September 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [asoc:topic/sunxi 2/2] sound/soc/sunxi/sun4i-codec.c:708:3-8: No need to set .owner here. The core will do it.
by kbuild test robot 17 Sep '15

17 Sep '15

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/sunxi head: 45fb6b6f2aa3f6b22b81078db0dba4b26c9d0bdb commit: 45fb6b6f2aa3f6b22b81078db0dba4b26c9d0bdb [2/2] ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs coccinelle warnings: (new ones prefixed by >>) >> sound/soc/sunxi/sun4i-codec.c:708:3-8: No need to set .owner here. The core will do it. Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH v2] ASoC: add ak4613 support
by Kuninori Morimoto 17 Sep '15

17 Sep '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - fix for checkpatch - adds ak4613 in SND_SOC_ALL_CODECS - ak4613_write() -> snd_soc_write() - ak4613_read() was removed - spin lock -> mutex - ak4613_resume() returns regcache_sync()'s result Documentation/devicetree/bindings/sound/ak4613.txt | 17 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4613.c | 469 +++++++++++++++++++++ 4 files changed, 493 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ak4613.txt create mode 100644 sound/soc/codecs/ak4613.c diff --git a/Documentation/devicetree/bindings/sound/ak4613.txt b/Documentation/devicetree/bindings/sound/ak4613.txt new file mode 100644 index 0000000..15a9195 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4613.txt @@ -0,0 +1,17 @@ +AK4613 I2C transmitter + +This device supports I2C mode only. + +Required properties: + +- compatible : "asahi-kasei,ak4613" +- reg : The chip select number on the I2C bus + +Example: + +&i2c { + ak4613: ak4613@0x10 { + compatible = "asahi-kasei,ak4613"; + reg = <0x10>; + }; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0c9733e..a92e4d4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -36,6 +36,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C select SND_SOC_AK4554 + select SND_SOC_AK4613 if I2C select SND_SOC_AK4641 if I2C select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C @@ -319,6 +320,10 @@ config SND_SOC_AK4535 config SND_SOC_AK4554 tristate "AKM AK4554 CODEC" +config SND_SOC_AK4613 + tristate "AKM AK4613 CODEC" + depends on I2C + config SND_SOC_AK4641 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 4a32077..5b6c8af 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -26,6 +26,7 @@ snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4554-objs := ak4554.o +snd-soc-ak4613-objs := ak4613.o snd-soc-ak4641-objs := ak4641.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o @@ -216,6 +217,7 @@ obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4554) += snd-soc-ak4554.o +obj-$(CONFIG_SND_SOC_AK4613) += snd-soc-ak4613.o obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c new file mode 100644 index 0000000..fd96a8f --- /dev/null +++ b/sound/soc/codecs/ak4613.c @@ -0,0 +1,469 @@ +/* + * ak4613.c -- Asahi Kasei ALSA Soc Audio driver + * + * Copyright (C) 2015 Renesas Electronics Corporation + * Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + * + * Based on ak4642.c by Kuninori Morimoto + * Based on wm8731.c by Richard Purdie + * Based on ak4535.c by Richard Purdie + * Based on wm8753.c by Liam Girdwood + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/clk.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/of_device.