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.

4 3

[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

2 1

[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

2 1

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

1 0

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.

1 0

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

3 10

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

1 0

[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

2 1

[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

3 3

[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

2 1

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

1 0

[PATCH v4] soundwire: intel: move to auxiliary bus
by Bard Liao 14 Jun '21

14 Jun '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Now that the auxiliary_bus exists, there's no reason to use platform devices as children of a PCI device any longer. This patch refactors the code by extending a basic auxiliary device with Intel link-specific structures that need to be passed between controller and link levels. This refactoring is much cleaner with no need for cross-pointers between device and link structures. Note that the auxiliary bus API has separate init and add steps, which requires more attention in the error unwinding paths. The main loop needs to deal with kfree() and auxiliary_device_uninit() for the current iteration before jumping to the common label which releases everything allocated in prior iterations. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- v2: - add link_dev_register for all kzalloc, device_init, and device_add. v3: - Modify the function description of sdw_intel_probe() which was missing in previous version. v4: - Move intel_link_dev_unregister(ldev) before pm_runtime_put_noidle( ldev->link_res.dev) to fix use-after-free reported by KASAN. --- drivers/soundwire/Kconfig | 1 + drivers/soundwire/intel.c | 56 ++++--- drivers/soundwire/intel.h | 14 +- drivers/soundwire/intel_init.c | 232 +++++++++++++++++++--------- include/linux/soundwire/sdw_intel.h | 6 +- 5 files changed, 202 insertions(+), 107 deletions(-) diff --git a/drivers/soundwire/Kconfig b/drivers/soundwire/Kconfig index 016e74230bb7..2b7795233282 100644 --- a/drivers/soundwire/Kconfig +++ b/drivers/soundwire/Kconfig @@ -25,6 +25,7 @@ config SOUNDWIRE_INTEL tristate "Intel SoundWire Master driver" select SOUNDWIRE_CADENCE select SOUNDWIRE_GENERIC_ALLOCATION + select AUXILIARY_BUS depends on ACPI && SND_SOC help SoundWire Intel Master driver. diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index fd95f94630b1..c11e3d8cd308 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -11,7 +11,7 @@ #include <linux/module.h> #include <linux/interrupt.h> #include <linux/io.h> -#include <linux/platform_device.h> +#include <linux/auxiliary_bus.h> #include <sound/pcm_params.h> #include <linux/pm_runtime.h> #include <sound/soc.h> @@ -1327,11 +1327,14 @@ static int intel_init(struct sdw_intel *sdw) } /* - * probe and init + * probe and init (aux_dev_id argument is required by function prototype but not used) */ -static int intel_master_probe(struct platform_device *pdev) +static int intel_link_probe(struct auxiliary_device *auxdev, + const struct auxiliary_device_id *aux_dev_id) + { - struct device *dev = &pdev->dev; + struct device *dev = &auxdev->dev; + struct sdw_intel_link_dev *ldev = auxiliary_dev_to_sdw_intel_link_dev(auxdev); struct sdw_intel *sdw; struct sdw_cdns *cdns; struct sdw_bus *bus; @@ -1344,14 +1347,14 @@ static int intel_master_probe(struct platform_device *pdev) cdns = &sdw->cdns; bus = &cdns->bus; - sdw->instance = pdev->id; - sdw->link_res = dev_get_platdata(dev); + sdw->instance = auxdev->id; + sdw->link_res = &ldev->link_res; cdns->dev = dev; cdns->registers = sdw->link_res->registers; cdns->instance = sdw->instance; cdns->msg_count = 0; - bus->link_id = pdev->id; + bus->link_id = auxdev->id; sdw_cdns_probe(cdns); @@ -1384,10 +1387,10 @@ static int intel_master_probe(struct platform_device *pdev) return 0; } -int intel_master_startup(struct platform_device *pdev) +int intel_link_startup(struct auxiliary_device *auxdev) { struct sdw_cdns_stream_config config; - struct device *dev = &pdev->dev; + struct device *dev = &auxdev->dev; struct sdw_cdns *cdns = dev_get_drvdata(dev); struct sdw_intel *sdw = cdns_to_intel(cdns); struct sdw_bus *bus = &cdns->bus; @@ -1524,9 +1527,9 @@ int intel_master_startup(struct platform_device *pdev) return ret; } -static int intel_master_remove(struct platform_device *pdev) +static void intel_link_remove(struct auxiliary_device *auxdev) { - struct device *dev = &pdev->dev; + struct device *dev = &auxdev->dev; struct sdw_cdns *cdns = dev_get_drvdata(dev); struct sdw_intel *sdw = cdns_to_intel(cdns); struct sdw_bus *bus = &cdns->bus; @@ -1542,19 +1545,17 @@ static int intel_master_remove(struct platform_device *pdev) snd_soc_unregister_component(dev); } sdw_bus_master_delete(bus); - - return 0; } -int intel_master_process_wakeen_event(struct platform_device *pdev) +int intel_link_process_wakeen_event(struct auxiliary_device *auxdev) { - struct device *dev = &pdev->dev; + struct device *dev = &auxdev->dev; struct sdw_intel *sdw; struct sdw_bus *bus; void __iomem *shim; u16 wake_sts; - sdw = platform_get_drvdata(pdev); + sdw = dev_get_drvdata(dev); bus = &sdw->cdns.bus; if (bus->prop.hw_disabled) { @@ -1976,17 +1977,22 @@ static const struct dev_pm_ops intel_pm = { SET_RUNTIME_PM_OPS(intel_suspend_runtime, intel_resume_runtime, NULL) }; -static struct platform_driver sdw_intel_drv = { - .probe = intel_master_probe, - .remove = intel_master_remove, +static const struct auxiliary_device_id intel_link_id_table[] = { + { .name = "soundwire_intel.link" }, + {}, +}; +MODULE_DEVICE_TABLE(auxiliary, intel_link_id_table); + +static struct auxiliary_driver sdw_intel_drv = { + .probe = intel_link_probe, + .remove = intel_link_remove, .driver = { - .name = "intel-sdw", + /* auxiliary_driver_register() sets .name to be the modname */ .pm = &intel_pm, - } + }, + .id_table = intel_link_id_table }; - -module_platform_driver(sdw_intel_drv); +module_auxiliary_driver(sdw_intel_drv); MODULE_LICENSE("Dual BSD/GPL"); -MODULE_ALIAS("platform:intel-sdw"); -MODULE_DESCRIPTION("Intel Soundwire Master Driver"); +MODULE_DESCRIPTION("Intel Soundwire Link Driver"); diff --git a/drivers/soundwire/intel.h b/drivers/soundwire/intel.h index 06bac8ba14e9..0b47b148da3f 100644 --- a/drivers/soundwire/intel.h +++ b/drivers/soundwire/intel.h @@ -7,7 +7,6 @@ /** * struct sdw_intel_link_res - Soundwire Intel link resource structure, * typically populated by the controller driver. - * @pdev: platform_device * @mmio_base: mmio base of SoundWire registers * @registers: Link IO registers base * @shim: Audio shim pointer @@ -23,7 +22,6 @@ * @list: used to walk-through all masters exposed by the same controller */ struct sdw_intel_link_res { - struct platform_device *pdev; void __iomem *mmio_base; /* not strictly needed, useful for debug */ void __iomem *registers; void __iomem *shim; @@ -48,7 +46,15 @@ struct sdw_intel { #endif }; -int intel_master_startup(struct platform_device *pdev); -int intel_master_process_wakeen_event(struct platform_device *pdev); +int intel_link_startup(struct auxiliary_device *auxdev); +int intel_link_process_wakeen_event(struct auxiliary_device *auxdev); + +struct sdw_intel_link_dev { + struct auxiliary_device auxdev; + struct sdw_intel_link_res link_res; +}; + +#define auxiliary_dev_to_sdw_intel_link_dev(auxiliary_dev) \ + container_of(auxiliary_dev, struct sdw_intel_link_dev, auxdev) #endif /* __SDW_INTEL_LOCAL_H */ diff --git a/drivers/soundwire/intel_init.c b/drivers/soundwire/intel_init.c index 30ce95ec2d70..9e283bef53d2 100644 --- a/drivers/soundwire/intel_init.c +++ b/drivers/soundwire/intel_init.c @@ -12,7 +12,7 @@ #include <linux/interrupt.h> #include <linux/io.h> #include <linux/module.h> -#include <linux/platform_device.h> +#include <linux/auxiliary_bus.h> #include <linux/pm_runtime.h> #include <linux/soundwire/sdw_intel.h> #include "cadence_master.h" @@ -24,28 +24,108 @@ #define SDW_LINK_BASE 0x30000 #define SDW_LINK_SIZE 0x10000 +static void intel_link_dev_release(struct device *dev) +{ + struct auxiliary_device *auxdev = to_auxiliary_dev(dev); + struct sdw_intel_link_dev *ldev = auxiliary_dev_to_sdw_intel_link_dev(auxdev); + + kfree(ldev); +} + +/* alloc, init and add link devices */ +static struct sdw_intel_link_dev *intel_link_dev_register(struct sdw_intel_res *res, + struct sdw_intel_ctx *ctx, + struct fwnode_handle *fwnode, + const char *name, + int link_id) +{ + struct sdw_intel_link_dev *ldev; + struct sdw_intel_link_res *link; + struct auxiliary_device *auxdev; + int ret; + + ldev = kzalloc(sizeof(*ldev), GFP_KERNEL); + if (!ldev) + return ERR_PTR(-ENOMEM); + + auxdev = &ldev->auxdev; + auxdev->name = name; + auxdev->dev.parent = res->parent; + auxdev->dev.fwnode = fwnode; + auxdev->dev.release = intel_link_dev_release; + + /* we don't use an IDA since we already have a link ID */ + auxdev->id = link_id; + + /* + * keep a handle on the allocated memory, to be used in all other functions. + * Since the same pattern is used to skip links that are not enabled, there is + * no need to check if ctx->ldev[i] is NULL later on. + */ + ctx->ldev[link_id] = ldev; + + /* Add link information used in the driver probe */ + link = &ldev->link_res; + link->mmio_base = res->mmio_base; + link->registers = res->mmio_base + SDW_LINK_BASE + + (SDW_LINK_SIZE * link_id); + link->shim = res->mmio_base + SDW_SHIM_BASE; + link->alh = res->mmio_base + SDW_ALH_BASE; + + link->ops = res->ops; + link->dev = res->dev; + + link->clock_stop_quirks = res->clock_stop_quirks; + link->shim_lock = &ctx->shim_lock; + link->shim_mask = &ctx->shim_mask; + link->link_mask = ctx->link_mask; + + /* now follow the two-step init/add sequence */ + ret = auxiliary_device_init(auxdev); + if (ret < 0) { + dev_err(res->parent, "failed to initialize link dev %s link_id %d\n", + name, link_id); + kfree(ldev); + return ERR_PTR(ret); + } + + ret = auxiliary_device_add(&ldev->auxdev); + if (ret < 0) { + dev_err(res->parent, "failed to add link dev %s link_id %d\n", + ldev->auxdev.name, link_id); + /* ldev will be freed with the put_device() and .release sequence */ + auxiliary_device_uninit(&ldev->auxdev); + return ERR_PTR(ret); + } + + return ldev; +} + +static void intel_link_dev_unregister(struct sdw_intel_link_dev *ldev) +{ + auxiliary_device_delete(&ldev->auxdev); + auxiliary_device_uninit(&ldev->auxdev); +} + static int sdw_intel_cleanup(struct sdw_intel_ctx *ctx) { - struct sdw_intel_link_res *link = ctx->links; + struct sdw_intel_link_dev *ldev; u32 link_mask; int i; - if (!link) - return 0; - link_mask = ctx->link_mask; - for (i = 0; i < ctx->count; i++, link++) { + for (i = 0; i < ctx->count; i++) { if (!(link_mask & BIT(i))) continue; - if (link->pdev) { - pm_runtime_disable(&link->pdev->dev); - platform_device_unregister(link->pdev); - } + ldev = ctx->ldev[i]; - if (!link->clock_stop_quirks) - pm_runtime_put_noidle(link->dev); + pm_runtime_disable(&ldev->auxdev.dev); + if (!ldev->link_res.clock_stop_quirks) + pm_runtime_put_noidle(ldev->link_res.dev); + + intel_link_dev_unregister(ldev); } return 0; @@ -91,9 +171,8 @@ EXPORT_SYMBOL_NS(sdw_intel_thread, SOUNDWIRE_INTEL_INIT); static struct sdw_intel_ctx *sdw_intel_probe_controller(struct sdw_intel_res *res) { - struct platform_device_info pdevinfo; - struct platform_device *pdev; struct sdw_intel_link_res *link; + struct sdw_intel_link_dev *ldev; struct sdw_intel_ctx *ctx; struct acpi_device *adev; struct sdw_slave *slave; @@ -116,67 +195,60 @@ static struct sdw_intel_ctx count = res->count; dev_dbg(&adev->dev, "Creating %d SDW Link devices\n", count); - ctx = devm_kzalloc(&adev->dev, sizeof(*ctx), GFP_KERNEL); + /* + * we need to alloc/free memory manually and can't use devm: + * this routine may be called from a workqueue, and not from + * the parent .probe. + * If devm_ was used, the memory might never be freed on errors. + */ + ctx = kzalloc(sizeof(*ctx), GFP_KERNEL); if (!ctx) return NULL; ctx->count = count; - ctx->links = devm_kcalloc(&adev->dev, ctx->count, - sizeof(*ctx->links), GFP_KERNEL); - if (!ctx->links) + + /* + * allocate the array of pointers. The link-specific data is allocated + * as part of the first loop below and released with the auxiliary_device_uninit(). + * If some links are disabled, the link pointer will remain NULL. Given that the + * number of links is small, this is simpler than using a list to keep track of links. + */ + ctx->ldev = kcalloc(ctx->count, sizeof(*ctx->ldev), GFP_KERNEL); + if (!ctx->ldev) { + kfree(ctx); return NULL; + } - ctx->count = count; ctx->mmio_base = res->mmio_base; ctx->link_mask = res->link_mask; ctx->handle = res->handle; mutex_init(&ctx->shim_lock); - link = ctx->links; link_mask = ctx->link_mask; INIT_LIST_HEAD(&ctx->link_list); - /* Create SDW Master devices */ - for (i = 0; i < count; i++, link++) { - if (!(link_mask & BIT(i))) { - dev_dbg(&adev->dev, - "Link %d masked, will not be enabled\n", i); + for (i = 0; i < count; i++) { + if (!(link_mask & BIT(i))) continue; - } - link->mmio_base = res->mmio_base; - link->registers = res->mmio_base + SDW_LINK_BASE - + (SDW_LINK_SIZE * i); - link->shim = res->mmio_base + SDW_SHIM_BASE; - link->alh = res->mmio_base + SDW_ALH_BASE; - - link->ops = res->ops; - link->dev = res->dev; - - link->clock_stop_quirks = res->clock_stop_quirks; - link->shim_lock = &ctx->shim_lock; - link->shim_mask = &ctx->shim_mask; - link->link_mask = link_mask; - - memset(&pdevinfo, 0, sizeof(pdevinfo)); - - pdevinfo.parent = res->parent; - pdevinfo.name = "intel-sdw"; - pdevinfo.id = i; - pdevinfo.fwnode = acpi_fwnode_handle(adev); - pdevinfo.data = link; - pdevinfo.size_data = sizeof(*link); - - pdev = platform_device_register_full(&pdevinfo); - if (IS_ERR(pdev)) { - dev_err(&adev->dev, - "platform device creation failed: %ld\n", - PTR_ERR(pdev)); + /* + * init and add a device for each link + * + * The name of the device will be soundwire_intel.link.[i], + * with the "soundwire_intel" module prefix automatically added + * by the auxiliary bus core. + */ + ldev = intel_link_dev_register(res, + ctx, + acpi_fwnode_handle(adev), + "link", + i); + if (IS_ERR(ldev)) goto err; - } - link->pdev = pdev; - link->cdns = platform_get_drvdata(pdev); + + link = &ldev->link_res; + link->cdns = dev_get_drvdata(&ldev->auxdev.dev); if (!link->cdns) { dev_err(&adev->dev, "failed to get link->cdns\n"); @@ -194,8 +266,7 @@ static struct sdw_intel_ctx num_slaves++; } - ctx->ids = devm_kcalloc(&adev->dev, num_slaves, - sizeof(*ctx->ids), GFP_KERNEL); + ctx->ids = kcalloc(num_slaves, sizeof(*ctx->ids), GFP_KERNEL); if (!ctx->ids) goto err; @@ -213,8 +284,14 @@ static struct sdw_intel_ctx return ctx; err: - ctx->count = i; - sdw_intel_cleanup(ctx); + while (i--) { + if (!(link_mask & BIT(i))) + continue; + ldev = ctx->ldev[i]; + intel_link_dev_unregister(ldev); + } + kfree(ctx->ldev); + kfree(ctx); return NULL; } @@ -222,7 +299,7 @@ static int sdw_intel_startup_controller(struct sdw_intel_ctx *ctx) { struct acpi_device *adev; - struct sdw_intel_link_res *link; + struct sdw_intel_link_dev *ldev; u32 caps; u32 link_mask; int i; @@ -241,27 +318,28 @@ sdw_intel_startup_controller(struct sdw_intel_ctx *ctx) return -EINVAL; } - if (!ctx->links) + if (!ctx->ldev) return -EINVAL; - link = ctx->links; link_mask = ctx->link_mask; /* Startup SDW Master devices */ - for (i = 0; i < ctx->count; i++, link++) { + for (i = 0; i < ctx->count; i++) { if (!(link_mask & BIT(i))) continue; - intel_master_startup(link->pdev); + ldev = ctx->ldev[i]; - if (!link->clock_stop_quirks) { + intel_link_startup(&ldev->auxdev); + + if (!ldev->link_res.clock_stop_quirks) { /* * we need to prevent the parent PCI device * from entering pm_runtime suspend, so that * power rails to the SoundWire IP are not * turned off. */ - pm_runtime_get_noresume(link->dev); + pm_runtime_get_noresume(ldev->link_res.dev); } } @@ -272,8 +350,8 @@ sdw_intel_startup_controller(struct sdw_intel_ctx *ctx) * sdw_intel_probe() - SoundWire Intel probe routine * @res: resource data * - * This registers a platform device for each Master handled by the controller, - * and SoundWire Master and Slave devices will be created by the platform + * This registers an auxiliary device for each Master handled by the controller, + * and SoundWire Master and Slave devices will be created by the auxiliary * device probe. All the information necessary is stored in the context, and * the res argument pointer can be freed after this step. * This function will be called after sdw_intel_acpi_scan() by SOF probe. @@ -306,27 +384,31 @@ EXPORT_SYMBOL_NS(sdw_intel_startup, SOUNDWIRE_INTEL_INIT); void sdw_intel_exit(struct sdw_intel_ctx *ctx) { sdw_intel_cleanup(ctx); + kfree(ctx->ids); + kfree(ctx->ldev); + kfree(ctx); } EXPORT_SYMBOL_NS(sdw_intel_exit, SOUNDWIRE_INTEL_INIT); void sdw_intel_process_wakeen_event(struct sdw_intel_ctx *ctx) { - struct sdw_intel_link_res *link; + struct sdw_intel_link_dev *ldev; u32 link_mask; int i; - if (!ctx->links) + if (!ctx->ldev) return; - link = ctx->links; link_mask = ctx->link_mask; /* Startup SDW Master devices */ - for (i = 0; i < ctx->count; i++, link++) { + for (i = 0; i < ctx->count; i++) { if (!(link_mask & BIT(i))) continue; - intel_master_process_wakeen_event(link->pdev); + ldev = ctx->ldev[i]; + + intel_link_process_wakeen_event(&ldev->auxdev); } } EXPORT_SYMBOL_NS(sdw_intel_process_wakeen_event, SOUNDWIRE_INTEL_INIT); diff --git a/include/linux/soundwire/sdw_intel.h b/include/linux/soundwire/sdw_intel.h index 3a5446ac014a..1ebea7764011 100644 --- a/include/linux/soundwire/sdw_intel.h +++ b/include/linux/soundwire/sdw_intel.h @@ -58,7 +58,7 @@ struct sdw_intel_acpi_info { u32 link_mask; }; -struct sdw_intel_link_res; +struct sdw_intel_link_dev; /* Intel clock-stop/pm_runtime quirk definitions */ @@ -109,7 +109,7 @@ struct sdw_intel_slave_id { * Controller * @num_slaves: total number of devices exposed across all enabled links * @handle: ACPI parent handle - * @links: information for each link (controller-specific and kept + * @ldev: information for each link (controller-specific and kept * opaque here) * @ids: array of slave_id, representing Slaves exposed across all enabled * links @@ -123,7 +123,7 @@ struct sdw_intel_ctx { u32 link_mask; int num_slaves; acpi_handle handle; - struct sdw_intel_link_res *links; + struct sdw_intel_link_dev **ldev; struct sdw_intel_slave_id *ids; struct list_head link_list; struct mutex shim_lock; /* lock for access to shared SHIM registers */ -- 2.17.1

