[alsa-devel] [PATCH v2] ASoC: Add support for cs42l73 codec
This patch adds support for the Cirrus Logic CS42L73 low power stereo codec.
This patch has cleared checkpatch.pl with no warnings or errors. Code changes requested were implemented. ASoC API changes requested were implemented.
Signed-off-by: Brian Austin brian.austin@cirrus.com --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs42l73.c | 1047 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs42l73.h | 223 ++++++++++ 4 files changed, 1276 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 3449431..9107c48 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 787881b..e89e84b 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..18a49d8 --- /dev/null +++ b/sound/soc/codecs/cs42l73.c @@ -0,0 +1,1047 @@ +/* + * cs42l73.c -- CS42L73 ALSA Soc Audio driver + * + * Copyright 2011 Cirrus Logic, Inc. + * + * Authors: Georgi Vlaev, Nucleus Systems Ltd, office@nucleusys.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 <linux/gpio.h> + +#include "cs42l73.h" + +struct sp_config { + u8 spc, mmcc, spfs; + u32 srate; +}; + +struct cs42l73_private { + enum snd_soc_control_type control_type; + void *control_data; + 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, +/*10*/ 0x00, 0x15, 0x00, 0x06, +/*14*/ 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, +/*60*/ 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(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 s42l73_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 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 cs42l73_mono_mixer_text[] = { + "Left", "Right", "Mono Mix"}; + +static const struct soc_enum spk_asp_mono_mixer_enum = + SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 6, + ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text); + +static const struct soc_enum spk_xsp_mono_mixer_enum = + SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 4, + ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text); + +static const struct soc_enum esl_asp_mono_mixer_enum = + SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 2, + ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text); + +static const struct soc_enum esl_xsp_mono_mixer_enum = + SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 0, + ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text); + +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 vspout_mixer_enum = + SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 5, + ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text); + +static const struct soc_enum xspout_mixer_enum = + SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 4, + ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text); + +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_DOUBLE_S8_TLV("Speakerphone Digital Playback Volume", + CS42L73_SPKDVOL, 0x34, 0xE4, hl_tlv), + + SOC_DOUBLE_S8_TLV("Ear Speaker Digital Playback Volume", + CS42L73_ESLDVOL, 0x34, 0xE4, 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("Invert ADC Signal Polarity Switch", CS42L73_ADCIPC, 1, 5, 1, + 0), + SOC_DOUBLE("ADC Boost Switch", CS42L73_ADCIPC, 2, 6, 1, 0), + + SOC_SINGLE("Charge Pump Frequency Volume", CS42L73_CPFCHC, 4, 15, 0), + + SOC_SINGLE("HL Limiter Attack Rate Volume", CS42L73_LIMARATEHL, 0, 0x3F, + 0), + SOC_SINGLE("HL Limiter Release Rate Volume", 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, 1, + limiter_tlv), + SOC_SINGLE_TLV("ALC Min Threshold Volume", CS42L73_ALCMINMAX, 2, 7, 1, + 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 Attenuation Volume", + CS42L73_XSPAIPAA, CS42L73_XSPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-XSP Attenuation Volume", + CS42L73_XSPAXSPAA, CS42L73_XSPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-ASP Attenuation Volume", + CS42L73_XSPAASPAA, CS42L73_XSPAASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-VSP Attenuation Volume", + CS42L73_XSPAVSPMA, CS42L73_XSPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("ASP-IP Attenuation Volume", + CS42L73_ASPAIPAA, CS42L73_ASPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-XSP Attenuation Volume", + CS42L73_ASPAXSPAA, CS42L73_ASPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-ASP Attenuation Volume", + CS42L73_ASPAASPAA, CS42L73_ASPBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-VSP Attenuation Volume", + CS42L73_ASPAVSPMA, CS42L73_ASPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("VSP-IP Attenuation Volume", + CS42L73_VSPAIPAA, CS42L73_VSPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-XSP Attenuation Volume", + CS42L73_VSPAXSPAA, CS42L73_VSPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-ASP Attenuation Volume", + CS42L73_VSPAASPAA, CS42L73_VSPBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-VSP Attenuation Volume", + CS42L73_VSPAVSPMA, CS42L73_VSPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("HL-IP Attenuation Volume", + CS42L73_HLAIPAA, CS42L73_HLBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-XSP Attenuation Volume", + CS42L73_HLAXSPAA, CS42L73_HLBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-ASP Attenuation Volume", + CS42L73_HLAASPAA, CS42L73_HLBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-VSP Attenuation Volume", + CS42L73_HLAVSPMA, CS42L73_HLBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_SINGLE_TLV("SPK-IP Mono Attenuation Volume", + CS42L73_SPKMIPMA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-XSP Mono Attenuation Volume", + CS42L73_SPKMXSPA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-ASP Mono Attenuation Volume", + CS42L73_SPKMASPA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-VSP Mono Attenuation Volume", + CS42L73_SPKMVSPMA, 0, 0x3F, 1, attn_tlv), + + SOC_SINGLE_TLV("ESL-IP Mono Attenuation Volume", + CS42L73_ESLMIPMA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-XSP Mono Attenuation Volume", + CS42L73_ESLMXSPA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-ASP Mono Attenuation Volume", + CS42L73_ESLMASPA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-VSP Mono Attenuation Volume", + CS42L73_ESLMVSPMA, 0, 0x3F, 1, attn_tlv), + + SOC_ENUM("IP Digital Swap/Mono Select", ip_swap_enum), + + SOC_ENUM("ESL-XSP Mono Mixer Select", esl_xsp_mono_mixer_enum), + SOC_ENUM("ESL-ASP Mono Mixer Select", esl_asp_mono_mixer_enum), + + SOC_ENUM("SPK-ASP Mono Mixer Select", spk_asp_mono_mixer_enum), + SOC_ENUM("SPK-XSP Mono Mixer Select", spk_xsp_mono_mixer_enum), + + SOC_ENUM("VSP Output Mixer Select", vspout_mixer_enum), + SOC_ENUM("XSP Output Mixer Select", xspout_mixer_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_INPUT("DMICA"), + SND_SOC_DAPM_INPUT("DMICB"), + + SND_SOC_DAPM_AIF_OUT("XSPOUT", "XSP Capture", 0, + CS42L73_PWRCTL2, 1, 1), + SND_SOC_DAPM_AIF_OUT("ASPOUT", "ASP Capture", 0, + CS42L73_PWRCTL2, 3, 1), + SND_SOC_DAPM_AIF_OUT("VSPOUT", "VSP Capture", 0, + CS42L73_PWRCTL2, 4, 1), + + SND_SOC_DAPM_AIF_IN("XSPIN", "XSP Playback", 0, + CS42L73_PWRCTL2, 0, 1), + SND_SOC_DAPM_AIF_IN("ASPIN", "ASP Playback", 0, + CS42L73_PWRCTL2, 2, 1), + SND_SOC_DAPM_AIF_IN("VSPIN", "VSP Playback", 0, + CS42L73_PWRCTL2, 4, 1), + + SND_SOC_DAPM_ADC("ADC Left", NULL, CS42L73_PWRCTL1, 5, 1), + SND_SOC_DAPM_ADC("ADC Right", NULL, CS42L73_PWRCTL1, 7, 1), + SND_SOC_DAPM_DAC("DAC Left", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC Right", NULL, SND_SOC_NOPM, 0, 0), + + 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 Mux Left", SND_SOC_NOPM, 0, 0, &pgaa_mux), + SND_SOC_DAPM_MUX("PGA Mux Right", SND_SOC_NOPM, 0, 0, &pgab_mux), + + SND_SOC_DAPM_PGA("HP Amp Left", CS42L73_PWRCTL3, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("HP Amp Right", CS42L73_PWRCTL3, 0, 1, NULL, 0), + + SND_SOC_DAPM_PGA("LO Amp Left", CS42L73_PWRCTL3, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("LO Amp Right", CS42L73_PWRCTL3, 1, 1, NULL, 0), + + SND_SOC_DAPM_PGA("SPK Amp", CS42L73_PWRCTL3, 2, 1, NULL, 0), + + SND_SOC_DAPM_PGA("EAR Amp", CS42L73_PWRCTL3, 3, 1, NULL, 0), + + SND_SOC_DAPM_PGA("SPKLO Amp", CS42L73_PWRCTL3, 4, 1, NULL, 0), + + 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[] = { + {"HPOUTA", NULL, "HP Amp Left"}, + {"HPOUTB", NULL, "HP Amp Right"}, + {"LINEOUTA", NULL, "LO Amp Left"}, + {"LINEOUTB", NULL, "LO Amp Right"}, + {"SPKOUT", NULL, "SPK Amp"}, + {"EAROUT", NULL, "EAR Amp"}, + {"SPKLINEOUT", NULL, "SPKLO Amp"}, + + {"HP Amp Left", "DAC", "DAC Left"}, + {"HP Amp Right", "DAC", "DAC Right"}, + {"LO Amp Left", "DAC", "DAC Left"}, + {"LO Amp Right", "DAC", "DAC Right"}, + {"SPK Amp", "DAC", "DAC Left"}, + {"SPKLO Amp", "DAC", "DAC Right"}, + {"EAR Amp", "DAC", "DAC Right"}, + + {"PGA Mux Left", NULL, "LINEINA"}, + {"PGA Mux Right", NULL, "LINEINB"}, + {"PGA Mux Left", NULL, "MIC1"}, + {"PGA Mux Right", NULL, "MIC2"}, + + {"PGA Left", NULL, "PGA Mux Left"}, + {"PGA Right", NULL, "PGA Mux Right"}, + {"ADC Left", "ADC", "PGA Left"}, + {"ADC Right", "ADC", "PGA Right"}, + + {"XSPOUT", NULL, "ADC Left"}, + {"XSPOUT", NULL, "ADC Right"}, + {"DAC Left", NULL, "XSPIN"}, + {"DAC Right", NULL, "XSPIN"}, + + {"ASPOUT", NULL, "ADC Left"}, + {"ASPOUT", NULL, "ADC Right"}, + {"DAC Left", NULL, "ASPIN"}, + {"DAC Right", NULL, "ASPIN"}, + + {"VSPOUT", NULL, "ADC Left"}, + {"VSPOUT", NULL, "ADC Right"}, + {"DAC