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

[alsa-devel] [TRIVIAL PATCH V2] ASoC: atmel_ssc_dai: Correct misuse of 0x%<decimal>
by Joe Perches 04 Aug '15

04 Aug '15

Correct misuse of 0x%d in logging message. Signed-off-by: Joe Perches <joe(a)perches.com> --- Resend with later, non-broken version of Evolution email client (3.16 -> 3.16.3) sound/soc/atmel/atmel_ssc_dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 841d059..ba8def5 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -290,7 +290,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, int dir, dir_mask; int ret; - pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n", + pr_debug("atmel_ssc_startup: SSC_SR=0x%x\n", ssc_readl(ssc_p->ssc->regs, SR)); /* Enable PMC peripheral clock for this SSC */

2 1

[alsa-devel] [PATCH] ASoC: topology: fix typo in soc_tplg_kcontrol_bind_io()
by Liam Girdwood 04 Aug '15

04 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index df3ddf0..4dadb5e 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -534,7 +534,7 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr, k->put = bops[i].put; if (k->get == NULL && bops[i].id == hdr->ops.get) k->get = bops[i].get; - if (k->info == NULL && ops[i].id == hdr->ops.info) + if (k->info == NULL && bops[i].id == hdr->ops.info) k->info = bops[i].info; } -- 2.1.4

2 1

[alsa-devel] [PATCH] ASoC: nau8825: Add driver for headset chip Nuvoton 8825
by Anatol Pomozov 04 Aug '15

04 Aug '15

Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- .../devicetree/bindings/sound/nau8825.txt | 18 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8825.c | 874 +++++++++++++++++++++ sound/soc/codecs/nau8825.h | 262 ++++++ 5 files changed, 1160 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nau8825.txt create mode 100644 sound/soc/codecs/nau8825.c create mode 100644 sound/soc/codecs/nau8825.h diff --git a/Documentation/devicetree/bindings/sound/nau8825.txt b/Documentation/devicetree/bindings/sound/nau8825.txt new file mode 100644 index 0000000..c3beb37 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8825.txt @@ -0,0 +1,18 @@ +Nuvoton NAU8825 audio codec + +This device supports I2C only. + +Required properties: + - compatible : Must be "nuvoton,nau8825" + - reg : the I2C address of the device. This is either 0x1a (CSB=0) or 0x1b (CSB=1). + +Optional properties: + - nuvoton,jkdet-pullup: Pull-up JKDET pin. Boolean. + +Example: + + codec: nau8825@1a { + compatible = "nuvoton,nau8825"; + reg = <0x1a>; + nuvoton,jkdet-pullup = "true"; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7fcbc22..1a3639c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -78,6 +78,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C + select SND_SOC_NAU8825 if I2C select SND_SOC_HDMI_CODEC select SND_SOC_PCM1681 if I2C select SND_SOC_PCM1792A if SPI_MASTER @@ -892,6 +893,9 @@ config SND_SOC_MC13783 config SND_SOC_ML26124 tristate +config SND_SOC_NAU8825 + tristate + config SND_SOC_TPA6130A2 tristate "Texas Instruments TPA6130A2 headphone amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5b46c8f..4874881 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -71,6 +71,7 @@ snd-soc-max98925-objs := max98925.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o +snd-soc-nau8825-objs := nau8825.o snd-soc-hdmi-codec-objs := hdmi.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o @@ -262,6 +263,7 @@ obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o +obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c new file mode 100644 index 0000000..db24465 --- /dev/null +++ b/sound/soc/codecs/nau8825.c @@ -0,0 +1,874 @@ +/* + * NAU8825 audio codec driver + * + * Copyright 2015 Nuvoton Technology Corp. + * Copyright 2015 Google Chromium project. + * Author: Anatol Pomozov <anatol(a)chromium.org> + * + * Licensed under the GPL-2. + */ + +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <linux/input.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> + +#include "nau8825.h" + +struct nau8825 { + struct device *dev; + struct regmap *regmap; + struct snd_soc_dapm_context *dapm; + int irq; + struct snd_soc_jack *jack; + int button_pressed; + bool jkdet_pullup; +}; + +static const struct reg_default nau8825_reg_defaults[] = { + { NAU8825_REG_ENA_CTRL, 0x00ff }, + { NAU8825_REG_CLK_DIVIDER, 0x0050 }, + { NAU8825_REG_FLL1, 0x0 }, + { NAU8825_REG_FLL2, 0x3126 }, + { NAU8825_REG_FLL3, 0x0008 }, + { NAU8825_REG_FLL4, 0x0010 }, + { NAU8825_REG_FLL5, 0x0 }, + { NAU8825_REG_FLL6, 0x6000 }, + { NAU8825_REG_FLL_VCO_RSV, 0xf13c }, + { NAU8825_REG_HSD_CTRL, 0x000c }, + { NAU8825_REG_JACK_DET_CTRL, 0x0 }, + { NAU8825_REG_INTERRUPT_MASK, 0x0 }, + { NAU8825_REG_INTERRUPT_DIS_CTRL, 0xffff }, + { NAU8825_REG_SAR_CTRL, 0x0015 }, + { NAU8825_REG_KEYDET_CTRL, 0x0110 }, + { NAU8825_REG_VDET_THRESHOLD_1, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_2, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_3, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_4, 0x0 }, + { NAU8825_REG_GPIO34_CTRL, 0x0 }, + { NAU8825_REG_GPIO12_CTRL, 0x0 }, + { NAU8825_REG_TDM_CTRL, 0x0 }, + { NAU8825_REG_I2S_PCM_CTRL1, 0x000b }, + { NAU8825_REG_I2S_PCM_CTRL2, 0x8010 }, + { NAU8825_REG_LEFT_TIME_SLOT, 0x0 }, + { NAU8825_REG_RIGHT_TIME_SLOT, 0x0 }, + { NAU8825_REG_BIQ_CTRL, 0x0 }, + { NAU8825_REG_BIQ_COF1, 0x0 }, + { NAU8825_REG_BIQ_COF2, 0x0 }, + { NAU8825_REG_BIQ_COF3, 0x0 }, + { NAU8825_REG_BIQ_COF4, 0x0 }, + { NAU8825_REG_BIQ_COF5, 0x0 }, + { NAU8825_REG_BIQ_COF6, 0x0 }, + { NAU8825_REG_BIQ_COF7, 0x0 }, + { NAU8825_REG_BIQ_COF8, 0x0 }, + { NAU8825_REG_BIQ_COF9, 0x0 }, + { NAU8825_REG_BIQ_COF10, 0x0 }, + { NAU8825_REG_ADC_RATE, 0x0010 }, + { NAU8825_REG_DAC_CTRL1, 0x0001 }, + { NAU8825_REG_DAC_CTRL2, 0x0 }, + { NAU8825_REG_DAC_DGAIN_CTRL, 0x0 }, + { NAU8825_REG_ADC_DGAIN_CTRL, 0x00cf }, + { NAU8825_REG_MUTE_CTRL, 0x0 }, + { NAU8825_REG_HSVOL_CTRL, 0x0 }, + { NAU8825_REG_DACL_CTRL, 0x02cf }, + { NAU8825_REG_DACR_CTRL, 0x00cf }, + { NAU8825_REG_ADC_DRC_KNEE_IP12, 0x1486 }, + { NAU8825_REG_ADC_DRC_KNEE_IP34, 0x0f12 }, + { NAU8825_REG_ADC_DRC_SLOPES, 0x25ff }, + { NAU8825_REG_ADC_DRC_ATKDCY, 0x3457 }, + { NAU8825_REG_DAC_DRC_KNEE_IP12, 0x1486 }, + { NAU8825_REG_DAC_DRC_KNEE_IP34, 0x0f12 }, + { NAU8825_REG_DAC_DRC_SLOPES, 0x25f9 }, + { NAU8825_REG_DAC_DRC_ATKDCY, 0x3457 }, + { NAU8825_REG_IMM_MODE_CTRL, 0x0 }, + { NAU8825_REG_CLASSG_CTRL, 0x0 }, + { NAU8825_REG_OPT_EFUSE_CTRL, 0x0 }, + { NAU8825_REG_MISC_CTRL, 0x0 }, + { NAU8825_REG_BIAS_ADJ, 0x0 }, + { NAU8825_REG_TRIM_SETTINGS, 0x0 }, + { NAU8825_REG_ANALOG_CONTROL_1, 0x0 }, + { NAU8825_REG_ANALOG_CONTROL_2, 0x0 }, + { NAU8825_REG_ANALOG_ADC_1, 0x0011 }, + { NAU8825_REG_ANALOG_ADC_2, 0x0020 }, + { NAU8825_REG_RDAC, 0x0008 }, + { NAU8825_REG_MIC_BIAS, 0x0006 }, + { NAU8825_REG_BOOST, 0x0 }, + { NAU8825_REG_FEPGA, 0x0 }, + { NAU8825_REG_POWER_UP_CONTROL, 0x0 }, + { NAU8825_REG_CHARGE_PUMP, 0x0 }, +}; + +static bool nau8825_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_ENA_CTRL: + case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV: + case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL: + case NAU8825_REG_INTERRUPT_MASK ... NAU8825_REG_KEYDET_CTRL: + case NAU8825_REG_VDET_THRESHOLD_1 ... NAU8825_REG_DACR_CTRL: + case NAU8825_REG_ADC_DRC_KNEE_IP12 ... NAU8825_REG_ADC_DRC_ATKDCY: + case NAU8825_REG_DAC_DRC_KNEE_IP12 ... NAU8825_REG_DAC_DRC_ATKDCY: + case NAU8825_REG_IMM_MODE_CTRL ... NAU8825_REG_IMM_RMS_R: + case NAU8825_REG_CLASSG_CTRL ... NAU8825_REG_OPT_EFUSE_CTRL: + case NAU8825_REG_MISC_CTRL: + case NAU8825_REG_I2C_DEVICE_ID ... NAU8825_REG_SARDOUT_RAM_STATUS: + case NAU8825_REG_BIAS_ADJ: + case NAU8825_REG_TRIM_SETTINGS ... NAU8825_REG_ANALOG_CONTROL_2: + case NAU8825_REG_ANALOG_ADC_1 ... NAU8825_REG_MIC_BIAS: + case NAU8825_REG_BOOST ... NAU8825_REG_FEPGA: + case NAU8825_REG_POWER_UP_CONTROL ... NAU8825_REG_GENERAL_STATUS: + return true; + default: + return false; + } + +} + +static bool nau8825_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_RESET ... NAU8825_REG_ENA_CTRL: + case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV: + case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL: + case NAU8825_REG_INTERRUPT_MASK: + case NAU8825_REG_INT_CLR_KEY_STATUS ... NAU8825_REG_KEYDET_CTRL: + case NAU8825_REG_VDET_THRESHOLD_1 ... NAU8825_REG_DACR_CTRL: + case NAU8825_REG_ADC_DRC_KNEE_IP12 ... NAU8825_REG_ADC_DRC_ATKDCY: + case NAU8825_REG_DAC_DRC_KNEE_IP12 ... NAU8825_REG_DAC_DRC_ATKDCY: + case NAU8825_REG_IMM_MODE_CTRL: + case NAU8825_REG_CLASSG_CTRL ... NAU8825_REG_OPT_EFUSE_CTRL: + case NAU8825_REG_MISC_CTRL: + case NAU8825_REG_BIAS_ADJ: + case NAU8825_REG_TRIM_SETTINGS ... NAU8825_REG_ANALOG_CONTROL_2: + case NAU8825_REG_ANALOG_ADC_1 ... NAU8825_REG_MIC_BIAS: + case NAU8825_REG_BOOST ... NAU8825_REG_FEPGA: + case NAU8825_REG_POWER_UP_CONTROL ... NAU8825_REG_CHARGE_PUMP: + return true; + default: + return false; + } +} + +static bool nau8825_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_RESET: + case NAU8825_REG_IRQ_STATUS: + case NAU8825_REG_INT_CLR_KEY_STATUS: + case NAU8825_REG_IMM_RMS_L: + case NAU8825_REG_IMM_RMS_R: + case NAU8825_REG_I2C_DEVICE_ID: + case NAU8825_REG_SARDOUT_RAM_STATUS: + case NAU8825_REG_CHARGE_PUMP_INPUT_READ: + case NAU8825_REG_GENERAL_STATUS: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_MINMAX_MUTE(adc_vol_tlv, -10300, 2400); +static const DECLARE_TLV_DB_MINMAX_MUTE(sidetone_vol_tlv, -4200, 0); +static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -5400, 0); +static const DECLARE_TLV_DB_MINMAX(fepga_gain_tlv, -100, 3600); + +static const struct snd_kcontrol_new nau8825_snd_controls[] = { + SOC_SINGLE_TLV("MIC Volume", NAU8825_REG_ADC_DGAIN_CTRL, + 0, 0xff, 0, adc_vol_tlv), + SOC_DOUBLE_TLV("HP Sidetone Volume", NAU8825_REG_ADC_DGAIN_CTRL, + 12, 8, 0x0f, 0, sidetone_vol_tlv), + SOC_DOUBLE_TLV("HP Volume", NAU8825_REG_HSVOL_CTRL, + 6, 0, 0x3f, 1, dac_vol_tlv), + SOC_SINGLE_TLV("Frontend PGA Gain", NAU8825_REG_POWER_UP_CONTROL, + 8, 37, 0, fepga_gain_tlv), +}; + +static int nau8825_charge_pump_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + struct regmap *regmap = nau8825->regmap; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + regmap_update_bits(regmap, NAU8825_REG_CHARGE_PUMP, + NAU8825_JAMNODCLOW, NAU8825_JAMNODCLOW); + } else { + regmap_update_bits(regmap, NAU8825_REG_CHARGE_PUMP, + NAU8825_JAMNODCLOW, 0); + } + + return 0; +} + +static int nau8825_output_driver_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + struct regmap *regmap = nau8825->regmap; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + /* Power Up L and R Output Drivers */ + regmap_update_bits(regmap, NAU8825_REG_POWER_UP_CONTROL, 0x3c, + 0x3c); + /* Power up Main Output Drivers (The main driver must be turned + on after the predriver to avoid pops) */ + regmap_update_bits(regmap, NAU8825_REG_POWER_UP_CONTROL, 0x3, + 0x3); + } else { + /* Power Down L and R Output Drivers */ + regmap_update_bits(regmap, NAU8825_REG_POWER_UP_CONTROL, 0x3f, + 0x0); + } + + return 0; +} + +/* DAC Mux 0x33[9] and 0x34[9] */ +static const char * const nau8825_dac_src[] = { + "DACL", "DACR", +}; + +static SOC_ENUM_SINGLE_DECL( + nau8825_dacl_enum, NAU8825_REG_DACL_CTRL, + NAU8825_DACL_CH_SEL_SFT, nau8825_dac_src); + +static SOC_ENUM_SINGLE_DECL( + nau8825_dacr_enum, NAU8825_REG_DACR_CTRL, + NAU8825_DACR_CH_SEL_SFT, nau8825_dac_src); + +static const struct snd_kcontrol_new nau8825_dacl_mux = + SOC_DAPM_ENUM("DACL Source", nau8825_dacl_enum); + +static const struct snd_kcontrol_new nau8825_dacr_mux = + SOC_DAPM_ENUM("DACR Source", nau8825_dacr_enum); + +static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("MIC"), + SND_SOC_DAPM_MICBIAS("MICBIAS", NAU8825_REG_MIC_BIAS, 8, 0), + + SND_SOC_DAPM_PGA("Frontend PGA", NAU8825_REG_POWER_UP_CONTROL, 14, 0, + NULL, 0), + + SND_SOC_DAPM_ADC("ADC", NULL, NAU8825_REG_ENA_CTRL, 8, 0), + SND_SOC_DAPM_SUPPLY("ADC Clock", NAU8825_REG_ENA_CTRL, 7, 0, NULL, 0), + + /* ADC for button press detection */ + SND_SOC_DAPM_ADC("SAR", NULL, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_ADC_EN_SFT, 0), + + SND_SOC_DAPM_DAC("ADACL", NULL, NAU8825_REG_RDAC, 12, 0), + SND_SOC_DAPM_DAC("ADACR", NULL, NAU8825_REG_RDAC, 13, 0), + SND_SOC_DAPM_SUPPLY("ADACL Clock", NAU8825_REG_RDAC, 8, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADACR Clock", NAU8825_REG_RDAC, 9, 0, NULL, 0), + + SND_SOC_DAPM_DAC("DDACR", NULL, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACR_SFT, 0), + SND_SOC_DAPM_DAC("DDACL", NULL, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACL_SFT, 0), + SND_SOC_DAPM_SUPPLY("DDAC Clock", NAU8825_REG_ENA_CTRL, 6, 0, NULL, 0), + + SND_SOC_DAPM_MUX("DACL Mux", SND_SOC_NOPM, 0, 0, + &nau8825_dacl_mux), + SND_SOC_DAPM_MUX("DACR Mux", SND_SOC_NOPM, 0, 0, + &nau8825_dacr_mux), + + SND_SOC_DAPM_PGA("HP amp L", NAU8825_REG_CLASSG_CTRL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP amp R", NAU8825_REG_CLASSG_CTRL, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HP amp power", NAU8825_REG_CLASSG_CTRL, 0, 0, NULL, + 0), + + SND_SOC_DAPM_SUPPLY("Change Pump", NAU8825_REG_CHARGE_PUMP, 5, 0, + nau8825_charge_pump_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_SUPPLY("DACL Power", NAU8825_REG_CHARGE_PUMP, 8, 1, NULL, + 0), + SND_SOC_DAPM_SUPPLY("DACR Power", NAU8825_REG_CHARGE_PUMP, 9, 1, NULL, + 0), + + SND_SOC_DAPM_OUT_DRV_E("Output Driver", SND_SOC_NOPM, 0, 0, NULL, 0, + nau8825_output_driver_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route nau8825_dapm_routes[] = { + {"Frontend PGA", NULL, "MIC"}, + {"ADC", NULL, "Frontend PGA"}, + {"ADC", NULL, "ADC Clock"}, + {"Capture", NULL, "ADC"}, + + {"DDACL", NULL, "Playback"}, + {"DDACR", NULL, "Playback"}, + {"DDACL", NULL, "DDAC Clock"}, + {"DDACR", NULL, "DDAC Clock"}, + {"DDACL", NULL, "DACL Power"}, + {"DDACR", NULL, "DACR Power"}, + {"DACL Mux", "DACL", "DDACL"}, + {"DACL Mux", "DACR", "DDACR"}, + {"DACR Mux", "DACL", "DDACL"}, + {"DACR Mux", "DACR", "DDACR"}, + {"HP amp L", NULL, "Change Pump"}, + {"HP amp R", NULL, "Change Pump"}, + {"HP amp L", NULL, "DACL Mux"}, + {"HP amp R", NULL, "DACR Mux"}, + {"HP amp L", NULL, "HP amp power"}, + {"HP amp R", NULL, "HP amp power"}, + {"ADACL", NULL, "HP amp L"}, + {"ADACR", NULL, "HP amp R"}, + {"ADACL", NULL, "ADACL Clock"}, + {"ADACR", NULL, "ADACR Clock"}, + {"Output Driver", NULL, "ADACL"}, + {"Output Driver", NULL, "ADACR"}, + {"HPOL", NULL, "Output Driver"}, + {"HPOR", NULL, "Output Driver"}, +}; + +static int nau8825_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 nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0; + + switch (params_width(params)) { + case 16: + val_len |= NAU8825_I2S_DL_16; + break; + case 20: + val_len |= NAU8825_I2S_DL_20; + break; + case 24: + val_len |= NAU8825_I2S_DL_24; + break; + case 32: + val_len |= NAU8825_I2S_DL_32; + break; + default: + return -EINVAL; + } + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1, + NAU8825_I2S_DL_MASK, val_len); + + return 0; +} + +static int nau8825_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + int reg_val; + + reg_val = mute ? NAU8825_HP_MUTE : 0; + regmap_update_bits(nau8825->regmap, NAU8825_REG_HSVOL_CTRL, + NAU8825_HP_MUTE, reg_val); + + return 0; +} + +static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + unsigned int ctrl1_val = 0, ctrl2_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl2_val |= NAU8825_I2S_MS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1_val |= NAU8825_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1_val |= NAU8825_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1_val |= NAU8825_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl1_val |= NAU8825_I2S_DF_RIGTH; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1_val |= NAU8825_I2S_DF_PCM_AB; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl1_val |= NAU8825_I2S_DF_PCM_AB; + ctrl1_val |= NAU8825_I2S_PCMB_EN; + break; + default: + return -EINVAL; + } + + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1, + NAU8825_I2S_DL_MASK | NAU8825_I2S_DF_MASK | + NAU8825_I2S_BP_MASK | NAU8825_I2S_PCMB_MASK, + ctrl1_val); + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, ctrl2_val); + + return 0; +} + +static const struct snd_soc_dai_ops nau8825_dai_ops = { + .hw_params = nau8825_hw_params, + .digital_mute = nau8825_mute, + .set_fmt = nau8825_set_dai_fmt, +}; + +#define NAU8825_RATES SNDRV_PCM_RATE_8000_192000 +#define NAU8825_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver nau8825_dai = { + .name = "nau8825-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8825_RATES, + .formats = NAU8825_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = NAU8825_RATES, + .formats = NAU8825_FORMATS, + }, + .ops = &nau8825_dai_ops, +}; + +/** + * nau8825_enable_jack_detect - Specify a jack for event reporting + * + * @component: component to register the jack with + * @jack: jack to use to report headset and button events on + * + * After this function has been called the headset insert/remove and button + * events will be routed to the given jack. Jack can be null to stop + * reporting. + */ +int nau8825_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + + nau8825->jack = jack; + + return 0; +} +EXPORT_SYMBOL_GPL(nau8825_enable_jack_detect); + + +static bool nau8825_is_jack_inserted(struct regmap *regmap) +{ + int status; + + regmap_read(regmap, NAU8825_REG_I2C_DEVICE_ID, &status); + return !(status & NAU8825_GPIO2JD1); +} + +static void nau8825_restart_jack_detection(struct regmap *regmap) +{ + /* this will restart the entire jack detection process including MIC/GND + * switching and create interrupts. We have to go from 0 to 1 and back + * to 0 to restart. + */ + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_RESTART, NAU8825_JACK_DET_RESTART); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_RESTART, 0); +} + +static void nau8825_eject_jack(struct nau8825 *nau8825) +{ + struct snd_soc_dapm_context *dapm = nau8825->dapm; + struct regmap *regmap = nau8825->regmap; + + snd_soc_dapm_disable_pin(dapm, "SAR"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + /* Detach 2kOhm Resistors from MICBIAS to MICGND1/2 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, 0); + /* ground HPL/HPR, MICGRND1/2 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0xf, 0xf); + + snd_soc_dapm_sync(dapm); +} + +static int nau8825_button_decode(int value) +{ + int buttons; + + if (value & BIT(0)) + buttons |= SND_JACK_BTN_0; + if (value & BIT(1)) + buttons |= SND_JACK_BTN_1; + if (value & BIT(2)) + buttons |= SND_JACK_BTN_2; + if (value & BIT(3)) + buttons |= SND_JACK_BTN_3; + + return buttons; +} + +static int nau8825_jack_insert(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + struct snd_soc_dapm_context *dapm = nau8825->dapm; + int jack_status_reg, mic_detected; + int type; + + regmap_read(regmap, NAU8825_REG_GENERAL_STATUS, &jack_status_reg); + mic_detected = (jack_status_reg >> 10) & 3; + + switch (mic_detected) { + case 0: + /* no mic */ + type = SND_JACK_HEADPHONE; + break; + case 1: + dev_dbg(nau8825->dev, "OMTP (micgnd1) mic connected\n"); + type = SND_JACK_HEADSET; + + /* Unground MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 3 << 2, + 1 << 2); + /* Unground HPL/R */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0x3, 0); + /* Attach 2kOhm Resistor from MICBIAS to MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, + NAU8825_MICBIAS_JKR2); + /* Attach SARADC to MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_INPUT_MASK, + NAU8825_SAR_INPUT_JKR2); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin(dapm, "SAR"); + snd_soc_dapm_sync(dapm); + break; + case 2: + case 3: + dev_dbg(nau8825->dev, "CTIA (micgnd2) mic connected\n"); + type = SND_JACK_HEADSET; + + /* Unground MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 3 << 2, + 2 << 2); + /* Unground HPL/R */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0x3, 0); + /* Attach 2kOhm Resistor from MICBIAS to MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, + NAU8825_MICBIAS_JKSLV); + /* Attach SARADC to MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_INPUT_MASK, + NAU8825_SAR_INPUT_JKSLV); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin(dapm, "SAR"); + snd_soc_dapm_sync(dapm); + break; + } + + return type; +} + +#define NAU8825_BUTTONS (SND_JACK_BTN_0 | SND_JACK_BTN_1 | \ + SND_JACK_BTN_2 | SND_JACK_BTN_3) + +static irqreturn_t nau8825_interrupt(int irq, void *data) +{ + struct nau8825 *nau8825 = (struct nau8825 *)data; + struct regmap *regmap = nau8825->regmap; + int active_irq, clear_irq = 0, event = 0, event_mask = 0; + + regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq); + + if ((active_irq & NAU8825_JACK_EJECTION_IRQ_MASK) == + NAU8825_JACK_EJECTION_DETECTED) { + + nau8825_eject_jack(nau8825); + event_mask |= SND_JACK_HEADSET; + clear_irq = NAU8825_JACK_EJECTION_IRQ_MASK; + } else if (active_irq & NAU8825_KEY_SHORT_PRESS_IRQ) { + int key_status; + + regmap_read(regmap, NAU8825_REG_INT_CLR_KEY_STATUS, + &key_status); + + /* upper 8 bits of the register are for short pressed keys, + lower 8 bits - for long pressed buttons */ + nau8825->button_pressed = nau8825_button_decode( + key_status >> 8); + + event |= nau8825->button_pressed; + event_mask |= NAU8825_BUTTONS; + clear_irq = NAU8825_KEY_SHORT_PRESS_IRQ; + } else if (active_irq & NAU8825_KEY_RELEASE_IRQ) { + event_mask = NAU8825_BUTTONS; + clear_irq = NAU8825_KEY_RELEASE_IRQ; + } else if (active_irq & NAU8825_HEADSET_COMPLETION_IRQ) { + if (nau8825_is_jack_inserted(regmap)) { + event |= nau8825_jack_insert(nau8825); + } else { + dev_warn(nau8825->dev, "Headset completion IRQ fired but no headset connected\n"); + nau8825_eject_jack(nau8825); + } + + event_mask |= SND_JACK_HEADSET; + clear_irq = NAU8825_HEADSET_COMPLETION_IRQ; + } + + if (!clear_irq) + clear_irq = active_irq; + /* clears the rightmost interruption */ + regmap_write(regmap, NAU8825_REG_INT_CLR_KEY_STATUS, clear_irq); + + if (event_mask) + snd_soc_jack_report(nau8825->jack, event, event_mask); + + return IRQ_HANDLED; +} + +static void nau8825_setup_buttons(struct regmap *regmap) +{ + /* Setup Button Detect (Debounce, number of buttons, and Hysteresis) */ + regmap_write(regmap, NAU8825_REG_KEYDET_CTRL, 0x7311); + + /* Setup 4 buttons impedane according to Android specification + * https://source.android.com/accessories/headset-spec.html + * Button 0 - 0-70 Ohm + * Button 1 - 110-180 Ohm + * Button 2 - 210-290 Ohm + * Button 3 - 360-680 Ohm + */ + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_1, 0x0f1f); + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_2, 0x325f); +} + +static void nau8825_init_regs(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + int jkdet_pullup_val; + + /* Power down DACs for power savings */ + regmap_write(regmap, NAU8825_REG_CHARGE_PUMP, 0x0300); + /* DAC Zero Crossing Enable */ + regmap_write(regmap, NAU8825_REG_MUTE_CTRL, 0x1000); + /* VMID Enable and Tieoff */ + regmap_write(regmap, NAU8825_REG_BIAS_ADJ, 0x0060); + /* Analog Bias Enable, Disable Boost Driver, + Automatic Short circuit protection enable */ + regmap_write(regmap, NAU8825_REG_BOOST, 0x3140); + /* Ground HP Outputs[1:0], + Enable Automatic Mic/Gnd switching reading on insert interrupt[6] */ + regmap_write(regmap, NAU8825_REG_HSD_CTRL, 0x004f); + + jkdet_pullup_val = nau8825->jkdet_pullup ? + NAU8825_JKDET_PULL_SELECT : 0; + regmap_update_bits(regmap, NAU8825_REG_GPIO12_CTRL, + NAU8825_JKDET_PULL_SELECT, jkdet_pullup_val); + + regmap_write(regmap, NAU8825_REG_JACK_DET_CTRL, 0x01e0); + /* IRQ Output Enable, Mask Insert Interrupt */ + regmap_write(regmap, NAU8825_REG_INTERRUPT_MASK, 0x0801); + + /* Setup SAR ADC */ + regmap_write(regmap, NAU8825_REG_SAR_CTRL, 0x0280); + nau8825_setup_buttons(regmap); + + /* Setup ADC x128 OSR */ + regmap_write(regmap, NAU8825_REG_ADC_RATE, 0x0002); + /* Setup DAC x128 OSR */ + regmap_write(regmap, NAU8825_REG_DAC_CTRL1, 0x0082); + /* DAC bias settings */ + regmap_write(regmap, NAU8825_REG_ANALOG_CONTROL_2, 0x1003); + /* ADC Bias Settings */ + regmap_write(regmap, NAU8825_REG_ANALOG_ADC_2, 0x0260); + /* Maximize Jack Insert Debounce */ + regmap_write(regmap, NAU8825_REG_JACK_DET_CTRL, 0x00e0); + + /* Setup clocks */ + regmap_write(regmap, NAU8825_REG_ENA_CTRL, 0x0416); + + /* Mask jack insert interruption */ + regmap_write(regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0x0011); +} + +static const struct regmap_config nau8825_regmap_config = { + .val_bits = 16, + .reg_bits = 16, + + .max_register = NAU8825_REG_MAX, + .readable_reg = nau8825_readable_reg, + .writeable_reg = nau8825_writeable_reg, + .volatile_reg = nau8825_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8825_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(nau8825_reg_defaults), +}; + +static int nau8825_codec_probe(struct snd_soc_codec *codec) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + nau8825->dapm = &codec->dapm; + + /* The interrupt clock is gated by x1[10:8], + * one of them needs to be enabled all the time for + * interrupts to happen. */ + snd_soc_dapm_force_enable_pin(&codec->dapm, "DDACR"); + snd_soc_dapm_sync(&codec->dapm); + + /* Chip needs one FSCLK cycle in order to generate interrupts, + * as we cannot guarantee one will be provided by the system. Turning + * master mode on then off enables us to generate that FSCLK cycle + * with a minimum of contention on the clock bus. + * Enables master mode and correctly divides BCLK and FSCLK from MCLK. + */ + regmap_write(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, 0x2019); + regmap_write(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, 0x0010); + + nau8825_restart_jack_detection(nau8825->regmap); + + return 0; +} + +static struct snd_soc_codec_driver nau8825_codec_driver = { + .probe = nau8825_codec_probe, + + .controls = nau8825_snd_controls, + .num_controls = ARRAY_SIZE(nau8825_snd_controls), + .dapm_widgets = nau8825_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8825_dapm_widgets), + .dapm_routes = nau8825_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8825_dapm_routes), +}; + +static void nau8825_reset_chip(struct regmap *regmap) +{ + regmap_write(regmap, NAU8825_REG_RESET, 0x00); + regmap_write(regmap, NAU8825_REG_RESET, 0x00); +} + +static int nau8825_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct nau8825 *nau8825; + int ret, value; + + nau8825 = devm_kzalloc(dev, sizeof(*nau8825), GFP_KERNEL); + if (!nau8825) + return -ENOMEM; + + i2c_set_clientdata(i2c, nau8825); + + nau8825->regmap = devm_regmap_init_i2c(i2c, &nau8825_regmap_config); + if (IS_ERR(nau8825->regmap)) + return PTR_ERR(nau8825->regmap); + nau8825->irq = i2c->irq; + nau8825->dev = dev; + + nau8825->jkdet_pullup = device_property_read_bool(dev, + "nuvoton,jkdet-pullup"); + + nau8825_reset_chip(nau8825->regmap); + ret = regmap_read(nau8825->regmap, NAU8825_REG_I2C_DEVICE_ID, &value); + if (ret < 0) { + dev_err(dev, "Failed to read device id from the NAU8825: %d\n", + ret); + return ret; + } + if ((value & NAU8825_SOFTWARE_ID_MASK) != + NAU8825_SOFTWARE_ID_NAU8825) { + dev_err(dev, "Not a NAU8825 chip\n"); + return -ENODEV; + } + + snd_soc_register_codec(&i2c->dev, &nau8825_codec_driver, + &nau8825_dai, 1); + + nau8825_init_regs(nau8825); + + if (nau8825->irq) { + int ret = devm_request_threaded_irq(dev, nau8825->irq, NULL, + nau8825_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "nau8825", nau8825); + + if (ret) { + dev_err(dev, "Cannot request irq %d (%d)\n", + nau8825->irq, ret); + return ret; + } + } + + return 0; +} + +static int nau8825_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id nau8825_i2c_ids[] = { + { "nau8825", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, nau8825_i2c_ids); + +static struct i2c_driver nau8825_driver = { + .driver = { + .name = "nau8825", + .owner = THIS_MODULE, + }, + .probe = nau8825_i2c_probe, + .remove = nau8825_i2c_remove, + .id_table = nau8825_i2c_ids, +}; +module_i2c_driver(nau8825_driver); + +MODULE_DESCRIPTION("ASoC nau8825 driver"); +MODULE_AUTHOR("Anatol Pomozov <anatol(a)chromium.org>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h new file mode 100644 index 0000000..4516d64 --- /dev/null +++ b/sound/soc/codecs/nau8825.h @@ -0,0 +1,262 @@ +/* + * NAU8825 ALSA SoC audio driver + * + * Copyright 2015 Google Inc. + * Author: Anatol Pomozov <anatol.pomozov(a)chrominium.org> + * + * 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. + */ + +#ifndef __NAU8825_H__ +#define __NAU8825_H__ + +#define NAU8825_REG_RESET 0x00 +#define NAU8825_REG_ENA_CTRL 0x01 +#define NAU8825_REG_CLK_DIVIDER 0x03 +#define NAU8825_REG_FLL1 0x04 +#define NAU8825_REG_FLL2 0x05 +#define NAU8825_REG_FLL3 0x06 +#define NAU8825_REG_FLL4 0x07 +#define NAU8825_REG_FLL5 0x08 +#define NAU8825_REG_FLL6 0x09 +#define NAU8825_REG_FLL_VCO_RSV 0x0a +#define NAU8825_REG_HSD_CTRL 0x0c +#define NAU8825_REG_JACK_DET_CTRL 0x0d +#define NAU8825_REG_INTERRUPT_MASK 0x0f +#define NAU8825_REG_IRQ_STATUS 0x10 +#define NAU8825_REG_INT_CLR_KEY_STATUS 0x11 +#define NAU8825_REG_INTERRUPT_DIS_CTRL 0x12 +#define NAU8825_REG_SAR_CTRL 0x13 +#define NAU8825_REG_KEYDET_CTRL 0x14 +#define NAU8825_REG_VDET_THRESHOLD_1 0x15 +#define NAU8825_REG_VDET_THRESHOLD_2 0x16 +#define NAU8825_REG_VDET_THRESHOLD_3 0x17 +#define NAU8825_REG_VDET_THRESHOLD_4 0x18 +#define NAU8825_REG_GPIO34_CTRL 0x19 +#define NAU8825_REG_GPIO12_CTRL 0x1a +#define NAU8825_REG_TDM_CTRL 0x1b +#define NAU8825_REG_I2S_PCM_CTRL1 0x1c +#define NAU8825_REG_I2S_PCM_CTRL2 0x1d +#define NAU8825_REG_LEFT_TIME_SLOT 0x1e +#define NAU8825_REG_RIGHT_TIME_SLOT 0x1f +#define NAU8825_REG_BIQ_CTRL 0x20 +#define NAU8825_REG_BIQ_COF1 0x21 +#define NAU8825_REG_BIQ_COF2 0x22 +#define NAU8825_REG_BIQ_COF3 0x23 +#define NAU8825_REG_BIQ_COF4 0x24 +#define NAU8825_REG_BIQ_COF5 0x25 +#define NAU8825_REG_BIQ_COF6 0x26 +#define NAU8825_REG_BIQ_COF7 0x27 +#define NAU8825_REG_BIQ_COF8 0x28 +#define NAU8825_REG_BIQ_COF9 0x29 +#define NAU8825_REG_BIQ_COF10 0x2a +#define NAU8825_REG_ADC_RATE 0x2b +#define NAU8825_REG_DAC_CTRL1 0x2c +#define NAU8825_REG_DAC_CTRL2 0x2d +#define NAU8825_REG_DAC_DGAIN_CTRL 0x2f +#define NAU8825_REG_ADC_DGAIN_CTRL 0x30 +#define NAU8825_REG_MUTE_CTRL 0x31 +#define NAU8825_REG_HSVOL_CTRL 0x32 +#define NAU8825_REG_DACL_CTRL 0x33 +#define NAU8825_REG_DACR_CTRL 0x34 +#define NAU8825_REG_ADC_DRC_KNEE_IP12 0x38 +#define NAU8825_REG_ADC_DRC_KNEE_IP34 0x39 +#define NAU8825_REG_ADC_DRC_SLOPES 0x3a +#define NAU8825_REG_ADC_DRC_ATKDCY 0x3b +#define NAU8825_REG_DAC_DRC_KNEE_IP12 0x45 +#define NAU8825_REG_DAC_DRC_KNEE_IP34 0x46 +#define NAU8825_REG_DAC_DRC_SLOPES 0x47 +#define NAU8825_REG_DAC_DRC_ATKDCY 0x48 +#define NAU8825_REG_IMM_MODE_CTRL 0x4c +#define NAU8825_REG_IMM_RMS_L 0x4d +#define NAU8825_REG_IMM_RMS_R 0x4e +#define NAU8825_REG_CLASSG_CTRL 0x50 +#define NAU8825_REG_OPT_EFUSE_CTRL 0x51 +#define NAU8825_REG_MISC_CTRL 0x55 +#define NAU8825_REG_I2C_DEVICE_ID 0x58 +#define NAU8825_REG_SARDOUT_RAM_STATUS 0x59 +#define NAU8825_REG_BIAS_ADJ 0x66 +#define NAU8825_REG_TRIM_SETTINGS 0x68 +#define NAU8825_REG_ANALOG_CONTROL_1 0x69 +#define NAU8825_REG_ANALOG_CONTROL_2 0x6a +#define NAU8825_REG_ANALOG_ADC_1 0x71 +#define NAU8825_REG_ANALOG_ADC_2 0x72 +#define NAU8825_REG_RDAC 0x73 +#define NAU8825_REG_MIC_BIAS 0x74 +#define NAU8825_REG_BOOST 0x76 +#define NAU8825_REG_FEPGA 0x77 +#define NAU8825_REG_POWER_UP_CONTROL 0x7f +#define NAU8825_REG_CHARGE_PUMP 0x80 +#define NAU8825_REG_CHARGE_PUMP_INPUT_READ 0x81 +#define NAU8825_REG_GENERAL_STATUS 0x82 +#define NAU8825_REG_MAX NAU8825_REG_GENERAL_STATUS + +/* ENA_CTRL (0x1) */ +#define NAU8825_ENABLE_DACR_SFT 10 +#define NAU8825_ENABLE_DACL_SFT 9 +#define NAU8825_ENABLE_ADC_SFT 8 +#define NAU8825_ENABLE_SAR_SFT 1 + +/* CLK_DIVIDER (0x3) */ +#define NAU8825_CLK_SRC_SFT 15 +#define NAU8825_CLK_SRC_MASK (1 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_SRC_VCO (1 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_SRC_MCLK (0 << NAU8825_CLK_SRC_SFT) + +/* HSD_CTRL (0xc) */ +#define NAU8825_HSD_AUTO_MODE (1 << 6) +#define NAU8825_SPKR_DWN1R_SHORT (1 << 1) +#define NAU8825_SPKR_DWN1L_SHORT (1 << 0) + +/* JACK_DET_CTRL (0xd) */ +#define NAU8825_JACK_DET_RESTART (1 << 9) + +/* IRQ_STATUS (0x10) */ +#define NAU8825_HEADSET_COMPLETION_IRQ (1 << 10) +#define NAU8825_SHORT_CIRCUIT_IRQ (1 << 9) +#define NAU8825_IMPEDANCE_MEAS_IRQ (1 << 8) +#define NAU8825_KEY_IRQ_MASK (0x7 << 5) +#define NAU8825_KEY_RELEASE_IRQ (1 << 7) +#define NAU8825_KEY_LONG_PRESS_IRQ (1 << 6) +#define NAU8825_KEY_SHORT_PRESS_IRQ (1 << 5) +#define NAU8825_MIC_DETECTION_IRQ (1 << 4) +#define NAU8825_JACK_EJECTION_IRQ_MASK (3 << 2) +#define NAU8825_JACK_EJECTION_DETECTED (1 << 2) +#define NAU8825_JACK_INSERTION_IRQ_MASK (3 << 0) +#define NAU8825_JACK_INSERTION_DETECTED (1 << 0) + +/* SAR_CTRL (0x13) */ +#define NAU8825_SAR_ADC_EN_SFT 12 +#define NAU8825_SAR_ADC_EN (1 << NAU8825_SAR_ADC_EN_SFT) +#define NAU8825_SAR_INPUT_MASK (1 << 11) +#define NAU8825_SAR_INPUT_JKSLV (1 << 11) +#define NAU8825_SAR_INPUT_JKR2 (0 << 11) + +/* KEYDET_CTRL (0x14) */ +#define NAU8825_KEYDET_LEVELS_NR_SFT 8 +#define NAU8825_KEYDET_LEVELS_NR_MASK (0x7 << 8) + +/* GPIO12_CTRL (0x1a) */ +/* 0 - pull down, 1 - pull up */ +#define NAU8825_JKDET_PULL_SELECT (1 << 11) + +/* I2S_PCM_CTRL1 (0x1c) */ +#define NAU8825_I2S_BP_SFT 7 +#define NAU8825_I2S_BP_MASK (1 << NAU8825_I2S_BP_SFT) +#define NAU8825_I2S_BP_INV (1 << NAU8825_I2S_BP_SFT) +#define NAU8825_I2S_PCMB_SFT 6 +#define NAU8825_I2S_PCMB_MASK (1 << NAU8825_I2S_PCMB_SFT) +#define NAU8825_I2S_PCMB_EN (1 << NAU8825_I2S_PCMB_SFT) +#define NAU8825_I2S_DL_SFT 2 +#define NAU8825_I2S_DL_MASK (0x3 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_16 (0 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_20 (1 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_24 (2 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_32 (3 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DF_SFT 0 +#define NAU8825_I2S_DF_MASK (0x3 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_RIGTH (0 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_LEFT (1 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_I2S (2 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_PCM_AB (3 << NAU8825_I2S_DF_SFT) + +/* I2S_PCM_CTRL2 (0x1d) */ +#define NAU8825_I2S_MS_SFT 3 +#define NAU8825_I2S_MS_MASK (1 << NAU8825_I2S_MS_SFT) +#define NAU8825_I2S_MS_MASTER (1 << NAU8825_I2S_MS_SFT) +#define NAU8825_I2S_MS_SLAVE (0 << NAU8825_I2S_MS_SFT) + +/* ADC_RATE (0x2b) */ +#define NAU8825_ADC_SINC4 (1 << 4) +#define NAU8825_ADC_SINC_DOWN_MASK 3 +#define NAU8825_ADC_SINC_DOWN_32 0 +#define NAU8825_ADC_SINC_DOWN_64 1 +#define NAU8825_ADC_SINC_DOWN_128 2 +#define NAU8825_ADC_SINC_DOWN_256 3 + +/* DAC_CTRL1 (0x2c) */ +#define NAU8825_DAC_CLIP_OFF (1 << 7) +#define NAU8825_DAC_OVERSAMPLE_MASK 7 +#define NAU8825_DAC_OVERSAMPLE_64 0 +#define NAU8825_DAC_OVERSAMPLE_256 1 +#define NAU8825_DAC_OVERSAMPLE_128 2 +#define NAU8825_DAC_OVERSAMPLE_32 4 + +/* MUTE_CTRL (0x31) */ +#define NAU8825_DAC_ZERO_CROSSING_EN (1 << 9) +#define NAU8825_DAC_SOFT_MUTE (1 << 9) + +/* HSVOL_CTRL (0x32) */ +#define NAU8825_HP_MUTE (1 << 15) + +/* DACL_CTRL (0x33) */ +#define NAU8825_DACL_CH_SEL_SFT 9 + +/* DACR_CTRL (0x34) */ +#define NAU8825_DACR_CH_SEL_SFT 9 + +/* I2C_DEVICE_ID (0x58) */ +#define NAU8825_GPIO2JD1 (1 << 7) +#define NAU8825_SOFTWARE_ID_MASK 0x3 +#define NAU8825_SOFTWARE_ID_NAU8825 0x0 + +/* BIAS_ADJ (0x66) */ +#define NAU8825_BIAS_VMID (1 << 6) +#define NAU8825_BIAS_VMID_SEL_MASK (3 << 4) +#define NAU8825_BIAS_VMID_SEL_OPEN (0 << 4) +#define NAU8825_BIAS_VMID_SEL_25KOHM (1 << 4) +#define NAU8825_BIAS_VMID_SEL_125KOHM (2 << 4) +#define NAU8825_BIAS_VMID_SEL_2_5KOHM (3 << 4) + +/* ANALOG_CONTROL_2 (0x6a) */ +#define NAU8825_HP_NON_CLASSG_CURRENT_2xADJ (1 << 12) +#define NAU8825_DAC_CAPACITOR_MSB (1 << 1) +#define NAU8825_DAC_CAPACITOR_LSB (1 << 0) + +/* ANALOG_ADC_2 (0x72) */ +#define NAU8825_ADC_VREFSEL_MASK (0x3 << 8) +#define NAU8825_ADC_VREFSEL_ANALOG (0 << 8) +#define NAU8825_ADC_VREFSEL_VMID (1 << 8) +#define NAU8825_ADC_VREFSEL_VMID_PLUS_0_5DB (2 << 8) +#define NAU8825_ADC_VREFSEL_VMID_PLUS_1DB (3 << 8) +#define NAU8825_POWERUP_ADCL (1 << 6) + +/* MIC_BIAS (0x74) */ +#define NAU8825_MICBIAS_JKSLV (1 << 14) +#define NAU8825_MICBIAS_JKR2 (1 << 12) +#define NAU8825_MICBIAS_POWERUP_SFT 8 + +/* BOOST (0x76) */ +#define NAU8825_PRECHARGE_DIS (1 << 13) +#define NAU8825_GLOBAL_BIAS_EN (1 << 12) +#define NAU8825_HP_BOOST_DIS (1 << 12) +#define NAU8825_SHORT_SHUTDOWN_EN (1 << 6) + +/* POWER_UP_CONTROL (0x7f) */ +#define NAU8825_POWERUP_HP_DRV_L (1 << 0) +#define NAU8825_POWERUP_HP_DRV_R (1 << 1) +#define NAU8825_POWERUP_DRV_IN_L (1 << 2) +#define NAU8825_POWERUP_DRV_IN_R (1 << 3) +#define NAU8825_POWERUP_INTEGR_L (1 << 4) +#define NAU8825_POWERUP_INTEGR_R (1 << 5) + +/* CHARGE_PUMP_AND_POWER_DOWN_CONTROL (0x80) */ +#define NAU8825_JAMNODCLOW (1 << 10) +#define NAU8825_POWER_DOWN_DACR (1 << 9) +#define NAU8825_POWER_DOWN_DACL (1 << 8) +#define NAU8825_CHANRGE_PUMP_EN (1 << 5) + + +/* System Clock Source */ +enum { + NAU8825_CLK_MCLK = 0, + NAU8825_CLK_INTERNAL, +}; + +int nau8825_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); + + +#endif /* __NAU8825_H__ */ -- 2.5.0

2 1

[alsa-devel] [PATCH v2] topology: Add command line topology tool to build topology binaries
by Liam Girdwood 04 Aug '15

04 Aug '15

Add a command line tool that will parse topology text files and convert to the binary topology data as used by the kernel. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Makefile.am | 3 ++ configure.ac | 5 ++- topology/Makefile.am | 10 +++++ topology/topology.c | 117 +++++++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 134 insertions(+), 1 deletion(-) create mode 100644 topology/Makefile.am create mode 100644 topology/topology.c diff --git a/Makefile.am b/Makefile.am index 5bbe588..613f62d 100644 --- a/Makefile.am +++ b/Makefile.am @@ -25,6 +25,9 @@ endif if HAVE_UCM SUBDIRS += alsaucm endif +if HAVE_TOPOLOGY +SUBDIRS += topology +endif EXTRA_DIST= TODO gitcompile AUTOMAKE_OPTIONS=foreign diff --git a/configure.ac b/configure.ac index f09aa54..4c279a9 100644 --- a/configure.ac +++ b/configure.ac @@ -50,6 +50,8 @@ AC_CHECK_HEADERS([alsa/seq.h], [have_seq="yes"], [have_seq="no"], [#include <alsa/asoundlib.h>]) AC_CHECK_HEADERS([alsa/use-case.h], [have_ucm="yes"], [have_ucm="no"], [#include <alsa/asoundlib.h>]) +AC_CHECK_HEADERS([alsa/topology.h], [have_topology="yes"], [have_topology="no"], + [#include <alsa/asoundlib.h>]) AC_CHECK_HEADERS([samplerate.h], [have_samplerate="yes"], [have_samplerate="no"], [#include <samplerate.h>]) @@ -58,6 +60,7 @@ AM_CONDITIONAL(HAVE_MIXER, test "$have_mixer" = "yes") AM_CONDITIONAL(HAVE_RAWMIDI, test "$have_rawmidi" = "yes") AM_CONDITIONAL(HAVE_SEQ, test "$have_seq" = "yes") AM_CONDITIONAL(HAVE_UCM, test "$have_ucm" = "yes") +AM_CONDITIONAL(HAVE_TOPOLOGY, test "$have_topology" = "yes") AM_CONDITIONAL(HAVE_SAMPLERATE, test "$have_samplerate" = "yes") dnl Check for librt @@ -358,7 +361,7 @@ AC_OUTPUT(Makefile alsactl/Makefile alsactl/init/Makefile \ m4/Makefile po/Makefile.in \ alsaconf/alsaconf alsaconf/Makefile \ alsaconf/po/Makefile \ - alsaucm/Makefile \ + alsaucm/Makefile topology/Makefile \ aplay/Makefile include/Makefile iecset/Makefile utils/Makefile \ utils/alsa-utils.spec seq/Makefile seq/aconnect/Makefile \ seq/aplaymidi/Makefile seq/aseqdump/Makefile seq/aseqnet/Makefile \ diff --git a/topology/Makefile.am b/topology/Makefile.am new file mode 100644 index 0000000..c370b14 --- /dev/null +++ b/topology/Makefile.am @@ -0,0 +1,10 @@ +bin_PROGRAMS = \ + alsatplg + +alsatplg_SOURCES = topology.c + +AM_CPPFLAGS = \ + -Wall -I$(top_srcdir)/include + +alsatplg_LDADD = -lasound + diff --git a/topology/topology.c b/topology/topology.c new file mode 100644 index 0000000..33c3276 --- /dev/null +++ b/topology/topology.c @@ -0,0 +1,117 @@ +/* + Copyright(c) 2014-2015 Intel Corporation + Copyright(c) 2010-2011 Texas Instruments Incorporated, + All rights reserved. + + This program is free software; you can redistribute it and/or modify + it under the terms of version 2 of the GNU General Public License as + published by the Free Software Foundation. + + This program is distributed in the hope that it will be useful, but + WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + General Public License for more details. + + You should have received a copy of the GNU General Public License + along with this program; if not, write to the Free Software + Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA. + The full GNU General Public License is included in this distribution + in the file called LICENSE.GPL. +*/ + +#include <stdlib.h> +#include <stdio.h> +#include <stdint.h> +#include <fcntl.h> +#include <unistd.h> +#include <errno.h> +#include <string.h> +#include <sys/stat.h> +#include <dlfcn.h> +#include <getopt.h> +#include <assert.h> + +#include <alsa/asoundlib.h> +#include <alsa/topology.h> +#include "gettext.h" + +static snd_output_t *log; + +static void usage(char *name) +{ + printf( +_("Usage: %s [OPTIONS]...\n" +"\n" +"-h, --help help\n" +"-c, --compile=FILE compile file\n" +"-v, --verbose=LEVEL set verbosity level (0...1)\n" +"-o, --output=FILE set output file\n" +), name); +} + +int main(int argc, char *argv[]) +{ + snd_tplg_t *snd_tplg; + static const char short_options[] = "hc:v:o:"; + static const struct option long_options[] = { + {"help", 0, 0, 'h'}, + {"verbose", 0, 0, 'v'}, + {"compile", 0, 0, 'c'}, + {"output", 0, 0, 'o'}, + {0, 0, 0, 0}, + }; + char *source_file = NULL, *output_file = NULL; + int c, err, verbose = 0, option_index; + +#ifdef ENABLE_NLS + setlocale(LC_ALL, ""); + textdomain(PACKAGE); +#endif + + err = snd_output_stdio_attach(&log, stderr, 0); + assert(err >= 0); + + while ((c = getopt_long(argc, argv, short_options, long_options, &option_index)) != -1) { + switch (c) { + case 'h': + usage(argv[0]); + return 0; + case 'v': + verbose = atoi(optarg); + break; + case 'c': + source_file = optarg; + break; + case 'o': + output_file = optarg; + break; + default: + fprintf(stderr, _("Try `%s --help' for more information.\n"), argv[0]); + return 1; + } + } + + if (source_file == NULL || output_file == NULL) { + usage(argv[0]); + return 1; + } + + snd_tplg = snd_tplg_new(); + if (snd_tplg == NULL) { + fprintf(stderr, _("failed to create new topology context\n")); + return 1; + } + + snd_tplg_verbose(snd_tplg, verbose); + + err = snd_tplg_build_file(snd_tplg, source_file, output_file); + if (err < 0) { + fprintf(stderr, _("failed to compile context %s\n"), source_file); + snd_tplg_free(snd_tplg); + return 1; + } + + snd_tplg_free(snd_tplg); + return 0; +} + -- 2.1.4

2 1

[alsa-devel] [PATCH] topology: Add command line topology tool to build topology binaries
by Liam Girdwood 04 Aug '15

04 Aug '15

Add a command line tool that will parse topology text files and convert to the binary topology data as used by the kernel. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Makefile.am | 3 ++ configure.ac | 5 ++- topology/Makefile.am | 13 ++++++ topology/topology.c | 117 +++++++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 137 insertions(+), 1 deletion(-) create mode 100644 topology/Makefile.am create mode 100644 topology/topology.c diff --git a/Makefile.am b/Makefile.am index 5bbe588..613f62d 100644 --- a/Makefile.am +++ b/Makefile.am @@ -25,6 +25,9 @@ endif if HAVE_UCM SUBDIRS += alsaucm endif +if HAVE_TOPOLOGY +SUBDIRS += topology +endif EXTRA_DIST= TODO gitcompile AUTOMAKE_OPTIONS=foreign diff --git a/configure.ac b/configure.ac index f09aa54..4c279a9 100644 --- a/configure.ac +++ b/configure.ac @@ -50,6 +50,8 @@ AC_CHECK_HEADERS([alsa/seq.h], [have_seq="yes"], [have_seq="no"], [#include <alsa/asoundlib.h>]) AC_CHECK_HEADERS([alsa/use-case.h], [have_ucm="yes"], [have_ucm="no"], [#include <alsa/asoundlib.h>]) +AC_CHECK_HEADERS([alsa/topology.h], [have_topology="yes"], [have_topology="no"], + [#include <alsa/asoundlib.h>]) AC_CHECK_HEADERS([samplerate.h], [have_samplerate="yes"], [have_samplerate="no"], [#include <samplerate.h>]) @@ -58,6 +60,7 @@ AM_CONDITIONAL(HAVE_MIXER, test "$have_mixer" = "yes") AM_CONDITIONAL(HAVE_RAWMIDI, test "$have_rawmidi" = "yes") AM_CONDITIONAL(HAVE_SEQ, test "$have_seq" = "yes") AM_CONDITIONAL(HAVE_UCM, test "$have_ucm" = "yes") +AM_CONDITIONAL(HAVE_TOPOLOGY, test "$have_topology" = "yes") AM_CONDITIONAL(HAVE_SAMPLERATE, test "$have_samplerate" = "yes") dnl Check for librt @@ -358,7 +361,7 @@ AC_OUTPUT(Makefile alsactl/Makefile alsactl/init/Makefile \ m4/Makefile po/Makefile.in \ alsaconf/alsaconf alsaconf/Makefile \ alsaconf/po/Makefile \ - alsaucm/Makefile \ + alsaucm/Makefile topology/Makefile \ aplay/Makefile include/Makefile iecset/Makefile utils/Makefile \ utils/alsa-utils.spec seq/Makefile seq/aconnect/Makefile \ seq/aplaymidi/Makefile seq/aseqdump/Makefile seq/aseqnet/Makefile \ diff --git a/topology/Makefile.am b/topology/Makefile.am new file mode 100644 index 0000000..f59ce01 --- /dev/null +++ b/topology/Makefile.am @@ -0,0 +1,13 @@ +bin_PROGRAMS = \ + alsatplg + +alsatplg_SOURCES = topology.c + +AM_CPPFLAGS = \ + -Wall -I$(top_srcdir)/include + +alsatplg_LDADD = -lasound + +# local build +AM_CPPFLAGS += -I$(top_srcdir)/../alsa-lib/include +alsatplg_LDADD = -L$(top_srcdir)/../alsa-lib/src/.libs diff --git a/topology/topology.c b/topology/topology.c new file mode 100644 index 0000000..7049bb7 --- /dev/null +++ b/topology/topology.c @@ -0,0 +1,117 @@ +/* + Copyright(c) 2014-2015 Intel Corporation + Copyright(c) 2010-2011 Texas Instruments Incorporated, + All rights reserved. + + This program is free software; you can redistribute it and/or modify + it under the terms of version 2 of the GNU General Public License as + published by the Free Software Foundation. + + This program is distributed in the hope that it will be useful, but + WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + General Public License for more details. + + You should have received a copy of the GNU General Public License + along with this program; if not, write to the Free Software + Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA. + The full GNU General Public License is included in this distribution + in the file called LICENSE.GPL. +*/ + +#include <stdlib.h> +#include <stdio.h> +#include <stdint.h> +#include <fcntl.h> +#include <unistd.h> +#include <errno.h> +#include <string.h> +#include <sys/stat.h> +#include <dlfcn.h> +#include <getopt.h> +#include <alsa/asoundlib.h> +#include <assert.h> + +#include "topology.h" +#include "gettext.h" + +static snd_output_t *log; + +static void usage(char *name) +{ + printf( +_("Usage: %s [OPTIONS]...\n" +"\n" +"-h, --help help\n" +"-c, --compile=FILE compile file\n" +"-v, --verbose=LEVEL set verbosity level (0...1)\n" +"-o, --output=FILE set output file\n" +), name); +} + +int main(int argc, char *argv[]) +{ + snd_tplg_t *snd_tplg; + static const char short_options[] = "hc:v:o:"; + static const struct option long_options[] = { + {"help", 0, 0, 'h'}, + {"verbose", 0, 0, 'v'}, + {"compile", 0, 0, 'c'}, + {"output", 0, 0, 'o'}, + {0, 0, 0, 0}, + }; + char *source_file = NULL, *output_file = NULL; + int c, err, verbose = 0, option_index; + +#ifdef ENABLE_NLS + setlocale(LC_ALL, ""); + textdomain(PACKAGE); +#endif + + err = snd_output_stdio_attach(&log, stderr, 0); + assert(err >= 0); + + while ((c = getopt_long(argc, argv, short_options, long_options, &option_index)) != -1) { + switch (c) { + case 'h': + usage(argv[0]); + return 0; + case 'v': + verbose = atoi(optarg); + break; + case 'c': + source_file = optarg; + break; + case 'o': + output_file = optarg; + break; + default: + fprintf(stderr, _("Try `%s --help' for more information.\n"), argv[0]); + return 1; + } + } + + if (source_file == NULL || output_file == NULL) { + usage(argv[0]); + return 1; + } + + snd_tplg = snd_tplg_new(); + if (snd_tplg == NULL) { + fprintf(stderr, _("failed to create new topology context\n")); + return 1; + } + + snd_tplg_verbose(snd_tplg, verbose); + + err = snd_tplg_build_file(snd_tplg, source_file, output_file); + if (err < 0) { + fprintf(stderr, _("failed to compile context %s\n"), source_file); + snd_tplg_free(snd_tplg); + return 1; + } + + snd_tplg_free(snd_tplg); + return 0; +} + -- 2.1.4

2 1

[alsa-devel] [PATCH 0/3] ALSA: hdac: stream fixes
by Vinod Koul 04 Aug '15

04 Aug '15

Hi Takashi, I had few off fixes on hdac piled up so time to send them out. First one is a fix for stream assignment for decoupled mode used in SKL driver. Second one is static checker fix reported by Dan Carpenter Last is a potential issue which our team reported so a WA for that. The command DMAs do not finish instantaneously, so we should per spec wait till DMA finishes Jeeja KP (2): ALSA: HDA: Fix stream assignment for host in decoupled mode ALSA: HDA: Dont check return for snd_hdac_chip_readl Vinod Koul (1): ALSA: HDA: wait for RIRB, CORB DMA to finish sound/hda/ext/hdac_ext_controller.c | 6 ------ sound/hda/ext/hdac_ext_stream.c | 2 +- sound/hda/hdac_controller.c | 14 ++++++++++++++ 3 files changed, 15 insertions(+), 7 deletions(-) -- 1.9.1

2 7

[alsa-devel] Question about Component and Card
by Kuninori Morimoto 04 Aug '15

04 Aug '15

Hi Lars Cc Mark I noticed my sound card (which is not upstreamed) can't probe in next branch. I think it is because this patch 1b7c12316982f74a5b8e7704c24cf5524d0723a3 (ASoC: Prevent components from being bound to multiple cards) In my case, my sound device (= CPU) has some DAIs, and these are connected to some codecs. It is shown as below. And I would like to use these as 2 sound cards. ex) aplay -D hw:0 xxx aplay -D hw:1 xxx But, 2nd sound card can't probe because of above patch now. (component->card indicates 1st card) It can be probeed if I used DPCM card (= 1 CPU + 2 Codec), but... But, is my understanding wrong ? codec0: codec@0 { xxx }; codec1: codec@1 { xxx }; sound: sound@0xxxxxx { #sound-dai-cells = <1>; rcar_sound,dai { dai0 { xxxx }; dai1 { xxxx }; }; }; card@0 { compatible = "simple-audio-card"; simple-audio-card,cpu { sound-dai = <&sound 0>; }; simple-audio-card,codec { sound-dai = <&codec0>; }; }; card@1 { compatible = "simple-audio-card"; simple-audio-card,cpu { sound-dai = <&sound 1>; }; simple-audio-card,codec { sound-dai = <&codec1>; }; };

3 5

[alsa-devel] [PATCH v14 0/3] ASoC: tda998x: add a codec to the HDMI transmitter
by Jean-Francois Moine 03 Aug '15

03 Aug '15

This patch series asks for Russell King's patch http://article.gmane.org/gmane.linux.ports.arm.kernel/411574 "drm/edid: add function to help find SADs" to be applied. v14: (Mark Brown) - protect the CODEC from edid change - change some structure names - remove the static pointer to the HDMI driver functions - get the ELD from a pointer in the device private area - prevent audio from streaming many sources at the same time - build the CODEC DAI's from two different samples - use EXPORT_SYMBOL_GPL for exported functions - simplify exchanges between audio and video by using common data in the device private area v13: - rebase on 4.2.0-rc3 - remove Russell's patches v12: - use Russell King's DRM ELD helper (Mark Brown) v11: - reduce the patch series to adding the tda998x codec only v10: - add the generic dt-card - define the audio ports from a DT graph of ports (Russell King) - reuse HDMI constants (Andrew Jackson - Jyri Sarha) - alloc rate_constraints in codec (Jyri Sarha) - fix bad number of channels (Jyri Sarha) - correct codec generation from config (Russell King - Jyri Sarha) - no module init/exit (Russell King) v9: - back to a TDA998x specific CODEC - more comments - change magic values to constants v8: - change some comments about the patches v7: - remove the change of the K predivider (Jyri Sarha) - add S24_3LE and S32_LE as possible audio formats (Jyri Sarha) - don't move the struct priv2 definition and use the slave encoder private data as the device private data (Russell King) - remove the useless request_module (Russell King/Mark Brown) - don't lock the HDMI module (Russell King) - use platform_device_unregister to remove the codec (Russell King) v6: - extend the HDMI CODEC instead of using a specific CODEC v5: - use the TDA998x private data instead of a specific area for the CODEC interface - the CODEC is TDA998x specific (Mark Brown) v4: - remove all the TDA998x specific stuff from the CODEC - move the EDID scan from the CODEC to the TDA998x - move the CODEC to sound/soc (Mark Brown) - update the audio_sample_rate from the EDID (Andrew Jackson) v3: fix bad rate (Andrew Jackson) v2: check double stream start (Mark Brown) Jean-Francois Moine (3): drm/i2c: tda998x: Add support of a DT graph of ports drm/i2c: tda998x: Change drvdata for audio extension ASoC: tda998x: add a codec to the HDMI transmitter .../devicetree/bindings/drm/i2c/tda998x.txt | 51 +++++++ drivers/gpu/drm/i2c/tda998x_drv.c | 163 +++++++++++++++++++-- include/sound/tda998x.h | 18 +++ sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tda998x.c | 141 ++++++++++++++++++ 6 files changed, 370 insertions(+), 11 deletions(-) create mode 100644 include/sound/tda998x.h create mode 100644 sound/soc/codecs/tda998x.c -- 2.4.6

3 6

[alsa-devel] [TRIVIAL PATCH] ASoC: atmel_ssc_dai: Correct misuse of 0x%<decimal>
by Joe Perches 03 Aug '15

03 Aug '15

Correct misuse of 0x%d in logging message. Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/soc/atmel/atmel_ssc_dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 841d059..ba8def5 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -290,7 +290,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, int dir, dir_mask; int ret; - pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n", + pr_debug("atmel_ssc_startup: SSC_SR=0x%x\n", ssc_readl(ssc_p->ssc->regs, SR)); /* Enable PMC peripheral clock for this SSC */

2 2

[alsa-devel] [PATCH] ASoC: rt5645: Fix lost pin setting for DMIC1
by Bard Liao 03 Aug '15

03 Aug '15

I2S2_DAC pin can be used for I2S or GPIO. We should set it as GPIO if we use GPIO5 as DMIC1 data pin. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 2 ++ sound/soc/codecs/rt5645.h | 4 ++++ 2 files changed, 6 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9d54b34..c34c498 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3356,6 +3356,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, break; case RT5645_DMIC_DATA_GPIO5: + regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1, + RT5645_I2S2_DAC_PIN_MASK, RT5645_I2S2_DAC_PIN_GPIO); regmap_update_bits(rt5645->regmap, RT5645_DMIC_CTRL1, RT5645_DMIC_1_DP_MASK, RT5645_DMIC_1_DP_GPIO5); regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1, diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 199b22f..dffdda2 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -1693,6 +1693,10 @@ #define RT5645_GP6_PIN_SFT 6 #define RT5645_GP6_PIN_GPIO6 (0x0 << 6) #define RT5645_GP6_PIN_DMIC2_SDA (0x1 << 6) +#define RT5645_I2S2_DAC_PIN_MASK (0x1 << 4) +#define RT5645_I2S2_DAC_PIN_SFT 4 +#define RT5645_I2S2_DAC_PIN_I2S (0x0 << 4) +#define RT5645_I2S2_DAC_PIN_GPIO (0x1 << 4) #define RT5645_GP8_PIN_MASK (0x1 << 3) #define RT5645_GP8_PIN_SFT 3 #define RT5645_GP8_PIN_GPIO8 (0x0 << 3) -- 1.8.1.1.439.g50a6b54

2 1

Re: [alsa-devel] [beagleboard] Audio cap & Linux version 4.1.3-ti-r6.2
by Micka 03 Aug '15

03 Aug '15

Ok, some news : I find out where it's blocked : I've compiled the alsa-lib, and I've put a lot of printf. In the file pcm.c, in the function : snd_pcm_wait_nocheck This line : err_poll = poll(pfd, npfds, timeout); is called, but I never go out .... my trace show : struct pollfd *pfd: 0xbed64780, npfds: 1, timeout: -1 I don't know if it's ok ? Micka, Le lun. 3 août 2015 à 09:41, Micka <mickamusset(a)gmail.com> a écrit : > From what I see, it looks like that the alsa lib is calling multiple time > SNDRV_PCM_IOCTL_SYNC_PTR > > And it's not calling SNDRV_PCM_IOCTL_WRITEN_FRAMES .... > > Robert, I'm having issue to find out the version of alsa lib, do you have > any idea ? > > Maybe I'm going to compile it > > Le lun. 3 août 2015 à 08:58, Micka <mickamusset(a)gmail.com> a écrit : > >> Hi, >> >> Thx to this site : >> http://www.programdevelop.com/4763397/ >> >> I understand better the flow of ALSA :) >> >> this is my new log : >> >> [ 84.688930] snd_soc_dai_digital_mute 0 0 >> [ 84.689000] snd_pcm_playback_ioctl >> [ 84.689032] snd_pcm_playback_ioctl1 3229892899 >> [ 84.689057] call snd_pcm_common_ioctl1 >> [ 84.689084] snd_pcm_common_ioctl1 3229892899 >> [ 84.689109] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.689624] snd_pcm_playback_ioctl >> [ 84.689664] snd_pcm_playback_ioctl1 3228057875 >> [ 84.689689] call snd_pcm_common_ioctl1 >> [ 84.689715] snd_pcm_common_ioctl1 3228057875 >> [ 84.695219] snd_pcm_playback_ioctl >> [ 84.695269] snd_pcm_playback_ioctl1 3229892899 >> [ 84.695296] call snd_pcm_common_ioctl1 >> [ 84.695322] snd_pcm_common_ioctl1 3229892899 >> [ 84.695347] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.695400] snd_pcm_playback_ioctl >> [ 84.695427] snd_pcm_playback_ioctl1 3229892899 >> [ 84.695451] call snd_pcm_common_ioctl1 >> [ 84.695476] snd_pcm_common_ioctl1 3229892899 >> [ 84.695499] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.696332] snd_pcm_playback_ioctl >> [ 84.696364] snd_pcm_playback_ioctl1 3229892899 >> [ 84.696389] call snd_pcm_common_ioctl1 >> [ 84.696414] snd_pcm_common_ioctl1 3229892899 >> [ 84.696438] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.696474] snd_pcm_playback_ioctl >> [ 84.696501] snd_pcm_playback_ioctl1 3229892899 >> [ 84.696524] call snd_pcm_common_ioctl1 >> [ 84.696548] snd_pcm_common_ioctl1 3229892899 >> [ 84.696572] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.729249] snd_pcm_playback_ioctl >> [ 84.729289] snd_pcm_playback_ioctl1 3229892899 >> [ 84.729303] call snd_pcm_common_ioctl1 >> [ 84.729317] snd_pcm_common_ioctl1 3229892899 >> [ 84.729329] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.729355] snd_pcm_playback_ioctl >> [ 84.729368] snd_pcm_playback_ioctl1 3229892899 >> [ 84.729378] call snd_pcm_common_ioctl1 >> [ 84.729390] snd_pcm_common_ioctl1 3229892899 >> [ 84.729401] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.729787] snd_pcm_playback_ioctl >> [ 84.729801] snd_pcm_playback_ioctl1 3229892899 >> [ 84.729812] call snd_pcm_common_ioctl1 >> [ 84.729824] snd_pcm_common_ioctl1 3229892899 >> [ 84.729835] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.729852] snd_pcm_playback_ioctl >> [ 84.729865] snd_pcm_playback_ioctl1 3229892899 >> [ 84.729875] call snd_pcm_common_ioctl1 >> [ 84.729887] snd_pcm_common_ioctl1 3229892899 >> [ 84.729898] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.731902] snd_pcm_playback_ioctl >> [ 84.731914] snd_pcm_playback_ioctl1 3229892899 >> [ 84.731925] call snd_pcm_common_ioctl1 >> [ 84.731937] snd_pcm_common_ioctl1 3229892899 >> [ 84.731947] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.731962] snd_pcm_playback_ioctl >> [ 84.731974] snd_pcm_playback_ioctl1 3229892899 >> [ 84.731984] call snd_pcm_common_ioctl1 >> [ 84.731996] snd_pcm_common_ioctl1 3229892899 >> [ 84.732007] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.732314] snd_pcm_playback_ioctl >> [ 84.732328] snd_pcm_playback_ioctl1 3229892899 >> [ 84.732338] call snd_pcm_common_ioctl1 >> [ 84.732350] snd_pcm_common_ioctl1 3229892899 >> [ 84.732361] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.732375] snd_pcm_playback_ioctl >> [ 84.732387] snd_pcm_playback_ioctl1 3229892899 >> [ 84.732398] call snd_pcm_common_ioctl1 >> [ 84.732410] snd_pcm_common_ioctl1 3229892899 >> [ 84.732421] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.734421] snd_pcm_playback_ioctl >> [ 84.734434] snd_pcm_playback_ioctl1 3229892899 >> [ 84.734444] call snd_pcm_common_ioctl1 >> [ 84.734456] snd_pcm_common_ioctl1 3229892899 >> [ 84.734467] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.734481] snd_pcm_playback_ioctl >> [ 84.734493] snd_pcm_playback_ioctl1 3229892899 >> [ 84.734503] call snd_pcm_common_ioctl1 >> [ 84.734515] snd_pcm_common_ioctl1 3229892899 >> [ 84.734526] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.734826] snd_pcm_playback_ioctl >> [ 84.734839] snd_pcm_playback_ioctl1 3229892899 >> [ 84.734850] call snd_pcm_common_ioctl1 >> [ 84.734862] snd_pcm_common_ioctl1 3229892899 >> [ 84.734872] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.734887] snd_pcm_playback_ioctl >> [ 84.734899] snd_pcm_playback_ioctl1 3229892899 >> [ 84.734909] call snd_pcm_common_ioctl1 >> [ 84.734921] snd_pcm_common_ioctl1 3229892899 >> [ 84.734932] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.734974] snd_pcm_playback_ioctl >> [ 84.734987] snd_pcm_playback_ioctl1 3229892899 >> [ 84.734998] call snd_pcm_common_ioctl1 >> [ 84.735010] snd_pcm_common_ioctl1 3229892899 >> [ 84.735020] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.735034] snd_pcm_playback_ioctl >> [ 84.735046] snd_pcm_playback_ioctl1 16706 >> [ 84.735057] call snd_pcm_common_ioctl1 >> [ 84.735068] snd_pcm_common_ioctl1 16706 >> [ 84.735080] --SNDRV_PCM_IOCTL_START: >> [ 84.735096] snd_pcm_do_start >> [ 84.735108] soc_pcm_trigger >> [ 84.737206] snd_pcm_playback_ioctl >> [ 84.737221] snd_pcm_playback_ioctl1 3229892899 >> [ 84.737232] call snd_pcm_common_ioctl1 >> [ 84.737244] snd_pcm_common_ioctl1 3229892899 >> [ 84.737254] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.737280] snd_pcm_playback_ioctl >> [ 84.737293] snd_pcm_playback_ioctl1 3229892899 >> [ 84.737303] call snd_pcm_common_ioctl1 >> [ 84.737315] snd_pcm_common_ioctl1 3229892899 >> [ 84.737326] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.747620] snd_pcm_playback_ioctl >> [ 84.747647] snd_pcm_playback_ioctl1 3229892899 >> [ 84.747660] call snd_pcm_common_ioctl1 >> [ 84.747673] snd_pcm_common_ioctl1 3229892899 >> [ 84.747685] SNDRV_PCM_IOCTL_SYNC_PTR: >> >> Le ven. 31 juil. 2015 à 14:27, <g4(a)novadsp.com> a écrit : >> >>> Hi Micka >>> >>> > looks like that my config is good ..... But I can't figure out, where >>> i'm blocked .... >>> >>> I am having the same problem with 4.1.0 >>> >>> > Do you know who call the function soc_pcm_prepare ? >>> >>> http://lxr.free-electrons.com/source/sound/soc/soc-pcm.c#L698 >>> >>> I will try and repeat the same exercise but it will be early next week. >>> >>> >>> -- >>> For more options, visit http://beagleboard.org/discuss >>> --- >>> You received this message because you are subscribed to the Google >>> Groups "BeagleBoard" group. >>> To unsubscribe from this group and stop receiving emails from it, send >>> an email to beagleboard+unsubscribe(a)googlegroups.com. >>> For more options, visit https://groups.google.com/d/optout. >>> >>

1 0

[alsa-devel] Alsa issue faced in port audio
by Srinivasan S 03 Aug '15

03 Aug '15

Dear ALSA Community, Am running port-audio(pjproject-2.4 & 2.3) on linux kernel version 3.14 & alsa-lib verison 1.0.27, but some how portaudio ALSA handles under/overrun in blocking read/write: it is never reported back to the application: handleXrun tries to recover, usually this works and leaves the stream in STATE_PREPARED. The xrun flag is set but neither the portaudio ALSA read nor the write methods look at this when they are in the read/write while loop. In case of playback this causes bad "framesAvail" values. The read funtion enters an endless loop. In port audio in writeframes for each time for read samples playback/capture poll will happen in this poll(snd_poll_descriptor_revents) we will get read/write event from device in the success case, when Xrun occurs the poll always returns 0 or NULL, which indicates A value of 0 indicates that the call timed out and no file descriptors were ready.But we traced the source code file descriptor is available & not corrupted Could you please let us know how this can be fixed in ALSA Kindly do the needful as early as possible Srinivasan S

1 1

[alsa-devel] EAPD enable for HP ALC221
by Kailang 03 Aug '15

03 Aug '15

Hi Takashi, HP ALC221 platform needs to enable EAPD for speaker output. Attach patch was enable the EAPD for speaker. Thanks. BR, Kailang

2 6

[alsa-devel] [PATCH] ALSA: hda - remove no physical connection pins from pin_quirk table
by Hui Wang 03 Aug '15

03 Aug '15

To cleanup the pin_quirk table: - rewrite the pin_config_match(), comparing all pins on the machine with the corresponding pins in the quirk table. - remove all 0x4xxxxxxx pin configurations from pin_quirk table - after removing the 0x4xxxxxxx pin configurations, some pin tables are exactly same, so removing the redudant pin tables. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 27 +++- sound/pci/hda/patch_realtek.c | 299 ++++------------------------------------ 2 files changed, 48 insertions(+), 278 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 03b7399..7f57a14 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -887,11 +887,32 @@ EXPORT_SYMBOL_GPL(snd_hda_apply_fixup); static bool pin_config_match(struct hda_codec *codec, const struct hda_pintbl *pins) { - for (; pins->nid; pins++) { - u32 def_conf = snd_hda_codec_get_pincfg(codec, pins->nid); - if (pins->val != def_conf) + int i; + + for (i = 0; i < codec->init_pins.used; i++) { + struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); + hda_nid_t nid = pin->nid; + u32 cfg = pin->cfg; + const struct hda_pintbl *t_pins; + int found; + + t_pins = pins; + found = 0; + for (; t_pins->nid; t_pins++) { + if (t_pins->nid == nid) { + found = 1; + if (t_pins->val == cfg) + break; + else if ((cfg & 0xf0000000) == 0x40000000 && (t_pins->val & 0xf0000000) == 0x40000000) + break; + else + return false; + } + } + if (!found && (cfg & 0xf0000000) != 0x40000000) return false; } + return true; } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 18ae17e..d9a8b9f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5386,399 +5386,192 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {} }; -#define ALC255_STANDARD_PINS \ - {0x18, 0x411111f0}, \ - {0x19, 0x411111f0}, \ - {0x1a, 0x411111f0}, \ - {0x1b, 0x411111f0}, \ - {0x1e, 0x411111f0} - #define ALC256_STANDARD_PINS \ {0x12, 0x90a60140}, \ {0x14, 0x90170110}, \ - {0x19, 0x411111f0}, \ - {0x1a, 0x411111f0}, \ - {0x1b, 0x411111f0}, \ {0x21, 0x02211020} #define ALC282_STANDARD_PINS \ - {0x14, 0x90170110}, \ - {0x18, 0x411111f0}, \ - {0x1a, 0x411111f0}, \ - {0x1b, 0x411111f0}, \ - {0x1e, 0x411111f0} - -#define ALC288_STANDARD_PINS \ - {0x17, 0x411111f0}, \ - {0x18, 0x411111f0}, \ - {0x19, 0x411111f0}, \ - {0x1a, 0x411111f0}, \ - {0x1e, 0x411111f0} + {0x14, 0x90170110} #define ALC290_STANDARD_PINS \ - {0x12, 0x99a30130}, \ - {0x13, 0x40000000}, \ - {0x16, 0x411111f0}, \ - {0x17, 0x411111f0}, \ - {0x19, 0x411111f0}, \ - {0x1b, 0x411111f0}, \ - {0x1e, 0x411111f0} + {0x12, 0x99a30130} #define ALC292_STANDARD_PINS \ {0x14, 0x90170110}, \ - {0x15, 0x0221401f}, \ - {0x1a, 0x411111f0}, \ - {0x1b, 0x411111f0}, \ - {0x1d, 0x40700001} - -#define ALC298_STANDARD_PINS \ - {0x18, 0x411111f0}, \ - {0x19, 0x411111f0}, \ - {0x1a, 0x411111f0}, \ - {0x1e, 0x411111f0}, \ - {0x1f, 0x411111f0} + {0x15, 0x0221401f} static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE, - ALC255_STANDARD_PINS, - {0x12, 0x40300000}, {0x14, 0x90170110}, - {0x17, 0x411111f0}, - {0x1d, 0x40538029}, {0x21, 0x02211020}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC255_STANDARD_PINS, {0x12, 0x90a60140}, {0x14, 0x90170110}, - {0x17, 0x40000000}, - {0x1d, 0x40700001}, {0x21, 0x02211020}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC255_STANDARD_PINS, {0x12, 0x90a60160}, {0x14, 0x90170120}, - {0x17, 0x40000000}, - {0x1d, 0x40700001}, {0x21, 0x02211030}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - {0x12, 0x40000000}, {0x14, 0x90170130}, - {0x17, 0x411111f0}, - {0x18, 0x411111f0}, - {0x19, 0x411111f0}, - {0x1a, 0x411111f0}, {0x1b, 0x01014020}, - {0x1d, 0x4054c029}, - {0x1e, 0x411111f0}, {0x21, 0x0221103f}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - {0x12, 0x40000000}, {0x14, 0x90170150}, - {0x17, 0x411111f0}, - {0x18, 0x411111f0}, - {0x19, 0x411111f0}, - {0x1a, 0x411111f0}, {0x1b, 0x02011020}, - {0x1d, 0x4054c029}, - {0x1e, 0x411111f0}, {0x21, 0x0221105f}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - {0x12, 0x40000000}, {0x14, 0x90170110}, - {0x17, 0x411111f0}, - {0x18, 0x411111f0}, - {0x19, 0x411111f0}, - {0x1a, 0x411111f0}, {0x1b, 0x01014020}, - {0x1d, 0x4054c029}, - {0x1e, 0x411111f0}, {0x21, 0x0221101f}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60160}, {0x14, 0x90170120}, {0x17, 0x90170140}, - {0x18, 0x40000000}, - {0x19, 0x411111f0}, - {0x1a, 0x411111f0}, - {0x1b, 0x411111f0}, - {0x1d, 0x41163b05}, - {0x1e, 0x411111f0}, {0x21, 0x0321102f}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC255_STANDARD_PINS, {0x12, 0x90a60160}, {0x14, 0x90170130}, - {0x17, 0x40000000}, - {0x1d, 0x40700001}, {0x21, 0x02211040}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC255_STANDARD_PINS, {0x12, 0x90a60160}, {0x14, 0x90170140}, - {0x17, 0x40000000}, - {0x1d, 0x40700001}, {0x21, 0x02211050}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC255_STANDARD_PINS, {0x12, 0x90a60170}, {0x14, 0x90170120}, - {0x17, 0x40000000}, - {0x1d, 0x40700001}, {0x21, 0x02211030}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC255_STANDARD_PINS, {0x12, 0x90a60170}, {0x14, 0x90170130}, - {0x17, 0x40000000}, - {0x1d, 0x40700001}, {0x21, 0x02211040}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC255_STANDARD_PINS, {0x12, 0x90a60170}, {0x14, 0x90170140}, - {0x17, 0x40000000}, - {0x1d, 0x40700001}, {0x21, 0x02211050}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell Inspiron 5548", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC255_STANDARD_PINS, {0x12, 0x90a60180}, {0x14, 0x90170130}, - {0x17, 0x40000000}, - {0x1d, 0x40700001}, {0x21, 0x02211040}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC255_STANDARD_PINS, {0x12, 0x90a60160}, {0x14, 0x90170120}, - {0x17, 0x40000000}, - {0x1d, 0x40700001}, {0x21, 0x02211030}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC256_STANDARD_PINS, - {0x13, 0x40000000}, - {0x1d, 0x40700001}, - {0x1e, 0x411111f0}), - SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC256_STANDARD_PINS, - {0x13, 0x411111f0}, - {0x1d, 0x40700001}, - {0x1e, 0x411111f0}), - SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC256_STANDARD_PINS, - {0x13, 0x411111f0}, - {0x1d, 0x4077992d}, - {0x1e, 0x411111ff}), + ALC256_STANDARD_PINS), SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4, {0x12, 0x90a60130}, - {0x13, 0x40000000}, {0x14, 0x90170110}, {0x15, 0x0421101f}, - {0x16, 0x411111f0}, - {0x17, 0x411111f0}, - {0x18, 0x411111f0}, - {0x19, 0x411111f0}, - {0x1a, 0x04a11020}, - {0x1b, 0x411111f0}, - {0x1d, 0x40748605}, - {0x1e, 0x411111f0}), + {0x1a, 0x04a11020}), SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED, {0x12, 0x90a60140}, - {0x13, 0x40000000}, {0x14, 0x90170110}, {0x15, 0x0421101f}, - {0x16, 0x411111f0}, - {0x17, 0x411111f0}, {0x18, 0x02811030}, - {0x19, 0x411111f0}, {0x1a, 0x04a1103f}, - {0x1b, 0x02011020}, - {0x1d, 0x40700001}, - {0x1e, 0x411111f0}), + {0x1b, 0x02011020}), SND_HDA_PIN_QUIRK(0x10ec0282, 0x103c, "HP 15 Touchsmart", ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC282_STANDARD_PINS, {0x12, 0x99a30130}, - {0x17, 0x40000000}, {0x19, 0x03a11020}, - {0x1d, 0x40f41905}, {0x21, 0x0321101f}), SND_HDA_PIN_QUIRK(0x10ec0282, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC282_STANDARD_PINS, {0x12, 0x99a30130}, - {0x17, 0x40020008}, {0x19, 0x03a11020}, - {0x1d, 0x40e00001}, {0x21, 0x03211040}), SND_HDA_PIN_QUIRK(0x10ec0282, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC282_STANDARD_PINS, {0x12, 0x99a30130}, - {0x17, 0x40000000}, {0x19, 0x03a11030}, - {0x1d, 0x40e00001}, {0x21, 0x03211020}), SND_HDA_PIN_QUIRK(0x10ec0282, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC282_STANDARD_PINS, {0x12, 0x99a30130}, - {0x17, 0x40000000}, - {0x19, 0x03a11030}, - {0x1d, 0x40f00001}, - {0x21, 0x03211020}), - SND_HDA_PIN_QUIRK(0x10ec0282, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, - ALC282_STANDARD_PINS, - {0x12, 0x99a30130}, - {0x17, 0x40000000}, {0x19, 0x04a11020}, - {0x1d, 0x40f00001}, {0x21, 0x0421101f}), - SND_HDA_PIN_QUIRK(0x10ec0282, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, - ALC282_STANDARD_PINS, - {0x12, 0x99a30130}, - {0x17, 0x40000000}, - {0x19, 0x03a11030}, - {0x1d, 0x40f00001}, - {0x21, 0x04211020}), SND_HDA_PIN_QUIRK(0x10ec0282, 0x103c, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED, ALC282_STANDARD_PINS, {0x12, 0x90a60140}, - {0x17, 0x40000000}, {0x19, 0x04a11030}, - {0x1d, 0x40f00001}, {0x21, 0x04211020}), SND_HDA_PIN_QUIRK(0x10ec0283, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, ALC282_STANDARD_PINS, {0x12, 0x90a60130}, - {0x17, 0x40020008}, - {0x19, 0x411111f0}, - {0x1d, 0x40e00001}, {0x21, 0x0321101f}), SND_HDA_PIN_QUIRK(0x10ec0283, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60160}, {0x14, 0x90170120}, - {0x17, 0x40000000}, - {0x18, 0x411111f0}, - {0x19, 0x411111f0}, - {0x1a, 0x411111f0}, - {0x1b, 0x411111f0}, - {0x1d, 0x40700001}, - {0x1e, 0x411111f0}, {0x21, 0x02211030}), SND_HDA_PIN_QUIRK(0x10ec0283, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, ALC282_STANDARD_PINS, {0x12, 0x90a60130}, - {0x17, 0x40020008}, {0x19, 0x03a11020}, - {0x1d, 0x40e00001}, {0x21, 0x0321101f}), SND_HDA_PIN_QUIRK(0x10ec0288, 0x1028, "Dell", ALC288_FIXUP_DELL_XPS_13_GPIO6, - ALC288_STANDARD_PINS, {0x12, 0x90a60120}, - {0x13, 0x40000000}, {0x14, 0x90170110}, - {0x1d, 0x4076832d}, {0x21, 0x0321101f}), SND_HDA_PIN_QUIRK(0x10ec0290, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC290_STANDARD_PINS, - {0x14, 0x411111f0}, {0x15, 0x04211040}, {0x18, 0x90170112}, - {0x1a, 0x04a11020}, - {0x1d, 0x4075812d}), + {0x1a, 0x04a11020}), SND_HDA_PIN_QUIRK(0x10ec0290, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC290_STANDARD_PINS, - {0x14, 0x411111f0}, {0x15, 0x04211040}, {0x18, 0x90170110}, - {0x1a, 0x04a11020}, - {0x1d, 0x4075812d}), + {0x1a, 0x04a11020}), SND_HDA_PIN_QUIRK(0x10ec0290, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC290_STANDARD_PINS, - {0x14, 0x411111f0}, {0x15, 0x0421101f}, - {0x18, 0x411111f0}, - {0x1a, 0x04a11020}, - {0x1d, 0x4075812d}), + {0x1a, 0x04a11020}), SND_HDA_PIN_QUIRK(0x10ec0290, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC290_STANDARD_PINS, - {0x14, 0x411111f0}, {0x15, 0x04211020}, - {0x18, 0x411111f0}, - {0x1a, 0x04a11040}, - {0x1d, 0x4076a12d}), + {0x1a, 0x04a11040}), SND_HDA_PIN_QUIRK(0x10ec0290, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC290_STANDARD_PINS, {0x14, 0x90170110}, {0x15, 0x04211020}, - {0x18, 0x411111f0}, - {0x1a, 0x04a11040}, - {0x1d, 0x4076a12d}), + {0x1a, 0x04a11040}), SND_HDA_PIN_QUIRK(0x10ec0290, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC290_STANDARD_PINS, {0x14, 0x90170110}, {0x15, 0x04211020}, - {0x18, 0x411111f0}, - {0x1a, 0x04a11020}, - {0x1d, 0x4076a12d}), + {0x1a, 0x04a11020}), SND_HDA_PIN_QUIRK(0x10ec0290, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC290_STANDARD_PINS, {0x14, 0x90170110}, {0x15, 0x0421101f}, - {0x18, 0x411111f0}, - {0x1a, 0x04a11020}, - {0x1d, 0x4075812d}), + {0x1a, 0x04a11020}), SND_HDA_PIN_QUIRK(0x10ec0292, 0x1028, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, ALC292_STANDARD_PINS, {0x12, 0x90a60140}, - {0x13, 0x411111f0}, {0x16, 0x01014020}, - {0x18, 0x411111f0}, - {0x19, 0x01a19030}, - {0x1e, 0x411111f0}), + {0x19, 0x01a19030}), SND_HDA_PIN_QUIRK(0x10ec0292, 0x1028, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, ALC292_STANDARD_PINS, {0x12, 0x90a60140}, - {0x13, 0x411111f0}, {0x16, 0x01014020}, {0x18, 0x02a19031}, - {0x19, 0x01a1903e}, - {0x1e, 0x411111f0}), + {0x19, 0x01a1903e}), SND_HDA_PIN_QUIRK(0x10ec0292, 0x1028, "Dell", ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, ALC292_STANDARD_PINS, - {0x12, 0x90a60140}, - {0x13, 0x411111f0}, - {0x16, 0x411111f0}, - {0x18, 0x411111f0}, - {0x19, 0x411111f0}, - {0x1e, 0x411111f0}), + {0x12, 0x90a60140}), SND_HDA_PIN_QUIRK(0x10ec0293, 0x1028, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_STANDARD_PINS, - {0x12, 0x40000000}, {0x13, 0x90a60140}, {0x16, 0x21014020}, - {0x18, 0x411111f0}, - {0x19, 0x21a19030}, - {0x1e, 0x411111f0}), + {0x19, 0x21a19030}), SND_HDA_PIN_QUIRK(0x10ec0293, 0x1028, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_STANDARD_PINS, - {0x12, 0x40000000}, - {0x13, 0x90a60140}, - {0x16, 0x411111f0}, - {0x18, 0x411111f0}, - {0x19, 0x411111f0}, - {0x1e, 0x411111f0}), - SND_HDA_PIN_QUIRK(0x10ec0293, 0x1028, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC292_STANDARD_PINS, - {0x12, 0x40000000}, - {0x13, 0x90a60140}, - {0x16, 0x21014020}, - {0x18, 0x411111f0}, - {0x19, 0x21a19030}, - {0x1e, 0x411111ff}), + {0x13, 0x90a60140}), SND_HDA_PIN_QUIRK(0x10ec0298, 0x1028, "Dell", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC298_STANDARD_PINS, {0x12, 0x90a60130}, - {0x13, 0x40000000}, - {0x14, 0x411111f0}, {0x17, 0x90170140}, - {0x1d, 0x4068a36d}, {0x21, 0x03211020}), {} }; @@ -6672,77 +6465,33 @@ static const struct hda_model_fixup alc662_fixup_models[] = { static const struct snd_hda_pin_quirk alc662_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0662, 0x1028, "Dell", ALC662_FIXUP_DELL_MIC_NO_PRESENCE, - {0x12, 0x4004c000}, {0x14, 0x01014010}, - {0x15, 0x411111f0}, - {0x16, 0x411111f0}, {0x18, 0x01a19020}, - {0x19, 0x411111f0}, {0x1a, 0x0181302f}, - {0x1b, 0x0221401f}, - {0x1c, 0x411111f0}, - {0x1d, 0x4054c601}, - {0x1e, 0x411111f0}), + {0x1b, 0x0221401f}), SND_HDA_PIN_QUIRK(0x10ec0668, 0x1028, "Dell", ALC668_FIXUP_AUTO_MUTE, {0x12, 0x99a30130}, {0x14, 0x90170110}, {0x15, 0x0321101f}, - {0x16, 0x03011020}, - {0x18, 0x40000008}, - {0x19, 0x411111f0}, - {0x1a, 0x411111f0}, - {0x1b, 0x411111f0}, - {0x1d, 0x41000001}, - {0x1e, 0x411111f0}, - {0x1f, 0x411111f0}), + {0x16, 0x03011020}), SND_HDA_PIN_QUIRK(0x10ec0668, 0x1028, "Dell", ALC668_FIXUP_AUTO_MUTE, {0x12, 0x99a30140}, {0x14, 0x90170110}, {0x15, 0x0321101f}, - {0x16, 0x03011020}, - {0x18, 0x40000008}, - {0x19, 0x411111f0}, - {0x1a, 0x411111f0}, - {0x1b, 0x411111f0}, - {0x1d, 0x41000001}, - {0x1e, 0x411111f0}, - {0x1f, 0x411111f0}), + {0x16, 0x03011020}), SND_HDA_PIN_QUIRK(0x10ec0668, 0x1028, "Dell", ALC668_FIXUP_AUTO_MUTE, {0x12, 0x99a30150}, {0x14, 0x90170110}, {0x15, 0x0321101f}, - {0x16, 0x03011020}, - {0x18, 0x40000008}, - {0x19, 0x411111f0}, - {0x1a, 0x411111f0}, - {0x1b, 0x411111f0}, - {0x1d, 0x41000001}, - {0x1e, 0x411111f0}, - {0x1f, 0x411111f0}), + {0x16, 0x03011020}), SND_HDA_PIN_QUIRK(0x10ec0668, 0x1028, "Dell", ALC668_FIXUP_AUTO_MUTE, - {0x12, 0x411111f0}, {0x14, 0x90170110}, {0x15, 0x0321101f}, - {0x16, 0x03011020}, - {0x18, 0x40000008}, - {0x19, 0x411111f0}, - {0x1a, 0x411111f0}, - {0x1b, 0x411111f0}, - {0x1d, 0x41000001}, - {0x1e, 0x411111f0}, - {0x1f, 0x411111f0}), + {0x16, 0x03011020}), SND_HDA_PIN_QUIRK(0x10ec0668, 0x1028, "Dell XPS 15", ALC668_FIXUP_AUTO_MUTE, {0x12, 0x90a60130}, {0x14, 0x90170110}, - {0x15, 0x0321101f}, - {0x16, 0x40000000}, - {0x18, 0x411111f0}, - {0x19, 0x411111f0}, - {0x1a, 0x411111f0}, - {0x1b, 0x411111f0}, - {0x1d, 0x40d6832d}, - {0x1e, 0x411111f0}, - {0x1f, 0x411111f0}), + {0x15, 0x0321101f}), {} }; -- 1.9.1