5 14

[PATCH] ASoC: rsnd: adg: clearly handle clock error / NULL case
by Kuninori Morimoto 14 Jun '21

14 Jun '21

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This driver is assuming that all adg->clk[i] is not NULL. Because of this prerequisites, for_each_rsnd_clk() is possible to work for all clk without checking NULL. In other words, all adg->clk[i] should not NULL. Some SoC might doesn't have clk_a/b/c/i. devm_clk_get() returns error in such case. This driver calls rsnd_adg_null_clk_get() and use null_clk instead of NULL in such cases. But devm_clk_get() might returns NULL even though such clocks exist, but it doesn't mean error (user deliberately chose to disable the feature). NULL clk itself is not error from clk point of view, but is error from this driver point of view because it is not assuming such case. But current code is using IS_ERR() which doesn't care NULL. This driver uses IS_ERR_OR_NULL() instead of IS_ERR() for clk check. And it uses ERR_CAST() to clarify null_clk error. One concern here is that it unconditionally uses null_clk if clk_a/b/c/i was error. It is correct if it doesn't exist, but is not correct if it returns error even though it exist. It needs to check "clock-names" from DT before calling devm_clk_get() to handling such case. But let's assume it is overkill so far. Link: https://lore.kernel.org/r/YMCmhfQUimHCSH/n@mwanda Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 0ebee1ed06a9..abe9d539709b 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -393,7 +393,7 @@ static struct clk *rsnd_adg_create_null_clk(struct rsnd_priv *priv, clk = clk_register_fixed_rate(dev, name, parent, 0, 0); if (IS_ERR(clk)) { dev_err(dev, "create null clk error\n"); - return NULL; + return ERR_CAST(clk); } return clk; @@ -430,9 +430,9 @@ static int rsnd_adg_get_clkin(struct rsnd_priv *priv) for (i = 0; i < CLKMAX; i++) { clk = devm_clk_get(dev, clk_name[i]); - if (IS_ERR(clk)) + if (IS_ERR_OR_NULL(clk)) clk = rsnd_adg_null_clk_get(priv); - if (IS_ERR(clk)) + if (IS_ERR_OR_NULL(clk)) goto err; adg->clk[i] = clk; -- 2.25.1

