November 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] wm8753: Skip noop reconfiguration of DAI mode
by Timo Juhani Lindfors 17 Nov '11

17 Nov '11

This patch makes it possible to set DAI mode to its currently applied value even if codec is active. This is necessary to allow aplay -t raw -r 44100 -f S16_LE -c 2 < /dev/urandom & alsactl store -f backup.state alsactl restore -f backup.state to work without returning errors. This patch is based on a patch sent by Klaus Kurzmann <mok(a)fluxnetz.de>. Signed-off-by: Timo Juhani Lindfors <timo.lindfors(a)iki.fi> --- sound/soc/codecs/wm8753.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index a950471..3a629d0 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -190,6 +190,9 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol, struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); u16 ioctl; + if (wm8753->dai_func == ucontrol->value.integer.value[0]) + return 0; + if (codec->active) return -EBUSY; -- 1.7.2.5

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[alsa-devel] [PATCH v7] ASoC:Add support for CS42L73 codec support
by Brian Austin 17 Nov '11

17 Nov '11

Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> Signed-off-by: Georgi Vlaev <joe(a)nucleusys.com> This patch adds support for the Cirrus Logic CS42L73 low power stereo codec --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs42l73.c | 1457 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs42l73.h | 227 +++++++ 4 files changed, 1690 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 684cc15..686f45a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -29,6 +29,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ALC5632 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 @@ -178,6 +179,9 @@ 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 af64905..62b01e4 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 @@ -117,6 +118,7 @@ obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.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..6fe259a --- /dev/null +++ b/sound/soc/codecs/cs42l73.c @@ -0,0 +1,1457 @@ +/* + * cs42l73.c -- CS42L73 ALSA Soc Audio driver + * + * Copyright 2011 Cirrus Logic, Inc. + * + * Authors: Georgi Vlaev, Nucleus Systems Ltd, <joe(a)nucleusys.com> + * Brian Austin, Cirrus Logic Inc, <brian.austin(a)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/regmap.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 { + struct sp_config config[3]; + struct regmap *regmap; + u32 sysclk; + u8 mclksel; + u32 mclk; +}; + +struct reg_default cs42l73_reg_defaults[] = { + { 1, 0x42 }, /* r01 - Device ID A&B */ + { 2, 0xA7 }, /* r02 - Device ID C&D */ + { 3, 0x30 }, /* r03 - Device ID E */ + { 6, 0xF1 }, /* r06 - Power Ctl 1 */ + { 7, 0xDF }, /* r07 - Power Ctl 2 */ + { 8, 0x3F }, /* r08 - Power Ctl 3 */ + { 9, 0x50 }, /* r09 - Charge Pump Freq */ + { 10, 0x53 }, /* r0A - Output Load MicBias Short Detect */ + { 11, 0x00 }, /* r0B - DMIC Master Clock Ctl */ + { 12, 0x00 }, /* r0C - Aux PCM Ctl */ + { 13, 0x15 }, /* r0D - Aux PCM Master Clock Ctl */ + { 14, 0x00 }, /* r0E - Audio PCM Ctl */ + { 15, 0x15 }, /* r0F - Audio PCM Master Clock Ctl */ + { 16, 0x00 }, /* r10 - Voice PCM Ctl */ + { 17, 0x15 }, /* r11 - Voice PCM Master Clock Ctl */ + { 18, 0x00 }, /* r12 - Voice/Aux Sample Rate */ + { 19, 0x06 }, /* r13 - Misc I/O Path Ctl */ + { 20, 0x00 }, /* r14 - ADC Input Path Ctl */ + { 21, 0x00 }, /* r15 - MICA Preamp, PGA Volume */ + { 22, 0x00 }, /* r16 - MICB Preamp, PGA Volume */ + { 23, 0x00 }, /* r17 - Input Path A Digital Volume */ + { 24, 0x00 }, /* r18 - Input Path B Digital Volume */ + { 25, 0x00 }, /* r19 - Playback Digital Ctl */ + { 26, 0x00 }, /* r1A - HP/LO Left Digital Volume */ + { 27, 0x00 }, /* r1B - HP/LO Right Digital Volume */ + { 28, 0x00 }, /* r1C - Speakerphone Digital Volume */ + { 29, 0x00 }, /* r1D - Ear/SPKLO Digital Volume */ + { 30, 0x00 }, /* r1E - HP Left Analog Volume */ + { 31, 0x00 }, /* r1F - HP Right Analog Volume */ + { 32, 0x00 }, /* r20 - LO Left Analog Volume */ + { 33, 0x00 }, /* r21 - LO Right Analog Volume */ + { 34, 0x00 }, /* r22 - Stereo Input Path Advisory Volume */ + { 35, 0x00 }, /* r23 - Aux PCM Input Advisory Volume */ + { 36, 0x00 }, /* r24 - Audio PCM Input Advisory Volume */ + { 37, 0x00 }, /* r25 - Voice PCM Input Advisory Volume */ + { 38, 0x00 }, /* r26 - Limiter Attack Rate HP/LO */ + { 39, 0x7F }, /* r27 - Limter Ctl, Release Rate HP/LO */ + { 40, 0x00 }, /* r28 - Limter Threshold HP/LO */ + { 41, 0x00 }, /* r29 - Limiter Attack Rate Speakerphone */ + { 42, 0x3F }, /* r2A - Limter Ctl, Release Rate Speakerphone */ + { 43, 0x00 }, /* r2B - Limter Threshold Speakerphone */ + { 44, 0x00 }, /* r2C - Limiter Attack Rate Ear/SPKLO */ + { 45, 0x3F }, /* r2D - Limter Ctl, Release Rate Ear/SPKLO */ + { 46, 0x00 }, /* r2E - Limter Threshold Ear/SPKLO */ + { 47, 0x00 }, /* r2F - ALC Enable, Attack Rate Left/Right */ + { 48, 0x3F }, /* r30 - ALC Release Rate Left/Right */ + { 49, 0x00 }, /* r31 - ALC Threshold Left/Right */ + { 50, 0x00 }, /* r32 - Noise Gate Ctl Left/Right */ + { 51, 0x00 }, /* r33 - ALC/NG Misc Ctl */ + { 52, 0x18 }, /* r34 - Mixer Ctl */ + { 53, 0x3F }, /* r35 - HP/LO Left Mixer Input Path Volume */ + { 54, 0x3F }, /* r36 - HP/LO Right Mixer Input Path Volume */ + { 55, 0x3F }, /* r37 - HP/LO Left Mixer Aux PCM Volume */ + { 56, 0x3F }, /* r38 - HP/LO Right Mixer Aux PCM Volume */ + { 57, 0x3F }, /* r39 - HP/LO Left Mixer Audio PCM Volume */ + { 58, 0x3F }, /* r3A - HP/LO Right Mixer Audio PCM Volume */ + { 59, 0x3F }, /* r3B - HP/LO Left Mixer Voice PCM Mono Volume */ + { 60, 0x3F }, /* r3C - HP/LO Right Mixer Voice PCM Mono Volume */ + { 61, 0x3F }, /* r3D - Aux PCM Left Mixer Input Path Volume */ + { 62, 0x3F }, /* r3E - Aux PCM Right Mixer Input Path Volume */ + { 63, 0x3F }, /* r3F - Aux PCM Left Mixer Volume */ + { 64, 0x3F }, /* r40 - Aux PCM Left Mixer Volume */ + { 65, 0x3F }, /* r41 - Aux PCM Left Mixer Audio PCM L Volume */ + { 66, 0x3F }, /* r42 - Aux PCM Right Mixer Audio PCM R Volume */ + { 67, 0x3F }, /* r43 - Aux PCM Left Mixer Voice PCM Volume */ + { 68, 0x3F }, /* r44 - Aux PCM Right Mixer Voice PCM Volume */ + { 69, 0x3F }, /* r45 - Audio PCM Left Input Path Volume */ + { 70, 0x3F }, /* r46 - Audio PCM Right Input Path Volume */ + { 71, 0x3F }, /* r47 - Audio PCM Left Mixer Aux PCM L Volume */ + { 72, 0x3F }, /* r48 - Audio PCM Right Mixer Aux PCM R Volume */ + { 73, 0x3F }, /* r49 - Audio PCM Left Mixer Volume */ + { 74, 0x3F }, /* r4A - Audio PCM Right Mixer Volume */ + { 75, 0x3F }, /* r4B - Audio PCM Left Mixer Voice PCM Volume */ + { 76, 0x3F }, /* r4C - Audio PCM Right Mixer Voice PCM Volume */ + { 77, 0x3F }, /* r4D - Voice PCM Left Input Path Volume */ + { 78, 0x3F }, /* r4E - Voice PCM Right Input Path Volume */ + { 79, 0x3F }, /* r4F - Voice PCM Left Mixer Aux PCM L Volume */ + { 80, 0x3F }, /* r50 - Voice PCM Right Mixer Aux PCM R Volume */ + { 81, 0x3F }, /* r51 - Voice PCM Left Mixer Audio PCM L Volume */ + { 82, 0x3F }, /* r52 - Voice PCM Right Mixer Audio PCM R Volume */ + { 83, 0x3F }, /* r53 - Voice PCM Left Mixer Voice PCM Volume */ + { 84, 0x3F }, /* r54 - Voice PCM Right Mixer Voice PCM Volume */ + { 85, 0xAA }, /* r55 - Mono Mixer Ctl */ + { 86, 0x3F }, /* r56 - SPK Mono Mixer Input Path Volume */ + { 87, 0x3F }, /* r57 - SPK Mono Mixer Aux PCM Mono/L/R Volume */ + { 88, 0x3F }, /* r58 - SPK Mono Mixer Audio PCM Mono/L/R Volume */ + { 89, 0x3F }, /* r59 - SPK Mono Mixer Voice PCM Mono Volume */ + { 90, 0x3F }, /* r5A - SPKLO Mono Mixer Input Path Mono Volume */ + { 91, 0x3F }, /* r5B - SPKLO Mono Mixer Aux Mono/L/R Volume */ + { 92, 0x3F }, /* r5C - SPKLO Mono Mixer Audio Mono/L/R Volume */ + { 93, 0x3F }, /* r5D - SPKLO Mono Mixer Voice Mono Volume */ + { 94, 0x00 }, /* r5E - Interrupt Mask 1 */ + { 95, 0x00 }, /* r5F - Interrupt Mask 2 */ +}; + +static bool cs42l73_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42L73_IS1: + case CS42L73_IS2: + return true; + default: + return false; + } +} + +static bool cs42l73_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42L73_DEVID_AB: + case CS42L73_DEVID_CD: + case CS42L73_DEVID_E: + case CS42L73_REVID: + case CS42L73_PWRCTL1: + case CS42L73_PWRCTL2: + case CS42L73_PWRCTL3: + case CS42L73_CPFCHC: + case CS42L73_OLMBMSDC: + case CS42L73_DMMCC: + case CS42L73_XSPC: + case CS42L73_XSPMMCC: + case CS42L73_ASPC: + case CS42L73_ASPMMCC: + case CS42L73_VSPC: + case CS42L73_VSPMMCC: + case CS42L73_VXSPFS: + case CS42L73_MIOPC: + case CS42L73_ADCIPC: + case CS42L73_MICAPREPGAAVOL: + case CS42L73_MICBPREPGABVOL: + case CS42L73_IPADVOL: + case CS42L73_IPBDVOL: + case CS42L73_PBDC: + case CS42L73_HLADVOL: + case CS42L73_HLBDVOL: + case CS42L73_SPKDVOL: + case CS42L73_ESLDVOL: + case CS42L73_HPAAVOL: + case CS42L73_HPBAVOL: + case CS42L73_LOAAVOL: + case CS42L73_LOBAVOL: + case CS42L73_STRINV: + case CS42L73_XSPINV: + case CS42L73_ASPINV: + case CS42L73_VSPINV: + case CS42L73_LIMARATEHL: + case CS42L73_LIMRRATEHL: + case CS42L73_LMAXHL: + case CS42L73_LIMARATESPK: + case CS42L73_LIMRRATESPK: + case CS42L73_LMAXSPK: + case CS42L73_LIMARATEESL: + case CS42L73_LIMRRATEESL: + case CS42L73_LMAXESL: + case CS42L73_ALCARATE: + case CS42L73_ALCRRATE: + case CS42L73_ALCMINMAX: + case CS42L73_NGCAB: + case CS42L73_ALCNGMC: + case CS42L73_MIXERCTL: + case CS42L73_HLAIPAA: + case CS42L73_HLBIPBA: + case CS42L73_HLAXSPAA: + case CS42L73_HLBXSPBA: + case CS42L73_HLAASPAA: + case CS42L73_HLBASPBA: + case CS42L73_HLAVSPMA: + case CS42L73_HLBVSPMA: + case CS42L73_XSPAIPAA: + case CS42L73_XSPBIPBA: + case CS42L73_XSPAXSPAA: + case CS42L73_XSPBXSPBA: + case CS42L73_XSPAASPAA: + case CS42L73_XSPAASPBA: + case CS42L73_XSPAVSPMA: + case CS42L73_XSPBVSPMA: + case CS42L73_ASPAIPAA: + case CS42L73_ASPBIPBA: + case CS42L73_ASPAXSPAA: + case CS42L73_ASPBXSPBA: + case CS42L73_ASPAASPAA: + case CS42L73_ASPBASPBA: + case CS42L73_ASPAVSPMA: + case CS42L73_ASPBVSPMA: + case CS42L73_VSPAIPAA: + case CS42L73_VSPBIPBA: + case CS42L73_VSPAXSPAA: + case CS42L73_VSPBXSPBA: + case CS42L73_VSPAASPAA: + case CS42L73_VSPBASPBA: + case CS42L73_VSPAVSPMA: + case CS42L73_VSPBVSPMA: + case CS42L73_MMIXCTL: + case CS42L73_SPKMIPMA: + case CS42L73_SPKMXSPA: + case CS42L73_SPKMASPA: + case CS42L73_SPKMVSPMA: + case CS42L73_ESLMIPMA: + case CS42L73_ESLMXSPA: + case CS42L73_ESLMASPA: + case CS42L73_ESLMVSPMA: + case CS42L73_IM1: + case CS42L73_IM2: + return true; + default: + return false; + } +} + +static const unsigned int hpaloa_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 13, TLV_DB_SCALE_ITEM(-7600, 200, 0), + 14, 75, TLV_DB_SCALE_ITEM(-4900, 100, 0), +}; + +static DECLARE_TLV_DB_SCALE(adc_boost_tlv, 0, 2500, 0); + +static DECLARE_TLV_DB_SCALE(hl_tlv, -10200, 50, 0); + +static DECLARE_TLV_DB_SCALE(ipd_tlv, -9600, 100, 0); + +static DECLARE_TLV_DB_SCALE(micpga_tlv, -600, 50, 0); + +static const unsigned int limiter_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0), + 3, 7, TLV_DB_SCALE_ITEM(-1200, 300, 0), +}; + +static const DECLARE_TLV_DB_SCALE(attn_tlv, -6300, 100, 1); + +static const char * const cs42l73_pgaa_text[] = { "Line A", "Mic 1" }; +static const char * const cs42l73_pgab_text[] = { "Line B", "Mic 2" }; + +static const struct soc_enum pgaa_enum = + SOC_ENUM_SINGLE(CS42L73_ADCIPC, 3, + ARRAY_SIZE(cs42l73_pgaa_text), cs42l73_pgaa_text); + +static const struct soc_enum pgab_enum = + SOC_ENUM_SINGLE(CS42L73_ADCIPC, 7, + ARRAY_SIZE(cs42l73_pgab_text), cs42l73_pgab_text); + +static const struct snd_kcontrol_new pgaa_mux = + SOC_DAPM_ENUM("Left Analog Input Capture Mux", pgaa_enum); + +static const struct snd_kcontrol_new pgab_mux = + SOC_DAPM_ENUM("Right Analog Input Capture Mux", pgab_enum); + +static const struct snd_kcontrol_new input_left_mixer[] = { + SOC_DAPM_SINGLE("ADC Left Input", CS42L73_PWRCTL1, + 5, 1, 1), + SOC_DAPM_SINGLE("DMIC Left Input", CS42L73_PWRCTL1, + 4, 1, 1), +}; + +static const struct snd_kcontrol_new input_right_mixer[] = { + SOC_DAPM_SINGLE("ADC Right Input", CS42L73_PWRCTL1, + 7, 1, 1), + SOC_DAPM_SINGLE("DMIC Right Input", CS42L73_PWRCTL1, + 6, 1, 1), +}; + +static const char * const cs42l73_ng_delay_text[] = { + "50ms", "100ms", "150ms", "200ms" }; + +static const struct soc_enum ng_delay_enum = + SOC_ENUM_SINGLE(CS42L73_NGCAB, 0, + ARRAY_SIZE(cs42l73_ng_delay_text), cs42l73_ng_delay_text); + +static const char * const charge_pump_freq_text[] = { + "0", "1", "2", "3", "4", + "5", "6", "7", "8", "9", + "10", "11", "12", "13", "14", "15" }; + +static const struct soc_enum charge_pump_enum = + SOC_ENUM_SINGLE(CS42L73_CPFCHC, 4, + ARRAY_SIZE(charge_pump_freq_text), charge_pump_freq_text); + +static const char * const cs42l73_mono_mix_texts[] = { + "Left", "Right", "Mono Mix"}; + +static const unsigned int cs42l73_mono_mix_values[] = { 0, 1, 2 }; + +static const struct soc_enum spk_asp_enum = + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 6, 1, + ARRAY_SIZE(cs42l73_mono_mix_texts), + cs42l73_mono_mix_texts, + cs42l73_mono_mix_values); + +static const struct snd_kcontrol_new spk_asp_mixer = + SOC_DAPM_ENUM("Route", spk_asp_enum); + +static const struct soc_enum spk_xsp_enum = + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 4, 3, + ARRAY_SIZE(cs42l73_mono_mix_texts), + cs42l73_mono_mix_texts, + cs42l73_mono_mix_values); + +static const struct snd_kcontrol_new spk_xsp_mixer = + SOC_DAPM_ENUM("Route", spk_xsp_enum); + +static const struct soc_enum esl_asp_enum = + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 2, 5, + ARRAY_SIZE(cs42l73_mono_mix_texts), + cs42l73_mono_mix_texts, + cs42l73_mono_mix_values); + +static const struct snd_kcontrol_new esl_asp_mixer = + SOC_DAPM_ENUM("Route", esl_asp_enum); + +static const struct soc_enum esl_xsp_enum = + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 0, 7, + ARRAY_SIZE(cs42l73_mono_mix_texts), + cs42l73_mono_mix_texts, + cs42l73_mono_mix_values); + +static const struct snd_kcontrol_new esl_xsp_mixer = + SOC_DAPM_ENUM("Route", esl_xsp_enum); + +static const char * const cs42l73_ip_swap_text[] = { + "Stereo", "Mono A", "Mono B", "Swap A-B"}; + +static const struct soc_enum ip_swap_enum = + SOC_ENUM_SINGLE(CS42L73_MIOPC, 6, + ARRAY_SIZE(cs42l73_ip_swap_text), cs42l73_ip_swap_text); + +static const char * const cs42l73_spo_mixer_text[] = {"Mono", "Stereo"}; + +static const struct soc_enum vsp_output_mux_enum = + SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 5, + ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text); + +static const struct soc_enum xsp_output_mux_enum = + SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 4, + ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text); + +static const struct snd_kcontrol_new vsp_output_mux = + SOC_DAPM_ENUM("Route", vsp_output_mux_enum); + +static const struct snd_kcontrol_new xsp_output_mux = + SOC_DAPM_ENUM("Route", xsp_output_mux_enum); + +static const struct snd_kcontrol_new hp_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 0, 1, 1); + +static const struct snd_kcontrol_new lo_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 1, 1, 1); + +static const struct snd_kcontrol_new spk_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 2, 1, 1); + +static const struct snd_kcontrol_new spklo_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 4, 1, 1); + +static const struct snd_kcontrol_new ear_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 3, 1, 1); + +static const struct snd_kcontrol_new cs42l73_snd_controls[] = { + SOC_DOUBLE_R_SX_TLV("Headphone Analog Playback Volume", + CS42L73_HPAAVOL, CS42L73_HPBAVOL, 7, + 0xffffffC1, 0x0C, hpaloa_tlv), + + SOC_DOUBLE_R_SX_TLV("LineOut Analog Playback Volume", CS42L73_LOAAVOL, + CS42L73_LOBAVOL, 7, 0xffffffC1, 0x0C, hpaloa_tlv), + + SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL, + CS42L73_MICBPREPGABVOL, 5, 0xffffff35, + 0x34, micpga_tlv), + + SOC_DOUBLE_R("MIC Preamp Switch", CS42L73_MICAPREPGAAVOL, + CS42L73_MICBPREPGABVOL, 