April 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v5] ASoC: cs53l30: Add support for Cirrus Logic CS53L30
by tim.howe＠cirrus.com 06 Jun '16

06 Jun '16

From: Tim Howe <tim.howe(a)cirrus.com> Signed-off-by: Tim Howe <tim.howe(a)cirrus.com> --- .../devicetree/bindings/sound/cs53l30.txt | 21 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs53l30.c | 943 +++++++++++++++++++++ sound/soc/codecs/cs53l30.h | 282 ++++++ 5 files changed, 1254 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs53l30.txt create mode 100644 sound/soc/codecs/cs53l30.c create mode 100644 sound/soc/codecs/cs53l30.h diff --git a/Documentation/devicetree/bindings/sound/cs53l30.txt b/Documentation/devicetree/bindings/sound/cs53l30.txt new file mode 100644 index 0000000..a2dff95 +++ b/Documentation/devicetree/bindings/sound/cs53l30.txt @@ -0,0 +1,21 @@ +CS53L30 audio CODEC + +Required properties: + + - compatible : "cirrus,cs53l30" + + - reg : the I2C address of the device + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. + + +Example: + +codec: cs53l30@48 { + compatible = "cirrus,cs53l30"; + reg = <0x48>; + reset-gpios = <&gpio 54 0>; +}; + diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b282169..2dd97d4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -60,6 +60,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271_SPI if SPI_MASTER select SND_SOC_CS42XX8_I2C if I2C select SND_SOC_CS4349 if I2C + select SND_SOC_CS53L30 if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI select SND_SOC_DA7213 if I2C @@ -449,6 +450,11 @@ config SND_SOC_CS4349 tristate "Cirrus Logic CS4349 CODEC" depends on I2C +# Cirrus Logic Quad-Channel ADC +config SND_SOC_CS53L30 + tristate "Cirrus Logic CS53L30 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 81324bc..88dff59 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -52,6 +52,7 @@ 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-cs53l30-objs := cs53l30.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -252,6 +253,7 @@ 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-$(CONFIG_SND_SOC_CS53L30) += snd-soc-cs53l30.o obj-$(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/cs53l30.c b/sound/soc/codecs/cs53l30.c new file mode 100644 index 0000000..2002ecd +++ b/sound/soc/codecs/cs53l30.c @@ -0,0 +1,943 @@ +/* + * cs53l30.c -- CS53l30 ALSA Soc Audio driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Authors: Paul Handrigan <Paul.Handrigan(a)cirrus.com>, + * 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/gpio/consumer.h> +#include <linux/of_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 "cs53l30.h" + +struct cs53l30_private { + struct regmap *regmap; + struct gpio_desc *reset_gpio; + u8 asp_config_ctl; + u32 mclk; +}; + +static const struct reg_default cs53l30_reg_defaults[] = { + { CS53L30_PWRCTL, CS53L30_THMS_PDN }, + { CS53L30_MCLKCTL, CS53L30_MCLK_DIV_DFLT }, + { CS53L30_INT_SR_CTL, CS53L30_INTRNL_FS_DFLT }, + { CS53L30_MICBIAS_CTL, CS53L30_MIC_BIAS_DFLT }, + { CS53L30_ASPCFG_CTL, CS53L30_ASP_RATE_48K }, + { CS53L30_ASP1_CTL, CS53L30_ASP1_TDM_PDN }, + { CS53L30_ASP1_TDMTX_CTL1, CS53L30_ASP1_CHTX_SLT47 }, + { CS53L30_ASP1_TDMTX_CTL2, CS53L30_ASP1_CHTX_SLT47 }, + { CS53L30_ASP1_TDMTX_CTL3, CS53L30_ASP1_CHTX_SLT47 }, + { CS53L30_ASP1_TDMTX_CTL4, CS53L30_ASP1_CHTX_SLT47 }, + { CS53L30_ASP1_TDMTX_EN1, CS53L30_ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN2, CS53L30_ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN3, CS53L30_ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN4, CS53L30_ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN5, CS53L30_ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN6, CS53L30_ASP_TX_DISABLED }, + { CS53L30_ASP2_CTL, CS53L30_ASP2_CTRL_DFLT }, + { CS53L30_SFT_RAMP, CS53L30_SFT_RMP_DFLT }, + { CS53L30_LRCLK_CTL1, CS53L30_LRCK_CTLX_DFLT }, + { CS53L30_LRCLK_CTL2, CS53L30_LRCK_CTLX_DFLT }, + { CS53L30_MUTEP_CTL1, CS53L30_MUTE_CTRL1_DFLT }, + { CS53L30_MUTEP_CTL2, CS53L30_MUTE_PDN_ULP }, + { CS53L30_INBIAS_CTL1, CS53L30_INBIAS_X_DFLT }, + { CS53L30_INBIAS_CTL2, CS53L30_INBIAS_X_DFLT }, + { CS53L30_DMIC1_STR_CTL, CS53L30_DMIC1_ST_DFLT }, + { CS53L30_DMIC2_STR_CTL, CS53L30_DMIC2_ST_DFLT }, + { CS53L30_ADCDMIC1_CTL1, CS53L30_ADC1_ON_AB_IN }, + { CS53L30_ADCDMIC1_CTL2, CS53L30_A1_D1_CTL2_DFLT }, + { CS53L30_ADC1_CTL3, CS53L30_ADC1_HPF_EN }, + { CS53L30_ADC1_NG_CTL, CS53L30_ADCX_ZERO_DFLT }, + { CS53L30_ADC1A_AFE_CTL, CS53L30_ADCX_ZERO_DFLT }, + { CS53L30_ADC1B_AFE_CTL, CS53L30_ADCX_ZERO_DFLT }, + { CS53L30_ADC1A_DIG_VOL, CS53L30_ADCX_ZERO_DFLT }, + { CS53L30_ADC1B_DIG_VOL, CS53L30_ADCX_ZERO_DFLT }, + { CS53L30_ADCDMIC2_CTL1, CS53L30_ADC2_ON_AB_IN }, + { CS53L30_ADCDMIC2_CTL2, CS53L30_ADCX_ZERO_DFLT }, + { CS53L30_ADC2_CTL3, CS53L30_ADC2_HPF_EN }, + { CS53L30_ADC2_NG_CTL, CS53L30_ADCX_ZERO_DFLT }, + { CS53L30_ADC2A_AFE_CTL, CS53L30_ADCX_ZERO_DFLT }, + { CS53L30_ADC2B_AFE_CTL, CS53L30_ADCX_ZERO_DFLT }, + { CS53L30_ADC2A_DIG_VOL, CS53L30_ADCX_ZERO_DFLT }, + { CS53L30_ADC2B_DIG_VOL, CS53L30_ADCX_ZERO_DFLT }, + { CS53L30_INT_MASK, CS53L30_DEVICE_INT_MASK }, +}; + +static bool cs53l30_volatile_register(struct device *dev, unsigned int reg) +{ + if (reg == CS53L30_IS) + return true; + else + return false; +} + +static bool cs53l30_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS53L30_DEVID_AB: + case CS53L30_DEVID_CD: + case CS53L30_DEVID_E: + case CS53L30_REVID: + case CS53L30_PWRCTL: + case CS53L30_MCLKCTL: + case CS53L30_INT_SR_CTL: + case CS53L30_MICBIAS_CTL: + case CS53L30_ASPCFG_CTL: + case CS53L30_ASP1_CTL: + case CS53L30_ASP1_TDMTX_CTL1: + case CS53L30_ASP1_TDMTX_CTL2: + case CS53L30_ASP1_TDMTX_CTL3: + case CS53L30_ASP1_TDMTX_CTL4: + case CS53L30_ASP1_TDMTX_EN1: + case CS53L30_ASP1_TDMTX_EN2: + case CS53L30_ASP1_TDMTX_EN3: + case CS53L30_ASP1_TDMTX_EN4: + case CS53L30_ASP1_TDMTX_EN5: + case CS53L30_ASP1_TDMTX_EN6: + case CS53L30_ASP2_CTL: + case CS53L30_SFT_RAMP: + case CS53L30_LRCLK_CTL1: + case CS53L30_LRCLK_CTL2: + case CS53L30_MUTEP_CTL1: + case CS53L30_MUTEP_CTL2: + case CS53L30_INBIAS_CTL1: + case CS53L30_INBIAS_CTL2: + case CS53L30_DMIC1_STR_CTL: + case CS53L30_DMIC2_STR_CTL: + case CS53L30_ADCDMIC1_CTL1: + case CS53L30_ADCDMIC1_CTL2: + case CS53L30_ADC1_CTL3: + case CS53L30_ADC1_NG_CTL: + case CS53L30_ADC1A_AFE_CTL: + case CS53L30_ADC1B_AFE_CTL: + case CS53L30_ADC1A_DIG_VOL: + case CS53L30_ADC1B_DIG_VOL: + case CS53L30_ADCDMIC2_CTL1: + case CS53L30_ADCDMIC2_CTL2: + case CS53L30_ADC2_CTL3: + case CS53L30_ADC2_NG_CTL: + case CS53L30_ADC2A_AFE_CTL: + case CS53L30_ADC2B_AFE_CTL: + case CS53L30_ADC2A_DIG_VOL: + case CS53L30_ADC2B_DIG_VOL: + case CS53L30_INT_MASK: + return true; + default: + return false; + } +} + +static DECLARE_TLV_DB_SCALE(adc_boost_tlv, 0, 2000, 0); +static DECLARE_TLV_DB_SCALE(adc_ng_boost_tlv, 0, 3000, 0); +static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); + +static DECLARE_TLV_DB_SCALE(dig_tlv, -9600, 100, 1); + +static const char * const input1_sel_text[] = { "DMIC1 On AB In", + "DMIC1 On A In", "DMIC1 On B In", "ADC1 On AB In", "ADC1 On A In", + "ADC1 On B In", "DMIC1 Off ADC1 Off", }; + +unsigned int const input1_sel_values[] = { CS53L30_DMIC1_ON_AB_IN, + CS53L30_DMIC1_ON_A_IN, CS53L30_DMIC1_ON_B_IN, CS53L30_ADC1_ON_AB_IN, + CS53L30_ADC1_ON_A_IN, CS53L30_ADC1_ON_B_IN, CS53L30_D1_OFF_A1_OFF, }; + +static const char * const input2_sel_text[] = { "DMIC2 On AB In", + "DMIC2 On A In", "DMIC2 On B In", "ADC2 On AB In", "ADC2 On A In", + "ADC2 On B In", "DMIC2 Off ADC2 Off", }; + +unsigned int const input2_sel_values[] = { CS53L30_DMIC2_ON_AB_IN, + CS53L30_DMIC2_ON_A_IN, CS53L30_DMIC2_ON_B_IN, CS53L30_ADC2_ON_AB_IN, + CS53L30_ADC2_ON_A_IN, CS53L30_ADC2_ON_B_IN, CS53L30_D2_OFF_A2_OFF, }; + +static const char * const input1_route_sel_text[] = { "ADC1_SEL", + "DMIC1_SEL" }; + +static const struct soc_enum input1_route_sel_enum = + SOC_ENUM_SINGLE(CS53L30_ADCDMIC1_CTL1, 0, + ARRAY_SIZE(input1_route_sel_text), input1_route_sel_text); + +static SOC_VALUE_ENUM_SINGLE_DECL(input1_sel_enum, CS53L30_ADCDMIC1_CTL1, 0, + CS53L30_A1_D1_PDN_MASK, input1_sel_text, input1_sel_values); + +static const struct snd_kcontrol_new input1_route_sel_mux = + SOC_DAPM_ENUM("Input 1 Route", input1_route_sel_enum); + +static const char * const input2_route_sel_text[] = { "ADC2_SEL", + "DMIC2_SEL" }; + +/* Note: CS53L30_ADCDMIC1_CTL1 CH_TYPE controls inputs 1 and 2 */ +static const struct soc_enum input2_route_sel_enum = + SOC_ENUM_SINGLE(CS53L30_ADCDMIC1_CTL1, 0, + ARRAY_SIZE(input2_route_sel_text), input2_route_sel_text); + +static SOC_VALUE_ENUM_SINGLE_DECL(input2_sel_enum, CS53L30_ADCDMIC2_CTL1, 0, + CS53L30_A1_D1_PDN_MASK, input2_sel_text, input2_sel_values); + +static const struct snd_kcontrol_new input2_route_sel_mux = + SOC_DAPM_ENUM("Input 2 Route", input2_route_sel_enum); + +/* + * TB = 6144*(MCLK(int) scaling factor)/MCLK(internal) + * TB - Time base + * NOTE: If MCLK_INT_SCALE = 0, then TB=1 + */ +static const char * const cs53l30_ng_delay_text[] = { + "TB*50ms", "TB*100ms", "TB*150ms", "TB*200ms" }; + +static const struct soc_enum adc1_ng_delay_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1_NG_CTL, 0, + ARRAY_SIZE(cs53l30_ng_delay_text), cs53l30_ng_delay_text); + +static const struct soc_enum adc2_ng_delay_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2_NG_CTL, 0, + ARRAY_SIZE(cs53l30_ng_delay_text), cs53l30_ng_delay_text); + +/* The noise gate threshold selected will depend on NG Boost */ +static const char * const cs53l30_ng_thres_text[] = { + "-64dB/-34dB", "-66dB/-36dB", "-70dB/-40dB", "-73dB/-43dB", + "-76dB/-46dB", "-82dB/-52dB", "-58dB", "-64dB"}; + +static const struct soc_enum adc1_ng_thres_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1_NG_CTL, 2, + ARRAY_SIZE(cs53l30_ng_thres_text), cs53l30_ng_thres_text); + +static const struct soc_enum adc2_ng_thres_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2_NG_CTL, 2, + ARRAY_SIZE(cs53l30_ng_thres_text), cs53l30_ng_thres_text); + +/* ADC Preamp gain select */ +static const char * const cs53l30_preamp_gain_sel_text[] = { + "0dB", "10dB", "20dB"}; + +static const struct soc_enum adc1a_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1A_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +static const struct soc_enum adc1b_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1B_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +static const struct soc_enum adc2a_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2A_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +static const struct soc_enum adc2b_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2B_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +/* Set MIC Bias Voltage Control */ +static const char * const cs53l30_micbias_text[] = { + "HiZ", "1.8V", "2.75V"}; + +static const struct soc_enum micbias_enum = + SOC_ENUM_SINGLE(CS53L30_MICBIAS_CTL, 0, + ARRAY_SIZE(cs53l30_micbias_text), + cs53l30_micbias_text); + +/* Corner frequencies are with an Fs of 48kHz. */ +static const char * const hpf_corner_freq_text[] = { + "1.86Hz", "120Hz", "235Hz", "466Hz"}; + +static const struct soc_enum adc1_hpf_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1_CTL3, 1, + ARRAY_SIZE(hpf_corner_freq_text), hpf_corner_freq_text); + +static const struct soc_enum adc2_hpf_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2_CTL3, 1, + ARRAY_SIZE(hpf_corner_freq_text), hpf_corner_freq_text); + +static const struct snd_kcontrol_new cs53l30_snd_controls[] = { + SOC_SINGLE("Digital Soft-Ramp Switch", CS53L30_SFT_RAMP, 5, 1, 0), + SOC_SINGLE("ADC1 Noise Gate Ganging Switch", + CS53L30_ADC1_CTL3, 0, 1, 0), + SOC_SINGLE("ADC2 Noise Gate Ganging Switch", + CS53L30_ADC2_CTL3, 0, 1, 0), + SOC_SINGLE("ADC1A Noise Gate Enable Switch", + CS53L30_ADC1_NG_CTL, 6, 1, 0), + SOC_SINGLE("ADC1B Noise Gate Enable Switch", + CS53L30_ADC1_NG_CTL, 7, 1, 0), + SOC_SINGLE("ADC2A Noise Gate Enable Switch", + CS53L30_ADC2_NG_CTL, 6, 1, 0), + SOC_SINGLE("ADC2B Noise Gate Enable Switch", + CS53L30_ADC2_NG_CTL, 7, 1, 0), + SOC_SINGLE("ADC1 Notch Filter Switch", + CS53L30_ADCDMIC1_CTL2, 7, 1, 1), + SOC_SINGLE("ADC2 Notch Filter Switch", + CS53L30_ADCDMIC2_CTL2, 7, 1, 1), + SOC_SINGLE("ADC1A Invert Switch", + CS53L30_ADCDMIC1_CTL2, 4, 1, 0), + SOC_SINGLE("ADC1B Invert Switch", + CS53L30_ADCDMIC1_CTL2, 5, 1, 0), + SOC_SINGLE("ADC2A Invert Switch", + CS53L30_ADCDMIC2_CTL2, 4, 1, 0), + SOC_SINGLE("ADC2B Invert Switch", + CS53L30_ADCDMIC2_CTL2, 5, 1, 0), + + SOC_SINGLE_TLV("ADC1A Digital Boost Volume", + CS53L30_ADCDMIC1_CTL2, 0, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC1B Digital Boost Volume", + CS53L30_ADCDMIC1_CTL2, 1, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC2A Digital Boost Volume", + CS53L30_ADCDMIC2_CTL2, 0, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC2B Digital Boost Volume", + CS53L30_ADCDMIC2_CTL2, 1, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC1 NG Boost Volume", + CS53L30_ADC1_NG_CTL, 5, 1, 0, adc_ng_boost_tlv), + SOC_SINGLE_TLV("ADC2 NG Boost Volume", + CS53L30_ADC2_NG_CTL, 5, 1, 0, adc_ng_boost_tlv), + + SOC_ENUM("Input 1 Channel Select", input1_sel_enum), + SOC_ENUM("Input 