2 1

Re: [alsa-devel] [PATCH] Fix logical-not-parentheses warning
by Takashi Iwai 03 Aug '15

03 Aug '15

On Sun, 02 Aug 2015 11:08:57 +0200, Tomer Barletz wrote: > > This fixes the following warning, that is seen with gcc 5.1: > warning: logical not is only applied to the left hand side of comparison [-Wlogical-not-parentheses]. > > Signed-off-by: Tomer Barletz <barletz(a)gmail.com> Applied, thanks. Takashi > --- > sound/pci/oxygen/oxygen_mixer.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c > index 6492bca..4ca1266 100644 > --- a/sound/pci/oxygen/oxygen_mixer.c > +++ b/sound/pci/oxygen/oxygen_mixer.c > @@ -88,7 +88,7 @@ static int dac_mute_put(struct snd_kcontrol *ctl, > int changed; > > mutex_lock(&chip->mutex); > - changed = !value->value.integer.value[0] != chip->dac_mute; > + changed = (!value->value.integer.value[0]) != chip->dac_mute; > if (changed) { > chip->dac_mute = !value->value.integer.value[0]; > chip->model.update_dac_mute(chip); > -- > 2.4.3 > >

1 0

[alsa-devel] Beaglebone black : Audio cap & Linux version 4.1.3-ti-r6.2
by Micka 03 Aug '15

03 Aug '15

HI everyone, I'm trying to test the sound output with the driver tlv320aic3104. I'm stuck somewhere and I can't figure it out ! I Put a lot of printk and I don't know where to put other printk. Do you know who call the function soc_pcm_prepare ? 693 <http://lxr.free-electrons.com/source/sound/soc/soc-pcm.c#L693> */**694 <http://lxr.free-electrons.com/source/sound/soc/soc-pcm.c#L694> * * Called by ALSA when the PCM substream is prepared, can set format, sample*695 <http://lxr.free-electrons.com/source/sound/soc/soc-pcm.c#L695> * * rate, etc. This function is non atomic and can be called multiple times,*696 <http://lxr.free-electrons.com/source/sound/soc/soc-pcm.c#L696> * * it can refer to the runtime info.*697 <http://lxr.free-electrons.com/source/sound/soc/soc-pcm.c#L697> * */*698 <http://lxr.free-electrons.com/source/sound/soc/soc-pcm.c#L698> static int soc_pcm_prepare <http://lxr.free-electrons.com/ident?i=soc_pcm_prepare>(struct snd_pcm_substream <http://lxr.free-electrons.com/ident?i=snd_pcm_substream> *substream) Looks like that it's ALSA, but I can't find where ? I tried to find a line in the kernel calling the function: rtd->ops <http://lxr.free-electrons.com/ident?i=ops>.prepare(.... ) can't find it .... my trace : [ 550.269286] ASoC: tlv320aic3x-hifi <-> 48038000.mcasp info: [ 550.269363] ASoC: rate mask 0x7fe [ 550.269398] ASoC: min ch 2 max ch 2 [ 550.269431] ASoC: min rate 8000 max rate 96000 [ 550.275809] tlv aic3x_hw_params [ 550.275862] snd_soc_read: snd soc component read: ret 0 reg 9 val 64 [ 550.275934] snd_soc_component_write reg 9 val 64 [ 550.275987] snd_soc_component_write reg 101 val 0 [ 550.276030] snd_soc_component_write reg 7 val 10 [ 550.276063] snd_soc_component_write reg 2 val 0 [ 550.289260] snd_soc_component_write reg 11 val 1 [ 550.289339] snd_soc_component_write reg 4 val 32 [ 550.289375] snd_soc_component_write reg 5 val 30 [ 550.289407] snd_soc_component_write reg 6 val 0 [ 550.289952] soc_pcm_prepare [ 550.290003] snd_soc_component_write reg 10 val 0 [ 550.290043] soc_dapm_stream_event 1 [ 550.290107] soc_dapm_stream_event: dapm_power_widgets:start dapm_power_widgets 1 [ 550.290150] trace_snd_soc_dapm_start [ 550.290176] dapm reset [ 550.290226] trace_snd_soc_dapm_widget_power [ 550.290257] trace_snd_soc_dapm_widget_power [ 550.290284] trace_snd_soc_dapm_widget_power [ 550.290311] trace_snd_soc_dapm_widget_power [ 550.290339] trace_snd_soc_dapm_widget_power [ 550.290370] trace_snd_soc_dapm_widget_power [ 550.290395] trace_snd_soc_dapm_widget_power [ 550.290423] trace_snd_soc_dapm_widget_power [ 550.290448] trace_snd_soc_dapm_widget_power [ 550.290477] trace_snd_soc_dapm_widget_power [ 550.290503] trace_snd_soc_dapm_widget_power [ 550.290531] trace_snd_soc_dapm_widget_power [ 550.290557] trace_snd_soc_dapm_widget_power [ 550.290582] trace_snd_soc_dapm_widget_power [ 550.290608] trace_snd_soc_dapm_widget_power [ 550.290634] trace_snd_soc_dapm_widget_power [ 550.290659] trace_snd_soc_dapm_widget_power [ 550.290686] trace_snd_soc_dapm_widget_power [ 550.290712] trace_snd_soc_dapm_widget_power [ 550.290738] trace_snd_soc_dapm_widget_power [ 550.290765] trace_snd_soc_dapm_widget_power [ 550.290791] trace_snd_soc_dapm_widget_power [ 550.290817] trace_snd_soc_dapm_widget_power [ 550.290842] trace_snd_soc_dapm_widget_power [ 550.290879] dap set bias to off [ 550.290910] dapm_pre_sequence_async off [ 550.290937] snd soc dapm set bias level 1 [ 550.290962] trace_snd_soc_bias_level_start 1 [ 550.290988] trace_snd_soc_bias_level_done 1 [ 550.291011] dapm_pre_sequence_async onsnd soc dapm set bias level 2 [ 550.291047] trace_snd_soc_bias_level_start 2 [ 550.291073] trace_snd_soc_bias_level_done 2 [ 550.291318] dapm_pre_sequence_async off [ 550.291354] snd soc dapm set bias level 1 [ 550.291381] trace_snd_soc_bias_level_start 1 [ 550.291405] soc set bias level [ 550.291432] tlv set bias level 1 [ 550.291456] tlv standby [ 550.291480] tlv set power [ 550.300504] snd_soc_read: snd soc component read: ret 0 reg 5 val 30 [ 550.300583] snd_soc_read: snd soc component read: ret 0 reg 6 val 0 [ 550.300631] snd_soc_component_write reg 5 val 30 [ 550.301103] snd_soc_component_write reg 6 val 0 [ 550.301577] trace_snd_soc_bias_level_done 1 [ 550.301611] dapm_pre_sequence_async onsnd soc dapm set bias level 2 [ 550.301650] trace_snd_soc_bias_level_start 2 [ 550.301675] soc set bias level [ 550.301701] tlv set bias level 2 [ 550.301724] tlv prepare [ 550.301757] trace_snd_soc_bias_level_done 2 [ 550.301864] dapm event WILL_PMU [ 550.301897] dapm event WILL_PMU [ 550.301923] dapm event WILL_PMU [ 550.301948] dapm event WILL_PMU [ 550.301972] dapm event WILL_PMU [ 550.301997] dapm event WILL_PMU [ 550.302021] dapm event WILL_PMU [ 550.302045] dapm event WILL_PMU [ 550.302069] dapm event WILL_PMU [ 550.302094] dapm event WILL_PMU [ 550.302119] dapm event WILL_PMU [ 550.302143] dapm event WILL_PMU [ 550.302168] dapm event WILL_PMU [ 550.302192] dapm event WILL_PMU [ 550.302216] dapm event WILL_PMU [ 550.302240] dapm event WILL_PMU [ 550.302265] dapm event WILL_PMU [ 550.302289] dapm event WILL_PMU [ 550.302313] dapm event WILL_PMU [ 550.302337] dapm event WILL_PMU [ 550.302362] dapm event WILL_PMU [ 550.302386] dapm event WILL_PMU [ 550.302410] dapm event WILL_PMU [ 550.302435] dapm event WILL_PMU [ 550.302460] dapm sq run 1 [ 550.302489] dapm sq run 1 [ 550.302523] dapm event PRE_PMU [ 550.302550] dapm event PRE_PMD [ 550.302575] dapm event PRE_PMU [ 550.302599] dapm event PRE_PMD [ 550.302623] dapm event PRE_PMU [ 550.302647] dapm event PRE_PMD [ 550.302671] dapm event PRE_PMU [ 550.302696] dapm event PRE_PMD [ 550.302720] dapm event PRE_PMU [ 550.302744] dapm event PRE_PMD [ 550.302769] dapm event PRE_PMU [ 550.302793] dapm event PRE_PMD [ 550.302817] dapm event POST_PMU [ 550.302842] dapm event POST_PMD [ 550.302867] dapm event POST_PMU [ 550.302891] dapm event POST_PMD [ 550.302915] dapm event POST_PMU [ 550.302939] dapm event POST_PMD [ 550.302964] dapm event POST_PMU [ 550.302988] dapm event POST_PMD [ 550.303012] dapm event POST_PMU [ 550.303036] dapm event POST_PMD [ 550.303060] dapm event POST_PMU [ 550.303084] dapm event POST_PMD [ 550.303110] dapm event PRE_PMU [ 550.303134] dapm event PRE_PMD [ 550.303159] dapm event POST_PMU [ 550.303183] dapm event POST_PMD [ 550.303209] dapm event PRE_PMU [ 550.303234] dapm event PRE_PMD [ 550.303258] dapm event PRE_PMU [ 550.303282] dapm event PRE_PMD [ 550.303306] dapm event PRE_PMU [ 550.303331] dapm event PRE_PMD [ 550.303355] dapm event PRE_PMU [ 550.303379] dapm event PRE_PMD [ 550.303403] dapm event POST_PMU [ 550.303427] dapm event POST_PMD [ 550.303451] dapm event POST_PMU [ 550.303475] dapm event POST_PMD [ 550.303499] dapm event POST_PMU [ 550.303524] dapm event POST_PMD [ 550.303548] dapm event POST_PMU [ 550.303572] dapm event POST_PMD [ 550.303598] dapm event PRE_PMU [ 550.303622] dapm event PRE_PMD [ 550.303647] dapm event PRE_PMU [ 550.303671] dapm event PRE_PMD [ 550.304162] dapm event POST_PMU [ 550.304199] dapm event POST_PMD [ 550.304226] dapm event POST_PMU [ 550.304250] dapm event POST_PMD [ 550.304278] dapm event PRE_PMU [ 550.304303] dapm event PRE_PMD [ 550.304328] dapm event PRE_PMU [ 550.304352] dapm event PRE_PMD [ 550.304377] dapm event PRE_PMU [ 550.304401] dapm event PRE_PMD [ 550.304426] dapm event PRE_PMU [ 550.304450] dapm event PRE_PMD [ 550.304474] dapm event POST_PMU [ 550.304499] dapm event POST_PMD [ 550.304523] dapm event POST_PMU [ 550.304547] dapm event POST_PMD [ 550.304572] dapm event POST_PMU [ 550.304596] dapm event POST_PMD [ 550.304620] dapm event POST_PMU [ 550.304644] dapm event POST_PMD [ 550.304670] dapm event PRE_PMU [ 550.304695] dapm event PRE_PMD [ 550.305156] dapm event POST_PMU [ 550.305191] dapm event POST_PMD [ 550.305219] dapm event PRE_PMU [ 550.305245] dapm event PRE_PMD [ 550.305696] dapm event POST_PMU [ 550.305730] dapm event POST_PMD [ 550.305757] dapm event PRE_PMU [ 550.305782] dapm event PRE_PMD [ 550.306275] dapm event POST_PMU [ 550.306312] dapm event POST_PMD [ 550.306341] dapm event PRE_PMU [ 550.306366] dapm event PRE_PMD [ 550.306849] dapm event POST_PMU [ 550.306885] dapm event POST_PMD [ 550.306914] dapm event PRE_PMU [ 550.306939] dapm event PRE_PMD [ 550.317926] dapm event POST_PMU [ 550.317958] dapm event POST_PMD [ 550.317977] dapm event PRE_PMU [ 550.317991] dapm event PRE_PMD [ 550.318401] dapm event POST_PMU [ 550.318422] dapm event POST_PMD [ 550.318439] dapm event PRE_PMU [ 550.318454] dapm event PRE_PMD [ 550.318467] dapm event POST_PMU [ 550.318480] dapm event POST_PMD [ 550.318496] dapm_power_widgets run all the bias changes in parrallel [ 550.321470] dapm_post_sequence_async prepare 1 [ 550.321498] snd soc dapm set bias level 3 [ 550.321513] trace_snd_soc_bias_level_start 3 [ 550.321527] soc set bias level [ 550.321543] tlv set bias level 3 [ 550.321555] tlv bias on [ 550.321570] trace_snd_soc_bias_level_done 3 [ 550.321649] dapm_power_widgets dapm_power_widgets [ 550.321666] DAPM sequencing finished [ 550.321680] trace_snd_soc_dapm_done [ 550.321695] pcm prepare call snd_soc_dai_digital_mute [ 550.321711] snd_soc_dai_digital_mute 0 0 [ 550.321725] tlv aic3x_mute [ 550.321739] snd_soc_read: snd soc component read: ret 0 reg 43 val 168 [ 550.321772] snd_soc_read: snd soc component read: ret 0 reg 44 val 168 [ 550.321798] snd_soc_component_write reg 43 val 40 [ 550.322224] snd_soc_component_write reg 44 val 40 [ 550.322630] pcm prepare call snd_soc_dai_digital_mute for cpu_dai [ 550.322651] snd_soc_dai_digital_mute 0 0 > >

1 1

[alsa-devel] [PATCH] ALSA: hda - Fix a wrong busy check in alt PCM open
by Takashi Iwai 03 Aug '15

03 Aug '15

Currently, the alt PCM open callback returns -EBUSY when an independent HP is turned off, supposing that it conflicts with the main PCM. However, obviously, this check is wrong when the independent HP itself isn't enabled but the alt PCM was explicitly created via alc_dac_nid by a codec driver. Reported-and-tested-by: Kailang Yang <kailang(a)realtek.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index ac0db1679f09..b077bb644434 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -5175,7 +5175,7 @@ static int alt_playback_pcm_open(struct hda_pcm_stream *hinfo, int err = 0; mutex_lock(&spec->pcm_mutex); - if (!spec->indep_hp_enabled) + if (spec->indep_hp && !spec->indep_hp_enabled) err = -EBUSY; else spec->active_streams |= 1 << STREAM_INDEP_HP; -- 2.4.5

2 3

[alsa-devel] [PATCH] ALSA: hda - Add pin quirk for the headset mic jack detection on Dell laptop
by woodrow.shen＠canonical.com 03 Aug '15

03 Aug '15

From: Woodrow Shen <woodrow.shen(a)canonical.com> The new Dell laptop with codec 256 can't detect headset mic when headset was inserted on the machine. From alsa-info, we check init_pin_configs and need to define the new register value for pin 0x1d & 0x1e because the original macro ALC256_STANDARD_PINS can't match pin definition. Also, the macro ALC256_STANDARD_PINS is simplified by removing them. This makes headset mic works on laptop. Codec: Realtek ALC256 Vendor Id: 0x10ec0256 Subsystem Id: 0x102806f2 BugLink: https://bugs.launchpad.net/bugs/1478497 Signed-off-by: Woodrow Shen <woodrow.shen(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d94c0e3..4ae877c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5398,8 +5398,6 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {0x19, 0x411111f0}, \ {0x1a, 0x411111f0}, \ {0x1b, 0x411111f0}, \ - {0x1d, 0x40700001}, \ - {0x1e, 0x411111f0}, \ {0x21, 0x02211020} #define ALC282_STANDARD_PINS \ @@ -5556,10 +5554,19 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x21, 0x02211030}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, ALC256_STANDARD_PINS, - {0x13, 0x40000000}), + {0x13, 0x40000000}, + {0x1d, 0x40700001}, + {0x1e, 0x411111f0}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, ALC256_STANDARD_PINS, - {0x13, 0x411111f0}), + {0x13, 0x411111f0}, + {0x1d, 0x40700001}, + {0x1e, 0x411111f0}), + SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC256_STANDARD_PINS, + {0x13, 0x411111f0}, + {0x1d, 0x4077992d}, + {0x1e, 0x411111ff}), SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4, {0x12, 0x90a60130}, {0x13, 0x40000000}, -- 2.1.4

3 6

[alsa-devel] [PATCH 3/6] ASoC: fsl_ssi: enable AC'97 asymmetric rates
by Maciej S. Szmigiero 02 Aug '15

02 Aug '15

AC'97 bus can support asymmetric playback/capture rates so enable them in this case in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index a83b900..7f4f0b9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1377,7 +1377,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Are the RX and the TX clocks locked? */ if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) { - ssi_private->cpu_dai_drv.symmetric_rates = 1; + if (!fsl_ssi_is_ac97(ssi_private)) + ssi_private->cpu_dai_drv.symmetric_rates = 1; + ssi_private->cpu_dai_drv.symmetric_channels = 1; ssi_private->cpu_dai_drv.symmetric_samplebits = 1; }

2 4

[alsa-devel] [PATCH 5/6] ASoC: fsl_ssi: instantiate AC'97 CODEC
by Maciej S. Szmigiero 02 Aug '15

02 Aug '15

Instantiate AC'97 CODEC in fsl_ssi driver AC'97 mode. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 21 +++++++++++++++++++++ 1 files changed, 21 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 154bcf6..8e5ff5e 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1460,6 +1460,27 @@ done: _fsl_ssi_set_dai_fmt(&pdev->dev, ssi_private, ssi_private->dai_fmt); + if (fsl_ssi_is_ac97(ssi_private)) { + u32 ssi_idx; + + ret = of_property_read_u32(np, "cell-index", &ssi_idx); + if (ret) { + dev_err(&pdev->dev, "cannot get SSI index property\n"); + goto error_sound_card; + } + + ssi_private->pdev = + platform_device_register_data(NULL, + "ac97-codec", ssi_idx, NULL, 0); + if (IS_ERR(ssi_private->pdev)) { + ret = PTR_ERR(ssi_private->pdev); + dev_err(&pdev->dev, + "failed to register AC97 codec platform: %d\n", + ret); + goto error_sound_card; + } + } + return 0; error_sound_card:

2 3

Re: [alsa-devel] [PATCH] Fix logical-not-parentheses warning
by Joe Perches 02 Aug '15

02 Aug '15

On Sun, 2015-08-02 at 02:08 -0700, Tomer Barletz wrote: > This fixes the following warning, that is seen with gcc 5.1: > warning: logical not is only applied to the left hand side of > comparison [-Wlogical-not-parentheses]. [] > diff --git a/sound/pci/oxygen/oxygen_mixer.c [] > @@ -88,7 +88,7 @@ static int dac_mute_put(struct snd_kcontrol *ctl, > int changed; > > mutex_lock(&chip->mutex); > - changed = !value->value.integer.value[0] != chip->dac_mute; > + changed = (!value->value.integer.value[0]) != chip->dac_mute; It seems this is because dac_mute's value is inverted 0 means true, 1 is false. I think it'd be simpler if dac_mute was converted to a standard bool (or perhaps renamed to not_muted) and this test was rewritten. Using !!value->value.integer.value[0] would also be more readable and intelligible.

1 0

Re: [alsa-devel] [PATCH] Fix logical-not-parentheses warning
by Takashi Iwai 02 Aug '15

02 Aug '15

On Sun, 02 Aug 2015 07:43:22 +0200, Tomer Barletz wrote: > > Signed-off-by: Tomer Barletz <barletz(a)gmail.com> This is rather a gcc5 problem that doesn't recognize the expression correctly. Though, this change make things worse, so I'm willing to take it. But please give the warning message itself in changelog to show exactly what is the issue. The subject doesn't enough to understand why this change is needed. thanks, Takashi > --- > sound/pci/oxygen/oxygen_mixer.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c > index 6492bca..4ca1266 100644 > --- a/sound/pci/oxygen/oxygen_mixer.c > +++ b/sound/pci/oxygen/oxygen_mixer.c > @@ -88,7 +88,7 @@ static int dac_mute_put(struct snd_kcontrol *ctl, > int changed; > > mutex_lock(&chip->mutex); > - changed = !value->value.integer.value[0] != chip->dac_mute; > + changed = (!value->value.integer.value[0]) != chip->dac_mute; > if (changed) { > chip->dac_mute = !value->value.integer.value[0]; > chip->model.update_dac_mute(chip); > -- > 2.4.3 > >

1 0

Re: [alsa-devel] [pulseaudio-discuss] Strange hardware: XFX nForce 780i SLI MCP
by Raymond Yau 02 Aug '15

02 Aug '15

> > Yesterday I got an interesting email related to dcaenc support. > > The person has an "XFX nForce 780i SLI" MCP board: http://www.nvidia.com/object/product_nforce_780i_sli_us.html , http://www.evga.com/support/manuals/files/132-CK-NF78.pdf > > As you see on page 18 of the manual, this board has no HDMI output, but there is an optical SPDIF. Strangely enough, this SPDIF is his "hdmi:CARD=NVidia,DEV=0" device. > > Given the recent discussions related to dynamically creating PCM devices for HDMI, I think you will be interested in this alsa-info output. > > Is there anything here that would need special treatment by ALSA or PulseAudio? Or, is this just a misnumbered/misnamed codec? > This can also be regarded as bug of hda_auto_parser.c which does not determine type of digital pin correctly Those internal spdif connector which can connect to an optional spdif headser or spdif input header of the graphic card for hdmi audio output http://www.intel.com/support/motherboards/desktop/sb/CS-032871.htm But the pincap of these kind of digital pin usually does not support DETECT nor HDMI

1 0

[alsa-devel] [PATCH 0/6] ASoC: Intel: Skylake: Add DSP management routines
by Vinod Koul 01 Aug '15

01 Aug '15

This series adds NHLT table support in the driver. This also adds support for dsp init, modules configuration and messaging support Hardik T Shah (1): ASoC: Intel: Skylake: Add helpers for SRC and converter modules Jeeja KP (5): ASoC: Intel: Skylake: Add NHLT support to get BE config ASoC: Intel: Skylake: Add dsp and ipc init helpers ASoC: Intel: Skylake: Add functions for DSP module configuration ASoC: Intel: Skylake: Add DSP module init and binding routines ASoC: Intel: Skylake: Add pipe management helpers sound/soc/intel/skylake/Makefile | 2 +- sound/soc/intel/skylake/skl-messages.c | 828 +++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-nhlt.c | 141 +++++ sound/soc/intel/skylake/skl-nhlt.h | 116 ++++ sound/soc/intel/skylake/skl-topology.h | 276 +++++++++ sound/soc/intel/skylake/skl-tplg-interface.h | 88 +++ sound/soc/intel/skylake/skl.h | 13 + 7 files changed, 1463 insertions(+), 1 deletion(-) create mode 100644 sound/soc/intel/skylake/skl-messages.c create mode 100644 sound/soc/intel/skylake/skl-nhlt.c create mode 100644 sound/soc/intel/skylake/skl-nhlt.h create mode 100644 sound/soc/intel/skylake/skl-topology.h create mode 100644 sound/soc/intel/skylake/skl-tplg-interface.h -- 1.9.1

2 18

[alsa-devel] [PATCH 00/27] Export I2C and OF module aliases in missing drivers
by Javier Martinez Canillas 01 Aug '15

01 Aug '15

Hello, Short version: This series add the missing MODULE_DEVICE_TABLE() for OF and I2C tables to export that information so modules have the correct aliases built-in and autoloading works correctly. Longer version: Currently it's mandatory for I2C drivers to have an I2C device ID table regardless if the device was registered using platform data or OF. This is because the I2C core needs an I2C device ID table for two reasons: 1) Match the I2C client with a I2C device ID so a struct i2c_device_id is passed to the I2C driver probe() function. 2) Export the module aliases from the I2C device ID table so userspace can auto-load the correct module. This is because i2c_device_uevent always reports a MODALIAS of the form i2c:<client->name>. Lee Jones posted a patch series [0] to solve 1) by allowing the I2C drivers to have a probe() function that does not get a i2c_device_id. The problem is that his series didn't take into account 2) so if that was merged and the I2C ID table is removed from all the drivers that don't needed it, module auto-loading will break for those. But even now there are many I2C drivers were module auto-loading is not working because of the fact that the I2C core always reports the MODALIAS as i2c:<client->name> and many developers didn't expect this. I've identified I2C drivers with 3 types of different issues: a) Those that have an i2c_table but are not exported. The match works and the correct i2c_device_id is passed on probe but since the ID table is not exported, module auto-load won't work. b) Those that have a of_table but are not exported. This is currently not an issue since even when the of_table is used to match the dev with the driver, an OF modalias is not reported by the I2C core. But if the I2C core is changed to report the MODALIAS of the form of:N*T*C as it's made by other subsystems, then module auto-load will break for these drivers. c) Those that don't have a of_table but should since are OF drivers with DT bindings doc for them. Since the I2C core does not report a OF modalias and since i2c_device_match() fallbacks to match the device part of the compatible string with the I2C device ID table, many OF drivers don't have an of_table to match. After all having a I2C device ID table is mandatory so it works without a of_table. So, in order to not make mandatory to have a I2C device ID table, at least a) and b) needs to be addressed, this series does that. c) should be fixed too since it seems wrong that a driver with a DT binding document, does not have a OF table and export it to modules. Also stripping the vendor part from the compatible string to match with the I2C devices ID table and reporting only the device name to user-space doesn't seem to be correct. I've identified at least two drivers that have the same name on their I2C device ID table so the manufacturer prefix is important. But I've not tried to fix c) yet since that is not so easy to automate due drivers not having all the information (i.e: the device name can match a documented compatible string device part but without the vendor prefix is hard to tell). I split the changes so the patches in this series are independent and can be picked individually by subsystem maintainers. Patch #27 changes the logic of i2c_device_uevent() to report an OF modalias if the device was registered using OF. But this patch is included in the series only as an RFC for illustration purposes since changing that without fixing c) will break module auto-loading for the drivers of devices registered with OF but that don't have a of_match_table. Although arguably, those drivers were relying on the assumption that a MODALIAS=i2c:<foo> would always be reported even for the OF case which is not the true on other subsystems. [0]: https://lkml.org/lkml/2014/8/28/283 Best regards, Javier Javier Martinez Canillas (27): mfd: stw481x: Export I2C module alias information spi: xcomm: Export I2C module alias information iio: Export I2C module alias information in missing drivers [media] Export I2C module alias information in missing drivers macintosh: therm_windtunnel: Export I2C module alias information misc: eeprom: Export I2C module alias information in missing drivers Input: Export I2C module alias information in missing drivers power: Export I2C module alias information in missing drivers i2c: core: Export I2C module alias information in dummy driver backlight: tosa: Export I2C module alias information [media] staging: media: lirc: Export I2C module alias information usb: phy: isp1301: Export I2C module alias information ALSA: ppc: keywest: Export I2C module alias information hwmon: (nct7904) Export I2C module alias information regulator: fan53555: Export I2C module alias information mfd: Export OF module alias information in missing drivers iio: Export OF module alias information in missing drivers hwmon: (g762) Export OF module alias information extcon: Export OF module alias information in missing drivers ASoC: Export OF module alias information in missing codec drivers rtc: Export OF module alias information in missing drivers macintosh: therm_windtunnel: Export OF module alias information leds: Export OF module alias information in missing drivers [media] smiapp: Export OF module alias information Input: touchscreen - Export OF module alias information regulator: isl9305: Export OF module alias information i2c: (RFC, don't apply) report OF style modalias when probing using DT drivers/extcon/extcon-rt8973a.c | 1 + drivers/extcon/extcon-sm5502.c | 1 + drivers/hwmon/g762.c | 1 + drivers/hwmon/nct7904.c | 1 + drivers/i2c/i2c-core.c | 9 +++++++++ drivers/iio/accel/mma8452.c | 1 + drivers/iio/accel/stk8312.c | 1 + drivers/iio/accel/stk8ba50.c | 1 + drivers/iio/light/cm32181.c | 1 + drivers/iio/light/cm3232.c | 1 + drivers/iio/light/cm36651.c | 1 + drivers/iio/light/gp2ap020a00f.c | 1 + drivers/iio/light/stk3310.c | 1 + drivers/input/misc/gp2ap002a00f.c | 1 + drivers/input/touchscreen/egalax_ts.c | 1 + drivers/input/touchscreen/goodix.c | 1 + drivers/input/touchscreen/mms114.c | 1 + drivers/leds/leds-pca963x.c | 1 + drivers/leds/leds-tca6507.c | 1 + drivers/macintosh/therm_windtunnel.c | 2 ++ drivers/media/i2c/ir-kbd-i2c.c | 1 + drivers/media/i2c/s5k6a3.c | 1 + drivers/media/i2c/smiapp/smiapp-core.c | 1 + drivers/mfd/rt5033.c | 1 + drivers/mfd/stw481x.c | 1 + drivers/mfd/tps65217.c | 1 + drivers/mfd/tps65218.c | 1 + drivers/misc/eeprom/eeprom.c | 1 + drivers/misc/eeprom/max6875.c | 1 + drivers/power/bq24190_charger.c | 1 + drivers/power/rt5033_battery.c | 2 +- drivers/regulator/fan53555.c | 1 + drivers/regulator/isl9305.c | 1 + drivers/rtc/rtc-ab-b5ze-s3.c | 1 + drivers/rtc/rtc-isl12022.c | 1 + drivers/rtc/rtc-isl12057.c | 1 + drivers/spi/spi-xcomm.c | 1 + drivers/staging/media/lirc/lirc_zilog.c | 1 + drivers/usb/phy/phy-isp1301.c | 1 + drivers/video/backlight/tosa_bl.c | 1 + sound/ppc/keywest.c | 1 + sound/soc/codecs/da9055.c | 1 + sound/soc/codecs/wm8510.c | 1 + sound/soc/codecs/wm8523.c | 1 + sound/soc/codecs/wm8580.c | 1 + 45 files changed, 54 insertions(+), 1 deletion(-) -- 2.4.3

3 7

Re: [alsa-devel] Abouut adv7511 driver
by Lars-Peter Clausen 01 Aug '15

01 Aug '15

Cc: alsa-ml On 07/31/2015 07:38 AM, Kuninori Morimoto wrote: > > Hi Lars > Cc Mark > > I noticed Analog Devices has adv7511 support on > http://wiki.analog.com/resources/tools-software/linux-drivers/drm/adv7511 > > And I tried this driver on my board, and it works. > > if my understanding was correct, latest linux kernel doesn't have > adv7511 "sound" support yet. > Thus, I would like to have it on linux kernel. > According to this driver/file, it is under GPL-2, so it is no problem > from license point of view. > > According to git log Signd-off-by, Lars/Mark are listed on this adv7511 sound driver. > but I wonder why it is not on upstream ? Are there some reason ?? > If you don't mind, I would like to send it to ML. > This is OK ? The reason why the driver is not upstream at the moment is that the way it implements things is definitely not the right way. But there is still a lot of discussion on what actually is the right way to implement audio support for HDMI devices. Once that has been sorted out the driver will hopefully go upstream. - Lars -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in

4 3

[alsa-devel] AudioFire4 quirk
by Johannes Oertel 01 Aug '15

01 Aug '15

Hi, when I try to use the snd-fireworks driver with my AudioFire4 (see the output of /proc/asound/cardX/firewire/firmware below) on my machine (kernel 4.1), it doesn't work. I am experiencing exactly the same problems as Harry with his AudioFire2 [1]; dmesg was reporting discontinuities. A few days ago, I saw Takashi Sakamoto's most recent patch for making the AudioFire2 work in this case [2]. I adapted his patch (see below) to apply to my situation with the AudioFire4 and now it is working flawlessly! However, according to the README in Takashi's GitHub repository [3], he is testing the driver with an AudioFire4 as well, so I asked him why I need this patch and he does not. He told me he is using firmware version 5.5 while I'm on 5.7.3. He said he is going to test his device with different firmware versions soon and write a patch, if needed. Finally, he asked me to post this whole issue to this list as well and that's what I have just done. :) Best regards, Johannes [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-August/080379.html [2] http://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=9c68… [3] https://github.com/takaswie/snd-firewire-improve /proc/asound/cardX/firewire/firmware: guid_hi: 0x148604 guid_lo: 0x5BA8D7BF type: 0xAF4 version: 0x0 vendor_name: Echo Digital Audio model_name: AudioFire4 dsp_version: 0x0 arm_version: 0x5070300 fpga_version: 0x3000200 flags: 0xE1 max_sample_rate: 0x17700 min_sample_rate: 0x7D00 supported_clock: 0x9 phys out: 0x6 phys in: 0x6 phys in grps: 0x2 phys in grp[0]: type 0x0, count 0x4 phys in grp[1]: type 0x1, count 0x2 phys out grps: 0x2 phys out grps[0]: type 0x0, count 0x4 phys out grps[1]: type 0x1, count 0x2 amdtp rx pcm channels 1x: 0x6 amdtp tx pcm channels 1x: 0x6 amdtp rx pcm channels 2x: 0x6 amdtp tx pcm channels 2x: 0x6 amdtp rx pcm channels 4x: 0x6 amdtp tx pcm channels 4x: 0x6 midi out ports: 0x1 midi in ports: 0x1 mixer playback channels: 0x6 mixer capture channels: 0x6 The patch: --- sound/firewire/fireworks/fireworks.c | 2 ++ sound/firewire/fireworks/fireworks.h | 1 + sound/firewire/fireworks/fireworks_stream.c | 4 ++-- 3 files changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index c670db4..544bdbf 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -250,6 +250,8 @@ efw_probe(struct fw_unit *unit, goto error; if (entry->model_id == MODEL_ECHO_AUDIOFIRE_2) efw->is_af2 = true; + if (entry->model_id == MODEL_ECHO_AUDIOFIRE_4) + efw->is_af4 = true; if (entry->model_id == MODEL_ECHO_AUDIOFIRE_9) efw->is_af9 = true; diff --git a/sound/firewire/fireworks/fireworks.h b/sound/firewire/fireworks/fireworks.h index 9493122..9a7dcbb 100644 --- a/sound/firewire/fireworks/fireworks.h +++ b/sound/firewire/fireworks/fireworks.h @@ -74,6 +74,7 @@ struct snd_efw { /* for quirks */ bool is_af2; + bool is_af4; bool is_af9; u32 firmware_version; diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c index a0762dd..182b96f 100644 --- a/sound/firewire/fireworks/fireworks_stream.c +++ b/sound/firewire/fireworks/fireworks_stream.c @@ -172,8 +172,8 @@ int snd_efw_stream_init_duplex(struct snd_efw *efw) efw->tx_stream.flags |= CIP_DBC_IS_END_EVENT; /* Fireworks reset dbc at bus reset. */ efw->tx_stream.flags |= CIP_SKIP_DBC_ZERO_CHECK; - /* AudioFire2 starts packets with non-zero dbc. */ - if (efw->is_af2) + /* AudioFire2 and AudioFire4 start packets with non-zero dbc. */ + if (efw->is_af2 || efw->is_af4) efw->tx_stream.flags |= CIP_SKIP_INIT_DBC_CHECK; /* AudioFire9 always reports wrong dbs. */ if (efw->is_af9) -- 2.4.6

2 1

[alsa-devel] Laptop internal speaker volume is apparently controlled by the headphone control (Intel HDA Panther Point)
by Dang Sananikone 01 Aug '15

01 Aug '15

Hi, I've just noticed that the internal speaker volume on my laptop is now controlled by the headphone control in alsamixer. I think this may be a recent change although I can't be sure. I'm using Arch Linux with kernel 4.1.3-1. I've uploaded the alsa info here: http://www.alsa-project.org/db/?f=f04e506dad863823b18f8f9e9d0261fdacd72695 Let me know if I can help. Kind Regards, Dang.

2 1

[alsa-devel] Using regmap_multi_reg_write() with regcache_sync()
by Sudheer 31 Jul '15

31 Jul '15

Hi Mark, We are trying to measure cold start output latency for audio playback usecase on our platform and observing that the latency is high. As a part of optimization, we have profiled regcache_sync() API when the codec is out of reset and observed that regcache_sync() takes little long time for synching all of the codec registers (around 100 registers). But, the bus connecting the codec has the capability to send multiple register writes at once. It seems regmap_multi_reg_write() can be used for triggering multiple register writes at once. So, we would like to know, if it is possible to use regmap_multi_reg_write() or a similar function with regcache_sync that intern calls bus driver's multi_reg_write API to sync to the HW, that will help in reducing the latency. Your comments/suggestions are highly appreciated. Thanks & Regards, Sudheer Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

2 1

[alsa-devel] [PATCH v3 00/13] alsa-lib topology support
by Liam Girdwood 31 Jul '15

31 Jul '15

This series adds support for parsing topology data text files and converting to the binary topology format accepted by the kernel. Changes since V2 :- o Added documentation in doxygen format describing API and text file format. o use SNDERR macro for error reporting. o Added text copy helper. o Make all parsed keywords lowercase o Assignment for some structure copies instead of memcpy(). o Fixed the private data parser to handle byte, short and word data. Changes since V1 :- o Terminate all strncpy() strings with NULL. o Use snd_pcm_format_value() instead of local lookup. o Set a maximum size for private data files. o Set capture/playback stream direction for new PCMs o Only set TLV access flags for TLV CTLs. Liam Girdwood (13): topology: uapi: Add UAPI headers for topology ABI topology: Add topology core parser. topology: Add text section parser. topology: Add PCM parser. topology: Add operations parser topology: Add private data parser topology: Add DAPM object parser topology: Add CTL parser topology: Add Channel map parser. topology: Add binary file builder. topology: autotools: Add build support for topology core topology: doxygen: Add doxygen support for topology core. conf: topology: Add topology file for broadwell audio DSP configure.ac | 11 +- doc/doxygen.cfg.in | 7 +- doc/index.doxygen | 1 + include/Makefile.am | 4 + include/sound/Makefile.am | 3 +- include/sound/asoc.h | 388 ++++++++++++++++++ include/sound/tlv.h | 23 ++ include/topology.h | 497 ++++++++++++++++++++++ src/Makefile.am | 7 + src/conf/Makefile.am | 2 +- src/conf/topology/Makefile.am | 1 + src/conf/topology/broadwell/Makefile.am | 4 + src/conf/topology/broadwell/broadwell.conf | 375 +++++++++++++++++ src/topology/Makefile.am | 19 + src/topology/builder.c | 327 +++++++++++++++ src/topology/channel.c | 122 ++++++ src/topology/ctl.c | 613 +++++++++++++++++++++++++++ src/topology/dapm.c | 562 +++++++++++++++++++++++++ src/topology/data.c | 396 ++++++++++++++++++ src/topology/elem.c | 187 +++++++++ src/topology/ops.c | 84 ++++ src/topology/parser.c | 359 ++++++++++++++++ src/topology/pcm.c | 639 +++++++++++++++++++++++++++++ src/topology/text.c | 88 ++++ src/topology/tplg_local.h | 234 +++++++++++ 25 files changed, 4948 insertions(+), 5 deletions(-) create mode 100644 include/sound/asoc.h create mode 100644 include/sound/tlv.h create mode 100644 include/topology.h create mode 100644 src/conf/topology/Makefile.am create mode 100644 src/conf/topology/broadwell/Makefile.am create mode 100644 src/conf/topology/broadwell/broadwell.conf create mode 100644 src/topology/Makefile.am create mode 100644 src/topology/builder.c create mode 100644 src/topology/channel.c create mode 100644 src/topology/ctl.c create mode 100644 src/topology/dapm.c create mode 100644 src/topology/data.c create mode 100644 src/topology/elem.c create mode 100644 src/topology/ops.c create mode 100644 src/topology/parser.c create mode 100644 src/topology/pcm.c create mode 100644 src/topology/text.c create mode 100644 src/topology/tplg_local.h -- 2.1.4

2 16

[alsa-devel] [PATCH v2 0/22] On-demand device probing
by Tomeu Vizoso 31 Jul '15

31 Jul '15

Hello, I have a problem with the panel on my Tegra Chromebook taking longer than expected to be ready during boot (Stéphane Marchesin reported what is basically the same issue in [0]), and have looked into ordered probing as a better way of solving this than moving nodes around in the DT or playing with initcall levels and linking order. While reading the thread [1] that Alexander Holler started with his series to make probing order deterministic, it occurred to me that it should be possible to achieve the same by probing devices as they are referenced by other devices. This basically reuses the information that is already implicit in the probe() implementations, saving us from refactoring existing drivers or adding information to DTBs. During review of v1 of this series Linus Walleij suggested that it should be the device driver core to make sure that dependencies are ready before probing a device. I gave this idea a try [2] but Mark Brown pointed out to the logic duplication between the resource acquisition and dependency discovery code paths (though I think it's fairly minor). To address that code duplication I experimented with Arnd's devm_probe [3] concept of having drivers declare their dependencies instead of acquiring them during probe, and while it worked [4], I don't think we end up winning anything when compared to just probing devices on-demand from resource getters. One remaining objection is to the "sprinkling" of calls to fwnode_ensure_device() in the resource getters of each subsystem, but I think it's the right thing to do given that the storage of resources is currently subsystem-specific. We could avoid the above by moving resource storage into the core, but I don't think there's a compelling case for that. I have tested this on boards with Tegra, iMX.6, Exynos and OMAP SoCs, and these patches were enough to eliminate all the deferred probes (except one in PandaBoard because omap_dma_system doesn't have a firmware node as of yet). With this series I get the kernel to output to the panel in 0.5s, instead of 2.8s. Regards, Tomeu [0] http://lists.freedesktop.org/archives/dri-devel/2014-August/066527.html [1] https://lkml.org/lkml/2014/5/12/452 [2] https://lkml.org/lkml/2015/6/17/305 [3] http://article.gmane.org/gmane.linux.ports.arm.kernel/277689 [4] https://lkml.org/lkml/2015/7/21/441 Changes in v2: - Move delay to platform.c - Use set_primary_fwnode() - Use of_node_full_name() - Move the logic for finding a platform device from its firmware node to of/platform.c as it's not needed for ACPI nodes. - Add acpi_dev_get_device() - Add fwnode_ensure_device() so the mechanism for probing devices on demand is independent of the firmware format. - Acquire regulator device lock before returning from regulator_dev_lookup() Tomeu Vizoso (22): platform: delay device-driver matches until late_initcall of/platform: Set fwnode field for new devices device property: add fwnode_get_name() of/platform: add of_platform_device_find() ACPI: add acpi_dev_get_device() device property: add fwnode_ensure_device() gpio: Probe GPIO drivers on demand gpio: Probe pinctrl devices on demand regulator: core: Reduce critical area in _regulator_get regulator: core: Probe regulators on demand drm: Probe panels on demand drm/tegra: Probe dpaux devices on demand i2c: core: Probe i2c master devices on demand pwm: Probe PWM chip devices on demand backlight: Probe backlight devices on demand usb: phy: Probe phy devices on demand clk: Probe clk providers on demand pinctrl: Probe pinctrl devices on demand phy: core: Probe phy providers on demand dma: of: Probe DMA controllers on demand power-supply: Probe power supplies on demand ASoC: core: Probe components on demand drivers/base/platform.c | 28 +++++++++++ drivers/base/property.c | 73 +++++++++++++++++++++++++++ drivers/clk/clk.c | 3 ++ drivers/dma/of-dma.c | 2 + drivers/gpio/gpiolib-of.c | 4 ++ drivers/gpu/drm/drm_panel.c | 2 + drivers/gpu/drm/tegra/dpaux.c | 2 + drivers/i2c/i2c-core.c | 2 + drivers/of/platform.c | 61 +++++++++++++++++++++++ drivers/phy/phy-core.c | 2 + drivers/pinctrl/devicetree.c | 1 + drivers/power/power_supply_core.c | 2 + drivers/pwm/core.c | 2 + drivers/regulator/core.c | 99 +++++++++++++++++++++---------------- drivers/usb/phy/phy.c | 2 + drivers/video/backlight/backlight.c | 2 + include/linux/acpi.h | 10 ++++ include/linux/of_platform.h | 1 + include/linux/property.h | 4 ++ sound/soc/soc-core.c | 6 ++- 20 files changed, 264 insertions(+), 44 deletions(-) -- 2.4.3

3 4

[alsa-devel] [PATCH] ASoC: cs4349: Fix up setting PWR_DWN bit
by Axel Lin 30 Jul '15

30 Jul '15

The PWR_DWN is Bit 7, so current code does not set the PWR_DWN bit. Fix it. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs4349.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c index 852be85..0d010c2 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -326,7 +326,7 @@ static int cs4349_runtime_suspend(struct device *dev) struct cs4349_private *cs4349 = dev_get_drvdata(dev); int ret; - ret = regmap_update_bits(cs4349->regmap, CS4349_MISC, PWR_DWN, 1); + ret = regmap_update_bits(cs4349->regmap, CS4349_MISC, PWR_DWN, PWR_DWN); if (ret < 0) return ret; -- 2.1.0

3 2

[alsa-devel] [PATCH] ASoC: cs4349: Remove unneeded NULL test for cs4349->reset_gpio
by Axel Lin 30 Jul '15

30 Jul '15

It's safe to call gpiod_set_value_cansleep() with NULL desc. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs4349.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c index a7538ae..852be85 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -300,8 +300,7 @@ static int cs4349_i2c_probe(struct i2c_client *client, if (IS_ERR(cs4349->reset_gpio)) return PTR_ERR(cs4349->reset_gpio); - if (cs4349->reset_gpio) - gpiod_set_value_cansleep(cs4349->reset_gpio, 1); + gpiod_set_value_cansleep(cs4349->reset_gpio, 1); i2c_set_clientdata(client, cs4349); @@ -316,8 +315,7 @@ static int cs4349_i2c_remove(struct i2c_client *client) snd_soc_unregister_codec(&client->dev); /* Hold down reset */ - if (cs4349->reset_gpio) - gpiod_set_value_cansleep(cs4349->reset_gpio, 0); + gpiod_set_value_cansleep(cs4349->reset_gpio, 0); return 0; } @@ -335,8 +333,7 @@ static int cs4349_runtime_suspend(struct device *dev) regcache_cache_only(cs4349->regmap, true); /* Hold down reset */ - if (cs4349->reset_gpio) - gpiod_set_value_cansleep(cs4349->reset_gpio, 0); + gpiod_set_value_cansleep(cs4349->reset_gpio, 0); return 0; } @@ -350,8 +347,7 @@ static int cs4349_runtime_resume(struct device *dev) if (ret < 0) return ret; - if (cs4349->reset_gpio) - gpiod_set_value_cansleep(cs4349->reset_gpio, 1); + gpiod_set_value_cansleep(cs4349->reset_gpio, 1); regcache_cache_only(cs4349->regmap, false); regcache_sync(cs4349->regmap); -- 2.1.0

3 2

[alsa-devel] [PATCH] ASoC: cs4349: Fix setting digital interface format
by Axel Lin 30 Jul '15

30 Jul '15

Mode Control - Register 02h Digital Interface Format (DIF[2:0]) Bits 6-4 DIF2 DIF1 DIF0 Description 0 0 0 Left-Justified, up to 24-bit data 0 0 1 I²S, up to 24-bit data 0 1 0 Right-Justified, 16-bit data 0 1 1 Right-Justified, 24-bit data 1 0 0 TDM slot 0 1 0 1 TDM slot 1 1 1 0 TDM slot 2 1 1 1 TDM slot 3 The DIF_MASK is 0x70, so current code does not correctly set the DIFx setting. Fix it. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- I assume the path "ASoC: cs4349: Don't use rtd->codec" will be applied first, so this patch is on top of that patch. sound/soc/codecs/cs4349.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c index 7f5c3c70..989289d 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -98,17 +98,16 @@ static int cs4349_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct cs4349_private *cs4349 = snd_soc_codec_get_drvdata(codec); - int mode, fmt, ret; + int fmt, ret; - mode = snd_soc_read(codec, CS4349_MODE); cs4349->rate = params_rate(params); switch (cs4349->mode) { case SND_SOC_DAIFMT_I2S: - mode |= MODE_FORMAT(DIF_I2S); + fmt = DIF_I2S; break; case SND_SOC_DAIFMT_LEFT_J: - mode |= MODE_FORMAT(DIF_LEFT_JST); + fmt = DIF_LEFT_JST; break; case SND_SOC_DAIFMT_RIGHT_J: switch (params_width(params)) { @@ -121,13 +120,13 @@ static int cs4349_pcm_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } - mode |= MODE_FORMAT(fmt); break; default: return -EINVAL; } - ret = snd_soc_write(codec, CS4349_MODE, mode); + ret = snd_soc_update_bits(codec, CS4349_MODE, DIF_MASK, + MODE_FORMAT(fmt)); if (ret < 0) return ret; -- 2.1.0

3 2

[alsa-devel] [RFC][PATCH] ALSA: control: ensure returning from blocked system-call after closing
by Takashi Sakamoto 30 Jul '15

30 Jul '15

Some operations with blocking flag for ALSA control character devices don't return to userspace when closing a file descriptor corresponding to the character device. I think this is a bug of ALSA core. While, I find no resources to assist this aspect. In a manual of close(2)/read(2), there'no description about it. On SUSv2, there's descriptions for pipe or FIFO for this behaviour, while nothing for character devices. On Linux kernel documentation, I cannot find this kind of explaination. Thus, I just stand on my experiences to use these operations for files on filesystems, sockets and virtual character devices. It may merely be due to my misunderstanding. Well, additionally, in unsubscribed state, read operation for the character device returns with EBADFD. On the other hand, in subscribed state and once the operations are going to be blocked, they don't return to userspace even if any unsubscribe ioctl(2) operations are executed. This losts a consistency about the actual effects of unsubscribe operation. It's my pleasure to discuss about these bugs (and my patch candidate to fix them). You can see these bugs with this sample program. = Sample program = #include <stdio.h> #include <stdlib.h> #include <errno.h> #include <string.h> #include <sys/types.h> #include <sys/stat.h> #include <fcntl.h> #include <unistd.h> #include <sys/ioctl.h> #include <sound/asound.h> #include <pthread.h> static void *thread_run(void *arg) { int fd = *(int *)arg; char buf[512] = {}; while (read(fd, buf, sizeof(buf)) >= 0) ; return (void *)NULL; } int main(int argc, char *argv[]) { pthread_t thread = pthread_self(); int fd; int subscribe; fd = open("/dev/snd/controlC0", O_RDONLY); if (fd < 0) { printf("%s\n", strerror(errno)); return EXIT_FAILURE; } subscribe = 1; if (ioctl(fd, SNDRV_CTL_IOCTL_SUBSCRIBE_EVENTS, &subscribe) < 0) { printf("%s\n", strerror(errno)); close(fd); return EXIT_FAILURE; } if (pthread_create(&thread, NULL, thread_run, (void *)&fd) < 0) { printf("%s\n", strerror(errno)); close(fd); return EXIT_FAILURE; } sleep(1); if (argc == 1) { close(fd); } else { subscribe = 0; if (ioctl(fd, SNDRV_CTL_IOCTL_SUBSCRIBE_EVENTS, &subscribe) < 0) { printf("%s\n", strerror(errno)); close(fd); return EXIT_FAILURE; } } /* Here, I expect blocking state of read(2) should be released. But... */ pthread_join(thread, NULL); return EXIT_SUCCESS; } Takashi Sakamoto (1): ALSA: control: add .flush operation to release blocking operation sound/core/control.c | 38 +++++++++++++++++++++++++++++++------- 1 file changed, 31 insertions(+), 7 deletions(-) -- 2.1.4

3 4

[alsa-devel] [PATCH 1/2] ASoC: cs4349: Fix suspend/resume
by Lars-Peter Clausen 30 Jul '15

30 Jul '15

dev_get_drvdata() will not return the snd_soc_runtime to which this CODEC is attached, so the current code will result in undefined behavior. To fix this just use regmap_update_bits(cs4349->regmap, ...) directly instead of snd_soc_update_bits(rtd->codec, ...). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/cs4349.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c index a8df8a74..db18799 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -328,10 +328,9 @@ static int cs4349_i2c_remove(struct i2c_client *client) static int cs4349_runtime_suspend(struct device *dev) { struct cs4349_private *cs4349 = dev_get_drvdata(dev); - struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); int ret; - ret = snd_soc_update_bits(rtd->codec, CS4349_MISC, PWR_DWN, 1); + ret = regmap_update_bits(cs4349->regmap, CS4349_MISC, PWR_DWN, 1); if (ret < 0) return ret; @@ -347,10 +346,9 @@ static int cs4349_runtime_suspend(struct device *dev) static int cs4349_runtime_resume(struct device *dev) { struct cs4349_private *cs4349 = dev_get_drvdata(dev); - struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); int ret; - ret = snd_soc_update_bits(rtd->codec, CS4349_MISC, PWR_DWN, 0); + ret = regmap_update_bits(cs4349->regmap, CS4349_MISC, PWR_DWN, 0); if (ret < 0) return ret; -- 2.1.4

3 4

[alsa-devel] [PATCH 1/2] ASoC: cs4349: Set .writeable_reg for cs4349_regmap
by Axel Lin 30 Jul '15

30 Jul '15

The first valid register index is 1 rather than 0, and the CS4349_CHIPID is readonly. So set .writeable_reg to avoid writing to these registers. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs4349.c | 20 ++++++++++++-------- 1 file changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c index f4fccc6..eb55889 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -54,14 +54,17 @@ struct cs4349_private { static bool cs4349_readable_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS4349_CHIPID: - case CS4349_MODE: - case CS4349_VMI: - case CS4349_MUTE: - case CS4349_VOLA: - case CS4349_VOLB: - case CS4349_RMPFLT: - case CS4349_MISC: + case CS4349_CHIPID ... CS4349_MISC: + return true; + default: + return false; + } +} + +static bool cs4349_writeable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS4349_MODE ... CS4349_MISC: return true; default: return false; @@ -272,6 +275,7 @@ static const struct regmap_config cs4349_regmap = { .reg_defaults = cs4349_reg_defaults, .num_reg_defaults = ARRAY_SIZE(cs4349_reg_defaults), .readable_reg = cs4349_readable_register, + .writeable_reg = cs4349_writeable_register, .cache_type = REGCACHE_RBTREE, }; -- 2.1.0

3 4

[alsa-devel] [PATCH] ALSA: hda - Fix MacBook Pro 5,2 quirk
by Takashi Iwai 30 Jul '15

30 Jul '15

MacBook Pro 5,2 with ALC889 codec had already a fixup entry, but this seems not working correctly, a fix for pin NID 0x15 is needed in addition. It's equivalent with the fixup for MacBook Air 1,1, so use this instead. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=102131 Reported-and-tested-by: Jeffery Miller <jefferym(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 18ae17ebb356..c456c04e0928 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2222,7 +2222,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x4300, "iMac 9,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC889_FIXUP_IMAC91_VREF), - SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_IMAC91_VREF), + SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_MBA11_VREF), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), -- 2.4.6

1 0

[alsa-devel] [PATCH 1/6] ASoC: fsl_ssi: enable IPG clock during AC'97 reg access
by Maciej S. Szmigiero 30 Jul '15

30 Jul '15

IPG clock have to be enabled during AC'97 CODEC register access in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 19 +++++++++++++++++++ 1 files changed, 19 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 484ff20..8185edc 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1127,10 +1127,17 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, struct regmap *regs = fsl_ac97_data->regs; unsigned int lreg; unsigned int lval; + int ret; if (reg > 0x7f) return; + ret = clk_prepare_enable(fsl_ac97_data->clk); + if (ret) { + pr_err("ac97 write clk_prepare_enable failed: %d\n", + ret); + return; + } lreg = reg << 12; regmap_write(regs, CCSR_SSI_SACADD, lreg); @@ -1141,6 +1148,8 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, regmap_update_bits(regs, CCSR_SSI_SACNT, CCSR_SSI_SACNT_RDWR_MASK, CCSR_SSI_SACNT_WR); udelay(100); + + clk_disable_unprepare(fsl_ac97_data->clk); } static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, @@ -1151,6 +1160,14 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, unsigned short val = -1; u32 reg_val; unsigned int lreg; + int ret; + + ret = clk_prepare_enable(fsl_ac97_data->clk); + if (ret) { + pr_err("ac97 read clk_prepare_enable failed: %d\n", + ret); + return -1; + } lreg = (reg & 0x7f) << 12; regmap_write(regs, CCSR_SSI_SACADD, lreg); @@ -1162,6 +1179,8 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, regmap_read(regs, CCSR_SSI_SACDAT, &reg_val); val = (reg_val >> 4) & 0xffff; + clk_disable_unprepare(fsl_ac97_data->clk); + return val; }

2 2

[alsa-devel] [PATCH 4/6] ASoC: fsl_ssi: add AC'97 ops setting check and cleanup
by Maciej S. Szmigiero 30 Jul '15

30 Jul '15

Check whether setting AC'97 ops succeeded and clean them on removal so the fsl_ssi driver can be reloaded. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 9 ++++++++- 1 files changed, 8 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 7f4f0b9..154bcf6 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1340,7 +1340,11 @@ static int fsl_ssi_probe(struct platform_device *pdev) fsl_ac97_data = ssi_private; - snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + if (ret) { + dev_err(&pdev->dev, "could not set AC'97 ops\n"); + return ret; + } } else { /* Initialize this copy of the CPU DAI driver structure */ memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template, @@ -1480,6 +1484,9 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); + if (fsl_ssi_is_ac97(ssi_private)) + snd_soc_set_ac97_ops(NULL); + return 0; }

1 0

[alsa-devel] [PATCH 2/6] ASoC: fsl_ssi: AC'97 DAI driver needs probe method too
by Maciej S. Szmigiero 30 Jul '15

30 Jul '15

AC'97 DAI driver struct need the same probe method as I2S one to setup DMA params in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 8185edc..a83b900 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1101,6 +1101,7 @@ static const struct snd_soc_component_driver fsl_ssi_component = { static struct snd_soc_dai_driver fsl_ssi_ac97_dai = { .bus_control = true, + .probe = fsl_ssi_dai_probe, .playback = { .stream_name = "AC97 Playback", .channels_min = 2,

1 0

[alsa-devel] [RFC PATCH 0/4] better support for bursty DMA usages
by Pierre-Louis Bossart 30 Jul '15

30 Jul '15

Set of patches to fix issues with hw_ptr fuzziness [1] and increased buffering w/ DSPs 1. disable rewinds to allow for new HDaudio SPIB DMA functionality (fetch up to the application pointer, rewinds not supported) 2. report max in-flight bytes to avoid problems with stale data (late wake-ups, rewinds) [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/093646.html TODO: 1. fixes and alsa-lib updates (compile-tested only for now) 2. get feedback 3. if supported, set DMA buffering based on negotiation between driver and app (capabilities vs. latency needs) Pierre-Louis Bossart (4): ALSA: core: let low-level driver or userspace disable rewinds ALSA: core: add .notify callback for pcm ops ALSA: core: add report of max dma burst ALSA: hda: add default value for max_dma_burst include/sound/pcm.h | 5 +++++ include/uapi/sound/asound.h | 6 ++++-- sound/core/pcm_lib.c | 19 +++++++++++++++++++ sound/core/pcm_native.c | 33 ++++++++++++++++++++++++++++++++- sound/pci/hda/hda_controller.c | 1 + 5 files changed, 61 insertions(+), 3 deletions(-) -- 1.9.1

6 33

Re: [alsa-devel] [ALC668]: Asus N751JK - Incorrect default pin assignment for external base speaker and external microphone not working
by Raymond Yau 30 Jul '15

30 Jul '15

> > "... after you replaced speakers.conf and lineout.conf by subwoofer.conf ..." > > I must have missed something again. I was under the impression that changing the configuration files was a workaround when the patch would be insufficient. Later today or tomorrow, I will reread this part and apply changes to the configuration files and report back. Ports: analog-output-speaker: Speakers (priority: 10000) analog-output-headphones: Headphones (priority: 9000, not available) Active Port: analog-output-speaker When you select model=dell-headset-multi, the driver create "Headphone Mic Jack" kctl with type 1 which is SND_JACK_HEADPHONE type 2 is SND_JACK_MICROPHONE type 3 is SND_JACK_HEADSET but hda jack does not use this type https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/include/s… JACK created Headphone Mic, type 1 CTRL: add: Headphone Mic Jack:0 JACK report Headphone Mic, status 0 JACK created Headset Mic Phantom, type 2 CTRL: add: Headset Mic Phantom Jack:0 JACK report Headset Mic Phantom, status 2 CTL Notify: Headset Mic Phantom Jack:0, mask=1 JACK created Internal Mic Phantom, type 2 CTRL: add: Internal Mic Phantom Jack:0 JACK report Internal Mic Phantom, status 2 CTL Notify: Internal Mic Phantom Jack:0, mask=1 JACK created Speaker Front Phantom, type 4 CTRL: add: Speaker Front Phantom Jack:0 JACK report Speaker Front Phantom, status 4 CTL Notify: Speaker Front Phantom Jack:0, mask=1 JACK created External Subwoofer, type 4 CTRL: add: External Subwoofer Jack:0 The problem of hda jack kctls are those retaskable I/O jack controls 1) multi io jacks in those 3 stack desktop 2) headphone/mic jack in asus netbook 3) dell-headset-multi combo jack the role of the jack in 1) change from input to output when "Channel mode" change from "2ch" to "4ch" or "6ch" the role of the jack in 2) and 3) change from Output to Input when "Capture Source" change from Internal Mic/Headset Mic to Headphone Mic In your case, Headphone Mic Jack kctl return the presence of HP jack except when the combo jack is set to support conventional mic (i.e. when the driver change pin ctl of hp pin node 0x15 to zero and set pin ctl of mic pin to IN), The driver already unmute the internal speaker when you change capture source to Headphone Mic The only point is whether pulseaudio still regard this Headphone Mic Jack as the availability of the headphone port when the capture source is Headphone Mic http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/?id=ca4942e89cf462…

1 0

[alsa-devel] [PATCH] ALSA: hda - Fix race between PM ops and HDA init/probe
by U. Artie Eoff 29 Jul '15

29 Jul '15

PM ops could be triggered before HDA is done initializing and cause PM to set HDA controller to D3Hot. This can result in "CORB reset timeout#2, CORBRP = 65535" and "no codecs initialized". Additionally, PM ops can be triggered before azx_probe_continue finishes (async probe). This can result in a NULL deref kernel crash. To fix this, avoid PM ops if !chip->running. Signed-off-by: U. Artie Eoff <ullysses.a.eoff(a)intel.com> --- sound/pci/hda/hda_intel.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 735bdcb04ce8..c38c68f57938 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -867,7 +867,7 @@ static int azx_suspend(struct device *dev) chip = card->private_data; hda = container_of(chip, struct hda_intel, chip); - if (chip->disabled || hda->init_failed) + if (chip->disabled || hda->init_failed || !chip->running) return 0; bus = azx_bus(chip); @@ -902,7 +902,7 @@ static int azx_resume(struct device *dev) chip = card->private_data; hda = container_of(chip, struct hda_intel, chip); - if (chip->disabled || hda->init_failed) + if (chip->disabled || hda->init_failed || !chip->running) return 0; if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL @@ -1027,7 +1027,7 @@ static int azx_runtime_idle(struct device *dev) return 0; if (!power_save_controller || !azx_has_pm_runtime(chip) || - azx_bus(chip)->codec_powered) + azx_bus(chip)->codec_powered || !chip->running) return -EBUSY; return 0; -- 2.1.0

3 3

[alsa-devel] [PATCH v2] ASoC: rt5677: Return error if devm_gpiod_get_optional return ERR_PTR
by Axel Lin 29 Jul '15

29 Jul '15

If devm_gpiod_get_optional() return ERR_PTR, it means something wrong so request gpio fails. We had better return error in such case. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/rt5677.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 8d70911..83e25a3 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -5133,14 +5133,14 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, if (IS_ERR(rt5677->pow_ldo2)) { ret = PTR_ERR(rt5677->pow_ldo2); dev_err(&i2c->dev, "Failed to request POW_LDO2: %d\n", ret); - rt5677->pow_ldo2 = 0; + return ret; } rt5677->reset_pin = devm_gpiod_get_optional(&i2c->dev, "realtek,reset", GPIOD_OUT_HIGH); if (IS_ERR(rt5677->reset_pin)) { ret = PTR_ERR(rt5677->reset_pin); dev_err(&i2c->dev, "Failed to request RESET: %d\n", ret); - rt5677->reset_pin = 0; + return ret; } if (rt5677->pow_ldo2 || rt5677->reset_pin) { -- 2.1.0

2 1

[alsa-devel] [PATCH] ASoC: tas2552: Remove unneeded NULL test for tas2552->enable_gpio
by Axel Lin 29 Jul '15

29 Jul '15

It's safe to call gpiod_set_value() with NULL desc. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/tas2552.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 2f6a65a..e3a0bca 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -493,8 +493,7 @@ static int tas2552_runtime_suspend(struct device *dev) regcache_cache_only(tas2552->regmap, true); regcache_mark_dirty(tas2552->regmap); - if (tas2552->enable_gpio) - gpiod_set_value(tas2552->enable_gpio, 0); + gpiod_set_value(tas2552->enable_gpio, 0); return 0; } @@ -503,8 +502,7 @@ static int tas2552_runtime_resume(struct device *dev) { struct tas2552_data *tas2552 = dev_get_drvdata(dev); - if (tas2552->enable_gpio) - gpiod_set_value(tas2552->enable_gpio, 1); + gpiod_set_value(tas2552->enable_gpio, 1); tas2552_sw_shutdown(tas2552, 0); @@ -585,8 +583,7 @@ static int tas2552_codec_probe(struct snd_soc_codec *codec) return ret; } - if (tas2552->enable_gpio) - gpiod_set_value(tas2552->enable_gpio, 1); + gpiod_set_value(tas2552->enable_gpio, 1); ret = pm_runtime_get_sync(codec->dev); if (ret < 0) { @@ -610,8 +607,7 @@ static int tas2552_codec_probe(struct snd_soc_codec *codec) return 0; probe_fail: - if (tas2552->enable_gpio) - gpiod_set_value(tas2552->enable_gpio, 0); + gpiod_set_value(tas2552->enable_gpio, 0); regulator_bulk_disable(ARRAY_SIZE(tas2552->supplies), tas2552->supplies); @@ -624,8 +620,7 @@ static int tas2552_codec_remove(struct snd_soc_codec *codec) pm_runtime_put(codec->dev); - if (tas2552->enable_gpio) - gpiod_set_value(tas2552->enable_gpio, 0); + gpiod_set_value(tas2552->enable_gpio, 0); return 0; }; -- 2.1.0

3 2

[alsa-devel] [RFT][PATCH 1/2] ASoC: max98088: Get rid of max98088_access table
by Axel Lin 29 Jul '15

29 Jul '15

The max98088_access table is used for look up readable/writable/volatile attributes of registers. The readable/writable/volatile registers are mostly in continuous ranges, so we can replace the max98088_access table entirely by using case range. Below is a summary of the readable/writeable/volatile registers: readable registers: 0x00 ~ 0xC9, 0xFF writeable registers: 0x03 ~ 0xC9, 0xFF volatile registers: 0x00 ~ 0x03, 0xFF Note, 0x00 should be read-only according to the datasheet. This patch reworks the implement for .readable and .volatile and also add implementation for .writable callback. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/max98088.c | 305 ++++---------------------------------------- 1 file changed, 25 insertions(+), 280 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 99c2daa..2c2df17 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -258,292 +258,36 @@ static const struct reg_default max98088_reg[] = { { 0xc9, 0x00 }, /* C9 DAI2 biquad */ }; -static struct { - int readable; - int writable; - int vol; -} max98088_access[M98088_REG_CNT] = { - { 0xFF, 0xFF, 1 }, /* 00 IRQ status */ - { 0xFF, 0x00, 1 }, /* 01 MIC status */ - { 0xFF, 0x00, 1 }, /* 02 jack status */ - { 0x1F, 0x1F, 1 }, /* 03 battery voltage */ - { 0xFF, 0xFF, 0 }, /* 04 */ - { 0xFF, 0xFF, 0 }, /* 05 */ - { 0xFF, 0xFF, 0 }, /* 06 */ - { 0xFF, 0xFF, 0 }, /* 07 */ - { 0xFF, 0xFF, 0 }, /* 08 */ - { 0xFF, 0xFF, 0 }, /* 09 */ - { 0xFF, 0xFF, 0 }, /* 0A */ - { 0xFF, 0xFF, 0 }, /* 0B */ - { 0xFF, 0xFF, 0 }, /* 0C */ - { 0xFF, 0xFF, 0 }, /* 0D */ - { 0xFF, 0xFF, 0 }, /* 0E */ - { 0xFF, 0xFF, 0 }, /* 0F interrupt enable */ - - { 0xFF, 0xFF, 0 }, /* 10 master clock */ - { 0xFF, 0xFF, 0 }, /* 11 DAI1 clock mode */ - { 0xFF, 0xFF, 0 }, /* 12 DAI1 clock control */ - { 0xFF, 0xFF, 0 }, /* 13 DAI1 clock control */ - { 0xFF, 0xFF, 0 }, /* 14 DAI1 format */ - { 0xFF, 0xFF, 0 }, /* 15 DAI1 clock */ - { 0xFF, 0xFF, 0 }, /* 16 DAI1 config */ - { 0xFF, 0xFF, 0 }, /* 17 DAI1 TDM */ - { 0xFF, 0xFF, 0 }, /* 18 DAI1 filters */ - { 0xFF, 0xFF, 0 }, /* 19 DAI2 clock mode */ - { 0xFF, 0xFF, 0 }, /* 1A DAI2 clock control */ - { 0xFF, 0xFF, 0 }, /* 1B DAI2 clock control */ - { 0xFF, 0xFF, 0 }, /* 1C DAI2 format */ - { 0xFF, 0xFF, 0 }, /* 1D DAI2 clock */ - { 0xFF, 0xFF, 0 }, /* 1E DAI2 config */ - { 0xFF, 0xFF, 0 }, /* 1F DAI2 TDM */ - - { 0xFF, 0xFF, 0 }, /* 20 DAI2 filters */ - { 0xFF, 0xFF, 0 }, /* 21 data config */ - { 0xFF, 0xFF, 0 }, /* 22 DAC mixer */ - { 0xFF, 0xFF, 0 }, /* 23 left ADC mixer */ - { 0xFF, 0xFF, 0 }, /* 24 right ADC mixer */ - { 0xFF, 0xFF, 0 }, /* 25 left HP mixer */ - { 0xFF, 0xFF, 0 }, /* 26 right HP mixer */ - { 0xFF, 0xFF, 0 }, /* 27 HP control */ - { 0xFF, 0xFF, 0 }, /* 28 left REC mixer */ - { 0xFF, 0xFF, 0 }, /* 29 right REC mixer */ - { 0xFF, 0xFF, 0 }, /* 2A REC control */ - { 0xFF, 0xFF, 0 }, /* 2B left SPK mixer */ - { 0xFF, 0xFF, 0 }, /* 2C right SPK mixer */ - { 0xFF, 0xFF, 0 }, /* 2D SPK control */ - { 0xFF, 0xFF, 0 }, /* 2E sidetone */ - { 0xFF, 0xFF, 0 }, /* 2F DAI1 playback level */ - - { 0xFF, 0xFF, 0 }, /* 30 DAI1 playback level */ - { 0xFF, 0xFF, 0 }, /* 31 DAI2 playback level */ - { 0xFF, 0xFF, 0 }, /* 32 DAI2 playbakc level */ - { 0xFF, 0xFF, 0 }, /* 33 left ADC level */ - { 0xFF, 0xFF, 0 }, /* 34 right ADC level */ - { 0xFF, 0xFF, 0 }, /* 35 MIC1 level */ - { 0xFF, 0xFF, 0 }, /* 36 MIC2 level */ - { 0xFF, 0xFF, 0 }, /* 37 INA level */ - { 0xFF, 0xFF, 0 }, /* 38 INB level */ - { 0xFF, 0xFF, 0 }, /* 39 left HP volume */ - { 0xFF, 0xFF, 0 }, /* 3A right HP volume */ - { 0xFF, 0xFF, 0 }, /* 3B left REC volume */ - { 0xFF, 0xFF, 0 }, /* 3C right REC volume */ - { 0xFF, 0xFF, 0 }, /* 3D left SPK volume */ - { 0xFF, 0xFF, 0 }, /* 3E right SPK volume */ - { 0xFF, 0xFF, 0 }, /* 3F MIC config */ - - { 0xFF, 0xFF, 0 }, /* 40 MIC threshold */ - { 0xFF, 0xFF, 0 }, /* 41 excursion limiter filter */ - { 0xFF, 0xFF, 0 }, /* 42 excursion limiter threshold */ - { 0xFF, 0xFF, 0 }, /* 43 ALC */ - { 0xFF, 0xFF, 0 }, /* 44 power limiter threshold */ - { 0xFF, 0xFF, 0 }, /* 45 power limiter config */ - { 0xFF, 0xFF, 0 }, /* 46 distortion limiter config */ - { 0xFF, 0xFF, 0 }, /* 47 audio input */ - { 0xFF, 0xFF, 0 }, /* 48 microphone */ - { 0xFF, 0xFF, 0 }, /* 49 level control */ - { 0xFF, 0xFF, 0 }, /* 4A bypass switches */ - { 0xFF, 0xFF, 0 }, /* 4B jack detect */ - { 0xFF, 0xFF, 0 }, /* 4C input enable */ - { 0xFF, 0xFF, 0 }, /* 4D output enable */ - { 0xFF, 0xFF, 0 }, /* 4E bias control */ - { 0xFF, 0xFF, 0 }, /* 4F DAC power */ - - { 0xFF, 0xFF, 0 }, /* 50 DAC power */ - { 0xFF, 0xFF, 0 }, /* 51 system */ - { 0xFF, 0xFF, 0 }, /* 52 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 53 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 54 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 55 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 56 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 57 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 58 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 59 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 5A DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 5B DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 5C DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 5D DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 5E DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 5F DAI1 EQ2 */ - - { 0xFF, 0xFF, 0 }, /* 60 DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 61 DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 62 DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 63 DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 64 DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 65 DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 66 DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 67 DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 68 DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 69 DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 6A DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 6B DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 6C DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 6D DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 6E DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 6F DAI1 EQ3 */ - - { 0xFF, 0xFF, 0 }, /* 70 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 71 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 72 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 73 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 74 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 75 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 76 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 77 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 78 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 79 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 7A DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 7B DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 7C DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 7D DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 7E DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 7F DAI1 EQ5 */ - - { 0xFF, 0xFF, 0 }, /* 80 DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 81 DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 82 DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 83 DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 84 DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 85 DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 86 DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 87 DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 88 DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 89 DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 8A DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 8B DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 8C DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 8D DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 8E DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 8F DAI2 EQ2 */ - - { 0xFF, 0xFF, 0 }, /* 90 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 91 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 92 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 93 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 94 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 95 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 96 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 97 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 98 DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 99 DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 9A DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 9B DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 9C DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 9D DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 9E DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 9F DAI2 EQ3 */ - - { 0xFF, 0xFF, 0 }, /* A0 DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* A1 DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* A2 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A3 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A4 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A5 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A6 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A7 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A8 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A9 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* AA DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* AB DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* AC DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* AD DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* AE DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* AF DAI2 EQ5 */ - - { 0xFF, 0xFF, 0 }, /* B0 DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* B1 DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* B2 DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* B3 DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* B4 DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* B5 DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* B6 DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* B7 DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* B8 DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* B9 DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* BA DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* BB DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* BC DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* BD DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* BE DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* BF DAI1 biquad */ - - { 0xFF, 0xFF, 0 }, /* C0 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C1 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C2 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C3 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C4 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C5 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C6 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C7 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C8 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C9 DAI2 biquad */ - { 0x00, 0x00, 0 }, /* CA */ - { 0x00, 0x00, 0 }, /* CB */ - { 0x00, 0x00, 0 }, /* CC */ - { 0x00, 0x00, 0 }, /* CD */ - { 0x00, 0x00, 0 }, /* CE */ - { 0x00, 0x00, 0 }, /* CF */ - - { 0x00, 0x00, 0 }, /* D0 */ - { 0x00, 0x00, 0 }, /* D1 */ - { 0x00, 0x00, 0 }, /* D2 */ - { 0x00, 0x00, 0 }, /* D3 */ - { 0x00, 0x00, 0 }, /* D4 */ - { 0x00, 0x00, 0 }, /* D5 */ - { 0x00, 0x00, 0 }, /* D6 */ - { 0x00, 0x00, 0 }, /* D7 */ - { 0x00, 0x00, 0 }, /* D8 */ - { 0x00, 0x00, 0 }, /* D9 */ - { 0x00, 0x00, 0 }, /* DA */ - { 0x00, 0x00, 0 }, /* DB */ - { 0x00, 0x00, 0 }, /* DC */ - { 0x00, 0x00, 0 }, /* DD */ - { 0x00, 0x00, 0 }, /* DE */ - { 0x00, 0x00, 0 }, /* DF */ - - { 0x00, 0x00, 0 }, /* E0 */ - { 0x00, 0x00, 0 }, /* E1 */ - { 0x00, 0x00, 0 }, /* E2 */ - { 0x00, 0x00, 0 }, /* E3 */ - { 0x00, 0x00, 0 }, /* E4 */ - { 0x00, 0x00, 0 }, /* E5 */ - { 0x00, 0x00, 0 }, /* E6 */ - { 0x00, 0x00, 0 }, /* E7 */ - { 0x00, 0x00, 0 }, /* E8 */ - { 0x00, 0x00, 0 }, /* E9 */ - { 0x00, 0x00, 0 }, /* EA */ - { 0x00, 0x00, 0 }, /* EB */ - { 0x00, 0x00, 0 }, /* EC */ - { 0x00, 0x00, 0 }, /* ED */ - { 0x00, 0x00, 0 }, /* EE */ - { 0x00, 0x00, 0 }, /* EF */ - - { 0x00, 0x00, 0 }, /* F0 */ - { 0x00, 0x00, 0 }, /* F1 */ - { 0x00, 0x00, 0 }, /* F2 */ - { 0x00, 0x00, 0 }, /* F3 */ - { 0x00, 0x00, 0 }, /* F4 */ - { 0x00, 0x00, 0 }, /* F5 */ - { 0x00, 0x00, 0 }, /* F6 */ - { 0x00, 0x00, 0 }, /* F7 */ - { 0x00, 0x00, 0 }, /* F8 */ - { 0x00, 0x00, 0 }, /* F9 */ - { 0x00, 0x00, 0 }, /* FA */ - { 0x00, 0x00, 0 }, /* FB */ - { 0x00, 0x00, 0 }, /* FC */ - { 0x00, 0x00, 0 }, /* FD */ - { 0x00, 0x00, 0 }, /* FE */ - { 0xFF, 0x00, 1 }, /* FF */ -}; - static bool max98088_readable_register(struct device *dev, unsigned int reg) { - return max98088_access[reg].readable; + switch (reg) { + case M98088_REG_00_IRQ_STATUS ... 0xC9: + case M98088_REG_FF_REV_ID: + return true; + default: + return false; + } +} + +static bool max98088_writeable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case M98088_REG_03_BATTERY_VOLTAGE ... 0xC9: + return true; + default: + return false; + } } static bool max98088_volatile_register(struct device *dev, unsigned int reg) { - return max98088_access[reg].vol; + switch (reg) { + case M98088_REG_00_IRQ_STATUS ... M98088_REG_03_BATTERY_VOLTAGE: + case M98088_REG_FF_REV_ID: + return true; + default: + return false; + } } static const struct regmap_config max98088_regmap = { @@ -551,6 +295,7 @@ static const struct regmap_config max98088_regmap = { .val_bits = 8, .readable_reg = max98088_readable_register, + .writeable_reg = max98088_writeable_register, .volatile_reg = max98088_volatile_register, .max_register = 0xff, -- 2.1.0

2 3

[alsa-devel] [PATCH] ASoC: wm8983: Get rid of wm8983_access_masks table
by Axel Lin 29 Jul '15

29 Jul '15

The max8983_access table is used for look up readable/writable attributes of registers. The writable registers are mostly in continuous ranges, so we can replace max8983_access table by using case range. The read fields are all 0, so just drop implement of .readable callback. Also set .max_register setting. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/wm8983.c | 80 +++++++++-------------------------------------- 1 file changed, 14 insertions(+), 66 deletions(-) diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index f6861cc..f3193fb 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -84,66 +84,6 @@ static const struct reg_default wm8983_defaults[] = { { 0x3D, 0x0000 }, /* R61 - BIAS CTRL */ }; -static const struct wm8983_reg_access { - u16 read; /* Mask of readable bits */ - u16 write; /* Mask of writable bits */ -} wm8983_access_masks[WM8983_MAX_REGISTER + 1] = { - [0x00] = { 0x0000, 0x01FF }, /* R0 - Software Reset */ - [0x01] = { 0x0000, 0x01FF }, /* R1 - Power management 1 */ - [0x02] = { 0x0000, 0x01FF }, /* R2 - Power management 2 */ - [0x03] = { 0x0000, 0x01EF }, /* R3 - Power management 3 */ - [0x04] = { 0x0000, 0x01FF }, /* R4 - Audio Interface */ - [0x05] = { 0x0000, 0x003F }, /* R5 - Companding control */ - [0x06] = { 0x0000, 0x01FD }, /* R6 - Clock Gen control */ - [0x07] = { 0x0000, 0x000F }, /* R7 - Additional control */ - [0x08] = { 0x0000, 0x003F }, /* R8 - GPIO Control */ - [0x09] = { 0x0000, 0x0070 }, /* R9 - Jack Detect Control 1 */ - [0x0A] = { 0x0000, 0x004F }, /* R10 - DAC Control */ - [0x0B] = { 0x0000, 0x01FF }, /* R11 - Left DAC digital Vol */ - [0x0C] = { 0x0000, 0x01FF }, /* R12 - Right DAC digital vol */ - [0x0D] = { 0x0000, 0x00FF }, /* R13 - Jack Detect Control 2 */ - [0x0E] = { 0x0000, 0x01FB }, /* R14 - ADC Control */ - [0x0F] = { 0x0000, 0x01FF }, /* R15 - Left ADC Digital Vol */ - [0x10] = { 0x0000, 0x01FF }, /* R16 - Right ADC Digital Vol */ - [0x12] = { 0x0000, 0x017F }, /* R18 - EQ1 - low shelf */ - [0x13] = { 0x0000, 0x017F }, /* R19 - EQ2 - peak 1 */ - [0x14] = { 0x0000, 0x017F }, /* R20 - EQ3 - peak 2 */ - [0x15] = { 0x0000, 0x017F }, /* R21 - EQ4 - peak 3 */ - [0x16] = { 0x0000, 0x007F }, /* R22 - EQ5 - high shelf */ - [0x18] = { 0x0000, 0x01FF }, /* R24 - DAC Limiter 1 */ - [0x19] = { 0x0000, 0x007F }, /* R25 - DAC Limiter 2 */ - [0x1B] = { 0x0000, 0x01FF }, /* R27 - Notch Filter 1 */ - [0x1C] = { 0x0000, 0x017F }, /* R28 - Notch Filter 2 */ - [0x1D] = { 0x0000, 0x017F }, /* R29 - Notch Filter 3 */ - [0x1E] = { 0x0000, 0x017F }, /* R30 - Notch Filter 4 */ - [0x20] = { 0x0000, 0x01BF }, /* R32 - ALC control 1 */ - [0x21] = { 0x0000, 0x00FF }, /* R33 - ALC control 2 */ - [0x22] = { 0x0000, 0x01FF }, /* R34 - ALC control 3 */ - [0x23] = { 0x0000, 0x000F }, /* R35 - Noise Gate */ - [0x24] = { 0x0000, 0x001F }, /* R36 - PLL N */ - [0x25] = { 0x0000, 0x003F }, /* R37 - PLL K 1 */ - [0x26] = { 0x0000, 0x01FF }, /* R38 - PLL K 2 */ - [0x27] = { 0x0000, 0x01FF }, /* R39 - PLL K 3 */ - [0x29] = { 0x0000, 0x000F }, /* R41 - 3D control */ - [0x2A] = { 0x0000, 0x01E7 }, /* R42 - OUT4 to ADC */ - [0x2B] = { 0x0000, 0x01BF }, /* R43 - Beep control */ - [0x2C] = { 0x0000, 0x0177 }, /* R44 - Input ctrl */ - [0x2D] = { 0x0000, 0x01FF }, /* R45 - Left INP PGA gain ctrl */ - [0x2E] = { 0x0000, 0x01FF }, /* R46 - Right INP PGA gain ctrl */ - [0x2F] = { 0x0000, 0x0177 }, /* R47 - Left ADC BOOST ctrl */ - [0x30] = { 0x0000, 0x0177 }, /* R48 - Right ADC BOOST ctrl */ - [0x31] = { 0x0000, 0x007F }, /* R49 - Output ctrl */ - [0x32] = { 0x0000, 0x01FF }, /* R50 - Left mixer ctrl */ - [0x33] = { 0x0000, 0x01FF }, /* R51 - Right mixer ctrl */ - [0x34] = { 0x0000, 0x01FF }, /* R52 - LOUT1 (HP) volume ctrl */ - [0x35] = { 0x0000, 0x01FF }, /* R53 - ROUT1 (HP) volume ctrl */ - [0x36] = { 0x0000, 0x01FF }, /* R54 - LOUT2 (SPK) volume ctrl */ - [0x37] = { 0x0000, 0x01FF }, /* R55 - ROUT2 (SPK) volume ctrl */ - [0x38] = { 0x0000, 0x004F }, /* R56 - OUT3 mixer ctrl */ - [0x39] = { 0x0000, 0x00FF }, /* R57 - OUT4 (MONO) mix ctrl */ - [0x3D] = { 0x0000, 0x0100 } /* R61 - BIAS CTRL */ -}; - /* vol/gain update regs */ static const int vol_update_regs[] = { WM8983_LEFT_DAC_DIGITAL_VOL, @@ -605,12 +545,19 @@ static int eqmode_put(struct snd_kcontrol *kcontrol, return 0; } -static bool wm8983_readable(struct device *dev, unsigned int reg) +static bool wm8983_writeable(struct device *dev, unsigned int reg) { - if (reg > WM8983_MAX_REGISTER) - return 0; - - return wm8983_access_masks[reg].read != 0; + switch (reg) { + case WM8983_SOFTWARE_RESET ... WM8983_RIGHT_ADC_DIGITAL_VOL: + case WM8983_EQ1_LOW_SHELF ... WM8983_DAC_LIMITER_2: + case WM8983_NOTCH_FILTER_1 ... WM8983_NOTCH_FILTER_4: + case WM8983_ALC_CONTROL_1 ... WM8983_PLL_K_3: + case WM8983_3D_CONTROL ... WM8983_OUT4_MONO_MIX_CTRL: + case WM8983_BIAS_CTRL: + return true; + default: + return false; + } } static int wm8983_dac_mute(struct snd_soc_dai *dai, int mute) @@ -1048,8 +995,9 @@ static const struct regmap_config wm8983_regmap = { .reg_defaults = wm8983_defaults, .num_reg_defaults = ARRAY_SIZE(wm8983_defaults), .cache_type = REGCACHE_RBTREE, + .max_register = WM8983_MAX_REGISTER, - .readable_reg = wm8983_readable, + .writeable_reg = wm8983_writeable, }; #if defined(CONFIG_SPI_MASTER) -- 2.1.0

