[alsa-devel] [PATCH v3] ASoC: Add support for cs42l73 codec
Signed-off-by:Brian Austin brian.austin@cirrus.com Signed-off-by:Georgi Vlaev joe@nucleusys.com
This patch adds support for Cirrus Logic CS42L73 low power stereo codec.
Applied coding style changes and added correct SoB emails Digital mute to be implemented after more testing --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs42l73.c | 1076 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs42l73.h | 227 ++++++++++ 4 files changed, 1309 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..609cebf --- /dev/null +++ b/sound/soc/codecs/cs42l73.c @@ -0,0 +1,1076 @@ +/* + * cs42l73.c -- CS42L73 ALSA Soc Audio driver + * + * Copyright 2011 Cirrus Logic, Inc. + * + * Authors: Georgi Vlaev, Nucleus Systems Ltd, joe@nucleusys.com + * Brian Austin, Cirrus Logic Inc, brian.austin@cirrus.com + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include "cs42l73.h" + +struct sp_config { + u8 spc, mmcc, spfs; + u32 srate; +}; +struct cs42l73_private { + u32 sysclk; + u8 mclksel; + u32 mclk; + struct sp_config config[3]; +}; + +static const u8 cs42l73_reg[] = { +/*0*/ 0x00, 0x42, 0xA7, 0x30, +/*4*/ 0x00, 0x00, 0xF1, 0xDF, +/*8*/ 0x3F, 0x57, 0x53, 0x00, +/*C*/ 0x00, 0x15, 0x00, 0x15, +/*A*/ 0x00, 0x15, 0x00, 0x06, +/*E*/ 0x00, 0x00, 0x00, 0x00, +/*18*/ 0x00, 0x00, 0x00, 0x00, +/*1C*/ 0x00, 0x00, 0x00, 0x00, +/*20*/ 0x00, 0x00, 0x00, 0x00, +/*24*/ 0x00, 0x00, 0x00, 0x7F, +/*28*/ 0x00, 0x00, 0x3F, 0x00, +/*2C*/ 0x00, 0x3F, 0x00, 0x00, +/*30*/ 0x3F, 0x00, 0x00, 0x00, +/*34*/ 0x18, 0x3F, 0x3F, 0x3F, +/*38*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*3C*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*40*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*44*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*48*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*4C*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*50*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*54*/ 0x3F, 0xAA, 0x3F, 0x3F, +/*58*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*5C*/ 0x3F, 0x3F, 0x00, 0x00, +/*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(adc_boost_tlv, 0, 2500, 0); + +static DECLARE_TLV_DB_SCALE(hl_tlv, -10200, 50, 0); + +static DECLARE_TLV_DB_SCALE(ipd_tlv, -9600, 100, 0); + +static DECLARE_TLV_DB_SCALE(micpga_tlv, -600, 50, 0); + +static const unsigned int limiter_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0), + 3, 7, TLV_DB_SCALE_ITEM(-1200, 300, 0), +}; + +static const DECLARE_TLV_DB_SCALE(attn_tlv, -6300, 100, 1); + +static const char * const cs42l73_pgaa_text[] = { "Line A", "Mic 1" }; +static const char * const cs42l73_pgab_text[] = { "Line B", "Mic 2" }; + +static const struct soc_enum pgaa_enum = + SOC_ENUM_SINGLE(CS42L73_ADCIPC, 3, + ARRAY_SIZE(cs42l73_pgaa_text), cs42l73_pgaa_text); + +static const struct soc_enum pgab_enum = + SOC_ENUM_SINGLE(CS42L73_ADCIPC, 7, + ARRAY_SIZE(cs42l73_pgab_text), cs42l73_pgab_text); + +static const struct snd_kcontrol_new pgaa_mux = + SOC_DAPM_ENUM("Left Analog Input Capture Mux", pgaa_enum); + +static const struct snd_kcontrol_new pgab_mux = + SOC_DAPM_ENUM("Right Analog Input Capture Mux", pgab_enum); + +static const char * const cs42l73_ng_delay_text[] = { + "50ms", "100ms", "150ms", "200ms" }; + +static const struct soc_enum ng_delay_enum = + SOC_ENUM_SINGLE(CS42L73_NGCAB, 0, + ARRAY_SIZE(cs42l73_ng_delay_text), cs42l73_ng_delay_text); + +static const char * const charge_pump_freq_text[] = { + "0", "1", "2", "3", "4", + "5", "6", "7", "8", "9", + "10", "11", "12", "13", "14", "15" }; + +static const struct soc_enum charge_pump_enum = + SOC_ENUM_SINGLE(CS42L73_CPFCHC, 4, + ARRAY_SIZE(charge_pump_freq_text), charge_pump_freq_text); + +static const char * const cs42l73_mono_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 dac1_output_mixer[] = { + SOC_DAPM_SINGLE("Headphone Switch", CS42L73_PWRCTL3, 0, 1, 1), + SOC_DAPM_SINGLE("LineOut Switch", CS42L73_PWRCTL3, 1, 1, 1), +}; + +static const struct snd_kcontrol_new dac2_output_mixer[] = { + SOC_DAPM_SINGLE("Speakerphone Switch", CS42L73_PWRCTL3, 2, 1, 1), + SOC_DAPM_SINGLE("Speakerphone LineOut Switch", CS42L73_PWRCTL3, + 4, 1, 1), + SOC_DAPM_SINGLE("EarSpeaker Switch", CS42L73_PWRCTL3, 3, 1, 1), +}; + +static const struct snd_kcontrol_new cs42l73_snd_controls[] = { + SOC_DOUBLE_R_SX_TLV("Headphone Analog Playback Volume", CS42L73_HPAAVOL, + CS42L73_HPBAVOL, 7, 0xffffffC1, 0x0C, hpaloa_tlv), + + SOC_DOUBLE_R_SX_TLV("LineOut Analog Playback Volume", CS42L73_LOAAVOL, + CS42L73_LOBAVOL, 7, 0xffffffC1, 0x0C, hpaloa_tlv), + + SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL, + CS42L73_MICBPREPGABVOL, 5, 0xffffff35, + 0x34, micpga_tlv), + + SOC_DOUBLE_R("MIC Preamp Switch", CS42L73_MICAPREPGAAVOL, + CS42L73_MICBPREPGABVOL, 6, 1, 1), + + SOC_DOUBLE_R_SX_TLV("Input Path Digital Volume", CS42L73_IPADVOL, + CS42L73_IPBDVOL, 7, 0xffffffA0, 0xA0, ipd_tlv), + + SOC_DOUBLE_R_SX_TLV("HL Digital Playback Volume", + CS42L73_HLADVOL, CS42L73_HLBDVOL, 7, 0xffffffE5, + 0xE4, hl_tlv), + + SOC_SINGLE_TLV("ADC A Boost Volume", + CS42L73_ADCIPC, 2, 0x01, 1, adc_boost_tlv), + + SOC_SINGLE_TLV("ADC B Boost Volume", + CS42L73_ADCIPC, 6, 0x01, 1, adc_boost_tlv), + + SOC_SINGLE_TLV("Speakerphone Digital Playback Volume", + CS42L73_SPKDVOL, 0, 0xE4, 1, hl_tlv), + + SOC_SINGLE_TLV("Ear Speaker Digital Playback Volume", + CS42L73_ESLDVOL, 0, 0xE4, 1, hl_tlv), + + SOC_DOUBLE_R("Headphone Analog Playback Switch", CS42L73_HPAAVOL, + CS42L73_HPBAVOL, 7, 1, 1), + + SOC_DOUBLE_R("LineOut Analog Playback Switch", CS42L73_LOAAVOL, + CS42L73_LOBAVOL, 7, 1, 1), + SOC_DOUBLE("Input Path Digital Switch", CS42L73_ADCIPC, 0, 4, 1, 1), + SOC_DOUBLE("HL Digital Playback Switch", CS42L73_PBDC, 0, + 1, 1, 1), + SOC_SINGLE("Speakerphone Digital Playback Switch", CS42L73_PBDC, 2, 1, + 1), + SOC_SINGLE("Ear Speaker Digital Playback Switch", CS42L73_PBDC, 3, 1, + 1), + + SOC_SINGLE("PGA Soft-Ramp Switch", CS42L73_MIOPC, 3, 1, 0), + SOC_SINGLE("Analog Zero Cross Switch", CS42L73_MIOPC, 2, 1, 0), + SOC_SINGLE("Digital Soft-Ramp Switch", CS42L73_MIOPC, 1, 1, 0), + SOC_SINGLE("Analog Output Soft-Ramp Switch", CS42L73_MIOPC, 0, 1, 0), + + SOC_DOUBLE("ADC Signal Polarity Switch", CS42L73_ADCIPC, 1, 5, 1, + 0), + + SOC_SINGLE("HL Limiter Attack Rate", CS42L73_LIMARATEHL, 0, 0x3F, + 0), + SOC_SINGLE("HL Limiter Release Rate", CS42L73_LIMRRATEHL, 0, + 0x3F, 0), + + + SOC_SINGLE("HL Limiter Switch", CS42L73_LIMRRATEHL, 7, 1, 0), + SOC_SINGLE("HL Limiter All Channels Switch", CS42L73_LIMRRATEHL, 6, 1, + 