This patch is adding support for alc562[123] codecs. It's based on the source code available in HP source code and other places.

Signed-off-by: Arnaud Patard arnaud.patard@rtp-net.org

Index: sound-2.6/sound/soc/codecs/Kconfig =================================================================== --- sound-2.6.orig/sound/soc/codecs/Kconfig 2010-10-17 11:07:13.000000000 +0200 +++ sound-2.6/sound/soc/codecs/Kconfig 2010-10-17 11:07:35.000000000 +0200 @@ -22,6 +22,7 @@ select SND_SOC_AK4535 if I2C select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C + select SND_SOC_ALC562 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C select SND_SOC_CS4270 if I2C @@ -129,6 +130,9 @@ config SND_SOC_AK4671 tristate

+config SND_SOC_ALC5623 + tristate + config SND_SOC_CQ0093VC tristate

@@ -317,3 +321,4 @@

config SND_SOC_WM9090 tristate + Index: sound-2.6/sound/soc/codecs/Makefile =================================================================== --- sound-2.6.orig/sound/soc/codecs/Makefile 2010-10-17 11:07:13.000000000 +0200 +++ sound-2.6/sound/soc/codecs/Makefile 2010-10-17 11:07:35.000000000 +0200 @@ -17,6 +17,7 @@ snd-soc-l3-objs := l3.o snd-soc-max98088-objs := max98088.o snd-soc-pcm3008-objs := pcm3008.o +snd-soc-alc5623-objs := alc5623.o snd-soc-spdif-objs := spdif_transciever.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-stac9766-objs := stac9766.o @@ -92,6 +93,7 @@ obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o +obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o Index: sound-2.6/sound/soc/codecs/alc5623.c =================================================================== --- /dev/null 1970-01-01 00:00:00.000000000 +0000 +++ sound-2.6/sound/soc/codecs/alc5623.c 2010-10-17 11:07:35.000000000 +0200 @@ -0,0 +1,1148 @@ +/* + * alc5623.c -- alc562[123] ALSA Soc Audio driver + * + * Copyright 2008 Realtek Microelectronics + * Author: flove flove@realtek.com Ethan eku@marvell.com + * + * Copyright 2010 Arnaud Patard arnaud.patard@rtp-net.org + * + * + * Based on WM8753.c + * + * 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/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/alc5623.h> + +#include "alc5623.h" + +static int caps_charge = 2000; +module_param(caps_charge, int, 0); +MODULE_PARM_DESC(caps_charge, "ALC5623 cap charge time (msecs)"); + +/* codec private data */ +struct alc5623_priv { + enum snd_soc_control_type control_type; + void *control_data; + struct mutex mutex; + u8 id; + unsigned int sysclk; + u16 reg_cache[ALC5623_VENDOR_ID2+2]; + unsigned int add_ctrl; + unsigned int jack_det_ctrl; +}; + +static void alc5623_fill_cache(struct snd_soc_codec *codec) +{ + int i, step = codec->driver->reg_cache_step; + u16 *cache = codec->reg_cache; + + /* not really efficient ... */ + for (i = 0 ; i < codec->driver->reg_cache_size ; i += step) + cache[i] = codec->hw_read(codec, i); +} + +static inline int alc5623_reset(struct snd_soc_codec *codec) +{ + return snd_soc_write(codec, ALC5623_RESET, 0); +} + +static int amp_mixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + /* to power-on/off class-d amp generators/speaker */ + /* need to write to 'index-46h' register : */ + /* so write index num (here 0x46) to reg 0x6a */ + /* and then 0xffff/0 to reg 0x6c */ + snd_soc_write(w->codec, ALC5623_HID_CTRL_INDEX, 0x46); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0xFFFF); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0); + break; + } + + return 0; +} + +/* + * ALC5623 Controls + */ + +static const DECLARE_TLV_DB_SCALE(vol_tlv, -3450, 150, 0); +static const DECLARE_TLV_DB_SCALE(hp_tlv, -4650, 150, 0); +static const DECLARE_TLV_DB_SCALE(adc_rec_tlv, -1650, 150, 0); +static const unsigned int boost_tlv[] = { + TLV_DB_RANGE_HEAD(3), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), +}; +static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0); + +static const struct snd_kcontrol_new rt5621_vol_snd_controls[] = { + SOC_DOUBLE_TLV("Speaker Playback Volume", + ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Speaker Playback Switch", + ALC5623_SPK_OUT_VOL, 15, 7, 1, 1), + SOC_DOUBLE_TLV("Headphone Playback Volume", + ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Headphone Playback Switch", + ALC5623_HP_OUT_VOL, 15, 7, 1, 1), +}; + +static const struct snd_kcontrol_new rt5622_vol_snd_controls[] = { + SOC_DOUBLE_TLV("Speaker Playback Volume", + ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Speaker Playback Switch", + ALC5623_SPK_OUT_VOL, 15, 7, 1, 1), + SOC_DOUBLE_TLV("Line Playback Volume", + ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Line Playback Switch", + ALC5623_HP_OUT_VOL, 15, 7, 1, 1), +}; + +static const struct snd_kcontrol_new alc5623_vol_snd_controls[] = { + SOC_DOUBLE_TLV("Line Playback Volume", + ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Line Playback Switch", + ALC5623_SPK_OUT_VOL, 15, 7, 1, 1), + SOC_DOUBLE_TLV("Headphone Playback Volume", + ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Headphone Playback Switch", + ALC5623_HP_OUT_VOL, 15, 7, 1, 1), +}; + +static const struct snd_kcontrol_new alc5623_snd_controls[] = { + SOC_DOUBLE_TLV("Auxout Playback Volume", + ALC5623_MONO_AUX_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Auxout Playback Switch", + ALC5623_MONO_AUX_OUT_VOL, 15, 7, 1, 1), + SOC_DOUBLE_TLV("PCM Playback Volume", + ALC5623_STEREO_DAC_VOL, 8, 0, 31, 1, vol_tlv), + SOC_DOUBLE_TLV("AuxI Capture Volume", + ALC5623_AUXIN_VOL, 8, 0, 31, 1, vol_tlv), + SOC_DOUBLE_TLV("LineIn Capture Volume", + ALC5623_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv), + SOC_SINGLE_TLV("Mic1 Capture Volume", + ALC5623_MIC_VOL, 8, 31, 1, vol_tlv), + SOC_SINGLE_TLV("Mic2 Capture Volume", + ALC5623_MIC_VOL, 0, 31, 1, vol_tlv), + SOC_DOUBLE_TLV("Rec Capture Volume", + ALC5623_ADC_REC_GAIN, 7, 0, 31, 0, adc_rec_tlv), + SOC_SINGLE_TLV("Mic 1 Boost