October 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] soundwire: slave: Fix unused function warning on !ACPI
by Michal Suchanek 15 Oct '19

15 Oct '19

Fixes the following warning on !ACPI systems: drivers/soundwire/slave.c:16:12: warning: ‘sdw_slave_add’ defined but not used [-Wunused-function] static int sdw_slave_add(struct sdw_bus *bus, ^~~~~~~~~~~~~ Signed-off-by: Michal Suchanek <msuchanek(a)suse.de> --- drivers/soundwire/slave.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/slave.c b/drivers/soundwire/slave.c index f39a5815e25d..34c7e65831d1 100644 --- a/drivers/soundwire/slave.c +++ b/drivers/soundwire/slave.c @@ -6,6 +6,7 @@ #include <linux/soundwire/sdw_type.h> #include "bus.h" +#if IS_ENABLED(CONFIG_ACPI) static void sdw_slave_release(struct device *dev) { struct sdw_slave *slave = dev_to_sdw_dev(dev); @@ -60,7 +61,6 @@ static int sdw_slave_add(struct sdw_bus *bus, return ret; } -#if IS_ENABLED(CONFIG_ACPI) /* * sdw_acpi_find_slaves() - Find Slave devices in Master ACPI node * @bus: SDW bus instance -- 2.22.0

6 15

[alsa-devel] [PATCH] soundwire: qcom: fix devm_platform_ioremap_resource.cocci warnings
by Julia Lawall 15 Oct '19

15 Oct '19

From: kbuild test robot <lkp(a)intel.com> Use devm_platform_ioremap_resource helper which wraps platform_get_resource() and devm_ioremap_resource() together. Generated by: scripts/coccinelle/api/devm_platform_ioremap_resource.cocci Fixes: 4a789cff1848 ("soundwire: qcom: add support for SoundWire controller") CC: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Julia Lawall <julia.lawall(a)lip6.fr> --- url: https://github.com/0day-ci/linux/commits/Srinivas-Kandagatla/soundwire-Add-… :::::: branch date: 9 hours ago :::::: commit date: 9 hours ago qcom.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -784,7 +784,6 @@ static int qcom_swrm_probe(struct platfo struct sdw_master_prop *prop; struct sdw_bus_params *params; struct qcom_swrm_ctrl *ctrl; - struct resource *res; int ret; u32 val; @@ -801,8 +800,7 @@ static int qcom_swrm_probe(struct platfo } else { ctrl->reg_read = qcom_swrm_mmio_reg_read; ctrl->reg_write = qcom_swrm_mmio_reg_write; - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - ctrl->base = devm_ioremap_resource(dev, res); + ctrl->base = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(ctrl->base)) return PTR_ERR(ctrl->base); }

1 0

[alsa-devel] [PATCH] ASoC: simple-card: fixup simple_dai_link_of_dpcm() comment
by Kuninori Morimoto 15 Oct '19

15 Oct '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> The comment is wrong. This patch fixup it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-card.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index fc9c753..10b82bf 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -149,7 +149,7 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, if (li->cpu) { int is_single_links = 0; - /* BE is dummy */ + /* Codec is dummy */ codecs->of_node = NULL; codecs->dai_name = "snd-soc-dummy-dai"; codecs->name = "snd-soc-dummy"; @@ -179,7 +179,7 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, } else { struct snd_soc_codec_conf *cconf; - /* FE is dummy */ + /* CPU is dummy */ cpus->of_node = NULL; cpus->dai_name = "snd-soc-dummy-dai"; cpus->name = "snd-soc-dummy"; -- 2.7.4

3 2

[alsa-devel] [PATCH v2] ASoC: Intel: eve: Enable mclk and ssp sclk early
by Brent Lu 15 Oct '19

15 Oct '19

From: Naveen M <naveen.m(a)intel.com> rt5663 and rt5514 needs mclk/sclk early to synchronize its internal clocks. Signed-off-by: Naveen M <naveen.m(a)intel.com> Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 1 + .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 94 ++++++++++++++++++++++ 2 files changed, 95 insertions(+) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 5c27f7a..5e0e7db 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -320,6 +320,7 @@ config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH select SND_SOC_RT5514_SPI select SND_SOC_MAX98927 select SND_SOC_HDAC_HDMI + select SND_SOC_INTEL_SKYLAKE_SSP_CLK help This adds support for ASoC Onboard Codec I2S machine driver. This will create an alsa sound card for RT5663 + RT5514 + MAX98927. diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 74dda87..dc09a85 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -22,6 +22,9 @@ #include "../../codecs/rt5514.h" #include "../../codecs/rt5663.h" #include "../../codecs/hdac_hdmi.h" +#include <linux/clk.h> +#include <linux/clk-provider.h> +#include <linux/clkdev.h> #define KBL_REALTEK_CODEC_DAI "rt5663-aif" #define KBL_REALTEK_DMIC_CODEC_DAI "rt5514-aif1" @@ -50,6 +53,8 @@ struct kbl_codec_private { struct snd_soc_jack kabylake_headset; struct list_head hdmi_pcm_list; struct snd_soc_jack kabylake_hdmi[2]; + struct clk *mclk; + struct clk *sclk; }; enum { @@ -71,6 +76,61 @@ static const struct snd_kcontrol_new kabylake_controls[] = { SOC_DAPM_PIN_SWITCH("DMIC"), }; +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct kbl_codec_private *priv = snd_soc_card_get_drvdata(card); + int ret = 0; + + /* + * MCLK/SCLK need to be ON early for a successful synchronization of + * codec internal clock. And the clocks are turned off during + * POST_PMD after the stream is stopped. + */ + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + /* Enable MCLK */ + ret = clk_set_rate(priv->mclk, 24000000); + if (ret < 0) { + dev_err(card->dev, "Can't set rate for mclk, err: %d\n", + ret); + return ret; + } + + ret = clk_prepare_enable(priv->mclk); + if (ret < 0) { + dev_err(card->dev, "Can't enable mclk, err: %d\n", ret); + return ret; + } + + /* Enable SCLK */ + ret = clk_set_rate(priv->sclk, 3072000); + if (ret < 0) { + dev_err(card->dev, "Can't set rate for sclk, err: %d\n", + ret); + clk_disable_unprepare(priv->mclk); + return ret; + } + + ret = clk_prepare_enable(priv->sclk); + if (ret < 0) { + dev_err(card->dev, "Can't enable sclk, err: %d\n", ret); + clk_disable_unprepare(priv->mclk); + } + break; + case SND_SOC_DAPM_POST_PMD: + clk_disable_unprepare(priv->mclk); + clk_disable_unprepare(priv->sclk); + break; + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -79,11 +139,15 @@ static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), SND_SOC_DAPM_SPK("HDMI1", NULL), SND_SOC_DAPM_SPK("HDMI2", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route kabylake_map[] = { /* Headphones */ + { "Headphone Jack", NULL, "Platform Clock" }, { "Headphone Jack", NULL, "HPOL" }, { "Headphone Jack", NULL, "HPOR" }, @@ -92,6 +156,7 @@ static const struct snd_soc_dapm_route kabylake_map[] = { { "Right Spk", NULL, "Right BE_OUT" }, /* other jacks */ + { "Headset Mic", NULL, "Platform Clock" }, { "IN1P", NULL, "Headset Mic" }, { "IN1N", NULL, "Headset Mic" }, @@ -643,6 +708,7 @@ static int kabylake_audio_probe(struct platform_device *pdev) { struct kbl_codec_private *ctx; struct snd_soc_acpi_mach *mach; + int ret = 0; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -658,6 +724,34 @@ static int kabylake_audio_probe(struct platform_device *pdev) dmic_constraints = mach->mach_params.dmic_num == 2 ? &constraints_dmic_2ch : &constraints_dmic_channels; + ctx->mclk = devm_clk_get(&pdev->dev, "ssp1_mclk"); + if (IS_ERR(ctx->mclk)) { + ret = PTR_ERR(ctx->mclk); + if (ret == -ENOENT) { + dev_info(&pdev->dev, + "Failed to get ssp1_mclk, defer probe\n"); + return -EPROBE_DEFER; + } + + dev_err(&pdev->dev, "Failed to get ssp1_mclk with err:%d\n", + ret); + return ret; + } + + ctx->sclk = devm_clk_get(&pdev->dev, "ssp1_sclk"); + if (IS_ERR(ctx->sclk)) { + ret = PTR_ERR(ctx->sclk); + if (ret == -ENOENT) { + dev_info(&pdev->dev, + "Failed to get ssp1_sclk, defer probe\n"); + return -EPROBE_DEFER; + } + + dev_err(&pdev->dev, "Failed to get ssp1_sclk with err:%d\n", + ret); + return ret; + } + return devm_snd_soc_register_card(&pdev->dev, &kabylake_audio_card); } -- 2.7.4