0), + + SOC_SINGLE_TLV("HL Limiter Max Threshold Volume", CS42L73_LMAXHL, 5, 7, + 1, limiter_tlv), + + SOC_SINGLE_TLV("HL Limiter Cushion Volume", CS42L73_LMAXHL, 2, 7, 1, + limiter_tlv), + + SOC_SINGLE("SPK Limiter Attack Rate Volume", CS42L73_LIMARATESPK, 0, + 0x3F, 0), + SOC_SINGLE("SPK Limiter Release Rate Volume", CS42L73_LIMRRATESPK, 0, + 0x3F, 0), + SOC_SINGLE("SPK Limiter Switch", CS42L73_LIMRRATESPK, 7, 1, 0), + SOC_SINGLE("SPK Limiter All Channels Switch", CS42L73_LIMRRATESPK, 6, 1, + 0), + SOC_SINGLE_TLV("SPK Limiter Max Threshold Volume", CS42L73_LMAXSPK, 5, + 7, 1, limiter_tlv), + + SOC_SINGLE_TLV("SPK Limiter Cushion Volume", CS42L73_LMAXSPK, 2, 7, 1, + limiter_tlv), + + SOC_SINGLE("ESL Limiter Attack Rate Volume", CS42L73_LIMARATEESL, 0, + 0x3F, 0), + SOC_SINGLE("ESL Limiter Release Rate Volume", CS42L73_LIMRRATEESL, 0, + 0x3F, 0), + SOC_SINGLE("ESL Limiter Switch", CS42L73_LIMRRATEESL, 7, 1, 0), + SOC_SINGLE_TLV("ESL Limiter Max Threshold Volume", CS42L73_LMAXESL, 5, + 7, 1, limiter_tlv), + + SOC_SINGLE_TLV("ESL Limiter Cushion Volume", CS42L73_LMAXESL, 2, 7, 1, + limiter_tlv), + + SOC_SINGLE("ALC Attack Rate Volume", CS42L73_ALCARATE, 0, 0x3F, 0), + SOC_SINGLE("ALC Release Rate Volume", CS42L73_ALCRRATE, 0, 0x3F, 0), + SOC_DOUBLE("ALC Switch", CS42L73_ALCARATE, 6, 7, 1, 0), + SOC_SINGLE_TLV("ALC Max Threshold Volume", CS42L73_ALCMINMAX, 5, 7, 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 Volume", + CS42L73_XSPAIPAA, CS42L73_XSPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-XSP Volume", + CS42L73_XSPAXSPAA, CS42L73_XSPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-ASP Volume", + CS42L73_XSPAASPAA, CS42L73_XSPAASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-VSP Volume", + CS42L73_XSPAVSPMA, CS42L73_XSPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("ASP-IP Volume", + CS42L73_ASPAIPAA, CS42L73_ASPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-XSP Volume", + CS42L73_ASPAXSPAA, CS42L73_ASPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-ASP Volume", + CS42L73_ASPAASPAA, CS42L73_ASPBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-VSP Volume", + CS42L73_ASPAVSPMA, CS42L73_ASPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("VSP-IP Volume", + CS42L73_VSPAIPAA, CS42L73_VSPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-XSP Volume", + CS42L73_VSPAXSPAA, CS42L73_VSPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-ASP Volume", + CS42L73_VSPAASPAA, CS42L73_VSPBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-VSP Volume", + CS42L73_VSPAVSPMA, CS42L73_VSPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("HL-IP Volume", + CS42L73_HLAIPAA, CS42L73_HLBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-XSP Volume", + CS42L73_HLAXSPAA, CS42L73_HLBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-ASP Volume", + CS42L73_HLAASPAA, CS42L73_HLBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-VSP Volume", + CS42L73_HLAVSPMA, CS42L73_HLBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_SINGLE_TLV("SPK-IP Mono Volume", + CS42L73_SPKMIPMA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-XSP Mono Volume", + CS42L73_SPKMXSPA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-ASP Mono Volume", + CS42L73_SPKMASPA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-VSP Mono Volume", + CS42L73_SPKMVSPMA, 0, 0x3F, 1, attn_tlv), + + SOC_SINGLE_TLV("ESL-IP Mono Volume", + CS42L73_ESLMIPMA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-XSP Mono Volume", + CS42L73_ESLMXSPA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-ASP Mono Volume", + CS42L73_ESLMASPA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-VSP Mono 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_MIXER("DAC1 Mixer", SND_SOC_NOPM, 0, 0, + dac1_output_mixer, ARRAY_SIZE(dac1_output_mixer)), + SND_SOC_DAPM_MIXER("DAC2 Mixer", SND_SOC_NOPM, 0, 0, + dac2_output_mixer, ARRAY_SIZE(dac2_output_mixer)), + + SND_SOC_DAPM_PGA("DAC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DAC2", SND_SOC_NOPM, 0, 0, NULL, 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", CS42L73_PWRCTL3, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("LO Amp", 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("HPOUT"), + SND_SOC_DAPM_OUTPUT("LINEOUT"), + 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[] = { + {"HPOUT", NULL, "HP Amp"}, + {"LINEOUT", NULL, "LO Amp"}, + {"SPKOUT", NULL, "SPK Amp"}, + {"EAROUT", NULL, "EAR Amp"}, + {"SPKLINEOUT", NULL, "SPKLO Amp"}, + + {"HP Amp", NULL, "DAC1"}, + {"LO Amp", NULL, "DAC1"}, + {"SPK Amp", NULL, "DAC2"}, + {"SPKLO Amp", NULL, "DAC2"}, + {"EAR Amp", NULL, "DAC2"}, + + {"DAC1", NULL, "DAC1 Mixer"}, + {"DAC2", NULL, "DAC2 Mixer"}, + + {"DAC1 Mixer", "Headphone Switch", "XSPIN"}, + {"DAC1 Mixer", "LineOut Switch", "XSPIN"}, + {"DAC2 Mixer", "Speakerphone Switch", "XSPIN"}, + {"DAC2 Mixer", "Speakerphone LineOut Switch", "XSPIN"}, + {"DAC2 Mixer", "EarSpeaker Switch", "XSPIN"}, + + {"DAC1 Mixer", "Headphone Switch", "ASPIN"}, + {"DAC1 Mixer", "LineOut Switch", "ASPIN"}, + {"DAC2 Mixer", "Speakerphone Switch", "ASPIN"}, + {"DAC2 Mixer", "Speakerphone LineOut Switch", "ASPIN"}, + {"DAC2 Mixer", "EarSpeaker Switch", "ASPIN"}, + + {"DAC1 Mixer", "Headphone Switch", "VSPIN"}, + {"DAC1 Mixer", "LineOut Switch", "VSPIN"}, + {"DAC2 Mixer", "Speakerphone Switch", "VSPIN"}, + {"DAC2 Mixer", "Speakerphone LineOut Switch", "VSPIN"}, + {"DAC2 Mixer", "EarSpeaker Switch", "VSPIN"}, + + {"PGA Mux Left", "Line A", "LINEINA"}, + {"PGA Mux Right", "Line B", "LINEINB"}, + + {"PGA Mux Left", "Mic 1", "MIC1"}, + {"PGA Mux Right", "Mic 2", "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"}, + + {"ASPOUT", NULL, "ADC Left"}, + {"ASPOUT", NULL, "ADC Right"}, + + {"VSPOUT", NULL, "ADC Left"}, + {"VSPOUT", NULL, "ADC Right"}, + +}; + +struct cs42l73_mclk_div { + u32 mclk; + u32 srate; + u8 mmcc; +}; + +static struct cs42l73_mclk_div cs42l73_mclk_coeffs[] = { + /* MCLK, Sample Rate, xMMCC[5:0] */ + {5644800, 11025, 0x30}, + {5644800, 22050, 0x20}, + {5644800, 44100, 0x10}, + + {6000000, 8000, 0x39}, + {6000000, 11025, 0x33}, + {6000000, 12000, 0x31}, + {6000000, 16000, 0x29}, + {6000000, 22050, 0x23}, + {6000000, 24000, 0x21}, + {6000000, 32000, 0x19}, + {6000000, 44100, 0x13}, + {6000000, 48000, 0x11}, + + {6144000, 8000, 0x38}, + {6144000, 12000, 0x30}, + {6144000, 16000, 0x28}, + {6144000, 24000, 0x20}, + {6144000, 32000, 0x18}, + {6144000, 48000, 0x10}, + + {6500000, 8000, 0x3C}, + {6500000, 11025, 0x35}, + {6500000, 12000, 0x34}, + {6500000, 16000, 0x2C}, + {6500000, 22050, 