March 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2] ASoC: simple-card: overwrite cpu_dai->fmt with codec_dai->fmt
by Nicolin Chen 18 Mar '14

18 Mar '14

The current simple-card driver separates the daimft for cpu_dai and codec_dai. So we might get different values for them (0x4003 and 0x1003 for example): asoc-simple-card sound-cs42888.12: cpu : 2024000.esai / 4003 / 132000000 asoc-simple-card sound-cs42888.12: codec : cs42888 / 1003 / 24576000 asoc-simple-card sound-cs42888.12: cs42888 <-> 2024000.esai mapping ok This is not allowed at all as we need to keep the DAIFMT settings identical for both the ends of the link. Thus this patch fixes it by overwriting the cpu_dai->fmt with codec_dai->fmt since we defined the DAIFMT_MASTER basing on CODEC at the first place while the other bits are same. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- Changelog v2: * Keep the fmt identical for both CPU and CODEC sides. * Appended warning to binding doc. .../devicetree/bindings/sound/simple-card.txt | 6 ++++++ sound/soc/generic/simple-card.c | 20 ++++++++++++++------ 2 files changed, 20 insertions(+), 6 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index b30c222..881914b1 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -43,6 +43,12 @@ Optional CPU/CODEC subnodes properties: clock node (= common clock), or "system-clock-frequency" (if system doens't support common clock) +Note: + * For 'format', 'frame-master', 'bitclock-master', 'bitclock-inversion' and + 'frame-inversion', the simple card will use the settings of CODEC for both + CPU and CODEC sides as we need to keep the settings identical for both ends + of the link. + Example: sound { diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 5dd4769..09591ab 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -157,6 +157,8 @@ static int asoc_simple_card_parse_of(struct device_node *node, struct device *dev) { struct snd_soc_dai_link *dai_link = priv->snd_card.dai_link; + struct asoc_simple_dai *codec_dai = &priv->codec_dai; + struct asoc_simple_dai *cpu_dai = &priv->cpu_dai; struct device_node *np; char *name; int ret; @@ -189,7 +191,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, np = of_get_child_by_name(node, "simple-audio-card,cpu"); if (np) ret = asoc_simple_card_sub_parse_of(np, priv->daifmt, - &priv->cpu_dai, + cpu_dai, &dai_link->cpu_of_node, &dai_link->cpu_dai_name); if (ret < 0) @@ -200,12 +202,18 @@ static int asoc_simple_card_parse_of(struct device_node *node, np = of_get_child_by_name(node, "simple-audio-card,codec"); if (np) ret = asoc_simple_card_sub_parse_of(np, priv->daifmt, - &priv->codec_dai, + codec_dai, &dai_link->codec_of_node, &dai_link->codec_dai_name); if (ret < 0) return ret; + /* + * overwrite cpu_dai->fmt as its DAIFMT_MASTER bit is based on CODEC + * while the other bits should be identical unless buggy SW/HW design. + */ + cpu_dai->fmt = codec_dai->fmt; + if (!dai_link->cpu_dai_name || !dai_link->codec_dai_name) return -EINVAL; @@ -227,12 +235,12 @@ static int asoc_simple_card_parse_of(struct device_node *node, dev_dbg(dev, "platform : %04x\n", priv->daifmt); dev_dbg(dev, "cpu : %s / %04x / %d\n", dai_link->cpu_dai_name, - priv->cpu_dai.fmt, - priv->cpu_dai.sysclk); + cpu_dai->fmt, + cpu_dai->sysclk); dev_dbg(dev, "codec : %s / %04x / %d\n", dai_link->codec_dai_name, - priv->codec_dai.fmt, - priv->codec_dai.sysclk); + codec_dai->fmt, + codec_dai->sysclk); /* * soc_bind_dai_link() will check cpu name -- 1.8.4

3 4

[alsa-devel] [PATCH v3] ASoC: cs42888: Add codec driver support
by Nicolin Chen 18 Mar '14

