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-rw-r--r--sound/soc/codecs/alc5632.c1153
1 files changed, 1153 insertions, 0 deletions
diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c
new file mode 100644
index 00000000000..ee6a497b5e7
--- /dev/null
+++ b/sound/soc/codecs/alc5632.c
@@ -0,0 +1,1153 @@
+/*
+* alc5632.c -- ALC5632 ALSA SoC Audio Codec
+*
+* Copyright (C) 2011 The AC100 Kernel Team <ac100@lists.lauchpad.net>
+*
+* Authors: Leon Romanovsky <leon@leon.nu>
+* Andrey Danin <danindrey@mail.ru>
+* Ilya Petrov <ilya.muromec@gmail.com>
+* Marc Dietrich <marvin24@gmx.de>
+*
+* Based on alc5623.c by Arnaud Patard
+*
+* 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 <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+
+#include "alc5632.h"
+
+/*
+ * ALC5632 register cache
+ */
+static const u16 alc5632_reg_defaults[] = {
+ 0x59B4, 0x0000, 0x8080, 0x0000, /* 0 */
+ 0x8080, 0x0000, 0x8080, 0x0000, /* 4 */
+ 0xC800, 0x0000, 0xE808, 0x0000, /* 8 */
+ 0x1010, 0x0000, 0x0808, 0x0000, /* 12 */
+ 0xEE0F, 0x0000, 0xCBCB, 0x0000, /* 16 */
+ 0x7F7F, 0x0000, 0x0000, 0x0000, /* 20 */
+ 0xE010, 0x0000, 0x0000, 0x0000, /* 24 */
+ 0x8008, 0x0000, 0x0000, 0x0000, /* 28 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 32 */
+ 0x00C0, 0x0000, 0xEF00, 0x0000, /* 36 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 40 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 44 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 48 */
+ 0x8000, 0x0000, 0x0000, 0x0000, /* 52 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 56 */
+ 0x0000, 0x0000, 0x8000, 0x0000, /* 60 */
+ 0x0C0A, 0x0000, 0x0000, 0x0000, /* 64 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 68 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 72 */
+ 0xBE3E, 0x0000, 0xBE3E, 0x0000, /* 76 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 80 */
+ 0x803A, 0x0000, 0x0000, 0x0000, /* 84 */
+ 0x0000, 0x0000, 0x0009, 0x0000, /* 88 */
+ 0x0000, 0x0000, 0x3000, 0x0000, /* 92 */
+ 0x3075, 0x0000, 0x1010, 0x0000, /* 96 */
+ 0x3110, 0x0000, 0x0000, 0x0000, /* 100 */
+ 0x0553, 0x0000, 0x0000, 0x0000, /* 104 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 108 */
+};
+
+/* codec private data */
+struct alc5632_priv {
+ enum snd_soc_control_type control_type;
+ void *control_data;
+ struct mutex mutex;
+ u8 id;
+ unsigned int sysclk;
+};
+
+static int alc5632_volatile_register(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ switch (reg) {
+ case ALC5632_RESET:
+ case ALC5632_PWR_DOWN_CTRL_STATUS:
+ case ALC5632_GPIO_PIN_STATUS:
+ case ALC5632_OVER_CURR_STATUS:
+ case ALC5632_HID_CTRL_DATA:
+ case ALC5632_EQ_CTRL:
+ return 1;
+
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static inline int alc5632_reset(struct snd_soc_codec *codec)
+{
+ snd_soc_write(codec, ALC5632_RESET, 0);
+ return snd_soc_read(codec, ALC5632_RESET);
+}
+
+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, ALC5632_HID_CTRL_INDEX, 0x46);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ snd_soc_write(w->codec, ALC5632_HID_CTRL_DATA, 0xFFFF);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ snd_soc_write(w->codec, ALC5632_HID_CTRL_DATA, 0);
+ break;
+ }
+
+ return 0;
+}
+
+/*
+ * ALC5632 Controls
+ */
+
+/* -34.5db min scale, 1.5db steps, no mute */
+static const DECLARE_TLV_DB_SCALE(vol_tlv, -3450, 150, 0);
+/* -46.5db min scale, 1.5db steps, no mute */
+static const DECLARE_TLV_DB_SCALE(hp_tlv, -4650, 150, 0);
+/* -16.5db min scale, 1.5db steps, no mute */
+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),