0x25}, + {6500000, 24000, 0x24}, + {6500000, 32000, 0x1C}, + {6500000, 44100, 0x15}, + {6500000, 48000, 0x14}, + + {6400000, 8000, 0x3E}, + {6400000, 11025, 0x37}, + {6400000, 12000, 0x36}, + {6400000, 16000, 0x2E}, + {6400000, 22050, 0x27}, + {6400000, 24000, 0x26}, + {6400000, 32000, 0x1E}, + {6400000, 44100, 0x17}, + {6400000, 48000, 0x16}, +}; + +struct cs42l73_mclkx_div { + u32 mclkx; + u8 ratio; + u8 mclkdiv; +}; + +static struct cs42l73_mclkx_div cs42l73_mclkx_coeffs[] = { + {5644800, 1, 0}, /* 5644800 */ + {6000000, 1, 0}, /* 6000000 */ + {6144000, 1, 0}, /* 6144000 */ + {11289600, 2, 2}, /* 5644800 */ + {12288000, 2, 2}, /* 6144000 */ + {12000000, 2, 2}, /* 6000000 */ + {13000000, 2, 2}, /* 6500000 */ + {19200000, 3, 3}, /* 6400000 */ + {24000000, 4, 4}, /* 6000000 */ + {26000000, 4, 4}, /* 6500000 */ + {38400000, 6, 5} /* 6400000 */ +}; + +static int cs42l73_get_mclkx_coeff(int mclkx) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs42l73_mclkx_coeffs); i++) { + if (cs42l73_mclkx_coeffs[i].mclkx == mclkx) + return i; + } + return -EINVAL; +} + +static int cs42l73_get_mclk_coeff(int mclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs42l73_mclk_coeffs); i++) { + if (cs42l73_mclk_coeffs[i].mclk == mclk && + cs42l73_mclk_coeffs[i].srate == srate) + return i; + } + return -EINVAL; + +} + +static int cs42l73_set_mclk(struct snd_soc_dai *dai, unsigned int freq) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + + int mclkx_coeff; + u32 mclk = 0; + u8 dmmcc = 0; + + /* MCLKX -> MCLK */ + mclkx_coeff = cs42l73_get_mclkx_coeff(freq); + + mclk = cs42l73_mclkx_coeffs[mclkx_coeff].mclkx / + cs42l73_mclkx_coeffs[mclkx_coeff].ratio; + + dev_dbg(codec->dev, "MCLK%u %u <-> internal MCLK %u\n", + priv->mclksel + 1, cs42l73_mclkx_coeffs[mclkx_coeff].mclkx, + mclk); + + dmmcc = (priv->mclksel << 4) | + (cs42l73_mclkx_coeffs[mclkx_coeff].mclkdiv << 1); + + snd_soc_write(codec, CS42L73_DMMCC, dmmcc); + + priv->sysclk = mclkx_coeff; + priv->mclk = mclk; + + return 0; +} + +static int cs42l73_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case CS42L73_CLKID_MCLK1: + break; + case CS42L73_CLKID_MCLK2: + break; + default: + return -EINVAL; + } + + if ((cs42l73_set_mclk(dai, freq)) < 0) { + dev_err(codec->dev, "Unable to set MCLK for dai %s\n", + dai->name); + return -EINVAL; + } + + priv->mclksel = clk_id; + + return 0; +} + +static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + u8 id = codec_dai->id; + u8 inv, format; + u8 spc, mmcc; + + spc = snd_soc_read(codec, CS42L73_SPC(id)); + mmcc = snd_soc_read(codec, CS42L73_MMCC(id)); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + mmcc |= MS_MASTER; + break; + + case SND_SOC_DAIFMT_CBS_CFS: + mmcc &= ~MS_MASTER; + break; + + default: + return -EINVAL; + } + + format = (fmt & SND_SOC_DAIFMT_FORMAT_MASK); + inv = (fmt & SND_SOC_DAIFMT_INV_MASK); + + switch (format) { + case SND_SOC_DAIFMT_I2S: + spc &= ~SPDIF_PCM; + break; + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + if (mmcc & MS_MASTER) { + dev_err(codec->dev, + "PCM format 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 |= SPDIF_PCM; + break; + default: + return -EINVAL; + } + + if (spc & SPDIF_PCM) { + spc &= (31 << 3); /* Clear PCM mode, set MSB->LSB */ + switch (format) { + case SND_SOC_DAIFMT_DSP_B: + if (inv == SND_SOC_DAIFMT_IB_IF) + spc |= (PCM_MODE0 << 4); + if (inv == SND_SOC_DAIFMT_IB_NF) + spc |= (PCM_MODE1 << 4); + break; + case SND_SOC_DAIFMT_DSP_A: + if (inv == SND_SOC_DAIFMT_IB_IF) + spc |= (PCM_MODE1 << 4); + break; + default: + return -EINVAL; + } + } + + priv->config[id].spc = spc; + priv->config[id].mmcc = mmcc; + + return 0; +} + +static u32 cs42l73_asrc_rates[] = { + 8000, 11025, 12000, 16000, 22050, + 24000, 32000, 44100, 48000 +}; + +static unsigned int cs42l73_get_xspfs_coeff(u32 rate) +{ + int i; + for (i = 0; i < ARRAY_SIZE(cs42l73_asrc_rates); i++) { + if (cs42l73_asrc_rates[i] == rate) + return i + 1; + } + return 0; /* 0 = Don't know */ +} + +static void cs42l73_update_asrc(struct snd_soc_codec *codec, int id, int srate) +{ + u8 spfs = 0; + + if (srate > 0) + spfs = cs42l73_get_xspfs_coeff(srate); + + switch (id) { + case CS42L73_XSP: + snd_soc_update_bits(codec, CS42L73_VXSPFS, 0x0f, spfs); + break; + case CS42L73_ASP: + snd_soc_update_bits(codec, CS42L73_ASPC, 0x3c, spfs << 2); + break; + case CS42L73_VSP: + snd_soc_update_bits(codec, CS42L73_VXSPFS, 0xf0, spfs << 4); + break; + default: + break; + } +} + +static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + int id = dai->id; + int mclk_coeff; + int srate = params_rate(params); + + if (priv->config[id].mmcc & MS_MASTER) { + /* CS42L73 Master */ + /* MCLK -> srate */ + mclk_coeff = + cs42l73_get_mclk_coeff(priv->mclk, srate); + + if (mclk_coeff < 0) + return -EINVAL; + + dev_dbg(codec->dev, + "DAI[%d]: MCLK %u, srate %u, MMCC[5:0] = %x\n", + id, priv->mclk, srate, + cs42l73_mclk_coeffs[mclk_coeff].mmcc); + + priv->config[id].mmcc &= 0xC0; + priv->config[id].mmcc |= cs42l73_mclk_coeffs[mclk_coeff].mmcc; + priv->config[id].spc &= 0xFC; + priv->config[id].spc |= MCK_SCLK_64FS; + + } else { + /* CS42L73 Slave */ + dev_dbg(codec->dev, "DAI[%d]: Slave\n", id); + priv->config[id].spc &= 0xFC; + priv->config[id].spc |= MCK_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 const struct snd_soc_dai_ops cs42l73_ops = { + .startup = cs42l73_pcm_startup, + .hw_params = cs42l73_pcm_hw_params, + .set_fmt = cs42l73_set_dai_fmt, + .set_sysclk = cs42l73_set_sysclk, + .set_tristate = cs42l73_set_tristate, +}; + +static struct snd_soc_dai_driver cs42l73_dai[] = { + { + .name = "cs42l73-xsp", + .id = CS42L73_XSP, + .playback = { + .stream_name = "XSP Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .capture = { + .stream_name = "XSP Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .ops = &cs42l73_ops, + .symmetric_rates = 1, + }, + { + .name = "cs42l73-asp", + .id = CS42L73_ASP, + .playback = { + .stream_name = "ASP Playback", + .channels_min = 2, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .capture = { + .stream_name = "ASP Capture", + .channels_min = 2, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .ops = &cs42l73_ops, + .symmetric_rates = 1, + }, + { + .name = "cs42l73-vsp", + .id = CS42L73_VSP, + .playback = { + .stream_name = "VSP Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .capture = { + .stream_name = "VSP Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .ops = &cs42l73_ops, + .symmetric_rates = 1, + } +}; + +static int cs42l73_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int cs42l73_resume(struct