Left", NULL, "VSPIN"}, + {"DAC Right", NULL, "VSPIN"}, +}; + +struct cs42l73_mclk_div { + u32 mclk; + u32 srate; + u8 mmcc; +}; + +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; +}; + +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 */ +}; + +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; +} + +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) +{ + 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(priv->sysclk); + + 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->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); + + if (clk_id != CS42L73_CLKID_MCLK1 && clk_id != CS42L73_CLKID_MCLK2) { + dev_err(codec->dev, "Invalid clk_id %u\n", clk_id); + return -EINVAL; + } + + if ((cs42l73_get_mclkx_coeff(freq) < 0)) { + dev_err(codec->dev, "Invalid sysclk %u\n", freq); + return -EINVAL; + } + + if ((cs42l73_set_mclk(dai)) < 0) { + dev_err(codec->dev, "Unable to set MCLK for dai %s\n", + dai->name); + return -EINVAL; + } + + priv->sysclk = freq; + 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); + int id = codec_dai->id; + int 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 &= ~xSPDIF_PCM; + break; + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + if (mmcc & MS_MASTER) { + dev_err(codec->dev, + "PCM format is supported only in slave mode\n"); + return -EINVAL; + } + if (id == CS42L73_ASP) { + dev_err(codec->dev, + "PCM format is not supported on ASP port\n"); + return -EINVAL; + } + spc |= xSPDIF_PCM; + break; + default: + return -EINVAL; + } + + if (spc & xSPDIF_PCM) { + spc &= (31 << 3); /* Clear PCM mode, set MSB->LSB */ + if (format == SND_SOC_DAIFMT_DSP_B + && inv == SND_SOC_DAIFMT_IB_IF) + spc |= (xPCM_MODE0 << 4); + else + + if (format == SND_SOC_DAIFMT_DSP_B + && inv == SND_SOC_DAIFMT_IB_NF) + spc |= (xPCM_MODE1 << 4); + else + + if (format == SND_SOC_DAIFMT_DSP_A + && inv == SND_SOC_DAIFMT_IB_IF) + spc |= (xPCM_MODE1 << 4); + else + 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 |= xMCK_SCLK_64FS; + + } else { + /* CS42L73 Slave */ + dev_dbg(codec->dev, "DAI[%d]: Slave\n", id); + priv->config[id].spc &= 0xFC; + priv->config[id].spc |= xMCK_SCLK_64FS; + } + /* Update ASRCs */ + priv->config[id].srate = srate; + cs42l73_update_asrc(codec, id, srate); + snd_soc_write(codec, CS42L73_SPC(id), priv->config[id].spc); + snd_soc_write(codec, CS42L73_MMCC(id), priv->config[id].mmcc); + 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 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, +}; + +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); + + codec->control_data = cs42l73->control_data; + codec->hw_write = (hw_write_t)i2c_master_send; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, cs42l73->control_type); + 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; + + + snd_soc_add_controls(codec, cs42l73_snd_controls, + ARRAY_SIZE(cs42l73_snd_controls)); + + return ret; +} + +static int cs42l73_remove(struct snd_soc_codec *codec) +{ + cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +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), +}; + +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(struct cs42l73_private), GFP_KERNEL); + if (!cs42l73) { + dev_err(&i2c_client->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + i2c_set_clientdata(i2c_client, cs42l73); + cs42l73->control_data = i2c_client; + cs42l73->control_type = SND_SOC_I2C; + + + 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, office@nucleusys.com"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42l73.h b/sound/soc/codecs/cs42l73.h new file mode 100644 index 0000000..aa09c2e --- /dev/null +++ b/sound/soc/codecs/cs42l73.h @@ -0,0 +1,223 @@ +/* + * ALSA SoC CS42L73 codec driver + * + * Copyright 2011 Cirrus Logic, Inc. + * + * Author: Georgi Vlaev office@nucleusys.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 xSP_3ST (1 << 7) +#define xSPDIF_I2S 0 +#define xSPDIF_PCM (1 << 6) +#define xPCM_MODE0 0 +#define xPCM_MODE1 1 +#define xPCM_MODE2 2 +#define xPCM_BO_MSBLSB 0 +#define xPCM_BO_LSBMSB 1 +#define xMCK_SCLK_64FS 0 +#define xMCK_SCLK_MCLK 2 +#define xMCK_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 Mute */ +#define HPMUTE (1 << 7) +/* LO A/B Mute */ +#define LOMUTE (1 << 7) +/* SPK Digital Mute */ +#define SPKDMUTE (1 << 2) + +/* 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__ */
On Fri, 2011-09-30 at 11:41 -0500, Brian Austin wrote:
This patch adds support for the Cirrus Logic CS42L73 low power stereo codec.
This patch has cleared checkpatch.pl with no warnings or errors. Code changes requested were implemented. ASoC API changes requested were implemented.
Stuff like version changes can be below SoB
Signed-off-by: Brian Austin brian.austin@cirrus.com
sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs42l73.c | 1047 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs42l73.h | 223 ++++++++++ 4 files changed, 1276 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 3449431..9107c48 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 787881b..e89e84b 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..18a49d8 --- /dev/null +++ b/sound/soc/codecs/cs42l73.c @@ -0,0 +1,1047 @@ +/*
- cs42l73.c -- CS42L73 ALSA Soc Audio driver
- Copyright 2011 Cirrus Logic, Inc.
- Authors: Georgi Vlaev, Nucleus Systems Ltd, office@nucleusys.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 <linux/gpio.h>
gpio should be before sound, would make sense to sort this
+#include "cs42l73.h"
no need to empty line before this
+struct sp_config {
- u8 spc, mmcc, spfs;
- u32 srate;
+};
+struct cs42l73_private {
- enum snd_soc_control_type control_type;
- void *control_data;
- 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, +/*10*/ 0x00, 0x15, 0x00, 0x06,
/* abc */ pls
+/*14*/ 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, +/*60*/ 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(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 s42l73_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 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 cs42l73_mono_mixer_text[] = {
- "Left", "Right", "Mono Mix"};
+static const struct soc_enum spk_asp_mono_mixer_enum =
- SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 6,
ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text);+static const struct soc_enum spk_xsp_mono_mixer_enum =
- SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 4,
ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text);+static const struct soc_enum esl_asp_mono_mixer_enum =
- SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 2,
ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text);+static const struct soc_enum esl_xsp_mono_mixer_enum =
- SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 0,
ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text);+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 vspout_mixer_enum =
- SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 5,
ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text);+static const struct soc_enum xspout_mixer_enum =
- SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 4,
ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text);+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_DOUBLE_S8_TLV("Speakerphone Digital Playback Volume",
CS42L73_SPKDVOL, 0x34, 0xE4, hl_tlv),- SOC_DOUBLE_S8_TLV("Ear Speaker Digital Playback Volume",
CS42L73_ESLDVOL, 0x34, 0xE4, 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("Invert ADC Signal Polarity Switch", CS42L73_ADCIPC, 1, 5, 1,
0),- SOC_DOUBLE("ADC Boost Switch", CS42L73_ADCIPC, 2, 6, 1, 0),
- SOC_SINGLE("Charge Pump Frequency Volume", CS42L73_CPFCHC, 4, 15, 0),
- SOC_SINGLE("HL Limiter Attack Rate Volume", CS42L73_LIMARATEHL, 0, 0x3F,
0),- SOC_SINGLE("HL Limiter Release Rate Volume", 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, 1,
limiter_tlv),- SOC_SINGLE_TLV("ALC Min Threshold Volume", CS42L73_ALCMINMAX, 2, 7, 1,
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 selectsbetween 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 Attenuation Volume",
CS42L73_XSPAIPAA, CS42L73_XSPBIPBA, 0, 0x3F, 1,attn_tlv),- SOC_DOUBLE_R_TLV("XSP-XSP Attenuation Volume",
CS42L73_XSPAXSPAA, CS42L73_XSPBXSPBA, 0, 0x3F, 1,attn_tlv),- SOC_DOUBLE_R_TLV("XSP-ASP Attenuation Volume",
CS42L73_XSPAASPAA, CS42L73_XSPAASPBA, 0, 0x3F, 1,attn_tlv),- SOC_DOUBLE_R_TLV("XSP-VSP Attenuation Volume",
CS42L73_XSPAVSPMA, CS42L73_XSPBVSPMA, 0, 0x3F, 1,attn_tlv),- SOC_DOUBLE_R_TLV("ASP-IP Attenuation Volume",
CS42L73_ASPAIPAA, CS42L73_ASPBIPBA, 0, 0x3F, 1,attn_tlv),- SOC_DOUBLE_R_TLV("ASP-XSP Attenuation Volume",
CS42L73_ASPAXSPAA, CS42L73_ASPBXSPBA, 0, 0x3F, 1,attn_tlv),- SOC_DOUBLE_R_TLV("ASP-ASP Attenuation Volume",
CS42L73_ASPAASPAA, CS42L73_ASPBASPBA, 0, 0x3F, 1,attn_tlv),- SOC_DOUBLE_R_TLV("ASP-VSP Attenuation Volume",
CS42L73_ASPAVSPMA, CS42L73_ASPBVSPMA, 0, 0x3F, 1,attn_tlv),- SOC_DOUBLE_R_TLV("VSP-IP Attenuation Volume",
CS42L73_VSPAIPAA, CS42L73_VSPBIPBA, 0, 0x3F, 1,attn_tlv),- SOC_DOUBLE_R_TLV("VSP-XSP Attenuation Volume",
CS42L73_VSPAXSPAA, CS42L73_VSPBXSPBA, 0, 0x3F, 1,attn_tlv),- SOC_DOUBLE_R_TLV("VSP-ASP Attenuation Volume",
CS42L73_VSPAASPAA, CS42L73_VSPBASPBA, 0, 0x3F, 1,attn_tlv),- SOC_DOUBLE_R_TLV("VSP-VSP Attenuation Volume",
