* [PATCH v4] AM43xx-ePOS-EVM audio support with TLV320AIC31XX driver @ 2014-03-10 8:52 Jyri Sarha 2014-03-10 8:52 ` [PATCH v4] ASoC: tlv320aic31xx: Add basic codec driver implementation Jyri Sarha 2014-03-10 9:13 ` [PATCH v4] AM43xx-ePOS-EVM audio support with TLV320AIC31XX driver Mark Brown 0 siblings, 2 replies; 15+ messages in thread From: Jyri Sarha @ 2014-03-10 8:52 UTC (permalink / raw) To: alsa-devel, devicetree, linux-omap Cc: peter.ujfalusi, broonie, liam.r.girdwood, bcousson, detheridge, Jyri Sarha Since the v3 version of the patches: - Dropped "ASoC: tlv320aic32x4: Sort Makefile in alphabetic order" out - Changed DAMP widget "Mic Bias" to "MICBIAS" Best regards, Jyri Jyri Sarha (1): ASoC: tlv320aic31xx: Add basic codec driver implementation .../devicetree/bindings/sound/tlv320aic31xx.txt | 61 + include/dt-bindings/sound/tlv320aic31xx-micbias.h | 9 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic31xx.c | 1343 ++++++++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 258 ++++ 6 files changed, 1677 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt create mode 100644 include/dt-bindings/sound/tlv320aic31xx-micbias.h create mode 100644 sound/soc/codecs/tlv320aic31xx.c create mode 100644 sound/soc/codecs/tlv320aic31xx.h -- 1.7.9.5 ^ permalink raw reply [flat|nested] 15+ messages in thread
* [PATCH v4] ASoC: tlv320aic31xx: Add basic codec driver implementation 2014-03-10 8:52 [PATCH v4] AM43xx-ePOS-EVM audio support with TLV320AIC31XX driver Jyri Sarha @ 2014-03-10 8:52 ` Jyri Sarha 2014-03-10 9:39 ` Lokesh Vutla ` (2 more replies) 2014-03-10 9:13 ` [PATCH v4] AM43xx-ePOS-EVM audio support with TLV320AIC31XX driver Mark Brown 1 sibling, 3 replies; 15+ messages in thread From: Jyri Sarha @ 2014-03-10 8:52 UTC (permalink / raw) To: alsa-devel, devicetree, linux-omap Cc: peter.ujfalusi, broonie, liam.r.girdwood, bcousson, detheridge, Jyri Sarha 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> --- .../devicetree/bindings/sound/tlv320aic31xx.txt | 61 + include/dt-bindings/sound/tlv320aic31xx-micbias.h | 9 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic31xx.c | 1343 ++++++++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 258 ++++ 6 files changed, 1677 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt create mode 100644 include/dt-bindings/sound/tlv320aic31xx-micbias.h create mode 100644 sound/soc/codecs/tlv320aic31xx.c create mode 100644 sound/soc/codecs/tlv320aic31xx.h diff --git a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt new file mode 100644 index 0000000..0109df1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt @@ -0,0 +1,61 @@ +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 setting + 0 or MICBIAS_OFF - MICBIAS output it not powered + 1 or MICBIAS_2_0V - MICBIAS output is powered to 2.0V + 2 or MICBIAS_2_5V - MICBIAS output is powered to 2.5V + 3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD + If this node is not mentioned or if the value is unknown, then + micbias is set to 2.0V. +- 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: +#include <dt-bindings/sound/tlv320aic31xx-micbias.h> + +tlv320aic31xx: tlv320aic31xx@18 { + compatible = "ti,tlv320aic311x"; + reg = <0x18>; + + ai31xx-micbias-vg = <MICBIAS_OFF>; + + HPVDD-supply = <®ulator>; + SPRVDD-supply = <®ulator>; + SPLVDD-supply = <®ulator>; + AVDD-supply = <®ulator>; + IOVDD-supply = <®ulator>; + DVDD-supply = <®ulator>; +}; diff --git a/include/dt-bindings/sound/tlv320aic31xx-micbias.h b/include/dt-bindings/sound/tlv320aic31xx-micbias.h new file mode 100644 index 0000000..da9360f --- /dev/null +++ b/include/dt-bindings/sound/tlv320aic31xx-micbias.h @@ -0,0 +1,9 @@ +#ifndef __DT_TLV320AIC31XX_MICBIAS_H +#define __DT_TLV320AIC31XX_MICBIAS_H + +#define MICBIAS_OFF 0 +#define MICBIAS_2_0V 1 +#define MICBIAS_2_5V 2 +#define MICBIAS_AVDDV 3 + +#endif /* __DT_TLV320AIC31XX_MICBIAS_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e19b64f..af3c049 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -83,6 +83,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TAS5086 if I2C select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER + select SND_SOC_TLV320AIC31XX if I2C select SND_SOC_TLV320AIC32X4 if I2C select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C @@ -428,6 +429,9 @@ config SND_SOC_TLV320AIC26 tristate depends on SPI +config SND_SOC_TLV320AIC31XX + tristate + config SND_SOC_TLV320AIC32X4 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 90ab24b..018e3f3 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -75,6 +75,7 @@ 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-tlv320aic32x4-objs := tlv320aic32x4.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-tlv320dac33-objs := tlv320dac33.o @@ -215,6 +216,7 @@ 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_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c new file mode 100644 index 0000000..1d3d2416 --- /dev/null +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -0,0 +1,1343 @@ +/* + * 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 <dt-bindings/sound/tlv320aic31xx-micbias.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_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case AIC31XX_PAGECTL: /* regmap implementation requires this */ + case AIC31XX_RESET: /* always clears after write */ + case AIC31XX_OT_FLAG: + case AIC31XX_ADCFLAG: + case AIC31XX_DACFLAG1: + case AIC31XX_DACFLAG2: + case AIC31XX_OFFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRDACFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRADCFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRDACFLAG2: + case AIC31XX_INTRADCFLAG2: + return true; + } + return false; +} + +static bool aic31xx_writeable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case AIC31XX_OT_FLAG: + case AIC31XX_ADCFLAG: + case AIC31XX_DACFLAG1: + case AIC31XX_DACFLAG2: + case AIC31XX_OFFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRDACFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRADCFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRDACFLAG2: + case AIC31XX_INTRADCFLAG2: + return false; + } + return true; +} + +static const struct regmap_range_cfg aic31xx_ranges[] = { + { + .