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

[alsa-devel] [PATCH] ASoC: tlv320aic31xx: Remove snd_soc_codec_set_cache_io() call
by Jyri Sarha 13 Mar '14

13 Mar '14

Remove snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP) call and codec->control_data = aic31xx->regmap assignment since that already done by core. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- This patch should fix the last minute breakkage of v3.15 ASoC updates build. sound/soc/codecs/tlv320aic31xx.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index c9f6989..625b184 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1051,18 +1051,9 @@ static int aic31xx_codec_probe(struct snd_soc_codec *codec) 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; -- 1.7.9.5

2 1

[alsa-devel] [PATCH v6] ASoC: tlv320aic31xx: Fix unused variable warning from aic31xx_clk_off
by Jyri Sarha 13 Mar '14

13 Mar '14

Fix "warning: unused variable 'aic31xx'" from function 'aic31xx_clk_off'. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- sound/soc/codecs/tlv320aic31xx.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index e626b56..c9f6989 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -945,7 +945,6 @@ static void aic31xx_clk_on(struct snd_soc_codec *codec) 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; -- 1.7.9.5

2 1

[alsa-devel] [PATCH v6] ASoC: tlv320aic31xx: Add basic codec driver implementation
by Jyri Sarha 13 Mar '14

13 Mar '14

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(a)ti.com> --- Since v5 of this patch: - Fix "warning: unused variable 'aic31xx'" from function 'aic31xx_clk_off' .../devicetree/bindings/sound/tlv320aic31xx.txt | 61 + include/dt-bindings/sound/tlv320aic31xx-micbias.h | 8 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic31xx.c | 1294 ++++++++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 258 ++++ 6 files changed, 1627 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..74c66de --- /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 + 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 + * MICBIAS + +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 = <&regulator>; + SPRVDD-supply = <&regulator>; + SPLVDD-supply = <&regulator>; + AVDD-supply = <&regulator>; + IOVDD-supply = <&regulator>; + DVDD-supply = <&regulator>; +}; diff --git a/include/dt-bindings/sound/tlv320aic31xx-micbias.h b/include/dt-bindings/sound/tlv320aic31xx-micbias.h new file mode 100644 index 0000000..f5cb772 --- /dev/null +++ b/include/dt-bindings/sound/tlv320aic31xx-micbias.h @@ -0,0 +1,8 @@ +#ifndef __DT_TLV320AIC31XX_MICBIAS_H +#define __DT_TLV320AIC31XX_MICBIAS_H + +#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..bdc0d8b --- /dev/null +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -0,0 +1,1294 @@ +/* + * ALSA SoC TLV320AIC31XX codec driver + * + * Copyright (C) 2014 Texas Instruments, Inc. + * + * Author: Jyri Sarha <jsarha(a)ti.com> + * + * Based on ground work by: Ajit Kulkarni <x0175765(a)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; +}; + +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; + + 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); + default: + 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 */ + 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: + /* turn mic bias off */ + snd_soc_update_bits(codec, AIC31XX_MICBIAS, + AIC31XX_MICBIAS_MASK, 0); + dev_dbg(codec->dev, "%s: turned off\n", __func__); + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAC IN", "DAC Playback", 0, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("DAC Left Input", + SND_SOC_NOPM, 0, 0, &ldac_in_control), + SND_SOC_DAPM_MUX("DAC Right Input", + SND_SOC_NOPM, 0, 0, &rdac_in_control), + /* DACs */ + SND_SOC_DAPM_DAC_E("DAC Left", "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"), + + /* Inputs */ + SND_SOC_DAPM_INPUT("MIC1LP"), + SND_SOC_DAPM_INPUT("MIC1RP"), + SND_SOC_DAPM_INPUT("MIC1LM"), +}; + +static const struct snd_soc_dapm_widget aic311x_dapm_widgets[] = { + /* AIC3111 and AIC3110 have stereo class-D amplifier */ + SND_SOC_DAPM_OUT_DRV_E("SPL ClassD", AIC31XX_SPKAMP, 7, 0, NULL, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUT_DRV_E("SPR ClassD", AIC31XX_SPKAMP, 6, 0, NULL, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SWITCH("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[] = { + /* 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"}, + + /* 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 ret = 0; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + + if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT) + ret = snd_soc_add_codec_controls( + codec, aic311x_snd_controls, + ARRAY_SIZE(aic311x_snd_controls)); + else + ret = snd_soc_add_codec_controls( + codec, aic310x_snd_controls, + ARRAY_SIZE(aic310x_snd_controls)); + + return ret; +} + +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) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, aic311x_audio_map, + ARRAY_SIZE(aic311x_audio_map)); + if (ret) + return ret; + } else { + ret = snd_soc_dapm_new_controls( + dapm, aic310x_dapm_widgets, + ARRAY_SIZE(aic310x_dapm_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, aic310x_audio_map, + ARRAY_SIZE(aic310x_audio_map)); + if (ret) + return 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; + } + + /* 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 width %d rate %d\n", + __func__, params_format(params), params_width(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. + */ + if (gpio_is_valid(aic31xx->pdata.gpio_reset)) + gpio_set_value(aic31xx->pdata.gpio_reset, 0); + + 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); +} + +static void aic31xx_clk_off(struct snd_soc_codec *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); +} + +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); + + ret = aic31xx_add_controls(codec); + if (ret) + return ret; + + ret = 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_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