2 2

Some minor fixes for Topology2.0
by GitHub pull_request - opened 13 Jun '21

13 Jun '21

alsa-project/alsa-utils pull request #98 was opened from ranj063: Request URL : https://github.com/alsa-project/alsa-utils/pull/98 Patch URL : https://github.com/alsa-project/alsa-utils/pull/98.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH -next 0/9] ASoC: fsl: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 13 Jun '21

13 Jun '21

patch #1 ~ #8: Use devm_platform_get_and_ioremap_resource() patch #9 check return value of platform_get_resource_byname() Yang Yingliang (9): ASoC: fsl_asrc: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_aud2htx: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_easrc: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_esai: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_micfil: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_sai: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_spdif: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_ssi: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_xcvr: check return value after calling platform_get_resource_byname() sound/soc/fsl/fsl_asrc.c | 3 +-- sound/soc/fsl/fsl_aud2htx.c | 3 +-- sound/soc/fsl/fsl_easrc.c | 3 +-- sound/soc/fsl/fsl_esai.c | 3 +-- sound/soc/fsl/fsl_micfil.c | 3 +-- sound/soc/fsl/fsl_sai.c | 3 +-- sound/soc/fsl/fsl_spdif.c | 3 +-- sound/soc/fsl/fsl_ssi.c | 3 +-- sound/soc/fsl/fsl_xcvr.c | 4 ++++ 9 files changed, 12 insertions(+), 16 deletions(-) -- 2.25.1

2 10

alsa-lib 1.2.5 breaks espeak
by GitHub issues - opened 13 Jun '21

13 Jun '21

alsa-project/alsa-lib issue #155 was opened from MrBitBucket: In archlinux amd64 upgrading alsa-lib from 1.2.4 - 1.2.5 breaks the espeak application https://sourceforge.net/projects/espeak/. I see extra errors in espeak output eg ```Expression 'framesAvail' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 3629 Expression 'PaAlsaStreamComponent_GetAvailableFrames( self, &framesAvail, xrun )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 3989 Expression 'PaAlsaStreamComponent_RegisterChannels( &self->playback, &self->bufferProcessor, &playbackFrames, &xrun )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 4118 Expression 'PaAlsaStream_SetUpBuffers( stream, &framesGot, &xrun )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 4359``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/155 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 0/6] ALSA: bebob/fireworks: device entry corrections
by Takashi Sakamoto 12 Jun '21

12 Jun '21

Hi, After working for hardware database of systemd[1], I realize that current device entries of ALSA bebob/fireworks drivers includes some issues. This patchset corrects them. [1] https://github.com/systemd/systemd/pull/19124 Takashi Sakamoto (6): ALSA: bebob: fulfil device entries ALSA: fireworks: fulfil device entries ALSA: bebob: correct device entry for Mackie D.2 FireWire option card ALSA: bebob: correct device entry for Acoustic Reality eAR Master One, Eroica, Figaro, and Ciaccona ALSA: bebob: code refactoring for M-Audio models ALSA: bebob: correct device entries for Phonic Helix Board and FireFly series sound/firewire/bebob/bebob.c | 92 ++++++++++++++-------------- sound/firewire/bebob/bebob.h | 9 --- sound/firewire/fireworks/fireworks.c | 15 +++++ sound/firewire/fireworks/fireworks.h | 8 --- 4 files changed, 61 insertions(+), 63 deletions(-) -- 2.30.2

2 7

[PATCH][next] ALSA: i2c: tea6330t: Remove redundant initialization of variable err
by Colin King 12 Jun '21

12 Jun '21

From: Colin Ian King <colin.king(a)canonical.com> The variable err is being initialized with a value that is never read, it is being updated later on. The assignment is redundant and can be removed. Addresses-Coverity: ("Unused value") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/i2c/tea6330t.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/i2c/tea6330t.c b/sound/i2c/tea6330t.c index 742d0f724375..037d6293f728 100644 --- a/sound/i2c/tea6330t.c +++ b/sound/i2c/tea6330t.c @@ -284,7 +284,7 @@ int snd_tea6330t_update_mixer(struct snd_card *card, struct tea6330t *tea; const struct snd_kcontrol_new *knew; unsigned int idx; - int err = -ENOMEM; + int err; u8 default_treble, default_bass; unsigned char bytes[7]; -- 2.31.1

2 1

[PATCH] ALSA: firewire-lib: replace in_interrupt() with in_softirq()
by Takashi Sakamoto 12 Jun '21

12 Jun '21

Tasklet is used to handle isochronous context of 1394 OHCI. The explicit usage of in_softirq() may be preferable than in_interrupt(). Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream-trace.h | 2 +- sound/firewire/amdtp-stream.c | 6 +++--- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/firewire/amdtp-stream-trace.h b/sound/firewire/amdtp-stream-trace.h index aa53c13b89d3..5fd2aeccdfc2 100644 --- a/sound/firewire/amdtp-stream-trace.h +++ b/sound/firewire/amdtp-stream-trace.h @@ -49,7 +49,7 @@ TRACE_EVENT(amdtp_packet, __entry->data_blocks = data_blocks; __entry->data_block_counter = data_block_counter, __entry->packet_index = packet_index; - __entry->irq = !!in_interrupt(); + __entry->irq = !!in_softirq(); __entry->index = index; ), TP_printk( diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 1d9bc7b07df1..aad9778d1c4d 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -610,7 +610,7 @@ static void update_pcm_pointers(struct amdtp_stream *s, // buffer associated to PCM substream to process PCM frames in the buffer, instead // of receiving notification of period elapsed by poll wait. if (!pcm->runtime->no_period_wakeup) { - if (in_interrupt()) { + if (in_softirq()) { // In software IRQ context for 1394 OHCI. snd_pcm_period_elapsed(pcm); } else { @@ -1026,7 +1026,7 @@ static void generate_pkt_descs(struct amdtp_stream *s, const __be32 *ctx_header, static inline void cancel_stream(struct amdtp_stream *s) { s->packet_index = -1; - if (in_interrupt()) + if (in_softirq()) amdtp_stream_pcm_abort(s); WRITE_ONCE(s->pcm_buffer_pointer, SNDRV_PCM_POS_XRUN); } @@ -1738,7 +1738,7 @@ unsigned long amdtp_domain_stream_pcm_pointer(struct amdtp_domain *d, if (irq_target && amdtp_stream_running(irq_target)) { // In software IRQ context, the call causes dead-lock to disable the tasklet // synchronously. - if (!in_interrupt()) + if (!in_softirq()) fw_iso_context_flush_completions(irq_target->context); } -- 2.30.2

2 1

[PATCH -next] ALSA: n64: check return value after calling platform_get_resource()
by Yang Yingliang 12 Jun '21

12 Jun '21

It will cause null-ptr-deref if platform_get_resource() returns NULL, we need check the return value. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/mips/snd-n64.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/mips/snd-n64.c b/sound/mips/snd-n64.c index e35e93157755..463a6fe589eb 100644 --- a/sound/mips/snd-n64.c +++ b/sound/mips/snd-n64.c @@ -338,6 +338,10 @@ static int __init n64audio_probe(struct platform_device *pdev) strcpy(card->longname, "N64 Audio"); res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); + if (!res) { + err = -EINVAL; + goto fail_dma_alloc; + } if (devm_request_irq(&pdev->dev, res->start, n64audio_isr, IRQF_SHARED, "N64 Audio", priv)) { err = -EBUSY; -- 2.25.1

2 1

[PATCH -next] ALSA: hda/tegra: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 12 Jun '21

12 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/pci/hda/hda_tegra.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 6f2b743b9d75..5aeef6123781 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -235,11 +235,9 @@ static int hda_tegra_init_chip(struct azx *chip, struct platform_device *pdev) { struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); struct hdac_bus *bus = azx_bus(chip); - struct device *dev = hda->dev; struct resource *res; - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - hda->regs = devm_ioremap_resource(dev, res); + hda->regs = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(hda->regs)) return PTR_ERR(hda->regs); -- 2.25.1

2 1

[PATCH -next] ALSA: trident: Fix build error
by YueHaibing 12 Jun '21

12 Jun '21

sound/pci/trident/trident_memory.c: In function ‘set_tlb_bus’: sound/pci/trident/trident_memory.c:85:35: error: ‘pagetr’ undeclared (first use in this function); did you mean ‘page’? for (i = 0; i < UNIT_PAGES; i++, pagetr++) { ^~~~~~ page Fixes: 74fb98311c4e ("ALSA: trident: Drop shadow TLB pointer table") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/pci/trident/trident_memory.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/trident/trident_memory.c b/sound/pci/trident/trident_memory.c index f831ec02702f..4ad3855101c9 100644 --- a/sound/pci/trident/trident_memory.c +++ b/sound/pci/trident/trident_memory.c @@ -82,7 +82,7 @@ static inline void set_tlb_bus(struct snd_trident *trident, int page, { int i; page *= UNIT_PAGES; - for (i = 0; i < UNIT_PAGES; i++, pagetr++) { + for (i = 0; i < UNIT_PAGES; i++, page++) { __set_tlb_bus(trident, page, addr); addr += SNDRV_TRIDENT_PAGE_SIZE; } -- 2.17.1

2 1

Question about upmix plugin
by GitHub issues - opened 11 Jun '21

11 Jun '21

alsa-project/alsa-plugins issue #25 was opened from krumpfwylg: Hi, I've noticed the [latest commits](https://github.com/alsa-project/alsa-plugins/pull/2) done to the upmix plugin, but there's something I don't understand. Line 24 of pcm_upmix.c is _#define UPMIX_PCM_FORMAT SND_PCM_FORMAT_S16_ Does that mean the upmix plugin is still only capable of using S16_LE format ? Or does the UPMIX_PCM_FORMAT value can get changed to S32_LE by tweaking it in asoundrc file ? Issue URL : https://github.com/alsa-project/alsa-plugins/issues/25 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

Config parsing of pcm_rate broken since 1.1.7 (at least for a52 pcm plugin)
by GitHub issues - opened 11 Jun '21

11 Jun '21

alsa-project/alsa-lib issue #154 was opened from quequotion: As noted [in the mailing list](https://mailman.alsa-project.org/pipermail/alsa-devel/2019-December/1… back in 2019, config parsing (for alsa-plugins a52 pcm at least) has been broken since alsa-lib 1.1.7. As far as I know this was never bisected to a single commit in either alsa-lib or alsa-plugins. Looking at the mailing list discussion, it occurs to me that it may have given the impression that the config parser was broken when using [a patch that ports alsa-plugins to libswresample](https://git.harting.dev/anonfunc/alsa-plugins/commit/9cdbbb9874757b6f8fda7fb4ac2e3fc59da65946). Config parsing is broken *regardless* of that patch. @IdleGandalf worked around this by applying a patch to alsa-lib, in [src/pcm/pcm_rate.c](https://git.harting.dev/anonfunc/alsa-lib/commit/646bc80a63b0220300678a38e0c26a9a13af6fc0), but never got a response if this was an appropriate method to resolve the issue. Should this be fixed in alsa-lib or in alsa-plugins? Note: above links are to mirrors of the original patches, also in a repository owned by @IdleGandalf as far as I can tell. The links in the mailing list discussion are now 404. Issue URL : https://github.com/alsa-project/alsa-lib/issues/154 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

SNDRV_PCM_IOCTL_READI_FRAMES returns EIO
by Muni Sekhar 11 Jun '21

11 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". I noticed recording(arecord) audio data on a USB sound card fails at SNDRV_PCM_IOCTL_READI_FRAMES is giving EIO. Can someone please clarify when the kernel returns EIO for Ioctl SNDRV_PCM_IOCTL_READI_FRAMES? and is there a workaround for this? Output of the alsamixer is attached. -- Thanks, Sekhar