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, + .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[AIC31XX_NUM_SUPPLIES] = { + "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]; + 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_S_TLV("DAC Playback Volume", AIC31XX_LDACVOL, + AIC31XX_RDACVOL, 0, -127, 48, 7, 0, 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), + SOC_DOUBLE_R_S_TLV("ADC Capture Volume", AIC31XX_ADCVOL, AIC31XX_ADCVOL, + 0, -24, 40, 6, 0, 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("Speaker Driver Playback Switch", AIC31XX_SPLGAIN, + AIC31XX_SPRGAIN, 2, 1, 0), + SOC_DOUBLE_R_TLV("Speaker Driver Playback Volume", AIC31XX_SPLGAIN, + AIC31XX_SPRGAIN, 3, 3, 0, class_D_drv_tlv), + + SOC_DOUBLE_R_TLV("Speaker Analog Playback Volume", AIC31XX_LANALOGSPL, + AIC31XX_RANALOGSPR, 0, 0x7F, 1, sp_vol_tlv), +}; + +static const struct snd_kcontrol_new aic310x_snd_controls[] = { + SOC_SINGLE("Speaker Driver Playback Switch", AIC31XX_SPLGAIN, + 2, 1, 0), + SOC_SINGLE_TLV("Speaker Driver Playback Volume", AIC31XX_SPLGAIN, + 3, 3, 0, class_D_drv_tlv), + + SOC_SINGLE_TLV("Speaker 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; +} + + +#define WIDGET_BIT(reg, shift) (((shift) << 8) | (reg)) + +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); + + switch (WIDGET_BIT(w->reg, w->shift)) { + case WIDGET_BIT(AIC31XX_DACSETUP, 7): + mask = AIC31XX_LDACPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_DACSETUP, 6): + mask = AIC31XX_RDACPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_HPDRIVER, 7): + mask = AIC31XX_HPLDRVPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_HPDRIVER, 6): + mask = AIC31XX_HPRDRVPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_SPKAMP, 7): + mask = AIC31XX_SPLDRVPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_SPKAMP, 6): + mask = AIC31XX_SPRDRVPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_ADCSETUP, 7): + mask = AIC31XX_ADCPWRSTATUS_MASK; + reg = AIC31XX_ADCFLAG; + break; + default: + dev_err(w->codec->dev, "Unknown widget '%s' calling %s/n", + w->name, __func__); + return -EINVAL; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + return aic31xx_wait_bits(aic31xx, reg, mask, mask, 5000, 100); + case 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 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 != 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 != MICBIAS_OFF) { + snd_soc_update_bits(codec, AIC31XX_MICBIAS, + AIC31XX_MICBIAS_MASK, MICBIAS_OFF); + 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", "Left Playback", + 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", "Right Playback", + 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), + + /* Mic Bias */ + SND_SOC_DAPM_SUPPLY("MICBIAS", 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"), + + SND_SOC_DAPM_OUTPUT("Internal DAC Sink"), + + /* Inputs */ + SND_SOC_DAPM_INPUT("MIC1LP"), + SND_SOC_DAPM_INPUT("MIC1RP"), + SND_SOC_DAPM_INPUT("MIC1LM"), + + SND_SOC_DAPM_INPUT("Internal ADC Source"), +}; + +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("Speaker Left", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_spl_switch), + SND_SOC_DAPM_SWITCH("Speaker 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[] = { + /* Make DACs turn on when playing even if not mixed into any outputs */ + {"Internal DAC Sink", NULL, "DAC Left"}, + {"Internal DAC Sink", NULL, "DAC Right"}, + + /* Make ADC turn on when recording even if not mixed from any inputs */ + {"ADC", NULL, "Internal ADC Source"}, + + /* 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, "MICBIAS"}, + + /* 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 */ + {"Speaker Left", "Switch", "Output Left"}, + {"SPL ClassD", NULL, "Speaker Left"}, + {"SPL", NULL, "SPL ClassD"}, + + /* SP R path */ + {"Speaker Right", "Switch", "Output Right"}, + {"SPR ClassD", NULL, "Speaker 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; + u8 data = 0; + + dev_dbg(codec->dev, "## %s: format %d rate %d\n", + __func__, params_format(params), params_rate(params)); + + switch (params_width(params)) { + case 16: + break; + case 20: + data = (AIC31XX_WORD_LEN_20BITS << + AIC31XX_IFACE1_DATALEN_SHIFT); + break; + case 24: + data = (AIC31XX_WORD_LEN_24BITS << + AIC31XX_IFACE1_DATALEN_SHIFT); + break; + case 32: + 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; + 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; + 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. It may be to late to try + * soft reset, but it should not do any harm. + * Resetting the codec helps in minimizing leakage + * in case some of the regulators remains turned on. + */ + if (gpio_is_valid(aic31xx->pdata.gpio_reset)) + gpio_set_value(aic31xx->pdata.gpio_reset, 0); + else + snd_soc_write(aic31xx->codec, AIC31XX_RESET, 0x01); + regcache_mark_dirty(aic31xx->regmap); + dev_dbg(aic31xx->dev, "## %s: DISABLE received\n", __func__); + } + + return 0; +} + +static void aic31xx_clk_on(struct snd_soc_codec *codec) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + u8 mask = AIC31XX_PM_MASK; + u8 on = AIC31XX_PM_MASK; + + 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; +} + +static void aic31xx_clk_off(struct snd_soc_codec *codec) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + u8 mask = AIC31XX_PM_MASK; + u8 off = 0; + + 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; +} + +static int aic31xx_power_on(struct snd_soc_codec *codec) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + ret = regulator_bulk_enable(ARRAY_SIZE(aic31xx->supplies), + aic31xx->supplies); + if (ret) + return ret; + + if (gpio_is_valid(aic31xx->pdata.gpio_reset)) { + gpio_set_value(aic31xx->pdata.gpio_reset, 1); + 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; + } + return 0; +} + +static int aic31xx_power_off(struct snd_soc_codec *codec) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + regcache_cache_only(aic31xx->regmap, true); + ret = regulator_bulk_disable(ARRAY_SIZE(aic31xx->supplies), + aic31xx->supplies); + + 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) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) + aic31xx_clk_on(codec); + break; + case