3 2

[alsa-devel] [PATCH v2] ASoC: pcm3168a: Fix serial mode dependent format support
by Peter Ujfalusi 15 Oct '19

15 Oct '19

fmt 0 is perfectly valid (PCM3168A_FMT_I2S). Remove the return in case fmt == 0. Fixes: ("ASoC: pcm3168a: Use fixup instead of constraint for channels and formats") Reported-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, Changes since v1: - Remove one extra blank line also Regards, Peter sound/soc/codecs/pcm3168a.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 708dac27feff..313500ab36df 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -322,9 +322,6 @@ static void pcm3168a_update_fixup_pcm_stream(struct snd_soc_dai *dai) u64 formats = SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE; unsigned int channel_max = dai->id == PCM3168A_DAI_DAC ? 8 : 6; - if (!pcm3168a->io_params[dai->id].fmt) - return; - if (pcm3168a->io_params[dai->id].fmt == PCM3168A_FMT_RIGHT_J) { /* S16_LE is only supported in RIGHT_J mode */ formats |= SNDRV_PCM_FMTBIT_S16_LE; -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 1

[alsa-devel] [PATCH] ASoC: audio-graph: fixup graph_dai_link_of_dpcm() comment
by Kuninori Morimoto 15 Oct '19

15 Oct '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> The comment is wrong. This patch fixup it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/audio-graph-card.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 6007e63..9ad35d9 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -232,7 +232,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, if (li->cpu) { int is_single_links = 0; - /* BE is dummy */ + /* Codec is dummy */ codecs->of_node = NULL; codecs->dai_name = "snd-soc-dummy-dai"; codecs->name = "snd-soc-dummy"; @@ -263,7 +263,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, } else { struct snd_soc_codec_conf *cconf; - /* FE is dummy */ + /* CPU is dummy */ cpus->of_node = NULL; cpus->dai_name = "snd-soc-dummy-dai"; cpus->name = "snd-soc-dummy"; -- 2.7.4

3 2

[alsa-devel] [PATCH] ASoC: pcm3168a: Fix serial mode dependent format support
by Peter Ujfalusi 15 Oct '19

15 Oct '19

fmt 0 is perfectly valid (PCM3168A_FMT_I2S). Remove the return in case fmt == 0. Fixes: ("ASoC: pcm3168a: Use fixup instead of constraint for channels and formats") Reported-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/pcm3168a.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 708dac27feff..df8395b361d4 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -322,8 +322,6 @@ static void pcm3168a_update_fixup_pcm_stream(struct snd_soc_dai *dai) u64 formats = SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE; unsigned int channel_max = dai->id == PCM3168A_DAI_DAC ? 8 : 6; - if (!pcm3168a->io_params[dai->id].fmt) - return; if (pcm3168a->io_params[dai->id].fmt == PCM3168A_FMT_RIGHT_J) { /* S16_LE is only supported in RIGHT_J mode */ -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

1 1

[alsa-devel] [PATCH v3] drm/bridge: dw-hdmi: Restore audio when setting a mode
by Cheng-Yi Chiang 15 Oct '19

15 Oct '19

From: Daniel Kurtz <djkurtz(a)chromium.org> When setting a new display mode, dw_hdmi_setup() calls dw_hdmi_enable_video_path(), which disables all hdmi clocks, including the audio clock. We should only (re-)enable the audio clock if audio was already enabled when setting the new mode. Without this patch, on RK3288, there will be HDMI audio on some monitors if i2s was played to headphone when the monitor was plugged. ACER H277HU and ASUS PB278 are two of the monitors showing this issue. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> Signed-off-by: Daniel Kurtz <djkurtz(a)chromium.org> Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> --- Change from v2 to v3: - Remove spinlock around setting clock. drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c index a15fbf71b9d7..af060162b0af 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c @@ -2054,7 +2054,7 @@ static int dw_hdmi_setup(struct dw_hdmi *hdmi, struct drm_display_mode *mode) /* HDMI Initialization Step E - Configure audio */ hdmi_clk_regenerator_update_pixel_clock(hdmi); - hdmi_enable_audio_clk(hdmi, true); + hdmi_enable_audio_clk(hdmi, hdmi->audio_enable); } /* not for DVI mode */ -- 2.23.0.581.g78d2f28ef7-goog