6, 1, 1), + + SOC_DOUBLE_R_SX_TLV("Input Path Digital Volume", CS42L73_IPADVOL, + CS42L73_IPBDVOL, 7, 0xffffffA0, 0xA0, ipd_tlv), + + SOC_DOUBLE_R_SX_TLV("HL Digital Playback Volume", + CS42L73_HLADVOL, CS42L73_HLBDVOL, 7, 0xffffffE5, + 0xE4, hl_tlv), + + SOC_SINGLE_TLV("ADC A Boost Volume", + CS42L73_ADCIPC, 2, 0x01, 1, adc_boost_tlv), + + SOC_SINGLE_TLV("ADC B Boost Volume", + CS42L73_ADCIPC, 6, 0x01, 1, adc_boost_tlv), + + SOC_SINGLE_TLV("Speakerphone Digital Playback Volume", + CS42L73_SPKDVOL, 0, 0xE4, 1, hl_tlv), + + SOC_SINGLE_TLV("Ear Speaker Digital Playback Volume", + CS42L73_ESLDVOL, 0, 0xE4, 1, hl_tlv), + + SOC_DOUBLE_R("Headphone Analog Playback Switch", CS42L73_HPAAVOL, + CS42L73_HPBAVOL, 7, 1, 1), + + SOC_DOUBLE_R("LineOut Analog Playback Switch", CS42L73_LOAAVOL, + CS42L73_LOBAVOL, 7, 1, 1), + SOC_DOUBLE("Input Path Digital Switch", CS42L73_ADCIPC, 0, 4, 1, 1), + SOC_DOUBLE("HL Digital Playback Switch", CS42L73_PBDC, 0, + 1, 1, 1), + SOC_SINGLE("Speakerphone Digital Playback Switch", CS42L73_PBDC, 2, 1, + 1), + SOC_SINGLE("Ear Speaker Digital Playback Switch", CS42L73_PBDC, 3, 1, + 1), + + SOC_SINGLE("PGA Soft-Ramp Switch", CS42L73_MIOPC, 3, 1, 0), + SOC_SINGLE("Analog Zero Cross Switch", CS42L73_MIOPC, 2, 1, 0), + SOC_SINGLE("Digital Soft-Ramp Switch", CS42L73_MIOPC, 1, 1, 0), + SOC_SINGLE("Analog Output Soft-Ramp Switch", CS42L73_MIOPC, 0, 1, 0), + + SOC_DOUBLE("ADC Signal Polarity Switch", CS42L73_ADCIPC, 1, 5, 1, + 0), + + SOC_SINGLE("HL Limiter Attack Rate", CS42L73_LIMARATEHL, 0, 0x3F, + 0), + SOC_SINGLE("HL Limiter Release Rate", CS42L73_LIMRRATEHL, 0, + 0x3F, 0), + + + SOC_SINGLE("HL Limiter Switch", CS42L73_LIMRRATEHL, 7, 1, 0), + SOC_SINGLE("HL Limiter All Channels Switch", CS42L73_LIMRRATEHL, 6, 1, + 0), + + SOC_SINGLE_TLV("HL Limiter Max Threshold Volume", CS42L73_LMAXHL, 5, 7, + 1, limiter_tlv), + + SOC_SINGLE_TLV("HL Limiter Cushion Volume", CS42L73_LMAXHL, 2, 7, 1, + limiter_tlv), + + SOC_SINGLE("SPK Limiter Attack Rate Volume", CS42L73_LIMARATESPK, 0, + 0x3F, 0), + SOC_SINGLE("SPK Limiter Release Rate Volume", CS42L73_LIMRRATESPK, 0, + 0x3F, 0), + SOC_SINGLE("SPK Limiter Switch", CS42L73_LIMRRATESPK, 7, 1, 0), + SOC_SINGLE("SPK Limiter All Channels Switch", CS42L73_LIMRRATESPK, + 6, 1, 0), + SOC_SINGLE_TLV("SPK Limiter Max Threshold Volume", CS42L73_LMAXSPK, 5, + 7, 1, limiter_tlv), + + SOC_SINGLE_TLV("SPK Limiter Cushion Volume", CS42L73_LMAXSPK, 2, 7, 1, + limiter_tlv), + + SOC_SINGLE("ESL Limiter Attack Rate Volume", CS42L73_LIMARATEESL, 0, + 0x3F, 0), + SOC_SINGLE("ESL Limiter Release Rate Volume", CS42L73_LIMRRATEESL, 0, + 0x3F, 0), + SOC_SINGLE("ESL Limiter Switch", CS42L73_LIMRRATEESL, 7, 1, 0), + SOC_SINGLE_TLV("ESL Limiter Max Threshold Volume", CS42L73_LMAXESL, 5, + 7, 1, limiter_tlv), + + SOC_SINGLE_TLV("ESL Limiter Cushion Volume", CS42L73_LMAXESL, 2, 7, 1, + limiter_tlv), + + SOC_SINGLE("ALC Attack Rate Volume", CS42L73_ALCARATE, 0, 0x3F, 0), + SOC_SINGLE("ALC Release Rate Volume", CS42L73_ALCRRATE, 0, 0x3F, 0), + SOC_DOUBLE("ALC Switch", CS42L73_ALCARATE, 6, 7, 1, 0), + SOC_SINGLE_TLV("ALC Max Threshold Volume", CS42L73_ALCMINMAX, 5, 7, 0, + limiter_tlv), + SOC_SINGLE_TLV("ALC Min Threshold Volume", CS42L73_ALCMINMAX, 2, 7, 0, + limiter_tlv), + + SOC_DOUBLE("NG Enable Switch", CS42L73_NGCAB, 6, 7, 1, 0), + SOC_SINGLE("NG Boost Switch", CS42L73_NGCAB, 5, 1, 0), + /* + NG Threshold depends on NG_BOOTSAB, which selects + between two threshold scales in decibels. + Set linear values for now .. + */ + SOC_SINGLE("NG Threshold", CS42L73_NGCAB, 2, 7, 0), + SOC_ENUM("NG Delay", ng_delay_enum), + + SOC_ENUM("Charge Pump Frequency", charge_pump_enum), + + SOC_DOUBLE_R_TLV("XSP-IP Volume", + CS42L73_XSPAIPAA, CS42L73_XSPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-XSP Volume", + CS42L73_XSPAXSPAA, CS42L73_XSPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-ASP Volume", + CS42L73_XSPAASPAA, CS42L73_XSPAASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-VSP Volume", + CS42L73_XSPAVSPMA, CS42L73_XSPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("ASP-IP Volume", + CS42L73_ASPAIPAA, CS42L73_ASPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-XSP Volume", + CS42L73_ASPAXSPAA, CS42L73_ASPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-ASP Volume", + CS42L73_ASPAASPAA, CS42L73_ASPBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-VSP Volume", + CS42L73_ASPAVSPMA, CS42L73_ASPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("VSP-IP Volume", + CS42L73_VSPAIPAA, CS42L73_VSPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-XSP Volume", + CS42L73_VSPAXSPAA, CS42L73_VSPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-ASP Volume", + CS42L73_VSPAASPAA, CS42L73_VSPBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-VSP Volume", + CS42L73_VSPAVSPMA, CS42L73_VSPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("HL-IP Volume", + CS42L73_HLAIPAA, CS42L73_HLBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-XSP Volume", + CS42L73_HLAXSPAA, CS42L73_HLBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-ASP Volume", + CS42L73_HLAASPAA, CS42L73_HLBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-VSP Volume", + CS42L73_HLAVSPMA, CS42L73_HLBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_SINGLE_TLV("SPK-IP Mono Volume", + CS42L73_SPKMIPMA, 0, 0x3E, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-XSP Mono Volume", + CS42L73_SPKMXSPA, 0, 0x3E, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-ASP Mono Volume", + CS42L73_SPKMASPA, 0, 0x3E, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-VSP Mono Volume", + CS42L73_SPKMVSPMA, 0, 0x3E, 1, attn_tlv), + + SOC_SINGLE_TLV("ESL-IP Mono Volume", + CS42L73_ESLMIPMA, 0, 0x3E, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-XSP Mono Volume", + CS42L73_ESLMXSPA, 0, 0x3E, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-ASP Mono Volume", + CS42L73_ESLMASPA, 0, 0x3E, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-VSP Mono Volume", + CS42L73_ESLMVSPMA, 0, 0x3E, 1, attn_tlv), + + SOC_ENUM("IP Digital Swap/Mono Select", ip_swap_enum), + + SOC_ENUM("VSPOUT Mono/Stereo Select", vsp_output_mux_enum), + SOC_ENUM("XSPOUT Mono/Stereo Select", xsp_output_mux_enum), +}; + +static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("LINEINA"), + SND_SOC_DAPM_INPUT("LINEINB"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_SUPPLY("MIC1 Bias", CS42L73_PWRCTL2, 6, 1, NULL, 0), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_SUPPLY("MIC2 Bias", CS42L73_PWRCTL2, 7, 1, NULL, 0), + + SND_SOC_DAPM_AIF_OUT("XSPOUTL", "XSP Capture", 0, + CS42L73_PWRCTL2, 1, 1), + SND_SOC_DAPM_AIF_OUT("XSPOUTR", "XSP Capture", 0, + CS42L73_PWRCTL2, 1, 1), + SND_SOC_DAPM_AIF_OUT("ASPOUTL", "ASP Capture", 0, + CS42L73_PWRCTL2, 3, 1), + SND_SOC_DAPM_AIF_OUT("ASPOUTR", "ASP Capture", 0, + CS42L73_PWRCTL2, 3, 1), + SND_SOC_DAPM_AIF_OUT("VSPOUTL", "VSP Capture", 0, + CS42L73_PWRCTL2, 4, 1), + SND_SOC_DAPM_AIF_OUT("VSPOUTR", "VSP Capture", 0, + CS42L73_PWRCTL2, 4, 1), + + SND_SOC_DAPM_PGA("PGA Left", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PGA Right", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MUX("PGA Left Mux", SND_SOC_NOPM, 0, 0, &pgaa_mux), + SND_SOC_DAPM_MUX("PGA Right Mux", SND_SOC_NOPM, 0, 0, &pgab_mux), + + SND_SOC_DAPM_ADC("ADC Left", NULL, CS42L73_PWRCTL1, 7, 1), + SND_SOC_DAPM_ADC("ADC Right", NULL, CS42L73_PWRCTL1, 5, 1), + SND_SOC_DAPM_ADC("DMIC Left", NULL, CS42L73_PWRCTL1, 6, 1), + SND_SOC_DAPM_ADC("DMIC Right", NULL, CS42L73_PWRCTL1, 4, 1), + + SND_SOC_DAPM_MIXER_NAMED_CTL("Input Left Capture", SND_SOC_NOPM, + 0, 0, input_left_mixer, + ARRAY_SIZE(input_left_mixer)), + + SND_SOC_DAPM_MIXER_NAMED_CTL("Input Right Capture", SND_SOC_NOPM, + 0, 