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> + +#define PW_MGMT1 0x00 /* Power Management 1 */ +#define PW_MGMT2 0x01 /* Power Management 2 */ +#define PW_MGMT3 0x02 /* Power Management 3 */ +#define CTRL1 0x03 /* Control 1 */ +#define CTRL2 0x04 /* Control 2 */ +#define DEMP1 0x05 /* De-emphasis1 */ +#define DEMP2 0x06 /* De-emphasis2 */ +#define OFD 0x07 /* Overflow Detect */ +#define ZRD 0x08 /* Zero Detect */ +#define ICTRL 0x09 /* Input Control */ +#define OCTRL 0x0a /* Output Control */ +#define LOUT1 0x0b /* LOUT1 Volume Control */ +#define ROUT1 0x0c /* ROUT1 Volume Control */ +#define LOUT2 0x0d /* LOUT2 Volume Control */ +#define ROUT2 0x0e /* ROUT2 Volume Control */ +#define LOUT3 0x0f /* LOUT3 Volume Control */ +#define ROUT3 0x10 /* ROUT3 Volume Control */ +#define LOUT4 0x11 /* LOUT4 Volume Control */ +#define ROUT4 0x12 /* ROUT4 Volume Control */ +#define LOUT5 0x13 /* LOUT5 Volume Control */ +#define ROUT5 0x14 /* ROUT5 Volume Control */ +#define LOUT6 0x15 /* LOUT6 Volume Control */ +#define ROUT6 0x16 /* ROUT6 Volume Control */ + +/* PW_MGMT1 */ +#define RSTN BIT(0) +#define PMDAC BIT(1) +#define PMADC BIT(2) +#define PMVR BIT(3) + +/* PW_MGMT2 */ +#define PMAD_ALL 0x7 + +/* PW_MGMT3 */ +#define PMDA_ALL 0x3f + +/* CTRL1 */ +#define DIF0 BIT(3) +#define DIF1 BIT(4) +#define DIF2 BIT(5) +#define TDM0 BIT(6) +#define TDM1 BIT(7) +#define NO_FMT (0xff) +#define FMT_MASK (0xf8) + +/* CTRL2 */ +#define DFS_NORMAL_SPEED (0 << 2) +#define DFS_DOUBLE_SPEED (1 << 2) +#define DFS_QUAD_SPEED (2 << 2) + +struct ak4613_priv { + struct mutex lock; + + unsigned int fmt; + u8 fmt_ctrl; + int cnt; +}; + +struct ak4613_formats { + unsigned int width; + unsigned int fmt; +}; + +struct ak4613_interface { + struct ak4613_formats capture; + struct ak4613_formats playback; +}; + +static const struct reg_default ak4613_reg[] = { + { 0x0, 0x0f }, { 0x1, 0x07 }, { 0x2, 0x3f }, { 0x3, 0x20 }, + { 0x4, 0x20 }, { 0x5, 0x55 }, { 0x6, 0x05 }, { 0x7, 0x07 }, + { 0x8, 0x0f }, { 0x9, 0x07 }, { 0xa, 0x3f }, { 0xb, 0x00 }, + { 0xc, 0x00 }, { 0xd, 0x00 }, { 0xe, 0x00 }, { 0xf, 0x00 }, + { 0x10, 0x00 }, { 0x11, 0x00 }, { 0x12, 0x00 }, { 0x13, 0x00 }, + { 0x14, 0x00 }, { 0x15, 0x00 }, { 0x16, 0x00 }, +}; + +#define AUDIO_IFACE_IDX_TO_VAL(i) (i << 3) +#define AUDIO_IFACE(b, fmt) { b, SND_SOC_DAIFMT_##fmt } +static const struct ak4613_interface ak4613_iface[] = { + /* capture */ /* playback */ + [0] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(16, RIGHT_J) }, + [1] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(20, RIGHT_J) }, + [2] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, RIGHT_J) }, + [3] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, LEFT_J) }, + [4] = { AUDIO_IFACE(24, I2S), AUDIO_IFACE(24, I2S) }, +}; + +static const struct regmap_config ak4613_regmap_cfg = { + .reg_bits = 8, + .val_bits = 8, + .max_register = 0x16, + .reg_defaults = ak4613_reg, + .num_reg_defaults = ARRAY_SIZE(ak4613_reg), +}; + +static const struct of_device_id ak4613_of_match[] = { + { .compatible = "asahi-kasei,ak4613", .data = &ak4613_regmap_cfg }, + {}, +}; +MODULE_DEVICE_TABLE(of, ak4613_of_match); + +static const struct i2c_device_id ak4613_i2c_id[] = { + { "ak4613", (kernel_ulong_t)&ak4613_regmap_cfg }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ak4613_i2c_id); + +static const struct snd_soc_dapm_widget