2 5

[PATCH 0/3] ALSA: bebob: remove workarounds for suppressed quirks
by Takashi Sakamoto 11 Jun '21

11 Jun '21

Hi, As a result to apply the commit 1bd1b3be8655 ("ALSA: bebob: perform sequence replay for media clock recovery"), some quirks seem to be suppressed. This patchset removes workarounds for the suppressed quirks and includes code cleanup. Takashi Sakamoto (3): ALSA: bebob: dismiss sleep after breaking connections ALSA: bebob: delete workaround for protocol version 3 ALSA: bebob: code refactoring for model-dependent quirks sound/firewire/bebob/bebob.c | 41 ++++++++++++++++++++--------- sound/firewire/bebob/bebob.h | 10 ++++--- sound/firewire/bebob/bebob_stream.c | 32 ++++++---------------- 3 files changed, 42 insertions(+), 41 deletions(-) -- 2.27.0

2 4

Add support for automatic attributes in Topology2.0
by GitHub pull_request - opened 11 Jun '21

11 Jun '21

alsa-project/alsa-utils pull request #97 was opened from ranj063: Add support for automatic attributes in Topology2.0 and also add support for processing objects that are simply used to update the parent object's private data. Request URL : https://github.com/alsa-project/alsa-utils/pull/97 Patch URL : https://github.com/alsa-project/alsa-utils/pull/97.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

ALSA lib pcm_dmix.c:1063:(snd_pcm_dmix_open) unable to create IPC semaphore
by GitHub issues - edited 11 Jun '21

11 Jun '21

alsa-project/alsa-lib issue #153 was edited from Cean1024: alsa info: alsa-config-1.0 alsa-lib-1.1.5 alsa-plugins-1.1.5 alsa-utils-1.1.5 asound.conf: ``` pcm.dmixer { type dmix ipc_key 1024 slave { pcm "hw:0,0" period_time 0 period_size 1024 buffer_size 4096 rate 48000 format "S32_LE" channels 2 } bindings { 0 0 1 1 } } ``` test cmd: ``` arecord -f dat | aplay -f dat -D plug:dmixer ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/153 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ALSA lib pcm_dmix.c:1063:(snd_pcm_dmix_open) unable to create IPC semaphore
by GitHub issues - edited 11 Jun '21

11 Jun '21

alsa-project/alsa-lib issue #153 was edited from Cean1024: alsa info: alsa-config-1.0 alsa-lib-1.1.5 alsa-plugins-1.1.5 alsa-utils-1.1.5 asound.conf: pcm.dmixer { type dmix ipc_key 1024 slave { pcm "hw:0,0" period_time 0 period_size 1024 buffer_size 4096 rate 48000 format "S32_LE" channels 2 } bindings { 0 0 1 1 } } test cmd: arecord -f dat | aplay -f dat -D plug:dmixer Issue URL : https://github.com/alsa-project/alsa-lib/issues/153 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2 0/3] ALSA: pcm:firewire: allow to operate for period elapse event in process context
by Takashi Sakamoto 11 Jun '21

11 Jun '21

Hi, This patchset is revised version of my previous one: * https://lore.kernel.org/alsa-devel/20210609012244.24296-1-o-takashi@sakamoc… All of drivers in ALSA firewire stack have two chances to process isochronous packets of any isochronous context; in software IRQ context for 1394 OHCI, and in process context of ALSA PCM application. In the process context, callbacks of .pointer and .ack are utilized. The callbacks are done by ALSA PCM core under acquiring lock of PCM substream, In design of ALSA PCM core, call of snd_pcm_period_elapsed() is used for drivers to awaken user processes from waiting for available frames. The function voluntarily acquires lock of PCM substream, therefore it is not called in the process context since it causes dead lock. As a workaround to avoid the dead lock, all of drivers in ALSA firewire stack use workqueue to delegate the call. This patchset is my attempt for the issue. A variant of 'snd_pcm_period_elapsed()' without lock acquisition is going to be added, named 'snd_pcm_period_elapsed_under_stream_lock()'. The call is available in callbacks of .pointer and .ack of snd_pcm_ops structure. Changes from v1: * fix context section of kernel API documentation Takashi Sakamoto (3): ALSA: pcm: add snd_pcm_period_elapsed() variant without acquiring lock of PCM substream ALSA: firewire-lib: operate for period elapse event in process context ALSA: firewire-lib: obsolete workqueue for period update include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 71 +++++++++++++++++++++++++++-------- sound/firewire/amdtp-stream.c | 46 +++++++---------------- sound/firewire/amdtp-stream.h | 1 - 4 files changed, 71 insertions(+), 48 deletions(-) -- 2.27.0

2 16

ALSA lib pcm_dmix.c:1063:(snd_pcm_dmix_open) unable to create IPC semaphore
by GitHub issues - opened 11 Jun '21

11 Jun '21

alsa-project/alsa-lib issue #153 was opened from Cean1024: Issue URL : https://github.com/alsa-project/alsa-lib/issues/153 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Re: [PATCH] ASoC:fsl_spdif:Remove superfluous error message around platform_get_irq()
by Shengjiu Wang 11 Jun '21

11 Jun '21

On Thu, Jun 10, 2021 at 12:02 PM Zhongjun Tan <hbut_tan(a)163.com> wrote: > > From: Tan Zhongjun <tanzhongjun(a)yulong.com> > > The platform_get_irq() prints error message telling that interrupt is > missing, hence there is no need to duplicated that message. > > Signed-off-by: Tan Zhongjun <tanzhongjun(a)yulong.com> Acked-by: Shengjiu Wang <shengjiu.wang(a)gmail.com> > --- > sound/soc/fsl/fsl_spdif.c | 4 +--- > 1 file changed, 1 insertion(+), 3 deletions(-) > > diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c > index 2a76714eb8e6..29cefd459241 100644 > --- a/sound/soc/fsl/fsl_spdif.c > +++ b/sound/soc/fsl/fsl_spdif.c > @@ -1368,10 +1368,8 @@ static int fsl_spdif_probe(struct platform_device *pdev) > > for (i = 0; i < spdif_priv->soc->interrupts; i++) { > irq = platform_get_irq(pdev, i); > - if (irq < 0) { > - dev_err(&pdev->dev, "no irq for node %s\n", pdev->name); > + if (irq < 0) > return irq; > - } > > ret = devm_request_irq(&pdev->dev, irq, spdif_isr, 0, > dev_name(&pdev->dev), spdif_priv); > -- > 2.17.1 >

1 0

Re: [PATCH] ASoC:fsl_easrc:Remove superfluous error message around platform_get_irq()
by Shengjiu Wang 11 Jun '21

11 Jun '21

On Thu, Jun 10, 2021 at 8:52 PM Zhongjun Tan <hbut_tan(a)163.com> wrote: > > From: Tan Zhongjun <tanzhongjun(a)yulong.com> > > 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. > > Signed-off-by: Tan Zhongjun <tanzhongjun(a)yulong.com> Acked-by: Shengjiu Wang <shengjiu.wang(a)gmail.com> > --- > sound/soc/fsl/fsl_easrc.c | 4 +--- > 1 file changed, 1 insertion(+), 3 deletions(-) > > diff --git a/sound/soc/fsl/fsl_easrc.c b/sound/soc/fsl/fsl_easrc.c > index b1765c7d3bcd..25747433916e 100644 > --- a/sound/soc/fsl/fsl_easrc.c > +++ b/sound/soc/fsl/fsl_easrc.c > @@ -1901,10 +1901,8 @@ static int fsl_easrc_probe(struct platform_device *pdev) > } > > irq = platform_get_irq(pdev, 0); > - if (irq < 0) { > - dev_err(dev, "no irq for node %pOF\n", np); > + if (irq < 0) > return irq; > - } > > ret = devm_request_irq(&pdev->dev, irq, fsl_easrc_isr, 0, > dev_name(dev), easrc); > -- > 2.17.1 > >

1 0

[PATCH] ALSA: seq: Fix race of snd_seq_timer_open()
by Takashi Iwai 10 Jun '21

10 Jun '21

The timer instance per queue is exclusive, and snd_seq_timer_open() should have managed the concurrent accesses. It looks as if it's checking the already existing timer instance at the beginning, but it's not right, because there is no protection, hence any later concurrent call of snd_seq_timer_open() may override the timer instance easily. This may result in UAF, as the leftover timer instance can keep running while the queue itself gets closed, as spotted by syzkaller recently. For avoiding the race, add a proper check at the assignment of tmr->timeri again, and return -EBUSY if it's been already registered. Reported-by: syzbot+ddc1260a83ed1cbf6fb5(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/000000000000dce34f05c42f110c@google.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_timer.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c index 1645e4142e30..9863be6fd43e 100644 --- a/sound/core/seq/seq_timer.c +++ b/sound/core/seq/seq_timer.c @@ -297,8 +297,16 @@ int snd_seq_timer_open(struct snd_seq_queue *q) return err; } spin_lock_irq(&tmr->lock); - tmr->timeri = t; + if (tmr->timeri) + err = -EBUSY; + else + tmr->timeri = t; spin_unlock_irq(&tmr->lock); + if (err < 0) { + snd_timer_close(t); + snd_timer_instance_free(t); + return err; + } return 0; } -- 2.26.2

1 0

pulseaudio can't load configuration for snd_hda_intel
by GitHub issues - edited 10 Jun '21

10 Jun '21

alsa-project/alsa-ucm-conf issue #99 was edited from hzhuang1: In v1.2.2, the content of HDA-Intel.conf is in below. v1.2.2 is widely used, such as in ubuntu 20.04. ``` Syntax 3 Define.Use "" # a non-empty string to use UCM configuration for HDA devices Define.AcpCardId "$${CardIdByName:acp}" ``` With this content, pulseaudio can't load configuration from UCM. Then I tried to edit the content. Change it to ``` Define.Use "2" ``` or ``` Define.Use "3" ``` Pulseaudio worked with "options snd-hda-intel probe_mask=1" on DELL laptop. My question is what's the real meaning of **Define.Use** at here. Why the default value of **Define.Use** is empty string? Even in latest code, the **Define.Use** is also an empty string. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/99 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

pulseaudio can't load configuration for snd_hda_intel
by GitHub issues - opened 10 Jun '21

10 Jun '21

alsa-project/alsa-ucm-conf issue #99 was opened from hzhuang1: In v1.2.2, the content of HDA-Intel.conf is in below. v1.2.2 is widely used, such as in ubuntu 20.04. ``` Syntax 3 Define.Use "" # a non-empty string to use UCM configuration for HDA devices Define.AcpCardId "$${CardIdByName:acp}" ``` With this content, pulseaudio can't load configuration from UCM. Then I tried to edit the content. Change it to ``` Define.Use "2" ``` or ``` Define.Use "3" ``` Pulseaudio worked with "options snd-hda-intel probe_mask=1" on DELL laptop. My question is what's the real meaning of **Define.Use** at here. Why the default value of **Define.Use** is empty string? Even in v1.2.4, the **Define.Use** is also an empty string. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/99 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Logitech C922x microphone intermittently working
by GitHub issues - opened 10 Jun '21

10 Jun '21

alsa-project/alsa-lib issue #152 was opened from sambo57u: I have a Logitech C922X (detected as C922) USB webcam. Running latest updates on Fedora 34. The microphone of this device only works about 1 out of 5 reboots. When not working if I install another microphone, activate that one, the unplug and plug C922 back and switch to it, it starts to work. Some kind of a timing issue that is very annoying. Don't have this problem with Logitech C270. The camera always works fine. Issue URL : https://github.com/alsa-project/alsa-lib/issues/152 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Undeclarated constant
by GitHub issues - opened 10 Jun '21