2 1

[alsa-devel] ASoC updates for v3.15
by Mark Brown 13 Mar '14

13 Mar '14

The following changes since commit deeed33850c8a376addabbf971df433b2a1ba74c: Merge remote-tracking branches 'asoc/topic/wm8996', 'asoc/topic/wm9081' and 'asoc/topic/wm9705' into asoc-next (2014-03-12 23:05:31 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.15-2 for you to fetch changes up to 56fe48976db38ed3984e0e68b71e503e8179dd89: Merge remote-tracking branch 'asoc/topic/tlv320aic31xx' into asoc-next (2014-03-13 14:19:54 +0000) ---------------------------------------------------------------- ASoC: Updates for v3.15 This is mostly a few additional fixes from Lars-Peter, a new driver and cleaning up a git failure with merging the Intel branch (combined with an xargs failure to pay attention to error codes). The history lists a bunch of additional commits for the branch but the content of those commits is actually present already but not recorded in history due to git failing. Unfortunately xargs is used in the merge script and it doesn't do a good job of noticing errors from the commands it invokes. ---------------------------------------------------------------- Dan Carpenter (4): ASoC: Intel: sst-firmware: missing curly braces (harmless) ASoC: intel: restore IRQs on error ASoC: intel: incorrect sizeof() ASoC: Baytrail: fix error handling in sst_byt_dsp_init() Jarkko Nikula (15): ASoC: Intel: Add common SST driver loader on ACPI systems ASoC: Intel: Move extended fw base and size fields in struct sst_pdata ASoC: Intel: sst-acpi: Request firmware before SST platform driver probing ASoC: Intel: sst-acpi: Add support for multiple machine drivers per platform ASoC: Intel: Add Baytrail SST ID and Baytrail specific register bits ASoC: Intel: Add Intel Baytrail SST DSP support ASoC: Intel: Add Intel Baytrail SST DSP IPC support ASoC: Intel: Add Intel Baytrail SST PCM platform driver ASoC: Intel: Add machine driver for Baytrail SST with RT5640 codec ASoC: Intel: Add Baytrail SST and byt-rt5640 machine driver probing ASoC: Intel: Add build support for Baytrail SST ASoC: Intel: sst-acpi: Fix Oops in case of missing firmware ASoC: Intel: byt-rt5640: Update internal mic and speaker kcontrol names ASoC: Intel: byt-rt5640: Use init time DAI format ASoC: Intel: Baytrail: Fix implicit declaration of function 'memcpy_fromio' Jyri Sarha (1): ASoC: tlv320aic31xx: Add basic codec driver implementation Lars-Peter Clausen (5): ASoC: ams-delta: Fix compile error ASoC: snappercl15: Convert to table based DAPM setup ASoC: pxa: magician: Convert to table based DAPM and control setup ASoC: pxa: tosa: Convert to table based DAPM and control setup ASoC: s6105-ipcam: Convert to table based DAPM setup Liam Girdwood (18): ASoC: Intel: Add a mfld prefix to Intel SST drivers. ASoC: Intel: Add Intel SST audio DSP low level shim driver. ASoC: Intel: Add Intel SST audio DSP Firmware loader. ASoC: Intel: Add trace support for generic SST IPC messages. ASoC: Intel: Add build support for Intel SST DSP core. ASoC: Intel: Add GFP_KERNEL flag to firmware DMA buffer. ASoC: Intel: Rename SST trace event header to be less generic. ASoC: Intel: Fix sparse warnings for firmware loader ASoC: Intel: Fix build for sst-dsp.c on PPC architecture ASoC: Intel: Add support for Haswell/Broadwell DSP ASoC: Intel: Add Haswell/Broadwell IPC ASoC: Intel: Add Haswell and Broadwell PCM platform driver ASoC: Intel: Add trace support for Haswell/Broadwell SST IPC messages. ASoC: Intel: Add build support for Haswell ADSP ASoC: Intel: Add Haswell Machine support ASoC: Intel: Check Haswell IPC process_reply/notification return value. ASoC: Intel: Use .dai_fmt for setting Haswell BE format. ASoC: Intel: Clean up indentation for Haswell machine driver/Kconfig Mark Brown (26): MAINTAINERS: Add entry for Cirrus CODEC drivers ASoC: io: Remove SPI support Merge remote-tracking branches 'asoc/topic/ak4671', 'asoc/topic/cs42l51' and 'asoc/topic/alc5623' into asoc-io ASoC: io: Remove SND_SOC_I2C ASoC: da732x: Replace hw_read usage with snd_soc_read() ASoC: io: Remove hw_read() operation Merge branches 'topic/ad193x', 'topic/tlv320aic23', 'topic/tlv320aic32x4', 'topic/wm8991', 'fix/si476x' and 'fix/88pm860' of git://git.kernel.org/.../broonie/sound into asoc-io Merge branch 'topic/dapm' of git://git.kernel.org/.../broonie/sound into asoc-pxa Merge tag 'asoc-v3.15' into asoc-intel Merge tag 'asoc-v3.14-rc6' into asoc-linus Merge remote-tracking branch 'asoc/fix/spear' into asoc-linus Merge tag 'asoc-v3.15' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/cirrus' into asoc-next Merge remote-tracking branch 'asoc/topic/intel' into asoc-next Merge remote-tracking branch 'asoc/topic/io' into asoc-next Merge remote-tracking branch 'asoc/topic/jack' into asoc-next Merge remote-tracking branch 'asoc/topic/kirkwood' into asoc-next Merge remote-tracking branch 'asoc/topic/lm4857' into asoc-next Merge remote-tracking branch 'asoc/topic/lm49453' into asoc-next Merge remote-tracking branch 'asoc/topic/maintainers' into asoc-next Merge remote-tracking branch 'asoc/topic/max98088' into asoc-next Merge remote-tracking branch 'asoc/topic/max98095' into asoc-next Merge remote-tracking branch 'asoc/topic/pxa' into asoc-next Merge remote-tracking branch 'asoc/topic/s6000' into asoc-next Merge remote-tracking branch 'asoc/topic/tlv320aic31xx' into asoc-next Stephen Warren (1): ASoC: Intel: don't select RT5640 if !I2C Takashi Iwai (4): ASoC: lm4857: Use SOC_ENUM_SINGLE_EXT_DECL() ASoC: max98088: Use SOC_*_ENUM_SINGLE_DECL() ASoC: max98095: Use SOC_ENUM_SINGLE_DECL() ASoC: lm49453: Use SOC_ENUM_SINGLE_DECL() Thomas Petazzoni (2): ASoC: kirkwood: enable Kirkwood driver for mvebu platforms sound: ASoC: add ASoC board driver for Armada 370 DB Xiubo Li (6): ASoC: io: Clean up snd_soc_codec_set_cache_io() ASoC: core: Set the default I/O up try regmap. ASoC: sgtl5000: Simplify ASoC probe code ASoC: codec: Simplify ASoC probe code. ASoC: io: New signature for snd_soc_codec_set_cache_io() ASoC: core: Fix check before setting default I/O up try regmap xiangxiao (2): ASoC: add data field into snd_soc_jack_gpio ASoC: delay the initial jack detect by debounce_time .../bindings/sound/armada-370db-audio.txt | 24 + .../devicetree/bindings/sound/mvebu-audio.txt | 1 + .../devicetree/bindings/sound/tlv320aic31xx.txt | 61 + MAINTAINERS | 7 + include/dt-bindings/sound/tlv320aic31xx-micbias.h | 8 + include/sound/soc.h | 13 +- sound/soc/cirrus/snappercl15.c | 18 +- sound/soc/codecs/88pm860x-codec.c | 3 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ad193x.c | 10 +- sound/soc/codecs/adau1373.c | 7 - sound/soc/codecs/adav80x.c | 7 - sound/soc/codecs/ak4535.c | 9 - sound/soc/codecs/ak4641.c | 8 - sound/soc/codecs/ak4642.c | 8 - sound/soc/codecs/ak4671.c | 12 +- sound/soc/codecs/alc5623.c | 7 - sound/soc/codecs/alc5632.c | 8 - sound/soc/codecs/cq93vc.c | 3 +- sound/soc/codecs/cs4270.c | 9 - sound/soc/codecs/cs42l51.c | 6 - sound/soc/codecs/cs42l52.c | 9 +- sound/soc/codecs/cs42l73.c | 11 +- sound/soc/codecs/da7210.c | 8 - sound/soc/codecs/da7213.c | 8 - sound/soc/codecs/da732x.c | 29 +- sound/soc/codecs/da9055.c | 8 - sound/soc/codecs/isabelle.c | 16 - sound/soc/codecs/lm4857.c | 3 +- sound/soc/codecs/lm49453.c | 31 +- sound/soc/codecs/max9768.c | 5 - sound/soc/codecs/max98088.c | 45 +- sound/soc/codecs/max98090.c | 8 - sound/soc/codecs/max98095.c | 56 +- sound/soc/codecs/max9850.c | 8 - sound/soc/codecs/mc13783.c | 4 +- sound/soc/codecs/ml26124.c | 10 - sound/soc/codecs/rt5631.c | 9 - sound/soc/codecs/rt5640.c | 8 - sound/soc/codecs/sgtl5000.c | 8 - sound/soc/codecs/si476x.c | 6 +- sound/soc/codecs/sn95031.c | 2 - sound/soc/codecs/ssm2518.c | 10 - sound/soc/codecs/ssm2602.c | 7 - sound/soc/codecs/sta32x.c | 14 - sound/soc/codecs/sta529.c | 10 - sound/soc/codecs/tlv320aic23.c | 8 - sound/soc/codecs/tlv320aic26.c | 2 - sound/soc/codecs/tlv320aic31xx.c | 1295 ++++++++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 258 ++++ sound/soc/codecs/tlv320aic32x4.c | 2 - sound/soc/codecs/tlv320aic3x.c | 6 - sound/soc/codecs/tlv320dac33.c | 1 - sound/soc/codecs/wm2000.c | 2 - sound/soc/codecs/wm2200.c | 7 - sound/soc/codecs/wm5100.c | 7 - sound/soc/codecs/wm5102.c | 4 +- sound/soc/codecs/wm5110.c | 3 +- sound/soc/codecs/wm8350.c | 4 +- sound/soc/codecs/wm8400.c | 3 +- sound/soc/codecs/wm8510.c | 10 +- sound/soc/codecs/wm8523.c | 7 - sound/soc/codecs/wm8580.c | 6 - sound/soc/codecs/wm8711.c | 6 - sound/soc/codecs/wm8728.c | 11 +- sound/soc/codecs/wm8731.c | 7 - sound/soc/codecs/wm8737.c | 6 - sound/soc/codecs/wm8741.c | 6 - sound/soc/codecs/wm8750.c | 6 - sound/soc/codecs/wm8753.c | 7 - sound/soc/codecs/wm8770.c | 6 - sound/soc/codecs/wm8776.c | 6 - sound/soc/codecs/wm8804.c | 8 - sound/soc/codecs/wm8900.c | 8 +- sound/soc/codecs/wm8903.c | 10 +- sound/soc/codecs/wm8904.c | 9 - sound/soc/codecs/wm8940.c | 6 - sound/soc/codecs/wm8955.c | 8 - sound/soc/codecs/wm8960.c | 6 - sound/soc/codecs/wm8961.c | 7 - sound/soc/codecs/wm8962.c | 7 - sound/soc/codecs/wm8971.c | 6 - sound/soc/codecs/wm8974.c | 6 - sound/soc/codecs/wm8978.c | 8 +- sound/soc/codecs/wm8983.c | 6 - sound/soc/codecs/wm8985.c | 7 - sound/soc/codecs/wm8988.c | 8 - sound/soc/codecs/wm8990.c | 8 - sound/soc/codecs/wm8991.c | 8 - sound/soc/codecs/wm8993.c | 7 - sound/soc/codecs/wm8994.c | 3 +- sound/soc/codecs/wm8995.c | 7 - sound/soc/codecs/wm8996.c | 12 +- sound/soc/codecs/wm8997.c | 4 +- sound/soc/codecs/wm9081.c | 11 +- sound/soc/codecs/wm9090.c | 10 - sound/soc/intel/Kconfig | 4 +- sound/soc/kirkwood/Kconfig | 10 +- sound/soc/kirkwood/Makefile | 2 + sound/soc/kirkwood/armada-370-db.c | 120 ++ sound/soc/kirkwood/kirkwood-i2s.c | 1 + sound/soc/omap/ams-delta.c | 2 +- sound/soc/pxa/magician.c | 34 +- sound/soc/pxa/tosa.c | 35 +- sound/soc/s6000/s6105-ipcam.c | 28 +- sound/soc/soc-core.c | 14 +- sound/soc/soc-io.c | 64 +- sound/soc/soc-jack.c | 5 +- 109 files changed, 1958 insertions(+), 795 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/armada-370db-audio.txt 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 create mode 100644 sound/soc/kirkwood/armada-370-db.c