2 1

[alsa-devel] [RFC PATCH 0/2] Introduce for-dpcm DT property
by Daniel Baluta 15 Oct '19

15 Oct '19

We need to be able to create DPCM links even if we have a single CPU DAI or just a dummy CPU DAI. Daniel Baluta (2): ASoC: simple-card: Introduce force-dpcm DT property ASoC: simple-card: Add documentation for force-dpcm property .../devicetree/bindings/sound/simple-card.txt | 1 + include/sound/simple_card_utils.h | 4 +++ sound/soc/generic/simple-card-utils.c | 17 +++++++++++++ sound/soc/generic/simple-card.c | 25 +++++++++++++++++-- 4 files changed, 45 insertions(+), 2 deletions(-) -- 2.17.1

4 7

Re: [alsa-devel] [PATCH 2/2] ASoC: simple_card_utils.h: Fix potential multiple redefinition error
by Kuninori Morimoto 15 Oct '19

15 Oct '19

Hi > asoc_simple_debug_info and asoc_simple_debug_dai must be static > otherwise we might a compilation error if the compiler decides > not to inline the given function. > > Fixes: 0580dde59438686d ("ASoC: simple-card-utils: add asoc_simple_debug_info()") > Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> > --- Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Thank you for your help !! Best regards --- Kuninori Morimoto

1 0

[alsa-devel] [PATCH] ASoC: Intel: eve: Enable mclk and ssp sclk early
by Brent Lu 14 Oct '19

14 Oct '19

From: Naveen M <naveen.m(a)intel.com> rt5663 and rt5514 needs mclk/sclk early to synchronize its internal clocks. Signed-off-by: Naveen M <naveen.m(a)intel.com> Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 1 + .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 100 +++++++++++++++++++++ 2 files changed, 101 insertions(+) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 5c27f7a..d5f167e 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -315,6 +315,7 @@ config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST depends on SPI + select SND_SOC_INTEL_SKYLAKE_SSP_CLK select SND_SOC_RT5663 select SND_SOC_RT5514 select SND_SOC_RT5514_SPI diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 74dda87..7b6e70b 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -22,6 +22,9 @@ #include "../../codecs/rt5514.h" #include "../../codecs/rt5663.h" #include "../../codecs/hdac_hdmi.h" +#include <linux/clk.h> +#include <linux/clk-provider.h> +#include <linux/clkdev.h> #define KBL_REALTEK_CODEC_DAI "rt5663-aif" #define KBL_REALTEK_DMIC_CODEC_DAI "rt5514-aif1" @@ -50,6 +53,8 @@ struct kbl_codec_private { struct snd_soc_jack kabylake_headset; struct list_head hdmi_pcm_list; struct snd_soc_jack kabylake_hdmi[2]; + struct clk *mclk; + struct clk *sclk; }; enum { @@ -71,6 +76,67 @@ static const struct snd_kcontrol_new kabylake_controls[] = { SOC_DAPM_PIN_SWITCH("DMIC"), }; +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct kbl_codec_private *priv = snd_soc_card_get_drvdata(card); + int ret = 0; + + /* + * MCLK/SCLK need to be ON early for a successful synchronization of + * codec internal clock. And the clocks are turned off during + * POST_PMD after the stream is stopped. + */ + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (__clk_is_enabled(priv->mclk)) + return 0; + + /* Enable MCLK */ + ret = clk_set_rate(priv->mclk, 24000000); + if (ret < 0) { + dev_err(card->dev, "Can't set rate for mclk, err: %d\n", + ret); + return ret; + } + + ret = clk_prepare_enable(priv->mclk); + if (ret < 0) { + dev_err(card->dev, "Can't enable mclk, err: %d\n", ret); + return ret; + } + + /* Enable SCLK */ + ret = clk_set_rate(priv->sclk, 3072000); + if (ret < 0) { + dev_err(card->dev, "Can't set rate for sclk, err: %d\n", + ret); + clk_disable_unprepare(priv->mclk); + return ret; + } + + ret = clk_prepare_enable(priv->sclk); + if (ret < 0) { + dev_err(card->dev, "Can't enable sclk, err: %d\n", ret); + clk_disable_unprepare(priv->mclk); + } + break; + case SND_SOC_DAPM_POST_PMD: + if (!__clk_is_enabled(priv->mclk)) + return 0; + + clk_disable_unprepare(priv->mclk); + clk_disable_unprepare(priv->sclk); + break; + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -79,11 +145,15 @@ static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), SND_SOC_DAPM_SPK("HDMI1", NULL), SND_SOC_DAPM_SPK("HDMI2", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route kabylake_map[] = { /* Headphones */ + { "Headphone Jack", NULL, "Platform Clock" }, { "Headphone Jack", NULL, "HPOL" }, { "Headphone Jack", NULL, "HPOR" }, @@ -92,6 +162,7 @@ static const struct snd_soc_dapm_route kabylake_map[] = { { "Right Spk", NULL, "Right BE_OUT" }, /* other jacks */ + { "Headset Mic", NULL, "Platform Clock" }, { "IN1P", NULL, "Headset Mic" }, { "IN1N", NULL, "Headset Mic" }, @@ -643,6 +714,7 @@ static int kabylake_audio_probe(struct platform_device *pdev) { struct kbl_codec_private *ctx; struct snd_soc_acpi_mach *mach; + int ret = 0; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -658,6 +730,34 @@ static int kabylake_audio_probe(struct platform_device *pdev) dmic_constraints = mach->mach_params.dmic_num == 2 ? &constraints_dmic_2ch : &constraints_dmic_channels; + ctx->mclk = devm_clk_get(&pdev->dev, "ssp1_mclk"); + if (IS_ERR(ctx->mclk)) { + ret = PTR_ERR(ctx->mclk); + if (ret == -ENOENT) { + dev_info(&pdev->dev, + "Failed to get ssp1_mclk, defer probe\n"); + return -EPROBE_DEFER; + } + + dev_err(&pdev->dev, "Failed to get ssp1_mclk with err:%d\n", + ret); + return ret; + } + + ctx->sclk = devm_clk_get(&pdev->dev, "ssp1_sclk"); + if (IS_ERR(ctx->sclk)) { + ret = PTR_ERR(ctx->sclk); + if (ret == -ENOENT) { + dev_info(&pdev->dev, + "Failed to get ssp1_sclk, defer probe\n"); + return -EPROBE_DEFER; + } + + dev_err(&pdev->dev, "Failed to get ssp1_sclk with err:%d\n", + ret); + return ret; + } + return devm_snd_soc_register_card(&pdev->dev, &kabylake_audio_card); } -- 2.7.4