2 Channel Select", input2_sel_enum), + + SOC_ENUM("ADC1 HPF Select", adc1_hpf_enum), + SOC_ENUM("ADC2 HPF Select", adc2_hpf_enum), + SOC_ENUM("ADC1 NG Threshold", adc1_ng_thres_enum), + SOC_ENUM("ADC2 NG Threshold", adc2_ng_thres_enum), + SOC_ENUM("ADC1 NG Delay", adc1_ng_delay_enum), + SOC_ENUM("ADC2 NG Delay", adc2_ng_delay_enum), + SOC_ENUM("ADC1A Pre Amp Gain", adc1a_preamp_gain_enum), + SOC_ENUM("ADC1B Pre Amp Gain", adc1b_preamp_gain_enum), + SOC_ENUM("ADC2A Pre Amp Gain", adc2a_preamp_gain_enum), + SOC_ENUM("ADC2B Pre Amp Gain", adc2b_preamp_gain_enum), + SOC_ENUM("Mic Bias Voltage Select", micbias_enum), + + SOC_SINGLE_SX_TLV("ADC1A PGA Volume", + CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + SOC_SINGLE_SX_TLV("ADC1B PGA Volume", + CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + SOC_SINGLE_SX_TLV("ADC2A PGA Volume", + CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + SOC_SINGLE_SX_TLV("ADC2B PGA Volume", + CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + + SOC_SINGLE_SX_TLV("ADC1A Digital Volume", + CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + SOC_SINGLE_SX_TLV("ADC1B Digital Volume", + CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + SOC_SINGLE_SX_TLV("ADC2A Digital Volume", + CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + SOC_SINGLE_SX_TLV("ADC2B Digital Volume", + CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), +}; + +static int cs53l30_asp_sdout_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 cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(priv->regmap, CS53L30_ASP1_CTL, + CS53L30_ASP1_3ST, 0); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(priv->regmap, CS53L30_ASP1_CTL, + CS53L30_ASP1_3ST, 1); + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dapm_widget cs53l30_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("IN1_DMIC1"), + SND_SOC_DAPM_INPUT("IN2"), + SND_SOC_DAPM_INPUT("IN3_DMIC2"), + SND_SOC_DAPM_INPUT("IN4"), + SND_SOC_DAPM_SUPPLY("MIC1 Bias", CS53L30_MICBIAS_CTL, 4, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC2 Bias", CS53L30_MICBIAS_CTL, 5, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC3 Bias", CS53L30_MICBIAS_CTL, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC4 Bias", CS53L30_MICBIAS_CTL, 7, 1, NULL, 0), + + SND_SOC_DAPM_AIF_OUT_E("ASP_SDOUT1", NULL, 0, CS53L30_ASP1_CTL, + CS53L30_ASP1_SDOUT_PDN, 1, + cs53l30_asp_sdout_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)), + SND_SOC_DAPM_AIF_OUT_E("ASP_SDOUT2", NULL, 0, CS53L30_ASP2_CTL, + CS53L30_ASP2_SDOUT_PDN, 1, + cs53l30_asp_sdout_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)), + + SND_SOC_DAPM_MUX("Input Mux 1", SND_SOC_NOPM, 0, 0, + &input1_route_sel_mux), + SND_SOC_DAPM_MUX("Input Mux 2", SND_SOC_NOPM, 0, 0, + &input2_route_sel_mux), + + SND_SOC_DAPM_ADC("ADC1A", NULL, CS53L30_ADCDMIC1_CTL1, 6, 1), + SND_SOC_DAPM_ADC("ADC1B", NULL, CS53L30_ADCDMIC1_CTL1, 7, 1), + SND_SOC_DAPM_ADC("ADC2A", NULL, CS53L30_ADCDMIC2_CTL1, 6, 1), + SND_SOC_DAPM_ADC("ADC2B", NULL, CS53L30_ADCDMIC2_CTL1, 7, 1), + SND_SOC_DAPM_ADC("DMIC1", NULL, CS53L30_ADCDMIC1_CTL1, 2, 1), + SND_SOC_DAPM_ADC("DMIC2", NULL, CS53L30_ADCDMIC2_CTL1, 2, 1), +}; + +static const struct snd_soc_dapm_route cs53l30_audio_map[] = { + + /* ADC Input Paths */ + {"ADC1A", NULL, "IN1_DMIC1"}, + {"Input Mux 1", "ADC1_SEL", "ADC1A"}, + {"ADC1B", NULL, "IN2"}, + + {"ADC2A", NULL, "IN3_DMIC2"}, + {"Input Mux 2", "ADC2_SEL", "ADC2A"}, + {"ADC2B", NULL, "IN4"}, + + /* MIC Bias Paths */ + {"ADC1A", NULL, "MIC1 Bias"}, + {"ADC1B", NULL, "MIC2 Bias"}, + {"ADC2A", NULL, "MIC3 Bias"}, + {"ADC2B", NULL, "MIC4 Bias"}, + + /* DMIC Paths */ + {"DMIC1", NULL, "IN1_DMIC1"}, + {"Input Mux 1", "DMIC1_SEL", "DMIC1"}, + + {"DMIC2", NULL, "IN3_DMIC2"}, + {"Input Mux 2", "DMIC2_SEL", "DMIC2"}, + + /* Output Paths */ + {"ASP_SDOUT1", NULL, "ADC1A" }, + {"ASP_SDOUT1", NULL, "Input Mux 1"}, + {"ASP_SDOUT1", NULL, "ADC1B"}, + + {"ASP_SDOUT2", NULL, "ADC2A"}, + {"ASP_SDOUT2", NULL, "Input Mux 2"}, + {"ASP_SDOUT2", NULL, "ADC2B"}, + + {"ASP1 Capture", NULL, "ASP_SDOUT1"}, + {"ASP2 Capture", NULL, "ASP_SDOUT2"}, +}; + +struct cs53l30_mclk_div { + u32 mclk; + u32 srate; + u8 asp_rate; + u8 internal_fs_ratio; + u8 mclk_int_scale; +}; + +static struct cs53l30_mclk_div cs53l30_mclk_coeffs[] = { + /* NOTE: Enable MCLK_INT_SCALE to save power. */ + + /* MCLK, Sample Rate, asp_rate, internal_fs_ratio, mclk_int_scale */ + {5644800, 11025, 0x4, 1, 1}, + {5644800, 22050, 0x8, 1, 1}, + {5644800, 44100, 0xC, 1, 1}, + + {6000000, 8000, 0x1, 0, 1}, + {6000000, 11025, 0x2, 0, 1}, + {6000000, 12000, 0x4, 0, 1}, + {6000000, 16000, 0x5, 0, 1}, + {6000000, 22050, 0x6, 0, 1}, + {6000000, 24000, 0x8, 0, 1}, + {6000000, 32000, 0x9, 0, 1}, + {6000000, 44100, 0xA, 0, 1}, + {6000000, 48000, 0xC, 0, 1}, + + {6144000, 8000, 0x1, 1, 1}, + {6144000, 11025, 0x2, 1, 1}, + {6144000, 12000, 0x4, 1, 1}, + {6144000, 16000, 0x5, 1, 1}, + {6144000, 22050, 0x6, 1, 1}, + {6144000, 24000, 0x8, 1, 1}, + {6144000, 32000, 0x9, 1, 1}, + {6144000, 44100, 0xA, 1, 1}, + {6144000, 48000, 0xC, 1, 1}, + + {6400000, 8000, 0x1, 1, 1}, + {6400000, 11025, 0x2, 1, 1}, + {6400000, 12000, 0x4, 1, 1}, + {6400000, 16000, 0x5, 1, 1}, + {6400000, 22050, 0x6, 1, 1}, + {6400000, 24000, 0x8, 1, 1}, + {6400000, 32000, 0x9, 1, 1}, + {6400000, 44100, 0xA, 1, 1}, + {6400000, 48000, 0xC, 1, 1}, +}; + +struct cs53l30_mclkx_div { + u32 mclkx; + u8 ratio; + u8 mclkdiv; +}; + +static struct cs53l30_mclkx_div cs53l30_mclkx_coeffs[] = { + {5644800, 1, 0}, + {6000000, 1, 0}, + {6144000, 1, 0}, + {11289600, 2, 1}, + {12288000, 2, 1}, + {12000000, 2, 1}, + {19200000, 3, 2}, +}; + +static int cs53l30_get_mclkx_coeff(int mclkx) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs53l30_mclkx_coeffs); i++) { + if (cs53l30_mclkx_coeffs[i].mclkx == mclkx) + return i; + } + return -EINVAL; +} + +static int cs53l30_get_mclk_coeff(int mclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs53l30_mclk_coeffs); i++) { + if (cs53l30_mclk_coeffs[i].mclk == mclk && + cs53l30_mclk_coeffs[i].srate == srate) + return i; + } + return -EINVAL; + +} + +static int cs53l30_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + + int mclkx_coeff; + u32 mclk; + unsigned int mclk_ctl; + + /* MCLKX -> MCLK */ + mclkx_coeff = cs53l30_get_mclkx_coeff(freq); + if (mclkx_coeff < 0) + return mclkx_coeff; + + mclk = cs53l30_mclkx_coeffs[mclkx_coeff].mclkx / + cs53l30_mclkx_coeffs[mclkx_coeff].ratio; + + regmap_read(priv->regmap, CS53L30_MCLKCTL, &mclk_ctl); + mclk_ctl &= ~CS53L30_MCLK_DIV; + mclk_ctl |= cs53l30_mclkx_coeffs[mclkx_coeff].mclkdiv; + + regmap_update_bits(priv->regmap, CS53L30_MCLKCTL, CS53L30_MCLK_DIV, + mclk_ctl << CS53L30_MCLK_DIV); + priv->mclk = mclk; + return 0; +} + +static int cs53l30_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int asp_config_ctl; + + regmap_read(priv->regmap, CS53L30_ASPCFG_CTL, &asp_config_ctl); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + asp_config_ctl |= CS53L30_ASP_MS; + break; + + case SND_SOC_DAIFMT_CBS_CFS: + asp_config_ctl &= ~CS53L30_ASP_MS; + break; + + default: + return -EINVAL; + } + + /* Check to see if the SCLK is inverted */ + if (fmt & (SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_IB_IF)) + asp_config_ctl |= CS53L30_ASP_SCLK_INV; + else + asp_config_ctl &= ~CS53L30_ASP_SCLK_INV; + + priv->asp_config_ctl = asp_config_ctl; + return 0; +} + +static int cs53l30_pcm_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 cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + int mclk_coeff; + int srate = params_rate(params); + unsigned int int_sr_ctl, mclk_ctl; + + /* MCLK -> srate */ + mclk_coeff = cs53l30_get_mclk_coeff(priv->mclk, srate); + if (mclk_coeff < 0) + return -EINVAL; + + regmap_read(priv->regmap, CS53L30_INT_SR_CTL, &int_sr_ctl); + if (cs53l30_mclk_coeffs[mclk_coeff].internal_fs_ratio) + int_sr_ctl |= CS53L30_INTRNL_FS_RATIO; + else + int_sr_ctl &= ~CS53L30_INTRNL_FS_RATIO; + regmap_update_bits(priv->regmap, CS53L30_INT_SR_CTL, + CS53L30_INTL_FS_RAT_MSK, int_sr_ctl << + CS53L30_INTL_FS_RAT_SFT); + + regmap_read(priv->regmap, CS53L30_MCLKCTL, &mclk_ctl); + if (cs53l30_mclk_coeffs[mclk_coeff].mclk_int_scale) + mclk_ctl |= CS53L30_MCLK_INT_SCALE; + else + mclk_ctl &= ~CS53L30_MCLK_INT_SCALE; + regmap_update_bits(priv->regmap, CS53L30_MCLKCTL, + CS53L30_MCK_INT_SCL_MSK, mclk_ctl << + CS53L30_MCK_INT_SCL_SFT); + + priv->asp_config_ctl &= CS53L30_ASP_CNFG_MASK; + priv->asp_config_ctl |= (cs53l30_mclk_coeffs[mclk_coeff].asp_rate + & CS53L30_ASP_RATE_MASK); + regmap_update_bits(priv->regmap, CS53L30_ASPCFG_CTL, + CS53L30_ASP_RATE_MASK, priv->asp_config_ctl); + + return 0; +} + +static int cs53l30_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int reg; + int i, inter_max_check; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + CS53L30_PDN_LP, 0); + break; + case SND_SOC_BIAS_STANDBY: + if (dapm->bias_level == SND_SOC_BIAS_OFF) { + regmap_update_bits(priv->regmap, CS53L30_MCLKCTL, + CS53L30_MCLK_DIS, 0); + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + CS53L30_PDN_ULP, 0); + msleep(50); + } else { + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + CS53L30_PDN_LP, CS53L30_PDN_LP); + } + break; + + case SND_SOC_BIAS_OFF: + regmap_update_bits(priv->regmap, CS53L30_INT_MASK, + CS53L30_PDN_DONE, 0); + /* If digital softramp is set, the amount of time required + * for power down increases and depends on the digital + * volume setting. + */ + + /* Set the max possible time if digsft is set */ + regmap_read(priv->regmap, CS53L30_SFT_RAMP, &reg); + if (reg & CS53L30_DIGSFT) + inter_max_check = CS53L30_PDN_POLL_MAX; + else + inter_max_check = 10; + + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + CS53L30_PDN_ULP, CS53L30_PDN_ULP); + msleep(20); /* PDN_DONE will take a min of 20ms to be set.*/ + regmap_read(priv->regmap, CS53L30_IS, &reg); /* Clr status */ + for (i = 0; i < inter_max_check; i++) { + if (inter_max_check < 10) { + usleep_range(1000, 1100); + regmap_read(priv->regmap, CS53L30_IS, &reg); + if (reg & CS53L30_PDN_DONE) + break; + } else { + usleep_range(10000, 10100); + regmap_read(priv->regmap, CS53L30_IS, &reg); + if (reg & CS53L30_PDN_DONE) + break; + } + } + /* PDN_DONE is set. We now can disable the MCLK */ + regmap_update_bits(priv->regmap, CS53L30_INT_MASK, + CS53L30_PDN_DONE, CS53L30_PDN_DONE); + regmap_update_bits(priv->regmap, CS53L30_MCLKCTL, + CS53L30_MCLK_DIS, CS53L30_MCLK_DIS); + break; + } + + return 0; +} + +static int cs53l30_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + + return regmap_update_bits(priv->regmap, CS53L30_ASP1_CTL, + CS53L30_ASP1_3ST, + (CS53L30_ASP1_3ST_VAL(tristate) & + CS53L30_ASP1_3ST)); +} + +unsigned int const cs53l30_src_rates[] = { + 8000, 11025, 12000, 16000, 22050, + 24000, 32000, 44100, 48000 +}; + +static struct snd_pcm_hw_constraint_list src_constraints = { + .count = ARRAY_SIZE(cs53l30_src_rates), + .list = cs53l30_src_rates, +}; + +static int cs53l30_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &src_constraints); + + return 0; +} + +/* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */ +#define CS53L30_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) + +#define CS53L30_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops cs53l30_ops = { + .startup = cs53l30_pcm_startup, + .hw_params = cs53l30_pcm_hw_params, + .set_fmt = cs53l30_set_dai_fmt, + .set_sysclk = cs53l30_set_sysclk, + .set_tristate = cs53l30_set_tristate, +}; + +static struct snd_soc_dai_driver cs53l30_dai[] = { + { + .name = "cs53l30-asp1", + .id = CS53L30_ASP1, + .capture = { + .stream_name = "ASP1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS53L30_RATES, + .formats = CS53L30_FORMATS, + }, + .ops = &cs53l30_ops, + .symmetric_rates = 1, + }, + { + .name = "cs53l30-asp2", + .id = CS53L30_ASP2, + .capture = { + .stream_name = "ASP2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS53L30_RATES, + .formats = CS53L30_FORMATS, + }, + .ops = &cs53l30_ops, + .symmetric_rates = 1, + } +}; + +static struct snd_soc_codec_driver soc_codec_dev_cs53l30 = { + .set_bias_level = cs53l30_set_bias_level, + + .dapm_widgets = cs53l30_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs53l30_dapm_widgets), + .dapm_routes = cs53l30_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs53l30_audio_map), + + .controls = cs53l30_snd_controls, + .num_controls = ARRAY_SIZE(cs53l30_snd_controls), +}; + +static struct regmap_config cs53l30_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS53L30_MAX_REGISTER, + .reg_defaults = cs53l30_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs53l30_reg_defaults), + .volatile_reg = cs53l30_volatile_register, + .readable_reg = cs53l30_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs53l30_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct cs53l30_private *cs53l30; + int ret = 0; + unsigned int devid = 0; + unsigned int reg; + + cs53l30 = devm_kzalloc(&client->dev, + sizeof(struct cs53l30_private), GFP_KERNEL); + if (!cs53l30) { + dev_err(&client->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + /* Reset the Device */ + cs53l30->reset_gpio = devm_gpiod_get_optional(&client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs53l30->reset_gpio)) + return PTR_ERR(cs53l30->reset_gpio); + + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 1); + + i2c_set_clientdata(client, cs53l30); + + cs53l30->mclk = 0; + + cs53l30->regmap = devm_regmap_init_i2c(client, &cs53l30_regmap); + if (IS_ERR(cs53l30->regmap)) { + ret = PTR_ERR(cs53l30->regmap); + dev_err(&client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + /* initialize codec */ + ret = regmap_read(cs53l30->regmap, CS53L30_DEVID_AB, &reg); + devid = reg << 12; + + ret = regmap_read(cs53l30->regmap, CS53L30_DEVID_CD, &reg); + devid |= reg << 4; + + ret = regmap_read(cs53l30->regmap, CS53L30_DEVID_E, &reg); + devid |= (reg & 0xF0) >> 4; + + if (devid != CS53L30_DEVID) { + ret = -ENODEV; + dev_err(&client->dev, + "CS53L30 Device ID (%X). Expected %X\n", + devid, CS53L30_DEVID); + return ret; + } + + ret = regmap_read(cs53l30->regmap, CS53L30_REVID, &reg); + if (ret < 0) { + dev_err(&client->dev, "Get Revision ID failed\n"); + return ret; + } + + dev_info(&client->dev, + "Cirrus Logic CS53L30, Revision: %02X\n", reg & 0xFF); + + ret = snd_soc_register_codec(&client->dev, + &soc_codec_dev_cs53l30, cs53l30_dai, + ARRAY_SIZE(cs53l30_dai)); + return ret; +} + +static int cs53l30_i2c_remove(struct i2c_client *client) +{ + struct cs53l30_private *cs53l30 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + /* Hold down reset */ + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 0); + + return 0; +} + +#ifdef CONFIG_PM +static int cs53l30_runtime_suspend(struct device *dev) +{ + struct cs53l30_private *cs53l30 = dev_get_drvdata(dev); + + regcache_cache_only(cs53l30->regmap, true); + + /* Hold down reset */ + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 0); + + return 0; +} + +static int cs53l30_runtime_resume(struct device *dev) +{ + struct cs53l30_private *cs53l30 = dev_get_drvdata(dev); + + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 1); + + regcache_cache_only(cs53l30->regmap, false); + regcache_sync(cs53l30->regmap); + + return 0; +} +#endif + +static const struct dev_pm_ops cs53l30_runtime_pm = { + SET_RUNTIME_PM_OPS(cs53l30_runtime_suspend, cs53l30_runtime_resume, + NULL) +}; + +static const struct of_device_id cs53l30_of_match[] = { + { .compatible = "cirrus,cs53l30", }, + {}, +}; + +MODULE_DEVICE_TABLE(of, cs53l30_of_match); + +static const struct i2c_device_id cs53l30_id[] = { + {"cs53l30", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs53l30_id); + +static struct i2c_driver cs53l30_i2c_driver = { + .driver = { + .name = "cs53l30", + }, + .id_table = cs53l30_id, + .probe = cs53l30_i2c_probe, + .remove = cs53l30_i2c_remove, + +}; + +module_i2c_driver(cs53l30_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS53L30 driver"); +MODULE_AUTHOR("Paul Handrigan, Cirrus Logic Inc, <Paul.Handrigan(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs53l30.h b/sound/soc/codecs/cs53l30.h new file mode 100644 index 0000000..116e358 +++ b/sound/soc/codecs/cs53l30.h @@ -0,0 +1,282 @@ +/* + * ALSA SoC CS53L30 codec driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <Paul.Handrigan(a)cirrus.com>, + * 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. + * + */ + +#ifndef __CS53L30_H__ +#define __CS53L30_H__ + +/* I2C Registers */ +#define CS53L30_DEVID_AB 0x01 /* Device ID A & B [RO]. */ +#define CS53L30_DEVID_CD 0x02 /* Device ID C & D [RO]. */ +#define CS53L30_DEVID_E 0x03 /* Device ID E [RO]. */ +#define CS53L30_REVID 0x05 /* Revision ID [RO]. */ +#define CS53L30_PWRCTL 0x06 /* Power Control. */ +#define CS53L30_MCLKCTL 0x07 /* MCLK Control. */ +#define CS53L30_INT_SR_CTL 0x08 /* Internal Sample Rate Control. */ +#define CS53L30_MICBIAS_CTL 0x0A /* Mic Bias Control. */ +#define CS53L30_ASPCFG_CTL 0x0C /* ASP Config Control. */ +#define CS53L30_ASP1_CTL 0x0D /* ASP1 Control. */ +#define CS53L30_ASP1_TDMTX_CTL1 0x0E /* ASP1 TDM TX Control 1 */ +#define CS53L30_ASP1_TDMTX_CTL2 0x0F /* ASP1 TDM TX Control 2 */ +#define CS53L30_ASP1_TDMTX_CTL3 0x10 /* ASP1 TDM TX Control 3 */ +#define CS53L30_ASP1_TDMTX_CTL4 0x11 /* ASP1 TDM TX Control 4 */ +#define CS53L30_ASP1_TDMTX_EN1 0x12 /* ASP1 TDM TX Enable 1 */ +#define CS53L30_ASP1_TDMTX_EN2 0x13 /* ASP1 TDM TX Enable 2 */ +#define CS53L30_ASP1_TDMTX_EN3 0x14 /* ASP1 TDM TX Enable 3 */ +#define CS53L30_ASP1_TDMTX_EN4 0x15 /* ASP1 TDM TX Enable 4 */ +#define CS53L30_ASP1_TDMTX_EN5 0x16 /* ASP1 TDM TX Enable 5 */ +#define CS53L30_ASP1_TDMTX_EN6 0x17 /* ASP1 TDM TX Enable 6 */ +#define CS53L30_ASP2_CTL 0x18 /* ASP2 Control. */ +#define CS53L30_SFT_RAMP 0x1A /* Soft Ramp Control. */ +#define CS53L30_LRCLK_CTL1 0x1B /* LRCLK Control 1. */ +#define CS53L30_LRCLK_CTL2 0x1C /* LRCLK Control 2. */ +#define CS53L30_MUTEP_CTL1 0x1F /* Mute Pin Control 1. */ +#define CS53L30_MUTEP_CTL2 0x20 /* Mute Pin Control 2. */ +#define CS53L30_INBIAS_CTL1 0x21 /* Input Bias Control 1. */ +#define CS53L30_INBIAS_CTL2 0x22 /* Input Bias Control 2. */ +#define CS53L30_DMIC1_STR_CTL 0x23 /* DMIC1 Stereo Control. */ +#define CS53L30_DMIC2_STR_CTL 0x24 /* DMIC2 Stereo Control. */ +#define CS53L30_ADCDMIC1_CTL1 0x25 /* ADC1/DMIC1 Control 1. */ +#define CS53L30_ADCDMIC1_CTL2 0x26 /* ADC1/DMIC1 Control 2. */ +#define CS53L30_ADC1_CTL3 0x27 /* ADC1 Control 3. */ +#define CS53L30_ADC1_NG_CTL 0x28 /* ADC1 Noise Gate Control. */ +#define CS53L30_ADC1A_AFE_CTL 0x29 /* ADC1A AFE Control. */ +#define CS53L30_ADC1B_AFE_CTL 0x2A /* ADC1B AFE Control. */ +#define CS53L30_ADC1A_DIG_VOL 0x2B /* ADC1A Digital Volume. */ +#define CS53L30_ADC1B_DIG_VOL 0x2C /* ADC1B Digital Volume. */ +#define CS53L30_ADCDMIC2_CTL1 0x2D /* ADC2/DMIC2 Control 1. */ +#define CS53L30_ADCDMIC2_CTL2 0x2E /* ADC2/DMIC2 Control 2. */ +#define CS53L30_ADC2_CTL3 0x2F /* ADC2 Control 3. */ +#define CS53L30_ADC2_NG_CTL 0x30 /* ADC2 Noise Gate Control. */ +#define CS53L30_ADC2A_AFE_CTL 0x31 /* ADC2A AFE Control. */ +#define CS53L30_ADC2B_AFE_CTL 0x32 /* ADC2B AFE Control. */ +#define CS53L30_ADC2A_DIG_VOL 0x33 /* ADC2A Digital Volume. */ +#define CS53L30_ADC2B_DIG_VOL 0x34 /* ADC2B Digital Volume. */ +#define CS53L30_INT_MASK 0x35 /* Interrupt Mask. */ +#define CS53L30_IS 0x36 /* Interrupt Status. */ +#define CS53L30_MAX_REGISTER 0x36 + +/* Device ID */ +#define CS53L30_DEVID 0x53A30 + +/* PDN_DONE Poll Maximum + * If soft ramp is set it will take much longer to power down + * the system. + */ +#define CS53L30_PDN_POLL_MAX 90 + +/* Bitfield Definitions */ + +/* CS53L30_PWRCTL */ +#define CS53L30_PDN_ULP (1 << 7) +#define