.c +++ b/sound/soc/adi/axi-i2s.c @@ -198,8 +198,7 @@ static int axi_i2s_probe(struct platform_device *pdev) axi_i2s_parse_of(i2s, pdev->dev.of_node); - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_ioremap_resource(&pdev->dev, res); + base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(base)) return PTR_ERR(base); -- 2.25.1

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[PATCH -next] ASoC: atmel-classd: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 14 Jun '21

14 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/atmel/atmel-classd.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c index 6023369e0f1a..a9f9f449c48c 100644 --- a/sound/soc/atmel/atmel-classd.c +++ b/sound/soc/atmel/atmel-classd.c @@ -558,8 +558,7 @@ static int atmel_classd_probe(struct platform_device *pdev) return ret; } - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - io_base = devm_ioremap_resource(dev, res); + io_base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(io_base)) return PTR_ERR(io_base); -- 2.25.1

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Re: [PATCH] ASoC:fsl_spdif:Remove superfluous error message around platform_get_irq()
by Mark Brown 14 Jun '21

14 Jun '21

On Thu, 10 Jun 2021 12:00:37 +0800, Zhongjun Tan wrote: > The platform_get_irq() prints error message telling that interrupt is > missing, hence there is no need to duplicated that message. Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/1] ASoC:fsl_spdif:Remove superfluous error message around platform_get_irq() commit: 2e8a8adb96a335a04f1697dd4314f5569521328f 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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ALSA 1.2.5.1 bugfix release
by Jaroslav Kysela 14 Jun '21

14 Jun '21

Hello all, the ALSA userspace alsa-lib, alsa-utils and alsa-ucm-conf packages were released. You may download them from the ALSA website http://www.alsa-project.org or directly: HTTP: https://www.alsa-project.org/files/pub FTP: ftp://ftp.alsa-project.org/pub Full list of changes: https://www.alsa-project.org/wiki/Changes_v1.2.5_v1.2.5.1 The fingerprint of the public signing key is: F04D F507 37AC 1A88 4C4B 3D71 8380 596D A6E5 9C91 Have fun, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

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USB sound card issues
by Muni Sekhar 14 Jun '21

14 Jun '21

Hi all, I am using a USB sound card(snd-usb-audio), and it fails while doing the audio recording "arecord: pcm_read:2032: read error: Input/output error". What could be the problem? and is there a workaround for this? $ arecord -l **** List of CAPTURE Hardware Devices **** card 1: BT_AG [BT_002], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 $ arecord --device hw:1,0 --channels 1 --format S16_LE --rate 32000_Hz x.wav Recording WAVE 'x.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono arecord: pcm_read:2032: read error: Input/output error -- Thanks, Sekhar

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added an option to aseqnet to optionally set the midi process name
by GitHub pull_request - reopened 14 Jun '21

14 Jun '21

alsa-project/alsa-utils pull request #95 was reopened from BrainDamage: This option allows to run multiple instances of aseqnet without having to double check the assigned port number, since each one can get spawned with a unique name. The patch is pretty minimal and clean, thopefully here shouldn't be troubles merging it. Request URL : https://github.com/alsa-project/alsa-utils/pull/95 Patch URL : https://github.com/alsa-project/alsa-utils/pull/95.patch Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH] alsa: control_led - fix initialization in the mode show callback
by Jaroslav Kysela 14 Jun '21

14 Jun '21

The str variable should be always initialized before use even if the switch covers all cases. This is a minimalistic fix: Assign NULL, the sprintf() may print '(null)' if something is corrupted. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/core/control_led.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/control_led.c b/sound/core/control_led.c index 861f71f2fab5..764058cc345d 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -397,7 +397,7 @@ static ssize_t mode_show(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_ctl_led *led = container_of(dev, struct snd_ctl_led, dev); - const char *str; + const char *str = NULL; switch (led->mode) { case MODE_FOLLOW_MUTE: str = "follow-mute"; break; -- 2.30.2

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[PATCH 0/7] Prepare EP93xx drivers for Common Clock Framework
by Alexander Sverdlin 14 Jun '21