18 Mar '14

This patch adds support for the Cirrus Logic CS42888 Audio CODEC that has four 24-bit A/D and eight 24-bit D/A converters. [ CS42888 supports both I2C and SPI control ports. As initial patch, this patch only adds the support for I2C. ] Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- Changelog v3: * Refined document at 'clocks' section. v2: * Separated driver into I2C and main part drivers. * Added error out if failed to get clock. * Enabled Auto-mute function as default. * Disabled runtime-pm in remove(). * Added return value checking for regcache_sync(). * Fixed some trival parts like 'const' as Lars mentioned. .../devicetree/bindings/sound/cs42888.txt | 28 ++ sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs42888-i2c.c | 68 +++ sound/soc/codecs/cs42888.c | 536 +++++++++++++++++++++ sound/soc/codecs/cs42888.h | 213 ++++++++ 6 files changed, 859 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs42888.txt create mode 100644 sound/soc/codecs/cs42888-i2c.c create mode 100644 sound/soc/codecs/cs42888.c create mode 100644 sound/soc/codecs/cs42888.h diff --git a/Documentation/devicetree/bindings/sound/cs42888.txt b/Documentation/devicetree/bindings/sound/cs42888.txt new file mode 100644 index 0000000..b20bd78 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs42888.txt @@ -0,0 +1,28 @@ +CS42888 audio CODEC + +Required properties: + + - compatible : "cirrus,cs42888" + + - reg : the I2C address of the device for I2C + + - clocks : a list of phandles + clock-specifiers, one for each entry in + clock-names + + - clock-names : must contain "mclk" + + - VA-supply, VD-supply, VLS-supply, VLC-supply: power supplies for the device, + as covered in Documentation/devicetree/bindings/regulator/regulator.txt + +Example: + +codec: cs42888@48 { + compatible = "cirrus,cs42888"; + reg = <0x48>; + clocks = <&codec_mclk 0>; + clock-names = "mclk"; + VA-supply = <&reg_audio>; + VD-supply = <&reg_audio>; + VLS-supply = <&reg_audio>; + VLC-supply = <&reg_audio>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 32d7a6f..1642578 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -44,6 +44,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS42L73 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI + select SND_SOC_CS42888_I2C if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if I2C select SND_SOC_DA7213 if I2C @@ -303,6 +304,15 @@ config SND_SOC_CS4271 tristate "Cirrus Logic CS4271 CODEC" depends on SND_SOC_I2C_AND_SPI +config SND_SOC_CS42888 + tristate + +config SND_SOC_CS42888_I2C + tristate "Cirrus Logic CS42888 CODEC (I2C)" + depends on I2C + select SND_SOC_CS42888 + select REGMAP_I2C + config SND_SOC_CX20442 tristate depends on TTY diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index cb46c4c..f5612fb 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -30,6 +30,8 @@ snd-soc-cs42l52-objs := cs42l52.o snd-soc-cs42l73-objs := cs42l73.o snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o +snd-soc-cs42888-objs := cs42888.o +snd-soc-cs42888-i2c-objs := cs42888-i2c.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -178,6 +180,8 @@ obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o +obj-$(CONFIG_SND_SOC_CS42888) += snd-soc-cs42888.o +obj-$(CONFIG_SND_SOC_CS42888_I2C) += snd-soc-cs42888-i2c.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o diff --git a/sound/soc/codecs/cs42888-i2c.c b/sound/soc/codecs/cs42888-i2c.c new file mode 100644 index 0000000..54813bc --- /dev/null +++ b/sound/soc/codecs/cs42888-i2c.c @@ -0,0 +1,68 @@ +/* + * Cirrus Logic CS42888 Audio CODEC Digital Audio Interface (DAI) I2C driver + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen(a)freescale.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/pm_runtime.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "cs42888.h" + +static int cs42888_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + u32 ret = cs42888_probe(&i2c->dev, + devm_regmap_init_i2c(i2c, &cs42888_regmap_config)); + if (ret) + return ret; + + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + + return 0; +} + +static int cs42888_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + pm_runtime_disable(&i2c->dev); + + return 0; +} + +static struct i2c_device_id cs42888_i2c_id[] = { + {"cs42888", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs42888_i2c_id); + +static struct i2c_driver cs42888_i2c_driver = { + .driver = { + .name = "cs42888", + .owner = THIS_MODULE, + .pm = &cs42888_pm, + }, + .probe = cs42888_i2c_probe, + .remove = cs42888_i2c_remove, + .id_table = cs42888_i2c_id, +}; + +module_i2c_driver(cs42888_i2c_driver); + +MODULE_DESCRIPTION("Cirrus Logic CS42888 ALSA SoC Codec I2C Driver"); +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42888.c b/sound/soc/codecs/cs42888.c new file mode 100644 index 0000000..8cd3d4a --- /dev/null +++ b/sound/soc/codecs/cs42888.c @@ -0,0 +1,536 @@ +/* + * Cirrus Logic CS42888 Audio CODEC Digital Audio Interface (DAI) driver + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen(a)freescale.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/module.h> +#include <linux/pm_runtime.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "cs42888.h" + +#define CS42888_NUM_SUPPLIES 4 +static const char *const cs42888_supply_names[CS42888_NUM_SUPPLIES] = { + "VA", + "VD", + "VLS", + "VLC", +}; + +#define CS42888_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* codec private data */ +struct cs42888_priv { + struct regulator_bulk_data supplies[CS42888_NUM_SUPPLIES]; + struct regmap *regmap; + struct clk *clk; + + bool slave_mode; + unsigned long sysclk; +}; + +/* -127.5dB to 0dB with step of 0.5dB */ +static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); +/* -64dB to 24dB with step of 0.5dB */ +static const DECLARE_TLV_DB_SCALE(adc_tlv, -6400, 50, 0); + +static const char *const cs42888_adc_single[] = { "Differential", "Single-Ended" }; +static const char *const cs42888_szc[] = { "Immediate Change", "Zero Cross", + "Soft Ramp", "Soft Ramp on Zero Cross" }; + +static const struct soc_enum adc1_single_enum = + SOC_ENUM_SINGLE(CS42888_ADCCTL, 4, 2, cs42888_adc_single); +static const struct soc_enum adc2_single_enum = + SOC_ENUM_SINGLE(CS42888_ADCCTL, 3, 2, cs42888_adc_single); +static const struct soc_enum dac_szc_enum = + SOC_ENUM_SINGLE(CS42888_TXCTL, 5, 4, cs42888_szc); +static const struct soc_enum adc_szc_enum = + SOC_ENUM_SINGLE(CS42888_TXCTL, 0, 4, cs42888_szc); + +static const struct snd_kcontrol_new cs42888_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC1 Playback Volume", CS42888_VOLAOUT1, + CS42888_VOLAOUT2, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC2 Playback Volume", CS42888_VOLAOUT3, + CS42888_VOLAOUT4, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC3 Playback Volume", CS42888_VOLAOUT5, + CS42888_VOLAOUT6, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC4 Playback Volume", CS42888_VOLAOUT7, + CS42888_VOLAOUT8, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_S_TLV("ADC1 Capture Volume", CS42888_VOLAIN1, + CS42888_VOLAIN2, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE_R_S_TLV("ADC2 Capture Volume", CS42888_VOLAIN3, + CS42888_VOLAIN4, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE("DAC1 Invert Switch", CS42888_DACINV, 0, 1, 1, 0), + SOC_DOUBLE("DAC2 Invert Switch", CS42888_DACINV, 2, 3, 1, 0), + SOC_DOUBLE("DAC3 Invert Switch", CS42888_DACINV, 4, 5, 1, 0), + SOC_DOUBLE("DAC4 Invert Switch", CS42888_DACINV, 6, 7, 1, 0), + SOC_DOUBLE("ADC1 Invert Switch", CS42888_ADCINV, 0, 1, 1, 0), + SOC_DOUBLE("ADC2 Invert Switch", CS42888_ADCINV, 2, 3, 1, 0), + SOC_SINGLE("ADC High-Pass Filter Switch", CS42888_ADCCTL, 