CS53L30_PDN_LP (1 << 6) +#define CS53L30_DISCHARGE_FILT (1 << 5) +#define CS53L30_THMS_PDN (1 << 4) + +/* CS53L30_MCLKCTL */ +#define CS53L30_MCLK_DIS (1 << 7) +#define CS53L30_MCLK_INT_SCALE (1 << 6) +#define CS53L30_DMIC_DRIVE (1 << 5) +#define CS53L30_MCLK_DIV (3 << 2) +#define CS53L30_MCLK_DIV_DFLT (1 << 2) +#define CS53L30_SYNC_EN (1 << 1) +#define CS53L30_MCK_INT_SCL_MSK 0x40 +#define CS53L30_MCK_INT_SCL_SFT 6 + +/* CS53L30_INT_SR_CTL */ +#define CS53L30_INTRNL_FS_RATIO (1 << 4) +#define CS53L30_INTRNL_FS_DFLT (7 << 2) +#define CS53L30_MCLK_19MHZ_EN (1 << 0) +#define CS53L30_INTL_FS_RAT_MSK 0x10 +#define CS53L30_INTL_FS_RAT_SFT 4 + +/* CS53L30_MICBIAS_CTL */ +#define CS53L30_MIC4_BIAS_PDN (1 << 7) +#define CS53L30_MIC3_BIAS_PDN (1 << 6) +#define CS53L30_MIC2_BIAS_PDN (1 << 5) +#define CS53L30_MIC1_BIAS_PDN (1 << 4) +#define CS53L30_VP_MIN (1 << 2) +#define CS53L30_MIC_BIAS_CTRL (3 << 0) +#define CS53L30_BIAS_ALL_PDN 0xF0 +#define CS53L30_MIC_BIAS_DFLT (CS53L30_BIAS_ALL_PDN | CS53L30_VP_MIN) + +/* CS53L30_ASPCFG_CTL */ +#define CS53L30_ASP_MS (1 << 7) +#define CS53L30_ASP_SCLK_INV (1 << 4) +#define CS53L30_ASP_RATE_48K (3 << 2) +#define CS53L30_ASP_RATE_MASK 0x0F +#define CS53L30_ASP_CNFG_MASK 0xF0 + +/* CS53L30_ASP1_CTL */ +#define CS53L30_ASP1_TDM_PDN (1 << 7) +#define CS53L30_ASP1_SDOUT_PDN (1 << 6) +#define CS53L30_ASP1_3ST (1 << 5) +#define CS53L30_SHIFT_LEFT (1 << 4) +#define CS53L30_ASP1_DRIVE (1 << 0) +#define CS53L30_ASP1_3ST_VAL(x) ((x) << 5) + +/* CS53L30_ASP1_TDMTX_CTL */ +#define CS53L30_ASP1_CHX_TX_ST (1 << 7) +#define CS53L30_ASP1_CHX_TX_LOC 0x3F +#define CS53L30_ASP1_CHTX_SLT47 0x2F +#define CS53L30_ASP_TX_DISABLED 0x00 + +/* CS53L30_ASP2_CTL */ +#define CS53L30_ASP2_SDOUT_PDN (1 << 6) +#define CS53L30_ASP2_DRIVE (1 << 0) +#define CS53L30_ASP2_CTRL_DFLT 0x00 + +/* CS53L30_SFT_RAMP */ +#define CS53L30_DIGSFT (1 << 5) +#define CS53L30_SFT_RMP_DFLT 0x00 + +/* CS53L30_LRCLK_CTL2 */ +#define CS53L30_LRCK_50_NPW (1 << 3) +#define CS53L30_LRCK_TPWH (7 << 0) +#define CS53L30_LRCK_CTLX_DFLT 0x00 + +/* CS53L30_MUTEP_CTL */ +#define CS53L30_MUTE_PDN_ULP (1 << 7) +#define CS53L30_MUTE_PDN_LP (1 << 6) +#define CS53L30_MUTE_M4B_PDN (1 << 4) +#define CS53L30_MUTE_M3B_PDN (1 << 3) +#define CS53L30_MUTE_M2B_PDN (1 << 2) +#define CS53L30_MUTE_M1B_PDN (1 << 1) +#define CS53L30_MUTE_MB_ALL_PDN (1 << 0) +#define CS53L30_MUTE_CTRL1_DFLT 0x00 + +/* CS53L30_MUTEP_CTL2 */ +#define CS53L30_MUTE_PIN_PLRTY (1 << 7) +#define CS53L30_MUTE_ASPTDM_PDN (1 << 6) +#define CS53L30_MTE_ASPSDO2_PDN (1 << 5) +#define CS53L30_MTE_ASPSDO1_PDN (1 << 4) +#define CS53L30_MUTE_ADC2B_PDN (1 << 3) +#define CS53L30_MUTE_ADC2A_PDN (1 << 2) +#define CS53L30_MUTE_ADC1B_PDN (1 << 1) +#define CS53L30_MUTE_ADC1A_PDN (1 << 0) + +/* CS53L30_INBIAS_CTL1 */ +#define CS53L30_IN4M_BIAS (3 << 6) +#define CS53L30_IN4P_BIAS (3 << 4) +#define CS53L30_IN3M_BIAS (3 << 2) +#define CS53L30_IN3P_BIAS (3 << 0) + +/* CS53L30_INBIAS_CTL2 */ +#define CS53L30_IN2M_BIAS (3 << 6) +#define CS53L30_IN2P_BIAS (3 << 4) +#define CS53L30_IN1M_BIAS (3 << 2) +#define CS53L30_IN1P_BIAS (3 << 0) +#define CS53L30_INBIAS_X_DFLT 0xAA + +/* CS53L30_DMIC1_STR_CTL */ +#define CS53L30_DMIC1_STERO_EN (1 << 5) +#define CS53L30_DMIC1_ST_DFLT 0xA8 + +/* CS53L30_DMIC2_STR_CTL */ +#define CS53L30_DMIC2_STERO_EN (1 << 5) +#define CS53L30_DMIC2_ST_DFLT 0xEC + +/* CS53L30_ADCDMIC1_CTL1 */ +#define CS53L30_ADC1B_PDN (1 << 7) +#define CS53L30_ADC1A_PDN (1 << 6) +#define CS53L30_DMIC1_PDN (1 << 2) +#define CS53L30_DMIC1_SCLK_DIV (1 << 1) +#define CS53L30_CH_TYPE (1 << 0) +#define CS53L30_DMIC1_ON_AB_IN (CS53L30_CH_TYPE) +#define CS53L30_DMIC1_ON_A_IN (CS53L30_ADC1B_PDN | CS53L30_CH_TYPE) +#define CS53L30_DMIC1_ON_B_IN (CS53L30_ADC1A_PDN | CS53L30_CH_TYPE) +#define CS53L30_ADC1_ON_AB_IN (CS53L30_DMIC1_PDN) +#define CS53L30_ADC1_ON_A_IN (CS53L30_ADC1B_PDN | CS53L30_DMIC1_PDN) +#define CS53L30_ADC1_ON_B_IN (CS53L30_ADC1A_PDN | CS53L30_DMIC1_PDN) +#define CS53L30_D1_OFF_A1_OFF (CS53L30_ADC1A_PDN | \ + CS53L30_ADC1B_PDN | \ + CS53L30_DMIC1_PDN) +#define CS53L30_A1_D1_PDN_MASK 0xFF + +/* CS53L30_ADCDMIC1_CTL2 */ +#define CS53L30_ADC1_NOTCH_DIS (1 << 7) +#define CS53L30_ADC1B_INV (1 << 5) +#define CS53L30_ADC1A_INV (1 << 4) +#define CS53L30_ADC1B_DIG_BOOST (1 << 1) +#define CS53L30_ADC1A_DIG_BOOST (1 << 0) +#define CS53L30_A1_D1_CTL2_DFLT 0x00 + +/* CS53L30_ADC1_CTL3 */ +#define CS53L30_ADC1_HPF_EN (1 << 3) +#define CS53L30_ADC1_HPF_CF (3 << 1) +#define CS53L30_ADC1_NG_ALL (1 << 0) + +/* CS53L30_ADC1_NG_CTL */ +#define CS53L30_ADC1B_NG (1 << 7) +#define CS53L30_ADC1A_NG (1 << 6) +#define CS53L30_ADC1_NG_BOOST (1 << 5) +#define CS53L30_ADC1_NG_THRESH (7 << 2) +#define CS53L30_ADC1_NG_DELAY (3 << 0) +#define CS53L30_ADCX_ZERO_DFLT 0x00 + +/* CS53L30_ADC1A_AFE_CTL */ +#define CS53L30_ADC1A_PREAMP (3 << 6) +#define CS53L30_ADC1A_PGA_VOL 0x3F + +/* CS53L30_ADC1B_AFE_CTL */ +#define CS53L30_ADC1B_PREAMP (3 << 6) +#define CS53L30_ADC1B_PGA_VOL 0x3F + +/* CS53L30_ADCXX_DIG_VOL */ +#define CS53L30_MUTE_DIG_OUT (1 << 7) + +/* CS53L30_ADCDMIC2_CTL1 */ +#define CS53L30_ADC2B_PDN (1 << 7) +#define CS53L30_ADC2A_PDN (1 << 6) +#define CS53L30_DMIC2_PDN (1 << 2) +#define CS53L30_DMIC2_CLKDIV (1 << 1) +#define CS53L30_DMIC2_ON_AB_IN 0x00 /* CH_TYPE must = 1 */ +#define CS53L30_DMIC2_ON_A_IN (CS53L30_ADC2B_PDN) /* CH_TYPE must = 1 */ +#define CS53L30_DMIC2_ON_B_IN (CS53L30_ADC2A_PDN) /* CH_TYPE must = 1 */ +#define CS53L30_ADC2_ON_AB_IN (CS53L30_DMIC2_PDN) /* CH_TYPE must = 0 */ +#define CS53L30_ADC2_ON_A_IN (CS53L30_ADC2B_PDN | \ + CS53L30_DMIC2_PDN) +#define CS53L30_ADC2_ON_B_IN (CS53L30_ADC2A_PDN | \ + CS53L30_DMIC2_PDN) +#define CS53L30_D2_OFF_A2_OFF (CS53L30_ADC2A_PDN | \ + CS53L30_ADC2B_PDN | \ + CS53L30_DMIC2_PDN) + +/* CS53L30_ADCDMIC2_CTL2 */ +#define CS53L30_ADC2_NOTCH_DIS (1 << 7) +#define CS53L30_ADC2B_INV (1 << 5) +#define CS53L30_ADC2A_INV (1 << 4) +#define CS53L30_ADC2B_DIG_BOOST (1 << 1) +#define CS53L30_ADC2A_DIG_BOOST (1 << 0) + +/* CS53L30_ADC2_CTL3 */ +#define CS53L30_ADC2_HPF_EN (1 << 3) +#define CS53L30_ADC2_HPF_CF (3 << 1) +#define CS53L30_ADC2_NG_ALL (1 << 0) + +/* CS53L30_INT */ +#define CS53L30_PDN_DONE (1 << 7) +#define CS53L30_THMS_TRIP (1 << 6) +#define CS53L30_SYNC_DONE (1 << 5) +#define CS53L30_ADC2B_OVFL (1 << 4) +#define CS53L30_ADC2A_OVFL (1 << 3) +#define CS53L30_ADC1B_OVFL (1 << 2) +#define CS53L30_ADC1A_OVFL (1 << 1) +#define CS53L30_MUTE_PIN (1 << 0) +#define CS53L30_DEVICE_INT_MASK 0xFF + +/* Serial Ports */ +#define CS53L30_ASP1 0 +#define CS53L30_ASP2 1 + +#endif /* __CS53L30_H__ */ -- 2.4.5