0, input_right_mixer, + ARRAY_SIZE(input_right_mixer)), + + SND_SOC_DAPM_MIXER("ASPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("ASPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("XSPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("XSPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("VSPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("VSPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_AIF_IN("XSPINL", "XSP Playback", 0, + CS42L73_PWRCTL2, 0, 1), + SND_SOC_DAPM_AIF_IN("XSPINR", "XSP Playback", 0, + CS42L73_PWRCTL2, 0, 1), + SND_SOC_DAPM_AIF_IN("XSPINM", "XSP Playback", 0, + CS42L73_PWRCTL2, 0, 1), + + SND_SOC_DAPM_AIF_IN("ASPINL", "ASP Playback", 0, + CS42L73_PWRCTL2, 2, 1), + SND_SOC_DAPM_AIF_IN("ASPINR", "ASP Playback", 0, + CS42L73_PWRCTL2, 2, 1), + SND_SOC_DAPM_AIF_IN("ASPINM", "ASP Playback", 0, + CS42L73_PWRCTL2, 2, 1), + + SND_SOC_DAPM_AIF_IN("VSPIN", "VSP Playback", 0, + CS42L73_PWRCTL2, 4, 1), + + SND_SOC_DAPM_MIXER("HL Left Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("HL Right Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SPK Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("ESL Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MUX("ESL-XSP Mux", SND_SOC_NOPM, + 0, 0, &esl_xsp_mixer), + + SND_SOC_DAPM_MUX("ESL-ASP Mux", SND_SOC_NOPM, + 0, 0, &esl_asp_mixer), + + SND_SOC_DAPM_MUX("SPK-ASP Mux", SND_SOC_NOPM, + 0, 0, &spk_asp_mixer), + + SND_SOC_DAPM_MUX("SPK-XSP Mux", SND_SOC_NOPM, + 0, 0, &spk_xsp_mixer), + + SND_SOC_DAPM_PGA("HL Left DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("HL Right DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("SPK DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("ESL DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("HP Amp", CS42L73_PWRCTL3, 0, 1, + &hp_amp_ctl), + SND_SOC_DAPM_SWITCH("LO Amp", CS42L73_PWRCTL3, 1, 1, + &lo_amp_ctl), + SND_SOC_DAPM_SWITCH("SPK Amp", CS42L73_PWRCTL3, 2, 1, + &spk_amp_ctl), + SND_SOC_DAPM_SWITCH("EAR Amp", CS42L73_PWRCTL3, 3, 1, + &ear_amp_ctl), + SND_SOC_DAPM_SWITCH("SPKLO Amp", CS42L73_PWRCTL3, 4, 1, + &spklo_amp_ctl), + + SND_SOC_DAPM_OUTPUT("HPOUTA"), + SND_SOC_DAPM_OUTPUT("HPOUTB"), + SND_SOC_DAPM_OUTPUT("LINEOUTA"), + SND_SOC_DAPM_OUTPUT("LINEOUTB"), + SND_SOC_DAPM_OUTPUT("EAROUT"), + SND_SOC_DAPM_OUTPUT("SPKOUT"), + SND_SOC_DAPM_OUTPUT("SPKLINEOUT"), +}; + +static const struct snd_soc_dapm_route cs42l73_audio_map[] = { + + /* SPKLO EARSPK Paths */ + {"EAROUT", NULL, "EAR Amp"}, + {"SPKLINEOUT", NULL, "SPKLO Amp"}, + + {"EAR Amp", "Switch", "ESL DAC"}, + {"SPKLO Amp", "Switch", "ESL DAC"}, + + {"ESL DAC", "ESL-ASP Mono Volume", "ESL Mixer"}, + {"ESL DAC", "ESL-XSP Mono Volume", "ESL Mixer"}, + {"ESL DAC", "ESL-VSP Mono Volume", "VSPIN"}, + /* Loopback */ + {"ESL DAC", "ESL-IP Mono Volume", "Input Left Capture"}, + {"ESL DAC", "ESL-IP Mono Volume", "Input Right Capture"}, + + {"ESL Mixer", NULL, "ESL-ASP Mux"}, + {"ESL Mixer", NULL, "ESL-XSP Mux"}, + + {"ESL-ASP Mux", "Left", "ASPINL"}, + {"ESL-ASP Mux", "Right", "ASPINR"}, + {"ESL-ASP Mux", "Mono Mix", "ASPINM"}, + + {"ESL-XSP Mux", "Left", "XSPINL"}, + {"ESL-XSP Mux", "Right", "XSPINR"}, + {"ESL-XSP Mux", "Mono Mix", "XSPINM"}, + + /* Speakerphone Paths */ + {"SPKOUT", NULL, "SPK Amp"}, + {"SPK Amp", "Switch", "SPK DAC"}, + + {"SPK DAC", "SPK-ASP Mono Volume", "SPK Mixer"}, + {"SPK DAC", "SPK-XSP Mono Volume", "SPK Mixer"}, + {"SPK DAC", "SPK-VSP Mono Volume", "VSPIN"}, + /* Loopback */ + {"SPK DAC", "SPK-IP Mono Volume", "Input Left Capture"}, + {"SPK DAC", "SPK-IP Mono Volume", "Input Right Capture"}, + + {"SPK Mixer", NULL, "SPK-ASP Mux"}, + {"SPK Mixer", NULL, "SPK-XSP Mux"}, + + {"SPK-ASP Mux", "Left", "ASPINL"}, + {"SPK-ASP Mux", "Mono Mix", "ASPINM"}, + {"SPK-ASP Mux", "Right", "ASPINR"}, + + {"SPK-XSP Mux", "Left", "XSPINL"}, + {"SPK-XSP Mux", "Mono Mix", "XSPINM"}, + {"SPK-XSP Mux", "Right", "XSPINR"}, + + /* HP LineOUT Paths */ + {"HPOUTA", NULL, "HP Amp"}, + {"HPOUTB", NULL, "HP Amp"}, + {"LINEOUTA", NULL, "LO Amp"}, + {"LINEOUTB", NULL, "LO Amp"}, + + {"HP Amp", "Switch", "HL Left DAC"}, + {"HP Amp", "Switch", "HL Right DAC"}, + {"LO Amp", "Switch", "HL Left DAC"}, + {"LO Amp", "Switch", "HL Right DAC"}, + + {"HL Left DAC", "HL-XSP Volume", "HL Left Mixer"}, + {"HL Right DAC", "HL-XSP Volume", "HL Right Mixer"}, + {"HL Left DAC", "HL-ASP Volume", "HL Left Mixer"}, + {"HL Right DAC", "HL-ASP Volume", "HL Right Mixer"}, + {"HL Left DAC", "HL-VSP Volume", "HL Left Mixer"}, + {"HL Right DAC", "HL-VSP Volume", "HL Right Mixer"}, + /* Loopback */ + {"HL Left DAC", "HL-IP Volume", "HL Left Mixer"}, + {"HL Right DAC", "HL-IP Volume", "HL Right Mixer"}, + {"HL Left Mixer", NULL, "Input Left Capture"}, + {"HL Right Mixer", NULL, "Input Right Capture"}, + + {"HL Left Mixer", NULL, "ASPINL"}, + {"HL Right Mixer", NULL, "ASPINR"}, + {"HL Left Mixer", NULL, "XSPINL"}, + {"HL Right Mixer", NULL, "XSPINR"}, + {"HL Left Mixer", NULL, "VSPIN"}, + {"HL Right Mixer", NULL, "VSPIN"}, + + /* Capture Paths */ + {"MIC1", NULL, "MIC1 Bias"}, + {"PGA Left Mux", "Mic 1", "MIC1"}, + {"MIC2", NULL, "MIC2 Bias"}, + {"PGA Right Mux", "Mic 2", "MIC2"}, + + {"PGA Left Mux", "Line A", "LINEINA"}, + {"PGA Right Mux", "Line B", "LINEINB"}, + + {"PGA Left", NULL, "PGA Left Mux"}, + {"PGA Right", NULL, "PGA Right Mux"}, + + {"ADC Left", NULL, "PGA Left"}, + {"ADC Right", NULL, "PGA Right"}, + + {"Input Left Capture", "ADC Left Input", "ADC Left"}, + {"Input Right Capture", "ADC Right Input", "ADC Right"}, + {"Input Left Capture", "DMIC Left Input", "DMIC Left"}, + {"Input Right Capture", "DMIC Right Input", "DMIC Right"}, + + /* Audio Capture */ + {"ASPL Output Mixer", NULL, "Input Left Capture"}, + {"ASPR Output Mixer", NULL, "Input Right Capture"}, + + {"ASPOUTL", "ASP-IP Volume", "ASPL Output Mixer"}, + {"ASPOUTR", "ASP-IP Volume", "ASPR Output Mixer"}, + + /* Auxillary Capture */ + {"XSPL Output Mixer", NULL, "Input Left Capture"}, + {"XSPR Output Mixer", NULL, "Input Right Capture"}, + + {"XSPOUTL", "XSP-IP Volume", "XSPL Output Mixer"}, + {"XSPOUTR", "XSP-IP Volume", "XSPR Output Mixer"}, + + {"XSPOUTL", NULL, "XSPL Output Mixer"}, + {"XSPOUTR", NULL, "XSPR Output Mixer"}, + + /* Voice Capture */ + {"VSPL Output Mixer", NULL, "Input Left Capture"}, + {"VSPR Output Mixer", NULL, "Input Left Capture"}, + + {"VSPOUTL", "VSP-IP Volume", "VSPL Output Mixer"}, + {"VSPOUTR", "VSP-IP Volume", "VSPR Output Mixer"}, + + {"VSPOUTL", NULL, "VSPL Output Mixer"}, + {"VSPOUTR", NULL, "VSPR Output Mixer"}, +}; + +struct cs42l73_mclk_div { + u32 mclk; + u32 srate; + u8 mmcc; +}; + +static struct cs42l73_mclk_div cs42l73_mclk_coeffs[] = { + /* MCLK, Sample Rate, xMMCC[5:0] */ + {5644800, 11025, 0x30}, + {5644800, 22050, 0x20}, + {5644800, 44100, 0x10}, + + {6000000, 8000, 0x39}, + {6000000, 11025, 0x33}, + {6000000, 12000, 0x31}, + {6000000, 16000, 0x29}, + {6000000, 22050, 0x23}, + {6000000, 24000, 0x21}, + {6000000, 32000, 0x19}, + {6000000, 44100, 0x13}, + {6000000, 48000, 0x11}, + + {6144000, 8000, 0x38}, + {6144000, 12000, 0x30}, + {6144000, 16000, 0x28}, + {6144000, 24000, 0x20}, + {6144000, 32000, 0x18}, + {6144000, 48000, 0x10}, + + {6500000, 8000, 0x3C}, + {6500000, 11025, 0x35}, + {6500000, 12000, 0x34}, + {6500000, 16000, 0x2C}, + {6500000, 22050, 0x25}, + {6500000, 24000, 0x24}, + {6500000, 32000, 0x1C}, + {6500000, 44100, 0x15}, + {6500000, 48000, 0x14}, + + {6400000, 8000, 0x3E}, + {6400000, 11025, 0x37}, + {6400000, 12000, 0x36}, + {6400000, 16000, 0x2E}, + {6400000, 22050, 0x27}, + {6400000, 24000, 0x26}, + {6400000, 32000, 0x1E}, + {6400000, 44100, 0x17}, + {6400000, 48000, 0x16}, +}; + +struct cs42l73_mclkx_div { + u32 mclkx; + u8 ratio; + u8 mclkdiv; +}; + +static struct cs42l73_mclkx_div cs42l73_mclkx_coeffs[] = { + {5644800, 1, 0}, /* 5644800 */ + {6000000, 1, 0}, /* 6000000 */ + {6144000, 1, 0}, /* 6144000 */ + {11289600, 2, 2}, /* 5644800 */ + {12288000, 2, 2}, /* 6144000 */ + {12000000, 2, 2}, /* 6000000 */ + {13000000, 2, 2}, /* 6500000 */ + {19200000, 3, 3}, /* 6400000 */ + {24000000, 4, 4}, /* 6000000 */ + {26000000, 4, 4}, /* 6500000 */ + {38400000, 6, 5} /* 6400000 */ +}; + +static int cs42l73_get_mclkx_coeff(int