ak4613_dapm_widgets[] = { + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("LOUT1"), + SND_SOC_DAPM_OUTPUT("LOUT2"), + SND_SOC_DAPM_OUTPUT("LOUT3"), + SND_SOC_DAPM_OUTPUT("LOUT4"), + SND_SOC_DAPM_OUTPUT("LOUT5"), + SND_SOC_DAPM_OUTPUT("LOUT6"), + + SND_SOC_DAPM_OUTPUT("ROUT1"), + SND_SOC_DAPM_OUTPUT("ROUT2"), + SND_SOC_DAPM_OUTPUT("ROUT3"), + SND_SOC_DAPM_OUTPUT("ROUT4"), + SND_SOC_DAPM_OUTPUT("ROUT5"), + SND_SOC_DAPM_OUTPUT("ROUT6"), + + /* Inputs */ + SND_SOC_DAPM_INPUT("LIN1"), + SND_SOC_DAPM_INPUT("LIN2"), + + SND_SOC_DAPM_INPUT("RIN1"), + SND_SOC_DAPM_INPUT("RIN2"), + + /* DAC */ + SND_SOC_DAPM_DAC("DAC1", NULL, PW_MGMT3, 0, 0), + SND_SOC_DAPM_DAC("DAC2", NULL, PW_MGMT3, 1, 0), + SND_SOC_DAPM_DAC("DAC3", NULL, PW_MGMT3, 2, 0), + SND_SOC_DAPM_DAC("DAC4", NULL, PW_MGMT3, 3, 0), + SND_SOC_DAPM_DAC("DAC5", NULL, PW_MGMT3, 4, 0), + SND_SOC_DAPM_DAC("DAC6", NULL, PW_MGMT3, 5, 0), + + /* ADC */ + SND_SOC_DAPM_ADC("ADC1", NULL, PW_MGMT2, 0, 0), + SND_SOC_DAPM_ADC("ADC2", NULL, PW_MGMT2, 1, 0), +}; + +static const struct snd_soc_dapm_route ak4613_intercon[] = { + {"LOUT1", NULL, "DAC1"}, + {"LOUT2", NULL, "DAC2"}, + {"LOUT3", NULL, "DAC3"}, + {"LOUT4", NULL, "DAC4"}, + {"LOUT5", NULL, "DAC5"}, + {"LOUT6", NULL, "DAC6"}, + + {"ROUT1", NULL, "DAC1"}, + {"ROUT2", NULL, "DAC2"}, + {"ROUT3", NULL, "DAC3"}, + {"ROUT4", NULL, "DAC4"}, + {"ROUT5", NULL, "DAC5"}, + {"ROUT6", NULL, "DAC6"}, + + {"DAC1", NULL, "Playback"}, + {"DAC2", NULL, "Playback"}, + {"DAC3", NULL, "Playback"}, + {"DAC4", NULL, "Playback"}, + {"DAC5", NULL, "Playback"}, + {"DAC6", NULL, "Playback"}, + + {"Capture", NULL, "ADC1"}, + {"Capture", NULL, "ADC2"}, + + {"ADC1", NULL, "LIN1"}, + {"ADC2", NULL, "LIN2"}, + + {"ADC1", NULL, "RIN1"}, + {"ADC2", NULL, "RIN2"}, +}; + +static void ak4613_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + struct device *dev = codec->dev; + + mutex_lock(&priv->lock); + priv->cnt--; + if (priv->cnt < 0) { + dev_err(dev, "unexpected counter error\n"); + priv->cnt = 0; + } + if (!priv->cnt) + priv->fmt_ctrl = NO_FMT; + mutex_unlock(&priv->lock); +} + +static int ak4613_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + + fmt &= SND_SOC_DAIFMT_FORMAT_MASK; + + switch (fmt) { + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_I2S: + priv->fmt = fmt; + + break; + default: + return -EINVAL; + } + + return 0; +} + +static int ak4613_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + const struct ak4613_formats *fmts; + struct device *dev = codec->dev; + unsigned int width = params_width(params); + unsigned int fmt = priv->fmt; + unsigned int rate; + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int i, ret; + u8 fmt_ctrl, ctrl2; + + rate = params_rate(params); + switch (rate) { + case 32000: + case 44100: + case 48000: + ctrl2 = DFS_NORMAL_SPEED; + break; + case 88200: + case 96000: + ctrl2 = DFS_DOUBLE_SPEED; + break; + case 176400: + case 192000: + ctrl2 = DFS_QUAD_SPEED; + break; + default: + return -EINVAL; + } + + /* + * FIXME + * + * It doesn't support TDM at this point + */ + fmt_ctrl = NO_FMT; + for (i = 0; i < ARRAY_SIZE(ak4613_iface); i++) { + fmts = (is_play) ? &ak4613_iface[i].playback : + &ak4613_iface[i].capture; + + if (fmts->fmt != fmt) + continue; + + if (fmt == SND_SOC_DAIFMT_RIGHT_J) { + if (fmts->width != width) + continue; + } else { + if (fmts->width < width) + continue; + } + + fmt_ctrl = AUDIO_IFACE_IDX_TO_VAL(i); + break; + } + + ret = -EINVAL; + if (fmt_ctrl == NO_FMT) + goto hw_params_end; + + mutex_lock(&priv->lock); + if ((priv->fmt_ctrl == NO_FMT) || + (priv->fmt_ctrl == fmt_ctrl)) { + priv->fmt_ctrl = fmt_ctrl; + priv->cnt++; + ret = 0; + } + mutex_unlock(&priv->lock); + + if (ret < 0) + goto hw_params_end; + + snd_soc_update_bits(codec, CTRL1, FMT_MASK, fmt_ctrl); + snd_soc_write(codec, CTRL2, ctrl2); + +hw_params_end: + if (ret < 0) + dev_warn(dev, "unsupported data width/format combination\n"); + + return ret; +} + +static int ak4613_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u8 mgmt1 = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + mgmt1 |= RSTN; + /* fall through */ + case SND_SOC_BIAS_PREPARE: + mgmt1 |= PMADC | PMDAC; + /* fall through */ + case SND_SOC_BIAS_STANDBY: + mgmt1 |= PMVR; + /* fall through */ + case SND_SOC_BIAS_OFF: + default: + break; + } + + snd_soc_write(codec, PW_MGMT1, mgmt1); + + return 0; +} + +static const struct snd_soc_dai_ops ak4613_dai_ops = { + .shutdown = ak4613_dai_shutdown, + .set_fmt = ak4613_dai_set_fmt, + .hw_params = ak4613_dai_hw_params, +}; + +#define AK4613_PCM_RATE (SNDRV_PCM_RATE_32000 |\ + SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_64000 |\ + SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000 |\ + SNDRV_PCM_RATE_176400 |\ + SNDRV_PCM_RATE_192000) +#define AK4613_PCM_FMTBIT (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver ak4613_dai = { + .name = "ak4613-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = AK4613_PCM_RATE, + .formats = AK4613_PCM_FMTBIT, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = AK4613_PCM_RATE, + .formats = AK4613_PCM_FMTBIT, + }, + .ops = &ak4613_dai_ops, + .symmetric_rates = 1, +}; + +static int ak4613_resume(struct snd_soc_codec *codec) +{ + struct regmap *regmap = dev_get_regmap(codec->dev, NULL); + + regcache_mark_dirty(regmap); + return regcache_sync(regmap); +} + +static struct snd_soc_codec_driver soc_codec_dev_ak4613 = { + .resume = ak4613_resume, + .set_bias_level = ak4613_set_bias_level, + .dapm_widgets = ak4613_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4613_dapm_widgets), + .dapm_routes = ak4613_intercon, + .num_dapm_routes = ARRAY_SIZE(ak4613_intercon), +}; + +static int ak4613_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct device_node *np = dev->of_node; + const struct regmap_config *regmap_cfg; + struct regmap *regmap; + struct ak4613_priv *priv; + + regmap_cfg = NULL; + if (np) { + const struct of_device_id *of_id; + + of_id = of_match_device(ak4613_of_match, dev); + if (of_id) + regmap_cfg = of_id->data; + } else { + regmap_cfg = (const struct regmap_config *)id->driver_data; + } + + if (!regmap_cfg) + return -EINVAL; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->fmt_ctrl = NO_FMT; + priv->cnt = 0; + + mutex_init(&priv->lock); + + i2c_set_clientdata(i2c, priv); + + regmap = devm_regmap_init_i2c(i2c, regmap_cfg); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return snd_soc_register_codec(dev, &soc_codec_dev_ak4613, + &ak4613_dai, 1); +} + +static int ak4613_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static struct i2c_driver ak4613_i2c_driver = { + .driver = { + .name = "ak4613-codec", + .owner = THIS_MODULE, + .of_match_table = ak4613_of_match, + }, + .probe = ak4613_i2c_probe, + .remove = ak4613_i2c_remove, + .id_table = ak4613_i2c_id, +}; + +module_i2c_driver(ak4613_i2c_driver); + +MODULE_DESCRIPTION("Soc AK4613 driver"); +MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>"); +MODULE_LICENSE("GPL v2"); -- 1.9.1

