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

[alsa-devel] [PATCH v4] ASoC: tlv320aic3x: Add support for high power analog output
by Saravanan Sekar 17 May '19

17 May '19

Add support to output level control for the analog high power output drivers HPOUT and HPCOM. Signed-off-by: Saravanan Sekar <sravanhome(a)gmail.com> --- Notes: Notes: Changes in V4: -Added separate mono playback volume control -grouped volume controls with corresponding switch Changes in V3: -Fixed compilation error Changes in V2: - Removed power control as it is handled by DAPM - Added level control for left channel sound/soc/codecs/tlv320aic3x.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 516d17cb2182..599e4ed3850b 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -324,6 +324,9 @@ static DECLARE_TLV_DB_SCALE(adc_tlv, 0, 50, 0); */ static DECLARE_TLV_DB_SCALE(output_stage_tlv, -5900, 50, 1); +/* Output volumes. From 0 to 9 dB in 1 dB steps */ +static const DECLARE_TLV_DB_SCALE(out_tlv, 0, 100, 0); + static const struct snd_kcontrol_new aic3x_snd_controls[] = { /* Output */ SOC_DOUBLE_R_TLV("PCM Playback Volume", @@ -386,11 +389,17 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { DACL1_2_HPLCOM_VOL, DACR1_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv), - /* Output pin mute controls */ + /* Output pin controls */ + SOC_DOUBLE_R_TLV("Line Playback Volume", LLOPM_CTRL, RLOPM_CTRL, 4, + 9, 0, out_tlv), SOC_DOUBLE_R("Line Playback Switch", LLOPM_CTRL, RLOPM_CTRL, 3, 0x01, 0), + SOC_DOUBLE_R_TLV("HP Playback Volume", HPLOUT_CTRL, HPROUT_CTRL, 4, + 9, 0, out_tlv), SOC_DOUBLE_R("HP Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3, 0x01, 0), + SOC_DOUBLE_R_TLV("HPCOM Playback Volume", HPLCOM_CTRL, HPRCOM_CTRL, + 4, 9, 0, out_tlv), SOC_DOUBLE_R("HPCOM Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3, 0x01, 0), @@ -472,6 +481,9 @@ static const struct snd_kcontrol_new aic3x_mono_controls[] = { 0, 118, 1, output_stage_tlv), SOC_SINGLE("Mono Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), + SOC_SINGLE_TLV("Mono Playback Volume", MONOLOPM_CTRL, 4, 9, 0, + out_tlv), + }; /* -- 2.17.1

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[alsa-devel] [PATCH V2] ASoC: cs42xx8: Add regcache mask dirty
by S.j. Wang 17 May '19

17 May '19

Add regcache_mark_dirty before regcache_sync for power of codec may be lost at suspend, then all the register need to be reconfigured. Fixes: 0c516b4ff85c ("ASoC: cs42xx8: Add codec driver support for CS42448/CS42888") Cc: <stable(a)vger.kernel.org> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Changs in V2 - Don't split Fixes tag. sound/soc/codecs/cs42xx8.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index ebb9e0cf8364..28a4ac36c4f8 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -558,6 +558,7 @@ static int cs42xx8_runtime_resume(struct device *dev) msleep(5); regcache_cache_only(cs42xx8->regmap, false); + regcache_mark_dirty(cs42xx8->regmap); ret = regcache_sync(cs42xx8->regmap); if (ret) { -- 2.21.0

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Re: [alsa-devel] [PATCH] ASoC: cs42xx8: Add regcache mask dirty
by S.j. Wang 17 May '19

17 May '19

Hi > Add regcache_mark_dirty before regcache_sync for power of codec may be > lost at suspend, then all the register need to be reconfigured. > > Fixes: 0c516b4ff85c ("ASoC: cs42xx8: Add codec driver support for > CS42448/CS42888") The Fixes tag is split, will send v2. Best regards Wang shengjiu

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[alsa-devel] [PATCH] ASoC: sun4i-codec: fix first delay on Speaker
by Georgii Staroselskii 17 May '19