mclkx) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs42l73_mclkx_coeffs); i++) { + if (cs42l73_mclkx_coeffs[i].mclkx == mclkx) + return i; + } + return -EINVAL; +} + +static int cs42l73_get_mclk_coeff(int mclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs42l73_mclk_coeffs); i++) { + if (cs42l73_mclk_coeffs[i].mclk == mclk && + cs42l73_mclk_coeffs[i].srate == srate) + return i; + } + return -EINVAL; + +} + +static int cs42l73_set_mclk(struct snd_soc_dai *dai, unsigned int freq) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + + int mclkx_coeff; + u32 mclk = 0; + u8 dmmcc = 0; + + /* MCLKX -> MCLK */ + mclkx_coeff = cs42l73_get_mclkx_coeff(freq); + + mclk = cs42l73_mclkx_coeffs[mclkx_coeff].mclkx / + cs42l73_mclkx_coeffs[mclkx_coeff].ratio; + + dev_dbg(codec->dev, "MCLK%u %u <-> internal MCLK %u\n", + priv->mclksel + 1, cs42l73_mclkx_coeffs[mclkx_coeff].mclkx, + mclk); + + dmmcc = (priv->mclksel << 4) | + (cs42l73_mclkx_coeffs[mclkx_coeff].mclkdiv << 1); + + snd_soc_write(codec, CS42L73_DMMCC, dmmcc); + + priv->sysclk = mclkx_coeff; + priv->mclk = mclk; + + return 0; +} + +static int cs42l73_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case CS42L73_CLKID_MCLK1: + break; + case CS42L73_CLKID_MCLK2: + break; + default: + return -EINVAL; + } + + if ((cs42l73_set_mclk(dai, freq)) < 0) { + dev_err(codec->dev, "Unable to set MCLK for dai %s\n", + dai->name); + return -EINVAL; + } + + priv->mclksel = clk_id; + + return 0; +} + +static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + u8 id = codec_dai->id; + u8 inv, format; + u8 spc, mmcc; + + spc = snd_soc_read(codec, CS42L73_SPC(id)); + mmcc = snd_soc_read(codec, CS42L73_MMCC(id)); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + mmcc |= MS_MASTER; + break; + + case SND_SOC_DAIFMT_CBS_CFS: + mmcc &= ~MS_MASTER; + break; + + default: + return -EINVAL; + } + + format = (fmt & SND_SOC_DAIFMT_FORMAT_MASK); + inv = (fmt & SND_SOC_DAIFMT_INV_MASK); + + switch (format) { + case SND_SOC_DAIFMT_I2S: + spc &= ~SPDIF_PCM; + break; + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + if (mmcc & MS_MASTER) { + dev_err(codec->dev, + "PCM format in slave mode only\n"); + return -EINVAL; + } + if (id == CS42L73_ASP) { + dev_err(codec->dev, + "PCM format is not supported on ASP port\n"); + return -EINVAL; + } + spc |= SPDIF_PCM; + break; + default: + return -EINVAL; + } + + if (spc & SPDIF_PCM) { + spc &= (31 << 3); /* Clear PCM mode, set MSB->LSB */ + switch (format) { + case SND_SOC_DAIFMT_DSP_B: + if (inv == SND_SOC_DAIFMT_IB_IF) + spc |= (PCM_MODE0 << 4); + if (inv == SND_SOC_DAIFMT_IB_NF) + spc |= (PCM_MODE1 << 4); + break; + case SND_SOC_DAIFMT_DSP_A: + if (inv == SND_SOC_DAIFMT_IB_IF) + spc |= (PCM_MODE1 << 4); + break; + default: + return -EINVAL; + } + } + + priv->config[id].spc = spc; + priv->config[id].mmcc = mmcc; + + return 0; +} + +static u32 cs42l73_asrc_rates[] = { + 8000, 11025, 12000, 16000, 22050, + 24000, 32000, 44100, 48000 +}; + +static unsigned int cs42l73_get_xspfs_coeff(u32 rate) +{ + int i; + for (i = 0; i < ARRAY_SIZE(cs42l73_asrc_rates); i++) { + if (cs42l73_asrc_rates[i] == rate) + return i + 1; + } + return 0; /* 0 = Don't know */ +} + +static void cs42l73_update_asrc(struct snd_soc_codec *codec, int id, int srate) +{ + u8 spfs = 0; + + if (srate > 0) + spfs = cs42l73_get_xspfs_coeff(srate); + + switch (id) { + case CS42L73_XSP: + snd_soc_update_bits(codec, CS42L73_VXSPFS, 0x0f, spfs); + break; + case CS42L73_ASP: + snd_soc_update_bits(codec, CS42L73_ASPC, 0x3c, spfs << 2); + break; + case CS42L73_VSP: + snd_soc_update_bits(codec, CS42L73_VXSPFS, 0xf0, spfs << 4); + break; + default: + break; + } +} + +static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + int id = dai->id; + int mclk_coeff; + int srate = params_rate(params); + + if (priv->config[id].mmcc & MS_MASTER) { + /* CS42L73 Master */ + /* MCLK -> srate */ + mclk_coeff = + cs42l73_get_mclk_coeff(priv->mclk, srate); + + if (mclk_coeff < 0) + return -EINVAL; + + dev_dbg(codec->dev, + "DAI[%d]: MCLK %u, srate %u, MMCC[5:0] = %x\n", + id, priv->mclk, srate, + cs42l73_mclk_coeffs[mclk_coeff].mmcc); + + priv->config[id].mmcc &= 0xC0; + priv->config[id].mmcc |= cs42l73_mclk_coeffs[mclk_coeff].mmcc; + priv->config[id].spc &= 0xFC; + priv->config[id].spc &= MCK_SCLK_64FS; + } else { + /* CS42L73 Slave */ + priv->config[id].spc &= 0xFC; + priv->config[id].spc |= MCK_SCLK_64FS; + } + /* Update ASRCs */ + priv->config[id].srate = srate; + + snd_soc_write(codec, CS42L73_SPC(id), priv->config[id].spc); + snd_soc_write(codec, CS42L73_MMCC(id), priv->config[id].mmcc); + + cs42l73_update_asrc(codec, id, srate); + + return 0; +} + +static int cs42l73_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(codec); + + 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) { + regcache_cache_only(cs42l73->regmap, false); + regcache_sync(cs42l73->regmap); + } + 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) +{ + + struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(codec); + regcache_sync(cs42l73->regmap); + + cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +static int cs42l73_probe(struct snd_soc_codec *codec) +{ + int ret; + struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(codec); + + codec->control_data = cs42l73->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + regcache_cache_only(cs42l73->regmap, true); + + cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + 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, + + .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 struct regmap_config cs42l73_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS42L73_MAX_REGISTER, + .reg_defaults = cs42l73_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs42l73_reg_defaults), + .volatile_reg = cs42l73_volatile_register, + .readable_reg = cs42l73_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs42l73_private *cs42l73; + int ret; + unsigned int devid = 0; + unsigned int reg; + + 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); + + cs42l73->regmap = regmap_init_i2c(i2c_client, &cs42l73_regmap); + if (IS_ERR(cs42l73->regmap)) { + ret = PTR_ERR(cs42l73->regmap); + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); + goto err; + } + /* initialize codec */ + ret = regmap_read(cs42l73->regmap, CS42L73_DEVID_AB, &reg); + devid = (reg & 0xFF) << 12; + + ret = regmap_read(cs42l73->regmap, CS42L73_DEVID_CD, &reg); + devid |= (reg & 0xFF) << 4; + + ret = regmap_read(cs42l73->regmap, CS42L73_DEVID_E, &reg); + devid |= (reg & 0xF0) >> 4; + + + if (devid != CS42L73_DEVID) { + dev_err(&i2c_client->dev, + "CS42L73 Device ID (%X). Expected %X\n", + devid, CS42L73_DEVID); + goto err_regmap; + } + + ret = regmap_read(cs42l73->regmap, CS42L73_REVID, &reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "Get Revision ID failed\n"); + goto err_regmap; + } + + dev_info(&i2c_client->dev, + "Cirrus Logic CS42L73, Revision: %02X\n", ret & 0xFF); + + regcache_cache_only(cs42l73->regmap, true); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs42l73, cs42l73_dai, + ARRAY_SIZE(cs42l73_dai)); + if (ret < 0) + goto err_regmap; + return 0; + +err_regmap: + regmap_exit(cs42l73->regmap); + +err: + 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); + regmap_exit(cs42l73->regmap); + + 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) { + pr_err("Failed to register CS42L73 I2C driver: %d\n", ret); + 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(a)nucleusys.com>"); +MODULE_AUTHOR("Brian Austin, Cirrus Logic Inc, <brian.austin(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42l73.h b/sound/soc/codecs/cs42l73.h new file mode 100644 index 0000000..7c3bf7f --- /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(a)nucleusys.com> + * Brian Austin <brian.austin(a)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]. */ +#define CS42L73_MAX_REGISTER 0x61 /* Total Registers */ +/* 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__ */ -- 1.7.7.1