2 1

[alsa-devel] [PATCH 1/2] ASoC: Intel: haswell: fix initialize 'NULL device *' issue
by Jie Yang 29 Jul '15

29 Jul '15

Add initialization for sst_hsw.dev at init stage, which fix the 'NULL device *' warning issues. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/haswell/sst-haswell-ipc.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index f95f271..f6efa9d 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -2119,6 +2119,8 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) if (hsw == NULL) return -ENOMEM; + hsw->dev = dev; + ipc = &hsw->ipc; ipc->dev = dev; ipc->ops.tx_msg = hsw_tx_msg; -- 1.9.1

2 3

[alsa-devel] [PATCH 0/6] ASoC: Fix sparse warnings
by Lars-Peter Clausen 29 Jul '15

29 Jul '15

Hi, This series fixes a couple of sparse warnings that have appeared recently in the ASoC tree. Non of them are critical mostly just missing annotations or storage qualifiers. It still makes sense to fix those though since it reduces the noise that has to be filtered through when checking new code for new (potentially critical) warnings. - Lars Lars-Peter Clausen (6): ASoC: wm8804: Drop dublicate const ASoC: lpass-ipq806x: Staticise local symbols ASoC: rcar ctu: Staticise local symbols ASoC: uniperf: Add missing __iomem annotation ASoC: uniperf: Staticise local symbols ASoC: xtfpga-i2s: Add missing __rcu annotation sound/soc/codecs/wm8804.c | 2 +- sound/soc/qcom/lpass-ipq806x.c | 2 +- sound/soc/sh/rcar/ctu.c | 2 +- sound/soc/sti/uniperif.h | 2 +- sound/soc/sti/uniperif_player.c | 2 +- sound/soc/xtensa/xtfpga-i2s.c | 2 +- 6 files changed, 6 insertions(+), 6 deletions(-) -- 2.1.4

7 18

[alsa-devel] [PATCH] ASoC: max98090: Simplify max98090_readable_register implementation
by Axel Lin 29 Jul '15

29 Jul '15

The readable registers are in consecutive ranges: 0x01 ~ 0x03, 0x0D ~ 0xD1, 0xFF So simplify the implementation by specifying a range of consecutive values in a single case label. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/max98090.c | 71 ++------------------------------------------- 1 file changed, 2 insertions(+), 69 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 91c9f9f..315c71f 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -267,75 +267,8 @@ static bool max98090_volatile_register(struct device *dev, unsigned int reg) static bool max98090_readable_register(struct device *dev, unsigned int reg) { switch (reg) { - case M98090_REG_DEVICE_STATUS: - case M98090_REG_JACK_STATUS: - case M98090_REG_INTERRUPT_S: - case M98090_REG_RESERVED: - case M98090_REG_LINE_INPUT_CONFIG: - case M98090_REG_LINE_INPUT_LEVEL: - case M98090_REG_INPUT_MODE: - case M98090_REG_MIC1_INPUT_LEVEL: - case M98090_REG_MIC2_INPUT_LEVEL: - case M98090_REG_MIC_BIAS_VOLTAGE: - case M98090_REG_DIGITAL_MIC_ENABLE: - case M98090_REG_DIGITAL_MIC_CONFIG: - case M98090_REG_LEFT_ADC_MIXER: - case M98090_REG_RIGHT_ADC_MIXER: - case M98090_REG_LEFT_ADC_LEVEL: - case M98090_REG_RIGHT_ADC_LEVEL: - case M98090_REG_ADC_BIQUAD_LEVEL: - case M98090_REG_ADC_SIDETONE: - case M98090_REG_SYSTEM_CLOCK: - case M98090_REG_CLOCK_MODE: - case M98090_REG_CLOCK_RATIO_NI_MSB: - case M98090_REG_CLOCK_RATIO_NI_LSB: - case M98090_REG_CLOCK_RATIO_MI_MSB: - case M98090_REG_CLOCK_RATIO_MI_LSB: - case M98090_REG_MASTER_MODE: - case M98090_REG_INTERFACE_FORMAT: - case M98090_REG_TDM_CONTROL: - case M98090_REG_TDM_FORMAT: - case M98090_REG_IO_CONFIGURATION: - case M98090_REG_FILTER_CONFIG: - case M98090_REG_DAI_PLAYBACK_LEVEL: - case M98090_REG_DAI_PLAYBACK_LEVEL_EQ: - case M98090_REG_LEFT_HP_MIXER: - case M98090_REG_RIGHT_HP_MIXER: - case M98090_REG_HP_CONTROL: - case M98090_REG_LEFT_HP_VOLUME: - case M98090_REG_RIGHT_HP_VOLUME: - case M98090_REG_LEFT_SPK_MIXER: - case M98090_REG_RIGHT_SPK_MIXER: - case M98090_REG_SPK_CONTROL: - case M98090_REG_LEFT_SPK_VOLUME: - case M98090_REG_RIGHT_SPK_VOLUME: - case M98090_REG_DRC_TIMING: - case M98090_REG_DRC_COMPRESSOR: - case M98090_REG_DRC_EXPANDER: - case M98090_REG_DRC_GAIN: - case M98090_REG_RCV_LOUTL_MIXER: - case M98090_REG_RCV_LOUTL_CONTROL: - case M98090_REG_RCV_LOUTL_VOLUME: - case M98090_REG_LOUTR_MIXER: - case M98090_REG_LOUTR_CONTROL: - case M98090_REG_LOUTR_VOLUME: - case M98090_REG_JACK_DETECT: - case M98090_REG_INPUT_ENABLE: - case M98090_REG_OUTPUT_ENABLE: - case M98090_REG_LEVEL_CONTROL: - case M98090_REG_DSP_FILTER_ENABLE: - case M98090_REG_BIAS_CONTROL: - case M98090_REG_DAC_CONTROL: - case M98090_REG_ADC_CONTROL: - case M98090_REG_DEVICE_SHUTDOWN: - case M98090_REG_EQUALIZER_BASE ... M98090_REG_EQUALIZER_BASE + 0x68: - case M98090_REG_RECORD_BIQUAD_BASE ... M98090_REG_RECORD_BIQUAD_BASE + 0x0E: - case M98090_REG_DMIC3_VOLUME: - case M98090_REG_DMIC4_VOLUME: - case M98090_REG_DMIC34_BQ_PREATTEN: - case M98090_REG_RECORD_TDM_SLOT: - case M98090_REG_SAMPLE_RATE: - case M98090_REG_DMIC34_BIQUAD_BASE ... M98090_REG_DMIC34_BIQUAD_BASE + 0x0E: + case M98090_REG_DEVICE_STATUS ... M98090_REG_INTERRUPT_S: + case M98090_REG_LINE_INPUT_CONFIG ... 0xD1: case M98090_REG_REVISION_ID: return true; default: -- 2.1.0

3 3

[alsa-devel] [PATCH] sound/usb: add dB range mapping for some devices.
by Yao-Wen Mao 29 Jul '15

29 Jul '15

Add the correct dB ranges of Bose Companion 5 and Drangonfly DAC 1.2. Signed-off-by: Yao-Wen Mao <yaowen(a)google.com> --- sound/usb/mixer_maps.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index e5000da..f0c9e49 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -339,6 +339,19 @@ static struct usbmix_name_map gamecom780_map[] = { static const struct usbmix_name_map scms_usb3318_map[] = { { 10, NULL }, { 0 } + +/* Bose companion 5, the dB conversion factor is 16 instead of 256 */ +static struct usbmix_dB_map bose_companion5_dB = {-5006, -6}; +static struct usbmix_name_map bose_companion5_map[] = { + { 3, NULL, .dB = &bose_companion5_dB }, + { 0 } /* terminator */ +}; + +/* Dragonfly DAC 1.2, the dB conversion factor is 1 instead of 256 */ +static struct usbmix_dB_map dragonfly_1_2_dB = {0, 5000}; +static struct usbmix_name_map dragonfly_1_2_map[] = { + { 7, NULL, .dB = &dragonfly_1_2_dB }, + { 0 } /* terminator */ }; /* @@ -451,6 +464,16 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x25c4, 0x0003), .map = scms_usb3318_map, }, + { + /* Bose Companion 5 */ + .id = USB_ID(0x05a7, 0x1020), + .map = bose_companion5_map, + }, + { + /* Dragonfly DAC 1.2 */ + .id = USB_ID(0x21b4, 0x0081), + .map = dragonfly_1_2_map, + }, { 0 } /* terminator */ }; -- 2.5.0.rc2.392.g76e840b

2 2

Re: [alsa-devel] Bug on Dell Vostro 5480 (Intel Wildcat) kernel driver
by Paulo Roberto de Oliveira Castro 29 Jul '15

29 Jul '15

I'm not an expert on applying patches to the kernel (never did that before), therefore, I don't know if I did everything right. What I did was: downloaded 4.2-rc4 from kernel.org[1], added the line by hand (couldn't apply the patch using patch), copied the config file from /boot/ (the config fedora 22 uses for its kernel) to .config on the root of the kernel directory, runned "make oldconfig" (typing ENTER for every question, is that right?), "make", "make modules_install", "make install". Rebooted, and chose the new kernel I just installed. The sound worked like a charm, the speakers worked as expected, and the volume was as loud as on Windows. Is the patch going to be applied on master? Any chance that fix reaches my Fedora 22 (with backport to older kernel versions)? Thanks for the help, Paulo [1] https://www.kernel.org/pub/linux/kernel/v4.x/testing/linux-4.2-rc4.tar.xz 2015-07-28 10:49 GMT-03:00 Paulo Roberto de Oliveira Castro < p.oliveira.castro(a)gmail.com>: > > > ---------- Forwarded message --------- > From: Takashi Iwai <tiwai(a)suse.de> > Date: Tue, Jul 28, 2015, 03:35 > Subject: Re: [alsa-devel] Bug on Dell Vostro 5480 (Intel Wildcat) kernel > driver > To: Paulo Roberto de Oliveira Castro <p.oliveira.castro(a)gmail.com> > Cc: <alsa-devel(a)alsa-project.org> > > > On Tue, 28 Jul 2015 05:35:34 +0200, > Paulo Roberto de Oliveira Castro wrote: > > > > Hi everyone, > > > > After installing Fedora 22 on my Dell Vostro 5480, the speakers don't > work, > > and the sound only comes out of the subwoofer. Here is the output of > > alsa-info.sh. After talking to "debianuser" on IRC, we found this > > workaround: > > > > sudo hda-verb /dev/snd/hwC1D0 0x14 SET_PIN_WIDGET_CONTROL 0x40 # enable, > > set it to OUT > > sudo hda-verb /dev/snd/hwC1D0 0x14 SET_CONNECT_SEL 0 # switch to 0x0c > > sudo hda-verb /dev/snd/hwC1D0 0x14 SET_AMP_GAIN_MUTE 0xa000 # unmute left > > sudo hda-verb /dev/snd/hwC1D0 0x14 SET_AMP_GAIN_MUTE 0x9000 # unmute > right > > > > We suspect that it's a kernel driver bug. This makes the speakers work > > alongside the headphones (both at the same time) but I need help making > it > > working when headphones are not plugged in. > > Does the patch below work? Try with 4.2-rc kernel at best. > > > Takashi > > --- > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index 4ae877c3b6a1..18ae17ebb356 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -5185,6 +5185,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] > = { > SND_PCI_QUIRK(0x1028, 0x064a, "Dell", > ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), > SND_PCI_QUIRK(0x1028, 0x064b, "Dell", > ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), > SND_PCI_QUIRK(0x1028, 0x0665, "Dell XPS 13", > ALC288_FIXUP_DELL_XPS_13), > + SND_PCI_QUIRK(0x1028, 0x069a, "Dell Vostro 5480", > ALC290_FIXUP_SUBWOOFER_HSJACK), > SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", > ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), > SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", > ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), > SND_PCI_QUIRK(0x1028, 0x06da, "Dell", > ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), >

2 1

[alsa-devel] [PATCH] ASoC: dapm: fix module unload/reload failed issue by reset kcontrol's widget list
by Jie Yang 29 Jul '15

29 Jul '15

We need reset the related kcontrol's widget list during the freeing of a widget, otherwise, the widget list may be outdated and reference to it may introduce errors(they usually occurs during driver modules unloading/reloading). Here adding a func dapm_update_kcontrols_of_freeing_widget and call it at dapm_update_kcontrols_of_freeing_widget, to fix those issues. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/soc-dapm.c | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index aa327c9..1ad8144 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2305,6 +2305,35 @@ static void dapm_free_path(struct snd_soc_dapm_path *path) kfree(path); } +/* update all dapm kcontrols that related to a widget which being freed*/ +static void dapm_update_kcontrols_of_freeing_widget(struct snd_soc_dapm_widget * w) +{ + struct dapm_kcontrol_data *data = NULL; + int i, j, n; + + if (w && w->kcontrols) { + for (i = 0; i < w->num_kcontrols; i++) { + if (w->kcontrols[i] == NULL) + continue; + data = snd_kcontrol_chip(w->kcontrols[i]); + if (data == NULL) + continue; + for (j = 0, n = 0; j < data->wlist->num_widgets; j++) { + if (data->wlist->widgets[j] == w) { + data->wlist->widgets[j] = NULL; + n++; + } + } + + if (n) { + data->wlist->num_widgets -= n; + data->widget = NULL; + } + } + } + +} + /* free all dapm widgets and resources */ static void dapm_free_widgets(struct snd_soc_dapm_context *dapm) { @@ -2326,6 +2355,8 @@ static void dapm_free_widgets(struct snd_soc_dapm_context *dapm) list_for_each_entry_safe(p, next_p, &w->sinks, list_source) dapm_free_path(p); + dapm_update_kcontrols_of_freeing_widget(w); + kfree(w->kcontrols); kfree(w->name); kfree(w); -- 1.9.1

4 4

[alsa-devel] [PATCH 1/2] ALSA: hda - skip runtime PM if async probe is not finished
by U. Artie Eoff 28 Jul '15

28 Jul '15

Crash can occur if runtime PM is triggered before the async probe finishes via the azx_firmware_cb when CONFIG_SND_HDA_PATCH_LOADER is defined. Don't execute PM ops if the async probe has not completed yet. The probe is finished when chip->running is true. Signed-off-by: U. Artie Eoff <ullysses.a.eoff(a)intel.com> --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 735bdcb04ce8..b729b25a6ad6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1023,7 +1023,7 @@ static int azx_runtime_idle(struct device *dev) chip = card->private_data; hda = container_of(chip, struct hda_intel, chip); - if (chip->disabled || hda->init_failed) + if (chip->disabled || hda->init_failed || !chip->running) return 0; if (!power_save_controller || !azx_has_pm_runtime(chip) || -- 1.9.3

4 9

[alsa-devel] [PATCH v13 0/3] ASoC: tda998x: add a codec to the HDMI transmitter
by Jean-Francois Moine 28 Jul '15

28 Jul '15

This patch series asks for Russell King's patch http://article.gmane.org/gmane.linux.ports.arm.kernel/411574 drm/edid: add function to help find SADs to be applayed. v13: - rebase on 4.2.0-rc3 - remove Russell's patches v12: - use Russell King's DRM ELD helper (Mark Brown) v11: - reduce the patch series to adding the tda998x codec only v10: - add the generic dt-card - define the audio ports from a DT graph of ports (Russell King) - reuse HDMI constants (Andrew Jackson - Jyri Sarha) - alloc rate_constraints in codec (Jyri Sarha) - fix bad number of channels (Jyri Sarha) - correct codec generation from config (Russell King - Jyri Sarha) - no module init/exit (Russell King) v9: - back to a TDA998x specific CODEC - more comments - change magic values to constants v8: - change some comments about the patches v7: - remove the change of the K predivider (Jyri Sarha) - add S24_3LE and S32_LE as possible audio formats (Jyri Sarha) - don't move the struct priv2 definition and use the slave encoder private data as the device private data (Russell King) - remove the useless request_module (Russell King/Mark Brown) - don't lock the HDMI module (Russell King) - use platform_device_unregister to remove the codec (Russell King) v6: - extend the HDMI CODEC instead of using a specific CODEC v5: - use the TDA998x private data instead of a specific area for the CODEC interface - the CODEC is TDA998x specific (Mark Brown) v4: - remove all the TDA998x specific stuff from the CODEC - move the EDID scan from the CODEC to the TDA998x - move the CODEC to sound/soc (Mark Brown) - update the audio_sample_rate from the EDID (Andrew Jackson) v3: fix bad rate (Andrew Jackson) v2: check double stream start (Mark Brown) Jean-Francois Moine (3): drm/i2c: tda998x: Add support of a DT graph of ports drm/i2c: tda998x: Change drvdata for audio extension ASoC: tda998x: add a codec to the HDMI transmitter .../devicetree/bindings/drm/i2c/tda998x.txt | 51 ++++++ drivers/gpu/drm/i2c/tda998x_drv.c | 178 +++++++++++++++++++-- include/sound/tda998x.h | 23 +++ sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tda998x.c | 132 +++++++++++++++ 6 files changed, 381 insertions(+), 11 deletions(-) create mode 100644 include/sound/tda998x.h create mode 100644 sound/soc/codecs/tda998x.c -- 2.1.4

4 12

[alsa-devel] rockchip rk3288 registration
by Michael Trimarchi 28 Jul '15

28 Jul '15

Hi I have created a simple sound card to test registration on i2s but I have some problems that after a while in SND_SOC_DAIFMT_CBM_CFM the device stop to register and receive a input/output error. It means that I can not receive anymore from the device arecord -c 2 -r 96000 -f S32_LE -D hw:0 > record.pcm Recording WAVE 'stdin' : Signed 32 bit Little Endian, Rate 96000 Hz, Stereo arecord: pcm_read:2031: read error: Input/output error The result size is a bit random. I have no problem on playback until now. Any idea? Michael

1 1

[alsa-devel] Running PJSIP for voip calls in alsa
by Srinivasan S 28 Jul '15

28 Jul '15

Dear Clemens, Could you please let me know whether the below issue is in PJSIP or alsa or driver or in the ti sdk8 BSP's (rootfs) 1. We are running PJSIP for establishing VOIP calls on ti sdk 8 BSP's (linux kernel 3.14), we are seeing the below error messages in the PJSIP, Could you please let me know what could be issue ? 08:32:42.082 Master/sound Underflow, buf_cnt=0, will generate 1 frame 08:32:42.083 Master/sound Underflow, buf_cnt=0, will generate 1 frame 08:32:42.086 Master/sound Underflow, buf_cnt=0, will generate 1 frame 08:32:42.088 Master/sound Underflow, buf_cnt=0, will generate 1 frame 08:32:42.088 Master/sound Underflow, buf_cnt=0, will generate 1 frame 08:32:42.091 Master/sound Underflow, buf_cnt=0, will generate 1 frame 08:32:42.112 Master/sound Underflow, buf_cnt=0, will generate 1 frame In failure call pjsip reports Master/sound Underflow continuously, At this point the control stays at pjsip and application will wait for it and all functionality in idle state at application and after this, no other feature will work until restart the application. The above problem observed with new rootfs (ti sdk 8) but when we tried with old rootfs (ti sdk 7) we are able to make calls with no problem. Few calls are long duration calls(33 and 55 minutes). With ti sdk 8 kernel, I tried replacing even the alsa libraries from ti sdk 7 rootfs to ti sdk 8 rootfs, but still the issue persists, /usr/lib/libasound.la /usr/lib/libasound.so /usr/lib/alsa-lib/smixer/smixer-ac97.la /usr/lib/alsa-lib/smixer/smixer-ac97.so /usr/lib/alsa-lib/smixer/smixer-hda.la /usr/lib/alsa-lib/smixer/smixer-hda.so /usr/lib/alsa-lib/smixer/smixer-sbase.la /usr/lib/alsa-lib/smixer/smixer-sbase.so Firstly could you please let me know why this issue is seen in ti sdk 8 BSP's (ie., in rootfs), & what could be the problem in ti sdk 8 rootfs ?? Apart from the above is there any other ALSA related configuration files, needs to copied from the ti sdk 7 rootfs to ti sdk 8 rootfs, inorder to resolve this issue If possible, could you please validate this PJSIP version 2.3 on ti sdk 8 BSP's on your EVK kit & let me know whether this issue is reproducible Awaiting for your replies, Many Thanks in advance

3 11

[alsa-devel] [PATCH] sound/usb: add dB range mapping quirks.
by Yao-Wen Mao 28 Jul '15

28 Jul '15

Some usb audio devices don't follow the specification of the volume range. Add dB_mapping_quirks to fix some devices' dB ranges so that the function snd_mixer_selem_set_playback_dB in alsa-lib can set the correct value for those devices. Signed-off-by: Yao-Wen Mao <yaowen(a)google.com> --- sound/usb/mixer.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 6b3acba..b6b27dc 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -933,6 +933,32 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, } } +/* dB mapping quirks */ +static void dB_mapping_quirks(struct usb_mixer_elem_info *cval, + struct snd_kcontrol *kctl) +{ + switch (cval->mixer->chip->usb_id) { + case USB_ID(0x05a7, 0x1020): /* Bose Companion 5 */ + if (strcmp(kctl->id.name, "PCM Playback Volume") == 0) { + cval->dBmin = (convert_signed_value(cval, cval->min) * 100) / 16; + cval->dBmax = (convert_signed_value(cval, cval->max) * 100) / 16; + } + break; + case USB_ID(0x21b4, 0x0081): /* Dragonfly DAC 1.2 */ + if (strcmp(kctl->id.name, "PCM Playback Volume") == 0) { + cval->dBmin = convert_signed_value(cval, cval->min) * 100; + cval->dBmax = convert_signed_value(cval, cval->max) * 100; + } + break; + case USB_ID(0x046d, 0x0a29): /* Logitech H800 */ + if (strcmp(kctl->id.name, "PCM Playback Volume") == 0) { + cval->dBmin = convert_signed_value(cval, cval->min); + cval->dBmax = convert_signed_value(cval, cval->max); + } + break; + } +} + /* * retrieve the minimum and maximum values for the specified control */ @@ -1027,6 +1053,10 @@ static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval, */ cval->dBmin = (convert_signed_value(cval, cval->min) * 100) / 256; cval->dBmax = (convert_signed_value(cval, cval->max) * 100) / 256; + + if (kctl) + dB_mapping_quirks(cval, kctl); + if (cval->dBmin > cval->dBmax) { /* something is wrong; assume it's either from/to 0dB */ if (cval->dBmin < 0) -- 2.5.0.rc2.392.g76e840b

2 5

[alsa-devel] Cannot open shared library /usr/lib/alsa-lib/libasound_module_pcm_jack.so
by Srinivasan S 28 Jul '15

28 Jul '15

Dear Clemens, Am trying to cross compile, could you pls help me out for cross_compiling Am facing some configuration issue as shown below srinivasan@tata-HP-Elite-7100-Microtower-PC:~/alsa-plugins-1.0.29$ ./configure --host=arm-linux CC=/opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin/arm-linux-gnueabihf-gcc --prefix=/home/srinivasan/alsa-plugins-1.0.29/ checking for a BSD-compatible install... /usr/bin/install -c checking whether build environment is sane... yes checking for arm-linux-strip... no checking for strip... strip checking for a thread-safe mkdir -p... /bin/mkdir -p checking for gawk... gawk checking whether make sets $(MAKE)... yes checking whether to enable maintainer-specific portions of Makefiles... yes checking for arm-linux-gcc... /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin/arm-linux-gnueabihf-gcc checking whether the C compiler works... yes checking for C compiler default output file name... a.out checking for suffix of executables... checking whether we are cross compiling... yes checking for suffix of object files... o checking whether we are using the GNU C compiler... yes checking whether /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin/arm-linux-gnueabihf-gcc accepts -g... yes checking for /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin/arm-linux-gnueabihf-gcc option to accept ISO C89... none needed checking for style of include used by make... GNU checking dependency style of /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin/arm-linux-gnueabihf-gcc... gcc3 checking build system type... x86_64-unknown-linux-gnu checking host system type... arm-unknown-linux-gnu checking how to print strings... printf checking for a sed that does not truncate output... /bin/sed checking for grep that handles long lines and -e... /bin/grep checking for egrep... /bin/grep -E checking for fgrep... /bin/grep -F checking for ld used by /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin/arm-linux-gnueabihf-gcc... /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/arm-linux-gnueabihf/bin/ld checking if the linker (/opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/arm-linux-gnueabihf/bin/ld) is GNU ld... yes checking for BSD- or MS-compatible name lister (nm)... no checking for arm-linux-dumpbin... no checking for arm-linux-link... no checking for dumpbin... no checking for link... link -dump configure: WARNING: using cross tools not prefixed with host triplet checking the name lister (nm) interface... BSD nm checking whether ln -s works... yes checking the maximum length of command line arguments... 1572864 checking whether the shell understands some XSI constructs... yes checking whether the shell understands "+="... yes checking how to convert x86_64-unknown-linux-gnu file names to arm-unknown-linux-gnu format... func_convert_file_noop checking how to convert x86_64-unknown-linux-gnu file names to toolchain format... func_convert_file_noop checking for /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/arm-linux-gnueabihf/bin/ld option to reload object files... -r checking for arm-linux-objdump... no checking for objdump... objdump checking how to recognize dependent libraries... pass_all checking for arm-linux-dlltool... no checking for dlltool... no checking how to associate runtime and link libraries... printf %s\n checking for arm-linux-ar... no checking for ar... ar checking for archiver @FILE support... @ checking for arm-linux-strip... strip checking for arm-linux-ranlib... no checking for ranlib... ranlib checking command to parse nm output from /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin/arm-linux-gnueabihf-gcc object... ok checking for sysroot... no checking for arm-linux-mt... no checking for mt... mt checking if mt is a manifest tool... no checking how to run the C preprocessor... /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin/arm-linux-gnueabihf-gcc -E checking for ANSI C header files... yes checking for sys/types.h... yes checking for sys/stat.h... yes checking for stdlib.h... yes checking for string.h... yes checking for memory.h... yes checking for strings.h... yes checking for inttypes.h... yes checking for stdint.h... yes checking for unistd.h... yes checking for dlfcn.h... yes checking for objdir... .libs checking if /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin/arm-linux-gnueabihf-gcc supports -fno-rtti -fno-exceptions... no checking for /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin/arm-linux-gnueabihf-gcc option to produce PIC... -fPIC -DPIC checking if /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin/arm-linux-gnueabihf-gcc PIC flag -fPIC -DPIC works... yes checking if /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin/arm-linux-gnueabihf-gcc static flag -static works... yes checking if /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin/arm-linux-gnueabihf-gcc supports -c -o file.o... yes checking if /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin/arm-linux-gnueabihf-gcc supports -c -o file.o... (cached) yes checking whether the /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin/arm-linux-gnueabihf-gcc linker (/opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/arm-linux-gnueabihf/bin/ld) supports shared libraries... yes checking whether -lc should be explicitly linked in... no checking dynamic linker characteristics... GNU/Linux ld.so checking how to hardcode library paths into programs... immediate checking whether stripping libraries is possible... yes checking if libtool supports shared libraries... yes checking whether to build shared libraries... yes checking whether to build static libraries... no checking for ANSI C header files... (cached) yes ./configure: line 11534: CC_NOUNDEFINED: command not found checking for arm-linux-pkg-config... no checking for pkg-config... /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin//pkg-config checking pkg-config is at least version 0.9.0... yes checking for ALSA... no configure: error: Package requirements (alsa >= 1.0.11) were not met: No package 'alsa' found Consider adjusting the PKG_CONFIG_PATH environment variable if you installed software in a non-standard prefix. Alternatively, you may set the environment variables ALSA_CFLAGS and ALSA_LIBS to avoid the need to call pkg-config. See the pkg-config man page for more details. srinivasan@tata-HP-Elite-7100-Microtower-PC:~/alsa-plugins-1.0.29$ Thanks Srini

2 6

[alsa-devel] pcm prepare
by Andy Ng 28 Jul '15

28 Jul '15

Hi, I am quite surprised that by sending hardware params with alsa-lib the prepare function is called automatically in the driver. In my case, I would like to open and configure pcmNp and pcmNc and then call prepare for pcmNp and pcmPc channels. It is quite imperative to execute specific configuration steps after both channels are opened and configured. I was wondering if there is any ioctl interface that can be hooked in alsa-lib to pass extra configuration info before start the streams. Any ideas? Andy

2 1

[alsa-devel] Bug on Dell Vostro 5480 (Intel Wildcat) kernel driver
by Paulo Roberto de Oliveira Castro 28 Jul '15

28 Jul '15

Hi everyone, After installing Fedora 22 on my Dell Vostro 5480, the speakers don't work, and the sound only comes out of the subwoofer. Here is the output of alsa-info.sh. After talking to "debianuser" on IRC, we found this workaround: sudo hda-verb /dev/snd/hwC1D0 0x14 SET_PIN_WIDGET_CONTROL 0x40 # enable, set it to OUT sudo hda-verb /dev/snd/hwC1D0 0x14 SET_CONNECT_SEL 0 # switch to 0x0c sudo hda-verb /dev/snd/hwC1D0 0x14 SET_AMP_GAIN_MUTE 0xa000 # unmute left sudo hda-verb /dev/snd/hwC1D0 0x14 SET_AMP_GAIN_MUTE 0x9000 # unmute right We suspect that it's a kernel driver bug. This makes the speakers work alongside the headphones (both at the same time) but I need help making it working when headphones are not plugged in. Thanks, Paulo

2 2

[alsa-devel] [PATCH v2] ALSA: hda - Fix race condition between HDA driver and runtime PM
by U. Artie Eoff 28 Jul '15

28 Jul '15

Don't execute runtime suspend if HDA is not finished initializing. Otherwise, the following errors can occur in hda: snd_hda_intel 0000:00:1b.0: CORB reset timeout#2, CORBRP = 65535 snd_hda_intel 0000:00:1b.0: no codecs initialized Debugged and root-cause found by zhuo-hao.lee(a)intel.com v2: remove unnecessary extra check for chip init (libin) Signed-off-by: U. Artie Eoff <ullysses.a.eoff(a)intel.com> --- sound/pci/hda/hda_intel.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b729b25a6ad6..15d4e28fc64e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1017,17 +1017,19 @@ static int azx_runtime_idle(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; struct hda_intel *hda; + struct hdac_bus *bus; if (!card) return 0; chip = card->private_data; hda = container_of(chip, struct hda_intel, chip); + bus = azx_bus(chip); if (chip->disabled || hda->init_failed || !chip->running) return 0; if (!power_save_controller || !azx_has_pm_runtime(chip) || - azx_bus(chip)->codec_powered) + bus->codec_powered || !bus->chip_init) return -EBUSY; return 0; -- 1.9.3

3 5

[alsa-devel] [PATCH v2 1/2] ASoC: max98088: Get rid of max98088_access table
by Axel Lin 28 Jul '15

28 Jul '15

The max98088_access table is used for look up readable/writable/volatile attributes of registers. The readable/writable/volatile registers are mostly in continuous ranges, so we can replace the max98088_access table entirely by using case range. Below is a summary of the readable/writeable/volatile registers: readable registers: 0x00 ~ 0xC9, 0xFF writeable registers: 0x03 ~ 0xC9 volatile registers: 0x00 ~ 0x03, 0xFF Note, 0x00 should be read-only according to the datasheet. This patch reworks the implement for .readable and .volatile and also add implementation for .writable callback. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- v2: no code change. Just update commit log, 0xff register is not writeable. sound/soc/codecs/max98088.c | 305 ++++---------------------------------------- 1 file changed, 25 insertions(+), 280 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 99c2daa..2c2df17 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -258,292 +258,36 @@ static const struct reg_default max98088_reg[] = { { 0xc9, 0x00 }, /* C9 DAI2 biquad */ }; -static struct { - int readable; - int writable; - int vol; -} max98088_access[M98088_REG_CNT] = { - { 0xFF, 0xFF, 1 }, /* 00 IRQ status */ - { 0xFF, 0x00, 1 }, /* 01 MIC status */ - { 0xFF, 0x00, 1 }, /* 02 jack status */ - { 0x1F, 0x1F, 1 }, /* 03 battery voltage */ - { 0xFF, 0xFF, 0 }, /* 04 */ - { 0xFF, 0xFF, 0 }, /* 05 */ - { 0xFF, 0xFF, 0 }, /* 06 */ - { 0xFF, 0xFF, 0 }, /* 07 */ - { 0xFF, 0xFF, 0 }, /* 08 */ - { 0xFF, 0xFF, 0 }, /* 09 */ - { 0xFF, 0xFF, 0 }, /* 0A */ - { 0xFF, 0xFF, 0 }, /* 0B */ - { 0xFF, 0xFF, 0 }, /* 0C */ - { 0xFF, 0xFF, 0 }, /* 0D */ - { 0xFF, 0xFF, 0 }, /* 0E */ - { 0xFF, 0xFF, 0 }, /* 0F interrupt enable */ - - { 0xFF, 0xFF, 0 }, /* 10 master clock */ - { 0xFF, 0xFF, 0 }, /* 11 DAI1 clock mode */ - { 0xFF, 0xFF, 0 }, /* 12 DAI1 clock control */ - { 0xFF, 0xFF, 0 }, /* 13 DAI1 clock control */ - { 0xFF, 0xFF, 0 }, /* 14 DAI1 format */ - { 0xFF, 0xFF, 0 }, /* 15 DAI1 clock */ - { 0xFF, 0xFF, 0 }, /* 16 DAI1 config */ - { 0xFF, 0xFF, 0 }, /* 17 DAI1 TDM */ - { 0xFF, 0xFF, 0 }, /* 18 DAI1 filters */ - { 0xFF, 0xFF, 0 }, /* 19 DAI2 clock mode */ - { 0xFF, 0xFF, 0 }, /* 1A DAI2 clock control */ - { 0xFF, 0xFF, 0 }, /* 1B DAI2 clock control */ - { 0xFF, 0xFF, 0 }, /* 1C DAI2 format */ - { 0xFF, 0xFF, 0 }, /* 1D DAI2 clock */ - { 0xFF, 0xFF, 0 }, /* 1E DAI2 config */ - { 0xFF, 0xFF, 0 }, /* 1F DAI2 TDM */ - - { 0xFF, 0xFF, 0 }, /* 20 DAI2 filters */ - { 0xFF, 0xFF, 0 }, /* 21 data config */ - { 0xFF, 0xFF, 0 }, /* 22 DAC mixer */ - { 0xFF, 0xFF, 0 }, /* 23 left ADC mixer */ - { 0xFF, 0xFF, 0 }, /* 24 right ADC mixer */ - { 0xFF, 0xFF, 0 }, /* 25 left HP mixer */ - { 0xFF, 0xFF, 0 }, /* 26 right HP mixer */ - { 0xFF, 0xFF, 0 }, /* 27 HP control */ - { 0xFF, 0xFF, 0 }, /* 28 left REC mixer */ - { 0xFF, 0xFF, 0 }, /* 29 right REC mixer */ - { 0xFF, 0xFF, 0 }, /* 2A REC control */ - { 0xFF, 0xFF, 0 }, /* 2B left SPK mixer */ - { 0xFF, 0xFF, 0 }, /* 2C right SPK mixer */ - { 0xFF, 0xFF, 0 }, /* 2D SPK control */ - { 0xFF, 0xFF, 0 }, /* 2E sidetone */ - { 0xFF, 0xFF, 0 }, /* 2F DAI1 playback level */ - - { 0xFF, 0xFF, 0 }, /* 30 DAI1 playback level */ - { 0xFF, 0xFF, 0 }, /* 31 DAI2 playback level */ - { 0xFF, 0xFF, 0 }, /* 32 DAI2 playbakc level */ - { 0xFF, 0xFF, 0 }, /* 33 left ADC level */ - { 0xFF, 0xFF, 0 }, /* 34 right ADC level */ - { 0xFF, 0xFF, 0 }, /* 35 MIC1 level */ - { 0xFF, 0xFF, 0 }, /* 36 MIC2 level */ - { 0xFF, 0xFF, 0 }, /* 37 INA level */ - { 0xFF, 0xFF, 0 }, /* 38 INB level */ - { 0xFF, 0xFF, 0 }, /* 39 left HP volume */ - { 0xFF, 0xFF, 0 }, /* 3A right HP volume */ - { 0xFF, 0xFF, 0 }, /* 3B left REC volume */ - { 0xFF, 0xFF, 0 }, /* 3C right REC volume */ - { 0xFF, 0xFF, 0 }, /* 3D left SPK volume */ - { 0xFF, 0xFF, 0 }, /* 3E right SPK volume */ - { 0xFF, 0xFF, 0 }, /* 3F MIC config */ - - { 0xFF, 0xFF, 0 }, /* 40 MIC threshold */ - { 0xFF, 0xFF, 0 }, /* 41 excursion limiter filter */ - { 0xFF, 0xFF, 0 }, /* 42 excursion limiter threshold */ - { 0xFF, 0xFF, 0 }, /* 43 ALC */ - { 0xFF, 0xFF, 0 }, /* 44 power limiter threshold */ - { 0xFF, 0xFF, 0 }, /* 45 power limiter config */ - { 0xFF, 0xFF, 0 }, /* 46 distortion limiter config */ - { 0xFF, 0xFF, 0 }, /* 47 audio input */ - { 0xFF, 0xFF, 0 }, /* 48 microphone */ - { 0xFF, 0xFF, 0 }, /* 49 level control */ - { 0xFF, 0xFF, 0 }, /* 4A bypass switches */ - { 0xFF, 0xFF, 0 }, /* 4B jack detect */ - { 0xFF, 0xFF, 0 }, /* 4C input enable */ - { 0xFF, 0xFF, 0 }, /* 4D output enable */ - { 0xFF, 0xFF, 0 }, /* 4E bias control */ - { 0xFF, 0xFF, 0 }, /* 4F DAC power */ - - { 0xFF, 0xFF, 0 }, /* 50 DAC power */ - { 0xFF, 0xFF, 0 }, /* 51 system */ - { 0xFF, 0xFF, 0 }, /* 52 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 53 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 54 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 55 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 56 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 57 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 58 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 59 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 5A DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 5B DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 5C DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 5D DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 5E DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 5F DAI1 EQ2 */ - - { 0xFF, 0xFF, 0 }, /* 60 DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 61 DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 62 DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 63 DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 64 DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 65 DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 66 DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 67 DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 68 DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 69 DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 6A DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 6B DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 6C DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 6D DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 6E DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 6F DAI1 EQ3 */ - - { 0xFF, 0xFF, 0 }, /* 70 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 71 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 72 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 73 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 74 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 75 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 76 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 77 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 78 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 79 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 7A DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 7B DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 7C DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 7D DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 7E DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 7F DAI1 EQ5 */ - - { 0xFF, 0xFF, 0 }, /* 80 DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 81 DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 82 DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 83 DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 84 DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 85 DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 86 DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 87 DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 88 DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 89 DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 8A DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 8B DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 8C DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 8D DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 8E DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 8F DAI2 EQ2 */ - - { 0xFF, 0xFF, 0 }, /* 90 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 91 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 92 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 93 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 94 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 95 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 96 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 97 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 98 DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 99 DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 9A DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 9B DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 9C DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 9D DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 9E DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 9F DAI2 EQ3 */ - - { 0xFF, 0xFF, 0 }, /* A0 DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* A1 DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* A2 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A3 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A4 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A5 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A6 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A7 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A8 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A9 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* AA DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* AB DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* AC DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* AD DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* AE DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* AF DAI2 EQ5 */ - - { 0xFF, 0xFF, 0 }, /* B0 DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* B1 DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* B2 DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* B3 DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* B4 DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* B5 DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* B6 DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* B7 DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* B8 DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* B9 DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* BA DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* BB DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* BC DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* BD DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* BE DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* BF DAI1 biquad */ - - { 0xFF, 0xFF, 0 }, /* C0 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C1 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C2 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C3 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C4 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C5 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C6 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C7 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C8 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C9 DAI2 biquad */ - { 0x00, 0x00, 0 }, /* CA */ - { 0x00, 0x00, 0 }, /* CB */ - { 0x00, 0x00, 0 }, /* CC */ - { 0x00, 0x00, 0 }, /* CD */ - { 0x00, 0x00, 0 }, /* CE */ - { 0x00, 0x00, 0 }, /* CF */ - - { 0x00, 0x00, 0 }, /* D0 */ - { 0x00, 0x00, 0 }, /* D1 */ - { 0x00, 0x00, 0 }, /* D2 */ - { 0x00, 0x00, 0 }, /* D3 */ - { 0x00, 0x00, 0 }, /* D4 */ - { 0x00, 0x00, 0 }, /* D5 */ - { 0x00, 0x00, 0 }, /* D6 */ - { 0x00, 0x00, 0 }, /* D7 */ - { 0x00, 0x00, 0 }, /* D8 */ - { 0x00, 0x00, 0 }, /* D9 */ - { 0x00, 0x00, 0 }, /* DA */ - { 0x00, 0x00, 0 }, /* DB */ - { 0x00, 0x00, 0 }, /* DC */ - { 0x00, 0x00, 0 }, /* DD */ - { 0x00, 0x00, 0 }, /* DE */ - { 0x00, 0x00, 0 }, /* DF */ - - { 0x00, 0x00, 0 }, /* E0 */ - { 0x00, 0x00, 0 }, /* E1 */ - { 0x00, 0x00, 0 }, /* E2 */ - { 0x00, 0x00, 0 }, /* E3 */ - { 0x00, 0x00, 0 }, /* E4 */ - { 0x00, 0x00, 0 }, /* E5 */ - { 0x00, 0x00, 0 }, /* E6 */ - { 0x00, 0x00, 0 }, /* E7 */ - { 0x00, 0x00, 0 }, /* E8 */ - { 0x00, 0x00, 0 }, /* E9 */ - { 0x00, 0x00, 0 }, /* EA */ - { 0x00, 0x00, 0 }, /* EB */ - { 0x00, 0x00, 0 }, /* EC */ - { 0x00, 0x00, 0 }, /* ED */ - { 0x00, 0x00, 0 }, /* EE */ - { 0x00, 0x00, 0 }, /* EF */ - - { 0x00, 0x00, 0 }, /* F0 */ - { 0x00, 0x00, 0 }, /* F1 */ - { 0x00, 0x00, 0 }, /* F2 */ - { 0x00, 0x00, 0 }, /* F3 */ - { 0x00, 0x00, 0 }, /* F4 */ - { 0x00, 0x00, 0 }, /* F5 */ - { 0x00, 0x00, 0 }, /* F6 */ - { 0x00, 0x00, 0 }, /* F7 */ - { 0x00, 0x00, 0 }, /* F8 */ - { 0x00, 0x00, 0 }, /* F9 */ - { 0x00, 0x00, 0 }, /* FA */ - { 0x00, 0x00, 0 }, /* FB */ - { 0x00, 0x00, 0 }, /* FC */ - { 0x00, 0x00, 0 }, /* FD */ - { 0x00, 0x00, 0 }, /* FE */ - { 0xFF, 0x00, 1 }, /* FF */ -}; - static bool max98088_readable_register(struct device *dev, unsigned int reg) { - return max98088_access[reg].readable; + switch (reg) { + case M98088_REG_00_IRQ_STATUS ... 0xC9: + case M98088_REG_FF_REV_ID: + return true; + default: + return false; + } +} + +static bool max98088_writeable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case M98088_REG_03_BATTERY_VOLTAGE ... 0xC9: + return true; + default: + return false; + } } static bool max98088_volatile_register(struct device *dev, unsigned int reg) { - return max98088_access[reg].vol; + switch (reg) { + case M98088_REG_00_IRQ_STATUS ... M98088_REG_03_BATTERY_VOLTAGE: + case M98088_REG_FF_REV_ID: + return true; + default: + return false; + } } static const struct regmap_config max98088_regmap = { @@ -551,6 +295,7 @@ static const struct regmap_config max98088_regmap = { .val_bits = 8, .readable_reg = max98088_readable_register, + .writeable_reg = max98088_writeable_register, .volatile_reg = max98088_volatile_register, .max_register = 0xff, -- 2.1.0

1 1

[alsa-devel] [PATCH v3] ASoC: Add support for NAU8825 codec to ASoC
by Chih-Chiang Chang 28 Jul '15

28 Jul '15

The NAU88L25 is an ultra-low power high performance audio codec designed for smartphone, tablet PC, and other portable devices by Nuvoton, now add linux driver support for it. Signed-off-by: Chih-Chiang Chang <ccchang12(a)nuvoton.com> --- v3->v2: - fix the wrong definition of reg_default - fix the flow of set_sys_clk() and nau8825_set_bias_level() - remove unnecessary code in nau8825_volatile_register() and nau8825_i2c_probe() - add trigger function for ADC/DAC control - add some register definitions - fix some coding style issues v2->v1: - fixes according to Lars-Peter Clausen's review comments - removes unused platform data file - corrects the naming of DAPM input widget - fixes some wrong coding of SOC widgets and other codes - adds definition and remark for config FLL clock - moves the code of reset hardware registers from codec_probe() to i2c_probe() - removes unused codes sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8825.c | 724 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/nau8825.h | 399 +++++++++++++++++++++++++ 4 files changed, 1130 insertions(+) create mode 100644 sound/soc/codecs/nau8825.c create mode 100644 sound/soc/codecs/nau8825.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index efaafce..1edb781 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -76,6 +76,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C + select SND_SOC_NAU8825 if I2C select SND_SOC_HDMI_CODEC select SND_SOC_PCM1681 if I2C select SND_SOC_PCM1792A if SPI_MASTER @@ -470,6 +471,10 @@ config SND_SOC_MAX98925 config SND_SOC_MAX9850 tristate +config SND_SOC_NAU8825 + tristate "Nuvoton NAU8825 CODEC" + depends on I2C + config SND_SOC_PCM1681 tristate "Texas Instruments PCM1681 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index cf160d9..db0a4ec 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -69,6 +69,7 @@ snd-soc-max98925-objs := max98925.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o +snd-soc-nau8825-objs := nau8825.o snd-soc-hdmi-codec-objs := hdmi.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o @@ -256,6 +257,7 @@ obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o +obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c new file mode 100644 index 0000000..d3b306a --- /dev/null +++ b/sound/soc/codecs/nau8825.c @@ -0,0 +1,724 @@ +/* + * nau8825.c + * + * Copyright 2015 Nuvoton Technology Corp. + * Author: Meng-Huang Kuo <mhkuo(a)nuvoton.com> + * + * 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/module.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <linux/device.h> +#include <linux/clk.h> +#include <linux/regmap.h> +#include <linux/acpi.h> +#include <asm/div64.h> +#include <sound/jack.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <linux/delay.h> +#include "nau8825.h" + +static const DECLARE_TLV_DB_SCALE(out_hp_vol_tlv, -5400, 100, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -10350, 50, 0); + +static const struct snd_kcontrol_new nau8825_snd_controls[] = { + SOC_SINGLE_TLV("MIC Volume", NAU8825_ADC_DGAIN_CTRL, + NAU8825_ADC_DGAIN_SFT, + NAU8825_ADC_VOL_RSCL_RANGE, 0, adc_vol_tlv), + SOC_DOUBLE_TLV("HP Volume", NAU8825_HSVOL_CTRL, + NAU8825_L_HSVOL_SFT, NAU8825_R_HSVOL_SFT, + NAU8825_VOL_RSCL_RANGE, 1, out_hp_vol_tlv), +}; + +static const struct snd_kcontrol_new nau8825_hpo_mix[] = { + SOC_DAPM_SINGLE("HP L Switch", NAU8825_HSVOL_CTRL, + NAU8825_L_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("HP R Switch", NAU8825_HSVOL_CTRL, + NAU8825_R_MUTE_SFT, 1, 1), +}; + +static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { + /* Input */ + SND_SOC_DAPM_INPUT("MIC"), + SND_SOC_DAPM_SUPPLY("MIC BIAS", NAU8825_BOOST, NAU8825_G_BIAS_SFT, 0, + NULL, 0), + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + /* DACs */ + SND_SOC_DAPM_DAC("DAC L1", NULL, NAU8825_DAC_CTRL, + NAU8825_DAC_L_SFT, 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, NAU8825_DAC_CTRL, + NAU8825_DAC_R_SFT, 0), + /* SPO/HPO/LOUT/Mono Mixer */ + SND_SOC_DAPM_MIXER("HPO MIX", SND_SOC_NOPM, 0, 0, nau8825_hpo_mix, + ARRAY_SIZE(nau8825_hpo_mix)), + SND_SOC_DAPM_SUPPLY("HP amp", NAU8825_CLASSG_CTRL, + NAU8825_CLASSG_EN_SHIFT, 0, NULL, 0), + /* Output */ + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route nau8825_dapm_routes[] = { + {"AIF1TX", NULL, "MIC"}, + {"DAC L1", NULL, "AIF1RX"}, + {"DAC R1", NULL, "AIF1RX"}, + {"HPO MIX", "HP L Switch", "DAC L1"}, + {"HPO MIX", "HP R Switch", "DAC R1"}, + {"HPOL", NULL, "HPO MIX"}, + {"HPOR", NULL, "HPO MIX"}, + {"HPOL", NULL, "HP amp"}, + {"HPOR", NULL, "HP amp"}, +}; + +static int nau8825_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 nau8825_priv *nau8825 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0; + + switch (params_width(params)) { + case 16: + break; + case 20: + val_len |= NAU8825_I2S_DL_20; + break; + case 24: + val_len |= NAU8825_I2S_DL_24; + break; + case 32: + val_len |= NAU8825_I2S_DL_32; + break; + default: + return -EINVAL; + } + regmap_update_bits(nau8825->regmap, NAU8825_I2S_PCM_CTRL_1, + NAU8825_I2S_DL_MASK, val_len); + return 0; +} + +static int nau8825_dac_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8825_priv *nau8825 = snd_soc_codec_get_drvdata(codec); + + if (mute) + regmap_update_bits(nau8825->regmap, NAU8825_DAC_MUTE_CTRL, + NAU8825_SOFT_MUTE_MASK, NAU8825_SOFT_MUTE_EN); + else + regmap_update_bits(nau8825->regmap, NAU8825_DAC_MUTE_CTRL, + NAU8825_SOFT_MUTE_MASK, NAU8825_SOFT_MUTE_DIS); + return 0; +} + +static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8825_priv *nau8825 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + dev_dbg(codec->dev, " %s ###nau8825_set_dai_fmt %x\n", __func__, fmt); + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val |= NAU8825_I2S_BP_INV; + break; + default: + dev_alert(codec->dev, "Invalid DAI interface format\n"); + return -EINVAL; + } + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + reg_val |= NAU8825_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= NAU8825_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_RIGHT_J: + reg_val |= NAU8825_I2S_DF_RIGHT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= NAU8825_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= NAU8825_I2S_DF_PCM_B; + reg_val |= NAU8825_I2S_PCMB_EN; + break; + default: + dev_alert(codec->dev, "Invalid DAI I2S/PCM format\n"); + return -EINVAL; + } + regmap_update_bits(nau8825->regmap, NAU8825_I2S_PCM_CTRL_1, + NAU8825_I2S_DL_MASK | NAU8825_I2S_DF_MASK + | NAU8825_I2S_BP_MASK | NAU8825_I2S_PCMB_MASK, + reg_val); + return 0; +} + +static void config_fll_clk_12m(struct snd_soc_codec *codec) +{ + struct nau8825_priv *nau8825 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(nau8825->regmap, NAU8825_CLK_DIVIDER, + NAU8825_CLK_MCLK_SRC_MASK, 0x0003); + regmap_update_bits(nau8825->regmap, NAU8825_FLL_1, + NAU8825_FLL_RATIO_MASK, 0x0001); + /* FLL 16-bit fractional input */ + regmap_write(nau8825->regmap, NAU8825_FLL_2, 0xc49b); + /* FLL 10-bit integer input */ + regmap_update_bits(nau8825->regmap, NAU8825_FLL_3, + NAU8825_FLL_INTEGER_MASK, 0x0020); + /* FLL pre-scaler */ + regmap_update_bits(nau8825->regmap, NAU8825_FLL_4, + NAU8825_FLL_REF_DIV_MASK, 0x0800); + /* select divied VCO input */ + regmap_update_bits(nau8825->regmap, NAU8825_FLL_5, + NAU8825_FLL_FILTER_SW_MASK, 0x0000); + /* FLL sigma delta modulator enable */ + regmap_update_bits(nau8825->regmap, NAU8825_FLL_6, + NAU8825_SDM_EN_MASK, 0x4000); +} + +static void set_sys_clk(struct snd_soc_codec *codec, int sys_clk) +{ + struct nau8825_priv *nau8825 = snd_soc_codec_get_drvdata(codec); + + pr_debug("%s :: sys_clk=%x\n", __func__, sys_clk); + switch (sys_clk) { + case NAU8825_INTERNALCLOCK: + regmap_update_bits(nau8825->regmap, NAU8825_CLK_DIVIDER, + NAU8825_SYSCLK_EN_MASK, NAU8825_SYSCLK_DIS); + regmap_update_bits(nau8825->regmap, NAU8825_FLL_6, + NAU8825_DCO_EN_MASK, NAU8825_DCO_EN); + regmap_update_bits(nau8825->regmap, NAU8825_CLK_DIVIDER, + NAU8825_SYSCLK_EN_MASK, NAU8825_SYSCLK_EN); + break; + case NAU8825_MCLK: + default: + regmap_update_bits(nau8825->regmap, NAU8825_FLL_6, + NAU8825_DCO_EN_MASK, NAU8825_DCO_DIS); + regmap_update_bits(nau8825->regmap, NAU8825_I2S_PCM_CTRL_2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); + /* FLL clock source from MCLK */ + regmap_update_bits(nau8825->regmap, NAU8825_CLK_DIVIDER, + NAU8825_SYSCLK_EN_MASK, NAU8825_SYSCLK_DIS); + mdelay(2); + regmap_update_bits(nau8825->regmap, NAU8825_CLK_DIVIDER, + NAU8825_SYSCLK_EN_MASK, NAU8825_SYSCLK_EN); + break; + } +} + +static int nau8825_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + + switch (clk_id) { + case NAU8825_MCLK: + config_fll_clk_12m(codec); + set_sys_clk(codec, clk_id); + break; + case NAU8825_INTERNALCLOCK: + set_sys_clk(codec, clk_id); + break; + default: + dev_err(codec->dev, "Wrong clock src\n"); + return -EINVAL; + } + return 0; +} + +static int nau8825_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct nau8825_priv *nau8825 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + /* All power is driven by DAPM system*/ + dev_dbg(codec->dev, "###nau8825_set_bias_level BIAS_ON\n"); + regmap_update_bits(nau8825->regmap, NAU8825_BIAS_ADJ, + NAU8825_VMIDSEL_MASK, NAU8825_VMIDSEL_125KOHM); + regmap_update_bits(nau8825->regmap, NAU8825_BIAS_ADJ, + NAU8825_VMID_MASK, NAU8825_VMID_EN); + regmap_update_bits(nau8825->regmap, NAU8825_BOOST, + NAU8825_G_BIAS_MASK, NAU8825_G_BIAS_EN); + break; + case SND_SOC_BIAS_OFF: + dev_dbg(codec->dev, "###nau8825_set_bias_level OFF\n"); + set_sys_clk(codec, NAU8825_INTERNALCLOCK); + regmap_update_bits(nau8825->regmap, NAU8825_BIAS_ADJ, + NAU8825_VMID_MASK, NAU8825_VMID_DIS); + regmap_update_bits(nau8825->regmap, NAU8825_BOOST, + NAU8825_G_BIAS_MASK, NAU8825_G_BIAS_DIS); + break; + default: + break; + } + codec->dapm.bias_level = level; + dev_dbg(codec->dev, "## nau8825_set_bias_level %d\n", level); + return 0; +} + + +static const struct reg_default nau8825_reg[] = { + {0x000, 0x0000}, + {0x001, 0x00ff}, + {0x003, 0x0050}, + {0x004, 0x0000}, + {0x005, 0x3126}, + {0x006, 0x0008}, + {0x007, 0x0010}, + {0x008, 0x0000}, + {0x009, 0x6000}, + {0x00a, 0xf13c}, + {0x00c, 0x000c}, + {0x00d, 0x0000}, + {0x00f, 0x0800}, + {0x010, 0x0000}, + {0x011, 0x0000}, + {0x012, 0x0010}, + {0x013, 0x0015}, + {0x014, 0x0110}, + {0x015, 0x0000}, + {0x016, 0x0000}, + {0x017, 0x0000}, + {0x018, 0x0000}, + {0x019, 0x0000}, + {0x01a, 0x0000}, + {0x01b, 0x0000}, + {0x01c, 0x000b}, + {0x01d, 0x8010}, + {0x01e, 0x0000}, + {0x01f, 0x0000}, + {0x020, 0x0000}, + {0x021, 0x0000}, + {0x022, 0x0000}, + {0x023, 0x0000}, + {0x024, 0x0000}, + {0x025, 0x0000}, + {0x026, 0x0000}, + {0x027, 0x0000}, + {0x028, 0x0000}, + {0x029, 0x0000}, + {0x02a, 0x0000}, + {0x02b, 0x0010}, + {0x02c, 0x0001}, + {0x02d, 0x0000}, + {0x02f, 0x0000}, + {0x030, 0x01cf}, + {0x031, 0x0000}, + {0x032, 0x0000}, + {0x033, 0x00cf}, + {0x034, 0x0000}, + {0x038, 0x1486}, + {0x039, 0x0f12}, + {0x03a, 0x25ff}, + {0x03b, 0x3457}, + {0x045, 0x1486}, + {0x046, 0x0f12}, + {0x047, 0x25f9}, + {0x048, 0x3457}, + {0x04c, 0x0000}, + {0x04d, 0x0000}, + {0x050, 0x0000}, + {0x051, 0x0000}, + {0x055, 0x0000}, + {0x058, 0x0000}, + {0x059, 0x0000}, + {0x066, 0x0000}, + {0x068, 0x0000}, + {0x069, 0x0000}, + {0x06a, 0x0020}, + {0x071, 0x0011}, + {0x072, 0x0020}, + {0x073, 0x0008}, + {0x074, 0x0006}, + {0x076, 0x0000}, + {0x077, 0x0000}, + {0x07f, 0x0000}, + {0x080, 0x0300}, + {0x081, 0x0000}, + {0x082, 0x0000}, +}; + +static int nau8825_codec_remove(struct snd_soc_codec *codec) +{ + struct nau8825_priv *nau8825 = snd_soc_codec_get_drvdata(codec); + + regmap_write(nau8825->regmap, NAU8825_RESET, 0x00); + regmap_write(nau8825->regmap, NAU8825_RESET, 0x00); + return 0; +} + +static int nau8825_codec_probe(struct snd_soc_codec *codec) +{ + struct nau8825_priv *nau8825 = snd_soc_codec_get_drvdata(codec); + + /* bias current settings */ + regmap_write(nau8825->regmap, 0x0072, 0x0260); + /* enable bias */ + nau8825_set_bias_level(codec, SND_SOC_BIAS_ON); + mdelay(10); + /* DAC digital default gain 0 dB */ + regmap_write(nau8825->regmap, 0x0033, 0x00cf); + regmap_write(nau8825->regmap, 0x0034, 0x02cf); + /* DAC driver default gain -29 dB */ + regmap_write(nau8825->regmap, 0x0032, 0x075d); + /* ADC digital default gain 8 dB */ + regmap_write(nau8825->regmap, 0x0030, 0x00d2); + /* ehance I2C SDA driver strength */ + regmap_write(nau8825->regmap, 0x0080, 0x0800); + /* enable ADC/DAC clocks */ + regmap_write(nau8825->regmap, 0x0001, 0x07fd); + /* headphone output */ + regmap_write(nau8825->regmap, 0x00C, 0x000c); + return 0; +} + +static const struct snd_soc_codec_driver soc_codec_driver_nau8825 = { + .probe = nau8825_codec_probe, + .remove = nau8825_codec_remove, + .suspend_bias_off = true, + .set_bias_level = nau8825_set_bias_level, + .controls = nau8825_snd_controls, + .num_controls = ARRAY_SIZE(nau8825_snd_controls), + .dapm_widgets = nau8825_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8825_dapm_widgets), + .dapm_routes = nau8825_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8825_dapm_routes), +}; + +static bool nau8825_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_RESET: + case NAU8825_ENA_CTRL: + case NAU8825_CLK_EN: + case NAU8825_CLK_DIVIDER: + case NAU8825_FLL_1: + case NAU8825_FLL_2: + case NAU8825_FLL_3: + case NAU8825_FLL_4: + case NAU8825_FLL_5: + case NAU8825_FLL_6: + case NAU8825_HEADSET_CTRL: + case NAU8825_JACK_DET_CTRL: + case NAU8825_IRQ_MASK: + case NAU8825_IRQ_STATUS: + case NAU8825_IRQ_CLEAR: + case NAU8825_IRQ_CTRL: + case NAU8825_SAR_ADC: + case NAU8825_VDET_COEFFICIENT: + case NAU8825_VDET_THRESHOLD_1: + case NAU8825_VDET_THRESHOLD_2: + case NAU8825_GPIO12_CTRL: + case NAU8825_GPIO34_CTRL: + case NAU8825_I2S_PCM_CTRL_1: + case NAU8825_I2S_PCM_CTRL_2: + case NAU8825_ADC_RATE: + case NAU8825_DAC_CTRL1: + case NAU8825_DAC_CTRL2: + case NAU8825_IMM_MODE_CTRL: + case NAU8825_IMM_RMS_VALUE: + case NAU8825_ADC_DGAIN_CTRL: + case NAU8825_DAC_MUTE_CTRL: + case NAU8825_HSVOL_CTRL: + case NAU8825_DACL_CTRL: + case NAU8825_DACR_CTRL: + case NAU8825_SAR_ADC_OUTPUT: + case NAU8825_ANALOG_CTRL_2: + case NAU8825_ANALOG_ADC_1: + case NAU8825_ANALOG_ADC_2: + case NAU8825_DAC_CTRL: + case NAU8825_MIC_BIAS: + case NAU8825_BOOST: + case NAU8825_CLASSG_CTRL: + case NAU8825_I2C_DEVICE_ID: + case NAU8825_BIAS_ADJ: + case NAU8825_POWER_UP_CTRL: + case NAU8825_CHARGE_BUMP_CTRL: + case NAU8825_CHARGE_BUMP_RD: + case NAU8825_GENERAL_STATUS: + return true; + default: + return false; + } +} + +static bool nau8825_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_RESET: + case NAU8825_CLK_DIVIDER: + case NAU8825_FLL_1: + case NAU8825_FLL_2: + case NAU8825_FLL_3: + case NAU8825_FLL_4: + case NAU8825_FLL_5: + case NAU8825_FLL_6: + case NAU8825_ANALOG_CTRL_2: + case NAU8825_ANALOG_ADC_1: + case NAU8825_ANALOG_ADC_2: + case NAU8825_DAC_CTRL: + case NAU8825_MIC_BIAS: + case NAU8825_BOOST: + case NAU8825_CLASSG_CTRL: + case NAU8825_I2C_DEVICE_ID: + case NAU8825_BIAS_ADJ: + case NAU8825_POWER_UP_CTRL: + case NAU8825_CHARGE_BUMP_CTRL: + case NAU8825_GENERAL_STATUS: + return true; + default: + return false; + } +} + +static const struct regmap_config nau8825_regmap = { + .reg_bits = 16, + .val_bits = 16, + .use_single_rw = true, + .max_register = NAU8825_MAX_REGISTER, + .volatile_reg = nau8825_volatile_register, + .readable_reg = nau8825_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8825_reg, + .num_reg_defaults = ARRAY_SIZE(nau8825_reg), +}; + +static int ena_adc(struct nau8825_priv *nau8825) +{ + /* adc & clock settings */ + regmap_update_bits(nau8825->regmap, NAU8825_ENA_CTRL, + NAU8825_ADC_CLK_MASK, NAU8825_ADC_CLK_EN); + regmap_update_bits(nau8825->regmap, NAU8825_ENA_CTRL, + NAU8825_ADC_MASK, NAU8825_ADC_EN); + /* adc PGA settings */ + regmap_update_bits(nau8825->regmap, NAU8825_POWER_UP_CTRL, + NAU8825_FEPGA_GAIN_MASK, 0x1b00); + regmap_update_bits(nau8825->regmap, NAU8825_POWER_UP_CTRL, + NAU8825_FEPGA_MASK, NAU8825_FEPGA_EN); + /* mic bias output level (1.1V) */ + regmap_update_bits(nau8825->regmap, NAU8825_MIC_BIAS, + NAU8825_MIC_BIAS_LVL_MSK, 0x04); + /* enable mic bias */ + regmap_update_bits(nau8825->regmap, NAU8825_MIC_BIAS, + NAU8825_MIC_POWERUP_MSK, NAU8825_MIC_POWERUP_EN); + mdelay(10); + return 0; +} + +static int dis_adc(struct nau8825_priv *nau8825) +{ + /* disable PGA */ + regmap_update_bits(nau8825->regmap, NAU8825_POWER_UP_CTRL, + NAU8825_FEPGA_MASK, NAU8825_FEPGA_DIS); + /* adc & clock settings */ + regmap_update_bits(nau8825->regmap, NAU8825_ENA_CTRL, + NAU8825_ADC_MASK, NAU8825_ADC_DIS); + regmap_update_bits(nau8825->regmap, NAU8825_ENA_CTRL, + NAU8825_ADC_CLK_MASK, NAU8825_ADC_CLK_DIS); + return 0; +} + +static int ena_dac(struct nau8825_priv *nau8825) +{ + /* charge bump settings */ + regmap_update_bits(nau8825->regmap, NAU8825_CHARGE_BUMP_CTRL, + 0x07ff, 0x720); + /* enable DAC clock */ + regmap_update_bits(nau8825->regmap, NAU8825_ENA_CTRL, + NAU8825_DAC_CLK_MASK, NAU8825_DAC_CLK_EN); + /* enable DAC channel */ + regmap_update_bits(nau8825->regmap, NAU8825_ENA_CTRL, + NAU8825_DAC_MASK, NAU8825_DAC_EN); + /* classG parameter settings */ + regmap_update_bits(nau8825->regmap, NAU8825_CLASSG_CTRL, + ~NAU8825_CLASSG_EN_MASK, 0x2007); + /* enable DAC */ + regmap_write(nau8825->regmap, NAU8825_DAC_CTRL, 0x332c); + /* output driver settings */ + regmap_write(nau8825->regmap, NAU8825_POWER_UP_CTRL, 0x073c); + /* power up main driver */ + regmap_update_bits(nau8825->regmap, NAU8825_POWER_UP_CTRL, + NAU8825_PUP_MAIN_MASK, NAU8825_PUP_MAIN_ENA); + mdelay(200); + /* enable charge bump */ + regmap_update_bits(nau8825->regmap, NAU8825_CHARGE_BUMP_CTRL, + NAU8825_PD_DAC_MASK, NAU8825_PD_DAC_DIS); + return 0; +} + +static int dis_dac(struct nau8825_priv *nau8825) +{ + /* disable charge bump */ + regmap_update_bits(nau8825->regmap, NAU8825_CHARGE_BUMP_CTRL, + NAU8825_PD_DAC_MASK, NAU8825_PD_DAC_ENA); + /* disable classG */ + regmap_update_bits(nau8825->regmap, NAU8825_CLASSG_CTRL, + NAU8825_CLASSG_PATH_MASK, NAU8825_CLASSG_PATH_DIS); + /* disable DAC */ + regmap_update_bits(nau8825->regmap, NAU8825_DAC_CTRL, + NAU8825_DAC_LR_MASK, NAU8825_DAC_LR_DIS); + regmap_update_bits(nau8825->regmap, NAU8825_DAC_CTRL, + NAU8825_DAC_CLK_LR_MASK, NAU8825_DAC_CLK_LR_DIS); + /* disable output driver */ + regmap_update_bits(nau8825->regmap, NAU8825_POWER_UP_CTRL, + 0x3f, 0x00); + /* disable DAC channel */ + regmap_update_bits(nau8825->regmap, NAU8825_ENA_CTRL, + NAU8825_L_DAC_MASK, NAU8825_L_DAC_DIS); + /* disable DAC clock */ + regmap_update_bits(nau8825->regmap, NAU8825_ENA_CTRL, + NAU8825_DAC_CLK_MASK, NAU8825_DAC_CLK_DIS); + return 0; +} + +static int nau8825_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct nau8825_priv *nau8825 = snd_soc_codec_get_drvdata(codec); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + config_fll_clk_12m(codec); + set_sys_clk(codec, NAU8825_MCLK); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ena_dac(nau8825); + else + ena_adc(nau8825); + break; + case SNDRV_PCM_TRIGGER_RESUME: + break; + case SNDRV_PCM_TRIGGER_STOP: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dis_dac(nau8825); + else + dis_adc(nau8825); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + break; + default: + break; + } + return 0; +} + +#define NAU8825_RATES SNDRV_PCM_RATE_8000_192000 +#define NAU8825_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +static const struct snd_soc_dai_ops nau8825_dai_ops = { + .trigger = nau8825_trigger, + .hw_params = nau8825_hw_params, + .set_sysclk = nau8825_dai_set_sysclk, + .set_fmt = nau8825_set_dai_fmt, + .digital_mute = nau8825_dac_mute, +}; + +static struct snd_soc_dai_driver nau8825_dai_driver[] = { + { + .name = "nau8825-aif1", + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8825_RATES, + .formats = NAU8825_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8825_RATES, + .formats = NAU8825_FORMATS, + }, + .ops = &nau8825_dai_ops, + } +}; + + +static int nau8825_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *i2c_id) +{ + struct nau8825_priv *nau8825; + int ret; + + nau8825 = devm_kzalloc(&i2c->dev, sizeof(*nau8825), + GFP_KERNEL); + if (nau8825 == NULL) + return -ENOMEM; + nau8825->i2c = i2c; + i2c_set_clientdata(i2c, nau8825); + nau8825->regmap = devm_regmap_init_i2c(i2c, &nau8825_regmap); + if (IS_ERR(nau8825->regmap)) { + ret = PTR_ERR(nau8825->regmap); + dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); + return ret; + } + /* software reset */ + regmap_write(nau8825->regmap, NAU8825_RESET, 0x00); + regmap_write(nau8825->regmap, NAU8825_RESET, 0x00); + + /* register sound card */ + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_driver_nau8825, + nau8825_dai_driver, + ARRAY_SIZE(nau8825_dai_driver)); + return ret; +} + +static int nau8825_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + return 0; +} + +static const struct i2c_device_id nau8825_i2c_id[] = { + {"nau8825", 0}, + { } +}; +MODULE_DEVICE_TABLE(i2c, nau8825_i2c_id); + +static const struct acpi_device_id nau8825_acpi_match[] = { + { "10508825", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, nau8825_acpi_match); + +static struct i2c_driver nau8825_i2c_driver = { + .driver = { + .name = "nau8825", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(nau8825_acpi_match), + }, + .probe = nau8825_i2c_probe, + .remove = (nau8825_i2c_remove), + .id_table = nau8825_i2c_id, +}; +module_i2c_driver(nau8825_i2c_driver); + +MODULE_DESCRIPTION("ASoC NAU8825 codec driver"); +MODULE_AUTHOR("Nuvoton"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h new file mode 100644 index 0000000..3050e7c --- /dev/null +++ b/sound/soc/codecs/nau8825.h @@ -0,0 +1,399 @@ +/* + * nau8825.h + * + * Copyright 2015 Nuvoton Technology Corp. + * Author: Meng-Huang Kuo <mhkuo(a)nuvoton.com> + * + * 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. + */ + +#ifndef _NAU8825_H +#define _NAU8825_H + +#define NAU8825_RESET 0x00 +#define NAU8825_ENA_CTRL 0x01 +#define NAU8825_CLK_EN 0x02 +#define NAU8825_CLK_DIVIDER 0x03 +#define NAU8825_FLL_1 0x04 +#define NAU8825_FLL_2 0x05 +#define NAU8825_FLL_3 0x06 +#define NAU8825_FLL_4 0x07 +#define NAU8825_FLL_5 0x08 +#define NAU8825_FLL_6 0x09 +#define NAU8825_HEADSET_CTRL 0x0C +#define NAU8825_JACK_DET_CTRL 0x0D +#define NAU8825_IRQ_MASK 0x0F +#define NAU8825_IRQ_STATUS 0x10 +#define NAU8825_IRQ_CLEAR 0x11 +#define NAU8825_BTN_STATUS 0x11 +#define NAU8825_IRQ_CTRL 0x12 +#define NAU8825_SAR_ADC 0x13 +#define NAU8825_VDET_COEFFICIENT 0x14 +#define NAU8825_VDET_THRESHOLD_1 0x15 +#define NAU8825_VDET_THRESHOLD_2 0x16 +#define NAU8825_GPIO34_CTRL 0x19 +#define NAU8825_GPIO12_CTRL 0x1A +#define NAU8825_TDM_CTRL 0x1B +#define NAU8825_I2S_PCM_CTRL_1 0x1C +#define NAU8825_I2S_PCM_CTRL_2 0x1D +#define NAU8825_BIQ_CTRL 0x20 +#define NAU8825_BIQ_COF1 0x21 +#define NAU8825_BIQ_COF2 0x22 +#define NAU8825_BIQ_COF3 0x23 +#define NAU8825_BIQ_COF4 0x24 +#define NAU8825_BIQ_COF5 0x25 +#define NAU8825_BIQ_COF6 0x26 +#define NAU8825_BIQ_COF7 0x27 +#define NAU8825_BIQ_COF8 0x28 +#define NAU8825_BIQ_COF9 0x29 +#define NAU8825_BIQ_COF10 0x2A +#define NAU8825_ADC_RATE 0x2B +#define NAU8825_DAC_CTRL1 0x2C +#define NAU8825_DAC_CTRL2 0x2D +#define NAU8825_DAC_DGAIN_CTRL 0x2F +#define NAU8825_ADC_DGAIN_CTRL 0x30 +#define NAU8825_DAC_MUTE_CTRL 0x31 +#define NAU8825_HSVOL_CTRL 0x32 +#define NAU8825_DACL_CTRL 0x33 +#define NAU8825_DACR_CTRL 0x34 +#define NAU8825_ADC_DRC_KNEE_IP12 0x38 +#define NAU8825_ADC_DRC_KNEE_IP34 0x39 +#define NAU8825_ADC_DRC_SLOPES 0x3A +#define NAU8825_ADC_DRC_ATKDCY 0x3B +#define NAU8825_DAC_DRC_KNEE_IP12 0x45 +#define NAU8825_DAC_DRC_KNEE_IP34 0x46 +#define NAU8825_DAC_DRC_SLOPES 0x47 +#define NAU8825_DAC_DRC_ATKDCY 0x48 +#define NAU8825_IMM_MODE_CTRL 0x4C +#define NAU8825_IMM_RMS_VALUE 0x4D +#define NAU8825_CLASSG_CTRL 0x50 +#define NAU8825_I2C_DEVICE_ID 0x58 +#define NAU8825_SAR_ADC_OUTPUT 0x59 +#define NAU8825_BIAS_ADJ 0x66 +#define NAU8825_ANALOG_CTRL_2 0x6A +#define NAU8825_ANALOG_ADC_1 0x71 +#define NAU8825_ANALOG_ADC_2 0x72 +#define NAU8825_DAC_CTRL 0x73 +#define NAU8825_MIC_BIAS 0x74 +#define NAU8825_BOOST 0x76 +#define NAU8825_POWER_UP_CTRL 0x7F +#define NAU8825_CHARGE_BUMP_CTRL 0x80 +#define NAU8825_CHARGE_BUMP_RD 0x81 +#define NAU8825_GENERAL_STATUS 0x82 +#define NAU8825_MAX_REGISTER 0xFF + +/* reg. NAU8825_ENA_CTRL (0x01) */ +#define NAU8825_R_DAC_MASK (0x1 << 10) +#define NAU8825_R_DAC_EN (0x1 << 10) +#define NAU8825_R_DAC_DIS (0x0 << 10) +#define NAU8825_L_DAC_MASK (0x1 << 9) +#define NAU8825_L_DAC_EN (0x1 << 9) +#define NAU8825_L_DAC_DIS (0x0 << 9) +#define NAU8825_DAC_MASK (0x3 << 9) +#define NAU8825_DAC_EN (0x3 << 9) +#define NAU8825_DAC_DIS (0x0 << 9) + +#define NAU8825_R_DAC_MASK (0x1 << 10) +#define NAU8825_R_DAC_EN (0x1 << 10) +#define NAU8825_R_DAC_DIS (0x0 << 10) +#define NAU8825_L_DAC_MASK (0x1 << 9) +#define NAU8825_L_DAC_EN (0x1 << 9) +#define NAU8825_L_DAC_DIS (0x0 << 9) +#define NAU8825_ADC_MASK (0x1 << 8) +#define NAU8825_ADC_EN (0x1 << 8) +#define NAU8825_ADC_DIS (0x0 << 8) +#define NAU8825_ADC_CLK_MASK (0x1 << 7) +#define NAU8825_ADC_CLK_EN (0x1 << 7) +#define NAU8825_ADC_CLK_DIS (0x0 << 7) +#define NAU8825_DAC_CLK_MASK (0x1 << 6) +#define NAU8825_DAC_CLK_EN (0x1 << 6) +#define NAU8825_DAC_CLK_DIS (0x0 << 6) +#define NAU8825_IMM_CLK_MASK (0x1 << 5) +#define NAU8825_IMM_CLK_EN (0x1 << 5) +#define NAU8825_IMM_CLK_DIS (0x0 << 5) +#define NAU8825_I2S_CLK_MASK (0x1 << 4) +#define NAU8825_I2S_CLK_EN (0x1 << 4) +#define NAU8825_I2S_CLK_DIS (0x0 << 4) +#define NAU8825_BIST_CLK_MASK (0x1 << 3) +#define NAU8825_BIST_CLK_EN (0x1 << 3) +#define NAU8825_BIST_CLK_DIS (0x0 << 3) +#define NAU8825_OTP_CLK_MASK (0x1 << 2) +#define NAU8825_OTP_CLK_EN (0x1 << 2) +#define NAU8825_OTP_CLK_DIS (0x0 << 2) +#define NAU8825_SAR_CLK_MASK (0x1 << 1) +#define NAU8825_SAR_CLK_EN (0x1 << 1) +#define NAU8825_SAR_CLK_DIS (0x0 << 1) +#define NAU8825_DRC_CLK_MASK (0x1 << 0) +#define NAU8825_DRC_CLK_EN (0x1 << 0) +#define NAU8825_DRC_CLK_DIS (0x0 << 0) + +/* reg. NAU8825_CLK_DIVIDER (0x03) */ +#define NAU8825_SYSCLK_EN_MASK (0x1 << 15) +#define NAU8825_SYSCLK_EN (0x1 << 15) +#define NAU8825_SYSCLK_DIS (0x0 << 15) +#define NAU8825_CLK_MCLK_SRC_MASK (0xF << 0) + +/* reg. NAU8825_FLL_1 (0x04) */ +#define NAU8825_FLL_RATIO_MASK (0x7F << 0) + +/* reg. NAU8825_FLL_3 (0x06) */ +#define NAU8825_FLL_INTEGER_MASK (0x3FF << 0) + +/* reg. NAU8825_FLL_4 (0x07) */ +#define NAU8825_FLL_REF_DIV_MASK (0x3 << 10) + +/* reg. NAU8825_FLL_5 (0x08) */ +#define NAU8825_FLL_FILTER_SW_MASK (0x1 << 14) + +/* reg. NAU8825_FLL_6 (0x09) */ +#define NAU8825_DCO_EN_MASK (0x1 << 15) +#define NAU8825_DCO_EN (0x1 << 15) +#define NAU8825_DCO_DIS (0x0 << 15) +#define NAU8825_SDM_EN_MASK (0x1 << 14) + +/* reg. NAU8825_HEADSET_CTRL (0x0c) */ +#define NAU8825_RESET_HSD_MASK (0x1 << 15) +#define NAU8825_RESET_HSD_EN (0x1 << 15) +#define NAU8825_RESET_HSD_DIS (0x0 << 15) +#define NAU8825_AUTO_DETECT_MASK (0x1 << 6) +#define NAU8825_AUTO_DETECT_EN (0x1 << 6) +#define NAU8825_AUTO_DETECT_DIS (0x0 << 6) +#define NAU8825_MANUAL_START_MASK (0x1 << 4) +#define NAU8825_MANUAL_START_EN (0x1 << 4) +#define NAU8825_MANUAL_START_DIS (0x0 << 4) +#define NAU8825_ENGND_MASK (0x3 << 2) + +/* reg. NAU8825_JACK_DET_CTRL (0x0d) */ +#define NAU8825_DB_BP_MODE_MASK (0x1 << 8) +#define NAU8825_DB_BP_MODE_EN (0x1 << 8) +#define NAU8825_DB_BP_MODE_DIS (0x0 << 8) + +/* reg. NAU8825_IRQ_MASK (0x0f) */ +#define NAU8825_IRQ_QE_MASK (0x1 << 11) +#define NAU8825_IRQ_QE_EN (0x1 << 11) +#define NAU8825_IRQ_QE_DIS (0x0 << 11) +#define NAU8825_IRQ_EJECT_MASK (0x1 << 2) +#define NAU8825_IRQ_EJECT_EN (0x1 << 2) +#define NAU8825_IRQ_EJECT_DIS (0x0 << 2) + + +/* reg. NAU8825_IRQ_STATUS (0x10) */ +#define NAU8825_HSD_COMPLETE_INT (0x1 << 10) +#define NAU8825_EMRG_INT (0x1 << 9) +#define NAU8825_RMS_INT (0x1 << 8) +#define NAU8825_BTN_RELEASE_INT (0x1 << 7) +#define NAU8825_LONG_BTN_INT (0x1 << 6) +#define NAU8825_SHORT_BTN_INT (0x1 << 5) +#define NAU8825_BTN_INT (0x7 << 5) +#define NAU8825_MIC_DET_INT (0x1 << 4) +#define NAU8825_JACK_EJCT_INT (0x3 << 2) +#define NAU8825_JACK_DET_INT (0x3 << 0) +#define NAU8825_JACK_OUT (0x1 << 2) +#define NAU8825_JACK_IN (0x1 << 0) +#define NAU8825_JACK_INT (0xF << 0) + +/* reg. NAU8825_BTN_STATUS (0x11) */ +#define NAU8825_SHORT_BTN_MASK (0xFF << 8) +#define NAU8825_LONG_BTN_MASK (0xFF << 0) +#define NAU8825_BTN_0 (0x01 << 0) +#define NAU8825_BTN_1 (0x01 << 1) +#define NAU8825_BTN_2 (0x01 << 2) +#define NAU8825_BTN_3 (0x01 << 3) + +/* reg. NAU8825_IRQ_CTRL (0x12) */ +#define NAU8825_HEADSET_INT_MASK (0x1 << 10) +#define NAU8825_HEADSET_INT_EN (0x0 << 10) +#define NAU8825_HEADSET_INT_DIS (0x1 << 10) +#define NAU8825_RMS_INT_MASK (0x1 << 8) +#define NAU8825_RMS_INT_EN (0x0 << 8) +#define NAU8825_RMS_INT_DIS (0x1 << 8) +#define NAU8825_KEYREL_INT_MASK (0x1 << 7) +#define NAU8825_KEYREL_INT_EN (0x0 << 7) +#define NAU8825_KEYREL_INT_DIS (0x1 << 7) +#define NAU8825_LONGKEY_INT_MASK (0x1 << 6) +#define NAU8825_LONGKEY_INT_EN (0x0 << 6) +#define NAU8825_LONGKEY_INT_DIS (0x1 << 6) +#define NAU8825_SHORTKEY_INT_MASK (0x1 << 5) +#define NAU8825_SHORTKEY_INT_EN (0x0 << 5) +#define NAU8825_SHORTKEY_INT_DIS (0x1 << 5) +#define NAU8825_EJECT_INT_MASK (0x1 << 2) +#define NAU8825_EJECT_INT_EN (0x0 << 2) +#define NAU8825_EJECT_INT_DIS (0x1 << 2) +#define NAU8825_JACK_IN_INT_MASK (0x1 << 0) +#define NAU8825_JACK_IN_INT_EN (0x0 << 0) +#define NAU8825_JACK_IN_INT_DIS (0x1 << 0) + +/* reg. NAU8825_SAR_ADC (0x13) */ +#define NAU8825_SAR_EN_MASK (0x1 << 12) +#define NAU8825_SAR_EN (0x1 << 12) +#define NAU8825_SAR_DIS (0x0 << 12) +#define NAU8825_JKSLV_MASK (0x1 << 11) +#define NAU8825_JKSLV_EN (0x1 << 11) +#define NAU8825_JKSLV_DIS (0x0 << 11) + +/* reg. NAU8825_I2S_PCM_CTRL_1 (0x1C) */ +#define NAU8825_I2S_TRI_STATE_MASK (0x1 << 15) +#define NAU8825_I2S_TRI_STATE_EN (0x1 << 15) +#define NAU8825_I2S_TRI_STATE_DIS (0x0 << 15) +#define NAU8825_I2S_BP_MASK (0x1 << 7) +#define NAU8825_I2S_BP_INV (0x1 << 7) +#define NAU8825_I2S_PCMB_MASK (0x1 << 6) +#define NAU8825_I2S_PCMB_EN (0x1 << 6) +#define NAU8825_I2S_DL_MASK (0x3 << 2) +#define NAU8825_I2S_DF_MASK 0x3 +#define NAU8825_I2S_DF_RIGHT 0x0 +#define NAU8825_I2S_DF_LEFT 0x1 +#define NAU8825_I2S_DF_I2S 0x2 +#define NAU8825_I2S_DF_PCM_A 0x3 +#define NAU8825_I2S_DF_PCM_B 0x3 + +/* reg. NAU8825_I2S_PCM_CTRL_2 (0x1D) */ +#define NAU8825_I2S_MS_MASK (0x1 << 3) +#define NAU8825_I2S_MS_MASTER (0x1 << 3) +#define NAU8825_I2S_MS_SLAVE (0x0 << 3) + +/* reg. NAU8825_BIQ_CTRL (0x20) */ +#define NAU8825_BIQ_WRT_MASK (0x1 << 4) +#define NAU8825_BIQ_WRT_EN (0x1 << 4) +#define NAU8825_BIQ_WRT_DIS (0x0 << 4) + +/* reg. NAU8825_ADC_DGAIN_CTRL (0x30) */ +#define NAU8825_ADC_DGAIN_SFT 0 + +/* reg. NAU8825_DAC_MUTE_CTRL (0x31) */ +#define NAU8825_SOFT_MUTE_MASK (0x1 << 9) +#define NAU8825_SOFT_MUTE_EN (0x1 << 9) +#define NAU8825_SOFT_MUTE_DIS (0x0 << 9) + +/* reg. NAU8825_HSVOL_CTRL (0x32) */ +#define NAU8825_R_MUTE (0x1 << 15) +#define NAU8825_R_MUTE_SFT 15 +#define NAU8825_L_MUTE (0x1 << 14) +#define NAU8825_L_MUTE_SFT 14 +#define NAU8825_L_HSVOL_SFT 6 +#define NAU8825_R_HSVOL_SFT 0 + +/* reg. NAU8825_IMM_MODE_CTRL (0x4C) */ +#define NAU8825_IMM_MODE_EN_MASK (0x1 << 3) +#define NAU8825_IMM_MODE_EN (0x1 << 3) +#define NAU8825_IMM_MODE_DIS (0x0 << 3) + +/* reg. NAU8825_CLASSG_CTRL (0x50) */ +#define NAU8825_CLASSG_PATH_MASK (0x3 << 1) +#define NAU8825_CLASSG_PATH_EN (0x3 << 1) +#define NAU8825_CLASSG_PATH_DIS (0x3 << 0) +#define NAU8825_CLASSG_LEFT_SHIFT 1 +#define NAU8825_CLASSG_EN_SHIFT 0 +#define NAU8825_CLASSG_EN_MASK (0x1 << 0) +#define NAU8825_CLASSG_EN_SHIFT 0 + +/*#define NAU8825_I2C_DEVICE_ID (0x58) */ +#define NAU8825_GPIO2JD1 (0x1 << 7) + +/* reg. NAU8825_BIAS_ADJ (0x66) */ +#define NAU8825_VMID_MASK (0x1 << 6) +#define NAU8825_VMID_EN (0x1 << 6) +#define NAU8825_VMID_DIS (0x0 << 6) +#define NAU8825_VMIDSEL_MASK (0x3 << 4) +#define NAU8825_VMIDSEL_125KOHM (0x2 << 4) + +/* reg. NAU8825_ANALOG_CTRL_2 (0x6A) */ +#define NAU8825_VMID_MASK (0x1 << 6) +#define NAU8825_VMID_EN (0x1 << 6) +#define NAU8825_VMID_DIS (0x0 << 6) + +/* reg. NAU8825_DAC_CTRL (0x73) */ +#define NAU8825_DAC_R_SFT 13 +#define NAU8825_DAC_L_SFT 12 +#define NAU8825_DAC_CLK_R_SFT 9 +#define NAU8825_DAC_CLK_L_SFT 8 +#define NAU8825_DAC_LR_MASK (0x3 << 12) +#define NAU8825_DAC_LR_EN (0x3 << 12) +#define NAU8825_DAC_LR_DIS (0x0 << 12) +#define NAU8825_DAC_CLK_LR_MASK (0x3 << 8) +#define NAU8825_DAC_CLK_LR_EN (0x3 << 8) +#define NAU8825_DAC_CLK_LR_DIS (0x0 << 8) +#define NAU8825_DAC_CLK_DELAY_MASK (0x7 << 4) +#define NAU8825_DAC_REF_VOLT_MASK (0x3 << 2) + +/* reg. NAU8825_MIC_BIAS (0x74) */ +#define NAU8825_INT2KB_MSK (0x1 << 14) +#define NAU8825_INT2KB_EN (0x1 << 14) +#define NAU8825_INT2KB_DIS (0x0 << 14) +#define NAU8825_INT2KA_MSK (0x1 << 12) +#define NAU8825_INT2KA_EN (0x1 << 12) +#define NAU8825_INT2KA_DIS (0x0 << 12) +#define NAU8825_MIC_POWERUP_MSK (0x1 << 8) +#define NAU8825_MIC_POWERUP_EN (0x1 << 8) +#define NAU8825_MIC_POWERUP_DIS (0x0 << 8) +#define NAU8825_MIC_BIAS_LVL_MSK (0x7 << 0) + +/* reg. NAU8825_BOOST (0x76) */ +#define NAU8825_G_BIAS_MASK (0x1 << 12) +#define NAU8825_G_BIAS_SFT 12 +#define NAU8825_G_BIAS_EN (0x1 << 12) +#define NAU8825_G_BIAS_DIS (0x0 << 12) +#define NAU8825_BOOST_DRV_MASK (0x1 << 9) +#define NAU8825_BOOST_DRV_EN (0x0 << 9) +#define NAU8825_BOOST_DRV_DIS (0x1 << 9) + +/* reg. NAU8825_POWER_UP_CTRL (0x7F) */ +#define NAU8825_FEPGA_MASK (0x1 << 14) +#define NAU8825_FEPGA_EN (0x1 << 14) +#define NAU8825_FEPGA_DIS (0x0 << 14) +#define NAU8825_FEPGA_GAIN_MASK (0x3F << 8) +#define NAU8825_OUTPUT_DRIVER_MASK (0x3F << 0) +#define NAU8825_OUTPUT_DRIVER_EN (0x3F << 0) +#define NAU8825_OUTPUT_DRIVER_DIS (0x0 << 0) +#define NAU8825_PUP_INTEG_MASK (0x3 << 4) +#define NAU8825_PUP_INTEG_ENA (0x3 << 4) +#define NAU8825_PUP_INTEG_DIS (0x0 << 4) +#define NAU8825_PUP_OUT_MASK (0x3 << 2) +#define NAU8825_PUP_OUT_ENA (0x3 << 2) +#define NAU8825_PUP_OUT_DIS (0x0 << 2) +#define NAU8825_PUP_MAIN_MASK (0x3 << 0) +#define NAU8825_PUP_MAIN_ENA (0x3 << 0) +#define NAU8825_PUP_MAIN_DIS (0x0 << 0) + + +/* reg. NAU8825_CHARGE_BUMP_CTRL (0x80) */ +#define NAU8825_PD_DAC_MASK (0x3 << 8) +#define NAU8825_PD_DAC_ENA (0x3 << 8) +#define NAU8825_PD_DAC_DIS (0x0 << 8) +#define NAU8825_CB_CLK_MASK (0x1 << 7) +#define NAU8825_CB_CLK_EN (0x1 << 7) +#define NAU8825_CB_CLK_DIS (0x0 << 7) +#define NAU8825_CB_MASK (0x1 << 5) +#define NAU8825_CB_EN (0x1 << 5) +#define NAU8825_CB_DIS (0x0 << 5) + +/* reg. NAU8825_GENERAL_STATUS (0x82) */ +#define NAU8825_OUT12_MASK (0x3 << 10) +#define NAU8825_OUT2EN_OUT1EN (0x3 << 10) +#define NAU8825_OUT2EN_OUT1DIS (0x2 << 10) +#define NAU8825_OUT2DIS_OUT1EN (0x1 << 10) +#define NAU8825_OUT2DIS_OUT1DIS (0x0 << 10) + +/* Volume Rescale */ +#define NAU8825_VOL_RSCL_RANGE 0x36 +#define NAU8825_ADC_VOL_RSCL_RANGE 0xFF + +/* Data format */ +#define NAU8825_I2S_DL_16 (0x0 << 2) +#define NAU8825_I2S_DL_20 (0x1 << 2) +#define NAU8825_I2S_DL_24 (0x2 << 2) +#define NAU8825_I2S_DL_32 (0x3 << 2) + +enum { + NAU8825_INTERNALCLOCK = 0, + NAU8825_MCLK, +}; + +struct nau8825_priv { + struct regmap *regmap; + struct i2c_client *i2c; +}; +#endif /* _NAU8825_H */ + -- 1.9.3