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[alsa-devel] [PATCH v3] ASoC: soc-core: Fix sparse warning in be32_to_cpup() call
by Jyri Sarha 17 Sep '15

17 Sep '15

Store return value of of_get_property() to a pointer of __be32 type as device tree has big endian type. This fixes a sparse warning couple of lines later when be32_to_cpup() is used to convert from big endian to cpu endian. The whole conversion is not really necessary, as we are only checking if the value is zero or not, but I wanted to add it to remind in the future that the data has to be converted before use. Compiler should optimize the unnecessary operations away. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- Damn, I wrote description ready but forgot to produce the patch again. So here is a v3 of the same one liner patch. sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1929f0e..af4999c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3294,7 +3294,7 @@ static int snd_soc_of_get_slot_mask(struct device_node *np, unsigned int *mask) { u32 val; - const u32 *of_slot_mask = of_get_property(np, prop_name, &val); + const __be32 *of_slot_mask = of_get_property(np, prop_name, &val); int i; if (!of_slot_mask) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: soc-core: Fix sparse warning in be32_to_cpup() call
by Jyri Sarha 17 Sep '15

17 Sep '15

Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1929f0e..614b831 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3301,7 +3301,7 @@ static int snd_soc_of_get_slot_mask(struct device_node *np, return 0; val /= sizeof(u32); for (i = 0; i < val; i++) - if (be32_to_cpup(&of_slot_mask[i])) + if (be32_to_cpup((__be32 *)&of_slot_mask[i])) *mask |= (1 << i); return val; -- 1.9.1

2 2

[alsa-devel] [PATCH 1/4] ASoC: arizona: Add default cases for event switches
by Charles Keepax 17 Sep '15

17 Sep '15

Since the addition of the WILL_PMU / WILL_PMD several of the switches in arizona.c no longer cover all cases or have a default case. Whilst this isn't causing any problems in the interests of robustness add default cases. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 8a2221a..1bc19f1 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -147,6 +147,8 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w, 0x4f5, 0x0da); } break; + default: + break; } return 0; @@ -725,6 +727,9 @@ int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, reg = snd_soc_read(codec, ARIZONA_INPUT_ENABLES); if (reg == 0) arizona_in_set_vu(codec, 0); + break; + default: + break; } return 0; @@ -806,6 +811,8 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, break; } break; + default: + break; } return 0; -- 1.7.2.5

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[alsa-devel] Test device for automated testing
by Uli Franke 17 Sep '15

17 Sep '15

Hi everybody, we're currently writing a lot of ALSA user space applications. In order to test this code with our unit testing and other test environments we'd like to know if there's a dummy PCM device which provides the fundamental operations like reading/writing and setting the common hardware parameters. It doesn't even have to supply some useful functionality like loop back or similar. Neither the docs nor the goo led us to something useful. Thanks and regards Uli

2 4

[alsa-devel] [PATCH 0/2] ASoC: atmel-classd: add the Audio Class D Amplifier
by Songjun Wu 17 Sep '15

17 Sep '15

The Audio Class D Amplifier driver includes two parts. 1) Driver code to implement the Audio Class D Amplifier function. 2) Device tree binding document, it describes how to add the Audio Class D Amplifier in device tree. Songjun Wu (2): ASoC: atmel-classd: add the Audio Class D Amplifier code ASoC: atmel-classd: DT binding for Class D audio amplifier driver .../devicetree/bindings/sound/atmel-classd.txt | 73 ++ sound/soc/atmel/Kconfig | 9 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel-classd.c | 801 ++++++++++++++++++++ sound/soc/atmel/atmel-classd.h | 120 +++ 5 files changed, 1005 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-classd.txt create mode 100644 sound/soc/atmel/atmel-classd.c create mode 100644 sound/soc/atmel/atmel-classd.h -- 1.7.9.5