+};
+/* 0db min scale, 6 db steps, no mute */
+static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0);
+/* 0db min scalem 0.75db steps, no mute */
+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",
+ ALC5632_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
+ /* bit 15 mutes left, bit 7 right */
+ SOC_DOUBLE("Line 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),
+ SOC_DOUBLE("Headphone Playback Switch",
+ ALC5632_HP_OUT_VOL, 15, 7, 1, 1),
+};
+
+static const struct snd_kcontrol_new alc5632_snd_controls[] = {
+ SOC_DOUBLE_TLV("Auxout Playback Volume",
+ ALC5632_AUX_OUT_VOL, 8, 0, 31, 1, hp_tlv),
+ SOC_DOUBLE("Auxout Playback Switch",
+ ALC5632_AUX_OUT_VOL, 15, 7, 1, 1),
+ SOC_SINGLE_TLV("Voice DAC Playback Volume",
+ ALC5632_VOICE_DAC_VOL, 0, 63, 0, vdac_tlv),
+ SOC_SINGLE_TLV("Phone Capture Volume",
+ 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",
+ ALC5632_STEREO_DAC_IN_VOL, 8, 0, 63, 1, vdac_tlv),
+ SOC_DOUBLE("Stereo DAC 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),
+ SOC_SINGLE_TLV("Mic2 Capture Volume",
+ ALC5632_MIC_VOL, 0, 31, 1, vol_tlv),
+ SOC_DOUBLE_TLV("Rec Capture Volume",
+ ALC5632_ADC_REC_GAIN, 8, 0, 31, 0, adc_rec_tlv),
+ SOC_SINGLE_TLV("Mic 1 Boost Volume",
+ ALC5632_MIC_CTRL, 10, 2, 0, boost_tlv),
+ SOC_SINGLE_TLV("Mic 2 Boost Volume",
+ ALC5632_MIC_CTRL, 8, 2, 0, boost_tlv),
+ SOC_SINGLE_TLV("Digital Boost Volume",
+ ALC5632_DIGI_BOOST_CTRL, 0, 7, 0, dig_tlv),
+};
+
+/*
+ * DAPM Controls
+ */
+static const struct snd_kcontrol_new alc5632_hp_mixer_controls[] = {
+SOC_DAPM_SINGLE("LI2HP Playback Switch", ALC5632_LINE_IN_VOL, 15, 1, 1),
+SOC_DAPM_SINGLE("PHONE2HP Playback Switch", ALC5632_PHONE_IN_VOL, 15, 1, 1),
+SOC_DAPM_SINGLE("MIC12HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 15, 1, 1),
+SOC_DAPM_SINGLE("MIC22HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 11, 1, 1),
+SOC_DAPM_SINGLE("VOICE2HP Playback Switch", ALC5632_VOICE_DAC_VOL, 15, 1, 1),
+};
+
+static const struct snd_kcontrol_new alc5632_hpl_mixer_controls[] = {
+SOC_DAPM_SINGLE("ADC2HP_L Playback Switch", ALC5632_ADC_REC_GAIN, 15, 1, 1),
+SOC_DAPM_SINGLE("DACL2HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 3, 1, 1),
+};
+
+static const struct snd_kcontrol_new alc5632_hpr_mixer_controls[] = {
+SOC_DAPM_SINGLE("ADC2HP_R Playback Switch", ALC5632_ADC_REC_GAIN, 7, 1, 1),
+SOC_DAPM_SINGLE("DACR2HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 2, 1, 1),
+};
+
+static const struct snd_kcontrol_new alc5632_mono_mixer_controls[] = {
+SOC_DAPM_SINGLE("ADC2MONO_L Playback Switch", ALC5632_ADC_REC_GAIN, 14, 1, 1),
+SOC_DAPM_SINGLE("ADC2MONO_R Playback Switch", ALC5632_ADC_REC_GAIN, 6, 1, 1),
+SOC_DAPM_SINGLE("LI2MONO Playback Switch", ALC5632_LINE_IN_VOL, 13, 1, 1),
+SOC_DAPM_SINGLE("MIC12MONO Playback Switch",
+ ALC5632_MIC_ROUTING_CTRL, 13, 1, 1),
+SOC_DAPM_SINGLE("MIC22MONO Playback Switch",
+ ALC5632_MIC_ROUTING_CTRL, 9, 1, 1),
+SOC_DAPM_SINGLE("DAC2MONO Playback Switch", ALC5632_MIC_ROUTING_CTRL, 0, 1, 1),
+SOC_DAPM_SINGLE("VOICE2MONO Playback Switch", ALC5632_VOICE_DAC_VOL, 13, 1, 1),
+};
+
+static const struct snd_kcontrol_new alc5632_speaker_mixer_controls[] = {
+SOC_DAPM_SINGLE("LI2SPK Playback Switch", ALC5632_LINE_IN_VOL, 14, 1, 1),
+SOC_DAPM_SINGLE("PHONE2SPK Playback Switch", ALC5632_PHONE_IN_VOL, 14, 1, 1),
+SOC_DAPM_SINGLE("MIC12SPK Playback Switch",
+ ALC5632_MIC_ROUTING_CTRL, 14, 1, 1),
+SOC_DAPM_SINGLE("MIC22SPK Playback Switch",
+ ALC5632_MIC_ROUTING_CTRL, 10, 1, 1),
+SOC_DAPM_SINGLE("DAC2SPK Playback Switch", ALC5632_MIC_ROUTING_CTRL, 1, 1, 1),