7, 1, 1), + SOC_SINGLE("DAC De-emphasis Switch", CS42888_ADCCTL, 5, 1, 0), + SOC_ENUM("ADC1 Single Ended Mode Switch", adc1_single_enum), + SOC_ENUM("ADC2 Single Ended Mode Switch", adc2_single_enum), + SOC_SINGLE("DAC Single Volume Control Switch", CS42888_TXCTL, 7, 1, 0), + SOC_ENUM("DAC Soft Ramp & Zero Cross Control Switch", dac_szc_enum), + SOC_SINGLE("DAC Auto Mute Switch", CS42888_TXCTL, 4, 1, 0), + SOC_SINGLE("Mute ADC Serial Port Switch", CS42888_TXCTL, 3, 1, 0), + SOC_SINGLE("ADC Single Volume Control Switch", CS42888_TXCTL, 2, 1, 0), + SOC_ENUM("ADC Soft Ramp & Zero Cross Control Switch", adc_szc_enum), +}; + +static const struct snd_soc_dapm_widget cs42888_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC1", "Playback", CS42888_PWRCTL, 1, 1), + SND_SOC_DAPM_DAC("DAC2", "Playback", CS42888_PWRCTL, 2, 1), + SND_SOC_DAPM_DAC("DAC3", "Playback", CS42888_PWRCTL, 3, 1), + SND_SOC_DAPM_DAC("DAC4", "Playback", CS42888_PWRCTL, 4, 1), + + SND_SOC_DAPM_OUTPUT("AOUT1L"), + SND_SOC_DAPM_OUTPUT("AOUT1R"), + SND_SOC_DAPM_OUTPUT("AOUT2L"), + SND_SOC_DAPM_OUTPUT("AOUT2R"), + SND_SOC_DAPM_OUTPUT("AOUT3L"), + SND_SOC_DAPM_OUTPUT("AOUT3R"), + SND_SOC_DAPM_OUTPUT("AOUT4L"), + SND_SOC_DAPM_OUTPUT("AOUT4R"), + + SND_SOC_DAPM_ADC("ADC1", "Capture", CS42888_PWRCTL, 5, 1), + SND_SOC_DAPM_ADC("ADC2", "Capture", CS42888_PWRCTL, 6, 1), + + SND_SOC_DAPM_INPUT("AIN1L"), + SND_SOC_DAPM_INPUT("AIN1R"), + SND_SOC_DAPM_INPUT("AIN2L"), + SND_SOC_DAPM_INPUT("AIN2R"), + + SND_SOC_DAPM_SUPPLY("PWR", CS42888_PWRCTL, 0, 1, NULL, 0), +}; + +static const struct snd_soc_dapm_route cs42888_dapm_routes[] = { + /* Playback */ + { "AOUT1L", NULL, "DAC1" }, + { "AOUT1R", NULL, "DAC1" }, + { "DAC1", NULL, "PWR" }, + + { "AOUT2L", NULL, "DAC2" }, + { "AOUT2R", NULL, "DAC2" }, + { "DAC2", NULL, "PWR" }, + + { "AOUT3L", NULL, "DAC3" }, + { "AOUT3R", NULL, "DAC3" }, + { "DAC3", NULL, "PWR" }, + + { "AOUT4L", NULL, "DAC4" }, + { "AOUT4R", NULL, "DAC4" }, + { "DAC4", NULL, "PWR" }, + + /* Capture */ + { "ADC1", NULL, "AIN1L" }, + { "ADC1", NULL, "AIN1R" }, + { "ADC1", NULL, "PWR" }, + + { "ADC2", NULL, "AIN2L" }, + { "ADC2", NULL, "AIN2R" }, + { "ADC2", NULL, "PWR" }, +}; + +struct cs42888_ratios { + unsigned int ratio; + unsigned char speed; + unsigned char mclk; +}; + +static const struct cs42888_ratios cs42888_ratios[] = { + { 64, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_256(4) }, + { 96, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_384(4) }, + { 128, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_512(4) }, + { 192, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_768(4) }, + { 256, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_256(1) }, + { 384, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_384(1) }, + { 512, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_512(1) }, + { 768, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_768(1) }, + { 1024, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_1024(1) } +}; + +static int cs42888_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + + cs42888->sysclk = freq; + + return 0; +} + +static int cs42888_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + u32 val; + + /* Set DAI format */ + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_LEFT_J: + val = CS42888_INTF_DAC_DIF_LEFTJ | CS42888_INTF_ADC_DIF_LEFTJ; + break; + case SND_SOC_DAIFMT_I2S: + val = CS42888_INTF_DAC_DIF_I2S | CS42888_INTF_ADC_DIF_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val = CS42888_INTF_DAC_DIF_RIGHTJ | CS42888_INTF_ADC_DIF_RIGHTJ; + break; + default: + dev_err(codec->dev, "unsupported dai format\n"); + return -EINVAL; + } + + regmap_update_bits(cs42888->regmap, CS42888_INTF, + CS42888_INTF_DAC_DIF_MASK | + CS42888_INTF_ADC_DIF_MASK, val); + + /* Set master/slave audio interface */ + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cs42888->slave_mode = true; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs42888->slave_mode = false; + break; + default: + dev_err(codec->dev, "unsupported master/slave mode\n"); + return -EINVAL; + } + + return 0; +} + +static int cs42888_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + u32 ratio = cs42888->sysclk / params_rate(params); + u32 i, fm, val, mask; + + for (i = 0; i < ARRAY_SIZE(cs42888_ratios); i++) { + if (cs42888_ratios[i].ratio == ratio) + break; + } + + if (i == ARRAY_SIZE(cs42888_ratios)) { + dev_err(codec->dev, "unsupported sysclk ratio\n"); + return -EINVAL; + } + + mask = CS42888_FUNCMOD_MFREQ_MASK; + val = cs42888_ratios[i].mclk; + + fm = cs42888->slave_mode ? CS42888_FM_AUTO : cs42888_ratios[i].speed; + + regmap_update_bits(cs42888->regmap, CS42888_FUNCMOD, + CS42888_FUNCMOD_xC_FM_MASK(tx) | mask, + CS42888_FUNCMOD_xC_FM(tx, fm) | val); + + return 0; +} + +static int cs42888_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(cs42888->regmap, CS42888_DACMUTE, + CS42888_DACMUTE_ALL, mute ? CS42888_DACMUTE_ALL : 0); + + return 0; +} + +static const struct snd_soc_dai_ops cs42888_dai_ops = { + .set_fmt = cs42888_set_dai_fmt, + .set_sysclk = cs42888_set_dai_sysclk, + .hw_params = cs42888_hw_params, + .digital_mute = cs42888_digital_mute, +}; + +static struct snd_soc_dai_driver cs42888_dai = { + .name = "cs42888", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = CS42888_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = CS42888_FORMATS, + }, + .ops = &cs42888_dai_ops, +}; + +static const struct reg_default cs42888_reg[] = { + { 0x01, 0x01 }, /* Chip I.D. and Revision Register */ + { 0x02, 0x00 }, /* Power Control */ + { 0x03, 0xF0 }, /* Functional Mode */ + { 0x04, 0x46 }, /* Interface Formats */ + { 0x05, 0x00 }, /* ADC Control & DAC De-Emphasis */ + { 0x06, 0x10 }, /* Transition Control */ + { 0x07, 0x00 }, /* DAC Channel Mute */ + { 0x08, 0x00 }, /* Volume Control AOUT1 */ + { 0x09, 0x00 }, /* Volume Control AOUT2 */ + { 0x0a, 0x00 }, /* Volume Control AOUT3 */ + { 0x0b, 0x00 }, /* Volume Control AOUT4 */ + { 0x0c, 0x00 }, /* Volume Control AOUT5 */ + { 0x0d, 0x00 }, /* Volume Control AOUT6 */ + { 0x0e, 0x00 }, /* Volume Control AOUT7 */ + { 0x0f, 0x00 }, /* Volume Control AOUT8 */ + { 0x10, 0x00 }, /* DAC Channel Invert */ + { 0x11, 0x00 }, /* Volume Control AIN1 */ + { 0x12, 0x00 }, /* Volume Control AIN2 */ + { 0x13, 0x00 }, /* Volume Control AIN3 */ + { 0x14, 0x00 }, /* Volume Control AIN4 */ + { 0x17, 0x00 }, /* ADC Channel Invert */ + { 0x18, 0x00 }, /* Status Control */ + { 0x1a, 0x00 }, /* Status Mask */ + { 0x1b, 0x00 }, /* MUTEC Pin Control */ +}; + +static bool cs42888_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42888_STATUS: + return true; + default: + return false; + } +} + +static bool cs42888_writeable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42888_CHIPID: + case CS42888_STATUS: + return false; + default: + return true; + } +} + +const struct regmap_config cs42888_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS42888_LASTREG, + .reg_defaults = cs42888_reg, + .num_reg_defaults = ARRAY_SIZE(cs42888_reg), + .volatile_reg = cs42888_volatile_register, + .writeable_reg = cs42888_writeable_register, + .cache_type = REGCACHE_RBTREE, +}; +EXPORT_SYMBOL_GPL(cs42888_regmap_config); + +static int cs42888_codec_probe(struct snd_soc_codec *codec) +{ + struct cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + + /* Mute all DAC channels */ + regmap_write(cs42888->regmap, CS42888_DACMUTE, CS42888_DACMUTE_ALL); + + return 0; +} + +static const struct snd_soc_codec_driver cs42888_driver = { + .probe = cs42888_codec_probe, + .idle_bias_off = true, + + .controls = cs42888_snd_controls, + .num_controls = ARRAY_SIZE(cs42888_snd_controls), + .dapm_widgets = cs42888_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42888_dapm_widgets), + .dapm_routes = cs42888_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(cs42888_dapm_routes), +}; + +int cs42888_probe(struct device *dev, struct regmap *regmap) +{ + struct cs42888_priv *cs42888; + int ret, val, i; + + cs42888 = devm_kzalloc(dev, sizeof(*cs42888), GFP_KERNEL); + if (cs42888 == NULL) + return -ENOMEM; + + dev_set_drvdata(dev, cs42888); + + cs42888->clk = devm_clk_get(dev, "mclk"); + if (IS_ERR(cs42888->clk)) { + dev_err(dev, "failed to get the clock: %ld\n", + PTR_ERR(cs42888->clk)); + return -EINVAL; + } + + cs42888->sysclk = clk_get_rate(cs42888->clk); + + for (i = 0; i < ARRAY_SIZE(cs42888->supplies); i++) + cs42888->supplies[i].supply = cs42888_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, + ARRAY_SIZE(cs42888->supplies), cs42888->supplies); + if (ret) { + dev_err(dev, "failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); + if (ret) { + dev_err(dev, "failed to enable supplies: %d\n", ret); + return ret; + } + + /* Make sure hardware reset done */ + msleep(5); + + cs42888->regmap = regmap; + if (IS_ERR(cs42888->regmap)) { + ret = PTR_ERR(cs42888->regmap); + dev_err(dev, "failed to allocate regmap: %d\n", ret); + goto err_enable; + } + + /* + * We haven't marked the chip revision as volatile due to + * sharing a register with the right input volume; explicitly + * bypass the cache to read it. + */ + regcache_cache_bypass(cs42888->regmap, true); + + /* Validate the chip ID */ + regmap_read(cs42888->regmap, CS42888_CHIPID, &val); + if (val < 0) { + dev_err(dev, "failed to get device ID: %x", val); + ret = -EINVAL; + goto err_enable; + } + + /* The top four bits of the chip ID should be 0000 */ + if ((val & CS42888_CHIPID_CHIP_ID_MASK) != 0x00) { + dev_err(dev, "unmatched chip ID: %d\n", + val & CS42888_CHIPID_CHIP_ID_MASK); + ret = -EINVAL; + goto err_enable; + } + + dev_info(dev, "found device, revision %X\n", + val & CS42888_CHIPID_REV_ID_MASK); + + regcache_cache_bypass(cs42888->regmap, false); + + ret = snd_soc_register_codec(dev, &cs42888_driver, &cs42888_dai, 1); + if (ret) { + dev_err(dev, "failed to register codec:%d\n", ret); + goto err_enable; + } + + regcache_cache_only(cs42888->regmap, true); + +err_enable: + regulator_bulk_disable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); + + return ret; +} +EXPORT_SYMBOL_GPL(cs42888_probe); + +#ifdef CONFIG_PM_RUNTIME +static int cs42888_runtime_resume(struct device *dev) +{ + struct cs42888_priv *cs42888 = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(cs42888->clk); + if (ret) { + dev_err(dev, "failed to enable mclk: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); + if (ret) { + dev_err(dev, "failed to enable supplies: %d\n", ret); + goto err_clk; + } + + /* Make sure hardware reset done */ + msleep(5); + + regcache_cache_only(cs42888->regmap, false); + + ret = regcache_sync(cs42888->regmap); + if (ret) { + dev_err(dev, "failed to sync regmap: %d\n", ret); + goto err_bulk; + } + + return 0; + +err_bulk: + regulator_bulk_disable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); +err_clk: + clk_disable_unprepare(cs42888->clk); + + return ret; +} + +static int cs42888_runtime_suspend(struct device *dev) +{ + struct cs42888_priv *cs42888 = dev_get_drvdata(dev); + + regcache_cache_only(cs42888->regmap, true); + + regulator_bulk_disable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); + + clk_disable_unprepare(cs42888->clk); + + return 0; +} +#endif + +const struct dev_pm_ops cs42888_pm = { + SET_RUNTIME_PM_OPS(cs42888_runtime_suspend, cs42888_runtime_resume, NULL) +}; +EXPORT_SYMBOL_GPL(cs42888_pm); + +const struct of_device_id cs42888_of_match[] = { + { .compatible = "cirrus,cs42888", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, cs42888_of_match); +EXPORT_SYMBOL_GPL(cs42888_of_match); + +MODULE_DESCRIPTION("Cirrus Logic CS42888 ALSA SoC Codec Driver"); +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42888.h b/sound/soc/codecs/cs42888.h new file mode 100644 index 0000000..eda55ac --- /dev/null +++ b/sound/soc/codecs/cs42888.h @@ -0,0 +1,213 @@ +/* + * cs42888.h - Cirrus Logic CS42888 Audio CODEC driver header file + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen(a)freescale.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#ifndef _CS42888_H +#define _CS42888_H + +extern const struct dev_pm_ops cs42888_pm; +extern const struct regmap_config cs42888_regmap_config; +int cs42888_probe(struct device *dev, struct regmap *regmap); + +/* CS42888 register map */ +#define CS42888_CHIPID 0x01 /* Chip ID */ +#define CS42888_PWRCTL 0x02 /* Power Control */ +#define CS42888_FUNCMOD 0x03 /* Functional Mode */ +#define CS42888_INTF 0x04 /* Interface Formats */ +#define CS42888_ADCCTL 0x05 /* ADC Control */ +#define CS42888_TXCTL 0x06 /* Transition Control */ +#define CS42888_DACMUTE 0x07 /* DAC Mute Control */ +#define CS42888_VOLAOUT1 0x08 /* Volume Control AOUT1 */ +#define CS42888_VOLAOUT2 0x09 /* Volume Control AOUT2 */ +#define CS42888_VOLAOUT3 0x0A /* Volume Control AOUT3 */ +#define CS42888_VOLAOUT4 0x0B /* Volume Control AOUT4 */ +#define CS42888_VOLAOUT5 0x0C /* Volume Control AOUT5 */ +#define CS42888_VOLAOUT6 0x0D /* Volume Control AOUT6 */ +#define CS42888_VOLAOUT7 0x0E /* Volume Control AOUT7 */ +#define CS42888_VOLAOUT8 0x0F /* Volume Control AOUT8 */ +#define CS42888_DACINV 0x10 /* DAC Channel Invert */ +#define CS42888_VOLAIN1 0x11 /* Volume Control AIN1 */ +#define CS42888_VOLAIN2 0x12 /* Volume Control AIN2 */ +#define CS42888_VOLAIN3 0x13 /* Volume Control AIN3 */ +#define CS42888_VOLAIN4 0x14 /* Volume Control AIN4 */ +#define CS42888_ADCINV 0x17 /* ADC Channel Invert */ +#define CS42888_STATUSCTL 0x18 /* Status Control */ +#define CS42888_STATUS 0x19 /* Status */ +#define CS42888_STATUSM 0x1A /* Status Mask */ +#define CS42888_MUTEC 0x1B /* MUTEC Pin Control */ + +#define CS42888_FIRSTREG CS42888_CHIPID +#define CS42888_LASTREG CS42888_MUTEC +#define CS42888_NUMREGS (CS42888_LASTREG - CS42888_FIRSTREG + 1) +#define CS42888_I2C_INCR 0x80 + +/* Chip I.D. and Revision Register (Address 01h) */ +#define CS42888_CHIPID_CHIP_ID_MASK 0xF0 +#define CS42888_CHIPID_REV_ID_MASK 0x0F + +/* Power Control (Address 02h) */ +#define CS42888_PWRCTL_PDN_ADC2_SHIFT 6 +#define CS42888_PWRCTL_PDN_ADC2_MASK (1 << CS42888_PWRCTL_PDN_ADC2_SHIFT) +#define CS42888_PWRCTL_PDN_ADC2 (1 << CS42888_PWRCTL_PDN_ADC2_SHIFT) +#define CS42888_PWRCTL_PDN_ADC1_SHIFT 5 +#define CS42888_PWRCTL_PDN_ADC1_MASK (1 << CS42888_PWRCTL_PDN_ADC1_SHIFT) +#define CS42888_PWRCTL_PDN_ADC1 (1 << CS42888_PWRCTL_PDN_ADC1_SHIFT) +#define CS42888_PWRCTL_PDN_DAC4_SHIFT 4 +#define CS42888_PWRCTL_PDN_DAC4_MASK (1 << CS42888_PWRCTL_PDN_DAC4_SHIFT) +#define CS42888_PWRCTL_PDN_DAC4 (1 << CS42888_PWRCTL_PDN_DAC4_SHIFT) +#define CS42888_PWRCTL_PDN_DAC3_SHIFT 3 +#define CS42888_PWRCTL_PDN_DAC3_MASK (1 << CS42888_PWRCTL_PDN_DAC3_SHIFT) +#define CS42888_PWRCTL_PDN_DAC3 (1 << CS42888_PWRCTL_PDN_DAC3_SHIFT) +#define CS42888_PWRCTL_PDN_DAC2_SHIFT 2 +#define CS42888_PWRCTL_PDN_DAC2_MASK (1 << CS42888_PWRCTL_PDN_DAC2_SHIFT) +#define CS42888_PWRCTL_PDN_DAC2 (1 << CS42888_PWRCTL_PDN_DAC2_SHIFT) +#define CS42888_PWRCTL_PDN_DAC1_SHIFT 1 +#define CS42888_PWRCTL_PDN_DAC1_MASK (1 << CS42888_PWRCTL_PDN_DAC1_SHIFT) +#define CS42888_PWRCTL_PDN_DAC1 (1 << CS42888_PWRCTL_PDN_DAC1_SHIFT) +#define CS42888_PWRCTL_PDN_SHIFT 0 +#define CS42888_PWRCTL_PDN_MASK (1 << CS42888_PWRCTL_PDN_SHIFT) +#define CS42888_PWRCTL_PDN (1 << CS42888_PWRCTL_PDN_SHIFT) + +/* Functional Mode (Address 03h) */ +#define CS42888_FUNCMOD_DAC_FM_SHIFT 6 +#define CS42888_FUNCMOD_DAC_FM_WIDTH 