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[alsa-devel] [asoc:topic/simple 1/1] sound/soc/soc-core.c:3306:47: sparse: incorrect type in argument 1 (different base types)
by kbuild test robot 16 Sep '15

16 Sep '15

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/simple head: 6131084a0bc966107021d8c89489f9cd1663b902 commit: 6131084a0bc966107021d8c89489f9cd1663b902 [1/1] ASoC: simple-card: Add tdm slot mask support to simple-card reproduce: # apt-get install sparse git checkout 6131084a0bc966107021d8c89489f9cd1663b902 make ARCH=x86_64 allmodconfig make C=1 CF=-D__CHECK_ENDIAN__ sparse warnings: (new ones prefixed by >>) >> sound/soc/soc-core.c:3306:47: sparse: incorrect type in argument 1 (different base types) sound/soc/soc-core.c:3306:47: expected restricted __be32 const [usertype] *p sound/soc/soc-core.c:3306:47: got unsigned int const [usertype] * vim +3306 sound/soc/soc-core.c 3290 return 0; 3291 } 3292 EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_simple_widgets); 3293 3294 static int snd_soc_of_get_slot_mask(struct device_node *np, 3295 const char *prop_name, 3296 unsigned int *mask) 3297 { 3298 u32 val; 3299 const u32 *of_slot_mask = of_get_property(np, prop_name, &val); 3300 int i; 3301 3302 if (!of_slot_mask) 3303 return 0; 3304 val /= sizeof(u32); 3305 for (i = 0; i < val; i++) > 3306 if (be32_to_cpup(&of_slot_mask[i])) 3307 *mask |= (1 << i); 3308 3309 return val; 3310 } 3311 3312 int snd_soc_of_parse_tdm_slot(struct device_node *np, 3313 unsigned int *tx_mask, 3314 unsigned int *rx_mask, --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH] ASoC: SPEAr: Make SND_SPEAR_SOC select SND_SOC_GENERIC_DMAENGINE_PCM
by Axel Lin 16 Sep '15

16 Sep '15

devm_snd_dmaengine_pcm_register() is guarded by CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/spear/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/spear/Kconfig b/sound/soc/spear/Kconfig index 0a53053..4fb9141 100644 --- a/sound/soc/spear/Kconfig +++ b/sound/soc/spear/Kconfig @@ -1,6 +1,6 @@ config SND_SPEAR_SOC tristate - select SND_DMAENGINE_PCM + select SND_SOC_GENERIC_DMAENGINE_PCM config SND_SPEAR_SPDIF_OUT tristate -- 2.1.0

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[alsa-devel] [PATCH RFC 0/5] ASoC: TDM stuff for simple-card, mcasp, and tlv320aic3x
by Jyri Sarha 16 Sep '15

16 Sep '15

The first patch is an independent fix. The symmetric_samplebits constraint could probably be removed too (with some changes to mcasp code too), but as tlv320aic3x family codecs won't support it, I could not test it. Jyri Sarha (5): ASoC: davinci-mcasp: Set .symmetric_rates = 1 in snd_soc_dai_driver ASoC: simple-card: Add tdm slot mask support to simple-card ASoC: davinci-mcasp: Add set_tdm_slots() support ASoC: davinci-mcasp: Get rid of bclk_lrclk_ratio in private data ASoC: tlv320aic3x: Improve tdm support .../devicetree/bindings/sound/tdm-slot.txt | 11 +- include/sound/simple_card.h | 2 + include/sound/soc.h | 2 + sound/soc/codecs/tlv320aic3x.c | 30 +- sound/soc/davinci/davinci-mcasp.c | 306 ++++++++++++++------- sound/soc/generic/simple-card.c | 8 +- sound/soc/soc-core.c | 25 ++ 7 files changed, 274 insertions(+), 110 deletions(-) -- 1.9.1

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