Signed-off-by:Brian Austin brian.austin@cirrus.com Signed-off-by:Georgi Vlaev joe@nucleusys.com

This patch adds support for Cirrus Logic CS42L73 low power stereo codec --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs42l73.c | 1255 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs42l73.h | 227 ++++++++ 4 files changed, 1488 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/cs42l73.c create mode 100644 sound/soc/codecs/cs42l73.h

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4584514..4f0ce61 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -28,6 +28,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ALC5623 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C + select SND_SOC_CS42L73 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI select SND_SOC_CX20442 @@ -175,6 +176,9 @@ config SND_SOC_CQ0093VC config SND_SOC_CS42L51 tristate

+config SND_SOC_CS42L73 + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a7c415d..a76475d 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -15,6 +15,7 @@ snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o +snd-soc-cs42l73-objs := cs42l73.o snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o snd-soc-cx20442-objs := cx20442.o @@ -115,6 +116,7 @@ obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o +obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c new file mode 100644 index 0000000..e191f44 --- /dev/null +++ b/sound/soc/codecs/cs42l73.c @@ -0,0 +1,1255 @@ +/* + * cs42l73.c -- CS42L73 ALSA Soc Audio driver + * + * Copyright 2011 Cirrus Logic, Inc. + * + * Authors: Georgi Vlaev, Nucleus Systems Ltd, joe@nucleusys.com + * Brian Austin, Cirrus Logic Inc, brian.austin@cirrus.com + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include "cs42l73.h" + +struct sp_config { + u8 spc, mmcc, spfs; + u32 srate; +}; +struct cs42l73_private { + u32 sysclk; + u8 mclksel; + u32 mclk; + struct sp_config config[3]; +}; + +static const u8 cs42l73_reg[] = { +/*0*/ 0x00, 0x42, 0xA7, 0x30, +/*4*/ 0x00, 0x00, 0xF1, 0xDF, +/*8*/ 0x3F, 0x57, 0x53, 0x00, +/*C*/ 0x00, 0x15, 0x00, 0x15, +/*A*/ 0x00, 0x15, 0x00, 0x06, +/*E*/ 0x00, 0x00, 0x00, 0x00, +/*18*/ 0x00, 0x00, 0x00, 0x00, +/*1C*/ 0x00, 0x00, 0x00, 0x00, +/*20*/ 0x00, 0x00, 0x00, 0x00, +/*24*/ 0x00, 0x00, 0x00, 0x7F, +/*28*/ 0x00, 0x00, 0x3F, 0x00, +/*2C*/ 0x00, 0x3F, 0x00, 0x00, +/*30*/ 0x3F, 0x00, 0x00, 0x00, +/*34*/ 0x18, 0x3F, 0x3F, 0x3F, +/*38*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*3C*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*40*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*44*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*48*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*4C*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*50*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*54*/ 0x3F, 0xAA, 0x3F, 0x3F, +/*58*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*5C*/ 0x3F, 0x3F, 0x00, 0x00, +}; + +static const unsigned int hpaloa_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 13, TLV_DB_SCALE_ITEM(-7600, 200, 0), + 14, 75, TLV_DB_SCALE_ITEM(-4900, 100, 0), +}; + +static DECLARE_TLV_DB_SCALE(adc_boost_tlv, 0, 2500, 0); + +static DECLARE_TLV_DB_SCALE(hl_tlv, -10200, 50, 0); + +static DECLARE_TLV_DB_SCALE(ipd_tlv, -9600, 100, 0); + +static DECLARE_TLV_DB_SCALE(micpga_tlv, -600, 50, 0); + +static const unsigned int limiter_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0), + 3, 7, TLV_DB_SCALE_ITEM(-1200, 300, 0), +}; + +static const DECLARE_TLV_DB_SCALE(attn_tlv, -6300, 100, 1); + +static const char * const cs42l73_pgaa_text[] = { "Line A", "Mic 1" }; +static const char * const cs42l73_pgab_text[] = { "Line B", "Mic 2" }; + +static const struct soc_enum pgaa_enum = + SOC_ENUM_SINGLE(CS42L73_ADCIPC, 3, + ARRAY_SIZE(cs42l73_pgaa_text), cs42l73_pgaa_text); + +static const struct soc_enum pgab_enum = + SOC_ENUM_SINGLE(CS42L73_ADCIPC, 7, + ARRAY_SIZE(cs42l73_pgab_text), cs42l73_pgab_text); + +static const struct snd_kcontrol_new pgaa_mux = + SOC_DAPM_ENUM("Left Analog Input Capture Mux", pgaa_enum); + +static const struct snd_kcontrol_new pgab_mux = + SOC_DAPM_ENUM("Right Analog Input Capture Mux", pgab_enum); + +static const struct snd_kcontrol_new input_left_mixer[] = { + SOC_DAPM_SINGLE("ADC Left Input", CS42L73_PWRCTL1, + 5, 1, 1), + SOC_DAPM_SINGLE("DMIC Left Input", CS42L73_PWRCTL1, + 4, 1, 1), +}; + +static const struct snd_kcontrol_new input_right_mixer[] = { + SOC_DAPM_SINGLE("ADC Right Input", CS42L73_PWRCTL1, + 7, 1, 1), + SOC_DAPM_SINGLE("DMIC Right Input", CS42L73_PWRCTL1, + 6, 1, 1), +}; + +static const char * const cs42l73_ng_delay_text[] = { + "50ms", "100ms", "150ms", "200ms" }; + +static const struct soc_enum ng_delay_enum = + SOC_ENUM_SINGLE(CS42L73_NGCAB, 0, + ARRAY_SIZE(cs42l73_ng_delay_text), cs42l73_ng_delay_text); + +static const char * const charge_pump_freq_text[] = { + "0", "1", "2", "3", "4", + "5", "6", "7", "8", "9", + "10", "11", "12", "13", "14", "15" }; + +static const struct soc_enum charge_pump_enum = + SOC_ENUM_SINGLE(CS42L73_CPFCHC, 4, + ARRAY_SIZE(charge_pump_freq_text), charge_pump_freq_text); + +static const char * const cs42l73_mono_mix_texts[] = { + "Left", "Right", "Mono Mix"}; + +static const unsigned int cs42l73_mono_mix_values[] = { 0, 1, 2 }; + +static const struct soc_enum spk_asp_enum = + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 6, 1, + ARRAY_SIZE(cs42l73_mono_mix_texts), + cs42l73_mono_mix_texts, + cs42l73_mono_mix_values); + +static