2 +#define CS42888_FUNCMOD_DAC_FM_MASK (((1 << CS42888_FUNCMOD_DAC_FM_WIDTH) - 1) << CS42888_FUNCMOD_DAC_FM_SHIFT) +#define CS42888_FUNCMOD_DAC_FM(v) ((v) << CS42888_FUNCMOD_DAC_FM_SHIFT) +#define CS42888_FUNCMOD_ADC_FM_SHIFT 4 +#define CS42888_FUNCMOD_ADC_FM_WIDTH 2 +#define CS42888_FUNCMOD_ADC_FM_MASK (((1 << CS42888_FUNCMOD_ADC_FM_WIDTH) - 1) << CS42888_FUNCMOD_ADC_FM_SHIFT) +#define CS42888_FUNCMOD_ADC_FM(v) ((v) << CS42888_FUNCMOD_ADC_FM_SHIFT) +#define CS42888_FUNCMOD_xC_FM_MASK(x) ((x) ? CS42888_FUNCMOD_DAC_FM_MASK : CS42888_FUNCMOD_ADC_FM_MASK) +#define CS42888_FUNCMOD_xC_FM(x, v) ((x) ? CS42888_FUNCMOD_DAC_FM(v) : CS42888_FUNCMOD_ADC_FM(v)) +#define CS42888_FUNCMOD_MFREQ_SHIFT 1 +#define CS42888_FUNCMOD_MFREQ_WIDTH 3 +#define CS42888_FUNCMOD_MFREQ_MASK (((1 << CS42888_FUNCMOD_MFREQ_WIDTH) - 1) << CS42888_FUNCMOD_MFREQ_SHIFT) +#define CS42888_FUNCMOD_MFREQ_256(s) ((0 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_384(s) ((1 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_512(s) ((2 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_768(s) ((3 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_1024(s) ((4 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) + +#define CS42888_FM_SINGLE 0 +#define CS42888_FM_DOUBLE 1 +#define CS42888_FM_QUAD 2 +#define CS42888_FM_AUTO 3 + +/* Interface Formats (Address 04h) */ +#define CS42888_INTF_FREEZE_SHIFT 7 +#define CS42888_INTF_FREEZE_MASK (1 << CS42888_INTF_FREEZE_SHIFT) +#define CS42888_INTF_FREEZE (1 << CS42888_INTF_FREEZE_SHIFT) +#define CS42888_INTF_AUX_DIF_SHIFT 6 +#define CS42888_INTF_AUX_DIF_MASK (1 << CS42888_INTF_AUX_DIF_SHIFT) +#define CS42888_INTF_AUX_DIF (1 << CS42888_INTF_AUX_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_SHIFT 3 +#define CS42888_INTF_DAC_DIF_WIDTH 3 +#define CS42888_INTF_DAC_DIF_MASK (((1 << CS42888_INTF_DAC_DIF_WIDTH) - 1) << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_LEFTJ (0 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_I2S (1 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_RIGHTJ (2 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_RIGHTJ_16 (3 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_ONELINE_20 (4 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_ONELINE_24 (6 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_TDM (7 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_SHIFT 0 +#define CS42888_INTF_ADC_DIF_WIDTH 3 +#define CS42888_INTF_ADC_DIF_MASK (((1 << CS42888_INTF_ADC_DIF_WIDTH) - 1) << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_LEFTJ (0 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_I2S (1 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_RIGHTJ (2 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_RIGHTJ_16 (3 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_ONELINE_20 (4 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_ONELINE_24 (6 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_TDM (7 << CS42888_INTF_ADC_DIF_SHIFT) + +/* ADC Control & DAC De-Emphasis (Address 05h) */ +#define CS42888_ADCCTL_ADC_HPF_FREEZE_SHIFT 7 +#define CS42888_ADCCTL_ADC_HPF_FREEZE_MASK (1 << CS42888_ADCCTL_ADC_HPF_FREEZE_SHIFT) +#define CS42888_ADCCTL_ADC_HPF_FREEZE (1 << CS42888_ADCCTL_ADC_HPF_FREEZE_SHIFT) +#define CS42888_ADCCTL_DAC_DEM_SHIFT 5 +#define CS42888_ADCCTL_DAC_DEM_MASK (1 << CS42888_ADCCTL_DAC_DEM_SHIFT) +#define CS42888_ADCCTL_DAC_DEM (1 << CS42888_ADCCTL_DAC_DEM_SHIFT) +#define CS42888_ADCCTL_ADC1_SINGLE_SHIFT 4 +#define CS42888_ADCCTL_ADC1_SINGLE_MASK (1 << CS42888_ADCCTL_ADC1_SINGLE_SHIFT) +#define CS42888_ADCCTL_ADC1_SINGLE (1 << CS42888_ADCCTL_ADC1_SINGLE_SHIFT) +#define CS42888_ADCCTL_ADC2_SINGLE_SHIFT 3 +#define CS42888_ADCCTL_ADC2_SINGLE_MASK (1 << CS42888_ADCCTL_ADC2_SINGLE_SHIFT) +#define CS42888_ADCCTL_ADC2_SINGLE (1 << CS42888_ADCCTL_ADC2_SINGLE_SHIFT) + +/* Transition Control (Address 06h) */ +#define CS42888_TXCTL_DAC_SNGVOL_SHIFT 7 +#define CS42888_TXCTL_DAC_SNGVOL_MASK (1 << CS42888_TXCTL_DAC_SNGVOL_SHIFT) +#define CS42888_TXCTL_DAC_SNGVOL (1 << CS42888_TXCTL_DAC_SNGVOL_SHIFT) +#define CS42888_TXCTL_DAC_SZC_SHIFT 5 +#define CS42888_TXCTL_DAC_SZC_WIDTH 2 +#define CS42888_TXCTL_DAC_SZC_MASK (((1 << CS42888_TXCTL_DAC_SZC_WIDTH) - 1) << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_IC (0 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_ZC (1 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_SR (2 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_SRZC (3 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_AMUTE_SHIFT 4 +#define CS42888_TXCTL_AMUTE_MASK (1 << CS42888_TXCTL_AMUTE_SHIFT) +#define CS42888_TXCTL_AMUTE (1 << CS42888_TXCTL_AMUTE_SHIFT) +#define CS42888_TXCTL_MUTE_ADC_SP_SHIFT 3 +#define CS42888_TXCTL_MUTE_ADC_SP_MASK (1 << CS42888_TXCTL_MUTE_ADC_SP_SHIFT) +#define CS42888_TXCTL_MUTE_ADC_SP (1 << CS42888_TXCTL_MUTE_ADC_SP_SHIFT) +#define CS42888_TXCTL_ADC_SNGVOL_SHIFT 2 +#define CS42888_TXCTL_ADC_SNGVOL_MASK (1 << CS42888_TXCTL_ADC_SNGVOL_SHIFT) +#define CS42888_TXCTL_ADC_SNGVOL (1 << CS42888_TXCTL_ADC_SNGVOL_SHIFT) +#define CS42888_TXCTL_ADC_SZC_SHIFT 0 +#define CS42888_TXCTL_ADC_SZC_MASK (((1 << CS42888_TXCTL_ADC_SZC_WIDTH) - 1) << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_IC (0 << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_ZC (1 << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_SR (2 << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_SRZC (3 << CS42888_TXCTL_ADC_SZC_SHIFT) + +/* DAC Channel Mute (Address 07h) */ +#define CS42888_DACMUTE_AOUT(n) (0x1 << n) +#define CS42888_DACMUTE_ALL 0xff + +/* Status Control (Address 18h)*/ +#define CS42888_STATUSCTL_INI_SHIFT 2 +#define CS42888_STATUSCTL_INI_WIDTH 2 +#define CS42888_STATUSCTL_INI_MASK (((1 << CS42888_STATUSCTL_INI_WIDTH) - 1) << CS42888_STATUSCTL_INI_SHIFT) +#define CS42888_STATUSCTL_INT_ACTIVE_HIGH (0 << CS42888_STATUSCTL_INI_SHIFT) +#define CS42888_STATUSCTL_INT_ACTIVE_LOW (1 << CS42888_STATUSCTL_INI_SHIFT) +#define CS42888_STATUSCTL_INT_OPEN_DRAIN (2 << CS42888_STATUSCTL_INI_SHIFT) + +/* Status (Address 19h)*/ +#define CS42888_STATUS_DAC_CLK_ERR_SHIFT 4 +#define CS42888_STATUS_DAC_CLK_ERR_MASK (1 << CS42888_STATUS_DAC_CLK_ERR_SHIFT) +#define CS42888_STATUS_ADC_CLK_ERR_SHIFT 3 +#define CS42888_STATUS_ADC_CLK_ERR_MASK (1 << CS42888_STATUS_ADC_CLK_ERR_SHIFT) +#define CS42888_STATUS_ADC2_OVFL_SHIFT 1 +#define CS42888_STATUS_ADC2_OVFL_MASK (1 << CS42888_STATUS_ADC2_OVFL_SHIFT) +#define CS42888_STATUS_ADC1_OVFL_SHIFT 0 +#define CS42888_STATUS_ADC1_OVFL_MASK (1 << CS42888_STATUS_ADC1_OVFL_SHIFT) + +/* Status Mask (Address 1Ah) */ +#define CS42888_STATUS_DAC_CLK_ERR_M_SHIFT 4 +#define CS42888_STATUS_DAC_CLK_ERR_M_MASK (1 << CS42888_STATUS_DAC_CLK_ERR_M_SHIFT) +#define CS42888_STATUS_ADC_CLK_ERR_M_SHIFT 3 +#define CS42888_STATUS_ADC_CLK_ERR_M_MASK (1 << CS42888_STATUS_ADC_CLK_ERR_M_SHIFT) +#define CS42888_STATUS_ADC2_OVFL_M_SHIFT 1 +#define CS42888_STATUS_ADC2_OVFL_M_MASK (1 << CS42888_STATUS_ADC2_OVFL_M_SHIFT) +#define CS42888_STATUS_ADC1_OVFL_M_SHIFT 0 +#define CS42888_STATUS_ADC1_OVFL_M_MASK (1 << CS42888_STATUS_ADC1_OVFL_M_SHIFT) + +/* MUTEC Pin Control (Address 1Bh) */ +#define CS42888_MUTEC_MCPOLARITY_SHIFT 1 +#define CS42888_MUTEC_MCPOLARITY_MASK (1 << CS42888_MUTEC_MCPOLARITY_SHIFT) +#define CS42888_MUTEC_MCPOLARITY_ACTIVE_LOW (0 << CS42888_MUTEC_MCPOLARITY_SHIFT) +#define CS42888_MUTEC_MCPOLARITY_ACTIVE_HIGH (1 << CS42888_MUTEC_MCPOLARITY_SHIFT) +#define CS42888_MUTEC_MUTEC_ACTIVE_SHIFT 0 +#define CS42888_MUTEC_MUTEC_ACTIVE_MASK (1 << CS42888_MUTEC_MUTEC_ACTIVE_SHIFT) +#define CS42888_MUTEC_MUTEC_ACTIVE (1 << CS42888_MUTEC_MUTEC_ACTIVE_SHIFT) +#endif /* _CS42888_H */ -- 1.8.4