14 Jun '21

Nikita posted a patch converting EP93xx to use Common Clock Framework. It turns out some cleanup is necessary in the EP93xx drivers to avoid "Enabling unprepared" clock warnings. Patches with stack traces in the commit messages are tested on EP9302. Link: https://lore.kernel.org/patchwork/patch/1435884/ Alexander Sverdlin (7): iio: ep93xx: Prepare clock before using it spi: spi-ep93xx: Prepare clock before using it Input: ep93xx_keypad: Prepare clock before using it video: ep93xx: Prepare clock before using it dmaengine: ep93xx: Prepare clock before using it ASoC: cirrus: i2s: Prepare clock before using it pwm: ep93xx: Prepare clock before using it drivers/dma/ep93xx_dma.c | 6 +++--- drivers/iio/adc/ep93xx_adc.c | 6 +++--- drivers/input/keyboard/ep93xx_keypad.c | 4 ++-- drivers/pwm/pwm-ep93xx.c | 12 ++++++------ drivers/spi/spi-ep93xx.c | 4 ++-- drivers/video/fbdev/ep93xx-fb.c | 4 ++-- sound/soc/cirrus/ep93xx-i2s.c | 12 ++++++------ 7 files changed, 24 insertions(+), 24 deletions(-) -- 2.32.0

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[PATCH] ALSA: firewire: fill Kconfig with entries for devices based on DICE ASICs
by Takashi Sakamoto 14 Jun '21

14 Jun '21

Nowadays a lot of devices are supported by ALSA dice driver. This commit fills Kconfig with entries for them. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/Kconfig | 21 +++++++++++++++++++-- 1 file changed, 19 insertions(+), 2 deletions(-) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index fc6858131b59..a08a0dbfc951 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -18,8 +18,25 @@ config SND_DICE select SND_HWDEP select SND_FIREWIRE_LIB help - Say Y here to include support for many DACs based on the DICE - chip family (DICE-II/Jr/Mini) which TC Applied Technologies produces. + Say Y here to include support for devices based on the DICE chip family + (DICE-II/TCD2210(Mini)/TCD2220(Jr.)) which TC Applied Technologies (TCAT) produced. + * Allen and Heath Zed R16 + * Alesis iO 14/26 FireWire, MasterControl, MultiMix 8/12/16 FireWire + * Avid Mbox 3 Pro + * FlexRadio Systems FLEX-3000, FLEX-5000 + * Focusrite Liquid Saffire 56 + * Focusrite Saffire Pro 14, Pro 24, Pro 24 DSP, Pro 26, Pro 40(TCD2220) + * Harman Music Group Lexicon I-ONIX FW810S + * Loud Technologies Mackie Onyx Blackbird, Onyx 820i/1220i/1620i/1640i (latter models) + * M-Audio ProFire 610/2626 + * Mytek Stereo192-DSD DAC + * Midas Klark Teknik VeniceF series + * PreSonus FireStudio, FireStudio Mobile, FireStudio Project, FireStudio Tube + * PreSonus StudioLive 16.4.2, 16.0.2, 24.4.2, 32.4.2 + * Solid State Logic Duende Classic, Duende Mini + * TC Electronic Studio Konnekt 48, Konnekt 24D, Konnekt Live, Impact Twin + * TC Electronic Digital Konnekt x32, Desktop Konnekt 6 + * Weiss Engineering ADC2, Vesta, Minerva, AFI1, DAC1, INT202, DAC202 To compile this driver as a module, choose M here: the module will be called snd-dice. -- 2.30.2

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Android porting patches
by GitHub pull_request - opened 14 Jun '21

14 Jun '21

alsa-project/alsa-lib pull request #156 was opened from cwhuang: This is a series of patches to make alsa-lib buildable with Android source tree. Test on with Android 8.1 and later. Request URL : https://github.com/alsa-project/alsa-lib/pull/156 Patch URL : https://github.com/alsa-project/alsa-lib/pull/156.patch Repository URL: https://github.com/alsa-project/alsa-lib

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