const struct snd_kcontrol_new spk_asp_mixer = + SOC_DAPM_ENUM("Route", spk_asp_enum); + +static const struct soc_enum spk_xsp_enum = + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 4, 3, + ARRAY_SIZE(cs42l73_mono_mix_texts), + cs42l73_mono_mix_texts, + cs42l73_mono_mix_values); + +static const struct snd_kcontrol_new spk_xsp_mixer = + SOC_DAPM_ENUM("Route", spk_xsp_enum); + +static const struct soc_enum esl_asp_enum = + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 2, 5, + ARRAY_SIZE(cs42l73_mono_mix_texts), + cs42l73_mono_mix_texts, + cs42l73_mono_mix_values); + +static const struct snd_kcontrol_new esl_asp_mixer = + SOC_DAPM_ENUM("Route", esl_asp_enum); + +static const struct soc_enum esl_xsp_enum = + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 0, 7, + ARRAY_SIZE(cs42l73_mono_mix_texts), + cs42l73_mono_mix_texts, + cs42l73_mono_mix_values); + +static const struct snd_kcontrol_new esl_xsp_mixer = + SOC_DAPM_ENUM("Route", esl_xsp_enum); + +static const char * const cs42l73_ip_swap_text[] = { + "Stereo", "Mono A", "Mono B", "Swap A-B"}; + +static const struct soc_enum ip_swap_enum = + SOC_ENUM_SINGLE(CS42L73_MIOPC, 6, + ARRAY_SIZE(cs42l73_ip_swap_text), cs42l73_ip_swap_text); + +static const char * const cs42l73_spo_mixer_text[] = {"Mono", "Stereo"}; + +static const struct soc_enum vsp_output_mux_enum = + SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 5, + ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text); + +static const struct soc_enum xsp_output_mux_enum = + SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 4, + ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text); + +static const struct snd_kcontrol_new vsp_output_mux = + SOC_DAPM_ENUM("VSPOUT Mux", vsp_output_mux_enum); + +static const struct snd_kcontrol_new xsp_output_mux = + SOC_DAPM_ENUM("XSPOUT Mux", xsp_output_mux_enum); + +static const struct snd_kcontrol_new hp_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 0, 1, 1); + +static const struct snd_kcontrol_new lo_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 1, 1, 1); + +static const struct snd_kcontrol_new spk_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 2, 1, 1); + +static const struct snd_kcontrol_new spklo_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 4, 1, 1); + +static const struct snd_kcontrol_new ear_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 3, 1, 1); + +static const struct snd_kcontrol_new cs42l73_snd_controls[] = { + SOC_DOUBLE_R_SX_TLV("Headphone Analog Playback Volume", + CS42L73_HPAAVOL, CS42L73_HPBAVOL, 7, + 0xffffffC1, 0x0C, hpaloa_tlv), + + SOC_DOUBLE_R_SX_TLV("LineOut Analog Playback Volume", CS42L73_LOAAVOL, + CS42L73_LOBAVOL, 7, 0xffffffC1, 0x0C, hpaloa_tlv), + + SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL, + CS42L73_MICBPREPGABVOL, 5, 0xffffff35, + 0x34, micpga_tlv), + + SOC_DOUBLE_R("MIC Preamp Switch", CS42L73_MICAPREPGAAVOL, + CS42L73_MICBPREPGABVOL, 6, 1, 1), + + SOC_DOUBLE_R_SX_TLV("Input Path Digital Volume", CS42L73_IPADVOL, + CS42L73_IPBDVOL, 7, 0xffffffA0, 0xA0, ipd_tlv), + + SOC_DOUBLE_R_SX_TLV("HL Digital Playback Volume", + CS42L73_HLADVOL, CS42L73_HLBDVOL, 7, 0xffffffE5, + 0xE4, hl_tlv), + + SOC_SINGLE_TLV("ADC A Boost Volume", + CS42L73_ADCIPC, 2, 0x01, 1, adc_boost_tlv), + + SOC_SINGLE_TLV("ADC B Boost Volume", + CS42L73_ADCIPC, 6, 0x01, 1, adc_boost_tlv), + + SOC_SINGLE_TLV("Speakerphone Digital Playback Volume", + CS42L73_SPKDVOL, 0, 0xE4, 1, hl_tlv), + + SOC_SINGLE_TLV("Ear Speaker Digital Playback Volume", + CS42L73_ESLDVOL, 0, 0xE4, 1, hl_tlv), + + SOC_DOUBLE_R("Headphone Analog Playback Switch", CS42L73_HPAAVOL, + CS42L73_HPBAVOL, 7, 1, 1), + + SOC_DOUBLE_R("LineOut Analog Playback Switch", CS42L73_LOAAVOL, + CS42L73_LOBAVOL, 7, 1, 1), + SOC_DOUBLE("Input Path Digital Switch", CS42L73_ADCIPC, 0, 4, 1, 1), + SOC_DOUBLE("HL Digital Playback Switch", CS42L73_PBDC, 0, + 1, 1, 1), + SOC_SINGLE("Speakerphone Digital Playback Switch", CS42L73_PBDC, 2, 1, + 1), + SOC_SINGLE("Ear Speaker Digital Playback Switch", CS42L73_PBDC, 3, 1, + 1), + + SOC_SINGLE("PGA Soft-Ramp Switch", CS42L73_MIOPC, 3, 1, 0), + SOC_SINGLE("Analog Zero Cross Switch", CS42L73_MIOPC, 2, 1, 0), + SOC_SINGLE("Digital Soft-Ramp Switch", CS42L73_MIOPC, 1, 1, 0), + SOC_SINGLE("Analog Output Soft-Ramp Switch", CS42L73_MIOPC, 0, 1, 0), + + SOC_DOUBLE("ADC Signal Polarity Switch", CS42L73_ADCIPC, 1, 5, 1, + 0), + + SOC_SINGLE("HL Limiter Attack Rate", CS42L73_LIMARATEHL, 0, 0x3F, + 0), + SOC_SINGLE("HL Limiter Release Rate", CS42L73_LIMRRATEHL, 0, + 0x3F, 0), + + + SOC_SINGLE("HL Limiter Switch", CS42L73_LIMRRATEHL, 7, 1, 0), + SOC_SINGLE("HL Limiter All Channels Switch", CS42L73_LIMRRATEHL, 6, 1, + 0), + + SOC_SINGLE_TLV("HL Limiter Max Threshold Volume", CS42L73_LMAXHL, 5, 7, + 1, limiter_tlv), + + SOC_SINGLE_TLV("HL Limiter Cushion Volume", CS42L73_LMAXHL, 2, 7, 1, + limiter_tlv), + + SOC_SINGLE("SPK Limiter Attack Rate Volume", CS42L73_LIMARATESPK, 0, + 0x3F, 0), + SOC_SINGLE("SPK Limiter Release Rate Volume", CS42L73_LIMRRATESPK, 0, + 0x3F, 0), + SOC_SINGLE("SPK Limiter Switch", CS42L73_LIMRRATESPK, 7, 1, 0), + SOC_SINGLE("SPK Limiter All Channels Switch", CS42L73_LIMRRATESPK, + 6, 1, 0), + SOC_SINGLE_TLV("SPK Limiter Max Threshold Volume", CS42L73_LMAXSPK, 5, + 7, 1, limiter_tlv), + + SOC_SINGLE_TLV("SPK Limiter Cushion Volume", CS42L73_LMAXSPK, 2, 7, 1, + limiter_tlv), + + SOC_SINGLE("ESL Limiter Attack Rate Volume", CS42L73_LIMARATEESL, 0, + 0x3F, 