4 3

[alsa-devel] [PATCH v5] ASoC: tlv320aic31xx: Add basic codec driver implementation
by Jyri Sarha 13 Mar '14

13 Mar '14

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(a)ti.com> --- Since v4 version: - Remove MICBIAS_OFF DT parameter - Remove logging and add missing default: to aic31xx_dapm_power_event() - Take control, widget, and route adding errors into account - Don't try soft reset in power event handler - Remove route to MICBIAS from codec driver damp routing table and add it as output pin to DT doc - Update year in copyright message and change triple newlines to double - Remove internal DAC/ADC routes that were still haunting in the previous patch - Allow PLL reprogramming on the fly as it appears to work just fine BR, Jyri .../devicetree/bindings/sound/tlv320aic31xx.txt | 61 + include/dt-bindings/sound/tlv320aic31xx-micbias.h | 8 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic31xx.c | 1295 ++++++++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 258 ++++ 6 files changed, 1628 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..74c66de --- /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 + 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 + * MICBIAS + +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 = <&regulator>; + SPRVDD-supply = <&regulator>; + SPLVDD-supply = <&regulator>; + AVDD-supply = <&regulator>; + IOVDD-supply = <&regulator>; + DVDD-supply = <&regulator>; +}; diff --git a/include/dt-bindings/sound/tlv320aic31xx-micbias.h b/include/dt-bindings/sound/tlv320aic31xx-micbias.h new file mode 100644 index 0000000..f5cb772 --- /dev/null +++ b/include/dt-bindings/sound/tlv320aic31xx-micbias.h @@ -0,0 +1,8 @@ +#ifndef __DT_TLV320AIC31XX_MICBIAS_H +#define __DT_TLV320AIC31XX_MICBIAS_H + +#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..e60e37b --- /dev/null +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -0,0 +1,1295 @@ +/* + * ALSA SoC TLV320AIC31XX codec driver + * + * Copyright (C) 2014 Texas Instruments, Inc. + * + * Author: Jyri Sarha <jsarha(a)ti.com> + * + * Based on ground work by: Ajit Kulkarni <x0175765(a)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; +}; + +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; + + 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); + default: + 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 */ + 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: + /* turn mic bias off */ + snd_soc_update_bits(codec, AIC31XX_MICBIAS, + AIC31XX_MICBIAS_MASK, 0); + dev_dbg(codec->dev, "%s: turned off\n", __func__); + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAC IN", "DAC Playback", 0, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("DAC Left Input", + SND_SOC_NOPM, 0, 0, &ldac_in_control), + SND_SOC_DAPM_MUX("DAC Right Input", + SND_SOC_NOPM, 0, 0, &rdac_in_control), + /* DACs */ + SND_SOC_DAPM_DAC_E("DAC Left", "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"), + + /* Inputs */ + SND_SOC_DAPM_INPUT("MIC1LP"), + SND_SOC_DAPM_INPUT("MIC1RP"), + SND_SOC_DAPM_INPUT("MIC1LM"), +}; + +static const struct snd_soc_dapm_widget aic311x_dapm_widgets[] = { + /* AIC3111 and AIC3110 have stereo class-D amplifier */ + SND_SOC_DAPM_OUT_DRV_E("SPL ClassD", AIC31XX_SPKAMP, 7, 0, NULL, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUT_DRV_E("SPR ClassD", AIC31XX_SPKAMP, 6, 0, NULL, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SWITCH("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[] = { + /* 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"}, + + /* 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 ret = 0; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + + if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT) + ret = snd_soc_add_codec_controls( + codec, aic311x_snd_controls, + ARRAY_SIZE(aic311x_snd_controls)); + else + ret = snd_soc_add_codec_controls( + codec, aic310x_snd_controls, + ARRAY_SIZE(aic310x_snd_controls)); + + return ret; +} + +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) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, aic311x_audio_map, + ARRAY_SIZE(aic311x_audio_map)); + if (ret) + return ret; + } else { + ret = snd_soc_dapm_new_controls( + dapm, aic310x_dapm_widgets, + ARRAY_SIZE(aic310x_dapm_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, aic310x_audio_map, + ARRAY_SIZE(aic310x_audio_map)); + if (ret) + return 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; + } + + /* 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 width %d rate %d\n", + __func__, params_format(params), params_width(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. + */ + if (gpio_is_valid(aic31xx->pdata.gpio_reset)) + gpio_set_value(aic31xx->pdata.gpio_reset, 0); + + 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); +} + +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); +} + +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); + + ret = aic31xx_add_controls(codec); + if (ret) + return ret; + + ret = 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_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

2 2

[alsa-devel] zoom r16 quirks
by James Miller 13 Mar '14

13 Mar '14

Hi, I am trying to get the zoom R16 working with alsa for full duplex operation. I have been able to get the 8 channels of capture working after following jmancine's advice and using the following quirk on ifnum 4: { .ifnum = 4, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_LE, .channels = 8, .iface = 1, .altsetting = 1, .altset_idx = 1, .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, .rate_min = 44100, .rate_max = 96000, .nr_rates = 4, .rate_table = (unsigned int[]) { 44100, 48000, 88200, 96000 } } }, However, what I discovered was that the ifnum 4 in this case is related to the udev matching rules for the device number, so when I plugged in the zoom, it was being assigned to device 4, which was then matched to the ifnum 4. If I unplugged and replugged the device, it was then designated as device 5 in the udev tree, and wasn't working. I read in quirks-table.h the following: (To work with * hotplugging, bDeviceClass must be set to USB_CLASS_PER_INTERFACE.) so I changed the matching rule macro to : USB_DEVICE_INTERFACE_CLASS(0x1686, 0x00dd, USB_CLASS_AUDIO) to attempt to match the device during udev assignment (in case the device class wasn't returned from the device properly) and then set .binterfaceclass = USB_CLASS_PER_INTERFACE below that before the quirk data. I think that I am heading in the correct direction, although I was wondering if you could confirm the following: Am I correct in assuming that the .ifnum in the quirk is related to udev device matching rules? When I write zoom-quirks.h in the way that I am currently (see below), I get a incorrect maxpacket assignment for interface 3 (midi) and I traced this to: dev_warn(ddev, "config %d interface %d altsetting %d " "bulk endpoint 0x%X has invalid maxpacket %d\n", in config.c line 259. The above bit of code is selected in conditions where the device is usb high speed but I have kern.log reporting that the device is class 1.1. Its a 24 bit chip, and jmancine reports that the device is being incorrectly assigned a 32bit le format for playback when it should be 24bit le, so perhaps this is because of a protocol mismatch? When I have the device successfully recognised the next time, I will check in format.c that it is being assigned the correct bit format, but I am uncertain as to how to confirm what bprotocol to use. Any ideas? Here is the zoom-quirks.h that I am currently using. { /* ZOOM R16 in USB 2.0 mode */ USB_DEVICE_INTERFACE_CLASS(0x1686, 0x00dd, USB_CLASS_AUDIO), .bDeviceClass = USB_CLASS_PER_INTERFACE, .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 1, /*PLAYBACK*/ .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_LE, .channels = 2, .iface = 1, .altsetting = 1, .altset_idx = 1, .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, .endpoint = 0x03, .ep_attr = (USB_ENDPOINT_XFER_ISOC && USB_ENDPOINT_SYNC_ADAPTIVE && SND_USB_ENDPOINT_TYPE_DATA), .rate_min = 44100, .rate_max = 96000, .nr_rates = 4, .rate_table = (unsigned int[]) { 44100, 48000, 88200, 96000 } } }, { .ifnum = 2, /*CAPTURE*/ .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_LE, .channels = 8, .iface = 2, .altsetting = 1, .altset_idx = 1, .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, .endpoint = 0x84, .ep_attr = (USB_ENDPOINT_XFER_ISOC && USB_ENDPOINT_SYNC_SYNC && USB_ENDPOINT_USAGE_DATA), .rate_min = 44100, .rate_max = 96000, .nr_rates = 4, .rate_table = (unsigned int[]) { 44100, 48000, 88200, 96000 } } }, { .ifnum = 3, .type = QUIRK_MIDI_FIXED_ENDPOINT, }, { .ifnum = -1, }, } } }, Thanks for any help!! -- James Stewart Miller Bsc(hons) Psych.

