[alsa-devel] [PATCH RFC 0/5] AM43xx-ePOS-EVM audio support with TLV320AIC31XX driver
This patch set adds audio support for AM43xx-ePOS-EVM. I'll mail the dts and defconfig changes once these have been merged.
Jyri Sarha (5): ASoC: tlv320aic31xx: Add basic codec driver implementation ASoC: tlv320aic31xx: Add codec driver to Makefile and Kconfig ASoC: tlv320aic31xx: Add DT binding document ASoC: davinci-evm: Add AM43xx-EPOS-EVM audio support ASoC: davinci: Add SND_AM43XX_SOC_EPOS_EVM build option
.../bindings/sound/davinci-evm-audio.txt | 9 +- .../devicetree/bindings/sound/tlv320aic31xx.txt | 57 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic31xx.c | 1360 ++++++++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 265 ++++ sound/soc/davinci/Kconfig | 12 + sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 41 + 9 files changed, 1748 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt create mode 100644 sound/soc/codecs/tlv320aic31xx.c create mode 100644 sound/soc/codecs/tlv320aic31xx.h
This commit adds a bare bones driver support for TLV320AIC31XX family audio codecs. The driver adds basic stereo playback trough headphone and speaker outputs and mono capture trough microphone inputs.
The driver is currently missing support at least for mini DSP features and jack detection. I have tested the driver only on TLV320AIC3111, but based on the data sheets TLV320AIC3100, TLV320AIC3110, and TLV320AIC3120 should work Ok too.
The base for the implementation was taken from: git@gitorious.org:ti-codecs/ti-codecs.git ajitk/topics/k3.10.1-aic31xx -branch at commit 77504eba0294764e9e63b4a0c696b44db187cd13.
Signed-off-by: Jyri Sarha jsarha@ti.com --- sound/soc/codecs/tlv320aic31xx.c | 1360 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 265 ++++++++ 2 files changed, 1625 insertions(+) create mode 100644 sound/soc/codecs/tlv320aic31xx.c create mode 100644 sound/soc/codecs/tlv320aic31xx.h
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c new file mode 100644 index 0000000..1f83779 --- /dev/null +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -0,0 +1,1360 @@ +/* + * ALSA SoC TLV320AIC31XX codec driver + * + * Copyright (C) 2013 Texas Instruments, Inc. + * + * Author: Jyri Sarha jsarha@ti.com + * + * Based on ground work by: Ajit Kulkarni x0175765@ti.com + * + * This package 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 PACKAGE IS PROVIDED AS IS AND WITHOUT ANY EXPRESS OR + * IMPLIED WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED + * WARRANTIES OF MERCHANTIBILITY AND FITNESS FOR A PARTICULAR PURPOSE. + * + * The TLV320AIC31xx series of audio codec is a low-power, highly integrated + * high performance codec which provides a stereo DAC, a mono ADC, + * and mono/stereo Class-D speaker driver. + */ + +#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/gpio.h> +#include <linux/regulator/consumer.h> +#include <linux/of_gpio.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "tlv320aic31xx.h" + +static const struct reg_default aic31xx_reg_defaults[] = { + { AIC31XX_CLKMUX, 0x00 }, + { AIC31XX_PLLPR, 0x11 }, + { AIC31XX_PLLJ, 0x04 }, + { AIC31XX_PLLDMSB, 0x00 }, + { AIC31XX_PLLDLSB, 0x00 }, + { AIC31XX_NDAC, 0x01 }, + { AIC31XX_MDAC, 0x01 }, + { AIC31XX_DOSRMSB, 0x00 }, + { AIC31XX_DOSRLSB, 0x80 }, + { AIC31XX_NADC, 0x01 }, + { AIC31XX_MADC, 0x01 }, + { AIC31XX_AOSR, 0x80 }, + { AIC31XX_IFACE1, 0x00 }, + { AIC31XX_DATA_OFFSET, 0x00 }, + { AIC31XX_IFACE2, 0x00 }, + { AIC31XX_BCLKN, 0x01 }, + { AIC31XX_DACSETUP, 0x14 }, + { AIC31XX_DACMUTE, 0x0c }, + { AIC31XX_LDACVOL, 0x00 }, + { AIC31XX_RDACVOL, 0x00 }, + { AIC31XX_ADCSETUP, 0x00 }, + { AIC31XX_ADCFGA, 0x80 }, + { AIC31XX_ADCVOL, 0x00 }, + { AIC31XX_HPDRIVER, 0x04 }, + { AIC31XX_SPKAMP, 0x06 }, + { AIC31XX_DACMIXERROUTE, 0x00 }, + { AIC31XX_LANALOGHPL, 0x7f }, + { AIC31XX_RANALOGHPR, 0x7f }, + { AIC31XX_LANALOGSPL, 0x7f }, + { AIC31XX_RANALOGSPR, 0x7f }, + { AIC31XX_HPLGAIN, 0x02 }, + { AIC31XX_HPRGAIN, 0x02 }, + { AIC31XX_SPLGAIN, 0x00 }, + { AIC31XX_SPRGAIN, 0x00 }, + { AIC31XX_MICBIAS, 0x00 }, + { AIC31XX_MICPGA, 0x80 }, + { AIC31XX_MICPGAPI, 0x00 }, + { AIC31XX_MICPGAMI, 0x00 }, +}; + +static bool aic31xx_precious(struct device *dev, unsigned int reg) +{ + switch (reg) { + case AIC31XX_OFFLAG: + case AIC31XX_INTRDACFLAG: + case AIC31XX_INTRADCFLAG: + return true; + } + return false; +} + +static bool aic31xx_real_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case AIC31XX_OT_FLAG: + case AIC31XX_ADCFLAG: + case AIC31XX_DACFLAG1: + case AIC31XX_DACFLAG2: + case AIC31XX_INTRDACFLAG2: + case AIC31XX_INTRADCFLAG2: + return true; + } + return false; +} + +static bool aic31xx_volatile(struct device *dev, unsigned int reg) +{ + if (aic31xx_precious(dev, reg) || aic31xx_real_volatile(dev, reg)) + return true; + + switch (reg) { + case AIC31XX_PAGECTL: /* regmap implementation requires this */ + case AIC31XX_RESET: /* always clears after write */ + return true; + } + return false; +} + +static bool aic31xx_writeable(struct device *dev, unsigned int reg) +{ + if (aic31xx_precious(dev, reg) || aic31xx_real_volatile(dev, reg)) + return false; + + return true; +} + +static const struct regmap_range_cfg aic31xx_ranges[] = { + { + .name = "codec-regmap", + .range_min = 0, + .range_max = 12 * 128, + .selector_reg = AIC31XX_PAGECTL, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = 128, + }, +}; + +struct regmap_config aic31xx_i2c_regmap = { + .reg_bits = 8, + .val_bits = 8, + .writeable_reg = aic31xx_writeable, + .volatile_reg = aic31xx_volatile, + .precious_reg = aic31xx_precious, + .reg_defaults = aic31xx_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(aic31xx_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .ranges = aic31xx_ranges, + .num_ranges = ARRAY_SIZE(aic31xx_ranges), + .max_register = 12 * 128, +}; + +#define AIC31XX_NUM_SUPPLIES 6 +static const char * const aic31xx_supply_names[] = { + "HPVDD", + "SPRVDD", + "SPLVDD", + "AVDD", + "IOVDD", + "DVDD", +}; + +struct aic31xx_disable_nb { + struct notifier_block nb; + struct aic31xx_priv *aic31xx; +}; + +struct aic31xx_priv { + struct snd_soc_codec *codec; + u8 i2c_regs_status; + struct device *dev; + struct regmap *regmap; + struct aic31xx_pdata pdata; + struct regulator_bulk_data supplies[AIC31XX_NUM_SUPPLIES]; + struct aic31xx_disable_nb disable_nb[AIC31XX_NUM_SUPPLIES]; + int power; + unsigned int sysclk; + int rate_div_line; + int codec_clk_on; +}; + +struct aic31xx_rate_divs { + u32 mclk; + u32 rate; + u8 p_val; + u8 pll_j; + u16 pll_d; + u16 dosr; + u8 ndac; + u8 mdac; + u8 aosr; + u8 nadc; + u8 madc; +}; + +/* ADC dividers can be disabled by cofiguring them to 0 */ +static const struct aic31xx_rate_divs aic31xx_divs[] = { + /* mclk rate pll: p j d dosr ndac mdac aors nadc madc */ + /* 8k rate */ + {12000000, 8000, 1, 8, 1920, 128, 48, 2, 128, 48, 2}, + {24000000, 8000, 2, 8, 1920, 128, 48, 2, 128, 48, 2}, + {25000000, 8000, 2, 7, 8643, 128, 48, 2, 128, 48, 2}, + /* 11.025k rate */ + {12000000, 11025, 1, 7, 5264, 128, 32, 2, 128, 32, 2}, + {24000000, 11025, 2, 7, 5264, 128, 32, 2, 128, 32, 2}, + {25000000, 11025, 2, 7, 2253, 128, 32, 2, 128, 32, 2}, + /* 16k rate */ + {12000000, 16000, 1, 8, 1920, 128, 24, 2, 128, 24, 2}, + {24000000, 16000, 2, 8, 1920, 128, 24, 2, 128, 24, 2}, + {25000000, 16000, 2, 7, 8643, 128, 24, 2, 128, 24, 2}, + /* 22.05k rate */ + {12000000, 22050, 1, 7, 5264, 128, 16, 2, 128, 16, 2}, + {24000000, 22050, 2, 7, 5264, 128, 16, 2, 128, 16, 2}, + {25000000, 22050, 2, 7, 2253, 128, 16, 2, 128, 16, 2}, + /* 32k rate */ + {12000000, 32000, 1, 8, 1920, 128, 12, 2, 128, 12, 2}, + {24000000, 32000, 2, 8, 1920, 128, 12, 2, 128, 12, 2}, + {25000000, 32000, 2, 7, 8643, 128, 12, 2, 128, 12, 2}, + /* 44.1k rate */ + {12000000, 44100, 1, 7, 5264, 128, 8, 2, 128, 8, 2}, + {24000000, 44100, 2, 7, 5264, 128, 8, 2, 128, 8, 2}, + {25000000, 44100, 2, 7, 2253, 128, 8, 2, 128, 8, 2}, + /* 48k rate */ + {12000000, 48000, 1, 8, 1920, 128, 8, 2, 128, 8, 2}, + {24000000, 