0), + SOC_SINGLE("ESL Limiter Release Rate Volume", CS42L73_LIMRRATEESL, 0, + 0x3F, 0), + SOC_SINGLE("ESL Limiter Switch", CS42L73_LIMRRATEESL, 7, 1, 0), + SOC_SINGLE_TLV("ESL Limiter Max Threshold Volume", CS42L73_LMAXESL, 5, + 7, 1, limiter_tlv), + + SOC_SINGLE_TLV("ESL Limiter Cushion Volume", CS42L73_LMAXESL, 2, 7, 1, + limiter_tlv), + + SOC_SINGLE("ALC Attack Rate Volume", CS42L73_ALCARATE, 0, 0x3F, 0), + SOC_SINGLE("ALC Release Rate Volume", CS42L73_ALCRRATE, 0, 0x3F, 0), + SOC_DOUBLE("ALC Switch", CS42L73_ALCARATE, 6, 7, 1, 0), + SOC_SINGLE_TLV("ALC Max Threshold Volume", CS42L73_ALCMINMAX, 5, 7, 0, + limiter_tlv), + SOC_SINGLE_TLV("ALC Min Threshold Volume", CS42L73_ALCMINMAX, 2, 7, 0, + limiter_tlv), + + SOC_DOUBLE("NG Enable Switch", CS42L73_NGCAB, 6, 7, 1, 0), + SOC_SINGLE("NG Boost Switch", CS42L73_NGCAB, 5, 1, 0), + /* + NG Threshold depends on NG_BOOTSAB, which selects + between two threshold scales in decibels. + Set linear values for now .. + */ + SOC_SINGLE("NG Threshold", CS42L73_NGCAB, 2, 7, 0), + SOC_ENUM("NG Delay", ng_delay_enum), + + SOC_DOUBLE_R_TLV("XSP-IP Volume", + CS42L73_XSPAIPAA, CS42L73_XSPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-XSP Volume", + CS42L73_XSPAXSPAA, CS42L73_XSPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-ASP Volume", + CS42L73_XSPAASPAA, CS42L73_XSPAASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-VSP Volume", + CS42L73_XSPAVSPMA, CS42L73_XSPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("ASP-IP Volume", + CS42L73_ASPAIPAA, CS42L73_ASPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-XSP Volume", + CS42L73_ASPAXSPAA, CS42L73_ASPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-ASP Volume", + CS42L73_ASPAASPAA, CS42L73_ASPBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-VSP Volume", + CS42L73_ASPAVSPMA, CS42L73_ASPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("VSP-IP Volume", + CS42L73_VSPAIPAA, CS42L73_VSPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-XSP Volume", + CS42L73_VSPAXSPAA, CS42L73_VSPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-ASP Volume", + CS42L73_VSPAASPAA, CS42L73_VSPBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-VSP Volume", + CS42L73_VSPAVSPMA, CS42L73_VSPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("HL-IP Volume", + CS42L73_HLAIPAA, CS42L73_HLBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-XSP Volume", + CS42L73_HLAXSPAA, CS42L73_HLBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-ASP Volume", + CS42L73_HLAASPAA, CS42L73_HLBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-VSP Volume", + CS42L73_HLAVSPMA, CS42L73_HLBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_SINGLE_TLV("SPK-IP Mono Volume", + CS42L73_SPKMIPMA, 0, 0x3E, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-XSP Mono Volume", + CS42L73_SPKMXSPA, 0, 0x3E, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-ASP Mono Volume", + CS42L73_SPKMASPA, 0, 0x3E, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-VSP Mono Volume", + CS42L73_SPKMVSPMA, 0, 0x3E, 1, attn_tlv), + + SOC_SINGLE_TLV("ESL-IP Mono Volume", + CS42L73_ESLMIPMA, 0, 0x3E, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-XSP Mono Volume", + CS42L73_ESLMXSPA, 0, 0x3E, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-ASP Mono Volume", + CS42L73_ESLMASPA, 0, 0x3E, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-VSP Mono Volume", + CS42L73_ESLMVSPMA, 0, 0x3E, 1, attn_tlv), + + SOC_ENUM("IP Digital Swap/Mono Select", ip_swap_enum), + +}; + +static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("LINEINA"), + SND_SOC_DAPM_INPUT("LINEINB"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_SUPPLY("MIC1 Bias", CS42L73_PWRCTL2, 6, 1, NULL, 0), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_SUPPLY("MIC2 Bias", CS42L73_PWRCTL2, 7, 1, NULL, 0), + + SND_SOC_DAPM_AIF_OUT("XSPOUTL", "XSP Capture", 0, + CS42L73_PWRCTL2, 1, 1), + SND_SOC_DAPM_AIF_OUT("XSPOUTR", "XSP Capture", 0, + CS42L73_PWRCTL2, 1, 1), + SND_SOC_DAPM_AIF_OUT("ASPOUTL", "ASP Capture", 0, + CS42L73_PWRCTL2, 3, 1), + SND_SOC_DAPM_AIF_OUT("ASPOUTR", "ASP Capture", 0, + CS42L73_PWRCTL2, 3, 1), + SND_SOC_DAPM_AIF_OUT("VSPOUTL", "VSP Capture", 0, + CS42L73_PWRCTL2, 4, 1), + SND_SOC_DAPM_AIF_OUT("VSPOUTR", "VSP Capture", 0, + CS42L73_PWRCTL2, 4, 1), + + SND_SOC_DAPM_PGA("PGA Left", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PGA Right", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MUX("PGA Left Mux", SND_SOC_NOPM, 0, 0, &pgaa_mux), + SND_SOC_DAPM_MUX("PGA Right Mux", SND_SOC_NOPM, 0, 0, &pgab_mux), + + SND_SOC_DAPM_ADC("ADC Left", NULL, CS42L73_PWRCTL1, 7, 1), + SND_SOC_DAPM_ADC("ADC Right", NULL, CS42L73_PWRCTL1, 5, 1), + SND_SOC_DAPM_ADC("DMIC Left", NULL, CS42L73_PWRCTL1, 6, 1), + SND_SOC_DAPM_ADC("DMIC Right", NULL, CS42L73_PWRCTL1, 4, 1), + + SND_SOC_DAPM_MIXER_NAMED_CTL("Input Left Capture", SND_SOC_NOPM, + 0, 0, input_left_mixer, + ARRAY_SIZE(input_left_mixer)), + + SND_SOC_DAPM_MIXER_NAMED_CTL("Input Right Capture", SND_SOC_NOPM, + 0, 0, input_right_mixer, + ARRAY_SIZE(input_right_mixer)), + + SND_SOC_DAPM_MIXER("ASPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("ASPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("XSPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("XSPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("VSPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("VSPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MUX("VSPOUT Mux", SND_SOC_NOPM, + 0, 