CS42L73_VSPAVSPMA, CS42L73_VSPBVSPMA, 0, 0x3F, 1,attn_tlv),- SOC_DOUBLE_R_TLV("HL-IP Attenuation Volume",
CS42L73_HLAIPAA, CS42L73_HLBIPBA, 0, 0x3F, 1,attn_tlv),- SOC_DOUBLE_R_TLV("HL-XSP Attenuation Volume",
CS42L73_HLAXSPAA, CS42L73_HLBXSPBA, 0, 0x3F, 1,attn_tlv),- SOC_DOUBLE_R_TLV("HL-ASP Attenuation Volume",
CS42L73_HLAASPAA, CS42L73_HLBASPBA, 0, 0x3F, 1,attn_tlv),- SOC_DOUBLE_R_TLV("HL-VSP Attenuation Volume",
CS42L73_HLAVSPMA, CS42L73_HLBVSPMA, 0, 0x3F, 1,attn_tlv),- SOC_SINGLE_TLV("SPK-IP Mono Attenuation Volume",
CS42L73_SPKMIPMA, 0, 0x3F, 1, attn_tlv),- SOC_SINGLE_TLV("SPK-XSP Mono Attenuation Volume",
CS42L73_SPKMXSPA, 0, 0x3F, 1, attn_tlv),- SOC_SINGLE_TLV("SPK-ASP Mono Attenuation Volume",
CS42L73_SPKMASPA, 0, 0x3F, 1, attn_tlv),- SOC_SINGLE_TLV("SPK-VSP Mono Attenuation Volume",
CS42L73_SPKMVSPMA, 0, 0x3F, 1, attn_tlv),- SOC_SINGLE_TLV("ESL-IP Mono Attenuation Volume",
CS42L73_ESLMIPMA, 0, 0x3F, 1, attn_tlv),- SOC_SINGLE_TLV("ESL-XSP Mono Attenuation Volume",
CS42L73_ESLMXSPA, 0, 0x3F, 1, attn_tlv),- SOC_SINGLE_TLV("ESL-ASP Mono Attenuation Volume",
CS42L73_ESLMASPA, 0, 0x3F, 1, attn_tlv),- SOC_SINGLE_TLV("ESL-VSP Mono Attenuation Volume",
CS42L73_ESLMVSPMA, 0, 0x3F, 1, attn_tlv),- SOC_ENUM("IP Digital Swap/Mono Select", ip_swap_enum),
- SOC_ENUM("ESL-XSP Mono Mixer Select", esl_xsp_mono_mixer_enum),
- SOC_ENUM("ESL-ASP Mono Mixer Select", esl_asp_mono_mixer_enum),
- SOC_ENUM("SPK-ASP Mono Mixer Select", spk_asp_mono_mixer_enum),
- SOC_ENUM("SPK-XSP Mono Mixer Select", spk_xsp_mono_mixer_enum),
- SOC_ENUM("VSP Output Mixer Select", vspout_mixer_enum),
- SOC_ENUM("XSP Output Mixer Select", xspout_mixer_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_INPUT("DMICA"),
- SND_SOC_DAPM_INPUT("DMICB"),
- SND_SOC_DAPM_AIF_OUT("XSPOUT", "XSP Capture", 0,
CS42L73_PWRCTL2, 1, 1),- SND_SOC_DAPM_AIF_OUT("ASPOUT", "ASP Capture", 0,
CS42L73_PWRCTL2, 3, 1),- SND_SOC_DAPM_AIF_OUT("VSPOUT", "VSP Capture", 0,
CS42L73_PWRCTL2, 4, 1),- SND_SOC_DAPM_AIF_IN("XSPIN", "XSP Playback", 0,
CS42L73_PWRCTL2, 0, 1),- SND_SOC_DAPM_AIF_IN("ASPIN", "ASP Playback", 0,
CS42L73_PWRCTL2, 2, 1),- SND_SOC_DAPM_AIF_IN("VSPIN", "VSP Playback", 0,
CS42L73_PWRCTL2, 4, 1),- SND_SOC_DAPM_ADC("ADC Left", NULL, CS42L73_PWRCTL1, 5, 1),
- SND_SOC_DAPM_ADC("ADC Right", NULL, CS42L73_PWRCTL1, 7, 1),
- SND_SOC_DAPM_DAC("DAC Left", NULL, SND_SOC_NOPM, 0, 0),
- SND_SOC_DAPM_DAC("DAC Right", NULL, SND_SOC_NOPM, 0, 0),
- 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 Mux Left", SND_SOC_NOPM, 0, 0, &pgaa_mux),
- SND_SOC_DAPM_MUX("PGA Mux Right", SND_SOC_NOPM, 0, 0, &pgab_mux),
- SND_SOC_DAPM_PGA("HP Amp Left", CS42L73_PWRCTL3, 0, 1, NULL, 0),
- SND_SOC_DAPM_PGA("HP Amp Right", CS42L73_PWRCTL3, 0, 1, NULL, 0),
- SND_SOC_DAPM_PGA("LO Amp Left", CS42L73_PWRCTL3, 1, 1, NULL, 0),
- SND_SOC_DAPM_PGA("LO Amp Right", CS42L73_PWRCTL3, 1, 1, NULL, 0),
- SND_SOC_DAPM_PGA("SPK Amp", CS42L73_PWRCTL3, 2, 1, NULL, 0),
- SND_SOC_DAPM_PGA("EAR Amp", CS42L73_PWRCTL3, 3, 1, NULL, 0),
- SND_SOC_DAPM_PGA("SPKLO Amp", CS42L73_PWRCTL3, 4, 1, NULL, 0),
- 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[] = {
- {"HPOUTA", NULL, "HP Amp Left"},
- {"HPOUTB", NULL, "HP Amp Right"},
- {"LINEOUTA", NULL, "LO Amp Left"},
- {"LINEOUTB", NULL, "LO Amp Right"},
- {"SPKOUT", NULL, "SPK Amp"},
- {"EAROUT", NULL, "EAR Amp"},
- {"SPKLINEOUT", NULL, "SPKLO Amp"},
- {"HP Amp Left", "DAC", "DAC Left"},
- {"HP Amp Right", "DAC", "DAC Right"},
- {"LO Amp Left", "DAC", "DAC Left"},
- {"LO Amp Right", "DAC", "DAC Right"},
- {"SPK Amp", "DAC", "DAC Left"},
- {"SPKLO Amp", "DAC", "DAC Right"},
- {"EAR Amp", "DAC", "DAC Right"},
This is not right, all amplifiers are connecting to DACs, whereas per spec you have Headset and speaker DACs and a mux before each DAC. Please change this, otherwise this code will result in turning of all Amps when DAC is active which is not right...
- {"PGA Mux Left", NULL, "LINEINA"},
- {"PGA Mux Right", NULL, "LINEINB"},
- {"PGA Mux Left", NULL, "MIC1"},
- {"PGA Mux Right", NULL, "MIC2"},
- {"PGA Left", NULL, "PGA Mux Left"},
- {"PGA Right", NULL, "PGA Mux Right"},
- {"ADC Left", "ADC", "PGA Left"},
- {"ADC Right", "ADC", "PGA Right"},
- {"XSPOUT", NULL, "ADC Left"},
- {"XSPOUT", NULL, "ADC Right"},
- {"DAC Left", NULL, "XSPIN"},
- {"DAC Right", NULL, "XSPIN"},
- {"ASPOUT", NULL, "ADC Left"},
- {"ASPOUT", NULL, "ADC Right"},
- {"DAC Left", NULL, "ASPIN"},
- {"DAC Right", NULL, "ASPIN"},
- {"VSPOUT", NULL, "ADC Left"},
- {"VSPOUT", NULL, "ADC Right"},
- {"DAC Left", NULL, "VSPIN"},
- {"DAC Right", NULL, "VSPIN"},
+};
+struct cs42l73_mclk_div {
- u32 mclk;
- u32 srate;
- u8 mmcc;
+};
+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;
+};
+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 */
+};
+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;
+}
+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) +{
- 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(priv->sysclk);
- 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->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);
- if (clk_id != CS42L73_CLKID_MCLK1 && clk_id != CS42L73_CLKID_MCLK2) {
dev_err(codec->dev, "Invalid clk_id %u\n", clk_id);return -EINVAL;- }
- if ((cs42l73_get_mclkx_coeff(freq) < 0)) {
dev_err(codec->dev, "Invalid sysclk %u\n", freq);return -EINVAL;- }
- if ((cs42l73_set_mclk(dai)) < 0) {
dev_err(codec->dev, "Unable to set MCLK for dai %s\n",dai->name);return -EINVAL;- }
- priv->sysclk = freq;
- 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);
- int id = codec_dai->id;
- int 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 &= ~xSPDIF_PCM;break;- case SND_SOC_DAIFMT_DSP_A:
- case SND_SOC_DAIFMT_DSP_B:
if (mmcc & MS_MASTER) {dev_err(codec->dev,"PCM format is supported only in slave mode\n");return -EINVAL;}if (id == CS42L73_ASP) {dev_err(codec->dev,"PCM format is not supported on ASP port\n");return -EINVAL;}spc |= xSPDIF_PCM;break;- default:
return -EINVAL;- }
- if (spc & xSPDIF_PCM) {
spc &= (31 << 3); /* Clear PCM mode, set MSB->LSB */if (format == SND_SOC_DAIFMT_DSP_B&& inv == SND_SOC_DAIFMT_IB_IF)spc |= (xPCM_MODE0 << 4);elseif (format == SND_SOC_DAIFMT_DSP_B&& inv == SND_SOC_DAIFMT_IB_NF)spc |= (xPCM_MODE1 << 4);elseif (format == SND_SOC_DAIFMT_DSP_A&& inv == SND_SOC_DAIFMT_IB_IF)spc |= (xPCM_MODE1 << 4);elsereturn -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 |= xMCK_SCLK_64FS;- } else {
/* CS42L73 Slave */dev_dbg(codec->dev, "DAI[%d]: Slave\n", id);priv->config[id].spc &= 0xFC;priv->config[id].spc |= xMCK_SCLK_64FS;- }
- /* Update ASRCs */
- priv->config[id].srate = srate;
- cs42l73_update_asrc(codec, id, srate);
- snd_soc_write(codec, CS42L73_SPC(id), priv->config[id].spc);
- snd_soc_write(codec, CS42L73_MMCC(id), priv->config[id].mmcc);
- 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);
indent? surely checkpatch should have detected this
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 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,
+};
+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);
- codec->control_data = cs42l73->control_data;
- codec->hw_write = (hw_write_t)i2c_master_send;
- ret = snd_soc_codec_set_cache_io(codec, 8, 8, cs42l73->control_type);
- 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;
- snd_soc_add_controls(codec, cs42l73_snd_controls,
ARRAY_SIZE(cs42l73_snd_controls));- return ret;
+}
+static int cs42l73_remove(struct snd_soc_codec *codec) +{
- cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
+}
+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),
+};
+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(struct cs42l73_private), GFP_KERNEL);
sizeof(*cs42l73)
- if (!cs42l73) {
dev_err(&i2c_client->dev, "could not allocate codec\n");return -ENOMEM;- }
- i2c_set_clientdata(i2c_client, cs42l73);
- cs42l73->control_data = i2c_client;
- cs42l73->control_type = SND_SOC_I2C;
- 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, office@nucleusys.com"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42l73.h b/sound/soc/codecs/cs42l73.h new file mode 100644 index 0000000..aa09c2e --- /dev/null +++ b/sound/soc/codecs/cs42l73.h @@ -0,0 +1,223 @@ +/*
- ALSA SoC CS42L73 codec driver
- Copyright 2011 Cirrus Logic, Inc.
- Author: Georgi Vlaev office@nucleusys.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 xSP_3ST (1 << 7)
??
+#define xSPDIF_I2S 0 +#define xSPDIF_PCM (1 << 6) +#define xPCM_MODE0 0 +#define xPCM_MODE1 1 +#define xPCM_MODE2 2 +#define xPCM_BO_MSBLSB 0 +#define xPCM_BO_LSBMSB 1 +#define xMCK_SCLK_64FS 0 +#define xMCK_SCLK_MCLK 2 +#define xMCK_SCLK_PREMCLK 3
+/* CS42L73_xSPMMCC */ +#define MS_MASTER (1 << 7)
All these should be properly namespaced..
+/* 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 Mute */ +#define HPMUTE (1 << 7) +/* LO A/B Mute */ +#define LOMUTE (1 << 7) +/* SPK Digital Mute */ +#define SPKDMUTE (1 << 2)
+/* 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__ */
now code shows, you dont use gpio, so why add the header?
On Sep 30, 2011, at 12:34 PM, Vinod Koul wrote:
On Fri, 2011-09-30 at 11:41 -0500, Brian Austin wrote:
This patch adds support for the Cirrus Logic CS42L73 low power stereo codec.
This patch has cleared checkpatch.pl with no warnings or errors. Code changes requested were implemented. ASoC API changes requested were implemented.