48000, 2, 8, 1920, 128, 8, 2, 128, 8, 2}, + {25000000, 48000, 2, 7, 8643, 128, 8, 2, 128, 8, 2}, + /* 88.2k rate */ + {12000000, 88200, 1, 7, 5264, 64, 8, 2, 64, 8, 2}, + {24000000, 88200, 2, 7, 5264, 64, 8, 2, 64, 8, 2}, + {25000000, 88200, 2, 7, 2253, 64, 8, 2, 64, 8, 2}, + /* 96k rate */ + {12000000, 96000, 1, 8, 1920, 64, 8, 2, 64, 8, 2}, + {24000000, 96000, 2, 8, 1920, 64, 8, 2, 64, 8, 2}, + {25000000, 96000, 2, 7, 8643, 64, 8, 2, 64, 8, 2}, + /* 176.4k rate */ + {12000000, 176400, 1, 7, 5264, 32, 8, 2, 32, 8, 2}, + {24000000, 176400, 2, 7, 5264, 32, 8, 2, 32, 8, 2}, + {25000000, 176400, 2, 7, 2253, 32, 8, 2, 32, 8, 2}, + /* 192k rate */ + {12000000, 192000, 1, 8, 1920, 32, 8, 2, 32, 8, 2}, + {24000000, 192000, 2, 8, 1920, 32, 8, 2, 32, 8, 2}, + {25000000, 192000, 2, 7, 8643, 32, 8, 2, 32, 8, 2}, +}; + +static const char * const ldac_in_text[] = { + "off", "Left Data", "Right Data", "Mono" +}; + +static const char * const rdac_in_text[] = { + "off", "Right Data", "Left Data", "Mono" +}; + +static SOC_ENUM_SINGLE_DECL(ldac_in_enum, AIC31XX_DACSETUP, 4, ldac_in_text); + +static SOC_ENUM_SINGLE_DECL(rdac_in_enum, AIC31XX_DACSETUP, 2, rdac_in_text); + +static const char * const mic_select_text[] = { + "off", "FFR 10 Ohm", "FFR 20 Ohm", "FFR 40 Ohm" +}; + +static const +SOC_ENUM_SINGLE_DECL(mic1lp_p_enum, AIC31XX_MICPGAPI, 6, mic_select_text); +static const +SOC_ENUM_SINGLE_DECL(mic1rp_p_enum, AIC31XX_MICPGAPI, 4, mic_select_text); +static const +SOC_ENUM_SINGLE_DECL(mic1lm_p_enum, AIC31XX_MICPGAPI, 2, mic_select_text); + +static const +SOC_ENUM_SINGLE_DECL(cm_m_enum, AIC31XX_MICPGAMI, 6, mic_select_text); +static const +SOC_ENUM_SINGLE_DECL(mic1lm_m_enum, AIC31XX_MICPGAMI, 4, mic_select_text); + +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(adc_fgain_tlv, 0, 10, 0); +static const DECLARE_TLV_DB_SCALE(adc_cgain_tlv, -2000, 50, 0); +static const DECLARE_TLV_DB_SCALE(mic_pga_tlv, 0, 50, 0); +static const DECLARE_TLV_DB_SCALE(hp_drv_tlv, 0, 100, 0); +static const DECLARE_TLV_DB_SCALE(class_D_drv_tlv, 600, 600, 0); +static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(sp_vol_tlv, -6350, 50, 0); + +/* + * controls to be exported to the user space + */ +static const struct snd_kcontrol_new aic31xx_snd_controls[] = { + SOC_DOUBLE_R_SX_TLV("DAC Playback Volume", AIC31XX_LDACVOL, + AIC31XX_RDACVOL, 0, 0x81, 0xaf, dac_vol_tlv), + + SOC_SINGLE_TLV("ADC Fine Capture Volume", AIC31XX_ADCFGA, 4, 4, 1, + adc_fgain_tlv), + + SOC_SINGLE("ADC Capture Switch", AIC31XX_ADCFGA, 7, 1, 1), + + /* ADC COARSE GAIN */ + /* Writes to AIC31XX_ADCVOL should be masked with 0x7F, but there + appears to be no side effect if writes are no masked. */ + SOC_SINGLE_SX_TLV("ADC Capture Volume", AIC31XX_ADCVOL, + 0, 0x68, 0x40, adc_cgain_tlv), + + SOC_SINGLE_TLV("Mic PGA Capture Volume", AIC31XX_MICPGA, 0, + 119, 0, mic_pga_tlv), + + SOC_DOUBLE_R("HP Driver Playback Switch", AIC31XX_HPLGAIN, + AIC31XX_HPRGAIN, 2, 1, 0), + SOC_DOUBLE_R_TLV("HP Driver Playback Volume", AIC31XX_HPLGAIN, + AIC31XX_HPRGAIN, 3, 0x09, 0, hp_drv_tlv), + + SOC_DOUBLE_R_TLV("HP Analog Playback Volume", AIC31XX_LANALOGHPL, + AIC31XX_RANALOGHPR, 0, 0x7F, 1, hp_vol_tlv), +}; + +static const struct snd_kcontrol_new aic311x_snd_controls[] = { + SOC_DOUBLE_R("SP Driver Playback Switch", AIC31XX_SPLGAIN, + AIC31XX_SPRGAIN, 2, 1, 0), + SOC_DOUBLE_R_TLV("SP Driver Playback Volume", AIC31XX_SPLGAIN, + AIC31XX_SPRGAIN, 3, 3, 0, class_D_drv_tlv), + + SOC_DOUBLE_R_TLV("SP Analog Playback Volume", AIC31XX_LANALOGSPL, + AIC31XX_RANALOGSPR, 0, 0x7F, 1, sp_vol_tlv), +}; + +static const struct snd_kcontrol_new aic310x_snd_controls[] = { + SOC_SINGLE("SP Driver Playback Switch", AIC31XX_SPLGAIN, + 2, 1, 0), + SOC_SINGLE_TLV("SP Driver Playback Volume", AIC31XX_SPLGAIN, + 3, 3, 0, class_D_drv_tlv), + + SOC_SINGLE_TLV("SP Analog Playback Volume", AIC31XX_LANALOGSPL, + 0, 0x7F, 1, sp_vol_tlv), +}; + +static const struct snd_kcontrol_new ldac_in_control = + SOC_DAPM_ENUM("DAC Left Input", ldac_in_enum); + +static const struct snd_kcontrol_new rdac_in_control = + SOC_DAPM_ENUM("DAC Right Input", rdac_in_enum); + +int aic31xx_wait_bits(struct aic31xx_priv *aic31xx, unsigned int reg, + unsigned int mask, unsigned int wbits, int sleep, + int count) +{ + unsigned int bits; + int counter = count; + int ret = regmap_read(aic31xx->regmap, reg, &bits); + while ((bits & mask) != wbits && counter && !ret) { + usleep_range(sleep, sleep * 2); + ret = regmap_read(aic31xx->regmap, reg, &bits); + counter--; + } + if ((bits & mask) != wbits) { + dev_err(aic31xx->dev, + "%s: Failed! 0x%x was 0x%x expected 0x%x (%d, 0x%x, %d us)\n", + __func__, reg, bits, wbits, ret, mask, + (count - counter) * sleep); + ret = -1; + } + return ret; +} + +static int aic31xx_dapm_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(w->codec); + unsigned int reg = AIC31XX_DACFLAG1; + unsigned int mask; + + dev_dbg(w->codec->dev, "%s -> %s (event: %d)\n", w->name, + event == SND_SOC_DAPM_POST_PMU ? "on" : + event == SND_SOC_DAPM_POST_PMD ? "off" : + "(unsupported event)", event); + + if (!strcmp(w->name, "DAC Left")) { + mask = AIC31XX_LDACPWRSTATUS_MASK; + } else if (!strcmp(w->name, "DAC Right")) { + mask = AIC31XX_RDACPWRSTATUS_MASK; + } else if (!strcmp(w->name, "HPL Driver")) { + mask = AIC31XX_HPLDRVPWRSTATUS_MASK; + } else if (!strcmp(w->name, "HPR Driver")) { + mask = AIC31XX_HPRDRVPWRSTATUS_MASK; + } else if (!strcmp(w->name, "SPL ClassD") || + !strcmp(w->name, "SPK ClassD")) { + mask = AIC31XX_SPLDRVPWRSTATUS_MASK; + } else if (!strcmp(w->name, "SPR ClassD")) { + mask = AIC31XX_SPRDRVPWRSTATUS_MASK; + } else if (!strcmp(w->name, "ADC")) { + mask = AIC31XX_ADCPWRSTATUS_MASK; + reg = AIC31XX_ADCFLAG; + } else { + dev_err(w->codec->dev, "Unknown widget '%s' calling %s/n", + w->name, __func__); + return -1; + } + + if (event == SND_SOC_DAPM_POST_PMU) + return aic31xx_wait_bits(aic31xx, reg, mask, mask, 5000, 100); + else if (event == SND_SOC_DAPM_POST_PMD) + return aic31xx_wait_bits(aic31xx, reg, mask, 0, 5000, 100); + + dev_dbg(w->codec->dev, "Unhandled dapm widget event %d from %s\n", + event, w->name); + return 0; +} + +static const struct snd_kcontrol_new left_output_switches[] = { + SOC_DAPM_SINGLE("From Left DAC", AIC31XX_DACMIXERROUTE, 6, 1, 0), + SOC_DAPM_SINGLE("From MIC1LP", AIC31XX_DACMIXERROUTE, 5, 1, 0), + SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 4, 1, 0), +}; + +static const struct snd_kcontrol_new right_output_switches[] = { + SOC_DAPM_SINGLE("From Right DAC", AIC31XX_DACMIXERROUTE, 2, 1, 0), + SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 1, 1, 0), +}; + +static const struct snd_kcontrol_new p_term_mic1lp = + SOC_DAPM_ENUM("MIC1LP P-Terminal", mic1lp_p_enum); + +static const struct snd_kcontrol_new p_term_mic1rp = + SOC_DAPM_ENUM("MIC1RP P-Terminal", mic1rp_p_enum); + +static const struct snd_kcontrol_new p_term_mic1lm = + SOC_DAPM_ENUM("MIC1LM P-Terminal", mic1lm_p_enum); + +static const struct snd_kcontrol_new m_term_mic1lm = + SOC_DAPM_ENUM("MIC1LM M-Terminal", mic1lm_m_enum); + +static const struct snd_kcontrol_new aic31xx_dapm_hpl_switch = + SOC_DAPM_SINGLE("Switch", AIC31XX_LANALOGHPL, 7, 1, 0); + +static const struct snd_kcontrol_new aic31xx_dapm_hpr_switch = + SOC_DAPM_SINGLE("Switch", AIC31XX_RANALOGHPR, 7, 1, 0); + +static const struct snd_kcontrol_new aic31xx_dapm_spl_switch = + SOC_DAPM_SINGLE("Switch", AIC31XX_LANALOGSPL, 7, 1, 0); + +static const struct snd_kcontrol_new aic31xx_dapm_spr_switch = + SOC_DAPM_SINGLE("Switch", AIC31XX_RANALOGSPR, 7, 1, 0); + +static int aic31xx_clk_pm_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + u8 mask = AIC31XX_PM_MASK; + u8 on = AIC31XX_PM_MASK; + u8 off = 0; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + dev_dbg(codec->dev, "codec clock -> on (rate %d)\n", + aic31xx_divs[aic31xx->rate_div_line].rate); + snd_soc_update_bits(codec, AIC31XX_PLLPR, mask, on); + mdelay(10); + snd_soc_update_bits(codec, AIC31XX_NDAC, mask, on); + snd_soc_update_bits(codec, AIC31XX_MDAC, mask, on); + if (aic31xx_divs[aic31xx->rate_div_line].nadc) + snd_soc_update_bits(codec, AIC31XX_NADC, mask, on); + if (aic31xx_divs[aic31xx->rate_div_line].madc) + snd_soc_update_bits(codec, AIC31XX_MADC, mask, on); + snd_soc_update_bits(codec, AIC31XX_BCLKN, mask, on); + aic31xx->codec_clk_on = 1; + } else if (SND_SOC_DAPM_EVENT_OFF(event)) { + dev_dbg(codec->dev, "codec clock -> off\n"); + snd_soc_update_bits(codec, AIC31XX_BCLKN, mask, off); + snd_soc_update_bits(codec, AIC31XX_MADC, mask, off); + snd_soc_update_bits(codec, AIC31XX_NADC, mask, off); + snd_soc_update_bits(codec, AIC31XX_MDAC, mask, off); + snd_soc_update_bits(codec, AIC31XX_NDAC, mask, off); + snd_soc_update_bits(codec, AIC31XX_PLLPR, mask, off); + aic31xx->codec_clk_on = 0; + } + return 0; +} + +static int mic_bias_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* change mic bias voltage to user defined */ + if (aic31xx->pdata.micbias_vg != AIC31XX_MICBIAS_OFF) { + snd_soc_update_bits(codec, AIC31XX_MICBIAS, + AIC31XX_MICBIAS_MASK, + aic31xx->pdata.micbias_vg << + AIC31XX_MICBIAS_SHIFT); + dev_dbg(codec->dev, "%s: turned on\n", __func__); + } + break; + case SND_SOC_DAPM_PRE_PMD: + if (aic31xx->pdata.micbias_vg != AIC31XX_MICBIAS_OFF) { + snd_soc_update_bits(codec, AIC31XX_MICBIAS, + AIC31XX_MICBIAS_MASK, 0); + dev_dbg(codec->dev, "%s: turned off\n", __func__); + } + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAC IN", "DAC Playback", 0, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("DAC Left Input", + SND_SOC_NOPM, 0, 0, &ldac_in_control), + SND_SOC_DAPM_MUX("DAC Right Input", + SND_SOC_NOPM, 0, 0, &rdac_in_control), + /* DACs */ + SND_SOC_DAPM_DAC_E("DAC Left", "DAC Left Input", + AIC31XX_DACSETUP, 7, 0, aic31xx_dapm_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_DAC_E("DAC Right", "DAC Right Input", + AIC31XX_DACSETUP, 6, 0, aic31xx_dapm_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Output Left", SND_SOC_NOPM, 0, 0, + left_output_switches, + ARRAY_SIZE(left_output_switches)), + SND_SOC_DAPM_MIXER("Output Right", SND_SOC_NOPM, 0, 0, + right_output_switches, + ARRAY_SIZE(right_output_switches)), + + SND_SOC_DAPM_SWITCH("HP Left", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_hpl_switch), + SND_SOC_DAPM_SWITCH("HP Right", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_hpr_switch), + + /* Output drivers */ + SND_SOC_DAPM_OUT_DRV_E("HPL Driver", AIC31XX_HPDRIVER, 7, 0, + NULL, 0, aic31xx_dapm_power_event, + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_OUT_DRV_E("HPR Driver", AIC31XX_HPDRIVER, 6, 0, + NULL, 0, aic31xx_dapm_power_event, + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU), + + /* ADC */ + SND_SOC_DAPM_ADC_E("ADC", "Capture", AIC31XX_ADCSETUP, 7, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + + /* Input Selection to MIC_PGA */ + SND_SOC_DAPM_MUX("MIC1LP P-Terminal", SND_SOC_NOPM, 0, 0, + &p_term_mic1lp), + SND_SOC_DAPM_MUX("MIC1RP P-Terminal", SND_SOC_NOPM, 0, 0, + &p_term_mic1rp), + SND_SOC_DAPM_MUX("MIC1LM P-Terminal", SND_SOC_NOPM, 0, 0, + &p_term_mic1lm), + + SND_SOC_DAPM_MUX("MIC1LM M-Terminal", SND_SOC_NOPM, 0, 0, + &m_term_mic1lm), + /* Enabling & Disabling MIC Gain Ctl */ + SND_SOC_DAPM_PGA("MIC_GAIN_CTL", AIC31XX_MICPGA, + 7, 1, NULL, 0), + + SND_SOC_DAPM_SUPPLY("CODEC_CLK_IN", SND_SOC_NOPM, 0, 0, + aic31xx_clk_pm_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* Mic Bias */ + SND_SOC_DAPM_SUPPLY("Mic Bias", SND_SOC_NOPM, 0, 0, mic_bias_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + + /* Inputs */ + SND_SOC_DAPM_INPUT("MIC1LP"), + SND_SOC_DAPM_INPUT("MIC1RP"), + SND_SOC_DAPM_INPUT("MIC1LM"), +}; + +static const struct snd_soc_dapm_widget aic311x_dapm_widgets[] = { + /* AIC3111 and AIC3110 have stereo class-D amplifier */ + SND_SOC_DAPM_OUT_DRV_E("SPL ClassD", AIC31XX_SPKAMP, 7, 0, NULL, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUT_DRV_E("SPR ClassD", AIC31XX_SPKAMP, 6, 0, NULL, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SWITCH("SP Left", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_spl_switch), + SND_SOC_DAPM_SWITCH("SP Right", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_spr_switch), + SND_SOC_DAPM_OUTPUT("SPL"), + SND_SOC_DAPM_OUTPUT("SPR"), +}; + +/* AIC3100 and AIC3120 have only mono class-D amplifier */ +static const struct snd_soc_dapm_widget aic310x_dapm_widgets[] = { + SND_SOC_DAPM_OUT_DRV_E("SPK ClassD", AIC31XX_SPKAMP, 7, 0, NULL, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SWITCH("Speaker", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_spl_switch), + SND_SOC_DAPM_OUTPUT("SPK"), +}; + +static const struct snd_soc_dapm_route +aic31xx_audio_map[] = { + /* Clocks for ADC */ + {"ADC", NULL, "CODEC_CLK_IN"}, + + /* DAC Input Routing */ + {"DAC Left Input", "Left Data", "DAC IN"}, + {"DAC Left Input", "Right Data", "DAC IN"}, + {"DAC Left Input", "Mono", "DAC IN"}, + {"DAC Right Input", "Left Data", "DAC IN"}, + {"DAC Right Input", "Right Data", "DAC IN"}, + {"DAC Right Input", "Mono", "DAC IN"}, + {"DAC Left", NULL, "DAC Left Input"}, + {"DAC Right", NULL, "DAC Right Input"}, + + /* Mic input */ + {"MIC1LP P-Terminal", "FFR 10 Ohm", "MIC1LP"}, + {"MIC1LP P-Terminal", "FFR 20 Ohm", "MIC1LP"}, + {"MIC1LP P-Terminal", "FFR 40 Ohm", "MIC1LP"}, + {"MIC1RP P-Terminal", "FFR 10 Ohm", "MIC1RP"}, + {"MIC1RP P-Terminal", "FFR 20 Ohm", "MIC1RP"}, + {"MIC1RP P-Terminal", "FFR 40 Ohm", "MIC1RP"}, + {"MIC1LM P-Terminal", "FFR 10 Ohm", "MIC1LM"}, + {"MIC1LM P-Terminal", "FFR 20 Ohm", "MIC1LM"}, + {"MIC1LM P-Terminal", "FFR 40 Ohm", "MIC1LM"}, + + {"MIC1LM M-Terminal", "FFR 10 Ohm", "MIC1LM"}, + {"MIC1LM M-Terminal", "FFR 20 Ohm", "MIC1LM"}, + {"MIC1LM M-Terminal", "FFR 40 Ohm", "MIC1LM"}, + + {"MIC_GAIN_CTL", NULL, "MIC1LP P-Terminal"}, + {"MIC_GAIN_CTL", NULL, "MIC1RP P-Terminal"}, + {"MIC_GAIN_CTL", NULL, "MIC1LM P-Terminal"}, + {"MIC_GAIN_CTL", NULL, "MIC1LM M-Terminal"}, + + {"ADC", NULL, "MIC_GAIN_CTL"}, + {"MIC_GAIN_CTL", NULL, "Mic Bias"}, + + /* Clocks for DAC */ + {"DAC Left", NULL, "CODEC_CLK_IN" }, + {"DAC Right", NULL, "CODEC_CLK_IN"}, + + /* Left Output */ + {"Output Left", "From Left DAC", "DAC Left"}, + {"Output Left", "From MIC1LP", "MIC1LP"}, + {"Output Left", "From MIC1RP", "MIC1RP"}, + + /* Right Output */ + {"Output Right", "From Right DAC", "DAC Right"}, + {"Output Right", "From MIC1RP", "MIC1RP"}, + + /* HPL path */ + {"HP Left", "Switch", "Output Left"}, + {"HPL Driver", NULL, "HP Left"}, + {"HPL", NULL, "HPL Driver"}, + + /* HPR path */ + {"HP Right", "Switch", "Output Right"}, + {"HPR Driver", NULL, "HP Right"}, + {"HPR", NULL, "HPR Driver"}, +}; + +static const struct snd_soc_dapm_route +aic311x_audio_map[] = { + /* SP L path */ + {"SP Left", "Switch", "Output Left"}, + {"SPL ClassD", NULL, "SP Left"}, + {"SPL", NULL, "SPL ClassD"}, + + /* SP R path */ + {"SP Right", "Switch", "Output Right"}, + {"SPR ClassD", NULL, "SP Right"}, + {"SPR", NULL, "SPR ClassD"}, +}; + +static const struct snd_soc_dapm_route +aic310x_audio_map[] = { + /* SP L path */ + {"Speaker", "Switch", "Output Left"}, + {"SPK ClassD", NULL, "Speaker"}, + {"SPK", NULL, "SPK ClassD"}, +}; + +static int aic31xx_add_controls(struct snd_soc_codec *codec) +{ + int err = 0; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + + if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT) { + err = snd_soc_add_codec_controls( + codec, aic311x_snd_controls, + ARRAY_SIZE(aic311x_snd_controls)); + if (err < 0) + dev_dbg(codec->dev, "Invalid control\n"); + + } else { + err = snd_soc_add_codec_controls( + codec, aic310x_snd_controls, + ARRAY_SIZE(aic310x_snd_controls)); + if (err < 0) + dev_dbg(codec->dev, "Invalid Control\n"); + } + return 0; +} + +static int aic31xx_add_widgets(struct snd_soc_codec *codec) +{ + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT) { + ret = snd_soc_dapm_new_controls( + dapm, aic311x_dapm_widgets, + ARRAY_SIZE(aic311x_dapm_widgets)); + if (ret) + dev_err(codec->dev, + "Adding %d dapm widgets failed: %d\n", + ARRAY_SIZE(aic311x_dapm_widgets), ret); + ret = snd_soc_dapm_add_routes(dapm, aic311x_audio_map, + ARRAY_SIZE(aic311x_audio_map)); + if (ret) + dev_err(codec->dev, + "Adding %d DAPM routes failed: %d\n", + ARRAY_SIZE(aic311x_audio_map), ret); + } else { + ret = snd_soc_dapm_new_controls( + dapm, aic310x_dapm_widgets, + ARRAY_SIZE(aic310x_dapm_widgets)); + if (ret) + dev_err(codec->dev, + "Adding %d dapm widgets failed: %d\n", + ARRAY_SIZE(aic310x_dapm_widgets), ret); + ret = snd_soc_dapm_add_routes(dapm, aic310x_audio_map, + ARRAY_SIZE(aic310x_audio_map)); + if (ret) + dev_err(codec->dev, + "Adding %d DAPM routes failed: %d\n", + ARRAY_SIZE(aic310x_audio_map), ret); + } + + return 0; +} + +static int aic31xx_setup_pll(struct snd_soc_codec *codec, + struct snd_pcm_hw_params *params) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int bclk_n = 0; + int i; + + /* Use PLL as CODEC_CLKIN and DAC_CLK as BDIV_CLKIN */ + snd_soc_update_bits(codec, AIC31XX_CLKMUX, + AIC31XX_CODEC_CLKIN_MASK, AIC31XX_CODEC_CLKIN_PLL); + snd_soc_update_bits(codec, AIC31XX_IFACE2, + AIC31XX_BDIVCLK_MASK, AIC31XX_DAC2BCLK); + + for (i = 0; i < ARRAY_SIZE(aic31xx_divs); i++) { + if (aic31xx_divs[i].rate == params_rate(params) && + aic31xx_divs[i].mclk == aic31xx->sysclk) + break; + } + + if (i == ARRAY_SIZE(aic31xx_divs)) { + dev_err(codec->dev, "%s: Sampling rate %u not supported\n", + __func__, params_rate(params)); + return -EINVAL; + } + + if (aic31xx->codec_clk_on) { + int old = aic31xx->rate_div_line; + if (aic31xx_divs[i].pll_j != aic31xx_divs[old].pll_j || + aic31xx_divs[i].pll_d != aic31xx_divs[old].pll_d || + aic31xx_divs[i].p_val != aic31xx_divs[old].p_val) { + dev_err(codec->dev, + "%s: Can't reconfigure PLL while it is on.\n", + __func__); + return -EBUSY; + } + } else { + /* PLL configuration */ + snd_soc_update_bits(codec, AIC31XX_PLLPR, AIC31XX_PLL_MASK, + (aic31xx_divs[i].p_val << 4) | 0x01); + snd_soc_write(codec, AIC31XX_PLLJ, aic31xx_divs[i].pll_j); + + snd_soc_write(codec, AIC31XX_PLLDMSB, + aic31xx_divs[i].pll_d >> 8); + snd_soc_write(codec, AIC31XX_PLLDLSB, + aic31xx_divs[i].pll_d & 0xff); + } + + /* DAC dividers configuration */ + snd_soc_update_bits(codec, AIC31XX_NDAC, AIC31XX_PLL_MASK, + aic31xx_divs[i].ndac); + snd_soc_update_bits(codec, AIC31XX_MDAC, AIC31XX_PLL_MASK, + aic31xx_divs[i].mdac); + + snd_soc_write(codec, AIC31XX_DOSRMSB, aic31xx_divs[i].dosr >> 8); + snd_soc_write(codec, AIC31XX_DOSRLSB, aic31xx_divs[i].dosr & 0xff); + + /* ADC dividers configuration. Write reset value 1 if not used. */ + snd_soc_update_bits(codec, AIC31XX_NADC, AIC31XX_PLL_MASK, + aic31xx_divs[i].nadc ? aic31xx_divs[i].nadc : 1); + snd_soc_update_bits(codec, AIC31XX_MADC, AIC31XX_PLL_MASK, + aic31xx_divs[i].madc ? aic31xx_divs[i].madc : 1); + + snd_soc_write(codec, AIC31XX_AOSR, aic31xx_divs[i].aosr); + + /* Bit clock divider configuration. */ + bclk_n = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac) + / snd_soc_params_to_frame_size(params); + if (bclk_n == 0) { + dev_err(codec->dev, "%s: Not enough BLCK bandwidth\n", + __func__); + return -EINVAL; + } + + snd_soc_update_bits(codec, AIC31XX_BCLKN, + AIC31XX_PLL_MASK, bclk_n); + + aic31xx->rate_div_line = i; + + dev_dbg(codec->dev, + "pll %d.%04d/%d dosr %d n %d m %d aosr %d n %d m %d bclk_n %d\n", + aic31xx_divs[i].pll_j, aic31xx_divs[i].pll_d, + aic31xx_divs[i].p_val, aic31xx_divs[i].dosr, + aic31xx_divs[i].ndac, aic31xx_divs[i].mdac, + aic31xx_divs[i].aosr, aic31xx_divs[i].nadc, + aic31xx_divs[i].madc, bclk_n); + + return 0; +} + +static int aic31xx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *tmp) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + u8 data = 0; + + dev_dbg(codec->dev, "## %s: format %d rate %d\n", + __func__, params_format(params), params_rate(params)); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + data = (AIC31XX_WORD_LEN_20BITS << + AIC31XX_IFACE1_DATALEN_SHIFT); + break; + case SNDRV_PCM_FORMAT_S24_3LE: + data = (AIC31XX_WORD_LEN_24BITS << + AIC31XX_IFACE1_DATALEN_SHIFT); + break; + case SNDRV_PCM_FORMAT_S32_LE: + data = (AIC31XX_WORD_LEN_32BITS << + AIC31XX_IFACE1_DATALEN_SHIFT); + break; + default: + dev_err(codec->dev, "%s: Unsupported format %d\n", + __func__, params_format(params)); + return -EINVAL; + } + + snd_soc_update_bits(codec, AIC31XX_IFACE1, + AIC31XX_IFACE1_DATALEN_MASK, + data); + + return aic31xx_setup_pll(codec, params); +} + +static int aic31xx_dac_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + if (mute) { + snd_soc_update_bits(codec, AIC31XX_DACMUTE, + AIC31XX_DACMUTE_MASK, + AIC31XX_DACMUTE_MASK); + } else { + snd_soc_update_bits(codec, AIC31XX_DACMUTE, + AIC31XX_DACMUTE_MASK, 0x0); + } + + return 0; +} + +static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + u8 iface_reg1 = 0; + u8 iface_reg3 = 0; + u8 dsp_a_val = 0; + + dev_dbg(codec->dev, "## %s: fmt = 0x%x\n", __func__, fmt); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface_reg1 |= AIC31XX_BCLK_MASTER | AIC31XX_WCLK_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBS_CFS: + dev_err(codec->dev, "Unsupported DAI master/slave interface\n"); + return -EINVAL; + default: + dev_alert(codec->dev, "Invalid DAI master/slave interface\n"); + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_DSP_A: + dsp_a_val = 0x1; + case SND_SOC_DAIFMT_DSP_B: + /* NOTE: BCLKINV bit value 1 equas NB and 0 equals IB */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + iface_reg3 |= AIC31XX_BCLKINV_MASK; + break; + case SND_SOC_DAIFMT_IB_NF: + break; + default: + return -EINVAL; + } + iface_reg1 |= (AIC31XX_DSP_MODE << + AIC31XX_IFACE1_DATATYPE_SHIFT); + break; + case SND_SOC_DAIFMT_RIGHT_J: + iface_reg1 |= (AIC31XX_RIGHT_JUSTIFIED_MODE << + AIC31XX_IFACE1_DATATYPE_SHIFT); + break; + case SND_SOC_DAIFMT_LEFT_J: + iface_reg1 |= (AIC31XX_LEFT_JUSTIFIED_MODE << + AIC31XX_IFACE1_DATATYPE_SHIFT); + break; + default: + dev_err(codec->dev, "Invalid DAI interface format\n"); + return -EINVAL; + } + + snd_soc_update_bits(codec, AIC31XX_IFACE1, + AIC31XX_IFACE1_DATATYPE_MASK | + AIC31XX_IFACE1_MASTER_MASK, + iface_reg1); + snd_soc_update_bits(codec, AIC31XX_DATA_OFFSET, + AIC31XX_DATA_OFFSET_MASK, + dsp_a_val); + snd_soc_update_bits(codec, AIC31XX_IFACE2, + AIC31XX_BCLKINV_MASK, + iface_reg3); + + return 0; +} + +static int aic31xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int i; + + dev_dbg(codec->dev, "## %s: clk_id = %d, freq = %d, dir = %d\n", + __func__, clk_id, freq, dir); + + for (i = 0; aic31xx_divs[i].mclk != freq; i++) + if (i == ARRAY_SIZE(aic31xx_divs)) { + dev_err(aic31xx->dev, "%s: Unsupported frequency %d\n", + __func__, freq); + return -EINVAL; + } + + /* set clock on MCLK, BCLK, or GPIO1 as PLL input */ + snd_soc_update_bits(codec, AIC31XX_CLKMUX, AIC31XX_PLL_CLKIN_MASK, + clk_id << AIC31XX_PLL_CLKIN_SHIFT); + + aic31xx->sysclk = freq; + return 0; +} + +static int aic31xx_regulator_event(struct notifier_block *nb, + unsigned long event, void *data) +{ + struct aic31xx_disable_nb *disable_nb = + container_of(nb, struct aic31xx_disable_nb, nb); + struct aic31xx_priv *aic31xx = disable_nb->aic31xx; + + if (event & REGULATOR_EVENT_DISABLE) { + /* + * Put codec to reset and as at least one + * of the supplies was disabled + */ + dev_dbg(aic31xx->dev, "## %s: DISABLE received\n", __func__); + if (gpio_is_valid(aic31xx->pdata.gpio_reset)) + gpio_set_value(aic31xx->pdata.gpio_reset, 0); + regcache_mark_dirty(aic31xx->regmap); + } + + return 0; +} + +static int aic31xx_set_power(struct snd_soc_codec *codec, int power) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int ret; + + dev_dbg(codec->dev, "## %s: %d -> %d\n", __func__, + aic31xx->power, power); + if (aic31xx->power == power) + return 0; + + if (power) { + ret = regulator_bulk_enable(ARRAY_SIZE(aic31xx->supplies), + aic31xx->supplies); + if (ret) + return ret; + aic31xx->power = 1; + + if (gpio_is_valid(aic31xx->pdata.gpio_reset)) { + gpio_set_value(aic31xx->pdata.gpio_reset, 