17 May '19

This commit fixes the same issue as bf14da7 but for another codepath. The issue can be triggered by having Speaker in codec audio routing. &codec { allwinner,pa-gpios = <&r_pio 0 11 GPIO_ACTIVE_HIGH>; /* PL11 */ allwinner,audio-routing = "Speaker", "LINEOUT"; status = "okay"; } Signed-off-by: Georgii Staroselskii <georgii.staroselskii(a)emlid.com> --- sound/soc/sunxi/sun4i-codec.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 15d08e3..e0099519 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1329,6 +1329,15 @@ static int sun4i_codec_spk_event(struct snd_soc_dapm_widget *w, gpiod_set_value_cansleep(scodec->gpio_pa, !!SND_SOC_DAPM_EVENT_ON(event)); + if (SND_SOC_DAPM_EVENT_ON(event)) { + /* + * Need a delay to have the amplifier up. 700ms seems the best + * compromise between the time to let the amplifier up and the + * time not to feel this delay while playing a sound. + */ + msleep(700); + } + return 0; } -- 2.7.4

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[alsa-devel] [PATCH v3 1/2] dt-bindings: sound: Convert Allwinner SPDIF binding to YAML
by Maxime Ripard 16 May '19

16 May '19

The Allwinner SoCs feature an SPDIF controller across multiple SoC generations. However, earlier generations were a bit simpler than the subsequent ones, and for example would always have RX and TX capabilities, and no reset lines. In order to express this, let's create two YAML schemas instead of the free form text we had before. Signed-off-by: Maxime Ripard <maxime.ripard(a)bootlin.com> --- Changes from v2: - Add comments Changes from v1: - Merged the two schemas together and used the draft-7 conditionals --- Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml | 101 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++- Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt | 42 +------------------------------ 2 files changed, 101 insertions(+), 42 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml delete mode 100644 Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml new file mode 100644 index 000000000000..5d72d48e923e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml @@ -0,0 +1,101 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun4i-a10-spdif.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner A10 S/PDIF Controller Device Tree Bindings + +maintainers: + - Chen-Yu Tsai <wens(a)csie.org> + - Liam Girdwood <lgirdwood(a)gmail.com> + - Mark Brown <broonie(a)kernel.org> + - Maxime Ripard <maxime.ripard(a)bootlin.com> + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + oneOf: + - const: allwinner,sun4i-a10-spdif + - const: allwinner,sun6i-a31-spdif + - const: allwinner,sun8i-h3-spdif + - items: + - const: allwinner,sun8i-a83t-spdif + - const: allwinner,sun8i-h3-spdif + - items: + - const: allwinner,sun50i-a64-spdif + - const: allwinner,sun8i-h3-spdif + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: apb + - const: spdif + + dmas: + items: + - description: RX DMA Channel + - description: TX DMA Channel + + dma-names: + items: + - const: rx + - const: tx + + # Even though it only applies to subschemas under the conditionals, + # not listing them here will trigger a warning because of the + # additionalsProperties set to false. + resets: + maxItems: 1 + +allOf: + - if: + properties: + compatible: + contains: + enum: + - allwinner,sun6i-a31-spdif + - allwinner,sun8i-h3-spdif + + then: + required: + - resets + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + spdif: spdif@1c21000 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun4i-a10-spdif"; + reg = <0x01c21000 0x40>; + interrupts = <13>; + clocks = <&apb0_gates 1>, <&spdif_clk>; + clock-names = "apb", "spdif"; + dmas = <&dma 0 2>, <&dma 0 2>; + dma-names = "rx", "tx"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt b/Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt deleted file mode 100644 index 0c64a209c2e9..000000000000 --- a/Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt +++ /dev/null @@ -1,42 +0,0 @@ -Allwinner Sony/Philips Digital Interface Format (S/PDIF) Controller - -The Allwinner S/PDIF audio block is a transceiver that allows the -processor to receive and transmit digital audio via an coaxial cable or -a fibre cable. -For now only playback is supported. - -Required properties: - - - compatible : should be one of the following: - - "allwinner,sun4i-a10-spdif": for the Allwinner A10 SoC - - "allwinner,sun6i-a31-spdif": for the Allwinner A31 SoC - - "allwinner,sun8i-h3-spdif": for the Allwinner H3 SoC - - - reg : Offset and length of the register set for the device. - - - interrupts : Contains the spdif interrupt. - - - dmas : Generic dma devicetree binding as described in - Documentation/devicetree/bindings/dma/dma.txt. - - - dma-names : Two dmas have to be defined, "tx" and "rx". - - - clocks : Contains an entry for each entry in clock-names. - - - clock-names : Includes the following entries: - "apb" clock for the spdif bus. - "spdif" clock for spdif controller. - - - resets : reset specifier for the ahb reset (A31 and newer only) - -Example: - -spdif: spdif@1c21000 { - compatible = "allwinner,sun4i-a10-spdif"; - reg = <0x01c21000 0x40>; - interrupts = <13>; - clocks = <&apb0_gates 1>, <&spdif_clk>; - clock-names = "apb", "spdif"; - dmas = <&dma 0 2>, <&dma 0 2>; - dma-names = "rx", "tx"; -}; base-commit: 00c3bc00f0cadbb48f2414370733892676511e79 -- git-series 0.9.1