+SOC_DAPM_SINGLE("VOICE2SPK Playback Switch", ALC5632_VOICE_DAC_VOL, 14, 1, 1),
+};
+
+/* Left Record Mixer */
+static const struct snd_kcontrol_new alc5632_captureL_mixer_controls[] = {
+SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5632_ADC_REC_MIXER, 14, 1, 1),
+SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5632_ADC_REC_MIXER, 13, 1, 1),
+SOC_DAPM_SINGLE("LineInL Capture Switch", ALC5632_ADC_REC_MIXER, 12, 1, 1),
+SOC_DAPM_SINGLE("Left Phone Capture Switch", ALC5632_ADC_REC_MIXER, 11, 1, 1),
+SOC_DAPM_SINGLE("HPMixerL Capture Switch", ALC5632_ADC_REC_MIXER, 10, 1, 1),
+SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5632_ADC_REC_MIXER, 9, 1, 1),
+SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5632_ADC_REC_MIXER, 8, 1, 1),
+};
+
+/* Right Record Mixer */
+static const struct snd_kcontrol_new alc5632_captureR_mixer_controls[] = {
+SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5632_ADC_REC_MIXER, 6, 1, 1),
+SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5632_ADC_REC_MIXER, 5, 1, 1),
+SOC_DAPM_SINGLE("LineInR Capture Switch", ALC5632_ADC_REC_MIXER, 4, 1, 1),
+SOC_DAPM_SINGLE("Right Phone Capture Switch", ALC5632_ADC_REC_MIXER, 3, 1, 1),
+SOC_DAPM_SINGLE("HPMixerR Capture Switch", ALC5632_ADC_REC_MIXER, 2, 1, 1),
+SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5632_ADC_REC_MIXER, 1, 1, 1),
+SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5632_ADC_REC_MIXER, 0, 1, 1),
+};
+
+static const char *alc5632_spk_n_sour_sel[] = {
+ "RN/-R", "RP/+R", "LN/-R", "Mute"};
+static const char *alc5632_hpl_out_input_sel[] = {
+ "Vmid", "HP Left Mix"};
+static const char *alc5632_hpr_out_input_sel[] = {
+ "Vmid", "HP Right Mix"};
+static const char *alc5632_spkout_input_sel[] = {
+ "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
+static const char *alc5632_aux_out_input_sel[] = {
+ "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
+
+/* auxout output mux */
+static const struct soc_enum alc5632_aux_out_input_enum =
+SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 6, 4, alc5632_aux_out_input_sel);
+static const struct snd_kcontrol_new alc5632_auxout_mux_controls =
+SOC_DAPM_ENUM("AuxOut Mux", alc5632_aux_out_input_enum);
+
+/* speaker output mux */
+static const struct soc_enum alc5632_spkout_input_enum =
+SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 10, 4, alc5632_spkout_input_sel);
+static const struct snd_kcontrol_new alc5632_spkout_mux_controls =
+SOC_DAPM_ENUM("SpeakerOut Mux", alc5632_spkout_input_enum);
+
+/* headphone left output mux */
+static const struct soc_enum alc5632_hpl_out_input_enum =
+SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 9, 2, alc5632_hpl_out_input_sel);
+static const struct snd_kcontrol_new alc5632_hpl_out_mux_controls =
+SOC_DAPM_ENUM("Left Headphone Mux", alc5632_hpl_out_input_enum);
+
+/* headphone right output mux */
+static const struct soc_enum alc5632_hpr_out_input_enum =
+SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 8, 2, alc5632_hpr_out_input_sel);
+static const struct snd_kcontrol_new alc5632_hpr_out_mux_controls =
+SOC_DAPM_ENUM("Right Headphone Mux", alc5632_hpr_out_input_enum);
+
+/* speaker output N select */
+static const struct soc_enum alc5632_spk_n_sour_enum =
+SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 14, 4, alc5632_spk_n_sour_sel);
+static const struct snd_kcontrol_new alc5632_spkoutn_mux_controls =
+SOC_DAPM_ENUM("SpeakerOut N Mux", alc5632_spk_n_sour_enum);
+
+/* speaker amplifier */
+static const char *alc5632_amp_names[] = {"AB Amp", "D Amp"};