10 Jun '21

alsa-project/alsa-plugins issue #24 was opened from unknown2045: Hi, I am trying to install the alsa-plugins, when I do the make command I get the following errors: ![Captura de pantalla de 2021-06-10 13-52-37](https://user-images.githubusercontent.com/85682900/121520376-229e… How can I solve it? Thanks in advance! Issue URL : https://github.com/alsa-project/alsa-plugins/issues/24 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[sound:for-next 164/168] include/sound/memalloc.h:63:17: error: use of undeclared identifier 'PAGE_SIZE'
by kernel test robot 10 Jun '21

10 Jun '21

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 83fbcaed24d797528de00c024674fd58e8f1634f commit: 37af81c5998f4b0f23fb452cffa4b8a1c00ce95b [164/168] ALSA: core: Abstract memory alloc helpers config: riscv-randconfig-r002-20210610 (attached as .config) compiler: clang version 13.0.0 (https://github.com/llvm/llvm-project d2012d965d60c3258b3a69d024491698f8aec386) reproduce (this is a W=1 build): wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross # install riscv cross compiling tool for clang build # apt-get install binutils-riscv64-linux-gnu # https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?id=… git remote add sound https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git git fetch --no-tags sound for-next git checkout 37af81c5998f4b0f23fb452cffa4b8a1c00ce95b # save the attached .config to linux build tree COMPILER_INSTALL_PATH=$HOME/0day COMPILER=clang make.cross ARCH=riscv If you fix the issue, kindly add following tag as appropriate Reported-by: kernel test robot <lkp(a)intel.com> All errors (new ones prefixed by >>): In file included from sound/core/pcm_iec958.c:8: In file included from include/sound/pcm.h:12: >> include/sound/memalloc.h:63:17: error: use of undeclared identifier 'PAGE_SIZE' return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; ^ >> include/sound/memalloc.h:63:35: error: use of undeclared identifier 'PAGE_SHIFT' return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; ^ 2 errors generated. vim +/PAGE_SIZE +63 include/sound/memalloc.h ^1da177e4c3f41 Linus Torvalds 2005-04-16 57 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 58 /* 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 59 * return the pages matching with the given byte size 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 60 */ 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 61 static inline unsigned int snd_sgbuf_aligned_pages(size_t size) 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 62 { 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 @63 return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 64 } 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 65 :::::: The code at line 63 was first introduced by commit :::::: 4cae99d9b5305ab8cccc839fccceb81ec9e5abda ALSA: memalloc: declare snd_sgbuf_aligned_pages() unconditionally :::::: TO: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY CI Kernel Test Service, Intel Corporation https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org

1 0

[PATCH] ALSA: core: Fix build error due to missing PAGE_SIZE
by Takashi Iwai 10 Jun '21

10 Jun '21

The recent refactoring of memalloc stuff removed the inclusion of asm/page.h for simplicity, but it turned out this caused a compile error due the lack of PAGE_SIZE definition on some architectures. Do a partial revert for recovering from that. Fixes: 37af81c5998f ("ALSA: core: Abstract memory alloc helpers") Reported-by: kernel test robot <lkp(a)intel.com> Link: https://lore.kernel.org/r/202106101858.PfXMMuAa-lkp@intel.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/memalloc.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index 1918c60f1f35..44d87775b352 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -9,8 +9,9 @@ #ifndef __SOUND_MEMALLOC_H #define __SOUND_MEMALLOC_H +#include <asm/page.h> + struct device; -struct page; struct vm_area_struct; /* -- 2.26.2

1 0

[sound:for-next 164/167] include/sound/memalloc.h:63:17: error: 'PAGE_SIZE' undeclared
by kernel test robot 10 Jun '21

10 Jun '21

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 7fdabab8f56239466275adea5d2f0a6fff27527b commit: 37af81c5998f4b0f23fb452cffa4b8a1c00ce95b [164/167] ALSA: core: Abstract memory alloc helpers config: h8300-randconfig-r006-20210610 (attached as .config) compiler: h8300-linux-gcc (GCC) 9.3.0 reproduce (this is a W=1 build): wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross # https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?id=… git remote add sound https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git git fetch --no-tags sound for-next git checkout 37af81c5998f4b0f23fb452cffa4b8a1c00ce95b # save the attached .config to linux build tree COMPILER_INSTALL_PATH=$HOME/0day COMPILER=gcc-9.3.0 make.cross ARCH=h8300 If you fix the issue, kindly add following tag as appropriate Reported-by: kernel test robot <lkp(a)intel.com> All errors (new ones prefixed by >>): In file included from include/sound/pcm.h:12, from sound/core/pcm_iec958.c:8: include/sound/memalloc.h: In function 'snd_sgbuf_aligned_pages': >> include/sound/memalloc.h:63:17: error: 'PAGE_SIZE' undeclared (first use in this function) 63 | return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; | ^~~~~~~~~ include/sound/memalloc.h:63:17: note: each undeclared identifier is reported only once for each function it appears in >> include/sound/memalloc.h:63:35: error: 'PAGE_SHIFT' undeclared (first use in this function) 63 | return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; | ^~~~~~~~~~ vim +/PAGE_SIZE +63 include/sound/memalloc.h ^1da177e4c3f41 Linus Torvalds 2005-04-16 57 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 58 /* 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 59 * return the pages matching with the given byte size 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 60 */ 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 61 static inline unsigned int snd_sgbuf_aligned_pages(size_t size) 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 62 { 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 @63 return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 64 } 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 65 :::::: The code at line 63 was first introduced by commit :::::: 4cae99d9b5305ab8cccc839fccceb81ec9e5abda ALSA: memalloc: declare snd_sgbuf_aligned_pages() unconditionally :::::: TO: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY CI Kernel Test Service, Intel Corporation https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org

1 0

[PATCH 0/1 v2] usb-audio: Add support for Denon DN-X1600
by Damien Zammit 10 Jun '21

10 Jun '21

Hi Takashi, You are correct, the patch still works without the hunk referring to the PITCH control. I have removed the hunk and was tested with this patch and it still functioned as expected. Regards, Damien Z

2 2

[PATCH] Docs: hda: Fix spelling mistakes
by huangjianghui 10 Jun '21

10 Jun '21

correct Reatek to Realtek Signed-off-by: huangjianghui <huangjianghui(a)uniontech.com> --- Documentation/sound/hd-audio/controls.rst | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/sound/hd-audio/controls.rst b/Documentation/sound/hd-audio/controls.rst index f2ebc4f79b44..dbe6483f4ff4 100644 --- a/Documentation/sound/hd-audio/controls.rst +++ b/Documentation/sound/hd-audio/controls.rst @@ -102,7 +102,7 @@ Conexant codecs --------------- Auto-Mute Mode - See Reatek codecs. + See Realtek codecs. Analog codecs -- 2.20.1

2 1

[PATCH] sound/arm: Remove unnecessary variables
by zuoqilin1＠163.com 10 Jun '21

10 Jun '21

From: zuoqilin <zuoqilin(a)yulong.com> There is no need to define the variable "ret" to receive. Signed-off-by: zuoqilin <zuoqilin(a)yulong.com> --- sound/arm/pxa2xx-ac97.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 6322e639..a67e668 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -47,9 +47,7 @@ static unsigned short pxa2xx_ac97_legacy_read(struct snd_ac97 *ac97, static void pxa2xx_ac97_legacy_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { - int __always_unused ret; - - ret = pxa2xx_ac97_write(ac97->num, reg, val); + pxa2xx_ac97_write(ac97->num, reg, val); } static const struct snd_ac97_bus_ops pxa2xx_ac97_ops = { -- 1.9.1

2 1

[PATCH v3 0/3] ALSA: pcm: add snd_pcm_period_elapsed() variant without acquiring lock of PCM substream
by Takashi Sakamoto 10 Jun '21

10 Jun '21

Hi, This patchset is revised version of my previous one: * https://lore.kernel.org/alsa-devel/20210609143145.146680-1-o-takashi@sakamo… All of drivers in ALSA firewire stack have two chances to process isochronous packets of any isochronous context; in software IRQ context for 1394 OHCI, and in process context of ALSA PCM application. In the process context, callbacks of .pointer and .ack are utilized. The callbacks are done by ALSA PCM core under acquiring lock of PCM substream, In design of ALSA PCM core, call of snd_pcm_period_elapsed() is used for drivers to awaken user processes from waiting for available frames. The function voluntarily acquires lock of PCM substream, therefore it is not called in the process context since it causes dead lock. As a workaround to avoid the dead lock, all of drivers in ALSA firewire stack use workqueue to delegate the call. This patchset is my attempt for the issue. A variant of 'snd_pcm_period_elapsed()' without lock acquisition is going to be added, named 'snd_pcm_period_elapsed_under_stream_lock()'. The call is available in callbacks of .pointer and .ack of snd_pcm_ops structure. Changes from v2: * delete context section from kernel API documentation Takashi Sakamoto (3): ALSA: pcm: add snd_pcm_period_elapsed() variant without acquiring lock of PCM substream ALSA: firewire-lib: operate for period elapse event in process context ALSA: firewire-lib: obsolete workqueue for period update include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 64 +++++++++++++++++++++++++++-------- sound/firewire/amdtp-stream.c | 46 ++++++++----------------- sound/firewire/amdtp-stream.h | 1 - 4 files changed, 64 insertions(+), 48 deletions(-) -- 2.27.0

2 4

[Bug] Duplication in Array Element not Handled in Parsing conf
by GitHub issues - opened 10 Jun '21

10 Jun '21

alsa-project/alsa-lib issue #151 was opened from aiChaoSONG: When parsing array element in conf files, an auto-increase number is used as config id(`snd_config_t->id`), and array element as the config value(`snd_config_t->u.string/integer/integer64/real`), duplication is not handle in `parse_array_defs/parse_array_def/parse_value` functions. Suppose we have a conf: ``` array [ "a" "b" "a" ] ``` the parsed result will be: ``` parent | ---- snd_config_t {id: "0", u.string: "a"} | ---- snd_config_t {id: "1", u.string: "b"} | ---- snd_config_t {id: "2", u.string: "a"} ``` The duplication in "a" may be desired. But please consider this condition: suppose we have four conf files A, B, C and D, where A is the conf above, let B include A, C include A, D include B and C, so D include A twice indirectly. File A will be processed twice when parsing D, because duplication is not handled, here comes the problem, the parsed result of A in D is: ``` parent | ---- snd_config_t {id: "0", u.string: "a"} | ---- snd_config_t {id: "1", u.string: "b"} | ---- snd_config_t {id: "2", u.string: "a"} | ---- snd_config_t {id: "3", u.string: "a"} | ---- snd_config_t {id: "4", u.string: "b"} | ---- snd_config_t {id: "5", u.string: "a"} ``` This is obviously wrong. I think handle duplication in parse_value is not the solution, because element duplication in an array is wanted. The key problem here is that we parse the same conf more than once. Issue URL : https://github.com/alsa-project/alsa-lib/issues/151 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[bug report] ASoC: rsnd: add null CLOCKIN support
by Dan Carpenter 10 Jun '21