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[alsa-devel] [PATCH -next] ASoC: SOF: Fix randbuild error
by YueHaibing 14 Oct '19

14 Oct '19

When LEDS_TRIGGER_AUDIO is m and SND_SOC_SOF is y, sound/soc/sof/control.o: In function `snd_sof_switch_put': control.c:(.text+0x587): undefined reference to `ledtrig_audio_set' control.c:(.text+0x593): undefined reference to `ledtrig_audio_set' Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: 5d43001ae436 ("ASoC: SOF: acpi led support for switch controls") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/sof/control.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c index 41551e8..2c4abd4 100644 --- a/sound/soc/sof/control.c +++ b/sound/soc/sof/control.c @@ -36,10 +36,12 @@ static void update_mute_led(struct snd_sof_control *scontrol, scontrol->led_ctl.led_value = temp; +#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO) if (!scontrol->led_ctl.direction) ledtrig_audio_set(LED_AUDIO_MUTE, temp ? LED_OFF : LED_ON); else ledtrig_audio_set(LED_AUDIO_MICMUTE, temp ? LED_OFF : LED_ON); +#endif } static inline u32 mixer_to_ipc(unsigned int value, u32 *volume_map, int size) -- 2.7.4

3 2

Re: [alsa-devel] linux-next: Tree for Oct 14 (sound/soc/sof/)
by Randy Dunlap 14 Oct '19

14 Oct '19

On 10/13/19 11:47 PM, Stephen Rothwell wrote: > Hi all, > > Changes since 20191011: > on i386: ld: sound/soc/sof/control.o: in function `snd_sof_switch_put': control.c:(.text+0x49a): undefined reference to `ledtrig_audio_set' ld: control.c:(.text+0x4d1): undefined reference to `ledtrig_audio_set' when CONFIG_LEDS_TRIGGER_AUDIO=m CONFIG_SND_SOC_SOF=y This code in <linux/leds.h> does not handle the config combo above: #if IS_ENABLED(CONFIG_LEDS_TRIGGER_AUDIO) enum led_brightness ledtrig_audio_get(enum led_audio type); void ledtrig_audio_set(enum led_audio type, enum led_brightness state); #else static inline enum led_brightness ledtrig_audio_get(enum led_audio type) { return LED_OFF; } static inline void ledtrig_audio_set(enum led_audio type, enum led_brightness state) { } #endif -- ~Randy

1 0

[alsa-devel] TypeError: 'device' is an invalid keyword argument for this function
by GitHub issues - edited 14 Oct '19

14 Oct '19

alsa-project/alsa-lib issue #13 was edited from paddygoat: Hello again! I upgraded Alsa lib to 1.1.7 and tested ok. A small error as below when running in Jupyter Notebook, Python 3.6:: `#inp = alsaaudio.PCM(alsaaudio.PCM_CAPTURE, alsaaudio.PCM_NONBLOCK) # UltraMic384K 16bit r0, USB Audio inp = alsaaudio.PCM(type = alsaaudio.PCM_CAPTURE, mode=alsaaudio.PCM_NORMAL, device='UltraMic384K 16bit r0') # class alsaaudio.PCM(type=PCM_PLAYBACK, mode=PCM_NORMAL, device='default', cardindex=-1)` --------------------------------------------------------------------------- TypeError Traceback (most recent call last) <ipython-input-6-947bfb637aff> in <module> 16 modelName="randomforestPaddyFalse" 17 modelType="randomforest" ---> 18 recordAnalyzeAudio(10, outputWavFile, 2.0, modelName, modelType) <ipython-input-5-8ddddfa5b724> in recordAnalyzeAudio(duration, outputWavFile, midTermBufferSizeSec, modelName, modelType) 87 # UltraMic384K 16bit r0, USB Audio 88 ---> 89 inp = alsaaudio.PCM(type = alsaaudio.PCM_CAPTURE, mode=alsaaudio.PCM_NORMAL, device='UltraMic384K 16bit r0') 90 # class alsaaudio.PCM(type=PCM_PLAYBACK, mode=PCM_NORMAL, device='default', cardindex=-1) 91 inp.setchannels(2) TypeError: 'device' is an invalid keyword argument for this function Thanks! Issue URL : https://github.com/alsa-project/alsa-lib/issues/13 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] TypeError: 'device' is an invalid keyword argument for this function
by GitHub issues - opened 14 Oct '19

14 Oct '19

alsa-project/alsa-lib issue #13 was opened from paddygoat: Hello again! I upgraded Alsa lib to 1.1.7 and tested ok. A small error as below when running in Jupyter Notebook, Python 3.6:: ` #inp = alsaaudio.PCM(alsaaudio.PCM_CAPTURE, alsaaudio.PCM_NONBLOCK) # UltraMic384K 16bit r0, USB Audio inp = alsaaudio.PCM(type = alsaaudio.PCM_CAPTURE, mode=alsaaudio.PCM_NORMAL, device='UltraMic384K 16bit r0') # class alsaaudio.PCM(type=PCM_PLAYBACK, mode=PCM_NORMAL, device='default', cardindex=-1)` --------------------------------------------------------------------------- TypeError Traceback (most recent call last) <ipython-input-6-947bfb637aff> in <module> 16 modelName="randomforestPaddyFalse" 17 modelType="randomforest" ---> 18 recordAnalyzeAudio(10, outputWavFile, 2.0, modelName, modelType) <ipython-input-5-8ddddfa5b724> in recordAnalyzeAudio(duration, outputWavFile, midTermBufferSizeSec, modelName, modelType) 87 # UltraMic384K 16bit r0, USB Audio 88 ---> 89 inp = alsaaudio.PCM(type = alsaaudio.PCM_CAPTURE, mode=alsaaudio.PCM_NORMAL, device='UltraMic384K 16bit r0') 90 # class alsaaudio.PCM(type=PCM_PLAYBACK, mode=PCM_NORMAL, device='default', cardindex=-1) 91 inp.setchannels(2) TypeError: 'device' is an invalid keyword argument for this function Thanks! Issue URL : https://github.com/alsa-project/alsa-lib/issues/13 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] [PATCH v8 0/3] ASoC: codecs: Add WSA881x Smart Speaker amplifier support
by Srinivas Kandagatla 14 Oct '19