4 5

[alsa-devel] [PATCH 0/7] ASoC: Cleanup AC'97 reset handling
by Lars-Peter Clausen 27 Jul '15

27 Jul '15

AC'97 devices require that they are properly reset before they can be used. Currently ASoC drivers do this by hand each on its own. This lead to many similar but slightly different implementations. E.g. Some supporting warm reset, some don't. And even though AC'97 do have a device ID which uniquely identifies a device most drivers only use the content of register 0 for device verification. This patch series introduces a new generic helper function in the ALSA AC'97 core that can be used to reset a device and verify its ID if required. And then converts drivers to use this new helper function instead of custom code. snd_soc_new_ac97_codec() is extended to be able to perform a reset between the allocation and registration of the AC'97 device, this will allow more drivers to use it. - Lars Lars-Peter Clausen (7): ALSA: ac97: Add helper function to reset the AC97 device ASoC: ac97: Add support for resetting device before registration ASoC: ad1980: Use core AC'97 reset helper ASoC: stac9766: Use core reset helper ASoC: wm9705: Use core AC'97 reset helper ASoC: wm9712: Use core AC'97 reset helper ASoC: wm9713: Use core AC'97 reset helper include/sound/ac97_codec.h | 2 ++ include/sound/soc.h | 3 ++- sound/ac97_bus.c | 62 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ad1980.c | 36 +++++++++----------------- sound/soc/codecs/stac9766.c | 57 +++++------------------------------------ sound/soc/codecs/wm9705.c | 40 ++++++----------------------- sound/soc/codecs/wm9712.c | 45 ++++++-------------------------- sound/soc/codecs/wm9713.c | 48 ++++++----------------------------- sound/soc/codecs/wm9713.h | 2 -- sound/soc/soc-ac97.c | 30 ++++++++++++++++++---- 10 files changed, 134 insertions(+), 191 deletions(-) -- 2.1.4

4 20

[alsa-devel] [ALC668]: Asus N751JK - Incorrect default pin assignment for external base speaker and external microphone not working
by Arthur Borsboom 27 Jul '15

27 Jul '15

Hi, While being in the process of making my audio work on my new notebook, I noticed that the audio hardware is partially working out-of-the-Linux-kernel-box v4.0.7 Arch. The following two things seem not to work. -------------------------------- 1. External base speaker -------------------------------- The Asus N751JK notebook (ALC668) has a small external 'base' speaker (jack 2,5mm 4 rings), which I only can get to work with the help of hdajackretask with the following setting: * Override unconnected pin 0x1a and connect it to Internal Speaker (LFE). In the sound kernel tree I have noticed PCI ID fixes which are more or less equal to this situation. https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/pci… This SND_PCI_QUIRK definition looks a lot like it: SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_BASS_1A), The Vendor/PCI ID is different for this laptop and I guess a new line should be added something like this: SND_PCI_QUIRK(0x1043, 0x17bd, "Asus N751JK", ALC662_FIXUP_BASS_1A), -------------------------------- 2. External microphone -------------------------------- The Asus N751JK notebook (ALC668) has a 3,5mm 4-ring combo jack for headphones and microphone. The headphone is detected correct and auto-mute works good (jack detection). The microphone is not detected and does not work. I was unable to get it to work with hdajackretask or any of the Asus model settings, such as: * options snd_hda_intel model=,asus-mode4 I was able to get the microphone to work with the following setting: * options snd_hda_intel model=,dell-headset-multi This setting creates two new items in PulseAudio making a total of three: * Internal microphone (works out of the box) * Headset microphone (works indeed with a headset) * Microphone (don't know if this works, since I don't have a single microphone, but it does not work with the headset) Unfortunately the auto-input-device-switch to the Headset microphone does not seem to work. When I plugin a headset, the headphones get detected and the Internal speakers are muted. The Input device does not automatically switch from Internal microphone to Headset microphone. I have to do this manually, after which it works. I don't know if the microphone has a jack detection. If it does not, maybe it is possible to use the headphones jack detection? I have added the alsa-info without any snd_intel / alsa modifications here: http://www.alsa-project.org/db/?f=0b5135bd12f28bb5cb182f83bfa21bc50022faa6 I don't mind to help to test, for example module options, patches and/or compile kernels. Who is able and willing to help to fix this issue? With kind regards, Arthur Borsboom. -- Arthur Borsboom Email: arthurborsboom(a)gmail.com

2 20

[alsa-devel] Possible GPL-licensed header included in alsa-lib
by Clemens Lang 27 Jul '15

27 Jul '15

Hi, I recently reviewed the alsa-lib 1.0.27.2 source code from a licensing perspective to ensure compliant usage and distribution. It seems the intention is for the libraries to be covered by LGPL. I had a look at include/list.h, which seems to be licensed under GPL-2.0. It does not have a license header or any other copyright information, but it states it was taken from Linux 2.4.0: include/list.h, line 5: | This code was taken from the Linux 2.4.0 kernel. [jaroslav] I'm not sure one could argue that the header was trivial, especially since it does not only contain constants, structures and interface definitions, but also implementations of inline functions. I found one previous inquiry about a similar issue[1], but it does not cover the include/list.h header. Two questions: 1. Have the copyright holders of the code in question agreed to re-license under a different, LGPL-preserving license and, if yes, is there some documentation about that? 2. If the header is indeed covered by GPL-2.0, would you accept patches to replace it with a different implementation, e.g. Rusty Russell's MIT-licensed list from ccan[2]? [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2011-March/038267.html [2] http://ccodearchive.net/info/list.html Thank you for your time, Clemens Lang -- BMW Car IT GmbH Clemens Lang Spezialist Entwicklung Lise-Meitner-Straße 14 89081 Ulm Mail: clemens.lang(a)bmw-carit.de Web: http://www.bmw-carit.de -------------------------------------------------------------------- BMW Car IT GmbH Geschäftsführer: Michael Würtenberger und Reinhard Stolle Sitz und Registergericht: München HRB 134810 --------------------------------------------------------------------

3 12

[alsa-devel] snd_pcm_set_params faild on set_buffer_time_near and set_period_time_near
by Martin Geier 27 Jul '15

27 Jul '15

Hallo, I tried to configure alsa device with snd_pcm_set_params function, requested parameters was: channel 2, rate: 22050, soft_resample: 1, latency: 50000 Function snd_pcm_hw_params_set_buffer_time_near and snd_pcm_hw_param_set_near is called with: min: 50000, max: 50000, mindir: 0, maxdir: -1 and input params was: ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 22050 PERIOD_TIME: (362 743039) PERIOD_SIZE: [8 16384] PERIOD_BYTES: [32 65536] PERIODS: [2 19] BUFFER_TIME: (725 1486078) BUFFER_SIZE: [16 32768] BUFFER_BYTES: [64 131072] TICK_TIME: ALL unfortunately, this function fails, on function snd_pcm_hw_param_set_first: ALSA ERROR hw_params: set_near (BUFFER_TIME) value = 50000 : Invalid argument ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 22050 PERIOD_TIME: (4580 5533) PERIOD_SIZE: NONE PERIOD_BYTES: [404 488] PERIODS: 10 BUFFER_TIME: (50022 50023) BUFFER_SIZE: 1103 BUFFER_BYTES: 4412 TICK_TIME: ALL This shouldn't be problem because in snd_pcm_set_params is snd_pcm_hw_params_set_period_time_near called. This function should be called with same params as snd_pcm_hw_params_set_buffer_time_near but the input params are: snd_pcm_hw_param_set_near: min: 12500, max: 12500, mindir: 0, maxdir: -1 hw_params: snd_pcm_hw_param_set_near (PERIOD_TIME) ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 22050 PERIOD_TIME: (4580 5533) PERIOD_SIZE: NONE PERIOD_BYTES: [404 488] PERIODS: 10 BUFFER_TIME: (50022 50023) BUFFER_SIZE: 1103 BUFFER_BYTES: 4412 TICK_TIME: ALL because in snd_pcm_hw_param_set_near is if (last) err = snd_pcm_hw_param_set_last(pcm, params, var, val, dir); else err = snd_pcm_hw_param_set_first(pcm, params, var, val, dir); if (err < 0) dump_hw_params(params, "set_near", var, *val, err); return err; and in error part is only debug output and not restore original parameters. Environment: alsa-lib 1.0.27.1 TI AM335x soc Simple fix is attached, but I am not sure if it is correct. BR Martin Geier -- *StreamUnlimited* High Tech Campus Vienna, Gutheil-Schoder-Gasse 10, A-1102 Vienna, Austria Levocska 9, 851 01 Bratislava, Slovakia Email: martin.geier(a)streamunlimited.com www.streamunlimited.com

3 9

[alsa-devel] [PATCH] ALSA: hda - Apply fixup for another Toshiba Satellite S50D
by Takashi Iwai 27 Jul '15

27 Jul '15

Toshiba Satellite S50D has another model with a different PCI SSID (1179:fa93) while the previous fixup was for 1179:fa91. Adjust the fixup entry with SND_PCI_QUIRK_MASK() to match with both devices. Reported-by: Tim Sample <timsample(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_sigmatel.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index dcc7fe91244c..9d947aef2c8b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2920,7 +2920,8 @@ static const struct snd_pci_quirk stac92hd83xxx_fixup_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x148a, "HP Mini", STAC_92HD83XXX_HP_LED), SND_PCI_QUIRK_VENDOR(PCI_VENDOR_ID_HP, "HP", STAC_92HD83XXX_HP), - SND_PCI_QUIRK(PCI_VENDOR_ID_TOSHIBA, 0xfa91, + /* match both for 0xfa91 and 0xfa93 */ + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_TOSHIBA, 0xfffd, 0xfa91, "Toshiba Satellite S50D", STAC_92HD83XXX_GPIO10_EAPD), {} /* terminator */ }; -- 2.4.6

1 0

[alsa-devel] [PATCH] ALSA: fireworks: add support for AudioFire2 quirk
by Takashi Sakamoto 27 Jul '15

27 Jul '15

Fireworks uses TSB43CB43(IceLynx-Micro) as its IEC 61883-1/6 interface. This chip includes ARM7 core, and loads and runs program. The firmware is stored in on-board memory and loaded every powering-on. Echo Audio ships several versions of firmwares for each model. These firmwares have each quirk and the quirk changes a sequence of packets. AudioFire2 has a quirk to transfer a first packet with non-zero in its dbc field. This causes ALSA Fireworks driver to detect discontinuity. As long as I investigated, firmware 5.7, 5.7.6 and 5.8 have this quirk. This commit adds a support for the quirk to handle AudioFire2 packets. For safe, CIP_SKIP_INIT_DBC_CHECK is applied to all versions of AudioFire2's firmwares. 02 00050002 90ffffff <- 42 0005000a 90013000 42 00050012 90014400 42 0005001a 90015800 02 0005001a 90ffffff 42 00050022 90019000 42 0005002a 9001a400 42 00050032 9001b800 02 00050032 90ffffff 42 0005003a 9001d000 42 00050042 9001e400 42 0005004a 9001f800 02 0005004a 90ffffff Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/fireworks/fireworks.c | 2 ++ sound/firewire/fireworks/fireworks.h | 1 + sound/firewire/fireworks/fireworks_stream.c | 3 +++ 3 files changed, 6 insertions(+) diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index 2682e7e..c670db4 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -248,6 +248,8 @@ efw_probe(struct fw_unit *unit, err = get_hardware_info(efw); if (err < 0) goto error; + if (entry->model_id == MODEL_ECHO_AUDIOFIRE_2) + efw->is_af2 = true; if (entry->model_id == MODEL_ECHO_AUDIOFIRE_9) efw->is_af9 = true; diff --git a/sound/firewire/fireworks/fireworks.h b/sound/firewire/fireworks/fireworks.h index 4f0201a..c33252b 100644 --- a/sound/firewire/fireworks/fireworks.h +++ b/sound/firewire/fireworks/fireworks.h @@ -70,6 +70,7 @@ struct snd_efw { bool resp_addr_changable; /* for quirks */ + bool is_af2; bool is_af9; u32 firmware_version; diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c index c55db1b..a0762dd 100644 --- a/sound/firewire/fireworks/fireworks_stream.c +++ b/sound/firewire/fireworks/fireworks_stream.c @@ -172,6 +172,9 @@ int snd_efw_stream_init_duplex(struct snd_efw *efw) efw->tx_stream.flags |= CIP_DBC_IS_END_EVENT; /* Fireworks reset dbc at bus reset. */ efw->tx_stream.flags |= CIP_SKIP_DBC_ZERO_CHECK; + /* AudioFire2 starts packets with non-zero dbc. */ + if (efw->is_af2) + efw->tx_stream.flags |= CIP_SKIP_INIT_DBC_CHECK; /* AudioFire9 always reports wrong dbs. */ if (efw->is_af9) efw->tx_stream.flags |= CIP_WRONG_DBS; -- 2.1.4

2 1

[alsa-devel] [PATCH v3 0/4] i915 to call hda driver on HDMI plug/unplug
by David Henningsson 27 Jul '15

27 Jul '15

Changes since v2 is that the patch set has diminished to not transfer connector/eld information, since it might be better that such information to be transferred by a separate call in the ordinary direction. That could be added in a later patch set. The audio_ptr is now a void* (instead of a hdac_bus*) so that the audio side can easily refactor the ptr to point to the codec instead of the bus if we would find that beneficial. David Henningsson (4): drm/i915: Add audio hotplug info callback drm/i915: Call audio hotplug notify function ALSA: hda - Dispatch incoming HDMI hotplug i915 callback ALSA: hda - Wake the codec up on hotplug notify events drivers/gpu/drm/i915/i915_drv.h | 1 + drivers/gpu/drm/i915/intel_audio.c | 23 ++++++++++++++++++++--- include/drm/i915_component.h | 5 +++++ include/sound/hdaudio.h | 4 ++++ sound/hda/hdac_i915.c | 23 +++++++++++++++++++++++ sound/pci/hda/patch_hdmi.c | 13 ++++++++++++- 6 files changed, 65 insertions(+), 4 deletions(-) -- 1.7.9.5

4 7

Re: [alsa-devel] [ALC668]: Asus N751JK - surround 21 and surround40 pulseaudio bug
by Raymond Yau 26 Jul '15

26 Jul '15

> > Alsa-info patched kernel: http://www.alsa-project.org/db/?f=cd23a3f0bb72e38d73224620e3e8ab57a73aa441 > > Chmap results: > > Type = FIXED, Channels = 2 > FL FR > Type = FIXED, Channels = 4 > FL FR NA LFE > > When unplugging the subwoofer, then chmap gives this and stays like this. > > Cannot open PCM stream hw:1 for PLAYBACK > > [Pavucontrol] -> [Configuration] -> [Built-in Audio] gives > > Analog Stereo Duplex > Analog Stereo Output > Analog Surround 2.1 Output + Analog Stereo Input > Analog Surround 2.1 Output > Analog Surround 4.0 Output + Analog Stereo Input > Analog Surround 4.0 Output > Analog Stereo Input > Off Seem pulseaudio bug since your laptop does not have surround40 chmap It is not clear how pulseaudio's 21 channel-map match with alsa's 2.1 chmap http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mix… [Mapping analog-surround-21] device-strings = surround21:%f channel-map = front-left,front-right,lfe paths-output = analog-output analog-output-lineout analog-output-speaker priority = 8 direction = output [Mapping analog-surround-40] device-strings = surround40:%f channel-map = front-left,front-right,rear-left,rear-right paths-output = analog-output analog-output-lineout analog-output-speaker priority = 7 direction = output > > Pulseaudio verbose log while plugging in the subwoofer: http://pastebin.com/Ya5u1qz6 https://bugs.freedesktop.org/enter_bug.cgi?product=PulseAudio

1 0

[alsa-devel] [PATCH v3] ALSA: hda - Fix the headset mic that will not work on Dell desktop machine
by woodrow.shen＠canonical.com 25 Jul '15

25 Jul '15

From: Woodrow Shen <woodrow.shen(a)canonical.com> When the headset was plugged in the Dell desktop, the mic of headset can't be detected and workable. According to the alsa-info, we found the differece between alsa and init_pin_configs on the machine, so we need to add the pin configs to make headset work. Codec: Realtek ALC3234 Vendor Id: 0x10ec0255 Subsystem Id: 0x102806bb BugLink: https://bugs.launchpad.net/bugs/1477900 Signed-off-by: Woodrow Shen <woodrow.shen(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 742fc62..d94c0e3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5473,6 +5473,28 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1e, 0x411111f0}, {0x21, 0x0221103f}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x40000000}, + {0x14, 0x90170150}, + {0x17, 0x411111f0}, + {0x18, 0x411111f0}, + {0x19, 0x411111f0}, + {0x1a, 0x411111f0}, + {0x1b, 0x02011020}, + {0x1d, 0x4054c029}, + {0x1e, 0x411111f0}, + {0x21, 0x0221105f}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x40000000}, + {0x14, 0x90170110}, + {0x17, 0x411111f0}, + {0x18, 0x411111f0}, + {0x19, 0x411111f0}, + {0x1a, 0x411111f0}, + {0x1b, 0x01014020}, + {0x1d, 0x4054c029}, + {0x1e, 0x411111f0}, + {0x21, 0x0221101f}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60160}, {0x14, 0x90170120}, {0x17, 0x90170140}, -- 2.1.4

2 1

Re: [alsa-devel] ALSA: hda - Use generic parser for Cirrus codec driver
by Dan Carpenter 25 Jul '15

25 Jul '15

Hello Takashi Iwai, The patch 1077a024812d: "ALSA: hda - Use generic parser for Cirrus codec driver" from Dec 19, 2012, leads to the following static checker warning: sound/pci/hda/patch_cirrus.c:1004 cs4210_spdif_automute() warn: we tested 'spdif_present' before and it was 'true' sound/pci/hda/patch_cirrus.c 984 static void cs4210_spdif_automute(struct hda_codec *codec, 985 struct hda_jack_callback *tbl) 986 { 987 struct cs_spec *spec = codec->spec; 988 bool spdif_present = false; 989 hda_nid_t spdif_pin = spec->gen.autocfg.dig_out_pins[0]; 990 991 /* detect on spdif is specific to CS4210 */ 992 if (!spec->spdif_detect || 993 spec->vendor_nid != CS4210_VENDOR_NID) 994 return; 995 996 spdif_present = snd_hda_jack_detect(codec, spdif_pin); 997 if (spdif_present == spec->spdif_present) 998 return; 999 1000 spec->spdif_present = spdif_present; 1001 /* SPDIF TX on/off */ 1002 if (spdif_present) ^^^^^^^^^^^^^ 1003 snd_hda_set_pin_ctl(codec, spdif_pin, 1004 spdif_present ? PIN_OUT : 0); ^^^^^^^^^^^^^ I'm not positive but I think this is buggy. We should remove the if statement. 1005 1006 cs_automute(codec); 1007 } regards, dan carpenter

2 3

[alsa-devel] [PATCH] ASoC: rt5677: Return error if devm_gpiod_get_optional return ERR_PTR
by Axel Lin 25 Jul '15

25 Jul '15

If devm_gpiod_get_optional() return ERR_PTR, it means something wrong so request gpio fails. We had better return error in such case. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/rt5677.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 8d70911..3a074b6 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -5133,14 +5133,14 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, if (IS_ERR(rt5677->pow_ldo2)) { ret = PTR_ERR(rt5677->pow_ldo2); dev_err(&i2c->dev, "Failed to request POW_LDO2: %d\n", ret); - rt5677->pow_ldo2 = 0; + return PTR_ERR(rt5677->pow_ldo2); } rt5677->reset_pin = devm_gpiod_get_optional(&i2c->dev, "realtek,reset", GPIOD_OUT_HIGH); if (IS_ERR(rt5677->reset_pin)) { ret = PTR_ERR(rt5677->reset_pin); dev_err(&i2c->dev, "Failed to request RESET: %d\n", ret); - rt5677->reset_pin = 0; + return PTR_ERR(rt5677->reset_pin); } if (rt5677->pow_ldo2 || rt5677->reset_pin) { -- 2.1.0

1 0

Re: [alsa-devel] writing an alsa driver
by Marco Bodega 24 Jul '15

24 Jul '15

Hi all, I own an Asus x205ta too and I really want to help to make this audio card works. I'm using linux next with Michele Curti's patch (as of 23 june) Here kernel .config: CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH=m CONFIG_SND_SOC_INTEL_BYTCR_RT5645_MACH=m # CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH is no CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH=m # CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH is not set # CONFIG_SND_SOC_INTEL_BROADWELL_MACH is not set CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH=m CONFIG_SND_SOC_INTEL_BYTCR_RT5645_MACH=m bo@X205TA:~$ uname -a Linux X205TA 4.2.0-rc2-next-20150717+ #3 SMP Wed Jul 22 15:31:59 CEST 2015 x86_64 x86_64 x86_64 GNU/Linux Output of lsmod: snd_soc_sst_bytcr_rt5645 16384 0 snd_intel_sst_acpi 16384 1 snd_intel_sst_core 73728 1 snd_intel_sst_acpi snd_soc_rt5645 139264 1 snd_soc_sst_mfld_platform 90112 3 snd_intel_sst_core snd_pcm 102400 4 snd_soc_rt5645,snd_soc_core,snd_soc_sst_mfld_platform,snd_soc_sst_bytcr_rt5645 bo@X205TA:~$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: baytrailcraudio [baytrailcraudio], device 0: Baytrail Audio (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 bo@@X205TA:~$ aplay -D hw:0 -c 2 -r 48000 -f S16_LE /dev/urandomPlaying raw data '/dev/urandom' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo aplay: set_params:1297: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 48000 PERIOD_TIME: 125000 PERIOD_SIZE: 6000 PERIOD_BYTES: 24000 PERIODS: 4 BUFFER_TIME: 500000 BUFFER_SIZE: 24000 BUFFER_BYTES: 96000 TICK_TIME: 0 bo@X205TA:~$ cat /proc/interrupts | grep sst 6: 0 0 0 0 IO-APIC 28-fasteoi intel_sst_driver bo@X205TA:~$ ll /lib/firmware/intel/fw* -rw-r--r-- 1 root root 647078 lug 22 17:56 /lib/firmware/intel/fw_sst_0f28.bin -rw-r--r-- 1 root root 265684 lug 22 17:56 /lib/firmware/intel/fw_sst_0f28.bin-48kHz_i2s_master -rw-r--r-- 1 root root 701614 lug 22 17:56 /lib/firmware/intel/fw_sst_22a8.bin I read that the firmware used by bytx205_rt5645 is intel/fw_sst_0f28.bin: in fact in sst_acpi i have: {"10EC5648", "X205", "bytx205_rt5645", NULL, "intel/fw_sst_0f28.bin", &byt_rvp_platform_data }, but it's bounded to sst1 as i see from disassembling acpi data: cat dsdt.dsl: [...] Device (SSP1) { Name (_ADR, Zero) // _ADR: Address Name (_HID, "SSPX0000" /* Intel SSP Device */) // _HID: Hardware ID Name (_CID, "SSPX0000" /* Intel SSP Device */) // _CID: Compatible ID [...] Interrupt (ResourceConsumer, Level, ActiveLow, Exclusive, ,, ) { 0x0000001D, } Interrupt (ResourceConsumer, Level, ActiveLow, Exclusive, ,, ) { 0x00000018, } Interrupt (ResourceConsumer, Level, ActiveLow, Exclusive, ,, ) { 0x00000019, } Interrupt (ResourceConsumer, Level, ActiveLow, Exclusive, ,, ) { 0x0000001A, } Interrupt (ResourceConsumer, Level, ActiveLow, Exclusive, ,, ) { 0x0000001B, } Interrupt (ResourceConsumer, Level, ActiveLow, Exclusive, ,, ) { 0x0000001C, } [...] however the fw is tied to work with SSP0. What can i do? I tried to connect parts in routing: #DSP routing amixer -c0 sset 'codec_out0 mix 0 pcm0_in' on amixer -c0 sset 'media0_out mix 0 media1_in' on amixer -c0 sset 'media1_in Gain 0' 80% amixer -c0 sset 'media1_in Gain 0 Ramp Delay' 50 amixer -c0 sset 'media1_in Gain 0' off amixer -c0 sset 'pcm0_in Gain 0' 80% amixer -c0 sset 'pcm0_in Gain 0 Ramp Delay' 50 amixer -c0 sset 'pcm0_in Gain 0' off amixer -c0 sset 'codec_out0 Gain 0' 80% amixer -c0 sset 'codec_out0 Gain 0 Ramp Delay' 50 amixer -c0 sset 'codec_out0 Gain 0' off #Codec routin, if you have different one please modfiy amixer -c0 sset "DAC MIXL INF1" on amixer -c0 sset "DAC MIXR INF1" on amixer -c0 sset "Stereo DAC MIXL DAC L1" on amixer -c0 sset "Stereo DAC MIXR DAC R1" on amixer -c0 sset "SPK MIXL DAC L1" on amixer -c0 sset "SPK MIXR DAC R1" on amixer -c0 sset "SPOL MIX SPKVOL L" on amixer -c0 sset "SPOR MIX SPKVOL R" on amixer -c0 sset "SPOL MIX DAC L1" on amixer -c0 sset "SPOR MIX DAC R1" on amixer -c0 sset "Speaker L" on amixer -c0 sset "Speaker R" on amixer -c0 sset "Headphone" off amixer -c0 sset "Ext Spk" on and the attached dmesg log appeared I get no sound but at least no heat :) Is everything that I've done correct? I see in these output logs many of the errors and bugs that the other users found. Thanks for the time spent Marco Output of dmesg:

2 1

[alsa-devel] [GIT PULL] ASoC fixes for v4.2
by Mark Brown 24 Jul '15

24 Jul '15

The following changes since commit 52721d9d3334c1cb1f76219a161084094ec634dc: Linux 4.2-rc3 (2015-07-19 14:45:02 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git asoc-fix-v4.2-rc3 for you to fetch changes up to 996034b117b467709dec7811ef134063934fa626: Merge remote-tracking branches 'asoc/fix/sgtl5000', 'asoc/fix/topology' and 'asoc/fix/zx' into asoc-linus (2015-07-24 16:18:22 +0100) ---------------------------------------------------------------- ASoC: Fixes for v4.2 A lot of small fixes here, a few to the core: - Fix for binding DAPM stream widgets on devices with prefixes assigned to them - Minor fixes for the newly added topology interfaces - Locking and memory leak fixes for DAPM - Driver specific fixes ---------------------------------------------------------------- Axel Lin (2): ASoC: sgtl5000: Fix up define for SGTL5000_SMALL_POP ASoC: pcm1681: Fix setting de-emphasis sampling rate selection Ben Zhang (1): ASoC: ssm4567: Keep TDM_BCLKS in ssm4567_set_dai_fmt Charles Keepax (1): ASoC: dapm: Fix deadlock on auto-disable mux controls Fang, Yang A (1): ASoC: Intel: fix incorrect widget name Jeeja KP (1): ASoC: topology: Fix to add dapm mixer info Juergen Borleis (1): ASoC: fsl-ssi: Fix bitclock calculation for master mode Koro Chen (2): ASoC: mediatek: Fix unbalanced calls to runtime suspend/resume ASoC: mediatek: Use platform_of_node for machine drivers Krzysztof Kozlowski (2): ASoC: zx: i2s: Fix devm_ioremap_resource return value check ASoC: zx: spdif: Fix devm_ioremap_resource return value check Lars-Peter Clausen (4): ASoC: dapm: Lock during userspace access ASoC: dapm: Fix kcontrol widget name memory management ASoC: Free card DAPM context on snd_soc_instantiate_card() error path ASoC: dapm: Don't add prefix to widget stream name Liam Girdwood (1): ASoC: topology: Fix TLV size calculation. Mark Brown (6): Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/pcm1681' into asoc-linus Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branch 'asoc/fix/ssm4567' into asoc-linus Merge remote-tracking branches 'asoc/fix/fsl-ssi', 'asoc/fix/intel', 'asoc/fix/intel-kconfig' and 'asoc/fix/mediatek' into asoc-linus Merge remote-tracking branches 'asoc/fix/sgtl5000', 'asoc/fix/topology' and 'asoc/fix/zx' into asoc-linus Nicolas Boichat (1): ASoC: rt5645: Check if codec is initialized in workqueue handler Shilpa Sreeramalu (1): ASoC: Intel: Get correct usage_count value to load firmware Vinod Koul (2): ASoC: topology: fix typos in topology header ASoC: Intel: use CONFIG_SND_SOC for intel boards .../devicetree/bindings/sound/mt8173-max98090.txt | 2 ++ .../bindings/sound/mt8173-rt5650-rt5676.txt | 2 ++ include/uapi/sound/asoc.h | 4 +-- sound/soc/codecs/pcm1681.c | 2 +- sound/soc/codecs/rt5645.c | 3 ++ sound/soc/codecs/sgtl5000.h | 2 +- sound/soc/codecs/ssm4567.c | 8 ++++- sound/soc/fsl/fsl_ssi.c | 2 +- sound/soc/intel/Makefile | 2 +- sound/soc/intel/atom/sst/sst_drv_interface.c | 14 ++++----- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 4 +-- sound/soc/mediatek/mt8173-max98090.c | 17 ++++++++--- sound/soc/mediatek/mt8173-rt5650-rt5676.c | 19 +++++++++--- sound/soc/mediatek/mtk-afe-pcm.c | 2 ++ sound/soc/soc-core.c | 1 + sound/soc/soc-dapm.c | 35 +++++++++++----------- sound/soc/soc-topology.c | 23 +++++++------- sound/soc/zte/zx296702-i2s.c | 4 +-- sound/soc/zte/zx296702-spdif.c | 4 +-- 19 files changed, 93 insertions(+), 57 deletions(-)

2 1

[alsa-devel] [PATCH RFT] ASoC: cs4265: Fix setting dai format for Left/Right Justified
by Axel Lin 24 Jul '15

24 Jul '15

The settings in current code does not match the datasheet, fix it. DAC Control - Address 03h DAC Digital Interface Format (Bits 5:4) DAC_DIF1 DAC_DIF0 Description 0 0 Left Justified, up to 24-bit data (default) 0 1 I²S, up to 24-bit data 1 0 Right-Justified, 16-bit Data 1 1 Right-Justified, 24-bit Data Transmitter Control 2 - Address 12h Transmitter Digital Interface Format (Bits 7:6) Tx_DIF1 Tx_DIF0 Description Format Figure 0 0 Left Justified, up to 24-bit data (default) 0 1 I²S, up to 24-bit data 1 0 Right-Justified, 16-bit Data 1 1 Right-Justified, 24-bit Data Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi, I don't have the h/w to test, so please test if this patch works. Thanks, Axel sound/soc/codecs/cs4265.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index d1a77c7..e42a5038 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -457,14 +457,14 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream, case SND_SOC_DAIFMT_RIGHT_J: if (params_width(params) == 16) { snd_soc_update_bits(codec, CS4265_DAC_CTL, - CS4265_DAC_CTL_DIF, (1 << 5)); + CS4265_DAC_CTL_DIF, (2 << 4)); snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, - CS4265_SPDIF_CTL2_DIF, (1 << 7)); + CS4265_SPDIF_CTL2_DIF, (2 << 6)); } else { snd_soc_update_bits(codec, CS4265_DAC_CTL, - CS4265_DAC_CTL_DIF, (3 << 5)); + CS4265_DAC_CTL_DIF, (3 << 4)); snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, - CS4265_SPDIF_CTL2_DIF, (1 << 7)); + CS4265_SPDIF_CTL2_DIF, (3 << 6)); } break; case SND_SOC_DAIFMT_LEFT_J: @@ -473,7 +473,7 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, CS4265_ADC_CTL, CS4265_ADC_DIF, 0); snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, - CS4265_SPDIF_CTL2_DIF, (1 << 6)); + CS4265_SPDIF_CTL2_DIF, 0); break; default: -- 2.1.0

2 1

[alsa-devel] [PATCH] ASoC: cs4265: CS4265_INT_STATUS is readable register
by Axel Lin 24 Jul '15

24 Jul '15

CS4265_INT_STATUS was missed in cs4265_readable_register, fix it. The register 0x01 ~ 0x12 are readable, use CS4265_CHIP_ID ... CS4265_SPDIF_CTL2 syntax for better readability. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs4265.c | 18 +----------------- 1 file changed, 1 insertion(+), 17 deletions(-) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index e42a5038..55db19d 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -60,23 +60,7 @@ static const struct reg_default cs4265_reg_defaults[] = { static bool cs4265_readable_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS4265_PWRCTL: - case CS4265_DAC_CTL: - case CS4265_ADC_CTL: - case CS4265_MCLK_FREQ: - case CS4265_SIG_SEL: - case CS4265_CHB_PGA_CTL: - case CS4265_CHA_PGA_CTL: - case CS4265_ADC_CTL2: - case CS4265_DAC_CHA_VOL: - case CS4265_DAC_CHB_VOL: - case CS4265_DAC_CTL2: - case CS4265_SPDIF_CTL1: - case CS4265_SPDIF_CTL2: - case CS4265_INT_MASK: - case CS4265_STATUS_MODE_MSB: - case CS4265_STATUS_MODE_LSB: - case CS4265_CHIP_ID: + case CS4265_CHIP_ID ... CS4265_SPDIF_CTL2: return true; default: return false; -- 2.1.0

2 1

[alsa-devel] [PATCH] ASoC: Use params_width() at appropriate places
by Axel Lin 24 Jul '15

24 Jul '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/max98925.c | 2 +- sound/soc/codecs/pcm512x.c | 4 ++-- sound/soc/codecs/wm2200.c | 2 +- sound/soc/codecs/wm5100.c | 2 +- sound/soc/codecs/wm8776.c | 4 ++-- sound/soc/codecs/wm8996.c | 2 +- 6 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c index ebb648a..5990de3 100644 --- a/sound/soc/codecs/max98925.c +++ b/sound/soc/codecs/max98925.c @@ -430,7 +430,7 @@ static int max98925_dai_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); - switch (snd_pcm_format_width(params_format(params))) { + switch (params_width(params)) { case 16: regmap_update_bits(max98925->regmap, MAX98925_FORMAT, diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index de16429..047c489 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -1117,7 +1117,7 @@ static int pcm512x_hw_params(struct snd_pcm_substream *substream, params_rate(params), params_channels(params)); - switch (snd_pcm_format_width(params_format(params))) { + switch (params_width(params)) { case 16: alen = PCM512x_ALEN_16; break; @@ -1132,7 +1132,7 @@ static int pcm512x_hw_params(struct snd_pcm_substream *substream, break; default: dev_err(codec->dev, "Bad frame size: %d\n", - snd_pcm_format_width(params_format(params))); + params_width(params)); return -EINVAL; } diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 35199fc..fd1439e 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1702,7 +1702,7 @@ static int wm2200_hw_params(struct snd_pcm_substream *substream, int *bclk_rates; /* Data sizes if not using TDM */ - wl = snd_pcm_format_width(params_format(params)); + wl = params_width(params); if (wl < 0) return wl; fl = snd_soc_params_to_frame_size(params); diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 8120b0c..c2cdcae 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1408,7 +1408,7 @@ static int wm5100_hw_params(struct snd_pcm_substream *substream, base = dai->driver->base; /* Data sizes if not using TDM */ - wl = snd_pcm_format_width(params_format(params)); + wl = params_width(params); if (wl < 0) return wl; fl = snd_soc_params_to_frame_size(params); diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index 592866d..183c9a4 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -265,7 +265,7 @@ static int wm8776_hw_params(struct snd_pcm_substream *substream, } /* Set word length */ - switch (snd_pcm_format_width(params_format(params))) { + switch (params_width(params)) { case 16: iface = 0; break; @@ -280,7 +280,7 @@ static int wm8776_hw_params(struct snd_pcm_substream *substream, break; default: dev_err(codec->dev, "Unsupported sample size: %i\n", - snd_pcm_format_width(params_format(params))); + params_width(params)); return -EINVAL; } diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 2a696db..f7ccd9f 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1780,7 +1780,7 @@ static int wm8996_hw_params(struct snd_pcm_substream *substream, wm8996->rx_rate[dai->id] = params_rate(params); /* Needs looking at for TDM */ - bits = snd_pcm_format_width(params_format(params)); + bits = params_width(params); if (bits < 0) return bits; aifdata |= (bits << WM8996_AIF1TX_WL_SHIFT) | bits; -- 2.1.0

2 1

[alsa-devel] [PATCH v2] ALSA: hda - Fix the headset mic that will not work on Dell desktop machine
by woodrow.shen＠canonical.com 24 Jul '15

24 Jul '15

When the headset was plugged in the Dell desktop, the mic of headset can't be detected and workable. According to the alsa-info, we found the differece between alsa and init_pin_configs on the machine, so we need to add the pin configs to make headset work. Codec: Realtek ALC3234 Vendor Id: 0x10ec0255 Subsystem Id: 0x102806bb BugLink: https://bugs.launchpad.net/bugs/1477900 Signed-off-by: Woodrow Shen <woodrow.shen(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 742fc62..d94c0e3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5473,6 +5473,28 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1e, 0x411111f0}, {0x21, 0x0221103f}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x40000000}, + {0x14, 0x90170150}, + {0x17, 0x411111f0}, + {0x18, 0x411111f0}, + {0x19, 0x411111f0}, + {0x1a, 0x411111f0}, + {0x1b, 0x02011020}, + {0x1d, 0x4054c029}, + {0x1e, 0x411111f0}, + {0x21, 0x0221105f}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x40000000}, + {0x14, 0x90170110}, + {0x17, 0x411111f0}, + {0x18, 0x411111f0}, + {0x19, 0x411111f0}, + {0x1a, 0x411111f0}, + {0x1b, 0x01014020}, + {0x1d, 0x4054c029}, + {0x1e, 0x411111f0}, + {0x21, 0x0221101f}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60160}, {0x14, 0x90170120}, {0x17, 0x90170140}, -- 2.1.4

2 1

[alsa-devel] [PATCH 1/2] ASoC: pcm1681: Fix setting de-emphasis sampling rate selection
by Axel Lin 24 Jul '15

24 Jul '15

The de-emphasis sampling rate selection is controlled by BIT[3:4] of PCM1681_DEEMPH_CONTROL register. Do proper left shift to set it. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/pcm1681.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index b2c990f..1011142 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -102,7 +102,7 @@ static int pcm1681_set_deemph(struct snd_soc_codec *codec) if (val != -1) { regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL, - PCM1681_DEEMPH_RATE_MASK, val); + PCM1681_DEEMPH_RATE_MASK, val << 3); enable = 1; } else enable = 0; -- 2.1.0

3 9

[alsa-devel] [PATCH] ASoC: sta529: Remove redundant I2C_FUNC_SMBUS_BYTE_DATA functionality check
by Axel Lin 24 Jul '15

24 Jul '15

This checking is done by regmap_get_i2c_bus() which is called in devm_regmap_init_i2c(). Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/sta529.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index 3430f44..2cdaca9 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -339,9 +339,6 @@ static int sta529_i2c_probe(struct i2c_client *i2c, struct sta529 *sta529; int ret; - if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA)) - return -EINVAL; - sta529 = devm_kzalloc(&i2c->dev, sizeof(struct sta529), GFP_KERNEL); if (!sta529) return -ENOMEM; -- 2.1.0

2 1

[alsa-devel] [PATCH] ASoC: ssm4567: Added ACPI entry for SSM4567 codec
by Harsha Priya 24 Jul '15

24 Jul '15

Added INT343B ACPI ID for the SSM4567 codec Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: M Naveen <naveen.m(a)intel.com> Signed-off-by: M R Sathya Prakash <sathya.prakash.m.r(a)intel.com> --- sound/soc/codecs/ssm4567.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c index 938d2cb..af536ae 100644 --- a/sound/soc/codecs/ssm4567.c +++ b/sound/soc/codecs/ssm4567.c @@ -10,6 +10,7 @@ * Licensed under the GPL-2. */ +#include <linux/acpi.h> #include <linux/module.h> #include <linux/init.h> #include <linux/i2c.h> @@ -450,10 +451,21 @@ static const struct i2c_device_id ssm4567_i2c_ids[] = { }; MODULE_DEVICE_TABLE(i2c, ssm4567_i2c_ids); +#ifdef CONFIG_ACPI + +static const struct acpi_device_id ssm4567_acpi_match[] = { + { "INT343B", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, ssm4567_acpi_match); + +#endif + static struct i2c_driver ssm4567_driver = { .driver = { .name = "ssm4567", .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(ssm4567_acpi_match), }, .probe = ssm4567_i2c_probe, .remove = ssm4567_i2c_remove, -- 1.9.1

4 3

[alsa-devel] [PATCH v1] ALSA: hda- Add headset mic pin quirk for a Dell device
by woodrow.shen＠canonical.com 24 Jul '15

24 Jul '15

From: Woodrow Shen <woodrow.shen(a)canonical.com> Fix the headset mic that will not work on Dell desktop machine. BugLink: https://bugs.launchpad.net/bugs/1475553 Signed-off-by: Woodrow Shen <woodrow.shen(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 742fc62..d94c0e3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5473,6 +5473,28 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1e, 0x411111f0}, {0x21, 0x0221103f}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x40000000}, + {0x14, 0x90170150}, + {0x17, 0x411111f0}, + {0x18, 0x411111f0}, + {0x19, 0x411111f0}, + {0x1a, 0x411111f0}, + {0x1b, 0x02011020}, + {0x1d, 0x4054c029}, + {0x1e, 0x411111f0}, + {0x21, 0x0221105f}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x40000000}, + {0x14, 0x90170110}, + {0x17, 0x411111f0}, + {0x18, 0x411111f0}, + {0x19, 0x411111f0}, + {0x1a, 0x411111f0}, + {0x1b, 0x01014020}, + {0x1d, 0x4054c029}, + {0x1e, 0x411111f0}, + {0x21, 0x0221101f}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60160}, {0x14, 0x90170120}, {0x17, 0x90170140}, -- 2.1.4

4 6

[alsa-devel] [ [PATCH] ALSA: hda- Add headset mic pin quirk for a Dell device]
by woodrow.shen＠canonical.com 24 Jul '15

24 Jul '15

From: woodrowshen <woodrow.shen(a)canonical.com> BugLink: https://bugs.launchpad.net/bugs/1475553 rigned-off-by: woodrowshen <woodrow.shen(a)canonical.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 742fc62..d94c0e3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5473,6 +5473,28 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1e, 0x411111f0}, {0x21, 0x0221103f}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x40000000}, + {0x14, 0x90170150}, + {0x17, 0x411111f0}, + {0x18, 0x411111f0}, + {0x19, 0x411111f0}, + {0x1a, 0x411111f0}, + {0x1b, 0x02011020}, + {0x1d, 0x4054c029}, + {0x1e, 0x411111f0}, + {0x21, 0x0221105f}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x40000000}, + {0x14, 0x90170110}, + {0x17, 0x411111f0}, + {0x18, 0x411111f0}, + {0x19, 0x411111f0}, + {0x1a, 0x411111f0}, + {0x1b, 0x01014020}, + {0x1d, 0x4054c029}, + {0x1e, 0x411111f0}, + {0x21, 0x0221101f}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60160}, {0x14, 0x90170120}, {0x17, 0x90170140}, -- 2.1.4

1 0

[alsa-devel] [PATCH v2 00/19] Update ALSA, and au0828 drivers to use Managed Media Controller API
by Shuah Khan 23 Jul '15

23 Jul '15

This patch series updates ALSA driver, and au0828 core driver to use Managed Media controller API to share tuner. Please note that Managed Media Controller API and DVB and V4L2 drivers updates to use Media Controller API have been added in a prior patch series. Media Controller API is enhanced with two new interfaces to register and unregister entity_notify hooks to allow drivers to take appropriate actions when as new entities get added to the shared media device. Tested exclusion between digital, analog, and audio to ensure when tuner has an active link to DVB FE, analog, and audio will detect and honor the tuner busy conditions and vice versa. Changes since v1: Link to v1: http://www.spinics.net/lists/linux-media/msg91697.html 1. Fixed Open Issue: ALSA has makes media_entity_pipeline_start() call in irq path. I am seeing warnings that the graph_mutex is unsafe irq lock. Media Controller API updates to start/stop pipeline to be irq safe might be necessary. Maybe there are other MC interfaces that need to be irq safe, but I haven't seen any problems with my testing. graph_mutex is changed to a spinlock. Changed drivers that directly hold the lock for graph walks. media_entity_setup_link() might need to be made IRQ safe. I am running more tests to ensure there is no lock warns when dvb, video, and audio apps. run at the sametime and use MC API (that holds graph_lock) to check for tuner availability. Initial testing looked good so far. 2. Add enable_source handler field to struct media_device Add a new field to enable handler to find source entity for the sink entity and check if it is available, and activate the link using media_entity_setup_link() interface. Bridge driver is expected to implement and set the handler when media_device is registered or when bridge driver finds the media_device during probe. This is to enable the use-case to find tuner entity connected to the decoder entity and check if it is available, and activate the using media_entity_setup_link() if it is available. This hanlder can be invoked from media core (v4l-core, dvb-core) as well as other drivers such as ALSA that control the media device. 3. Changes to v4l2-core, ALSA to use enable_source handler 4. Changes to au0828 bridge driver to implement enable_source handler Note: This series includes a revert to a patch that added media controller entity framework enhancements that implemented entity ops for entity_notify functionality. Entity ops for entity_notify doesn't handle and cover entity create ordering variations that could occur during boot. entity_notify list has been moved to media device level which makes the entity_notify calls to work correctly. History: This patch series has been updated to address comments from 3 previous versions of this series. Links to v3 version for reference are: https://www.mail-archive.com/linux-media%40vger.kernel.org/msg89491.html https://www.mail-archive.com/linux-media@vger.kernel.org/msg89492.html https://www.mail-archive.com/linux-media%40vger.kernel.org/msg89493.html Shuah Khan (19): Revert "[media] media: media controller entity framework enhancements for ALSA" media: Media Controller register/unregister entity_notify API media: Add ALSA Media Controller devnodes media: au8522 change to create MC pad for ALSA Audio Out media: Convert graph_mutex to a spinlock and call it graph_lock media: platform exynos4-is: Update graph_mutex to graph_lock spinlock media: platform omap3isp: Update graph_mutex to graph_lock spinlock media: platform s3c-camif: Update graph_mutex to graph_lock spinlock media: platform vsp1: Update graph_mutex to graph_lock spinlock media: platform xilinx: Update graph_mutex to graph_lock spinlock staging media: davinci_vpfe: Update graph_mutex to graph_lock spinlock staging media: omap4iss: Update graph_mutex to graph_lock spinlock media: Add irq safe Media Controller start/stop pipeline API media: Add enable_source handler field to struct media_device media: v4l-core add v4l_enable_media_tuner() to check for tuner availability media: Change v4l-core to check for tuner availability media: dvb-frontend change to check for tuner availability from open media: au0828 change to use Managed Media Controller API sound/usb: Update ALSA driver to use Managed Media Controller API drivers/media/dvb-core/dvb_frontend.c | 43 ++-- drivers/media/dvb-frontends/au8522.h | 8 + drivers/media/dvb-frontends/au8522_decoder.c | 1 + drivers/media/dvb-frontends/au8522_priv.h | 8 - drivers/media/media-device.c | 60 +++++- drivers/media/media-entity.c | 88 ++++++-- drivers/media/platform/exynos4-is/fimc-isp-video.c | 8 +- drivers/media/platform/exynos4-is/fimc-lite.c | 8 +- drivers/media/platform/exynos4-is/media-dev.c | 14 +- drivers/media/platform/exynos4-is/media-dev.h | 4 +- drivers/media/platform/omap3isp/isp.c | 4 +- drivers/media/platform/omap3isp/ispvideo.c | 4 +- drivers/media/platform/s3c-camif/camif-core.c | 6 +- drivers/media/platform/vsp1/vsp1_video.c | 4 +- drivers/media/platform/xilinx/xilinx-dma.c | 4 +- drivers/media/usb/au0828/au0828-core.c | 184 ++++++++++++----- drivers/media/usb/au0828/au0828-video.c | 72 ++----- drivers/media/usb/au0828/au0828.h | 5 + drivers/media/v4l2-core/v4l2-dev.c | 17 ++ drivers/media/v4l2-core/v4l2-ioctl.c | 29 +++ drivers/media/v4l2-core/videobuf2-core.c | 5 + drivers/staging/media/davinci_vpfe/vpfe_video.c | 12 +- drivers/staging/media/omap4iss/iss.c | 4 +- drivers/staging/media/omap4iss/iss_video.c | 4 +- include/media/media-device.h | 41 +++- include/media/media-entity.h | 7 +- include/media/v4l2-dev.h | 3 + include/uapi/linux/media.h | 5 + sound/usb/Makefile | 15 +- sound/usb/card.c | 5 + sound/usb/card.h | 1 + sound/usb/media.c | 227 +++++++++++++++++++++ sound/usb/media.h | 52 +++++ sound/usb/pcm.c | 15 +- sound/usb/quirks-table.h | 1 + sound/usb/quirks.c | 9 +- sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 1 + 38 files changed, 765 insertions(+), 215 deletions(-) create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.1.4

1 20

Re: [alsa-devel] linux-next: Tree for Jul 23 (sound/soc/codecs/sti-sas.c)
by Randy Dunlap 23 Jul '15

23 Jul '15

On 07/22/15 23:57, Stephen Rothwell wrote: > Hi all, > > Changes since 20150722: > on x86_64: ../sound/soc/codecs/sti-sas.c: In function 'stih416_dac_probe': ../sound/soc/codecs/sti-sas.c:227:2: error: implicit declaration of function 'devm_reset_control_get' [-Werror=implicit-function-declaration] dac->rst = devm_reset_control_get(codec->dev, "dac_rst"); ^ ../sound/soc/codecs/sti-sas.c:227:11: warning: assignment makes pointer from integer without a cast [enabled by default] dac->rst = devm_reset_control_get(codec->dev, "dac_rst"); ^ Full randconfig file is attached. -- ~Randy

1 0

[alsa-devel] [PATCH] ASoC: rl6231: add pll preset table
by Bard Liao 23 Jul '15

23 Jul '15

Currently, rl6231_pll_calc provide a working PLL parameters for given freq_in and freq_out. However, in some cases it is not the perfect parameter. For example if freq_in = 19200000 and freq_out = 24576000, the calculated parameter will gengrate 24.5647 MHz which is not exactly the same as what we need. But the PLL can output 24.576 MHz as exactly what we expect if we set the best PLL parameter. To improve it, we put the best match parameters in a preset table. We can search the preset table first, if there is no preset parameter for the given freq_in and freq_out, we can still calculate a working PLL parameter. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rl6231.c | 27 ++++++++++++++++++++++++++- 1 file changed, 26 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c index 56650d6..96f3e90 100644 --- a/sound/soc/codecs/rl6231.c +++ b/sound/soc/codecs/rl6231.c @@ -43,6 +43,19 @@ int rl6231_calc_dmic_clk(int rate) } EXPORT_SYMBOL_GPL(rl6231_calc_dmic_clk); +struct pll_calc_map { + unsigned int pll_in; + unsigned int pll_out; + int k; + int n; + int m; + bool m_bp; +}; + +static const struct pll_calc_map pll_preset_table[] = { + {19200000, 24576000, 3, 30, 3, false}, +}; + /** * rl6231_pll_calc - Calcualte PLL M/N/K code. * @freq_in: external clock provided to codec. @@ -57,7 +70,7 @@ int rl6231_pll_calc(const unsigned int freq_in, const unsigned int freq_out, struct rl6231_pll_code *pll_code) { int max_n = RL6231_PLL_N_MAX, max_m = RL6231_PLL_M_MAX; - int k, red, n_t, pll_out, in_t, out_t; + int i, k, red, n_t, pll_out, in_t, out_t; int n = 0, m = 0, m_t = 0; int red_t = abs(freq_out - freq_in); bool bypass = false; @@ -65,6 +78,18 @@ int rl6231_pll_calc(const unsigned int freq_in, if (RL6231_PLL_INP_MAX < freq_in || RL6231_PLL_INP_MIN > freq_in) return -EINVAL; + for (i = 0; i < ARRAY_SIZE(pll_preset_table); i++) { + if (freq_in == pll_preset_table[i].pll_in && + freq_out == pll_preset_table[i].pll_out) { + k = pll_preset_table[i].k; + m = pll_preset_table[i].m; + n = pll_preset_table[i].n; + bypass = pll_preset_table[i].m_bp; + pr_debug("Use preset PLL parameter table\n"); + goto code_find; + } + } + k = 100000000 / freq_out - 2; if (k > RL6231_PLL_K_MAX) k = RL6231_PLL_K_MAX; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: xtensa: Remove unnecessary snd_pcm_lib_preallocate_free_for_all()
by Vaishali Thakkar 23 Jul '15

23 Jul '15

The ALSA core takes care that all preallocated memory is freed when the PCM itself is freed. There is no need to do this manually in the driver. Signed-off-by: Vaishali Thakkar <vthakkar1994(a)gmail.com> --- sound/soc/xtensa/xtfpga-i2s.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/xtensa/xtfpga-i2s.c b/sound/soc/xtensa/xtfpga-i2s.c index 1cfb19e..039f65e 100644 --- a/sound/soc/xtensa/xtfpga-i2s.c +++ b/sound/soc/xtensa/xtfpga-i2s.c @@ -474,11 +474,6 @@ static int xtfpga_pcm_new(struct snd_soc_pcm_runtime *rtd) card->dev, size, size); } -static void xtfpga_pcm_free(struct snd_pcm *pcm) -{ - snd_pcm_lib_preallocate_free_for_all(pcm); -} - static const struct snd_pcm_ops xtfpga_pcm_ops = { .open = xtfpga_pcm_open, .close = xtfpga_pcm_close, @@ -490,7 +485,6 @@ static const struct snd_pcm_ops xtfpga_pcm_ops = { static const struct snd_soc_platform_driver xtfpga_soc_platform = { .pcm_new = xtfpga_pcm_new, - .pcm_free = xtfpga_pcm_free, .ops = &xtfpga_pcm_ops, }; -- 1.9.1

2 1

[alsa-devel] [PATCHv2 0/3] regmap: Use different lockdep class for each regmap init call
by Nicolas Boichat 23 Jul '15

23 Jul '15

Second version. Small fixup in the regmap patch itself [3/3] (the declaration of __devm_regmap_init_ac97 was missing lock_key/name parameters). Also, there are 2 drivers that are using function pointers named "regmap_init", which causes problems as it is now a macro: put these in parentheses to prevent the preprocessor from substituting them. Patches 1+2 should be pushed first to avoid breaking git bissect. Compile tested with allyesconfig on x86_64 and arm. Nicolas Boichat (3): mfd: vexpress: Add parentheses around bridge->ops->regmap_init call thermal: sti: Add parentheses around bridge->ops->regmap_init call regmap: Use different lockdep class for each regmap init call drivers/base/regmap/regmap-ac97.c | 22 +++-- drivers/base/regmap/regmap-i2c.c | 22 +++-- drivers/base/regmap/regmap-mmio.c | 27 ++++-- drivers/base/regmap/regmap-spi.c | 22 +++-- drivers/base/regmap/regmap-spmi.c | 44 +++++---- drivers/base/regmap/regmap.c | 31 +++--- drivers/bus/vexpress-config.c | 2 +- drivers/thermal/st/st_thermal.c | 2 +- include/linux/regmap.h | 192 ++++++++++++++++++++++++++++---------- 9 files changed, 252 insertions(+), 112 deletions(-) -- 2.4.3.573.g4eafbef

3 6

[alsa-devel] [PATCH] ASoC: rt5677: Remove NULL test for desc before gpiod_set_value_cansleep call
by Axel Lin 23 Jul '15

23 Jul '15

It's safe to call gpiod_set_value_cansleep() with NULL desc. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/rt5677.c | 18 ++++++------------ 1 file changed, 6 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 016db7d..8d70911 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4763,10 +4763,8 @@ static int rt5677_remove(struct snd_soc_codec *codec) struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); regmap_write(rt5677->regmap, RT5677_RESET, 0x10ec); - if (rt5677->pow_ldo2) - gpiod_set_value_cansleep(rt5677->pow_ldo2, 0); - if (rt5677->reset_pin) - gpiod_set_value_cansleep(rt5677->reset_pin, 0); + gpiod_set_value_cansleep(rt5677->pow_ldo2, 0); + gpiod_set_value_cansleep(rt5677->reset_pin, 0); return 0; } @@ -4780,10 +4778,8 @@ static int rt5677_suspend(struct snd_soc_codec *codec) regcache_cache_only(rt5677->regmap, true); regcache_mark_dirty(rt5677->regmap); - if (rt5677->pow_ldo2) - gpiod_set_value_cansleep(rt5677->pow_ldo2, 0); - if (rt5677->reset_pin) - gpiod_set_value_cansleep(rt5677->reset_pin, 0); + gpiod_set_value_cansleep(rt5677->pow_ldo2, 0); + gpiod_set_value_cansleep(rt5677->reset_pin, 0); } return 0; @@ -4794,10 +4790,8 @@ static int rt5677_resume(struct snd_soc_codec *codec) struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); if (!rt5677->dsp_vad_en) { - if (rt5677->pow_ldo2) - gpiod_set_value_cansleep(rt5677->pow_ldo2, 1); - if (rt5677->reset_pin) - gpiod_set_value_cansleep(rt5677->reset_pin, 1); + gpiod_set_value_cansleep(rt5677->pow_ldo2, 1); + gpiod_set_value_cansleep(rt5677->reset_pin, 1); if (rt5677->pow_ldo2 || rt5677->reset_pin) msleep(10); -- 2.1.0

2 1

[alsa-devel] [PATCH] ASoC: tas2552: Fix off-by-one for max_register setting
by Axel Lin 23 Jul '15

23 Jul '15

The latest valid register is TAS2552_VBAT_DATA. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/tas2552.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2552.h b/sound/soc/codecs/tas2552.h index 5746f8f..e34752b 100644 --- a/sound/soc/codecs/tas2552.h +++ b/sound/soc/codecs/tas2552.h @@ -42,7 +42,7 @@ #define TAS2552_BOOST_APT_CTRL 0x14 #define TAS2552_VER_NUM 0x16 #define TAS2552_VBAT_DATA 0x19 -#define TAS2552_MAX_REG 0x20 +#define TAS2552_MAX_REG TAS2552_VBAT_DATA /* CFG1 Register Masks */ #define TAS2552_DEV_RESET (1 << 0) -- 2.1.0

3 2

[alsa-devel] [PATCH] ASoC: tas5086: Improve the logic for de-emphasis sampling rate selection
by Axel Lin 23 Jul '15

23 Jul '15

Slightly improve the logic for de-emphasis sampling rate selection by break out the loop if the rate is matched. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/tas5086.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 0a49fc8..d49d25d 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -266,10 +266,14 @@ static int tas5086_set_deemph(struct snd_soc_codec *codec) struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); int i, val = 0; - if (priv->deemph) - for (i = 0; i < ARRAY_SIZE(tas5086_deemph); i++) - if (tas5086_deemph[i] == priv->rate) + if (priv->deemph) { + for (i = 0; i < ARRAY_SIZE(tas5086_deemph); i++) { + if (tas5086_deemph[i] == priv->rate) { val = i; + break; + } + } + } return regmap_update_bits(priv->regmap, TAS5086_SYS_CONTROL_1, TAS5086_DEEMPH_MASK, val); -- 2.1.0

3 2

Re: [alsa-devel] Intel Baytrail Linux Audio
by Vinod Koul 23 Jul '15

23 Jul '15

On Thu, Jul 09, 2015 at 02:57:58PM -0400, Tyler Yeomans wrote: > Antonio/Vinod, > > I have made the change in the sst-acpi.c file for the atom specific boards. > I've attached the dmesg output here with dynamic debug enabled. > > To clarify- > > snd-intel-sst-core is loaded > FE-BE fix has been applied > I'm using the ssp0 firmware. > > Interrupts are still 0 > > Still no sound. > > Would leaving the device on and providing remote access (i.e though > teamviewer) assist in any way? The fw download seems to fail on your device, which can also be due to incorrect interrupt mapping... [ 20.386850] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 20.386862] intel_sst_acpi 80860F28:00: fw returned err -16 [ 20.386873] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp2-port [ 20.752559] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 20.752570] intel_sst_acpi 80860F28:00: fw returned err -16 [ 21.163378] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x0 fw_state 0x1 [ 21.163388] intel_sst_acpi 80860F28:00: fw download failed -16 Can you tell me which device you have, what was preinstalled on it -- ~Vinod

2 4

[alsa-devel] [PATCH] ASoC: mediatek: Add suspend/resume callbacks
by Koro Chen 23 Jul '15

23 Jul '15

This adds suspend/resume callbacks, which are common for each DAI. To be able to continue the last playback/capture after resume when suspend was done during a playback/capture, in the callbacks we do backup/restore of registers which were set before prepare stage. Registers to be backup/restore are defined in a backup list array. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- sound/soc/mediatek/mtk-afe-common.h | 8 ---- sound/soc/mediatek/mtk-afe-pcm.c | 77 +++++++++++++++++++++++++++++++++++++ 2 files changed, 77 insertions(+), 8 deletions(-) diff --git a/sound/soc/mediatek/mtk-afe-common.h b/sound/soc/mediatek/mtk-afe-common.h index a88b175..cc4393c 100644 --- a/sound/soc/mediatek/mtk-afe-common.h +++ b/sound/soc/mediatek/mtk-afe-common.h @@ -98,12 +98,4 @@ struct mtk_afe_memif { const struct mtk_afe_irq_data *irqdata; }; -struct mtk_afe { - /* address for ioremap audio hardware register */ - void __iomem *base_addr; - struct device *dev; - struct regmap *regmap; - struct mtk_afe_memif memif[MTK_AFE_MEMIF_NUM]; - struct clk *clocks[MTK_CLK_NUM]; -}; #endif diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c index de38d2e..55471ee 100644 --- a/sound/soc/mediatek/mtk-afe-pcm.c +++ b/sound/soc/mediatek/mtk-afe-pcm.c @@ -45,18 +45,21 @@ /* Memory interface */ #define AFE_DL1_BASE 0x0040 #define AFE_DL1_CUR 0x0044 +#define AFE_DL1_END 0x0048 #define AFE_DL2_BASE 0x0050 #define AFE_DL2_CUR 0x0054 #define AFE_AWB_BASE 0x0070 #define AFE_AWB_CUR 0x007c #define AFE_VUL_BASE 0x0080 #define AFE_VUL_CUR 0x008c +#define AFE_VUL_END 0x0088 #define AFE_DAI_BASE 0x0090 #define AFE_DAI_CUR 0x009c #define AFE_MOD_PCM_BASE 0x0330 #define AFE_MOD_PCM_CUR 0x033c #define AFE_HDMI_OUT_BASE 0x0374 #define AFE_HDMI_OUT_CUR 0x0378 +#define AFE_HDMI_OUT_END 0x037c #define AFE_ADDA2_TOP_CON0 0x0600 @@ -127,6 +130,34 @@ enum afe_tdm_ch_start { AFE_TDM_CH_ZERO, }; +static const unsigned int mtk_afe_backup_list[] = { + AUDIO_TOP_CON0, + AFE_CONN1, + AFE_CONN2, + AFE_CONN7, + AFE_CONN8, + AFE_DAC_CON1, + AFE_DL1_BASE, + AFE_DL1_END, + AFE_VUL_BASE, + AFE_VUL_END, + AFE_HDMI_OUT_BASE, + AFE_HDMI_OUT_END, + AFE_HDMI_CONN0, + AFE_DAC_CON0, +}; + +struct mtk_afe { + /* address for ioremap audio hardware register */ + void __iomem *base_addr; + struct device *dev; + struct regmap *regmap; + struct mtk_afe_memif memif[MTK_AFE_MEMIF_NUM]; + struct clk *clocks[MTK_CLK_NUM]; + unsigned int backup_regs[ARRAY_SIZE(mtk_afe_backup_list)]; + bool suspended; +}; + static const struct snd_pcm_hardware mtk_afe_hardware = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP_VALID), @@ -722,11 +753,53 @@ static const struct snd_soc_dai_ops mtk_afe_hdmi_ops = { }; +static int mtk_afe_runtime_suspend(struct device *dev); +static int mtk_afe_runtime_resume(struct device *dev); + +static int mtk_afe_dai_suspend(struct snd_soc_dai *dai) +{ + struct mtk_afe *afe = snd_soc_dai_get_drvdata(dai); + int i; + + dev_dbg(afe->dev, "%s\n", __func__); + if (pm_runtime_status_suspended(afe->dev) || afe->suspended) + return 0; + + for (i = 0; i < ARRAY_SIZE(mtk_afe_backup_list); i++) + regmap_read(afe->regmap, mtk_afe_backup_list[i], + &afe->backup_regs[i]); + + afe->suspended = true; + mtk_afe_runtime_suspend(afe->dev); + return 0; +} + +static int mtk_afe_dai_resume(struct snd_soc_dai *dai) +{ + struct mtk_afe *afe = snd_soc_dai_get_drvdata(dai); + int i = 0; + + dev_dbg(afe->dev, "%s\n", __func__); + if (pm_runtime_status_suspended(afe->dev) || !afe->suspended) + return 0; + + mtk_afe_runtime_resume(afe->dev); + + for (i = 0; i < ARRAY_SIZE(mtk_afe_backup_list); i++) + regmap_write(afe->regmap, mtk_afe_backup_list[i], + afe->backup_regs[i]); + + afe->suspended = false; + return 0; +} + static struct snd_soc_dai_driver mtk_afe_pcm_dais[] = { /* FE DAIs: memory intefaces to CPU */ { .name = "DL1", /* downlink 1 */ .id = MTK_AFE_MEMIF_DL1, + .suspend = mtk_afe_dai_suspend, + .resume = mtk_afe_dai_resume, .playback = { .stream_name = "DL1", .channels_min = 1, @@ -738,6 +811,8 @@ static struct snd_soc_dai_driver mtk_afe_pcm_dais[] = { }, { .name = "VUL", /* voice uplink */ .id = MTK_AFE_MEMIF_VUL, + .suspend = mtk_afe_dai_suspend, + .resume = mtk_afe_dai_resume, .capture = { .stream_name = "VUL", .channels_min = 1, @@ -774,6 +849,8 @@ static struct snd_soc_dai_driver mtk_afe_hdmi_dais[] = { { .name = "HDMI", .id = MTK_AFE_MEMIF_HDMI, + .suspend = mtk_afe_dai_suspend, + .resume = mtk_afe_dai_resume, .playback = { .stream_name = "HDMI", .channels_min = 2, -- 1.8.1.1.dirty

2 1

[alsa-devel] [PATCH 1/2] ASoC: cs35l32: Fix define for CS35L32_SDOUT_3ST
by Axel Lin 23 Jul '15

23 Jul '15

According to the datasheet, the CS35L32_SDOUT_3ST is BIT(3). Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs35l32.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l32.h b/sound/soc/codecs/cs35l32.h index 31ab804..1d6c250 100644 --- a/sound/soc/codecs/cs35l32.h +++ b/sound/soc/codecs/cs35l32.h @@ -80,7 +80,7 @@ struct cs35l32_platform_data { #define CS35L32_GAIN_MGR_MASK 0x08 #define CS35L32_ADSP_SHARE_MASK 0x08 #define CS35L32_ADSP_DATACFG_MASK 0x30 -#define CS35L32_SDOUT_3ST 0x80 +#define CS35L32_SDOUT_3ST 0x08 #define CS35L32_BATT_REC_MASK 0x0E #define CS35L32_BATT_THRESH_MASK 0x30 -- 2.1.0

3 4

[alsa-devel] [PATCH] ASoC: cs42l73: Fix mask for setting CS42L73_SP_3ST bit
by Axel Lin 23 Jul '15

23 Jul '15

CS42L73_SP_3ST is BIT(7), so the mask field is wrong. Fix it. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs42l73.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 7cb1d70..be6477f 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1236,8 +1236,8 @@ static int cs42l73_set_tristate(struct snd_soc_dai *dai, int tristate) struct snd_soc_codec *codec = dai->codec; int id = dai->id; - return snd_soc_update_bits(codec, CS42L73_SPC(id), - 0x7F, tristate << 7); + return snd_soc_update_bits(codec, CS42L73_SPC(id), CS42L73_SP_3ST, + tristate << 7); } static const struct snd_pcm_hw_constraint_list constraints_12_24 = { -- 2.1.0

3 2

[alsa-devel] [PATCH 0/4] i915 to call hda driver on HDMI plug/unplug
by David Henningsson 23 Jul '15

23 Jul '15

This patch set aims to resolve three problems: * The first - and most serious one - is that the audio driver is not woken up properly when in power save modes, especially not when the HDA controller is in D3. By having the i915 driver call directly into the hda driver, the HDA driver is always notified that an HDMI hotplug event has happened. * Second, there is currently no way for userspace to match an HDMI audio output with an HDMI video output. We fix this by sending connector_type and connector_type_id in the HDMI hotplug callback. * Third, writing ELD info to the hardware just so the HDA driver can read it from the hardware seems a bit inefficient. We could just pass that information in the callback, too. The patch in its current form fixes the first of these problems and provides most of the infrastructure for the second and third problem. The patch set is based on 4.2rc2 + my recent codec wakeup patch. So far, it has been tested (and working) on one Skylake machine. David Henningsson (4): drm/i915: Add audio hotplug info struct drm/i915: Call audio hotplug notify function ALSA: hda - Dispatch incoming HDMI hotplug i915 callback ALSA: hda - Wake the codec up on hotplug notify events drivers/gpu/drm/i915/i915_drv.h | 1 + drivers/gpu/drm/i915/intel_audio.c | 46 ++++++++++++++++++++++++++++++++++++ include/drm/i915_component.h | 19 +++++++++++++++ include/sound/hdaudio.h | 4 ++++ sound/hda/hdac_i915.c | 24 +++++++++++++++++++ sound/pci/hda/patch_hdmi.c | 22 ++++++++++++++++- 6 files changed, 115 insertions(+), 1 deletion(-) -- 1.7.9.5

5 22

[alsa-devel] [PATCH v2 0/4] i915 to call hda driver on HDMI plug/unplug
by David Henningsson 22 Jul '15

22 Jul '15

Changes since previous version: * Renames and refactorings according to Daniel's and Takashi's suggestions. * Debug message slightly improved. * Braswell now has all three pins too, Baytrail still has only one pin. It would still be good with some confirmation from Intel that this is correct. David Henningsson (4): drm/i915: Add audio hotplug info struct drm/i915: Call audio hotplug notify function ALSA: hda - Dispatch incoming HDMI hotplug i915 callback ALSA: hda - Wake the codec up on hotplug notify events drivers/gpu/drm/i915/i915_drv.h | 1 + drivers/gpu/drm/i915/intel_audio.c | 41 ++++++++++++++++++++++++++++++++++++ include/drm/i915_component.h | 18 ++++++++++++++++ include/sound/hdaudio.h | 4 ++++ sound/hda/hdac_i915.c | 26 +++++++++++++++++++++++ sound/pci/hda/patch_hdmi.c | 13 +++++++++++- 6 files changed, 102 insertions(+), 1 deletion(-) -- 1.7.9.5

1 4

[alsa-devel] [RFC][PATCH 00/37] ALSA: firewire: support AMDTP variants
by Takashi Sakamoto 22 Jul '15

22 Jul '15

Hi, This patchset is for three new drivers and related modification to firewire-lib module. Overview: * Patch 01-02: for firewire-lib, to split AMDTP layer to two parts * Patch 03-07: for firewire-lib, to add MIDI asynchronous transaction * Patch 08-16: for firewire-digi00x, a new driver for Digi 002/003 family * Patch 17-26: for firewire-tascam, a new driver for TASCAM FireWire series * Patch 27-28: for firewire-lib, to add SPH field in CIP header support * Patch 29-37: for firewire-motu, a new driver for MOTU FireWire series Status: * firewire-digi00x: * well-prepared, except for final testing with Digi 003 series * firewire-tascam: * well-prepared, except for conversion of controll messages to MIDI * firewire-motu: * experimental, for preview to developers As the other my RFCs, I prepare for a branch in my backport repository, to make it easy to test this patchset. https://github.com/takaswie/snd-firewire-improve/tree/amdtp-variants I'm happy if receiving your comments, reviews, tests and your patches. I have a plan to posting patches again to merging these well-prepared drivers till next merge window for Linux 4.3. Cc: Damien Zammit <damien(a)zamaudio.com> Cc: Robin Gareus <robin(a)gareus.org> Cc: Yoshifuji Hideaki <yoshfuji(a)linux-ipv6.org> Regards Takashi Sakamoto (37): ALSA: firewire-lib: rename 'amdtp' to 'amdtp-stream' for functional separation ALSA: firewire-lib: functional separation between packet stream and data block format ALSA: firewire-lib: add helper functions for asynchronous MIDI port ALSA: firewire-lib: serialize request transaction and response transaction ALSA: firewire-lib: schedule tasklet again when MIDI substream has rest of MIDI messages ALSA: firewire-lib: add throttle for MIDI data rate ALSA: firewire-lib: avoid endless loop to transfer MIDI messages at fatal error ALSA: firewire-digi00x: add skelton for Digi 002/003 family ALSA: firewire-digi00x: add protocol layer ALSA: firewire-digi00x: add stream functionality ALSA: firewire-digi00x: add proc node to show clock status ALSA: firewire-digi00x: add PCM functionality ALSA: firewire-digi00x: add MIDI functionality ALSA: firewire-digi00x: add hwdep interface ALSA: firewire-digi00x: add support for asynchronous messaging ALSA: firewire-digi00x: add support for MIDI ports for machine control ALSA: firewire-tascam: add skeleton for TASCAM FireWire series ALSA: firewire-tascam: add a structure for model-dependent parameters. ALSA: firewire-tascam: add proc node to show firmware information ALSA: firewire-tascam: add specific protocol layer ALSA: firewire-tascam: add streaming functionality ALSA: firewire-tascam: add PCM functionality ALSA: firewire-tascam: add transaction functionality ALSA: firewire-tascam: add MIDI functionality ALSA: firewire-tascam: add hwdep interface ALSA: firewire-tascam: add MMAP support to show status and control message ALSA: firewire-lib: add support for source packet header field in CIP header ALSA: firewire-lib: enable protocol layer to handle current cycle count ALSA: firewire-motu: add skeleton for Mark of the unicorn (MOTU) FireWire series ALSA: firewire-motu: add a structure for model-dependent parameters. ALSA: firewire-motu: add MOTU specific protocol layer ALSA: firewire-motu: add stream functionality ALSA: firewire-motu: add transaction functionality ALSA: firewire-motu: add proc node to show the status of internal clock ALSA: firewire-motu: add PCM functionality ALSA: firewire-motu: add MIDI functionality ALSA: firewire-motu: add hwdep interface include/uapi/sound/asound.h | 5 +- include/uapi/sound/firewire.h | 16 +- sound/firewire/Kconfig | 38 + sound/firewire/Makefile | 5 +- sound/firewire/amdtp-am824.c | 432 ++++++++++ sound/firewire/amdtp-am824.h | 45 ++ sound/firewire/amdtp-stream.c | 882 ++++++++++++++++++++ sound/firewire/amdtp-stream.h | 253 ++++++ sound/firewire/amdtp.c | 1107 -------------------------- sound/firewire/amdtp.h | 296 ------- sound/firewire/bebob/bebob.h | 2 +- sound/firewire/bebob/bebob_midi.c | 16 +- sound/firewire/bebob/bebob_pcm.c | 10 +- sound/firewire/bebob/bebob_stream.c | 34 +- sound/firewire/dice/dice-midi.c | 16 +- sound/firewire/dice/dice-pcm.c | 10 +- sound/firewire/dice/dice-stream.c | 41 +- sound/firewire/dice/dice.h | 2 +- sound/firewire/digi00x/Makefile | 4 + sound/firewire/digi00x/amdtp-dot.c | 425 ++++++++++ sound/firewire/digi00x/digi00x-hwdep.c | 200 +++++ sound/firewire/digi00x/digi00x-midi.c | 198 +++++ sound/firewire/digi00x/digi00x-pcm.c | 355 +++++++++ sound/firewire/digi00x/digi00x-proc.c | 99 +++ sound/firewire/digi00x/digi00x-stream.c | 422 ++++++++++ sound/firewire/digi00x/digi00x-transaction.c | 137 ++++ sound/firewire/digi00x/digi00x.c | 174 ++++ sound/firewire/digi00x/digi00x.h | 155 ++++ sound/firewire/fcp.c | 2 +- sound/firewire/fireworks/fireworks.c | 12 +- sound/firewire/fireworks/fireworks.h | 2 +- sound/firewire/fireworks/fireworks_midi.c | 16 +- sound/firewire/fireworks/fireworks_pcm.c | 8 +- sound/firewire/fireworks/fireworks_stream.c | 8 +- sound/firewire/lib.c | 127 +++ sound/firewire/lib.h | 52 ++ sound/firewire/motu/Makefile | 3 + sound/firewire/motu/amdtp-motu.c | 442 ++++++++++ sound/firewire/motu/motu-hwdep.c | 210 +++++ sound/firewire/motu/motu-midi.c | 151 ++++ sound/firewire/motu/motu-pcm.c | 397 +++++++++ sound/firewire/motu/motu-proc.c | 73 ++ sound/firewire/motu/motu-stream.c | 670 ++++++++++++++++ sound/firewire/motu/motu-transaction.c | 97 +++ sound/firewire/motu/motu.c | 311 ++++++++ sound/firewire/motu/motu.h | 127 +++ sound/firewire/oxfw/oxfw-midi.c | 16 +- sound/firewire/oxfw/oxfw-pcm.c | 8 +- sound/firewire/oxfw/oxfw-stream.c | 11 +- sound/firewire/oxfw/oxfw.h | 2 +- sound/firewire/tascam/Makefile | 4 + sound/firewire/tascam/amdtp-tascam.c | 236 ++++++ sound/firewire/tascam/tascam-hwdep.c | 232 ++++++ sound/firewire/tascam/tascam-midi.c | 158 ++++ sound/firewire/tascam/tascam-pcm.c | 306 +++++++ sound/firewire/tascam/tascam-proc.c | 88 ++ sound/firewire/tascam/tascam-stream.c | 512 ++++++++++++ sound/firewire/tascam/tascam-transaction.c | 234 ++++++ sound/firewire/tascam/tascam.c | 232 ++++++ sound/firewire/tascam/tascam.h | 141 ++++ 60 files changed, 8760 insertions(+), 1507 deletions(-) create mode 100644 sound/firewire/amdtp-am824.c create mode 100644 sound/firewire/amdtp-am824.h create mode 100644 sound/firewire/amdtp-stream.c create mode 100644 sound/firewire/amdtp-stream.h delete mode 100644 sound/firewire/amdtp.c delete mode 100644 sound/firewire/amdtp.h create mode 100644 sound/firewire/digi00x/Makefile create mode 100644 sound/firewire/digi00x/amdtp-dot.c create mode 100644 sound/firewire/digi00x/digi00x-hwdep.c create mode 100644 sound/firewire/digi00x/digi00x-midi.c create mode 100644 sound/firewire/digi00x/digi00x-pcm.c create mode 100644 sound/firewire/digi00x/digi00x-proc.c create mode 100644 sound/firewire/digi00x/digi00x-stream.c create mode 100644 sound/firewire/digi00x/digi00x-transaction.c create mode 100644 sound/firewire/digi00x/digi00x.c create mode 100644 sound/firewire/digi00x/digi00x.h create mode 100644 sound/firewire/motu/Makefile create mode 100644 sound/firewire/motu/amdtp-motu.c create mode 100644 sound/firewire/motu/motu-hwdep.c create mode 100644 sound/firewire/motu/motu-midi.c create mode 100644 sound/firewire/motu/motu-pcm.c create mode 100644 sound/firewire/motu/motu-proc.c create mode 100644 sound/firewire/motu/motu-stream.c create mode 100644 sound/firewire/motu/motu-transaction.c create mode 100644 sound/firewire/motu/motu.c create mode 100644 sound/firewire/motu/motu.h create mode 100644 sound/firewire/tascam/Makefile create mode 100644 sound/firewire/tascam/amdtp-tascam.c create mode 100644 sound/firewire/tascam/tascam-hwdep.c create mode 100644 sound/firewire/tascam/tascam-midi.c create mode 100644 sound/firewire/tascam/tascam-pcm.c create mode 100644 sound/firewire/tascam/tascam-proc.c create mode 100644 sound/firewire/tascam/tascam-stream.c create mode 100644 sound/firewire/tascam/tascam-transaction.c create mode 100644 sound/firewire/tascam/tascam.c create mode 100644 sound/firewire/tascam/tascam.h -- 2.1.4

4 50

[alsa-devel] pulseaudio external mic jack detection - ASoC based card
by maruthi srinivas 22 Jul '15

22 Jul '15

Hi, I have a issue with mic jack status in Ubuntu "sound settings" UI. When external mic is removed, UI still shows Mic. When compared to HD Azalia driver, UI shouldn't show any external Mic jack information, if it is removed. Codec driver I use reports jack status properly and uses snd_soc_jack_report(). /proc/interrupts, /dev/input/eventX reflects jack activity. But the same is not reflected in UI. I want to know how pulseaudio detect jack status for drivers which follow ALSA SoC architecture. Does it read from any event from /dev/input/eventX ? Please help in understanding. Regards, Maruthi

4 4

[alsa-devel] [PATCH RFT] ASoC: sgtl5000: Fix up define for SGTL5000_SMALL_POP
by Axel Lin 22 Jul '15

22 Jul '15

Currently, below code actually does not update any bit because SGTL5000_SMALL_POP is 0. snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, SGTL5000_SMALL_POP, 1); The SGTL5000_SMALL_POP should be BIT(0) rather than 0, fix it. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi, I don't have h/w and datasheet, but the code looks obvious does not make sense. Can someone help to confirm and test if this patch? Thanks, Axel sound/soc/codecs/sgtl5000.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index bd7a344..1c317de 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -275,7 +275,7 @@ #define SGTL5000_BIAS_CTRL_MASK 0x000e #define SGTL5000_BIAS_CTRL_SHIFT 1 #define SGTL5000_BIAS_CTRL_WIDTH 3 -#define SGTL5000_SMALL_POP 0 +#define SGTL5000_SMALL_POP 1 /* * SGTL5000_CHIP_MIC_CTRL -- 2.1.0

5 5

[alsa-devel] [PATCH] ASoC: kirkwood: prevent double streaming
by Jean-Francois Moine 22 Jul '15

22 Jul '15

The kirkwood audio subsystem presents 2 PCM's for one source. Streaming on a second PCM while the first one is active cuts this last one. Then, ending the last stream gives a kernel trap in free_irq(). Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/kirkwood/kirkwood-dma.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index 4cf2245..dbfdfe9 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -148,10 +148,14 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) dram = mv_mbus_dram_info(); addr = substream->dma_buffer.addr; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (priv->substream_play) + return -EBUSY; priv->substream_play = substream; kirkwood_dma_conf_mbus_windows(priv->io, KIRKWOOD_PLAYBACK_WIN, addr, dram); } else { + if (priv->substream_rec) + return -EBUSY; priv->substream_rec = substream; kirkwood_dma_conf_mbus_windows(priv->io, KIRKWOOD_RECORD_WIN, addr, dram); -- 2.1.4

2 1

[alsa-devel] [PATCH 1/2] ASoC: dapm: Add helper function to free a widget
by Lars-Peter Clausen 22 Jul '15

22 Jul '15

snd_soc_tplg_widget_remove_all() has a verbatim copy of an older version of the widget freeing code from dapm_free_widgets(). Add a new helper function that takes care of freeing a widget and use it in both places. This removes the duplicated code and also makes sure that future changes to the widget freeing code only have to be made in one location. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Series depends on fix/dapm --- include/sound/soc-dapm.h | 1 + sound/soc/soc-dapm.c | 38 ++++++++++++++++++++++---------------- sound/soc/soc-topology.c | 25 +------------------------ 3 files changed, 24 insertions(+), 40 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 37d95a8..cadc7fc 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -397,6 +397,7 @@ int snd_soc_dapm_del_routes(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route, int num); int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route, int num); +void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w); /* dapm events */ void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1992568..7a57b01 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2315,30 +2315,36 @@ static void dapm_free_path(struct snd_soc_dapm_path *path) kfree(path); } +void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_path *p, *next_p; + + list_del(&w->list); + /* + * remove source and sink paths associated to this widget. + * While removing the path, remove reference to it from both + * source and sink widgets so that path is removed only once. + */ + list_for_each_entry_safe(p, next_p, &w->sources, list_sink) + dapm_free_path(p); + + list_for_each_entry_safe(p, next_p, &w->sinks, list_source) + dapm_free_path(p); + + kfree(w->kcontrols); + kfree(w->name); + kfree(w); +} + /* free all dapm widgets and resources */ static void dapm_free_widgets(struct snd_soc_dapm_context *dapm) { struct snd_soc_dapm_widget *w, *next_w; - struct snd_soc_dapm_path *p, *next_p; list_for_each_entry_safe(w, next_w, &dapm->card->widgets, list) { if (w->dapm != dapm) continue; - list_del(&w->list); - /* - * remove source and sink paths associated to this widget. - * While removing the path, remove reference to it from both - * source and sink widgets so that path is removed only once. - */ - list_for_each_entry_safe(p, next_p, &w->sources, list_sink) - dapm_free_path(p); - - list_for_each_entry_safe(p, next_p, &w->sinks, list_source) - dapm_free_path(p); - - kfree(w->kcontrols); - kfree(w->name); - kfree(w); + snd_soc_dapm_free_widget(w); } } diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 4a838e4..df3ddf0 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1734,7 +1734,6 @@ void snd_soc_tplg_widget_remove_all(struct snd_soc_dapm_context *dapm, u32 index) { struct snd_soc_dapm_widget *w, *next_w; - struct snd_soc_dapm_path *p, *next_p; list_for_each_entry_safe(w, next_w, &dapm->card->widgets, list) { @@ -1746,31 +1745,9 @@ void snd_soc_tplg_widget_remove_all(struct snd_soc_dapm_context *dapm, if (w->dobj.index != index && w->dobj.index != SND_SOC_TPLG_INDEX_ALL) continue; - - list_del(&w->list); - - /* - * remove source and sink paths associated to this widget. - * While removing the path, remove reference to it from both - * source and sink widgets so that path is removed only once. - */ - list_for_each_entry_safe(p, next_p, &w->sources, list_sink) { - list_del(&p->list_sink); - list_del(&p->list_source); - list_del(&p->list); - kfree(p); - } - list_for_each_entry_safe(p, next_p, &w->sinks, list_source) { - list_del(&p->list_sink); - list_del(&p->list_source); - list_del(&p->list); - kfree(p); - } /* check and free and dynamic widget kcontrols */ snd_soc_tplg_widget_remove(w); - kfree(w->kcontrols); - kfree(w->name); - kfree(w); + snd_soc_dapm_free_widget(w); } } EXPORT_SYMBOL_GPL(snd_soc_tplg_widget_remove_all); -- 2.1.4

2 3

[alsa-devel] [PATCH] ASoC: ssm4567: Keep TDM_BCLKS in ssm4567_set_dai_fmt
by Ben Zhang 22 Jul '15

22 Jul '15

The regmap_write in ssm4567_set_dai_fmt accidentally clears the TDM_BCLKS field which was set earlier by ssm4567_set_tdm_slot. This patch fixes it by using regmap_update_bits with proper mask. Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/ssm4567.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c index 4662587..a757ab8 100644 --- a/sound/soc/codecs/ssm4567.c +++ b/sound/soc/codecs/ssm4567.c @@ -315,7 +315,13 @@ static int ssm4567_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) if (invert_fclk) ctrl1 |= SSM4567_SAI_CTRL_1_FSYNC; - return regmap_write(ssm4567->regmap, SSM4567_REG_SAI_CTRL_1, ctrl1); + return regmap_update_bits(ssm4567->regmap, SSM4567_REG_SAI_CTRL_1, + SSM4567_SAI_CTRL_1_BCLK | + SSM4567_SAI_CTRL_1_FSYNC | + SSM4567_SAI_CTRL_1_LJ | + SSM4567_SAI_CTRL_1_TDM | + SSM4567_SAI_CTRL_1_PDM, + ctrl1); } static int ssm4567_set_power(struct ssm4567 *ssm4567, bool enable) -- 2.4.3.573.g4eafbef