+static const struct soc_enum alc5632_amp_enum =
+ SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 13, 2, alc5632_amp_names);
+static const struct snd_kcontrol_new alc5632_amp_mux_controls =
+ SOC_DAPM_ENUM("AB-D Amp Mux", alc5632_amp_enum);
+
+
+static const struct snd_soc_dapm_widget alc5632_dapm_widgets[] = {
+/* Muxes */
+SND_SOC_DAPM_MUX("AuxOut Mux", SND_SOC_NOPM, 0, 0,
+ &alc5632_auxout_mux_controls),
+SND_SOC_DAPM_MUX("SpeakerOut Mux", SND_SOC_NOPM, 0, 0,
+ &alc5632_spkout_mux_controls),
+SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0,
+ &alc5632_hpl_out_mux_controls),
+SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0,
+ &alc5632_hpr_out_mux_controls),
+SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0,
+ &alc5632_spkoutn_mux_controls),
+
+/* output mixers */
+SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0,
+ &alc5632_hp_mixer_controls[0],
+ ARRAY_SIZE(alc5632_hp_mixer_controls)),
+SND_SOC_DAPM_MIXER("HPR Mix", ALC5632_PWR_MANAG_ADD2, 4, 0,
+ &alc5632_hpr_mixer_controls[0],
+ ARRAY_SIZE(alc5632_hpr_mixer_controls)),
+SND_SOC_DAPM_MIXER("HPL Mix", ALC5632_PWR_MANAG_ADD2, 5, 0,
+ &alc5632_hpl_mixer_controls[0],
+ ARRAY_SIZE(alc5632_hpl_mixer_controls)),
+SND_SOC_DAPM_MIXER("HPOut Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
+SND_SOC_DAPM_MIXER("Mono Mix", ALC5632_PWR_MANAG_ADD2, 2, 0,
+ &alc5632_mono_mixer_controls[0],
+ ARRAY_SIZE(alc5632_mono_mixer_controls)),
+SND_SOC_DAPM_MIXER("Speaker Mix", ALC5632_PWR_MANAG_ADD2, 3, 0,
+ &alc5632_speaker_mixer_controls[0],
+ ARRAY_SIZE(alc5632_speaker_mixer_controls)),
+
+/* input mixers */
+SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5632_PWR_MANAG_ADD2, 1, 0,
+ &alc5632_captureL_mixer_controls[0],
+ ARRAY_SIZE(alc5632_captureL_mixer_controls)),
+SND_SOC_DAPM_MIXER("Right Capture Mix", ALC5632_PWR_MANAG_ADD2, 0, 0,
+ &alc5632_captureR_mixer_controls[0],
+ ARRAY_SIZE(alc5632_captureR_mixer_controls)),
+
+SND_SOC_DAPM_DAC("Left DAC", "HiFi Playback",
+ ALC5632_PWR_MANAG_ADD2, 9, 0),
+SND_SOC_DAPM_DAC("Right DAC", "HiFi Playback",
+ ALC5632_PWR_MANAG_ADD2, 8, 0),
+SND_SOC_DAPM_MIXER("DAC Left Channel", ALC5632_PWR_MANAG_ADD1, 15, 0, NULL, 0),
+SND_SOC_DAPM_MIXER("DAC Right Channel",
+ ALC5632_PWR_MANAG_ADD1, 14, 0, NULL, 0),
+SND_SOC_DAPM_MIXER("I2S Mix", ALC5632_PWR_MANAG_ADD1, 11, 0, NULL, 0),
+SND_SOC_DAPM_MIXER("Phone 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", "HiFi Capture",
+ ALC5632_PWR_MANAG_ADD2, 7, 0),
+SND_SOC_DAPM_ADC("Right ADC", "HiFi Capture",
+ ALC5632_PWR_MANAG_ADD2, 6, 0),
+SND_SOC_DAPM_PGA("Left Headphone", ALC5632_PWR_MANAG_ADD3, 11, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Right Headphone", ALC5632_PWR_MANAG_ADD3, 10, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Left Speaker", ALC5632_PWR_MANAG_ADD3, 13, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Right Speaker", ALC5632_PWR_MANAG_ADD3, 12, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Aux Out", ALC5632_PWR_MANAG_ADD3, 14, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Left LineIn", ALC5632_PWR_MANAG_ADD3, 7, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Right LineIn", ALC5632_PWR_MANAG_ADD3, 6, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Phone", ALC5632_PWR_MANAG_ADD3, 5, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Phone ADMix", ALC5632_PWR_MANAG_ADD3, 4, 0, NULL, 0),
+SND_SOC_DAPM_PGA("MIC1 PGA", ALC5632_PWR_MANAG_ADD3, 3, 0, NULL, 0),
+SND_SOC_DAPM_PGA("MIC2 PGA", ALC5632_PWR_MANAG_ADD3, 2, 0, NULL, 0),