3 4

[alsa-devel] HDA Firmware file question
by David Henningsson 17 Nov '11

17 Nov '11

Hi, As seen in the "Early Patching" section of the "More notes on HD-Audio driver" document, you're supposed to activate it like this: options snd-hda-intel patch=on-board-patch,hdmi-patch But how do we know the order here? The pci bus is probed in parallel (right?) so could this be racy, that on every other boot, the hdmi card is probed first instead? Also, since the patch file specifies a [codec] section, you could actually use the same patch file for almost all cases. I guess the case of having two card instances with the same codec at the same address is pretty unusual, right? -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

[alsa-devel] [PATCH 0/6] ASoC: Convert ALC5632 codec to use regmap API
by Leon Romanovsky 17 Nov '11

17 Nov '11

Leon Romanovsky (6): ASoC: Convert ALC5632 codec to use regmap API ASoC: alc5632: Fix compile without CONFIG_I2C ASoC: alc5632: Fix compile without CONFIG_PM ASoC: alc5632: Remove unrelevant registers and name the relevant ASoC: alc5632: Added support of two undocumented registers ASoC: alc5632: Update of i2c_probe function to use regmap API only sound/soc/codecs/alc5632.c | 222 ++++++++++++++++++++++++++------------------ sound/soc/codecs/alc5632.h | 2 + 2 files changed, 135 insertions(+), 89 deletions(-) -- 1.7.3.4

2 15

[alsa-devel] [PATCH 1/2] hda - fix ELD memory leak
by Wu Fengguang 17 Nov '11

17 Nov '11

memset(eld) clears eld->proc_entry which will leak the struct snd_info_entry when unloading the module. Fix it by - remove memset(eld) - set eld->eld_valid to true _after_ all eld fields have been filled - don't access the other eld fields when (eld->eld_valid == false) Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/hda_eld.c | 5 +---- sound/pci/hda/patch_hdmi.c | 11 +++++------ 2 files changed, 6 insertions(+), 10 deletions(-) --- linux.orig/sound/pci/hda/hda_eld.c 2011-11-15 21:02:25.000000000 +0800 +++ linux/sound/pci/hda/hda_eld.c 2011-11-15 21:13:46.000000000 +0800 @@ -297,10 +297,10 @@ static int hdmi_update_eld(struct hdmi_e buf + ELD_FIXED_BYTES + mnl + 3 * i); } + e->eld_valid = true; return 0; out_fail: - e->eld_ver = 0; return -EINVAL; } @@ -323,9 +323,6 @@ int snd_hdmi_get_eld(struct hdmi_eld *el * ELD is valid, actual eld_size is assigned in hdmi_update_eld() */ - if (!eld->eld_valid) - return -ENOENT; - size = snd_hdmi_get_eld_size(codec, nid); if (size == 0) { /* wfg: workaround for ASUS P5E-VM HDMI board */ --- linux.orig/sound/pci/hda/patch_hdmi.c 2011-11-15 21:02:25.000000000 +0800 +++ linux/sound/pci/hda/patch_hdmi.c 2011-11-15 21:13:42.000000000 +0800 @@ -980,20 +980,19 @@ static void hdmi_present_sense(struct hd * the unsolicited response to avoid custom WARs. */ int present = snd_hda_pin_sense(codec, pin_nid); + bool eld_valid = false; - memset(eld, 0, sizeof(*eld)); + eld->eld_valid = false; eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE); if (eld->monitor_present) - eld->eld_valid = !!(present & AC_PINSENSE_ELDV); - else - eld->eld_valid = 0; + eld_valid = !!(present & AC_PINSENSE_ELDV); printk(KERN_INFO "HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", - codec->addr, pin_nid, eld->monitor_present, eld->eld_valid); + codec->addr, pin_nid, eld->monitor_present, eld_valid); - if (eld->eld_valid) + if (eld_valid) if (!snd_hdmi_get_eld(eld, codec, pin_nid)) snd_hdmi_show_eld(eld);

3 18

[alsa-devel] [PATCH 1/2] Search-and-replace improve documentation language
by Andrew Eikum 16 Nov '11

16 Nov '11

--- src/pcm/pcm.c | 146 ++++++++++++++++++++++++++------------------------------- 1 files changed, 67 insertions(+), 79 deletions(-)

1 0

[alsa-devel] [PATCH] ALSA: hda - Send CONNECT_SEL on capture start
by Dylan Reid 16 Nov '11