Stuff like version changes can be below SoB
Signed-off-by: Brian Austin brian.austin@cirrus.com
sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs42l73.c | 1047 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs42l73.h | 223 ++++++++++ 4 files changed, 1276 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 3449431..9107c48 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 787881b..e89e84b 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..18a49d8 --- /dev/null +++ b/sound/soc/codecs/cs42l73.c @@ -0,0 +1,1047 @@ +/*
- cs42l73.c -- CS42L73 ALSA Soc Audio driver
- Copyright 2011 Cirrus Logic, Inc.
- Authors: Georgi Vlaev, Nucleus Systems Ltd, office@nucleusys.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 <linux/gpio.h>
gpio should be before sound, would make sense to sort this
This I can remove
+#include "cs42l73.h"
no need to empty line before this
+struct sp_config {
- u8 spc, mmcc, spfs;
- u32 srate;
+};
+struct cs42l73_private {
- enum snd_soc_control_type control_type;
- void *control_data;
- 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, +/*10*/ 0x00, 0x15, 0x00, 0x06,
/* abc */ pls
Thanks
+/*14*/ 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, +/*60*/ 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(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 s42l73_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 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 cs42l73_mono_mixer_text[] = {
- "Left", "Right", "Mono Mix"};
+static const struct soc_enum spk_asp_mono_mixer_enum =
- SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 6,
ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text);+static const struct soc_enum spk_xsp_mono_mixer_enum =
- SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 4,
ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text);+static const struct soc_enum esl_asp_mono_mixer_enum =
- SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 2,
ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text);+static const struct soc_enum esl_xsp_mono_mixer_enum =
- SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 0,
ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text);+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 vspout_mixer_enum =
- SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 5,
ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text);+static const struct soc_enum xspout_mixer_enum =
- SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 4,
ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text);+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_DOUBLE_S8_TLV("Speakerphone Digital Playback Volume",
CS42L73_SPKDVOL, 0x34, 0xE4, hl_tlv),- SOC_DOUBLE_S8_TLV("Ear Speaker Digital Playback Volume",
CS42L73_ESLDVOL, 0x34, 0xE4, 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("Invert ADC Signal Polarity Switch", CS42L73_ADCIPC, 1, 5, 1,
0),- SOC_DOUBLE("ADC Boost Switch", CS42L73_ADCIPC, 2, 6, 1, 0),
- SOC_SINGLE("Charge Pump Frequency Volume", CS42L73_CPFCHC, 4, 15, 0),
- SOC_SINGLE("HL Limiter Attack Rate Volume", CS42L73_LIMARATEHL, 0, 0x3F,
0),- SOC_SINGLE("HL Limiter Release Rate Volume", 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, 1,
limiter_tlv),- SOC_SINGLE_TLV("ALC Min Threshold Volume", CS42L73_ALCMINMAX, 2, 7, 1,
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 selectsbetween 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 Attenuation Volume",
CS42L73_XSPAIPAA, CS42L73_XSPBIPBA, 0, 0x3F, 1,attn_tlv),- SOC_DOUBLE_R_TLV("XSP-XSP Attenuation Volume",
CS42L73_XSPAXSPAA, CS42L73_XSPBXSPBA, 0, 0x3F, 1,attn_tlv),- SOC_DOUBLE_R_TLV("XSP-ASP Attenuation Volume",
CS42L73_XSPAASPAA, CS42L73_XSPAASPBA, 0, 0x3F, 1,attn_tlv),- SOC_DOUBLE_R_TLV("XSP-VSP Attenuation Volume",
CS42L73_XSPAVSPMA, CS42L73_XSPBVSPMA, 0, 0x3F, 1,attn_tlv),- SOC_DOUBLE_R_TLV("ASP-IP Attenuation Volume",
CS42L73_ASPAIPAA, CS42L73_ASPBIPBA, 0, 0x3F, 1,attn_tlv),- SOC_DOUBLE_R_TLV("ASP-XSP Attenuation Volume",
CS42L73_ASPAXSPAA, CS42L73_ASPBXSPBA, 0, 0x3F, 1,attn_tlv),- SOC_DOUBLE_R_TLV("ASP-ASP Attenuation Volume",
CS42L73_ASPAASPAA, CS42L73_ASPBASPBA, 0, 0x3F, 1,attn_tlv),- SOC_DOUBLE_R_TLV("ASP-VSP Attenuation Volume",
CS42L73_ASPAVSPMA, CS42L73_ASPBVSPMA, 0, 0x3F, 1,attn_tlv),- SOC_DOUBLE_R_TLV("VSP-IP Attenuation Volume",
CS42L73_VSPAIPAA, CS42L73_VSPBIPBA, 0, 0x3F, 1,attn_tlv),- SOC_DOUBLE_R_TLV("VSP-XSP Attenuation Volume",
CS42L73_VSPAXSPAA, CS42L73_VSPBXSPBA, 0, 0x3F, 1,attn_tlv),- SOC_DOUBLE_R_TLV("VSP-ASP Attenuation Volume",
CS42L73_VSPAASPAA, CS42L73_VSPBASPBA, 0, 0x3F, 1,attn_tlv),- SOC_DOUBLE_R_TLV("VSP-VSP Attenuation Volume",
CS42L73_VSPAVSPMA, CS42L73_VSPBVSPMA, 0, 0x3F, 1,attn_tlv),- SOC_DOUBLE_R_TLV("HL-IP Attenuation Volume",
CS42L73_HLAIPAA, CS42L73_HLBIPBA, 0, 0x3F, 1,attn_tlv),- SOC_DOUBLE_R_TLV("HL-XSP Attenuation Volume",
CS42L73_HLAXSPAA, CS42L73_HLBXSPBA, 0, 0x3F, 1,attn_tlv),- SOC_DOUBLE_R_TLV("HL-ASP Attenuation Volume",
CS42L73_HLAASPAA, CS42L73_HLBASPBA, 0, 0x3F, 1,attn_tlv),- SOC_DOUBLE_R_TLV("HL-VSP Attenuation Volume",
CS42L73_HLAVSPMA, CS42L73_HLBVSPMA, 0, 0x3F, 1,attn_tlv),- SOC_SINGLE_TLV("SPK-IP Mono Attenuation Volume",
CS42L73_SPKMIPMA, 0, 0x3F, 1, attn_tlv),- SOC_SINGLE_TLV("SPK-XSP Mono Attenuation Volume",
CS42L73_SPKMXSPA, 0, 0x3F, 1, attn_tlv),- SOC_SINGLE_TLV("SPK-ASP Mono Attenuation Volume",
CS42L73_SPKMASPA, 0, 0x3F, 1, attn_tlv),- SOC_SINGLE_TLV("SPK-VSP Mono Attenuation Volume",
CS42L73_SPKMVSPMA, 0, 0x3F, 1, attn_tlv),- SOC_SINGLE_TLV("ESL-IP Mono Attenuation Volume",
CS42L73_ESLMIPMA, 0, 0x3F, 1, attn_tlv),- SOC_SINGLE_TLV("ESL-XSP Mono Attenuation Volume",
CS42L73_ESLMXSPA, 0, 0x3F, 1, attn_tlv),- SOC_SINGLE_TLV("ESL-ASP Mono Attenuation Volume",
CS42L73_ESLMASPA, 0, 0x3F, 1, attn_tlv),- SOC_SINGLE_TLV("ESL-VSP Mono Attenuation Volume",
CS42L73_ESLMVSPMA, 0, 0x3F, 1, attn_tlv),- SOC_ENUM("IP Digital Swap/Mono Select", ip_swap_enum),
- SOC_ENUM("ESL-XSP Mono Mixer Select", esl_xsp_mono_mixer_enum),
- SOC_ENUM("ESL-ASP Mono Mixer Select", esl_asp_mono_mixer_enum),
- SOC_ENUM("SPK-ASP Mono Mixer Select", spk_asp_mono_mixer_enum),
- SOC_ENUM("SPK-XSP Mono Mixer Select", spk_xsp_mono_mixer_enum),
- SOC_ENUM("VSP Output Mixer Select", vspout_mixer_enum),
- SOC_ENUM("XSP Output Mixer Select", xspout_mixer_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_INPUT("DMICA"),
- SND_SOC_DAPM_INPUT("DMICB"),
- SND_SOC_DAPM_AIF_OUT("XSPOUT", "XSP Capture", 0,
CS42L73_PWRCTL2, 1, 1),- SND_SOC_DAPM_AIF_OUT("ASPOUT", "ASP Capture", 0,
CS42L73_PWRCTL2, 3, 1),- SND_SOC_DAPM_AIF_OUT("VSPOUT", "VSP Capture", 0,
CS42L73_PWRCTL2, 4, 1),- SND_SOC_DAPM_AIF_IN("XSPIN", "XSP Playback", 0,
CS42L73_PWRCTL2, 0, 1),- SND_SOC_DAPM_AIF_IN("ASPIN", "ASP Playback", 0,
CS42L73_PWRCTL2, 2, 1),- SND_SOC_DAPM_AIF_IN("VSPIN", "VSP Playback", 0,
CS42L73_PWRCTL2, 4, 1),- SND_SOC_DAPM_ADC("ADC Left", NULL, CS42L73_PWRCTL1, 5, 1),
- SND_SOC_DAPM_ADC("ADC Right", NULL, CS42L73_PWRCTL1, 7, 1),
- SND_SOC_DAPM_DAC("DAC Left", NULL, SND_SOC_NOPM, 0, 0),
- SND_SOC_DAPM_DAC("DAC Right", NULL, SND_SOC_NOPM, 0, 0),
- 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 Mux Left", SND_SOC_NOPM, 0, 0, &pgaa_mux),
- SND_SOC_DAPM_MUX("PGA Mux Right", SND_SOC_NOPM, 0, 0, &pgab_mux),
- SND_SOC_DAPM_PGA("HP Amp Left", CS42L73_PWRCTL3, 0, 1, NULL, 0),
- SND_SOC_DAPM_PGA("HP Amp Right", CS42L73_PWRCTL3, 0, 1, NULL, 0),
- SND_SOC_DAPM_PGA("LO Amp Left", CS42L73_PWRCTL3, 1, 1, NULL, 0),
- SND_SOC_DAPM_PGA("LO Amp Right", CS42L73_PWRCTL3, 1, 1, NULL, 0),
- SND_SOC_DAPM_PGA("SPK Amp", CS42L73_PWRCTL3, 2, 1, NULL, 0),
- SND_SOC_DAPM_PGA("EAR Amp", CS42L73_PWRCTL3, 3, 1, NULL, 0),
- SND_SOC_DAPM_PGA("SPKLO Amp", CS42L73_PWRCTL3, 4, 1, NULL, 0),
- 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[] = {
- {"HPOUTA", NULL, "HP Amp Left"},
- {"HPOUTB", NULL, "HP Amp Right"},
- {"LINEOUTA", NULL, "LO Amp Left"},
- {"LINEOUTB", NULL, "LO Amp Right"},
- {"SPKOUT", NULL, "SPK Amp"},
- {"EAROUT", NULL, "EAR Amp"},
- {"SPKLINEOUT", NULL, "SPKLO Amp"},
- {"HP Amp Left", "DAC", "DAC Left"},
- {"HP Amp Right", "DAC", "DAC Right"},
- {"LO Amp Left", "DAC", "DAC Left"},
- {"LO Amp Right", "DAC", "DAC Right"},
- {"SPK Amp", "DAC", "DAC Left"},
- {"SPKLO Amp", "DAC", "DAC Right"},
- {"EAR Amp", "DAC", "DAC Right"},
This is not right, all amplifiers are connecting to DACs, whereas per spec you have Headset and speaker DACs and a mux before each DAC. Please change this, otherwise this code will result in turning of all Amps when DAC is active which is not right…
Currently, I enable/disable the output pins in the machine driver. As we have discussed before, I am redoing this part of the codec driver to not rely on a machine driver to set the correct path. I was hoping to get the driver accepted and finalize the DAPM stuff later after more testing. If this is a blocker, I can resubmit when this is done.