SND_SOC_BIAS_STANDBY: + switch (codec->dapm.bias_level) { + case SND_SOC_BIAS_OFF: + aic31xx_power_on(codec); + break; + case SND_SOC_BIAS_PREPARE: + aic31xx_clk_off(codec); + break; + default: + BUG(); + } + break; + case SND_SOC_BIAS_OFF: + aic31xx_power_off(codec); + 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, + .symmetric_rates = 1, + } +}; + +#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 = MICBIAS_2_0V; + int ret; + + of_property_read_u32(np, "ai31xx-micbias-vg", &value); + switch (value) { + case MICBIAS_OFF: + case MICBIAS_2_0V: + case MICBIAS_2_5V: + case MICBIAS_AVDDV: + aic31xx->pdata.micbias_vg = value; + break; + default: + dev_err(aic31xx->dev, + "Bad ai31xx-micbias-vg value %d DT\n", + value); + aic31xx->pdata.micbias_vg = MICBIAS_2_0V; + } + + 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..52ed57c --- /dev/null +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -0,0 +1,258 @@ +/* + * 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), +}; + +struct aic31xx_pdata { + enum aic31xx_type codec_type; + unsigned int gpio_reset; + int 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 */ -- 1.7.9.5 ^ permalink raw reply related [flat|nested] 15+ messages in thread
* Re: [PATCH v4] ASoC: tlv320aic31xx: Add basic codec driver implementation 2014-03-10 8:52 ` [PATCH v4] ASoC: tlv320aic31xx: Add basic codec driver implementation Jyri Sarha @ 2014-03-10 9:39 ` Lokesh Vutla 2014-03-10 9:45 ` Mark Brown 2014-03-10 12:09 ` Mark Rutland 2014-03-10 12:58 ` Mark Brown 2 siblings, 1 reply; 15+ messages in thread From: Lokesh Vutla @ 2014-03-10 9:39 UTC (permalink / raw) To: Jyri Sarha Cc: alsa-devel, devicetree, linux-omap, peter.ujfalusi, broonie, liam.r.girdwood, bcousson, detheridge Hi Jyri, Just a few minor comments... On Monday 10 March 2014 02:22 PM, 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. > > 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> > --- > .../devicetree/bindings/sound/tlv320aic31xx.txt | 61 + > include/dt-bindings/sound/tlv320aic31xx-micbias.h | 9 + > sound/soc/codecs/Kconfig | 4 + > sound/soc/codecs/Makefile | 2 + > sound/soc/codecs/tlv320aic31xx.c | 1343 ++++++++++++++++++++ > sound/soc/codecs/tlv320aic31xx.h | 258 ++++ > 6 files changed, 1677 insertions(+) > create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt > create mode 100644 include/dt-bindings/sound/tlv320aic31xx-micbias.h > create mode 100644 sound/soc/codecs/tlv320aic31xx.c > create mode 100644 sound/soc/codecs/tlv320aic31xx.h > > diff --git a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt > new file mode 100644 > index 0000000..0109df1 > --- /dev/null > +++ b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt > @@ -0,0 +1,61 @@ > +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 setting > + 0 or MICBIAS_OFF - MICBIAS output it not powered > + 1 or MICBIAS_2_0V - MICBIAS output is powered to 2.0V > + 2 or MICBIAS_2_5V - MICBIAS output is powered to 2.5V > + 3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD > + If this node is not mentioned or if the value is unknown, then > + micbias is set to 2.0V. > +- 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: > +#include <dt-bindings/sound/tlv320aic31xx-micbias.h> > + > +tlv320aic31xx: tlv320aic31xx@18 { > + compatible = "ti,tlv320aic311x"; > + reg = <0x18>; > + > + ai31xx-micbias-vg = <MICBIAS_OFF>; > + > + HPVDD-supply = <®ulator>; > + SPRVDD-supply = <®ulator>; > + SPLVDD-supply = <®ulator>; > + AVDD-supply = <®ulator>; > + IOVDD-supply = <®ulator>; > + DVDD-supply = <®ulator>; > +}; > diff --git a/include/dt-bindings/sound/tlv320aic31xx-micbias.h b/include/dt-bindings/sound/tlv320aic31xx-micbias.h > new file mode 100644 > index 0000000..da9360f > --- /dev/null > +++ b/include/dt-bindings/sound/tlv320aic31xx-micbias.h > @@ -0,0 +1,9 @@ > +#ifndef __DT_TLV320AIC31XX_MICBIAS_H > +#define __DT_TLV320AIC31XX_MICBIAS_H > + > +#define MICBIAS_OFF 0 > +#define MICBIAS_2_0V 1 > +#define MICBIAS_2_5V 2 > +#define MICBIAS_AVDDV 3 > + > +#endif /* __DT_TLV320AIC31XX_MICBIAS_H */ > diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig > index e19b64f..af3c049 100644 > --- a/sound/soc/codecs/Kconfig > +++ b/sound/soc/codecs/Kconfig > @@ -83,6 +83,7 @@ config SND_SOC_ALL_CODECS > select SND_SOC_TAS5086 if I2C > select SND_SOC_TLV320AIC23 if I2C > select SND_SOC_TLV320AIC26 if SPI_MASTER > + select SND_SOC_TLV320AIC31XX if I2C > select SND_SOC_TLV320AIC32X4 if I2C > select SND_SOC_TLV320AIC3X if I2C > select SND_SOC_TPA6130A2 if I2C > @@ -428,6 +429,9 @@ config SND_SOC_TLV320AIC26 > tristate > depends on SPI > > +config SND_SOC_TLV320AIC31XX > + tristate > + > config SND_SOC_TLV320AIC32X4 > tristate > > diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile > index 90ab24b..018e3f3 100644 > --- a/sound/soc/codecs/Makefile > +++ b/sound/soc/codecs/Makefile > @@ -75,6 +75,7 @@ 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-tlv320aic32x4-objs := tlv320aic32x4.o > snd-soc-tlv320aic3x-objs := tlv320aic3x.o > snd-soc-tlv320dac33-objs := tlv320dac33.o > @@ -215,6 +216,7 @@ 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_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o > obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o > obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o > diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c > new file mode 100644 > index 0000000..1d3d2416 > --- /dev/null > +++ b/sound/soc/codecs/tlv320aic31xx.c > @@ -0,0 +1,1343 @@ > +/* > + * ALSA SoC TLV320AIC31XX codec driver > + * > + * Copyright (C) 2013 Texas Instruments, Inc. 2014..:) > + * > + * 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 <dt-bindings/sound/tlv320aic31xx-micbias.