Volume", + ALC5623_MIC_CTRL, 10, 2, 0, boost_tlv), + SOC_SINGLE_TLV("Mic 2 Boost Volume", + ALC5623_MIC_CTRL, 8, 2, 0, boost_tlv), + SOC_SINGLE_TLV("Digital Boost Volume", + ALC5623_ADD_CTRL_REG, 4, 3, 0, dig_tlv), +}; + +/* + * DAPM Controls + */ +static const struct snd_kcontrol_new alc5623_hp_mixer_controls[] = { +SOC_DAPM_SINGLE("LI2HP Playback Switch", ALC5623_LINE_IN_VOL, 15, 1, 1), +SOC_DAPM_SINGLE("AUXI2HP Playback Switch", ALC5623_AUXIN_VOL, 15, 1, 1), +SOC_DAPM_SINGLE("MIC12HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 15, 1, 1), +SOC_DAPM_SINGLE("MIC22HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 7, 1, 1), +SOC_DAPM_SINGLE("DAC2HP Playback Switch", ALC5623_STEREO_DAC_VOL, 15, 1, 1), +}; + +static const struct snd_kcontrol_new alc5623_hpl_mixer_controls[] = { +SOC_DAPM_SINGLE("ADC2HP_L Playback Switch", ALC5623_ADC_REC_GAIN, 15, 1, 1), +}; + +static const struct snd_kcontrol_new alc5623_hpr_mixer_controls[] = { +SOC_DAPM_SINGLE("ADC2HP_R Playback Switch", ALC5623_ADC_REC_GAIN, 14, 1, 1), +}; + +static const struct snd_kcontrol_new alc5623_mono_mixer_controls[] = { +SOC_DAPM_SINGLE("ADC2MONO_L Playback Switch", ALC5623_ADC_REC_GAIN, 13, 1, 1), +SOC_DAPM_SINGLE("ADC2MONO_R Playback Switch", ALC5623_ADC_REC_GAIN, 12, 1, 1), +SOC_DAPM_SINGLE("LI2MONO Playback Switch", ALC5623_LINE_IN_VOL, 13, 1, 1), +SOC_DAPM_SINGLE("AUXI2MONO Playback Switch", ALC5623_AUXIN_VOL, 13, 1, 1), +SOC_DAPM_SINGLE("MIC12MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 13, 1, 1), +SOC_DAPM_SINGLE("MIC22MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 5, 1, 1), +SOC_DAPM_SINGLE("DAC2MONO Playback Switch", ALC5623_STEREO_DAC_VOL, 13, 1, 1), +}; + +static const struct snd_kcontrol_new alc5623_speaker_mixer_controls[] = { +SOC_DAPM_SINGLE("LI2SPK Playback Switch", ALC5623_LINE_IN_VOL, 14, 1, 1), +SOC_DAPM_SINGLE("AUXI2SPK Playback Switch", ALC5623_AUXIN_VOL, 14, 1, 1), +SOC_DAPM_SINGLE("MIC12SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 14, 1, 1), +SOC_DAPM_SINGLE("MIC22SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 6, 1, 1), +SOC_DAPM_SINGLE("DAC2SPK Playback Switch", ALC5623_STEREO_DAC_VOL, 14, 1, 1), +}; + +/* Left Record Mixer */ +static const struct snd_kcontrol_new alc5623_captureL_mixer_controls[] = { +SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 14, 1, 1), +SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 13, 1, 1), +SOC_DAPM_SINGLE("LineInL Capture Switch", ALC5623_ADC_REC_MIXER, 12, 1, 1), +SOC_DAPM_SINGLE("Left AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 11, 1, 1), +SOC_DAPM_SINGLE("HPMixerL Capture Switch", ALC5623_ADC_REC_MIXER, 10, 1, 1), +SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 9, 1, 1), +SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 8, 1, 1), +}; + +/* Right Record Mixer */ +static const struct snd_kcontrol_new alc5623_captureR_mixer_controls[] = { +SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 6, 1, 1), +SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 5, 1, 1), +SOC_DAPM_SINGLE("LineInR Capture Switch", ALC5623_ADC_REC_MIXER, 4, 1, 1), +SOC_DAPM_SINGLE("Right AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 3, 1, 1), +SOC_DAPM_SINGLE("HPMixerR Capture Switch", ALC5623_ADC_REC_MIXER, 2, 1, 1), +SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 1, 1, 1), +SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 0, 1, 1), +}; + +static const char *alc5623_spk_n_sour_sel[] = { + "RN/-R", "RP/+R", "LN/-R", "Vmid" }; +static const char *alc5623_hpl_out_input_sel[] = { + "Vmid", "HP Left Mix"}; +static const char *alc5623_hpr_out_input_sel[] = { + "Vmid", "HP Right Mix"}; +static const char *alc5623_spkout_input_sel[] = { + "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; +static const char *alc5623_aux_out_input_sel[] = { + "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; + +/* auxout output mux */ +static const struct soc_enum alc5623_aux_out_input_enum = +SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 6, 4, alc5623_aux_out_input_sel); +static const struct snd_kcontrol_new alc5623_auxout_mux_controls = +SOC_DAPM_ENUM("Route", alc5623_aux_out_input_enum); + +/* speaker output mux */ +static const struct soc_enum alc5623_spkout_input_enum = +SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 10, 4, alc5623_spkout_input_sel); +static const struct snd_kcontrol_new alc5623_spkout_mux_controls = +SOC_DAPM_ENUM("Route", alc5623_spkout_input_enum); + +/* headphone left output mux */ +static const struct soc_enum alc5623_hpl_out_input_enum = +SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 9, 2, alc5623_hpl_out_input_sel); +static const struct snd_kcontrol_new alc5623_hpl_out_mux_controls = +SOC_DAPM_ENUM("Route", alc5623_hpl_out_input_enum); + +/* headphone right output mux */ +static const struct soc_enum alc5623_hpr_out_input_enum = +SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 8, 2, alc5623_hpr_out_input_sel); +static const struct snd_kcontrol_new alc5623_hpr_out_mux_controls = +SOC_DAPM_ENUM("Route", alc5623_hpr_out_input_enum); + +/* speaker output N select */ +static const struct soc_enum alc5623_spk_n_sour_enum = +SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 14, 4, alc5623_spk_n_sour_sel); +static const struct snd_kcontrol_new alc5623_spkoutn_mux_controls = +SOC_DAPM_ENUM("Route", alc5623_spk_n_sour_enum); + +static const struct snd_soc_dapm_widget alc5623_dapm_widgets[] = { +/* Muxes */ +SND_SOC_DAPM_MUX("AuxOut Mux", SND_SOC_NOPM, 0, 0, + &alc5623_auxout_mux_controls), +SND_SOC_DAPM_MUX("SpeakerOut Mux", SND_SOC_NOPM, 0, 0, + &alc5623_spkout_mux_controls), +SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, + &alc5623_hpl_out_mux_controls), +SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, + &alc5623_hpr_out_mux_controls), +SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0, + &alc5623_spkoutn_mux_controls), + +/* output