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[alsa-devel] [PATCH] ASoC: AK4458: add regulator for ak4458
by Viorel Suman 16 May '19

16 May '19

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> Add regulator for ak4458. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> --- sound/soc/codecs/ak4458.c | 27 ++++++++++++++++++++++++++- 1 file changed, 26 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index 7156215..06dcf13 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -12,6 +12,7 @@ #include <linux/of_device.h> #include <linux/of_gpio.h> #include <linux/pm_runtime.h> +#include <linux/regulator/consumer.h> #include <linux/slab.h> #include <sound/initval.h> #include <sound/pcm_params.h> @@ -21,6 +22,12 @@ #include "ak4458.h" +#define AK4458_NUM_SUPPLIES 2 +static const char *ak4458_supply_names[AK4458_NUM_SUPPLIES] = { + "DVDD", + "AVDD", +}; + struct ak4458_drvdata { struct snd_soc_dai_driver *dai_drv; const struct snd_soc_component_driver *comp_drv; @@ -37,6 +44,7 @@ struct ak4458_priv { int fmt; int slots; int slot_width; + struct regulator_bulk_data supplies[AK4458_NUM_SUPPLIES]; }; static const struct reg_default ak4458_reg_defaults[] = { @@ -666,7 +674,7 @@ static int ak4458_i2c_probe(struct i2c_client *i2c) { struct ak4458_priv *ak4458; const struct ak4458_drvdata *drvdata; - int ret; + int ret, i; ak4458 = devm_kzalloc(&i2c->dev, sizeof(*ak4458), GFP_KERNEL); if (!ak4458) @@ -691,6 +699,23 @@ static int ak4458_i2c_probe(struct i2c_client *i2c) if (IS_ERR(ak4458->mute_gpiod)) return PTR_ERR(ak4458->mute_gpiod); + for (i = 0; i < ARRAY_SIZE(ak4458->supplies); i++) + ak4458->supplies[i].supply = ak4458_supply_names[i]; + + ret = devm_regulator_bulk_get(ak4458->dev, ARRAY_SIZE(ak4458->supplies), + ak4458->supplies); + if (ret != 0) { + dev_err(ak4458->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(ak4458->supplies), + ak4458->supplies); + if (ret != 0) { + dev_err(ak4458->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + ret = devm_snd_soc_register_component(ak4458->dev, drvdata->comp_drv, drvdata->dai_drv, 1); if (ret < 0) { -- 2.7.4

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Re: [alsa-devel] [PATCH] ASoC: rt1308: Add RT1308 amplifier driver
by Mark Brown 16 May '19

16 May '19

On Wed, Apr 17, 2019 at 11:40:28AM +0000, Derek [方德義] wrote: Please fix your mail client to word wrap within paragraphs at something substantially less than 80 columns. Doing this makes your messages much easier to read and reply to. > > The ones here with comments look an awful lot like they're system specific config > > which should be left up to either platform data (or DT/ACPI configuration) or > > done at runtime (like gains). Some of the others like the pads configuration > > might also fit into that. > We don't open the DA gain setting for user at runtime, so have to set > a default level in the init list. That's not the way Linux drivers generally work... is there some technical reason for that or is it just some kind of policy decision? > > > +static void rt1308_reset(struct regmap *regmap) { > > > + regmap_write(regmap, RT1308_RESET, 0); } > > > > If we do need the init list shouldn't we also do a reninit every time we do a reset? > The init list is only set at driver probe, not every time after resetting. This is what I'm pointing out; if you don't restore these values then why set them at all?