1); + udelay(100); + } + snd_soc_write(codec, AIC31XX_RESET, 0x01); + udelay(100); + regcache_cache_only(aic31xx->regmap, false); + ret = regcache_sync(aic31xx->regmap); + if (ret != 0) { + dev_err(codec->dev, + "Failed to restore cache: %d\n", ret); + regcache_cache_only(aic31xx->regmap, true); + regulator_bulk_disable(ARRAY_SIZE(aic31xx->supplies), + aic31xx->supplies); + return ret; + } + } else { + /* + * Do soft reset to this codec instance in order to clear + * possible VDD leakage currents in case the supply regulators + * remain on + */ + + snd_soc_write(codec, AIC31XX_RESET, 0x01); + regcache_cache_only(aic31xx->regmap, true); + + if (gpio_is_valid(aic31xx->pdata.gpio_reset)) + gpio_set_value(aic31xx->pdata.gpio_reset, 0); + aic31xx->power = 0; + ret = regulator_bulk_disable(ARRAY_SIZE(aic31xx->supplies), + aic31xx->supplies); + regcache_mark_dirty(aic31xx->regmap); + } + + return ret; +} + +static int aic31xx_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + dev_dbg(codec->dev, "## %s: %d -> %d\n", __func__, + codec->dapm.bias_level, level); + + switch (level) { + /* full On */ + case SND_SOC_BIAS_ON: + /* All power is driven by DAPM system*/ + break; + /* partial On */ + case SND_SOC_BIAS_PREPARE: + break; + /* Off, with power */ + case SND_SOC_BIAS_STANDBY: + aic31xx_set_power(codec, 1); + break; + /* Off, without power */ + case SND_SOC_BIAS_OFF: + aic31xx_set_power(codec, 0); + break; + } + codec->dapm.bias_level = level; + + return 0; +} + + +static int aic31xx_suspend(struct snd_soc_codec *codec) +{ + aic31xx_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int aic31xx_resume(struct snd_soc_codec *codec) +{ + aic31xx_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +static int aic31xx_codec_probe(struct snd_soc_codec *codec) +{ + int ret = 0; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int i; + + dev_dbg(aic31xx->dev, "## %s\n", __func__); + + aic31xx = snd_soc_codec_get_drvdata(codec); + codec->control_data = aic31xx->regmap; + + aic31xx->codec = codec; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + + if (ret != 0) { + dev_err(codec->dev, "snd_soc_codec_set_cache_io failed %d\n", + ret); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) { + aic31xx->disable_nb[i].nb.notifier_call = + aic31xx_regulator_event; + aic31xx->disable_nb[i].aic31xx = aic31xx; + ret = regulator_register_notifier(aic31xx->supplies[i].consumer, + &aic31xx->disable_nb[i].nb); + if (ret) { + dev_err(codec->dev, + "Failed to request regulator notifier: %d\n", + ret); + return ret; + } + } + + regcache_cache_only(aic31xx->regmap, true); + regcache_mark_dirty(aic31xx->regmap); + + aic31xx_add_controls(codec); + aic31xx_add_widgets(codec); + + return ret; +} + +static int aic31xx_codec_remove(struct snd_soc_codec *codec) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int i; + /* power down chip */ + aic31xx_set_bias_level(codec, SND_SOC_BIAS_OFF); + + for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) + regulator_unregister_notifier(aic31xx->supplies[i].consumer, + &aic31xx->disable_nb[i].nb); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_driver_aic31xx = { + .probe = aic31xx_codec_probe, + .remove = aic31xx_codec_remove, + .suspend = aic31xx_suspend, + .resume = aic31xx_resume, + .set_bias_level = aic31xx_set_bias_level, + .controls = aic31xx_snd_controls, + .num_controls = ARRAY_SIZE(aic31xx_snd_controls), + .dapm_widgets = aic31xx_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(aic31xx_dapm_widgets), + .dapm_routes = aic31xx_audio_map, + .num_dapm_routes = ARRAY_SIZE(aic31xx_audio_map), +}; + +static 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, + .digital_mute = aic31xx_dac_mute, +}; + +static struct snd_soc_dai_driver aic31xx_dai_driver[] = { + { + .name = "tlv320aic31xx-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AIC31XX_RATES, + .formats = AIC31XX_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AIC31XX_RATES, + .formats = AIC31XX_FORMATS, + }, + .ops = &aic31xx_dai_ops, + } +}; + +#if defined(CONFIG_OF) +static const struct of_device_id tlv320aic31xx_of_match[] = { + { .compatible = "ti,tlv320aic310x" }, + { .compatible = "ti,tlv320aic311x" }, + { .compatible = "ti,tlv320aic3100" }, + { .compatible = "ti,tlv320aic3110" }, + { .compatible = "ti,tlv320aic3120" }, + { .compatible = "ti,tlv320aic3111" }, + {}, +}; +MODULE_DEVICE_TABLE(of, tlv320aic31xx_of_match); + +static void aic31xx_pdata_from_of(struct aic31xx_priv *aic31xx) +{ + struct device_node *np = aic31xx->dev->of_node; + unsigned int value = 0; + int ret; + + of_property_read_u32(np, "ai31xx-micbias-vg", &value); + switch (value) { + case 1: + aic31xx->pdata.micbias_vg = AIC31XX_MICBIAS_2_0V; + break; + case 2: + aic31xx->pdata.micbias_vg = AIC31XX_MICBIAS_2_5V; + break; + case 3: + aic31xx->pdata.micbias_vg = AIC31XX_MICBIAS_AVDDV; + break; + default: + dev_err(aic31xx->dev, + "Bad ai31xx-micbias-vg value %d DT\n", + value); + /* fallthrough */ + case 0: + aic31xx->pdata.micbias_vg = AIC31XX_MICBIAS_OFF; + } + + ret = of_get_named_gpio(np, "gpio-reset", 0); + if (ret > 0) + aic31xx->pdata.gpio_reset = ret; +} +#else /* CONFIG_OF */ +static void aic31xx_pdata_from_of(struct aic31xx_priv *aic31xx) +{ +} +#endif /* CONFIG_OF */ + +void aic31xx_device_init(struct aic31xx_priv *aic31xx) +{ + int ret, i; + + dev_set_drvdata(aic31xx->dev, aic31xx); + + if (dev_get_platdata(aic31xx->dev)) + memcpy(&aic31xx->pdata, dev_get_platdata(aic31xx->dev), + sizeof(aic31xx->pdata)); + else if (aic31xx->dev->of_node) + aic31xx_pdata_from_of(aic31xx); + + if (aic31xx->pdata.gpio_reset) { + ret = devm_gpio_request_one(aic31xx->dev, + aic31xx->pdata.gpio_reset, + GPIOF_OUT_INIT_HIGH, + "aic31xx-reset-pin"); + if (ret < 0) { + dev_err(aic31xx->dev, "not able to acquire gpio\n"); + return; + } + } + + for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) + aic31xx->supplies[i].supply = aic31xx_supply_names[i]; + + ret = devm_regulator_bulk_get(aic31xx->dev, + ARRAY_SIZE(aic31xx->supplies), + aic31xx->supplies); + if (ret != 0) + dev_err(aic31xx->dev, "Failed to request supplies: %d\n", ret); + +} + +static int aic31xx_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct aic31xx_priv *aic31xx; + int ret; + const struct regmap_config *regmap_config; + + dev_dbg(&i2c->dev, "## %s: %s codec_type = %d\n", __func__, + id->name, (int) id->driver_data); + + regmap_config = &aic31xx_i2c_regmap; + + aic31xx = devm_kzalloc(&i2c->dev, sizeof(*aic31xx), GFP_KERNEL); + if (aic31xx == NULL) + return -ENOMEM; + + aic31xx->regmap = devm_regmap_init_i2c(i2c, regmap_config); + + if (IS_ERR(aic31xx->regmap)) { + ret = PTR_ERR(aic31xx->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + aic31xx->dev = &i2c->dev; + + aic31xx->pdata.codec_type = id->driver_data; + + aic31xx_device_init(aic31xx); + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx, + aic31xx_dai_driver, + ARRAY_SIZE(aic31xx_dai_driver)); + + return ret; +} + +static int aic31xx_i2c_remove(struct i2c_client *i2c) +{ + struct aic31xx_priv *aic31xx = dev_get_drvdata(&i2c->dev); + + kfree(aic31xx); + return 0; +} + +static const struct i2c_device_id aic31xx_i2c_id[] = { + { "tlv320aic310x", AIC3100 }, + { "tlv320aic311x", AIC3110 }, + { "tlv320aic3100", AIC3100 }, + { "tlv320aic3110", AIC3110 }, + { "tlv320aic3120", AIC3120 }, + { "tlv320aic3111", AIC3111 }, + { } +}; +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, + .remove = (aic31xx_i2c_remove), + .id_table = aic31xx_i2c_id, +}; + +module_i2c_driver(aic31xx_i2c_driver); + +MODULE_DESCRIPTION("ASoC TLV320AIC3111 codec driver"); +MODULE_AUTHOR("Jyri Sarha"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h new file mode 100644 index 0000000..28d1326 --- /dev/null +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -0,0 +1,265 @@ +/* + * ALSA SoC TLV320AIC31XX codec driver + * + * Copyright (C) 2013 Texas Instruments, Inc. + * + * This package 