14 Oct '19

This patchset adds support to WSA8810/WSA8815 Class-D Smart Speaker Amplifier which is SoundWire interfaced. This patchset along with DB845c machine driver and WCD934x codec driver has been tested on SDM845 SoC based DragonBoard DB845c with two WSA8810 and Lenovo YOGA C630 Laptop based on SDM850 with WSA8815 speaker amplifiers. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Add thermal sensor support in WSA881x. Thanks, srini Changes since v7: - Added support to dai based on Discussion on mailing list. - Moved readable array to proper switch statement. - Remove unnecessary locks which can be added once tsens is supported. - Suffix gain to Volume as suggested by Mark. Srinivas Kandagatla (3): dt-bindings: ASoC: Add WSA881x bindings dt-bindings: ASoC: WSA881x: Add missing #sound-dai-cells ASoC: codecs: add wsa881x amplifier support .../bindings/sound/qcom,wsa881x.yaml | 68 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wsa881x.c | 1153 +++++++++++++++++ 4 files changed, 1233 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml create mode 100644 sound/soc/codecs/wsa881x.c -- 2.21.0

2 4

[alsa-devel] [PATCH -next] ASoC: adau7118: Fix Kconfig warning without CONFIG_I2C
by YueHaibing 14 Oct '19

14 Oct '19

When building a kernel without CONFIG_I2C, Kconfig warns: WARNING: unmet direct dependencies detected for REGMAP_I2C Depends on [n]: I2C [=n] Selected by [y]: - SND_SOC_ADAU7118_I2C [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] Add missing I2C dependency to SND_SOC_ADAU7118_I2C to fix this. Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: ca514c0f12b0 ("ASOC: Add ADAU7118 8 Channel PDM-to-I2S/TDM Converter driver") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d196752..6a28741 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -416,6 +416,7 @@ config SND_SOC_ADAU7118_HW config SND_SOC_ADAU7118_I2C tristate "Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM Converter - I2C" + depends on I2C select SND_SOC_ADAU7118 select REGMAP_I2C help -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: stm32: spdifrx: retry synchronization in sync state
by Olivier Moysan 14 Oct '19

14 Oct '19

When STM32 SPDIFRX is in sync state, allow multiple synchro attempts, instead of exiting on first unsuccessful trial. This is useful when spdif signal is not immediately available on input. This also allows Pulseaudio to check iec capture device availability when no signal is present. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/stm/stm32_spdifrx.c | 18 ++++++++++++++++-- 1 file changed, 16 insertions(+), 2 deletions(-) diff --git a/sound/soc/stm/stm32_spdifrx.c b/sound/soc/stm/stm32_spdifrx.c index cd4b235fce57..3fd28ee01675 100644 --- a/sound/soc/stm/stm32_spdifrx.c +++ b/sound/soc/stm/stm32_spdifrx.c @@ -351,6 +351,8 @@ static int stm32_spdifrx_start_sync(struct stm32_spdifrx_data *spdifrx) SPDIFRX_CR_CUMSK | SPDIFRX_CR_PTMSK | SPDIFRX_CR_RXSTEO; cr_mask = cr; + cr |= SPDIFRX_CR_NBTRSET(SPDIFRX_NBTR_63); + cr_mask |= SPDIFRX_CR_NBTR_MASK; cr |= SPDIFRX_CR_SPDIFENSET(SPDIFRX_SPDIFEN_SYNC); cr_mask |= SPDIFRX_CR_SPDIFEN_MASK; ret = regmap_update_bits(spdifrx->regmap, STM32_SPDIFRX_CR, @@ -666,7 +668,7 @@ static irqreturn_t stm32_spdifrx_isr(int irq, void *devid) struct snd_pcm_substream *substream = spdifrx->substream; struct platform_device *pdev = spdifrx->pdev; unsigned int cr, mask, sr, imr; - unsigned int flags; + unsigned int flags, sync_state; int err = 0, err_xrun = 0; regmap_read(spdifrx->regmap, STM32_SPDIFRX_SR, &sr); @@ -726,11 +728,23 @@ static irqreturn_t stm32_spdifrx_isr(int irq, void *devid) } if (err) { - /* SPDIFRX in STATE_STOP. Disable SPDIFRX to clear errors */ + regmap_read(spdifrx->regmap, STM32_SPDIFRX_CR, &cr); + sync_state = FIELD_GET(SPDIFRX_CR_SPDIFEN_MASK, cr) && + SPDIFRX_SPDIFEN_SYNC; + + /* SPDIFRX is in STATE_STOP. Disable SPDIFRX to clear errors */ cr = SPDIFRX_CR_SPDIFENSET(SPDIFRX_SPDIFEN_DISABLE); regmap_update_bits(spdifrx->regmap, STM32_SPDIFRX_CR, SPDIFRX_CR_SPDIFEN_MASK, cr); + /* If SPDIFRX was in STATE_SYNC, retry synchro */ + if (sync_state) { + cr = SPDIFRX_CR_SPDIFENSET(SPDIFRX_SPDIFEN_SYNC); + regmap_update_bits(spdifrx->regmap, STM32_SPDIFRX_CR, + SPDIFRX_CR_SPDIFEN_MASK, cr); + return IRQ_HANDLED; + } + if (substream) snd_pcm_stop(substream, SNDRV_PCM_STATE_DISCONNECTED); -- 2.17.1

2 1

[alsa-devel] [PATCH -next] ASoC: fsl_mqs: fix old-style function declaration
by YueHaibing 14 Oct '19

14 Oct '19

gcc warn about this: sound/soc/fsl/fsl_mqs.c:146:1: warning: static is not at beginning of declaration [-Wold-style-declaration] Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/fsl/fsl_mqs.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index f7fc44e..0c813a4 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -143,7 +143,7 @@ static void fsl_mqs_shutdown(struct snd_pcm_substream *substream, MQS_EN_MASK, 0); } -const static struct snd_soc_component_driver soc_codec_fsl_mqs = { +static const struct snd_soc_component_driver soc_codec_fsl_mqs = { .idle_bias_on = 1, .non_legacy_dai_naming = 1, }; -- 2.7.4