0, &vsp_output_mux), + + SND_SOC_DAPM_MUX("XSPOUT Mux", SND_SOC_NOPM, + 0, 0, &xsp_output_mux), + + SND_SOC_DAPM_AIF_IN("XSPINL", "XSP Playback", 0, + CS42L73_PWRCTL2, 0, 1), + SND_SOC_DAPM_AIF_IN("XSPINR", "XSP Playback", 0, + CS42L73_PWRCTL2, 0, 1), + SND_SOC_DAPM_AIF_IN("XSPINM", "XSP Playback", 0, + CS42L73_PWRCTL2, 0, 1), + + SND_SOC_DAPM_AIF_IN("ASPINL", "ASP Playback", 0, + CS42L73_PWRCTL2, 2, 1), + SND_SOC_DAPM_AIF_IN("ASPINR", "ASP Playback", 0, + CS42L73_PWRCTL2, 2, 1), + SND_SOC_DAPM_AIF_IN("ASPINM", "ASP Playback", 0, + CS42L73_PWRCTL2, 2, 1), + + SND_SOC_DAPM_AIF_IN("VSPIN", "VSP Playback", 0, + CS42L73_PWRCTL2, 4, 1), + + SND_SOC_DAPM_MIXER("HL Left Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("HL Right Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SPK Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("ESL Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MUX("ESL-XSP Mux", SND_SOC_NOPM, + 0, 0, &esl_xsp_mixer), + + SND_SOC_DAPM_MUX("ESL-ASP Mux", SND_SOC_NOPM, + 0, 0, &esl_asp_mixer), + + SND_SOC_DAPM_MUX("SPK-ASP Mux", SND_SOC_NOPM, + 0, 0, &spk_asp_mixer), + + SND_SOC_DAPM_MUX("SPK-XSP Mux", SND_SOC_NOPM, + 0, 0, &spk_xsp_mixer), + + SND_SOC_DAPM_PGA("HL Left DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("HL Right DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("SPK DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("ESL DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("HP Amp", CS42L73_PWRCTL3, 0, 1, + &hp_amp_ctl), + SND_SOC_DAPM_SWITCH("LO Amp", CS42L73_PWRCTL3, 1, 1, + &lo_amp_ctl), + SND_SOC_DAPM_SWITCH("SPK Amp", CS42L73_PWRCTL3, 2, 1, + &spk_amp_ctl), + SND_SOC_DAPM_SWITCH("EAR Amp", CS42L73_PWRCTL3, 3, 1, + &ear_amp_ctl), + SND_SOC_DAPM_SWITCH("SPKLO Amp", CS42L73_PWRCTL3, 4, 1, + &spklo_amp_ctl), + + SND_SOC_DAPM_OUTPUT("HPOUTA"), + SND_SOC_DAPM_OUTPUT("HPOUTB"), + SND_SOC_DAPM_OUTPUT("LINEOUTA"), + SND_SOC_DAPM_OUTPUT("LINEOUTB"), + SND_SOC_DAPM_OUTPUT("EAROUT"), + SND_SOC_DAPM_OUTPUT("SPKOUT"), + SND_SOC_DAPM_OUTPUT("SPKLINEOUT"), +}; + +static const struct snd_soc_dapm_route cs42l73_audio_map[] = { + + /* SPKLO EARSPK Paths */ + {"EAROUT", NULL, "EAR Amp"}, + {"SPKLINEOUT", NULL, "SPKLO Amp"}, + + {"EAR Amp", "Switch", "ESL DAC"}, + {"SPKLO Amp", "Switch", "ESL DAC"}, + + {"ESL DAC", "ESL-ASP Mono Volume", "ESL Mixer"}, + {"ESL DAC", "ESL-XSP Mono Volume", "ESL Mixer"}, + {"ESL DAC", "ESL-VSP Mono Volume", "VSPIN"}, + /* Loopback */ + {"ESL DAC", "ESL-IP Mono Volume", "Input Left Capture"}, + {"ESL DAC", "ESL-IP Mono Volume", "Input Right Capture"}, + + {"ESL Mixer", NULL, "ESL-ASP Mux"}, + {"ESL Mixer", NULL, "ESL-XSP Mux"}, + + {"ESL-ASP Mux", "Left", "ASPINL"}, + {"ESL-ASP Mux", "Right", "ASPINR"}, + {"ESL-ASP Mux", "Mono Mix", "ASPINM"}, + + {"ESL-XSP Mux", "Left", "XSPINL"}, + {"ESL-XSP Mux", "Right", "XSPINR"}, + {"ESL-XSP Mux", "Mono Mix", "XSPINM"}, + + /* Speakerphone Paths */ + {"SPKOUT", NULL, "SPK Amp"}, + {"SPK Amp", "Switch", "SPK DAC"}, + + {"SPK DAC", "SPK-ASP Mono Volume", "SPK Mixer"}, + {"SPK DAC", "SPK-XSP Mono Volume", "SPK Mixer"}, + {"SPK DAC", "SPK-VSP Mono Volume", "VSPIN"}, + /* Loopback */ + {"SPK DAC", "SPK-IP Mono Volume", "Input Left Capture"}, + {"SPK DAC", "SPK-IP Mono Volume", "Input Right Capture"}, + + {"SPK Mixer", NULL, "SPK-ASP Mux"}, + {"SPK Mixer", NULL, "SPK-XSP Mux"}, + + {"SPK-ASP Mux", "Left", "ASPINL"}, + {"SPK-ASP Mux", "Mono Mix", "ASPINM"}, + {"SPK-ASP Mux", "Right", "ASPINR"}, + + {"SPK-XSP Mux", "Left", "XSPINL"}, + {"SPK-XSP Mux", "Mono Mix", "XSPINM"}, + {"SPK-XSP Mux", "Right", "XSPINR"}, + + /* HP LineOUT Paths */ + {"HPOUTA", NULL, "HP Amp"}, + {"HPOUTB", NULL, "HP Amp"}, + {"LINEOUTA", NULL, "LO Amp"}, + {"LINEOUTB", NULL, "LO Amp"}, + + {"HP Amp", "Switch", "HL Left DAC"}, + {"HP Amp", "Switch", "HL Right DAC"}, + {"LO Amp", "Switch", "HL Left DAC"}, + {"LO Amp", "Switch", "HL Right DAC"}, + + {"HL Left DAC", "HL-XSP Volume", "HL Left Mixer"}, + {"HL Right DAC", "HL-XSP Volume", "HL Right Mixer"}, + {"HL Left DAC", "HL-ASP Volume", "HL Left Mixer"}, + {"HL Right DAC", "HL-ASP Volume", "HL Right Mixer"}, + {"HL Left DAC", "HL-VSP Volume", "HL Left Mixer"}, + {"HL Right DAC", "HL-VSP Volume", "HL Right Mixer"}, + /* Loopback */ + {"HL Left DAC", "HL-IP Volume", "HL Left Mixer"}, + {"HL Right DAC", "HL-IP Volume", "HL Right Mixer"}, + {"HL Left Mixer", NULL, "Input Left Capture"}, + {"HL Right Mixer", NULL, "Input Right Capture"}, + + {"HL Left Mixer", NULL, "ASPINL"}, + {"HL Right Mixer", NULL, "ASPINR"}, + {"HL Left Mixer", NULL, "XSPINL"}, + {"HL Right Mixer", NULL, "XSPINR"}, + {"HL Left Mixer", NULL, "VSPIN"}, + {"HL Right Mixer", NULL, "VSPIN"}, + + /* Capture Paths */ + {"MIC1", NULL, "MIC1 Bias"}, + {"PGA Left Mux", "Mic 1", "MIC1"}, + {"MIC2", NULL, "MIC2 Bias"}, + {"PGA Right Mux", "Mic 2", "MIC2"}, + + {"PGA Left Mux", "Line A", "LINEINA"}, + {"PGA Right Mux", "Line B", "LINEINB"}, + + {"PGA Left", NULL, "PGA Left Mux"}, + {"PGA Right", NULL, "PGA Right Mux"}, + + {"ADC Left", NULL, "PGA Left"}, + {"ADC Right", NULL, "PGA Right"}, + + {"Input Left Capture", "ADC Left Input", "ADC Left"}, + {"Input Right Capture", "ADC Right Input", "ADC Right"}, + {"Input Left Capture", "DMIC Left Input", "DMIC Left"}, + {"Input Right Capture", "DMIC Right Input", "DMIC Right"}, + + /* Audio Capture */ + {"ASPL Output Mixer", NULL, "Input Left Capture"}, + {"ASPR Output Mixer", NULL, "Input Right Capture"}, + + {"ASPOUTL", "ASP-IP Volume", "ASPL Output