- {"PGA Mux Left", NULL, "LINEINA"},
- {"PGA Mux Right", NULL, "LINEINB"},
- {"PGA Mux Left", NULL, "MIC1"},
- {"PGA Mux Right", NULL, "MIC2"},
- {"PGA Left", NULL, "PGA Mux Left"},
- {"PGA Right", NULL, "PGA Mux Right"},
- {"ADC Left", "ADC", "PGA Left"},
- {"ADC Right", "ADC", "PGA Right"},
- {"XSPOUT", NULL, "ADC Left"},
- {"XSPOUT", NULL, "ADC Right"},
- {"DAC Left", NULL, "XSPIN"},
- {"DAC Right", NULL, "XSPIN"},
- {"ASPOUT", NULL, "ADC Left"},
- {"ASPOUT", NULL, "ADC Right"},
- {"DAC Left", NULL, "ASPIN"},
- {"DAC Right", NULL, "ASPIN"},
- {"VSPOUT", NULL, "ADC Left"},
- {"VSPOUT", NULL, "ADC Right"},
- {"DAC Left", NULL, "VSPIN"},
- {"DAC Right", NULL, "VSPIN"},
+};
+struct cs42l73_mclk_div {
- u32 mclk;
- u32 srate;
- u8 mmcc;
+};
+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;
+};
+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 */
+};
+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;
+}
+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) +{
- 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(priv->sysclk);
- 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->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);
- if (clk_id != CS42L73_CLKID_MCLK1 && clk_id != CS42L73_CLKID_MCLK2) {
dev_err(codec->dev, "Invalid clk_id %u\n", clk_id);return -EINVAL;- }
- if ((cs42l73_get_mclkx_coeff(freq) < 0)) {
dev_err(codec->dev, "Invalid sysclk %u\n", freq);return -EINVAL;- }
- if ((cs42l73_set_mclk(dai)) < 0) {
dev_err(codec->dev, "Unable to set MCLK for dai %s\n",dai->name);return -EINVAL;- }
- priv->sysclk = freq;
- 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);
- int id = codec_dai->id;
- int 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 &= ~xSPDIF_PCM;break;- case SND_SOC_DAIFMT_DSP_A:
- case SND_SOC_DAIFMT_DSP_B:
if (mmcc & MS_MASTER) {dev_err(codec->dev,"PCM format is supported only in slave mode\n");return -EINVAL;}if (id == CS42L73_ASP) {dev_err(codec->dev,"PCM format is not supported on ASP port\n");return -EINVAL;}spc |= xSPDIF_PCM;break;- default:
return -EINVAL;- }
- if (spc & xSPDIF_PCM) {
spc &= (31 << 3); /* Clear PCM mode, set MSB->LSB */if (format == SND_SOC_DAIFMT_DSP_B&& inv == SND_SOC_DAIFMT_IB_IF)spc |= (xPCM_MODE0 << 4);elseif (format == SND_SOC_DAIFMT_DSP_B&& inv == SND_SOC_DAIFMT_IB_NF)spc |= (xPCM_MODE1 << 4);elseif (format == SND_SOC_DAIFMT_DSP_A&& inv == SND_SOC_DAIFMT_IB_IF)spc |= (xPCM_MODE1 << 4);elsereturn -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 |= xMCK_SCLK_64FS;- } else {
/* CS42L73 Slave */dev_dbg(codec->dev, "DAI[%d]: Slave\n", id);priv->config[id].spc &= 0xFC;priv->config[id].spc |= xMCK_SCLK_64FS;- }
- /* Update ASRCs */
- priv->config[id].srate = srate;
- cs42l73_update_asrc(codec, id, srate);
- snd_soc_write(codec, CS42L73_SPC(id), priv->config[id].spc);
- snd_soc_write(codec, CS42L73_MMCC(id), priv->config[id].mmcc);
- 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);indent? surely checkpatch should have detected this
It didn't catch this, but I can add one…
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 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,
+};
+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);
- codec->control_data = cs42l73->control_data;
- codec->hw_write = (hw_write_t)i2c_master_send;
- ret = snd_soc_codec_set_cache_io(codec, 8, 8, cs42l73->control_type);
- 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;
- snd_soc_add_controls(codec, cs42l73_snd_controls,
ARRAY_SIZE(cs42l73_snd_controls));- return ret;
+}
+static int cs42l73_remove(struct snd_soc_codec *codec) +{
- cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
+}
+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),
+};
+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(struct cs42l73_private), GFP_KERNEL);
sizeof(*cs42l73)
- if (!cs42l73) {
dev_err(&i2c_client->dev, "could not allocate codec\n");return -ENOMEM;- }
- i2c_set_clientdata(i2c_client, cs42l73);
- cs42l73->control_data = i2c_client;
- cs42l73->control_type = SND_SOC_I2C;
- 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, office@nucleusys.com"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42l73.h b/sound/soc/codecs/cs42l73.h new file mode 100644 index 0000000..aa09c2e --- /dev/null +++ b/sound/soc/codecs/cs42l73.h @@ -0,0 +1,223 @@ +/*
- ALSA SoC CS42L73 codec driver
- Copyright 2011 Cirrus Logic, Inc.
- Author: Georgi Vlaev office@nucleusys.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 xSP_3ST (1 << 7)
??
Tri-state Auxillary Serial Port Interface (XSP Port)
+#define xSPDIF_I2S 0 +#define xSPDIF_PCM (1 << 6) +#define xPCM_MODE0 0 +#define xPCM_MODE1 1 +#define xPCM_MODE2 2 +#define xPCM_BO_MSBLSB 0 +#define xPCM_BO_LSBMSB 1 +#define xMCK_SCLK_64FS 0 +#define xMCK_SCLK_MCLK 2 +#define xMCK_SCLK_PREMCLK 3
+/* CS42L73_xSPMMCC */ +#define MS_MASTER (1 << 7)
All these should be properly namespaced..
Thanks I missed that
+/* 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 Mute */ +#define HPMUTE (1 << 7) +/* LO A/B Mute */ +#define LOMUTE (1 << 7) +/* SPK Digital Mute */ +#define SPKDMUTE (1 << 2)
+/* 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__ */
now code shows, you dont use gpio, so why add the header?
On Fri, 2011-09-30 at 18:32 +0000, Austin, Brian wrote:
+/* 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 xSP_3ST (1 << 7)
??
Tri-state Auxillary Serial Port Interface (XSP Port)
Your formating is inconsistent here :)
On 09/30/2011 06:41 PM, Brian Austin wrote:
This patch adds support for the Cirrus Logic CS42L73 low power stereo codec.
This patch has cleared checkpatch.pl with no warnings or errors. Code changes requested were implemented. ASoC API changes requested were implemented.
Signed-off-by: Brian Austin brian.austin@cirrus.com
[...] +struct cs42l73_private {
extra space
[...]
+static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
- {"HPOUTA", NULL, "HP Amp Left"},
- {"HPOUTB", NULL, "HP Amp Right"},
- {"LINEOUTA", NULL, "LO Amp Left"},
- {"LINEOUTB", NULL, "LO Amp Right"},
- {"SPKOUT", NULL, "SPK Amp"},
- {"EAROUT", NULL, "EAR Amp"},
- {"SPKLINEOUT", NULL, "SPKLO Amp"},
- {"HP Amp Left", "DAC", "DAC Left"},
- {"HP Amp Right", "DAC", "DAC Right"},
- {"LO Amp Left", "DAC", "DAC Left"},
- {"LO Amp Right", "DAC", "DAC Right"},
- {"SPK Amp", "DAC", "DAC Left"},
- {"SPKLO Amp", "DAC", "DAC Right"},
- {"EAR Amp", "DAC", "DAC Right"},
I suppose this won't work as expected. The control element of the path will only have an effect if the sink is a mixer or a mux.
- {"PGA Mux Left", NULL, "LINEINA"},
- {"PGA Mux Right", NULL, "LINEINB"},
- {"PGA Mux Left", NULL, "MIC1"},
- {"PGA Mux Right", NULL, "MIC2"},
You probably want connect the inputs of the mux depending on its setting.
e.g. {"PGA Mux Left", "Line A", "LINEINA"},
[...]
+struct cs42l73_mclk_div cs42l73_mclk_coeffs[] = {
static
[...]
+struct cs42l73_mclkx_div cs42l73_mclkx_coeffs[] = {
static
[...] +int cs42l73_get_mclkx_coeff(int mclkx)
static
[...] +int cs42l73_get_mclk_coeff(int mclk, int srate)
static
+static int cs42l73_set_mclk(struct snd_soc_dai *dai) +{
- 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(priv->sysclk);
here you use priv->sysclk ...
+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);
- if (clk_id != CS42L73_CLKID_MCLK1 && clk_id != CS42L73_CLKID_MCLK2) {
dev_err(codec->dev, "Invalid clk_id %u\n", clk_id);return -EINVAL;- }
- if ((cs42l73_get_mclkx_coeff(freq) < 0)) {
dev_err(codec->dev, "Invalid sysclk %u\n", freq);return -EINVAL;- }
- if ((cs42l73_set_mclk(dai)) < 0) {
dev_err(codec->dev, "Unable to set MCLK for dai %s\n",dai->name);return -EINVAL;- }
- priv->sysclk = freq;
- priv->mclksel = clk_id;
... but here you only set it after having called cs42l73_set_mclk. Is this correct? Maybe pass it in as a parameter to cs42l73_set_mclk
- 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);
- int id = codec_dai->id;
- int inv, format;
unsigned int
- u8 spc, mmcc;
[...]
+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);
indention
[...]
+static struct snd_soc_dai_ops cs42l73_ops = {
const
- .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,
+};
+struct snd_soc_dai_driver cs42l73_dai[] = {
static
- {
.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,},
Indention looks a bit odd here
[...] +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);
- codec->control_data = cs42l73->control_data;
- codec->hw_write = (hw_write_t)i2c_master_send;
You don't need to initialize control_data and hw_write. snd_soc_codec_set_cache_io will do this for you. This also means that you can remove the control_data field from your driver private data struct.
- ret = snd_soc_codec_set_cache_io(codec, 8, 8, cs42l73->control_type);
If you CODEC only has a I2C interface you can use SND_SOC_I2C here directly instead of cs42l73->control_type
[...]