mixers */ +SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0, + &alc5623_hp_mixer_controls[0], + ARRAY_SIZE(alc5623_hp_mixer_controls)), +SND_SOC_DAPM_MIXER("HPR Mix", ALC5623_PWR_MANAG_ADD2, 4, 0, + &alc5623_hpr_mixer_controls[0], + ARRAY_SIZE(alc5623_hpr_mixer_controls)), +SND_SOC_DAPM_MIXER("HPL Mix", ALC5623_PWR_MANAG_ADD2, 5, 0, + &alc5623_hpl_mixer_controls[0], + ARRAY_SIZE(alc5623_hpl_mixer_controls)), +SND_SOC_DAPM_MIXER("HPOut Mix", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Mono Mix", ALC5623_PWR_MANAG_ADD2, 2, 0, + &alc5623_mono_mixer_controls[0], + ARRAY_SIZE(alc5623_mono_mixer_controls)), +SND_SOC_DAPM_MIXER("Speaker Mix", ALC5623_PWR_MANAG_ADD2, 3, 0, + &alc5623_speaker_mixer_controls[0], + ARRAY_SIZE(alc5623_speaker_mixer_controls)), + +/* input mixers */ +SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5623_PWR_MANAG_ADD2, 1, 0, + &alc5623_captureL_mixer_controls[0], + ARRAY_SIZE(alc5623_captureL_mixer_controls)), +SND_SOC_DAPM_MIXER("Right Capture Mix", ALC5623_PWR_MANAG_ADD2, 0, 0, + &alc5623_captureR_mixer_controls[0], + ARRAY_SIZE(alc5623_captureR_mixer_controls)), + +SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", + ALC5623_PWR_MANAG_ADD2, 9, 0), +SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", + ALC5623_PWR_MANAG_ADD2, 8, 0), +SND_SOC_DAPM_MIXER("I2S Mix", ALC5623_PWR_MANAG_ADD1, 15, 0, NULL, 0), +SND_SOC_DAPM_MIXER("AuxI Mix", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Line Mix", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", + ALC5623_PWR_MANAG_ADD2, 7, 0), +SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", + ALC5623_PWR_MANAG_ADD2, 6, 0), +SND_SOC_DAPM_PGA("Left Headphone", ALC5623_PWR_MANAG_ADD3, 10, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right Headphone", ALC5623_PWR_MANAG_ADD3, 9, 0, NULL, 0), +SND_SOC_DAPM_PGA("SpeakerOut", ALC5623_PWR_MANAG_ADD3, 12, 0, NULL, 0), +SND_SOC_DAPM_PGA("Left AuxOut", ALC5623_PWR_MANAG_ADD3, 14, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right AuxOut", ALC5623_PWR_MANAG_ADD3, 13, 0, NULL, 0), +SND_SOC_DAPM_PGA("Left LineIn", ALC5623_PWR_MANAG_ADD3, 7, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right LineIn", ALC5623_PWR_MANAG_ADD3, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("Left AuxI", ALC5623_PWR_MANAG_ADD3, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right AuxI", ALC5623_PWR_MANAG_ADD3, 4, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC1 PGA", ALC5623_PWR_MANAG_ADD3, 3, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC2 PGA", ALC5623_PWR_MANAG_ADD3, 2, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC1 Pre Amp", ALC5623_PWR_MANAG_ADD3, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC2 Pre Amp", ALC5623_PWR_MANAG_ADD3, 0, 0, NULL, 0), +SND_SOC_DAPM_MICBIAS("Mic Bias1", ALC5623_PWR_MANAG_ADD1, 11, 0), + +SND_SOC_DAPM_OUTPUT("AUXOUTL"), +SND_SOC_DAPM_OUTPUT("AUXOUTR"), +SND_SOC_DAPM_OUTPUT("HPL"), +SND_SOC_DAPM_OUTPUT("HPR"), +SND_SOC_DAPM_OUTPUT("SPKOUT"), +SND_SOC_DAPM_OUTPUT("SPKOUTN"), +SND_SOC_DAPM_INPUT("LINEINL"), +SND_SOC_DAPM_INPUT("LINEINR"), +SND_SOC_DAPM_INPUT("AUXINL"), +SND_SOC_DAPM_INPUT("AUXINR"), +SND_SOC_DAPM_INPUT("MIC1"), +SND_SOC_DAPM_INPUT("MIC2"), +SND_SOC_DAPM_VMID("Vmid"), +}; + +static const char *alc5623_amp_names[] = {"AB Amp", "D Amp"}; +static const struct soc_enum alc5623_amp_enum = + SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 13, 2, alc5623_amp_names); +static const struct snd_kcontrol_new alc5623_amp_mux_controls = + SOC_DAPM_ENUM("Route", alc5623_amp_enum); + +static const struct snd_soc_dapm_widget alc5623_dapm_amp_widgets[] = { +SND_SOC_DAPM_PGA_E("D Amp", ALC5623_PWR_MANAG_ADD2, 14, 0, NULL, 0, + amp_mixer_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_PGA("AB Amp", ALC5623_PWR_MANAG_ADD2, 15, 0, NULL, 0), +SND_SOC_DAPM_MUX("AB-D Amp Mux", SND_SOC_NOPM, 0, 0, + &alc5623_amp_mux_controls), +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* virtual mixer - mixes left & right channels */ + {"I2S Mix", NULL, "Left DAC"}, + {"I2S Mix", NULL, "Right DAC"}, + {"Line Mix", NULL, "Right LineIn"}, + {"Line Mix", NULL, "Left LineIn"}, + {"AuxI Mix", NULL, "Left AuxI"}, + {"AuxI Mix", NULL, "Right AuxI"}, + {"AUXOUTL", NULL, "Left AuxOut"}, + {"AUXOUTR", NULL, "Right AuxOut"}, + + /* HP mixer */ + {"HPL Mix", "ADC2HP_L Playback Switch", "Left Capture Mix"}, + {"HPL Mix", NULL, "HP Mix"}, + {"HPR Mix", "ADC2HP_R Playback Switch", "Right Capture Mix"}, + {"HPR Mix", NULL, "HP Mix"}, + {"HP Mix", "LI2HP Playback Switch", "Line Mix"}, + {"HP Mix", "AUXI2HP Playback Switch", "AuxI Mix"}, + {"HP Mix", "MIC12HP Playback Switch", "MIC1 PGA"}, + {"HP Mix", "MIC22HP Playback Switch", "MIC2 PGA"}, + {"HP Mix", "DAC2HP Playback Switch", "I2S Mix"}, + + /* speaker mixer */ + {"Speaker Mix", "LI2SPK Playback Switch", "Line Mix"}, + {"Speaker Mix", "AUXI2SPK Playback Switch", "AuxI Mix"}, + {"Speaker Mix", "MIC12SPK Playback Switch", "MIC1 PGA"}, + {"Speaker Mix", "MIC22SPK Playback Switch", "MIC2 PGA"}, + {"Speaker Mix", "DAC2SPK Playback Switch", "I2S Mix"}, + + /* mono mixer */ + {"Mono Mix", "ADC2MONO_L Playback Switch", "Left Capture Mix"}, + {"Mono Mix", "ADC2MONO_R Playback Switch", "Right Capture Mix"}, + {"Mono Mix", "LI2MONO Playback Switch", "Line Mix"}, + {"Mono Mix", "AUXI2MONO Playback Switch", "AuxI Mix"}, + {"Mono Mix", "MIC12MONO Playback Switch", "MIC1 PGA"}, + {"Mono Mix", "MIC22MONO Playback Switch", "MIC2 PGA"}, + {"Mono Mix", "DAC2MONO Playback Switch", "I2S Mix"}, + + /* Left record mixer */ + {"Left Capture Mix", "LineInL Capture Switch", "LINEINL"}, + {"Left Capture Mix", "Left AuxI Capture Switch", "AUXINL"}, + {"Left Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"}, + {"Left Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"}, + {"Left Capture Mix", "HPMixerL