snd_soc_codec *codec) +{ + + cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static int cs42l73_probe(struct snd_soc_codec *codec) +{ + int ret; + unsigned int devid = 0; + struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(codec); + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* initialize codec */ + ret = snd_soc_read(codec, CS42L73_DEVID_AB); + devid = (ret & 0xFF) << 12; + + ret = snd_soc_read(codec, CS42L73_DEVID_CD); + devid |= (ret & 0xFF) << 4; + + ret = snd_soc_read(codec, CS42L73_DEVID_E); + devid |= (ret & 0xF0) >> 4; + + + if (devid != CS42L73_DEVID) { + dev_err(codec->dev, + "CS42L73 Device ID (%X). Expected %X\n", + devid, CS42L73_DEVID); + return ret; + } + + ret = snd_soc_read(codec, CS42L73_REVID); + if (ret < 0) { + dev_err(codec->dev, "Get Revision ID failed\n"); + return ret; + } + + dev_info(codec->dev, + "Cirrus Logic CS42L73, Revision: %02X\n", ret & 0xFF); + + cs42l73->mclksel = CS42L73_CLKID_MCLK1; /* MCLK1 as master clk */ + cs42l73->mclk = 0; + + return ret; +} + +static int cs42l73_remove(struct snd_soc_codec *codec) +{ + cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs42l73 = { + .probe = cs42l73_probe, + .remove = cs42l73_remove, + .suspend = cs42l73_suspend, + .resume = cs42l73_resume, + .set_bias_level = cs42l73_set_bias_level, + .reg_cache_size = ARRAY_SIZE(cs42l73_reg), + .reg_cache_default = cs42l73_reg, + .reg_word_size = sizeof(u8), + .dapm_widgets = cs42l73_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42l73_dapm_widgets), + .dapm_routes = cs42l73_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs42l73_audio_map), + .controls = cs42l73_snd_controls, + .num_controls = ARRAY_SIZE(cs42l73_snd_controls), +}; + +static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs42l73_private *cs42l73; + int ret; + + cs42l73 = kzalloc((sizeof *cs42l73), GFP_KERNEL); + if (!cs42l73) { + dev_err(&i2c_client->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + i2c_set_clientdata(i2c_client, cs42l73); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs42l73, cs42l73_dai, + ARRAY_SIZE(cs42l73_dai)); + if (ret < 0) + kfree(cs42l73); + return ret; +} + +static __devexit int cs42l73_i2c_remove(struct i2c_client *client) +{ + struct cs42l73_private *cs42l73 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + kfree(cs42l73); + + return 0; +} + +static const struct i2c_device_id cs42l73_id[] = { + {"cs42l73", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs42l73_id); + +static struct i2c_driver cs42l73_i2c_driver = { + .driver = { + .name = "cs42l73", + .owner = THIS_MODULE, + }, + .id_table = cs42l73_id, + .probe = cs42l73_i2c_probe, + .remove = __devexit_p(cs42l73_i2c_remove), + +}; + +static int __init cs42l73_modinit(void) +{ + int ret; + ret = i2c_add_driver(&cs42l73_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "%s: can't add i2c driver\n", __func__); + return ret; + } + return 0; +} + +module_init(cs42l73_modinit); + +static void __exit cs42l73_exit(void) +{ + i2c_del_driver(&cs42l73_i2c_driver); +} + +module_exit(cs42l73_exit); + +MODULE_DESCRIPTION("ASoC CS42L73 driver"); +MODULE_AUTHOR("Georgi Vlaev, Nucleus Systems Ltd, joe@nucleusys.com"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42l73.h b/sound/soc/codecs/cs42l73.h new file mode 100644 index 0000000..188f06d --- /dev/null +++ b/sound/soc/codecs/cs42l73.h @@ -0,0 +1,227 @@ +/* + * ALSA SoC CS42L73 codec driver + * + * Copyright 2011 Cirrus Logic, Inc. + * + * Author: Georgi Vlaev joe@nucleusys.com + * Brian Austin brian.austin@cirrus.com + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __CS42L73_H__ +#define __CS42L73_H__ + +/* I2C Registers */ +/* I2C Address: 1001010[R/W] - 10010100 = 0x94(Write); 10010101 = 0x95(Read) */ +#define CS42L73_CHIP_ID 0x4a +#define CS42L73_DEVID_AB 0x01 /* Device ID A & B [RO]. */ +#define CS42L73_DEVID_CD 0x02 /* Device ID C & D [RO]. */ +#define CS42L73_DEVID_E 0x03 /* Device ID E [RO]. */ +#define CS42L73_REVID 0x05 /* Revision ID [RO]. */ +#define CS42L73_PWRCTL1 0x06 /* Power Control 1. */ +#define CS42L73_PWRCTL2 0x07 /* Power Control 2. */ +#define CS42L73_PWRCTL3 0x08 /* Power Control 3. */ +#define CS42L73_CPFCHC 0x09 /* Charge Pump Freq. Class H Ctl. */ +#define CS42L73_OLMBMSDC 0x0A /* Output Load, MIC Bias, MIC2 SDT */ +#define CS42L73_DMMCC 0x0B /* Digital MIC & Master Clock Ctl. */ +#define CS42L73_XSPC 0x0C /* Auxiliary Serial Port (XSP) Ctl. */ +#define CS42L73_XSPMMCC 0x0D /* XSP Master Mode Clocking Control. */ +#define CS42L73_ASPC 0x0E /* Audio Serial Port (ASP) Control. */ +#define CS42L73_ASPMMCC 0x0F /* ASP Master Mode Clocking Control. */ +#define CS42L73_VSPC 0x10 /* Voice Serial Port (VSP) Control. */ +#define CS42L73_VSPMMCC 0x11 /* VSP Master Mode Clocking Control. */ +#define CS42L73_VXSPFS 0x12 /* VSP & XSP Sample Rate. */ +#define CS42L73_MIOPC 0x13 /* Misc. Input & Output Path Control. */ +#define CS42L73_ADCIPC 0x14 /* ADC/IP Control. */ +#define CS42L73_MICAPREPGAAVOL 0x15 /* MIC 1 [A] PreAmp, PGAA Vol. */ +#define CS42L73_MICBPREPGABVOL 0x16 /* MIC 2 [B] PreAmp, PGAB Vol. */ +#define CS42L73_IPADVOL 0x17 /* Input Pat7h A Digital Volume. */ +#define CS42L73_IPBDVOL 0x18 /* Input Path B Digital Volume. */ +#define CS42L73_PBDC 0x19 /* Playback Digital Control. */ +#define CS42L73_HLADVOL 0x1A /* HP/Line A Out Digital Vol. */ +#define CS42L73_HLBDVOL 0x1B /* HP/Line B Out Digital Vol. */ +#define CS42L73_SPKDVOL 0x1C /* Spkphone Out [A] Digital Vol. */ +#define CS42L73_ESLDVOL 0x1D /* Ear/Spkphone LO [B] Digital */ +#define CS42L73_HPAAVOL 0x1E /* HP A Analog Volume. */ +#define CS42L73_HPBAVOL 0x1F /* HP B Analog Volume. */ +#define CS42L73_LOAAVOL 0x20 /* Line Out A Analog Volume. */ +#define CS42L73_LOBAVOL 0x21 /* Line Out B Analog Volume. */ +#define CS42L73_STRINV 0x22 /* Stereo Input Path Adv. Vol. */ +#define CS42L73_XSPINV 0x23 /* Auxiliary Port Input Advisory Vol. */ +#define CS42L73_ASPINV 0x24 /* Audio Port Input Advisory Vol. */ +#define CS42L73_VSPINV 0x25 /* Voice Port Input Advisory Vol. */ +#define CS42L73_LIMARATEHL 0x26 /* Lmtr Attack Rate HP/Line. */ +#define CS42L73_LIMRRATEHL 0x27 /* Lmtr Ctl, Rel.Rate HP/Line. */ +#define CS42L73_LMAXHL 0x28 /* Lmtr Thresholds HP/Line. */ +#define CS42L73_LIMARATESPK 0x29 /* Lmtr Attack Rate Spkphone [A]. */ +#define CS42L73_LIMRRATESPK 0x2A /* Lmtr Ctl,Release Rate Spk. [A]. */ +#define CS42L73_LMAXSPK 0x2B /* Lmtr Thresholds Spkphone [A]. */ +#define CS42L73_LIMARATEESL 0x2C /* Lmtr Attack Rate */ +#define CS42L73_LIMRRATEESL 0x2D /* Lmtr Ctl,Release Rate */ +#define CS42L73_LMAXESL 