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_volatile(struct device *dev, unsigned int reg) > +{ > + switch (reg) { > + case AIC31XX_PAGECTL: /* regmap implementation requires this */ > + case AIC31XX_RESET: /* always clears after write */ > + case AIC31XX_OT_FLAG: > + case AIC31XX_ADCFLAG: > + case AIC31XX_DACFLAG1: > + case AIC31XX_DACFLAG2: > + case AIC31XX_OFFLAG: /* Sticky interrupt flags */ > + case AIC31XX_INTRDACFLAG: /* Sticky interrupt flags */ > + case AIC31XX_INTRADCFLAG: /* Sticky interrupt flags */ > + case AIC31XX_INTRDACFLAG2: > + case AIC31XX_INTRADCFLAG2: > + return true; > + } > + return false; > +} > + > +static bool aic31xx_writeable(struct device *dev, unsigned int reg) > +{ > + switch (reg) { > + case AIC31XX_OT_FLAG: > + case AIC31XX_ADCFLAG: > + case AIC31XX_DACFLAG1: > + case AIC31XX_DACFLAG2: > + case AIC31XX_OFFLAG: /* Sticky interrupt flags */ > + case AIC31XX_INTRDACFLAG: /* Sticky interrupt flags */ > + case AIC31XX_INTRADCFLAG: /* Sticky interrupt flags */ > + case AIC31XX_INTRDACFLAG2: > + case AIC31XX_INTRADCFLAG2: > + return false; > + } > + return true; > +} > + > +static const struct regmap_range_cfg aic31xx_ranges[] = { > + { > + .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, > + .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[AIC31XX_NUM_SUPPLIES] = { > + "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]; > + 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_S_TLV("DAC Playback Volume", AIC31XX_LDACVOL, > + AIC31XX_RDACVOL, 0, -127, 48, 7, 0, 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), > + SOC_DOUBLE_R_S_TLV("ADC Capture Volume", AIC31XX_ADCVOL, AIC31XX_ADCVOL, > + 0, -24, 40, 6, 0, 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("Speaker Driver Playback Switch", AIC31XX_SPLGAIN, > + AIC31XX_SPRGAIN, 2, 1, 0), > + SOC_DOUBLE_R_TLV("Speaker Driver Playback Volume", AIC31XX_SPLGAIN, > + AIC31XX_SPRGAIN, 3, 3, 0, class_D_drv_tlv), > + > + SOC_DOUBLE_R_TLV("Speaker Analog Playback Volume", AIC31XX_LANALOGSPL, > + AIC31XX_RANALOGSPR, 0, 0x7F, 1, sp_vol_tlv), > +}; > + > +static const struct snd_kcontrol_new aic310x_snd_controls[] = { > + SOC_SINGLE("Speaker Driver Playback Switch", AIC31XX_SPLGAIN, > + 2, 1, 0), > + SOC_SINGLE_TLV("Speaker Driver Playback Volume", AIC31XX_SPLGAIN, > + 3, 3, 0, class_D_drv_tlv), > + > + SOC_SINGLE_TLV("Speaker 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; > +} > + > + extra line... > +#define WIDGET_BIT(reg, shift) (((shift) << 8) | (reg)) > + > +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); > + > + switch (WIDGET_BIT(w->reg, w->shift)) { > + case WIDGET_BIT(AIC31XX_DACSETUP, 7): > + mask = AIC31XX_LDACPWRSTATUS_MASK; > + break; > + case WIDGET_BIT(AIC31XX_DACSETUP, 6): > + mask = AIC31XX_RDACPWRSTATUS_MASK; > + break; > + case WIDGET_BIT(AIC31XX_HPDRIVER, 7): > + mask = AIC31XX_HPLDRVPWRSTATUS_MASK; > + break; > + case WIDGET_BIT(AIC31XX_HPDRIVER, 6): > + mask = AIC31XX_HPRDRVPWRSTATUS_MASK; > + break; > + case WIDGET_BIT(AIC31XX_SPKAMP, 7): > + mask = AIC31XX_SPLDRVPWRSTATUS_MASK; > + break; > + case WIDGET_BIT(AIC31XX_SPKAMP, 6): > + mask = AIC31XX_SPRDRVPWRSTATUS_MASK; > + break; > + case WIDGET_BIT(AIC31XX_ADCSETUP, 7): > + mask = AIC31XX_ADCPWRSTATUS_MASK; > + reg = AIC31XX_ADCFLAG; > + break; > + default: > + dev_err(w->codec->dev, "Unknown widget '%s' calling %s/n", > + w->name, __func__); > + return -EINVAL; > + } > + > + switch (event) { > + case SND_SOC_DAPM_POST_PMU: > + return aic31xx_wait_bits(aic31xx, reg, mask, mask, 5000, 100); > + case 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 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 != 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 != MICBIAS_OFF) { > + snd_soc_update_bits(codec, AIC31XX_MICBIAS, > + AIC31XX_MICBIAS_MASK, MICBIAS_OFF); > + 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", "Left Playback", > + 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", "Right Playback", > + 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), > + > + /* Mic Bias */ > + SND_SOC_DAPM_SUPPLY("MICBIAS", 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"), > + > + SND_SOC_DAPM_OUTPUT("Internal DAC Sink"), > + > + /* Inputs */ > + SND_SOC_DAPM_INPUT("MIC1LP"), > + SND_SOC_DAPM_INPUT("MIC1RP"), > + SND_SOC_DAPM_INPUT("MIC1LM"), > + > + SND_SOC_DAPM_INPUT("Internal ADC Source"), > +}; > + > +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("Speaker Left", SND_SOC_NOPM, 0, 0, > + &aic31xx_dapm_spl_switch), > + SND_SOC_DAPM_SWITCH("Speaker 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[] = { > + /* Make DACs turn on when playing even if not mixed into any outputs */ > + {"Internal DAC Sink", NULL, "DAC Left"}, > + {"Internal DAC Sink", NULL, "DAC Right"}, > + > + /* Make ADC turn on when recording even if not mixed from any inputs */ > + {"ADC", NULL, "Internal ADC Source"}, > + > + /* 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, "MICBIAS"}, > + > + /* 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 */ > + {"Speaker Left", "Switch", "Output Left"}, > + {"SPL ClassD", NULL, "Speaker Left"}, > + {"SPL", NULL, "SPL ClassD"}, > + > + /* SP R path */ > + {"Speaker Right", "Switch", "Output Right"}, > + {"SPR ClassD", NULL, "Speaker 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; > + u8 data = 0; > + > + dev_dbg(codec->dev, "## %s: format %d rate %d\n", > + __func__, params_format(params), params_rate(params)); > + > + switch (params_width(params)) { > + case 16: > + break; > + case 20: > + data = (AIC31XX_WORD_LEN_20BITS << > + AIC31XX_IFACE1_DATALEN_SHIFT); > + break; > + case 24: > + data = (AIC31XX_WORD_LEN_24BITS << > + AIC31XX_IFACE1_DATALEN_SHIFT); > + break; > + case 32: > + 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; > + 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; > + 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. It may be to late to try > + * soft reset, but it should not do any harm. > + * Resetting the codec helps in minimizing leakage > + * in case some of the regulators remains turned on. > + */ > + if (gpio_is_valid(aic31xx->pdata.gpio_reset)) > + gpio_set_value(aic31xx->pdata.gpio_reset, 0); > + else > + snd_soc_write(aic31xx->codec, AIC31XX_RESET, 0x01); > + regcache_mark_dirty(aic31xx->regmap); > + dev_dbg(aic31xx->dev, "## %s: DISABLE received\n", __func__); > + } > + > + return 0; > +} > + > +static void aic31xx_clk_on(struct snd_soc_codec *codec) > +{ > + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); > + u8 mask = AIC31XX_PM_MASK; > + u8 on = AIC31XX_PM_MASK; > + > + 