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 PACKAGE IS PROVIDED ``AS IS'' AND WITHOUT ANY EXPRESS OR + * IMPLIED WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED + * WARRANTIES OF MERCHANTIBILITY AND FITNESS FOR A PARTICULAR PURPOSE. + * + */ +#ifndef _TLV320AIC31XX_H +#define _TLV320AIC31XX_H + +#define AIC31XX_RATES SNDRV_PCM_RATE_8000_192000 + +#define AIC31XX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + + +#define AIC31XX_STEREO_CLASS_D_BIT 0x1 +#define AIC31XX_MINIDSP_BIT 0x2 + +enum aic31xx_type { + AIC3100 = 0, + AIC3110 = AIC31XX_STEREO_CLASS_D_BIT, + AIC3120 = AIC31XX_MINIDSP_BIT, + AIC3111 = (AIC31XX_STEREO_CLASS_D_BIT | AIC31XX_MINIDSP_BIT), +}; + +enum aic31xx_micbias_voltage { + AIC31XX_MICBIAS_OFF = 0, + AIC31XX_MICBIAS_2_0V = 1, + AIC31XX_MICBIAS_2_5V = 2, + AIC31XX_MICBIAS_AVDDV = 3, +}; + +struct aic31xx_pdata { + enum aic31xx_type codec_type; + unsigned int gpio_reset; + enum aic31xx_micbias_voltage micbias_vg; +}; + +/* Page Control Register */ +#define AIC31XX_PAGECTL 0x00 + +/* Page 0 Registers */ +/* Software reset register */ +#define AIC31XX_RESET 0x01 +/* OT FLAG register */ +#define AIC31XX_OT_FLAG 0x03 +/* Clock clock Gen muxing, Multiplexers*/ +#define AIC31XX_CLKMUX 0x04 +/* PLL P and R-VAL register */ +#define AIC31XX_PLLPR 0x05 +/* PLL J-VAL register */ +#define AIC31XX_PLLJ 0x06 +/* PLL D-VAL MSB register */ +#define AIC31XX_PLLDMSB 0x07 +/* PLL D-VAL LSB register */ +#define AIC31XX_PLLDLSB 0x08 +/* DAC NDAC_VAL register*/ +#define AIC31XX_NDAC 0x0B +/* DAC MDAC_VAL register */ +#define AIC31XX_MDAC 0x0C +/* DAC OSR setting register 1, MSB value */ +#define AIC31XX_DOSRMSB 0x0D +/* DAC OSR setting register 2, LSB value */ +#define AIC31XX_DOSRLSB 0x0E +#define AIC31XX_MINI_DSP_INPOL 0x10 +/* Clock setting register 8, PLL */ +#define AIC31XX_NADC 0x12 +/* Clock setting register 9, PLL */ +#define AIC31XX_MADC 0x13 +/* ADC Oversampling (AOSR) Register */ +#define AIC31XX_AOSR 0x14 +/* Clock setting register 9, Multiplexers */ +#define AIC31XX_CLKOUTMUX 0x19 +/* Clock setting register 10, CLOCKOUT M divider value */ +#define AIC31XX_CLKOUTMVAL 0x1A +/* Audio Interface Setting Register 1 */ +#define AIC31XX_IFACE1 0x1B +/* Audio Data Slot Offset Programming */ +#define AIC31XX_DATA_OFFSET 0x1C +/* Audio Interface Setting Register 2 */ +#define AIC31XX_IFACE2 0x1D +/* Clock setting register 11, BCLK N Divider */ +#define AIC31XX_BCLKN 0x1E +/* Audio Interface Setting Register 3, Secondary Audio Interface */ +#define AIC31XX_IFACESEC1 0x1F +/* Audio Interface Setting Register 4 */ +#define AIC31XX_IFACESEC2 0x20 +/* Audio Interface Setting Register 5 */ +#define AIC31XX_IFACESEC3 0x21 +/* I2C Bus Condition */ +#define AIC31XX_I2C 0x22 +/* ADC FLAG */ +#define AIC31XX_ADCFLAG 0x24 +/* DAC Flag Registers */ +#define AIC31XX_DACFLAG1 0x25 +#define AIC31XX_DACFLAG2 0x26 +/* Sticky Interrupt flag (overflow) */ +#define AIC31XX_OFFLAG 0x27 +/* Sticy DAC Interrupt flags */ +#define AIC31XX_INTRDACFLAG 0x2C +/* Sticy ADC Interrupt flags */ +#define AIC31XX_INTRADCFLAG 0x2D +/* DAC Interrupt flags 2 */ +#define AIC31XX_INTRDACFLAG2 0x2E +/* ADC Interrupt flags 2 */ +#define AIC31XX_INTRADCFLAG2 0x2F +/* INT1 interrupt control */ +#define AIC31XX_INT1CTRL 0x30 +/* INT2 interrupt control */ +#define AIC31XX_INT2CTRL 0x31 +/* GPIO1 control */ +#define AIC31XX_GPIO1 0x33 + +#define AIC31XX_DACPRB 0x3C +/* ADC Instruction Set Register */ +#define AIC31XX_ADCPRB 0x3D +/* DAC channel setup register */ +#define AIC31XX_DACSETUP 0x3F +/* DAC Mute and volume control register */ +#define AIC31XX_DACMUTE 0x40 +/* Left DAC channel digital volume control */ +#define AIC31XX_LDACVOL 0x41 +/* Right DAC channel digital volume control */ +#define AIC31XX_RDACVOL 0x42 +/* Headset detection */ +#define AIC31XX_HSDETECT 0x43 +/* ADC Digital Mic */ +#define AIC31XX_ADCSETUP 0x51 +/* ADC Digital Volume Control Fine Adjust */ +#define AIC31XX_ADCFGA 0x52 +/* ADC Digital Volume Control Coarse Adjust */ +#define AIC31XX_ADCVOL 0x53 + + +/* Page 1 Registers */ +/* Headphone drivers */ +#define AIC31XX_HPDRIVER 0x9F +/* Class-D Speakear Amplifier */ +#define AIC31XX_SPKAMP 0xA0 +/* HP Output Drivers POP Removal Settings */ +#define AIC31XX_HPPOP 0xA1 +/* Output Driver PGA Ramp-Down Period Control */ +#define AIC31XX_SPPGARAMP 0xA2 +/* DAC_L and DAC_R Output Mixer Routing */ +#define AIC31XX_DACMIXERROUTE 0xA3 +/* Left Analog Vol to HPL */ +#define AIC31XX_LANALOGHPL 0xA4 +/* Right Analog Vol to HPR */ +#define AIC31XX_RANALOGHPR 0xA5 +/* Left Analog Vol to SPL */ +#define AIC31XX_LANALOGSPL 0xA6 +/* Right Analog Vol to SPR */ +#define AIC31XX_RANALOGSPR 0xA7 +/* HPL Driver */ +#define AIC31XX_HPLGAIN 0xA8 +/* HPR Driver */ +#define AIC31XX_HPRGAIN 0xA9 +/* SPL Driver */ +#define AIC31XX_SPLGAIN 0xAA +/* SPR Driver */ +#define AIC31XX_SPRGAIN 0xAB +/* HP Driver Control */ +#define AIC31XX_HPCONTROL 0xAC +/* MIC Bias Control */ +#define AIC31XX_MICBIAS 0xAE +/* MIC PGA*/ +#define AIC31XX_MICPGA 0xAF +/* Delta-Sigma Mono ADC Channel Fine-Gain Input Selection for P-Terminal */ +#define AIC31XX_MICPGAPI 0xB0 +/* ADC Input Selection for M-Terminal */ +#define AIC31XX_MICPGAMI 0xB1 +/* Input CM Settings */ +#define AIC31XX_MICPGACM 0xB2 + +/* Bits, masks and shifts */ + +/* AIC31XX_CLKMUX */ +#define AIC31XX_PLL_CLKIN_MASK 0x0c +#define AIC31XX_PLL_CLKIN_SHIFT 2 +#define AIC31XX_PLL_CLKIN_MCLK 0 +#define AIC31XX_CODEC_CLKIN_MASK 0x03 +#define AIC31XX_CODEC_CLKIN_SHIFT 0 +#define AIC31XX_CODEC_CLKIN_PLL 3 +#define AIC31XX_CODEC_CLKIN_BCLK 1 + +/* AIC31XX_PLLPR, AIC31XX_NDAC, AIC31XX_MDAC, AIC31XX_NADC, AIC31XX_MADC, + AIC31XX_BCLKN */ +#define AIC31XX_PLL_MASK 0x7f +#define AIC31XX_PM_MASK 0x80 + +/* AIC31XX_IFACE1 */ +#define AIC31XX_WORD_LEN_16BITS 0x00 +#define AIC31XX_WORD_LEN_20BITS 0x01 +#define AIC31XX_WORD_LEN_24BITS 0x02 +#define AIC31XX_WORD_LEN_32BITS 0x03 +#define AIC31XX_IFACE1_DATALEN_MASK 0x30 +#define AIC31XX_IFACE1_DATALEN_SHIFT (4) +#define AIC31XX_IFACE1_DATATYPE_MASK 0xC0 +#define AIC31XX_IFACE1_DATATYPE_SHIFT (6) +#define AIC31XX_I2S_MODE 0x00 +#define AIC31XX_DSP_MODE 0x01 +#define AIC31XX_RIGHT_JUSTIFIED_MODE 0x02 +#define AIC31XX_LEFT_JUSTIFIED_MODE 0x03 +#define AIC31XX_IFACE1_MASTER_MASK 0x0C +#define AIC31XX_BCLK_MASTER 0x08 +#define AIC31XX_WCLK_MASTER 0x04 + +/* AIC31XX_DATA_OFFSET */ +#define AIC31XX_DATA_OFFSET_MASK 0xFF + +/* AIC31XX_IFACE2 */ +#define AIC31XX_BCLKINV_MASK 0x08 +#define AIC31XX_BDIVCLK_MASK 0x03 +#define AIC31XX_DAC2BCLK 0x00 +#define AIC31XX_DACMOD2BCLK 0x01 +#define AIC31XX_ADC2BCLK 0x02 +#define AIC31XX_ADCMOD2BCLK 0x03 + +/* AIC31XX_ADCFLAG */ +#define AIC31XX_ADCPWRSTATUS_MASK 0x40 + +/* AIC31XX_DACFLAG1 */ +#define AIC31XX_LDACPWRSTATUS_MASK 0x80 +#define AIC31XX_RDACPWRSTATUS_MASK 0x08 +#define AIC31XX_HPLDRVPWRSTATUS_MASK 0x20 +#define AIC31XX_HPRDRVPWRSTATUS_MASK 0x02 +#define AIC31XX_SPLDRVPWRSTATUS_MASK 0x10 +#define AIC31XX_SPRDRVPWRSTATUS_MASK 0x01 + +/* AIC31XX_INTRDACFLAG */ +#define AIC31XX_HPSCDETECT_MASK 0x80 +#define AIC31XX_BUTTONPRESS_MASK 0x20 +#define AIC31XX_HSPLUG_MASK 0x10 +#define AIC31XX_LDRCTHRES_MASK 0x08 +#define AIC31XX_RDRCTHRES_MASK 0x04 +#define AIC31XX_DACSINT_MASK 0x02 +#define AIC31XX_DACAINT_MASK 0x01 + +/* AIC31XX_INT1CTRL */ +#define AIC31XX_HSPLUGDET_MASK 0x80 +#define AIC31XX_BUTTONPRESSDET_MASK 0x40 +#define AIC31XX_DRCTHRES_MASK 0x20 +#define AIC31XX_AGCNOISE_MASK 0x10 +#define AIC31XX_OC_MASK 0x08 +#define AIC31XX_ENGINE_MASK 0x04 + +/* AIC31XX_DACSETUP */ +#define AIC31XX_SOFTSTEP_MASK 0x03 + +/* AIC31XX_DACMUTE */ +#define AIC31XX_DACMUTE_MASK 0x0C + +/* AIC31XX_MICBIAS */ +#define AIC31XX_MICBIAS_MASK 0x03 +#define AIC31XX_MICBIAS_SHIFT 0 + +#endif /* _TLV320AIC31XX_H */
On Wed, Feb 26, 2014 at 11:14:25AM +0200, Jyri Sarha wrote:
This commit adds a bare bones driver support for TLV320AIC31XX family audio codecs. The driver adds basic stereo playback trough headphone and speaker outputs and mono capture trough microphone inputs.