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[alsa-devel] [PATCH] ASoC: cs42l52: Fix mask bits for SOC_VALUE_ENUM_SINGLE
by Brian Austin 18 Mar '14

18 Mar '14

The mask bits values were wrong for the SOC_VALUE_ENUM_SINGLE for the PCM/ADC Swap controls Reported-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/cs42l52.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index be2ba1b..ab3ac7b 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -352,7 +352,7 @@ static const char * const right_swap_text[] = { static const unsigned int swap_values[] = { 0, 1, 3 }; static const struct soc_enum adca_swap_enum = - SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 2, 1, + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 2, 3, ARRAY_SIZE(left_swap_text), left_swap_text, swap_values); @@ -361,7 +361,7 @@ static const struct snd_kcontrol_new adca_mixer = SOC_DAPM_ENUM("Route", adca_swap_enum); static const struct soc_enum pcma_swap_enum = - SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 6, 1, + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 6, 3, ARRAY_SIZE(left_swap_text), left_swap_text, swap_values); @@ -370,7 +370,7 @@ static const struct snd_kcontrol_new pcma_mixer = SOC_DAPM_ENUM("Route", pcma_swap_enum); static const struct soc_enum adcb_swap_enum = - SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 0, 1, + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 0, 3, ARRAY_SIZE(right_swap_text), right_swap_text, swap_values); @@ -379,7 +379,7 @@ static const struct snd_kcontrol_new adcb_mixer = SOC_DAPM_ENUM("Route", adcb_swap_enum); static const struct soc_enum pcmb_swap_enum = - SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 4, 1, + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 4, 3, ARRAY_SIZE(right_swap_text), right_swap_text, swap_values); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: cs42l73: Fix mask bits for SOC_VALUE_ENUM_SINGLE
by Brian Austin 18 Mar '14