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[alsa-devel] [PATCH] ALSA: hda - Fix pop noise on Dell Inspiron 7537
by Takashi Iwai 13 Mar '14

13 Mar '14

Dell Inspiron 7537 shows loud pop noises at booting or shutdown. They can be reduced by avoiding the shutup call that is used for this codec. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=71861 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b50114176736..c31254a20331 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3868,6 +3868,7 @@ enum { ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, + ALC283_FIXUP_DELL1_SHUTUP, ALC269_FIXUP_HEADSET_MODE, ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, ALC269_FIXUP_ASUS_X101_FUNC, @@ -4087,6 +4088,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC283_FIXUP_DELL1_SHUTUP] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_no_shutup, + .chained = true, + .chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + }, [ALC269_FIXUP_HEADSET_MODE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode, @@ -4322,7 +4329,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f8, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x05f9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05f9, "Dell Inspiron 7537", ALC283_FIXUP_DELL1_SHUTUP), SND_PCI_QUIRK(0x1028, 0x05fb, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), -- 1.9.0

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[alsa-devel] [GIT PULL] ASoC updates for v3.15
by Mark Brown 13 Mar '14

13 Mar '14

The following changes since commit fa389e220254c69ffae0d403eac4146171062d08: Linux 3.14-rc6 (2014-03-09 19:41:57 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.15 for you to fetch changes up to deeed33850c8a376addabbf971df433b2a1ba74c: Merge remote-tracking branches 'asoc/topic/wm8996', 'asoc/topic/wm9081' and 'asoc/topic/wm9705' into asoc-next (2014-03-12 23:05:31 +0000) ---------------------------------------------------------------- ASoC: Updates for v3.15 Quite a busy release for ASoC this time, more on janitorial work than exciting new features but welcome nontheless: - Lots of cleanups from Takashi for enumerations; the original API for these was error prone so he's refactored lots of code to use more modern APIs which avoid issues. - Elimination of the ASoC level wrappers for I2C and SPI moving us closer to converting to regmap completely and avoiding some randconfig hassle. - Provide both manually and transparently locked DAPM APIs rather than a mix of the two fixing some concurrency issues. - Start converting CODEC drivers to use separate bus interface drivers rather than having them all in one file helping avoid dependency issues. - DPCM support for Intel Haswell and Bay Trail platforms. - Lots of work on improvements for simple-card, DaVinci and the Renesas rcar drivers. - New drivers for Analog Devices ADAU1977, TI PCM512x and parts of the CSR SiRF SoC. ---------------------------------------------------------------- Aaro Koskinen (1): ASoC: n810: fix init with DT boot Adam Thomson (1): ASoC: da9055: Add DT support for CODEC Alexander Shiyan (1): ASoC: cirrus: Remove excess dependencies on SND_SOC Bo Shen (2): ASoC: atmel_ssc_dai: make option to choose clock Binding: atmel-ssc: add option to choose clock Charles Keepax (33): ASoC: wm5102: Improve EQ coefficient controls ASoC: wm5110: Improve EQ coefficient controls ASoC: wm8997: Improve EQ coefficient controls ASoC: wm_adsp: Add debug print to note that the DSP has shutdown ASoC: wm5102: Correct typo in EQ coefficient sizes ASoC: wm5110: Correct type in EQ coefficient sizes ASoC: wm8997: Correct typo in EQ coefficient sizes ASoC: dapm: Add helpers to lock/unlock DAPM mutex ASoC: dapm: Add unlocked version of snd_soc_dapm_sync ASoC: adav80x: Update locking around use of DAPM pin API ASoC: wm5100: Update locking around use of DAPM pin API ASoC: wm8962: Update locking around use of DAPM pin API ASoC: wm8994: Update locking around use of DAPM pin API ASoC: wm8996: Update locking around use of DAPM pin API ASoC: mfld_machine: Update locking around use of DAPM pin API ASoC: ams-delta: Update locking around use of DAPM pin API ASoC: omap: n810: Update locking around use of DAPM pin API ASoC: omap: rx51: Update locking around use of DAPM pin API ASoC: pxa: corgi: Update locking around use of DAPM pin API ASoC: pxa: magician: Update locking around use of DAPM pin API ASoC: pxa: spitz: Update locking around use of DAPM pin API ASoC: pxa: tosa: Update locking around use of DAPM pin API ASoC: wm_adsp: Split firmware load into smaller chunks ASoC: wm_adsp: Correct type specifier in printf ASoC: arizona: An OUTDIV of 1 is not valid, avoid this ASoC: arizona: Add defines for FLL configuration constants ASoC: arizona: Move set of OUTDIV in to arizona_apply_fll ASoC: arizona: Move calculation of FLL configuration ASoC: arizona: Don't pass Fout into arizona_calc_fll ASoC: arizona: Calculate OUTDIV first ASoC: arizona: Calculate FLL gain last mfd: arizona: Add support for new fratio encoding ASoC: arizona: Support new fratio encoding on the wm5110 rev D Christian Engelmayer (3): ASoC: wm8991: Remove unused pointer in wm8991_probe() ASoC: wm8995: Remove unused pointer in hp_supply_event() ASoC: wm8993: Remove unused pointer in wm8993_remove() Daniel Mack (1): ASoC: cs4271: claim reset GPIO in bus probe function Denis Carikli (1): ASoC: eukrea-tlv320: Add DT support. Fabio Estevam (4): ASoC: wm8737: Use IS_ENABLED() macro ASoC: wm8983: Use IS_ENABLED() macro ASoC: fsl_esai: Check the return value from clk_prepare_enable() ASoC: fsl: imx-pcm-fiq: Remove unneeded 'out' label Jean-Francois Moine (8): ASoC: simple-card: simplify code ASoC: simple-card: simplify code ASoC: simple-card: simplify code ASoC: simple-card: simplify code ASoC: simple-card: simplify code ASoC: simple-card: simplify code ASoC: simple-card: simplify code ASoC: simple-card: simplify code Jyri Sarha (3): ASoC: davinci-evm: Add named clock reference to DT bindings ASoC: davinci-mcasp: Set BCLK divider if McASP is BCLK master ASoC: tlv320aic32x4: Sort Makefile in alphabetic order Kuninori Morimoto (58): ASoC: rsnd: tidyup register naming of BUSIF_MODE ASoC: rsnd: cleanup debug information method ASoC: rsnd: remove meaningless rsnd_ssi_non() ASoC: rsnd: control SCU ops in probe timing ASoC: rsnd: add rsnd_scu_init(), and separate init/start ASoC: rsnd: remove meaningless function parameter ASoC: rsnd: merge rsnd_scu_start/stop() and rsnd_scu_transfer_start/stop() ASoC: rsnd: rsnd_dai_is_clk_master() can be shared ASoC: rsnd: remove pin sync option ASoC: rsnd: SSI_MODE0/1 settings goes to scu.c ASoC: rsnd: remove ssiu from ssi.c ASoC: rsnd: rename rsnd_scu_convert_rate_ctrl() ASoC: rsnd: explain SRC bypass mode settings in comment ASoC: rsnd: remove duplicate *priv from rsnd_dma ASoC: rsnd: non 0 is error on probe ASoC: rsnd: fixup Gen2 module naming ASoC: rsnd: don't use schedule_work() when rsnd_dma_start() ASoC: rsnd: SCU should be called before SSI ASoC: rsnd: clarify scu.c area ASoC: rsnd: Merge macros in scu.c ASoC: rsnd: rsnd_scu_hpbif_is_enable() become macro ASoC: rsnd: merge SRC clock timing/setting ASoC: rsnd: add rsnd_ssi_is_play() ASoC: rsnd: extracts Gen1/Gen2 common parts ASoC: rsnd: remove SSI dependent DMAEngine callback ASoC: rsnd: rsnd_ssi_probe() goes forwarder than rsnd_scu_probe() ASoC: rsnd: add Gen2 SRC and DMAEngine support ASoC: rsnd: use device dependency clock ASoC: rsnd: tidyup original for_each_rsnd_xxx macro ASoC: rsnd: set DIV_EN register on rsnd_adg_set_convert_clk_gen2() ASoC: rsnd: print error if there is SRC settings mismatch ASoC: simple-card: cpu_dai_name creates confusion when DT case ASoC: rsnd: move priv member settings to upper side ASoC: rsnd: move rsnd_mod_call() macro ASoC: rsnd: remove verbose function parameter ASoC: rsnd: remove verbose debug message from scu/ssi ASoC: rsnd: unify rdai naming ASoC: rsnd: tidyup RSND_SSI_xxx flags ASoC: rsnd: run rsnd_path_init() when probe() timing ASoC: rsnd: use mod array instead of list on rdai ASoC: rsnd: get ssi/scu from rsnd_dai_stream ASoC: rsnd: use devm_clk_get() instead of clk_get() ASoC: rsnd: use function pointer for each probe ASoC: rsnd: remove unused SSI_CONTROL ASoC: rsnd: modify rsnd_adg_ssi_ws_timing_gen2() parameter ASoC: rsnd: share reg_field and reduce memory ASoC: rsnd: add struct rsnd_dai_platform_info ASoC: rsnd: Get correct SCU ID ASoC: rsnd: add rsnd_scu_enable_ssi_irq() ASoC: rsnd: call rsnd_scu_ssi_mode_init() from SSI ASoC: rsnd: add probe/remove callback on rsnd_mod_ops ASoC: rsnd: use mod probe method on SCU ASoC: rsnd: use mod probe method on SSI ASoC: rsnd: nothing to do on rsnd_dai_remove() ASoC: rsnd: remove all rsnd_xxx_remove() ASoC: rsnd: rename scu to src ASoC: simple-card: tidyup cpu/codec dai_fmt settings for non-DT ASoC: simple-card: card name can be option Lars-Peter Clausen (58): ASoC: h1940_uda1380: Don't modify runtime->hw ASoC: rx1950_uda1380: Don't modify runtime->hw ASoC: cs4271: Remove outdated comment ASoC: cs42l73: Don't mix SNDRV_PCM_RATE_KNOT with specific rates ASoC: cs42l73: Constify rate constraints ASoC: ssm2602: Constify rate constraints ASoC: twl6040: Constify rate constraints ASoC: wm8741: Constify rate constraints ASoC: wm8988: Constify rate constraints ASoC: dapm: Consistently use unsigned int for register values ASoC: ad193x: Split SPI and I2C code into different modules ASoC: ssm2602: Split SPI and I2C code into different modules ASoC: adav80x: Use devm_kzalloc() ASoC: adav80x: Split SPI and I2C code into different modules ASoC: Add ADAU1977 CODEC driver ASoC: pxa: Remove superfluous locking ASoC: da732x: Use da732x->regmap instead of codec->control_data ASoC: wm8996: Replace codec->control_data with wm8996->regmap ASoC: wm8962: Replace codec->control_data with wm8962->regmap ASoC: Remove unused 'reg2' field from soc_enum struct ASoC: Consolidate enum and value enum controls ASoC: dapm: Consolidate MUXs and value MUXs ASoC: Add macros for defining virtual enums ASoC: adau1373: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: max98090: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: mc13783: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: wm8994: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: wm8995: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: dapm: Consolidate MUXs and virtual MUXs ASoC: dapm: Break dapm_set_path_status() appart ASoC: neo1973_wm8753: Post gta01 support removal cleanup ASoC: neo1973_wm8753: Convert to table based setup ASoC: pxa: Pass correct DAPM context to {corgi,poodle,spitz}_ext_control ASoC: pxa: e740_wm9705: Convert to table based DAPM setup ASoC: pxa: e750_wm9705: Convert to table based DAPM setup ASoC: pxa: e800_wm9712: Convert to table based DAPM setup ASoC: pxa: mioa701_wm9713: Convert to table based DAPM setup ASoC: pxa: zylonite: Convert to table based DAPM setup ASoC: wm1133-ev1: Convert to table based DAPM setup ASoC: wm8753: Remove superfluous 'codec->cache_sync = 1' ASoC: wm{5102, 5110, 8997}: Replace codec->control_data with arizona->regmap ASoC: ak4104: Remove superfluous codec->control_data initialization ASoC: Handle ignore_pmdown_time for CODEC to CODEC links ASoC: Add helper functions for PCM runtime 'active' management ASoC: Fix active count tracking for CODEC to CODEC links ASoC: Add helper function to check whether a CODEC is active ASoC: Add component pointer to the DAI struct ASoC: Move active count from CODEC to component ASoC: Move ignore_pmdown_time from CODEC to component ASoC: sam9g20_wm8731: Convert to table based DAPM setup ASoC: si476x: Fix IO setup ASoC: 88pm860: Fix IO setup ASoC: Consolidate snd_soc_register_dai() and snd_soc_register_dais() ASoC: Pass CODEC to snd_soc_register_dais() ASoC: Add a per component dai list ASoC: migor: Convert to table based DAPM setup ASoC: spear: spdif_out: Fix mute control ASoC: Fix use after free Mark Brown (60): Merge branch 'fix/davinci' of git://git.kernel.org/.../broonie/sound into HEAD Merge branch 'topic/sign' of git://git.kernel.org/.../broonie/sound into asoc-tlv320aic32x4 ASoC: wm8962: Hold a runtime PM reference while handling interrupts ASoC: wm8962: Check if we runtime resume the device when starting FLL ASoC: wm8962: Clean up error handling for failed FLL start ASoC: wm8962: Reinitialise the IRQ completion rather than just trying it ASoC: ak4554: Add to SND_SOC_ALL_CODECS ASoC: pcm1681: Convert to params_width() ASoC: pcm1792a: Convert to params_width() ASoC: codecs: Make OF supported CODECs visible in Kconfig ASoC: pcm512x: Add PCM512x driver ASoC: pcm512x: More constification ASoC: codecs: Put the CODEC drivers in a menu ASoC: wm8770: Depend on SPI only ASoC: pcm512x: Implement paging support ASoC: pcm512x: Implement analogue volume control Merge branch 'topic/of' of git://git.kernel.org/.../broonie/sound into asoc-simple ASoC: tobermory: Stop the FLL if we fail to switch SYSCLK to it ASoC: cs42l51: Don't log if we fail to allocate memory ASoC: cs42l51: Convert to direct regmap API usage ASoC: ak4671: Convert to table based control init ASoC: ak4671: Convert to direct regmap API usage ASoC: pcm512x: Add regmap select Merge remote-tracking branch 'asoc/topic/adav80x' into asoc-adau1977 ALSA: Export snd_pcm_constraint_mask64() Merge branch 'fix/dapm' of git://git.kernel.org/.../broonie/sound into asoc-dapm ASoC: alc5623: Convert to direct regmap API usage Merge branch 'topic/of' of git://git.kernel.org/.../broonie/sound into asoc-simple ASoC: da732x: Remove leftover cache size setting Merge branches 'topic/dapm', 'topic/sign', 'topic/adau1373', 'topic/max98090', 'topic/mc13783', 'fix/wm8994' and 'topic/wm8995' of git://git.kernel.org/.../broonie/sound into asoc-enum ASoC: fsl: Don't select FIQ from Eukrea Merge branch 'topic/pcm' of git://git.kernel.org/.../broonie/sound into asoc-enum Merge tag 'parse-val' of git://git.kernel.org/.../broonie/regmap into asoc-core Merge branch 'topic/tlv320aic23' of git://git.kernel.org/.../broonie/sound into asoc-samsung Merge branch 'topic/tlv320aic23' of git://git.kernel.org/.../broonie/sound into asoc-fsl ASoC: pcm512x: Split out bus drivers ASoC: pcm512x: Fix duplicate const warning Merge remote-tracking branch 'asoc/fix/pcm' into asoc-linus Merge remote-tracking branches 'asoc/fix/88pm860', 'asoc/fix/omap' and 'asoc/fix/si476x' into asoc-linus Merge branch 'topic/pcm' of git://git.kernel.org/.../broonie/sound into asoc-core ASoC: dapm: Staticise dapm_mark_dirty() ASoC: dapm: Sprinkle lockdep asserts through the code ASoC: tlv320aic23: Remove spurious bits per word setting Merge tag 'asoc-v3.14-rc6' into asoc-linus Merge remote-tracking branch 'asoc/fix/spear' into asoc-linus Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/enum' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm1681' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm1792a' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm512x' into asoc-next Merge remote-tracking branches 'asoc/topic/88pm860x', 'asoc/topic/ad1836', 'asoc/topic/ad193x', 'asoc/topic/adau1977', 'asoc/topic/ak4104', 'asoc/topic/ak4641', 'asoc/topic/ak4671' and 'asoc/topic/alc5623' into asoc-next Merge remote-tracking branches 'asoc/topic/arizona', 'asoc/topic/atmel', 'asoc/topic/cirrus', 'asoc/topic/cs4271', 'asoc/topic/cs42l51', 'asoc/topic/cs42l52', 'asoc/topic/cs42l73' and 'asoc/topic/da7210' into asoc-next Merge remote-tracking branches 'asoc/topic/da7213', 'asoc/topic/da732x', 'asoc/topic/da9055', 'asoc/topic/davinci', 'asoc/topic/fsl', 'asoc/topic/fsl-esai', 'asoc/topic/fsl-sai' and 'asoc/topic/fsl-spdif' into asoc-next Merge remote-tracking branches 'asoc/topic/ml26124', 'asoc/topic/of', 'asoc/topic/omap', 'asoc/topic/pxa' and 'asoc/topic/rcar' into asoc-next Merge remote-tracking branches 'asoc/topic/samsung', 'asoc/topic/sgtl5000', 'asoc/topic/sh', 'asoc/topic/simple', 'asoc/topic/sirf', 'asoc/topic/sn95031', 'asoc/topic/ssm2602' and 'asoc/topic/stac9766' into asoc-next Merge remote-tracking branches 'asoc/topic/tlv320aic23', 'asoc/topic/tlv320aic26', 'asoc/topic/tlv320aic32x4', 'asoc/topic/tlv320aic3x', 'asoc/topic/tlv320dac33', 'asoc/topic/twl4030', 'asoc/topic/twl6040', 'asoc/topic/uda1380', 'asoc/topic/wl1273' and 'asoc/topic/wm2200' into asoc-next Merge remote-tracking branches 'asoc/topic/wm5100', 'asoc/topic/wm8523', 'asoc/topic/wm8731', 'asoc/topic/wm8737', 'asoc/topic/wm8741', 'asoc/topic/wm8753', 'asoc/topic/wm8903', 'asoc/topic/wm8904', 'asoc/topic/wm8940' and 'asoc/topic/wm8955' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8961', 'asoc/topic/wm8962', 'asoc/topic/wm8974', 'asoc/topic/wm8983', 'asoc/topic/wm8988', 'asoc/topic/wm8990', 'asoc/topic/wm8991', 'asoc/topic/wm8993' and 'asoc/topic/wm8994' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8996', 'asoc/topic/wm9081' and 'asoc/topic/wm9705' into asoc-next Markus Pargmann (9): ASoC: core: Add signed register volume control logic ASoC: core: Add SOC_DOUBLE_R_S_TLV ASoC: tlv320aic32x4: Use gpio_is_valid ASoC: tlv320aic32x4: Use signed int mixer controls ASoC: tlv320aic32x4: DT support ASoC: tlv320aic32x4: Support for master clock