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; > +} > + > +static void aic31xx_clk_off(struct snd_soc_codec *codec) > +{ > + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); > + u8 mask = AIC31XX_PM_MASK; > + u8 off = 0; > + > + 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; > +} > + > +static int aic31xx_power_on(struct snd_soc_codec *codec) > +{ > + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); > + int ret = 0; > + > + ret = regulator_bulk_enable(ARRAY_SIZE(aic31xx->supplies), > + aic31xx->supplies); > + if (ret) > + return ret; > + > + if (gpio_is_valid(aic31xx->pdata.gpio_reset)) { > + gpio_set_value(aic31xx->pdata.gpio_reset, 1); > + 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; > + } > + return 0; > +} > + > +static int aic31xx_power_off(struct snd_soc_codec *codec) > +{ > + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); > + int ret = 0; > + > + regcache_cache_only(aic31xx->regmap, true); > + ret = regulator_bulk_disable(ARRAY_SIZE(aic31xx->supplies), > + aic31xx->supplies); > + > + 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) { > + case SND_SOC_BIAS_ON: > + break; > + case SND_SOC_BIAS_PREPARE: > + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) > + aic31xx_clk_on(codec); > + break; > + case SND_SOC_BIAS_STANDBY: > + switch (codec->dapm.bias_level) { > + case SND_SOC_BIAS_OFF: > + aic31xx_power_on(codec); > + break; > + case SND_SOC_BIAS_PREPARE: > + aic31xx_clk_off(codec); > + break; > + default: > + BUG(); > + } > + break; > + case SND_SOC_BIAS_OFF: > + aic31xx_power_off(codec); > + break; > + } > + codec->dapm.bias_level = level; > + > + return 0; > +} > + > + extra line... Thanks and regards, Lokesh > +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, > + .symmetric_rates = 1, > + } > +}; > + > +#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 = MICBIAS_2_0V; > + int ret; > + > + of_property_read_u32(np, "ai31xx-micbias-vg", &value); > + switch (value) { > + case MICBIAS_OFF: > + case MICBIAS_2_0V: > + case MICBIAS_2_5V: > + case MICBIAS_AVDDV: > + aic31xx->pdata.micbias_vg = value; > + break; > + default: > + dev_err(aic31xx->dev, > + "Bad ai31xx-micbias-vg value %d DT\n", > + value); > + aic31xx->pdata.micbias_vg = MICBIAS_2_0V; > + } > + > + 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); devm_kzalloc will take care of cleaning aic31xx during driver removal. No need to do kfree here. > + 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..52ed57c > --- /dev/null > +++ b/sound/soc/codecs/tlv320aic31xx.h > @@ -0,0 +1,258 @@ > +/* > + * ALSA SoC TLV320AIC31XX codec driver > + * > + * Copyright (C) 2013 Texas Instruments, Inc. 2014.. > + * > + * 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), > +}; > + > +struct aic31xx_pdata { > + enum aic31xx_type codec_type; > + unsigned int gpio_reset; > + int 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 > + > + extra line here..:) > +/* 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 */ > ^ permalink raw reply [flat|nested] 15+ messages in thread
* Re: [PATCH v4] ASoC: tlv320aic31xx: Add basic codec driver implementation 2014-03-10 9:39 ` Lokesh Vutla @ 2014-03-10 9:45 ` Mark Brown 0 siblings, 0 replies; 15+ messages in thread From: Mark Brown @ 2014-03-10 9:45 UTC (permalink / raw) To: Lokesh Vutla Cc: Jyri Sarha, alsa-devel, devicetree, linux-omap, peter.ujfalusi, liam.r.girdwood, bcousson, detheridge [-- Attachment #1: Type: text/plain, Size: 627 bytes --] On Mon, Mar 10, 2014 at 03:09:40PM +0530, Lokesh Vutla wrote: > Hi Jyri, > > Just a few minor comments... > On Monday 10 March 2014 02:22 PM, 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. You've quoted the entire mail here but added only (as far as I can tell) three lines of content - don't do that, it makes it harder to find what you've added to the mail. Instead delete context that you're not replying to from the mail. [-- Attachment #2: Digital signature --] [-- Type: application/pgp-signature, Size: 836 bytes --] ^ permalink raw reply [flat|nested] 15+ messages in thread
* Re: [PATCH v4] ASoC: tlv320aic31xx: Add basic codec driver implementation 2014-03-10 8:52 ` [PATCH v4] ASoC: tlv320aic31xx: Add basic codec driver implementation Jyri Sarha 2014-03-10 9:39 ` Lokesh Vutla @ 2014-03-10 12:09 ` Mark Rutland 2014-03-10 12:11 ` Mark Brown 2014-03-10 13:15 ` Jyri Sarha 2014-03-10 12:58 ` Mark Brown 2 siblings, 2 replies; 15+ messages in thread From: Mark Rutland @ 2014-03-10 12:09 UTC (permalink / raw) To: Jyri Sarha Cc: alsa-devel@alsa-project.org, devicetree@vger.kernel.org, linux-omap@vger.kernel.org, peter.ujfalusi@ti.com, broonie@kernel.org, liam.r.girdwood@linux.intel.com, bcousson@baylibre.com, detheridge@ti.com On Mon, Mar 10, 2014 at 08:52:21AM +0000, 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. > > 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> > --- > .../devicetree/bindings/sound/tlv320aic31xx.txt | 61 + > include/dt-bindings/sound/tlv320aic31xx-micbias.h | 9 + > sound/soc/codecs/Kconfig | 4 + > sound/soc/codecs/Makefile | 2 + > sound/soc/codecs/tlv320aic31xx.c | 1343 ++++++++++++++++++++ > sound/soc/codecs/tlv320aic31xx.h | 258 ++++ > 6 files changed, 1677 insertions(+) > create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt > create mode 100644 include/dt-bindings/sound/tlv320aic31xx-micbias.h > create mode 100644 sound/soc/codecs/tlv320aic31xx.c > create mode 100644 sound/soc/codecs/tlv320aic31xx.h > > diff --git a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt > new file mode 100644 > index 0000000..0109df1 > --- /dev/null > +++ b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt > @@ -0,0 +1,61 @@ > +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 I believe calling this reset-gpio would be more in keeping with the gpio bindings. I'd get rid of "pin number" from the description. GPIOs are referred to with phandle + gpio-specifier pairs, and this makes it sound like a single integer value. > +- ai31xx-micbias-vg - MicBias Voltage setting This name is a bit terse. Perhaps s/vg/voltage/ ? > + 0 or MICBIAS_OFF - MICBIAS output it not powered s/it/is/ > + 1 or MICBIAS_2_0V - MICBIAS output is powered to 2.0V > + 2 or MICBIAS_2_5V - MICBIAS output is powered to 2.5V > + 3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD > + If this node is not mentioned or if the value is unknown, then > + micbias is set to 2.0V. Why does the dts write need to select these -- what prevents this getting selected dynamically? Cheers, Mark. ^ permalink raw reply [flat|nested] 15+ messages in thread