4 5

[alsa-devel] [PATCH 0/4] ASoC: SOF: fix static analysis warnings
by Pierre-Louis Bossart 14 Oct '19

14 Oct '19

Recent SOF commits generate warnings reported e.g. by cppcheck. Unfortunately those tools are too verbose to be used by our CI, so there's always a delay in detecting such issues. Pierre-Louis Bossart (4): ASoC: SOF: Intel: byt: fix operator precedence warnings ASoC: SOF: Intel: bdw: fix operator precedence warnings ASoC: SOF: topology: remove always-true redundant test ASoC: SOF: topology: check errors when parsing LED tokens sound/soc/sof/intel/bdw.c | 16 ++++++++-------- sound/soc/sof/intel/byt.c | 16 ++++++++-------- sound/soc/sof/topology.c | 12 ++++++++---- 3 files changed, 24 insertions(+), 20 deletions(-) -- 2.20.1

2 8

[alsa-devel] [PATCH v2 00/10] ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: support WoV
by Tzung-Bi Shih 14 Oct '19

14 Oct '19

This series makes mt6358, cros_ec_codec, and mt8183-mt6358-ts3a227-max98357 support WoV (wake on voice). The first 3 commits are some cleanups and refactors. It looks like breaking the existing interface. But please be noticed that, the cros_ec_codec has not used by any real device yet. The refactor is very necessary to keep the style consistent and for easier to further extend and maintain. platform/chrome: cros_ec: remove unused EC feature ASoC: cros_ec_codec: refactor I2S RX ASoC: cros_ec_codec: extract DMIC EC command from I2S RX The 4th commit extends the feature offered from EC codec. platform/chrome: cros_ec: add common commands for EC codec The 5th commit changes the behavior of setting and getting DMIC gains. ASoC: cros_ec_codec: read max DMIC gain from EC codec The 6th and 7th commit make cros_ec_codec support WoV. ASoC: dt-bindings: cros_ec_codec: add SHM bindings ASoC: cros_ec_codec: support WoV The 8th commit sets necessary registers on mt6358 to support WoV. ASoC: mediatek: mt6358: support WoV The last 2 commit make machine driver mt8183-mt6358-ts3a227-max98357 support WoV if ec-codec is in DTS. ASoC: dt-bindings: mt8183: add ec-codec ASoC: mediatek: mt8183: support WoV Changes from v1: - fix a compile error and make kbuild bot happy https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156315.h… Tzung-Bi Shih (10): platform/chrome: cros_ec: remove unused EC feature ASoC: cros_ec_codec: refactor I2S RX ASoC: cros_ec_codec: extract DMIC EC command from I2S RX platform/chrome: cros_ec: add common commands for EC codec ASoC: cros_ec_codec: read max DMIC gain from EC codec ASoC: dt-bindings: cros_ec_codec: add SHM bindings ASoC: cros_ec_codec: support WoV ASoC: mediatek: mt6358: support WoV ASoC: dt-bindings: mt8183: add ec-codec ASoC: mediatek: mt8183: support WoV .../bindings/sound/google,cros-ec-codec.txt | 24 +- .../sound/mt8183-mt6358-ts3a227-max98357.txt | 3 + drivers/platform/chrome/cros_ec_trace.c | 5 +- .../linux/platform_data/cros_ec_commands.h | 285 ++++- sound/soc/codecs/cros_ec_codec.c | 1125 +++++++++++++---- sound/soc/codecs/mt6358.c | 105 ++ sound/soc/mediatek/Kconfig | 1 + .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 70 +- 8 files changed, 1292 insertions(+), 326 deletions(-) -- 2.23.0.581.g78d2f28ef7-goog

4 18

[alsa-devel] [PATCH] firmware: vpd: Add an interface to read VPD value
by Cheng-Yi Chiang 14 Oct '19

14 Oct '19

Add an interface for other driver to query VPD value. This will be used for ASoC machine driver to query calibration data stored in VPD for smart amplifier speaker resistor calibration. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- drivers/firmware/google/vpd.c | 16 ++++++++++++++++ include/linux/firmware/google/google_vpd.h | 18 ++++++++++++++++++ 2 files changed, 34 insertions(+) create mode 100644 include/linux/firmware/google/google_vpd.h diff --git a/drivers/firmware/google/vpd.c b/drivers/firmware/google/vpd.c index db0812263d46..71e9d2da63be 100644 --- a/drivers/firmware/google/vpd.c +++ b/drivers/firmware/google/vpd.c @@ -65,6 +65,22 @@ static ssize_t vpd_attrib_read(struct file *filp, struct kobject *kobp, info->bin_attr.size); } +int vpd_attribute_read_value(bool ro, const char *key, + char **value, u32 value_len) +{ + struct vpd_attrib_info *info; + struct vpd_section *sec = ro ? &ro_vpd : &rw_vpd; + + list_for_each_entry(info, &sec->attribs, list) { + if (strcmp(info->key, key) == 0) { + *value = kstrndup(info->value, value_len, GFP_KERNEL); + return 0; + } + } + return -EINVAL; +} +EXPORT_SYMBOL(vpd_attribute_read_value); + /* * vpd_section_check_key_name() * diff --git a/include/linux/firmware/google/google_vpd.h b/include/linux/firmware/google/google_vpd.h new file mode 100644 index 000000000000..6f1160f28af8 --- /dev/null +++ b/include/linux/firmware/google/google_vpd.h @@ -0,0 +1,18 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Google VPD interface. + * + * Copyright 2019 Google Inc. + */ + +/* Interface for reading VPD value on Chrome platform. */ + +#ifndef __GOOGLE_VPD_H +#define __GOOGLE_VPD_H + +#include <linux/types.h> + +int vpd_attribute_read_value(bool ro, const char *key, + char **value, u32 value_len); + +#endif /* __GOOGLE_VPD_H */ -- 2.23.0.581.g78d2f28ef7-goog

6 10

[alsa-devel] [PATCH] ASoC: rt1011: export r0 and temperature config
by Shuming [范書銘] 14 Oct '19