18 Mar '14

The mask bits values were wrong for the SOC_VALUE_ENUM_SINGLE for the mono mix controls. Reported-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/cs42l73.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 61c0f64..4cb7652 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -325,7 +325,7 @@ static const char * const cs42l73_mono_mix_texts[] = { static const unsigned int cs42l73_mono_mix_values[] = { 0, 1, 2 }; static const struct soc_enum spk_asp_enum = - SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 6, 1, + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 6, 3, ARRAY_SIZE(cs42l73_mono_mix_texts), cs42l73_mono_mix_texts, cs42l73_mono_mix_values); @@ -343,7 +343,7 @@ static const struct snd_kcontrol_new spk_xsp_mixer = SOC_DAPM_ENUM("Route", spk_xsp_enum); static const struct soc_enum esl_asp_enum = - SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 2, 5, + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 2, 3, ARRAY_SIZE(cs42l73_mono_mix_texts), cs42l73_mono_mix_texts, cs42l73_mono_mix_values); @@ -352,7 +352,7 @@ static const struct snd_kcontrol_new esl_asp_mixer = SOC_DAPM_ENUM("Route", esl_asp_enum); static const struct soc_enum esl_xsp_enum = - SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 0, 7, + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 0, 3, ARRAY_SIZE(cs42l73_mono_mix_texts), cs42l73_mono_mix_texts, cs42l73_mono_mix_values); -- 1.7.9.5

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[alsa-devel] [PATCH v2 0/7] Audio support for Armada 370 DB
by Thomas Petazzoni 18 Mar '14

18 Mar '14

Hello, This series of patches enable audio support on the Marvell Armada 370 Development Board. Since both the I2S controller on the SoC side and the I2C audio codec are already supported by the kernel, the amount of work is fairly limited. Changes since v1: * Drop patches "sound: codec: add Device Tree binding to cs42l51" and " sound: soc: enable Kirkwood driver for mvebu platforms" since they have been applied by Mark Brown. * Set DAI format directly in the snd_soc_dai_link structure instead of separately calling snd_soc_dai_set_fmt(). Suggested by Mark Brown. * Do not call snd_soc_dapm_enable_pin(), as Mark Brown mentionned that DAPM widgets default to enabled. It allowed to entirely remove the dai_init operation. * Use devm_snd_soc_register_card() instead of snd_soc_register_card(), as suggested by Mark Brown. * Wrote a Device Tree binding document for the DT binding introduced by this driver. Requested by Mark Brown. * Use the DT to look up the controller and the codec, as suggested by Mark Brown. * Add S/PDIF support, which has been successfully tested. I've kept it as separate patches (for both the ASoC machine driver, and the Device Tree bits), so that it can be reviewed and applied separately. * Modified the commit title so that they contain "ASoC", as requested by Mark Brown. Patches 1 and 2 are to be reviewed/applied by the ASoC maintainer Mark Brown, while patches 3 to 7 are to be reviewed/applied by the ARM mvebu maintainers. Note that the audio support for Armada 370 also needs a fix to the CS42L51, which is being discussed with the author of the change that apparently introduced the problem (see discussion at http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071916.ht…). Thanks, Thomas Thomas Petazzoni (7): sound: ASoC: add ASoC board driver for Armada 370 DB sound: ASoC: add S/PDIF support to Armada 370 DB ASoC driver ARM: mvebu: add audio I2S controller to Armada 370 Device Tree ARM: mvebu: add I2C0 muxing option for Armada 370 SoC ARM: mvebu: add audio support to Armada 370 DB ARM: mvebu: enable S/PDIF audio in Armada 370 DB Device Tree ARM: mvebu: enable audio options in mvebu_defconfig .../bindings/sound/armada-370db-audio.txt | 27 ++++ arch/arm/boot/dts/armada-370-db.dts | 56 ++++++++ arch/arm/boot/dts/armada-370.dtsi | 28 ++++ arch/arm/configs/mvebu_defconfig | 5 + sound/soc/kirkwood/Kconfig | 9 ++ sound/soc/kirkwood/Makefile | 2 + sound/soc/kirkwood/armada-370-db.c | 148 +++++++++++++++++++++ 7 files changed, 275 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/armada-370db-audio.txt create mode 100644 sound/soc/kirkwood/armada-370-db.c -- 1.8.3.2

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[alsa-devel] [RFC] ASoC: samsung: don't build pcm and spdif on exynos
by Arnd Bergmann 18 Mar '14

18 Mar '14

In commit d37bdf736d9b "ASoC: samsung: Use ASoC dmaengine code where possible", Mark Brown changed the EXYNOS ASoC code to no longer use the private dma interfaces that are standing in the way of multiplatform enablement. However, in 313367e7bfa "ASoC: Samsung: Update Kconfig for I2S,SPDIF and PCM audio", two new users of that interface appeared, since it is now possible to enable SND_SOC_SAMSUNG_SMDK_SPDIF and SND_SOC_SMDK_WM8994_PCM on all Samsung SoCs including EXYNOS. This patch does a partial revert of 313367e7bfa by allowing these drivers on all samsung platforms except EXYNOS, so we can proceed with the multiplatform patches. If support for these drivers is actually needed on EXYNOS machines, the drivers should probably first be moved over to use the dmaengine code. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: Sangsu Park <sangsu4u.park(a)samsung.com> Cc: Sangbeom Kim <sbkim73(a)samsung.com> Cc: Padmavathi Venna <padma.v(a)samsung.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Padma, could you comment on this? When you did the patch that introduced the usage of these drivers on EXYNOS, did you actually need that for a particular machine that would get broken by this revert? diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 7b610a8..efc01e0 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -167,7 +167,7 @@ config SND_SOC_GONI_AQUILA_WM8994 config SND_SOC_SAMSUNG_SMDK_SPDIF tristate "SoC S/PDIF Audio support for SMDK" - depends on SND_SOC_SAMSUNG + depends on SND_SOC_SAMSUNG && !ARCH_EXYNOS select SND_SAMSUNG_SPDIF help Say Y if you want to add support for SoC S/PDIF audio on the SMDK. @@ -183,7 +183,7 @@ config SND_SOC_SMDK_WM8580_PCM config SND_SOC_SMDK_WM8994_PCM tristate "SoC PCM Audio support for WM8994 on SMDK" - depends on SND_SOC_SAMSUNG + depends on SND_SOC_SAMSUNG && !ARCH_EXYNOS depends on I2C=y select MFD_WM8994 select SND_SOC_WM8994

1 0

[alsa-devel] [PATCH] ASoC: codecs: Replace instances of rtd->codec with dai->codec
by Lars-Peter Clausen 18 Mar '14

18 Mar '14

With CODEC to CODEC links rtd->codec does not necessarily point to the driver's CODEC. CODEC drivers should always use dai->codec and never even look at the PCM runtime. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/isabelle.c | 3 +-- sound/soc/codecs/mc13783.c | 6 ++---- sound/soc/codecs/rt5640.c | 3 +-- sound/soc/codecs/sta529.c | 3 +-- sound/soc/codecs/tlv320aic31xx.c | 5 ++--- sound/soc/codecs/uda134x.c | 3 +-- sound/soc/codecs/uda1380.c | 3 +-- sound/soc/codecs/wm8580.c | 3 +-- 8 files changed, 10 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index 3e264a7..3a89ce6 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -918,8 +918,7 @@ static int isabelle_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 aif = 0; unsigned int fs_val = 0; diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 37d737e..2c59b1f 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -106,8 +106,7 @@ static int mc13783_pcm_hw_params_dac(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; unsigned int rate = params_rate(params); int i; @@ -126,8 +125,7 @@ static int mc13783_pcm_hw_params_codec(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; unsigned int rate = params_rate(params); unsigned int val; diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 13ccee4..0061ae6 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1594,8 +1594,7 @@ static int get_clk_info(int sclk, int rate) static int rt5640_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); unsigned int val_len = 0, val_clk, mask_clk; int dai_sel, pre_div, bclk_ms, frame_size; diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index a3c61d3..a40c4b0 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -193,8 +193,7 @@ static int sta529_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; int pdata, play_freq_val, record_freq_val; int bclk_to_fs_ratio; diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index d3517a9..cc1c9c2 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -753,10 +753,9 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec, static int aic31xx_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, - struct snd_soc_dai *tmp) + struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u8 data = 0; dev_dbg(codec->dev, "## %s: format %d width %d rate %d\n", diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index c94d4c1..edf27ac 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -203,8 +203,7 @@ static int uda134x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec); u8 hw_params; diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 4dadaa8..e62e707 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -566,8 +566,7 @@ static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream, static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); /* shut down WSPLL power if running from this clock */ diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 7558c83..af7ed8b 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -504,8 +504,7 @@ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8580_priv *wm8580 = snd_soc_codec_get_drvdata(codec); u16 paifa = 0; u16 paifb = 0; -- 1.8.0