* Re: [PATCH v4] ASoC: tlv320aic31xx: Add basic codec driver implementation 2014-03-10 12:09 ` Mark Rutland @ 2014-03-10 12:11 ` Mark Brown 2014-03-10 13:15 ` Jyri Sarha 1 sibling, 0 replies; 15+ messages in thread From: Mark Brown @ 2014-03-10 12:11 UTC (permalink / raw) To: Mark Rutland Cc: Jyri Sarha, alsa-devel@alsa-project.org, devicetree@vger.kernel.org, linux-omap@vger.kernel.org, peter.ujfalusi@ti.com, liam.r.girdwood@linux.intel.com, bcousson@baylibre.com, detheridge@ti.com [-- Attachment #1: Type: text/plain, Size: 652 bytes --] On Mon, Mar 10, 2014 at 12:09:38PM +0000, Mark Rutland wrote: > On Mon, Mar 10, 2014 at 08:52:21AM +0000, Jyri Sarha wrote: > > + 1 or MICBIAS_2_0V - MICBIAS output is powered to 2.0V > > + 2 or MICBIAS_2_5V - MICBIAS output is powered to 2.5V > > + 3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD > > + If this node is not mentioned or if the value is unknown, then > > + micbias is set to 2.0V. > Why does the dts write need to select these -- what prevents this > getting selected dynamically? It's a property of the system integration - the board will be designed to bias the microphone at some voltage. [-- Attachment #2: Digital signature --] [-- Type: application/pgp-signature, Size: 836 bytes --] ^ permalink raw reply [flat|nested] 15+ messages in thread
* Re: [PATCH v4] ASoC: tlv320aic31xx: Add basic codec driver implementation 2014-03-10 12:09 ` Mark Rutland 2014-03-10 12:11 ` Mark Brown @ 2014-03-10 13:15 ` Jyri Sarha 2014-03-10 13:31 ` Mark Rutland 1 sibling, 1 reply; 15+ messages in thread From: Jyri Sarha @ 2014-03-10 13:15 UTC (permalink / raw) To: Mark Rutland Cc: alsa-devel@alsa-project.org, devicetree@vger.kernel.org, linux-omap@vger.kernel.org, peter.ujfalusi@ti.com, broonie@kernel.org, liam.r.girdwood@linux.intel.com, bcousson@baylibre.com, detheridge@ti.com On 03/10/2014 02:09 PM, Mark Rutland wrote: > On Mon, Mar 10, 2014 at 08:52:21AM +0000, 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. >> >> 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> >> --- >> .../devicetree/bindings/sound/tlv320aic31xx.txt | 61 + >> include/dt-bindings/sound/tlv320aic31xx-micbias.h | 9 + >> sound/soc/codecs/Kconfig | 4 + >> sound/soc/codecs/Makefile | 2 + >> sound/soc/codecs/tlv320aic31xx.c | 1343 ++++++++++++++++++++ >> sound/soc/codecs/tlv320aic31xx.h | 258 ++++ >> 6 files changed, 1677 insertions(+) >> create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt >> create mode 100644 include/dt-bindings/sound/tlv320aic31xx-micbias.h >> create mode 100644 sound/soc/codecs/tlv320aic31xx.c >> create mode 100644 sound/soc/codecs/tlv320aic31xx.h >> >> diff --git a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt >> new file mode 100644 >> index 0000000..0109df1 >> --- /dev/null >> +++ b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt >> @@ -0,0 +1,61 @@ >> +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 > > I believe calling this reset-gpio would be more in keeping with the gpio > bindings. > > I'd get rid of "pin number" from the description. GPIOs are referred to > with phandle + gpio-specifier pairs, and this makes it sound like a > single integer value. > >> +- ai31xx-micbias-vg - MicBias Voltage setting > > This name is a bit terse. Perhaps s/vg/voltage/ ? > This parameter name is symmetric to already existing "ai3x-micbias-vg" in tlv320aic3x driver. Well... that is not a very good reason for not to change it. I'll change that to the next version of the patch. >> + 0 or MICBIAS_OFF - MICBIAS output it not powered > > s/it/is/ > To be fixed. Best regards, Jyri ^ permalink raw reply [flat|nested] 15+ messages in thread
* Re: [PATCH v4] ASoC: tlv320aic31xx: Add basic codec driver implementation 2014-03-10 13:15 ` Jyri Sarha @ 2014-03-10 13:31 ` Mark Rutland 0 siblings, 0 replies; 15+ messages in thread From: Mark Rutland @ 2014-03-10 13:31 UTC (permalink / raw) To: Jyri Sarha Cc: devicetree@vger.kernel.org, alsa-devel@alsa-project.org, liam.r.girdwood@linux.intel.com, peter.ujfalusi@ti.com, detheridge@ti.com, broonie@kernel.org, bcousson@baylibre.com, linux-omap@vger.kernel.org On Mon, Mar 10, 2014 at 01:15:56PM +0000, Jyri Sarha wrote: > On 03/10/2014 02:09 PM, Mark Rutland wrote: > > On Mon, Mar 10, 2014 at 08:52:21AM +0000, 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. > >> > >> 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> > >> --- > >> .../devicetree/bindings/sound/tlv320aic31xx.txt | 61 + > >> include/dt-bindings/sound/tlv320aic31xx-micbias.h | 9 + > >> sound/soc/codecs/Kconfig | 4 + > >> sound/soc/codecs/Makefile | 2 + > >> sound/soc/codecs/tlv320aic31xx.c | 1343 ++++++++++++++++++++ > >> sound/soc/codecs/tlv320aic31xx.h | 258 ++++ > >> 6 files changed, 1677 insertions(+) > >> create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt > >> create mode 100644 include/dt-bindings/sound/tlv320aic31xx-micbias.h > >> create mode 100644 sound/soc/codecs/tlv320aic31xx.c > >> create mode 100644 sound/soc/codecs/tlv320aic31xx.h > >> > >> diff --git a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt > >> new file mode 100644 > >> index 0000000..0109df1 > >> --- /dev/null > >> +++ b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt > >> @@ -0,0 +1,61 @@ > >> +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 > > > > I believe calling this reset-gpio would be more in keeping with the gpio > > bindings. > > > > I'd get rid of "pin number" from the description. GPIOs are referred to > > with phandle + gpio-specifier pairs, and this makes it sound like a > > single integer value. > > > >> +- ai31xx-micbias-vg - MicBias Voltage setting > > > > This name is a bit terse. Perhaps s/vg/voltage/ ? > > > > This parameter name is symmetric to already existing "ai3x-micbias-vg" > in tlv320aic3x driver. Well... that is not a very good reason for not to > change it. I'll change that to the next version of the patch. If we already have ai3x-micbias-vg then there's no reason to do differently here -- that'll just make things more confusing. > > >> + 0 or MICBIAS_OFF - MICBIAS output it not powered > > > > s/it/is/ > > > > To be fixed. Cheers. Mark. ^ permalink raw reply [flat|nested] 15+ messages in thread