Capture Switch", "HPL Mix"}, + {"Left Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"}, + {"Left Capture Mix", "MonoMixer Capture Switch", "Mono Mix"}, + + /*Right record mixer */ + {"Right Capture Mix", "LineInR Capture Switch", "LINEINR"}, + {"Right Capture Mix", "Right AuxI Capture Switch", "AUXINR"}, + {"Right Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"}, + {"Right Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"}, + {"Right Capture Mix", "HPMixerR Capture Switch", "HPR Mix"}, + {"Right Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"}, + {"Right Capture Mix", "MonoMixer Capture Switch", "Mono Mix"}, + + /* headphone left mux */ + {"Left Headphone Mux", "HP Left Mix", "HPL Mix"}, + {"Left Headphone Mux", "Vmid", "Vmid"}, + + /* headphone right mux */ + {"Right Headphone Mux", "HP Right Mix", "HPR Mix"}, + {"Right Headphone Mux", "Vmid", "Vmid"}, + + /* speaker out mux */ + {"SpeakerOut Mux", "Vmid", "Vmid"}, + {"SpeakerOut Mux", "HPOut Mix", "HPOut Mix"}, + {"SpeakerOut Mux", "Speaker Mix", "Speaker Mix"}, + {"SpeakerOut Mux", "Mono Mix", "Mono Mix"}, + + /* Mono/Aux Out mux */ + {"AuxOut Mux", "Vmid", "Vmid"}, + {"AuxOut Mux", "HPOut Mix", "HPOut Mix"}, + {"AuxOut Mux", "Speaker Mix", "Speaker Mix"}, + {"AuxOut Mux", "Mono Mix", "Mono Mix"}, + + /* output pga */ + {"HPL", NULL, "Left Headphone"}, + {"Left Headphone", NULL, "Left Headphone Mux"}, + {"HPR", NULL, "Right Headphone"}, + {"Right Headphone", NULL, "Right Headphone Mux"}, + {"Left AuxOut", NULL, "AuxOut Mux"}, + {"Right AuxOut", NULL, "AuxOut Mux"}, + + /* input pga */ + {"Left LineIn", NULL, "LINEINL"}, + {"Right LineIn", NULL, "LINEINR"}, + {"Left AuxI", NULL, "AUXINL"}, + {"Right AuxI", NULL, "AUXINR"}, + {"MIC1 Pre Amp", NULL, "MIC1"}, + {"MIC2 Pre Amp", NULL, "MIC2"}, + {"MIC1 PGA", NULL, "MIC1 Pre Amp"}, + {"MIC2 PGA", NULL, "MIC2 Pre Amp"}, + + /* left ADC */ + {"Left ADC", NULL, "Left Capture Mix"}, + + /* right ADC */ + {"Right ADC", NULL, "Right Capture Mix"}, + + {"SpeakerOut N Mux", "RN/-R", "SpeakerOut"}, + {"SpeakerOut N Mux", "RP/+R", "SpeakerOut"}, + {"SpeakerOut N Mux", "LN/-R", "SpeakerOut"}, + {"SpeakerOut N Mux", "Vmid", "Vmid"}, + + {"SPKOUT", NULL, "SpeakerOut"}, + {"SPKOUTN", NULL, "SpeakerOut N Mux"}, +}; + +static const struct snd_soc_dapm_route intercon_spk[] = { + {"SpeakerOut", NULL, "SpeakerOut Mux"}, +}; + +static const struct snd_soc_dapm_route intercon_amp_spk[] = { + {"AB Amp", NULL, "SpeakerOut Mux"}, + {"D Amp", NULL, "SpeakerOut Mux"}, + {"AB-D Amp Mux", "AB Amp", "AB Amp"}, + {"AB-D Amp Mux", "D Amp", "D Amp"}, + {"SpeakerOut", NULL, "AB-D Amp Mux"}, +}; + +/* PLL divisors */ +struct _pll_div { + u32 pll_in; + u32 pll_out; + u16 regvalue; +}; + +/* Note : pll code from original alc5623 driver. Not sure of how good it is */ +/* usefull only for master mode */ +static const struct _pll_div codec_master_pll_div[] = { + + { 2048000, 8192000, 0x0ea0}, + { 3686400, 8192000, 0x4e27}, + { 12000000, 8192000, 0x456b}, + { 13000000, 8192000, 0x495f}, + { 13100000, 8192000, 0x0320}, + { 2048000, 11289600, 0xf637}, + { 3686400, 11289600, 0x2f22}, + { 12000000, 11289600, 0x3e2f}, + { 13000000, 11289600, 0x4d5b}, + { 13100000, 11289600, 0x363b}, + { 2048000, 16384000, 0x1ea0}, + { 3686400, 16384000, 0x9e27}, + { 12000000, 16384000, 0x452b}, + { 13000000, 16384000, 0x542f}, + { 13100000, 16384000, 0x03a0}, + { 2048000, 16934400, 0xe625}, + { 3686400, 16934400, 0x9126}, + { 12000000, 16934400, 0x4d2c}, + { 13000000, 16934400, 0x742f}, + { 13100000, 16934400, 0x3c27}, + { 2048000, 22579200, 0x2aa0}, + { 3686400, 22579200, 0x2f20}, + { 12000000, 22579200, 0x7e2f}, + { 13000000, 22579200, 0x742f}, + { 13100000, 22579200, 0x3c27}, + { 2048000, 24576000, 0x2ea0}, + { 3686400, 24576000, 0xee27}, + { 12000000, 24576000, 0x2915}, + { 13000000, 24576000, 0x772e}, + { 13100000, 24576000, 0x0d20}, +}; + +static const struct _pll_div codec_slave_pll_div[] = { + + { 1024000, 16384000, 0x3ea0}, + { 1411200, 22579200, 0x3ea0}, + { 1536000, 24576000, 0x3ea0}, + { 2048000, 16384000, 0x1ea0}, + { 2822400, 22579200, 0x1ea0}, + { 3072000, 24576000, 0x1ea0}, + +}; + +static int alc5623_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + int i; + struct snd_soc_codec *codec = codec_dai->codec; + int gbl_clk = 0, pll_div = 0; + u16 reg; + + if (pll_id < ALC5623_PLL_FR_MCLK || pll_id > ALC5623_PLL_FR_BCK) + return -ENODEV; + + /* Disable PLL power */ + snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2, + ALC5623_PWR_ADD2_PLL, + 0); + + /* pll is not used in slave mode */ + reg = snd_soc_read(codec, ALC5623_DAI_CONTROL); + if (reg & ALC5623_DAI_SDP_SLAVE_MODE) + return 0; + + if (!freq_in || !freq_out) + return 0; + + switch (pll_id) { + case ALC5623_PLL_FR_MCLK: + for (i = 0; i < ARRAY_SIZE(codec_master_pll_div); i++) { + if (codec_master_pll_div[i].pll_in == freq_in + && codec_master_pll_div[i].pll_out == freq_out) { + /* PLL source from MCLK */ + pll_div = codec_master_pll_div[i].regvalue; + break; + } + } + break; + case ALC5623_PLL_FR_BCK: + for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) { + if (codec_slave_pll_div[i].pll_in == freq_in + && codec_slave_pll_div[i].pll_out == freq_out) { + /* PLL source from Bitclk */ + gbl_clk = ALC5623_GBL_CLK_PLL_SOUR_SEL_BITCLK; + pll_div = codec_slave_pll_div[i].regvalue; + break; + } + } + break; + default: + return -EINVAL; + } + + if (!pll_div) + return -EINVAL; + + snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk); + snd_soc_write(codec, ALC5623_PLL_CTRL, pll_div); + snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2, + ALC5623_PWR_ADD2_PLL, + ALC5623_PWR_ADD2_PLL); + gbl_clk |= ALC5623_GBL_CLK_SYS_SOUR_SEL_PLL; + snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk); + + return 0; +} + +struct _coeff_div { + u16 fs; + u16 regvalue; +}; + +/* codec hifi mclk (after PLL) clock divider