Mixer"}, + {"ASPOUTR", "ASP-IP Volume", "ASPR Output Mixer"}, + + /* Auxillary Capture */ + {"XSPL Output Mixer", NULL, "Input Left Capture"}, + {"XSPR Output Mixer", NULL, "Input Right Capture"}, + + {"XSPOUTL", "XSP-IP Volume", "XSPL Output Mixer"}, + {"XSPOUTR", "XSP-IP Volume", "XSPR Output Mixer"}, + + {"XSPOUT Mux", NULL, "XSPL Output Mixer"}, + {"XSPOUT Mux", NULL, "XSPR Output Mixer"}, + + {"XSPOUTL", "Mono", "XSPOUT Mux"}, + {"XSPOUTR", "Mono", "XSPOUT Mux"}, + + {"XSPOUTL", "Stereo", "XSPOUT Mux"}, + {"XSPOUTR", "Stereo", "XSPOUT Mux"}, + + /* Voice Capture */ + {"VSPL Output Mixer", NULL, "Input Left Capture"}, + {"VSPR Output Mixer", NULL, "Input Left Capture"}, + + {"VSPOUTL", "VSP-IP Volume", "VSPL Output Mixer"}, + {"VSPOUTR", "VSP-IP Volume", "VSPR Output Mixer"}, + + {"VSPOUT Mux", NULL, "VSPL Output Mixer"}, + {"VSPOUT Mux", NULL, "VSPR Output Mixer"}, + + {"VSPOUTL", "Mono", "VSPOUT Mux"}, + {"VSPOUTR", "Mono", "VSPOUT Mux"}, + + {"VSPOUTL", "Stereo", "VSPOUT Mux"}, + {"VSPOUTR", "Stereo", "VSPOUT Mux"}, +}; + +struct cs42l73_mclk_div { + u32 mclk; + u32 srate; + u8 mmcc; +}; + +static struct cs42l73_mclk_div cs42l73_mclk_coeffs[] = { + /* MCLK, Sample Rate, xMMCC[5:0] */ + {5644800, 11025, 0x30}, + {5644800, 22050, 0x20}, + {5644800, 44100, 0x10}, + + {6000000, 8000, 0x39}, + {6000000, 11025, 0x33}, + {6000000, 12000, 0x31}, + {6000000, 16000, 0x29}, + {6000000, 22050, 0x23}, + {6000000, 24000, 0x21}, + {6000000, 32000, 0x19}, + {6000000, 44100, 0x13}, + {6000000, 48000, 0x11}, + + {6144000, 8000, 0x38}, + {6144000, 12000, 0x30}, + {6144000, 16000, 0x28}, + {6144000, 24000, 0x20}, + {6144000, 32000, 0x18}, + {6144000, 48000, 0x10}, + + {6500000, 8000, 0x3C}, + {6500000, 11025, 0x35}, + {6500000, 12000, 0x34}, + {6500000, 16000, 0x2C}, + {6500000, 22050, 0x25}, + {6500000, 24000, 0x24}, + {6500000, 32000, 0x1C}, + {6500000, 44100, 0x15}, + {6500000, 48000, 0x14}, + + {6400000, 8000, 0x3E}, + {6400000, 11025, 0x37}, + {6400000, 12000, 0x36}, + {6400000, 16000, 0x2E}, + {6400000, 22050, 0x27}, + {6400000, 24000, 0x26}, + {6400000, 32000, 0x1E}, + {6400000, 44100, 0x17}, + {6400000, 48000, 0x16}, +}; + +struct cs42l73_mclkx_div { + u32 mclkx; + u8 ratio; + u8 mclkdiv; +}; + +static struct cs42l73_mclkx_div cs42l73_mclkx_coeffs[] = { + {5644800, 1, 0}, /* 5644800 */ + {6000000, 1, 0}, /* 6000000 */ + {6144000, 1, 0}, /* 6144000 */ + {11289600, 2, 2}, /* 5644800 */ + {12288000, 2, 2}, /* 6144000 */ + {12000000, 2, 2}, /* 6000000 */ + {13000000, 2, 2}, /* 6500000 */ + {19200000, 3, 3}, /* 6400000 */ + {24000000, 4, 4}, /* 6000000 */ + {26000000, 4, 4}, /* 6500000 */ + {38400000, 6, 5} /* 6400000 */ +}; + +static int cs42l73_get_mclkx_coeff(int mclkx) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs42l73_mclkx_coeffs); i++) { + if (cs42l73_mclkx_coeffs[i].mclkx == mclkx) + return i; + } + return -EINVAL; +} + +static int cs42l73_get_mclk_coeff(int mclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs42l73_mclk_coeffs); i++) { + if (cs42l73_mclk_coeffs[i].mclk == mclk && + cs42l73_mclk_coeffs[i].srate == srate) + return i; + } + return -EINVAL; + +} + +static int cs42l73_set_mclk(struct snd_soc_dai *dai, unsigned int freq) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + + int mclkx_coeff; + u32 mclk = 0; + u8 dmmcc = 0; + + /* MCLKX -> MCLK */ + mclkx_coeff = cs42l73_get_mclkx_coeff(freq); + + mclk = cs42l73_mclkx_coeffs[mclkx_coeff].mclkx / + cs42l73_mclkx_coeffs[mclkx_coeff].ratio; + + dev_dbg(codec->dev, "MCLK%u %u <-> internal MCLK %u\n", + priv->mclksel + 1, cs42l73_mclkx_coeffs[mclkx_coeff].mclkx, + mclk); + + dmmcc = (priv->mclksel << 4) | + (cs42l73_mclkx_coeffs[mclkx_coeff].mclkdiv << 1); + + snd_soc_write(codec, CS42L73_DMMCC, dmmcc); + + priv->sysclk = mclkx_coeff; + priv->mclk = mclk; + + return 0; +} + +static int cs42l73_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case CS42L73_CLKID_MCLK1: + break; + case CS42L73_CLKID_MCLK2: + break; + default: + return -EINVAL; + } + + if ((cs42l73_set_mclk(dai, freq)) < 0) { + dev_err(codec->dev, "Unable to set MCLK for dai %s\n", + dai->name); + return -EINVAL; + } + + priv->mclksel = clk_id; + + return 0; +} + +static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + u8 id = codec_dai->id; + u8 inv, format; + u8 spc, mmcc; + + spc = snd_soc_read(codec, CS42L73_SPC(id)); + mmcc = snd_soc_read(codec, CS42L73_MMCC(id)); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + mmcc |= MS_MASTER; + break; + + case SND_SOC_DAIFMT_CBS_CFS: + mmcc &= ~MS_MASTER; + break; + + default: + return -EINVAL; + } + + format = (fmt & SND_SOC_DAIFMT_FORMAT_MASK); + inv = (fmt & SND_SOC_DAIFMT_INV_MASK); + + switch (format) { + case SND_SOC_DAIFMT_I2S: + spc &= ~SPDIF_PCM; + break; + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + if (mmcc & MS_MASTER) { + dev_err(codec->dev, + "PCM format in slave mode only\n"); + return -EINVAL; + } + if (id == CS42L73_ASP) { + dev_err(codec->dev, + "PCM format is not supported on ASP port\n"); + return -EINVAL; + } + spc |= SPDIF_PCM; + break; + default: + return -EINVAL; + } + + if (spc & SPDIF_PCM) { + spc &= (31 << 3); /* Clear PCM mode, set MSB->LSB */ + switch (format) { + case SND_SOC_DAIFMT_DSP_B: + if (inv == SND_SOC_DAIFMT_IB_IF) + spc |= (PCM_MODE0 << 4); + if (inv == SND_SOC_DAIFMT_IB_NF) + spc |= (PCM_MODE1 << 4); + break; + case SND_SOC_DAIFMT_DSP_A: + if (inv == SND_SOC_DAIFMT_IB_IF) + spc |= (PCM_MODE1 << 4); + break; + default: + return -EINVAL; + } + } + + priv->config[id].spc = spc; + priv->config[id].mmcc = mmcc; + + return 0; +} + +static u32 cs42l73_asrc_rates[] = { + 8000, 11025, 12000, 16000, 22050, + 24000, 32000, 44100, 48000 +}; + +static unsigned int cs42l73_get_xspfs_coeff(u32 rate) +{ + int i; + for (i = 0; i < ARRAY_SIZE(cs42l73_asrc_rates); i++) { + if (cs42l73_asrc_rates[i] == rate) + return i + 1; + } + return 0; /* 0 = Don't know */ +} + +static void cs42l73_update_asrc(struct snd_soc_codec *codec, int id, int srate) +{ + u8 spfs = 0; + + if (srate > 0) + spfs = cs42l73_get_xspfs_coeff(srate); + + switch (id) { + case CS42L73_XSP: + snd_soc_update_bits(codec, CS42L73_VXSPFS, 0x0f, spfs); + break; + case CS42L73_ASP: + snd_soc_update_bits(codec, CS42L73_ASPC, 0x3c, spfs << 2); + break; + case CS42L73_VSP: + snd_soc_update_bits(codec, CS42L73_VXSPFS, 0xf0, spfs << 4); + break; + default: + break; + } +} + +static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + int id = dai->id; + int mclk_coeff; + int srate = params_rate(params); + + if (priv->config[id].mmcc & MS_MASTER) { + /* CS42L73 Master */ + /* MCLK -> srate */ + mclk_coeff = + cs42l73_get_mclk_coeff(priv->mclk, srate); + + if (mclk_coeff < 0) + return -EINVAL; + + dev_dbg(codec->dev, + "DAI[%d]: MCLK %u, srate %u, MMCC[5:0] = %x\n", + id, priv->mclk, srate, + cs42l73_mclk_coeffs[mclk_coeff].mmcc); + + priv->config[id].mmcc &= 0xC0; + priv->config[id].mmcc |= cs42l73_mclk_coeffs[mclk_coeff].mmcc; + priv->config[id].spc &= 0xFC; + priv->config[id].spc |= MCK_SCLK_64FS; + } else { + /* CS42L73 Slave */ + priv->config[id].spc &= 0xFC; + priv->config[id].spc |= MCK_SCLK_64FS; + } + /* Update ASRCs */ + priv->config[id].srate = srate; + + snd_soc_write(codec, CS42L73_SPC(id), priv->config[id].spc); + snd_soc_write(codec, CS42L73_MMCC(id), priv->config[id].mmcc); + + cs42l73_update_asrc(codec, id, srate); + + return 0; +} + +static int cs42l73_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_update_bits(codec, CS42L73_DMMCC, MCLKDIS, 0); + snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 0); + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = snd_soc_cache_sync(codec); + if (ret < 0) { + dev_err(codec->dev, + "Failed to sync cache: %d\n", ret); + return ret; + } + } + snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 1); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 1); + snd_soc_update_bits(codec, CS42L73_DMMCC, MCLKDIS, 1); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static int cs42l73_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + int id = dai->id; + + return snd_soc_update_bits(codec, CS42L73_SPC(id), + 0x7F, tristate << 7); +} + +static struct snd_pcm_hw_constraint_list constraints_12_24 = { + .count = ARRAY_SIZE(cs42l73_asrc_rates), + .list = cs42l73_asrc_rates, +}; + +static int cs42l73_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_12_24); + return 0; +} + +/* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */ +#define CS42L73_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) + + +#define CS42L73_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops cs42l73_ops = { + .startup = cs42l73_pcm_startup, + .hw_params = cs42l73_pcm_hw_params, + .set_fmt = cs42l73_set_dai_fmt, + .set_sysclk = cs42l73_set_sysclk, + .set_tristate = cs42l73_set_tristate, +}; + +static struct snd_soc_dai_driver cs42l73_dai[] = { + { + .name = "cs42l73-xsp", + .id = CS42L73_XSP, + .playback = { + .stream_name = "XSP Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .capture = { + .stream_name = "XSP Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .ops = &cs42l73_ops, + .symmetric_rates = 1, + }, + { + .name = "cs42l73-asp", + .id = CS42L73_ASP, + .playback = { + .stream_name = "ASP Playback", + .channels_min = 2, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .capture = { + .stream_name = "ASP Capture", + .channels_min = 2, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .ops = &cs42l73_ops, + .symmetric_rates = 1, + }, + { + .name = "cs42l73-vsp", + .id = CS42L73_VSP, + .playback = { + .stream_name = "VSP Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .capture = { + .stream_name = "VSP Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .ops = &cs42l73_ops, + .symmetric_rates = 1, + } +}; + +static int cs42l73_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int cs42l73_resume(struct snd_soc_codec *codec) +{ + + cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static int cs42l73_probe(struct snd_soc_codec *codec) +{ + int ret; + unsigned int devid = 0; + struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(codec); + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* initialize codec */ + ret = snd_soc_read(codec, CS42L73_DEVID_AB); + devid = (ret & 0xFF) << 12; + + ret = snd_soc_read(codec, CS42L73_DEVID_CD); + devid |= (ret & 0xFF) << 4; + + ret = snd_soc_read(codec, CS42L73_DEVID_E); + devid |= (ret & 0xF0) >> 4; + + + if (devid != CS42L73_DEVID) { + dev_err(codec->dev, + "CS42L73 Device ID (%X). Expected %X\n", + devid, CS42L73_DEVID); + return ret; + } + + ret = snd_soc_read(codec, CS42L73_REVID); + if (ret < 0) { + dev_err(codec->dev, "Get Revision ID failed\n"); + return ret; + } + + dev_info(codec->dev, + "Cirrus Logic CS42L73, Revision: %02X\n", ret & 0xFF); + + cs42l73->mclksel = CS42L73_CLKID_MCLK1; /* MCLK1 as master clk */ + cs42l73->mclk = 0; + + return ret; +} + +static int cs42l73_remove(struct snd_soc_codec *codec) +{ + cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs42l73 = { + .probe = cs42l73_probe, + .remove = cs42l73_remove, + .suspend = cs42l73_suspend, + .resume = cs42l73_resume, + .set_bias_level = cs42l73_set_bias_level, + .reg_cache_size = ARRAY_SIZE(cs42l73_reg), + .reg_cache_default = cs42l73_reg, + .reg_word_size = sizeof(u8), + .dapm_widgets = cs42l73_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42l73_dapm_widgets), + .dapm_routes = cs42l73_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs42l73_audio_map), + .controls = cs42l73_snd_controls, + .num_controls = ARRAY_SIZE(cs42l73_snd_controls), +}; + +static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs42l73_private *cs42l73; + int ret; + + cs42l73 = kzalloc((sizeof *cs42l73), GFP_KERNEL); + if (!cs42l73) { + dev_err(&i2c_client->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + i2c_set_clientdata(i2c_client, cs42l73); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs42l73, cs42l73_dai, + ARRAY_SIZE(cs42l73_dai)); + if (ret < 0) + kfree(cs42l73); + return ret; +} + +static __devexit int cs42l73_i2c_remove(struct i2c_client *client) +{ + struct cs42l73_private *cs42l73 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + kfree(cs42l73); + + return 0; +} + +static const struct i2c_device_id cs42l73_id[] = { + {"cs42l73", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs42l73_id); + +static struct i2c_driver cs42l73_i2c_driver = { + .driver = { + .name = "cs42l73", + .owner = THIS_MODULE, + }, + .id_table = cs42l73_id, + .probe = cs42l73_i2c_probe, + .remove = __devexit_p(cs42l73_i2c_remove), + +}; + +static int __init cs42l73_modinit(void) +{ + int ret; + ret = i2c_add_driver(&cs42l73_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "%s: can't add i2c driver\n", __func__); + return ret; + } + return 0; +} + +module_init(cs42l73_modinit); + +static void __exit cs42l73_exit(void) +{ + i2c_del_driver(&cs42l73_i2c_driver); +} + +module_exit(cs42l73_exit); + +MODULE_DESCRIPTION("ASoC CS42L73 driver"); +MODULE_AUTHOR("Georgi Vlaev, Nucleus Systems Ltd, joe@nucleusys.com"); +MODULE_AUTHOR("Brian Austin, Cirrus Logic Inc, brian.austin@cirrus.com"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42l73.h b/sound/soc/codecs/cs42l73.h new file mode 100644 index 0000000..188f06d --- /dev/null +++ b/sound/soc/codecs/cs42l73.h @@ -0,0 +1,227 @@ +/* + * ALSA SoC CS42L73 codec driver + * + * Copyright 2011 Cirrus Logic, Inc. + * + * Author: Georgi Vlaev joe@nucleusys.com + * Brian Austin brian.austin@cirrus.com + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __CS42L73_H__ +#define __CS42L73_H__ + +/* I2C Registers */ +/* I2C Address: 1001010[R/W] - 10010100 = 0x94(Write); 10010101 = 0x95(Read) */ +#define CS42L73_CHIP_ID 0x4a +#define CS42L73_DEVID_AB 0x01 /* Device ID A & B [RO]. */ +#define CS42L73_DEVID_CD 0x02 /* Device ID C & D [RO]. */ +#define CS42L73_DEVID_E 0x03 /* Device ID E [RO]. */ +#define CS42L73_REVID 0x05 /* Revision ID [RO]. */ +#define CS42L73_PWRCTL1 0x06 /* Power Control 1. */ +#define CS42L73_PWRCTL2 0x07 /* Power Control 2. */ +#define CS42L73_PWRCTL3 0x08 /* Power Control 3. */ +#define CS42L73_CPFCHC 0x09 /* Charge Pump Freq. Class H Ctl. */ +#define CS42L73_OLMBMSDC 0x0A /* Output Load, MIC Bias, MIC2 SDT */ +#define CS42L73_DMMCC 0x0B /* Digital MIC & Master Clock Ctl. */ +#define CS42L73_XSPC 0x0C /* Auxiliary Serial Port (XSP) Ctl. */ +#define CS42L73_XSPMMCC 0x0D /* XSP Master Mode Clocking Control. */ +#define CS42L73_ASPC 0x0E /* Audio Serial Port (ASP) Control. */ +#define CS42L73_ASPMMCC 0x0F /* ASP Master Mode Clocking Control. */ +#define CS42L73_VSPC 0x10 /* Voice Serial Port (VSP) Control. */ +#define CS42L73_VSPMMCC 0x11 /* VSP Master Mode Clocking Control. */ +#define CS42L73_VXSPFS 0x12 /* VSP & XSP Sample Rate. */ +#define CS42L73_MIOPC 0x13 /* Misc. Input & Output Path Control. */ +#define CS42L73_ADCIPC 0x14 /* ADC/IP Control. */ +#define CS42L73_MICAPREPGAAVOL 0x15 /* MIC 1 [A] PreAmp, PGAA Vol. */ +#define CS42L73_MICBPREPGABVOL 0x16 /* MIC 2 [B] PreAmp, PGAB Vol. */ +#define CS42L73_IPADVOL 0x17 /* Input Pat7h A Digital Volume. */ +#define CS42L73_IPBDVOL 0x18 /* Input Path B Digital Volume. */ +#define CS42L73_PBDC 0x19 /* Playback Digital Control. */ +#define CS42L73_HLADVOL 0x1A /* HP/Line A Out Digital Vol. */ +#define CS42L73_HLBDVOL 0x1B /* HP/Line B Out Digital Vol. */ +#define CS42L73_SPKDVOL 0x1C /* Spkphone Out [A] Digital Vol. */ +#define CS42L73_ESLDVOL 0x1D /* Ear/Spkphone LO [B] Digital */ +#define CS42L73_HPAAVOL 0x1E /* HP A Analog Volume. */ +#define CS42L73_HPBAVOL 0x1F /* HP B Analog Volume. */ +#define CS42L73_LOAAVOL 0x20 /* Line Out A Analog Volume. */ +#define CS42L73_LOBAVOL 0x21 /* Line Out B Analog Volume. */ +#define CS42L73_STRINV 0x22 /* Stereo Input Path Adv. Vol. */ +#define CS42L73_XSPINV 0x23 /* Auxiliary Port Input Advisory Vol. */ +#define CS42L73_ASPINV 0x24 /* Audio Port Input Advisory Vol. */ +#define CS42L73_VSPINV 0x25 /* Voice Port Input Advisory Vol. */ +#define CS42L73_LIMARATEHL 0x26 /* Lmtr Attack Rate HP/Line. */ +#define CS42L73_LIMRRATEHL 0x27 /* Lmtr Ctl, Rel.Rate HP/Line. */ +#define CS42L73_LMAXHL 0x28 /* Lmtr Thresholds HP/Line. */ +#define CS42L73_LIMARATESPK 0x29 /* Lmtr Attack Rate Spkphone [A]. */ +#define CS42L73_LIMRRATESPK 0x2A /* Lmtr Ctl,Release Rate Spk. [A]. */ +#define CS42L73_LMAXSPK 0x2B /* Lmtr Thresholds Spkphone [A]. */ +#define CS42L73_LIMARATEESL 0x2C /* Lmtr Attack Rate */ +#define CS42L73_LIMRRATEESL 0x2D /* Lmtr