4 3

[alsa-devel] [PATCH] ALSA: hda - Add headset mic pin quirk for a Dell device
by David Henningsson 22 Jul '15

22 Jul '15

Without this patch, the headset mic will not work on this machine. BugLink: https://bugs.launchpad.net/bugs/1476987 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index caba66b..11c8a74 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5463,6 +5463,17 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1d, 0x40700001}, {0x21, 0x02211030}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x40000000}, + {0x14, 0x90170130}, + {0x17, 0x411111f0}, + {0x18, 0x411111f0}, + {0x19, 0x411111f0}, + {0x1a, 0x411111f0}, + {0x1b, 0x01014020}, + {0x1d, 0x4054c029}, + {0x1e, 0x411111f0}, + {0x21, 0x0221103f}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60160}, {0x14, 0x90170120}, {0x17, 0x90170140}, -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: hda - remove one pin from ALC292_STANDARD_PINS
by Hui Wang 22 Jul '15

22 Jul '15

One more Dell laptop with alc293 codec needs ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, but the pin 0x1e does not match the corresponding one in the ALC292_STANDARD_PINS. To use this macro for this machine, we need to remove pin 0x1e from it. BugLink: https://bugs.launchpad.net/bugs/1476888 Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 26 +++++++++++++++++++------- 1 file changed, 19 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index caba66b..ee6f13a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5430,8 +5430,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {0x15, 0x0221401f}, \ {0x1a, 0x411111f0}, \ {0x1b, 0x411111f0}, \ - {0x1d, 0x40700001}, \ - {0x1e, 0x411111f0} + {0x1d, 0x40700001} #define ALC298_STANDARD_PINS \ {0x18, 0x411111f0}, \ @@ -5690,35 +5689,48 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x13, 0x411111f0}, {0x16, 0x01014020}, {0x18, 0x411111f0}, - {0x19, 0x01a19030}), + {0x19, 0x01a19030}, + {0x1e, 0x411111f0}), SND_HDA_PIN_QUIRK(0x10ec0292, 0x1028, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, ALC292_STANDARD_PINS, {0x12, 0x90a60140}, {0x13, 0x411111f0}, {0x16, 0x01014020}, {0x18, 0x02a19031}, - {0x19, 0x01a1903e}), + {0x19, 0x01a1903e}, + {0x1e, 0x411111f0}), SND_HDA_PIN_QUIRK(0x10ec0292, 0x1028, "Dell", ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, ALC292_STANDARD_PINS, {0x12, 0x90a60140}, {0x13, 0x411111f0}, {0x16, 0x411111f0}, {0x18, 0x411111f0}, - {0x19, 0x411111f0}), + {0x19, 0x411111f0}, + {0x1e, 0x411111f0}), SND_HDA_PIN_QUIRK(0x10ec0293, 0x1028, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_STANDARD_PINS, {0x12, 0x40000000}, {0x13, 0x90a60140}, {0x16, 0x21014020}, {0x18, 0x411111f0}, - {0x19, 0x21a19030}), + {0x19, 0x21a19030}, + {0x1e, 0x411111f0}), SND_HDA_PIN_QUIRK(0x10ec0293, 0x1028, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_STANDARD_PINS, {0x12, 0x40000000}, {0x13, 0x90a60140}, {0x16, 0x411111f0}, {0x18, 0x411111f0}, - {0x19, 0x411111f0}), + {0x19, 0x411111f0}, + {0x1e, 0x411111f0}), + SND_HDA_PIN_QUIRK(0x10ec0293, 0x1028, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC292_STANDARD_PINS, + {0x12, 0x40000000}, + {0x13, 0x90a60140}, + {0x16, 0x21014020}, + {0x18, 0x411111f0}, + {0x19, 0x21a19030}, + {0x1e, 0x411111ff}), SND_HDA_PIN_QUIRK(0x10ec0298, 0x1028, "Dell", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, ALC298_STANDARD_PINS, {0x12, 0x90a60130}, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: dapm: Don't add prefix to widget stream name
by Lars-Peter Clausen 21 Jul '15

21 Jul '15

Commit fdb6eb0a1287 ("ASoC: dapm: Modify widget stream name according to prefix") fixed the case where a DAPM route between a DAI widget and a DAC/ADC/AIF widget with a matching stream name was not created when the DAPM context was using a prefix. Unfortunately the patch introduced a few issues on its own like leaking the dynamically allocated stream name memory and also not checking whether the allocation succeeded in the first place. It is also incomplete in that it still does not handle the case where stream name of the widget is a substring of the stream name of the DAI, which is explicitly allowed and works fine if no DAPM prefix is used. Revert the commit and take a slightly different approach to solving the issue. Instead of comparing the widget's stream name to the name of the DAI widget compare it to the stream name of the DAI widget. The stream name of the DAI widget is identical to the name of the DAI widget except that it wont have the DAPM prefix added. So this approach behaves identical regardless to whether the DAPM context uses a prefix or not. We don't have to worry about potentially matching with a widget with the same stream name, but from a different DAPM context with a different prefix, since the code already makes sure that both the DAI widget and the matched widget are from the same DAPM context. Fixes: fdb6eb0a1287 ("ASoC: dapm: Modify widget stream name according to prefix") Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1628f0c..1992568 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3344,16 +3344,10 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, } prefix = soc_dapm_prefix(dapm); - if (prefix) { + if (prefix) w->name = kasprintf(GFP_KERNEL, "%s %s", prefix, widget->name); - if (widget->sname) - w->sname = kasprintf(GFP_KERNEL, "%s %s", prefix, - widget->sname); - } else { + else w->name = kasprintf(GFP_KERNEL, "%s", widget->name); - if (widget->sname) - w->sname = kasprintf(GFP_KERNEL, "%s", widget->sname); - } if (w->name == NULL) { kfree(w); return NULL; @@ -3802,7 +3796,7 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) break; } - if (!w->sname || !strstr(w->sname, dai_w->name)) + if (!w->sname || !strstr(w->sname, dai_w->sname)) continue; if (dai_w->id == snd_soc_dapm_dai_in) { -- 2.1.4

3 4

[alsa-devel] [patch] ASoC: max98925: reading beyond the end of the array
by Dan Carpenter 21 Jul '15

21 Jul '15

Debug prints are seldom useful and this one has an annoying thing where it reads one space beyond the end of the array on error. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c index e1753c8..ebb648a 100644 --- a/sound/soc/codecs/max98925.c +++ b/sound/soc/codecs/max98925.c @@ -271,8 +271,6 @@ static inline int max98925_rate_value(struct snd_soc_codec *codec, break; } } - dev_dbg(codec->dev, "%s: sample rate is %d, returning %d\n", - __func__, rate_table[i].rate, *value); return ret; }

2 4

Re: [alsa-devel] ASoC: rsnd: show debug message for SSI/SRC/DVC connection
by Dan Carpenter 21 Jul '15

21 Jul '15

Hello Kuninori Morimoto, This is a semi-automatic email about new static checker warnings. The patch 84e95355602c: "ASoC: rsnd: show debug message for SSI/SRC/DVC connection" from Jul 15, 2015, leads to the following Smatch complaint: sound/soc/sh/rcar/core.c:338 rsnd_dai_connect() warn: variable dereferenced before check 'mod' (see line 335) sound/soc/sh/rcar/core.c 334 { 335 struct rsnd_priv *priv = rsnd_mod_to_priv(mod); ^^^^^^^^^^^^^^^^^^^^^ Patch adds new dereference. 336 struct device *dev = rsnd_priv_to_dev(priv); 337 338 if (!mod) ^^^^ Existing code assumed "mod" could be NULL. 339 return -EIO; 340 regards, dan carpenter

3 2

[alsa-devel] [patch] ASoC: sti: error handling bug in sti_uniperiph_cpu_dai_of()
by Dan Carpenter 21 Jul '15

21 Jul '15

There is a stray '!' which means the condition is never true. Fixes: f3bd847eb0a7 ('ASoC: sti: Add uniperipheral dai driver') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c index 51f745c..dffabf3 100644 --- a/sound/soc/sti/sti_uniperif.c +++ b/sound/soc/sti/sti_uniperif.c @@ -175,7 +175,7 @@ static int sti_uniperiph_cpu_dai_of(struct device_node *node, UNIPERIF_FIFO_DATA_OFFSET(uni); uni->irq = platform_get_irq(priv->pdev, 0); - if (!uni->irq < 0) { + if (uni->irq < 0) { dev_err(dev, "Failed to get IRQ resource"); return -ENXIO; }

2 1

[alsa-devel] [PATCH] ALSA: hda - Add new GPU codec ID 0x10de007d to snd-hda
by Aaron Plattner 21 Jul '15

21 Jul '15

Vendor ID 0x10de007d is used by a yet-to-be-named GPU chip. Signed-off-by: Aaron Plattner <aplattner(a)nvidia.com> --- This chip also has the 2-ch audio swapping bug, so patch_nvhdmi is appropriate here. sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 95158914cc6c..a97db5fc8a15 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3512,6 +3512,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x10de0070, .name = "GPU 70 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de0071, .name = "GPU 71 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de0072, .name = "GPU 72 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de007d, .name = "GPU 7d HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x11069f80, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi }, { .id = 0x11069f81, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi }, @@ -3576,6 +3577,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0067"); MODULE_ALIAS("snd-hda-codec-id:10de0070"); MODULE_ALIAS("snd-hda-codec-id:10de0071"); MODULE_ALIAS("snd-hda-codec-id:10de0072"); +MODULE_ALIAS("snd-hda-codec-id:10de007d"); MODULE_ALIAS("snd-hda-codec-id:10de8001"); MODULE_ALIAS("snd-hda-codec-id:11069f80"); MODULE_ALIAS("snd-hda-codec-id:11069f81"); -- 2.4.6

2 2

[alsa-devel] [PATCH 0/7] Update ALSA, and au0828 drivers to use Managed Media Controller API
by Shuah Khan 20 Jul '15

20 Jul '15

s patch series updates ALSA driver, and au0828 core driver to use Managed Media controller API to share tuner. Please note that Managed Media Controller API and DVB and V4L2 drivers updates to use Media Controller API have been added in a prior patch series. Media Controller API is enhanced with two new interfaces to register and unregister entity_notify hooks to allow drivers to take appropriate actions when as new entities get added to the shared media device. Tested exclusion between digital, analog, and audio to ensure when tuner has an active link to DVB FE, analog, and audio will detect and honor the tuner busy conditions and vice versa. This patch series has been updated to address comments from 3 previous versions of this series. Link to v3 below for reference. https://www.mail-archive.com/linux-media%40vger.kernel.org/msg89491.html https://www.mail-archive.com/linux-media@vger.kernel.org/msg89492.html https://www.mail-archive.com/linux-media%40vger.kernel.org/msg89493.html Open Issues: ALSA has makes media_entity_pipeline_start() call in irq path. I am seeing warnings that the graph_mutex is unsafe irq lock. Media Controller API updates to start/stop pipeline to be irq safe might be necessary. Maybe there are other MC interfaces that need to be irq safe, but I haven't seen any problems with my testing. So as per options, graph_mutex could be changed to a spinlock. It looks like drivers hold this lock and it isn't abstracted to MC API. Unfortunate, this would require changes to drivers that directly hold the lock for graph walks if this mutex is changed to spinlock. This issue needs to be resolved. Note: This series includes a revert to a patch that added media controller entity framework enhancements that implemented entity ops for entity_notify functionality. Entity ops for entity_notify doesn't handle and cover entity create ordering variations that could occur during boot. entity_notify list has been moved to media device level which makes the entity_notify calls to work correctly. Shuah Khan (7): Revert "[media] media: media controller entity framework enhancements for ALSA" media: Media Controller register/unregister entity_notify API media: Add ALSA Media Controller devnodes media: change dvb-frontend to honor MC tuner enable error media: au8522 change to create MC pad for ALSA Audio Out media: au0828 change to use Managed Media Controller API sound/usb: Update ALSA driver to use Managed Media Controller API drivers/media/dvb-core/dvb_frontend.c | 5 +- drivers/media/dvb-frontends/au8522.h | 8 + drivers/media/dvb-frontends/au8522_decoder.c | 1 + drivers/media/dvb-frontends/au8522_priv.h | 8 - drivers/media/media-device.c | 46 ++++- drivers/media/usb/au0828/au0828-core.c | 132 ++++++++------ drivers/media/usb/au0828/au0828.h | 5 + include/media/media-device.h | 23 +++ include/media/media-entity.h | 4 - include/uapi/linux/media.h | 5 + sound/usb/Makefile | 15 +- sound/usb/card.c | 5 + sound/usb/card.h | 1 + sound/usb/media.c | 260 +++++++++++++++++++++++++++ sound/usb/media.h | 52 ++++++ sound/usb/pcm.c | 15 +- sound/usb/quirks-table.h | 1 + sound/usb/quirks.c | 9 +- sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 1 + 20 files changed, 523 insertions(+), 75 deletions(-) create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.1.4

3 14

[alsa-devel] [asoc:topic/rockchip 2/2] warning: (SND_SOC_ROCKCHIP_MAX98090 && ..) selects SND_SOC_ROCKCHIP_I2S which has unmet direct dependencies (SOUND && ..)
by kbuild test robot 20 Jul '15

20 Jul '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/rockchip head: 86059653ea7ca7b30ed25d6bec5807ba59a4f2e6 commit: 86059653ea7ca7b30ed25d6bec5807ba59a4f2e6 [2/2] ASoC: rockchip: Add machine driver for rt5645/rt5650 codec config: sparc64-allyesconfig (attached as .config) reproduce: wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 86059653ea7ca7b30ed25d6bec5807ba59a4f2e6 # save the attached .config to linux build tree make.cross ARCH=sparc64 All warnings (new ones prefixed by >>): warning: (SND_SOC_ROCKCHIP_MAX98090 && SND_SOC_ROCKCHIP_RT5645) selects SND_SOC_ROCKCHIP_I2S which has unmet direct dependencies (SOUND && !M68K && !UML && SND && SND_SOC && CLKDEV_LOOKUP && SND_SOC_ROCKCHIP) --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH 1/2] ASoC: cs4349: Fix max_register setting for cs4349_regmap
by Axel Lin 20 Jul '15

20 Jul '15

The max_register should be the maximum valid register index rather than number of registers. Also remove unused defines. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs4349.c | 2 +- sound/soc/codecs/cs4349.h | 3 --- 2 files changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c index a8df8a74..9273b06 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -268,7 +268,7 @@ static struct regmap_config cs4349_regmap = { .reg_bits = 8, .val_bits = 8, - .max_register = CS4349_NUMREGS, + .max_register = CS4349_MISC, .reg_defaults = cs4349_reg_defaults, .num_reg_defaults = ARRAY_SIZE(cs4349_reg_defaults), .readable_reg = cs4349_readable_register, diff --git a/sound/soc/codecs/cs4349.h b/sound/soc/codecs/cs4349.h index 3884a89..7effa0a 100644 --- a/sound/soc/codecs/cs4349.h +++ b/sound/soc/codecs/cs4349.h @@ -36,9 +36,6 @@ struct cs4349_platform_data { #define CS4349_RMPFLT 0x07 /* Ramp and Filter Control */ #define CS4349_MISC 0x08 /* Power Down,Freeze Control,Pop Stop*/ -#define CS4349_FIRSTREG 0x01 -#define CS4349_LASTREG 0x08 -#define CS4349_NUMREGS (CS4349_LASTREG - CS4349_FIRSTREG + 1) #define CS4349_I2C_INCR 0x80 -- 2.1.0

3 4

[alsa-devel] [PATCH] ASoC: rt5645: Remove return value in jack detect work
by Nicolas Boichat 20 Jul '15

20 Jul '15

"ASoC: rt5645: Check if codec is initialized in workqueue handler" adds a check if codec is NULL in rt5645_irq_detection, which returns an int. However, "ASoC: rt5645: Remove irq_jack_detection function" removes that function, and moves the code in jack_detect_work, which returns void. Remove the return value to fix compilation warning. Signed-off-by: Nicolas Boichat <drinkcat(a)chromium.org> --- Mark, Thanks for replacing "ASoC: rt5645: Check if codec is initialized in workqueue handler" in fix/rt5645, that makes this a bit easier. This patch applies on top of broonie/sound.git/for-next, and should be applied in topic/rt5645, after merging fix/rt5645 into topic/rt5645. sound/soc/codecs/rt5645.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 3a6756d..9d54b34 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2928,7 +2928,7 @@ static void rt5645_jack_detect_work(struct work_struct *work) int val, btn_type, gpio_state = 0, report = 0; if (!rt5645->codec) - return -EINVAL; + return; switch (rt5645->pdata.jd_mode) { case 0: /* Not using rt5645 JD */ -- 2.4.3.573.g4eafbef

2 1

[alsa-devel] [PATCH v3 0/2] Add codec machine driver for rockchip platform
by Xing Zheng 20 Jul '15

20 Jul '15

From: zhengxing <zhengxing(a)rock-chips.com> Hi, The simple-card is not common at present, soc maybe need own machine driver for jack detection. Add drivers for two families of rockchip-bases chromebooks. These machine drives don't use simplecard because we need custom jack detection plumbing. - use ts3a227e for ext jack detection with max98090 - call rt5645_set_jack_detect function via rt5645 codec driver Thanks. Changes in v3: Signed-off-by: zhengxing <zhengxing(a)rock-chips.com> zhengxing (2): ASoC: rockchip: Add machine driver for max98090 codec ASoC: rockchip: Add machine driver for rt5645/rt5650 codec .../bindings/sound/rockchip-max98090.txt | 19 ++ .../devicetree/bindings/sound/rockchip-rt5645.txt | 17 ++ sound/soc/rockchip/Kconfig | 19 ++ sound/soc/rockchip/Makefile | 6 + sound/soc/rockchip/rockchip_max98090.c | 236 ++++++++++++++++++++ sound/soc/rockchip/rockchip_rt5645.c | 225 +++++++++++++++++++ 6 files changed, 522 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip-max98090.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip-rt5645.txt create mode 100644 sound/soc/rockchip/rockchip_max98090.c create mode 100644 sound/soc/rockchip/rockchip_rt5645.c -- 1.7.9.5

2 4

[alsa-devel] [PATCH] ASoC: ux500: adjust devm usage
by Julia Lawall 20 Jul '15

20 Jul '15

The explicit call to devm_regulator_put in the probe and remove functions does not seem to be necessary. In particular, the functions prcmu_qos_remove_requirement and ux500_msp_i2s_cleanup_msp in the remove function seem to do nothing that can interfere with devm_regulator_put, making it safe to allow devm_regulator_put to occur after the end of the remove function. Convert the calls to clk_get to devm_clk_get, and remove the corresponding calls to clk_put in the probe and remove functions. Replace various gotos by direct returns, and drop unneeded labels. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/soc/ux500/ux500_msp_dai.c | 29 ++++++++++------------------- 1 file changed, 10 insertions(+), 19 deletions(-) diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 978f2d7..f5df08d 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -773,20 +773,22 @@ static int ux500_msp_drv_probe(struct platform_device *pdev) } prcmu_qos_add_requirement(PRCMU_QOS_APE_OPP, (char *)pdev->name, 50); - drvdata->pclk = clk_get(&pdev->dev, "apb_pclk"); + drvdata->pclk = devm_clk_get(&pdev->dev, "apb_pclk"); if (IS_ERR(drvdata->pclk)) { ret = (int)PTR_ERR(drvdata->pclk); - dev_err(&pdev->dev, "%s: ERROR: clk_get of pclk failed (%d)!\n", + dev_err(&pdev->dev, + "%s: ERROR: devm_clk_get of pclk failed (%d)!\n", __func__, ret); - goto err_pclk; + return ret; } - drvdata->clk = clk_get(&pdev->dev, NULL); + drvdata->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(drvdata->clk)) { ret = (int)PTR_ERR(drvdata->clk); - dev_err(&pdev->dev, "%s: ERROR: clk_get failed (%d)!\n", + dev_err(&pdev->dev, + "%s: ERROR: devm_clk_get failed (%d)!\n", __func__, ret); - goto err_clk; + return ret; } ret = ux500_msp_i2s_init_msp(pdev, &drvdata->msp, @@ -795,7 +797,7 @@ static int ux500_msp_drv_probe(struct platform_device *pdev) dev_err(&pdev->dev, "%s: ERROR: Failed to init MSP-struct (%d)!", __func__, ret); - goto err_init_msp; + return ret; } dev_set_drvdata(&pdev->dev, drvdata); @@ -804,7 +806,7 @@ static int ux500_msp_drv_probe(struct platform_device *pdev) if (ret < 0) { dev_err(&pdev->dev, "Error: %s: Failed to register MSP%d!\n", __func__, drvdata->msp->id); - goto err_init_msp; + return ret; } ret = ux500_pcm_register_platform(pdev); @@ -819,13 +821,6 @@ static int ux500_msp_drv_probe(struct platform_device *pdev) err_reg_plat: snd_soc_unregister_component(&pdev->dev); -err_init_msp: - clk_put(drvdata->clk); -err_clk: - clk_put(drvdata->pclk); -err_pclk: - devm_regulator_put(drvdata->reg_vape); - return ret; } @@ -837,12 +832,8 @@ static int ux500_msp_drv_remove(struct platform_device *pdev) snd_soc_unregister_component(&pdev->dev); - devm_regulator_put(drvdata->reg_vape); prcmu_qos_remove_requirement(PRCMU_QOS_APE_OPP, "ux500_msp_i2s"); - clk_put(drvdata->clk); - clk_put(drvdata->pclk); - ux500_msp_i2s_cleanup_msp(pdev, drvdata->msp); return 0;

3 2

[alsa-devel] Applied "ASoC: cs4349: include gpio/consumer.h" to the asoc tree
by Mark Brown 20 Jul '15

20 Jul '15

The patch ASoC: cs4349: include gpio/consumer.h has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 091571d071a435a517aec42c4f8c56883c5dc531 Mon Sep 17 00:00:00 2001 From: "akpm(a)linux-foundation.org" <akpm(a)linux-foundation.org> Date: Fri, 17 Jul 2015 15:43:46 -0700 Subject: [PATCH] ASoC: cs4349: include gpio/consumer.h s390 allmodconfig: sound/soc/codecs/cs4349.c: In function 'cs4349_i2c_probe': sound/soc/codecs/cs4349.c:300: error: implicit declaration of function 'devm_gpiod_get_optional' sound/soc/codecs/cs4349.c:301: error: 'GPIOD_OUT_LOW' undeclared (first use in this function) sound/soc/codecs/cs4349.c:301: error: (Each undeclared identifier is reported only once sound/soc/codecs/cs4349.c:301: error: for each function it appears in.) sound/soc/codecs/cs4349.c:306: error: implicit declaration of function 'gpiod_set_value_cansleep' Signed-off-by: Andrew Morton <akpm(a)linux-foundation.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/cs4349.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c index 2569010ee396..a7538aea7f99 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -16,6 +16,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/platform_device.h> #include <linux/pm.h> #include <linux/i2c.h> -- 2.1.4

1 0

[alsa-devel] Applied "ASoC: sti-uniperf: Fix implicit inclusion of pinctrl" to the asoc tree
by Mark Brown 20 Jul '15

20 Jul '15

The patch ASoC: sti-uniperf: Fix implicit inclusion of pinctrl has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From c7ab66836d2d31fe549b71c0b835c563d4419061 Mon Sep 17 00:00:00 2001 From: Mark Brown <broonie(a)kernel.org> Date: Mon, 20 Jul 2015 18:36:15 +0100 Subject: [PATCH] ASoC: sti-uniperf: Fix implicit inclusion of pinctrl Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sti/sti_uniperif.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c index 83a301f2aa66..10e8c3a0b85f 100644 --- a/sound/soc/sti/sti_uniperif.c +++ b/sound/soc/sti/sti_uniperif.c @@ -6,6 +6,7 @@ */ #include <linux/module.h> +#include <linux/pinctrl.h> #include "uniperif.h" -- 2.1.4

1 0

[alsa-devel] [PATCH v2 1/4] ASoC: fsl_sai: Add driver suspend and resume to support MEGA Fast
by Zidan Wang 20 Jul '15

20 Jul '15

For i.MX6 SoloX, there is a mode of the SoC to shutdown all power source of modules during system suspend and resume procedure. Thus, SAI needs to save all the values of registers before the system suspend and restore them after the system resume. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 33 +++++++++++++++++++++++++++++++++ 1 file changed, 33 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 5c73bea..62187e4 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -637,6 +637,8 @@ static bool fsl_sai_readable_reg(struct device *dev, unsigned int reg) static bool fsl_sai_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { + case FSL_SAI_TCSR: + case FSL_SAI_RCSR: case FSL_SAI_TFR: case FSL_SAI_RFR: case FSL_SAI_TDR: @@ -681,6 +683,7 @@ static const struct regmap_config fsl_sai_regmap_config = { .readable_reg = fsl_sai_readable_reg, .volatile_reg = fsl_sai_volatile_reg, .writeable_reg = fsl_sai_writeable_reg, + .cache_type = REGCACHE_RBTREE, }; static int fsl_sai_probe(struct platform_device *pdev) @@ -802,10 +805,40 @@ static const struct of_device_id fsl_sai_ids[] = { { /* sentinel */ } }; +#if CONFIG_PM_SLEEP +static int fsl_sai_suspend(struct device *dev) +{ + struct fsl_sai *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, true); + regcache_mark_dirty(sai->regmap); + + return 0; +} + +static int fsl_sai_resume(struct device *dev) +{ + struct fsl_sai *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, false); + regmap_write(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_SR); + regmap_write(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_SR); + msleep(1); + regmap_write(sai->regmap, FSL_SAI_TCSR, 0); + regmap_write(sai->regmap, FSL_SAI_RCSR, 0); + return regcache_sync(sai->regmap); +} +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops fsl_sai_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(fsl_sai_suspend, fsl_sai_resume) +}; + static struct platform_driver fsl_sai_driver = { .probe = fsl_sai_probe, .driver = { .name = "fsl-sai", + .pm = &fsl_sai_pm_ops, .of_match_table = fsl_sai_ids, }, }; -- 1.9.1

3 6

Re: [alsa-devel] ASoC: Add max98925 codec driver
by Dan Carpenter 20 Jul '15

20 Jul '15

Hello Anish Kumar, The patch 1ff2765182d1: "ASoC: Add max98925 codec driver" from Mar 9, 2015, leads to the following static checker warning: sound/soc/codecs/max98925.c:274 max98925_rate_value() error: buffer overflow 'rate_table' 9 <= 9 sound/soc/codecs/max98925.c 259 static inline int max98925_rate_value(struct snd_soc_codec *codec, 260 int rate, int clock, int *value, int *n, int *m) 261 { 262 int ret = -EINVAL; 263 int i; 264 265 for (i = 0; i < ARRAY_SIZE(rate_table); i++) { 266 if (rate_table[i].rate >= rate) { 267 *value = rate_table[i].sr; 268 *n = rate_table[i].divisors[clock][0]; 269 *m = rate_table[i].divisors[clock][1]; 270 ret = 0; 271 break; 272 } 273 } 274 dev_dbg(codec->dev, "%s: sample rate is %d, returning %d\n", 275 __func__, rate_table[i].rate, *value); ^^^^^^^^^^^^^^^^^^ If we don't break then we're off by one. 276 return ret; 277 } regards, dan carpenter

2 2

[alsa-devel] [PATCH] ALSA: hda: add new AMD PCI IDs with proper driver caps
by Maruthi Srinivas Bayyavarapu 20 Jul '15

20 Jul '15

Fixes audio problems on newer asics Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Cc: stable(a)vger.kernel.org --- sound/pci/hda/hda_intel.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ca151b4..cb7e8f8 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2182,6 +2182,8 @@ static const struct pci_device_id azx_ids[] = { /* ATI HDMI */ { PCI_DEVICE(0x1002, 0x1308), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + { PCI_DEVICE(0x1002, 0x157a), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0x793b), .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, { PCI_DEVICE(0x1002, 0x7919), @@ -2236,8 +2238,14 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0xaab0), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + { PCI_DEVICE(0x1002, 0xaac0), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0xaac8), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + { PCI_DEVICE(0x1002, 0xaad8), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + { PCI_DEVICE(0x1002, 0xaae8), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, /* VIA VT8251/VT8237A */ { PCI_DEVICE(0x1106, 0x3288), .driver_data = AZX_DRIVER_VIA | AZX_DCAPS_POSFIX_VIA }, -- 1.9.1

2 1

[alsa-devel] [PATCH v2] ALSA: hda - Fix Skylake codec timeout
by David Henningsson 20 Jul '15

20 Jul '15

When the controller is powered up but the HDMI codec is powered down on Skylake, the power well is turned off. When the codec is then powered up again, we need to poke the codec a little extra to make sure it wakes up. Otherwise we'll get sad "no response from codec" messages and broken audio. This also changes azx_runtime_resume to actually call snd_hdac_set_codec_wakeup for Skylake (before STATETS read). (Otherwise it would only have been called for Haswell and Broadwell, which both do not need it, so this probably was not the author's intention.) Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/hda/hdac_i915.c | 5 ++++- sound/pci/hda/hda_intel.c | 18 ++++++++++-------- 2 files changed, 14 insertions(+), 9 deletions(-) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 442500e..5676b84 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -56,8 +56,11 @@ int snd_hdac_display_power(struct hdac_bus *bus, bool enable) enable ? "enable" : "disable"); if (enable) { - if (!bus->i915_power_refcount++) + if (!bus->i915_power_refcount++) { acomp->ops->get_power(acomp->dev); + snd_hdac_set_codec_wakeup(bus, true); + snd_hdac_set_codec_wakeup(bus, false); + } } else { WARN_ON(!bus->i915_power_refcount); if (!--bus->i915_power_refcount) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ca151b4..9962237 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -979,14 +979,16 @@ static int azx_runtime_resume(struct device *dev) if (!azx_has_pm_runtime(chip)) return 0; - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL - && hda->need_i915_power) { - bus = azx_bus(chip); - snd_hdac_display_power(bus, true); - haswell_set_bclk(hda); - /* toggle codec wakeup bit for STATESTS read */ - snd_hdac_set_codec_wakeup(bus, true); - snd_hdac_set_codec_wakeup(bus, false); + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { + bus = azx_bus(chip); + if (hda->need_i915_power) { + snd_hdac_display_power(bus, true); + haswell_set_bclk(hda); + } else { + /* toggle codec wakeup bit for STATESTS read */ + snd_hdac_set_codec_wakeup(bus, true); + snd_hdac_set_codec_wakeup(bus, false); + } } /* Read STATESTS before controller reset */ -- 1.9.1

4 10

[alsa-devel] Beaglebone audio card problem diagnosis?
by g4＠novadsp.com 19 Jul '15

19 Jul '15

Hello all. I'm running mainline 4.1.0 on a Beaglebone Black (TI AM3359 SOC). The board has a standard TI TLV320AIC codec connected via the B3 rev B audio cape: (http://elinux.org/CircuitCo:Audio_Cape_RevB) Loading the nominally correct device tree overlay gets the following kernel log (1). An aplay -l listing (2) shows something recognized is present. Lsmod (3) shows what looks like a sane collection of prerequisite modules. Upon trying to play a test file the kernel gets a number of buffer underflow/underrun messages (4) There was a patch posted here (http://mailman.alsa-project.org/pipermail/alsa-devel/2014-November/083797.h tml) which appears to have made it into 4.1.0. Any thoughts on where to start looking to fix? More info can be provided if required and can also check hardware pin activity. Any pointers much appreciated. TAIA Jerry. -----------------------------------snip (1) root@arm:/home/debian# echo BB-BONE-AUDI-02 > /sys/devices/platform/bone_capemgr/slots [ 100.654019] bone_capemgr bone_capemgr: part_number 'BB-BONE-AUDI-02', version 'N/A' [ 100.662117] bone_capemgr bone_capemgr: slot #4: override [ 100.667840] bone_capemgr bone_capemgr: Using override eeprom data at slot 4 [ 100.675221] bone_capemgr bone_capemgr: slot #4: 'Override Board Name,00A0,Override Manuf,BB-BONE-AUDI-02' [ 100.712198] bone_capemgr bone_capemgr: slot #4: dtbo 'BB-BONE-AUDI-02-00A0.dtbo' loaded; overlay id #0 root@arm:/home/debian# [ 101.015982] davinci_evm ocp:sound: ASoC: CPU DAI (null) not registered [ 101.023017] davinci_evm ocp:sound: snd_soc_register_card failed (-517) [ 101.056114] 2-0018 supply IOVDD not found, using dummy regulator [ 101.067907] 2-0018 supply DVDD not found, using dummy regulator [ 101.074535] 2-0018 supply AVDD not found, using dummy regulator [ 101.080746] 2-0018 supply DRVDD not found, using dummy regulator [ 101.089316] davinci_evm ocp:sound: ASoC: CODEC DAI tlv320aic3x-hifi not registered [ 101.097310] davinci_evm ocp:sound: snd_soc_register_card failed (-517) [ 101.114161] davinci_evm ocp:sound: tlv320aic3x-hifi <-> 48038000.mcasp mapping ok -----------------------------------snip (2) root@arm:/home/debian# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: EVM [DA830 EVM], device 0: AIC3X tlv320aic3x-hifi-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 -----------------------------------snip (3) root@arm:/home/debian# lsmod Module Size Used by snd_soc_davinci_mcasp 14407 2 snd_soc_tlv320aic3x 40077 1 snd_soc_edma 1150 1 snd_soc_davinci_mcasp snd_soc_evm 4802 0 snd_soc_omap 2573 1 snd_soc_davinci_mcasp snd_soc_core 156839 5 snd_soc_davinci_mcasp,snd_soc_edma,snd_soc_omap,snd_soc_tlv320aic3x,snd_soc_ evm snd_compress 11858 1 snd_soc_core snd_pcm_dmaengine 5065 2 snd_soc_core,snd_soc_omap snd_pcm 85645 5 snd_soc_davinci_mcasp,snd_soc_core,snd_soc_omap,snd_soc_tlv320aic3x,snd_pcm_ dmaengine snd_timer 17058 1 snd_pcm snd 58469 4 snd_soc_core,snd_timer,snd_pcm,snd_compress soundcore 6861 1 snd omap_aes 13065 0 omap_sham 19198 0 tda998x 11695 1 tilcdc 27899 0 omap_rng 4354 0 drm_kms_helper 106679 3 tda998x,tilcdc rng_core 7266 1 omap_rng uio_pdrv_genirq 3317 0 uio 8350 1 uio_pdrv_genirq leds_gpio 3110 0 -----------------------------------snip (4) debian@arm:~$ aplay sine2s.wav Playing WAVE 'sine2s.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo [ 618.068355] davinci-mcasp 48038000.mcasp: Transmit buffer underflow underrun!!! (at least 0.410 ms long) [ 618.092511] davinci-mcasp 48038000.mcasp: Transmit buffer underflow underrun!!! (at least 11.824 ms long) [ 618.121692] davinci-mcasp 48038000.mcasp: Transmit buffer underflow underrun!!! (at least 3.200 ms long) [ 618.134323] davinci-mcasp 48038000.mcasp: Transmit buffer underflow

1 0

[alsa-devel] [PATCH] ALSA: hda - Add headset mic support for Acer Aspire V5-573G
by Takashi Iwai 19 Jul '15

19 Jul '15

From: Mateusz Sylwestrzak <matisec7(a)gmail.com> Acer Aspire V5 with the ALC282 codec is given the wrong value for the 0x19 PIN by the laptop's BIOS. Overriding it with the correct value adds support for the headset microphone which would not otherwise be visible in the system. The fix is based on commit 7819717b1134 with a similar quirk for Acer Aspire with the ALC269 codec. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=96201 Cc: <stable(a)vger.kernel.org> Signed-off-by: Mateusz Sylwestrzak <matisec7(a)gmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d35cf506a7db..caba66b2311c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5061,7 +5061,7 @@ static const struct hda_fixup alc269_fixups[] = { { 0x14, 0x90170110 }, { 0x17, 0x40000008 }, { 0x18, 0x411111f0 }, - { 0x19, 0x411111f0 }, + { 0x19, 0x01a1913c }, { 0x1a, 0x411111f0 }, { 0x1b, 0x411111f0 }, { 0x1d, 0x40f89b2d }, -- 2.4.5

1 0

[alsa-devel] [PATCH v3 0/2] Add codec machine driver for rockchip platform
by Xing Zheng 19 Jul '15

19 Jul '15

From: zhengxing <zhengxing(a)rock-chips.com> Hi, The simple-card is not common at present, soc maybe need own machine driver for jack detection. Add drivers for two families of rockchip-bases chromebooks. These machine drives don't use simplecard because we need custom jack detection plumbing. - use ts3a227e for ext jack detection with max98090 - call rt5645_set_jack_detect function via rt5645 codec driver Thanks. Changes in v3: Signed-off-by: zhengxing <zhengxing(a)rock-chips.com> zhengxing (2): ASoC: rockchip: Add machine driver for max98090 codec ASoC: rockchip: Add machine driver for rt5645/rt5650 codec .../bindings/sound/rockchip-max98090.txt | 19 ++ .../devicetree/bindings/sound/rockchip-rt5645.txt | 17 ++ sound/soc/rockchip/Kconfig | 19 ++ sound/soc/rockchip/Makefile | 6 + sound/soc/rockchip/rockchip_max98090.c | 236 ++++++++++++++++++++ sound/soc/rockchip/rockchip_rt5645.c | 225 +++++++++++++++++++ 6 files changed, 522 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip-max98090.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip-rt5645.txt create mode 100644 sound/soc/rockchip/rockchip_max98090.c create mode 100644 sound/soc/rockchip/rockchip_rt5645.c -- 1.7.9.5

1 2

[alsa-devel] [PATCH v2 0/2] Add codec machine driver for rockchip platform
by Xing Zheng 19 Jul '15

19 Jul '15

From: zhengxing <zhengxing(a)rock-chips.com> Hi, The simple-card is not common at present, soc maybe need own machine driver for jack detection. Add drivers for two families of rockchip-bases chromebooks. These machine drives don't use simplecard because we need custom jack detection plumbing. - use ts3a227e for ext jack detection with max98090 - call rt5645_set_jack_detect function via rt5645 codec driver Thanks. Changes in v2: Signed-off-by: zhengxing <zhengxing(a)rock-chips.com> zhengxing (2): ASoC: rockchip: Add machine driver for max98090 codec ASoC: rockchip: Add machine driver for rt5645/rt5650 codec .../bindings/sound/rockchip-max98090.txt | 19 ++ .../devicetree/bindings/sound/rockchip-rt5645.txt | 17 ++ sound/soc/rockchip/Kconfig | 19 ++ sound/soc/rockchip/Makefile | 6 + sound/soc/rockchip/rockchip_max98090.c | 247 ++++++++++++++++++++ sound/soc/rockchip/rockchip_rt5645.c | 236 +++++++++++++++++++ 6 files changed, 544 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip-max98090.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip-rt5645.txt create mode 100644 sound/soc/rockchip/rockchip_max98090.c create mode 100644 sound/soc/rockchip/rockchip_rt5645.c -- 1.7.9.5

2 4

Re: [alsa-devel] After kernel upgrade to v ~~> 4.1.0, no audio output unless UNPLUG/RE-PLUGIN of audio jack on card; OK with earlier kernels
by robert.devanna＠nospammail.net 19 Jul '15

19 Jul '15

On Mon, Jul 6, 2015, at 01:46 AM, Takashi Iwai wrote: > It's a known problem with VIA codecs, and the fix was already merged > in 4.2-rc1, the commit 735c75cf4d434862e38c01dcfb2ce8d2fcb9035f. > The next stable 4.1.x kernel should contain the fix. Upgraded to uname -rm 4.1.2-1.g5ccacb4-desktop x86_64 Fyi, the problem remains unchanged. To get sound, a physical unplug/replug is required.

2 1

[alsa-devel] [PATCH 0/2] Add codec machine driver for rockchip platform
by Xing Zheng 18 Jul '15

18 Jul '15

From: zhengxing <zhengxing(a)rock-chips.com> Hi, The simple-card is not common at present, soc maybe need own machine driver for jack detection. Add drivers for two families of rockchip-bases chromebooks. These machine drives don't use simplecard because we need custom jack detection plumbing. - use ts3a227e for ext jack detection with max98090 - call rt5645_set_jack_detect function via rt5645 codec driver Thanks. zhengxing (2): ASoC: rockchip: Add machine driver for max98090 codec ASoC: rockchip: Add machine driver for rt5645/rt5650 codec sound/soc/rockchip/Kconfig | 19 +++ sound/soc/rockchip/Makefile | 6 + sound/soc/rockchip/rockchip_max98090.c | 246 ++++++++++++++++++++++++++++++++ sound/soc/rockchip/rockchip_rt5645.c | 235 ++++++++++++++++++++++++++++++ 4 files changed, 506 insertions(+) create mode 100644 sound/soc/rockchip/rockchip_max98090.c create mode 100644 sound/soc/rockchip/rockchip_rt5645.c -- 1.7.9.5

5 23

[alsa-devel] [PATCH v2] arm64: dts: mt8173: Add afe device node
by Koro Chen 17 Jul '15

17 Jul '15

This adds afe (audio front end) device node to the MT8173 dtsi file. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- This patch is based on Matthias's tree: https://github.com/mbgg/linux-mediatek branch: v4.2-next/arm64 Changes since v1: - change node name to afe: audio-controller@11220000 --- arch/arm64/boot/dts/mediatek/mt8173.dtsi | 32 ++++++++++++++++++++++++++++++++ 1 file changed, 32 insertions(+) diff --git a/arch/arm64/boot/dts/mediatek/mt8173.dtsi b/arch/arm64/boot/dts/mediatek/mt8173.dtsi index 0696f8f..ce9255a 100644 --- a/arch/arm64/boot/dts/mediatek/mt8173.dtsi +++ b/arch/arm64/boot/dts/mediatek/mt8173.dtsi @@ -14,6 +14,7 @@ #include <dt-bindings/clock/mt8173-clk.h> #include <dt-bindings/interrupt-controller/irq.h> #include <dt-bindings/interrupt-controller/arm-gic.h> +#include <dt-bindings/power/mt8173-power.h> #include <dt-bindings/reset-controller/mt8173-resets.h> #include "mt8173-pinfunc.h" @@ -393,6 +394,37 @@ #size-cells = <0>; status = "disabled"; }; + + afe: audio-controller@11220000 { + compatible = "mediatek,mt8173-afe-pcm"; + reg = <0 0x11220000 0 0x1000>; + interrupts = <GIC_SPI 134 IRQ_TYPE_EDGE_FALLING>; + power-domains = <&scpsys MT8173_POWER_DOMAIN_AUDIO>; + clocks = <&infracfg CLK_INFRA_AUDIO>, + <&topckgen CLK_TOP_AUDIO_SEL>, + <&topckgen CLK_TOP_AUD_INTBUS_SEL>, + <&topckgen CLK_TOP_APLL1_DIV0>, + <&topckgen CLK_TOP_APLL2_DIV0>, + <&topckgen CLK_TOP_I2S0_M_SEL>, + <&topckgen CLK_TOP_I2S1_M_SEL>, + <&topckgen CLK_TOP_I2S2_M_SEL>, + <&topckgen CLK_TOP_I2S3_M_SEL>, + <&topckgen CLK_TOP_I2S3_B_SEL>; + clock-names = "infra_sys_audio_clk", + "top_pdn_audio", + "top_pdn_aud_intbus", + "bck0", + "bck1", + "i2s0_m", + "i2s1_m", + "i2s2_m", + "i2s3_m", + "i2s3_b"; + assigned-clocks = <&topckgen CLK_TOP_AUD_1_SEL>, + <&topckgen CLK_TOP_AUD_2_SEL>; + assigned-clock-parents = <&topckgen CLK_TOP_APLL1>, + <&topckgen CLK_TOP_APLL2>; + }; }; }; -- 1.8.1.1.dirty

3 3

[alsa-devel] [PATCH] ALSA: sparc: Add missing kfree in error path
by Christophe JAILLET 17 Jul '15

17 Jul '15

If 'of_ioremap' fails, then 'amd' should be freed, otherwise, there is a memory leak. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/sparc/amd7930.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c index 1b1a89e..784ceb8 100644 --- a/sound/sparc/amd7930.c +++ b/sound/sparc/amd7930.c @@ -956,6 +956,7 @@ static int snd_amd7930_create(struct snd_card *card, if (!amd->regs) { snd_printk(KERN_ERR "amd7930-%d: Unable to map chip registers.\n", dev); + kfree(amd); return -EIO; } -- 2.1.4

2 1

[alsa-devel] [PATCH 0/35] ASoC: rsnd: add CTU/MIX prototype support
by Kuninori Morimoto 17 Jul '15

17 Jul '15

Hi Mark, Simon These patches add CTU/MIX prototype support on Renesas sound card driver. It needs DPCM feature, so it is assuming to be used via rsrc-card sound card. These CTU/MIX is still prototype support. This Mixer works well if sound A and B are using same bit length, same sampling rate. 1) - 2) adds audio prefix support on DT which is needed on DPCM 3) - 29) cleanup/tidyup/workaround/prepare patches for CTU/MIX support 30) - 31) CTU/MIX support on driver 32) - 35) CTU/MIX support on SoC Simon, This time I didn't Cc:ed to SH-ML, since 32) - 35) need Document update (?). It is done on 30) - 31). I can post 32) - 35) if Mark accepted 1) - 31) Kuninori Morimoto (35): 1) ASoC: core: add snd_soc_of_parse_audio_prefix() 2) ASoC: rsrc-card: use snd_soc_of_parse_audio_route/prefix for routing 3) ASoC: rsnd: remove unnecessary 'out of memory' message from SSI 4) ASoC: rsnd: remove unnecessary 'out of memory' message from SRC 5) ASoC: rsnd: remove unnecessary 'out of memory' message from DVC 6) ASoC: rsnd: rename BUSIF_DALIGN to SSI_BUSIF_DALIGN 7) ASoC: rsnd: rename INT_ENABLE to SSI_INT_ENABLE 8) ASoC: rsnd: fixup each module counter on __rsnd_mod_call() 9) ASoC: rsnd: add workaround for SRC sync convert + DVC 10) ASoC: rsnd: rsnd_mod_id() return -1 if mod was NULL 11) ASoC: rsnd: move DVC specific macro into dvc.c 12) ASoC: rsnd: move SRC specific macro into src.c 13) ASoC: rsnd: dvc: make sure DVC soft reset 14) ASoC: rsnd: src: make sure SRC soft reset 15) ASoC: rsnd: enable module multi connection 16) ASoC: rsnd: rename rsnd_path_parse/break() into add/remove 17) ASoC: rsnd: add rsnd_path_parse() for CTU/MIX/DVC route setting 18) ASoC: rsnd: add rsnd_dvc_initialize_lock/unlock() 19) ASoC: rsnd: add rsnd_src_initialize_lock/unlock() 20) ASoC: rsnd: tidyup ADINR function name 21) ASoC: rsnd: add rsnd_get_adinr_chan() 22) ASoC: rsnd: tidyup data align position 23) ASoC: rsnd: show debug message for SSI/SRC/DVC connection 24) ASoC: rsnd: tidyup rsnd_dma_ops definition place 25) ASoC: rsnd: check the Gen1 at the beginning of DVC probe 26) ASoC: rsnd: dma: add DMA name on .ops 27) ASoC: rsnd: add rsnd_io_to_mod() 28) ASoC: rsnd: tidyup SRC position on each code 29) ASoC: rsnd: update Audio DMA path search method 30) ASoC: rsnd: add CTU (Channel Transfer Unit) prototype support 31) ASoC: rsnd: add MIX (Mixer) support 32) ARM: shmobile: r8a7790: Add Audio CTU support on DTSI 33) ARM: shmobile: r8a7790: Add Audio MIX support on DTSI 34) ARM: shmobile: r8a7791: Add Audio CTU support on DTSI 35) ARM: shmobile: r8a7791: Add Audio MIX support on DTSI Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 22 +++++++++++ Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt | 7 ++++ arch/arm/boot/dts/r8a7790.dtsi | 23 ++++++++++++ arch/arm/boot/dts/r8a7791.dtsi | 23 ++++++++++++ include/dt-bindings/clock/r8a7790-clock.h | 2 + include/dt-bindings/clock/r8a7791-clock.h | 2 + include/sound/rcar_snd.h | 14 +++++++ include/sound/soc.h | 4 ++ sound/soc/sh/rcar/Makefile | 2 +- sound/soc/sh/rcar/core.c | 194 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++------------ sound/soc/sh/rcar/ctu.c | 171 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/dma.c | 126 ++++++++++++++++++++++++++++++++++++++++++-------------------- sound/soc/sh/rcar/dvc.c | 73 ++++++++++++++++-------------------- sound/soc/sh/rcar/gen.c | 17 ++++++++- sound/soc/sh/rcar/mix.c | 200 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 107 ++++++++++++++++++++++++++++++++++++---------------- sound/soc/sh/rcar/rsrc-card.c | 22 +++++++++-- sound/soc/sh/rcar/src.c | 117 +++++++++++++++++++++++++++++++-------------------------- sound/soc/sh/rcar/ssi.c | 4 +- sound/soc/soc-core.c | 20 ++++++++++ 20 files changed, 952 insertions(+), 198 deletions(-) Best regards --- Kuninori Morimoto

2 67

[alsa-devel] [PATCH] speaker-test: Add option to specify signal scale
by Julian Scheel 17 Jul '15

17 Jul '15

Allow generated signals (sine and noise) to have a specified signal scale instead of using hardcoded limits of 80%. This can be handy for debugging, ie when analysing clipping issues with audio drivers. Signed-off-by: Julian Scheel <julian(a)jusst.de> --- speaker-test/speaker-test.c | 30 ++++++++++++++++++------------ 1 file changed, 18 insertions(+), 12 deletions(-) diff --git a/speaker-test/speaker-test.c b/speaker-test/speaker-test.c index ccf2671..25970af 100644 --- a/speaker-test/speaker-test.c +++ b/speaker-test/speaker-test.c @@ -100,6 +100,7 @@ static unsigned int period_time = 0; /* period time in us */ static unsigned int nperiods = 4; /* number of periods */ static double freq = 440.0; /* sinusoidal wave frequency in Hz */ static int test_type = TEST_PINK_NOISE; /* Test type. 1 = noise, 2 = sine wave */ +static float generator_scale = 0.8; /* Scale to use for sine volume */ static pink_noise_t pink; static snd_pcm_uframes_t buffer_size; static snd_pcm_uframes_t period_size; @@ -306,7 +307,7 @@ static void generate_sine(uint8_t *frames, int channel, int count, double *_phas switch (format) { case SND_PCM_FORMAT_S8: if (chn==channel) { - res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * 0x03fffffff; /* Don't use MAX volume */ + res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * generator_scale * 0x7fffffff; ires = res; *samp8++ = ires >> 24; } else { @@ -315,7 +316,7 @@ static void generate_sine(uint8_t *frames, int channel, int count, double *_phas break; case SND_PCM_FORMAT_S16_LE: if (chn==channel) { - res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * 0x03fffffff; /* Don't use MAX volume */ + res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * generator_scale * 0x7fffffff; ires = res; *samp16++ = LE_SHORT(ires >> 16); } else { @@ -324,7 +325,7 @@ static void generate_sine(uint8_t *frames, int channel, int count, double *_phas break; case SND_PCM_FORMAT_S16_BE: if (chn==channel) { - res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * 0x03fffffff; /* Don't use MAX volume */ + res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * generator_scale * 0x7fffffff; ires = res; *samp16++ = BE_SHORT(ires >> 16); } else { @@ -333,7 +334,7 @@ static void generate_sine(uint8_t *frames, int channel, int count, double *_phas break; case SND_PCM_FORMAT_FLOAT_LE: if (chn==channel) { - res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * 0.75 ; /* Don't use MAX volume */ + res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * generator_scale; fres = res; *samp_f++ = fres; } else { @@ -342,7 +343,7 @@ static void generate_sine(uint8_t *frames, int channel, int count, double *_phas break; case SND_PCM_FORMAT_S32_LE: if (chn==channel) { - res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * 0x03fffffff; /* Don't use MAX volume */ + res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * generator_scale * 0x7fffffff; ires = res; *samp32++ = LE_INT(ires); } else { @@ -351,7 +352,7 @@ static void generate_sine(uint8_t *frames, int channel, int count, double *_phas break; case SND_PCM_FORMAT_S32_BE: if (chn==channel) { - res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * 0x03fffffff; /* Don't use MAX volume */ + res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * generator_scale * 0x7fffffff; ires = res; *samp32++ = BE_INT(ires); } else { @@ -389,7 +390,7 @@ static void generate_pink_noise( uint8_t *frames, int channel, int count) { switch (format) { case SND_PCM_FORMAT_S8: if (chn==channel) { - res = generate_pink_noise_sample(&pink) * 0x03fffffff; /* Don't use MAX volume */ + res = generate_pink_noise_sample(&pink) * generator_scale * 0x07fffffff; ires = res; *samp8++ = ires >> 24; } else { @@ -398,7 +399,7 @@ static void generate_pink_noise( uint8_t *frames, int channel, int count) { break; case SND_PCM_FORMAT_S16_LE: if (chn==channel) { - res = generate_pink_noise_sample(&pink) * 0x03fffffff; /* Don't use MAX volume */ + res = generate_pink_noise_sample(&pink) * generator_scale * 0x07fffffff; ires = res; *samp16++ = LE_SHORT(ires >> 16); } else { @@ -407,7 +408,7 @@ static void generate_pink_noise( uint8_t *frames, int channel, int count) { break; case SND_PCM_FORMAT_S16_BE: if (chn==channel) { - res = generate_pink_noise_sample(&pink) * 0x03fffffff; /* Don't use MAX volume */ + res = generate_pink_noise_sample(&pink) * generator_scale * 0x07fffffff; ires = res; *samp16++ = BE_SHORT(ires >> 16); } else { @@ -416,7 +417,7 @@ static void generate_pink_noise( uint8_t *frames, int channel, int count) { break; case SND_PCM_FORMAT_S32_LE: if (chn==channel) { - res = generate_pink_noise_sample(&pink) * 0x03fffffff; /* Don't use MAX volume */ + res = generate_pink_noise_sample(&pink) * generator_scale * 0x07fffffff; ires = res; *samp32++ = LE_INT(ires); } else { @@ -425,7 +426,7 @@ static void generate_pink_noise( uint8_t *frames, int channel, int count) { break; case SND_PCM_FORMAT_S32_BE: if (chn==channel) { - res = generate_pink_noise_sample(&pink) * 0x03fffffff; /* Don't use MAX volume */ + res = generate_pink_noise_sample(&pink) * generator_scale * 0x07fffffff; ires = res; *samp32++ = BE_INT(ires); } else { @@ -1017,6 +1018,7 @@ static void help(void) "-W,--wavdir Specify the directory containing WAV files\n" "-m,--chmap Specify the channel map to override\n" "-X,--force-frequency force frequencies outside the 30-8000hz range\n" + "-S,--scale Scale of generated test tones in percent (default=80)\n" "\n")); printf(_("Recognized sample formats are:")); for (fmt = supported_formats; *fmt >= 0; fmt++) { @@ -1060,6 +1062,7 @@ int main(int argc, char *argv[]) { {"wavdir", 1, NULL, 'W'}, {"debug", 0, NULL, 'd'}, {"force-frequency", 0, NULL, 'X'}, + {"scale", 1, NULL, 'S'}, #ifdef CONFIG_SUPPORT_CHMAP {"chmap", 1, NULL, 'm'}, #endif @@ -1081,7 +1084,7 @@ int main(int argc, char *argv[]) { while (1) { int c; - if ((c = getopt_long(argc, argv, "hD:r:c:f:F:b:p:P:t:l:s:w:W:d:X" + if ((c = getopt_long(argc, argv, "hD:r:c:f:F:b:p:P:t:l:s:w:W:d:XS:" #ifdef CONFIG_SUPPORT_CHMAP "m:" #endif @@ -1182,6 +1185,9 @@ int main(int argc, char *argv[]) { chmap = optarg; break; #endif + case 'S': + generator_scale = atoi(optarg) / 100.0; + break; default: fprintf(stderr, _("Unknown option '%c'\n"), c); exit(EXIT_FAILURE); -- 2.4.3

2 2

[alsa-devel] [PATCH 1/2 V5] regmap: Use reg_sequence for multi_reg_write / register_patch
by Nariman Poushin 17 Jul '15

17 Jul '15

Separate the functionality using sequences of register writes from the functions that take register defaults. This change renames the arguments in order to support the extension of reg_sequence to take an optional delay to be applied after any given register in a sequence is written. This avoids adding an int to all register defaults, which could substantially increase memory usage for regmaps with large default tables. This also updates all the clients of multi_reg_write/register_patch. Signed-off-by: Nariman Poushin <nariman(a)opensource.wolfsonmicro.com> --- drivers/base/regmap/internal.h | 2 +- drivers/base/regmap/regmap.c | 22 +++++++++++----------- drivers/gpu/drm/i2c/adv7511.c | 2 +- drivers/input/misc/drv260x.c | 6 +++--- drivers/input/misc/drv2665.c | 2 +- drivers/input/misc/drv2667.c | 4 ++-- drivers/mfd/arizona-core.c | 2 +- drivers/mfd/twl6040.c | 2 +- drivers/mfd/wm5102-tables.c | 6 +++--- drivers/mfd/wm5110-tables.c | 6 +++--- drivers/mfd/wm8994-core.c | 8 ++++---- drivers/mfd/wm8997-tables.c | 2 +- include/linux/regmap.h | 17 ++++++++++++++--- sound/soc/codecs/arizona.c | 2 +- sound/soc/codecs/cs35l32.c | 2 +- sound/soc/codecs/cs42l52.c | 2 +- sound/soc/codecs/da7210.c | 4 ++-- sound/soc/codecs/rt5640.c | 2 +- sound/soc/codecs/rt5645.c | 4 ++-- sound/soc/codecs/rt5651.c | 2 +- sound/soc/codecs/rt5670.c | 2 +- sound/soc/codecs/rt5677.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/wm2200.c | 2 +- sound/soc/codecs/wm5100.c | 2 +- sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8993.c | 2 +- 27 files changed, 62 insertions(+), 51 deletions(-) diff --git a/drivers/base/regmap/internal.h b/drivers/base/regmap/internal.h index b2b2849..873ddf9 100644 --- a/drivers/base/regmap/internal.h +++ b/drivers/base/regmap/internal.h @@ -136,7 +136,7 @@ struct regmap { /* if set, the HW registers are known to match map->reg_defaults */ bool no_sync_defaults; - struct reg_default *patch; + struct reg_sequence *patch; int patch_regs; /* if set, converts bulk rw to single rw */ diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c index dd63bcb..0a849ee 100644 --- a/drivers/base/regmap/regmap.c +++ b/drivers/base/regmap/regmap.c @@ -1755,7 +1755,7 @@ EXPORT_SYMBOL_GPL(regmap_bulk_write); * relative. The page register has been written if that was neccessary. */ static int _regmap_raw_multi_reg_write(struct regmap *map, - const struct reg_default *regs, + const struct reg_sequence *regs, size_t num_regs) { int ret; @@ -1812,12 +1812,12 @@ static unsigned int _regmap_register_page(struct regmap *map, } static int _regmap_range_multi_paged_reg_write(struct regmap *map, - struct reg_default *regs, + struct reg_sequence *regs, size_t num_regs) { int ret; int i, n; - struct reg_default *base; + struct reg_sequence *base; unsigned int this_page = 0; /* * the set of registers are not neccessarily in order, but @@ -1855,7 +1855,7 @@ static int _regmap_range_multi_paged_reg_write(struct regmap *map, } static int _regmap_multi_reg_write(struct regmap *map, - const struct reg_default *regs, + const struct reg_sequence *regs, size_t num_regs) { int i; @@ -1907,8 +1907,8 @@ static int _regmap_multi_reg_write(struct regmap *map, struct regmap_range_node *range; range = _regmap_range_lookup(map, reg); if (range) { - size_t len = sizeof(struct reg_default)*num_regs; - struct reg_default *base = kmemdup(regs, len, + size_t len = sizeof(struct reg_sequence)*num_regs; + struct reg_sequence *base = kmemdup(regs, len, GFP_KERNEL); if (!base) return -ENOMEM; @@ -1941,7 +1941,7 @@ static int _regmap_multi_reg_write(struct regmap *map, * A value of zero will be returned on success, a negative errno will be * returned in error cases. */ -int regmap_multi_reg_write(struct regmap *map, const struct reg_default *regs, +int regmap_multi_reg_write(struct regmap *map, const struct reg_sequence *regs, int num_regs) { int ret; @@ -1974,7 +1974,7 @@ EXPORT_SYMBOL_GPL(regmap_multi_reg_write); * be returned in error cases. */ int regmap_multi_reg_write_bypassed(struct regmap *map, - const struct reg_default *regs, + const struct reg_sequence *regs, int num_regs) { int ret; @@ -2587,10 +2587,10 @@ EXPORT_SYMBOL_GPL(regmap_async_complete); * The caller must ensure that this function cannot be called * concurrently with either itself or regcache_sync(). */ -int regmap_register_patch(struct regmap *map, const struct reg_default *regs, +int regmap_register_patch(struct regmap *map, const struct reg_sequence *regs, int num_regs) { - struct reg_default *p; + struct reg_sequence *p; int ret; bool bypass; @@ -2599,7 +2599,7 @@ int regmap_register_patch(struct regmap *map, const struct reg_default *regs, return 0; p = krealloc(map->patch, - sizeof(struct reg_default) * (map->patch_regs + num_regs), + sizeof(struct reg_sequence) * (map->patch_regs + num_regs), GFP_KERNEL); if (p) { memcpy(p + map->patch_regs, regs, num_regs * sizeof(*regs)); diff --git a/drivers/gpu/drm/i2c/adv7511.c b/drivers/gpu/drm/i2c/adv7511.c index 2aaa3c8..00416f2 100644 --- a/drivers/gpu/drm/i2c/adv7511.c +++ b/drivers/gpu/drm/i2c/adv7511.c @@ -54,7 +54,7 @@ static struct adv7511 *encoder_to_adv7511(struct drm_encoder *encoder) } /* ADI recommended values for proper operation. */ -static const struct reg_default adv7511_fixed_registers[] = { +static const struct reg_sequence adv7511_fixed_registers[] = { { 0x98, 0x03 }, { 0x9a, 0xe0 }, { 0x9c, 0x30 }, diff --git a/drivers/input/misc/drv260x.c b/drivers/input/misc/drv260x.c index e5d60ec..f5c9cf2 100644 --- a/drivers/input/misc/drv260x.c +++ b/drivers/input/misc/drv260x.c @@ -313,14 +313,14 @@ static void drv260x_close(struct input_dev *input) gpiod_set_value(haptics->enable_gpio, 0); } -static const struct reg_default drv260x_lra_cal_regs[] = { +static const struct reg_sequence drv260x_lra_cal_regs[] = { { DRV260X_MODE, DRV260X_AUTO_CAL }, { DRV260X_CTRL3, DRV260X_NG_THRESH_2 }, { DRV260X_FEEDBACK_CTRL, DRV260X_FB_REG_LRA_MODE | DRV260X_BRAKE_FACTOR_4X | DRV260X_LOOP_GAIN_HIGH }, }; -static const struct reg_default drv260x_lra_init_regs[] = { +static const struct reg_sequence drv260x_lra_init_regs[] = { { DRV260X_MODE, DRV260X_RT_PLAYBACK }, { DRV260X_A_TO_V_CTRL, DRV260X_AUDIO_HAPTICS_PEAK_20MS | DRV260X_AUDIO_HAPTICS_FILTER_125HZ }, @@ -337,7 +337,7 @@ static const struct reg_default drv260x_lra_init_regs[] = { { DRV260X_CTRL4, DRV260X_AUTOCAL_TIME_500MS }, }; -static const struct reg_default drv260x_erm_cal_regs[] = { +static const struct reg_sequence drv260x_erm_cal_regs[] = { { DRV260X_MODE, DRV260X_AUTO_CAL }, { DRV260X_A_TO_V_MIN_INPUT, DRV260X_AUDIO_HAPTICS_MIN_IN_VOLT }, { DRV260X_A_TO_V_MAX_INPUT, DRV260X_AUDIO_HAPTICS_MAX_IN_VOLT }, diff --git a/drivers/input/misc/drv2665.c b/drivers/input/misc/drv2665.c index 0afaa33..924456e 100644 --- a/drivers/input/misc/drv2665.c +++ b/drivers/input/misc/drv2665.c @@ -132,7 +132,7 @@ static void drv2665_close(struct input_dev *input) "Failed to enter standby mode: %d\n", error); } -static const struct reg_default drv2665_init_regs[] = { +static const struct reg_sequence drv2665_init_regs[] = { { DRV2665_CTRL_2, 0 | DRV2665_10_MS_IDLE_TOUT }, { DRV2665_CTRL_1, DRV2665_25_VPP_GAIN }, }; diff --git a/drivers/input/misc/drv2667.c b/drivers/input/misc/drv2667.c index fc0fddf..047136a 100644 --- a/drivers/input/misc/drv2667.c +++ b/drivers/input/misc/drv2667.c @@ -262,14 +262,14 @@ static void drv2667_close(struct input_dev *input) "Failed to enter standby mode: %d\n", error); } -static const struct reg_default drv2667_init_regs[] = { +static const struct reg_sequence drv2667_init_regs[] = { { DRV2667_CTRL_2, 0 }, { DRV2667_CTRL_1, DRV2667_25_VPP_GAIN }, { DRV2667_WV_SEQ_0, 1 }, { DRV2667_WV_SEQ_1, 0 } }; -static const struct reg_default drv2667_page1_init[] = { +static const struct reg_sequence drv2667_page1_init[] = { { DRV2667_RAM_HDR_SZ, 0x05 }, { DRV2667_RAM_START_HI, 0x80 }, { DRV2667_RAM_START_LO, 0x06 }, diff --git a/drivers/mfd/arizona-core.c b/drivers/mfd/arizona-core.c index bebf58a..66d50be 100644 --- a/drivers/mfd/arizona-core.c +++ b/drivers/mfd/arizona-core.c @@ -392,7 +392,7 @@ err: * Register patch to some of the CODECs internal write sequences * to ensure a clean exit from the low power sleep state. */ -static const struct reg_default wm5110_sleep_patch[] = { +static const struct reg_sequence wm5110_sleep_patch[] = { { 0x337A, 0xC100 }, { 0x337B, 0x0041 }, { 0x3300, 0xA210 }, diff --git a/drivers/mfd/twl6040.c b/drivers/mfd/twl6040.c index c5265c1..583dc33 100644 --- a/drivers/mfd/twl6040.c +++ b/drivers/mfd/twl6040.c @@ -86,7 +86,7 @@ static const struct reg_default twl6040_defaults[] = { { 0x2E, 0x00 }, /* REG_STATUS (ro) */ }; -static struct reg_default twl6040_patch[] = { +static struct reg_sequence twl6040_patch[] = { /* * Select I2C bus access to dual access registers * Interrupt register is cleared on read diff --git a/drivers/mfd/wm5102-tables.c b/drivers/mfd/wm5102-tables.c index aeae6ec..423fb37 100644 --- a/drivers/mfd/wm5102-tables.c +++ b/drivers/mfd/wm5102-tables.c @@ -21,7 +21,7 @@ #define WM5102_NUM_AOD_ISR 2 #define WM5102_NUM_ISR 5 -static const struct reg_default wm5102_reva_patch[] = { +static const struct reg_sequence wm5102_reva_patch[] = { { 0x80, 0x0003 }, { 0x221, 0x0090 }, { 0x211, 0x0014 }, @@ -57,7 +57,7 @@ static const struct reg_default wm5102_reva_patch[] = { { 0x80, 0x0000 }, }; -static const struct reg_default wm5102_revb_patch[] = { +static const struct reg_sequence wm5102_revb_patch[] = { { 0x19, 0x0001 }, { 0x80, 0x0003 }, { 0x081, 0xE022 }, @@ -80,7 +80,7 @@ static const struct reg_default wm5102_revb_patch[] = { /* We use a function so we can use ARRAY_SIZE() */ int wm5102_patch(struct arizona *arizona) { - const struct reg_default *wm5102_patch; + const struct reg_sequence *wm5102_patch; int patch_size; switch (arizona->rev) { diff --git a/drivers/mfd/wm5110-tables.c b/drivers/mfd/wm5110-tables.c index 12cad94..26ce14f 100644 --- a/drivers/mfd/wm5110-tables.c +++ b/drivers/mfd/wm5110-tables.c @@ -21,7 +21,7 @@ #define WM5110_NUM_AOD_ISR 2 #define WM5110_NUM_ISR 5 -static const struct reg_default wm5110_reva_patch[] = { +static const struct reg_sequence wm5110_reva_patch[] = { { 0x80, 0x3 }, { 0x44, 0x20 }, { 0x45, 0x40 }, @@ -134,7 +134,7 @@ static const struct reg_default wm5110_reva_patch[] = { { 0x209, 0x002A }, }; -static const struct reg_default wm5110_revb_patch[] = { +static const struct reg_sequence wm5110_revb_patch[] = { { 0x80, 0x3 }, { 0x36e, 0x0210 }, { 0x370, 0x0210 }, @@ -224,7 +224,7 @@ static const struct reg_default wm5110_revb_patch[] = { { 0x80, 0x0 }, }; -static const struct reg_default wm5110_revd_patch[] = { +static const struct reg_sequence wm5110_revd_patch[] = { { 0x80, 0x3 }, { 0x80, 0x3 }, { 0x393, 0x27 }, diff --git a/drivers/mfd/wm8994-core.c b/drivers/mfd/wm8994-core.c index 53ae5af..0f4169a 100644 --- a/drivers/mfd/wm8994-core.c +++ b/drivers/mfd/wm8994-core.c @@ -243,21 +243,21 @@ static int wm8994_ldo_in_use(struct wm8994_pdata *pdata, int ldo) } #endif -static const struct reg_default wm8994_revc_patch[] = { +static const struct reg_sequence wm8994_revc_patch[] = { { 0x102, 0x3 }, { 0x56, 0x3 }, { 0x817, 0x0 }, { 0x102, 0x0 }, }; -static const struct reg_default wm8958_reva_patch[] = { +static const struct reg_sequence wm8958_reva_patch[] = { { 0x102, 0x3 }, { 0xcb, 0x81 }, { 0x817, 0x0 }, { 0x102, 0x0 }, }; -static const struct reg_default wm1811_reva_patch[] = { +static const struct reg_sequence wm1811_reva_patch[] = { { 0x102, 0x3 }, { 0x56, 0xc07 }, { 0x5d, 0x7e }, @@ -326,7 +326,7 @@ static int wm8994_device_init(struct wm8994 *wm8994, int irq) { struct wm8994_pdata *pdata; struct regmap_config *regmap_config; - const struct reg_default *regmap_patch = NULL; + const struct reg_sequence *regmap_patch = NULL; const char *devname; int ret, i, patch_regs = 0; int pulls = 0; diff --git a/drivers/mfd/wm8997-tables.c b/drivers/mfd/wm8997-tables.c index c0c25d75..cab2c68 100644 --- a/drivers/mfd/wm8997-tables.c +++ b/drivers/mfd/wm8997-tables.c @@ -17,7 +17,7 @@ #include "arizona.h" -static const struct reg_default wm8997_reva_patch[] = { +static const struct reg_sequence wm8997_reva_patch[] = { { 0x80, 0x0003 }, { 0x214, 0x0008 }, { 0x458, 0x0000 }, diff --git a/include/linux/regmap.h b/include/linux/regmap.h index 519c962..4a67590 100644 --- a/include/linux/regmap.h +++ b/include/linux/regmap.h @@ -50,6 +50,17 @@ struct reg_default { unsigned int def; }; +/** + * Register/value pairs for sequences of writes + * + * @reg: Register address. + * @def: Register value. + */ +struct reg_sequence { + unsigned int reg; + unsigned int def; +}; + #ifdef CONFIG_REGMAP enum regmap_endian { @@ -410,10 +421,10 @@ int regmap_raw_write(struct regmap *map, unsigned int reg, const void *val, size_t val_len); int regmap_bulk_write(struct regmap *map, unsigned int reg, const void *val, size_t val_count); -int regmap_multi_reg_write(struct regmap *map, const struct reg_default *regs, +int regmap_multi_reg_write(struct regmap *map, const struct reg_sequence *regs, int num_regs); int regmap_multi_reg_write_bypassed(struct regmap *map, - const struct reg_default *regs, + const struct reg_sequence *regs, int num_regs); int regmap_raw_write_async(struct regmap *map, unsigned int reg, const void *val, size_t val_len); @@ -452,7 +463,7 @@ void regcache_mark_dirty(struct regmap *map); bool regmap_check_range_table(struct regmap *map, unsigned int reg, const struct regmap_access_table *table); -int regmap_register_patch(struct regmap *map, const struct reg_default *regs, +int regmap_register_patch(struct regmap *map, const struct reg_sequence *regs, int num_regs); int regmap_parse_val(struct regmap *map, const void *buf, unsigned int *val); diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 802e05e..5edd33f 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1366,7 +1366,7 @@ static void arizona_wm5102_set_dac_comp(struct snd_soc_codec *codec, { struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); struct arizona *arizona = priv->arizona; - struct reg_default dac_comp[] = { + struct reg_sequence dac_comp[] = { { 0x80, 0x3 }, { ARIZONA_DAC_COMP_1, 0 }, { ARIZONA_DAC_COMP_2, 0 }, diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 8f40025..2813a1b 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -276,7 +276,7 @@ static const struct snd_soc_codec_driver soc_codec_dev_cs35l32 = { }; /* Current and threshold powerup sequence Pg37 in datasheet */ -static const struct reg_default cs35l32_monitor_patch[] = { +static const struct reg_sequence cs35l32_monitor_patch[] = { { 0x00, 0x99 }, { 0x48, 0x17 }, diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 4de52c9..8b2d059 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1118,7 +1118,7 @@ static const struct snd_soc_codec_driver soc_codec_dev_cs42l52 = { }; /* Current and threshold powerup sequence Pg37 */ -static const struct reg_default cs42l52_threshold_patch[] = { +static const struct reg_sequence cs42l52_threshold_patch[] = { { 0x00, 0x99 }, { 0x3E, 0xBA }, diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 21810e5..bf0fb3d 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -1182,7 +1182,7 @@ static struct snd_soc_codec_driver soc_codec_dev_da7210 = { #if IS_ENABLED(CONFIG_I2C) -static struct reg_default da7210_regmap_i2c_patch[] = { +static struct reg_sequence da7210_regmap_i2c_patch[] = { /* System controller master disable */ { DA7210_STARTUP1, 0x00 }, @@ -1269,7 +1269,7 @@ static struct i2c_driver da7210_i2c_driver = { #if defined(CONFIG_SPI_MASTER) -static struct reg_default da7210_regmap_spi_patch[] = { +static struct reg_sequence da7210_regmap_spi_patch[] = { /* Dummy read to give two pulses over nCS for SPI */ { DA7210_AUX2, 0x00 }, { DA7210_AUX2, 0x00 }, diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 9bc78e5..1ed1f88 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -51,7 +51,7 @@ static const struct regmap_range_cfg rt5640_ranges[] = { .window_len = 0x1, }, }; -static const struct reg_default init_list[] = { +static const struct reg_sequence init_list[] = { {RT5640_PR_BASE + 0x3d, 0x3600}, {RT5640_PR_BASE + 0x12, 0x0aa8}, {RT5640_PR_BASE + 0x14, 0x0aaa}, diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9ce311e..c0f4be4 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -54,7 +54,7 @@ static const struct regmap_range_cfg rt5645_ranges[] = { }, }; -static const struct reg_default init_list[] = { +static const struct reg_sequence init_list[] = { {RT5645_PR_BASE + 0x3d, 0x3600}, {RT5645_PR_BASE + 0x1c, 0xfd20}, {RT5645_PR_BASE + 0x20, 0x611f}, @@ -63,7 +63,7 @@ static const struct reg_default init_list[] = { }; #define RT5645_INIT_REG_LEN ARRAY_SIZE(init_list) -static const struct reg_default rt5650_init_list[] = { +static const struct reg_sequence rt5650_init_list[] = { {0xf6, 0x0100}, }; diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index a3506e1..db9b866 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -46,7 +46,7 @@ static const struct regmap_range_cfg rt5651_ranges[] = { .window_len = 0x1, }, }; -static struct reg_default init_list[] = { +static struct reg_sequence init_list[] = { {RT5651_PR_BASE + 0x3d, 0x3e00}, }; diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index a9123d4..462a91f 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -51,7 +51,7 @@ static const struct regmap_range_cfg rt5670_ranges[] = { .window_len = 0x1, }, }; -static const struct reg_default init_list[] = { +static const struct reg_sequence init_list[] = { { RT5670_PR_BASE + 0x14, 0x9a8a }, { RT5670_PR_BASE + 0x38, 0x3ba1 }, { RT5670_PR_BASE + 0x3d, 0x3640 }, diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 31d969a..b897752 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -54,7 +54,7 @@ static const struct regmap_range_cfg rt5677_ranges[] = { }, }; -static const struct reg_default init_list[] = { +static const struct reg_sequence init_list[] = { {RT5677_ASRC_12, 0x0018}, {RT5677_PR_BASE + 0x3d, 0x364d}, {RT5677_PR_BASE + 0x17, 0x4fc0}, diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index a7cf19b..83ae1eb 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1668,7 +1668,7 @@ static const struct i2c_device_id aic3x_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id); -static const struct reg_default aic3007_class_d[] = { +static const struct reg_sequence aic3007_class_d[] = { /* Class-D speaker driver init; datasheet p. 46 */ { AIC3X_PAGE_SELECT, 0x0D }, { 0xD, 0x0D }, diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index c830832..6c60792 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -897,7 +897,7 @@ static bool wm2200_readable_register(struct device *dev, unsigned int reg) } } -static const struct reg_default wm2200_reva_patch[] = { +static const struct reg_sequence wm2200_reva_patch[] = { { 0x07, 0x0003 }, { 0x102, 0x0200 }, { 0x203, 0x0084 }, diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 4c10cd8..26d79bb 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1247,7 +1247,7 @@ static const struct snd_soc_dapm_route wm5100_dapm_routes[] = { { "PWM2", NULL, "PWM2 Driver" }, }; -static const struct reg_default wm5100_reva_patches[] = { +static const struct reg_sequence wm5100_reva_patches[] = { { WM5100_AUDIO_IF_1_10, 0 }, { WM5100_AUDIO_IF_1_11, 1 }, { WM5100_AUDIO_IF_1_12, 2 }, diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index c5748fd..05492e8 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3495,7 +3495,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8962 = { }; /* Improve power consumption for IN4 DC measurement mode */ -static const struct reg_default wm8962_dc_measure[] = { +static const struct reg_sequence wm8962_dc_measure[] = { { 0xfd, 0x1 }, { 0xcc, 0x40 }, { 0xfd, 0 }, diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 8a8db86..52ec64d 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1595,7 +1595,7 @@ static int wm8993_resume(struct snd_soc_codec *codec) #endif /* Tune DC servo configuration */ -static struct reg_default wm8993_regmap_patch[] = { +static struct reg_sequence wm8993_regmap_patch[] = { { 0x44, 3 }, { 0x56, 3 }, { 0x44, 0 }, -- 2.1.4

2 3

[alsa-devel] Applied "ASoC: wm5110: Use reg_sequence for multi_reg_write/register_patch" to the asoc tree
by Mark Brown 17 Jul '15

17 Jul '15

The patch ASoC: wm5110: Use reg_sequence for multi_reg_write/register_patch has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From dc606e05468fc5e3b125bffdd54253f853bfbb8e Mon Sep 17 00:00:00 2001 From: Nariman Poushin <nariman(a)opensource.wolfsonmicro.com> Date: Fri, 17 Jul 2015 15:11:10 +0100 Subject: [PATCH] ASoC: wm5110: Use reg_sequence for multi_reg_write/register_patch Introduced by: commit 8019ff6cfc04 ("regmap: Use reg_sequence for multi_reg_write / register_patch") Interacting with: commit d1acd31883d7 ("ASoC: wm5110: Add special DRE on/off handling for the headphone path") Signed-off-by: Nariman Poushin <nariman(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wm5110.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index b5c201b1e454..1f29991c1f71 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -185,7 +185,7 @@ static int wm5110_sysclk_ev(struct snd_soc_dapm_widget *w, return 0; } -static const struct reg_default wm5110_no_dre_left_enable[] = { +static const struct reg_sequence wm5110_no_dre_left_enable[] = { { 0x3024, 0xE410 }, { 0x3025, 0x0056 }, { 0x301B, 0x0224 }, @@ -203,7 +203,7 @@ static const struct reg_default wm5110_no_dre_left_enable[] = { { 0x3039, 0x3080 }, }; -static const struct reg_default wm5110_dre_left_enable[] = { +static const struct reg_sequence wm5110_dre_left_enable[] = { { 0x3024, 0x0231 }, { 0x3025, 0x0B00 }, { 0x301B, 0x0227 }, @@ -221,7 +221,7 @@ static const struct reg_default wm5110_dre_left_enable[] = { { 0x3039, 0x0B00 }, }; -static const struct reg_default wm5110_no_dre_right_enable[] = { +static const struct reg_sequence wm5110_no_dre_right_enable[] = { { 0x3074, 0xE414 }, { 0x3075, 0x0056 }, { 0x306B, 0x0224 }, @@ -239,7 +239,7 @@ static const struct reg_default wm5110_no_dre_right_enable[] = { { 0x3089, 0x3080 }, }; -static const struct reg_default wm5110_dre_right_enable[] = { +static const struct reg_sequence wm5110_dre_right_enable[] = { { 0x3074, 0x0231 }, { 0x3075, 0x0B00 }, { 0x306B, 0x0227 }, @@ -263,7 +263,7 @@ static int wm5110_hp_pre_enable(struct snd_soc_dapm_widget *w) struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); struct arizona *arizona = priv->arizona; unsigned int val = snd_soc_read(codec, ARIZONA_DRE_ENABLE); - const struct reg_default *wseq; + const struct reg_sequence *wseq; int nregs; switch (w->shift) { @@ -354,7 +354,7 @@ static int wm5110_hp_ev(struct snd_soc_dapm_widget *w, static int wm5110_clear_pga_volume(struct arizona *arizona, int output) { - struct reg_default clear_pga = { + struct reg_sequence clear_pga = { ARIZONA_OUTPUT_PATH_CONFIG_1L + output * 4, 0x80 }; int ret; -- 2.1.4

1 0

[alsa-devel] [PATCH] ALSA: hda - Check the return value from pm_runtime_get/put*()
by Takashi Iwai 17 Jul '15

17 Jul '15

This patch changes the return type of snd_hdac_power_up/down() and variants to pass the error code from the underlying pm_runtime_get/put() calls. Currently they are ignored, but in most places, these should be handled properly. As an example, the regmap handler is updated to check the return value and accesses the register only when the wakeup succeeds. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hdaudio.h | 16 ++++++++-------- sound/hda/hdac_device.c | 26 ++++++++++++++++++-------- sound/hda/hdac_regmap.c | 10 ++++++---- 3 files changed, 32 insertions(+), 20 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 4caf1fde8a4f..288c7fa509d8 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -164,15 +164,15 @@ static inline int snd_hdac_read_parm(struct hdac_device *codec, hda_nid_t nid, } #ifdef CONFIG_PM -void snd_hdac_power_up(struct hdac_device *codec); -void snd_hdac_power_down(struct hdac_device *codec); -void snd_hdac_power_up_pm(struct hdac_device *codec); -void snd_hdac_power_down_pm(struct hdac_device *codec); +int snd_hdac_power_up(struct hdac_device *codec); +int snd_hdac_power_down(struct hdac_device *codec); +int snd_hdac_power_up_pm(struct hdac_device *codec); +int snd_hdac_power_down_pm(struct hdac_device *codec); #else -static inline void snd_hdac_power_up(struct hdac_device *codec) {} -static inline void snd_hdac_power_down(struct hdac_device *codec) {} -static inline void snd_hdac_power_up_pm(struct hdac_device *codec) {} -static inline void snd_hdac_power_down_pm(struct hdac_device *codec) {} +static inline int snd_hdac_power_up(struct hdac_device *codec) { return 0; } +static inline int snd_hdac_power_down(struct hdac_device *codec) { return 0; } +static inline int snd_hdac_power_up_pm(struct hdac_device *codec) { return 0; } +static inline int snd_hdac_power_down_pm(struct hdac_device *codec) { return 0; } #endif /* diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index cdee7103f649..df7039ebe1fa 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -501,23 +501,27 @@ EXPORT_SYMBOL_GPL(snd_hdac_get_connections); * This function calls the runtime PM helper to power up the given codec. * Unlike snd_hdac_power_up_pm(), you should call this only for the code * path that isn't included in PM path. Otherwise it gets stuck. + * + * Returns zero if successful, or a negative error code. */ -void snd_hdac_power_up(struct hdac_device *codec) +int snd_hdac_power_up(struct hdac_device *codec) { - pm_runtime_get_sync(&codec->dev); + return pm_runtime_get_sync(&codec->dev); } EXPORT_SYMBOL_GPL(snd_hdac_power_up); /** * snd_hdac_power_down - power down the codec * @codec: the codec object + * + * Returns zero if successful, or a negative error code. */ -void snd_hdac_power_down(struct hdac_device *codec) +int snd_hdac_power_down(struct hdac_device *codec) { struct device *dev = &codec->dev; pm_runtime_mark_last_busy(dev); - pm_runtime_put_autosuspend(dev); + return pm_runtime_put_autosuspend(dev); } EXPORT_SYMBOL_GPL(snd_hdac_power_down); @@ -529,11 +533,14 @@ EXPORT_SYMBOL_GPL(snd_hdac_power_down); * which may be called by PM suspend/resume again. OTOH, if a power-up * call must wake up the sleeper (e.g. in a kctl callback), use * snd_hdac_power_up() instead. + * + * Returns zero if successful, or a negative error code. */ -void snd_hdac_power_up_pm(struct hdac_device *codec) +int snd_hdac_power_up_pm(struct hdac_device *codec) { if (!atomic_inc_not_zero(&codec->in_pm)) - snd_hdac_power_up(codec); + return snd_hdac_power_up(codec); + return 0; } EXPORT_SYMBOL_GPL(snd_hdac_power_up_pm); @@ -543,11 +550,14 @@ EXPORT_SYMBOL_GPL(snd_hdac_power_up_pm); * * Like snd_hdac_power_up_pm(), this function is used in a recursive * code path like init code which may be called by PM suspend/resume again. + * + * Returns zero if successful, or a negative error code. */ -void snd_hdac_power_down_pm(struct hdac_device *codec) +int snd_hdac_power_down_pm(struct hdac_device *codec) { if (atomic_dec_if_positive(&codec->in_pm) < 0) - snd_hdac_power_down(codec); + return snd_hdac_power_down(codec); + return 0; } EXPORT_SYMBOL_GPL(snd_hdac_power_down_pm); #endif diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 1eabcdf69457..b0ed870ffb88 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -410,8 +410,9 @@ int snd_hdac_regmap_write_raw(struct hdac_device *codec, unsigned int reg, err = reg_raw_write(codec, reg, val); if (err == -EAGAIN) { - snd_hdac_power_up_pm(codec); - err = reg_raw_write(codec, reg, val); + err = snd_hdac_power_up_pm(codec); + if (!err) + err = reg_raw_write(codec, reg, val); snd_hdac_power_down_pm(codec); } return err; @@ -442,8 +443,9 @@ int snd_hdac_regmap_read_raw(struct hdac_device *codec, unsigned int reg, err = reg_raw_read(codec, reg, val); if (err == -EAGAIN) { - snd_hdac_power_up_pm(codec); - err = reg_raw_read(codec, reg, val); + err = snd_hdac_power_up_pm(codec); + if (!err) + err = reg_raw_read(codec, reg, val); snd_hdac_power_down_pm(codec); } return err; -- 2.4.5

1 0

[alsa-devel] [PATCH] ALSA: pcm: Fix lockdep warning with nonatomic PCM ops
by Takashi Iwai 17 Jul '15

17 Jul '15

With the nonatomic PCM ops, the system may spew lockdep warnings like: ============================================= [ INFO: possible recursive locking detected ] 4.2.0-rc1-jeejaval3 #12 Not tainted --------------------------------------------- aplay/4029 is trying to acquire lock: (snd_pcm_link_rwsem){.+.+.+}, at: [<ffffffff816fd473>] snd_pcm_stream_lock+0x43/0x60 but task is already holding lock: (snd_pcm_link_rwsem){.+.+.+}, at: [<ffffffff816fcf29>] snd_pcm_action_nonatomic+0x29/0x80 other info that might help us debug this: Possible unsafe locking scenario: CPU0 ---- lock(snd_pcm_link_rwsem); lock(snd_pcm_link_rwsem); Although this is false-positive as the rwsem is taken always as read-only for these code paths, it's certainly annoying to see this at any occasion. A simple fix is to use down_read_nested() in snd_pcm_stream_lock() that can be called inside another lock. Reported-by: Vinod Koul <vinod.koul(a)intel.com> Reported-by: Jeeja Kp <jeeja.kp(a)intel.com> Tested-by: Jeeja Kp <jeeja.kp(a)intel.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index d126c03361ae..75888dd38a7f 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -85,7 +85,7 @@ static DECLARE_RWSEM(snd_pcm_link_rwsem); void snd_pcm_stream_lock(struct snd_pcm_substream *substream) { if (substream->pcm->nonatomic) { - down_read(&snd_pcm_link_rwsem); + down_read_nested(&snd_pcm_link_rwsem, SINGLE_DEPTH_NESTING); mutex_lock(&substream->self_group.mutex); } else { read_lock(&snd_pcm_link_rwlock); -- 2.4.5

1 0

[alsa-devel] [asoc:for-next 6/18] sound/soc/codecs/rt5645.c:2931:3: warning: 'return' with a value, in function returning void
by kbuild test robot 17 Jul '15