+SND_SOC_DAPM_PGA("MIC1 Pre Amp", ALC5632_PWR_MANAG_ADD3, 1, 0, NULL, 0),
+SND_SOC_DAPM_PGA("MIC2 Pre Amp", ALC5632_PWR_MANAG_ADD3, 0, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("Mic Bias1", ALC5632_PWR_MANAG_ADD1, 3, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("Mic Bias2", ALC5632_PWR_MANAG_ADD1, 2, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA_E("D Amp", ALC5632_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", ALC5632_PWR_MANAG_ADD2, 15, 0, NULL, 0),
+SND_SOC_DAPM_MUX("AB-D Amp Mux", ALC5632_PWR_MANAG_ADD1, 10, 0,
+ &alc5632_amp_mux_controls),
+
+SND_SOC_DAPM_OUTPUT("AUXOUT"),
+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("PHONEP"),
+SND_SOC_DAPM_INPUT("PHONEN"),
+SND_SOC_DAPM_INPUT("MIC1"),
+SND_SOC_DAPM_INPUT("MIC2"),
+SND_SOC_DAPM_VMID("Vmid"),
+};
+
+
+static const struct snd_soc_dapm_route alc5632_dapm_routes[] = {
+ /* 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"},
+ {"Phone Mix", NULL, "Phone"},
+ {"Phone Mix", NULL, "Phone ADMix"},
+ {"AUXOUT", NULL, "Aux Out"},
+
+ /* DAC */
+ {"DAC Right Channel", NULL, "I2S Mix"},
+ {"DAC Left Channel", NULL, "I2S Mix"},
+
+ /* 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", "PHONE2HP Playback Switch", "Phone Mix"},
+ {"HP Mix", "MIC12HP Playback Switch", "MIC1 PGA"},
+ {"HP Mix", "MIC22HP Playback Switch", "MIC2 PGA"},
+
+ {"HPR Mix", "DACR2HP Playback Switch", "DAC Right Channel"},
+ {"HPL Mix", "DACL2HP Playback Switch", "DAC Left Channel"},
+
+ /* speaker mixer */
+ {"Speaker Mix", "LI2SPK Playback Switch", "Line Mix"},
+ {"Speaker Mix", "PHONE2SPK Playback Switch", "Phone Mix"},
+ {"Speaker Mix", "MIC12SPK Playback Switch", "MIC1 PGA"},
+ {"Speaker Mix", "MIC22SPK Playback Switch", "MIC2 PGA"},
+ {"Speaker Mix", "DAC2SPK Playback Switch", "DAC Left Channel"},
+
+
+
+ /* 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", "VOICE2MONO Playback Switch", "Phone Mix"},
+ {"Mono Mix", "MIC12MONO Playback Switch", "MIC1 PGA"},
+ {"Mono Mix", "MIC22MONO Playback Switch", "MIC2 PGA"},
+ {"Mono Mix", "DAC2MONO Playback Switch", "DAC Left Channel"},
+
+ /* Left record mixer */
+ {"Left Capture Mix", "LineInL Capture Switch", "LINEINL"},
+ {"Left Capture Mix", "Left Phone Capture Switch", "PHONEN"},
+ {"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 Phone Capture Switch", "PHONEP"},
+ {"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"},
+ {"Aux Out", NULL, "AuxOut Mux"},
+
+ /* input pga */
+ {"Left LineIn", NULL, "LINEINL"},
+ {"Right LineIn", NULL, "LINEINR"},
+ {"Phone", NULL, "PHONEP"},
+ {"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", "Left Speaker"},
+ {"SpeakerOut N Mux", "RP/+R", "Left Speaker"},
+ {"SpeakerOut N Mux", "LN/-R", "Left Speaker"},
+ {"SpeakerOut N Mux", "Mute", "Vmid"},
+
+ {"SpeakerOut N Mux", "RN/-R", "Right Speaker"},
+ {"SpeakerOut N Mux", "RP/+R", "Right Speaker"},
+ {"SpeakerOut N Mux", "LN/-R", "Right Speaker"},
+ {"SpeakerOut N Mux", "Mute", "Vmid"},
+
+ {"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"},
+ {"Left Speaker", NULL, "AB-D Amp Mux"},
+ {"Right Speaker", NULL, "AB-D Amp Mux"},
+
+ {"SPKOUT", NULL, "Left Speaker"},
+ {"SPKOUT", NULL, "Right Speaker"},
+
+ {"SPKOUTN", NULL, "SpeakerOut N Mux"},
+
+};
+
+/* PLL divisors */
+struct _pll_div {
+ u32 pll_in;
+ u32 pll_out;
+ u16 regvalue;
+};
+
+/* Note : pll code from original alc5632 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},
+};
+
+/* FOUT = MCLK*(N+2)/((M+2)*(K+2))
+ N: bit 15:8 (div 2 .. div 257)
+ K: bit 6:4 typical 2
+ M: bit 3:0 (div 2 .. div 17)
+
+ same as for 5623 - thanks!