- snd_soc_add_controls(codec, cs42l73_snd_controls,
ARRAY_SIZE(cs42l73_snd_controls));
Just put the controls in your codec driver struct instead of manually adding them.
- return ret;
+}
+static int cs42l73_remove(struct snd_soc_codec *codec) +{
- cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
+}
+struct snd_soc_codec_driver soc_codec_dev_cs42l73 = {
static
- .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),
+};
On Sep 30, 2011, at 1:50 PM, Lars-Peter Clausen wrote:
On 09/30/2011 06:41 PM, Brian Austin wrote:
This patch adds support for the Cirrus Logic CS42L73 low power stereo codec.
This patch has cleared checkpatch.pl with no warnings or errors. Code changes requested were implemented. ASoC API changes requested were implemented.
Signed-off-by: Brian Austin brian.austin@cirrus.com
[...] +struct cs42l73_private {
extra space
[...]
+static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
- {"HPOUTA", NULL, "HP Amp Left"},
- {"HPOUTB", NULL, "HP Amp Right"},
- {"LINEOUTA", NULL, "LO Amp Left"},
- {"LINEOUTB", NULL, "LO Amp Right"},
- {"SPKOUT", NULL, "SPK Amp"},
- {"EAROUT", NULL, "EAR Amp"},
- {"SPKLINEOUT", NULL, "SPKLO Amp"},
- {"HP Amp Left", "DAC", "DAC Left"},
- {"HP Amp Right", "DAC", "DAC Right"},
- {"LO Amp Left", "DAC", "DAC Left"},
- {"LO Amp Right", "DAC", "DAC Right"},
- {"SPK Amp", "DAC", "DAC Left"},
- {"SPKLO Amp", "DAC", "DAC Right"},
- {"EAR Amp", "DAC", "DAC Right"},
I suppose this won't work as expected. The control element of the path will only have an effect if the sink is a mixer or a mux.
- {"PGA Mux Left", NULL, "LINEINA"},
- {"PGA Mux Right", NULL, "LINEINB"},
- {"PGA Mux Left", NULL, "MIC1"},
- {"PGA Mux Right", NULL, "MIC2"},
You probably want connect the inputs of the mux depending on its setting.
e.g. {"PGA Mux Left", "Line A", "LINEINA"},
[...]
+struct cs42l73_mclk_div cs42l73_mclk_coeffs[] = {
static
[...]
+struct cs42l73_mclkx_div cs42l73_mclkx_coeffs[] = {
static
[...] +int cs42l73_get_mclkx_coeff(int mclkx)
static
[...] +int cs42l73_get_mclk_coeff(int mclk, int srate)
static
+static int cs42l73_set_mclk(struct snd_soc_dai *dai) +{
- 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(priv->sysclk);
here you use priv->sysclk ...
+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);
- if (clk_id != CS42L73_CLKID_MCLK1 && clk_id != CS42L73_CLKID_MCLK2) {
dev_err(codec->dev, "Invalid clk_id %u\n", clk_id);return -EINVAL;- }
- if ((cs42l73_get_mclkx_coeff(freq) < 0)) {
dev_err(codec->dev, "Invalid sysclk %u\n", freq);return -EINVAL;- }
- if ((cs42l73_set_mclk(dai)) < 0) {
dev_err(codec->dev, "Unable to set MCLK for dai %s\n",dai->name);return -EINVAL;- }
- priv->sysclk = freq;
- priv->mclksel = clk_id;
... but here you only set it after having called cs42l73_set_mclk. Is this correct? Maybe pass it in as a parameter to cs42l73_set_mclk
- 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);
- int id = codec_dai->id;
- int inv, format;
unsigned int
- u8 spc, mmcc;
[...]
+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);indention
[...]
+static struct snd_soc_dai_ops cs42l73_ops = {
const
- .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,
+};
+struct snd_soc_dai_driver cs42l73_dai[] = {
static
- {
.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,},Indention looks a bit odd here
[...] +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);
- codec->control_data = cs42l73->control_data;
- codec->hw_write = (hw_write_t)i2c_master_send;
You don't need to initialize control_data and hw_write. snd_soc_codec_set_cache_io will do this for you. This also means that you can remove the control_data field from your driver private data struct.
- ret = snd_soc_codec_set_cache_io(codec, 8, 8, cs42l73->control_type);
If you CODEC only has a I2C interface you can use SND_SOC_I2C here directly instead of cs42l73->control_type
[...]
- snd_soc_add_controls(codec, cs42l73_snd_controls,
ARRAY_SIZE(cs42l73_snd_controls));Just put the controls in your codec driver struct instead of manually adding them.
- return ret;
+}
+static int cs42l73_remove(struct snd_soc_codec *codec) +{
- cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
+}
+struct snd_soc_codec_driver soc_codec_dev_cs42l73 = {
static
- .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),
+};
Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
Thanks for the feedback. This is good. The DAPM Routing works with a machine driver, but i will have to make it work inside the codec I guess.
Thanks, Brian
-----Original Message----- From: alsa-devel-bounces@alsa-project.org [mailto:alsa-devel-bounces@alsa-project.org] On Behalf Of Brian Austin Sent: Friday, September 30, 2011 10:12 PM To: alsa-devel@alsa-project.org Cc: Brian Austin; vinod.koul@linux.intel.com; broonie@opensource.wolfsonmicro.com; lrg@ti.com; joe@nucleusys.com Subject: [alsa-devel] [PATCH v2] ASoC: Add support for cs42l73 codec
This patch adds support for the Cirrus Logic CS42L73 low power stereo codec.
This patch has cleared checkpatch.pl with no warnings or errors. Code changes requested were implemented. ASoC API changes requested were implemented.
Signed-off-by: Brian Austin brian.austin@cirrus.com
sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs42l73.c | 1047 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs42l73.h | 223 ++++++++++ 4 files changed, 1276 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 3449431..9107c48 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 787881b..e89e84b 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..18a49d8 --- /dev/null +++ b/sound/soc/codecs/cs42l73.c @@ -0,0 +1,1047 @@ +/*
- cs42l73.c -- CS42L73 ALSA Soc Audio driver
- Copyright 2011 Cirrus Logic, Inc.
- Authors: Georgi Vlaev, Nucleus Systems Ltd, office@nucleusys.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 <linux/gpio.h>
+#include "cs42l73.h"
+struct sp_config {
u8 spc, mmcc, spfs;u32 srate;+};
+struct cs42l73_private {
enum snd_soc_control_type control_type;void *control_data;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, +/*10*/ 0x00, 0x15, 0x00, 0x06, +/*14*/ 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, +/*60*/ 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(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 s42l73_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 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 cs42l73_mono_mixer_text[] = {
"Left", "Right", "Mono Mix"};+static const struct soc_enum spk_asp_mono_mixer_enum =
SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 6,ARRAY_SIZE(cs42l73_mono_mixer_text),cs42l73_mono_mixer_text);
+static const struct soc_enum spk_xsp_mono_mixer_enum =
SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 4,ARRAY_SIZE(cs42l73_mono_mixer_text),cs42l73_mono_mixer_text);
+static const struct soc_enum esl_asp_mono_mixer_enum =
SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 2,ARRAY_SIZE(cs42l73_mono_mixer_text),cs42l73_mono_mixer_text);
+static const struct soc_enum esl_xsp_mono_mixer_enum =
SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 0,ARRAY_SIZE(cs42l73_mono_mixer_text),cs42l73_mono_mixer_text);
+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 vspout_mixer_enum =
SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 5,ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text);+static const struct soc_enum xspout_mixer_enum =
SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 4,ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text);+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_DOUBLE_S8_TLV("Speakerphone Digital Playback Volume",CS42L73_SPKDVOL, 0x34, 0xE4, hl_tlv),SOC_DOUBLE_S8_TLV("Ear Speaker Digital Playback Volume",CS42L73_ESLDVOL, 0x34, 0xE4, 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("Invert ADC Signal Polarity Switch", CS42L73_ADCIPC, 1, 5,1,
0),SOC_DOUBLE("ADC Boost Switch", CS42L73_ADCIPC, 2, 6, 1, 0),SOC_SINGLE("Charge Pump Frequency Volume", CS42L73_CPFCHC, 4, 15,0),
SOC_SINGLE("HL Limiter Attack Rate Volume", CS42L73_LIMARATEHL, 0,0x3F,
0),SOC_SINGLE("HL Limiter Release Rate Volume", 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, 1,
limiter_tlv),SOC_SINGLE_TLV("ALC Min Threshold Volume", CS42L73_ALCMINMAX, 2,7, 1,