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[alsa-devel] [REGRESSION] Headphones no longer working on MacPro6, 1 with 4.4
by Laura Abbott 06 Jun '16

06 Jun '16

Hi, We received a bug report https://bugzilla.redhat.com/show_bug.cgi?id=1316119 that the headphone jack on a MacPro6,1 stopped working on an upgrade from 4.3 to 4.4. The bugzilla has the alsainfo, diffing shows that the Amp-Out vals are different. I tried a revert of 9f660a1c4 (" ALSA: hda/realtek - Fix silent headphone output on MacPro 4,1 (v2)") but that didn't help. Any ideas before asking for a bisect? Does this hardware version need to have the vref fixup as well? Thanks, Laura

2 9

[alsa-devel] mcasp: stream has more channels (2) than are enabled in mcasp (0)
by Rick Mann 03 Jun '16

03 Jun '16

I've got a Beaglebone Green with a custom cape (CODEC and amp and some other stuff). Kernel 4.4.6-bone-rt-r6. When I load the overlay that should set up my CODEC in the device tree, I get the following in the log. [ 367.172337] davinci-mcasp 48038000.mcasp: stream has more channels (2) than are enabled in mcasp (0) [ 367.172374] davinci-mcasp 48038000.mcasp: ASoC: can't set 48038000.mcasp hw params: -22 You can see a little more log context, along with the DTS, here: http://pastebin.com/UPB1xjPF Can anyone tell me what this means? Thanks! -- Rick Mann rmann(a)latencyzero.com

3 14

[alsa-devel] [PATCH] ASoC: tlv320aix31xx: Add ACPI match for Lenovo 100S
by Bastien Nocera 01 Jun '16

01 Jun '16

The Lenovo 100S netbook has a codec controller for which there is a driver, but doesn't know how to access the device. This adds the necessary ACPI table for the driver to find the device. Device (TTLV) { Name (_ADR, Zero) // _ADR: Address Name (_HID, "10TI3100") // _HID: Hardware ID Name (_CID, "10TI3100") // _CID: Compatible ID Name (_DDN, "TI TLV320AIC3100 Codec Controller ") // _DDN: DOS Device Name Name (_UID, One) // _UID: Unique ID Signed-off-by: Bastien Nocera <hadess(a)hadess.net> Tested-by: Jan Schmidt <jan(a)centricular.com> --- sound/soc/codecs/tlv320aic31xx.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index ee4def4..3c5e1df 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -28,6 +28,7 @@ #include <linux/i2c.h> #include <linux/gpio.h> #include <linux/regulator/consumer.h> +#include <linux/acpi.h> #include <linux/of.h> #include <linux/of_gpio.h> #include <linux/slab.h> @@ -1280,10 +1281,19 @@ static const struct i2c_device_id aic31xx_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id); +#ifdef CONFIG_ACPI +static const struct acpi_device_id aic31xx_acpi_match[] = { + { "10TI3100", 0 }, + { } +}; +MODULE_DEVICE_TABLE(acpi, aic31xx_acpi_match); +#endif + static struct i2c_driver aic31xx_i2c_driver = { .driver = { .name = "tlv320aic31xx-codec", .of_match_table = of_match_ptr(tlv320aic31xx_of_match), + .acpi_match_table = ACPI_PTR(aic31xx_acpi_match), }, .probe = aic31xx_i2c_probe, .remove = aic31xx_i2c_remove, -- 2.7.3

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[alsa-devel] [PATCH resend v5 0/3] ASoC: cygnus: Add audio support for Broadcom Cygnus SoC
by Simran Rai 31 May '16

31 May '16

Hi, This patchset contains audio support for Broadcom's Cygnus SoC. It contains DT bindings and core audio driver. The audio driver supports both capture and playback of Audio PCM samples over I2S/TDM interface and provides playback support over SPDIF interface. This patchset is derived from a previously submitted patchset: http://lkml.iu.edu/hypermail/linux/kernel/1503.3/05434.html This patchset has been tested on Cygnus wireless audio bcm958305K board. It is rebased on top of v4.6-rc1 and is available from github: repo: https://github.com/Broadcom/cygnus-linux/tree/cygnus-sound-v5 Changes from v4: - Fix power suspend function and add power resume function - Remove clock initialization code from audio driver to clock framework Changes from v3: - Fix the subject lines to match the style for the subsystem Changes from v2: - Split patchset 2/2 from v2 into patchsets 2/3 and 3/3. - Remove SND_SOC_CYGNUS_DIAG. Diagnostics can be performed using standard kernel trace infrastructure. - Fix interrupt handler. Acknowledge only those interrupts that are handledby ISR. - Modify configure_vco() and the pll_macro_entry() struct to make it better readable. The functionality did not change. - Remove casts on macros - Removed surround sound channel grouping from the driver. Changes from v1: - Address code review comments. Fixed print format of type size_t and pointer. Simran Rai (3): ASoC: cygnus: Add DT bindings for Broadcom Cygnus audio ASoC: cygnus: Add Cygnus audio DAI driver ASoC: cygnus: Add Cygnus audio DMA driver .../bindings/sound/brcm,cygnus-audio.txt | 67 + sound/soc/bcm/Kconfig | 9 + sound/soc/bcm/Makefile | 5 + sound/soc/bcm/cygnus-pcm.c | 861 +++++++++++ sound/soc/bcm/cygnus-ssp.c | 1513 ++++++++++++++++++++ sound/soc/bcm/cygnus-ssp.h | 139 ++ 6 files changed, 2594 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt create mode 100644 sound/soc/bcm/cygnus-pcm.c create mode 100644 sound/soc/bcm/cygnus-ssp.c create mode 100644 sound/soc/bcm/cygnus-ssp.h -- 1.7.9.5

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[alsa-devel] [PATCH v6 0/3] ASoC: cygnus: Add audio support for Broadcom Cygnus SoC
by Simran Rai 31 May '16

31 May '16

Hi, This patchset contains audio support for Broadcom's Cygnus SoC. It contains DT bindings and core audio driver. The audio driver supports both capture and playback of Audio PCM samples over I2S/TDM interface and provides playback support over SPDIF interface. This patchset is derived from a previously submitted patchset: http://lkml.iu.edu/hypermail/linux/kernel/1503.3/05434.html This patchset has been tested on Cygnus wireless audio bcm958305K board. It is based on v4.6-rc1 and is available from github: repo: https://github.com/Broadcom/cygnus-linux/tree/cygnus-sound-v6 Changes from v5: - Fix code style, e.g. change "if" statements to "switch" statements - Fix SPDIF output enable register field - Set BUFFER_PAIR_ENABLE for both mono, stereo and TDM modes - Reflect PCM bit formats for SPDIF, SSP and TDM as supported by hardware Changes from v4: - Fix power suspend function and add power resume function - Move clock initialization code from audio driver to clock framework Changes from v3: - Fix the subject lines to match the style for the subsystem Changes from v2: - Split patchset 2/2 from v2 into patchsets 2/3 and 3/3. - Remove SND_SOC_CYGNUS_DIAG. Diagnostics can be performed using standard kernel trace infrastructure. - Fix interrupt handler. Acknowledge only those interrupts that are handledby ISR. - Modify configure_vco() and the pll_macro_entry() struct to make it better readable. The functionality did not change. - Remove casts on macros - Remove surround sound channel grouping from the driver. Changes from v1: - Address code review comments. Fix print format of type size_t and pointer. Simran Rai (3): ASoC: cygnus: Add DT bindings for Broadcom Cygnus audio ASoC: cygnus: Add Cygnus audio DAI driver ASoC: cygnus: Add Cygnus audio DMA driver .../bindings/sound/brcm,cygnus-audio.txt | 67 + sound/soc/bcm/Kconfig | 9 + sound/soc/bcm/Makefile | 5 + sound/soc/bcm/cygnus-pcm.c | 861 +++++++++++ sound/soc/bcm/cygnus-ssp.c | 1529 ++++++++++++++++++++ sound/soc/bcm/cygnus-ssp.h | 139 ++ 6 files changed, 2610 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt create mode 100644 sound/soc/bcm/cygnus-pcm.c create mode 100644 sound/soc/bcm/cygnus-ssp.c create mode 100644 sound/soc/bcm/cygnus-ssp.h -- 1.7.9.5

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[alsa-devel] [PATCH v4 0/7] ASoC: davinci-mcbsp: add binding for McBSP
by Petr Kulhavy 19 May '16

19 May '16

This series of patches adds devicetree support for the davinci McBSP audio interface. Changes in version 4 of the patch: * minor correction in the binding example ("interrupt names") * re-added index note to the "mpu" property.