Always CC maintainers on patches please.
+static bool aic31xx_volatile(struct device *dev, unsigned int reg) +{
- if (aic31xx_precious(dev, reg) || aic31xx_real_volatile(dev, reg))
return true;
- switch (reg) {
- case AIC31XX_PAGECTL: /* regmap implementation requires this */
- case AIC31XX_RESET: /* always clears after write */
return true;
- }
- return false;
+}
This is more complex than you need, just write a standalone volatile function with some comments in it.
+static const struct regmap_range_cfg aic31xx_ranges[] = {
- {
.name = "codec-regmap",
Make this meaningful or omit it.
+#define AIC31XX_NUM_SUPPLIES 6 +static const char * const aic31xx_supply_names[] = {
Use the define in the array size to help check everything lines up.
+static const char * const ldac_in_text[] = {
- "off", "Left Data", "Right Data", "Mono"
+};
Off not off - be consistent both internally and with the general style.
+static const struct snd_kcontrol_new aic311x_snd_controls[] = {
- SOC_DOUBLE_R("SP Driver Playback Switch", AIC31XX_SPLGAIN,
AIC31XX_SPRGAIN, 2, 1, 0),
SP?
- if (!strcmp(w->name, "DAC Left")) {
mask = AIC31XX_LDACPWRSTATUS_MASK;
There's no overlap with the enable bits? In general it would seem nicer to have a switch based on the register bits used for the enable rather than the tree of string comparisions but it's probably not that big a deal overall.
dev_err(w->codec->dev, "Unknown widget '%s' calling %s/n",
w->name, __func__);
return -1;
Return real error codes.
- if (event == SND_SOC_DAPM_POST_PMU)
return aic31xx_wait_bits(aic31xx, reg, mask, mask, 5000, 100);
- else if (event == SND_SOC_DAPM_POST_PMD)
return aic31xx_wait_bits(aic31xx, reg, mask, 0, 5000, 100);
switch.
- SND_SOC_DAPM_DAC_E("DAC Right", "DAC Right Input",
AIC31XX_DACSETUP, 6, 0, aic31xx_dapm_power_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
Don't use stream names, use DAPM to route from the AIF to the widgets.
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
break;
params_width().
AIC31XX_IFACE1_DATALEN_MASK,
data);
- return aic31xx_setup_pll(codec, params);
This is going to mean that you have a symmetry constraint I think, if you try to set up different rates for playback and capture presumably things will break. Setting symmetric_rates will tell applications about that.
- case SND_SOC_DAIFMT_CBS_CFM:
- case SND_SOC_DAIFMT_CBM_CFS:
- case SND_SOC_DAIFMT_CBS_CFS:
dev_err(codec->dev, "Unsupported DAI master/slave interface\n");
return -EINVAL;
- default:
dev_alert(codec->dev, "Invalid DAI master/slave interface\n");
return -EINVAL;
Just have a default.
- for (i = 0; aic31xx_divs[i].mclk != freq; i++)
if (i == ARRAY_SIZE(aic31xx_divs)) {
dev_err(aic31xx->dev, "%s: Unsupported frequency %d\n",
__func__, freq);
return -EINVAL;
}
Braces around the for () too please.
+static int aic31xx_set_power(struct snd_soc_codec *codec, int power) +{
- struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
- int ret;
- dev_dbg(codec->dev, "## %s: %d -> %d\n", __func__,
aic31xx->power, power);
- if (aic31xx->power == power)
return 0;
This looks like you need sensible refcounting somewhere? Implementing this as the standard PM operations may be sensible.
if (gpio_is_valid(aic31xx->pdata.gpio_reset)) {
gpio_set_value(aic31xx->pdata.gpio_reset, 1);
udelay(100);
}
snd_soc_write(codec, AIC31XX_RESET, 0x01);
udelay(100);
regcache_cache_only(aic31xx->regmap, false);
You should be coming out of cache only mode before you issue the reset write. Is it not sensible to skip that if you have a physical reset line?
snd_soc_write(codec, AIC31XX_RESET, 0x01);
regcache_cache_only(aic31xx->regmap, true);
if (gpio_is_valid(aic31xx->pdata.gpio_reset))
gpio_set_value(aic31xx->pdata.gpio_reset, 0);
Likewise here. Also if you do reset the CODEC then the dance you did with the regulator notifiers becomes pointless - the goal with that is to avoid the need to resync the cache if the CODEC wasn't actually reset by a power cycle but since the CODEC is going to be explicitly reset before it's powered off there's no benefit.
- switch (level) {
- /* full On */
- case SND_SOC_BIAS_ON:
/* All power is driven by DAPM system*/
break;
- /* partial On */
Coding style, please.
On 03/03/2014 08:34 AM, Mark Brown wrote:
On Wed, Feb 26, 2014 at 11:14:25AM +0200, Jyri Sarha wrote:
This commit adds a bare bones driver support for TLV320AIC31XX family audio codecs. The driver adds basic stereo playback trough headphone and speaker outputs and mono capture trough microphone inputs.
Always CC maintainers on patches please.
Will do.
+static bool aic31xx_volatile(struct device *dev, unsigned int reg) +{
- if (aic31xx_precious(dev, reg) || aic31xx_real_volatile(dev, reg))
return true;
- switch (reg) {
- case AIC31XX_PAGECTL: /* regmap implementation requires this */
- case AIC31XX_RESET: /* always clears after write */
return true;
- }
- return false;
+}
This is more complex than you need, just write a standalone volatile function with some comments in it.
Replaced with straight forward aic31xx_volatile() and aic31xx_writable() functions.
+static const struct regmap_range_cfg aic31xx_ranges[] = {
- {
.name = "codec-regmap",
Make this meaningful or omit it.
Removed.
+#define AIC31XX_NUM_SUPPLIES 6 +static const char * const aic31xx_supply_names[] = {
Use the define in the array size to help check everything lines up.
Done.
+static const char * const ldac_in_text[] = {
- "off", "Left Data", "Right Data", "Mono"
+};
Off not off - be consistent both internally and with the general style.
Done.
+static const struct snd_kcontrol_new aic311x_snd_controls[] = {
- SOC_DOUBLE_R("SP Driver Playback Switch", AIC31XX_SPLGAIN,
AIC31XX_SPRGAIN, 2, 1, 0),
SP?
"SP" replaced with "Speaker", also in DAPM switches.
- if (!strcmp(w->name, "DAC Left")) {
mask = AIC31XX_LDACPWRSTATUS_MASK;
There's no overlap with the enable bits? In general it would seem nicer to have a switch based on the register bits used for the enable rather than the tree of string comparisions but it's probably not that big a deal overall.
Not with bits, but with regs. I decided this structure would be easier to implement and understand than having a switch(w->reg) with nested switch(w->shift) cases. I replaced the string comparisons with a single switch case using a binary combination of w->reg and w->shift.
dev_err(w->codec->dev, "Unknown widget '%s' calling %s/n",
w->name, __func__);
return -1; switch (w->shift) {
case 7: mask = AIC31XX_LDACPWRSTATUS_MASK;
Return real error codes.
Changed to -EINVAL.
- if (event == SND_SOC_DAPM_POST_PMU)
return aic31xx_wait_bits(aic31xx, reg, mask, mask, 5000, 100);
- else if (event == SND_SOC_DAPM_POST_PMD)
return aic31xx_wait_bits(aic31xx, reg, mask, 0, 5000, 100);
switch.
Done.
- SND_SOC_DAPM_DAC_E("DAC Right", "DAC Right Input",
AIC31XX_DACSETUP, 6, 0, aic31xx_dapm_power_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
Don't use stream names, use DAPM to route from the AIF to the widgets.
Renamed DAC stnames to "Left Playback" and "Right Playback".
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
break;
params_width().
Done.
AIC31XX_IFACE1_DATALEN_MASK,
data);
- return aic31xx_setup_pll(codec, params);
This is going to mean that you have a symmetry constraint I think, if you try to set up different rates for playback and capture presumably things will break. Setting symmetric_rates will tell applications about that.
Done.
- case SND_SOC_DAIFMT_CBS_CFM:
- case SND_SOC_DAIFMT_CBM_CFS:
- case SND_SOC_DAIFMT_CBS_CFS:
dev_err(codec->dev, "Unsupported DAI master/slave interface\n");
return -EINVAL;
- default:
dev_alert(codec->dev, "Invalid DAI master/slave interface\n");
return -EINVAL;
Just have a default.
Done.
- for (i = 0; aic31xx_divs[i].mclk != freq; i++)
if (i == ARRAY_SIZE(aic31xx_divs)) {
dev_err(aic31xx->dev, "%s: Unsupported frequency %d\n",
__func__, freq);
return -EINVAL;
}
Braces around the for () too please.
Done.
+static int aic31xx_set_power(struct snd_soc_codec *codec, int power) +{
- struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
- int ret;
- dev_dbg(codec->dev, "## %s: %d -> %d\n", __func__,
aic31xx->power, power);
- if (aic31xx->power == power)
return 0;
This looks like you need sensible refcounting somewhere? Implementing this as the standard PM operations may be sensible.
Not really. This was just avoid power on sequence when bias level goes from SND_SOC_BIAS_PREPARE back to SND_SOC_BIAS_STANDBY. I'll change this to a "if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)" in set_bias_level like other codecs do.
if (gpio_is_valid(aic31xx->pdata.gpio_reset)) {
gpio_set_value(aic31xx->pdata.gpio_reset, 1);
udelay(100);
}
snd_soc_write(codec, AIC31XX_RESET, 0x01);
udelay(100);
regcache_cache_only(aic31xx->regmap, false);
You should be coming out of cache only mode before you issue the reset write. Is it not sensible to skip that if you have a physical reset line?
snd_soc_write(codec, AIC31XX_RESET, 0x01);
regcache_cache_only(aic31xx->regmap, true);
if (gpio_is_valid(aic31xx->pdata.gpio_reset))
gpio_set_value(aic31xx->pdata.gpio_reset, 0);
Likewise here. Also if you do reset the CODEC then the dance you did with the regulator notifiers becomes pointless - the goal with that is to avoid the need to resync the cache if the CODEC wasn't actually reset by a power cycle but since the CODEC is going to be explicitly reset before it's powered off there's no benefit.