16 Nov '11

cur_adc is set to zero in cs_capture_pcm_prepare. This causes the SET_CONNECT_SEL verb to be sent to node zero when doing automic after recording. Delay sending the command until we are about to start capture and cur_adc is set to the correct NID. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/patch_cirrus.c | 16 ++++++++++------ 1 files changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 2a2d864..30f2f6a 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -250,6 +250,9 @@ static int cs_capture_pcm_prepare(struct hda_pcm_stream *hinfo, spec->cur_adc = spec->adc_nid[spec->cur_input]; spec->cur_adc_stream_tag = stream_tag; spec->cur_adc_format = format; + snd_hda_codec_write(codec, spec->cur_adc, 0, + AC_VERB_SET_CONNECT_SEL, + spec->adc_idx[spec->cur_input]); snd_hda_codec_setup_stream(codec, spec->cur_adc, stream_tag, 0, format); return 0; } @@ -685,13 +688,13 @@ static int change_cur_input(struct hda_codec *codec, unsigned int idx, /* stream is running, let's swap the current ADC */ __snd_hda_codec_cleanup_stream(codec, spec->cur_adc, 1); spec->cur_adc = spec->adc_nid[idx]; + snd_hda_codec_write(codec, spec->cur_adc, 0, + AC_VERB_SET_CONNECT_SEL, + spec->adc_idx[idx]); snd_hda_codec_setup_stream(codec, spec->cur_adc, spec->cur_adc_stream_tag, 0, spec->cur_adc_format); } - snd_hda_codec_write(codec, spec->cur_adc, 0, - AC_VERB_SET_CONNECT_SEL, - spec->adc_idx[idx]); spec->cur_input = idx; return 1; } @@ -974,9 +977,10 @@ static void cs_automic(struct hda_codec *codec) spec->cur_input = spec->last_input; } - snd_hda_codec_write_cache(codec, spec->cur_adc, 0, - AC_VERB_SET_CONNECT_SEL, - spec->adc_idx[spec->cur_input]); + if (spec->cur_adc) + snd_hda_codec_write_cache(codec, spec->cur_adc, 0, + AC_VERB_SET_CONNECT_SEL, + spec->adc_idx[spec->cur_input]); } else { if (present) change_cur_input(codec, spec->automic_idx, 0); -- 1.7.5.185.g0b9dee

2 3

[alsa-devel] [PATCH] ARM: pxa: add support for Raumfeld DDX
by Daniel Mack 16 Nov '11

16 Nov '11

This new product features a STA32x codec. To support it, some rework of the regulator initialization for the Raumfeld platform is necessary. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Cc: Eric Miao <eric.y.miao(a)gmail.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)ti.com> --- Mark, Liam, Eric - which tree would be good to the this merged? arch/arm/mach-pxa/raumfeld.c | 119 +++++++++++++++++--- sound/soc/pxa/Kconfig | 2 + sound/soc/pxa/raumfeld.c | 249 +++++++++++++++++++++++++++++++++++++++--- 3 files changed, 338 insertions(+), 32 deletions(-) diff --git a/arch/arm/mach-pxa/raumfeld.c b/arch/arm/mach-pxa/raumfeld.c index 7856fe4..22366ff 100644 --- a/arch/arm/mach-pxa/raumfeld.c +++ b/arch/arm/mach-pxa/raumfeld.c @@ -850,7 +850,7 @@ struct regulator_init_data audio_va_initdata = { }, }; -static struct fixed_voltage_config audio_va_config = { +static struct fixed_voltage_config cs4270_va_config = { .supply_name = "audio_va", .microvolts = 5000000, .gpio = GPIO_AUDIO_VA_ENABLE, @@ -859,47 +859,114 @@ static struct fixed_voltage_config audio_va_config = { .init_data = &audio_va_initdata, }; -static struct platform_device audio_va_device = { +static struct platform_device cs4270_va_device = { .name = "reg-fixed-voltage", .id = 0, .dev = { - .platform_data = &audio_va_config, + .platform_data = &cs4270_va_config, }, }; /* Dummy supplies for Codec's VD/VLC */ -static struct regulator_consumer_supply audio_dummy_supplies[] = { +static struct regulator_consumer_supply cs4270_dummy_supplies[] = { REGULATOR_SUPPLY("vd", "0-0048"), REGULATOR_SUPPLY("vlc", "0-0048"), }; -struct regulator_init_data audio_dummy_initdata = { - .consumer_supplies = audio_dummy_supplies, - .num_consumer_supplies = ARRAY_SIZE(audio_dummy_supplies), +struct regulator_init_data cs4270_dummy_initdata = { + .consumer_supplies = cs4270_dummy_supplies, + .num_consumer_supplies = ARRAY_SIZE(cs4270_dummy_supplies), .constraints = { .valid_ops_mask = REGULATOR_CHANGE_STATUS, }, }; -static struct fixed_voltage_config audio_dummy_config = { - .supply_name = "audio_vd", +static struct fixed_voltage_config cs4270_dummy_config = { + .supply_name = "cs4270_vd", .microvolts = 3300000, .gpio = -1, - .init_data = &audio_dummy_initdata, + .init_data = &cs4270_dummy_initdata, }; -static struct platform_device audio_supply_dummy_device = { +static struct platform_device cs4270_supply_dummy_device = { .name = "reg-fixed-voltage", .id = 1, .dev = { - .platform_data = &audio_dummy_config, + .platform_data = &cs4270_dummy_config, }, }; -static struct platform_device *audio_regulator_devices[] = { - &audio_va_device, - &audio_supply_dummy_device, +static struct platform_device *cs4270_regulator_devices[] = { + &cs4270_va_device, + &cs4270_supply_dummy_device, +}; + +/* Fixed regulator for sta32x Vdda supply + * 0-001a maps to the sta32x codec devname (derived from i2c bus num + addr) + */ + +static struct regulator_consumer_supply sta32x_va_consumer_supply = + REGULATOR_SUPPLY("Vdda", "0-001a"); + +struct regulator_init_data sta32x_va_initdata = { + .consumer_supplies = &sta32x_va_consumer_supply, + .num_consumer_supplies = 1, + .constraints = { + .valid_ops_mask = REGULATOR_CHANGE_STATUS, + }, +}; + +static struct fixed_voltage_config sta32x_va_config = { + .supply_name = "audio_va", + .microvolts = 3300000, + .gpio = GPIO_AUDIO_VA_ENABLE, + .enable_high = 1, + .enabled_at_boot = 0, + .init_data = &sta32x_va_initdata, +}; + +static struct platform_device sta32x_va_device = { + .name = "reg-fixed-voltage", + .id = 0, + .dev = { + .platform_data = &sta32x_va_config, + }, +}; + +/* Dummy supplies for sta32x codec's Vdd3/Vcc */ + +static struct regulator_consumer_supply sta32x_dummy_supplies[] = { + REGULATOR_SUPPLY("Vdd3", "0-001a"), + REGULATOR_SUPPLY("Vcc", "0-001a"), +}; + +struct regulator_init_data sta32x_dummy_initdata = { + .consumer_supplies = sta32x_dummy_supplies, + .num_consumer_supplies = ARRAY_SIZE(sta32x_dummy_supplies), + .constraints = { + .valid_ops_mask = REGULATOR_CHANGE_STATUS, + }, +}; + +static struct fixed_voltage_config sta32x_dummy_config = { + .supply_name = "audio_vd", + .microvolts = 3300000, + .gpio = -1, + .init_data = &sta32x_dummy_initdata, +}; + +static struct platform_device sta32x_supply_dummy_device = { + .name = "reg-fixed-voltage", + .id = 1, + .dev = { + .platform_data = &sta32x_dummy_config, + }, +}; + +static struct platform_device *sta32x_regulator_devices[] = { + &sta32x_va_device, + &sta32x_supply_dummy_device, }; /** @@ -948,6 +1015,11 @@ static struct i2c_board_info raumfeld_connector_i2c_board_info __initdata = { .addr = 0x48, }; +static struct i2c_board_info raumfeld_ddx_i2c_board_info __initdata = { + .type = "sta326", + .addr = 0x1a, +}; + static struct eeti_ts_platform_data eeti_ts_pdata = { .irq_active_high = 1, }; @@ -987,7 +1059,10 @@ static void __init raumfeld_audio_init(void) else gpio_direction_output(GPIO_MCLK_RESET, 1); - platform_add_devices(ARRAY_AND_SIZE(audio_regulator_devices)); + if ((system_rev & 0xff00) == 0x0400) + platform_add_devices(ARRAY_AND_SIZE(sta32x_regulator_devices)); + else + platform_add_devices(ARRAY_AND_SIZE(cs4270_regulator_devices)); } static void __init raumfeld_common_init(void) @@ -1060,7 +1135,11 @@ static void __init raumfeld_connector_init(void) { pxa3xx_mfp_config(ARRAY_AND_SIZE(raumfeld_connector_pin_config)); spi_register_board_info(ARRAY_AND_SIZE(connector_spi_devices)); - i2c_register_board_info(0, &raumfeld_connector_i2c_board_info, 1); + + if ((system_rev & 0xff00) == 0x0400) + i2c_register_board_info(0, &raumfeld_ddx_i2c_board_info, 1); + else + i2c_register_board_info(0, &raumfeld_connector_i2c_board_info, 1); platform_device_register(&smc91x_device); @@ -1072,7 +1151,11 @@ static void __init raumfeld_speaker_init(void) { pxa3xx_mfp_config(ARRAY_AND_SIZE(raumfeld_speaker_pin_config)); spi_register_board_info(ARRAY_AND_SIZE(speaker_spi_devices)); - i2c_register_board_info(0, &raumfeld_connector_i2c_board_info, 1); + + if ((system_rev & 0xff00) == 0x0400) + i2c_register_board_info(0, &raumfeld_ddx_i2c_board_info, 1); + else + i2c_register_board_info(0, &raumfeld_connector_i2c_board_info, 1); platform_device_register(&smc91x_device); platform_device_register(&rotary_encoder_device); diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 33ebc46..e6db56a 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -152,6 +152,8 @@ config SND_SOC_RAUMFELD select SND_PXA_SOC_SSP select SND_SOC_CS4270 select SND_SOC_AK4104 + select SND_SOC_STA32X + select SND_SOC_WM8782 help Say Y if you want to add support for SoC audio on Raumfeld devices diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 1a591f1..894731b 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -21,6 +21,7 @@ #include <linux/delay.h> #include <linux/gpio.h> #include <sound/pcm.h> +#include <sound/pcm_params.h> #include <sound/soc.h> #include <asm/mach-types.h> @@ -225,6 +226,177 @@ static struct snd_soc_ops raumfeld_ak4104_ops = { .hw_params = raumfeld_ak4104_hw_params, }; +/* STA32X */ +static int raumfeld_sta32x_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + + /* FIXME: currently only fixed MCLK of 12.288MHz is available */ + snd_pcm_hw_constraint_mask64(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000); + + /* PXA DMA cannot do zero extend for 24bit samples, + * thus only 16bit (two samples packet into 32bit word) + * or 32bit samples are possible + */ + snd_pcm_hw_constraint_mask64(substream->runtime, + SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE); + + /* FIXME: we have a fixed MCLK, set it here so ALSA knows + * the supported sample rates and can resample if necessary + */ + return snd_soc_dai_set_sysclk(codec_dai, 0, 12288000, 0); +} + +static void raumfeld_sta32x_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + + /* set freq to 0 to enable all possible codec sample rates */ + snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0); +} + +static int raumfeld_sta32x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + unsigned int fmt, clk = 0; + int ret = 0; + + // FIXME: currently only fixed MCLK of 12.288MHz is available + switch (params_rate(params)) { + case 32000: + case 48000: + case 96000: + case 192000: + clk = 12288000; + break; + default: + return -EINVAL; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S32_LE: + case SNDRV_PCM_FORMAT_S32_BE: + /* this enables network mode for 2 * 32bit samples */ + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 0, 2, 32); + if (ret < 0) + return ret; + break; + case SNDRV_PCM_FORMAT_S16_LE: + case SNDRV_PCM_FORMAT_S16_BE: + /* this disables network mode */ + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0, 0, 0, 16); + if (ret < 0) + return ret; + break; + default: + return -EINVAL; + } + + fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; + + /* setup the CODEC DAI */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, 0); + if (ret < 0) + return ret; + + /* setup the CPU DAI */ + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops raumfeld_sta32x_ops = { + .startup = raumfeld_sta32x_startup, + .shutdown = raumfeld_sta32x_shutdown, + .hw_params = raumfeld_sta32x_hw_params, +}; + +/* WM8782 */ +static int raumfeld_wm8782_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + unsigned int fmt, clk = 0; + int ret = 0; + + // FIXME: currently only fixed MCLK of 12.288MHz is available + switch (params_rate(params)) { + case 48000: + clk = 12288000; + break; + default: + return -EINVAL; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + case SNDRV_PCM_FORMAT_S16_BE: + break; + default: + return -EINVAL; + } + + fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; + + /* setup the CPU DAI */ + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops raumfeld_wm8782_ops = { + .hw_params = raumfeld_wm8782_hw_params, +}; + #define DAI_LINK_CS4270 \ { \ .name = "CS4270", \ @@ -247,6 +419,28 @@ static struct snd_soc_ops raumfeld_ak4104_ops = { .codec_name = "spi0.0", \ } +#define DAI_LINK_STA32X \ +{ \ + .name = "STA32X", \ + .stream_name = "Playback", \ + .cpu_dai_name = "pxa-ssp-dai.0", \ + .platform_name = "pxa-pcm-audio", \ + .codec_dai_name = "STA32X", \ + .ops = &raumfeld_sta32x_ops, \ + .codec_name = "sta32x.0-001a", \ +} + +#define DAI_LINK_WM8782 \ +{ \ + .name = "wm8782", \ + .stream_name = "wm8782", \ + .cpu_dai_name = "pxa-ssp-dai.0", \ + .platform_name = "pxa-pcm-audio", \ + .codec_dai_name = "wm8782", \ + .codec_name = "wm8782.0", \ + .ops = &raumfeld_wm8782_ops, \ +} + static struct snd_soc_dai_link snd_soc_raumfeld_connector_dai[] = { DAI_LINK_CS4270, @@ -258,6 +452,12 @@ static struct snd_soc_dai_link snd_soc_raumfeld_speaker_dai[] = DAI_LINK_CS4270, }; +static struct snd_soc_dai_link snd_soc_raumfeld_ddx_dai[] = +{ + DAI_LINK_STA32X, + DAI_LINK_WM8782, +}; + static struct snd_soc_card snd_soc_raumfeld_connector = { .name = "Raumfeld Connector", .dai_link = snd_soc_raumfeld_connector_dai, @@ -274,7 +474,15 @@ static struct snd_soc_card snd_soc_raumfeld_speaker = { .resume_pre = raumfeld_analog_resume, }; -static struct platform_device *raumfeld_audio_device; +static struct snd_soc_card snd_soc_raumfeld_ddx = { + .name = "Raumfeld DDX", + .dai_link = snd_soc_raumfeld_ddx_dai, + .num_links = ARRAY_SIZE(snd_soc_raumfeld_ddx_dai), + .suspend_post = raumfeld_analog_suspend, + .resume_pre = raumfeld_analog_resume, +}; + +static struct platform_device *raumfeld_audio_device, *wm8782_device; static int __init raumfeld_audio_init(void) { @@ -284,26 +492,39 @@ static int __init raumfeld_audio_init(void) !machine_is_raumfeld_connector()) return 0; - max9486_client = i2c_new_device(i2c_get_adapter(0), - &max9486_hwmon_info); + if ((system_rev & 0xff00) == 0x0400) { + wm8782_device = platform_device_alloc("wm8782", 0); + if (!wm8782_device) + return -ENOMEM; - if (!max9486_client) - return -ENOMEM; + platform_device_add(wm8782_device); + } else { + max9486_client = i2c_new_device(i2c_get_adapter(0), + &max9486_hwmon_info); + + if (!max9486_client) + return -ENOMEM; - set_max9485_clk(MAX9485_MCLK_FREQ_122880); + set_max9485_clk(MAX9485_MCLK_FREQ_122880); + } - /* Register analog device */ + /* Register audio device */ raumfeld_audio_device = platform_device_alloc("soc-audio", 0); if (!raumfeld_audio_device) return -ENOMEM; - if (machine_is_raumfeld_speaker()) + if ((system_rev & 0xff00) == 0x0400) platform_set_drvdata(raumfeld_audio_device, - &snd_soc_raumfeld_speaker); - - if (machine_is_raumfeld_connector()) - platform_set_drvdata(raumfeld_audio_device, - &snd_soc_raumfeld_connector); + &snd_soc_raumfeld_ddx); + else { + if (machine_is_raumfeld_speaker()) + platform_set_drvdata(raumfeld_audio_device, + &snd_soc_raumfeld_speaker); + + if (machine_is_raumfeld_connector()) + platform_set_drvdata(raumfeld_audio_device, + &snd_soc_raumfeld_connector); + } ret = platform_device_add(raumfeld_audio_device); if (ret < 0) @@ -317,8 +538,8 @@ static void __exit raumfeld_audio_exit(void) { raumfeld_enable_audio(false); + platform_device_unregister(wm8782_device); platform_device_unregister(raumfeld_audio_device); - i2c_unregister_device(max9486_client); gpio_free(GPIO_MCLK_RESET); -- 1.7.6.4