0x2E /* Lmtr Thresholds */ +#define CS42L73_ALCARATE 0x2F /* ALC Enable, Attack Rate AB. */ +#define CS42L73_ALCRRATE 0x30 /* ALC Release Rate AB. */ +#define CS42L73_ALCMINMAX 0x31 /* ALC Thresholds AB. */ +#define CS42L73_NGCAB 0x32 /* Noise Gate Ctl AB. */ +#define CS42L73_ALCNGMC 0x33 /* ALC & Noise Gate Misc Ctl. */ +#define CS42L73_MIXERCTL 0x34 /* Mixer Control. */ +#define CS42L73_HLAIPAA 0x35 /* HP/LO Left Mixer: L. */ +#define CS42L73_HLBIPBA 0x36 /* HP/LO Right Mixer: R. */ +#define CS42L73_HLAXSPAA 0x37 /* HP/LO Left Mixer: XSP L */ +#define CS42L73_HLBXSPBA 0x38 /* HP/LO Right Mixer: XSP R */ +#define CS42L73_HLAASPAA 0x39 /* HP/LO Left Mixer: ASP L */ +#define CS42L73_HLBASPBA 0x3A /* HP/LO Right Mixer: ASP R */ +#define CS42L73_HLAVSPMA 0x3B /* HP/LO Left Mixer: VSP. */ +#define CS42L73_HLBVSPMA 0x3C /* HP/LO Right Mixer: VSP */ +#define CS42L73_XSPAIPAA 0x3D /* XSP Left Mixer: Left */ +#define CS42L73_XSPBIPBA 0x3E /* XSP Rt. Mixer: Right */ +#define CS42L73_XSPAXSPAA 0x3F /* XSP Left Mixer: XSP L */ +#define CS42L73_XSPBXSPBA 0x40 /* XSP Rt. Mixer: XSP R */ +#define CS42L73_XSPAASPAA 0x41 /* XSP Left Mixer: ASP L */ +#define CS42L73_XSPAASPBA 0x42 /* XSP Rt. Mixer: ASP R */ +#define CS42L73_XSPAVSPMA 0x43 /* XSP Left Mixer: VSP */ +#define CS42L73_XSPBVSPMA 0x44 /* XSP Rt. Mixer: VSP */ +#define CS42L73_ASPAIPAA 0x45 /* ASP Left Mixer: Left */ +#define CS42L73_ASPBIPBA 0x46 /* ASP Rt. Mixer: Right */ +#define CS42L73_ASPAXSPAA 0x47 /* ASP Left Mixer: XSP L */ +#define CS42L73_ASPBXSPBA 0x48 /* ASP Rt. Mixer: XSP R */ +#define CS42L73_ASPAASPAA 0x49 /* ASP Left Mixer: ASP L */ +#define CS42L73_ASPBASPBA 0x4A /* ASP Rt. Mixer: ASP R */ +#define CS42L73_ASPAVSPMA 0x4B /* ASP Left Mixer: VSP */ +#define CS42L73_ASPBVSPMA 0x4C /* ASP Rt. Mixer: VSP */ +#define CS42L73_VSPAIPAA 0x4D /* VSP Left Mixer: Left */ +#define CS42L73_VSPBIPBA 0x4E /* VSP Rt. Mixer: Right */ +#define CS42L73_VSPAXSPAA 0x4F /* VSP Left Mixer: XSP L */ +#define CS42L73_VSPBXSPBA 0x50 /* VSP Rt. Mixer: XSP R */ +#define CS42L73_VSPAASPAA 0x51 /* VSP Left Mixer: ASP Left */ +#define CS42L73_VSPBASPBA 0x52 /* VSP Rt. Mixer: ASP Right */ +#define CS42L73_VSPAVSPMA 0x53 /* VSP Left Mixer: VSP */ +#define CS42L73_VSPBVSPMA 0x54 /* VSP Rt. Mixer: VSP */ +#define CS42L73_MMIXCTL 0x55 /* Mono Mixer Controls. */ +#define CS42L73_SPKMIPMA 0x56 /* SPK Mono Mixer: In. Path */ +#define CS42L73_SPKMXSPA 0x57 /* SPK Mono Mixer: XSP Mono/L/R Att. */ +#define CS42L73_SPKMASPA 0x58 /* SPK Mono Mixer: ASP Mono/L/R Att. */ +#define CS42L73_SPKMVSPMA 0x59 /* SPK Mono Mixer: VSP Mono Atten. */ +#define CS42L73_ESLMIPMA 0x5A /* Ear/SpLO Mono Mixer: */ +#define CS42L73_ESLMXSPA 0x5B /* Ear/SpLO Mono Mixer: XSP */ +#define CS42L73_ESLMASPA 0x5C /* Ear/SpLO Mono Mixer: ASP */ +#define CS42L73_ESLMVSPMA 0x5D /* Ear/SpLO Mono Mixer: VSP */ +#define CS42L73_IM1 0x5E /* Interrupt Mask 1. */ +#define CS42L73_IM2 0x5F /* Interrupt Mask 2. */ +#define CS42L73_IS1 0x60 /* Interrupt Status 1 [RO]. */ +#define CS42L73_IS2 0x61 /* Interrupt Status 2 [RO]. */ + +/* Bitfield Definitions */ + +/* CS42L73_PWRCTL1 */ +#define PDN_ADCB (1 << 7) +#define PDN_DMICB (1 << 6) +#define PDN_ADCA (1 << 5) +#define PDN_DMICA (1 << 4) +#define PDN_LDO (1 << 2) +#define DISCHG_FILT (1 << 1) +#define PDN (1 << 0) + +/* CS42L73_PWRCTL2 */ +#define PDN_MIC2_BIAS (1 << 7) +#define PDN_MIC1_BIAS (1 << 6) +#define PDN_VSP (1 << 4) +#define PDN_ASP_SDOUT (1 << 3) +#define PDN_ASP_SDIN (1 << 2) +#define PDN_XSP_SDOUT (1 << 1) +#define PDN_XSP_SDIN (1 << 0) + +/* CS42L73_PWRCTL3 */ +#define PDN_THMS (1 << 5) +#define PDN_SPKLO (1 << 4) +#define PDN_EAR (1 << 3) +#define PDN_SPK (1 << 2) +#define PDN_LO (1 << 1) +#define PDN_HP (1 << 0) + +/* Thermal Overload Detect. Requires interrupt ... */ +#define THMOVLD_150C 0 +#define THMOVLD_132C 1 +#define THMOVLD_115C 2 +#define THMOVLD_098C 3 + + +/* CS42L73_ASPC, CS42L73_XSPC, CS42L73_VSPC */ +#define SP_3ST (1 << 7) +#define SPDIF_I2S 0 +#define SPDIF_PCM (1 << 6) +#define PCM_MODE0 0 +#define PCM_MODE1 1 +#define PCM_MODE2 2 +#define PCM_BO_MSBLSB 0 +#define PCM_BO_LSBMSB 1 +#define MCK_SCLK_64FS 0 +#define MCK_SCLK_MCLK 2 +#define MCK_SCLK_PREMCLK 3 + +/* CS42L73_xSPMMCC */ +#define MS_MASTER (1 << 7) + + +/* CS42L73_DMMCC */ +#define MCLKDIS (1 << 0) +#define MCLKSEL_MCLK2 (1 << 4) +#define MCLKSEL_MCLK1 (0 << 4) + +/* CS42L73 MCLK derived from MCLK1 or MCLK2 */ +#define CS42L73_CLKID_MCLK1 0 +#define CS42L73_CLKID_MCLK2 1 + +#define CS42L73_MCLKXDIV 0 +#define CS42L73_MMCCDIV 1 + +#define CS42L73_XSP 0 +#define CS42L73_ASP 1 +#define CS42L73_VSP 2 + +/* IS1, IM1 */ +#define MIC2_SDET (1 << 6) +#define THMOVLD (1 << 4) +#define DIGMIXOVFL (1 << 3) +#define IPBOVFL (1 << 1) +#define IPAOVFL (1 << 0) + +/* Analog Softramp */ +#define ANLGOSFT (1 << 0) + +/* HP A/B Analog Mute */ +#define HPA_MUTE (1 << 7) +/* LO A/B Analog Mute */ +#define LOA_MUTE (1 << 7) +/* Digital Mute */ +#define HLAD_MUTE (1 << 0) +#define HLBD_MUTE (1 << 1) +#define SPKD_MUTE (1 << 2) +#define ESLD_MUTE (1 << 3) + +/* Misc defines for codec */ +#define CS42L73_RESET_GPIO 143 + +#define CS42L73_DEVID 0x00042A73 +#define CS42L73_MCLKX_MIN 5644800 +#define CS42L73_MCLKX_MAX 38400000 + +#define CS42L73_SPC(id) (CS42L73_XSPC + (id << 1)) +#define CS42L73_MMCC(id) (CS42L73_XSPMMCC + (id << 1)) +#define CS42L73_SPFS(id) ((id == CS42L73_ASP) ? CS42L73_ASPC : CS42L73_VXSPFS) + +#endif /* __CS42L73_H__ */
On Thu, Oct 06, 2011 at 11:19:05AM -0500, Brian Austin wrote:
- {"DAC1 Mixer", "Headphone Switch", "XSPIN"},
- {"DAC1 Mixer", "LineOut Switch", "XSPIN"},
- {"DAC2 Mixer", "Speakerphone Switch", "XSPIN"},
- {"DAC2 Mixer", "Speakerphone LineOut Switch", "XSPIN"},
- {"DAC2 Mixer", "EarSpeaker Switch", "XSPIN"},
- {"DAC1 Mixer", "Headphone Switch", "ASPIN"},
- {"DAC1 Mixer", "LineOut Switch", "ASPIN"},
- {"DAC2 Mixer", "Speakerphone Switch", "ASPIN"},
- {"DAC2 Mixer", "Speakerphone LineOut Switch", "ASPIN"},
- {"DAC2 Mixer", "EarSpeaker Switch", "ASPIN"},
- {"DAC1 Mixer", "Headphone Switch", "VSPIN"},
- {"DAC1 Mixer", "LineOut Switch", "VSPIN"},
- {"DAC2 Mixer", "Speakerphone Switch", "VSPIN"},
- {"DAC2 Mixer", "Speakerphone LineOut Switch", "VSPIN"},
- {"DAC2 Mixer", "EarSpeaker Switch", "VSPIN"},
This all looks totally wrong and I'm surprised the driver loads without errors. There's two issues:
- All the controls here are for the input to the DAC mixers but their names are for physical outputs. I'd expect the switches to name inputs (for example, "DAC1 Mixer VSPIN Switch"). What's actually going on here?