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 selectsbetween 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 Attenuation Volume",CS42L73_XSPAIPAA, CS42L73_XSPBIPBA, 0, 0x3F, 1,attn_tlv),SOC_DOUBLE_R_TLV("XSP-XSP Attenuation Volume",CS42L73_XSPAXSPAA, CS42L73_XSPBXSPBA, 0, 0x3F, 1,attn_tlv),SOC_DOUBLE_R_TLV("XSP-ASP Attenuation Volume",CS42L73_XSPAASPAA, CS42L73_XSPAASPBA, 0, 0x3F, 1,attn_tlv),SOC_DOUBLE_R_TLV("XSP-VSP Attenuation Volume",CS42L73_XSPAVSPMA, CS42L73_XSPBVSPMA, 0, 0x3F, 1,attn_tlv),SOC_DOUBLE_R_TLV("ASP-IP Attenuation Volume",CS42L73_ASPAIPAA, CS42L73_ASPBIPBA, 0, 0x3F, 1,attn_tlv),SOC_DOUBLE_R_TLV("ASP-XSP Attenuation Volume",CS42L73_ASPAXSPAA, CS42L73_ASPBXSPBA, 0, 0x3F, 1,attn_tlv),SOC_DOUBLE_R_TLV("ASP-ASP Attenuation Volume",CS42L73_ASPAASPAA, CS42L73_ASPBASPBA, 0, 0x3F, 1,attn_tlv),SOC_DOUBLE_R_TLV("ASP-VSP Attenuation Volume",CS42L73_ASPAVSPMA, CS42L73_ASPBVSPMA, 0, 0x3F, 1,attn_tlv),SOC_DOUBLE_R_TLV("VSP-IP Attenuation Volume",CS42L73_VSPAIPAA, CS42L73_VSPBIPBA, 0, 0x3F, 1,attn_tlv),SOC_DOUBLE_R_TLV("VSP-XSP Attenuation Volume",CS42L73_VSPAXSPAA, CS42L73_VSPBXSPBA, 0, 0x3F, 1,attn_tlv),SOC_DOUBLE_R_TLV("VSP-ASP Attenuation Volume",CS42L73_VSPAASPAA, CS42L73_VSPBASPBA, 0, 0x3F, 1,attn_tlv),SOC_DOUBLE_R_TLV("VSP-VSP Attenuation Volume",CS42L73_VSPAVSPMA, CS42L73_VSPBVSPMA, 0, 0x3F, 1,attn_tlv),SOC_DOUBLE_R_TLV("HL-IP Attenuation Volume",CS42L73_HLAIPAA, CS42L73_HLBIPBA, 0, 0x3F, 1,attn_tlv),SOC_DOUBLE_R_TLV("HL-XSP Attenuation Volume",CS42L73_HLAXSPAA, CS42L73_HLBXSPBA, 0, 0x3F, 1,attn_tlv),SOC_DOUBLE_R_TLV("HL-ASP Attenuation Volume",CS42L73_HLAASPAA, CS42L73_HLBASPBA, 0, 0x3F, 1,attn_tlv),SOC_DOUBLE_R_TLV("HL-VSP Attenuation Volume",CS42L73_HLAVSPMA, CS42L73_HLBVSPMA, 0, 0x3F, 1,attn_tlv),SOC_SINGLE_TLV("SPK-IP Mono Attenuation Volume",CS42L73_SPKMIPMA, 0, 0x3F, 1, attn_tlv),SOC_SINGLE_TLV("SPK-XSP Mono Attenuation Volume",CS42L73_SPKMXSPA, 0, 0x3F, 1, attn_tlv),SOC_SINGLE_TLV("SPK-ASP Mono Attenuation Volume",CS42L73_SPKMASPA, 0, 0x3F, 1, attn_tlv),SOC_SINGLE_TLV("SPK-VSP Mono Attenuation Volume",CS42L73_SPKMVSPMA, 0, 0x3F, 1, attn_tlv),SOC_SINGLE_TLV("ESL-IP Mono Attenuation Volume",CS42L73_ESLMIPMA, 0, 0x3F, 1, attn_tlv),SOC_SINGLE_TLV("ESL-XSP Mono Attenuation Volume",CS42L73_ESLMXSPA, 0, 0x3F, 1, attn_tlv),SOC_SINGLE_TLV("ESL-ASP Mono Attenuation Volume",CS42L73_ESLMASPA, 0, 0x3F, 1, attn_tlv),SOC_SINGLE_TLV("ESL-VSP Mono Attenuation Volume",CS42L73_ESLMVSPMA, 0, 0x3F, 1, attn_tlv),SOC_ENUM("IP Digital Swap/Mono Select", ip_swap_enum),SOC_ENUM("ESL-XSP Mono Mixer Select", esl_xsp_mono_mixer_enum),SOC_ENUM("ESL-ASP Mono Mixer Select", esl_asp_mono_mixer_enum),SOC_ENUM("SPK-ASP Mono Mixer Select", spk_asp_mono_mixer_enum),SOC_ENUM("SPK-XSP Mono Mixer Select", spk_xsp_mono_mixer_enum),SOC_ENUM("VSP Output Mixer Select", vspout_mixer_enum),SOC_ENUM("XSP Output Mixer Select", xspout_mixer_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_INPUT("DMICA"),SND_SOC_DAPM_INPUT("DMICB"),SND_SOC_DAPM_AIF_OUT("XSPOUT", "XSP Capture", 0,CS42L73_PWRCTL2, 1, 1),SND_SOC_DAPM_AIF_OUT("ASPOUT", "ASP Capture", 0,CS42L73_PWRCTL2, 3, 1),SND_SOC_DAPM_AIF_OUT("VSPOUT", "VSP Capture", 0,CS42L73_PWRCTL2, 4, 1),SND_SOC_DAPM_AIF_IN("XSPIN", "XSP Playback", 0,CS42L73_PWRCTL2, 0, 1),SND_SOC_DAPM_AIF_IN("ASPIN", "ASP Playback", 0,CS42L73_PWRCTL2, 2, 1),SND_SOC_DAPM_AIF_IN("VSPIN", "VSP Playback", 0,CS42L73_PWRCTL2, 4, 1),SND_SOC_DAPM_ADC("ADC Left", NULL, CS42L73_PWRCTL1, 5, 1),SND_SOC_DAPM_ADC("ADC Right", NULL, CS42L73_PWRCTL1, 7, 1),SND_SOC_DAPM_DAC("DAC Left", NULL, SND_SOC_NOPM, 0, 0),SND_SOC_DAPM_DAC("DAC Right", NULL, SND_SOC_NOPM, 0, 0),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 Mux Left", SND_SOC_NOPM, 0, 0,&pgaa_mux),
SND_SOC_DAPM_MUX("PGA Mux Right", SND_SOC_NOPM, 0, 0,&pgab_mux),
SND_SOC_DAPM_PGA("HP Amp Left", CS42L73_PWRCTL3, 0, 1, NULL, 0),SND_SOC_DAPM_PGA("HP Amp Right", CS42L73_PWRCTL3, 0, 1, NULL,0),
SND_SOC_DAPM_PGA("LO Amp Left", CS42L73_PWRCTL3, 1, 1, NULL, 0),SND_SOC_DAPM_PGA("LO Amp Right", CS42L73_PWRCTL3, 1, 1, NULL,0),
SND_SOC_DAPM_PGA("SPK Amp", CS42L73_PWRCTL3, 2, 1, NULL, 0),SND_SOC_DAPM_PGA("EAR Amp", CS42L73_PWRCTL3, 3, 1, NULL, 0),SND_SOC_DAPM_PGA("SPKLO Amp", CS42L73_PWRCTL3, 4, 1, NULL, 0),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[] = {
{"HPOUTA", NULL, "HP Amp Left"},{"HPOUTB", NULL, "HP Amp Right"},{"LINEOUTA", NULL, "LO Amp Left"},{"LINEOUTB", NULL, "LO Amp Right"},{"SPKOUT", NULL, "SPK Amp"},{"EAROUT", NULL, "EAR Amp"},{"SPKLINEOUT", NULL, "SPKLO Amp"},{"HP Amp Left", "DAC", "DAC Left"},{"HP Amp Right", "DAC", "DAC Right"},{"LO Amp Left", "DAC", "DAC Left"},{"LO Amp Right", "DAC", "DAC Right"},{"SPK Amp", "DAC", "DAC Left"},{"SPKLO Amp", "DAC", "DAC Right"},{"EAR Amp", "DAC", "DAC Right"},{"PGA Mux Left", NULL, "LINEINA"},{"PGA Mux Right", NULL, "LINEINB"},{"PGA Mux Left", NULL, "MIC1"},{"PGA Mux Right", NULL, "MIC2"},{"PGA Left", NULL, "PGA Mux Left"},{"PGA Right", NULL, "PGA Mux Right"},{"ADC Left", "ADC", "PGA Left"},{"ADC Right", "ADC", "PGA Right"},{"XSPOUT", NULL, "ADC Left"},{"XSPOUT", NULL, "ADC Right"},{"DAC Left", NULL, "XSPIN"},{"DAC Right", NULL, "XSPIN"},{"ASPOUT", NULL, "ADC Left"},{"ASPOUT", NULL, "ADC Right"},{"DAC Left", NULL, "ASPIN"},{"DAC Right", NULL, "ASPIN"},{"VSPOUT", NULL, "ADC Left"},{"VSPOUT", NULL, "ADC Right"},{"DAC Left", NULL, "VSPIN"},{"DAC Right", NULL, "VSPIN"},+};
+struct cs42l73_mclk_div {
u32 mclk;u32 srate;u8 mmcc;+};
+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;+};
+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 */+};
+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;+}
+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) +{
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(priv->sysclk);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->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);if (clk_id != CS42L73_CLKID_MCLK1 && clk_id != CS42L73_CLKID_MCLK2){
dev_err(codec->dev, "Invalid clk_id %u\n", clk_id);return -EINVAL;}if ((cs42l73_get_mclkx_coeff(freq) < 0)) {dev_err(codec->dev, "Invalid sysclk %u\n", freq);return -EINVAL;}if ((cs42l73_set_mclk(dai)) < 0) {dev_err(codec->dev, "Unable to set MCLK for dai %s\n",dai->name);return -EINVAL;}priv->sysclk = freq;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);int id = codec_dai->id;int 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 &= ~xSPDIF_PCM;break;case SND_SOC_DAIFMT_DSP_A:case SND_SOC_DAIFMT_DSP_B:if (mmcc & MS_MASTER) {dev_err(codec->dev,"PCM format is supported only in slave mode\n");return -EINVAL;}if (id == CS42L73_ASP) {dev_err(codec->dev,"PCM format is not supported on ASP port\n");return -EINVAL;}spc |= xSPDIF_PCM;break;default:return -EINVAL;}if (spc & xSPDIF_PCM) {spc &= (31 << 3); /* Clear PCM mode, set MSB->LSB */if (format == SND_SOC_DAIFMT_DSP_B&& inv == SND_SOC_DAIFMT_IB_IF)spc |= (xPCM_MODE0 << 4);elseif (format == SND_SOC_DAIFMT_DSP_B&& inv == SND_SOC_DAIFMT_IB_NF)spc |= (xPCM_MODE1 << 4);elseif (format == SND_SOC_DAIFMT_DSP_A&& inv == SND_SOC_DAIFMT_IB_IF)spc |= (xPCM_MODE1 << 4);elsereturn -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 */
You could return -EINVAL.
+}
+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 |= xMCK_SCLK_64FS;} else {/* CS42L73 Slave */dev_dbg(codec->dev, "DAI[%d]: Slave\n", id);priv->config[id].spc &= 0xFC;priv->config[id].spc |= xMCK_SCLK_64FS;}/* Update ASRCs */priv->config[id].srate = srate;cs42l73_update_asrc(codec, id, srate);snd_soc_write(codec, CS42L73_SPC(id), priv->config[id].spc);snd_soc_write(codec, CS42L73_MMCC(id), priv->config[id].mmcc);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 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,+};
+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);codec->control_data = cs42l73->control_data;codec->hw_write = (hw_write_t)i2c_master_send;ret = snd_soc_codec_set_cache_io(codec, 8, 8, cs42l73->control_type);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;snd_soc_add_controls(codec, cs42l73_snd_controls,ARRAY_SIZE(cs42l73_snd_controls));return ret;+}
+static int cs42l73_remove(struct snd_soc_codec *codec) +{
cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF);return 0;+}
+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),+};
+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(struct cs42l73_private), GFP_KERNEL);if (!cs42l73) {dev_err(&i2c_client->dev, "could not allocate codec\n");return -ENOMEM;}i2c_set_clientdata(i2c_client, cs42l73);cs42l73->control_data = i2c_client;cs42l73->control_type = SND_SOC_I2C;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, office@nucleusys.com"); +MODULE_LICENSE("GPL");
I don't see implementation of digital_mute function. Is it going to be added later?
diff --git a/sound/soc/codecs/cs42l73.h b/sound/soc/codecs/cs42l73.h new file mode 100644 index 0000000..aa09c2e --- /dev/null +++ b/sound/soc/codecs/cs42l73.h @@ -0,0 +1,223 @@ +/*
- ALSA SoC CS42L73 codec driver
- Copyright 2011 Cirrus Logic, Inc.