+*/
+
+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 alc5632_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 < ALC5632_PLL_FR_MCLK || pll_id > ALC5632_PLL_FR_VBCLK)
+ return -EINVAL;
+
+ /* Disable PLL power */
+ snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2,
+ ALC5632_PWR_ADD2_PLL1,
+ 0);
+ snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2,
+ ALC5632_PWR_ADD2_PLL2,
+ 0);
+
+ /* pll is not used in slave mode */
+ reg = snd_soc_read(codec, ALC5632_DAI_CONTROL);
+ if (reg & ALC5632_DAI_SDP_SLAVE_MODE)
+ return 0;
+
+ if (!freq_in || !freq_out)
+ return 0;
+
+ switch (pll_id) {
+ case ALC5632_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 ALC5632_PLL_FR_BCLK:
+ 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 = ALC5632_PLL_FR_BCLK;
+ pll_div = codec_slave_pll_div[i].regvalue;
+ break;
+ }
+ }
+ break;
+ case ALC5632_PLL_FR_VBCLK:
+ 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 voice clock */
+ gbl_clk = ALC5632_PLL_FR_VBCLK;
+ pll_div = codec_slave_pll_div[i].regvalue;
+ break;
+ }
+ }
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (!pll_div)
+ return -EINVAL;
+
+ /* choose MCLK/BCLK/VBCLK */
+ snd_soc_write(codec, ALC5632_GPCR2, gbl_clk);
+ /* choose PLL1 clock rate */
+ snd_soc_write(codec, ALC5632_PLL1_CTRL, pll_div);
+ /* enable PLL1 */
+ snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2,
+ ALC5632_PWR_ADD2_PLL1,
+ ALC5632_PWR_ADD2_PLL1);
+ /* enable PLL2 */
+ snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2,
+ ALC5632_PWR_ADD2_PLL2,
+ ALC5632_PWR_ADD2_PLL2);
+ /* use PLL1 as main SYSCLK */
+ snd_soc_update_bits(codec, ALC5632_GPCR1,
+ ALC5632_GPCR1_CLK_SYS_SRC_SEL_PLL1,
+ ALC5632_GPCR1_CLK_SYS_SRC_SEL_PLL1);
+
+ 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[] = {
+ {512*1, 0x3075},
+};
+
+static int get_coeff(struct snd_soc_codec *codec, int rate)
+{
+ struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec);
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
+ if (coeff_div[i].fs * rate == alc5632->sysclk)
+ return i;
+ }
+ return -EINVAL;
+}
+
+/*
+ * Clock after PLL and dividers
+ */
+static int alc5632_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 alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec);
+
+ switch (freq) {
+ case 8192000:
+ case 11289600:
+ case 12288000:
+ case 16384000:
+ case 16934400:
+ case 18432000:
+ case 22579200:
+ case 24576000:
+ alc5632->sysclk = freq;
+ return 0;
+ }
+ return -EINVAL;
+}
+
+static int alc5632_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 = ALC5632_DAI_SDP_MASTER_MODE;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ iface = ALC5632_DAI_SDP_SLAVE_MODE;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= ALC5632_DAI_I2S_DF_I2S;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface |= ALC5632_DAI_I2S_DF_LEFT;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface |= ALC5632_DAI_I2S_DF_PCM_A;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ iface |= ALC5632_DAI_I2S_DF_PCM_B;
+ 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 |= ALC5632_DAI_MAIN_I2S_BCLK_POL_CTRL;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ iface |= ALC5632_DAI_MAIN_I2S_BCLK_POL_CTRL;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return snd_soc_write(codec, ALC5632_DAI_CONTROL, iface);
+}
+
+static int alc5632_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;
+ int coeff, rate;
+ u16 iface;
+
+ iface = snd_soc_read(codec, ALC5632_DAI_CONTROL);
+ iface &= ~ALC5632_DAI_I2S_DL_MASK;
+