ASoC: tlv320aic32x4: Support for regulators ASoC: tlv320aic32x4: Rearrange clock tree shutdown ASoC: tlv320aic3x: Remove tlv320aic32x4 from compatibles of tlv320aic3x Max Filippov (3): ASoC: tlv320aic23: add support for SPI control mode ASoC: update Kconfig of AIC23 users to select I2C variant ASoC: tlv320aic23: add missing EXPORT_SYMBOLs Nenghua Cao (3): ASoC: core: fix coccinelle warnings regmap: add regmap_parse_val api ASoC: core: use regmap's parse_val to do endian translation Patrick Lai (1): ASoC: pcm: free path list before exiting from error conditions Paul Bolle (1): ASoC: smdk_wm8994: Fix typo "wm8894" Peter Ujfalusi (6): ASoC: davinci-mcasp: Return value handling cleanup for mcasp_common_hw_param() ASoC: davinci-mcasp: Move pm callbacks from platform device to soc_dai_driver ASoC: davinci-mcasp: Code cleanup in davinci_mcasp_hw_params() ASoC: davinci-mcasp: Rename platform data struct ASoC: davinci-mcasp: Remove long lines from suspend/resume callbacks ASoC: davinci-mcasp: Rename the pointer to davinci dma_params as dma_params Rongjun Ying (3): ASoC: sirf: Add SiRF internal audio codec driver ASoC: sirf: Add SiRF audio port driver is used by SiRF internal audio codec ASoC: sirf: Add SiRF audio card Sachin Kamat (6): ASoC: samsung: h1940_uda1380: Remove unused variable ASoC: samsung: h1940_uda1380: Fix build warning ASoC: samsung: rx1950_uda1380: Remove unused variable ASoC: cs42l51: Remove unused variable ASoC: samsung: Remove invalid file reference ASoC: s3c24xx: Remove invalid file reference Takashi Iwai (72): ASoC: da732x: Remove superfluous DA732X_SOC_ENUM_DOUBLE_R() ASoC: ssm2602: Omit superfluous elements in input select array ASoC: wm8990: Fix the wrong number of enum items ASoC: wm8991: Fix the wrong number of enum items ASoC: ad193x: Use SOC_ENUM_SINGLE_DECL() ASoC: Rename soc_enum.max field with items ASoC: twl4030: Clean up duplicated code ALSA: da732x: Remove superfluous DA732X_SOC_ENUM_DOUBLE_R() ASoC: Add const to SOC_ENUM_*_DECL() macros ASoC: adau1373: Remove superfluous const ASoC: lm49453: Remove superfluous const ASoC: mc13783: Remove superfluous const ASoC: rt5631: Remove superfluous const ASoC: rt5640: Remove superfluous const ASoC: ssm2518: Remove superfluous const ASoC: sta529: Remove superfluous const ASoC: wm8804: Remove superfluous const ASoC: wm8978: Remove superfluous const ASoC: wm8983: Remove superfluous const ASoC: wm8985: Remove superfluous const ASoC: wm8995: Remove superfluous const ASoC: arizona: Fix wrong number of items in enum ctls ASoC: adau1373: Use SOC_ENUM_SINGLE_DECL() ASoC: alc5623: Use SOC_ENUM_SINGLE_DECL() ASoC: cs42l52: Use SOC_ENUM_SINGLE_DECL() ASoC: da7210: Use SOC_ENUM_SINGLE_DECL() ASoC: da7213: Use SOC_ENUM_SINGLE_DECL() ASoC: da9055: Use SOC_ENUM_SINGLE_DECL() ASoC: max98090: Use SOC_ENUM_SINGLE_DECL() ASoC: mc13783: Use SOC_ENUM_SINGLE_DECL() ASoC: sgtl5000: Use SOC_ENUM_SINGLE_DECL() ASoC: sn95031: Use SOC_ENUM_SINGLE_DECL() ASoC: stac9766: Use SOC_ENUM_{SINGLE|DOUBLE}_DECL() ASoC: tlv320aic26: Use SOC_ENUM_SINGLE_DECL() ASoC: twl4030: Use SOC_ENUM_SINGLE_DECL() ASoC: twl6040: Use SOC_ENUM_SINGLE_DECL() ASoC: uda1380: Use SOC_ENUM_SINGLE_DECL() ASoC: wl1273: Use SOC_ENUM_SINGLE_EXT_DECL() ASoC: wm2200: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8523: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8731: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8737: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8903: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8904: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8940: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8961: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8962: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8974: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8993: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8995: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8996: Use SOC_ENUM_SINGLE_DECL() ASoC: wm_hubs: Use SOC_ENUM_SINGLE_DECL() ASoC: omap: Use SOC_ENUM_SINGLE_EXT_DECL() ASoC: 88pm860x: Use SOC_ENUM_SINGLE_DECL() ASoC: ak4641: Use SOC_ENUM_SINGLE_DECL() ASoC: ak4671: Use SOC_ENUM_SINGLE_DECL() ASoC: alc5632: Use SOC_ENUM_SINGLE_DECL() ASoC: wm2200: Use SOC_ENUM_SINGLE_*_DECL() ASoC: da732x: Use SOC_ENUM_SINGLE_DECL() ASoC: ml26124: Use SOC_ENUM_SINGLE_DECL() ASoC: tlv320aic23: Use SOC_ENUM_SINGLE_DECL() ASoC: wm5100: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8955: Use SOC_ENUM_SINGLE_DECL() ASoC: wm8988: Use SOC_ENUM_SINGLE_DECL() ASoC: wm9081: Use SOC_ENUM_SINGLE_DECL() ASoC: wm9705: Use SOC_ENUM_SINGLE_DECL() ASoC: ad1836: Use SOC_ENUM_SINGLE_DECL() ASoC: cs42l73: Use SOC_ENUM_SINGLE_DECL() ASoC: tlv320dac33: Use SOC_ENUM_SINGLE_*_DECL() ASoC: wm5102: Use ARRAY_SIZE() for SOC_VALUE_ENUM_SINGLE() ASoC: wm8997: Use ARRAY_SIZE() for SOC_VALUE_ENUM_SINGLE() ASoC: cs4271: Fix build error without CONFIG_SPI_MASTER Thomas Petazzoni (1): ASoC: cs42l51: add Device Tree binding to cs42l51 Xiang Xiao (2): ASoC: dapm: Power off all widgets in the snd_soc_dapm_shutdown ASoC: dapm: Reorder the bias update sequence Xiubo Li (19): ASoC: core: set_tdm_slot() will return -ENOTSUPP if no operation provided ASoC: fsl: use snd_soc_dai_init_dma_data() ASoC: simple-card: fix __asoc_simple_card_dai_init ASoC: simple-card: simplify the daifmt code ASoC: simple-card: Add snd_card's name parsing from DT node support ASoC: fsl-sai: convert to use regmap API for Freeacale SAI ASoC: add snd_soc_of_parse_audio_simple_widgets for DT ASoC: binding: add widgets.txt ASoC: simple-card: add off-codec widgets supports. ASoC: simple-card: for new properties documenting and usage ASoC: fsl: make sure the regmap-mmio is actually enabled ASoC: fsl-spdif: big-endian support ASoC: fsl-esai: big-endian support ASoC: binding: add tdm-slot.txt ASoC: core: add TDM slot parsing from DT supports ASoC: simple-card: add slot information parsing supports ASoC: fsl-sai: fix Freescale SAI DAI format setting. ASoC: fsl-sai: Add SND_SOC_DAIFMT_DSP_A/B support. ASoC: cache: Do the codec->reg_cache zero pionter check .../devicetree/bindings/i2c/trivial-devices.txt | 1 + .../devicetree/bindings/misc/atmel-ssc.txt | 8 + Documentation/devicetree/bindings/sound/da9055.txt | 22 + .../bindings/sound/davinci-evm-audio.txt | 9 +- .../devicetree/bindings/sound/eukrea-tlv320.txt | 21 + .../devicetree/bindings/sound/fsl,esai.txt | 5 + .../devicetree/bindings/sound/fsl,spdif.txt | 5 + .../devicetree/bindings/sound/pcm512x.txt | 30 + .../devicetree/bindings/sound/simple-card.txt | 17 +- .../devicetree/bindings/sound/sirf-audio-codec.txt | 17 + .../devicetree/bindings/sound/sirf-audio-port.txt | 20 + .../devicetree/bindings/sound/sirf-audio.txt | 41 + .../devicetree/bindings/sound/tdm-slot.txt | 20 + .../devicetree/bindings/sound/tlv320aic32x4.txt | 30 + .../devicetree/bindings/sound/tlv320aic3x.txt | 1 - .../devicetree/bindings/sound/widgets.txt | 20 + drivers/base/regmap/regmap.c | 12 + drivers/misc/atmel-ssc.c | 6 + include/linux/atmel-ssc.h | 1 + include/linux/mfd/arizona/registers.h | 6 +- include/linux/platform_data/adau1977.h | 45 + include/linux/platform_data/asoc-s3c.h | 3 +- include/linux/platform_data/asoc-s3c24xx_simtec.h | 3 +- include/linux/platform_data/davinci_asp.h | 4 +- include/linux/regmap.h | 9 + include/sound/rcar_snd.h | 36 +- include/sound/simple_card.h | 6 +- include/sound/soc-dai.h | 3 + include/sound/soc-dapm.h | 40 +- include/sound/soc.h | 126 +- include/trace/events/hswadsp.h | 384 +++++ include/trace/events/intel-sst.h | 148 ++ sound/core/pcm_lib.c | 1 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/atmel/Kconfig | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 13 +- sound/soc/atmel/sam9g20_wm8731.c | 20 +- sound/soc/blackfin/Kconfig | 9 +- sound/soc/cirrus/Kconfig | 4 +- sound/soc/codecs/88pm860x-codec.c | 119 +- sound/soc/codecs/Kconfig | 183 +- sound/soc/codecs/Makefile | 33 +- sound/soc/codecs/ad1836.c | 4 +- sound/soc/codecs/ad193x-i2c.c | 54 + sound/soc/codecs/ad193x-spi.c | 48 + sound/soc/codecs/ad193x.c | 144 +- sound/soc/codecs/ad193x.h | 7 + sound/soc/codecs/adau1373.c | 32 +- sound/soc/codecs/adau1977-i2c.c | 59 + sound/soc/codecs/adau1977-spi.c | 76 + sound/soc/codecs/adau1977.c | 1018 +++++++++++ sound/soc/codecs/adau1977.h | 37 + sound/soc/codecs/adav801.c | 53 + sound/soc/codecs/adav803.c | 50 + sound/soc/codecs/adav80x.c | 152 +- sound/soc/codecs/adav80x.h | 7 + sound/soc/codecs/ak4104.c | 2 - sound/soc/codecs/ak4641.c | 16 +- sound/soc/codecs/ak4671.c | 240 ++- sound/soc/codecs/ak4671.h | 2 - sound/soc/codecs/alc5623.c | 117 +- sound/soc/codecs/alc5632.c | 40 +- sound/soc/codecs/arizona.c | 325 ++-- sound/soc/codecs/cs4271.c | 63 +- sound/soc/codecs/cs42l51.c | 86 +- sound/soc/codecs/cs42l52.c | 92 +- sound/soc/codecs/cs42l73.c | 55 +- sound/soc/codecs/da7210.c | 20 +- sound/soc/codecs/da7213.c | 151 +- sound/soc/codecs/da732x.c | 167 +- sound/soc/codecs/da732x.h | 3 - sound/soc/codecs/da9055.c | 92 +- sound/soc/codecs/lm49453.c | 16 +- sound/soc/codecs/max98088.c | 2 +- sound/soc/codecs/max98090.c | 172 +- sound/soc/codecs/max98095.c | 4 +- sound/soc/codecs/mc13783.c | 20 +- sound/soc/codecs/ml26124.c | 12 +- sound/soc/codecs/pcm1681.c | 15 +- sound/soc/codecs/pcm1792a.c | 33 +- sound/soc/codecs/pcm512x-i2c.c | 71 + sound/soc/codecs/pcm512x-spi.c | 69 + sound/soc/codecs/pcm512x.c | 589 +++++++ sound/soc/codecs/pcm512x.h | 171 ++ sound/soc/codecs/rt5631.c | 75 +- sound/soc/codecs/rt5640.c | 75 +- sound/soc/codecs/sgtl5000.c | 10 +- sound/soc/codecs/si476x.c | 2 +- sound/soc/codecs/sirf-audio-codec.c | 533 ++++++ sound/soc/codecs/sirf-audio-codec.h | 75 + sound/soc/codecs/sn95031.c | 44 +- sound/soc/codecs/ssm2518.c | 14 +- sound/soc/codecs/ssm2602-i2c.c | 57 + sound/soc/codecs/ssm2602-spi.c | 41 + sound/soc/codecs/ssm2602.c | 173 +- sound/soc/codecs/ssm2602.h | 14 + sound/soc/codecs/sta529.c | 2 +- sound/soc/codecs/stac9766.c | 38 +- sound/soc/codecs/tlv320aic23-i2c.c | 59 + sound/soc/codecs/tlv320aic23-spi.c | 56 + sound/soc/codecs/tlv320aic23.c | 71 +- sound/soc/codecs/tlv320aic23.h | 6 + sound/soc/codecs/tlv320aic26.c | 5 +- sound/soc/codecs/tlv320aic32x4.c | 232 ++- sound/soc/codecs/tlv320dac33.c | 34 +- sound/soc/codecs/twl4030.c | 101 +- sound/soc/codecs/twl6040.c | 17 +- sound/soc/codecs/uda1380.c | 40 +- sound/soc/codecs/wl1273.c | 9 +- sound/soc/codecs/wm2200.c | 18 +- sound/soc/codecs/wm5100.c | 36 +- sound/soc/codecs/wm5102.c | 28 +- sound/soc/codecs/wm5110.c | 19 +- sound/soc/codecs/wm8523.c | 4 +- sound/soc/codecs/wm8711.c | 2 +- sound/soc/codecs/wm8731.c | 4 +- sound/soc/codecs/wm8737.c | 50 +- sound/soc/codecs/wm8741.c | 34 +- sound/soc/codecs/wm8753.c | 5 +- sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8903.c | 108 +- sound/soc/codecs/wm8904.c | 77 +- sound/soc/codecs/wm8940.c | 20 +- sound/soc/codecs/wm8955.c | 11 +- sound/soc/codecs/wm8958-dsp2.c | 8 +- sound/soc/codecs/wm8961.c | 16 +- sound/soc/codecs/wm8962.c | 80 +- sound/soc/codecs/wm8974.c | 4 +- sound/soc/codecs/wm8978.c | 30 +- sound/soc/codecs/wm8983.c | 45 +- sound/soc/codecs/wm8985.c | 39 +- sound/soc/codecs/wm8988.c | 62 +- sound/soc/codecs/wm8990.c | 41 +- sound/soc/codecs/wm8991.c | 44 +- sound/soc/codecs/wm8993.c | 66 +- sound/soc/codecs/wm8994.c | 52 +- sound/soc/codecs/wm8995.c | 43 +- sound/soc/codecs/wm8996.c | 75 +- sound/soc/codecs/wm8997.c | 25 +- sound/soc/codecs/wm9081.c | 23 +- sound/soc/codecs/wm9705.c | 12 +- sound/soc/codecs/wm_adsp.c | 50 +- sound/soc/codecs/wm_hubs.c | 16 +- sound/soc/davinci/davinci-evm.c | 58 +- sound/soc/davinci/davinci-mcasp.c | 196 ++- sound/soc/fsl/Kconfig | 12 +- sound/soc/fsl/eukrea-tlv320.c | 108 +- sound/soc/fsl/fsl_esai.c | 32 +- sound/soc/fsl/fsl_sai.c | 332 ++-- sound/soc/fsl/fsl_sai.h | 48 +- sound/soc/fsl/fsl_spdif.c | 9 +- sound/soc/fsl/imx-pcm-fiq.c | 7 +- sound/soc/fsl/wm1133-ev1.c | 11 +- sound/soc/generic/simple-card.c | 245 +-- sound/soc/intel/Kconfig | 42 +- sound/soc/intel/Makefile | 27 +- sound/soc/intel/byt-rt5640.c | 187 ++ sound/soc/intel/haswell.c | 235 +++ sound/soc/intel/mfld_machine.c | 65 +- sound/soc/intel/sst-acpi.c | 284 ++++ sound/soc/intel/sst-baytrail-dsp.c | 372 ++++ sound/soc/intel/sst-baytrail-ipc.c | 867 ++++++++++ sound/soc/intel/sst-baytrail-ipc.h | 69 + sound/soc/intel/sst-baytrail-pcm.c | 422 +++++ sound/soc/intel/sst-dsp-priv.h | 309 ++++ sound/soc/intel/sst-dsp.c | 385 +++++ sound/soc/intel/sst-dsp.h | 233 +++ sound/soc/intel/sst-firmware.c | 587 +++++++ sound/soc/intel/sst-haswell-dsp.c | 517 ++++++ sound/soc/intel/sst-haswell-ipc.c | 1785 ++++++++++++++++++++ sound/soc/intel/sst-haswell-ipc.h | 488 ++++++ sound/soc/intel/sst-haswell-pcm.c | 872 ++++++++++ sound/soc/intel/{sst_dsp.h => sst-mfld-dsp.h} | 8 +- .../intel/{sst_platform.c => sst-mfld-platform.c} | 8 +- .../intel/{sst_platform.h => sst-mfld-platform.h} | 4 +- sound/soc/omap/Kconfig | 4 +- sound/soc/omap/ams-delta.c | 55 +- sound/soc/omap/n810.c | 26 +- sound/soc/omap/rx51.c | 22 +- sound/soc/pxa/corgi.c | 49 +- sound/soc/pxa/e740_wm9705.c | 10 +- sound/soc/pxa/e750_wm9705.c | 10 +- sound/soc/pxa/e800_wm9712.c | 19 +- sound/soc/pxa/magician.c | 26 +- sound/soc/pxa/mioa701_wm9713.c | 19 +- sound/soc/pxa/poodle.c | 7 +- sound/soc/pxa/spitz.c | 58 +- sound/soc/pxa/tosa.c | 32 +- sound/soc/pxa/zylonite.c | 17 +- sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/h1940_uda1380.c | 7 +- sound/soc/samsung/neo1973_wm8753.c | 168 +- sound/soc/samsung/rx1950_uda1380.c | 5 - sound/soc/samsung/smdk_wm8994.c | 2 +- sound/soc/samsung/tobermory.c | 2 + sound/soc/sh/migor.c | 19 +- sound/soc/sh/rcar/Makefile | 2 +- sound/soc/sh/rcar/adg.c | 228 ++- sound/soc/sh/rcar/core.c | 306 ++-- sound/soc/sh/rcar/gen.c | 98 +- sound/soc/sh/rcar/rsnd.h | 195 ++- sound/soc/sh/rcar/scu.c | 384 ----- sound/soc/sh/rcar/src.c | 687 ++++++++ sound/soc/sh/rcar/ssi.c | 332 ++-- sound/soc/sirf/Kconfig | 14 + sound/soc/sirf/Makefile | 5 + sound/soc/sirf/sirf-audio-port.c | 194 +++ sound/soc/sirf/sirf-audio-port.h | 62 + sound/soc/sirf/sirf-audio.c | 156 ++ sound/soc/soc-cache.c | 13 +- sound/soc/soc-compress.c | 65 +- sound/soc/soc-core.c | 582 ++++--- sound/soc/soc-dapm.c | 465 ++--- sound/soc/soc-pcm.c | 112 +- sound/soc/spear/spdif_out.c | 10 +- sound/soc/tegra/Kconfig | 2 +- 217 files changed, 17528 insertions(+), 4570 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/da9055.txt create mode 100644 Documentation/devicetree/bindings/sound/eukrea-tlv320.txt create mode 100644 Documentation/devicetree/bindings/sound/pcm512x.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-port.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/tdm-slot.txt create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic32x4.txt create mode 100644 Documentation/devicetree/bindings/sound/widgets.txt create mode 100644 include/linux/platform_data/adau1977.h create mode 100644 include/trace/events/hswadsp.h create mode 100644 include/trace/events/intel-sst.h create mode 100644 sound/soc/codecs/ad193x-i2c.c create mode 100644 sound/soc/codecs/ad193x-spi.c create mode 100644 sound/soc/codecs/adau1977-i2c.c create mode 100644 sound/soc/codecs/adau1977-spi.c create mode 100644 sound/soc/codecs/adau1977.c create mode 100644 sound/soc/codecs/adau1977.h create mode 100644 sound/soc/codecs/adav801.c create mode 100644 sound/soc/codecs/adav803.c create mode 100644 sound/soc/codecs/pcm512x-i2c.c create mode 100644 sound/soc/codecs/pcm512x-spi.c create mode 100644 sound/soc/codecs/pcm512x.c create mode 100644 sound/soc/codecs/pcm512x.h create mode 100644 sound/soc/codecs/sirf-audio-codec.c create mode 100644 sound/soc/codecs/sirf-audio-codec.h create mode 100644 sound/soc/codecs/ssm2602-i2c.c create mode 100644 sound/soc/codecs/ssm2602-spi.c create mode 100644 sound/soc/codecs/tlv320aic23-i2c.c create mode 100644 sound/soc/codecs/tlv320aic23-spi.c create mode 100644 sound/soc/intel/byt-rt5640.c create mode 100644 sound/soc/intel/haswell.c create mode 100644 sound/soc/intel/sst-acpi.c create mode 100644 sound/soc/intel/sst-baytrail-dsp.c create mode 100644 sound/soc/intel/sst-baytrail-ipc.c create mode 100644 sound/soc/intel/sst-baytrail-ipc.h create mode 100644 sound/soc/intel/sst-baytrail-pcm.c create mode 100644 sound/soc/intel/sst-dsp-priv.h create mode 100644 sound/soc/intel/sst-dsp.c create mode 100644 sound/soc/intel/sst-dsp.h create mode 100644 sound/soc/intel/sst-firmware.c create mode 100644 sound/soc/intel/sst-haswell-dsp.c create mode 100644 sound/soc/intel/sst-haswell-ipc.c create mode 100644 sound/soc/intel/sst-haswell-ipc.h create mode 100644 sound/soc/intel/sst-haswell-pcm.c rename sound/soc/intel/{sst_dsp.h => sst-mfld-dsp.h} (96%) rename sound/soc/intel/{sst_platform.c => sst-mfld-platform.c} (99%) rename sound/soc/intel/{sst_platform.h => sst-mfld-platform.h} (97%) delete mode 100644 sound/soc/sh/rcar/scu.c create mode 100644 sound/soc/sh/rcar/src.c create mode 100644 sound/soc/sirf/Kconfig create mode 100644 sound/soc/sirf/Makefile create mode 100644 sound/soc/sirf/sirf-audio-port.c create mode 100644 sound/soc/sirf/sirf-audio-port.h create mode 100644 sound/soc/sirf/sirf-audio.c