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[alsa-devel] [PATCH] ASoC: Say how long short WM8958 DSP2 firmwares are
by Mark Brown 16 Nov '11

16 Nov '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8958-dsp2.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 0293763..39e9557 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -55,7 +55,8 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name, return 0; if (fw->size < 32) { - dev_err(codec->dev, "%s: firmware too short\n", name); + dev_err(codec->dev, "%s: firmware too short (%d bytes)\n", + name, fw->size); goto err; } -- 1.7.7.2

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[alsa-devel] [PATCH] ASoC: alc5632: rename volume/switch contols for master and speaker volumes.
by Leon Romanovsky 16 Nov '11

16 Nov '11

From: Andrey Danin <danindrey(a)mail.ru> Signed-off-by: Andrey Danin <danindrey(a)mail.ru> Signed-off-by: Leon Romanovsky <leon(a)leon.nu> --- sound/soc/codecs/alc5632.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 8a3bf71..71c8a8d 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -144,10 +144,10 @@ static const DECLARE_TLV_DB_SCALE(vdac_tlv, -3525, 075, 0); static const struct snd_kcontrol_new alc5632_vol_snd_controls[] = { /* left starts at bit 8, right at bit 0 */ /* 31 steps (5 bit), -46.5db scale */ - SOC_DOUBLE_TLV("Line Playback Volume", + SOC_DOUBLE_TLV("Speaker Playback Volume", ALC5632_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), /* bit 15 mutes left, bit 7 right */ - SOC_DOUBLE("Line Playback Switch", + SOC_DOUBLE("Speaker Playback Switch", ALC5632_SPK_OUT_VOL, 15, 7, 1, 1), SOC_DOUBLE_TLV("Headphone Playback Volume", ALC5632_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), @@ -166,9 +166,9 @@ static const struct snd_kcontrol_new alc5632_snd_controls[] = { ALC5632_PHONE_IN_VOL, 8, 31, 1, vol_tlv), SOC_DOUBLE_TLV("LineIn Capture Volume", ALC5632_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv), - SOC_DOUBLE_TLV("Stereo DAC Playback Volume", + SOC_DOUBLE_TLV("Master Playback Volume", ALC5632_STEREO_DAC_IN_VOL, 8, 0, 63, 1, vdac_tlv), - SOC_DOUBLE("Stereo DAC Playback Switch", + SOC_DOUBLE("Master Playback Switch", ALC5632_STEREO_DAC_IN_VOL, 15, 7, 1, 1), SOC_SINGLE_TLV("Mic1 Capture Volume", ALC5632_MIC_VOL, 8, 31, 1, vol_tlv), -- 1.7.4.4

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