Ctl,Release Rate */ +#define CS42L73_LMAXESL 0x2E /* Lmtr Thresholds */ +#define CS42L73_ALCARATE 0x2F /* ALC Enable, Attack Rate AB. */ +#define CS42L73_ALCRRATE 0x30 /* ALC Release Rate AB. */ +#define CS42L73_ALCMINMAX 0x31 /* ALC Thresholds AB. */ +#define CS42L73_NGCAB 0x32 /* Noise Gate Ctl AB. */ +#define CS42L73_ALCNGMC 0x33 /* ALC & Noise Gate Misc Ctl. */ +#define CS42L73_MIXERCTL 0x34 /* Mixer Control. */ +#define CS42L73_HLAIPAA 0x35 /* HP/LO Left Mixer: L. */ +#define CS42L73_HLBIPBA 0x36 /* HP/LO Right Mixer: R. */ +#define CS42L73_HLAXSPAA 0x37 /* HP/LO Left Mixer: XSP L */ +#define CS42L73_HLBXSPBA 0x38 /* HP/LO Right Mixer: XSP R */ +#define CS42L73_HLAASPAA 0x39 /* HP/LO Left Mixer: ASP L */ +#define CS42L73_HLBASPBA 0x3A /* HP/LO Right Mixer: ASP R */ +#define CS42L73_HLAVSPMA 0x3B /* HP/LO Left Mixer: VSP. */ +#define CS42L73_HLBVSPMA 0x3C /* HP/LO Right Mixer: VSP */ +#define CS42L73_XSPAIPAA 0x3D /* XSP Left Mixer: Left */ +#define CS42L73_XSPBIPBA 0x3E /* XSP Rt. Mixer: Right */ +#define CS42L73_XSPAXSPAA 0x3F /* XSP Left Mixer: XSP L */ +#define CS42L73_XSPBXSPBA 0x40 /* XSP Rt. Mixer: XSP R */ +#define CS42L73_XSPAASPAA 0x41 /* XSP Left Mixer: ASP L */ +#define CS42L73_XSPAASPBA 0x42 /* XSP Rt. Mixer: ASP R */ +#define CS42L73_XSPAVSPMA 0x43 /* XSP Left Mixer: VSP */ +#define CS42L73_XSPBVSPMA 0x44 /* XSP Rt. Mixer: VSP */ +#define CS42L73_ASPAIPAA 0x45 /* ASP Left Mixer: Left */ +#define CS42L73_ASPBIPBA 0x46 /* ASP Rt. Mixer: Right */ +#define CS42L73_ASPAXSPAA 0x47 /* ASP Left Mixer: XSP L */ +#define CS42L73_ASPBXSPBA 0x48 /* ASP Rt. Mixer: XSP R */ +#define CS42L73_ASPAASPAA 0x49 /* ASP Left Mixer: ASP L */ +#define CS42L73_ASPBASPBA 0x4A /* ASP Rt. Mixer: ASP R */ +#define CS42L73_ASPAVSPMA 0x4B /* ASP Left Mixer: VSP */ +#define CS42L73_ASPBVSPMA 0x4C /* ASP Rt. Mixer: VSP */ +#define CS42L73_VSPAIPAA 0x4D /* VSP Left Mixer: Left */ +#define CS42L73_VSPBIPBA 0x4E /* VSP Rt. Mixer: Right */ +#define CS42L73_VSPAXSPAA 0x4F /* VSP Left Mixer: XSP L */ +#define CS42L73_VSPBXSPBA 0x50 /* VSP Rt. Mixer: XSP R */ +#define CS42L73_VSPAASPAA 0x51 /* VSP Left Mixer: ASP Left */ +#define CS42L73_VSPBASPBA 0x52 /* VSP Rt. Mixer: ASP Right */ +#define CS42L73_VSPAVSPMA 0x53 /* VSP Left Mixer: VSP */ +#define CS42L73_VSPBVSPMA 0x54 /* VSP Rt. Mixer: VSP */ +#define CS42L73_MMIXCTL 0x55 /* Mono Mixer Controls. */ +#define CS42L73_SPKMIPMA 0x56 /* SPK Mono Mixer: In. Path */ +#define CS42L73_SPKMXSPA 0x57 /* SPK Mono Mixer: XSP Mono/L/R Att. */ +#define CS42L73_SPKMASPA 0x58 /* SPK Mono Mixer: ASP Mono/L/R Att. */ +#define CS42L73_SPKMVSPMA 0x59 /* SPK Mono Mixer: VSP Mono Atten. */ +#define CS42L73_ESLMIPMA 0x5A /* Ear/SpLO Mono Mixer: */ +#define CS42L73_ESLMXSPA 0x5B /* Ear/SpLO Mono Mixer: XSP */ +#define CS42L73_ESLMASPA 0x5C /* Ear/SpLO Mono Mixer: ASP */ +#define CS42L73_ESLMVSPMA 0x5D /* Ear/SpLO Mono Mixer: VSP */ +#define CS42L73_IM1 0x5E /* Interrupt Mask 1. */ +#define CS42L73_IM2 0x5F /* Interrupt Mask 2. */ +#define CS42L73_IS1 0x60 /* Interrupt Status 1 [RO]. */ +#define CS42L73_IS2 0x61 /* Interrupt Status 2 [RO]. */ + +/* Bitfield Definitions */ + +/* CS42L73_PWRCTL1 */ +#define PDN_ADCB (1 << 7) +#define PDN_DMICB (1 << 6) +#define PDN_ADCA (1 << 5) +#define PDN_DMICA (1 << 4) +#define PDN_LDO (1 << 2) +#define DISCHG_FILT (1 << 1) +#define PDN (1 << 0) + +/* CS42L73_PWRCTL2 */ +#define PDN_MIC2_BIAS (1 << 7) +#define PDN_MIC1_BIAS (1 << 6) +#define PDN_VSP (1 << 4) +#define PDN_ASP_SDOUT (1 << 3) +#define PDN_ASP_SDIN (1 << 2) +#define PDN_XSP_SDOUT (1 << 1) +#define PDN_XSP_SDIN (1 << 0) + +/* CS42L73_PWRCTL3 */ +#define PDN_THMS (1 << 5) +#define PDN_SPKLO (1 << 4) +#define PDN_EAR (1 << 3) +#define PDN_SPK (1 << 2) +#define PDN_LO (1 << 1) +#define PDN_HP (1 << 0) + +/* Thermal Overload Detect. Requires interrupt ... */ +#define THMOVLD_150C 0 +#define THMOVLD_132C 1 +#define THMOVLD_115C 2 +#define THMOVLD_098C 3 + + +/* CS42L73_ASPC, CS42L73_XSPC, CS42L73_VSPC */ +#define SP_3ST (1 << 7) +#define SPDIF_I2S 0 +#define SPDIF_PCM (1 << 6) +#define PCM_MODE0 0 +#define PCM_MODE1 1 +#define PCM_MODE2 2 +#define PCM_BO_MSBLSB 0 +#define PCM_BO_LSBMSB 1 +#define MCK_SCLK_64FS 0 +#define MCK_SCLK_MCLK 2 +#define MCK_SCLK_PREMCLK 3 + +/* CS42L73_xSPMMCC */ +#define MS_MASTER (1 << 7) + + +/* CS42L73_DMMCC */ +#define MCLKDIS (1 << 0) +#define MCLKSEL_MCLK2 (1 << 4) +#define MCLKSEL_MCLK1 (0 << 4) + +/* CS42L73 MCLK derived from MCLK1 or MCLK2 */ +#define CS42L73_CLKID_MCLK1 0 +#define CS42L73_CLKID_MCLK2 1 + +#define CS42L73_MCLKXDIV 0 +#define CS42L73_MMCCDIV 1 + +#define CS42L73_XSP 0 +#define CS42L73_ASP 1 +#define CS42L73_VSP 2 + +/* IS1, IM1 */ +#define MIC2_SDET (1 << 6) +#define THMOVLD (1 << 4) +#define DIGMIXOVFL (1 << 3) +#define IPBOVFL (1 << 1) +#define IPAOVFL (1 << 0) + +/* Analog Softramp */ +#define ANLGOSFT (1 << 0) + +/* HP A/B Analog Mute */ +#define HPA_MUTE (1 << 7) +/* LO A/B Analog Mute */ +#define LOA_MUTE (1 << 7) +/* Digital Mute */ +#define HLAD_MUTE (1 << 0) +#define HLBD_MUTE (1 << 1) +#define SPKD_MUTE (1 << 2) +#define ESLD_MUTE (1 << 3) + +/* Misc defines for codec */ +#define CS42L73_RESET_GPIO 143 + +#define CS42L73_DEVID 0x00042A73 +#define CS42L73_MCLKX_MIN 5644800 +#define CS42L73_MCLKX_MAX 38400000 + +#define CS42L73_SPC(id) (CS42L73_XSPC + (id << 1)) +#define CS42L73_MMCC(id) (CS42L73_XSPMMCC + (id << 1)) +#define CS42L73_SPFS(id) ((id == CS42L73_ASP) ? CS42L73_ASPC : CS42L73_VXSPFS) + +#endif /* __CS42L73_H__ */