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[alsa-devel] [PATCH] ASoC: Intel: don't select RT5640 if !I2C
by Stephen Warren 13 Mar '14

13 Mar '14

From: Stephen Warren <swarren(a)nvidia.com> The rt5640 driver won't compile without I2C enabled. Hence, the Intel Haswell and Baytrail+RT5640 ASoC drivers must also depend on I2C, since these select RT5640. This solves: sound/soc/codecs/rt5640.c:2220:1: warning: data definition has no type or storage class [enabled by default] sound/soc/codecs/rt5640.c:2220:1: error: type defaults to ‘int’ in declaration of ‘module_i2c_driver’ [-Werror=implicit-int] sound/soc/codecs/rt5640.c:2220:1: warning: parameter names (without types) in function declaration [enabled by default] sound/soc/codecs/rt5640.c:2210:26: warning: ‘rt5640_i2c_driver’ defined but not used [-Wunused-variable] Reported-by: Jim Davis <jim.epost(a)gmail.com> Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/intel/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 4577b69fcf2c..3c81b3891209 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -32,7 +32,7 @@ config SND_SOC_INTEL_BAYTRAIL config SND_SOC_INTEL_HASWELL_MACH tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint" - depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C select SND_SOC_INTEL_HASWELL select SND_SOC_RT5640 help @@ -43,7 +43,7 @@ config SND_SOC_INTEL_HASWELL_MACH config SND_SOC_INTEL_BYT_RT5640_MACH tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec" - depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C select SND_SOC_INTEL_BAYTRAIL select SND_SOC_RT5640 help -- 1.8.1.5

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[alsa-devel] [PATCH] ASoC: tlv320aic23: Remove spurious bits per word setting
by Mark Brown 12 Mar '14

12 Mar '14

From: Mark Brown <broonie(a)linaro.org> regmap should handle any byte ordering issues required, it is looking for a byte stream from the bus, so don't set 16 bits per word. This is likely to have tested out OK due to use of an unmerged SPI controller driver. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/tlv320aic23-spi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic23-spi.c b/sound/soc/codecs/tlv320aic23-spi.c index 585aea436c6a..3b387e41d75d 100644 --- a/sound/soc/codecs/tlv320aic23-spi.c +++ b/sound/soc/codecs/tlv320aic23-spi.c @@ -25,7 +25,6 @@ static int aic23_spi_probe(struct spi_device *spi) dev_dbg(&spi->dev, "probing tlv320aic23 spi device\n"); - spi->bits_per_word = 16; spi->mode = SPI_MODE_0; ret = spi_setup(spi); if (ret < 0) -- 1.9.0

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