I rewrote the bias-level/power logic a bit. It should now look more like the other codecs. I hope you like this version better.
- switch (level) {
- /* full On */
- case SND_SOC_BIAS_ON:
/* All power is driven by DAPM system*/
break;
- /* partial On */
Coding style, please.
Removed the redundant badly indented comments.
Fixed also the Kconfig and Makefile ordering and added dt-include for micbias and changed the default to 2.0V. All are now squashed to a single patch.
In addition to addressing your comments I also added an internal ADC input and DAC output for codec to turn on at playback/capture start even if the alsamixers are not set correctly.
I'll mail the patches shortly.
Best regards, Jyri
Signed-off-by: Jyri Sarha jsarha@ti.com --- sound/soc/codecs/Kconfig | 4 ++++ sound/soc/codecs/Makefile | 2 ++ 2 files changed, 6 insertions(+)
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 983d087a..ead9dc5 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -74,6 +74,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC32X4 if I2C + select SND_SOC_TLV320AIC31XX if I2C select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C @@ -364,6 +365,9 @@ config SND_SOC_TLV320AIC26 config SND_SOC_TLV320AIC32X4 tristate
+config SND_SOC_TLV320AIC31XX + tristate + config SND_SOC_TLV320AIC3X tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bc12676..23f8042 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -64,6 +64,7 @@ snd-soc-stac9766-objs := stac9766.o snd-soc-tas5086-objs := tas5086.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o +snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o snd-soc-tlv320dac33-objs := tlv320dac33.o @@ -194,6 +195,7 @@ obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o +obj-$(CONFIG_SND_SOC_TLV320AIC31XX) += snd-soc-tlv320aic31xx.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o
On 02/26/2014 11:14 AM, Jyri Sarha wrote:
Signed-off-by: Jyri Sarha jsarha@ti.com
sound/soc/codecs/Kconfig | 4 ++++ sound/soc/codecs/Makefile | 2 ++ 2 files changed, 6 insertions(+)
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 983d087a..ead9dc5 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -74,6 +74,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC32X4 if I2C
- select SND_SOC_TLV320AIC31XX if I2C
You should keep this file ordered.
select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C @@ -364,6 +365,9 @@ config SND_SOC_TLV320AIC26 config SND_SOC_TLV320AIC32X4 tristate
+config SND_SOC_TLV320AIC31XX
tristate
config SND_SOC_TLV320AIC3X tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bc12676..23f8042 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -64,6 +64,7 @@ snd-soc-stac9766-objs := stac9766.o snd-soc-tas5086-objs := tas5086.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o +snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o
Same here.
snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o snd-soc-tlv320dac33-objs := tlv320dac33.o @@ -194,6 +195,7 @@ obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o +obj-$(CONFIG_SND_SOC_TLV320AIC31XX) += snd-soc-tlv320aic31xx.o
and here
obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o
On Wed, Feb 26, 2014 at 11:14:26AM +0200, Jyri Sarha wrote:
--- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -74,6 +74,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC32X4 if I2C
- select SND_SOC_TLV320AIC31XX if I2C select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C
Keep these files sorted and don't send the build infrastructure as a separate patch unless the driver as a whole is split into multiple patches - there is no value unless the driver is being built up over several patches, it just raises a review question on the driver patch.
Initial version of tlv320aic31xx device tree bindings document.
Signed-off-by: Jyri Sarha jsarha@ti.com --- .../devicetree/bindings/sound/tlv320aic31xx.txt | 57 ++++++++++++++++++++ 1 file changed, 57 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt
diff --git a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt new file mode 100644 index 0000000..a512b4e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt @@ -0,0 +1,57 @@ +Texas Instruments - tlv320aic31xx Codec module + +The tlv320aic31xx serial control bus communicates through I2C protocols + +Required properties: + +- compatible - "string" - One of: + "ti,tlv320aic310x" - Generic TLV320AIC31xx with mono speaker amp + "ti,tlv320aic311x" - Generic TLV320AIC31xx with stereo speaker amp + "ti,tlv320aic3100" - TLV320AIC3100 (mono speaker amp, no MiniDSP) + "ti,tlv320aic3110" - TLV320AIC3110 (stereo speaker amp, no MiniDSP) + "ti,tlv320aic3120" - TLV320AIC3120 (mono speaker amp, MiniDSP) + "ti,tlv320aic3111" - TLV320AIC3111 (stereo speaker amp, MiniDSP) + +- reg - <int> - I2C slave address + + +Optional properties: + +- gpio-reset - gpio pin number used for codec reset +- ai31xx-micbias-vg - MicBias Voltage required. + 1 - MICBIAS output is powered to 2.0V, + 2 - MICBIAS output is powered to 2.5V, + 3 - MICBIAS output is connected to AVDD, + If this node is not mentioned or if the value is incorrect, then MicBias + is powered down. +- HPVDD-supply, SPRVDD-supply, SPLVDD-supply, AVDD-supply, IOVDD-supply, + DVDD-supply : power supplies for the device as covered in + Documentation/devicetree/bindings/regulator/regulator.txt + +CODEC output pins: + * HPL + * HPR + * SPL, devices with stereo speaker amp + * SPR, devices with stereo speaker amp + * SPK, devices with mono speaker amp + +CODEC input pins: + * MIC1LP + * MIC1RP + * MIC1LM + +The pins can be used in referring sound node's audio-routing property. + +Example: + +tlv320aic31xx: tlv320aic31xx@18 { + compatible = "ti,tlv320aic311x"; + reg = <0x18>; + + HPVDD-supply = <®ulator>; + SPRVDD-supply = <®ulator>; + SPLVDD-supply = <®ulator>; + AVDD-supply = <®ulator>; + IOVDD-supply = <®ulator>; + DVDD-supply = <®ulator>; +};
On Wed, Feb 26, 2014 at 11:14:27AM +0200, Jyri Sarha wrote:
+- ai31xx-micbias-vg - MicBias Voltage required.
- 1 - MICBIAS output is powered to 2.0V,
- 2 - MICBIAS output is powered to 2.5V,
- 3 - MICBIAS output is connected to AVDD,
- If this node is not mentioned or if the value is incorrect, then MicBias
- is powered down.
Please provide defines for these as part of the binding. It's not clear to me why leaving the micbias powered down is a sane default, it'll only be powered up if it's connected to something and used which doesn't seem altogether sane. I'd suggest picking 2.0V (the lowest voltage so least likely to cause damage).
Add machine driver support for AM43xx-ePOS-EVM and update associated device tree binding document.
Signed-off-by: Jyri Sarha jsarha@ti.com --- .../bindings/sound/davinci-evm-audio.txt | 9 +++-- sound/soc/davinci/davinci-evm.c | 41 ++++++++++++++++++++ 2 files changed, 47 insertions(+), 3 deletions(-)
diff --git a/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt b/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt index 865178d..356cba1 100644 --- a/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt @@ -2,8 +2,10 @@
Required properties: - compatible : "ti,da830-evm-audio" : forDM365/DA8xx/OMAPL1x/AM33xx + : "ti,am43xx-epos-evm-audio" : for am43xx-epos-evm - ti,model : The user-visible name of this sound complex. -- ti,audio-codec : The phandle of the TLV320AIC3x audio codec +- ti,audio-codec : The phandle of the TLV320AIC3x audio codec, + or the TLV320AIC31xx audio codec. - ti,mcasp-controller : The phandle of the McASP controller - ti,codec-clock-rate : The Codec Clock rate (in Hz) applied to the Codec - ti,audio-routing : A list of the connections between audio components. @@ -14,9 +16,10 @@ Required properties: Board connectors:
* Headphone Jack - * Line Out + * Line Out - "ti,da830-evm-audio" only * Mic Jack - * Line In + * Line In - "ti,da830-evm-audio" only + * Speaker - "ti,am43xx-epos-evm-audio" only
Example: diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 5e3bc3c..d4d965e 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -128,6 +128,33 @@ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) return 0; }
+static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +/* Logic for EVMs with an aic31xx */ +static int evm_aic31xx_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct device_node *np = codec->card->dev->of_node; + int ret; + + snd_soc_dapm_new_controls(dapm, aic31xx_dapm_widgets, + ARRAY_SIZE(aic31xx_dapm_widgets)); + + if (np) { + ret = snd_soc_of_parse_audio_routing(codec->card, + "ti,audio-routing"); + if (ret) + return ret; + } + + return 0; +} + /* davinci-evm digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link dm6446_evm_dai = { .name = "TLV320AIC3X", @@ -326,11 +353,25 @@ static struct snd_soc_dai_link evm_dai_tlv320aic3x = { SND_SOC_DAIFMT_IB_NF, };
+static struct snd_soc_dai_link evm_dai_tlv320aic3111 = { + .name = "TLV320AIC3111", + .stream_name = "AIC3111", + .codec_dai_name = "tlv320aic31xx-hifi", + .ops = &evm_ops, + .init = evm_aic31xx_init, + .dai_fmt = (SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_IB_NF), +}; + static const struct of_device_id davinci_evm_dt_ids[] = { { .compatible = "ti,da830-evm-audio", .data = (void *) &evm_dai_tlv320aic3x, }, + { + .compatible = "ti,am43xx-epos-evm-audio", + .data = &evm_dai_tlv320aic3111, + }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, davinci_evm_dt_ids);
Add support for am335x and am43x based boards with tlv320aic31xx compatible codec connected to McASP.
Signed-off-by: Jyri Sarha jsarha@ti.com --- sound/soc/davinci/Kconfig | 12 ++++++++++++ sound/soc/davinci/Makefile | 1 + 2 files changed, 13 insertions(+)
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index a8ec1fc..e1e3663 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -26,6 +26,18 @@ config SND_AM33XX_SOC_EVM AM335X-EVMSK, and BeagelBone with AudioCape boards have this setup.
+config SND_AM43XX_SOC_EPOS_EVM + tristate "SoC Audio for the AM33XX/AM43XX and TLV320AIC31XX based board" + depends on SND_DAVINCI_SOC + depends on SOC_AM33XX || SOC_AM43XX + select SND_SOC_TLV320AIC31XX + select SND_DAVINCI_SOC_MCASP + help + Say Y or M if you want to add support for SoC audio on + AM335X/AM43XX boards using McASP to connect to a codec of + TLV320AIC31XX series. For example AM43XX-EPOS-EVM has such a + setup. + config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" depends on SND_DAVINCI_SOC diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index 744d4d9..d61998f 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -13,3 +13,4 @@ obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o snd-soc-evm-objs := davinci-evm.o
obj-$(CONFIG_SND_DAVINCI_SOC_GENERIC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_AM43XX_SOC_EPOS_EVM) += snd-soc-evm.o
participants (3)
-
Jyri Sarha
-
Mark Brown
-
Peter Ujfalusi