10 Jun '21

Hello Kuninori Morimoto, The patch d6956a7dde6f: "ASoC: rsnd: add null CLOCKIN support" from May 24, 2021, leads to the following static checker warning: sound/soc/sh/rcar/adg.c:435 rsnd_adg_get_clkin() warn: 'clk' isn't an ERR_PTR sound/soc/sh/rcar/adg.c 423 static int rsnd_adg_get_clkin(struct rsnd_priv *priv) 424 { 425 struct rsnd_adg *adg = priv->adg; 426 struct device *dev = rsnd_priv_to_dev(priv); 427 struct clk *clk; 428 int i; 429 430 for (i = 0; i < CLKMAX; i++) { 431 clk = devm_clk_get(dev, clk_name[i]); If devm_clk_get() returns NULL that's not an error. 432 433 if (IS_ERR(clk)) 434 clk = rsnd_adg_null_clk_get(priv); But I think if rsnd_adg_null_clk_get() returns an error then it is an error. 435 if (IS_ERR(clk)) This is impossible. 436 goto err; So I think it should be: if (IS_ERR(clk)) { clk = rsnd_adg_null_clk_get(priv); if (!clk) goto err; } 437 438 adg->clk[i] = clk; 439 } 440 441 return 0; 442 443 err: 444 dev_err(dev, "adg clock IN get failed\n"); 445 446 rsnd_adg_null_clk_clean(priv); 447 448 return -EIO; 449 } regards, dan carpenter

2 3

[PATCH v3 0/3] ALSA: pcm: add snd_pcm_period_elapsed() variant without acquiring lock of PCM substream
by Takashi Sakamoto 10 Jun '21

10 Jun '21

Hi, This patchset is revised version of my previous one: * https://lore.kernel.org/alsa-devel/20210609143145.146680-1-o-takashi@sakamo… All of drivers in ALSA firewire stack have two chances to process isochronous packets of any isochronous context; in software IRQ context for 1394 OHCI, and in process context of ALSA PCM application. In the process context, callbacks of .pointer and .ack are utilized. The callbacks are done by ALSA PCM core under acquiring lock of PCM substream, In design of ALSA PCM core, call of snd_pcm_period_elapsed() is used for drivers to awaken user processes from waiting for available frames. The function voluntarily acquires lock of PCM substream, therefore it is not called in the process context since it causes dead lock. As a workaround to avoid the dead lock, all of drivers in ALSA firewire stack use workqueue to delegate the call. This patchset is my attempt for the issue. A variant of 'snd_pcm_period_elapsed()' without lock acquisition is going to be added, named 'snd_pcm_period_elapsed_under_stream_lock()'. The call is available in callbacks of .pointer and .ack of snd_pcm_ops structure. Changes from v2: * delete context section from kernel API documentation Takashi Sakamoto (3): ALSA: pcm: add snd_pcm_period_elapsed() variant without acquiring lock of PCM substream ALSA: firewire-lib: operate for period elapse event in process context ALSA: firewire-lib: obsolete workqueue for period update include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 64 +++++++++++++++++++++++++++-------- sound/firewire/amdtp-stream.c | 46 ++++++++----------------- sound/firewire/amdtp-stream.h | 1 - 4 files changed, 64 insertions(+), 48 deletions(-) -- 2.27.0

1 1

aplay change to use 16-bit samples by default is not backwards compatible and is not documented
by GitHub issues - opened 10 Jun '21

10 Jun '21

alsa-project/alsa-utils issue #96 was opened from nwoeanhinnogaehr: The change introduced in https://github.com/alsa-project/alsa-utils/commit/0c5948e98a6a8535c89b7bcab… breaks a lot of scripts and other programs I have which invoke `aplay` and expect the default sample format to be U8. While I think it's somewhat reasonable to make this change given that U8 is a rather low quality sample format, it's worth acknowledging that aplay has had this behavior for a very long time and many people probably don't expect it to change. Given that the commit log seems to only talk about the case of `arecord`, I wonder if we could reach a compromise and keep the format as U8 for `aplay` and switch it for `arecord`, since users are unlikely to complain about poor sound quality when *playing raw audio* (surely they would take the time to get the sample format set up correctly before attempting to play raw audio). Also, the man page has not been updated to reflect this change and still says that U8 is the default. Issue URL : https://github.com/alsa-project/alsa-utils/issues/96 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v2 0/8] ASoC: tidyup snd_soc_of_parse_daifmt()
by Kuninori Morimoto 10 Jun '21

10 Jun '21

Hi Mark These are v2 of parsing for daifmt. I want to add new audio-graph-card2 sound card driver, and this is last part of necessary soc-core cleanup for it. Current some drivers are using DT, and then, snd_soc_of_parse_daifmt() parses daifmt, but bitclock/frame provider parsing part is one of headache, because we are assuming below both cases. A) node { bitclock-master; frame-master; ... }; B) link { bitclock-master = <&xxx>; frame-master = <&xxx>; ... }; The original was style A), and style B) was added later. snd_soc_of_parse_daifmt() parses A) style as original style, and user need to update to B) style for clock_provider part if needed. To handle it more flexibile, this patch-set adds new functions which separates snd_soc_of_parse_daifmt() helper function. snd_soc_daifmt_parse_format() : format part snd_soc_daifmt_parse_clock_provider_as_flag() : clock part for style A) snd_soc_daifmt_parse_clock_provider_as_phandl() : clock part for style B) snd_soc_daifmt_parse_clock_provider_as_bitmap() : clock part use with _from_bitmap v1 -> v2 - tidyup parse_clock_provider functions to _as_flag/phandle/bitmap() - don't exchange code style on each drivers. Link: https://lore.kernel.org/r/875yypdxlm.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (8): ASoC: soc-core: add snd_soc_daifmt_clock_provider_from_bitmap() ASoC: soc-core: add snd_soc_daifmt_clock_provider_fliped() ASoC: soc-core: add snd_soc_daifmt_parse_format/clock_provider() ASoC: atmel: switch to use snd_soc_daifmt_parse_format/clock_provider() ASoC: fsl: switch to use snd_soc_daifmt_parse_format/clock_provider() ASoC: meson: switch to use snd_soc_daifmt_parse_format/clock_provider() ASoC: simple-card-utils: switch to use snd_soc_daifmt_parse_format/clock_provider() ASoC: soc-core: remove snd_soc_of_parse_daifmt() include/sound/soc.h | 21 ++++-- sound/soc/atmel/mikroe-proto.c | 16 +++-- sound/soc/fsl/fsl-asoc-card.c | 14 ++-- sound/soc/generic/simple-card-utils.c | 19 ++--- sound/soc/meson/meson-card-utils.c | 13 ++-- sound/soc/soc-core.c | 99 +++++++++++++++++---------- 6 files changed, 110 insertions(+), 72 deletions(-) -- 2.25.1

2 11

[PATCH 0/5] ALSA: More abstracition of memory alloc helpers
by Takashi Iwai 09 Jun '21

09 Jun '21

Hi, this is a patch set for the code cleanup and refactoring around the ALSA core memory allocation helpers. It begins with a cleanup of the non-used code, then followed by the refacotoring with the new ops, finally moved the memory-specific mmap handling code from PCM to the core memalloc. Takashi === Takashi Iwai (5): ALSA: trident: Drop shadow TLB pointer table ALSA: core: Drop snd_sgbuf_get_ptr() ALSA: core: Abstract memory alloc helpers ALSA: core: Move mmap handler into memalloc ops ALSA: core: Add continuous and vmalloc mmap ops include/sound/memalloc.h | 83 +----- include/sound/pcm.h | 19 -- sound/core/memalloc.c | 444 +++++++++++++++++++---------- sound/core/memalloc_local.h | 19 ++ sound/core/pcm_local.h | 5 - sound/core/pcm_memory.c | 21 -- sound/core/pcm_native.c | 39 +-- sound/core/sgbuf.c | 90 ++++-- sound/pci/trident/trident.h | 1 - sound/pci/trident/trident_main.c | 11 +- sound/pci/trident/trident_memory.c | 53 ++-- 11 files changed, 405 insertions(+), 380 deletions(-) create mode 100644 sound/core/memalloc_local.h -- 2.26.2

1 5

Current version not working with Focusrite Scarlett 18i20
by GitHub issues - opened 09 Jun '21

09 Jun '21

alsa-project/alsa-lib issue #150 was opened from TheBitFighter: Hello, I hope I am in the right repository for this. My arch system updated alsa-card-profiles and alsa-utils today which caused my interface, a Focusrite Scarlett 18i20 1st gen to stop working. The card was still showing up in pulse and showed output, however no sound was actually played. Input was still working fine. It sounded like the output stage of the interface had been disabled, and the problem could only be fixed by booting Windows and setting new routing settings in Scarlett MixControl. The interface had exhibited some resetting and popping behavior for a few versions now (when stopping playback, changing application and playing something else there), but continued to work (given a few instances where sample rate played mismatched sample rate sent which caused weird audio slowdowns). It only stopped working today. What I did initially was reinstall the older versions of these two packages, which did not fix the problem. I now downgraded alsa-libs (1.2.5-2 -> 1.2.4-2), alsa-card-profiles (1:0.3.29-2 -> 1:0.3.28-1), lib32-alsa-lib (1.2.5-2 -> 1.2.4-1), alsa-topology-conf (1.2.5-1 -> 1.2.4-1), and alsa-ucm-conf (1.2.5-1 -> 1.2.4-1) which fixed the problem. (alsa-utils is at 1.2.5-2) `cat /etc/asound/version` now reports `Advanced Linux Sound Architecture Driver Version k5.12.9-arch1-1.` This also fixed the popping/resetting issues I had before. Thanks for the help! If you need any further information I am happy to help. Issue URL : https://github.com/alsa-project/alsa-lib/issues/150 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

opensuse tumbleweed no sound after alsa lib update (tumbleweed 2010605)
by GitHub issues - opened 09 Jun '21

09 Jun '21

alsa-project/alsa-lib issue #149 was opened from pawanyadav: sound is not working alsa (never installed pipewire and pulseaudio) asound.conf defaults.pcm.card 1 defaults.pcm.device 1 aplay -l **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Generic [HD-Audio Generic], device 0: ALC887-VD Analog [ALC887-VD Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Generic [HD-Audio Generic], device 1: ALC887-VD Digital [ALC887-VD Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 cat /proc/asound/cards 0 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xfcf40000 irq 67 1 [Generic ]: HDA-Intel - HD-Audio Generic HD-Audio Generic at 0xfce00000 irq 69 speaker-test 1.2.4 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise ALSA lib confmisc.c:855:(parse_card) cannot find card '1dmix:1' ALSA lib conf.c:5091:(_snd_config_evaluate) function snd_func_card_id returned error: No such device ALSA lib confmisc.c:422:(snd_func_concat) error evaluating strings ALSA lib conf.c:5091:(_snd_config_evaluate) function snd_func_concat returned error: No such device ALSA lib confmisc.c:1334:(snd_func_refer) error evaluating name ALSA lib conf.c:5091:(_snd_config_evaluate) function snd_func_refer returned error: No such device ALSA lib conf.c:5579:(snd_config_expand) Evaluate error: No such device ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM dmix:1dmix:1 Playback open error: -19,No such device Issue URL : https://github.com/alsa-project/alsa-lib/issues/149 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2] ASoC: qcom: Add four speaker support on MI2S secondary
by Srinivasa Rao Mandadapu 09 Jun '21

09 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 V1: -- removed set_channel_map/get_channel_map implementation as default kcontrols added in pcm_new API. sound/soc/qcom/lpass-cpu.c | 32 ++++++++++++++++++++++++++++++++ sound/soc/qcom/lpass-sc7180.c | 1 + sound/soc/qcom/lpass.h | 2 ++ 3 files changed, 35 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index c62d2612e8f5..15d69e311b38 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -29,6 +29,14 @@ #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 +332,25 @@ 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 +883,11 @@ 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.