- You've got three copies of each of the controls for each mixer - I'm surprsied this loads without the core complaining that it can't create controls with duplicate names.
On Oct 6, 2011, at 1:57 PM, Mark Brown wrote:
On Thu, Oct 06, 2011 at 11:19:05AM -0500, Brian Austin wrote:
- {"DAC1 Mixer", "Headphone Switch", "XSPIN"},
- {"DAC1 Mixer", "LineOut Switch", "XSPIN"},
- {"DAC2 Mixer", "Speakerphone Switch", "XSPIN"},
- {"DAC2 Mixer", "Speakerphone LineOut Switch", "XSPIN"},
- {"DAC2 Mixer", "EarSpeaker Switch", "XSPIN"},
- {"DAC1 Mixer", "Headphone Switch", "ASPIN"},
- {"DAC1 Mixer", "LineOut Switch", "ASPIN"},
- {"DAC2 Mixer", "Speakerphone Switch", "ASPIN"},
- {"DAC2 Mixer", "Speakerphone LineOut Switch", "ASPIN"},
- {"DAC2 Mixer", "EarSpeaker Switch", "ASPIN"},
- {"DAC1 Mixer", "Headphone Switch", "VSPIN"},
- {"DAC1 Mixer", "LineOut Switch", "VSPIN"},
- {"DAC2 Mixer", "Speakerphone Switch", "VSPIN"},
- {"DAC2 Mixer", "Speakerphone LineOut Switch", "VSPIN"},
- {"DAC2 Mixer", "EarSpeaker Switch", "VSPIN"},
This all looks totally wrong and I'm surprised the driver loads without errors. There's two issues:
All the controls here are for the input to the DAC mixers but their names are for physical outputs. I'd expect the switches to name inputs (for example, "DAC1 Mixer VSPIN Switch"). What's actually going on here?
You've got three copies of each of the controls for each mixer - I'm surprsied this loads without the core complaining that it can't create controls with duplicate names.
This works fine on my setup with a beagleboard.
There are 3 routes for the HP/LO/SPK/SPKLO/EAR. What I am trying to accomplish is the following. Allow a stream from any serial port to go to any output.
On Thu, Oct 06, 2011 at 07:15:22PM +0000, Austin, Brian wrote:
On Oct 6, 2011, at 1:57 PM, Mark Brown wrote:
On Thu, Oct 06, 2011 at 11:19:05AM -0500, Brian Austin wrote:
- {"DAC1 Mixer", "Headphone Switch", "XSPIN"},
- {"DAC1 Mixer", "Headphone Switch", "ASPIN"},
- {"DAC1 Mixer", "Headphone Switch", "VSPIN"},
This all looks totally wrong and I'm surprised the driver loads without errors. There's two issues:
- All the controls here are for the input to the DAC mixers but their names are for physical outputs. I'd expect the switches to name inputs (for example, "DAC1 Mixer VSPIN Switch"). What's actually going on here?
- You've got three copies of each of the controls for each mixer - I'm surprsied this loads without the core complaining that it can't create controls with duplicate names.
This works fine on my setup with a beagleboard.
I'm frankly astonished. What are these controls called?
There are 3 routes for the HP/LO/SPK/SPKLO/EAR. What I am trying to accomplish is the following. Allow a stream from any serial port to go to any output.
That's perfectly reasonable but what you've written doesn't seem to do that. The DAPM routes are in the form:
{ destination, control, source }
so what the above example says is that audio is routed from XSPIN to DAC1 if the DAC1 Headphone switch is turned on.
On Fri, 2011-10-07 at 11:34 +0100, Mark Brown wrote:
On Thu, Oct 06, 2011 at 07:15:22PM +0000, Austin, Brian wrote:
On Oct 6, 2011, at 1:57 PM, Mark Brown wrote:
On Thu, Oct 06, 2011 at 11:19:05AM -0500, Brian Austin wrote:
- {"DAC1 Mixer", "Headphone Switch", "XSPIN"},
- {"DAC1 Mixer", "Headphone Switch", "ASPIN"},
- {"DAC1 Mixer", "Headphone Switch", "VSPIN"},
This all looks totally wrong and I'm surprised the driver loads without errors. There's two issues:
- All the controls here are for the input to the DAC mixers but their names are for physical outputs. I'd expect the switches to name inputs (for example, "DAC1 Mixer VSPIN Switch"). What's actually going on here?
- You've got three copies of each of the controls for each mixer - I'm surprsied this loads without the core complaining that it can't create controls with duplicate names.
This works fine on my setup with a beagleboard.
I'm frankly astonished. What are these controls called?
There are 3 routes for the HP/LO/SPK/SPKLO/EAR. What I am trying to accomplish is the following. Allow a stream from any serial port to go to any output.
That's perfectly reasonable but what you've written doesn't seem to do that. The DAPM routes are in the form:
{ destination, control, source }so what the above example says is that audio is routed from XSPIN to DAC1 if the DAC1 Headphone switch is turned on.
See below:
+static const struct snd_kcontrol_new dac1_output_mixer[] = { + SOC_DAPM_SINGLE("Headphone Switch", CS42L73_PWRCTL3, 0, 1, 1), + SOC_DAPM_SINGLE("LineOut Switch", CS42L73_PWRCTL3, 1, 1, 1), +}; + +static const struct snd_kcontrol_new dac2_output_mixer[] = { + SOC_DAPM_SINGLE("Speakerphone Switch", CS42L73_PWRCTL3, 2, 1, 1), + SOC_DAPM_SINGLE("Speakerphone LineOut Switch", CS42L73_PWRCTL3, + 4, 1, 1), + SOC_DAPM_SINGLE("EarSpeaker Switch", CS42L73_PWRCTL3, 3, 1, 1), +}; +
I think naming should be slightly better :)
On Fri, Oct 07, 2011 at 04:37:16PM +0530, Vinod Koul wrote:
See below:
+static const struct snd_kcontrol_new dac1_output_mixer[] = {
SOC_DAPM_SINGLE("Headphone Switch", CS42L73_PWRCTL3, 0, 1, 1),SOC_DAPM_SINGLE("LineOut Switch", CS42L73_PWRCTL3, 1, 1, 1),+};
+static const struct snd_kcontrol_new dac2_output_mixer[] = {
SOC_DAPM_SINGLE("Speakerphone Switch", CS42L73_PWRCTL3, 2, 1,1),
SOC_DAPM_SINGLE("Speakerphone LineOut Switch", CS42L73_PWRCTL3,4, 1, 1),SOC_DAPM_SINGLE("EarSpeaker Switch", CS42L73_PWRCTL3, 3, 1, 1),+};
I think naming should be slightly better :)
I'm not sure, it might be more than the naming. The fact that we're in PWRCTL registers could be confusing hardware design, or the structure of the hardware may not actually reflect what the driver is doing.
On Oct 7, 2011, at 6:42 AM, Mark Brown wrote:
On Fri, Oct 07, 2011 at 04:37:16PM +0530, Vinod Koul wrote:
See below:
+static const struct snd_kcontrol_new dac1_output_mixer[] = {
SOC_DAPM_SINGLE("Headphone Switch", CS42L73_PWRCTL3, 0, 1, 1),SOC_DAPM_SINGLE("LineOut Switch", CS42L73_PWRCTL3, 1, 1, 1),+};
+static const struct snd_kcontrol_new dac2_output_mixer[] = {
SOC_DAPM_SINGLE("Speakerphone Switch", CS42L73_PWRCTL3, 2, 1,1),
SOC_DAPM_SINGLE("Speakerphone LineOut Switch", CS42L73_PWRCTL3,4, 1, 1),SOC_DAPM_SINGLE("EarSpeaker Switch", CS42L73_PWRCTL3, 3, 1, 1),+};
I think naming should be slightly better :)
I'm not sure, it might be more than the naming. The fact that we're in PWRCTL registers could be confusing hardware design, or the structure of the hardware may not actually reflect what the driver is doing.