4 3

[alsa-devel] [PATCH v2 0/4] ASoC: davinci: edma dmaengine PCM and mcasp preparation
by Peter Ujfalusi 18 Mar '14

18 Mar '14

Hi, This is a cutdown version of the previous [1] series containing only the ASoC part of the edma dmaengine conversion, but without the actual switch to this new edma-pcm platform for the McASP since the edma arch and dmaengine core changes are needed for it to be usable. Changes since v1: - arch and dmaengine patches has been dropped from the ASoC series - Patches has been reordered so only one patch is going to be needed to switch AM335x/AM447x to use the edma-pcm platform driver - all comments from Lars-Peter Clausen has been addressed for the edma-pcm driver - no changes to Kconfig/Makefile - do not yet build the edma-pcm driver. It will be enabled and integrated when we can do the real switch. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074269.html Regards, Peter --- Peter Ujfalusi (4): ASoC: davinci: Add edma dmaengine platform driver ASoC: davinci-mcasp: Provide correct filter_data for dmaengine for non-DT boot ASoC: davinci-mcasp: Constraint on the period and buffer size based on FIFO usage ASoC: davinci-mcasp: Assign the dma_data earlier in dai_probe callback sound/soc/davinci/davinci-mcasp.c | 59 +++++++++++++++++++++++++++++++++------ sound/soc/davinci/edma-pcm.c | 57 +++++++++++++++++++++++++++++++++++++ sound/soc/davinci/edma-pcm.h | 25 +++++++++++++++++ 3 files changed, 132 insertions(+), 9 deletions(-) create mode 100644 sound/soc/davinci/edma-pcm.c create mode 100644 sound/soc/davinci/edma-pcm.h -- 1.9.0

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[alsa-devel] [PATCH v2] ASoC: tlv320aic31xx: Don't call kfree for memory allocated by devm_kzalloc
by Axel Lin 18 Mar '14

18 Mar '14

The kfree call is not necessary, but we need to call snd_soc_unregister_codec() in remove(). Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/tlv320aic31xx.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index d3517a9..d5d6213 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1228,7 +1228,6 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, return -ENOMEM; aic31xx->regmap = devm_regmap_init_i2c(i2c, regmap_config); - if (IS_ERR(aic31xx->regmap)) { ret = PTR_ERR(aic31xx->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", @@ -1241,18 +1240,14 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, aic31xx_device_init(aic31xx); - ret = snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx, + return snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx, aic31xx_dai_driver, ARRAY_SIZE(aic31xx_dai_driver)); - - return ret; } static int aic31xx_i2c_remove(struct i2c_client *i2c) { - struct aic31xx_priv *aic31xx = dev_get_drvdata(&i2c->dev); - - kfree(aic31xx); + snd_soc_unregister_codec(&i2c->dev); return 0; } @@ -1274,7 +1269,7 @@ static struct i2c_driver aic31xx_i2c_driver = { .of_match_table = of_match_ptr(tlv320aic31xx_of_match), }, .probe = aic31xx_i2c_probe, - .remove = (aic31xx_i2c_remove), + .remove = aic31xx_i2c_remove, .id_table = aic31xx_i2c_id, }; -- 1.8.1.2

3 2

[alsa-devel] [PATCH] regmap: Fix possible sleep-in-atomic in fast io mode
by Takashi Iwai 18 Mar '14

18 Mar '14

regmap deploys the spinlock for the protection when set up in fast_io mode. But some places are still left with possible sleep in the lock, typically allocating memory with GFP_KERNEL. This patch fixes these calls by moving the allocations out of the lock. The krealloc() call in regmap_register_patch() makes it unsafe for concurrent accesses, however. But such a use case must be very rare, thus we take rather simplicity over complexity, and add a note in the function description mentioning this restriction. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Only compile tested. drivers/base/regmap/regmap.c | 36 ++++++++++++++++++++---------------- 1 file changed, 20 insertions(+), 16 deletions(-) diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c index 6a19515f8a45..e6827cee29a3 100644 --- a/drivers/base/regmap/regmap.c +++ b/drivers/base/regmap/regmap.c @@ -1520,12 +1520,12 @@ int regmap_bulk_write(struct regmap *map, unsigned int reg, const void *val, if (reg % map->reg_stride) return -EINVAL; - map->lock(map->lock_arg); /* * Some devices don't support bulk write, for * them we have a series of single write operations. */ if (!map->bus || map->use_single_rw) { + map->lock(map->lock_arg); for (i = 0; i < val_count; i++) { unsigned int ival; @@ -1554,24 +1554,25 @@ int regmap_bulk_write(struct regmap *map, unsigned int reg, const void *val, if (ret != 0) goto out; } +out: + map->unlock(map->lock_arg); } else { void *wval; wval = kmemdup(val, val_count * val_bytes, GFP_KERNEL); if (!wval) { - ret = -ENOMEM; dev_err(map->dev, "Error in memory allocation\n"); - goto out; + return -ENOMEM; } + map->lock(map->lock_arg); for (i = 0; i < val_count * val_bytes; i += val_bytes) map->format.parse_inplace(wval + i); ret = _regmap_raw_write(map, reg, wval, val_bytes * val_count); + map->unlock(map->lock_arg); kfree(wval); } -out: - map->unlock(map->lock_arg); return ret; } EXPORT_SYMBOL_GPL(regmap_bulk_write); @@ -2173,6 +2174,10 @@ EXPORT_SYMBOL_GPL(regmap_async_complete); * apply them immediately. Typically this is used to apply * corrections to be applied to the device defaults on startup, such * as the updates some vendors provide to undocumented registers. + * + * Note that the function gives no protection over concurrent access + * to map->patch and map->patch_regs. If needed, apply some lock in + * the caller side. */ int regmap_register_patch(struct regmap *map, const struct reg_default *regs, int num_regs) @@ -2185,6 +2190,16 @@ int regmap_register_patch(struct regmap *map, const struct reg_default *regs, num_regs)) return 0; + p = krealloc(map->patch, + sizeof(struct reg_default) * (map->patch_regs + num_regs), + GFP_KERNEL); + if (!p) + return -ENOMEM; + + memcpy(p + map->patch_regs, regs, num_regs * sizeof(*regs)); + map->patch = p; + map->patch_regs += num_regs; + map->lock(map->lock_arg); bypass = map->cache_bypass; @@ -2202,17 +2217,6 @@ int regmap_register_patch(struct regmap *map, const struct reg_default *regs, } } - p = krealloc(map->patch, - sizeof(struct reg_default) * (map->patch_regs + num_regs), - GFP_KERNEL); - if (p) { - memcpy(p + map->patch_regs, regs, num_regs * sizeof(*regs)); - map->patch = p; - map->patch_regs += num_regs; - } else { - ret = -ENOMEM; - } - out: map->async = false; map->cache_bypass = bypass; -- 1.9.0

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