14 Oct '19

In chromebook case, the machine driver will get the r0 calibration data and temperature from VPD. Therefore, the codec exports r0 and temperature config API for it. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1011.c | 41 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt1011.h | 5 +++++ 2 files changed, 46 insertions(+) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index a92a0bacd812..a992bb2224b2 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1325,6 +1325,47 @@ static int rt1011_r0_load_info(struct snd_kcontrol *kcontrol, .put = rt1011_r0_load_mode_put \ } +int rt1011_r0_config(struct snd_soc_component *component, unsigned int r0) +{ + struct rt1011_priv *rt1011 = + snd_soc_component_get_drvdata(component); + struct device *dev; + unsigned int r0_integer, r0_factor, format; + + if (!component->card->instantiated) + return 0; + + if (!r0) + return -EINVAL; + + dev = regmap_get_device(rt1011->regmap); + rt1011->r0_reg = r0; + + format = 2147483648U; /* 2^24 * 128 */ + r0_integer = format / rt1011->r0_reg / 128; + r0_factor = ((format / rt1011->r0_reg * 100) / 128) + - (r0_integer * 100); + dev_dbg(dev, "New r0 resistance about %d.%02d ohm, reg=0x%X\n", + r0_integer, r0_factor, rt1011->r0_reg); + + if (rt1011->r0_reg) + rt1011_r0_load(rt1011); + + return 0; +} +EXPORT_SYMBOL_GPL(rt1011_r0_config); + +int rt1011_temperature_config(struct snd_soc_component *component, + unsigned int temp) +{ + snd_soc_component_update_bits(component, + RT1011_STP_INITIAL_RESISTANCE_TEMP, 0x3ff, + (temp << 2)); + + return 0; +} +EXPORT_SYMBOL_GPL(rt1011_temperature_config); + static const struct snd_kcontrol_new rt1011_snd_controls[] = { /* I2S Data In Selection */ SOC_ENUM("DIN Source", rt1011_din_source_enum), diff --git a/sound/soc/codecs/rt1011.h b/sound/soc/codecs/rt1011.h index 2d65983f3d0f..b99df334be14 100644 --- a/sound/soc/codecs/rt1011.h +++ b/sound/soc/codecs/rt1011.h @@ -670,4 +670,9 @@ struct rt1011_priv { int recv_spk_mode; }; +int rt1011_r0_config(struct snd_soc_component *component, unsigned int r0); +int rt1011_temperature_config(struct snd_soc_component *component, + unsigned int temp); + + #endif /* end of _RT1011_H_ */ -- 2.23.0

4 9

[alsa-devel] [PATCH RESEND] ASoC: rt1011: set tx/rx slots from tx/rx_mask in TDM case
by Shuming [范書銘] 14 Oct '19

14 Oct '19

The TX/RX slot configuration use tx/rx_mask which requested by the machine driver. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1011.c | 97 +++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/rt1011.h | 23 ++++++++++ 2 files changed, 115 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index be1e276e3631..a36144de11cf 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1805,8 +1805,8 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, struct snd_soc_component *component = dai->component; struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); - unsigned int val = 0, tdm_en = 0; - int ret = 0; + unsigned int val = 0, tdm_en = 0, rx_slotnum, tx_slotnum; + int ret = 0, first_bit, last_bit; snd_soc_dapm_mutex_lock(dapm); if (rx_mask || tx_mask) @@ -1850,6 +1850,91 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, ret = -EINVAL; } + /* Rx slot configuration */ + rx_slotnum = hweight_long(rx_mask); + first_bit = find_next_bit((unsigned long *)&rx_mask, 32, 0); + if (rx_slotnum > 1 || rx_slotnum == 0) { + ret = -EINVAL; + dev_dbg(component->dev, "too many rx slots or zero slot\n"); + goto _set_tdm_err_; + } + if (rx_slotnum) { + if (first_bit == 0 || first_bit == 2 || + first_bit == 4 || first_bit == 6) { + snd_soc_component_update_bits(component, + RT1011_CROSS_BQ_SET_1, RT1011_MONO_LR_SEL_MASK, + RT1011_MONO_L_CHANNEL); + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_4, + RT1011_TDM_I2S_TX_L_DAC1_1_MASK | + RT1011_TDM_I2S_TX_R_DAC1_1_MASK, + (first_bit << RT1011_TDM_I2S_TX_L_DAC1_1_SFT) | + ((first_bit+1) + << RT1011_TDM_I2S_TX_R_DAC1_1_SFT)); + } else { + snd_soc_component_update_bits(component, + RT1011_CROSS_BQ_SET_1, RT1011_MONO_LR_SEL_MASK, + RT1011_MONO_R_CHANNEL); + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_4, + RT1011_TDM_I2S_TX_L_DAC1_1_MASK | + RT1011_TDM_I2S_TX_R_DAC1_1_MASK, + ((first_bit-1) + << RT1011_TDM_I2S_TX_L_DAC1_1_SFT) | + (first_bit << RT1011_TDM_I2S_TX_R_DAC1_1_SFT)); + } + } + + /* Tx slot configuration */ + tx_slotnum = hweight_long(tx_mask); + first_bit = find_next_bit((unsigned long *)&tx_mask, 32, 0); + last_bit = find_last_bit((unsigned long *)&tx_mask, 32); + if (tx_slotnum > 2 || (last_bit-first_bit) > 1) { + ret = -EINVAL; + dev_dbg(component->dev, "too many tx slots or tx slot location error\n"); + goto _set_tdm_err_; + } + + if (tx_slotnum == 1) { + snd_soc_component_update_bits(component, RT1011_TDM1_SET_2, + RT1011_TDM_I2S_DOCK_ADCDAT_LEN_1_MASK | + RT1011_TDM_ADCDAT1_DATA_LOCATION, first_bit); + if (first_bit == 1) + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC1_1_MASK, + RT1011_TDM_I2S_RX_ADC1_1_LL); + else if (first_bit == 3) + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC2_1_MASK, + RT1011_TDM_I2S_RX_ADC2_1_LL); + else if (first_bit == 5) + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC3_1_MASK, + RT1011_TDM_I2S_RX_ADC3_1_LL); + else if (first_bit == 7) + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC4_1_MASK, + RT1011_TDM_I2S_RX_ADC4_1_LL); + } else if (tx_slotnum == 2) { + if (first_bit == 0 || first_bit == 2 || + first_bit == 4 || first_bit == 6) { + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_2, + RT1011_TDM_I2S_DOCK_ADCDAT_LEN_1_MASK | + RT1011_TDM_ADCDAT1_DATA_LOCATION, + RT1011_TDM_I2S_DOCK_ADCDAT_2CH | first_bit); + } else { + ret = -EINVAL; + dev_dbg(component->dev, + "tx slot location should be paired and start from slot0/2/4/6\n"); + goto _set_tdm_err_; + } + } + snd_soc_component_update_bits(component, RT1011_TDM1_SET_1, RT1011_I2S_CH_TX_MASK | RT1011_I2S_CH_RX_MASK | RT1011_I2S_CH_TX_LEN_MASK | RT1011_I2S_CH_RX_LEN_MASK, val); @@ -1860,10 +1945,12 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, RT1011_TDM_I2S_DOCK_EN_1_MASK, tdm_en); snd_soc_component_update_bits(component, RT1011_TDM2_SET_2, RT1011_TDM_I2S_DOCK_EN_2_MASK, tdm_en); - snd_soc_component_update_bits(component, RT1011_TDM_TOTAL_SET, - RT1011_ADCDAT1_PIN_CONFIG | RT1011_ADCDAT2_PIN_CONFIG, - RT1011_ADCDAT1_OUTPUT | RT1011_ADCDAT2_OUTPUT); + if (tx_slotnum) + snd_soc_component_update_bits(component, RT1011_TDM_TOTAL_SET, + RT1011_ADCDAT1_PIN_CONFIG | RT1011_ADCDAT2_PIN_CONFIG, + RT1011_ADCDAT1_OUTPUT | RT1011_ADCDAT2_OUTPUT); +_set_tdm_err_: snd_soc_dapm_mutex_unlock(dapm); return ret; } diff --git a/sound/soc/codecs/rt1011.h b/sound/soc/codecs/rt1011.h index 2d65983f3d0f..3f7dea1191ad 100644 --- a/sound/soc/codecs/rt1011.h +++ b/sound/soc/codecs/rt1011.h @@ -460,6 +460,23 @@ #define RT1011_TDM_I2S_DOCK_EN_1_MASK (0x1 << 3) #define RT1011_TDM_I2S_DOCK_EN_1_SFT 3 #define RT1011_TDM_I2S_DOCK_EN_1 (0x1 << 3) +#define RT1011_TDM_ADCDAT1_DATA_LOCATION (0x7 << 0) + +/* TDM1 Setting-3 (0x0118) */ +#define RT1011_TDM_I2S_RX_ADC1_1_MASK (0x3 << 6) +#define RT1011_TDM_I2S_RX_ADC2_1_MASK (0x3 << 4) +#define RT1011_TDM_I2S_RX_ADC3_1_MASK (0x3 << 2) +#define RT1011_TDM_I2S_RX_ADC4_1_MASK (0x3 << 0) +#define RT1011_TDM_I2S_RX_ADC1_1_LL (0x2 << 6) +#define RT1011_TDM_I2S_RX_ADC2_1_LL (0x2 << 4) +#define RT1011_TDM_I2S_RX_ADC3_1_LL (0x2 << 2) +#define RT1011_TDM_I2S_RX_ADC4_1_LL (0x2 << 0) + +/* TDM1 Setting-4 (0x011a) */ +#define RT1011_TDM_I2S_TX_L_DAC1_1_MASK (0x7 << 12) +#define RT1011_TDM_I2S_TX_R_DAC1_1_MASK (0x7 << 8) +#define RT1011_TDM_I2S_TX_L_DAC1_1_SFT 12 +#define RT1011_TDM_I2S_TX_R_DAC1_1_SFT 8 /* TDM2 Setting-2 (0x0120) */ #define RT1011_TDM_I2S_DOCK_ADCDAT_LEN_2_MASK (0x7 << 13) @@ -585,6 +602,12 @@ #define RT1011_STP_T0_EN_BIT 6 #define RT1011_STP_T0_EN (0x1 << 6) +/* Cross Biquad Setting-1 (0x0702) */ +#define RT1011_MONO_LR_SEL_MASK (0x3 << 5) +#define RT1011_MONO_L_CHANNEL (0x0 << 5) +#define RT1011_MONO_R_CHANNEL (0x1 << 5) +#define RT1011_MONO_LR_MIX_CHANNEL (0x2 << 5) + /* ClassD Internal Setting-1 (0x1300) */ #define RT1011_DRIVER_READY_SPK (0x1 << 12) #define RT1011_DRIVER_READY_SPK_BIT 12 -- 2.23.0