* Re: [PATCH v4] ASoC: tlv320aic31xx: Add basic codec driver implementation 2014-03-10 8:52 ` [PATCH v4] ASoC: tlv320aic31xx: Add basic codec driver implementation Jyri Sarha 2014-03-10 9:39 ` Lokesh Vutla 2014-03-10 12:09 ` Mark Rutland @ 2014-03-10 12:58 ` Mark Brown [not found] ` <20140310125848.GI28112-GFdadSzt00ze9xe1eoZjHA@public.gmane.org> 2 siblings, 1 reply; 15+ messages in thread From: Mark Brown @ 2014-03-10 12:58 UTC (permalink / raw) To: Jyri Sarha Cc: alsa-devel, devicetree, linux-omap, peter.ujfalusi, liam.r.girdwood, bcousson, detheridge [-- Attachment #1: Type: text/plain, Size: 2645 bytes --] On Mon, Mar 10, 2014 at 10:52:21AM +0200, Jyri Sarha wrote: > +- ai31xx-micbias-vg - MicBias Voltage setting > + 0 or MICBIAS_OFF - MICBIAS output it not powered So, on every other version of this patch set I've suggested removing this as there's no reason why the bias would be wired up but disabled. Each time you seem to agree that the option should be removed yet here it is again... > +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); The core should already be logging well enough for this. > + switch (event) { > + case SND_SOC_DAPM_POST_PMU: > + return aic31xx_wait_bits(aic31xx, reg, mask, mask, 5000, 100); > + case 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); switch with a default would be more idiomatic here. > +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"); > + } Those error messages are real errors - you should print them out and also return the error code. Similarly for DAPM. > + if (event & REGULATOR_EVENT_DISABLE) { > + /* > + * Put codec to reset and as at least one of the > + * supplies was disabled. It may be to late to try > + * soft reset, but it should not do any harm. > + * Resetting the codec helps in minimizing leakage > + * in case some of the regulators remains turned on. > + */ > + if (gpio_is_valid(aic31xx->pdata.gpio_reset)) > + gpio_set_value(aic31xx->pdata.gpio_reset, 0); > + else > + snd_soc_write(aic31xx->codec, AIC31XX_RESET, 0x01); I'd expect this write may fail (potentially loudly and annoyingly) if the device really has just had its power pulled. [-- Attachment #2: Digital signature --] [-- Type: application/pgp-signature, Size: 836 bytes --] ^ permalink raw reply [flat|nested] 15+ messages in thread
[parent not found: <20140310125848.GI28112-GFdadSzt00ze9xe1eoZjHA@public.gmane.org>]
* Re: [PATCH v4] ASoC: tlv320aic31xx: Add basic codec driver implementation [not found] ` <20140310125848.GI28112-GFdadSzt00ze9xe1eoZjHA@public.gmane.org> @ 2014-03-10 13:25 ` Jyri Sarha 2014-03-10 15:47 ` Mark Brown 0 siblings, 1 reply; 15+ messages in thread From: Jyri Sarha @ 2014-03-10 13:25 UTC (permalink / raw) To: Mark Brown Cc: alsa-devel-K7yf7f+aM1XWsZ/bQMPhNw, devicetree-u79uwXL29TY76Z2rM5mHXA, linux-omap-u79uwXL29TY76Z2rM5mHXA, peter.ujfalusi-l0cyMroinI0, liam.r.girdwood-VuQAYsv1563Yd54FQh9/CA, bcousson-rdvid1DuHRBWk0Htik3J/w, detheridge-l0cyMroinI0 On 03/10/2014 02:58 PM, Mark Brown wrote: > On Mon, Mar 10, 2014 at 10:52:21AM +0200, Jyri Sarha wrote: > >> >+- ai31xx-micbias-vg - MicBias Voltage setting >> >+ 0 or MICBIAS_OFF - MICBIAS output it not powered > So, on every other version of this patch set I've suggested removing > this as there's no reason why the bias would be wired up but disabled. > Each time you seem to agree that the option should be removed yet here > it is again... > I agreed on not to have it as the default. I can still imagine couple of reason to have this option around. For instance, if the board designer decides to pull the bias voltage from outside of the chip completely, or like in my test setup when I feed the mic input from a headset output of another device, because I do not have a microphone around ATM. But if you want it removed, I'll do it. Best regards, Jyri -- To unsubscribe from this list: send the line "unsubscribe devicetree" in the body of a message to majordomo-u79uwXL29TY76Z2rM5mHXA@public.gmane.org More majordomo info at http://vger.kernel.org/majordomo-info.html ^ permalink raw reply [flat|nested] 15+ messages in thread
* Re: [PATCH v4] ASoC: tlv320aic31xx: Add basic codec driver implementation 2014-03-10 13:25 ` Jyri Sarha @ 2014-03-10 15:47 ` Mark Brown 2014-03-10 16:05 ` Jyri Sarha 0 siblings, 1 reply; 15+ messages in thread From: Mark Brown @ 2014-03-10 15:47 UTC (permalink / raw) To: Jyri Sarha Cc: alsa-devel, devicetree, linux-omap, peter.ujfalusi, liam.r.girdwood, bcousson, detheridge [-- Attachment #1: Type: text/plain, Size: 1094 bytes --] On Mon, Mar 10, 2014 at 03:25:41PM +0200, Jyri Sarha wrote: > On 03/10/2014 02:58 PM, Mark Brown wrote: > >On Mon, Mar 10, 2014 at 10:52:21AM +0200, Jyri Sarha wrote: > >>>+- ai31xx-micbias-vg - MicBias Voltage setting > >>>+ 0 or MICBIAS_OFF - MICBIAS output it not powered > >So, on every other version of this patch set I've suggested removing > >this as there's no reason why the bias would be wired up but disabled. > >Each time you seem to agree that the option should be removed yet here > >it is again... > I agreed on not to have it as the default. I can still imagine > couple of reason to have this option around. For instance, if the > board designer decides to pull the bias voltage from outside of the > chip completely, or like in my test setup when I feed the mic input > from a headset output of another device, because I do not have a > microphone around ATM. To repeat what I said first time around in those cases the pin wouldn't be connected at all so why would it ever be enabled? You're not specifying that it's going to be always enabled at a given voltage. [-- Attachment #2: Digital signature --] [-- Type: application/pgp-signature, Size: 836 bytes --] ^ permalink raw reply [flat|nested] 15+ messages in thread