- Author: Georgi Vlaev office@nucleusys.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 xSP_3ST (1 << 7) +#define xSPDIF_I2S 0 +#define xSPDIF_PCM (1 << 6) +#define xPCM_MODE0 0 +#define xPCM_MODE1 1 +#define xPCM_MODE2 2 +#define xPCM_BO_MSBLSB 0 +#define xPCM_BO_LSBMSB 1 +#define xMCK_SCLK_64FS 0 +#define xMCK_SCLK_MCLK 2 +#define xMCK_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 Mute */ +#define HPMUTE (1 << 7) +/* LO A/B Mute */ +#define LOMUTE (1 << 7) +/* SPK Digital Mute */ +#define SPKDMUTE (1 << 2)
+/* 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__ */
1.7.6.3
Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
+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 */You could return -EINVAL.
This is a correct value to return, not an error. It just means that it takes longer to lock the port if we do not know the sample rate
I don't see implementation of digital_mute function. Is it going to be added later?
I forgot that. I can add it before I submit.
Thanks, Brian
On Fri, Sep 30, 2011 at 11:41:52AM -0500, Brian Austin wrote:
This patch adds support for the Cirrus Logic CS42L73 low power stereo codec.
A few comments here beyond those others made (though I may be replicating some of what they said) but overall this is shaping up well - hopefully next version we can get this merged.
Folks: when replying to patches (or mails in general) you should delete unneeded context. This makes it much easier to find the text you wrote, all the stuff about cutting text improving bandwidth usage applies just as much to the humans reading your mail as to the computers moving it.
This patch has cleared checkpatch.pl with no warnings or errors. Code changes requested were implemented. ASoC API changes requested were implemented.
Stuff like this should go after the ---.
- cs42l73.c -- CS42L73 ALSA Soc Audio driver
- Copyright 2011 Cirrus Logic, Inc.
- Authors: Georgi Vlaev, Nucleus Systems Ltd, office@nucleusys.com
Who's not CCed on the patch, and who hasn't signed it off?
- SOC_DOUBLE("Input Path Digital Switch", CS42L73_ADCIPC, 0, 4, 1, 1),
s/Input Path/Capture/
- SOC_DOUBLE("Invert ADC Signal Polarity Switch", CS42L73_ADCIPC, 1, 5, 1,
0),
ADC Signal Polarity Switch.
- SOC_DOUBLE("ADC Boost Switch", CS42L73_ADCIPC, 2, 6, 1, 0),
This should be a Volume control with TLV information assuming it does what it sounds like and makes the signal louder.
- SOC_SINGLE("Charge Pump Frequency Volume", CS42L73_CPFCHC, 4, 15, 0),
No Volume (clearly this isn't a volume control), this should probably be an enum listing the rates.
- SOC_SINGLE("HL Limiter Attack Rate Volume", CS42L73_LIMARATEHL, 0, 0x3F,
0),- SOC_SINGLE("HL Limiter Release Rate Volume", CS42L73_LIMRRATEHL, 0,
0x3F, 0),
Similarly here, you probably want an enumaration showing the rates.
- SOC_DOUBLE_R_TLV("XSP-IP Attenuation Volume",
CS42L73_XSPAIPAA, CS42L73_XSPBIPBA, 0, 0x3F, 1,attn_tlv),
Attenuation Volume makes no sense - they're approximate synonyms. Just Volume.
- if (clk_id != CS42L73_CLKID_MCLK1 && clk_id != CS42L73_CLKID_MCLK2) {
dev_err(codec->dev, "Invalid clk_id %u\n", clk_id);return -EINVAL;- }
A switch statement would be more idomatic.
- if (spc & xSPDIF_PCM) {
spc &= (31 << 3); /* Clear PCM mode, set MSB->LSB */if (format == SND_SOC_DAIFMT_DSP_B&& inv == SND_SOC_DAIFMT_IB_IF)spc |= (xPCM_MODE0 << 4);elseif (format == SND_SOC_DAIFMT_DSP_B&& inv == SND_SOC_DAIFMT_IB_NF)spc |= (xPCM_MODE1 << 4);elseif (format == SND_SOC_DAIFMT_DSP_A&& inv == SND_SOC_DAIFMT_IB_IF)spc |= (xPCM_MODE1 << 4);elsereturn -EINVAL;
I can't help but feel this would be clearer with a switch statement.
+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);
This is going to fail if someone passes a non-boolean value in - better make sure that tristate is actually 1 or 0.
.ops = &cs42l73_ops,.symmetric_rates = 1,},- {
Please use the kernel coding style for indentation.
+/* 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)
All the identifiers in the header need to be namespaced.
On Oct 2, 2011, at 1:40 PM, Mark Brown wrote:
On Fri, Sep 30, 2011 at 11:41:52AM -0500, Brian Austin wrote:
This patch adds support for the Cirrus Logic CS42L73 low power stereo codec.
A few comments here beyond those others made (though I may be replicating some of what they said) but overall this is shaping up well - hopefully next version we can get this merged.
Folks: when replying to patches (or mails in general) you should delete unneeded context. This makes it much easier to find the text you wrote, all the stuff about cutting text improving bandwidth usage applies just as much to the humans reading your mail as to the computers moving it.
This patch has cleared checkpatch.pl with no warnings or errors. Code changes requested were implemented. ASoC API changes requested were implemented.
Stuff like this should go after the ---.
- cs42l73.c -- CS42L73 ALSA Soc Audio driver
- Copyright 2011 Cirrus Logic, Inc.
- Authors: Georgi Vlaev, Nucleus Systems Ltd, office@nucleusys.com
Who's not CCed on the patch, and who hasn't signed it off?
Wrong email in the file.
- SOC_DOUBLE("Input Path Digital Switch", CS42L73_ADCIPC, 0, 4, 1, 1),
s/Input Path/Capture/
got it
- SOC_DOUBLE("Invert ADC Signal Polarity Switch", CS42L73_ADCIPC, 1, 5, 1,
0),ADC Signal Polarity Switch.
- SOC_DOUBLE("ADC Boost Switch", CS42L73_ADCIPC, 2, 6, 1, 0),
This should be a Volume control with TLV information assuming it does what it sounds like and makes the signal louder.
This is just 0dB or 25dB so now it's just an enum if that's OK.
- SOC_SINGLE("Charge Pump Frequency Volume", CS42L73_CPFCHC, 4, 15, 0),
No Volume (clearly this isn't a volume control), this should probably be an enum listing the rates.
Now enumerated
- SOC_SINGLE("HL Limiter Attack Rate Volume", CS42L73_LIMARATEHL, 0, 0x3F,
0),- SOC_SINGLE("HL Limiter Release Rate Volume", CS42L73_LIMRRATEHL, 0,
0x3F, 0),Similarly here, you probably want an enumaration showing the rates.
This is a range of values that are used to set the rate by setting the step period size. It's values are 0 - 3F. I'm hoping to leave it as an SOC_SINGLE control if possible
- SOC_DOUBLE_R_TLV("XSP-IP Attenuation Volume",
CS42L73_XSPAIPAA, CS42L73_XSPBIPBA, 0, 0x3F, 1,attn_tlv),Attenuation Volume makes no sense - they're approximate synonyms. Just Volume.
- if (clk_id != CS42L73_CLKID_MCLK1 && clk_id != CS42L73_CLKID_MCLK2) {
dev_err(codec->dev, "Invalid clk_id %u\n", clk_id);return -EINVAL;- }
A switch statement would be more idomatic.
- if (spc & xSPDIF_PCM) {
spc &= (31 << 3); /* Clear PCM mode, set MSB->LSB */if (format == SND_SOC_DAIFMT_DSP_B&& inv == SND_SOC_DAIFMT_IB_IF)spc |= (xPCM_MODE0 << 4);elseif (format == SND_SOC_DAIFMT_DSP_B&& inv == SND_SOC_DAIFMT_IB_NF)spc |= (xPCM_MODE1 << 4);elseif (format == SND_SOC_DAIFMT_DSP_A&& inv == SND_SOC_DAIFMT_IB_IF)spc |= (xPCM_MODE1 << 4);elsereturn -EINVAL;I can't help but feel this would be clearer with a switch statement.
I'm trying to make this look better...
+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);
This is going to fail if someone passes a non-boolean value in - better make sure that tristate is actually 1 or 0.
.ops = &cs42l73_ops,.symmetric_rates = 1,},- {
Please use the kernel coding style for indentation.
+/* 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)
All the identifiers in the header need to be namespaced.
- if (spc & xSPDIF_PCM) {
spc &= (31 << 3); /* Clear PCM mode, set MSB->LSB */if (format == SND_SOC_DAIFMT_DSP_B&& inv == SND_SOC_DAIFMT_IB_IF)spc |= (xPCM_MODE0 << 4);elseif (format == SND_SOC_DAIFMT_DSP_B&& inv == SND_SOC_DAIFMT_IB_NF)spc |= (xPCM_MODE1 << 4);elseif (format == SND_SOC_DAIFMT_DSP_A&& inv == SND_SOC_DAIFMT_IB_IF)spc |= (xPCM_MODE1 << 4);elsereturn -EINVAL;I can't help but feel this would be clearer with a switch statement.
I'm trying to make this look better...
if (spc & xSPDIF_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 |= (xPCM_MODE0 << 4);
if (inv == SND_SOC_DAIFMT_IB_NF) spc |= (xPCM_MODE1 << 4); break; case SND_SOC_DAIFMT_DSP_A: if (inv == SND_SOC_DAIFMT_IB_IF) spc |= (xPCM_MODE1 << 4); break; default: return -EINVAL; } }
On Tue, Oct 04, 2011 at 07:17:16PM +0000, Austin, Brian wrote:
if (spc & xSPDIF_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 |= (xPCM_MODE0 << 4); if (inv == SND_SOC_DAIFMT_IB_NF) spc |= (xPCM_MODE1 << 4); break; case SND_SOC_DAIFMT_DSP_A: if (inv == SND_SOC_DAIFMT_IB_IF) spc |= (xPCM_MODE1 << 4); break; default: return -EINVAL; } }
This looks a lot better modulo the coding style issue (switch is not a function). Using snd_soc_update_bits() would be even better.
On Tue, Oct 04, 2011 at 05:53:10PM +0000, Austin, Brian wrote:
On Oct 2, 2011, at 1:40 PM, Mark Brown wrote:
On Fri, Sep 30, 2011 at 11:41:52AM -0500, Brian Austin wrote:
- SOC_DOUBLE("ADC Boost Switch", CS42L73_ADCIPC, 2, 6, 1, 0),
This should be a Volume control with TLV information assuming it does what it sounds like and makes the signal louder.
This is just 0dB or 25dB so now it's just an enum if that's OK.
No, you can still show the gain with TLV information so it renders nicely in UIs.
- SOC_SINGLE("HL Limiter Release Rate Volume", CS42L73_LIMRRATEHL, 0,
0x3F, 0),
Similarly here, you probably want an enumaration showing the rates.
This is a range of values that are used to set the rate by setting the step period size. It's values are 0 - 3F. I'm hoping to leave it as an SOC_SINGLE control if possible
Well, you should certainly drop the "Volume" from the name.
participants (6)
-
Austin, Brian -
Babu, Ramesh -
Brian Austin
-
Lars-Peter Clausen -
Mark Brown -
Vinod Koul