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ iface |= ALC5632_DAI_I2S_DL_16;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ iface |= ALC5632_DAI_I2S_DL_20;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ iface |= ALC5632_DAI_I2S_DL_24;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* set iface & srate */
+ snd_soc_write(codec, ALC5632_DAI_CONTROL, iface);
+ rate = params_rate(params);
+ coeff = get_coeff(codec, rate);
+ if (coeff < 0)
+ return -EINVAL;
+
+ coeff = coeff_div[coeff].regvalue;
+ snd_soc_write(codec, ALC5632_DAC_CLK_CTRL1, coeff);
+
+ return 0;
+}
+
+static int alc5632_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 hp_mute = ALC5632_MISC_HP_DEPOP_MUTE_L \
+ |ALC5632_MISC_HP_DEPOP_MUTE_R;
+ u16 mute_reg = snd_soc_read(codec, ALC5632_MISC_CTRL) & ~hp_mute;
+
+ if (mute)
+ mute_reg |= hp_mute;
+
+ return snd_soc_write(codec, ALC5632_MISC_CTRL, mute_reg);
+}
+
+#define ALC5632_ADD2_POWER_EN (ALC5632_PWR_ADD2_VREF)
+
+#define ALC5632_ADD3_POWER_EN (ALC5632_PWR_ADD3_MIC1_BOOST_AD)
+
+#define ALC5632_ADD1_POWER_EN \
+ (ALC5632_PWR_ADD1_DAC_REF \
+ | ALC5632_PWR_ADD1_SOFTGEN_EN \
+ | ALC5632_PWR_ADD1_HP_OUT_AMP \
+ | ALC5632_PWR_ADD1_HP_OUT_ENH_AMP \
+ | ALC5632_PWR_ADD1_MAIN_BIAS)
+
+static void enable_power_depop(struct snd_soc_codec *codec)
+{
+ snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD1,
+ ALC5632_PWR_ADD1_SOFTGEN_EN,
+ ALC5632_PWR_ADD1_SOFTGEN_EN);
+
+ snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD3,
+ ALC5632_ADD3_POWER_EN,
+ ALC5632_ADD3_POWER_EN);
+
+ snd_soc_update_bits(codec, ALC5632_MISC_CTRL,
+ ALC5632_MISC_HP_DEPOP_MODE2_EN,
+ ALC5632_MISC_HP_DEPOP_MODE2_EN);
+
+ /* "normal" mode: 0 @ 26 */
+ /* set all PR0-7 mixers to 0 */
+ snd_soc_update_bits(codec, ALC5632_PWR_DOWN_CTRL_STATUS,
+ ALC5632_PWR_DOWN_CTRL_STATUS_MASK,
+ 0);
+
+ msleep(500);
+
+ snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2,
+ ALC5632_ADD2_POWER_EN,
+ ALC5632_ADD2_POWER_EN);
+
+ snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD1,
+ ALC5632_ADD1_POWER_EN,
+ ALC5632_ADD1_POWER_EN);
+
+ /* disable HP Depop2 */
+ snd_soc_update_bits(codec, ALC5632_MISC_CTRL,
+ ALC5632_MISC_HP_DEPOP_MODE2_EN,
+ 0);
+
+}
+
+static int alc5632_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_update_bits(codec, ALC5632_PWR_MANAG_ADD1,
+ ALC5632_PWR_MANAG_ADD1_MASK,
+ ALC5632_PWR_ADD1_MAIN_BIAS);
+ snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2,
+ ALC5632_PWR_MANAG_ADD2_MASK,
+ ALC5632_PWR_ADD2_VREF);
+ /* "normal" mode: 0 @ 26 */
+ snd_soc_update_bits(codec, ALC5632_PWR_DOWN_CTRL_STATUS,
+ ALC5632_PWR_DOWN_CTRL_STATUS_MASK,
+ 0xffff ^ (ALC5632_PWR_VREF_PR3
+ | ALC5632_PWR_VREF_PR2));
+ break;
+ case SND_SOC_BIAS_OFF:
+ /* everything off, dac mute, inactive */
+ snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2,
+ ALC5632_PWR_MANAG_ADD2_MASK, 0);
+ snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD3,
+ ALC5632_PWR_MANAG_ADD3_MASK, 0);
+ snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD1,
+ ALC5632_PWR_MANAG_ADD1_MASK, 0);
+ break;
+ }
+ codec->dapm.bias_level = level;
+ return 0;
+}
+
+#define ALC5632_FORMATS (SNDRV_PCM_FMTBIT_S16_LE \
+ | SNDRV_PCM_FMTBIT_S24_LE \
+ | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops alc5632_dai_ops = {
+ .hw_params = alc5632_pcm_hw_params,
+ .digital_mute = alc5632_mute,
+ .set_fmt = alc5632_set_dai_fmt,
+ .set_sysclk = alc5632_set_dai_sysclk,
+ .set_pll = alc5632_set_dai_pll,
+};
+
+static struct snd_soc_dai_driver alc5632_dai = {
+ .name = "alc5632-hifi",
+ .playback = {
+ .stream_name = "HiFi Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = ALC5632_FORMATS,},
+ .capture = {