The way to enable the output is to enable the output pin. I have no control in SW over the DAC's. The pins are power controlled of course. I know it looks strange, but without using a machine driver to enable/disable the pins for a certain route, this seems to be the best solution.
I am of course open to other ideas :)
On Fri, Oct 07, 2011 at 02:05:57PM +0000, Austin, Brian wrote:
The way to enable the output is to enable the output pin. I have no control in SW over the DAC's. The pins are power controlled of course. I know it looks strange, but without using a machine driver to enable/disable the pins for a certain route, this seems to be the best solution.
Right, which is why I'm saying this routing makes no sense. You've got code which has a mixer mixing input signals down into the DAC but what you've actually got is a single output from the DAC which can then be sent to the outputs separately. You should be able to draw a graph of the audio routes through the CODEC which directly corresponds to the DAPM graph.
On Fri, 2011-10-07 at 14:05 +0000, Austin, Brian wrote:
On Oct 7, 2011, at 6:42 AM, Mark Brown wrote:
On Fri, Oct 07, 2011 at 04:37:16PM +0530, Vinod Koul wrote:
See below:
+static const struct snd_kcontrol_new dac1_output_mixer[] = {
SOC_DAPM_SINGLE("Headphone Switch", CS42L73_PWRCTL3, 0, 1, 1),SOC_DAPM_SINGLE("LineOut Switch", CS42L73_PWRCTL3, 1, 1, 1),+};
+static const struct snd_kcontrol_new dac2_output_mixer[] = {
SOC_DAPM_SINGLE("Speakerphone Switch", CS42L73_PWRCTL3, 2, 1,1),
SOC_DAPM_SINGLE("Speakerphone LineOut Switch", CS42L73_PWRCTL3,4, 1, 1),SOC_DAPM_SINGLE("EarSpeaker Switch", CS42L73_PWRCTL3, 3, 1, 1),+};
I think naming should be slightly better :)
I'm not sure, it might be more than the naming. The fact that we're in PWRCTL registers could be confusing hardware design, or the structure of the hardware may not actually reflect what the driver is doing.
The way to enable the output is to enable the output pin. I have no control in SW over the DAC's. The pins are power controlled of course. I know it looks strange, but without using a machine driver to enable/disable the pins for a certain route, this seems to be the best solution.
I am of course open to other ideas :)
What i saw in the spec is that this codec has three PCM ports and each one can be input to the DACs. The DAC output goes to output pins and in some cases shared (like DAC2 for ep/spkr). For output you have amp switches at end, so we can control where output is going as well. So we can have is APCM -> Mixer's -> DAC -> out switches -> pins All this is codec domain so why should machine driver be bothered about this stuff. Anything external to codec should be handled in machine driver, otherwise we end up creating same code in all machine drivers which use this codec.
On Oct 7, 2011, at 9:29 AM, Vinod Koul wrote:
On Fri, 2011-10-07 at 14:05 +0000, Austin, Brian wrote:
On Oct 7, 2011, at 6:42 AM, Mark Brown wrote:
On Fri, Oct 07, 2011 at 04:37:16PM +0530, Vinod Koul wrote:
See below:
+static const struct snd_kcontrol_new dac1_output_mixer[] = {
SOC_DAPM_SINGLE("Headphone Switch", CS42L73_PWRCTL3, 0, 1, 1),SOC_DAPM_SINGLE("LineOut Switch", CS42L73_PWRCTL3, 1, 1, 1),+};
+static const struct snd_kcontrol_new dac2_output_mixer[] = {
SOC_DAPM_SINGLE("Speakerphone Switch", CS42L73_PWRCTL3, 2, 1,1),
SOC_DAPM_SINGLE("Speakerphone LineOut Switch", CS42L73_PWRCTL3,4, 1, 1),SOC_DAPM_SINGLE("EarSpeaker Switch", CS42L73_PWRCTL3, 3, 1, 1),+};
I think naming should be slightly better :)
I'm not sure, it might be more than the naming. The fact that we're in PWRCTL registers could be confusing hardware design, or the structure of the hardware may not actually reflect what the driver is doing.
The way to enable the output is to enable the output pin. I have no control in SW over the DAC's. The pins are power controlled of course. I know it looks strange, but without using a machine driver to enable/disable the pins for a certain route, this seems to be the best solution.
I am of course open to other ideas :)
What i saw in the spec is that this codec has three PCM ports and each one can be input to the DACs. The DAC output goes to output pins and in some cases shared (like DAC2 for ep/spkr). For output you have amp switches at end, so we can control where output is going as well. So we can have is APCM -> Mixer's -> DAC -> out switches -> pins All this is codec domain so why should machine driver be bothered about this stuff. Anything external to codec should be handled in machine driver, otherwise we end up creating same code in all machine drivers which use this codec.
So more of a "mixer" for the PCM's to the DAC's and then the switches for the output control. I'll try that
Thanks
On Oct 7, 2011, at 9:29 AM, Vinod Koul wrote:
What i saw in the spec is that this codec has three PCM ports and each one can be input to the DACs. The DAC output goes to output pins and in some cases shared (like DAC2 for ep/spkr).
This is controlled through the 'HL-XSP/ASP/VSP' mixer control for Headphone/LineOut.
For output you have amp switches at end, so we can control where output is going as well. So we can have is APCM -> Mixer's -> DAC -> out switches -> pins
XPCM (hw:0,1) -> Mixers (HL-XSP) -> DAC(NULL) -> HP Amp (DAC1 Mixer Headphone) - > HPOUT.
All this is codec domain so why should machine driver be bothered about this stuff. Anything external to codec should be handled in machine driver, otherwise we end up creating same code in all machine drivers which use this codec.
-- ~Vinod
With the code I have I can do this and it works as intended, at least in my mind. I can enable and disable the outputs to send the stream to wherever I want and only power what is being used.
On Oct 12, 2011, at 4:37 PM, Austin, Brian wrote:
On Oct 7, 2011, at 9:29 AM, Vinod Koul wrote:
What i saw in the spec is that this codec has three PCM ports and each one can be input to the DACs. The DAC output goes to output pins and in some cases shared (like DAC2 for ep/spkr).
This is controlled through the 'HL-XSP/ASP/VSP' mixer control for Headphone/LineOut.
For output you have amp switches at end, so we can control where output is going as well. So we can have is APCM -> Mixer's -> DAC -> out switches -> pins
XPCM (hw:0,1) -> Mixers (HL-XSP) -> DAC(NULL) -> HP Amp (DAC1 Mixer Headphone) - > HPOUT.
All this is codec domain so why should machine driver be bothered about this stuff. Anything external to codec should be handled in machine driver, otherwise we end up creating same code in all machine drivers which use this codec.
-- ~Vinod
With the code I have I can do this and it works as intended, at least in my mind. I can enable and disable the outputs to send the stream to wherever I want and only power what is being used.
I see the problem now…
If you have multiple DAI's on the same serial port you get duplicate entries. I'll fix it. Sorry
Brian
On Oct 12, 2011, at 4:50 PM, Austin, Brian wrote:
On Oct 12, 2011, at 4:37 PM, Austin, Brian wrote:
On Oct 7, 2011, at 9:29 AM, Vinod Koul wrote:
What i saw in the spec is that this codec has three PCM ports and each one can be input to the DACs. The DAC output goes to output pins and in some cases shared (like DAC2 for ep/spkr).
This is controlled through the 'HL-XSP/ASP/VSP' mixer control for Headphone/LineOut.
For output you have amp switches at end, so we can control where output is going as well. So we can have is APCM -> Mixer's -> DAC -> out switches -> pins
XPCM (hw:0,1) -> Mixers (HL-XSP) -> DAC(NULL) -> HP Amp (DAC1 Mixer Headphone) - > HPOUT.
All this is codec domain so why should machine driver be bothered about this stuff. Anything external to codec should be handled in machine driver, otherwise we end up creating same code in all machine drivers which use this codec.
-- ~Vinod
With the code I have I can do this and it works as intended, at least in my mind. I can enable and disable the outputs to send the stream to wherever I want and only power what is being used.
I see the problem now…
If you have multiple DAI's on the same serial port you get duplicate entries. I'll fix it. Sorry
Brian
Disregard this email please.