17 Jul '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next head: db91b047f7131c7e8fe00ca9e38d14c0c103363f commit: 947d4dc4cd5be5da8e5e82f314d5cfb85a3eb3cc [6/18] Merge remote-tracking branch 'asoc/topic/rt5645' into asoc-next config: i386-randconfig-x009-201528 (attached as .config) reproduce: git checkout 947d4dc4cd5be5da8e5e82f314d5cfb85a3eb3cc # save the attached .config to linux build tree make ARCH=i386 All warnings (new ones prefixed by >>): sound/soc/codecs/rt5645.c: In function 'rt5645_jack_detect_work': >> sound/soc/codecs/rt5645.c:2931:3: warning: 'return' with a value, in function returning void return -EINVAL; ^ vim +/return +2931 sound/soc/codecs/rt5645.c 6e747d53 Bard Liao 2015-04-28 2915 regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL1, 6e747d53 Bard Liao 2015-04-28 2916 RT5645_DIG_GATE_CTRL, RT5645_DIG_GATE_CTRL); 6e747d53 Bard Liao 2015-04-28 2917 } 345b0f50 John Lin 2015-05-18 2918 rt5645_irq(0, rt5645); f3fa1bbd Oder Chiou 2014-09-19 2919 f3fa1bbd Oder Chiou 2014-09-19 2920 return 0; f3fa1bbd Oder Chiou 2014-09-19 2921 } f3fa1bbd Oder Chiou 2014-09-19 2922 EXPORT_SYMBOL_GPL(rt5645_set_jack_detect); f3fa1bbd Oder Chiou 2014-09-19 2923 cd6e82b8 Oder Chiou 2014-10-07 2924 static void rt5645_jack_detect_work(struct work_struct *work) cd6e82b8 Oder Chiou 2014-10-07 2925 { cd6e82b8 Oder Chiou 2014-10-07 2926 struct rt5645_priv *rt5645 = cd6e82b8 Oder Chiou 2014-10-07 2927 container_of(work, struct rt5645_priv, jack_detect_work.work); 6e747d53 Bard Liao 2015-04-28 2928 int val, btn_type, gpio_state = 0, report = 0; 6e747d53 Bard Liao 2015-04-28 2929 f2a5ded3 Nicolas Boichat 2015-07-17 2930 if (!rt5645->codec) f2a5ded3 Nicolas Boichat 2015-07-17 @2931 return -EINVAL; f2a5ded3 Nicolas Boichat 2015-07-17 2932 6e747d53 Bard Liao 2015-04-28 2933 switch (rt5645->pdata.jd_mode) { 6e747d53 Bard Liao 2015-04-28 2934 case 0: /* Not using rt5645 JD */ 0b0cefc8 Oder Chiou 2015-06-10 2935 if (rt5645->gpiod_hp_det) { 0b0cefc8 Oder Chiou 2015-06-10 2936 gpio_state = gpiod_get_value(rt5645->gpiod_hp_det); 0b0cefc8 Oder Chiou 2015-06-10 2937 dev_dbg(rt5645->codec->dev, "gpio_state = %d\n", 0b0cefc8 Oder Chiou 2015-06-10 2938 gpio_state); 0b0cefc8 Oder Chiou 2015-06-10 2939 report = rt5645_jack_detect(rt5645->codec, gpio_state); :::::: The code at line 2931 was first introduced by commit :::::: f2a5ded38592e5936a099ea6535ad5d3addcbc9d ASoC: rt5645: Check if codec is initialized in workqueue handler :::::: TO: Nicolas Boichat <drinkcat(a)chromium.org> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] HDMI hotplug on Skylake when power well is off
by David Henningsson 17 Jul '15

17 Jul '15

Hi, I'm trying to debug an issue here where the HDMI hotplug events are not delivered to the audio side when the power well is off. This is on a Skylake machine (running in HDA mode). I'm not sure whether the problem is upstream or due to my own patches while testing, so I was wondering how this is supposed to be working, so I can troubleshoot further? Should there be an IRQ on the HDA controller even if the power well is off, and if not, how should the audio driver be notified that an HDMI hotplug event has happened? -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

3 13

[alsa-devel] [PATCH v4] ASoC: rt5645: Check if codec is initialized in workqueue handler
by Nicolas Boichat 17 Jul '15

17 Jul '15

This fixes kernel panic on boot, if rt5645->codec is NULL when rt5645_jack_detect_work is first called. rt5645_jack_detect_work needs rt5645->codec to be initialized to setup dapm pins. Also, reporting jack events is useless, as the jacks cannot be set before the codec is ready. Since we manually call the interrupt handler in rt5645_set_jack_detect, the initial jack state will be reported correctly, and dapm pins will be setup at that time. Signed-off-by: Nicolas Boichat <drinkcat(a)chromium.org> --- This replaces 3c2234569cfff83718792f810a9bb855b78c2473, and applies on top of Linux 4.2-rc1, so that this patch can be applied in its own branch, without triggering a compile warning. Also, reworded the message to make it more obvious this can cause a kernel panic. Thanks! sound/soc/codecs/rt5645.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9ce311e..e9cc3aa 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2943,6 +2943,9 @@ static int rt5645_irq_detection(struct rt5645_priv *rt5645) { int val, btn_type, gpio_state = 0, report = 0; + if (!rt5645->codec) + return -EINVAL; + switch (rt5645->pdata.jd_mode) { case 0: /* Not using rt5645 JD */ if (rt5645->gpiod_hp_det) { -- 2.4.3.573.g4eafbef

2 3

[alsa-devel] [PATCH] ALSA: hda: fix kstrdup return value
by Heloise NH 17 Jul '15

17 Jul '15

From: kernelpatch_update <kernelpatch_update(a)163.com> In kstrdup we should return -ENOMEM when it reports an memory allocation failure, while the -ENODEV is referred to a failure in finding the cpu node in the device tree. Signed-off-by: Heloise NH <kernelpatch_update(a)163.com> --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5de3c5d..d78fa71 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -975,7 +975,7 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, if (codec->bus->modelname) { codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL); if (!codec->modelname) { - err = -ENODEV; + err = -ENOMEM; goto error; } } -- 1.9.1

2 1

[alsa-devel] bug-report: Play Bad Sound After Test Runs for 6 hrs
by Yadi 17 Jul '15

17 Jul '15

Hi, everyone, I encounter a strange issue that sound would be bad after running below testcase on intel target over 7 hours. composite: libsdl1.2-dev 1.2.14-4ubuntu1.1 libsdl-mixer1.2 1.2.8-6build1 alsa-lib-1.0.29 this issue would disappear if I try to attach process by gdb or adding print in alsa-lib/pcm. any comments would be appreciated. Yadi

2 1

[alsa-devel] drm/edid: sound/core: pcm_drm_eld.c compilation error in 4.2
by Jean-Francois Moine 17 Jul '15

17 Jul '15

The file sound/core/pcm_drm_eld.c which was added by the commit 838d1631b766529213684f07dd71cdf2e92f0623 ALSA: pcm: add DRM ELD helper lacks the function drm_eld_sad() which was defined by Russell's patch http://article.gmane.org/gmane.linux.ports.arm.kernel/411574 drm/edid: add function to help find SADs This patch has not been applied. -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

3 2

[alsa-devel] [PATCH] ASoC: rt5645: remove unused variable
by Sudip Mukherjee 17 Jul '15

17 Jul '15

We were getting a build warning about unused variable. This variable was a leftover from: a4e3c5fa955c ("ASoC: rt5645: Simplify rt5645_enable_push_button_irq") Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/soc/codecs/rt5645.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index d24c979..ff091d6 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2763,7 +2763,6 @@ static void rt5645_enable_push_button_irq(struct snd_soc_codec *codec, bool enable) { struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); - struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); if (enable) { snd_soc_dapm_force_enable_pin(dapm, "ADC L power"); -- 1.8.1.2

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: rt5645: Fix missing free_irq
by Koro Chen 17 Jul '15

17 Jul '15

The driver does not free irq when snd_soc_register_codec returns error. It does not return error when request irq failed, either. Add return when request irq failed, and free_irq if snd_soc_register_codec failed. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- Change since v1: - use free_irq instead of devm_request_threaded_irq --- sound/soc/codecs/rt5645.c | 17 ++++++++++++++--- 1 file changed, 14 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 2cedf0d..56e8c31 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3401,12 +3401,23 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, ret = request_threaded_irq(rt5645->i2c->irq, NULL, rt5645_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5645", rt5645); - if (ret) + if (ret) { dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + return ret; + } } - return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5645, - rt5645_dai, ARRAY_SIZE(rt5645_dai)); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5645, + rt5645_dai, ARRAY_SIZE(rt5645_dai)); + if (ret) + goto err_irq; + + return 0; + +err_irq: + if (rt5645->i2c->irq) + free_irq(rt5645->i2c->irq, rt5645); + return ret; } static int rt5645_i2c_remove(struct i2c_client *i2c) -- 1.8.1.1.dirty

1 1

[alsa-devel] [PATCH] ASoC: rt5645: Remove unused rt5645 variable
by Nicolas Boichat 17 Jul '15

17 Jul '15

"ASoC: rt5645: Simplify rt5645_enable_push_button_irq" removes the test that accessed rt5645->pdata.jd_mode (that test is now done in rt5645_jack_detect only), so we do not need that variable anymore. Signed-off-by: Nicolas Boichat <drinkcat(a)chromium.org> --- Applies on top of topic/rt5645. Sorry for not catching this earlier. sound/soc/codecs/rt5645.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 2cedf0d..c959ee0 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2763,7 +2763,6 @@ static void rt5645_enable_push_button_irq(struct snd_soc_codec *codec, bool enable) { struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); - struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); if (enable) { snd_soc_dapm_force_enable_pin(dapm, "ADC L power"); -- 2.4.3.573.g4eafbef

1 0

[alsa-devel] [PATCH] ASoC: Constify dev_pm_ops variables
by Axel Lin 17 Jul '15

17 Jul '15

The dev_pm_ops variables are not modified after initialization in these drivers, so make them const. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/wm2200.c | 2 +- sound/soc/codecs/wm5100.c | 2 +- sound/soc/codecs/wm8962.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index cdaa175..878d43a 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2481,7 +2481,7 @@ static int wm2200_runtime_resume(struct device *dev) } #endif -static struct dev_pm_ops wm2200_pm = { +static const struct dev_pm_ops wm2200_pm = { SET_RUNTIME_PM_OPS(wm2200_runtime_suspend, wm2200_runtime_resume, NULL) }; diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index e5572d2..eb494d0 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2706,7 +2706,7 @@ static int wm5100_runtime_resume(struct device *dev) } #endif -static struct dev_pm_ops wm5100_pm = { +static const struct dev_pm_ops wm5100_pm = { SET_RUNTIME_PM_OPS(wm5100_runtime_suspend, wm5100_runtime_resume, NULL) }; diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index c67e304..cbd1886 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3859,7 +3859,7 @@ static int wm8962_runtime_suspend(struct device *dev) } #endif -static struct dev_pm_ops wm8962_pm = { +static const struct dev_pm_ops wm8962_pm = { SET_RUNTIME_PM_OPS(wm8962_runtime_suspend, wm8962_runtime_resume, NULL) }; -- 2.1.0

1 0

[alsa-devel] Applied "ASoC: sti: sti_uniperiph_dai_create_ctrl() can be static" to the asoc tree
by Mark Brown 17 Jul '15

17 Jul '15

The patch ASoC: sti: sti_uniperiph_dai_create_ctrl() can be static has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From f2da4542268503d22869617b7dbcb40d364c78bb Mon Sep 17 00:00:00 2001 From: kbuild test robot <fengguang.wu(a)intel.com> Date: Fri, 17 Jul 2015 07:44:09 +0800 Subject: [PATCH] ASoC: sti: sti_uniperiph_dai_create_ctrl() can be static Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sti/sti_uniperif.c | 2 +- sound/soc/sti/uniperif_player.c | 4 ++-- sound/soc/sti/uniperif_reader.c | 4 ++-- 3 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c index 749e6b294184..83a301f2aa66 100644 --- a/sound/soc/sti/sti_uniperif.c +++ b/sound/soc/sti/sti_uniperif.c @@ -14,7 +14,7 @@ * This function is used to create Ctrl associated to DAI but also pcm device. * Request is done by front end to associate ctrl with pcm device id */ -int sti_uniperiph_dai_create_ctrl(struct snd_soc_dai *dai) +static int sti_uniperiph_dai_create_ctrl(struct snd_soc_dai *dai) { struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); struct uniperif *uni = priv->dai_data.uni; diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index d990d2c81a5d..f609089a6e00 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -43,7 +43,7 @@ * Note: snd_pcm_hardware is linked to DMA controller but is declared here to * integrate DAI_CPU capability in term of rate and supported channels */ -const struct snd_pcm_hardware uni_player_pcm_hw = { +static const struct snd_pcm_hardware uni_player_pcm_hw = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID, @@ -175,7 +175,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) return ret; } -int uni_player_clk_set_rate(struct uniperif *player, unsigned long rate) +static int uni_player_clk_set_rate(struct uniperif *player, unsigned long rate) { int rate_adjusted, rate_achieved, delta, ret; int adjustment = player->clk_adj; diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index 7d83827ba660..c502626f339b 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -17,7 +17,7 @@ * Note: snd_pcm_hardware is linked to DMA controller but is declared here to * integrate unireader capability in term of rate and supported channels */ -const struct snd_pcm_hardware uni_reader_pcm_hw = { +static const struct snd_pcm_hardware uni_reader_pcm_hw = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID, @@ -324,7 +324,7 @@ static int uni_reader_parse_dt(struct platform_device *pdev, return 0; } -const struct snd_soc_dai_ops uni_reader_dai_ops = { +static const struct snd_soc_dai_ops uni_reader_dai_ops = { .shutdown = uni_reader_shutdown, .prepare = uni_reader_prepare, .trigger = uni_reader_trigger, -- 2.1.4

1 0

[alsa-devel] [PATCH v2 0/12] Discover and probe dependencies
by Tomeu Vizoso 17 Jul '15

17 Jul '15

Hi, this is version 2 of a series that probes devices in dependency order so as to avoid deferred probes. While deferred probing is a powerful solution that makes sure that you eventually get a working system at the end of the boot, can make it very time consuming to find out why a device didn't probe and can also introduce big delays in when a device actually probes by sending it to the end of the deferred queue. So far I have only tested on a Tegra124 Chromebook. Thanks, Tomeu Changes in v2: - Instead of delaying all probes until late_initcall, only delay matches of platform devices that have a firmware node attached. - Allow bindings implementations register a function instead of using class callbacks, as not only subsystems implement firmware bindings. - Move strends to string.h - Document that consumers of backlight devices can use the 'backlight' property to hold a phandle to the backlight device. - Allocate the list of dependencies and pass it to the function that fills it. Tomeu Vizoso (12): device: property: delay device-driver matches device: property: find dependencies of a firmware node string: Introduce strends() gpio: register dependency parser for firmware nodes gpu: host1x: register dependency parser for firmware nodes backlight: Document consumers of backlight nodes backlight: register dependency parser for firmware nodes USB: EHCI: register dependency parser for firmware nodes regulator: register dependency parser for firmware nodes pwm: register dependency parser for firmware nodes ASoC: tegra: register dependency parser for firmware nodes driver-core: probe dependencies before probing .../bindings/video/backlight/backlight.txt | 22 ++++ drivers/base/dd.c | 139 +++++++++++++++++++++ drivers/base/property.c | 120 ++++++++++++++++++ drivers/gpio/gpiolib.c | 54 ++++++++ drivers/gpu/host1x/dev.c | 26 ++++ drivers/pwm/core.c | 28 +++++ drivers/regulator/core.c | 27 ++++ drivers/usb/host/ehci-tegra.c | 16 +++ drivers/video/backlight/backlight.c | 16 +++ include/linux/fwnode.h | 5 + include/linux/property.h | 12 ++ include/linux/string.h | 13 ++ sound/soc/tegra/tegra_max98090.c | 42 ++++++- 13 files changed, 519 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/video/backlight/backlight.txt -- 2.4.1

4 28

[alsa-devel] [PATCH 1/4] ASoC: max9877: Make driver global regmap struct local
by Lars-Peter Clausen 16 Jul '15

16 Jul '15

Use a stack local variable to handle function local state rather than a global static variable. The later has a potential for race conditions if the probe function runs for two devices concurrently. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/max9877.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c index e1df06f..b469e1c 100644 --- a/sound/soc/codecs/max9877.c +++ b/sound/soc/codecs/max9877.c @@ -20,8 +20,6 @@ #include "max9877.h" -static struct regmap *regmap; - static const struct reg_default max9877_regs[] = { { 0, 0x40 }, { 1, 0x00 }, @@ -145,6 +143,7 @@ static const struct regmap_config max9877_regmap = { static int max9877_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { + struct regmap *regmap; int i; regmap = devm_regmap_init_i2c(client, &max9877_regmap); -- 2.1.4

2 7

[alsa-devel] [PATCH 0/6] ASoC: Add audio for sti platforms round 2
by Arnaud Pouliquen 16 Jul '15

16 Jul '15

New set of patches after partial integration of [PATCH v3 0/9] ASoC: Add audio for sti platforms New patches: ASoC: sti: minor corrections for uniplayer Patches rebased from [PATCH v3 0/9] asoc: Add audio for sti platforms ASoC: sti: Add CPU DAI driver for capture ( IRQ enable /disable cleaned) ASoC: sti: Add uniperipheral dai driver ASoC: Add ability to build sti drivers ASoC: sti: Add clock adjustement control ASoC: sti: Add IEC control Arnaud Pouliquen (6): ASoC: sti: minor corrections for uniplayer ASoC: sti: Add CPU DAI driver for capture ASoC: sti: Add uniperipheral dai driver ASoC: Add ability to build sti drivers ASoC: sti: Add clock adjustement control ASoC: sti: Add IEC control sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/sti/Kconfig | 11 ++ sound/soc/sti/Makefile | 4 + sound/soc/sti/sti_uniperif.c | 253 ++++++++++++++++++++++++++++ sound/soc/sti/uniperif.h | 14 ++ sound/soc/sti/uniperif_player.c | 226 ++++++++++++++++++++++++- sound/soc/sti/uniperif_reader.c | 362 ++++++++++++++++++++++++++++++++++++++++ 8 files changed, 869 insertions(+), 3 deletions(-) create mode 100644 sound/soc/sti/Kconfig create mode 100644 sound/soc/sti/Makefile create mode 100644 sound/soc/sti/sti_uniperif.c create mode 100644 sound/soc/sti/uniperif_reader.c -- 1.9.1

2 10

[alsa-devel] [PATCH 0/7] ASoC: Intel: Skylake: Add code loader DMA support
by Vinod Koul 16 Jul '15

16 Jul '15

This series add code loader DMA support to Skylake driver and then adds the IPC init routines along with few smaller changes Omair M Abdullah (1): ASoC: Intel: Skylake: Print error code in IPC for SKL Subhransu S. Prusty (3): ASoC: Intel: Skylake: Add dsp loader ops ASoC: Intel: Skylake: Process code loader DMA interrupt ASoC: Intel: Skylake: Add SKL DSP initialization Vinod Koul (3): ASoC: Intel: Skylake: Add code loader DMA registers ASoC: Intel: Skylake: add code loader DMA operations ASoC: Intel: Skylake: Add code loader DMA APIs sound/soc/intel/common/sst-dsp-priv.h | 3 + sound/soc/intel/skylake/Makefile | 3 +- sound/soc/intel/skylake/skl-sst-cldma.c | 327 ++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-sst-cldma.h | 251 ++++++++++++++++++++++++ sound/soc/intel/skylake/skl-sst-dsp.c | 5 + sound/soc/intel/skylake/skl-sst-dsp.h | 26 +++ sound/soc/intel/skylake/skl-sst-ipc.c | 11 ++ sound/soc/intel/skylake/skl-sst.c | 280 +++++++++++++++++++++++++++ 8 files changed, 905 insertions(+), 1 deletion(-) create mode 100644 sound/soc/intel/skylake/skl-sst-cldma.c create mode 100644 sound/soc/intel/skylake/skl-sst-cldma.h create mode 100644 sound/soc/intel/skylake/skl-sst.c -- 1.9.1

2 15

[alsa-devel] [PATCH v4] ASoC: cs4349: Add support for Cirrus Logic CS4349
by timothyc.howe＠gmail.com 16 Jul '15

16 Jul '15

From: Tim Howe <tim.howe(a)cirrus.com> Signed-off-by: Tim Howe <tim.howe(a)cirrus.com> --- Documentation/devicetree/bindings/sound/cs4349.txt | 19 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs4349.c | 401 +++++++++++++++++++++ sound/soc/codecs/cs4349.h | 146 ++++++++ 5 files changed, 574 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs4349.txt create mode 100644 sound/soc/codecs/cs4349.c create mode 100644 sound/soc/codecs/cs4349.h diff --git a/Documentation/devicetree/bindings/sound/cs4349.txt b/Documentation/devicetree/bindings/sound/cs4349.txt new file mode 100644 index 0000000..54c117b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4349.txt @@ -0,0 +1,19 @@ +CS4349 audio CODEC + +Required properties: + + - compatible : "cirrus,cs4349" + + - reg : the I2C address of the device for I2C + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. + +Example: + +codec: cs4349@48 { + compatible = "cirrus,cs4349"; + reg = <0x48>; + reset-gpios = <&gpio 54 0>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7223c7f..4a78606 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -54,6 +54,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271_I2C if I2C select SND_SOC_CS4271_SPI if SPI_MASTER select SND_SOC_CS42XX8_I2C if I2C + select SND_SOC_CS4349 if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI select SND_SOC_DA7213 if I2C @@ -413,6 +414,11 @@ config SND_SOC_CS42XX8_I2C select SND_SOC_CS42XX8 select REGMAP_I2C +# Cirrus Logic CS4349 HiFi DAC +config SND_SOC_CS4349 + tristate "Cirrus Logic CS4349 CODEC" + depends on I2C + config SND_SOC_CX20442 tristate depends on TTY diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 9f7136c..36a9f37 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -46,6 +46,7 @@ snd-soc-cs4271-i2c-objs := cs4271-i2c.o snd-soc-cs4271-spi-objs := cs4271-spi.o snd-soc-cs42xx8-objs := cs42xx8.o snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o +snd-soc-cs4349-objs := cs4349.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -237,6 +238,7 @@ obj-$(CONFIG_SND_SOC_CS4271_I2C) += snd-soc-cs4271-i2c.o obj-$(CONFIG_SND_SOC_CS4271_SPI) += snd-soc-cs4271-spi.o obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o +obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o obj-$(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/cs4349.c b/sound/soc/codecs/cs4349.c new file mode 100644 index 0000000..a8df8a74 --- /dev/null +++ b/sound/soc/codecs/cs4349.c @@ -0,0 +1,401 @@ +/* + * cs4349.c -- CS4349 ALSA Soc Audio driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Authors: Tim Howe <Tim.Howe(a)cirrus.com> + * + * 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/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/platform_device.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/of_device.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include "cs4349.h" + + +static const struct reg_default cs4349_reg_defaults[] = { + { 2, 0x00 }, /* r02 - Mode Control */ + { 3, 0x09 }, /* r03 - Volume, Mixing and Inversion Control */ + { 4, 0x81 }, /* r04 - Mute Control */ + { 5, 0x00 }, /* r05 - Channel A Volume Control */ + { 6, 0x00 }, /* r06 - Channel B Volume Control */ + { 7, 0xB1 }, /* r07 - Ramp and Filter Control */ + { 8, 0x1C }, /* r08 - Misc. Control */ +}; + +/* Private data for the CS4349 */ +struct cs4349_private { + struct regmap *regmap; + struct cs4349_platform_data pdata; + struct gpio_desc *reset_gpio; + unsigned int mode; + int rate; +}; + +static bool cs4349_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS4349_CHIPID: + case CS4349_MODE: + case CS4349_VMI: + case CS4349_MUTE: + case CS4349_VOLA: + case CS4349_VOLB: + case CS4349_RMPFLT: + case CS4349_MISC: + return true; + default: + return false; + } +} + +static int cs4349_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4349_private *cs4349 = snd_soc_codec_get_drvdata(codec); + unsigned int fmt; + + fmt = format & SND_SOC_DAIFMT_FORMAT_MASK; + + switch (fmt) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + cs4349->mode = format & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + return -EINVAL; + } + + return 0; +} + +static int cs4349_pcm_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 cs4349_private *cs4349 = snd_soc_codec_get_drvdata(codec); + int mode, fmt, ret; + + mode = snd_soc_read(codec, CS4349_MODE); + cs4349->rate = params_rate(params); + + switch (cs4349->mode) { + case SND_SOC_DAIFMT_I2S: + mode |= MODE_FORMAT(DIF_I2S); + break; + case SND_SOC_DAIFMT_LEFT_J: + mode |= MODE_FORMAT(DIF_LEFT_JST); + break; + case SND_SOC_DAIFMT_RIGHT_J: + switch (params_width(params)) { + case 16: + fmt = DIF_RGHT_JST16; + break; + case 24: + fmt = DIF_RGHT_JST24; + break; + default: + return -EINVAL; + } + mode |= MODE_FORMAT(fmt); + break; + default: + return -EINVAL; + } + + ret = snd_soc_write(codec, CS4349_MODE, mode); + if (ret < 0) + return ret; + + return 0; +} + +static int cs4349_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + int reg; + + reg = 0; + if (mute) + reg = MUTE_AB_MASK; + + return snd_soc_update_bits(codec, CS4349_MUTE, MUTE_AB_MASK, reg); +} + +static DECLARE_TLV_DB_SCALE(dig_tlv, -12750, 50, 0); + +static const char * const chan_mix_texts[] = { + "Mute", "MuteA", "MuteA SwapB", "MuteA MonoB", "SwapA MuteB", + "BothR", "Swap", "SwapA MonoB", "MuteB", "Normal", "BothL", + "MonoB", "MonoA MuteB", "MonoA", "MonoA SwapB", "Mono", + /*Normal == Channel A = Left, Channel B = Right*/ +}; + +static const char * const fm_texts[] = { + "Auto", "Single", "Double", "Quad", +}; + +static const char * const deemph_texts[] = { + "None", "44.1k", "48k", "32k", +}; + +static const char * const softr_zeroc_texts[] = { + "Immediate", "Zero Cross", "Soft Ramp", "SR on ZC", +}; + +static int deemph_values[] = { + 0, 4, 8, 12, +}; + +static int softr_zeroc_values[] = { + 0, 64, 128, 192, +}; + +static const struct soc_enum chan_mix_enum = + SOC_ENUM_SINGLE(CS4349_VMI, 0, + ARRAY_SIZE(chan_mix_texts), + chan_mix_texts); + +static const struct soc_enum fm_mode_enum = + SOC_ENUM_SINGLE(CS4349_MODE, 0, + ARRAY_SIZE(fm_texts), + fm_texts); + +static SOC_VALUE_ENUM_SINGLE_DECL(deemph_enum, CS4349_MODE, 0, DEM_MASK, + deemph_texts, deemph_values); + +static SOC_VALUE_ENUM_SINGLE_DECL(softr_zeroc_enum, CS4349_RMPFLT, 0, + SR_ZC_MASK, softr_zeroc_texts, + softr_zeroc_values); + +static const struct snd_kcontrol_new cs4349_snd_controls[] = { + SOC_DOUBLE_R_TLV("Master Playback Volume", + CS4349_VOLA, CS4349_VOLB, 0, 0xFF, 1, dig_tlv), + SOC_ENUM("Functional Mode", fm_mode_enum), + SOC_ENUM("De-Emphasis Control", deemph_enum), + SOC_ENUM("Soft Ramp Zero Cross Control", softr_zeroc_enum), + SOC_ENUM("Channel Mixer", chan_mix_enum), + SOC_SINGLE("VolA = VolB Switch", CS4349_VMI, 7, 1, 0), + SOC_SINGLE("InvertA Switch", CS4349_VMI, 6, 1, 0), + SOC_SINGLE("InvertB Switch", CS4349_VMI, 5, 1, 0), + SOC_SINGLE("Auto-Mute Switch", CS4349_MUTE, 7, 1, 0), + SOC_SINGLE("MUTEC A = B Switch", CS4349_MUTE, 5, 1, 0), + SOC_SINGLE("Soft Ramp Up Switch", CS4349_RMPFLT, 5, 1, 0), + SOC_SINGLE("Soft Ramp Down Switch", CS4349_RMPFLT, 4, 1, 0), + SOC_SINGLE("Slow Roll Off Filter Switch", CS4349_RMPFLT, 2, 1, 0), + SOC_SINGLE("Freeze Switch", CS4349_MISC, 5, 1, 0), + SOC_SINGLE("Popguard Switch", CS4349_MISC, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget cs4349_dapm_widgets[] = { + SND_SOC_DAPM_DAC("HiFi DAC", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_OUTPUT("OutputA"), + SND_SOC_DAPM_OUTPUT("OutputB"), +}; + +static const struct snd_soc_dapm_route cs4349_routes[] = { + {"DAC Playback", NULL, "OutputA"}, + {"DAC Playback", NULL, "OutputB"}, + + {"OutputA", NULL, "HiFi DAC"}, + {"OutputB", NULL, "HiFi DAC"}, +}; + +#define CS4349_PCM_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define CS4349_PCM_RATES SNDRV_PCM_RATE_8000_192000 + +static const struct snd_soc_dai_ops cs4349_dai_ops = { + .hw_params = cs4349_pcm_hw_params, + .set_fmt = cs4349_set_dai_fmt, + .digital_mute = cs4349_digital_mute, +}; + +static struct snd_soc_dai_driver cs4349_dai = { + .name = "cs4349_hifi", + .playback = { + .stream_name = "DAC Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS4349_PCM_RATES, + .formats = CS4349_PCM_FORMATS, + }, + .ops = &cs4349_dai_ops, + .symmetric_rates = 1, +}; + +static struct snd_soc_codec_driver soc_codec_dev_cs4349 = { + .controls = cs4349_snd_controls, + .num_controls = ARRAY_SIZE(cs4349_snd_controls), + + .dapm_widgets = cs4349_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs4349_dapm_widgets), + .dapm_routes = cs4349_routes, + .num_dapm_routes = ARRAY_SIZE(cs4349_routes), +}; + +static struct regmap_config cs4349_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS4349_NUMREGS, + .reg_defaults = cs4349_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4349_reg_defaults), + .readable_reg = cs4349_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs4349_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct cs4349_private *cs4349; + struct cs4349_platform_data *pdata = dev_get_platdata(&client->dev); + int ret = 0; + + cs4349 = devm_kzalloc(&client->dev, sizeof(*cs4349), GFP_KERNEL); + if (!cs4349) + return -ENOMEM; + + cs4349->regmap = devm_regmap_init_i2c(client, &cs4349_regmap); + if (IS_ERR(cs4349->regmap)) { + ret = PTR_ERR(cs4349->regmap); + dev_err(&client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + if (pdata) + cs4349->pdata = *pdata; + + /* Reset the Device */ + cs4349->reset_gpio = devm_gpiod_get_optional(&client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs4349->reset_gpio)) + return PTR_ERR(cs4349->reset_gpio); + + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 1); + + i2c_set_clientdata(client, cs4349); + + return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4349, + &cs4349_dai, 1); +} + +static int cs4349_i2c_remove(struct i2c_client *client) +{ + struct cs4349_private *cs4349 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + /* Hold down reset */ + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 0); + + return 0; +} + +#ifdef CONFIG_PM +static int cs4349_runtime_suspend(struct device *dev) +{ + struct cs4349_private *cs4349 = dev_get_drvdata(dev); + struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); + int ret; + + ret = snd_soc_update_bits(rtd->codec, CS4349_MISC, PWR_DWN, 1); + if (ret < 0) + return ret; + + regcache_cache_only(cs4349->regmap, true); + + /* Hold down reset */ + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 0); + + return 0; +} + +static int cs4349_runtime_resume(struct device *dev) +{ + struct cs4349_private *cs4349 = dev_get_drvdata(dev); + struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); + int ret; + + ret = snd_soc_update_bits(rtd->codec, CS4349_MISC, PWR_DWN, 0); + if (ret < 0) + return ret; + + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 1); + + regcache_cache_only(cs4349->regmap, false); + regcache_sync(cs4349->regmap); + + return 0; +} +#endif + +static const struct dev_pm_ops cs4349_runtime_pm = { + SET_RUNTIME_PM_OPS(cs4349_runtime_suspend, cs4349_runtime_resume, + NULL) +}; + +static const struct of_device_id cs4349_of_match[] = { + { .compatible = "cirrus,cs4349", }, + {}, +}; + +MODULE_DEVICE_TABLE(of, cs4349_of_match); + +static const struct i2c_device_id cs4349_i2c_id[] = { + {"cs4349", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs4349_i2c_id); + +static struct i2c_driver cs4349_i2c_driver = { + .driver = { + .name = "cs4349", + .owner = THIS_MODULE, + .of_match_table = cs4349_of_match, + }, + .id_table = cs4349_i2c_id, + .probe = cs4349_i2c_probe, + .remove = cs4349_i2c_remove, +}; + +module_i2c_driver(cs4349_i2c_driver); + +MODULE_AUTHOR("Tim Howe <tim.howe(a)cirrus.com>"); +MODULE_DESCRIPTION("Cirrus Logic CS4349 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs4349.h b/sound/soc/codecs/cs4349.h new file mode 100644 index 0000000..3884a89 --- /dev/null +++ b/sound/soc/codecs/cs4349.h @@ -0,0 +1,146 @@ +/* + * ALSA SoC CS4349 codec driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Author: Tim Howe <Tim.Howe(a)cirrus.com> + * + * 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. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + */ + +#ifndef __CS4349_H__ +#define __CS4349_H__ + +struct cs4349_platform_data { + + /* GPIO for Reset */ + unsigned int gpio_nreset; + +}; + +/* CS4349 registers addresses */ +#define CS4349_CHIPID 0x01 /* Device and Rev ID, Read Only */ +#define CS4349_MODE 0x02 /* Mode Control */ +#define CS4349_VMI 0x03 /* Volume, Mixing, Inversion Control */ +#define CS4349_MUTE 0x04 /* Mute Control */ +#define CS4349_VOLA 0x05 /* DAC Channel A Volume Control */ +#define CS4349_VOLB 0x06 /* DAC Channel B Volume Control */ +#define CS4349_RMPFLT 0x07 /* Ramp and Filter Control */ +#define CS4349_MISC 0x08 /* Power Down,Freeze Control,Pop Stop*/ + +#define CS4349_FIRSTREG 0x01 +#define CS4349_LASTREG 0x08 +#define CS4349_NUMREGS (CS4349_LASTREG - CS4349_FIRSTREG + 1) +#define CS4349_I2C_INCR 0x80 + + +/* Device and Revision ID */ +#define CS4349_REVA 0xF0 /* Rev A */ +#define CS4349_REVB 0xF1 /* Rev B */ +#define CS4349_REVC2 0xFF /* Rev C2 */ + + +/* PDN_DONE Poll Maximum + * If soft ramp is set it will take much longer to power down + * the system. + */ +#define PDN_POLL_MAX 900 + + +/* Bitfield Definitions */ + +/* CS4349_MODE */ +/* (Digital Interface Format, De-Emphasis Control, Functional Mode */ +#define DIF2 (1 << 6) +#define DIF1 (1 << 5) +#define DIF0 (1 << 4) +#define DEM1 (1 << 3) +#define DEM0 (1 << 2) +#define FM1 (1 << 1) +#define DIF_LEFT_JST 0x00 +#define DIF_I2S 0x01 +#define DIF_RGHT_JST16 0x02 +#define DIF_RGHT_JST24 0x03 +#define DIF_TDM0 0x04 +#define DIF_TDM1 0x05 +#define DIF_TDM2 0x06 +#define DIF_TDM3 0x07 +#define DIF_MASK 0x70 +#define MODE_FORMAT(x) (((x)&7)<<4) +#define DEM_MASK 0x0C +#define NO_DEM 0x00 +#define DEM_441 0x04 +#define DEM_48K 0x08 +#define DEM_32K 0x0C +#define FM_AUTO 0x00 +#define FM_SNGL 0x01 +#define FM_DBL 0x02 +#define FM_QUAD 0x03 +#define FM_SNGL_MIN 30000 +#define FM_SNGL_MAX 54000 +#define FM_DBL_MAX 108000 +#define FM_QUAD_MAX 216000 +#define FM_MASK 0x03 + +/* CS4349_VMI (VMI = Volume, Mixing and Inversion Controls) */ +#define VOLBISA (1 << 7) +#define VOLAISB (1 << 7) +/* INVERT_A only available for Left Jstfd, Right Jstfd16 and Right Jstfd24 */ +#define INVERT_A (1 << 6) +/* INVERT_B only available for Left Jstfd, Right Jstfd16 and Right Jstfd24 */ +#define INVERT_B (1 << 5) +#define ATAPI3 (1 << 3) +#define ATAPI2 (1 << 2) +#define ATAPI1 (1 << 1) +#define ATAPI0 (1 << 0) +#define MUTEAB 0x00 +#define MUTEA_RIGHTB 0x01 +#define MUTEA_LEFTB 0x02 +#define MUTEA_SUMLRDIV2B 0x03 +#define RIGHTA_MUTEB 0x04 +#define RIGHTA_RIGHTB 0x05 +#define RIGHTA_LEFTB 0x06 +#define RIGHTA_SUMLRDIV2B 0x07 +#define LEFTA_MUTEB 0x08 +#define LEFTA_RIGHTB 0x09 /* Default */ +#define LEFTA_LEFTB 0x0A +#define LEFTA_SUMLRDIV2B 0x0B +#define SUMLRDIV2A_MUTEB 0x0C +#define SUMLRDIV2A_RIGHTB 0x0D +#define SUMLRDIV2A_LEFTB 0x0E +#define SUMLRDIV2_AB 0x0F +#define CHMIX_MASK 0x0F + +/* CS4349_MUTE */ +#define AUTOMUTE (1 << 7) +#define MUTEC_AB (1 << 5) +#define MUTE_A (1 << 4) +#define MUTE_B (1 << 3) +#define MUTE_AB_MASK 0x18 + +/* CS4349_RMPFLT (Ramp and Filter Control) */ +#define SCZ1 (1 << 7) +#define SCZ0 (1 << 6) +#define RMP_UP (1 << 5) +#define RMP_DN (1 << 4) +#define FILT_SEL (1 << 2) +#define IMMDT_CHNG 0x31 +#define ZEROCRSS 0x71 +#define SOFT_RMP 0xB1 +#define SFTRMP_ZEROCRSS 0xF1 +#define SR_ZC_MASK 0xC0 + +/* CS4349_MISC */ +#define PWR_DWN (1 << 7) +#define FREEZE (1 << 5) +#define POPG_EN (1 << 4) + +#endif /* __CS4349_H__ */ -- 2.4.5

2 1

[alsa-devel] [PATCH 1/2] V4 regmap: Use reg_sequence for multi_reg_write / register_patch
by Nariman Poushin 16 Jul '15

16 Jul '15

Add support for writing sequences of registers / patches with specified delays (in microseconds). Logically separates the functionality using sequences of register writes from the functions that take register defaults, as adding a delay field on the reg_defaults can increase memory usage substantially. This also updates all the clients of multi_reg_write/register_patch. Signed-off-by: Nariman Poushin <nariman(a)opensource.wolfsonmicro.com> --- drivers/base/regmap/internal.h | 2 +- drivers/base/regmap/regmap.c | 22 +++++++++++----------- drivers/gpu/drm/i2c/adv7511.c | 2 +- drivers/input/misc/drv260x.c | 6 +++--- drivers/input/misc/drv2665.c | 2 +- drivers/input/misc/drv2667.c | 4 ++-- drivers/mfd/arizona-core.c | 2 +- drivers/mfd/twl6040.c | 2 +- drivers/mfd/wm5102-tables.c | 6 +++--- drivers/mfd/wm5110-tables.c | 6 +++--- drivers/mfd/wm8994-core.c | 8 ++++---- drivers/mfd/wm8997-tables.c | 2 +- include/linux/regmap.h | 21 ++++++++++++++++++--- sound/soc/codecs/arizona.c | 2 +- sound/soc/codecs/cs35l32.c | 2 +- sound/soc/codecs/cs42l52.c | 2 +- sound/soc/codecs/da7210.c | 4 ++-- sound/soc/codecs/rt5640.c | 2 +- sound/soc/codecs/rt5645.c | 4 ++-- sound/soc/codecs/rt5651.c | 2 +- sound/soc/codecs/rt5670.c | 2 +- sound/soc/codecs/rt5677.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/wm2200.c | 2 +- sound/soc/codecs/wm5100.c | 2 +- sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8993.c | 2 +- 27 files changed, 66 insertions(+), 51 deletions(-) diff --git a/drivers/base/regmap/internal.h b/drivers/base/regmap/internal.h index b2b2849..873ddf9 100644 --- a/drivers/base/regmap/internal.h +++ b/drivers/base/regmap/internal.h @@ -136,7 +136,7 @@ struct regmap { /* if set, the HW registers are known to match map->reg_defaults */ bool no_sync_defaults; - struct reg_default *patch; + struct reg_sequence *patch; int patch_regs; /* if set, converts bulk rw to single rw */ diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c index dd63bcb..0a849ee 100644 --- a/drivers/base/regmap/regmap.c +++ b/drivers/base/regmap/regmap.c @@ -1755,7 +1755,7 @@ EXPORT_SYMBOL_GPL(regmap_bulk_write); * relative. The page register has been written if that was neccessary. */ static int _regmap_raw_multi_reg_write(struct regmap *map, - const struct reg_default *regs, + const struct reg_sequence *regs, size_t num_regs) { int ret; @@ -1812,12 +1812,12 @@ static unsigned int _regmap_register_page(struct regmap *map, } static int _regmap_range_multi_paged_reg_write(struct regmap *map, - struct reg_default *regs, + struct reg_sequence *regs, size_t num_regs) { int ret; int i, n; - struct reg_default *base; + struct reg_sequence *base; unsigned int this_page = 0; /* * the set of registers are not neccessarily in order, but @@ -1855,7 +1855,7 @@ static int _regmap_range_multi_paged_reg_write(struct regmap *map, } static int _regmap_multi_reg_write(struct regmap *map, - const struct reg_default *regs, + const struct reg_sequence *regs, size_t num_regs) { int i; @@ -1907,8 +1907,8 @@ static int _regmap_multi_reg_write(struct regmap *map, struct regmap_range_node *range; range = _regmap_range_lookup(map, reg); if (range) { - size_t len = sizeof(struct reg_default)*num_regs; - struct reg_default *base = kmemdup(regs, len, + size_t len = sizeof(struct reg_sequence)*num_regs; + struct reg_sequence *base = kmemdup(regs, len, GFP_KERNEL); if (!base) return -ENOMEM; @@ -1941,7 +1941,7 @@ static int _regmap_multi_reg_write(struct regmap *map, * A value of zero will be returned on success, a negative errno will be * returned in error cases. */ -int regmap_multi_reg_write(struct regmap *map, const struct reg_default *regs, +int regmap_multi_reg_write(struct regmap *map, const struct reg_sequence *regs, int num_regs) { int ret; @@ -1974,7 +1974,7 @@ EXPORT_SYMBOL_GPL(regmap_multi_reg_write); * be returned in error cases. */ int regmap_multi_reg_write_bypassed(struct regmap *map, - const struct reg_default *regs, + const struct reg_sequence *regs, int num_regs) { int ret; @@ -2587,10 +2587,10 @@ EXPORT_SYMBOL_GPL(regmap_async_complete); * The caller must ensure that this function cannot be called * concurrently with either itself or regcache_sync(). */ -int regmap_register_patch(struct regmap *map, const struct reg_default *regs, +int regmap_register_patch(struct regmap *map, const struct reg_sequence *regs, int num_regs) { - struct reg_default *p; + struct reg_sequence *p; int ret; bool bypass; @@ -2599,7 +2599,7 @@ int regmap_register_patch(struct regmap *map, const struct reg_default *regs, return 0; p = krealloc(map->patch, - sizeof(struct reg_default) * (map->patch_regs + num_regs), + sizeof(struct reg_sequence) * (map->patch_regs + num_regs), GFP_KERNEL); if (p) { memcpy(p + map->patch_regs, regs, num_regs * sizeof(*regs)); diff --git a/drivers/gpu/drm/i2c/adv7511.c b/drivers/gpu/drm/i2c/adv7511.c index 2aaa3c8..00416f2 100644 --- a/drivers/gpu/drm/i2c/adv7511.c +++ b/drivers/gpu/drm/i2c/adv7511.c @@ -54,7 +54,7 @@ static struct adv7511 *encoder_to_adv7511(struct drm_encoder *encoder) } /* ADI recommended values for proper operation. */ -static const struct reg_default adv7511_fixed_registers[] = { +static const struct reg_sequence adv7511_fixed_registers[] = { { 0x98, 0x03 }, { 0x9a, 0xe0 }, { 0x9c, 0x30 }, diff --git a/drivers/input/misc/drv260x.c b/drivers/input/misc/drv260x.c index e5d60ec..f5c9cf2 100644 --- a/drivers/input/misc/drv260x.c +++ b/drivers/input/misc/drv260x.c @@ -313,14 +313,14 @@ static void drv260x_close(struct input_dev *input) gpiod_set_value(haptics->enable_gpio, 0); } -static const struct reg_default drv260x_lra_cal_regs[] = { +static const struct reg_sequence drv260x_lra_cal_regs[] = { { DRV260X_MODE, DRV260X_AUTO_CAL }, { DRV260X_CTRL3, DRV260X_NG_THRESH_2 }, { DRV260X_FEEDBACK_CTRL, DRV260X_FB_REG_LRA_MODE | DRV260X_BRAKE_FACTOR_4X | DRV260X_LOOP_GAIN_HIGH }, }; -static const struct reg_default drv260x_lra_init_regs[] = { +static const struct reg_sequence drv260x_lra_init_regs[] = { { DRV260X_MODE, DRV260X_RT_PLAYBACK }, { DRV260X_A_TO_V_CTRL, DRV260X_AUDIO_HAPTICS_PEAK_20MS | DRV260X_AUDIO_HAPTICS_FILTER_125HZ }, @@ -337,7 +337,7 @@ static const struct reg_default drv260x_lra_init_regs[] = { { DRV260X_CTRL4, DRV260X_AUTOCAL_TIME_500MS }, }; -static const struct reg_default drv260x_erm_cal_regs[] = { +static const struct reg_sequence drv260x_erm_cal_regs[] = { { DRV260X_MODE, DRV260X_AUTO_CAL }, { DRV260X_A_TO_V_MIN_INPUT, DRV260X_AUDIO_HAPTICS_MIN_IN_VOLT }, { DRV260X_A_TO_V_MAX_INPUT, DRV260X_AUDIO_HAPTICS_MAX_IN_VOLT }, diff --git a/drivers/input/misc/drv2665.c b/drivers/input/misc/drv2665.c index 0afaa33..924456e 100644 --- a/drivers/input/misc/drv2665.c +++ b/drivers/input/misc/drv2665.c @@ -132,7 +132,7 @@ static void drv2665_close(struct input_dev *input) "Failed to enter standby mode: %d\n", error); } -static const struct reg_default drv2665_init_regs[] = { +static const struct reg_sequence drv2665_init_regs[] = { { DRV2665_CTRL_2, 0 | DRV2665_10_MS_IDLE_TOUT }, { DRV2665_CTRL_1, DRV2665_25_VPP_GAIN }, }; diff --git a/drivers/input/misc/drv2667.c b/drivers/input/misc/drv2667.c index fc0fddf..047136a 100644 --- a/drivers/input/misc/drv2667.c +++ b/drivers/input/misc/drv2667.c @@ -262,14 +262,14 @@ static void drv2667_close(struct input_dev *input) "Failed to enter standby mode: %d\n", error); } -static const struct reg_default drv2667_init_regs[] = { +static const struct reg_sequence drv2667_init_regs[] = { { DRV2667_CTRL_2, 0 }, { DRV2667_CTRL_1, DRV2667_25_VPP_GAIN }, { DRV2667_WV_SEQ_0, 1 }, { DRV2667_WV_SEQ_1, 0 } }; -static const struct reg_default drv2667_page1_init[] = { +static const struct reg_sequence drv2667_page1_init[] = { { DRV2667_RAM_HDR_SZ, 0x05 }, { DRV2667_RAM_START_HI, 0x80 }, { DRV2667_RAM_START_LO, 0x06 }, diff --git a/drivers/mfd/arizona-core.c b/drivers/mfd/arizona-core.c index bebf58a..66d50be 100644 --- a/drivers/mfd/arizona-core.c +++ b/drivers/mfd/arizona-core.c @@ -392,7 +392,7 @@ err: * Register patch to some of the CODECs internal write sequences * to ensure a clean exit from the low power sleep state. */ -static const struct reg_default wm5110_sleep_patch[] = { +static const struct reg_sequence wm5110_sleep_patch[] = { { 0x337A, 0xC100 }, { 0x337B, 0x0041 }, { 0x3300, 0xA210 }, diff --git a/drivers/mfd/twl6040.c b/drivers/mfd/twl6040.c index c5265c1..583dc33 100644 --- a/drivers/mfd/twl6040.c +++ b/drivers/mfd/twl6040.c @@ -86,7 +86,7 @@ static const struct reg_default twl6040_defaults[] = { { 0x2E, 0x00 }, /* REG_STATUS (ro) */ }; -static struct reg_default twl6040_patch[] = { +static struct reg_sequence twl6040_patch[] = { /* * Select I2C bus access to dual access registers * Interrupt register is cleared on read diff --git a/drivers/mfd/wm5102-tables.c b/drivers/mfd/wm5102-tables.c index aeae6ec..423fb37 100644 --- a/drivers/mfd/wm5102-tables.c +++ b/drivers/mfd/wm5102-tables.c @@ -21,7 +21,7 @@ #define WM5102_NUM_AOD_ISR 2 #define WM5102_NUM_ISR 5 -static const struct reg_default wm5102_reva_patch[] = { +static const struct reg_sequence wm5102_reva_patch[] = { { 0x80, 0x0003 }, { 0x221, 0x0090 }, { 0x211, 0x0014 }, @@ -57,7 +57,7 @@ static const struct reg_default wm5102_reva_patch[] = { { 0x80, 0x0000 }, }; -static const struct reg_default wm5102_revb_patch[] = { +static const struct reg_sequence wm5102_revb_patch[] = { { 0x19, 0x0001 }, { 0x80, 0x0003 }, { 0x081, 0xE022 }, @@ -80,7 +80,7 @@ static const struct reg_default wm5102_revb_patch[] = { /* We use a function so we can use ARRAY_SIZE() */ int wm5102_patch(struct arizona *arizona) { - const struct reg_default *wm5102_patch; + const struct reg_sequence *wm5102_patch; int patch_size; switch (arizona->rev) { diff --git a/drivers/mfd/wm5110-tables.c b/drivers/mfd/wm5110-tables.c index 12cad94..26ce14f 100644 --- a/drivers/mfd/wm5110-tables.c +++ b/drivers/mfd/wm5110-tables.c @@ -21,7 +21,7 @@ #define WM5110_NUM_AOD_ISR 2 #define WM5110_NUM_ISR 5 -static const struct reg_default wm5110_reva_patch[] = { +static const struct reg_sequence wm5110_reva_patch[] = { { 0x80, 0x3 }, { 0x44, 0x20 }, { 0x45, 0x40 }, @@ -134,7 +134,7 @@ static const struct reg_default wm5110_reva_patch[] = { { 0x209, 0x002A }, }; -static const struct reg_default wm5110_revb_patch[] = { +static const struct reg_sequence wm5110_revb_patch[] = { { 0x80, 0x3 }, { 0x36e, 0x0210 }, { 0x370, 0x0210 }, @@ -224,7 +224,7 @@ static const struct reg_default wm5110_revb_patch[] = { { 0x80, 0x0 }, }; -static const struct reg_default wm5110_revd_patch[] = { +static const struct reg_sequence wm5110_revd_patch[] = { { 0x80, 0x3 }, { 0x80, 0x3 }, { 0x393, 0x27 }, diff --git a/drivers/mfd/wm8994-core.c b/drivers/mfd/wm8994-core.c index 53ae5af..0f4169a 100644 --- a/drivers/mfd/wm8994-core.c +++ b/drivers/mfd/wm8994-core.c @@ -243,21 +243,21 @@ static int wm8994_ldo_in_use(struct wm8994_pdata *pdata, int ldo) } #endif -static const struct reg_default wm8994_revc_patch[] = { +static const struct reg_sequence wm8994_revc_patch[] = { { 0x102, 0x3 }, { 0x56, 0x3 }, { 0x817, 0x0 }, { 0x102, 0x0 }, }; -static const struct reg_default wm8958_reva_patch[] = { +static const struct reg_sequence wm8958_reva_patch[] = { { 0x102, 0x3 }, { 0xcb, 0x81 }, { 0x817, 0x0 }, { 0x102, 0x0 }, }; -static const struct reg_default wm1811_reva_patch[] = { +static const struct reg_sequence wm1811_reva_patch[] = { { 0x102, 0x3 }, { 0x56, 0xc07 }, { 0x5d, 0x7e }, @@ -326,7 +326,7 @@ static int wm8994_device_init(struct wm8994 *wm8994, int irq) { struct wm8994_pdata *pdata; struct regmap_config *regmap_config; - const struct reg_default *regmap_patch = NULL; + const struct reg_sequence *regmap_patch = NULL; const char *devname; int ret, i, patch_regs = 0; int pulls = 0; diff --git a/drivers/mfd/wm8997-tables.c b/drivers/mfd/wm8997-tables.c index c0c25d75..cab2c68 100644 --- a/drivers/mfd/wm8997-tables.c +++ b/drivers/mfd/wm8997-tables.c @@ -17,7 +17,7 @@ #include "arizona.h" -static const struct reg_default wm8997_reva_patch[] = { +static const struct reg_sequence wm8997_reva_patch[] = { { 0x80, 0x0003 }, { 0x214, 0x0008 }, { 0x458, 0x0000 }, diff --git a/include/linux/regmap.h b/include/linux/regmap.h index 519c962..ee5106e 100644 --- a/include/linux/regmap.h +++ b/include/linux/regmap.h @@ -50,6 +50,21 @@ struct reg_default { unsigned int def; }; +/** + * Register / Value pairs for sequences of writes, incorporating an optional + * delay in microseconds. + * + * @reg: Register address. + * @def: Register default value. + * @delay_us: Delay in microseconds + */ + +struct reg_sequence { + unsigned reg; + unsigned def; + unsigned delay_us; +}; + #ifdef CONFIG_REGMAP enum regmap_endian { @@ -410,10 +425,10 @@ int regmap_raw_write(struct regmap *map, unsigned int reg, const void *val, size_t val_len); int regmap_bulk_write(struct regmap *map, unsigned int reg, const void *val, size_t val_count); -int regmap_multi_reg_write(struct regmap *map, const struct reg_default *regs, +int regmap_multi_reg_write(struct regmap *map, const struct reg_sequence *regs, int num_regs); int regmap_multi_reg_write_bypassed(struct regmap *map, - const struct reg_default *regs, + const struct reg_sequence *regs, int num_regs); int regmap_raw_write_async(struct regmap *map, unsigned int reg, const void *val, size_t val_len); @@ -452,7 +467,7 @@ void regcache_mark_dirty(struct regmap *map); bool regmap_check_range_table(struct regmap *map, unsigned int reg, const struct regmap_access_table *table); -int regmap_register_patch(struct regmap *map, const struct reg_default *regs, +int regmap_register_patch(struct regmap *map, const struct reg_sequence *regs, int num_regs); int regmap_parse_val(struct regmap *map, const void *buf, unsigned int *val); diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 802e05e..5edd33f 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1366,7 +1366,7 @@ static void arizona_wm5102_set_dac_comp(struct snd_soc_codec *codec, { struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); struct arizona *arizona = priv->arizona; - struct reg_default dac_comp[] = { + struct reg_sequence dac_comp[] = { { 0x80, 0x3 }, { ARIZONA_DAC_COMP_1, 0 }, { ARIZONA_DAC_COMP_2, 0 }, diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 8f40025..2813a1b 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -276,7 +276,7 @@ static const struct snd_soc_codec_driver soc_codec_dev_cs35l32 = { }; /* Current and threshold powerup sequence Pg37 in datasheet */ -static const struct reg_default cs35l32_monitor_patch[] = { +static const struct reg_sequence cs35l32_monitor_patch[] = { { 0x00, 0x99 }, { 0x48, 0x17 }, diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 4de52c9..8b2d059 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1118,7 +1118,7 @@ static const struct snd_soc_codec_driver soc_codec_dev_cs42l52 = { }; /* Current and threshold powerup sequence Pg37 */ -static const struct reg_default cs42l52_threshold_patch[] = { +static const struct reg_sequence cs42l52_threshold_patch[] = { { 0x00, 0x99 }, { 0x3E, 0xBA }, diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 21810e5..bf0fb3d 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -1182,7 +1182,7 @@ static struct snd_soc_codec_driver soc_codec_dev_da7210 = { #if IS_ENABLED(CONFIG_I2C) -static struct reg_default da7210_regmap_i2c_patch[] = { +static struct reg_sequence da7210_regmap_i2c_patch[] = { /* System controller master disable */ { DA7210_STARTUP1, 0x00 }, @@ -1269,7 +1269,7 @@ static struct i2c_driver da7210_i2c_driver = { #if defined(CONFIG_SPI_MASTER) -static struct reg_default da7210_regmap_spi_patch[] = { +static struct reg_sequence da7210_regmap_spi_patch[] = { /* Dummy read to give two pulses over nCS for SPI */ { DA7210_AUX2, 0x00 }, { DA7210_AUX2, 0x00 }, diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 9bc78e5..1ed1f88 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -51,7 +51,7 @@ static const struct regmap_range_cfg rt5640_ranges[] = { .window_len = 0x1, }, }; -static const struct reg_default init_list[] = { +static const struct reg_sequence init_list[] = { {RT5640_PR_BASE + 0x3d, 0x3600}, {RT5640_PR_BASE + 0x12, 0x0aa8}, {RT5640_PR_BASE + 0x14, 0x0aaa}, diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9ce311e..c0f4be4 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -54,7 +54,7 @@ static const struct regmap_range_cfg rt5645_ranges[] = { }, }; -static const struct reg_default init_list[] = { +static const struct reg_sequence init_list[] = { {RT5645_PR_BASE + 0x3d, 0x3600}, {RT5645_PR_BASE + 0x1c, 0xfd20}, {RT5645_PR_BASE + 0x20, 0x611f}, @@ -63,7 +63,7 @@ static const struct reg_default init_list[] = { }; #define RT5645_INIT_REG_LEN ARRAY_SIZE(init_list) -static const struct reg_default rt5650_init_list[] = { +static const struct reg_sequence rt5650_init_list[] = { {0xf6, 0x0100}, }; diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index a3506e1..db9b866 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -46,7 +46,7 @@ static const struct regmap_range_cfg rt5651_ranges[] = { .window_len = 0x1, }, }; -static struct reg_default init_list[] = { +static struct reg_sequence init_list[] = { {RT5651_PR_BASE + 0x3d, 0x3e00}, }; diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index a9123d4..462a91f 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -51,7 +51,7 @@ static const struct regmap_range_cfg rt5670_ranges[] = { .window_len = 0x1, }, }; -static const struct reg_default init_list[] = { +static const struct reg_sequence init_list[] = { { RT5670_PR_BASE + 0x14, 0x9a8a }, { RT5670_PR_BASE + 0x38, 0x3ba1 }, { RT5670_PR_BASE + 0x3d, 0x3640 }, diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 31d969a..b897752 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -54,7 +54,7 @@ static const struct regmap_range_cfg rt5677_ranges[] = { }, }; -static const struct reg_default init_list[] = { +static const struct reg_sequence init_list[] = { {RT5677_ASRC_12, 0x0018}, {RT5677_PR_BASE + 0x3d, 0x364d}, {RT5677_PR_BASE + 0x17, 0x4fc0}, diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index a7cf19b..83ae1eb 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1668,7 +1668,7 @@ static const struct i2c_device_id aic3x_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id); -static const struct reg_default aic3007_class_d[] = { +static const struct reg_sequence aic3007_class_d[] = { /* Class-D speaker driver init; datasheet p. 46 */ { AIC3X_PAGE_SELECT, 0x0D }, { 0xD, 0x0D }, diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index c830832..6c60792 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -897,7 +897,7 @@ static bool wm2200_readable_register(struct device *dev, unsigned int reg) } } -static const struct reg_default wm2200_reva_patch[] = { +static const struct reg_sequence wm2200_reva_patch[] = { { 0x07, 0x0003 }, { 0x102, 0x0200 }, { 0x203, 0x0084 }, diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 4c10cd8..26d79bb 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1247,7 +1247,7 @@ static const struct snd_soc_dapm_route wm5100_dapm_routes[] = { { "PWM2", NULL, "PWM2 Driver" }, }; -static const struct reg_default wm5100_reva_patches[] = { +static const struct reg_sequence wm5100_reva_patches[] = { { WM5100_AUDIO_IF_1_10, 0 }, { WM5100_AUDIO_IF_1_11, 1 }, { WM5100_AUDIO_IF_1_12, 2 }, diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index c5748fd..05492e8 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3495,7 +3495,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8962 = { }; /* Improve power consumption for IN4 DC measurement mode */ -static const struct reg_default wm8962_dc_measure[] = { +static const struct reg_sequence wm8962_dc_measure[] = { { 0xfd, 0x1 }, { 0xcc, 0x40 }, { 0xfd, 0 }, diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 8a8db86..52ec64d 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1595,7 +1595,7 @@ static int wm8993_resume(struct snd_soc_codec *codec) #endif /* Tune DC servo configuration */ -static struct reg_default wm8993_regmap_patch[] = { +static struct reg_sequence wm8993_regmap_patch[] = { { 0x44, 3 }, { 0x56, 3 }, { 0x44, 0 }, -- 2.1.4

3 5

[alsa-devel] [asoc:fix/rt5645 1/1] sound/soc/codecs/rt5645.c:2947:3: warning: 'return' with no value, in function returning non-void
by kbuild test robot 16 Jul '15

16 Jul '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git fix/rt5645 head: 3c2234569cfff83718792f810a9bb855b78c2473 commit: 3c2234569cfff83718792f810a9bb855b78c2473 [1/1] ASoC: rt5645: Check if codec is initialized in workqueue handler config: i386-randconfig-x009-201528 (attached as .config) reproduce: git checkout 3c2234569cfff83718792f810a9bb855b78c2473 # save the attached .config to linux build tree make ARCH=i386 All warnings (new ones prefixed by >>): sound/soc/codecs/rt5645.c: In function 'rt5645_irq_detection': >> sound/soc/codecs/rt5645.c:2947:3: warning: 'return' with no value, in function returning non-void [-Wreturn-type] return; ^ vim +/return +2947 sound/soc/codecs/rt5645.c 2931 { 2932 int btn_type, val; 2933 2934 val = snd_soc_read(codec, RT5650_4BTN_IL_CMD1); 2935 pr_debug("val=0x%x\n", val); 2936 btn_type = val & 0xfff0; 2937 snd_soc_write(codec, RT5650_4BTN_IL_CMD1, val); 2938 2939 return btn_type; 2940 } 2941 2942 static int rt5645_irq_detection(struct rt5645_priv *rt5645) 2943 { 2944 int val, btn_type, gpio_state = 0, report = 0; 2945 2946 if (!rt5645->codec) > 2947 return; 2948 2949 switch (rt5645->pdata.jd_mode) { 2950 case 0: /* Not using rt5645 JD */ 2951 if (rt5645->gpiod_hp_det) { 2952 gpio_state = gpiod_get_value(rt5645->gpiod_hp_det); 2953 dev_dbg(rt5645->codec->dev, "gpio_state = %d\n", 2954 gpio_state); 2955 report = rt5645_jack_detect(rt5645->codec, gpio_state); --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

3 2

[alsa-devel] [PATCH] ALSA: hda/realtek: Enable headphone amp on HP Folio 9480m
by Keith Packard 16 Jul '15

16 Jul '15

This laptop needs GPIO4 pulled high to enable the headphone amplifier. I modelled the patch on the existing GPIO4 code which pulls the line low for the same purpose. Signed-off-by: Keith Packard <keithp(a)keithp.com> --- sound/pci/hda/patch_realtek.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6d01045..9060f1f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3436,6 +3436,23 @@ static void alc280_fixup_hp_gpio4(struct hda_codec *codec, } } +static void alc280_fixup_hp_gpio4_hp_amp(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + /* Pull GPIO4 high to enable headphone amp */ + static const struct hda_verb gpio_init[] = { + { 0x01, AC_VERB_SET_GPIO_MASK, 0x10 }, + { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x10 }, + { 0x01, AC_VERB_SET_GPIO_DATA, 0x10 }, + {} + }; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + snd_hda_add_verbs(codec, gpio_init); + } +} + static void gpio2_mic_hotkey_event(struct hda_codec *codec, struct hda_jack_callback *event) { @@ -4512,6 +4529,7 @@ enum { ALC286_FIXUP_HP_GPIO_LED, ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY, ALC280_FIXUP_HP_DOCK_PINS, + ALC280_FIXUP_HP_GPIO4_HP_AMP, ALC288_FIXUP_DELL_HEADSET_MODE, ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, ALC288_FIXUP_DELL_XPS_13_GPIO6, @@ -5012,6 +5030,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC280_FIXUP_HP_GPIO4 }, + [ALC280_FIXUP_HP_GPIO4_HP_AMP] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc280_fixup_hp_gpio4_hp_amp, + }, [ALC288_FIXUP_DELL_HEADSET_MODE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode_dell_alc288, @@ -5103,6 +5125,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x22b7, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x22bf, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x22cf, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22db, "HP", ALC280_FIXUP_HP_GPIO4_HP_AMP), SND_PCI_QUIRK(0x103c, 0x22dc, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x22fb, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), /* ALC290 */ -- 2.1.4

2 7

[alsa-devel] [asoc:topic/rt5645 7/7] sound/soc/codecs/rt5645.c:2766:22: warning: unused variable 'rt5645'
by kbuild test robot 16 Jul '15

16 Jul '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/rt5645 head: a4e3c5fa955ce0934774f3f3684377dbf2c20eca commit: a4e3c5fa955ce0934774f3f3684377dbf2c20eca [7/7] ASoC: rt5645: Simplify rt5645_enable_push_button_irq config: i386-randconfig-x009-201528 (attached as .config) reproduce: git checkout a4e3c5fa955ce0934774f3f3684377dbf2c20eca # save the attached .config to linux build tree make ARCH=i386 All warnings (new ones prefixed by >>): sound/soc/codecs/rt5645.c: In function 'rt5645_enable_push_button_irq': >> sound/soc/codecs/rt5645.c:2766:22: warning: unused variable 'rt5645' [-Wunused-variable] struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); ^ vim +/rt5645 +2766 sound/soc/codecs/rt5645.c d12d6c4e John Lin 2015-05-12 2750 regmap_write(rt5645->regmap, RT5645_DEPOP_M2, 0x1140); d12d6c4e John Lin 2015-05-12 2751 regmap_write(rt5645->regmap, RT5645_DEPOP_M1, 0x0000); d12d6c4e John Lin 2015-05-12 2752 regmap_write(rt5645->regmap, RT5645_GEN_CTRL2, 0x4020); d12d6c4e John Lin 2015-05-12 2753 regmap_write(rt5645->regmap, RT5645_PWR_ANLG2, 0x0006); d12d6c4e John Lin 2015-05-12 2754 regmap_write(rt5645->regmap, RT5645_PWR_DIG2, 0x0000); d12d6c4e John Lin 2015-05-12 2755 msleep(350); d12d6c4e John Lin 2015-05-12 2756 d12d6c4e John Lin 2015-05-12 2757 regcache_cache_bypass(rt5645->regmap, false); d12d6c4e John Lin 2015-05-12 2758 d12d6c4e John Lin 2015-05-12 2759 return ret; d12d6c4e John Lin 2015-05-12 2760 } d12d6c4e John Lin 2015-05-12 2761 6e747d53 Bard Liao 2015-04-28 2762 static void rt5645_enable_push_button_irq(struct snd_soc_codec *codec, 6e747d53 Bard Liao 2015-04-28 2763 bool enable) f3fa1bbd Oder Chiou 2014-09-19 2764 { e2ada818 Lars-Peter Clausen 2015-07-06 2765 struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); f3fa1bbd Oder Chiou 2014-09-19 @2766 struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); f3fa1bbd Oder Chiou 2014-09-19 2767 6e747d53 Bard Liao 2015-04-28 2768 if (enable) { a4e3c5fa Nicolas Boichat 2015-07-16 2769 snd_soc_dapm_force_enable_pin(dapm, "ADC L power"); a4e3c5fa Nicolas Boichat 2015-07-16 2770 snd_soc_dapm_force_enable_pin(dapm, "ADC R power"); a4e3c5fa Nicolas Boichat 2015-07-16 2771 snd_soc_dapm_sync(dapm); 22f5d9f8 Nicolas Boichat 2015-06-10 2772 6e747d53 Bard Liao 2015-04-28 2773 snd_soc_update_bits(codec, 6e747d53 Bard Liao 2015-04-28 2774 RT5645_INT_IRQ_ST, 0x8, 0x8); :::::: The code at line 2766 was first introduced by commit :::::: f3fa1bbd836a7d6efb2abd506ed8e24096f39062 ASoC: rt5645: Add headset detect function :::::: TO: Oder Chiou <oder_chiou(a)realtek.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH v3] ASoC: cs4349: Add support for Cirrus Logic CS4349
by timothyc.howe＠gmail.com 16 Jul '15

16 Jul '15

From: Tim Howe <tim.howe(a)cirrus.com> Signed-off-by: Tim Howe <tim.howe(a)cirrus.com> --- Documentation/devicetree/bindings/sound/cs4349.txt | 19 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs4349.c | 401 +++++++++++++++++++++ sound/soc/codecs/cs4349.h | 146 ++++++++ 5 files changed, 574 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs4349.txt create mode 100644 sound/soc/codecs/cs4349.c create mode 100644 sound/soc/codecs/cs4349.h diff --git a/Documentation/devicetree/bindings/sound/cs4349.txt b/Documentation/devicetree/bindings/sound/cs4349.txt new file mode 100644 index 0000000..54c117b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4349.txt @@ -0,0 +1,19 @@ +CS4349 audio CODEC + +Required properties: + + - compatible : "cirrus,cs4349" + + - reg : the I2C address of the device for I2C + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. + +Example: + +codec: cs4349@48 { + compatible = "cirrus,cs4349"; + reg = <0x48>; + reset-gpios = <&gpio 54 0>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7223c7f..4a78606 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -54,6 +54,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271_I2C if I2C select SND_SOC_CS4271_SPI if SPI_MASTER select SND_SOC_CS42XX8_I2C if I2C + select SND_SOC_CS4349 if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI select SND_SOC_DA7213 if I2C @@ -413,6 +414,11 @@ config SND_SOC_CS42XX8_I2C select SND_SOC_CS42XX8 select REGMAP_I2C +# Cirrus Logic CS4349 HiFi DAC +config SND_SOC_CS4349 + tristate "Cirrus Logic CS4349 CODEC" + depends on I2C + config SND_SOC_CX20442 tristate depends on TTY diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 9f7136c..36a9f37 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -46,6 +46,7 @@ snd-soc-cs4271-i2c-objs := cs4271-i2c.o snd-soc-cs4271-spi-objs := cs4271-spi.o snd-soc-cs42xx8-objs := cs42xx8.o snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o +snd-soc-cs4349-objs := cs4349.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -237,6 +238,7 @@ obj-$(CONFIG_SND_SOC_CS4271_I2C) += snd-soc-cs4271-i2c.o obj-$(CONFIG_SND_SOC_CS4271_SPI) += snd-soc-cs4271-spi.o obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o +obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o obj-$(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/cs4349.c b/sound/soc/codecs/cs4349.c new file mode 100644 index 0000000..fb8d1e1 --- /dev/null +++ b/sound/soc/codecs/cs4349.c @@ -0,0 +1,401 @@ +/* + * cs4349.c -- CS4349 ALSA Soc Audio driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Authors: Tim Howe <Tim.Howe(a)cirrus.com> + * + * 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/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/platform_device.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/of_device.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include "cs4349.h" + + +static const struct reg_default cs4349_reg_defaults[] = { + { 2, 0x00 }, /* r02 - Mode Control */ + { 3, 0x09 }, /* r03 - Volume, Mixing and Inversion Control */ + { 4, 0x81 }, /* r04 - Mute Control */ + { 5, 0x00 }, /* r05 - Channel A Volume Control */ + { 6, 0x00 }, /* r06 - Channel B Volume Control */ + { 7, 0xB1 }, /* r07 - Ramp and Filter Control */ + { 8, 0x1C }, /* r08 - Misc. Control */ +}; + +/* Private data for the CS4349 */ +struct cs4349_private { + struct regmap *regmap; + struct cs4349_platform_data pdata; + struct gpio_desc *reset_gpio; + unsigned int mode; + int rate; +}; + +static bool cs4349_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS4349_CHIPID: + case CS4349_MODE: + case CS4349_VMI: + case CS4349_MUTE: + case CS4349_VOLA: + case CS4349_VOLB: + case CS4349_RMPFLT: + case CS4349_MISC: + return true; + default: + return false; + } +} + +static int cs4349_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4349_private *cs4349 = snd_soc_codec_get_drvdata(codec); + unsigned int fmt; + + fmt = format & SND_SOC_DAIFMT_FORMAT_MASK; + + switch (fmt) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + cs4349->mode = format & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + return -EINVAL; + } + + return 0; +} + +static int cs4349_pcm_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 cs4349_private *cs4349 = snd_soc_codec_get_drvdata(codec); + int mode, fmt, ret; + + mode = snd_soc_read(codec, CS4349_MODE); + cs4349->rate = params_rate(params); + + switch (cs4349->mode) { + case SND_SOC_DAIFMT_I2S: + mode |= MODE_FORMAT(DIF_I2S); + break; + case SND_SOC_DAIFMT_LEFT_J: + mode |= MODE_FORMAT(DIF_LEFT_JST); + break; + case SND_SOC_DAIFMT_RIGHT_J: + switch (params_width(params)) { + case 16: + fmt = DIF_RGHT_JST16; + break; + case 24: + fmt = DIF_RGHT_JST24; + break; + default: + return -EINVAL; + } + mode |= MODE_FORMAT(fmt); + break; + default: + return -EINVAL; + } + + ret = snd_soc_write(codec, CS4349_MODE, mode); + if (ret < 0) + return ret; + + return 0; +} + +static int cs4349_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + int reg; + + reg = 0; + if (mute) + reg = MUTE_AB_MASK; + + return snd_soc_update_bits(codec, CS4349_MUTE, MUTE_AB_MASK, reg); +} + +static DECLARE_TLV_DB_SCALE(dig_tlv, -12750, 50, 0); + +static const char * const chan_mix_texts[] = { + "Mute", "MuteA", "MuteA SwapB", "MuteA MonoB", "SwapA MuteB", + "BothR", "Swap", "SwapA MonoB", "MuteB", "Normal", "BothL", + "MonoB", "MonoA MuteB", "MonoA", "MonoA SwapB", "Mono", + /*Normal == Channel A = Left, Channel B = Right*/ +}; + +static const char * const fm_texts[] = { + "Auto", "Single", "Double", "Quad", +}; + +static const char * const deemph_texts[] = { + "None", "44.1k", "48k", "32k", +}; + +static const char * const softr_zeroc_texts[] = { + "Immediate", "Zero Cross", "Soft Ramp", "SR on ZC", +}; + +static int deemph_values[] = { + 0, 4, 8, 12, +}; + +static int softr_zeroc_values[] = { + 0, 64, 128, 192, +}; + +static const struct soc_enum chan_mix_enum = + SOC_ENUM_SINGLE(CS4349_VMI, 0, + ARRAY_SIZE(chan_mix_texts), + chan_mix_texts); + +static const struct soc_enum fm_mode_enum = + SOC_ENUM_SINGLE(CS4349_MODE, 0, + ARRAY_SIZE(fm_texts), + fm_texts); + +static SOC_VALUE_ENUM_SINGLE_DECL(deemph_enum, CS4349_MODE, 0, DEM_MASK, + deemph_texts, deemph_values); + +static SOC_VALUE_ENUM_SINGLE_DECL(softr_zeroc_enum, CS4349_RMPFLT, 0, + SR_ZC_MASK, softr_zeroc_texts, + softr_zeroc_values); + +static const struct snd_kcontrol_new cs4349_snd_controls[] = { + SOC_DOUBLE_R_TLV("Master Playback Volume", + CS4349_VOLA, CS4349_VOLB, 0, 0xFF, 1, dig_tlv), + SOC_ENUM("Functional Mode", fm_mode_enum), + SOC_ENUM("De-Emphasis Control", deemph_enum), + SOC_ENUM("Soft Ramp Zero Cross Control", softr_zeroc_enum), + SOC_ENUM("Channel Mixer", chan_mix_enum), + SOC_SINGLE("VolA = VolB Switch", CS4349_VMI, 7, 1, 0), + SOC_SINGLE("InvertA Switch", CS4349_VMI, 6, 1, 0), + SOC_SINGLE("InvertB Switch", CS4349_VMI, 5, 1, 0), + SOC_SINGLE("Auto-Mute Switch", CS4349_MUTE, 7, 1, 0), + SOC_SINGLE("MUTEC A = B Switch", CS4349_MUTE, 5, 1, 0), + SOC_SINGLE("Soft Ramp Up Switch", CS4349_RMPFLT, 5, 1, 0), + SOC_SINGLE("Soft Ramp Down Switch", CS4349_RMPFLT, 4, 1, 0), + SOC_SINGLE("Slow Roll Off Filter Switch", CS4349_RMPFLT, 2, 1, 0), + SOC_SINGLE("Freeze Switch", CS4349_MISC, 5, 1, 0), + SOC_SINGLE("Popguard Switch", CS4349_MISC, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget cs4349_dapm_widgets[] = { + SND_SOC_DAPM_DAC("HiFi DAC", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_OUTPUT("OutputA"), + SND_SOC_DAPM_OUTPUT("OutputB"), +}; + +static const struct snd_soc_dapm_route cs4349_routes[] = { + {"DAC Playback", NULL, "OutputA"}, + {"DAC Playback", NULL, "OutputB"}, + + {"OutputA", NULL, "HiFi DAC"}, + {"OutputB", NULL, "HiFi DAC"}, +}; + +#define CS4349_PCM_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE) + +#define CS4349_PCM_RATES SNDRV_PCM_RATE_8000_192000 + +static const struct snd_soc_dai_ops cs4349_dai_ops = { + .hw_params = cs4349_pcm_hw_params, + .set_fmt = cs4349_set_dai_fmt, + .digital_mute = cs4349_digital_mute, +}; + +static struct snd_soc_dai_driver cs4349_dai = { + .name = "cs4349_hifi", + .playback = { + .stream_name = "DAC Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS4349_PCM_RATES, + .formats = CS4349_PCM_FORMATS, + }, + .ops = &cs4349_dai_ops, + .symmetric_rates = 1, +}; + +static struct snd_soc_codec_driver soc_codec_dev_cs4349 = { + .controls = cs4349_snd_controls, + .num_controls = ARRAY_SIZE(cs4349_snd_controls), + + .dapm_widgets = cs4349_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs4349_dapm_widgets), + .dapm_routes = cs4349_routes, + .num_dapm_routes = ARRAY_SIZE(cs4349_routes), +}; + +static struct regmap_config cs4349_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS4349_NUMREGS, + .reg_defaults = cs4349_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4349_reg_defaults), + .readable_reg = cs4349_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs4349_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct cs4349_private *cs4349; + struct cs4349_platform_data *pdata = dev_get_platdata(&client->dev); + int ret = 0; + + cs4349 = devm_kzalloc(&client->dev, sizeof(*cs4349), GFP_KERNEL); + if (!cs4349) + return -ENOMEM; + + cs4349->regmap = devm_regmap_init_i2c(client, &cs4349_regmap); + if (IS_ERR(cs4349->regmap)) { + ret = PTR_ERR(cs4349->regmap); + dev_err(&client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + if (pdata) + cs4349->pdata = *pdata; + + /* Reset the Device */ + cs4349->reset_gpio = devm_gpiod_get_optional(&client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs4349->reset_gpio)) + return PTR_ERR(cs4349->reset_gpio); + + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 1); + + i2c_set_clientdata(client, cs4349); + + return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4349, + &cs4349_dai, 1); +} + +static int cs4349_i2c_remove(struct i2c_client *client) +{ + struct cs4349_private *cs4349 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + /* Hold down reset */ + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 0); + + return 0; +} + +#ifdef CONFIG_PM +static int cs4349_runtime_suspend(struct device *dev) +{ + struct cs4349_private *cs4349 = dev_get_drvdata(dev); + struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); + int ret; + + ret = snd_soc_update_bits(rtd->codec, CS4349_MISC, PWR_DWN, 1); + if (ret < 0) + return ret; + + regcache_sync(cs4349->regmap); + regcache_cache_only(cs4349->regmap, true); + + /* Hold down reset */ + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 0); + + return 0; +} + +static int cs4349_runtime_resume(struct device *dev) +{ + struct cs4349_private *cs4349 = dev_get_drvdata(dev); + struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); + int ret; + + ret = snd_soc_update_bits(rtd->codec, CS4349_MISC, PWR_DWN, 0); + if (ret < 0) + return ret; + + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 1); + + regcache_cache_only(cs4349->regmap, false); + regcache_sync(cs4349->regmap); + + return 0; +} +#endif + +static const struct dev_pm_ops cs4349_runtime_pm = { + SET_RUNTIME_PM_OPS(cs4349_runtime_suspend, cs4349_runtime_resume, + NULL) +}; + +static const struct of_device_id cs4349_of_match[] = { + { .compatible = "cirrus,cs4349", }, + {}, +}; + +MODULE_DEVICE_TABLE(of, cs4349_of_match); + +static const struct i2c_device_id cs4349_i2c_id[] = { + {"cs4349", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs4349_i2c_id); + +static struct i2c_driver cs4349_i2c_driver = { + .driver = { + .name = "cs4349", + .owner = THIS_MODULE, + .of_match_table = cs4349_of_match, + }, + .id_table = cs4349_i2c_id, + .probe = cs4349_i2c_probe, + .remove = cs4349_i2c_remove, +}; + +module_i2c_driver(cs4349_i2c_driver); + +MODULE_AUTHOR("Tim Howe <tim.howe(a)cirrus.com>"); +MODULE_DESCRIPTION("Cirrus Logic CS4349 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs4349.h b/sound/soc/codecs/cs4349.h new file mode 100644 index 0000000..3884a89 --- /dev/null +++ b/sound/soc/codecs/cs4349.h @@ -0,0 +1,146 @@ +/* + * ALSA SoC CS4349 codec driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Author: Tim Howe <Tim.Howe(a)cirrus.com> + * + * 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. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + */ + +#ifndef __CS4349_H__ +#define __CS4349_H__ + +struct cs4349_platform_data { + + /* GPIO for Reset */ + unsigned int gpio_nreset; + +}; + +/* CS4349 registers addresses */ +#define CS4349_CHIPID 0x01 /* Device and Rev ID, Read Only */ +#define CS4349_MODE 0x02 /* Mode Control */ +#define CS4349_VMI 0x03 /* Volume, Mixing, Inversion Control */ +#define CS4349_MUTE 0x04 /* Mute Control */ +#define CS4349_VOLA 0x05 /* DAC Channel A Volume Control */ +#define CS4349_VOLB 0x06 /* DAC Channel B Volume Control */ +#define CS4349_RMPFLT 0x07 /* Ramp and Filter Control */ +#define CS4349_MISC 0x08 /* Power Down,Freeze Control,Pop Stop*/ + +#define CS4349_FIRSTREG 0x01 +#define CS4349_LASTREG 0x08 +#define CS4349_NUMREGS (CS4349_LASTREG - CS4349_FIRSTREG + 1) +#define CS4349_I2C_INCR 0x80 + + +/* Device and Revision ID */ +#define CS4349_REVA 0xF0 /* Rev A */ +#define CS4349_REVB 0xF1 /* Rev B */ +#define CS4349_REVC2 0xFF /* Rev C2 */ + + +/* PDN_DONE Poll Maximum + * If soft ramp is set it will take much longer to power down + * the system. + */ +#define PDN_POLL_MAX 900 + + +/* Bitfield Definitions */ + +/* CS4349_MODE */ +/* (Digital Interface Format, De-Emphasis Control, Functional Mode */ +#define DIF2 (1 << 6) +#define DIF1 (1 << 5) +#define DIF0 (1 << 4) +#define DEM1 (1 << 3) +#define DEM0 (1 << 2) +#define FM1 (1 << 1) +#define DIF_LEFT_JST 0x00 +#define DIF_I2S 0x01 +#define DIF_RGHT_JST16 0x02 +#define DIF_RGHT_JST24 0x03 +#define DIF_TDM0 0x04 +#define DIF_TDM1 0x05 +#define DIF_TDM2 0x06 +#define DIF_TDM3 0x07 +#define DIF_MASK 0x70 +#define MODE_FORMAT(x) (((x)&7)<<4) +#define DEM_MASK 0x0C +#define NO_DEM 0x00 +#define DEM_441 0x04 +#define DEM_48K 0x08 +#define DEM_32K 0x0C +#define FM_AUTO 0x00 +#define FM_SNGL 0x01 +#define FM_DBL 0x02 +#define FM_QUAD 0x03 +#define FM_SNGL_MIN 30000 +#define FM_SNGL_MAX 54000 +#define FM_DBL_MAX 108000 +#define FM_QUAD_MAX 216000 +#define FM_MASK 0x03 + +/* CS4349_VMI (VMI = Volume, Mixing and Inversion Controls) */ +#define VOLBISA (1 << 7) +#define VOLAISB (1 << 7) +/* INVERT_A only available for Left Jstfd, Right Jstfd16 and Right Jstfd24 */ +#define INVERT_A (1 << 6) +/* INVERT_B only available for Left Jstfd, Right Jstfd16 and Right Jstfd24 */ +#define INVERT_B (1 << 5) +#define ATAPI3 (1 << 3) +#define ATAPI2 (1 << 2) +#define ATAPI1 (1 << 1) +#define ATAPI0 (1 << 0) +#define MUTEAB 0x00 +#define MUTEA_RIGHTB 0x01 +#define MUTEA_LEFTB 0x02 +#define MUTEA_SUMLRDIV2B 0x03 +#define RIGHTA_MUTEB 0x04 +#define RIGHTA_RIGHTB 0x05 +#define RIGHTA_LEFTB 0x06 +#define RIGHTA_SUMLRDIV2B 0x07 +#define LEFTA_MUTEB 0x08 +#define LEFTA_RIGHTB 0x09 /* Default */ +#define LEFTA_LEFTB 0x0A +#define LEFTA_SUMLRDIV2B 0x0B +#define SUMLRDIV2A_MUTEB 0x0C +#define SUMLRDIV2A_RIGHTB 0x0D +#define SUMLRDIV2A_LEFTB 0x0E +#define SUMLRDIV2_AB 0x0F +#define CHMIX_MASK 0x0F + +/* CS4349_MUTE */ +#define AUTOMUTE (1 << 7) +#define MUTEC_AB (1 << 5) +#define MUTE_A (1 << 4) +#define MUTE_B (1 << 3) +#define MUTE_AB_MASK 0x18 + +/* CS4349_RMPFLT (Ramp and Filter Control) */ +#define SCZ1 (1 << 7) +#define SCZ0 (1 << 6) +#define RMP_UP (1 << 5) +#define RMP_DN (1 << 4) +#define FILT_SEL (1 << 2) +#define IMMDT_CHNG 0x31 +#define ZEROCRSS 0x71 +#define SOFT_RMP 0xB1 +#define SFTRMP_ZEROCRSS 0xF1 +#define SR_ZC_MASK 0xC0 + +/* CS4349_MISC */ +#define PWR_DWN (1 << 7) +#define FREEZE (1 << 5) +#define POPG_EN (1 << 4) + +#endif /* __CS4349_H__ */ -- 2.4.5

2 1

[alsa-devel] [PATCH] ASoC: rt298: remove meanless pr_info
by Bard Liao 16 Jul '15

16 Jul '15

This line doesn't include any information. So, remove it. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt298.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 7c4bcb6..6ff68a3 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -1144,8 +1144,6 @@ static int rt298_i2c_probe(struct i2c_client *i2c, const struct acpi_device_id *acpiid; int i, ret; - pr_info("%s\n", __func__); - rt298 = devm_kzalloc(&i2c->dev, sizeof(*rt298), GFP_KERNEL); if (NULL == rt298) -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH 1/2] ASoC: rt5645: Update dapm pins when the card is not instantiated yet
by Nicolas Boichat 16 Jul '15

16 Jul '15

This makes sure the dapm state is consistent when the card is instantiated. However, if the card is not instantiated yet, we still update the registers manually in the "jack in" case, so that we can immediately report if a mic is present or not. Disabling "Mic Det Power" after detection, and on jack out, can wait until the card gets instantiated. Signed-off-by: Nicolas Boichat <drinkcat(a)chromium.org> --- sound/soc/codecs/rt5645.c | 36 +++++++++++------------------------- 1 file changed, 11 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 093a41d..a56bca6 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2805,12 +2805,11 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) if (jack_insert) { regmap_write(rt5645->regmap, RT5645_CHARGE_PUMP, 0x0006); - if (codec->component.card->instantiated) { - /* for jack type detect */ - snd_soc_dapm_force_enable_pin(dapm, "LDO2"); - snd_soc_dapm_force_enable_pin(dapm, "Mic Det Power"); - snd_soc_dapm_sync(dapm); - } else { + /* for jack type detect */ + snd_soc_dapm_force_enable_pin(dapm, "LDO2"); + snd_soc_dapm_force_enable_pin(dapm, "Mic Det Power"); + snd_soc_dapm_sync(dapm); + if (!dapm->card->instantiated) { /* Power up necessary bits for JD if dapm is not ready yet */ regmap_update_bits(rt5645->regmap, RT5645_PWR_ANLG1, @@ -2841,12 +2840,8 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) rt5645_enable_push_button_irq(codec, true); } } else { - if (codec->component.card->instantiated) { - snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); - snd_soc_dapm_sync(dapm); - } else - regmap_update_bits(rt5645->regmap, - RT5645_PWR_VOL, RT5645_PWR_MIC_DET, 0); + snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); + snd_soc_dapm_sync(dapm); rt5645->jack_type = SND_JACK_HEADPHONE; } @@ -2855,19 +2850,10 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) if (rt5645->en_button_func) rt5645_enable_push_button_irq(codec, false); else { - if (codec->component.card->instantiated) { - if (rt5645->pdata.jd_mode == 0) - snd_soc_dapm_disable_pin(dapm, "LDO2"); - snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); - snd_soc_dapm_sync(dapm); - } else { - if (rt5645->pdata.jd_mode == 0) - regmap_update_bits(rt5645->regmap, - RT5645_PWR_MIXER, - RT5645_PWR_LDO2, 0); - regmap_update_bits(rt5645->regmap, - RT5645_PWR_VOL, RT5645_PWR_MIC_DET, 0); - } + if (rt5645->pdata.jd_mode == 0) + snd_soc_dapm_disable_pin(dapm, "LDO2"); + snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); + snd_soc_dapm_sync(dapm); } } -- 2.4.3.573.g4eafbef

2 3

[alsa-devel] Applied "ASoC: rt5645: Check if codec is initialized in workqueue handler" to the asoc tree
by Mark Brown 16 Jul '15

16 Jul '15

The patch ASoC: rt5645: Check if codec is initialized in workqueue handler has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 3c2234569cfff83718792f810a9bb855b78c2473 Mon Sep 17 00:00:00 2001 From: Nicolas Boichat <drinkcat(a)chromium.org> Date: Thu, 16 Jul 2015 11:30:58 +0800 Subject: [PATCH] ASoC: rt5645: Check if codec is initialized in workqueue handler rt5645_jack_detect_work needs rt5645->codec to be initialized to setup dapm pins. Also, reporting jack events is useless, as the jacks cannot be set before the codec is ready. Since we manually call the interrupt handler in rt5645_set_jack_detect, the initial jack state will be reported correctly, and dapm pins will be setup at that time. Signed-off-by: Nicolas Boichat <drinkcat(a)chromium.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5645.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9ce311e..d1d49e8 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2943,6 +2943,9 @@ static int rt5645_irq_detection(struct rt5645_priv *rt5645) { int val, btn_type, gpio_state = 0, report = 0; + if (!rt5645->codec) + return; + switch (rt5645->pdata.jd_mode) { case 0: /* Not using rt5645 JD */ if (rt5645->gpiod_hp_det) { -- 2.1.4

1 0

[alsa-devel] [PATCH] ALSA: hda: fix kstrdup return value
by kernelpatch_update 16 Jul '15

16 Jul '15

In kstrdup we should return -ENOMEM when it reports an memory allocation failure, while the -ENODEV is referred to a failure in finding the cpu node in the device tree. Signed-off-by: kernelpatch_update <kernelpatch_update(a)163.com> --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5de3c5d..d78fa71 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -975,7 +975,7 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, if (codec->bus->modelname) { codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL); if (!codec->modelname) { - err = -ENODEV; + err = -ENOMEM; goto error; } } -- 1.9.1

2 2

[alsa-devel] [PATCH 1/1] ASoC: sti-sas: fix x86 compilation issue.
by Arnaud Pouliquen 16 Jul '15