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[alsa-devel] [asoc:for-5.3 34/40] sound/soc/mediatek/common/mtk-afe-fe-dai.c:268:2: error: too few arguments to function 'mtk_regmap_update_bits'
by kbuild test robot 16 May '19

16 May '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.3 head: c8609f3870f7078fc7922eb816ed4908a9bd44f3 commit: 1628fc3f47717fc3c8d9cd5fd6c7bc7d3796927d [34/40] ASoC: Mediatek: add memory interface data align config: arm-allmodconfig (attached as .config) compiler: arm-linux-gnueabi-gcc (Debian 7.2.0-11) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 1628fc3f47717fc3c8d9cd5fd6c7bc7d3796927d # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=arm If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> All errors (new ones prefixed by >>): sound/soc/mediatek/common/mtk-afe-fe-dai.c: In function 'mtk_afe_fe_prepare': >> sound/soc/mediatek/common/mtk-afe-fe-dai.c:268:2: error: too few arguments to function 'mtk_regmap_update_bits' mtk_regmap_update_bits(afe->regmap, memif->data->hd_align_reg, ^~~~~~~~~~~~~~~~~~~~~~ sound/soc/mediatek/common/mtk-afe-fe-dai.c:19:12: note: declared here static int mtk_regmap_update_bits(struct regmap *map, int reg, ^~~~~~~~~~~~~~~~~~~~~~ vim +/mtk_regmap_update_bits +268 sound/soc/mediatek/common/mtk-afe-fe-dai.c 236 237 int mtk_afe_fe_prepare(struct snd_pcm_substream *substream, 238 struct snd_soc_dai *dai) 239 { 240 struct snd_soc_pcm_runtime *rtd = substream->private_data; 241 struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); 242 struct mtk_base_afe_memif *memif = &afe->memif[rtd->cpu_dai->id]; 243 int hd_audio = 0; 244 int hd_align = 1; 245 246 /* set hd mode */ 247 switch (substream->runtime->format) { 248 case SNDRV_PCM_FORMAT_S16_LE: 249 hd_audio = 0; 250 break; 251 case SNDRV_PCM_FORMAT_S32_LE: 252 hd_audio = 1; 253 hd_align = 1; 254 break; 255 case SNDRV_PCM_FORMAT_S24_LE: 256 hd_audio = 1; 257 hd_align = 0; 258 break; 259 default: 260 dev_err(afe->dev, "%s() error: unsupported format %d\n", 261 __func__, substream->runtime->format); 262 break; 263 } 264 265 mtk_regmap_update_bits(afe->regmap, memif->data->hd_reg, 266 1, hd_audio, memif->data->hd_shift); 267 > 268 mtk_regmap_update_bits(afe->regmap, memif->data->hd_align_reg, 269 memif->data->hd_align_mshift, 270 hd_align ? memif->data->hd_align_mshift : 0); 271 272 return 0; 273 } 274 EXPORT_SYMBOL_GPL(mtk_afe_fe_prepare); 275 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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Re: [alsa-devel] [PATCH V4] ASoC: cs42xx8: add reset-gpios in binding document
by S.j. Wang 16 May '19

16 May '19

Hi > On Thu, May 16, 2019 at 8:36 AM S.j. Wang <shengjiu.wang(a)nxp.com> > wrote: > > > cs42888: codec@48 { > > @@ -25,4 +30,5 @@ cs42888: codec@48 { > > VD-supply = <&reg_audio>; > > VLS-supply = <&reg_audio>; > > VLC-supply = <&reg_audio>; > > + reset-gpios = <&pca9557_b 1 1>; > > reset-gpios = <&pca9557_b 1 GPIO_ACTIVE_LOW>; please Ok, sent v5. Best regards Wang shengjiu

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[alsa-devel] [PATCH V4] ASoC: cs42xx8: add reset-gpios in binding document
by S.j. Wang 16 May '19

16 May '19

Add reset-gpios property, which is optional. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Changes in V4 - use gpios instead of gpio Changes in RESEND v3 - send updated binding document only Changes in v3 - update binding document. Changes in v2 - use devm_gpiod_get_optional instead of of_get_named_gpio Documentation/devicetree/bindings/sound/cs42xx8.txt | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/cs42xx8.txt b/Documentation/devicetree/bindings/sound/cs42xx8.txt index 8619a156d038..0138ef559cc4 100644 --- a/Documentation/devicetree/bindings/sound/cs42xx8.txt +++ b/Documentation/devicetree/bindings/sound/cs42xx8.txt @@ -14,6 +14,11 @@ Required properties: - VA-supply, VD-supply, VLS-supply, VLC-supply: power supplies for the device, as covered in Documentation/devicetree/bindings/regulator/regulator.txt +Optional properties: + + - reset-gpios : a GPIO spec to define which pin is connected to the chip's + !RESET pin + Example: cs42888: codec@48 { @@ -25,4 +30,5 @@ cs42888: codec@48 { VD-supply = <&reg_audio>; VLS-supply = <&reg_audio>; VLC-supply = <&reg_audio>; + reset-gpios = <&pca9557_b 1 1>; }; -- 2.21.0

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