coefficients */ +/* values inspired from column BCLK=32Fs of Appendix A table */ +static const struct _coeff_div coeff_div[] = { + {256*8, 0x3a69}, + {384*8, 0x3c6b}, + {256*4, 0x2a69}, + {384*4, 0x2c6b}, + {256*2, 0x1a69}, + {384*2, 0x1c6b}, + {256*1, 0x0a69}, + {384*1, 0x0c6b}, +}; + +static int get_coeff(struct snd_soc_codec *codec, int rate) +{ + struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].fs * rate == alc5623->sysclk) + return i; + } + return -EINVAL; +} + +/* + * Clock after PLL and dividers + */ +static int alc5623_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); + + switch (freq) { + case 8192000: + case 11289600: + case 12288000: + case 16384000: + case 16934400: + case 18432000: + case 22579200: + case 24576000: + alc5623->sysclk = freq; + return 0; + } + return -EINVAL; +} + +static int alc5623_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface = ALC5623_DAI_SDP_MASTER_MODE; + break; + case SND_SOC_DAIFMT_CBS_CFS: + iface = ALC5623_DAI_SDP_SLAVE_MODE; + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= ALC5623_DAI_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + iface |= ALC5623_DAI_I2S_DF_RIGHT; + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= ALC5623_DAI_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= ALC5623_DAI_I2S_DF_PCM; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= ALC5623_DAI_I2S_DF_PCM | ALC5623_DAI_I2S_PCM_MODE; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL; + break; + case SND_SOC_DAIFMT_NB_IF: + break; + default: + return -EINVAL; + } + + return snd_soc_write(codec, ALC5623_DAI_CONTROL, iface); +} + +static int alc5623_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 alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); + int coeff, rate; + u16 iface; + + iface = snd_soc_read(codec, ALC5623_DAI_CONTROL); + iface &= ~ALC5623_DAI_I2S_DL_MASK; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + iface |= ALC5623_DAI_I2S_DL_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= ALC5623_DAI_I2S_DL_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= ALC5623_DAI_I2S_DL_24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= ALC5623_DAI_I2S_DL_32; + break; + default: + return -EINVAL; + } + + /* set iface & srate */ + snd_soc_write(codec, ALC5623_DAI_CONTROL, iface); + rate = params_rate(params); + coeff = get_coeff(codec, rate); + if (coeff < 0) + return -EINVAL; + + coeff = coeff_div[coeff].regvalue; + dev_dbg(codec->dev, "%s: sysclk=%d,rate=%d,coeff=0x%04x\n", + __func__, alc5623->sysclk, rate, coeff); + snd_soc_write(codec, ALC5623_STEREO_AD_DA_CLK_CTRL, coeff); + + return 0; +} + +static int alc5623_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 hp_mute = ALC5623_MISC_M_DAC_L_INPUT | ALC5623_MISC_M_DAC_R_INPUT; + u16 mute_reg = snd_soc_read(codec, ALC5623_MISC_CTRL) & ~hp_mute; + + if (mute) + mute_reg |= hp_mute; + + return snd_soc_write(codec, ALC5623_MISC_CTRL, mute_reg); +} + +#define ALC5623_ADD2_POWER_EN (ALC5623_PWR_ADD2_VREF \ + | ALC5623_PWR_ADD2_DAC_REF_CIR) + +#define ALC5623_ADD3_POWER_EN (ALC5623_PWR_ADD3_MAIN_BIAS \ + | ALC5623_PWR_ADD3_MIC1_BOOST_AD) + +#define ALC5623_ADD1_POWER_EN \ + (ALC5623_PWR_ADD1_SHORT_CURR_DET_EN | ALC5623_PWR_ADD1_SOFTGEN_EN \ + | ALC5623_PWR_ADD1_DEPOP_BUF_HP | ALC5623_PWR_ADD1_HP_OUT_AMP \ + | ALC5623_PWR_ADD1_HP_OUT_ENH_AMP) + +#define ALC5623_ADD1_POWER_EN_5622 \ + (ALC5623_PWR_ADD1_SHORT_CURR_DET_EN \ + | ALC5623_PWR_ADD1_HP_OUT_AMP) + +static void enable_power_depop(struct snd_soc_codec *codec) +{ + struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); + + snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD1, + ALC5623_PWR_ADD1_SOFTGEN_EN, + ALC5623_PWR_ADD1_SOFTGEN_EN); + + snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, ALC5623_ADD3_POWER_EN); + + snd_soc_update_bits(codec, ALC5623_MISC_CTRL, + ALC5623_MISC_HP_DEPOP_MODE2_EN, + ALC5623_MISC_HP_DEPOP_MODE2_EN); + + msleep(500); + + snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, ALC5623_ADD2_POWER_EN); + + /* avoid writing '1' into 5622 reserved bits */ + if (alc5623->id == 0x22) + snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, + ALC5623_ADD1_POWER_EN_5622); + else + snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, + ALC5623_ADD1_POWER_EN); + + /* disable HP Depop2 */ + snd_soc_update_bits(codec, ALC5623_MISC_CTRL, + ALC5623_MISC_HP_DEPOP_MODE2_EN, + 0); + +} + +static int alc5623_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + enable_power_depop(codec); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* everything off except vref/vmid, */ + snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, + ALC5623_PWR_ADD2_VREF); + snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, + ALC5623_PWR_ADD3_MAIN_BIAS); + break; + case SND_SOC_BIAS_OFF: + /* everything off, dac mute, inactive */ + snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, 0); + snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, 0); + snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, 0); + break; + } + codec->bias_level = level; + return 0; +} + +#define ALC5623_FORMATS (SNDRV_PCM_FMTBIT_S16_LE \ + | SNDRV_PCM_FMTBIT_S24_LE \ + | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops alc5623_dai_ops = { + .hw_params = alc5623_pcm_hw_params, + .digital_mute = alc5623_mute, + .set_fmt = alc5623_set_dai_fmt, + .set_sysclk = alc5623_set_dai_sysclk, + .set_pll = alc5623_set_dai_pll, +}; + +static struct snd_soc_dai_driver alc5623_dai = { + .name = "alc5623-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ALC5623_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ALC5623_FORMATS,}, + + .ops = &alc5623_dai_ops, +}; + +static void alc5623_work(struct work_struct *work) +{ + struct snd_soc_codec *codec = + container_of(work, struct snd_soc_codec, delayed_work.work); + alc5623_set_bias_level(codec, codec->bias_level); +} + +static