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Re: [alsa-devel] [PATCH -next] ASoC: atmel: select SND_ATMEL_SOC_DMA for SND_ATMEL_SOC_SSC
by Alexandre Belloni 12 Oct '19

12 Oct '19

On 12/10/2019 10:42:30+0800, Mao Wenan wrote: > If SND_ATMEL_SOC_SSC_PDC=y and SND_ATMEL_SOC_SSC_DMA=m, > below errors can be found: > sound/soc/atmel/atmel_ssc_dai.o: In function > `atmel_ssc_set_audio': > atmel_ssc_dai.c:(.text+0x6fe): undefined reference to > `atmel_pcm_dma_platform_register' > make: *** [vmlinux] Error 1 > > After commit 18291410557f ("ASoC: atmel: enable > SOC_SSC_PDC and SOC_SSC_DMA in Kconfig"), SND_ATMEL_SOC_DMA > and SND_ATMEL_SOC_SSC are selected by SND_ATMEL_SOC_SSC_DMA, > SND_ATMEL_SOC_SSC is also selected by SND_ATMEL_SOC_SSC_PDC, > the results are SND_ATMEL_SOC_DMA=m but SND_ATMEL_SOC_SSC=y, > so the errors happen. > > This patch make SND_ATMEL_SOC_SSC select SND_ATMEL_SOC_DMA. > > Fixes: 18291410557f ("ASoC: atmel: enable SOC_SSC_PDC and SOC_SSC_DMA in Kconfig") > Signed-off-by: Mao Wenan <maowenan(a)huawei.com> > --- > sound/soc/atmel/Kconfig | 1 + > 1 file changed, 1 insertion(+) > > diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig > index f118c22..2938f6b 100644 > --- a/sound/soc/atmel/Kconfig > +++ b/sound/soc/atmel/Kconfig > @@ -19,6 +19,7 @@ config SND_ATMEL_SOC_DMA > > config SND_ATMEL_SOC_SSC > tristate > + select SND_ATMEL_SOC_DMA > This is not the solution because this doesn't allow to compile out DMA and use only PDC. I think Arnd already submitted a proper patch. > config SND_ATMEL_SOC_SSC_PDC > tristate "SoC PCM DAI support for AT91 SSC controller using PDC" > -- > 2.7.4 > -- Alexandre Belloni, Bootlin Embedded Linux and Kernel engineering https://bootlin.com

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