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[alsa-devel] [PATCH] alsa-lib:Make snd_atomic_write_* truly atomic
by Andreas x Larsson 17 May '16

17 May '16

Under some usecases a race condition appears inside the snd_atomic_write_* functions. The 'begin' and 'end' variables are updated with the ++ operator which is not atomic but needs to be. This can be achieved with the gcc atomic_* built-ins. Combined with __ATOMIC_SEQ_CST as the memory model, memory barriers are introduced so those can also be removed from the code. Signed-off-by: Andreas x Larsson <andrelan(a)axis.com> Signed-off-by: Arvid Nihlgård Lindell <arvidnl(a)axis.com> --- alsa-lib/include/iatomic.h | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/alsa-lib/include/iatomic.h b/alsa-lib/include/iatomic.h index acdd3e2..7fafe20 100644 --- a/alsa-lib/include/iatomic.h +++ b/alsa-lib/include/iatomic.h @@ -140,14 +140,12 @@ static __inline__ void snd_atomic_write_init(snd_atomic_write_t *w) static __inline__ void snd_atomic_write_begin(snd_atomic_write_t *w) { - w->begin++; - wmb(); + __atomic_add_fetch(&w->begin, 1,__ATOMIC_SEQ_CST); } static __inline__ void snd_atomic_write_end(snd_atomic_write_t *w) { - wmb(); - w->end++; + __atomic_add_fetch(&w->end, 1, __ATOMIC_SEQ_CST); } static __inline__ void snd_atomic_read_init(snd_atomic_read_t *r, snd_atomic_write_t *w) -- 2.1.4

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[alsa-devel] ADAU1761 default register value problem
by Ricard Wanderlof 16 May '16

16 May '16

Hi Lars, A while ago I posted a question to the list regarding the default values set up for the ADAU1761 (and similar CODECs), but I didn't get a response. I really need to try and figure out a way forward here though so I'll bring this up again. The problem I have is that the driver assumes that the CODEC has been initialized with the default register settings when the system boots. However, if the system has not been power cycled but just rebooted, this is not necessarily the case, as the CODEC has no external reset pin but derives its reset signal internally from the power level. Furthermore, if the system upon reboot tries to set a register value to one of the default settings (for example as the result of a hw_params() call), then the regcache mechanism will disregard the write as it assumes the default value is already set, which is not necessarily the case. As I see it the solution to this problem is simply not to assume anything about the register settings, so that the first write to a given register always takes effect. I can work on a patch to accomplish this, however, since so much work apparently has been laid down in getting the default values into the driver in the first place, I'm assuming there's something more to the story that I'm missing, so I'd like to get a better grip on this before I dive in. /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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[alsa-devel] [PATCH v2 1/2] ASoC: dapm: support user-defined stop condition in dai_get_connected_widgets
by Piotr Stankiewicz 13 May '16

13 May '16

Certain situations may warrant examining DAPM paths only to a certain arbitrary point, as opposed to always following them to the end. For instance, when establishing a connection between a front-end DAI link and a back-end DAI link in a DPCM path, it does not make sense to walk the DAPM graph beyond the first widget associated with a back-end link. This patch introduces a mechanism which lets a user of dai_get_connected_widgets supply a function which will be called for every node during the graph walk. When invoked, this function can execute arbitrary logic to decide whether the walk, given a DAPM widget and walk direction, should be terminated at that point or continued as normal. Signed-off-by: Piotr Stankiewicz <piotrs(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 5 +++- sound/soc/soc-dapm.c | 68 +++++++++++++++++++++++++++++++++++++--------- sound/soc/soc-pcm.c | 3 +- 3 files changed, 61 insertions(+), 15 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 9706946..fe5bc93 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -357,6 +357,7 @@ struct snd_soc_dapm_context; struct regulator; struct snd_soc_dapm_widget_list; struct snd_soc_dapm_update; +enum snd_soc_dapm_direction; int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); @@ -450,7 +451,9 @@ void dapm_mark_endpoints_dirty(struct snd_soc_card *card); /* dapm path query */ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, - struct snd_soc_dapm_widget_list **list); + struct snd_soc_dapm_widget_list **list, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)); struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm( struct snd_kcontrol *kcontrol); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 801ae1a..075f0d7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1073,7 +1073,11 @@ static int dapm_widget_list_create(struct snd_soc_dapm_widget_list **list, */ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, struct list_head *list, enum snd_soc_dapm_direction dir, - int (*fn)(struct snd_soc_dapm_widget *, struct list_head *)) + int (*fn)(struct snd_soc_dapm_widget *, struct list_head *, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)), + int (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)) { enum snd_soc_dapm_direction rdir = SND_SOC_DAPM_DIR_REVERSE(dir); struct snd_soc_dapm_path *path; @@ -1088,6 +1092,13 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, if (list) list_add_tail(&widget->work_list, list); + if (custom_stop_condition) { + con = custom_stop_condition(widget, dir); + if (con >= 0) + return con; + con = 0; + } + if ((widget->is_ep & SND_SOC_DAPM_DIR_TO_EP(dir)) && widget->connected) { widget->endpoints[dir] = snd_soc_dapm_suspend_check(widget); return widget->endpoints[dir]; @@ -1106,7 +1117,7 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, if (path->connect) { path->walking = 1; - con += fn(path->node[dir], list); + con += fn(path->node[dir], list, custom_stop_condition); path->walking = 0; } } @@ -1119,23 +1130,41 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, /* * Recursively check for a completed path to an active or physically connected * output widget. Returns number of complete paths. + * + * Optionally, can be supplied with a function acting as a stopping condition. + * This function takes the dapm widget currently being examined and the walk + * direction as an arguments, and should return an integer, as follows: + * - >=0: if the walk is to be stopped (this should indicate the number of + * endpoints the widget has), + * - <0 : if the walk is to be continued as normal. */ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, - struct list_head *list) + struct list_head *list, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *i, + enum snd_soc_dapm_direction)) { return is_connected_ep(widget, list, SND_SOC_DAPM_DIR_OUT, - is_connected_output_ep); + is_connected_output_ep, custom_stop_condition); } /* * Recursively check for a completed path to an active or physically connected * input widget. Returns number of complete paths. + * + * Optionally, can be supplied with a function acting as a stopping condition. + * This function takes the dapm widget currently being examined and the walk + * direction as an arguments, and should return an integer, as follows: + * - >=0: if the walk is to be stopped (this should indicate the number of + * endpoints the widget has), + * - <0 : if the walk is to be continued as normal. */ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, - struct list_head *list) + struct list_head *list, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *i, + enum snd_soc_dapm_direction)) { return is_connected_ep(widget, list, SND_SOC_DAPM_DIR_IN, - is_connected_input_ep); + is_connected_input_ep, custom_stop_condition); } /** @@ -1143,15 +1172,26 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, * @dai: the soc DAI. * @stream: stream direction. * @list: list of active widgets for this stream. + * @custom_stop_condition: (optional) a function meant to stop the widget graph + * walk based on custom logic. * * Queries DAPM graph as to whether an valid audio stream path exists for * the initial stream specified by name. This takes into account * current mixer and mux kcontrol settings. Creates list of valid widgets. * + * Optionally, can be supplied with a function acting as a stopping condition. + * This function takes the dapm widget currently being examined and the walk + * direction as an arguments, and should return an integer, as follows: + * - >=0: if the walk is to be stopped (this should indicate the number of + * endpoints the widget has), + * - <0 : if the walk is to be continued as normal. + * * Returns the number of valid paths or negative error. */ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, - struct snd_soc_dapm_widget_list **list) + struct snd_soc_dapm_widget_list **list, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)) { struct snd_soc_card *card = dai->component->card; struct snd_soc_dapm_widget *w; @@ -1171,9 +1211,11 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, } if (stream == SNDRV_PCM_STREAM_PLAYBACK) - paths = is_connected_output_ep(dai->playback_widget, &widgets); + paths = is_connected_output_ep(dai->playback_widget, &widgets, + custom_stop_condition); else - paths = is_connected_input_ep(dai->capture_widget, &widgets); + paths = is_connected_input_ep(dai->capture_widget, &widgets, + custom_stop_condition); /* Drop starting point */ list_del(widgets.next); @@ -1268,8 +1310,8 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); - in = is_connected_input_ep(w, NULL); - out = is_connected_output_ep(w, NULL); + in = is_connected_input_ep(w, NULL, NULL); + out = is_connected_output_ep(w, NULL, NULL); return out != 0 && in != 0; } @@ -1928,8 +1970,8 @@ static ssize_t dapm_widget_power_read_file(struct file *file, in = 0; out = 0; } else { - in = is_connected_input_ep(w, NULL); - out = is_connected_output_ep(w, NULL); + in = is_connected_input_ep(w, NULL, NULL); + out = is_connected_output_ep(w, NULL, NULL); } ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index aa99dac..c2b0aa8 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1294,7 +1294,8 @@ int dpcm_path_get(struct snd_soc_pcm_runtime *fe, int paths; /* get number of valid DAI paths and their widgets */ - paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, list); + paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, list, + NULL); dev_dbg(fe->dev, "ASoC: found %d audio %s paths\n", paths, stream ? "capture" : "playback"); -- 1.7.10.4

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