Sorry for the confusion
On Oct 7, 2011, at 9:29 AM, Vinod Koul wrote:
On Fri, 2011-10-07 at 14:05 +0000, Austin, Brian wrote:
On Oct 7, 2011, at 6:42 AM, Mark Brown wrote:
On Fri, Oct 07, 2011 at 04:37:16PM +0530, Vinod Koul wrote:
See below:
+static const struct snd_kcontrol_new dac1_output_mixer[] = {
SOC_DAPM_SINGLE("Headphone Switch", CS42L73_PWRCTL3, 0, 1, 1),SOC_DAPM_SINGLE("LineOut Switch", CS42L73_PWRCTL3, 1, 1, 1),+};
+static const struct snd_kcontrol_new dac2_output_mixer[] = {
SOC_DAPM_SINGLE("Speakerphone Switch", CS42L73_PWRCTL3, 2, 1,1),
SOC_DAPM_SINGLE("Speakerphone LineOut Switch", CS42L73_PWRCTL3,4, 1, 1),SOC_DAPM_SINGLE("EarSpeaker Switch", CS42L73_PWRCTL3, 3, 1, 1),+};
I think naming should be slightly better :)
I'm not sure, it might be more than the naming. The fact that we're in PWRCTL registers could be confusing hardware design, or the structure of the hardware may not actually reflect what the driver is doing.
The way to enable the output is to enable the output pin. I have no control in SW over the DAC's. The pins are power controlled of course. I know it looks strange, but without using a machine driver to enable/disable the pins for a certain route, this seems to be the best solution.
I am of course open to other ideas :)
What i saw in the spec is that this codec has three PCM ports and each one can be input to the DACs. The DAC output goes to output pins and in some cases shared (like DAC2 for ep/spkr). For output you have amp switches at end, so we can control where output is going as well. So we can have is APCM -> Mixer's -> DAC -> out switches -> pins All this is codec domain so why should machine driver be bothered about this stuff. Anything external to codec should be handled in machine driver, otherwise we end up creating same code in all machine drivers which use this codec.
-- ~Vinod
I have changed the DAPM widgets for the Amp's to be Switches. They are on or off based on the mixer control. so now it looks as follows.
static const struct snd_kcontrol_new hp_amp_ctl = SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 0, 1, 1);
static const struct snd_kcontrol_new lo_amp_ctl = SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 1, 1, 1);
static const struct snd_kcontrol_new spk_amp_ctl = SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 2, 1, 1);
static const struct snd_kcontrol_new spklo_amp_ctl = SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 4, 1, 1);
static const struct snd_kcontrol_new ear_amp_ctl = SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 3, 1, 1);
SND_SOC_DAPM_SWITCH("HP Amp", CS42L73_PWRCTL3, 0, 1, &hp_amp_ctl), SND_SOC_DAPM_SWITCH("LO Amp", CS42L73_PWRCTL3, 1, 1, &lo_amp_ctl), SND_SOC_DAPM_SWITCH("SPK Amp", CS42L73_PWRCTL3, 2, 1, &spk_amp_ctl), SND_SOC_DAPM_SWITCH("EAR Amp", CS42L73_PWRCTL3, 3, 1, &ear_amp_ctl), SND_SOC_DAPM_SWITCH("SPKLO Amp", CS42L73_PWRCTL3, 4, 1, &spklo_amp_ctl),
{"HP Amp", "Switch", "DAC1"}, {"LO Amp", "Switch", "DAC1"}, {"SPK Amp", "Switch", "DAC2"}, {"SPKLO Amp", "Switch", "DAC2"}, {"EAR Amp", "Switch", "DAC2"},
This allows me to route the streams from any PCM to any output. IF this is OK I would like to resubmit.
Thanks, Brian
On Mon, Oct 17, 2011 at 07:11:39PM +0000, Austin, Brian wrote:
{"HP Amp", "Switch", "DAC1"}, {"LO Amp", "Switch", "DAC1"}, {"SPK Amp", "Switch", "DAC2"}, {"SPKLO Amp", "Switch", "DAC2"}, {"EAR Amp", "Switch", "DAC2"},
This allows me to route the streams from any PCM to any output. IF this is OK I would like to resubmit.
This looks reasonable.
[Please fix your mail client to word wrap within paragraphs.]
On Thu, 2011-10-06 at 19:57 +0100, Mark Brown wrote:
On Thu, Oct 06, 2011 at 11:19:05AM -0500, Brian Austin wrote:
- {"DAC1 Mixer", "Headphone Switch", "XSPIN"},
- {"DAC1 Mixer", "LineOut Switch", "XSPIN"},
- {"DAC2 Mixer", "Speakerphone Switch", "XSPIN"},
- {"DAC2 Mixer", "Speakerphone LineOut Switch", "XSPIN"},
- {"DAC2 Mixer", "EarSpeaker Switch", "XSPIN"},
- {"DAC1 Mixer", "Headphone Switch", "ASPIN"},
- {"DAC1 Mixer", "LineOut Switch", "ASPIN"},
- {"DAC2 Mixer", "Speakerphone Switch", "ASPIN"},
- {"DAC2 Mixer", "Speakerphone LineOut Switch", "ASPIN"},
- {"DAC2 Mixer", "EarSpeaker Switch", "ASPIN"},
- {"DAC1 Mixer", "Headphone Switch", "VSPIN"},
- {"DAC1 Mixer", "LineOut Switch", "VSPIN"},
- {"DAC2 Mixer", "Speakerphone Switch", "VSPIN"},
- {"DAC2 Mixer", "Speakerphone LineOut Switch", "VSPIN"},
- {"DAC2 Mixer", "EarSpeaker Switch", "VSPIN"},
This all looks totally wrong and I'm surprised the driver loads without errors. There's two issues:
All the controls here are for the input to the DAC mixers but their names are for physical outputs. I'd expect the switches to name inputs (for example, "DAC1 Mixer VSPIN Switch"). What's actually going on here?
You've got three copies of each of the controls for each mixer - I'm surprsied this loads without the core complaining that it can't create controls with duplicate names.
This codec has two DACs, three PCM ports. Each DAC has a mixer to allow any port to be routed to it
I recommended this to be done as it helps DAPM to power off unused paths.
On Thu, Oct 6, 2011 at 11:19 AM, Brian Austin brian.austin@cirrus.com wrote:
+static const u8 cs42l73_reg[] = { +/*0*/ 0x00, 0x42, 0xA7, 0x30, +/*4*/ 0x00, 0x00, 0xF1, 0xDF, +/*8*/ 0x3F, 0x57, 0x53, 0x00, +/*C*/ 0x00, 0x15, 0x00, 0x15, +/*A*/ 0x00, 0x15, 0x00, 0x06, +/*E*/ 0x00, 0x00, 0x00, 0x00, +/*18*/ 0x00, 0x00, 0x00, 0x00, +/*1C*/ 0x00, 0x00, 0x00, 0x00, +/*20*/ 0x00, 0x00, 0x00, 0x00, +/*24*/ 0x00, 0x00, 0x00, 0x7F, +/*28*/ 0x00, 0x00, 0x3F, 0x00, +/*2C*/ 0x00, 0x3F, 0x00, 0x00, +/*30*/ 0x3F, 0x00, 0x00, 0x00, +/*34*/ 0x18, 0x3F, 0x3F, 0x3F, +/*38*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*3C*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*40*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*44*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*48*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*4C*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*50*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*54*/ 0x3F, 0xAA, 0x3F, 0x3F, +/*58*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*5C*/ 0x3F, 0x3F, 0x00, 0x00, +/*60*/ 0x00, 0x00 +};
The comments should go on the right side, not the left.
+static struct cs42l73_mclkx_div cs42l73_mclkx_coeffs[] = {
- {5644800, 1, 0}, /* 5644800 */
- {6000000, 1, 0}, /* 6000000 */
- {6144000, 1, 0}, /* 6144000 */
- {11289600, 2, 2}, /* 5644800 */
- {12288000, 2, 2}, /* 6144000 */
- {12000000, 2, 2}, /* 6000000 */
- {13000000, 2, 2}, /* 6500000 */
- {19200000, 3, 3}, /* 6400000 */
- {24000000, 4, 4}, /* 6000000 */
- {26000000, 4, 4}, /* 6500000 */
- {38400000, 6, 5} /* 6400000 */
What do these comments mean?
In fact, I think there should be more comments in this file overall. For example, almost none of the functions are commented.
+/* I2C Registers */ +/* I2C Address: 1001010[R/W] - 10010100 = 0x94(Write); 10010101 = 0x95(Read) */
This chip has separate I2C addresses for reading and writing?
On Fri, Oct 07, 2011 at 02:13:32AM +0000, Tabi Timur-B04825 wrote:
+/* I2C Registers */ +/* I2C Address: 1001010[R/W] - 10010100 = 0x94(Write); 10010101 = 0x95(Read) */
This chip has separate I2C addresses for reading and writing?
This is standard - the I2C address can be quoted in either 8 bit form (where the lowest bit is a read write flag) or 7 bit form. The confusing thing here is that the Linux code uses the 7 bit form exclusively.
participants (6)
-
Austin, Brian -
Brian Austin
-
Mark Brown -
Tabi Timur-B04825 -
Vinod Koul -
Vinod Koul