2 2

[PATCH] usb-audio: Add support for Denon DN-X1600
by Damien Zammit 09 Jun '21

09 Jun '21

This patch applies clean to tiwai/sound.git repo on hash 82b954a96c1550bae2cb1f1f4f084425ab237cd7 and was tested on this version. This provides support for Denon DN-X1600 hardware mixer. The device itself supports 44100, 48000 and 96000 (Hz) sample rates, but switching rates via software is currently not working. Therefore, this patch hardcodes the sample rate to 48000Hz which enables all 8 channels to function correctly when the correct sample rate is selected on the hardware itself. MIDI also tested and works. Signed-off-by: Damien Zammit <damien(a)zamaudio.com> Tested-by: xalmoxis(a)gmail.com --- sound/usb/clock.c | 7 +++++ sound/usb/pcm.c | 4 +++ sound/usb/quirks-table.h | 70 ++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 81 insertions(+) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 0afae839d..52de52288 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -534,6 +534,13 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, * rate. */ clock = snd_usb_clock_find_source(chip, fmt, false); + + /* Denon DN-X1600 hardcoded + * Sample rate seems to be set on the hardware itself + */ + if (chip->usb_id == USB_ID(0x154e, 0x500e)) + return 0; + if (clock < 0) return clock; } diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index c66831ee1..6913a5b02 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -197,6 +197,10 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, if (!(fmt->attributes & UAC_EP_CS_ATTR_PITCH_CONTROL)) return 0; + /* Denon DN-X1600 doesn't properly handle this, bail out too */ + if (chip->usb_id == USB_ID(0x154e, 0x500e)) + return 0; + usb_audio_dbg(chip, "enable PITCH for EP 0x%x\n", fmt->endpoint); switch (fmt->protocol) { diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index bdba37d0f..19bb499c1 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3044,6 +3044,76 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } }, +/* Denon DN-X1600 */ +{ + USB_AUDIO_DEVICE(0x154e, 0x500e), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Denon", + .product_name = "DN-X1600", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]){ + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE, + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0x0, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ADAPTIVE, + .maxpacksize = 0x138, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { + 48000 + } + } + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0x0, + .endpoint = 0x85, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ADAPTIVE, + .maxpacksize = 0x138, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { + 48000 + } + } + }, + { + .ifnum = 4, + .type = QUIRK_MIDI_STANDARD_INTERFACE, + }, + { + .ifnum = -1 + } + } + } +}, + /* Microsoft XboxLive Headset/Xbox Communicator */ { USB_DEVICE(0x045e, 0x0283), -- 2.13.1

2 1

[PATCH 0/3] ALSA: pcm:firewire: allow to operate for period elapse event in process context
by Takashi Sakamoto 09 Jun '21

09 Jun '21

Hi, This patchset comes from my former RFC: [RFC][PATCH 0/3] ALSA: pcm/firewire: allow to queue period elapse event in process context * https://lore.kernel.org/alsa-devel/20210606091838.80812-1-o-takashi@sakamoc… All of drivers in ALSA firewire stack have two chances to process isochronous packets of any isochronous context; in software IRQ context for 1394 OHCI, and in process context of ALSA PCM application. In the process context, callbacks of .pointer and .ack are utilized. The callbacks are done by ALSA PCM core under acquiring lock of PCM substream, In design of ALSA PCM core, call of snd_pcm_period_elapsed() is used for drivers to awaken user processes from waiting for available frames. The function voluntarily acquires lock of PCM substream, therefore it is not called in the process context since it causes dead lock. As a workaround to avoid the dead lock, all of drivers in ALSA firewire stack use workqueue to delegate the call. This patchset is my attempt for the issue. A variant of 'snd_pcm_period_elapsed()' without lock acquisition is going to be added, named 'snd_pcm_period_elapsed_under_stream_lock()'. The call is available in callbacks of .pointer and .ack of snd_pcm_ops structure. Changes from RFC: * dismiss inlining * rename function name * improve function comments I tested the patchset with ALSA OXFW driver and ftrace, covering axfer and jackd cases since pulseaudio and pipewire are programmed with timer-based scheduling model and ALSA runtime expects drivers not to call snd_pcm_period_elapsed(). The configuration of tracer and filters is: ``` $ sudo trace-cmd record -p function_graph \ -l :mod:snd_firewire_lib \ -l fw_iso_context_queue:mod:firewire_core \ -l snd_pcm_ioctl:mod:snd_pcm \ -l snd_pcm_period_elapsed*:mod:snd_pcm ``` The runtime of axfer without '-M' option can often call .pointer and .ack. Below is a sample about .pointer case via SNDRV_PCM_IOCT_HWSYNC. The new snd_pcm_period_elapsed_under_stream_lock() should be called. ``` 3929.769359: funcgraph_entry: | snd_pcm_ioctl() { 3929.769360: funcgraph_entry: | amdtp_domain_stream_pcm_pointer() { 3929.769361: funcgraph_entry: | irq_target_callback() { 3929.769361: funcgraph_entry: | process_rx_packets() { 3929.769362: funcgraph_entry: | process_ctx_payloads() { 3929.769363: funcgraph_entry: 0.391 us | process_it_ctx_payloads(); 3929.769363: funcgraph_entry: | snd_pcm_period_elapsed_under_stream_lock() { 3929.769364: funcgraph_entry: 0.411 us | amdtp_domain_stream_pcm_pointer(); 3929.769365: funcgraph_exit: 1.723 us | } 3929.769365: funcgraph_exit: 3.106 us | } 3929.769366: funcgraph_entry: | queue_packet() { 3929.769366: funcgraph_entry: 0.561 us | fw_iso_context_queue(); 3929.769367: funcgraph_exit: 1.122 us | } 3929.769367: funcgraph_exit: 5.731 us | } 3929.769367: funcgraph_entry: | process_ctxs_in_domain() { 3929.769368: funcgraph_entry: | process_tx_packets() { 3929.769369: funcgraph_entry: | process_ctx_payloads() { 3929.769369: funcgraph_entry: 0.321 us | process_ir_ctx_payloads(); 3929.769369: funcgraph_exit: 0.962 us | } 3929.769370: funcgraph_entry: 0.491 us | fw_iso_context_queue(); 3929.769371: funcgraph_exit: 2.364 us | } 3929.769371: funcgraph_exit: 3.427 us | } 3929.769371: funcgraph_exit: 10.038 us | } 3929.769372: funcgraph_exit: 11.271 us | } 3929.769372: funcgraph_exit: 13.606 us | } ``` The runtime of jackd heavily relies on period wakeup scheduled in invocation of interrupt handler. Below is a sample about software IRQ case of 1394 OHCI. The snd_pcm_period_elapsed() should be called. ``` 5318.980502: funcgraph_entry: | irq_target_callback() { 5318.980503: funcgraph_entry: | process_rx_packets() { 5318.980503: funcgraph_entry: | process_ctx_payloads() { 5318.980503: funcgraph_entry: 0.531 us | process_it_ctx_payloads(); 5318.980505: funcgraph_entry: | snd_pcm_period_elapsed() { 5318.980505: funcgraph_entry: | snd_pcm_period_elapsed_under_stream_lock() { 5318.980505: funcgraph_entry: 0.180 us | amdtp_domain_stream_pcm_pointer(); 5318.980508: funcgraph_exit: 3.106 us | } 5318.980508: funcgraph_exit: 3.497 us | } 5318.980508: funcgraph_exit: 4.990 us | } 5318.980508: funcgraph_entry: | queue_packet() { 5318.980509: funcgraph_entry: 0.350 us | fw_iso_context_queue(); 5318.980509: funcgraph_exit: 0.702 us | } 5318.980509: funcgraph_entry: | queue_packet() { 5318.980509: funcgraph_entry: 0.291 us | fw_iso_context_queue(); 5318.980510: funcgraph_exit: 0.651 us | } 5318.980510: funcgraph_entry: | queue_packet() { 5318.980510: funcgraph_entry: 0.280 us | fw_iso_context_queue(); 5318.980511: funcgraph_exit: 0.641 us | } 5318.980511: funcgraph_exit: 8.416 us | } 5318.980511: funcgraph_entry: | process_ctxs_in_domain() { 5318.980512: funcgraph_entry: | process_tx_packets() { 5318.980512: funcgraph_entry: | process_ctx_payloads() { 5318.980512: funcgraph_entry: 0.531 us | process_ir_ctx_payloads(); 5318.980513: funcgraph_entry: | snd_pcm_period_elapsed() { 5318.980513: funcgraph_entry: | snd_pcm_period_elapsed_under_stream_lock() { 5318.980513: funcgraph_entry: 0.170 us | amdtp_domain_stream_pcm_pointer(); 5318.980514: funcgraph_exit: 0.842 us | } 5318.980514: funcgraph_exit: 1.242 us | } 5318.980514: funcgraph_exit: 2.335 us | } 5318.980515: funcgraph_entry: 0.301 us | fw_iso_context_queue(); 5318.980515: funcgraph_entry: 0.291 us | fw_iso_context_queue(); 5318.980516: funcgraph_entry: 0.290 us | fw_iso_context_queue(); 5318.980516: funcgraph_exit: 4.198 us | } 5318.980516: funcgraph_exit: 5.119 us | } 5318.980516: funcgraph_exit: 14.077 us | } ``` ALSA OXFW driver works well to select appropriate kernel API by distinguishing running context. I also tested for error cases that the sequence of tx packets is invalid or packet queueing fails. In both cases, the error doesn't bring system corruption. Takashi Sakamoto (3): ALSA: pcm: add snd_pcm_period_elapsed() variant without acquiring lock of PCM substream ALSA: firewire-lib: operate for period elapse event in process context ALSA: firewire-lib: obsolete workqueue for period update include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 68 +++++++++++++++++++++++++++-------- sound/firewire/amdtp-stream.c | 46 ++++++++---------------- sound/firewire/amdtp-stream.h | 1 - 4 files changed, 68 insertions(+), 48 deletions(-) -- 2.27.0

2 5

[PATCH] Asoc: qcom: Add four speaker support on MI2S secondary
by Srinivasa Rao Mandadapu 09 Jun '21