* Re: [PATCH v4] ASoC: tlv320aic31xx: Add basic codec driver implementation 2014-03-10 15:47 ` Mark Brown @ 2014-03-10 16:05 ` Jyri Sarha 2014-03-10 19:23 ` Jyri Sarha 0 siblings, 1 reply; 15+ messages in thread From: Jyri Sarha @ 2014-03-10 16:05 UTC (permalink / raw) To: Mark Brown Cc: alsa-devel, devicetree, linux-omap, peter.ujfalusi, liam.r.girdwood, bcousson, detheridge On 03/10/2014 05:47 PM, Mark Brown wrote: > On Mon, Mar 10, 2014 at 03:25:41PM +0200, Jyri Sarha wrote: >> On 03/10/2014 02:58 PM, Mark Brown wrote: >>> On Mon, Mar 10, 2014 at 10:52:21AM +0200, Jyri Sarha wrote: ... > >> I agreed on not to have it as the default. I can still imagine >> couple of reason to have this option around. For instance, if the >> board designer decides to pull the bias voltage from outside of the >> chip completely, or like in my test setup when I feed the mic input >> from a headset output of another device, because I do not have a >> microphone around ATM. > > To repeat what I said first time around in those cases the pin wouldn't > be connected at all so why would it ever be enabled? You're not > specifying that it's going to be always enabled at a given voltage. > What ever. My laptop headset output does not seem to mind about bias voltage being fed to it. In the first case the pin wouldn't be connected so it does not matter. I'll remove the option. Best regards, Jyri ^ permalink raw reply [flat|nested] 15+ messages in thread
* Re: [PATCH v4] ASoC: tlv320aic31xx: Add basic codec driver implementation 2014-03-10 16:05 ` Jyri Sarha @ 2014-03-10 19:23 ` Jyri Sarha 0 siblings, 0 replies; 15+ messages in thread From: Jyri Sarha @ 2014-03-10 19:23 UTC (permalink / raw) To: Mark Brown Cc: alsa-devel, devicetree, linux-omap, peter.ujfalusi, liam.r.girdwood, bcousson, detheridge On 03/10/2014 06:05 PM, Jyri Sarha wrote: > On 03/10/2014 05:47 PM, Mark Brown wrote: >> On Mon, Mar 10, 2014 at 03:25:41PM +0200, Jyri Sarha wrote: >>> On 03/10/2014 02:58 PM, Mark Brown wrote: >>>> On Mon, Mar 10, 2014 at 10:52:21AM +0200, Jyri Sarha wrote: > ... >> >>> I agreed on not to have it as the default. I can still imagine >>> couple of reason to have this option around. For instance, if the >>> board designer decides to pull the bias voltage from outside of the >>> chip completely, or like in my test setup when I feed the mic input >>> from a headset output of another device, because I do not have a >>> microphone around ATM. >> >> To repeat what I said first time around in those cases the pin wouldn't >> be connected at all so why would it ever be enabled? You're not >> specifying that it's going to be always enabled at a given voltage. >> > > What ever. My laptop headset output does not seem to mind about bias > voltage being fed to it. In the first case the pin wouldn't be connected > so it does not matter. I'll remove the option. > Damn, I finally understood what you meant. I removed the explicit route to MICBIAS from the codec driver and left it to be added by a machine driver. Also removed all references to MIBIAS_OFF DT parameter. Best regards, Jyri ^ permalink raw reply [flat|nested] 15+ messages in thread
* Re: [PATCH v4] AM43xx-ePOS-EVM audio support with TLV320AIC31XX driver 2014-03-10 8:52 [PATCH v4] AM43xx-ePOS-EVM audio support with TLV320AIC31XX driver Jyri Sarha 2014-03-10 8:52 ` [PATCH v4] ASoC: tlv320aic31xx: Add basic codec driver implementation Jyri Sarha @ 2014-03-10 9:13 ` Mark Brown 2014-03-10 9:26 ` Jyri Sarha 1 sibling, 1 reply; 15+ messages in thread From: Mark Brown @ 2014-03-10 9:13 UTC (permalink / raw) To: Jyri Sarha Cc: alsa-devel, devicetree, linux-omap, peter.ujfalusi, liam.r.girdwood, bcousson, detheridge [-- Attachment #1: Type: text/plain, Size: 319 bytes --] On Mon, Mar 10, 2014 at 10:52:20AM +0200, Jyri Sarha wrote: > Since the v3 version of the patches: > - Dropped "ASoC: tlv320aic32x4: Sort Makefile in alphabetic order" out Please don't send cover letters for single patches, if there's anything important put it in the patch after the --- (or in the changelog). [-- Attachment #2: Digital signature --] [-- Type: application/pgp-signature, Size: 836 bytes --] ^ permalink raw reply [flat|nested] 15+ messages in thread
* Re: [PATCH v4] AM43xx-ePOS-EVM audio support with TLV320AIC31XX driver 2014-03-10 9:13 ` [PATCH v4] AM43xx-ePOS-EVM audio support with TLV320AIC31XX driver Mark Brown @ 2014-03-10 9:26 ` Jyri Sarha 0 siblings, 0 replies; 15+ messages in thread From: Jyri Sarha @ 2014-03-10 9:26 UTC (permalink / raw) To: Mark Brown Cc: alsa-devel, devicetree, linux-omap, peter.ujfalusi, liam.r.girdwood, bcousson, detheridge On 03/10/2014 11:13 AM, Mark Brown wrote: > On Mon, Mar 10, 2014 at 10:52:20AM +0200, Jyri Sarha wrote: >> Since the v3 version of the patches: >> - Dropped "ASoC: tlv320aic32x4: Sort Makefile in alphabetic order" out > > Please don't send cover letters for single patches, if there's anything > important put it in the patch after the --- (or in the changelog). > Just wanted to indicate that this is still the same series, even if there is only one patch left... but won't do that anymore. BR, Jyri ^ permalink raw reply [flat|nested] 15+ messages in thread
end of thread, other threads:[~2014-03-10 19:23 UTC | newest]
Thread overview: 15+ messages (download: mbox.gz follow: Atom feed
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2014-03-10 8:52 [PATCH v4] AM43xx-ePOS-EVM audio support with TLV320AIC31XX driver Jyri Sarha
2014-03-10 8:52 ` [PATCH v4] ASoC: tlv320aic31xx: Add basic codec driver implementation Jyri Sarha
2014-03-10 9:39 ` Lokesh Vutla
2014-03-10 9:45 ` Mark Brown
2014-03-10 12:09 ` Mark Rutland
2014-03-10 12:11 ` Mark Brown
2014-03-10 13:15 ` Jyri Sarha
2014-03-10 13:31 ` Mark Rutland
2014-03-10 12:58 ` Mark Brown
[not found] ` <20140310125848.GI28112-GFdadSzt00ze9xe1eoZjHA@public.gmane.org>
2014-03-10 13:25 ` Jyri Sarha
2014-03-10 15:47 ` Mark Brown
2014-03-10 16:05 ` Jyri Sarha
2014-03-10 19:23 ` Jyri Sarha
2014-03-10 9:13 ` [PATCH v4] AM43xx-ePOS-EVM audio support with TLV320AIC31XX driver Mark Brown
2014-03-10 9:26 ` Jyri Sarha
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