16 Jul '15

Fixes for compilation warnings and errors reported by Kbuild test robot for x86_64 and i386. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- sound/soc/codecs/sti-sas.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c index 32db2c2..39f3c8b 100644 --- a/sound/soc/codecs/sti-sas.c +++ b/sound/soc/codecs/sti-sas.c @@ -227,8 +227,8 @@ static int stih416_dac_probe(struct snd_soc_dai *dai) dac->rst = devm_reset_control_get(codec->dev, "dac_rst"); if (IS_ERR(dac->rst)) { dev_err(dai->codec->dev, - "%s: ERROR: DAC reset control not defined (%d)!\n", - __func__, (int)dac->rst); + "%s: ERROR: DAC reset control not defined !\n", + __func__); dac->rst = NULL; return -EFAULT; } @@ -547,6 +547,7 @@ static int sti_sas_driver_probe(struct platform_device *pdev) { struct device_node *pnode = pdev->dev.of_node; struct sti_sas_data *drvdata; + const struct of_device_id *of_id; /* Allocate device structure */ drvdata = devm_kzalloc(&pdev->dev, sizeof(struct sti_sas_data), @@ -555,12 +556,13 @@ static int sti_sas_driver_probe(struct platform_device *pdev) return -ENOMEM; /* Populate data structure depending on compatibility */ - if (!of_match_node(sti_sas_dev_match, pnode)->data) { + of_id = of_match_node(sti_sas_dev_match, pnode); + if (!of_id->data) { dev_err(&pdev->dev, "data associated to device is missing"); return -EINVAL; } - drvdata->dev_data = of_match_node(sti_sas_dev_match, pnode)->data; + drvdata->dev_data = (struct sti_sas_dev_data *)of_id->data; /* Initialise device structure */ drvdata->dev = &pdev->dev; -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: fsi: Remove obsolete sh_fsi2 platform_device_id entry
by Geert Uytterhoeven 16 Jul '15

16 Jul '15

Since the removal of the r8a7740 legacy board code in commit 1fa59bda21c7fa36 ("ARM: shmobile: Remove legacy board code for Armadillo-800 EVA"), all former users of the "sh_fsi2" platform device name are only supported in generic DT-only ARM multi-platform builds. The driver doesn't need to match platform devices by name anymore, hence remove the corresponding platform_device_id entry. Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- The commit mentioned above is in -next, destined for v4.3. --- sound/soc/sh/fsi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 142c066eaee2ecb1..0215c78cbddf5491 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1911,7 +1911,6 @@ MODULE_DEVICE_TABLE(of, fsi_of_match); static const struct platform_device_id fsi_id_table[] = { { "sh_fsi", (kernel_ulong_t)&fsi1_core }, - { "sh_fsi2", (kernel_ulong_t)&fsi2_core }, {}, }; MODULE_DEVICE_TABLE(platform, fsi_id_table); -- 1.9.1

4 3

[alsa-devel] [PATCH v3] ASoC: rt5645: Check if codec is initialized in workqueue handler
by Nicolas Boichat 16 Jul '15

16 Jul '15

rt5645_jack_detect_work needs rt5645->codec to be initialized to setup dapm pins. Also, reporting jack events is useless, as the jacks cannot be set before the codec is ready. Since we manually call the interrupt handler in rt5645_set_jack_detect, the initial jack state will be reported correctly, and dapm pins will be setup at that time. Signed-off-by: Nicolas Boichat <drinkcat(a)chromium.org> --- sound/soc/codecs/rt5645.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 093a41d..ff75d04 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2918,6 +2918,9 @@ static void rt5645_jack_detect_work(struct work_struct *work) container_of(work, struct rt5645_priv, jack_detect_work.work); int val, btn_type, gpio_state = 0, report = 0; + if (!rt5645->codec) + return; + switch (rt5645->pdata.jd_mode) { case 0: /* Not using rt5645 JD */ if (rt5645->gpiod_hp_det) { -- 2.4.3.573.g4eafbef

1 0

[alsa-devel] [PATCH v2 0/3] ASoC: rt5645: Remove codec dependency in workqueue handler
by Nicolas Boichat 16 Jul '15

16 Jul '15

Same issue as https://patchwork.kernel.org/patch/6694351/, but reworked fix. The first 2 patches are preparatory work for the 3rd one, which fixes occasional kernel panics on boot. Nicolas Boichat (3): ASoC: rt5645: Simplify rt5645_enable_push_button_irq ASoC: rt5645: Remove irq_jack_detection function ASoC: rt5645: Remove codec dependency in workqueue handler sound/soc/codecs/rt5645.c | 185 +++++++++++++++++++++++----------------------- sound/soc/codecs/rt5645.h | 1 + 2 files changed, 95 insertions(+), 91 deletions(-) -- 2.4.3.573.g4eafbef

3 14

[alsa-devel] [PATCH] ALSA: pcm_dmaengine: Run complete cb only if runtime check passes
by Misael Lopez Cruz 15 Jul '15

15 Jul '15

The DMA complete callback uses substream's runtime data, so make sure the runtime check passes before going forward. The callback could run while the stream is closing or after it's already closed. By then, the substream's runtime data might already been freed and cause NULL pointer dereferences. Signed-off-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Reviewed-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/core/pcm_dmaengine.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index fba365a..061a8f7 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -133,7 +133,12 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_set_config_from_dai_data); static void dmaengine_pcm_dma_complete(void *arg) { struct snd_pcm_substream *substream = arg; - struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + struct dmaengine_pcm_runtime_data *prtd; + + if (PCM_RUNTIME_CHECK(substream)) + return; + + prtd = substream_to_prtd(substream); prtd->pos += snd_pcm_lib_period_bytes(substream); if (prtd->pos >= snd_pcm_lib_buffer_bytes(substream)) -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: Intel: Get correct usage_count value to load firmware
by yang.a.fang＠intel.com 15 Jul '15

15 Jul '15

From: Shilpa Sreeramalu <shilpa.sreeramalu(a)intel.com> The usage_count variable was read before it was set to the correct value, due to which the firmware load was failing. Because of this IPC messages sent to the firmware were timing out causing a delay of about 1 second while playing audio from the internal speakers. With this patch the usage_count is read after the function call pm_runtime_get_sync which will increment the usage_count variable and the firmware load is successful and all the IPC messages are processed correctly. Cc: stable(a)vger.kernel.org Signed-off-by: Shilpa Sreeramalu <shilpa.sreeramalu(a)intel.com> --- sound/soc/intel/atom/sst/sst_drv_interface.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c index 620da1d..440b953 100644 --- a/sound/soc/intel/atom/sst/sst_drv_interface.c +++ b/sound/soc/intel/atom/sst/sst_drv_interface.c @@ -42,6 +42,11 @@ #define MIN_FRAGMENT_SIZE (50 * 1024) #define MAX_FRAGMENT_SIZE (1024 * 1024) #define SST_GET_BYTES_PER_SAMPLE(pcm_wd_sz) (((pcm_wd_sz + 15) >> 4) << 1) +#ifdef PM_CONFIG +#define GET_USAGE_COUNT(dev) (atomic_read(&dev->power.usage_count)) +#else +#define GET_USAGE_COUNT(dev) 1 +#endif int free_stream_context(struct intel_sst_drv *ctx, unsigned int str_id) { @@ -141,15 +146,9 @@ static int sst_power_control(struct device *dev, bool state) int ret = 0; int usage_count = 0; -#ifdef CONFIG_PM - usage_count = atomic_read(&dev->power.usage_count); -#else - usage_count = 1; -#endif - if (state == true) { ret = pm_runtime_get_sync(dev); - + usage_count = GET_USAGE_COUNT(dev); dev_dbg(ctx->dev, "Enable: pm usage count: %d\n", usage_count); if (ret < 0) { dev_err(ctx->dev, "Runtime get failed with err: %d\n", ret); @@ -164,6 +163,7 @@ static int sst_power_control(struct device *dev, bool state) } } } else { + usage_count = GET_USAGE_COUNT(dev); dev_dbg(ctx->dev, "Disable: pm usage count: %d\n", usage_count); return sst_pm_runtime_put(ctx); } -- 1.7.9.5

4 6

[alsa-devel] [PATCH] Softvol: Allow per process volume control.
by Jimmy 15 Jul '15

15 Jul '15

This change uses ephemeral pcm controls to provide per process volume control in conjunction with dmix. It basically generates a new control every time a softvol PCM is opened and deletes it afterwards. It also needs to generate a unique name for the control and I have included one method of doing that. This patch is totally non-breaking and backwards compatible, apart from one call to snd_ctl_elem_unlock (see below). Changes to asoundrc: * pcm.softvol.per_process BOOL has been added to enable these changes, defaults to false. * when per_process is true then bits of text in pcm.softvol.control.name surrounded by curly braces ('{' and '}') are treated as substitution variables (eg "softvol.{pid}") so that uniquely named controls can be generated. Changes to pcm_softvol.c * new int (really a bool) in snd_pcm_softvol_t that is read in _...open() and passed through to the internal open method because that seemed to be the thing to do and then used in softvol_load_control(). * a big ugly per_proc_format_ctl_name() function has been added that generates the new hopefully unique names. It talks to /proc/ and does a bunch string manipulation. There probably are no off-by-one errors left in there. * per process controls are closed along with the pcm. * I added a call to snd_ctl_elem_unlock() when a new control is created and *didn't* put a guard on it because I don't understand why anyone would want the control locked anyway ... It seems that the first time that softvol is opened you can't use the control so you have to close and open it again. Presumably there is a reason for this because unlock isn't called anywhere else, but I don't know what it is. Nothing else calls unlock either, maybe nothing else makes new on-demand ctls. I'm open to suggestions like "get rid of the templating" and "use better config variable names" but as far as I can tell the basic idea is useful and simple. --- src/pcm/pcm_softvol.c | 148 ++++++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 144 insertions(+), 4 deletions(-) diff --git a/src/pcm/pcm_softvol.c b/src/pcm/pcm_softvol.c index c6cfd8896b26..a03b672a6f42 100644 --- a/src/pcm/pcm_softvol.c +++ b/src/pcm/pcm_softvol.c @@ -31,6 +31,11 @@ #include "pcm_local.h" #include "pcm_plugin.h" +#include <sys/types.h> +#include <unistd.h> +#include <errno.h> +#include <libgen.h> //basename + #ifndef PIC /* entry for static linking */ const char *_snd_module_pcm_softvol = ""; @@ -51,6 +56,7 @@ typedef struct { double min_dB; double max_dB; unsigned int *dB_value; + int per_process; } snd_pcm_softvol_t; #define VOL_SCALE_SHIFT 16 @@ -408,6 +414,17 @@ static void softvol_free(snd_pcm_softvol_t *svol) static int snd_pcm_softvol_close(snd_pcm_t *pcm) { snd_pcm_softvol_t *svol = pcm->private_data; + snd_ctl_elem_info_t *cinfo; + + if (svol->per_process) { + snd_ctl_elem_info_alloca(&cinfo); + snd_ctl_elem_info_set_id(cinfo, &svol->elem.id); + // Ignore non-existence at this point. + if (!snd_ctl_elem_info(svol->ctl, cinfo)) { + snd_ctl_elem_remove(svol->ctl, &cinfo->id); + } + } + softvol_free(svol); return 0; } @@ -689,6 +706,92 @@ static int add_user_ctl(snd_pcm_softvol_t *svol, snd_ctl_elem_info_t *cinfo, int return snd_ctl_elem_write(svol->ctl, &svol->elem); } +/* Takes a template string of the form "softvol {cmd} ({pid})" and writes to + * outbuf a the same string with the text within curly braces replaced with + * relevant information. Returns the number of chars written to outbuf + * including a terminating '\0' or a negative number on error. + * + * Currently supported values for substition are: + * pid The current processes pid + * cmd The basename of the processes argv[0], from /proc/self/cmdline + * exe The basename of the processes binary, from following /proc/self/exe + * { The literal open curly brace '{' + * ob The literal open curly brace '{' + * cb The literal open curly brace '}' + */ +static int per_proc_format_ctl_name(char *outbuf, size_t outbuf_len, char *template) { + int template_len = strlen(template), n, proc_fd; + char *outbuf_p = outbuf, *open_b, *close_b, *template_p = template; + char proc_buf[50]; // max pid can be up to 22 bits ... + char resolved_path[PATH_MAX]; + + // walk through template_p pushing onto outbuf_p + while (outbuf_p < outbuf+outbuf_len) { + open_b = strchr(template_p, '{'); + if (open_b == NULL) { + // Point the end of the current literal to the end + open_b = template + template_len; + } + memcpy(outbuf_p, template_p, open_b-template_p); + outbuf_p += open_b-template_p; + template_p = open_b+1; + + if (template_p >= template + template_len - 1) { + *outbuf_p = '\0'; + break; + } + + if ((close_b = strchr(open_b, '}')) == NULL) { + SNDERR("Invalid control name template: Unmatched '{'"); + return -EINVAL; + } + *close_b = '\0'; + if (strcmp(template_p, "pid") == 0) { + n = snprintf(outbuf_p, outbuf+outbuf_len-outbuf_p, "%d", getpid()); + } else if (strcmp(template_p, "cmd") == 0) { + n = snprintf(proc_buf, 49, "/proc/%d/cmdline", getpid()); + proc_fd = open(proc_buf, O_RDONLY); + if (proc_fd < 0) { + // Maybe a short lived process? + n = errno; + SNDERR("Unable to complete control name template: open: %s: %s", strerror(errno), proc_buf); + return -n; + } + n = read(proc_fd, proc_buf, 49); + proc_buf[n] = '\0'; + n = basename(proc_buf)-proc_buf; + open_b = strchr(proc_buf+n, ' '); + if (open_b != NULL) *open_b = '\0'; + n = snprintf(outbuf_p, outbuf+outbuf_len-outbuf_p, "%s", proc_buf+n); + } else if (strcmp(template_p, "exe") == 0) { + n = snprintf(proc_buf, 49, "/proc/%d/exe", getpid()); + if (realpath(proc_buf, resolved_path) == NULL) { + // Maybe a short lived process? + n = errno; + SNDERR("Unable to complete control name template: open: %s: %s", strerror(errno), proc_buf); + return -n; + } + n = basename(resolved_path)-resolved_path; + n = snprintf(outbuf_p, outbuf+outbuf_len-outbuf_p, "%s", basename(resolved_path)); + } else if (strcmp(template_p, "{") == 0 || strcmp(template_p, "ob") == 0) { + n = 1; + *outbuf_p = '{'; + } else if (strcmp(template_p, "cb") == 0) { + n = 1; + *outbuf_p = '}'; + } else if (strlen(template_p) == 0) { + n = 0; + } else { + SNDERR("Invalid control name template: Unknown substitution: %s", template_p); + return -EINVAL; + } + + outbuf_p = outbuf_p + n; + template_p = close_b + 1; + } + return strlen(outbuf)+1; +} + /* * load and set up user-control * returns 0 if the user-control is found or created, @@ -700,7 +803,7 @@ static int softvol_load_control(snd_pcm_t *pcm, snd_pcm_softvol_t *svol, int cchannels, double min_dB, double max_dB, int resolution) { - char tmp_name[32]; + char tmp_name[32], tmp_template[sizeof(ctl_id->name)]; snd_pcm_info_t *info; snd_ctl_elem_info_t *cinfo; int err; @@ -724,6 +827,15 @@ static int softvol_load_control(snd_pcm_t *pcm, snd_pcm_softvol_t *svol, return err; } + if (svol->per_process) { + err = per_proc_format_ctl_name(tmp_template, sizeof(ctl_id->name), ctl_id->name); + if (err < 1) { // Shouldn't be zero length output... + SNDERR("Cannot open CTL %s: Control name template error %d", tmp_name, err); + return -EINVAL; + } + memcpy(&ctl_id->name, tmp_template, sizeof(ctl_id->name)); + } + svol->elem.id = *ctl_id; svol->max_val = resolution - 1; svol->min_dB = min_dB; @@ -747,6 +859,7 @@ static int softvol_load_control(snd_pcm_t *pcm, snd_pcm_softvol_t *svol, SNDERR("Cannot add a control"); return err; } + snd_ctl_elem_unlock(svol->ctl, &cinfo->id); } else { if (! (cinfo->access & SNDRV_CTL_ELEM_ACCESS_USER)) { /* hardware control exists */ @@ -836,6 +949,10 @@ static const snd_pcm_ops_t snd_pcm_softvol_ops = { * \param resolution resolution of control * \param slave Slave PCM handle * \param close_slave When set, the slave PCM handle is closed with copy PCM + * \param per_process When set, a new ephemeral control is created for each + * softvol PCM. This allows per process volume control when used with + * dmix. Make sure the control name has eg {pid}, {cmd}, {exe} in it so + * the generated names are unique. * \retval zero on success otherwise a negative error code * \warning Using of this function might be dangerous in the sense * of compatibility reasons. The prototype might be freely @@ -846,7 +963,7 @@ int snd_pcm_softvol_open(snd_pcm_t **pcmp, const char *name, int ctl_card, snd_ctl_elem_id_t *ctl_id, int cchannels, double min_dB, double max_dB, int resolution, - snd_pcm_t *slave, int close_slave) + snd_pcm_t *slave, int close_slave, int per_process) { snd_pcm_t *pcm; snd_pcm_softvol_t *svol; @@ -862,6 +979,7 @@ int snd_pcm_softvol_open(snd_pcm_t **pcmp, const char *name, svol = calloc(1, sizeof(*svol)); if (! svol) return -ENOMEM; + svol->per_process = per_process; err = softvol_load_control(slave, svol, ctl_card, ctl_id, cchannels, min_dB, max_dB, resolution); if (err < 0) { @@ -929,6 +1047,17 @@ If the control already exists and it's a system control (i.e. no user-defined control), the plugin simply passes its slave without any changes. +If per_process is set you can (and should) user certain substition variables +in the control name so that uniquely named controls can be generated per-process. +Substition variables are within curly braces, everything else is used literaly, eg +{cmd} ({pid}). Currently supported variables are: +- pid The current processes pid +- cmd The basename of the processes argv[0], from /proc/self/cmdline +- exe The basename of the processes binary, from following /proc/self/exe +- { The literal open curly brace '{' +- ob The literal open curly brace '{' +- cb The literal open curly brace '}' + \code pcm.name { type softvol # Soft Volume conversion PCM @@ -953,6 +1082,8 @@ pcm.name { [max_dB REAL] # maximal dB value (default: 0.0) [resolution INT] # resolution (default: 256) # resolution = 2 means a mute switch + [per_process BOOL] # generate one control per process (default: false) + # useful with dmix. } \endcode @@ -992,7 +1123,7 @@ int _snd_pcm_softvol_open(snd_pcm_t **pcmp, const char *name, int resolution = PRESET_RESOLUTION; double min_dB = PRESET_MIN_DB; double max_dB = ZERO_DB; - int card = -1, cchannels = 2; + int card = -1, cchannels = 2, per_process = 0; snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); @@ -1035,6 +1166,15 @@ int _snd_pcm_softvol_open(snd_pcm_t **pcmp, const char *name, } continue; } + if (strcmp(id, "per_process") == 0) { + err = snd_config_get_bool(n); + if (err < 0) { + SNDERR("Invalid boolean value in per_process"); + return err; + } + per_process = err; + continue; + } SNDERR("Unknown field %s", id); return -EINVAL; } @@ -1092,7 +1232,7 @@ int _snd_pcm_softvol_open(snd_pcm_t **pcmp, const char *name, return err; } err = snd_pcm_softvol_open(pcmp, name, sformat, card, ctl_id, cchannels, - min_dB, max_dB, resolution, spcm, 1); + min_dB, max_dB, resolution, spcm, 1, per_process); if (err < 0) snd_pcm_close(spcm); } -- 2.1.4

3 2

[alsa-devel] [PATCH 1/2][linux-next] ASoC: Fix warning while make xmldocs caused by soc-core.c
by Masanari Iida 15 Jul '15

15 Jul '15

This patch fix following warning while make xmldocs. Warning(.//sound/soc/soc-core.c:2148): No description found for parameter 'ratio' Add missing ":" Signed-off-by: Masanari Iida <standby24x7(a)gmail.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 8c8ce65..691fcc4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2140,7 +2140,7 @@ EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll); /** * snd_soc_dai_set_bclk_ratio - configure BCLK to sample rate ratio. * @dai: DAI - * @ratio Ratio of BCLK to Sample rate. + * @ratio: Ratio of BCLK to Sample rate. * * Configures the DAI for a preset BCLK to sample rate ratio. */ -- 2.5.0.rc2.13.g961abca

1 1

[alsa-devel] [PATCH] Drop owner assignment from i2c_driver (and platform left-overs)
by Krzysztof Kozlowski 15 Jul '15

15 Jul '15

3 6

[alsa-devel] [patch] ASoC: fsl: fsl_spdif: signedness bug in fsl_spdif_startup()
by Dan Carpenter 15 Jul '15

15 Jul '15

We need "i" to be signed or it leads to a forever loop in the error handling. Fixes: fa3be9208dcb ('ASoC: fsl: fsl_spdif: Check for clk_prepare_enable() error') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 3ee4539..7a92f30 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -454,7 +454,8 @@ static int fsl_spdif_startup(struct snd_pcm_substream *substream, struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); struct platform_device *pdev = spdif_priv->pdev; struct regmap *regmap = spdif_priv->regmap; - u32 scr, mask, i; + u32 scr, mask; + int i; int ret; /* Reset module and interrupts only for first initialization */

3 2

[alsa-devel] [PATCH] ASoC: Constify snd_soc_dai_ops variables
by Axel Lin 15 Jul '15

15 Jul '15

The snd_soc_dai_ops variables are not modified after initialization in these drivers, so make them const. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs42l52.c | 2 +- sound/soc/codecs/cs42l56.c | 2 +- sound/soc/codecs/isabelle.c | 8 ++++---- sound/soc/codecs/lm49453.c | 10 +++++----- sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/max98357a.c | 2 +- sound/soc/codecs/mc13783.c | 6 +++--- sound/soc/codecs/rt5645.c | 2 +- sound/soc/codecs/rt5670.c | 2 +- sound/soc/codecs/rt5677.c | 2 +- sound/soc/codecs/si476x.c | 2 +- sound/soc/codecs/tas2552.c | 2 +- sound/soc/codecs/tlv320aic31xx.c | 2 +- 13 files changed, 22 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 4de52c9..c8f4af2 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -919,7 +919,7 @@ static int cs42l52_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE) -static struct snd_soc_dai_ops cs42l52_ops = { +static const struct snd_soc_dai_ops cs42l52_ops = { .hw_params = cs42l52_pcm_hw_params, .digital_mute = cs42l52_digital_mute, .set_fmt = cs42l52_set_fmt, diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 1e11ba4..22bc2f3 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -989,7 +989,7 @@ static int cs42l56_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops cs42l56_ops = { +static const struct snd_soc_dai_ops cs42l56_ops = { .hw_params = cs42l56_pcm_hw_params, .digital_mute = cs42l56_digital_mute, .set_fmt = cs42l56_set_dai_fmt, diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index ac7dfe7..4013864 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -1016,25 +1016,25 @@ static int isabelle_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) #define ISABELLE_FORMATS (SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops isabelle_hs_dai_ops = { +static const struct snd_soc_dai_ops isabelle_hs_dai_ops = { .hw_params = isabelle_hw_params, .set_fmt = isabelle_set_dai_fmt, .digital_mute = isabelle_hs_mute, }; -static struct snd_soc_dai_ops isabelle_hf_dai_ops = { +static const struct snd_soc_dai_ops isabelle_hf_dai_ops = { .hw_params = isabelle_hw_params, .set_fmt = isabelle_set_dai_fmt, .digital_mute = isabelle_hf_mute, }; -static struct snd_soc_dai_ops isabelle_line_dai_ops = { +static const struct snd_soc_dai_ops isabelle_line_dai_ops = { .hw_params = isabelle_hw_params, .set_fmt = isabelle_set_dai_fmt, .digital_mute = isabelle_line_mute, }; -static struct snd_soc_dai_ops isabelle_ul_dai_ops = { +static const struct snd_soc_dai_ops isabelle_ul_dai_ops = { .hw_params = isabelle_hw_params, .set_fmt = isabelle_set_dai_fmt, }; diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index ef432aa..4e76992 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1287,35 +1287,35 @@ static int lm49453_set_bias_level(struct snd_soc_codec *codec, #define LM49453_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops lm49453_headset_dai_ops = { +static const struct snd_soc_dai_ops lm49453_headset_dai_ops = { .hw_params = lm49453_hw_params, .set_sysclk = lm49453_set_dai_sysclk, .set_fmt = lm49453_set_dai_fmt, .digital_mute = lm49453_hp_mute, }; -static struct snd_soc_dai_ops lm49453_speaker_dai_ops = { +static const struct snd_soc_dai_ops lm49453_speaker_dai_ops = { .hw_params = lm49453_hw_params, .set_sysclk = lm49453_set_dai_sysclk, .set_fmt = lm49453_set_dai_fmt, .digital_mute = lm49453_ls_mute, }; -static struct snd_soc_dai_ops lm49453_haptic_dai_ops = { +static const struct snd_soc_dai_ops lm49453_haptic_dai_ops = { .hw_params = lm49453_hw_params, .set_sysclk = lm49453_set_dai_sysclk, .set_fmt = lm49453_set_dai_fmt, .digital_mute = lm49453_ha_mute, }; -static struct snd_soc_dai_ops lm49453_ep_dai_ops = { +static const struct snd_soc_dai_ops lm49453_ep_dai_ops = { .hw_params = lm49453_hw_params, .set_sysclk = lm49453_set_dai_sysclk, .set_fmt = lm49453_set_dai_fmt, .digital_mute = lm49453_ep_mute, }; -static struct snd_soc_dai_ops lm49453_lineout_dai_ops = { +static const struct snd_soc_dai_ops lm49453_lineout_dai_ops = { .hw_params = lm49453_hw_params, .set_sysclk = lm49453_set_dai_sysclk, .set_fmt = lm49453_set_dai_fmt, diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 1697340..a2cdf52 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2386,7 +2386,7 @@ EXPORT_SYMBOL_GPL(max98090_mic_detect); #define MAX98090_RATES SNDRV_PCM_RATE_8000_96000 #define MAX98090_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops max98090_dai_ops = { +static const struct snd_soc_dai_ops max98090_dai_ops = { .set_sysclk = max98090_dai_set_sysclk, .set_fmt = max98090_dai_set_fmt, .set_tdm_slot = max98090_set_tdm_slot, diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index e88119a..f5e3dce 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -79,7 +79,7 @@ static struct snd_soc_codec_driver max98357a_codec_driver = { .num_dapm_routes = ARRAY_SIZE(max98357a_dapm_routes), }; -static struct snd_soc_dai_ops max98357a_dai_ops = { +static const struct snd_soc_dai_ops max98357a_dai_ops = { .trigger = max98357a_daiops_trigger, }; diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 3d44fc5..3e770cb 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -650,14 +650,14 @@ static int mc13783_remove(struct snd_soc_codec *codec) #define MC13783_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops mc13783_ops_dac = { +static const struct snd_soc_dai_ops mc13783_ops_dac = { .hw_params = mc13783_pcm_hw_params_dac, .set_fmt = mc13783_set_fmt_async, .set_sysclk = mc13783_set_sysclk_dac, .set_tdm_slot = mc13783_set_tdm_slot_dac, }; -static struct snd_soc_dai_ops mc13783_ops_codec = { +static const struct snd_soc_dai_ops mc13783_ops_codec = { .hw_params = mc13783_pcm_hw_params_codec, .set_fmt = mc13783_set_fmt_async, .set_sysclk = mc13783_set_sysclk_codec, @@ -698,7 +698,7 @@ static struct snd_soc_dai_driver mc13783_dai_async[] = { }, }; -static struct snd_soc_dai_ops mc13783_ops_sync = { +static const struct snd_soc_dai_ops mc13783_ops_sync = { .hw_params = mc13783_pcm_hw_params_sync, .set_fmt = mc13783_set_fmt_sync, .set_sysclk = mc13783_set_sysclk_sync, diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9dfa431..9b1ad6b 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3106,7 +3106,7 @@ static int rt5645_resume(struct snd_soc_codec *codec) #define RT5645_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) -static struct snd_soc_dai_ops rt5645_aif_dai_ops = { +static const struct snd_soc_dai_ops rt5645_aif_dai_ops = { .hw_params = rt5645_hw_params, .set_fmt = rt5645_set_dai_fmt, .set_sysclk = rt5645_set_dai_sysclk, diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index a791d7e..132d012 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2720,7 +2720,7 @@ static int rt5670_resume(struct snd_soc_codec *codec) #define RT5670_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) -static struct snd_soc_dai_ops rt5670_aif_dai_ops = { +static const struct snd_soc_dai_ops rt5670_aif_dai_ops = { .hw_params = rt5670_hw_params, .set_fmt = rt5670_set_dai_fmt, .set_sysclk = rt5670_set_dai_sysclk, diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 13b871f..151db60 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4861,7 +4861,7 @@ static int rt5677_write(void *context, unsigned int reg, unsigned int val) #define RT5677_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) -static struct snd_soc_dai_ops rt5677_aif_dai_ops = { +static const struct snd_soc_dai_ops rt5677_aif_dai_ops = { .hw_params = rt5677_hw_params, .set_fmt = rt5677_set_dai_fmt, .set_sysclk = rt5677_set_dai_sysclk, diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 3e72964..a8402d0 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -208,7 +208,7 @@ out: return err; } -static struct snd_soc_dai_ops si476x_dai_ops = { +static const struct snd_soc_dai_ops si476x_dai_ops = { .hw_params = si476x_codec_hw_params, .set_fmt = si476x_codec_set_dai_fmt, }; diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 71cfae0..f2ae302 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -520,7 +520,7 @@ static const struct dev_pm_ops tas2552_pm = { NULL) }; -static struct snd_soc_dai_ops tas2552_speaker_dai_ops = { +static const struct snd_soc_dai_ops tas2552_speaker_dai_ops = { .hw_params = tas2552_hw_params, .prepare = tas2552_prepare, .set_sysclk = tas2552_set_dai_sysclk, diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index c4c960f..df01fec 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1121,7 +1121,7 @@ static struct snd_soc_codec_driver soc_codec_driver_aic31xx = { .num_dapm_routes = ARRAY_SIZE(aic31xx_audio_map), }; -static struct snd_soc_dai_ops aic31xx_dai_ops = { +static const struct snd_soc_dai_ops aic31xx_dai_ops = { .hw_params = aic31xx_hw_params, .set_sysclk = aic31xx_set_dai_sysclk, .set_fmt = aic31xx_set_dai_fmt, -- 2.1.0

2 1

[alsa-devel] [PATCH] ASoC: da732x: Merge da732x_dai1_ops and da732x_dai2_ops
by Axel Lin 15 Jul '15

15 Jul '15

da732x_dai1_ops is identical to da732x_dai2_ops, so merge them to da732x_dai_ops. The da732x_dai_ops is not modified after initialization, so make it const. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/da732x.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index b1cb465..29251b0 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1196,13 +1196,7 @@ static int da732x_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, #define DA732X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops da732x_dai1_ops = { - .hw_params = da732x_hw_params, - .set_fmt = da732x_set_dai_fmt, - .set_sysclk = da732x_set_dai_sysclk, -}; - -static struct snd_soc_dai_ops da732x_dai2_ops = { +static const struct snd_soc_dai_ops da732x_dai_ops = { .hw_params = da732x_hw_params, .set_fmt = da732x_set_dai_fmt, .set_sysclk = da732x_set_dai_sysclk, @@ -1227,7 +1221,7 @@ static struct snd_soc_dai_driver da732x_dai[] = { .rates = DA732X_RATES, .formats = DA732X_FORMATS, }, - .ops = &da732x_dai1_ops, + .ops = &da732x_dai_ops, }, { .name = "DA732X_AIFB", @@ -1247,7 +1241,7 @@ static struct snd_soc_dai_driver da732x_dai[] = { .rates = DA732X_RATES, .formats = DA732X_FORMATS, }, - .ops = &da732x_dai2_ops, + .ops = &da732x_dai_ops, }, }; -- 2.1.0

3 2

[alsa-devel] [PATCH] ASoC: sirf-audio-codec: Staticise local symbols
by Axel Lin 15 Jul '15

15 Jul '15

Also make sirf_audio_codec_dai_ops const. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/sirf-audio-codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c index 29cb442..6bfd25c 100644 --- a/sound/soc/codecs/sirf-audio-codec.c +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -370,11 +370,11 @@ static int sirf_audio_codec_trigger(struct snd_pcm_substream *substream, return 0; } -struct snd_soc_dai_ops sirf_audio_codec_dai_ops = { +static const struct snd_soc_dai_ops sirf_audio_codec_dai_ops = { .trigger = sirf_audio_codec_trigger, }; -struct snd_soc_dai_driver sirf_audio_codec_dai = { +static struct snd_soc_dai_driver sirf_audio_codec_dai = { .name = "sirf-audio-codec", .playback = { .stream_name = "Playback", -- 2.1.0

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: Drop owner assignment from i2c_driver
by Krzysztof Kozlowski 15 Jul '15

15 Jul '15

i2c_driver does not need to set an owner because i2c_register_driver() will set it. Signed-off-by: Krzysztof Kozlowski <k.kozlowski(a)samsung.com> --- The coccinelle script which generated the patch was sent here: http://www.spinics.net/lists/kernel/msg2029903.html Changes since v1: 1. Rebased on next-20150714 2. Removed owner from rt298.c --- sound/soc/codecs/adau1373.c | 1 - sound/soc/codecs/adau1701.c | 1 - sound/soc/codecs/adau1761-i2c.c | 1 - sound/soc/codecs/adau1781-i2c.c | 1 - sound/soc/codecs/adau1977-i2c.c | 1 - sound/soc/codecs/adav803.c | 1 - sound/soc/codecs/ak4535.c | 1 - sound/soc/codecs/ak4641.c | 1 - sound/soc/codecs/ak4642.c | 1 - sound/soc/codecs/ak4671.c | 1 - sound/soc/codecs/alc5623.c | 1 - sound/soc/codecs/alc5632.c | 1 - sound/soc/codecs/cs35l32.c | 1 - sound/soc/codecs/cs4265.c | 1 - sound/soc/codecs/cs4270.c | 1 - sound/soc/codecs/cs4271-i2c.c | 1 - sound/soc/codecs/cs42l51-i2c.c | 1 - sound/soc/codecs/cs42l52.c | 1 - sound/soc/codecs/cs42l56.c | 1 - sound/soc/codecs/cs42l73.c | 1 - sound/soc/codecs/cs42xx8-i2c.c | 1 - sound/soc/codecs/da7210.c | 1 - sound/soc/codecs/da7213.c | 1 - sound/soc/codecs/da732x.c | 1 - sound/soc/codecs/da9055.c | 1 - sound/soc/codecs/isabelle.c | 1 - sound/soc/codecs/lm4857.c | 1 - sound/soc/codecs/lm49453.c | 1 - sound/soc/codecs/max9768.c | 1 - sound/soc/codecs/max98088.c | 1 - sound/soc/codecs/max98090.c | 1 - sound/soc/codecs/max98095.c | 1 - sound/soc/codecs/max9850.c | 1 - sound/soc/codecs/max9877.c | 1 - sound/soc/codecs/max98925.c | 1 - sound/soc/codecs/ml26124.c | 1 - sound/soc/codecs/pcm1681.c | 1 - sound/soc/codecs/pcm512x-i2c.c | 1 - sound/soc/codecs/rt286.c | 1 - sound/soc/codecs/rt298.c | 1 - sound/soc/codecs/rt5631.c | 1 - sound/soc/codecs/rt5640.c | 1 - sound/soc/codecs/rt5645.c | 1 - sound/soc/codecs/rt5651.c | 1 - sound/soc/codecs/rt5670.c | 1 - sound/soc/codecs/rt5677.c | 1 - sound/soc/codecs/sgtl5000.c | 1 - sound/soc/codecs/ssm2518.c | 1 - sound/soc/codecs/ssm2602-i2c.c | 1 - sound/soc/codecs/ssm4567.c | 1 - sound/soc/codecs/sta32x.c | 1 - sound/soc/codecs/sta350.c | 1 - sound/soc/codecs/sta529.c | 1 - sound/soc/codecs/tas2552.c | 1 - sound/soc/codecs/tas5086.c | 1 - sound/soc/codecs/tfa9879.c | 1 - sound/soc/codecs/tlv320aic31xx.c | 1 - sound/soc/codecs/tlv320aic32x4.c | 1 - sound/soc/codecs/tlv320aic3x.c | 1 - sound/soc/codecs/tlv320dac33.c | 1 - sound/soc/codecs/tpa6130a2.c | 1 - sound/soc/codecs/ts3a227e.c | 1 - sound/soc/codecs/uda1380.c | 1 - sound/soc/codecs/wm1250-ev1.c | 1 - sound/soc/codecs/wm2000.c | 1 - sound/soc/codecs/wm2200.c | 1 - sound/soc/codecs/wm5100.c | 1 - sound/soc/codecs/wm8510.c | 1 - sound/soc/codecs/wm8523.c | 1 - sound/soc/codecs/wm8580.c | 1 - sound/soc/codecs/wm8711.c | 1 - sound/soc/codecs/wm8728.c | 1 - sound/soc/codecs/wm8731.c | 1 - sound/soc/codecs/wm8737.c | 1 - sound/soc/codecs/wm8741.c | 1 - sound/soc/codecs/wm8750.c | 1 - sound/soc/codecs/wm8753.c | 1 - sound/soc/codecs/wm8776.c | 1 - sound/soc/codecs/wm8804-i2c.c | 1 - sound/soc/codecs/wm8900.c | 1 - sound/soc/codecs/wm8903.c | 1 - sound/soc/codecs/wm8904.c | 1 - sound/soc/codecs/wm8940.c | 1 - sound/soc/codecs/wm8955.c | 1 - sound/soc/codecs/wm8960.c | 1 - sound/soc/codecs/wm8961.c | 1 - sound/soc/codecs/wm8962.c | 1 - sound/soc/codecs/wm8971.c | 1 - sound/soc/codecs/wm8974.c | 1 - sound/soc/codecs/wm8978.c | 1 - sound/soc/codecs/wm8983.c | 1 - sound/soc/codecs/wm8985.c | 1 - sound/soc/codecs/wm8988.c | 1 - sound/soc/codecs/wm8990.c | 1 - sound/soc/codecs/wm8991.c | 1 - sound/soc/codecs/wm8993.c | 1 - sound/soc/codecs/wm8995.c | 1 - sound/soc/codecs/wm8996.c | 1 - sound/soc/codecs/wm9081.c | 1 - sound/soc/codecs/wm9090.c | 1 - 100 files changed, 100 deletions(-) diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index a43160254929..6c96860f46de 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1534,7 +1534,6 @@ MODULE_DEVICE_TABLE(i2c, adau1373_i2c_id); static struct i2c_driver adau1373_i2c_driver = { .driver = { .name = "adau1373", - .owner = THIS_MODULE, }, .probe = adau1373_i2c_probe, .remove = adau1373_i2c_remove, diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index ff7f846e3b76..de53c0d7bf10 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -915,7 +915,6 @@ MODULE_DEVICE_TABLE(i2c, adau1701_i2c_id); static struct i2c_driver adau1701_i2c_driver = { .driver = { .name = "adau1701", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(adau1701_dt_ids), }, .probe = adau1701_i2c_probe, diff --git a/sound/soc/codecs/adau1761-i2c.c b/sound/soc/codecs/adau1761-i2c.c index 862796dec693..348ccb17d3cc 100644 --- a/sound/soc/codecs/adau1761-i2c.c +++ b/sound/soc/codecs/adau1761-i2c.c @@ -47,7 +47,6 @@ MODULE_DEVICE_TABLE(i2c, adau1761_i2c_ids); static struct i2c_driver adau1761_i2c_driver = { .driver = { .name = "adau1761", - .owner = THIS_MODULE, }, .probe = adau1761_i2c_probe, .remove = adau1761_i2c_remove, diff --git a/sound/soc/codecs/adau1781-i2c.c b/sound/soc/codecs/adau1781-i2c.c index 2ce4362ccec1..0e32bba92339 100644 --- a/sound/soc/codecs/adau1781-i2c.c +++ b/sound/soc/codecs/adau1781-i2c.c @@ -45,7 +45,6 @@ MODULE_DEVICE_TABLE(i2c, adau1781_i2c_ids); static struct i2c_driver adau1781_i2c_driver = { .driver = { .name = "adau1781", - .owner = THIS_MODULE, }, .probe = adau1781_i2c_probe, .remove = adau1781_i2c_remove, diff --git a/sound/soc/codecs/adau1977-i2c.c b/sound/soc/codecs/adau1977-i2c.c index 9700e8c838c9..21e7394a972a 100644 --- a/sound/soc/codecs/adau1977-i2c.c +++ b/sound/soc/codecs/adau1977-i2c.c @@ -46,7 +46,6 @@ MODULE_DEVICE_TABLE(i2c, adau1977_i2c_ids); static struct i2c_driver adau1977_i2c_driver = { .driver = { .name = "adau1977", - .owner = THIS_MODULE, }, .probe = adau1977_i2c_probe, .remove = adau1977_i2c_remove, diff --git a/sound/soc/codecs/adav803.c b/sound/soc/codecs/adav803.c index 66d9fce34e62..52881faedcf6 100644 --- a/sound/soc/codecs/adav803.c +++ b/sound/soc/codecs/adav803.c @@ -36,7 +36,6 @@ static int adav803_remove(struct i2c_client *client) static struct i2c_driver adav803_driver = { .driver = { .name = "adav803", - .owner = THIS_MODULE, }, .probe = adav803_probe, .remove = adav803_remove, diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 8670861e5bec..54428c64387b 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -444,7 +444,6 @@ MODULE_DEVICE_TABLE(i2c, ak4535_i2c_id); static struct i2c_driver ak4535_i2c_driver = { .driver = { .name = "ak4535", - .owner = THIS_MODULE, }, .probe = ak4535_i2c_probe, .remove = ak4535_i2c_remove, diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 2d0ff4595ea0..b14176f8d884 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -609,7 +609,6 @@ MODULE_DEVICE_TABLE(i2c, ak4641_i2c_id); static struct i2c_driver ak4641_i2c_driver = { .driver = { .name = "ak4641", - .owner = THIS_MODULE, }, .probe = ak4641_i2c_probe, .remove = ak4641_i2c_remove, diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index fe963e17ceb4..4a90143d0e90 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -645,7 +645,6 @@ MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id); static struct i2c_driver ak4642_i2c_driver = { .driver = { .name = "ak4642-codec", - .owner = THIS_MODULE, .of_match_table = ak4642_of_match, }, .probe = ak4642_i2c_probe, diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 0e59063aeb6f..c73a9f6914b6 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -663,7 +663,6 @@ MODULE_DEVICE_TABLE(i2c, ak4671_i2c_id); static struct i2c_driver ak4671_i2c_driver = { .driver = { .name = "ak4671-codec", - .owner = THIS_MODULE, }, .probe = ak4671_i2c_probe, .remove = ak4671_i2c_remove, diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 0fc24e0d518c..cf99c4e90acc 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -1085,7 +1085,6 @@ MODULE_DEVICE_TABLE(of, alc5623_of_match); static struct i2c_driver alc5623_i2c_driver = { .driver = { .name = "alc562x-codec", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(alc5623_of_match), }, .probe = alc5623_i2c_probe, diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 29a26455ce25..ef6de511dc7e 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -1183,7 +1183,6 @@ MODULE_DEVICE_TABLE(of, alc5632_of_match); static struct i2c_driver alc5632_i2c_driver = { .driver = { .name = "alc5632", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(alc5632_of_match), }, .probe = alc5632_i2c_probe, diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 8f40025b7e7c..76564dc752d3 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -607,7 +607,6 @@ MODULE_DEVICE_TABLE(i2c, cs35l32_id); static struct i2c_driver cs35l32_i2c_driver = { .driver = { .name = "cs35l32", - .owner = THIS_MODULE, .pm = &cs35l32_runtime_pm, .of_match_table = cs35l32_of_match, }, diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index d7ec4756e45b..d1a77c7489d8 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -658,7 +658,6 @@ MODULE_DEVICE_TABLE(i2c, cs4265_id); static struct i2c_driver cs4265_i2c_driver = { .driver = { .name = "cs4265", - .owner = THIS_MODULE, .of_match_table = cs4265_of_match, }, .id_table = cs4265_id, diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index e6d4ff9fd992..e07807d96b68 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -751,7 +751,6 @@ MODULE_DEVICE_TABLE(i2c, cs4270_id); static struct i2c_driver cs4270_i2c_driver = { .driver = { .name = "cs4270", - .owner = THIS_MODULE, .of_match_table = cs4270_of_match, }, .id_table = cs4270_id, diff --git a/sound/soc/codecs/cs4271-i2c.c b/sound/soc/codecs/cs4271-i2c.c index b264da030340..dcb3223d7d8f 100644 --- a/sound/soc/codecs/cs4271-i2c.c +++ b/sound/soc/codecs/cs4271-i2c.c @@ -48,7 +48,6 @@ MODULE_DEVICE_TABLE(i2c, cs4271_i2c_id); static struct i2c_driver cs4271_i2c_driver = { .driver = { .name = "cs4271", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(cs4271_dt_ids), }, .probe = cs4271_i2c_probe, diff --git a/sound/soc/codecs/cs42l51-i2c.c b/sound/soc/codecs/cs42l51-i2c.c index c40428f25ba5..9bad478474fa 100644 --- a/sound/soc/codecs/cs42l51-i2c.c +++ b/sound/soc/codecs/cs42l51-i2c.c @@ -45,7 +45,6 @@ static int cs42l51_i2c_remove(struct i2c_client *i2c) static struct i2c_driver cs42l51_i2c_driver = { .driver = { .name = "cs42l51", - .owner = THIS_MODULE, .of_match_table = cs42l51_of_match, }, .probe = cs42l51_i2c_probe, diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 4de52c9957ac..b82d8e5b76ed 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1285,7 +1285,6 @@ MODULE_DEVICE_TABLE(i2c, cs42l52_id); static struct i2c_driver cs42l52_i2c_driver = { .driver = { .name = "cs42l52", - .owner = THIS_MODULE, .of_match_table = cs42l52_of_match, }, .id_table = cs42l52_id, diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 1e11ba45a79f..4ae793365985 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -1408,7 +1408,6 @@ MODULE_DEVICE_TABLE(i2c, cs42l56_id); static struct i2c_driver cs42l56_i2c_driver = { .driver = { .name = "cs42l56", - .owner = THIS_MODULE, .of_match_table = cs42l56_of_match, }, .id_table = cs42l56_id, diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index b7853b9d3a60..7cb1d7091dae 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1491,7 +1491,6 @@ MODULE_DEVICE_TABLE(i2c, cs42l73_id); static struct i2c_driver cs42l73_i2c_driver = { .driver = { .name = "cs42l73", - .owner = THIS_MODULE, .of_match_table = cs42l73_of_match, }, .id_table = cs42l73_id, diff --git a/sound/soc/codecs/cs42xx8-i2c.c b/sound/soc/codecs/cs42xx8-i2c.c index 5a71c9e5b441..800c1d549347 100644 --- a/sound/soc/codecs/cs42xx8-i2c.c +++ b/sound/soc/codecs/cs42xx8-i2c.c @@ -49,7 +49,6 @@ MODULE_DEVICE_TABLE(i2c, cs42xx8_i2c_id); static struct i2c_driver cs42xx8_i2c_driver = { .driver = { .name = "cs42xx8", - .owner = THIS_MODULE, .pm = &cs42xx8_pm, .of_match_table = cs42xx8_of_match, }, diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index a44ead615b67..457ed82d3e10 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -1259,7 +1259,6 @@ MODULE_DEVICE_TABLE(i2c, da7210_i2c_id); static struct i2c_driver da7210_i2c_driver = { .driver = { .name = "da7210", - .owner = THIS_MODULE, }, .probe = da7210_i2c_probe, .remove = da7210_i2c_remove, diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index baa36f6fd405..47fc3bec8a9c 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1585,7 +1585,6 @@ MODULE_DEVICE_TABLE(i2c, da7213_i2c_id); static struct i2c_driver da7213_i2c_driver = { .driver = { .name = "da7213", - .owner = THIS_MODULE, }, .probe = da7213_i2c_probe, .remove = da7213_remove, diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index b1cb4656a49c..0868568429a8 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1572,7 +1572,6 @@ MODULE_DEVICE_TABLE(i2c, da732x_i2c_id); static struct i2c_driver da732x_i2c_driver = { .driver = { .name = "da7320", - .owner = THIS_MODULE, }, .probe = da732x_i2c_probe, .remove = da732x_i2c_remove, diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index 925d82278249..04c76f37e1fd 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -1538,7 +1538,6 @@ static const struct of_device_id da9055_of_match[] = { static struct i2c_driver da9055_i2c_driver = { .driver = { .name = "da9055-codec", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(da9055_of_match), }, .probe = da9055_i2c_probe, diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index ac7dfe79c53b..34dc0f362482 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -1149,7 +1149,6 @@ MODULE_DEVICE_TABLE(i2c, isabelle_i2c_id); static struct i2c_driver isabelle_i2c_driver = { .driver = { .name = "isabelle", - .owner = THIS_MODULE, }, .probe = isabelle_i2c_probe, .remove = isabelle_i2c_remove, diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c index 99ffc49aa779..558de1053f73 100644 --- a/sound/soc/codecs/lm4857.c +++ b/sound/soc/codecs/lm4857.c @@ -142,7 +142,6 @@ MODULE_DEVICE_TABLE(i2c, lm4857_i2c_id); static struct i2c_driver lm4857_i2c_driver = { .driver = { .name = "lm4857", - .owner = THIS_MODULE, }, .probe = lm4857_i2c_probe, .id_table = lm4857_i2c_id, diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index ef432aa8c639..6d3e8dbe34f5 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1456,7 +1456,6 @@ MODULE_DEVICE_TABLE(i2c, lm49453_i2c_id); static struct i2c_driver lm49453_i2c_driver = { .driver = { .name = "lm49453", - .owner = THIS_MODULE, }, .probe = lm49453_i2c_probe, .remove = lm49453_i2c_remove, diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c index e3ccea229160..f987497a8411 100644 --- a/sound/soc/codecs/max9768.c +++ b/sound/soc/codecs/max9768.c @@ -242,7 +242,6 @@ MODULE_DEVICE_TABLE(i2c, max9768_i2c_id); static struct i2c_driver max9768_i2c_driver = { .driver = { .name = "max9768", - .owner = THIS_MODULE, }, .probe = max9768_i2c_probe, .remove = max9768_i2c_remove, diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index d0f45348bfbb..99c2daa0eebf 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -2011,7 +2011,6 @@ MODULE_DEVICE_TABLE(i2c, max98088_i2c_id); static struct i2c_driver max98088_i2c_driver = { .driver = { .name = "max98088", - .owner = THIS_MODULE, }, .probe = max98088_i2c_probe, .remove = max98088_i2c_remove, diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 1697340c3842..af6d282c5e2d 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2717,7 +2717,6 @@ MODULE_DEVICE_TABLE(acpi, max98090_acpi_match); static struct i2c_driver max98090_i2c_driver = { .driver = { .name = "max98090", - .owner = THIS_MODULE, .pm = &max98090_pm, .of_match_table = of_match_ptr(max98090_of_match), .acpi_match_table = ACPI_PTR(max98090_acpi_match), diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 66afd2b781e9..7e3814e65c87 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -2434,7 +2434,6 @@ MODULE_DEVICE_TABLE(of, max98095_of_match); static struct i2c_driver max98095_i2c_driver = { .driver = { .name = "max98095", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(max98095_of_match), }, .probe = max98095_i2c_probe, diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 481d58f1cb3f..7c990521638c 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -352,7 +352,6 @@ MODULE_DEVICE_TABLE(i2c, max9850_i2c_id); static struct i2c_driver max9850_i2c_driver = { .driver = { .name = "max9850", - .owner = THIS_MODULE, }, .probe = max9850_i2c_probe, .remove = max9850_i2c_remove, diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c index 8df99fb69ab1..e1df06fab756 100644 --- a/sound/soc/codecs/max9877.c +++ b/sound/soc/codecs/max9877.c @@ -174,7 +174,6 @@ MODULE_DEVICE_TABLE(i2c, max9877_i2c_id); static struct i2c_driver max9877_i2c_driver = { .driver = { .name = "max9877", - .owner = THIS_MODULE, }, .probe = max9877_i2c_probe, .remove = max9877_i2c_remove, diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c index aad664225dc3..ce551eecbf95 100644 --- a/sound/soc/codecs/max98925.c +++ b/sound/soc/codecs/max98925.c @@ -639,7 +639,6 @@ MODULE_DEVICE_TABLE(of, max98925_of_match); static struct i2c_driver max98925_i2c_driver = { .driver = { .name = "max98925", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(max98925_of_match), .pm = NULL, }, diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c index 1e8ede8454af..f561c78b9e0e 100644 --- a/sound/soc/codecs/ml26124.c +++ b/sound/soc/codecs/ml26124.c @@ -597,7 +597,6 @@ MODULE_DEVICE_TABLE(i2c, ml26124_i2c_id); static struct i2c_driver ml26124_i2c_driver = { .driver = { .name = "ml26124", - .owner = THIS_MODULE, }, .probe = ml26124_i2c_probe, .remove = ml26124_i2c_remove, diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index 477e13d30971..b2c990f08aa3 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -330,7 +330,6 @@ static int pcm1681_i2c_remove(struct i2c_client *client) static struct i2c_driver pcm1681_i2c_driver = { .driver = { .name = "pcm1681", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(pcm1681_dt_ids), }, .id_table = pcm1681_i2c_id, diff --git a/sound/soc/codecs/pcm512x-i2c.c b/sound/soc/codecs/pcm512x-i2c.c index dcdfac0ffeb1..dbff416e38be 100644 --- a/sound/soc/codecs/pcm512x-i2c.c +++ b/sound/soc/codecs/pcm512x-i2c.c @@ -67,7 +67,6 @@ static struct i2c_driver pcm512x_i2c_driver = { .id_table = pcm512x_i2c_id, .driver = { .name = "pcm512x", - .owner = THIS_MODULE, .of_match_table = pcm512x_of_match, .pm = &pcm512x_pm_ops, }, diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 76d88bad195c..bd9365885f73 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1259,7 +1259,6 @@ static int rt286_i2c_remove(struct i2c_client *i2c) static struct i2c_driver rt286_i2c_driver = { .driver = { .name = "rt286", - .owner = THIS_MODULE, .acpi_match_table = ACPI_PTR(rt286_acpi_match), }, .probe = rt286_i2c_probe, diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 7c4bcb65ef2c..75e5679dfef8 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -1259,7 +1259,6 @@ static int rt298_i2c_remove(struct i2c_client *i2c) static struct i2c_driver rt298_i2c_driver = { .driver = { .name = "rt298", - .owner = THIS_MODULE, .acpi_match_table = ACPI_PTR(rt298_acpi_match), }, .probe = rt298_i2c_probe, diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 058167c80d71..d8771020299d 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1725,7 +1725,6 @@ static int rt5631_i2c_remove(struct i2c_client *client) static struct i2c_driver rt5631_i2c_driver = { .driver = { .name = "rt5631", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(rt5631_i2c_dt_ids), }, .probe = rt5631_i2c_probe, diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 0cbdac653041..187470bf926d 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2242,7 +2242,6 @@ static int rt5640_i2c_remove(struct i2c_client *i2c) static struct i2c_driver rt5640_i2c_driver = { .driver = { .name = "rt5640", - .owner = THIS_MODULE, .acpi_match_table = ACPI_PTR(rt5640_acpi_match), .of_match_table = of_match_ptr(rt5640_of_match), }, diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9dfa431b337d..b4e9b6b60778 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3455,7 +3455,6 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) static struct i2c_driver rt5645_i2c_driver = { .driver = { .name = "rt5645", - .owner = THIS_MODULE, .acpi_match_table = ACPI_PTR(rt5645_acpi_match), }, .probe = rt5645_i2c_probe, diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 9f3bc5a5b7b9..2c9eb8983042 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -1806,7 +1806,6 @@ static int rt5651_i2c_remove(struct i2c_client *i2c) static struct i2c_driver rt5651_i2c_driver = { .driver = { .name = "rt5651", - .owner = THIS_MODULE, }, .probe = rt5651_i2c_probe, .remove = rt5651_i2c_remove, diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index a791d7e133f2..e84e223b4b86 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -3043,7 +3043,6 @@ static int rt5670_i2c_remove(struct i2c_client *i2c) static struct i2c_driver rt5670_i2c_driver = { .driver = { .name = "rt5670", - .owner = THIS_MODULE, .acpi_match_table = ACPI_PTR(rt5670_acpi_match), }, .probe = rt5670_i2c_probe, diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 13b871f11023..c69351dfa03a 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -5241,7 +5241,6 @@ static int rt5677_i2c_remove(struct i2c_client *i2c) static struct i2c_driver rt5677_i2c_driver = { .driver = { .name = "rt5677", - .owner = THIS_MODULE, }, .probe = rt5677_i2c_probe, .remove = rt5677_i2c_remove, diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index e673f6ceb521..f3e646757215 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1601,7 +1601,6 @@ MODULE_DEVICE_TABLE(of, sgtl5000_dt_ids); static struct i2c_driver sgtl5000_i2c_driver = { .driver = { .name = "sgtl5000", - .owner = THIS_MODULE, .of_match_table = sgtl5000_dt_ids, }, .probe = sgtl5000_i2c_probe, diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index f30de7639bb9..5d94d6c7ad33 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -815,7 +815,6 @@ MODULE_DEVICE_TABLE(i2c, ssm2518_i2c_ids); static struct i2c_driver ssm2518_driver = { .driver = { .name = "ssm2518", - .owner = THIS_MODULE, }, .probe = ssm2518_i2c_probe, .remove = ssm2518_i2c_remove, diff --git a/sound/soc/codecs/ssm2602-i2c.c b/sound/soc/codecs/ssm2602-i2c.c index 0d9779d6bfda..173ba85ff59e 100644 --- a/sound/soc/codecs/ssm2602-i2c.c +++ b/sound/soc/codecs/ssm2602-i2c.c @@ -52,7 +52,6 @@ MODULE_DEVICE_TABLE(of, ssm2602_of_match); static struct i2c_driver ssm2602_i2c_driver = { .driver = { .name = "ssm2602", - .owner = THIS_MODULE, .of_match_table = ssm2602_of_match, }, .probe = ssm2602_i2c_probe, diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c index 938d2cb6d78b..f3f1f6874c72 100644 --- a/sound/soc/codecs/ssm4567.c +++ b/sound/soc/codecs/ssm4567.c @@ -453,7 +453,6 @@ MODULE_DEVICE_TABLE(i2c, ssm4567_i2c_ids); static struct i2c_driver ssm4567_driver = { .driver = { .name = "ssm4567", - .owner = THIS_MODULE, }, .probe = ssm4567_i2c_probe, .remove = ssm4567_i2c_remove, diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 60eff36260cb..a9844b2ac829 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -1144,7 +1144,6 @@ MODULE_DEVICE_TABLE(i2c, sta32x_i2c_id); static struct i2c_driver sta32x_i2c_driver = { .driver = { .name = "sta32x", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(st32x_dt_ids), }, .probe = sta32x_i2c_probe, diff --git a/sound/soc/codecs/sta350.c b/sound/soc/codecs/sta350.c index bd819a3f205a..33a4612f0a07 100644 --- a/sound/soc/codecs/sta350.c +++ b/sound/soc/codecs/sta350.c @@ -1264,7 +1264,6 @@ MODULE_DEVICE_TABLE(i2c, sta350_i2c_id); static struct i2c_driver sta350_i2c_driver = { .driver = { .name = "sta350", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(st350_dt_ids), }, .probe = sta350_i2c_probe, diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index 4f70378b2cfb..3430f444c1ae 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -379,7 +379,6 @@ MODULE_DEVICE_TABLE(i2c, sta529_i2c_id); static struct i2c_driver sta529_i2c_driver = { .driver = { .name = "sta529", - .owner = THIS_MODULE, }, .probe = sta529_i2c_probe, .remove = sta529_i2c_remove, diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 71cfae0de3ee..d36afcbbd354 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -769,7 +769,6 @@ MODULE_DEVICE_TABLE(of, tas2552_of_match); static struct i2c_driver tas2552_i2c_driver = { .driver = { .name = "tas2552", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(tas2552_of_match), .pm = &tas2552_pm, }, diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 32942bed34b1..0a49fc851577 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -994,7 +994,6 @@ static int tas5086_i2c_remove(struct i2c_client *i2c) static struct i2c_driver tas5086_i2c_driver = { .driver = { .name = "tas5086", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(tas5086_dt_ids), }, .id_table = tas5086_i2c_id, diff --git a/sound/soc/codecs/tfa9879.c b/sound/soc/codecs/tfa9879.c index bf7596ba49f3..cb5310d89c0f 100644 --- a/sound/soc/codecs/tfa9879.c +++ b/sound/soc/codecs/tfa9879.c @@ -314,7 +314,6 @@ MODULE_DEVICE_TABLE(i2c, tfa9879_i2c_id); static struct i2c_driver tfa9879_i2c_driver = { .driver = { .name = "tfa9879", - .owner = THIS_MODULE, }, .probe = tfa9879_i2c_probe, .remove = tfa9879_i2c_remove, diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index c4c960f592a1..48dd9b2cd0fa 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1283,7 +1283,6 @@ MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id); static struct i2c_driver aic31xx_i2c_driver = { .driver = { .name = "tlv320aic31xx-codec", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(tlv320aic31xx_of_match), }, .probe = aic31xx_i2c_probe, diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index ad6cb90e5f9b..f2d3191961e1 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -871,7 +871,6 @@ MODULE_DEVICE_TABLE(of, aic32x4_of_id); static struct i2c_driver aic32x4_i2c_driver = { .driver = { .name = "tlv320aic32x4", - .owner = THIS_MODULE, .of_match_table = aic32x4_of_id, }, .probe = aic32x4_i2c_probe, diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index a7cf19b53fb2..125a93517cdb 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1825,7 +1825,6 @@ MODULE_DEVICE_TABLE(of, tlv320aic3x_of_match); static struct i2c_driver aic3x_i2c_driver = { .driver = { .name = "tlv320aic3x-codec", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(tlv320aic3x_of_match), }, .probe = aic3x_i2c_probe, diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index d67a311f0e75..781398fb2841 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1585,7 +1585,6 @@ MODULE_DEVICE_TABLE(i2c, tlv320dac33_i2c_id); static struct i2c_driver tlv320dac33_i2c_driver = { .driver = { .name = "tlv320dac33-codec", - .owner = THIS_MODULE, }, .probe = dac33_i2c_probe, .remove = dac33_i2c_remove, diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 6fac9e034c48..265c4c38804d 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -488,7 +488,6 @@ MODULE_DEVICE_TABLE(of, tpa6130a2_of_match); static struct i2c_driver tpa6130a2_i2c_driver = { .driver = { .name = "tpa6130a2", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(tpa6130a2_of_match), }, .probe = tpa6130a2_probe, diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c index ffc6f3028cc5..43568435c208 100644 --- a/sound/soc/codecs/ts3a227e.c +++ b/sound/soc/codecs/ts3a227e.c @@ -377,7 +377,6 @@ MODULE_DEVICE_TABLE(of, ts3a227e_of_match); static struct i2c_driver ts3a227e_driver = { .driver = { .name = "ts3a227e", - .owner = THIS_MODULE, .pm = &ts3a227e_pm, .of_match_table = of_match_ptr(ts3a227e_of_match), }, diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 6e159f59d219..5cafb16ccf1a 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -810,7 +810,6 @@ MODULE_DEVICE_TABLE(i2c, uda1380_i2c_id); static struct i2c_driver uda1380_i2c_driver = { .driver = { .name = "uda1380-codec", - .owner = THIS_MODULE, }, .probe = uda1380_i2c_probe, .remove = uda1380_i2c_remove, diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index 048f00568260..ec45c5b220b1 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -251,7 +251,6 @@ MODULE_DEVICE_TABLE(i2c, wm1250_ev1_i2c_id); static struct i2c_driver wm1250_ev1_i2c_driver = { .driver = { .name = "wm1250-ev1", - .owner = THIS_MODULE, }, .probe = wm1250_ev1_probe, .remove = wm1250_ev1_remove, diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 21d5402e343f..786abd02b140 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -942,7 +942,6 @@ MODULE_DEVICE_TABLE(i2c, wm2000_i2c_id); static struct i2c_driver wm2000_i2c_driver = { .driver = { .name = "wm2000", - .owner = THIS_MODULE, }, .probe = wm2000_i2c_probe, .remove = wm2000_i2c_remove, diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 01aeb75dd774..cdaa175283df 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2495,7 +2495,6 @@ MODULE_DEVICE_TABLE(i2c, wm2200_i2c_id); static struct i2c_driver wm2200_i2c_driver = { .driver = { .name = "wm2200", - .owner = THIS_MODULE, .pm = &wm2200_pm, }, .probe = wm2200_i2c_probe, diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 98495dd61239..e5572d28395d 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2720,7 +2720,6 @@ MODULE_DEVICE_TABLE(i2c, wm5100_i2c_id); static struct i2c_driver wm5100_i2c_driver = { .driver = { .name = "wm5100", - .owner = THIS_MODULE, .pm = &wm5100_pm, }, .probe = wm5100_i2c_probe, diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index dac5beb4d023..3cff5a699e57 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -690,7 +690,6 @@ MODULE_DEVICE_TABLE(i2c, wm8510_i2c_id); static struct i2c_driver wm8510_i2c_driver = { .driver = { .name = "wm8510", - .owner = THIS_MODULE, .of_match_table = wm8510_of_match, }, .probe = wm8510_i2c_probe, diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 43ea8ae5f871..5f8fde56e68b 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -534,7 +534,6 @@ MODULE_DEVICE_TABLE(i2c, wm8523_i2c_id); static struct i2c_driver wm8523_i2c_driver = { .driver = { .name = "wm8523", - .owner = THIS_MODULE, .of_match_table = wm8523_of_match, }, .probe = wm8523_i2c_probe, diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 759a7928ac3e..abf60355f7f7 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -978,7 +978,6 @@ MODULE_DEVICE_TABLE(i2c, wm8580_i2c_id); static struct i2c_driver wm8580_i2c_driver = { .driver = { .name = "wm8580", - .owner = THIS_MODULE, .of_match_table = wm8580_of_match, }, .probe = wm8580_i2c_probe, diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index cc8251f09f8a..44b9e0ae7451 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -478,7 +478,6 @@ MODULE_DEVICE_TABLE(i2c, wm8711_i2c_id); static struct i2c_driver wm8711_i2c_driver = { .driver = { .name = "wm8711", - .owner = THIS_MODULE, .of_match_table = wm8711_of_match, }, .probe = wm8711_i2c_probe, diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index f1a173e6ec33..cd7b02413ccf 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -319,7 +319,6 @@ MODULE_DEVICE_TABLE(i2c, wm8728_i2c_id); static struct i2c_driver wm8728_i2c_driver = { .driver = { .name = "wm8728", - .owner = THIS_MODULE, .of_match_table = wm8728_of_match, }, .probe = wm8728_i2c_probe, diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 628d50c40cc4..c8622329f86d 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -802,7 +802,6 @@ MODULE_DEVICE_TABLE(i2c, wm8731_i2c_id); static struct i2c_driver wm8731_i2c_driver = { .driver = { .name = "wm8731", - .owner = THIS_MODULE, .of_match_table = wm8731_of_match, }, .probe = wm8731_i2c_probe, diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index 6ad606fd8b69..b54a7b78f562 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -657,7 +657,6 @@ MODULE_DEVICE_TABLE(i2c, wm8737_i2c_id); static struct i2c_driver wm8737_i2c_driver = { .driver = { .name = "wm8737", - .owner = THIS_MODULE, .of_match_table = wm8737_of_match, }, .probe = wm8737_i2c_probe, diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index b34623786e35..430fa7ded713 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -633,7 +633,6 @@ MODULE_DEVICE_TABLE(i2c, wm8741_i2c_id); static struct i2c_driver wm8741_i2c_driver = { .driver = { .name = "wm8741", - .owner = THIS_MODULE, .of_match_table = wm8741_of_match, }, .probe = wm8741_i2c_probe, diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 56d89b0865fa..873933a7966f 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -826,7 +826,6 @@ MODULE_DEVICE_TABLE(i2c, wm8750_i2c_id); static struct i2c_driver wm8750_i2c_driver = { .driver = { .name = "wm8750", - .owner = THIS_MODULE, .of_match_table = wm8750_of_match, }, .probe = wm8750_i2c_probe, diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index feb2997a377a..b5e50ffbf99f 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1609,7 +1609,6 @@ MODULE_DEVICE_TABLE(i2c, wm8753_i2c_id); static struct i2c_driver wm8753_i2c_driver = { .driver = { .name = "wm8753", - .owner = THIS_MODULE, .of_match_table = wm8753_of_match, }, .probe = wm8753_i2c_probe, diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index ece9b4456767..592866dd60c7 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -536,7 +536,6 @@ MODULE_DEVICE_TABLE(i2c, wm8776_i2c_id); static struct i2c_driver wm8776_i2c_driver = { .driver = { .name = "wm8776", - .owner = THIS_MODULE, .of_match_table = wm8776_of_match, }, .probe = wm8776_i2c_probe, diff --git a/sound/soc/codecs/wm8804-i2c.c b/sound/soc/codecs/wm8804-i2c.c index 6596f5f3a0c3..f27464c2c5ba 100644 --- a/sound/soc/codecs/wm8804-i2c.c +++ b/sound/soc/codecs/wm8804-i2c.c @@ -49,7 +49,6 @@ MODULE_DEVICE_TABLE(of, wm8804_of_match); static struct i2c_driver wm8804_i2c_driver = { .driver = { .name = "wm8804", - .owner = THIS_MODULE, .pm = &wm8804_pm, .of_match_table = wm8804_of_match, }, diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index f3759ec5a863..98900aa66dc3 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1312,7 +1312,6 @@ MODULE_DEVICE_TABLE(i2c, wm8900_i2c_id); static struct i2c_driver wm8900_i2c_driver = { .driver = { .name = "wm8900", - .owner = THIS_MODULE, }, .probe = wm8900_i2c_probe, .remove = wm8900_i2c_remove, diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index b5322c1544fb..b011253459af 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -2193,7 +2193,6 @@ MODULE_DEVICE_TABLE(i2c, wm8903_i2c_id); static struct i2c_driver wm8903_i2c_driver = { .driver = { .name = "wm8903", - .owner = THIS_MODULE, .of_match_table = wm8903_of_match, }, .probe = wm8903_i2c_probe, diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 07dc40022d4f..b783743dc97e 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2294,7 +2294,6 @@ MODULE_DEVICE_TABLE(i2c, wm8904_i2c_id); static struct i2c_driver wm8904_i2c_driver = { .driver = { .name = "wm8904", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(wm8904_of_match), }, .probe = wm8904_i2c_probe, diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 98ef0ba5c2a4..f6f9395ea38e 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -787,7 +787,6 @@ MODULE_DEVICE_TABLE(i2c, wm8940_i2c_id); static struct i2c_driver wm8940_i2c_driver = { .driver = { .name = "wm8940", - .owner = THIS_MODULE, }, .probe = wm8940_i2c_probe, .remove = wm8940_i2c_remove, diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 2d591c24704b..12e4435f00f8 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -1009,7 +1009,6 @@ MODULE_DEVICE_TABLE(i2c, wm8955_i2c_id); static struct i2c_driver wm8955_i2c_driver = { .driver = { .name = "wm8955", - .owner = THIS_MODULE, }, .probe = wm8955_i2c_probe, .remove = wm8955_i2c_remove, diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 94c5c4681ce5..1ed0720b41f0 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -1216,7 +1216,6 @@ MODULE_DEVICE_TABLE(of, wm8960_of_match); static struct i2c_driver wm8960_i2c_driver = { .driver = { .name = "wm8960", - .owner = THIS_MODULE, .of_match_table = wm8960_of_match, }, .probe = wm8960_i2c_probe, diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index a057662632ff..20a010344fd5 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -982,7 +982,6 @@ MODULE_DEVICE_TABLE(i2c, wm8961_i2c_id); static struct i2c_driver wm8961_i2c_driver = { .driver = { .name = "wm8961", - .owner = THIS_MODULE, }, .probe = wm8961_i2c_probe, .remove = wm8961_i2c_remove, diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 8edc6d203404..c67e3047f022 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3878,7 +3878,6 @@ MODULE_DEVICE_TABLE(of, wm8962_of_match); static struct i2c_driver wm8962_i2c_driver = { .driver = { .name = "wm8962", - .owner = THIS_MODULE, .of_match_table = wm8962_of_match, .pm = &wm8962_pm, }, diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index b51184c4e816..2cdde32c43c6 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -710,7 +710,6 @@ MODULE_DEVICE_TABLE(i2c, wm8971_i2c_id); static struct i2c_driver wm8971_i2c_driver = { .driver = { .name = "wm8971", - .owner = THIS_MODULE, }, .probe = wm8971_i2c_probe, .remove = wm8971_i2c_remove, diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 33b16a7ba82e..0a60677397b3 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -634,7 +634,6 @@ MODULE_DEVICE_TABLE(i2c, wm8974_i2c_id); static struct i2c_driver wm8974_i2c_driver = { .driver = { .name = "wm8974", - .owner = THIS_MODULE, }, .probe = wm8974_i2c_probe, .remove = wm8974_i2c_remove, diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index cfc8cdf49970..d36d6001fbb7 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -1072,7 +1072,6 @@ MODULE_DEVICE_TABLE(i2c, wm8978_i2c_id); static struct i2c_driver wm8978_i2c_driver = { .driver = { .name = "wm8978", - .owner = THIS_MODULE, }, .probe = wm8978_i2c_probe, .remove = wm8978_i2c_remove, diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index 2fdd2c6cc09d..f6861cc4c321 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -1133,7 +1133,6 @@ MODULE_DEVICE_TABLE(i2c, wm8983_i2c_id); static struct i2c_driver wm8983_i2c_driver = { .driver = { .name = "wm8983", - .owner = THIS_MODULE, }, .probe = wm8983_i2c_probe, .remove = wm8983_i2c_remove, diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index 8a85f5004d41..9c3c1517a4f3 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -1144,7 +1144,6 @@ MODULE_DEVICE_TABLE(i2c, wm8985_i2c_id); static struct i2c_driver wm8985_i2c_driver = { .driver = { .name = "wm8985", - .owner = THIS_MODULE, }, .probe = wm8985_i2c_probe, .remove = wm8985_i2c_remove, diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index f13a995af277..c88ce99ce9e1 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -919,7 +919,6 @@ MODULE_DEVICE_TABLE(i2c, wm8988_i2c_id); static struct i2c_driver wm8988_i2c_driver = { .driver = { .name = "wm8988", - .owner = THIS_MODULE, }, .probe = wm8988_i2c_probe, .remove = wm8988_i2c_remove, diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 1993fd2a6f15..83d72d8bc51b 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1356,7 +1356,6 @@ MODULE_DEVICE_TABLE(i2c, wm8990_i2c_id); static struct i2c_driver wm8990_i2c_driver = { .driver = { .name = "wm8990", - .owner = THIS_MODULE, }, .probe = wm8990_i2c_probe, .remove = wm8990_i2c_remove, diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 44a677720828..709d16e338d9 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -1363,7 +1363,6 @@ MODULE_DEVICE_TABLE(i2c, wm8991_i2c_id); static struct i2c_driver wm8991_i2c_driver = { .driver = { .name = "wm8991", - .owner = THIS_MODULE, }, .probe = wm8991_i2c_probe, .remove = wm8991_i2c_remove, diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index f2d5211a9a96..9638accb74b9 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1742,7 +1742,6 @@ MODULE_DEVICE_TABLE(i2c, wm8993_i2c_id); static struct i2c_driver wm8993_i2c_driver = { .driver = { .name = "wm8993", - .owner = THIS_MODULE, }, .probe = wm8993_i2c_probe, .remove = wm8993_i2c_remove, diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 505b65f5734f..eda52a96c1fa 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -2298,7 +2298,6 @@ MODULE_DEVICE_TABLE(i2c, wm8995_i2c_id); static struct i2c_driver wm8995_i2c_driver = { .driver = { .name = "wm8995", - .owner = THIS_MODULE, }, .probe = wm8995_i2c_probe, .remove = wm8995_i2c_remove, diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index c600c8cdba0b..2a696dbb73af 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -3098,7 +3098,6 @@ MODULE_DEVICE_TABLE(i2c, wm8996_i2c_id); static struct i2c_driver wm8996_i2c_driver = { .driver = { .name = "wm8996", - .owner = THIS_MODULE, }, .probe = wm8996_i2c_probe, .remove = wm8996_i2c_remove, diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index ec91ce8a7ca8..45223a20331f 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1378,7 +1378,6 @@ MODULE_DEVICE_TABLE(i2c, wm9081_i2c_id); static struct i2c_driver wm9081_i2c_driver = { .driver = { .name = "wm9081", - .owner = THIS_MODULE, }, .probe = wm9081_i2c_probe, .remove = wm9081_i2c_remove, diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 13d23fc797db..12baf725d4a5 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -636,7 +636,6 @@ MODULE_DEVICE_TABLE(i2c, wm9090_id); static struct i2c_driver wm9090_i2c_driver = { .driver = { .name = "wm9090", - .owner = THIS_MODULE, }, .probe = wm9090_i2c_probe, .remove = wm9090_i2c_remove, -- 1.9.1

2 4

[alsa-devel] [PATCH v2] ASoC: cs4349: Add support for Cirrus Logic CS4349
by timothyc.howe＠gmail.com 15 Jul '15

15 Jul '15

From: Tim Howe <tim.howe(a)cirrus.com> Signed-off-by: Tim Howe <tim.howe(a)cirrus.com> --- Documentation/devicetree/bindings/sound/cs4349.txt | 19 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs4349.c | 414 +++++++++++++++++++++ sound/soc/codecs/cs4349.h | 146 ++++++++ 5 files changed, 587 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs4349.txt create mode 100644 sound/soc/codecs/cs4349.c create mode 100644 sound/soc/codecs/cs4349.h diff --git a/Documentation/devicetree/bindings/sound/cs4349.txt b/Documentation/devicetree/bindings/sound/cs4349.txt new file mode 100644 index 0000000..54c117b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4349.txt @@ -0,0 +1,19 @@ +CS4349 audio CODEC + +Required properties: + + - compatible : "cirrus,cs4349" + + - reg : the I2C address of the device for I2C + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. + +Example: + +codec: cs4349@48 { + compatible = "cirrus,cs4349"; + reg = <0x48>; + reset-gpios = <&gpio 54 0>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7223c7f..4a78606 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -54,6 +54,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271_I2C if I2C select SND_SOC_CS4271_SPI if SPI_MASTER select SND_SOC_CS42XX8_I2C if I2C + select SND_SOC_CS4349 if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI select SND_SOC_DA7213 if I2C @@ -413,6 +414,11 @@ config SND_SOC_CS42XX8_I2C select SND_SOC_CS42XX8 select REGMAP_I2C +# Cirrus Logic CS4349 HiFi DAC +config SND_SOC_CS4349 + tristate "Cirrus Logic CS4349 CODEC" + depends on I2C + config SND_SOC_CX20442 tristate depends on TTY diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 9f7136c..36a9f37 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -46,6 +46,7 @@ snd-soc-cs4271-i2c-objs := cs4271-i2c.o snd-soc-cs4271-spi-objs := cs4271-spi.o snd-soc-cs42xx8-objs := cs42xx8.o snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o +snd-soc-cs4349-objs := cs4349.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -237,6 +238,7 @@ obj-$(CONFIG_SND_SOC_CS4271_I2C) += snd-soc-cs4271-i2c.o obj-$(CONFIG_SND_SOC_CS4271_SPI) += snd-soc-cs4271-spi.o obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o +obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o obj-$(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/cs4349.c b/sound/soc/codecs/cs4349.c new file mode 100644 index 0000000..ee6b676 --- /dev/null +++ b/sound/soc/codecs/cs4349.c @@ -0,0 +1,414 @@ +/* + * cs4349.c -- CS4349 ALSA Soc Audio driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Authors: Tim Howe <Tim.Howe(a)cirrus.com> + * + * 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/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/platform_device.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/of_device.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include "cs4349.h" + + +static const struct reg_default cs4349_reg_defaults[] = { + { 2, 0x00 }, /* r02 - Mode Control */ + { 3, 0x09 }, /* r03 - Volume, Mixing and Inversion Control */ + { 4, 0x81 }, /* r04 - Mute Control */ + { 5, 0x00 }, /* r05 - Channel A Volume Control */ + { 6, 0x00 }, /* r06 - Channel B Volume Control */ + { 7, 0xB1 }, /* r07 - Ramp and Filter Control */ + { 8, 0x1C }, /* r08 - Misc. Control */ +}; + +/* Private data for the CS4349 */ +struct cs4349_private { + struct regmap *regmap; + struct cs4349_platform_data pdata; + struct gpio_desc *reset_gpio; + unsigned int mclk; + unsigned int mode; + int rate; +}; + +static bool cs4349_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS4349_CHIPID: + case CS4349_MODE: + case CS4349_VMI: + case CS4349_MUTE: + case CS4349_VOLA: + case CS4349_VOLB: + case CS4349_RMPFLT: + case CS4349_MISC: + return true; + default: + return false; + } +} + +static int cs4349_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) +{ + struct cs4349_private *cs4349 = snd_soc_codec_get_drvdata(codec); + + cs4349->mclk = freq; + + return 0; +} + +static int cs4349_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4349_private *cs4349 = snd_soc_codec_get_drvdata(codec); + unsigned int fmt; + + fmt = format & SND_SOC_DAIFMT_FORMAT_MASK; + + switch (fmt) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + cs4349->mode = format & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + return -EINVAL; + } + + return 0; +} + +static int cs4349_pcm_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 cs4349_private *cs4349 = snd_soc_codec_get_drvdata(codec); + int mode, fmt, ret; + + mode = snd_soc_read(codec, CS4349_MODE); + cs4349->rate = params_rate(params); + + switch (cs4349->mode) { + case SND_SOC_DAIFMT_I2S: + mode |= MODE_FORMAT(DIF_I2S); + break; + case SND_SOC_DAIFMT_LEFT_J: + mode |= MODE_FORMAT(DIF_LEFT_JST); + break; + case SND_SOC_DAIFMT_RIGHT_J: + switch (params_width(params)) { + case 16: + fmt = DIF_RGHT_JST16; + break; + case 24: + fmt = DIF_RGHT_JST24; + break; + default: + return -EINVAL; + } + mode |= MODE_FORMAT(fmt); + break; + default: + return -EINVAL; + } + + ret = snd_soc_write(codec, CS4349_MODE, mode); + if (ret < 0) + return ret; + + return 0; +} + +static int cs4349_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + int reg; + + reg = 0; + if (mute) + reg = MUTE_AB_MASK; + + return snd_soc_update_bits(codec, CS4349_MUTE, MUTE_AB_MASK, reg); +} + +static DECLARE_TLV_DB_SCALE(dig_tlv, -12750, 50, 0); + +static const char * const chan_mix_texts[] = { + "Mute", "MuteA", "MuteA SwapB", "MuteA MonoB", "SwapA MuteB", + "BothR", "Swap", "SwapA MonoB", "MuteB", "Normal", "BothL", + "MonoB", "MonoA MuteB", "MonoA", "MonoA SwapB", "Mono", + /*Normal == Channel A = Left, Channel B = Right*/ +}; + +static const char * const fm_texts[] = { + "Auto", "Single", "Double", "Quad", +}; + +static const char * const deemph_texts[] = { + "None", "44.1k", "48k", "32k", +}; + +static const char * const softr_zeroc_texts[] = { + "Immediate", "Zero Cross", "Soft Ramp", "SR on ZC", +}; + +static int deemph_values[] = { + 0, 4, 8, 12, +}; + +static int softr_zeroc_values[] = { + 0, 64, 128, 192, +}; + +static const struct soc_enum chan_mix_enum = + SOC_ENUM_SINGLE(CS4349_VMI, 0, + ARRAY_SIZE(chan_mix_texts), + chan_mix_texts); + +static const struct soc_enum fm_mode_enum = + SOC_ENUM_SINGLE(CS4349_MODE, 0, + ARRAY_SIZE(fm_texts), + fm_texts); + +static SOC_VALUE_ENUM_SINGLE_DECL(deemph_enum, CS4349_MODE, 0, DEM_MASK, + deemph_texts, deemph_values); + +static SOC_VALUE_ENUM_SINGLE_DECL(softr_zeroc_enum, CS4349_RMPFLT, 0, + SR_ZC_MASK, softr_zeroc_texts, + softr_zeroc_values); + +static const struct snd_kcontrol_new cs4349_snd_controls[] = { + SOC_DOUBLE_R_TLV("Master Playback Volume", + CS4349_VOLA, CS4349_VOLB, 0, 0xFF, 1, dig_tlv), + SOC_ENUM("Functional Mode", fm_mode_enum), + SOC_ENUM("De-Emphasis Control", deemph_enum), + SOC_ENUM("Soft Ramp Zero Cross Control", softr_zeroc_enum), + SOC_ENUM("Channel Mixer", chan_mix_enum), + SOC_SINGLE("VolA = VolB Switch", CS4349_VMI, 7, 1, 0), + SOC_SINGLE("InvertA Switch", CS4349_VMI, 6, 1, 0), + SOC_SINGLE("InvertB Switch", CS4349_VMI, 5, 1, 0), + SOC_SINGLE("Auto-Mute Switch", CS4349_MUTE, 7, 1, 0), + SOC_SINGLE("MUTEC A = B Switch", CS4349_MUTE, 5, 1, 0), + SOC_SINGLE("Soft Ramp Up Switch", CS4349_RMPFLT, 5, 1, 0), + SOC_SINGLE("Soft Ramp Down Switch", CS4349_RMPFLT, 4, 1, 0), + SOC_SINGLE("Slow Roll Off Filter Switch", CS4349_RMPFLT, 2, 1, 0), + SOC_SINGLE("Freeze Switch", CS4349_MISC, 5, 1, 0), + SOC_SINGLE("Popguard Switch", CS4349_MISC, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget cs4349_dapm_widgets[] = { + SND_SOC_DAPM_DAC("HiFi DAC", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_OUTPUT("OutputA"), + SND_SOC_DAPM_OUTPUT("OutputB"), +}; + +static const struct snd_soc_dapm_route cs4349_routes[] = { + {"DAC Playback", NULL, "OutputA"}, + {"DAC Playback", NULL, "OutputB"}, + + {"OutputA", NULL, "HiFi DAC"}, + {"OutputB", NULL, "HiFi DAC"}, +}; + +#define CS4349_PCM_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE) + +#define CS4349_PCM_RATES SNDRV_PCM_RATE_8000_192000 + +static const struct snd_soc_dai_ops cs4349_dai_ops = { + .hw_params = cs4349_pcm_hw_params, + .set_fmt = cs4349_set_dai_fmt, + .digital_mute = cs4349_digital_mute, +}; + +static struct snd_soc_dai_driver cs4349_dai = { + .name = "cs4349_hifi", + .playback = { + .stream_name = "DAC Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS4349_PCM_RATES, + .formats = CS4349_PCM_FORMATS, + }, + .ops = &cs4349_dai_ops, + .symmetric_rates = 1, +}; + +static struct snd_soc_codec_driver soc_codec_dev_cs4349 = { + .set_sysclk = cs4349_codec_set_sysclk, + + .controls = cs4349_snd_controls, + .num_controls = ARRAY_SIZE(cs4349_snd_controls), + + .dapm_widgets = cs4349_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs4349_dapm_widgets), + .dapm_routes = cs4349_routes, + .num_dapm_routes = ARRAY_SIZE(cs4349_routes), +}; + +static struct regmap_config cs4349_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS4349_NUMREGS, + .reg_defaults = cs4349_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4349_reg_defaults), + .readable_reg = cs4349_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs4349_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct cs4349_private *cs4349; + struct cs4349_platform_data *pdata = dev_get_platdata(&client->dev); + int ret = 0; + + cs4349 = devm_kzalloc(&client->dev, sizeof(*cs4349), GFP_KERNEL); + if (!cs4349) + return -ENOMEM; + + cs4349->regmap = devm_regmap_init_i2c(client, &cs4349_regmap); + if (IS_ERR(cs4349->regmap)) { + ret = PTR_ERR(cs4349->regmap); + dev_err(&client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + if (pdata) + cs4349->pdata = *pdata; + + /* Reset the Device */ + cs4349->reset_gpio = devm_gpiod_get_optional(&client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs4349->reset_gpio)) + return PTR_ERR(cs4349->reset_gpio); + + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 1); + + i2c_set_clientdata(client, cs4349); + + return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4349, + &cs4349_dai, 1); +} + +static int cs4349_i2c_remove(struct i2c_client *client) +{ + struct cs4349_private *cs4349 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + /* Hold down reset */ + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 0); + + return 0; +} + +#ifdef CONFIG_PM +static int cs4349_runtime_suspend(struct device *dev) +{ + struct cs4349_private *cs4349 = dev_get_drvdata(dev); + struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); + int ret; + + ret = snd_soc_update_bits(rtd->codec, CS4349_MISC, PWR_DWN, 1); + if (ret < 0) + return ret; + + regcache_cache_only(cs4349->regmap, true); + regcache_sync(cs4349->regmap); + + /* Hold down reset */ + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 0); + + return 0; +} + +static int cs4349_runtime_resume(struct device *dev) +{ + struct cs4349_private *cs4349 = dev_get_drvdata(dev); + struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); + int ret; + + ret = snd_soc_update_bits(rtd->codec, CS4349_MISC, PWR_DWN, 0); + if (ret < 0) + return ret; + + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 1); + + regcache_cache_only(cs4349->regmap, false); + regcache_sync(cs4349->regmap); + + return 0; +} +#endif + +static const struct dev_pm_ops cs4349_runtime_pm = { + SET_RUNTIME_PM_OPS(cs4349_runtime_suspend, cs4349_runtime_resume, + NULL) +}; + +static const struct of_device_id cs4349_of_match[] = { + { .compatible = "cirrus,cs4349", }, + {}, +}; + +MODULE_DEVICE_TABLE(of, cs4349_of_match); + +static const struct i2c_device_id cs4349_i2c_id[] = { + {"cs4349", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs4349_i2c_id); + +static struct i2c_driver cs4349_i2c_driver = { + .driver = { + .name = "cs4349", + .owner = THIS_MODULE, + .of_match_table = cs4349_of_match, + }, + .id_table = cs4349_i2c_id, + .probe = cs4349_i2c_probe, + .remove = cs4349_i2c_remove, +}; + +module_i2c_driver(cs4349_i2c_driver); + +MODULE_AUTHOR("Tim Howe <tim.howe(a)cirrus.com>"); +MODULE_DESCRIPTION("Cirrus Logic CS4349 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs4349.h b/sound/soc/codecs/cs4349.h new file mode 100644 index 0000000..3884a89 --- /dev/null +++ b/sound/soc/codecs/cs4349.h @@ -0,0 +1,146 @@ +/* + * ALSA SoC CS4349 codec driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Author: Tim Howe <Tim.Howe(a)cirrus.com> + * + * 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. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + */ + +#ifndef __CS4349_H__ +#define __CS4349_H__ + +struct cs4349_platform_data { + + /* GPIO for Reset */ + unsigned int gpio_nreset; + +}; + +/* CS4349 registers addresses */ +#define CS4349_CHIPID 0x01 /* Device and Rev ID, Read Only */ +#define CS4349_MODE 0x02 /* Mode Control */ +#define CS4349_VMI 0x03 /* Volume, Mixing, Inversion Control */ +#define CS4349_MUTE 0x04 /* Mute Control */ +#define CS4349_VOLA 0x05 /* DAC Channel A Volume Control */ +#define CS4349_VOLB 0x06 /* DAC Channel B Volume Control */ +#define CS4349_RMPFLT 0x07 /* Ramp and Filter Control */ +#define CS4349_MISC 0x08 /* Power Down,Freeze Control,Pop Stop*/ + +#define CS4349_FIRSTREG 0x01 +#define CS4349_LASTREG 0x08 +#define CS4349_NUMREGS (CS4349_LASTREG - CS4349_FIRSTREG + 1) +#define CS4349_I2C_INCR 0x80 + + +/* Device and Revision ID */ +#define CS4349_REVA 0xF0 /* Rev A */ +#define CS4349_REVB 0xF1 /* Rev B */ +#define CS4349_REVC2 0xFF /* Rev C2 */ + + +/* PDN_DONE Poll Maximum + * If soft ramp is set it will take much longer to power down + * the system. + */ +#define PDN_POLL_MAX 900 + + +/* Bitfield Definitions */ + +/* CS4349_MODE */ +/* (Digital Interface Format, De-Emphasis Control, Functional Mode */ +#define DIF2 (1 << 6) +#define DIF1 (1 << 5) +#define DIF0 (1 << 4) +#define DEM1 (1 << 3) +#define DEM0 (1 << 2) +#define FM1 (1 << 1) +#define DIF_LEFT_JST 0x00 +#define DIF_I2S 0x01 +#define DIF_RGHT_JST16 0x02 +#define DIF_RGHT_JST24 0x03 +#define DIF_TDM0 0x04 +#define DIF_TDM1 0x05 +#define DIF_TDM2 0x06 +#define DIF_TDM3 0x07 +#define DIF_MASK 0x70 +#define MODE_FORMAT(x) (((x)&7)<<4) +#define DEM_MASK 0x0C +#define NO_DEM 0x00 +#define DEM_441 0x04 +#define DEM_48K 0x08 +#define DEM_32K 0x0C +#define FM_AUTO 0x00 +#define FM_SNGL 0x01 +#define FM_DBL 0x02 +#define FM_QUAD 0x03 +#define FM_SNGL_MIN 30000 +#define FM_SNGL_MAX 54000 +#define FM_DBL_MAX 108000 +#define FM_QUAD_MAX 216000 +#define FM_MASK 0x03 + +/* CS4349_VMI (VMI = Volume, Mixing and Inversion Controls) */ +#define VOLBISA (1 << 7) +#define VOLAISB (1 << 7) +/* INVERT_A only available for Left Jstfd, Right Jstfd16 and Right Jstfd24 */ +#define INVERT_A (1 << 6) +/* INVERT_B only available for Left Jstfd, Right Jstfd16 and Right Jstfd24 */ +#define INVERT_B (1 << 5) +#define ATAPI3 (1 << 3) +#define ATAPI2 (1 << 2) +#define ATAPI1 (1 << 1) +#define ATAPI0 (1 << 0) +#define MUTEAB 0x00 +#define MUTEA_RIGHTB 0x01 +#define MUTEA_LEFTB 0x02 +#define MUTEA_SUMLRDIV2B 0x03 +#define RIGHTA_MUTEB 0x04 +#define RIGHTA_RIGHTB 0x05 +#define RIGHTA_LEFTB 0x06 +#define RIGHTA_SUMLRDIV2B 0x07 +#define LEFTA_MUTEB 0x08 +#define LEFTA_RIGHTB 0x09 /* Default */ +#define LEFTA_LEFTB 0x0A +#define LEFTA_SUMLRDIV2B 0x0B +#define SUMLRDIV2A_MUTEB 0x0C +#define SUMLRDIV2A_RIGHTB 0x0D +#define SUMLRDIV2A_LEFTB 0x0E +#define SUMLRDIV2_AB 0x0F +#define CHMIX_MASK 0x0F + +/* CS4349_MUTE */ +#define AUTOMUTE (1 << 7) +#define MUTEC_AB (1 << 5) +#define MUTE_A (1 << 4) +#define MUTE_B (1 << 3) +#define MUTE_AB_MASK 0x18 + +/* CS4349_RMPFLT (Ramp and Filter Control) */ +#define SCZ1 (1 << 7) +#define SCZ0 (1 << 6) +#define RMP_UP (1 << 5) +#define RMP_DN (1 << 4) +#define FILT_SEL (1 << 2) +#define IMMDT_CHNG 0x31 +#define ZEROCRSS 0x71 +#define SOFT_RMP 0xB1 +#define SFTRMP_ZEROCRSS 0xF1 +#define SR_ZC_MASK 0xC0 + +/* CS4349_MISC */ +#define PWR_DWN (1 << 7) +#define FREEZE (1 << 5) +#define POPG_EN (1 << 4) + +#endif /* __CS4349_H__ */ -- 2.4.5

2 1

[alsa-devel] armv7 raspberry pi kernel compilation and snd_usb_us122l tascam us122 module requires x86
by James Miller 15 Jul '15

15 Jul '15

hi - I am compiling a rt kernel for my raspberry pi, and I notice I am unable to select the us122l driver for inclusion as a module as it is listed as requiring x86 to be selected. Many of the other modules are available and use the same codebase as the us122l driver, so I was wondering if the x86 requirement is strictly necessary? Is it possible to check that the us122l module requires x86 and is incompatible with arm architecture please. Many thanks James -- James Stewart Miller Bsc(hons) Psych.