09 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> --- sound/soc/qcom/lpass-cpu.c | 106 +++++++++++++++++++++++++++++++++- sound/soc/qcom/lpass-sc7180.c | 1 + sound/soc/qcom/lpass.h | 5 ++ 3 files changed, 111 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index c62d2612e8f5..88a038a107a7 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -28,6 +28,19 @@ #define LPASS_CPU_I2S_SD2_3_MASK GENMASK(3, 2) #define LPASS_CPU_I2S_SD0_1_2_MASK GENMASK(2, 0) #define LPASS_CPU_I2S_SD0_1_2_3_MASK GENMASK(3, 0) +#define LPASS_CHMAP_IDX_UNKNOWN -1 + +/* + * Channel maps for multi-channel playbacks on MI2S Secondary, up to 4 n_ch + */ +static struct snd_pcm_chmap_elem lpass_chmaps[] = { + { .channels = 2, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, + { } +}; static int lpass_cpu_init_i2sctl_bitfields(struct device *dev, struct lpaif_i2sctl *i2sctl, struct regmap *map) @@ -86,6 +99,9 @@ static int lpass_cpu_daiops_startup(struct snd_pcm_substream *substream, clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); return ret; } + if (drvdata->chmap_info && drvdata->chmap_info->max_channels == 4) + drvdata->chmap_info->chmap = lpass_chmaps; + return 0; } @@ -96,6 +112,7 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream, clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); + drvdata->chmap_idx = LPASS_CHMAP_IDX_UNKNOWN; } static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, @@ -224,9 +241,14 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, ret); return ret; } - if (channels >= 2) + if (channels >= 2) { ret = regmap_fields_write(i2sctl->spkmono, id, LPAIF_I2SCTL_SPKMONO_STEREO); + if (channels == 4) + drvdata->chmap_idx = 1; + else + drvdata->chmap_idx = 0; + } else ret = regmap_fields_write(i2sctl->spkmono, id, LPAIF_I2SCTL_SPKMONO_MONO); @@ -324,6 +346,84 @@ const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); +static int lpass_cpu_chmap_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + unsigned const char *map; + unsigned int i; + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + struct lpass_data *drvdata = info->private_data; + + if (info->max_channels == 2) + map = &lpass_chmaps[0].map[0]; + else if (info->max_channels == 4) + map = &lpass_chmaps[1].map[0]; + + for (i = 0; i < info->max_channels; i++) { + if (drvdata->chmap_idx == LPASS_CHMAP_IDX_UNKNOWN) + ucontrol->value.integer.value[i] = 0; + else + ucontrol->value.integer.value[i] = map[i]; + } + + return 0; +} + +static int lpass_cpu_chmap_ctl_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + unsigned int i; + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + struct lpass_data *drvdata = info->private_data; + unsigned char *map; + + if (info->max_channels == 2) + map = &lpass_chmaps[0].map[0]; + else if (info->max_channels == 4) + map = &lpass_chmaps[1].map[0]; + + for (i = 0; i < info->max_channels; i++) { + if (drvdata->chmap_idx == LPASS_CHMAP_IDX_UNKNOWN) + map[i] = 0; + else + map[i] = ucontrol->value.integer.value[i]; + } + + return 0; +} + +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); + + ret = snd_pcm_add_chmap_ctls(rtd->pcm, SNDRV_PCM_STREAM_PLAYBACK, + snd_pcm_alt_chmaps, drv->playback.channels_max, 0, + &drvdata->chmap_info); + if (ret < 0) + return ret; + + + /* + * override handlers + */ + drvdata->chmap_info->private_data = drvdata; + drvdata->chmap_info->kctl->get = lpass_cpu_chmap_ctl_get; + drvdata->chmap_info->kctl->put = lpass_cpu_chmap_ctl_set; + + /* + * default chmap supported is stereo + */ + drvdata->chmap_info->chmap = lpass_chmaps; + drvdata->chmap_idx = LPASS_CHMAP_IDX_UNKNOWN; + + 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 +956,10 @@ 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..fb82591d550c 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -76,6 +76,9 @@ struct lpass_data { /* regmap backed by the low-power audio interface (LPAIF) registers */ struct regmap *lpaif_map; struct regmap *hdmiif_map; + struct snd_pcm_chmap *chmap_info; + + int chmap_idx; /* interrupts from the low-power audio interface (LPAIF) */ int lpaif_irq; @@ -259,5 +262,7 @@ 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.

2 2

[PATCH 0/9] ASoC: tidyup snd_soc_of_parse_daifmt()
by Kuninori Morimoto 09 Jun '21

09 Jun '21

Hi Mark I want to add new audio-graph-card2 sound card driver, and this is last part of necessary soc-core cleanup for it. Current some drivers are using DT, and Then, snd_soc_of_parse_daifmt() parses daifmt, but bitclock/frame provider parsing part is one of headache, because we are assuming below both cases. A) node { bitclock-master; frame-master; ... }; B) link { bitclock-master = <&xxx>; frame-master = <&xxx>; ... }; The original was style A), and style B) was added later. snd_soc_of_parse_daifmt() parses A) style as original style, and user need to update to B) style for clock_provider part if needed. In such case, user need to re-parse it, like below. daifmt = snd_soc_of_parse_daifmt(..., &bitclkmaster, &framemaster); daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK; if (codec == bitclkmaster) daifmt |= (codec == framemaster) ? SND_SOC_DAIFMT_CBM_CFM : SND_SOC_DAIFMT_CBM_CFS; else daifmt |= (codec == framemaster) ? SND_SOC_DAIFMT_CBS_CFM : SND_SOC_DAIFMT_CBS_CFS; This patch-set adds new functions, and handle these more simply. Unfortunately, there are too many use-case, do it by 1 function was implessible. style A) bit_frame = snd_soc_daifmt_parse_clock_provider(); daifmt = snd_soc_daifmt_parse_format(...) | /* format part */ snd_soc_daifmt_clock_provider_pickup(bit_frame); /* clock part */ style B) snd_soc_daifmt_parse_clock_provider(..., &bit, &frame); daifmt = snd_soc_daifmt_parse_format(...) | /* format part */ snd_soc_daifmt_clock_provider_pickup( /* clock part */ ((codec == bit) << 4) + (codec == frame)); Kuninori Morimoto (9): ASoC: soc-core: don't use discriminatory terms on snd_soc_runtime_get_dai_fmt() ASoC: soc-core: add snd_soc_daifmt_clock_provider_pickup() ASoC: soc-core: add snd_soc_daifmt_clock_provider_fliped() ASoC: soc-core: add snd_soc_daifmt_parse_format/clock_provider() ASoC: atmel: switch to use snd_soc_daifmt_parse_format/clock_provider() ASoC: fsl: switch to use snd_soc_daifmt_parse_format/clock_provider() ASoC: meson: switch to use snd_soc_daifmt_parse_format/clock_provider() ASoC: simple-card-utils: switch to use snd_soc_daifmt_parse_format/clock_provider() ASoC: soc-core: remove snd_soc_of_parse_daifmt() include/sound/soc.h | 13 +++- sound/soc/atmel/mikroe-proto.c | 18 ++--- sound/soc/fsl/fsl-asoc-card.c | 16 +--- sound/soc/generic/simple-card-utils.c | 19 ++--- sound/soc/meson/meson-card-utils.c | 15 ++-- sound/soc/soc-core.c | 103 ++++++++++++++++---------- 6 files changed, 99 insertions(+), 85 deletions(-) -- 2.25.1

3 13

[bug report] media: sound/usb: Use Media Controller API to share media resources
by Dan Carpenter 08 Jun '21

08 Jun '21

Hello Shuah Khan, The patch 66354f18fe5f: "media: sound/usb: Use Media Controller API to share media resources" from Apr 1, 2019, leads to the following static checker warning: sound/usb/media.c:287 snd_media_device_create() warn: 'mdev' can also be NULL sound/usb/media.c 270 271 mdev = media_device_usb_allocate(usbdev, KBUILD_MODNAME, THIS_MODULE); ^^^^ If CONFIG_MEDIA_CONTROLLER is disabled then "mdev" is NULL. 272 if (IS_ERR(mdev)) 273 return -ENOMEM; 274 275 /* save media device - avoid lookups */ 276 chip->media_dev = mdev; 277 278 snd_mixer_init: 279 /* Create media entities for mixer and control dev */ 280 ret = snd_media_mixer_init(chip); 281 /* media_device might be registered, print error and continue */ 282 if (ret) 283 dev_err(&usbdev->dev, 284 "Couldn't create media mixer entities. Error: %d\n", 285 ret); 286 287 if (!media_devnode_is_registered(mdev->devnode)) { ^^^^^^^^^^^^^ dereferenced without checking here 288 /* dont'register if snd_media_mixer_init() failed */ 289 if (ret) 290 goto create_fail; 291 292 /* register media_device */ 293 ret = media_device_register(mdev); 294 create_fail: 295 if (ret) { 296 snd_media_mixer_delete(chip); 297 media_device_delete(mdev, KBUILD_MODNAME, THIS_MODULE); 298 /* clear saved media_dev */ 299 chip->media_dev = NULL; 300 dev_err(&usbdev->dev, 301 "Couldn't register media device. Error: %d\n", 302 ret); 303 return ret; 304 } 305 } 306 307 return ret; 308 } regards, dan carpenter

2 3

alsamixer: use background color instead of COLOR_BLACK
by GitHub pull_request - opened 08 Jun '21

08 Jun '21

alsa-project/alsa-utils pull request #77 was opened from r-burns: Hi! I have a cosmetic "improvement" I'd like to share. Scare quotes because this is possibly opinionated, but I'd still like to know what you think. I think on typical terminals, COLOR_BLACK is the same or quite similar to the background color, so this does not make much of a difference. But on "solarized" color schemes and similar, the COLOR_BLACK may be quite different from the background color, so in my opinion it looks odd to have the alsamixer panel stand out this way. For example, I'm using the "Nord" color scheme on a pitch black #000000 background, and here's how the mixer looks with/without this patch (blank terminal, htop, and ncmpcpp also provided for comparison): ![image](https://user-images.githubusercontent.com/52847440/107131140-dbf46500-6888-11eb-938a-9a3258bf7184.png) As you can see, the current alsamixer in the center looks a bit strange and sticks out against my chosen background color and the other TUIs I'm using. Htop and ncmpcpp agree on using the builtin `-1` background color for their own backgrounds, but alsamixer uses COLOR_BLACK which stands out from the actual background. The alsamixer on the bottom right is using this patch, and blends in seamlessly. Again, this is just my opinion! Feel free to close if this is wrong or superfluous. Nonetheless, I thought I'd share this patch, in case anyone else feels the same way. Request URL : https://github.com/alsa-project/alsa-utils/pull/77 Patch URL : https://github.com/alsa-project/alsa-utils/pull/77.patch Repository URL: https://github.com/alsa-project/alsa-utils

2 1

[PATCH 00/15] ASoC: fixes for SoundWire codec drivers
by Pierre-Louis Bossart 08 Jun '21

08 Jun '21

The SOF CI exposed a set of issues with suspend/resume, error handling, register access and mixer values. These fixes were indentified with a tag so that they can be applied by linux-stable and distributions. Thanks to the Realtek and Maxim teams for their help. Bard Liao (3): ASoC: rt5682-sdw: set regcache_cache_only false before reading RT5682_DEVICE_ID ASoC: rt711-sdca-sdw: add readable for SDW_SDCA_CTL() registers ASoC: rt711-sdca: handle mbq_regmap in rt711_sdca_io_init Jack Yu (1): ASoC: rt715-sdca: fix clock stop prepare timeout issue Oder Chiou (1): ASoC: rt5682: Fix a problem with error handling in the io init function of the soundwire Pierre-Louis Bossart (10): ASoC: max98373-sdw: add missing memory allocation check ASoC: max98373-sdw: use first_hw_init flag on resume ASoC: rt1308-sdw: use first_hw_init flag on resume ASoC: rt1316-sdw: use first_hw_init flag on resume ASoC: rt5682-sdw: use first_hw_init flag on resume ASoC: rt700-sdw: use first_hw_init flag on resume ASoC: rt711-sdca-sdw: use first_hw_init flag on resume ASoC: rt711-sdw: use first_hw_init flag on resume ASoC: rt715-sdca-sdw: use first_hw_init flag on resume ASoC: rt715-sdw: use first_hw_init flag on resume sound/soc/codecs/max98373-sdw.c | 14 ++++++++------ sound/soc/codecs/max98373.h | 2 +- sound/soc/codecs/rt1308-sdw.c | 2 +- sound/soc/codecs/rt1316-sdw.c | 2 +- sound/soc/codecs/rt5682-sdw.c | 19 +++++++++++-------- sound/soc/codecs/rt700-sdw.c | 2 +- sound/soc/codecs/rt711-sdca-sdw.c | 12 +++++++++++- sound/soc/codecs/rt711-sdca.c | 4 ++++ sound/soc/codecs/rt711-sdw.c | 2 +- sound/soc/codecs/rt715-sdca-sdw.c | 3 ++- sound/soc/codecs/rt715-sdca-sdw.h | 1 + sound/soc/codecs/rt715-sdca.c | 9 ++++++--- sound/soc/codecs/rt715-sdca.h | 3 ++- sound/soc/codecs/rt715-sdw.c | 2 +- 14 files changed, 51 insertions(+), 26 deletions(-) -- 2.25.1

2 16