int alc5623_suspend(struct snd_soc_codec *codec, pm_message_t mesg) +{ + alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int alc5623_resume(struct snd_soc_codec *codec) +{ + int i, step = codec->driver->reg_cache_step; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 2 ; i < codec->driver->reg_cache_size ; i += step) + snd_soc_write(codec, i, cache[i]); + + alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* charge alc5623 caps */ + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) { + alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + codec->bias_level = SND_SOC_BIAS_ON; + alc5623_set_bias_level(codec, codec->bias_level); + } + + return 0; +} + +static int alc5623_probe(struct snd_soc_codec *codec) +{ + struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); + int ret; + + INIT_DELAYED_WORK(&codec->delayed_work, alc5623_work); + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, alc5623->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + alc5623_reset(codec); + alc5623_fill_cache(codec); + + /* power on device */ + alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + if (alc5623->add_ctrl) { + snd_soc_write(codec, ALC5623_ADD_CTRL_REG, + alc5623->add_ctrl); + } + + if (alc5623->jack_det_ctrl) { + snd_soc_write(codec, ALC5623_JACK_DET_CTRL, + alc5623->jack_det_ctrl); + } + + switch (alc5623->id) { + default: + case 0x21: + snd_soc_add_controls(codec, rt5621_vol_snd_controls, + ARRAY_SIZE(rt5621_vol_snd_controls)); + break; + case 0x22: + snd_soc_add_controls(codec, rt5622_vol_snd_controls, + ARRAY_SIZE(rt5622_vol_snd_controls)); + break; + case 0x23: + snd_soc_add_controls(codec, alc5623_vol_snd_controls, + ARRAY_SIZE(alc5623_vol_snd_controls)); + break; + } + + snd_soc_add_controls(codec, alc5623_snd_controls, + ARRAY_SIZE(alc5623_snd_controls)); + + snd_soc_dapm_new_controls(codec, alc5623_dapm_widgets, + ARRAY_SIZE(alc5623_dapm_widgets)); + + /* set up audio path interconnects */ + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + switch (alc5623->id) { + default: + case 0x21: + case 0x22: + snd_soc_dapm_new_controls(codec, alc5623_dapm_amp_widgets, + ARRAY_SIZE(alc5623_dapm_amp_widgets)); + snd_soc_dapm_add_routes(codec, intercon_amp_spk, + ARRAY_SIZE(intercon_amp_spk)); + break; + case 0x23: + snd_soc_dapm_add_routes(codec, intercon_spk, + ARRAY_SIZE(intercon_spk)); + break; + } + + return ret; +} + +/* + * This function forces any delayed work to be queued and run. + */ +static int run_delayed_work(struct delayed_work *dwork) +{ + int ret; + + /* cancel any work waiting to be queued. */ + ret = cancel_delayed_work(dwork); + + /* if there was any work waiting then we run it now and + * wait for it's completion */ + if (ret) { + schedule_delayed_work(dwork, 0); + flush_scheduled_work(); + } + return ret; +} + +/* power down chip */ +static int alc5623_remove(struct snd_soc_codec *codec) +{ + run_delayed_work(&codec->delayed_work); + alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_device_alc5623 = { + .probe = alc5623_probe, + .remove = alc5623_remove, + .suspend = alc5623_suspend, + .resume = alc5623_resume, + .set_bias_level = alc5623_set_bias_level, + .reg_cache_size = ALC5623_VENDOR_ID2+2, + .reg_word_size = sizeof(u16), + .reg_cache_step = 2, +}; + +/* + * ALC5623 2 wire address is determined by A1 pin + * state during powerup. + * low = 0x1a + * high = 0x1b + */ +static int alc5623_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct alc5623_platform_data *pdata; + struct alc5623_priv *alc5623; + int ret, vid1, vid2; + + vid1 = i2c_smbus_read_word_data(client, ALC5623_VENDOR_ID1); + if (vid1 < 0) { + dev_err(&client->dev, "failed to read I2C\n"); + return -EIO; + } + vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8); + + vid2 = i2c_smbus_read_byte_data(client, ALC5623_VENDOR_ID2); + if (vid2 < 0) { + dev_err(&client->dev, "failed to read I2C\n"); + return -EIO; + } + + if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) { + dev_err(&client->dev, "unknown or wrong codec\n"); + dev_err(&client->dev, "Expected %x:%lx, got %x:%x\n", + 0x10ec, id->driver_data, + vid1, vid2); + return -ENODEV; + } + + dev_dbg(&client->dev, "Found codec id : alc56%02x\n", vid2); + + alc5623 = kzalloc(sizeof(struct alc5623_priv), GFP_KERNEL); + if (alc5623 == NULL) { + ret = -ENOMEM; + goto err; + } + + pdata = client->dev.platform_data; + if (pdata) { + alc5623->add_ctrl = pdata->add_ctrl; + alc5623->jack_det_ctrl = pdata->jack_det_ctrl; + } + + alc5623->id = vid2; + switch (alc5623->id) { + case 0x21: + alc5623_dai.name = "alc5621-hifi"; + break; + case 0x22: + alc5623_dai.name = "alc5622-hifi"; + break; + default: + case 0x23: + alc5623_dai.name = "alc5623-hifi"; + break; + } + + i2c_set_clientdata(client, alc5623); + alc5623->control_data = client; + alc5623->control_type = SND_SOC_I2C; + mutex_init(&alc5623->mutex); + + ret = snd_soc_register_codec(&client->dev, + &soc_codec_device_alc5623, &alc5623_dai, 1); + if (ret != 0) { + dev_err(&client->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + return 0; + +err: + return ret; +} + +static int alc5623_i2c_remove(struct i2c_client *client) +{ + struct alc5623_priv *alc5623 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + kfree(alc5623); + return 0; +} + +static const struct i2c_device_id alc5623_i2c_table[] = { + {"alc5621", 0x21}, + {"alc5622", 0x22}, + {"alc5623", 0x23}, + {} +}; +MODULE_DEVICE_TABLE(i2c, alc5623_i2c_table); + +/* i2c codec control layer */ +static struct i2c_driver alc5623_i2c_driver = { + .driver = { + .name = "alc562x-codec", + .owner = THIS_MODULE, + }, + .probe = alc5623_i2c_probe, + .remove = __devexit_p(alc5623_i2c_remove), + .id_table = alc5623_i2c_table, +}; + +static int __init alc5623_modinit(void) +{ + int ret; + + ret = i2c_add_driver(&alc5623_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "%s: can't add i2c driver", __func__); + return ret; + } + + return ret; +} +module_init(alc5623_modinit); + +static void __exit alc5623_modexit(void) +{ + i2c_del_driver(&alc5623_i2c_driver); +} +module_exit(alc5623_modexit); + +MODULE_DESCRIPTION("ASoC alc5621/2/3 driver"); +MODULE_AUTHOR("Arnaud Patard arnaud.patard@rtp-net.org"); +MODULE_LICENSE("GPL"); + Index: sound-2.6/sound/soc/codecs/alc5623.h =================================================================== --- /dev/null 1970-01-01 00:00:00.000000000 +0000 +++ sound-2.6/sound/soc/codecs/alc5623.h 2010-10-17 11:07:35.000000000 +0200 @@ -0,0 +1,161 @@ +/* + * alc5623.h -- alc562[123] ALSA Soc Audio driver + * + * Copyright 2008 Realtek Microelectronics + * Copyright 2010 Arnaud Patard arnaud.patard@rtp-net.org + * + * Author: flove flove@realtek.com + * Arnaud Patard arnaud.patard@rtp-net.org + * + * 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. + * + */ + +#ifndef _ALC5623_H +#define _ALC5623_H + +#define ALC5623_RESET 0x00 +/* 5621 5622 5623 */ +/* speaker output vol 2 2 */ +/* line output vol 4 2 */ +/* HP output vol 4 0 4 */ +#define ALC5623_SPK_OUT_VOL 0x02 +#define ALC5623_HP_OUT_VOL 0x04 +#define ALC5623_MONO_AUX_OUT_VOL 0x06 +#define ALC5623_AUXIN_VOL 0x08 +#define ALC5623_LINE_IN_VOL 0x0A +#define ALC5623_STEREO_DAC_VOL 0x0C +#define ALC5623_MIC_VOL 0x0E +#define ALC5623_MIC_ROUTING_CTRL 0x10 +#define ALC5623_ADC_REC_GAIN 0x12 +#define ALC5623_ADC_REC_MIXER 0x14 +#define ALC5623_SOFT_VOL_CTRL_TIME 0x16 +/* ALC5623_OUTPUT_MIXER_CTRL : */ +/* same remark as for reg 2 line vs speaker */ +#define ALC5623_OUTPUT_MIXER_CTRL 0x1C +#define ALC5623_MIC_CTRL 0x22 + +#define ALC5623_DAI_CONTROL 0x34 +#define ALC5623_DAI_SDP_MASTER_MODE (0 << 15) +#define ALC5623_DAI_SDP_SLAVE_MODE (1 << 15) +#define ALC5623_DAI_I2S_PCM_MODE (1 << 14) +#define ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL (1 << 7) +#define ALC5623_DAI_ADC_DATA_L_R_SWAP (1 << 5) +#define ALC5623_DAI_DAC_DATA_L_R_SWAP (1 << 4) +#define ALC5623_DAI_I2S_DL_MASK (3 << 2) +#define ALC5623_DAI_I2S_DL_32 (3 << 2) +#define ALC5623_DAI_I2S_DL_24 (2 << 2) +#define ALC5623_DAI_I2S_DL_20 (1 << 2) +#define ALC5623_DAI_I2S_DL_16 (0 << 2) +#define ALC5623_DAI_I2S_DF_PCM (3 << 0) +#define ALC5623_DAI_I2S_DF_LEFT (2 << 0) +#define ALC5623_DAI_I2S_DF_RIGHT (1 << 0) +#define ALC5623_DAI_I2S_DF_I2S (0 << 0) + +#define ALC5623_STEREO_AD_DA_CLK_CTRL 0x36 +#define ALC5623_COMPANDING_CTRL 0x38 + +#define ALC5623_PWR_MANAG_ADD1 0x3A +#define ALC5623_PWR_ADD1_MAIN_I2S_EN (1 << 15) +#define ALC5623_PWR_ADD1_ZC_DET_PD_EN (1 << 14) +#define ALC5623_PWR_ADD1_MIC1_BIAS_EN (1 << 11) +#define ALC5623_PWR_ADD1_SHORT_CURR_DET_EN (1 << 10) +#define ALC5623_PWR_ADD1_SOFTGEN_EN (1 << 8) /* rsvd on 5622 */ +#define ALC5623_PWR_ADD1_DEPOP_BUF_HP (1 << 6) /* rsvd on 5622 */ +#define ALC5623_PWR_ADD1_HP_OUT_AMP (1 << 5) +#define ALC5623_PWR_ADD1_HP_OUT_ENH_AMP (1 << 4) /* rsvd on 5622 */ +#define ALC5623_PWR_ADD1_DEPOP_BUF_AUX (1 << 2) +#define ALC5623_PWR_ADD1_AUX_OUT_AMP (1 << 1) +#define ALC5623_PWR_ADD1_AUX_OUT_ENH_AMP (1 << 0) /* rsvd on 5622 */ + +#define ALC5623_PWR_MANAG_ADD2 0x3C +#define ALC5623_PWR_ADD2_LINEOUT (1 << 15) /* rt5623 */ +#define ALC5623_PWR_ADD2_CLASS_AB (1 << 15) /* rt5621 */ +#define ALC5623_PWR_ADD2_CLASS_D (1 << 14) /* rt5621 */ +#define ALC5623_PWR_ADD2_VREF (1 << 13) +#define ALC5623_PWR_ADD2_PLL (1 << 12) +#define ALC5623_PWR_ADD2_DAC_REF_CIR (1 << 10) +#define ALC5623_PWR_ADD2_L_DAC_CLK (1 << 9) +#define ALC5623_PWR_ADD2_R_DAC_CLK (1 << 8) +#define ALC5623_PWR_ADD2_L_ADC_CLK_GAIN (1 << 7) +#define ALC5623_PWR_ADD2_R_ADC_CLK_GAIN (1 << 6) +#define ALC5623_PWR_ADD2_L_HP_MIXER (1 << 5) +#define ALC5623_PWR_ADD2_R_HP_MIXER (1 << 4) +#define ALC5623_PWR_ADD2_SPK_MIXER (1 << 3) +#define ALC5623_PWR_ADD2_MONO_MIXER (1 << 2) +#define ALC5623_PWR_ADD2_L_ADC_REC_MIXER (1 << 1) +#define ALC5623_PWR_ADD2_R_ADC_REC_MIXER (1 << 0) + +#define ALC5623_PWR_MANAG_ADD3 0x3E +#define ALC5623_PWR_ADD3_MAIN_BIAS (1 << 15) +#define ALC5623_PWR_ADD3_AUXOUT_L_VOL_AMP (1 << 14) +#define ALC5623_PWR_ADD3_AUXOUT_R_VOL_AMP (1 << 13) +#define ALC5623_PWR_ADD3_SPK_OUT (1 << 12) +#define ALC5623_PWR_ADD3_HP_L_OUT_VOL (1 << 10) +#define ALC5623_PWR_ADD3_HP_R_OUT_VOL (1 << 9) +#define ALC5623_PWR_ADD3_LINEIN_L_VOL (1 << 7) +#define ALC5623_PWR_ADD3_LINEIN_R_VOL (1 << 6) +#define ALC5623_PWR_ADD3_AUXIN_L_VOL (1 << 5) +#define ALC5623_PWR_ADD3_AUXIN_R_VOL (1 << 4) +#define ALC5623_PWR_ADD3_MIC1_FUN_CTRL (1 << 3) +#define ALC5623_PWR_ADD3_MIC2_FUN_CTRL (1 << 2) +#define ALC5623_PWR_ADD3_MIC1_BOOST_AD (1 << 1) +#define ALC5623_PWR_ADD3_MIC2_BOOST_AD (1 << 0) + +#define ALC5623_ADD_CTRL_REG 0x40 + +#define ALC5623_GLOBAL_CLK_CTRL_REG 0x42 +#define ALC5623_GBL_CLK_SYS_SOUR_SEL_PLL (1 << 15) +#define ALC5623_GBL_CLK_SYS_SOUR_SEL_MCLK (0 << 15) +#define ALC5623_GBL_CLK_PLL_SOUR_SEL_BITCLK (1 << 14) +#define ALC5623_GBL_CLK_PLL_SOUR_SEL_MCLK (0 << 14) +#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV8 (3 << 1) +#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV4 (2 << 1) +#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV2 (1 << 1) +#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV1 (0 << 1) +#define ALC5623_GBL_CLK_PLL_PRE_DIV2 (1 << 0) +#define ALC5623_GBL_CLK_PLL_PRE_DIV1 (0 << 0) + +#define ALC5623_PLL_CTRL 0x44 +#define ALC5623_PLL_CTRL_N_VAL(n) (((n)&0xff) << 8) +#define ALC5623_PLL_CTRL_K_VAL(k) (((k)&0x7) << 4) +#define ALC5623_PLL_CTRL_M_VAL(m) ((m)&0xf) + +#define ALC5623_GPIO_OUTPUT_PIN_CTRL 0x4A +#define ALC5623_GPIO_PIN_CONFIG 0x4C +#define ALC5623_GPIO_PIN_POLARITY 0x4E +#define ALC5623_GPIO_PIN_STICKY 0x50 +#define ALC5623_GPIO_PIN_WAKEUP 0x52 +#define ALC5623_GPIO_PIN_STATUS 0x54 +#define ALC5623_GPIO_PIN_SHARING 0x56 +#define ALC5623_OVER_CURR_STATUS 0x58 +#define ALC5623_JACK_DET_CTRL 0x5A + +#define ALC5623_MISC_CTRL 0x5E +#define ALC5623_MISC_DISABLE_FAST_VREG (1 << 15) +#define ALC5623_MISC_SPK_CLASS_AB_OC_PD (1 << 13) /* 5621 */ +#define ALC5623_MISC_SPK_CLASS_AB_OC_DET (1 << 12) /* 5621 */ +#define ALC5623_MISC_HP_DEPOP_MODE3_EN (1 << 10) +#define ALC5623_MISC_HP_DEPOP_MODE2_EN (1 << 9) +#define ALC5623_MISC_HP_DEPOP_MODE1_EN (1 << 8) +#define ALC5623_MISC_AUXOUT_DEPOP_MODE3_EN (1 << 6) +#define ALC5623_MISC_AUXOUT_DEPOP_MODE2_EN (1 << 5) +#define ALC5623_MISC_AUXOUT_DEPOP_MODE1_EN (1 << 4) +#define ALC5623_MISC_M_DAC_L_INPUT (1 << 3) +#define ALC5623_MISC_M_DAC_R_INPUT (1 << 2) +#define ALC5623_MISC_IRQOUT_INV_CTRL (1 << 0) + +#define ALC5623_PSEDUEO_SPATIAL_CTRL 0x60 +#define ALC5623_EQ_CTRL 0x62 +#define ALC5623_EQ_MODE_ENABLE 0x66 +#define ALC5623_AVC_CTRL 0x68 +#define ALC5623_HID_CTRL_INDEX 0x6A +#define ALC5623_HID_CTRL_DATA 0x6C +#define ALC5623_VENDOR_ID1 0x7C +#define ALC5623_VENDOR_ID2 0x7E + +#define ALC5623_PLL_FR_MCLK 0 +#define ALC5623_PLL_FR_BCK 1 +#endif Index: sound-2.6/include/sound/alc5623.h =================================================================== --- /dev/null 1970-01-01 00:00:00.000000000 +0000 +++ sound-2.6/include/sound/alc5623.h 2010-10-17 11:07:35.000000000 +0200 @@ -0,0 +1,15 @@ +#ifndef _INCLUDE_SOUND_ALC5623_H +#define _INCLUDE_SOUND_ALC5623_H +struct alc5623_platform_data { + /* configure : */ + /* Lineout/Speaker Amps Vmid ratio control */ + /* enable/disable adc/dac high pass filters */ + unsigned int add_ctrl; + /* configure : */ + /* output to enable when jack is low */ + /* output to enable when jack is high */ + /* jack detect (gpio/nc/jack detect [12] */ + unsigned int jack_det_ctrl; +}; +#endif +

This patch is adding support for hp t5325 thin clients. There's a alc5623 codec connected to the i2s interface.

Signed-off-by: Arnaud Patard arnaud.patard@rtp-net.org

Index: sound-2.6/sound/soc/kirkwood/Kconfig =================================================================== --- sound-2.6.orig/sound/soc/kirkwood/Kconfig 2010-10-13 19:47:43.000000000 +0200 +++ sound-2.6/sound/soc/kirkwood/Kconfig 2010-10-13 19:52:16.000000000 +0200 @@ -18,3 +18,12 @@ Say Y if you want to add support for SoC audio on Openrd Client.

+config SND_KIRKWOOD_SOC_T5325 + tristate "SoC Audio support for HP t5325" + depends on SND_KIRKWOOD_SOC && MACH_T5325 + select SND_KIRKWOOD_SOC_I2S + select SND_SOC_ALC5623 + help + Say Y if you want to add support for SoC audio on + the HP t5325 thin client. + Index: sound-2.6/sound/soc/kirkwood/Makefile =================================================================== --- sound-2.6.orig/sound/soc/kirkwood/Makefile 2010-10-13 19:47:43.000000000 +0200 +++ sound-2.6/sound/soc/kirkwood/Makefile 2010-10-13 19:52:16.000000000 +0200 @@ -5,5 +5,7 @@ obj-$(CONFIG_SND_KIRKWOOD_SOC_I2S) += snd-soc-kirkwood-i2s.o

snd-soc-openrd-objs := kirkwood-openrd.o +snd-soc-t5325-objs := kirkwood-t5325.o

obj-$(CONFIG_SND_KIRKWOOD_SOC_OPENRD) += snd-soc-openrd.o +obj-$(CONFIG_SND_KIRKWOOD_SOC_T5325) += snd-soc-t5325.o Index: sound-2.6/sound/soc/kirkwood/kirkwood-t5325.c =================================================================== --- /dev/null 1970-01-01 00:00:00.000000000 +0000 +++ sound-2.6/sound/soc/kirkwood/kirkwood-t5325.c 2010-10-13 19:52:16.000000000 +0200 @@ -0,0 +1,141 @@ +/* + * kirkwood-t5325.c + * + * (c) 2010 Arnaud Patard arnaud.patard@rtp-net.org + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <mach/kirkwood.h> +#include <plat/audio.h> +#include <asm/mach-types.h> +#include "../codecs/alc5623.h" + +static int t5325_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; + int ret; + unsigned int freq, fmt; + + fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS; + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) + return ret; + + freq = params_rate(params) * 256; + + return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN); + +} + +static struct snd_soc_ops t5325_ops = { + .hw_params = t5325_hw_params, +}; + +static const struct snd_soc_dapm_widget t5325_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static const struct snd_soc_dapm_route t5325_route[] = { + { "Headphone Jack", NULL, "HPL" }, + { "Headphone Jack", NULL, "HPR" }, + + {"Speaker", NULL, "SPKOUT"}, + {"Speaker", NULL, "SPKOUTN"}, + + { "MIC1", NULL, "Mic Jack" }, + { "MIC2", NULL, "Mic Jack" }, +}; + +static int t5325_dai_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + + snd_soc_dapm_new_controls(codec, t5325_dapm_widgets, + ARRAY_SIZE(t5325_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, t5325_route, ARRAY_SIZE(t5325_route)); + + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Speaker"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link t5325_dai[] = { +{ + .name = "ALC5621", + .stream_name = "ALC5621 HiFi", + .cpu_dai_name = "kirkwood-i2s", + .platform_name = "kirkwood-pcm-audio", + .codec_dai_name = "alc5621-hifi", + .codec_name = "alc562x-codec.0-001a", + .ops = &t5325_ops, + .init = t5325_dai_init, +}, +}; + + +static struct snd_soc_card t5325 = { + .name = "t5325", + .dai_link = t5325_dai, + .num_links = ARRAY_SIZE(t5325_dai), +}; + +static struct platform_device *t5325_snd_device; + +static int __init t5325_init(void) +{ + int ret; + + if (!machine_is_t5325()) + return 0; + + t5325_snd_device = platform_device_alloc("soc-audio", -1); + if (!t5325_snd_device) + return -ENOMEM; + + platform_set_drvdata(t5325_snd_device, + &t5325); + + ret = platform_device_add(t5325_snd_device); + if (ret) { + printk(KERN_ERR "%s: platform_device_add failed\n", __func__); + platform_device_put(t5325_snd_device); + } + + return ret; +} +module_init(t5325_init); + +static void __exit t5325_exit(void) +{ + platform_device_unregister(t5325_snd_device); +} +module_exit(t5325_exit); + +MODULE_AUTHOR("Arnaud Patard arnaud.patard@rtp-net.org"); +MODULE_DESCRIPTION("ALSA SoC t5325 audio client"); +MODULE_LICENSE("GPL");