2 1

[alsa-devel] Issue in alsa when dma complete race with pcm release
by Shengjiu Wang 15 Jul '15

15 Jul '15

Hi alsa-devel There maybe a issue in ALSA when dma complete race with snd_pcm_release. The pcm release and dma complete are in different thread. There is occasion that dmaengine_pcm_dma_complete() is called too late, some memory has been freed, the prtd is null. Then there is kernel dump. Is there any solution for this issue? Thanks. Best regards wang shengjiu

2 9

[alsa-devel] [PATCH] ALSA: hda - Fix Skylake codec timeouts
by David Henningsson 15 Jul '15

15 Jul '15

When the controller is powered up but the HDMI codec is powered down on Skylake, the power well is turned off. When the codec is then powered up again, we need to poke the codec a little extra to make sure it wakes up. Otherwise we'll get sad "no response from codec" messages and broken audio. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> Tested-by: Woodrow Shen <woodrow.shen(a)canonical.com> --- * It would good to have an ack from Intel on this patch, since they have better hardware knowledge than I. * Also I haven't really kept track of all recent reorganisation of the hda driver so I'm not totally sure how many kernels back (if any) that this applies to sound/pci/hda/hda_intel.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ca151b4..872e9a7 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -567,9 +567,17 @@ static int azx_position_check(struct azx *chip, struct azx_dev *azx_dev) /* Enable/disable i915 display power for the link */ static int azx_intel_link_power(struct azx *chip, bool enable) { + int err; struct hdac_bus *bus = azx_bus(chip); - return snd_hdac_display_power(bus, enable); + err = snd_hdac_display_power(bus, enable); + if (err < 0) + return err; + if (enable && (chip->driver_caps & AZX_DCAPS_I915_POWERWELL)) { + snd_hdac_set_codec_wakeup(bus, true); + snd_hdac_set_codec_wakeup(bus, false); + } + return 0; } /* -- 1.9.1

2 3

[alsa-devel] [PATCH v2] ASoC: cs4349: Add support for Cirrus Logic CS4349
by timothyc.howe＠gmail.com 15 Jul '15

15 Jul '15

From: Tim Howe <tim.howe(a)cirrus.com> Signed-off-by: Tim Howe <tim.howe(a)cirrus.com> --- sound/soc/codecs/cs4349.c | 27 ++++++++------------------- 1 file changed, 8 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c index c9a96b6..ee6b676 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -305,19 +305,8 @@ static int cs4349_i2c_probe(struct i2c_client *client, return ret; } - if (pdata) { + if (pdata) cs4349->pdata = *pdata; - } else { - pdata = devm_kzalloc(&client->dev, - sizeof(struct cs4349_platform_data), - GFP_KERNEL); - if (!pdata) { - dev_err(&client->dev, - "could not allocate pdata\n"); - return -ENOMEM; - } - cs4349->pdata = *pdata; - } /* Reset the Device */ cs4349->reset_gpio = devm_gpiod_get_optional(&client->dev, @@ -330,9 +319,6 @@ static int cs4349_i2c_probe(struct i2c_client *client, i2c_set_clientdata(client, cs4349); - dev_info(&client->dev, - "Cirrus Logic CS4349\n"); - return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4349, &cs4349_dai, 1); } @@ -357,14 +343,17 @@ static int cs4349_runtime_suspend(struct device *dev) struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); int ret; - /* Hold down reset */ - if (cs4349->reset_gpio) - gpiod_set_value_cansleep(cs4349->reset_gpio, 0); - ret = snd_soc_update_bits(rtd->codec, CS4349_MISC, PWR_DWN, 1); if (ret < 0) return ret; + regcache_cache_only(cs4349->regmap, true); + regcache_sync(cs4349->regmap); + + /* Hold down reset */ + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 0); + return 0; } -- 2.4.5

2 1

[alsa-devel] [GIT PULL] Braswell firmware update
by Vinod Koul 14 Jul '15

14 Jul '15

Hello Linux firmware maintainers, Please PULL to receive the updated BSW firmware binary which contains power management fixes. We have also updated the version of binary in WHENCE file The following changes since commit 3161bfa479d5e9ed4f46b57df9bcecbbc4f8eb3c: linux-firmware: Update firmware patch for Intel Bluetooth 7260 (B3/B4) (2015-05-13 11:12:48 -0400) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git BSW for you to fetch changes up to e5fa14afd09909124fc7618fc4a0ac00fa4f9347: linux-firmware: update audio firmware for Braswell platform (2015-06-17 11:21:47 +0530) ---------------------------------------------------------------- Vinod Koul (1): linux-firmware: update audio firmware for Braswell platform WHENCE | 1 + intel/fw_sst_22a8.bin | Bin 647390 -> 701614 bytes 2 files changed, 1 insertion(+) Thanks -- ~Vinod

2 2

[alsa-devel] [PATCH] cht_bsw_rt5645: Fix Kernel crash with strcpy
by Jenny TC 14 Jul '15

14 Jul '15

Kernel crash is observed when strcpy is used with const char*. The crash is reported only if driver is loaded as a module or the probe is called in the late stage of Kernel boot (EPROBE_DEFER). This is because Kernel enables read only protection near the end. Issue is addressed by replacing the strcpy with pointer assignment Signed-off-by: Jenny TC <jenny.tc(a)intel.com> --- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index bdcaf46..3665d6c 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -367,7 +367,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) card->dev = &pdev->dev; sprintf(codec_name, "i2c-%s:00", drv->acpi_card->codec_id); /* set correct codec name */ - strcpy((char *)card->dai_link[2].codec_name, codec_name); + card->dai_link[2].codec_name = codec_name; snd_soc_card_set_drvdata(card, drv); ret_val = devm_snd_soc_register_card(&pdev->dev, card); if (ret_val) { -- 1.7.9.5

1 0

[alsa-devel] [PATCH] ALSA: line6: Fix -EBUSY error during active monitoring
by Takashi Iwai 14 Jul '15

14 Jul '15

When a monitor stream is active, the next PCM stream access results in EBUSY error because of the check in line6_stream_start(). Fix this by just skipping the submission of pending URBs when the stream is already running instead. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=101431 Cc: <stable(a)vger.kernel.org> # v4.0+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/line6/pcm.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/usb/line6/pcm.c b/sound/usb/line6/pcm.c index 8461d6bf992f..204cc074adb9 100644 --- a/sound/usb/line6/pcm.c +++ b/sound/usb/line6/pcm.c @@ -186,12 +186,8 @@ static int line6_stream_start(struct snd_line6_pcm *line6pcm, int direction, int ret = 0; spin_lock_irqsave(&pstr->lock, flags); - if (!test_and_set_bit(type, &pstr->running)) { - if (pstr->active_urbs || pstr->unlink_urbs) { - ret = -EBUSY; - goto error; - } - + if (!test_and_set_bit(type, &pstr->running) && + !(pstr->active_urbs || pstr->unlink_urbs)) { pstr->count = 0; /* Submit all currently available URBs */ if (direction == SNDRV_PCM_STREAM_PLAYBACK) @@ -199,7 +195,6 @@ static int line6_stream_start(struct snd_line6_pcm *line6pcm, int direction, else ret = line6_submit_audio_in_all_urbs(line6pcm); } - error: if (ret < 0) clear_bit(type, &pstr->running); spin_unlock_irqrestore(&pstr->lock, flags); -- 2.4.5

1 0

[alsa-devel] ASoC Multiple outputs
by Matt Redfearn 14 Jul '15

14 Jul '15

Hi list, I am writing an ALSA driver for an SoC / board which has a single I2S source. The source may be connected internally to either a codec outputting to analogue headphones, or to an HDMI transmitter. At the moment I have set up two snd_soc_dai_link within the same card connecting the source to each sink. static struct snd_soc_dai_link ci20_dai_link[] = { { .name = "ci20", .stream_name = "headphones", .cpu_dai_name = "jz4780-i2s", .platform_name = "jz4780-i2s", .codec_dai_name = "jz4780-hifi", .codec_name = "jz4780-codec", .init = ci20_init, .ops = &ci20_audio_dai_ops, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, }, { .name = "ci20 HDMI", .stream_name = "hdmi", .cpu_dai_name = "jz4780-i2s", .platform_name = "jz4780-i2s", .codec_dai_name = "dw-hdmi-hifi", .codec_name = "dw-hdmi-audio", .init = ci20_hdmi_init, .ops = &ci20_audio_dai_ops, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, } }; This works and creates 2 devices, and I am able to get mplayer to play sound out to either the headphones or HDMI using the argument "-ao alsa:device=hw=0.0" or "-ao alsa:device=hw=0.1" aplay -l **** List of PLAYBACK Hardware Devices **** card 0: ci20 [ci20], device 0: headphones jz4780-hifi-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: ci20 [ci20], device 1: hdmi jz4780-hdmi-hifi-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 However, alsamixer does not show anything about the second device (HDMI), only the analogue output associated with device 0. Please could someone advise what is the "correct" way to set up multiple links in such a system such that the headphones / HDMI outputs appear in the mixer and are available to pulseaudio. Should I create multiple cards? Should I create subdevices for the different outputs? If so, what structures should be set up within the driver to achieve this? Suggestions of any documentation / websites that may help would be great! Thanks! Matt

2 1

[alsa-devel] [PATCH] ASoC: topology: Fix to add dapm mixer info
by Subhransu S. Prusty 14 Jul '15

14 Jul '15

From: Jeeja KP <jeeja.kp(a)intel.com> Mixer control for widgets can't be created if the info is NULL. So assign the correct info for this. Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Acked-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index f38f111..70bcfe9 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -144,7 +144,7 @@ static const struct snd_soc_tplg_kcontrol_ops io_ops[] = { {SND_SOC_TPLG_CTL_STROBE, snd_soc_get_strobe, snd_soc_put_strobe, NULL}, {SND_SOC_TPLG_DAPM_CTL_VOLSW, snd_soc_dapm_get_volsw, - snd_soc_dapm_put_volsw, NULL}, + snd_soc_dapm_put_volsw, snd_soc_info_volsw}, {SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE, snd_soc_dapm_get_enum_double, snd_soc_dapm_put_enum_double, snd_soc_info_enum_double}, {SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT, snd_soc_dapm_get_enum_double, -- 2.4.3

2 1

[alsa-devel] [PATCH] ASoC: max98925: Remove control_data initialization
by Lars-Peter Clausen 14 Jul '15

14 Jul '15

The control_data field of the snd_soc_codec struct is no longer used by ASoC drivers using standard regmap IO. There is no need to initialize the field to point to the drivers regmap struct, so drop that. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/max98925.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c index aad6642..eddf8bc 100644 --- a/sound/soc/codecs/max98925.c +++ b/sound/soc/codecs/max98925.c @@ -523,7 +523,6 @@ static int max98925_probe(struct snd_soc_codec *codec) struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); max98925->codec = codec; - codec->control_data = max98925->regmap; regmap_write(max98925->regmap, MAX98925_GLOBAL_ENABLE, 0x00); /* It's not the default but we need to set DAI_DLY */ regmap_write(max98925->regmap, -- 2.1.4

2 1

[alsa-devel] [PATCH 1/3] ASoC: max98357a: Make 'sdmode-gpios' dts property optional
by Anatol Pomozov 14 Jul '15

14 Jul '15

The option is not needed if chip is always on or managed by some other part of system like platform card driver. Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- Documentation/devicetree/bindings/sound/max98357a.txt | 6 +++++- sound/soc/codecs/max98357a.c | 5 ++++- 2 files changed, 9 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/max98357a.txt b/Documentation/devicetree/bindings/sound/max98357a.txt index a7a149a..28645a2 100644 --- a/Documentation/devicetree/bindings/sound/max98357a.txt +++ b/Documentation/devicetree/bindings/sound/max98357a.txt @@ -4,7 +4,11 @@ This node models the Maxim MAX98357A DAC. Required properties: - compatible : "maxim,max98357a" -- sdmode-gpios : GPIO specifier for the GPIO -> DAC SDMODE pin + +Optional properties: +- sdmode-gpios : GPIO specifier for the chip's SD_MODE pin. + If this option is not specified then driver does not manage + the pin state (e.g. chip is always on). Example: diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 3a2fda0..fa1b793 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -31,6 +31,9 @@ static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, { struct gpio_desc *sdmode = snd_soc_dai_get_drvdata(dai); + if (!sdmode) + return 0; + switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: @@ -60,7 +63,7 @@ static int max98357a_codec_probe(struct snd_soc_codec *codec) { struct gpio_desc *sdmode; - sdmode = devm_gpiod_get(codec->dev, "sdmode", GPIOD_OUT_LOW); + sdmode = devm_gpiod_get_optional(codec->dev, "sdmode", GPIOD_OUT_LOW); if (IS_ERR(sdmode)) { dev_err(codec->dev, "%s() unable to get sdmode GPIO: %ld\n", __func__, PTR_ERR(sdmode)); -- 2.4.5

3 6

[alsa-devel] [PATCH - CS4349 2/2] Add devicetree bindings for Cirrus Logic CS4349
by timothyc.howe＠gmail.com 13 Jul '15

13 Jul '15

From: Tim Howe <tim.howe(a)cirrus.com> Signed-off-by: Tim Howe <tim.howe(a)cirrus.com> --- Documentation/devicetree/bindings/sound/cs4349.txt | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs4349.txt diff --git a/Documentation/devicetree/bindings/sound/cs4349.txt b/Documentation/devicetree/bindings/sound/cs4349.txt new file mode 100644 index 0000000..54c117b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4349.txt @@ -0,0 +1,19 @@ +CS4349 audio CODEC + +Required properties: + + - compatible : "cirrus,cs4349" + + - reg : the I2C address of the device for I2C + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. + +Example: + +codec: cs4349@48 { + compatible = "cirrus,cs4349"; + reg = <0x48>; + reset-gpios = <&gpio 54 0>; +}; -- 2.4.5

2 1

[alsa-devel] [PATCH - CS4349 1/2] Add support for Cirrus Logic CS4349
by timothyc.howe＠gmail.com 13 Jul '15

13 Jul '15

From: Tim Howe <tim.howe(a)cirrus.com> Signed-off-by: Tim Howe <tim.howe(a)cirrus.com> --- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs4349.c | 425 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs4349.h | 146 ++++++++++++++++ 4 files changed, 579 insertions(+) create mode 100644 sound/soc/codecs/cs4349.c create mode 100644 sound/soc/codecs/cs4349.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7223c7f..4a78606 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -54,6 +54,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271_I2C if I2C select SND_SOC_CS4271_SPI if SPI_MASTER select SND_SOC_CS42XX8_I2C if I2C + select SND_SOC_CS4349 if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI select SND_SOC_DA7213 if I2C @@ -413,6 +414,11 @@ config SND_SOC_CS42XX8_I2C select SND_SOC_CS42XX8 select REGMAP_I2C +# Cirrus Logic CS4349 HiFi DAC +config SND_SOC_CS4349 + tristate "Cirrus Logic CS4349 CODEC" + depends on I2C + config SND_SOC_CX20442 tristate depends on TTY diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 9f7136c..36a9f37 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -46,6 +46,7 @@ snd-soc-cs4271-i2c-objs := cs4271-i2c.o snd-soc-cs4271-spi-objs := cs4271-spi.o snd-soc-cs42xx8-objs := cs42xx8.o snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o +snd-soc-cs4349-objs := cs4349.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -237,6 +238,7 @@ obj-$(CONFIG_SND_SOC_CS4271_I2C) += snd-soc-cs4271-i2c.o obj-$(CONFIG_SND_SOC_CS4271_SPI) += snd-soc-cs4271-spi.o obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o +obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o obj-$(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/cs4349.c b/sound/soc/codecs/cs4349.c new file mode 100644 index 0000000..81fc2a0 --- /dev/null +++ b/sound/soc/codecs/cs4349.c @@ -0,0 +1,425 @@ +/* + * cs4349.c -- CS4349 ALSA Soc Audio driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Authors: Tim Howe <Tim.Howe(a)cirrus.com> + * + * 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/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/platform_device.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/of_device.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include "cs4349.h" + + +static const struct reg_default cs4349_reg_defaults[] = { + { 2, 0x00 }, /* r02 - Mode Control */ + { 3, 0x09 }, /* r03 - Volume, Mixing and Inversion Control */ + { 4, 0x81 }, /* r04 - Mute Control */ + { 5, 0x00 }, /* r05 - Channel A Volume Control */ + { 6, 0x00 }, /* r06 - Channel B Volume Control */ + { 7, 0xB1 }, /* r07 - Ramp and Filter Control */ + { 8, 0x1C }, /* r08 - Misc. Control */ +}; + +/* Private data for the CS4349 */ +struct cs4349_private { + struct regmap *regmap; + struct cs4349_platform_data pdata; + struct gpio_desc *reset_gpio; + unsigned int mclk; + unsigned int mode; + int rate; +}; + +static bool cs4349_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS4349_CHIPID: + case CS4349_MODE: + case CS4349_VMI: + case CS4349_MUTE: + case CS4349_VOLA: + case CS4349_VOLB: + case CS4349_RMPFLT: + case CS4349_MISC: + return true; + default: + return false; + } +} + +static int cs4349_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) +{ + struct cs4349_private *cs4349 = snd_soc_codec_get_drvdata(codec); + + cs4349->mclk = freq; + + return 0; +} + +static int cs4349_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4349_private *cs4349 = snd_soc_codec_get_drvdata(codec); + unsigned int fmt; + + fmt = format & SND_SOC_DAIFMT_FORMAT_MASK; + + switch (fmt) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + cs4349->mode = format & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + return -EINVAL; + } + + return 0; +} + +static int cs4349_pcm_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 cs4349_private *cs4349 = snd_soc_codec_get_drvdata(codec); + int mode, fmt, ret; + + mode = snd_soc_read(codec, CS4349_MODE); + cs4349->rate = params_rate(params); + + switch (cs4349->mode) { + case SND_SOC_DAIFMT_I2S: + mode |= MODE_FORMAT(DIF_I2S); + break; + case SND_SOC_DAIFMT_LEFT_J: + mode |= MODE_FORMAT(DIF_LEFT_JST); + break; + case SND_SOC_DAIFMT_RIGHT_J: + switch (params_width(params)) { + case 16: + fmt = DIF_RGHT_JST16; + break; + case 24: + fmt = DIF_RGHT_JST24; + break; + default: + return -EINVAL; + } + mode |= MODE_FORMAT(fmt); + break; + default: + return -EINVAL; + } + + ret = snd_soc_write(codec, CS4349_MODE, mode); + if (ret < 0) + return ret; + + return 0; +} + +static int cs4349_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + int reg; + + reg = 0; + if (mute) + reg = MUTE_AB_MASK; + + return snd_soc_update_bits(codec, CS4349_MUTE, MUTE_AB_MASK, reg); +} + +static DECLARE_TLV_DB_SCALE(dig_tlv, -12750, 50, 0); + +static const char * const chan_mix_texts[] = { + "Mute", "MuteA", "MuteA SwapB", "MuteA MonoB", "SwapA MuteB", + "BothR", "Swap", "SwapA MonoB", "MuteB", "Normal", "BothL", + "MonoB", "MonoA MuteB", "MonoA", "MonoA SwapB", "Mono", + /*Normal == Channel A = Left, Channel B = Right*/ +}; + +static const char * const fm_texts[] = { + "Auto", "Single", "Double", "Quad", +}; + +static const char * const deemph_texts[] = { + "None", "44.1k", "48k", "32k", +}; + +static const char * const softr_zeroc_texts[] = { + "Immediate", "Zero Cross", "Soft Ramp", "SR on ZC", +}; + +static int deemph_values[] = { + 0, 4, 8, 12, +}; + +static int softr_zeroc_values[] = { + 0, 64, 128, 192, +}; + +static const struct soc_enum chan_mix_enum = + SOC_ENUM_SINGLE(CS4349_VMI, 0, + ARRAY_SIZE(chan_mix_texts), + chan_mix_texts); + +static const struct soc_enum fm_mode_enum = + SOC_ENUM_SINGLE(CS4349_MODE, 0, + ARRAY_SIZE(fm_texts), + fm_texts); + +static SOC_VALUE_ENUM_SINGLE_DECL(deemph_enum, CS4349_MODE, 0, DEM_MASK, + deemph_texts, deemph_values); + +static SOC_VALUE_ENUM_SINGLE_DECL(softr_zeroc_enum, CS4349_RMPFLT, 0, + SR_ZC_MASK, softr_zeroc_texts, + softr_zeroc_values); + +static const struct snd_kcontrol_new cs4349_snd_controls[] = { + SOC_DOUBLE_R_TLV("Master Playback Volume", + CS4349_VOLA, CS4349_VOLB, 0, 0xFF, 1, dig_tlv), + SOC_ENUM("Functional Mode", fm_mode_enum), + SOC_ENUM("De-Emphasis Control", deemph_enum), + SOC_ENUM("Soft Ramp Zero Cross Control", softr_zeroc_enum), + SOC_ENUM("Channel Mixer", chan_mix_enum), + SOC_SINGLE("VolA = VolB Switch", CS4349_VMI, 7, 1, 0), + SOC_SINGLE("InvertA Switch", CS4349_VMI, 6, 1, 0), + SOC_SINGLE("InvertB Switch", CS4349_VMI, 5, 1, 0), + SOC_SINGLE("Auto-Mute Switch", CS4349_MUTE, 7, 1, 0), + SOC_SINGLE("MUTEC A = B Switch", CS4349_MUTE, 5, 1, 0), + SOC_SINGLE("Soft Ramp Up Switch", CS4349_RMPFLT, 5, 1, 0), + SOC_SINGLE("Soft Ramp Down Switch", CS4349_RMPFLT, 4, 1, 0), + SOC_SINGLE("Slow Roll Off Filter Switch", CS4349_RMPFLT, 2, 1, 0), + SOC_SINGLE("Freeze Switch", CS4349_MISC, 5, 1, 0), + SOC_SINGLE("Popguard Switch", CS4349_MISC, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget cs4349_dapm_widgets[] = { + SND_SOC_DAPM_DAC("HiFi DAC", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_OUTPUT("OutputA"), + SND_SOC_DAPM_OUTPUT("OutputB"), +}; + +static const struct snd_soc_dapm_route cs4349_routes[] = { + {"DAC Playback", NULL, "OutputA"}, + {"DAC Playback", NULL, "OutputB"}, + + {"OutputA", NULL, "HiFi DAC"}, + {"OutputB", NULL, "HiFi DAC"}, +}; + +#define CS4349_PCM_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE) + +#define CS4349_PCM_RATES SNDRV_PCM_RATE_8000_192000 + +static const struct snd_soc_dai_ops cs4349_dai_ops = { + .hw_params = cs4349_pcm_hw_params, + .set_fmt = cs4349_set_dai_fmt, + .digital_mute = cs4349_digital_mute, +}; + +static struct snd_soc_dai_driver cs4349_dai = { + .name = "cs4349_hifi", + .playback = { + .stream_name = "DAC Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS4349_PCM_RATES, + .formats = CS4349_PCM_FORMATS, + }, + .ops = &cs4349_dai_ops, + .symmetric_rates = 1, +}; + +static struct snd_soc_codec_driver soc_codec_dev_cs4349 = { + .set_sysclk = cs4349_codec_set_sysclk, + + .controls = cs4349_snd_controls, + .num_controls = ARRAY_SIZE(cs4349_snd_controls), + + .dapm_widgets = cs4349_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs4349_dapm_widgets), + .dapm_routes = cs4349_routes, + .num_dapm_routes = ARRAY_SIZE(cs4349_routes), +}; + +static struct regmap_config cs4349_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS4349_NUMREGS, + .reg_defaults = cs4349_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4349_reg_defaults), + .readable_reg = cs4349_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs4349_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct cs4349_private *cs4349; + struct cs4349_platform_data *pdata = dev_get_platdata(&client->dev); + int ret = 0; + + cs4349 = devm_kzalloc(&client->dev, sizeof(*cs4349), GFP_KERNEL); + if (!cs4349) + return -ENOMEM; + + cs4349->regmap = devm_regmap_init_i2c(client, &cs4349_regmap); + if (IS_ERR(cs4349->regmap)) { + ret = PTR_ERR(cs4349->regmap); + dev_err(&client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + if (pdata) { + cs4349->pdata = *pdata; + } else { + pdata = devm_kzalloc(&client->dev, + sizeof(struct cs4349_platform_data), + GFP_KERNEL); + if (!pdata) { + dev_err(&client->dev, + "could not allocate pdata\n"); + return -ENOMEM; + } + cs4349->pdata = *pdata; + } + + /* Reset the Device */ + cs4349->reset_gpio = devm_gpiod_get_optional(&client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs4349->reset_gpio)) + return PTR_ERR(cs4349->reset_gpio); + + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 1); + + i2c_set_clientdata(client, cs4349); + + dev_info(&client->dev, + "Cirrus Logic CS4349\n"); + + return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4349, + &cs4349_dai, 1); +} + +static int cs4349_i2c_remove(struct i2c_client *client) +{ + struct cs4349_private *cs4349 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + /* Hold down reset */ + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 0); + + return 0; +} + +#ifdef CONFIG_PM +static int cs4349_runtime_suspend(struct device *dev) +{ + struct cs4349_private *cs4349 = dev_get_drvdata(dev); + struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); + int ret; + + /* Hold down reset */ + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 0); + + ret = snd_soc_update_bits(rtd->codec, CS4349_MISC, PWR_DWN, 1); + if (ret < 0) + return ret; + + return 0; +} + +static int cs4349_runtime_resume(struct device *dev) +{ + struct cs4349_private *cs4349 = dev_get_drvdata(dev); + struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); + int ret; + + ret = snd_soc_update_bits(rtd->codec, CS4349_MISC, PWR_DWN, 0); + if (ret < 0) + return ret; + + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 1); + + regcache_cache_only(cs4349->regmap, false); + regcache_sync(cs4349->regmap); + + return 0; +} +#endif + +static const struct dev_pm_ops cs4349_runtime_pm = { + SET_RUNTIME_PM_OPS(cs4349_runtime_suspend, cs4349_runtime_resume, + NULL) +}; + +static const struct of_device_id cs4349_of_match[] = { + { .compatible = "cirrus,cs4349", }, + {}, +}; + +MODULE_DEVICE_TABLE(of, cs4349_of_match); + +static const struct i2c_device_id cs4349_i2c_id[] = { + {"cs4349", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs4349_i2c_id); + +static struct i2c_driver cs4349_i2c_driver = { + .driver = { + .name = "cs4349", + .owner = THIS_MODULE, + .of_match_table = cs4349_of_match, + }, + .id_table = cs4349_i2c_id, + .probe = cs4349_i2c_probe, + .remove = cs4349_i2c_remove, +}; + +module_i2c_driver(cs4349_i2c_driver); + +MODULE_AUTHOR("Tim Howe <tim.howe(a)cirrus.com>"); +MODULE_DESCRIPTION("Cirrus Logic CS4349 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs4349.h b/sound/soc/codecs/cs4349.h new file mode 100644 index 0000000..3884a89 --- /dev/null +++ b/sound/soc/codecs/cs4349.h @@ -0,0 +1,146 @@ +/* + * ALSA SoC CS4349 codec driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Author: Tim Howe <Tim.Howe(a)cirrus.com> + * + * 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. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + */ + +#ifndef __CS4349_H__ +#define __CS4349_H__ + +struct cs4349_platform_data { + + /* GPIO for Reset */ + unsigned int gpio_nreset; + +}; + +/* CS4349 registers addresses */ +#define CS4349_CHIPID 0x01 /* Device and Rev ID, Read Only */ +#define CS4349_MODE 0x02 /* Mode Control */ +#define CS4349_VMI 0x03 /* Volume, Mixing, Inversion Control */ +#define CS4349_MUTE 0x04 /* Mute Control */ +#define CS4349_VOLA 0x05 /* DAC Channel A Volume Control */ +#define CS4349_VOLB 0x06 /* DAC Channel B Volume Control */ +#define CS4349_RMPFLT 0x07 /* Ramp and Filter Control */ +#define CS4349_MISC 0x08 /* Power Down,Freeze Control,Pop Stop*/ + +#define CS4349_FIRSTREG 0x01 +#define CS4349_LASTREG 0x08 +#define CS4349_NUMREGS (CS4349_LASTREG - CS4349_FIRSTREG + 1) +#define CS4349_I2C_INCR 0x80 + + +/* Device and Revision ID */ +#define CS4349_REVA 0xF0 /* Rev A */ +#define CS4349_REVB 0xF1 /* Rev B */ +#define CS4349_REVC2 0xFF /* Rev C2 */ + + +/* PDN_DONE Poll Maximum + * If soft ramp is set it will take much longer to power down + * the system. + */ +#define PDN_POLL_MAX 900 + + +/* Bitfield Definitions */ + +/* CS4349_MODE */ +/* (Digital Interface Format, De-Emphasis Control, Functional Mode */ +#define DIF2 (1 << 6) +#define DIF1 (1 << 5) +#define DIF0 (1 << 4) +#define DEM1 (1 << 3) +#define DEM0 (1 << 2) +#define FM1 (1 << 1) +#define DIF_LEFT_JST 0x00 +#define DIF_I2S 0x01 +#define DIF_RGHT_JST16 0x02 +#define DIF_RGHT_JST24 0x03 +#define DIF_TDM0 0x04 +#define DIF_TDM1 0x05 +#define DIF_TDM2 0x06 +#define DIF_TDM3 0x07 +#define DIF_MASK 0x70 +#define MODE_FORMAT(x) (((x)&7)<<4) +#define DEM_MASK 0x0C +#define NO_DEM 0x00 +#define DEM_441 0x04 +#define DEM_48K 0x08 +#define DEM_32K 0x0C +#define FM_AUTO 0x00 +#define FM_SNGL 0x01 +#define FM_DBL 0x02 +#define FM_QUAD 0x03 +#define FM_SNGL_MIN 30000 +#define FM_SNGL_MAX 54000 +#define FM_DBL_MAX 108000 +#define FM_QUAD_MAX 216000 +#define FM_MASK 0x03 + +/* CS4349_VMI (VMI = Volume, Mixing and Inversion Controls) */ +#define VOLBISA (1 << 7) +#define VOLAISB (1 << 7) +/* INVERT_A only available for Left Jstfd, Right Jstfd16 and Right Jstfd24 */ +#define INVERT_A (1 << 6) +/* INVERT_B only available for Left Jstfd, Right Jstfd16 and Right Jstfd24 */ +#define INVERT_B (1 << 5) +#define ATAPI3 (1 << 3) +#define ATAPI2 (1 << 2) +#define ATAPI1 (1 << 1) +#define ATAPI0 (1 << 0) +#define MUTEAB 0x00 +#define MUTEA_RIGHTB 0x01 +#define MUTEA_LEFTB 0x02 +#define MUTEA_SUMLRDIV2B 0x03 +#define RIGHTA_MUTEB 0x04 +#define RIGHTA_RIGHTB 0x05 +#define RIGHTA_LEFTB 0x06 +#define RIGHTA_SUMLRDIV2B 0x07 +#define LEFTA_MUTEB 0x08 +#define LEFTA_RIGHTB 0x09 /* Default */ +#define LEFTA_LEFTB 0x0A +#define LEFTA_SUMLRDIV2B 0x0B +#define SUMLRDIV2A_MUTEB 0x0C +#define SUMLRDIV2A_RIGHTB 0x0D +#define SUMLRDIV2A_LEFTB 0x0E +#define SUMLRDIV2_AB 0x0F +#define CHMIX_MASK 0x0F + +/* CS4349_MUTE */ +#define AUTOMUTE (1 << 7) +#define MUTEC_AB (1 << 5) +#define MUTE_A (1 << 4) +#define MUTE_B (1 << 3) +#define MUTE_AB_MASK 0x18 + +/* CS4349_RMPFLT (Ramp and Filter Control) */ +#define SCZ1 (1 << 7) +#define SCZ0 (1 << 6) +#define RMP_UP (1 << 5) +#define RMP_DN (1 << 4) +#define FILT_SEL (1 << 2) +#define IMMDT_CHNG 0x31 +#define ZEROCRSS 0x71 +#define SOFT_RMP 0xB1 +#define SFTRMP_ZEROCRSS 0xF1 +#define SR_ZC_MASK 0xC0 + +/* CS4349_MISC */ +#define PWR_DWN (1 << 7) +#define FREEZE (1 << 5) +#define POPG_EN (1 << 4) + +#endif /* __CS4349_H__ */ -- 2.4.5

2 1

[alsa-devel] randconfig build error with next-20150713, in sound/soc/codecs/sti-sas.c
by Jim Davis 13 Jul '15

13 Jul '15

Building with the attached random configuration file, warning: (INTEL_SOC_DTS_IOSF_CORE && PUNIT_ATOM_DEBUG) selects IOSF_MBI which has unmet direct dependencies (PCI) warning: (SND_SOC_INTEL_BYTCR_RT5640_MACH && SND_SOC_INTEL_CHT_BSW_RT5672_MACH && SND_SOC_INTEL_CHT_BSW_RT5645_MACH && SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) selects SND_SST_IPC_ACPI which has unmet direct dependencies (SOUND && !M68K && !UML && SND && SND_SOC && ACPI) warning: (SND_SOC_ALL_CODECS) selects SND_SOC_STI_SAS which has unmet direct dependencies (SOUND && !M68K && !UML && SND && SND_SOC && SND_SOC_STI) sound/soc/codecs/sti-sas.c: In function ‘stih416_dac_probe’: sound/soc/codecs/sti-sas.c:227:13: error: implicit declaration of function ‘devm_reset_control_get’ [-Werror=implicit-function-declaration] dac->rst = devm_reset_control_get(codec->dev, "dac_rst"); ^ sound/soc/codecs/sti-sas.c:227:11: warning: assignment makes pointer from integer without a cast [-Wint-conversion] dac->rst = devm_reset_control_get(codec->dev, "dac_rst"); ^ sound/soc/codecs/sti-sas.c:231:14: warning: cast from pointer to integer of different size [-Wpointer-to-int-cast] __func__, (int)dac->rst); ^ cc1: some warnings being treated as errors

1 0

[alsa-devel] [PATCH 1/2] ASoC: rt5645: Fix missing free_irq
by Koro Chen 13 Jul '15

13 Jul '15

The driver does not free irq if snd_soc_register_codec fails. It does not return error when request irq failed, either. Fix this by using devm_request_threaded_irq(), and returns when error. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- sound/soc/codecs/rt5645.c | 15 ++++++++------- 1 file changed, 8 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9dfa431..f9f2db8 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3427,11 +3427,15 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, INIT_DELAYED_WORK(&rt5645->jack_detect_work, rt5645_jack_detect_work); if (rt5645->i2c->irq) { - ret = request_threaded_irq(rt5645->i2c->irq, NULL, rt5645_irq, - IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING - | IRQF_ONESHOT, "rt5645", rt5645); - if (ret) + ret = devm_request_threaded_irq(&i2c->dev, rt5645->i2c->irq, + NULL, rt5645_irq, + IRQF_TRIGGER_RISING | + IRQF_TRIGGER_FALLING | + IRQF_ONESHOT, "rt5645", rt5645); + if (ret) { dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + return ret; + } } return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5645, @@ -3442,9 +3446,6 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) { struct rt5645_priv *rt5645 = i2c_get_clientdata(i2c); - if (i2c->irq) - free_irq(i2c->irq, rt5645); - cancel_delayed_work_sync(&rt5645->jack_detect_work); snd_soc_unregister_codec(&i2c->dev); -- 1.8.1.1.dirty

2 6

[alsa-devel] [PATCH] ASoC: add rt298 codec driver
by Bard Liao 13 Jul '15

13 Jul '15

It is the initial version of ALC298 codec driver. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- include/sound/rt298.h | 20 + sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt298.c | 1274 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt298.h | 206 ++++++++ 5 files changed, 1509 insertions(+) create mode 100644 include/sound/rt298.h create mode 100644 sound/soc/codecs/rt298.c create mode 100644 sound/soc/codecs/rt298.h diff --git a/include/sound/rt298.h b/include/sound/rt298.h new file mode 100644 index 0000000..7fffeaa --- /dev/null +++ b/include/sound/rt298.h @@ -0,0 +1,20 @@ +/* + * linux/sound/rt286.h -- Platform data for RT286 + * + * Copyright 2013 Realtek Microelectronics + * + * 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. + */ + +#ifndef __LINUX_SND_RT298_H +#define __LINUX_SND_RT298_H + +struct rt298_platform_data { + bool cbj_en; /*combo jack enable*/ + bool gpio2_en; /*GPIO2 enable*/ + bool suspend_power_off; /* power is off during suspend */ +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index efaafce..76125a2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -83,6 +83,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM512x_I2C if I2C select SND_SOC_PCM512x_SPI if SPI_MASTER select SND_SOC_RT286 if I2C + select SND_SOC_RT298 if I2C select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C select SND_SOC_RT5645 if I2C @@ -512,12 +513,18 @@ config SND_SOC_RL6231 config SND_SOC_RL6347A tristate default y if SND_SOC_RT286=y + default y if SND_SOC_RT298=y default m if SND_SOC_RT286=m + default m if SND_SOC_RT298=m config SND_SOC_RT286 tristate depends on I2C +config SND_SOC_RT298 + tristate + depends on I2C + config SND_SOC_RT5631 tristate "Realtek ALC5631/RT5631 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index cf160d9..3b58c45 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -79,6 +79,7 @@ snd-soc-pcm512x-spi-objs := pcm512x-spi.o snd-soc-rl6231-objs := rl6231.o snd-soc-rl6347a-objs := rl6347a.o snd-soc-rt286-objs := rt286.o +snd-soc-rt298-objs := rt298.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o snd-soc-rt5645-objs := rt5645.o @@ -266,6 +267,7 @@ obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o +obj-$(CONFIG_SND_SOC_RT298) += snd-soc-rt298.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_RT5645) += snd-soc-rt5645.o diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c new file mode 100644 index 0000000..7c4bcb6 --- /dev/null +++ b/sound/soc/codecs/rt298.c @@ -0,0 +1,1274 @@ +/* + * rt298.c -- RT298 ALSA SoC audio codec driver + * + * Copyright 2015 Realtek Semiconductor Corp. + * Author: Bard Liao <bardliao(a)realtek.com> + * + * 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/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/acpi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/jack.h> +#include <linux/workqueue.h> +#include <sound/rt298.h> +#include <sound/hda_verbs.h> + +#include "rl6347a.h" +#include "rt298.h" + +#define RT298_VENDOR_ID 0x10ec0298 + +struct rt298_priv { + struct reg_default *index_cache; + int index_cache_size; + struct regmap *regmap; + struct snd_soc_codec *codec; + struct rt298_platform_data pdata; + struct i2c_client *i2c; + struct snd_soc_jack *jack; + struct delayed_work jack_detect_work; + int sys_clk; + int clk_id; + int is_hp_in; +}; + +static struct reg_default rt298_index_def[] = { + { 0x01, 0xaaaa }, + { 0x02, 0x8aaa }, + { 0x03, 0x0002 }, + { 0x04, 0xaf01 }, + { 0x08, 0x000d }, + { 0x09, 0xd810 }, + { 0x0a, 0x0120 }, + { 0x0b, 0x0000 }, + { 0x0d, 0x2800 }, + { 0x0f, 0x0000 }, + { 0x19, 0x0a17 }, + { 0x20, 0x0020 }, + { 0x33, 0x0208 }, + { 0x46, 0x0300 }, + { 0x49, 0x0004 }, + { 0x4f, 0x50e9 }, + { 0x50, 0x2000 }, + { 0x63, 0x2902 }, + { 0x67, 0x1111 }, + { 0x68, 0x1016 }, + { 0x69, 0x273f }, +}; +#define INDEX_CACHE_SIZE ARRAY_SIZE(rt298_index_def) + +static const struct reg_default rt298_reg[] = { + { 0x00170500, 0x00000400 }, + { 0x00220000, 0x00000031 }, + { 0x00239000, 0x0000007f }, + { 0x0023a000, 0x0000007f }, + { 0x00270500, 0x00000400 }, + { 0x00370500, 0x00000400 }, + { 0x00870500, 0x00000400 }, + { 0x00920000, 0x00000031 }, + { 0x00935000, 0x000000c3 }, + { 0x00936000, 0x000000c3 }, + { 0x00970500, 0x00000400 }, + { 0x00b37000, 0x00000097 }, + { 0x00b37200, 0x00000097 }, + { 0x00b37300, 0x00000097 }, + { 0x00c37000, 0x00000000 }, + { 0x00c37100, 0x00000080 }, + { 0x01270500, 0x00000400 }, + { 0x01370500, 0x00000400 }, + { 0x01371f00, 0x411111f0 }, + { 0x01439000, 0x00000080 }, + { 0x0143a000, 0x00000080 }, + { 0x01470700, 0x00000000 }, + { 0x01470500, 0x00000400 }, + { 0x01470c00, 0x00000000 }, + { 0x01470100, 0x00000000 }, + { 0x01837000, 0x00000000 }, + { 0x01870500, 0x00000400 }, + { 0x02050000, 0x00000000 }, + { 0x02139000, 0x00000080 }, + { 0x0213a000, 0x00000080 }, + { 0x02170100, 0x00000000 }, + { 0x02170500, 0x00000400 }, + { 0x02170700, 0x00000000 }, + { 0x02270100, 0x00000000 }, + { 0x02370100, 0x00000000 }, + { 0x01870700, 0x00000020 }, + { 0x00830000, 0x000000c3 }, + { 0x00930000, 0x000000c3 }, + { 0x01270700, 0x00000000 }, +}; + +static bool rt298_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0 ... 0xff: + case RT298_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID): + case RT298_GET_HP_SENSE: + case RT298_GET_MIC1_SENSE: + case RT298_PROC_COEF: + case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT298_MIC1, 0): + case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT298_SPK_OUT, 0): + case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT298_HP_OUT, 0): + return true; + default: + return true; + } + + +} + +static bool rt298_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0 ... 0xff: + case RT298_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID): + case RT298_GET_HP_SENSE: + case RT298_GET_MIC1_SENSE: + case RT298_SET_AUDIO_POWER: + case RT298_SET_HPO_POWER: + case RT298_SET_SPK_POWER: + case RT298_SET_DMIC1_POWER: + case RT298_SPK_MUX: + case RT298_HPO_MUX: + case RT298_ADC0_MUX: + case RT298_ADC1_MUX: + case RT298_SET_MIC1: + case RT298_SET_PIN_HPO: + case RT298_SET_PIN_SPK: + case RT298_SET_PIN_DMIC1: + case RT298_SPK_EAPD: + case RT298_SET_AMP_GAIN_HPO: + case RT298_SET_DMIC2_DEFAULT: + case RT298_DACL_GAIN: + case RT298_DACR_GAIN: + case RT298_ADCL_GAIN: + case RT298_ADCR_GAIN: + case RT298_MIC_GAIN: + case RT298_SPOL_GAIN: + case RT298_SPOR_GAIN: + case RT298_HPOL_GAIN: + case RT298_HPOR_GAIN: + case RT298_F_DAC_SWITCH: + case RT298_F_RECMIX_SWITCH: + case RT298_REC_MIC_SWITCH: + case RT298_REC_I2S_SWITCH: + case RT298_REC_LINE_SWITCH: + case RT298_REC_BEEP_SWITCH: + case RT298_DAC_FORMAT: + case RT298_ADC_FORMAT: + case RT298_COEF_INDEX: + case RT298_PROC_COEF: + case RT298_SET_AMP_GAIN_ADC_IN1: + case RT298_SET_AMP_GAIN_ADC_IN2: + case RT298_SET_POWER(RT298_DAC_OUT1): + case RT298_SET_POWER(RT298_DAC_OUT2): + case RT298_SET_POWER(RT298_ADC_IN1): + case RT298_SET_POWER(RT298_ADC_IN2): + case RT298_SET_POWER(RT298_DMIC2): + case RT298_SET_POWER(RT298_MIC1): + case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT298_MIC1, 0): + case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT298_SPK_OUT, 0): + case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT298_HP_OUT, 0): + return true; + default: + return false; + } +} + +#ifdef CONFIG_PM +static void rt298_index_sync(struct snd_soc_codec *codec) +{ + struct rt298_priv *rt298 = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < INDEX_CACHE_SIZE; i++) { + snd_soc_write(codec, rt298->index_cache[i].reg, + rt298->index_cache[i].def); + } +} +#endif + +static int rt298_support_power_controls[] = { + RT298_DAC_OUT1, + RT298_DAC_OUT2, + RT298_ADC_IN1, + RT298_ADC_IN2, + RT298_MIC1, + RT298_DMIC1, + RT298_DMIC2, + RT298_SPK_OUT, + RT298_HP_OUT, +}; +#define RT298_POWER_REG_LEN ARRAY_SIZE(rt298_support_power_controls) + +static int rt298_jack_detect(struct rt298_priv *rt298, bool *hp, bool *mic) +{ + struct snd_soc_dapm_context *dapm; + unsigned int val, buf; + + *hp = false; + *mic = false; + + if (!rt298->codec) + return -EINVAL; + + dapm = snd_soc_codec_get_dapm(rt298->codec); + + if (rt298->pdata.cbj_en) { + regmap_read(rt298->regmap, RT298_GET_HP_SENSE, &buf); + *hp = buf & 0x80000000; + if (*hp == rt298->is_hp_in) + return -1; + rt298->is_hp_in = *hp; + if (*hp) { + /* power on HV,VERF */ + regmap_update_bits(rt298->regmap, + RT298_DC_GAIN, 0x200, 0x200); + + snd_soc_dapm_force_enable_pin(dapm, "HV"); + snd_soc_dapm_force_enable_pin(dapm, "VREF"); + /* power LDO1 */ + snd_soc_dapm_force_enable_pin(dapm, "LDO1"); + snd_soc_dapm_sync(dapm); + + regmap_write(rt298->regmap, RT298_SET_MIC1, 0x24); + msleep(50); + + regmap_update_bits(rt298->regmap, + RT298_CBJ_CTRL1, 0xfcc0, 0xd400); + msleep(300); + regmap_read(rt298->regmap, RT298_CBJ_CTRL2, &val); + + if (0x0070 == (val & 0x0070)) { + *mic = true; + } else { + regmap_update_bits(rt298->regmap, + RT298_CBJ_CTRL1, 0xfcc0, 0xe400); + msleep(300); + regmap_read(rt298->regmap, + RT298_CBJ_CTRL2, &val); + if (0x0070 == (val & 0x0070)) + *mic = true; + else + *mic = false; + } + regmap_update_bits(rt298->regmap, + RT298_DC_GAIN, 0x200, 0x0); + + } else { + *mic = false; + regmap_write(rt298->regmap, RT298_SET_MIC1, 0x20); + } + } else { + regmap_read(rt298->regmap, RT298_GET_HP_SENSE, &buf); + *hp = buf & 0x80000000; + regmap_read(rt298->regmap, RT298_GET_MIC1_SENSE, &buf); + *mic = buf & 0x80000000; + } + + snd_soc_dapm_disable_pin(dapm, "HV"); + snd_soc_dapm_disable_pin(dapm, "VREF"); + if (!*hp) + snd_soc_dapm_disable_pin(dapm, "LDO1"); + snd_soc_dapm_sync(dapm); + + pr_debug("*hp = %d *mic = %d\n", *hp, *mic); + + return 0; +} + +static void rt298_jack_detect_work(struct work_struct *work) +{ + struct rt298_priv *rt298 = + container_of(work, struct rt298_priv, jack_detect_work.work); + int status = 0; + bool hp = false; + bool mic = false; + + if (rt298_jack_detect(rt298, &hp, &mic) < 0) + return; + + if (hp == true) + status |= SND_JACK_HEADPHONE; + + if (mic == true) + status |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(rt298->jack, status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); +} + +int rt298_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) +{ + struct rt298_priv *rt298 = snd_soc_codec_get_drvdata(codec); + + rt298->jack = jack; + + /* Send an initial empty report */ + snd_soc_jack_report(rt298->jack, 0, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + + return 0; +} +EXPORT_SYMBOL_GPL(rt298_mic_detect); + +static int is_mclk_mode(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); + struct rt298_priv *rt298 = snd_soc_codec_get_drvdata(codec); + + if (rt298->clk_id == RT298_SCLK_S_MCLK) + return 1; + else + return 0; +} + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); + +static const struct snd_kcontrol_new rt298_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC0 Playback Volume", RT298_DACL_GAIN, + RT298_DACR_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_DOUBLE_R_TLV("ADC0 Capture Volume", RT298_ADCL_GAIN, + RT298_ADCR_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_SINGLE_TLV("AMIC Volume", RT298_MIC_GAIN, + 0, 0x3, 0, mic_vol_tlv), + SOC_DOUBLE_R("Speaker Playback Switch", RT298_SPOL_GAIN, + RT298_SPOR_GAIN, RT298_MUTE_SFT, 1, 1), +}; + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt298_front_mix[] = { + SOC_DAPM_SINGLE("DAC Switch", RT298_F_DAC_SWITCH, + RT298_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("RECMIX Switch", RT298_F_RECMIX_SWITCH, + RT298_MUTE_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt298_rec_mix[] = { + SOC_DAPM_SINGLE("Mic1 Switch", RT298_REC_MIC_SWITCH, + RT298_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("I2S Switch", RT298_REC_I2S_SWITCH, + RT298_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("Line1 Switch", RT298_REC_LINE_SWITCH, + RT298_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("Beep Switch", RT298_REC_BEEP_SWITCH, + RT298_MUTE_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new spo_enable_control = + SOC_DAPM_SINGLE("Switch", RT298_SET_PIN_SPK, + RT298_SET_PIN_SFT, 1, 0); + +static const struct snd_kcontrol_new hpol_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT298_HPOL_GAIN, + RT298_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hpor_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT298_HPOR_GAIN, + RT298_MUTE_SFT, 1, 1); + +/* ADC0 source */ +static const char * const rt298_adc_src[] = { + "Mic", "RECMIX", "Dmic" +}; + +static const int rt298_adc_values[] = { + 0, 4, 5, +}; + +static SOC_VALUE_ENUM_SINGLE_DECL( + rt298_adc0_enum, RT298_ADC0_MUX, RT298_ADC_SEL_SFT, + RT298_ADC_SEL_MASK, rt298_adc_src, rt298_adc_values); + +static const struct snd_kcontrol_new rt298_adc0_mux = + SOC_DAPM_ENUM("ADC 0 source", rt298_adc0_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL( + rt298_adc1_enum, RT298_ADC1_MUX, RT298_ADC_SEL_SFT, + RT298_ADC_SEL_MASK, rt298_adc_src, rt298_adc_values); + +static const struct snd_kcontrol_new rt298_adc1_mux = + SOC_DAPM_ENUM("ADC 1 source", rt298_adc1_enum); + +static const char * const rt298_dac_src[] = { + "Front", "Surround" +}; +/* HP-OUT source */ +static SOC_ENUM_SINGLE_DECL(rt298_hpo_enum, RT298_HPO_MUX, + 0, rt298_dac_src); + +static const struct snd_kcontrol_new rt298_hpo_mux = +SOC_DAPM_ENUM("HPO source", rt298_hpo_enum); + +/* SPK-OUT source */ +static SOC_ENUM_SINGLE_DECL(rt298_spo_enum, RT298_SPK_MUX, + 0, rt298_dac_src); + +static const struct snd_kcontrol_new rt298_spo_mux = +SOC_DAPM_ENUM("SPO source", rt298_spo_enum); + +static int rt298_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_write(codec, + RT298_SPK_EAPD, RT298_SET_EAPD_HIGH); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_write(codec, + RT298_SPK_EAPD, RT298_SET_EAPD_LOW); + break; + + default: + return 0; + } + + return 0; +} + +static int rt298_set_dmic1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_write(codec, RT298_SET_PIN_DMIC1, 0x20); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_write(codec, RT298_SET_PIN_DMIC1, 0); + break; + default: + return 0; + } + + return 0; +} + +static int rt298_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + unsigned int nid; + + nid = (w->reg >> 20) & 0xff; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), + 0x7080, 0x7000); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), + 0x7080, 0x7080); + break; + default: + return 0; + } + + return 0; +} + +static int rt298_mic1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(codec, + RT298_A_BIAS_CTRL3, 0xc000, 0x8000); + snd_soc_update_bits(codec, + RT298_A_BIAS_CTRL2, 0xc000, 0x8000); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_update_bits(codec, + RT298_A_BIAS_CTRL3, 0xc000, 0x0000); + snd_soc_update_bits(codec, + RT298_A_BIAS_CTRL2, 0xc000, 0x0000); + break; + default: + return 0; + } + + return 0; +} + +static int rt298_vref_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(codec, + RT298_CBJ_CTRL1, 0x0400, 0x0000); + mdelay(50); + break; + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt298_dapm_widgets[] = { + + SND_SOC_DAPM_SUPPLY_S("HV", 1, RT298_POWER_CTRL1, + 12, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("VREF", RT298_POWER_CTRL1, + 0, 1, rt298_vref_event, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY_S("BG_MBIAS", 1, RT298_POWER_CTRL2, + 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("LDO1", 1, RT298_POWER_CTRL2, + 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("LDO2", 1, RT298_POWER_CTRL2, + 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("VREF1", 1, RT298_POWER_CTRL2, + 4, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("LV", 2, RT298_POWER_CTRL1, + 13, 1, NULL, 0), + + + SND_SOC_DAPM_SUPPLY("MCLK MODE", RT298_PLL_CTRL1, + 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC1 Input Buffer", SND_SOC_NOPM, + 0, 0, rt298_mic1_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC1 Pin"), + SND_SOC_DAPM_INPUT("DMIC2 Pin"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("LINE1"), + SND_SOC_DAPM_INPUT("Beep"), + + /* DMIC */ + SND_SOC_DAPM_PGA_E("DMIC1", RT298_SET_POWER(RT298_DMIC1), 0, 1, + NULL, 0, rt298_set_dmic1_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA("DMIC2", RT298_SET_POWER(RT298_DMIC2), 0, 1, + NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC Receiver", SND_SOC_NOPM, + 0, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX", SND_SOC_NOPM, 0, 0, + rt298_rec_mix, ARRAY_SIZE(rt298_rec_mix)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC 0", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX_E("ADC 0 Mux", RT298_SET_POWER(RT298_ADC_IN1), 0, 1, + &rt298_adc0_mux, rt298_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MUX_E("ADC 1 Mux", RT298_SET_POWER(RT298_ADC_IN2), 0, 1, + &rt298_adc1_mux, rt298_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + + /* Output Side */ + /* DACs */ + SND_SOC_DAPM_DAC("DAC 0", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC 1", NULL, SND_SOC_NOPM, 0, 0), + + /* Output Mux */ + SND_SOC_DAPM_MUX("SPK Mux", SND_SOC_NOPM, 0, 0, &rt298_spo_mux), + SND_SOC_DAPM_MUX("HPO Mux", SND_SOC_NOPM, 0, 0, &rt298_hpo_mux), + + SND_SOC_DAPM_SUPPLY("HP Power", RT298_SET_PIN_HPO, + RT298_SET_PIN_SFT, 0, NULL, 0), + + /* Output Mixer */ + SND_SOC_DAPM_MIXER("Front", RT298_SET_POWER(RT298_DAC_OUT1), 0, 1, + rt298_front_mix, ARRAY_SIZE(rt298_front_mix)), + SND_SOC_DAPM_PGA("Surround", RT298_SET_POWER(RT298_DAC_OUT2), 0, 1, + NULL, 0), + + /* Output Pga */ + SND_SOC_DAPM_SWITCH_E("SPO", SND_SOC_NOPM, 0, 0, + &spo_enable_control, rt298_spk_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SWITCH("HPO L", SND_SOC_NOPM, 0, 0, + &hpol_enable_control), + SND_SOC_DAPM_SWITCH("HPO R", SND_SOC_NOPM, 0, 0, + &hpor_enable_control), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("SPOL"), + SND_SOC_DAPM_OUTPUT("SPOR"), + SND_SOC_DAPM_OUTPUT("HPO Pin"), + SND_SOC_DAPM_OUTPUT("SPDIF"), +}; + +static const struct snd_soc_dapm_route rt298_dapm_routes[] = { + + {"ADC 0", NULL, "MCLK MODE", is_mclk_mode}, + {"ADC 1", NULL, "MCLK MODE", is_mclk_mode}, + {"Front", NULL, "MCLK MODE", is_mclk_mode}, + {"Surround", NULL, "MCLK MODE", is_mclk_mode}, + + {"HP Power", NULL, "LDO1"}, + {"HP Power", NULL, "LDO2"}, + {"HP Power", NULL, "LV"}, + {"HP Power", NULL, "VREF1"}, + {"HP Power", NULL, "BG_MBIAS"}, + + {"MIC1", NULL, "LDO1"}, + {"MIC1", NULL, "LDO2"}, + {"MIC1", NULL, "HV"}, + {"MIC1", NULL, "LV"}, + {"MIC1", NULL, "VREF"}, + {"MIC1", NULL, "VREF1"}, + {"MIC1", NULL, "BG_MBIAS"}, + {"MIC1", NULL, "MIC1 Input Buffer"}, + + {"SPO", NULL, "LDO1"}, + {"SPO", NULL, "LDO2"}, + {"SPO", NULL, "HV"}, + {"SPO", NULL, "LV"}, + {"SPO", NULL, "VREF"}, + {"SPO", NULL, "VREF1"}, + {"SPO", NULL, "BG_MBIAS"}, + + {"DMIC1", NULL, "DMIC1 Pin"}, + {"DMIC2", NULL, "DMIC2 Pin"}, + {"DMIC1", NULL, "DMIC Receiver"}, + {"DMIC2", NULL, "DMIC Receiver"}, + + {"RECMIX", "Beep Switch", "Beep"}, + {"RECMIX", "Line1 Switch", "LINE1"}, + {"RECMIX", "Mic1 Switch", "MIC1"}, + + {"ADC 0 Mux", "Dmic", "DMIC1"}, + {"ADC 0 Mux", "RECMIX", "RECMIX"}, + {"ADC 0 Mux", "Mic", "MIC1"}, + {"ADC 1 Mux", "Dmic", "DMIC2"}, + {"ADC 1 Mux", "RECMIX", "RECMIX"}, + {"ADC 1 Mux", "Mic", "MIC1"}, + + {"ADC 0", NULL, "ADC 0 Mux"}, + {"ADC 1", NULL, "ADC 1 Mux"}, + + {"AIF1TX", NULL, "ADC 0"}, + {"AIF2TX", NULL, "ADC 1"}, + + {"DAC 0", NULL, "AIF1RX"}, + {"DAC 1", NULL, "AIF2RX"}, + + {"Front", "DAC Switch", "DAC 0"}, + {"Front", "RECMIX Switch", "RECMIX"}, + + {"Surround", NULL, "DAC 1"}, + + {"SPK Mux", "Front", "Front"}, + {"SPK Mux", "Surround", "Surround"}, + + {"HPO Mux", "Front", "Front"}, + {"HPO Mux", "Surround", "Surround"}, + + {"SPO", "Switch", "SPK Mux"}, + {"HPO L", "Switch", "HPO Mux"}, + {"HPO R", "Switch", "HPO Mux"}, + {"HPO L", NULL, "HP Power"}, + {"HPO R", NULL, "HP Power"}, + + {"SPOL", NULL, "SPO"}, + {"SPOR", NULL, "SPO"}, + {"HPO Pin", NULL, "HPO L"}, + {"HPO Pin", NULL, "HPO R"}, +}; + +static int rt298_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 rt298_priv *rt298 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + int d_len_code; + + switch (params_rate(params)) { + /* bit 14 0:48K 1:44.1K */ + case 44100: + case 48000: + break; + default: + dev_err(codec->dev, "Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + switch (rt298->sys_clk) { + case 12288000: + case 24576000: + if (params_rate(params) != 48000) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt298->sys_clk); + return -EINVAL; + } + break; + case 11289600: + case 22579200: + if (params_rate(params) != 44100) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt298->sys_clk); + return -EINVAL; + } + break; + } + + if (params_channels(params) <= 16) { + /* bit 3:0 Number of Channel */ + val |= (params_channels(params) - 1); + } else { + dev_err(codec->dev, "Unsupported channels %d\n", + params_channels(params)); + return -EINVAL; + } + + d_len_code = 0; + switch (params_width(params)) { + /* bit 6:4 Bits per Sample */ + case 16: + d_len_code = 0; + val |= (0x1 << 4); + break; + case 32: + d_len_code = 2; + val |= (0x4 << 4); + break; + case 20: + d_len_code = 1; + val |= (0x2 << 4); + break; + case 24: + d_len_code = 2; + val |= (0x3 << 4); + break; + case 8: + d_len_code = 3; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, + RT298_I2S_CTRL1, 0x0018, d_len_code << 3); + dev_dbg(codec->dev, "format val = 0x%x\n", val); + + snd_soc_update_bits(codec, RT298_DAC_FORMAT, 0x407f, val); + snd_soc_update_bits(codec, RT298_ADC_FORMAT, 0x407f, val); + + return 0; +} + +static int rt298_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + snd_soc_update_bits(codec, + RT298_I2S_CTRL1, 0x800, 0x800); + break; + case SND_SOC_DAIFMT_CBS_CFS: + snd_soc_update_bits(codec, + RT298_I2S_CTRL1, 0x800, 0x0); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + snd_soc_update_bits(codec, + RT298_I2S_CTRL1, 0x300, 0x0); + break; + case SND_SOC_DAIFMT_LEFT_J: + snd_soc_update_bits(codec, + RT298_I2S_CTRL1, 0x300, 0x1 << 8); + break; + case SND_SOC_DAIFMT_DSP_A: + snd_soc_update_bits(codec, + RT298_I2S_CTRL1, 0x300, 0x2 << 8); + break; + case SND_SOC_DAIFMT_DSP_B: + snd_soc_update_bits(codec, + RT298_I2S_CTRL1, 0x300, 0x3 << 8); + break; + default: + return -EINVAL; + } + /* bit 15 Stream Type 0:PCM 1:Non-PCM */ + snd_soc_update_bits(codec, RT298_DAC_FORMAT, 0x8000, 0); + snd_soc_update_bits(codec, RT298_ADC_FORMAT, 0x8000, 0); + + return 0; +} + +static int rt298_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt298_priv *rt298 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s freq=%d\n", __func__, freq); + + if (RT298_SCLK_S_MCLK == clk_id) { + snd_soc_update_bits(codec, + RT298_I2S_CTRL2, 0x0100, 0x0); + snd_soc_update_bits(codec, + RT298_PLL_CTRL1, 0x20, 0x20); + } else { + snd_soc_update_bits(codec, + RT298_I2S_CTRL2, 0x0100, 0x0100); + snd_soc_update_bits(codec, + RT298_PLL_CTRL, 0x4, 0x4); + snd_soc_update_bits(codec, + RT298_PLL_CTRL1, 0x20, 0x0); + } + + switch (freq) { + case 19200000: + if (RT298_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + snd_soc_update_bits(codec, + RT298_I2S_CTRL2, 0x40, 0x40); + break; + case 24000000: + if (RT298_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + snd_soc_update_bits(codec, + RT298_I2S_CTRL2, 0x40, 0x0); + break; + case 12288000: + case 11289600: + snd_soc_update_bits(codec, + RT298_I2S_CTRL2, 0x8, 0x0); + snd_soc_update_bits(codec, + RT298_CLK_DIV, 0xfc1e, 0x0004); + break; + case 24576000: + case 22579200: + snd_soc_update_bits(codec, + RT298_I2S_CTRL2, 0x8, 0x8); + snd_soc_update_bits(codec, + RT298_CLK_DIV, 0xfc1e, 0x5406); + break; + default: + dev_err(codec->dev, "Unsupported system clock\n"); + return -EINVAL; + } + + rt298->sys_clk = freq; + rt298->clk_id = clk_id; + + return 0; +} + +static int rt298_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_codec *codec = dai->codec; + + dev_dbg(codec->dev, "%s ratio=%d\n", __func__, ratio); + if (50 == ratio) + snd_soc_update_bits(codec, + RT298_I2S_CTRL1, 0x1000, 0x1000); + else + snd_soc_update_bits(codec, + RT298_I2S_CTRL1, 0x1000, 0x0); + + + return 0; +} + +static int rt298_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (SND_SOC_BIAS_STANDBY == + snd_soc_codec_get_bias_level(codec)) { + snd_soc_write(codec, + RT298_SET_AUDIO_POWER, AC_PWRST_D0); + snd_soc_update_bits(codec, 0x0d, 0x200, 0x200); + snd_soc_update_bits(codec, 0x52, 0x80, 0x0); + mdelay(20); + snd_soc_update_bits(codec, 0x0d, 0x200, 0x0); + snd_soc_update_bits(codec, 0x52, 0x80, 0x80); + } + break; + + case SND_SOC_BIAS_ON: + mdelay(30); + snd_soc_update_bits(codec, + RT298_CBJ_CTRL1, 0x0400, 0x0400); + + break; + + case SND_SOC_BIAS_STANDBY: + snd_soc_write(codec, + RT298_SET_AUDIO_POWER, AC_PWRST_D3); + snd_soc_update_bits(codec, + RT298_CBJ_CTRL1, 0x0400, 0x0000); + break; + + default: + break; + } + + return 0; +} + +static irqreturn_t rt298_irq(int irq, void *data) +{ + struct rt298_priv *rt298 = data; + bool hp = false; + bool mic = false; + int ret, status = 0; + + ret = rt298_jack_detect(rt298, &hp, &mic); + + /* Clear IRQ */ + regmap_update_bits(rt298->regmap, RT298_IRQ_CTRL, 0x1, 0x1); + + if (ret == 0) { + if (hp == true) + status |= SND_JACK_HEADPHONE; + + if (mic == true) + status |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(rt298->jack, status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + + pm_wakeup_event(&rt298->i2c->dev, 300); + } + + return IRQ_HANDLED; +} + +static int rt298_probe(struct snd_soc_codec *codec) +{ + struct rt298_priv *rt298 = snd_soc_codec_get_drvdata(codec); + + rt298->codec = codec; + + if (rt298->i2c->irq) { + regmap_update_bits(rt298->regmap, + RT298_IRQ_CTRL, 0x2, 0x2); + + INIT_DELAYED_WORK(&rt298->jack_detect_work, + rt298_jack_detect_work); + schedule_delayed_work(&rt298->jack_detect_work, + msecs_to_jiffies(1250)); + } + + return 0; +} + +static int rt298_remove(struct snd_soc_codec *codec) +{ + struct rt298_priv *rt298 = snd_soc_codec_get_drvdata(codec); + + cancel_delayed_work_sync(&rt298->jack_detect_work); + + return 0; +} + +#ifdef CONFIG_PM +static int rt298_suspend(struct snd_soc_codec *codec) +{ + struct rt298_priv *rt298 = snd_soc_codec_get_drvdata(codec); + + rt298->is_hp_in = -1; + regcache_cache_only(rt298->regmap, true); + regcache_mark_dirty(rt298->regmap); + + return 0; +} + +static int rt298_resume(struct snd_soc_codec *codec) +{ + struct rt298_priv *rt298 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt298->regmap, false); + rt298_index_sync(codec); + regcache_sync(rt298->regmap); + + return 0; +} +#else +#define rt298_suspend NULL +#define rt298_resume NULL +#endif + +#define RT298_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define RT298_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static const struct snd_soc_dai_ops rt298_aif_dai_ops = { + .hw_params = rt298_hw_params, + .set_fmt = rt298_set_dai_fmt, + .set_sysclk = rt298_set_dai_sysclk, + .set_bclk_ratio = rt298_set_bclk_ratio, +}; + +static struct snd_soc_dai_driver rt298_dai[] = { + { + .name = "rt298-aif1", + .id = RT298_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT298_STEREO_RATES, + .formats = RT298_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT298_STEREO_RATES, + .formats = RT298_FORMATS, + }, + .ops = &rt298_aif_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "rt298-aif2", + .id = RT298_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT298_STEREO_RATES, + .formats = RT298_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT298_STEREO_RATES, + .formats = RT298_FORMATS, + }, + .ops = &rt298_aif_dai_ops, + .symmetric_rates = 1, + }, + +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt298 = { + .probe = rt298_probe, + .remove = rt298_remove, + .suspend = rt298_suspend, + .resume = rt298_resume, + .set_bias_level = rt298_set_bias_level, + .idle_bias_off = true, + .controls = rt298_snd_controls, + .num_controls = ARRAY_SIZE(rt298_snd_controls), + .dapm_widgets = rt298_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt298_dapm_widgets), + .dapm_routes = rt298_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt298_dapm_routes), +}; + +static const struct regmap_config rt298_regmap = { + .reg_bits = 32, + .val_bits = 32, + .max_register = 0x02370100, + .volatile_reg = rt298_volatile_register, + .readable_reg = rt298_readable_register, + .reg_write = rl6347a_hw_write, + .reg_read = rl6347a_hw_read, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt298_reg, + .num_reg_defaults = ARRAY_SIZE(rt298_reg), +}; + +static const struct i2c_device_id rt298_i2c_id[] = { + {"rt298", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt298_i2c_id); + +static const struct acpi_device_id rt298_acpi_match[] = { + { "INT343A", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt298_acpi_match); + +static int rt298_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt298_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt298_priv *rt298; + struct device *dev = &i2c->dev; + const struct acpi_device_id *acpiid; + int i, ret; + + pr_info("%s\n", __func__); + + rt298 = devm_kzalloc(&i2c->dev, sizeof(*rt298), + GFP_KERNEL); + if (NULL == rt298) + return -ENOMEM; + + rt298->regmap = devm_regmap_init(&i2c->dev, NULL, i2c, &rt298_regmap); + if (IS_ERR(rt298->regmap)) { + ret = PTR_ERR(rt298->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt298->regmap, + RT298_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &ret); + if (ret != RT298_VENDOR_ID) { + dev_err(&i2c->dev, + "Device with ID register %#x is not rt298\n", ret); + return -ENODEV; + } + + rt298->index_cache = rt298_index_def; + rt298->index_cache_size = INDEX_CACHE_SIZE; + rt298->i2c = i2c; + i2c_set_clientdata(i2c, rt298); + + /* restore codec default */ + for (i = 0; i < INDEX_CACHE_SIZE; i++) + regmap_write(rt298->regmap, rt298->index_cache[i].reg, + rt298->index_cache[i].def); + for (i = 0; i < ARRAY_SIZE(rt298_reg); i++) + regmap_write(rt298->regmap, rt298_reg[i].reg, + rt298_reg[i].def); + + if (pdata) + rt298->pdata = *pdata; + + /* enable jack combo mode on supported devices */ + acpiid = acpi_match_device(dev->driver->acpi_match_table, dev); + if (acpiid) { + rt298->pdata = *(struct rt298_platform_data *) + acpiid->driver_data; + } + + /* VREF Charging */ + regmap_update_bits(rt298->regmap, 0x04, 0x80, 0x80); + regmap_update_bits(rt298->regmap, 0x1b, 0x860, 0x860); + /* Vref2 */ + regmap_update_bits(rt298->regmap, 0x08, 0x20, 0x20); + + regmap_write(rt298->regmap, RT298_SET_AUDIO_POWER, AC_PWRST_D3); + + for (i = 0; i < RT298_POWER_REG_LEN; i++) + regmap_write(rt298->regmap, + RT298_SET_POWER(rt298_support_power_controls[i]), + AC_PWRST_D1); + + if (!rt298->pdata.cbj_en) { + regmap_write(rt298->regmap, RT298_CBJ_CTRL2, 0x0000); + regmap_write(rt298->regmap, RT298_MIC1_DET_CTRL, 0x0816); + regmap_update_bits(rt298->regmap, + RT298_CBJ_CTRL1, 0xf000, 0xb000); + } else { + regmap_update_bits(rt298->regmap, + RT298_CBJ_CTRL1, 0xf000, 0x5000); + } + + mdelay(10); + + if (!rt298->pdata.gpio2_en) + regmap_write(rt298->regmap, RT298_SET_DMIC2_DEFAULT, 0x4000); + else + regmap_write(rt298->regmap, RT298_SET_DMIC2_DEFAULT, 0); + + mdelay(10); + + regmap_write(rt298->regmap, RT298_MISC_CTRL1, 0x0000); + regmap_update_bits(rt298->regmap, + RT298_WIND_FILTER_CTRL, 0x0082, 0x0082); + regmap_update_bits(rt298->regmap, RT298_IRQ_CTRL, 0x2, 0x2); + rt298->is_hp_in = -1; + + if (rt298->i2c->irq) { + ret = request_threaded_irq(rt298->i2c->irq, NULL, rt298_irq, + IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt298", rt298); + if (ret != 0) { + dev_err(&i2c->dev, + "Failed to reguest IRQ: %d\n", ret); + return ret; + } + } + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt298, + rt298_dai, ARRAY_SIZE(rt298_dai)); + + return ret; +} + +static int rt298_i2c_remove(struct i2c_client *i2c) +{ + struct rt298_priv *rt298 = i2c_get_clientdata(i2c); + + if (i2c->irq) + free_irq(i2c->irq, rt298); + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + + +static struct i2c_driver rt298_i2c_driver = { + .driver = { + .name = "rt298", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt298_acpi_match), + }, + .probe = rt298_i2c_probe, + .remove = rt298_i2c_remove, + .id_table = rt298_i2c_id, +}; + +module_i2c_driver(rt298_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT298 driver"); +MODULE_AUTHOR("Bard Liao <bardliao(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt298.h b/sound/soc/codecs/rt298.h new file mode 100644 index 0000000..31da162 --- /dev/null +++ b/sound/soc/codecs/rt298.h @@ -0,0 +1,206 @@ +/* + * rt298.h -- RT298 ALSA SoC audio driver + * + * Copyright 2011 Realtek Microelectronics + * Author: Johnny Hsu <johnnyhsu(a)realtek.com> + * + * 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. + */ + +#ifndef __RT298_H__ +#define __RT298_H__ + +#define VERB_CMD(V, N, D) ((N << 20) | (V << 8) | D) + +#define RT298_AUDIO_FUNCTION_GROUP 0x01 +#define RT298_DAC_OUT1 0x02 +#define RT298_DAC_OUT2 0x03 +#define RT298_DIG_CVT 0x06 +#define RT298_ADC_IN1 0x09 +#define RT298_ADC_IN2 0x08 +#define RT298_MIXER_IN 0x0b +#define RT298_MIXER_OUT1 0x0c +#define RT298_MIXER_OUT2 0x0d +#define RT298_DMIC1 0x12 +#define RT298_DMIC2 0x13 +#define RT298_SPK_OUT 0x14 +#define RT298_MIC1 0x18 +#define RT298_LINE1 0x1a +#define RT298_BEEP 0x1d +#define RT298_SPDIF 0x1e +#define RT298_VENDOR_REGISTERS 0x20 +#define RT298_HP_OUT 0x21 +#define RT298_MIXER_IN1 0x22 +#define RT298_MIXER_IN2 0x23 + +#define RT298_SET_PIN_SFT 6 +#define RT298_SET_PIN_ENABLE 0x40 +#define RT298_SET_PIN_DISABLE 0 +#define RT298_SET_EAPD_HIGH 0x2 +#define RT298_SET_EAPD_LOW 0 + +#define RT298_MUTE_SFT 7 + +/* Verb commands */ +#define RT298_GET_PARAM(NID, PARAM) VERB_CMD(AC_VERB_PARAMETERS, NID, PARAM) +#define RT298_SET_POWER(NID) VERB_CMD(AC_VERB_SET_POWER_STATE, NID, 0) +#define RT298_SET_AUDIO_POWER RT298_SET_POWER(RT298_AUDIO_FUNCTION_GROUP) +#define RT298_SET_HPO_POWER RT298_SET_POWER(RT298_HP_OUT) +#define RT298_SET_SPK_POWER RT298_SET_POWER(RT298_SPK_OUT) +#define RT298_SET_DMIC1_POWER RT298_SET_POWER(RT298_DMIC1) +#define RT298_SPK_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT298_SPK_OUT, 0) +#define RT298_HPO_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT298_HP_OUT, 0) +#define RT298_ADC0_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT298_MIXER_IN1, 0) +#define RT298_ADC1_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT298_MIXER_IN2, 0) +#define RT298_SET_MIC1\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT298_MIC1, 0) +#define RT298_SET_PIN_HPO\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT298_HP_OUT, 0) +#define RT298_SET_PIN_SPK\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT298_SPK_OUT, 0) +#define RT298_SET_PIN_DMIC1\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT298_DMIC1, 0) +#define RT298_SET_PIN_SPDIF\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT298_SPDIF, 0) +#define RT298_SET_PIN_DIG_CVT\ + VERB_CMD(AC_VERB_SET_DIGI_CONVERT_1, RT298_DIG_CVT, 0) +#define RT298_SPK_EAPD\ + VERB_CMD(AC_VERB_SET_EAPD_BTLENABLE, RT298_SPK_OUT, 0) +#define RT298_SET_AMP_GAIN_HPO\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_HP_OUT, 0) +#define RT298_SET_AMP_GAIN_ADC_IN1\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_ADC_IN1, 0) +#define RT298_SET_AMP_GAIN_ADC_IN2\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_ADC_IN2, 0) +#define RT298_GET_HP_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT298_HP_OUT, 0) +#define RT298_GET_MIC1_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT298_MIC1, 0) +#define RT298_SET_DMIC2_DEFAULT\ + VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT298_DMIC2, 0) +#define RT298_SET_SPDIF_DEFAULT\ + VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT298_SPDIF, 0) +#define RT298_DACL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_DAC_OUT1, 0xa000) +#define RT298_DACR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_DAC_OUT1, 0x9000) +#define RT298_ADCL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_ADC_IN1, 0x6000) +#define RT298_ADCR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_ADC_IN1, 0x5000) +#define RT298_MIC_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_MIC1, 0x7000) +#define RT298_SPOL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_SPK_OUT, 0xa000) +#define RT298_SPOR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_SPK_OUT, 0x9000) +#define RT298_HPOL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_HP_OUT, 0xa000) +#define RT298_HPOR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_HP_OUT, 0x9000) +#define RT298_F_DAC_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_MIXER_OUT1, 0x7000) +#define RT298_F_RECMIX_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_MIXER_OUT1, 0x7100) +#define RT298_REC_MIC_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_MIXER_IN, 0x7000) +#define RT298_REC_I2S_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_MIXER_IN, 0x7100) +#define RT298_REC_LINE_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_MIXER_IN, 0x7200) +#define RT298_REC_BEEP_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_MIXER_IN, 0x7300) +#define RT298_DAC_FORMAT\ + VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT298_DAC_OUT1, 0) +#define RT298_ADC_FORMAT\ + VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT298_ADC_IN1, 0) +#define RT298_COEF_INDEX\ + VERB_CMD(AC_VERB_SET_COEF_INDEX, RT298_VENDOR_REGISTERS, 0) +#define RT298_PROC_COEF\ + VERB_CMD(AC_VERB_SET_PROC_COEF, RT298_VENDOR_REGISTERS, 0) + +/* Index registers */ +#define RT298_A_BIAS_CTRL1 0x01 +#define RT298_A_BIAS_CTRL2 0x02 +#define RT298_POWER_CTRL1 0x03 +#define RT298_A_BIAS_CTRL3 0x04 +#define RT298_POWER_CTRL2 0x08 +#define RT298_I2S_CTRL1 0x09 +#define RT298_I2S_CTRL2 0x0a +#define RT298_CLK_DIV 0x0b +#define RT298_DC_GAIN 0x0d +#define RT298_POWER_CTRL3 0x0f +#define RT298_MIC1_DET_CTRL 0x19 +#define RT298_MISC_CTRL1 0x20 +#define RT298_IRQ_CTRL 0x33 +#define RT298_WIND_FILTER_CTRL 0x46 +#define RT298_PLL_CTRL1 0x49 +#define RT298_CBJ_CTRL1 0x4f +#define RT298_CBJ_CTRL2 0x50 +#define RT298_PLL_CTRL 0x63 +#define RT298_DEPOP_CTRL1 0x66 +#define RT298_DEPOP_CTRL2 0x67 +#define RT298_DEPOP_CTRL3 0x68 +#define RT298_DEPOP_CTRL4 0x69 + +/* SPDIF (0x06) */ +#define RT298_SPDIF_SEL_SFT 0 +#define RT298_SPDIF_SEL_PCM0 0 +#define RT298_SPDIF_SEL_PCM1 1 +#define RT298_SPDIF_SEL_SPOUT 2 +#define RT298_SPDIF_SEL_PP 3 + +/* RECMIX (0x0b) */ +#define RT298_M_REC_BEEP_SFT 0 +#define RT298_M_REC_LINE1_SFT 1 +#define RT298_M_REC_MIC1_SFT 2 +#define RT298_M_REC_I2S_SFT 3 + +/* Front (0x0c) */ +#define RT298_M_FRONT_DAC_SFT 0 +#define RT298_M_FRONT_REC_SFT 1 + +/* SPK-OUT (0x14) */ +#define RT298_M_SPK_MUX_SFT 14 +#define RT298_SPK_SEL_MASK 0x1 +#define RT298_SPK_SEL_SFT 0 +#define RT298_SPK_SEL_F 0 +#define RT298_SPK_SEL_S 1 + +/* HP-OUT (0x21) */ +#define RT298_M_HP_MUX_SFT 14 +#define RT298_HP_SEL_MASK 0x1 +#define RT298_HP_SEL_SFT 0 +#define RT298_HP_SEL_F 0 +#define RT298_HP_SEL_S 1 + +/* ADC (0x22) (0x23) */ +#define RT298_ADC_SEL_MASK 0x7 +#define RT298_ADC_SEL_SFT 0 +#define RT298_ADC_SEL_SURR 0 +#define RT298_ADC_SEL_FRONT 1 +#define RT298_ADC_SEL_DMIC 2 +#define RT298_ADC_SEL_BEEP 4 +#define RT298_ADC_SEL_LINE1 5 +#define RT298_ADC_SEL_I2S 6 +#define RT298_ADC_SEL_MIC1 7 + +#define RT298_SCLK_S_MCLK 0 +#define RT298_SCLK_S_PLL 1 + +enum { + RT298_AIF1, + RT298_AIF2, + RT298_AIFS, +}; + +int rt298_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); + +#endif /* __RT298_H__ */ + -- 1.8.1.1.439.g50a6b54

2 4

[alsa-devel] [PATCH] terminate readlink result string
by Tobias Stoeckmann 13 Jul '15

13 Jul '15

readlink does not guarantee that its result string is nul-terminated. Instead, increase the buffer by one byte to make sure that we can add '\0' at the end. --- alsactl/init_sysfs.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/alsactl/init_sysfs.c b/alsactl/init_sysfs.c index 0cbada2..5c789b6 100644 --- a/alsactl/init_sysfs.c +++ b/alsactl/init_sysfs.c @@ -108,11 +108,11 @@ static char *sysfs_attr_get_value(const char *devpath, const char *attr_name) if (S_ISLNK(statbuf.st_mode)) { /* links return the last element of the target path */ - char link_target[PATH_SIZE]; + char link_target[PATH_SIZE + 1]; int len; const char *pos; - len = readlink(path_full, link_target, sizeof(link_target)); + len = readlink(path_full, link_target, sizeof(link_target) - 1); if (len > 0) { link_target[len] = '\0'; pos = strrchr(link_target, '/'); -- 2.4.5

2 1

[alsa-devel] [PATCH 1/5] regmap: Add better support for devices without readback support
by Lars-Peter Clausen 13 Jul '15

13 Jul '15

Currently regmap requires that a reg_read callback is supplied, otherwise a warning is emitted each time regmap_read() is called. This means a device or bus without readback support needs to supply dummy reg_read callback. Apart from that regmap_read() will still work fine if a cache is used. Remove the warning and let regmap_readable() return false if not reg_read callback is supplied. This means a device no longer has to supply a dummy callback if it does not support readback and it also doesn't have to have a readable_reg callback that always returns false since this is now implicit. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- drivers/base/regmap/regmap.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c index dd63bcb..8acc837 100644 --- a/drivers/base/regmap/regmap.c +++ b/drivers/base/regmap/regmap.c @@ -93,6 +93,9 @@ bool regmap_writeable(struct regmap *map, unsigned int reg) bool regmap_readable(struct regmap *map, unsigned int reg) { + if (!map->reg_read) + return false; + if (map->max_register && reg > map->max_register) return false; @@ -2109,8 +2112,6 @@ static int _regmap_read(struct regmap *map, unsigned int reg, int ret; void *context = _regmap_map_get_context(map); - WARN_ON(!map->reg_read); - if (!map->cache_bypass) { ret = regcache_read(map, reg, val); if (ret == 0) -- 2.1.4

2 9

[alsa-devel] [PATCH 1/2] ASoC: sti-sas: Fix checking return value for ERR_PTR
by Axel Lin 13 Jul '15

13 Jul '15

Both devm_regmap_init and syscon_regmap_lookup_by_phandle return ERR_PTR on failure. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/sti-sas.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c index 32db2c2..4168b88 100644 --- a/sound/soc/codecs/sti-sas.c +++ b/sound/soc/codecs/sti-sas.c @@ -568,17 +568,17 @@ static int sti_sas_driver_probe(struct platform_device *pdev) /* Request the DAC & SPDIF registers memory region */ drvdata->dac.virt_regmap = devm_regmap_init(&pdev->dev, NULL, drvdata, drvdata->dev_data->regmap); - if (!drvdata->dac.virt_regmap) { + if (IS_ERR(drvdata->dac.virt_regmap)) { dev_err(&pdev->dev, "audio registers not enabled\n"); - return -EFAULT; + return PTR_ERR(drvdata->dac.virt_regmap); } /* Request the syscon region */ drvdata->dac.regmap = syscon_regmap_lookup_by_phandle(pnode, "st,syscfg"); - if (!drvdata->dac.regmap) { + if (IS_ERR(drvdata->dac.regmap)) { dev_err(&pdev->dev, "syscon registers not available\n"); - return -EFAULT; + return PTR_ERR(drvdata->dac.regmap); } drvdata->spdif.regmap = drvdata->dac.regmap; -- 2.1.0

2 3

[alsa-devel] [PATCH] ASoC: sti-sas: fix platform_no_drv_owner.cocci warnings
by kbuild test robot 13 Jul '15

13 Jul '15

sound/soc/codecs/sti-sas.c:616:3-8: No need to set .owner here. The core will do it. Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci CC: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- sti-sas.c | 1 - 1 file changed, 1 deletion(-) --- a/sound/soc/codecs/sti-sas.c +++ b/sound/soc/codecs/sti-sas.c @@ -613,7 +613,6 @@ static int sti_sas_driver_remove(struct static struct platform_driver sti_sas_platform_driver = { .driver = { .name = "sti-sas-codec", - .owner = THIS_MODULE, .of_match_table = sti_sas_dev_match, }, .probe = sti_sas_driver_probe,

2 1