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

[alsa-devel] [PATCH] sound: open brace should be on the previous line
by parna.naveenkumar＠gmail.com 17 May '19

17 May '19

From: Naveen Kumar Parna <parna.naveenkumar(a)gmail.com> Resolved open brace { should be on the previous line checkpatch.pl error. While addressing this error, also corrected the affected code for below mentioned checkpatch errors. ERROR: spaces required around that '<' (ctx:VxV) ERROR: spaces required around that '==' (ctx:VxV) ERROR: space required before the open parenthesis '(' Signed-off-by: Naveen Kumar Parna <parna.naveenkumar(a)gmail.com> --- sound/sound_core.c | 21 +++++++-------------- 1 file changed, 7 insertions(+), 14 deletions(-) diff --git a/sound/sound_core.c b/sound/sound_core.c index 40ad000c2e3c..ce794a2afc6b 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -111,8 +111,7 @@ module_exit(cleanup_soundcore); #define SOUND_STEP 16 -struct sound_unit -{ +struct sound_unit { int unit_minor; const struct file_operations *unit_fops; struct sound_unit *next; @@ -151,8 +150,7 @@ module_param(preclaim_oss, int, 0444); static int soundcore_open(struct inode *, struct file *); -static const struct file_operations soundcore_fops = -{ +static const struct file_operations soundcore_fops = { /* We must have an owner or the module locking fails */ .owner = THIS_MODULE, .open = soundcore_open, @@ -173,8 +171,7 @@ static int __sound_insert_unit(struct sound_unit * s, struct sound_unit **list, while (*list && (*list)->unit_minor<n) list=&((*list)->next); - while(n<top) - { + while (n < top) { /* Found a hole ? */ if(*list==NULL || (*list)->unit_minor>n) break; @@ -219,11 +216,9 @@ static int __sound_insert_unit(struct sound_unit * s, struct sound_unit **list, static struct sound_unit *__sound_remove_unit(struct sound_unit **list, int unit) { - while(*list) - { + while (*list) { struct sound_unit *p=*list; - if(p->unit_minor==unit) - { + if (p->unit_minor == unit) { *list=p->next; return p; } @@ -528,8 +523,7 @@ static struct sound_unit *__look_for_unit(int chain, int unit) struct sound_unit *s; s=chains[chain]; - while(s && s->unit_minor <= unit) - { + while (s && s->unit_minor <= unit) { if(s->unit_minor==unit) return s; s=s->next; @@ -545,8 +539,7 @@ static int soundcore_open(struct inode *inode, struct file *file) const struct file_operations *new_fops = NULL; chain=unit&0x0F; - if(chain==4 || chain==5) /* dsp/audio/dsp16 */ - { + if (chain == 4 || chain == 5) { /* dsp/audio/dsp16 */ unit&=0xF0; unit|=3; chain=3; -- 2.17.1

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[alsa-devel] [PATCH V2] ASoC: cs42xx8: Add reset gpio handling
by S.j. Wang 17 May '19

17 May '19

Handle the reset GPIO and reset the device every time we start it. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Changes in v2 - use devm_gpiod_get_optional instead of of_get_named_gpio sound/soc/codecs/cs42xx8.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index 28a4ac36c4f8..b377cddaf2e6 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -14,6 +14,7 @@ #include <linux/delay.h> #include <linux/module.h> #include <linux/of_device.h> +#include <linux/of_gpio.h> #include <linux/pm_runtime.h> #include <linux/regulator/consumer.h> #include <sound/pcm_params.h> @@ -45,6 +46,7 @@ struct cs42xx8_priv { bool slave_mode; unsigned long sysclk; u32 tx_channels; + struct gpio_desc *gpiod_reset; }; /* -127.5dB to 0dB with step of 0.5dB */ @@ -467,6 +469,13 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap) return -EINVAL; } + cs42xx8->gpiod_reset = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_HIGH); + if (IS_ERR(cs42xx8->gpiod_reset)) + return PTR_ERR(cs42xx8->gpiod_reset); + + gpiod_set_value_cansleep(cs42xx8->gpiod_reset, 0); + cs42xx8->clk = devm_clk_get(dev, "mclk"); if (IS_ERR(cs42xx8->clk)) { dev_err(dev, "failed to get the clock: %ld\n", @@ -547,6 +556,8 @@ static int cs42xx8_runtime_resume(struct device *dev) return ret; } + gpiod_set_value_cansleep(cs42xx8->gpiod_reset, 0); + ret = regulator_bulk_enable(ARRAY_SIZE(cs42xx8->supplies), cs42xx8->supplies); if (ret) { @@ -586,6 +597,8 @@ static int cs42xx8_runtime_suspend(struct device *dev) regulator_bulk_disable(ARRAY_SIZE(cs42xx8->supplies), cs42xx8->supplies); + gpiod_set_value_cansleep(cs42xx8->gpiod_reset, 1); + clk_disable_unprepare(cs42xx8->clk); return 0; -- 2.21.0

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[alsa-devel] [PATCH] 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> --- 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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[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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