+ .stream_name = "HiFi Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = ALC5632_FORMATS,},
+
+ .ops = &alc5632_dai_ops,
+ .symmetric_rates = 1,
+};
+
+static int alc5632_suspend(struct snd_soc_codec *codec, pm_message_t mesg)
+{
+ alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int alc5632_resume(struct snd_soc_codec *codec)
+{
+ int ret;
+
+ /* mark cache as needed to sync */
+ codec->cache_sync = 1;
+
+ ret = snd_soc_cache_sync(codec);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to sync cache: %d\n", ret);
+ return ret;
+ }
+
+ alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ return 0;
+}
+
+#define ALC5632_REC_UNMUTE (ALC5632_ADC_REC_MIC2 \
+ | ALC5632_ADC_REC_LINE_IN | ALC5632_ADC_REC_AUX \
+ | ALC5632_ADC_REC_HP | ALC5632_ADC_REC_SPK \
+ | ALC5632_ADC_REC_MONOMIX)
+
+#define ALC5632_MIC_ROUTE (ALC5632_MIC_ROUTE_HP \
+ | ALC5632_MIC_ROUTE_SPK \
+ | ALC5632_MIC_ROUTE_MONOMIX)
+
+#define ALC5632_PWR_DEFAULT (ALC5632_PWR_ADC_STATUS \
+ | ALC5632_PWR_DAC_STATUS \
+ | ALC5632_PWR_AMIX_STATUS \
+ | ALC5632_PWR_VREF_STATUS)
+
+#define ALC5632_ADC_REC_GAIN_COMP(x) (int)((x - ALC5632_ADC_REC_GAIN_BASE) \
+ / ALC5632_ADC_REC_GAIN_STEP)
+
+#define ALC5632_MIC_BOOST_COMP(x) (int)(x / ALC5632_MIC_BOOST_STEP)
+
+#define ALC5632_SPK_OUT_VOL_COMP(x) (int)(x / ALC5632_SPK_OUT_VOL_STEP)
+
+static int alc5632_probe(struct snd_soc_codec *codec)
+{
+ struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ ret = snd_soc_codec_set_cache_io(codec, 8, 16, alc5632->control_type);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ return ret;
+ }
+
+ alc5632_reset(codec);
+
+ /* power on device */
+ alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ switch (alc5632->id) {
+ case 0x5c:
+ snd_soc_add_controls(codec, alc5632_vol_snd_controls,
+ ARRAY_SIZE(alc5632_vol_snd_controls));
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return ret;
+}
+
+/* power down chip */
+static int alc5632_remove(struct snd_soc_codec *codec)
+{
+ alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_device_alc5632 = {
+ .probe = alc5632_probe,
+ .remove = alc5632_remove,
+ .suspend = alc5632_suspend,
+ .resume = alc5632_resume,
+ .set_bias_level = alc5632_set_bias_level,
+ .reg_word_size = sizeof(u16),
+ .reg_cache_step = 2,
+ .reg_cache_default = alc5632_reg_defaults,
+ .reg_cache_size = ARRAY_SIZE(alc5632_reg_defaults),
+ .volatile_register = alc5632_volatile_register,
+ .controls = alc5632_snd_controls,
+ .num_controls = ARRAY_SIZE(alc5632_snd_controls),
+ .dapm_widgets = alc5632_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(alc5632_dapm_widgets),
+ .dapm_routes = alc5632_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(alc5632_dapm_routes),
+};
+
+/*
+ * alc5632 2 wire address is determined by A1 pin
+ * state during powerup.
+ * low = 0x1a
+ * high = 0x1b
+ */
+static int alc5632_i2c_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ struct alc5632_priv *alc5632;
+ int ret, vid1, vid2;
+
+ vid1 = i2c_smbus_read_word_data(client, ALC5632_VENDOR_ID1);
+ if (vid1 < 0) {
+ dev_err(&client->dev, "failed to read I2C\n");
+ return -EIO;
+ } else {
+ dev_info(&client->dev, "got vid1: %x\n", vid1);
+ }
+ vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8);
+
+ vid2 = i2c_smbus_read_word_data(client, ALC5632_VENDOR_ID2);
+ if (vid2 < 0) {
+ dev_err(&client->dev, "failed to read I2C\n");
+ return -EIO;
+ } else {
+ dev_info(&client->dev, "got vid2: %x\n", vid2);
+ }
+ vid2 = (vid2 & 0xff);
+
+ 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;
+ }
+
+ alc5632 = devm_kzalloc(&client->dev,
+ sizeof(struct alc5632_priv), GFP_KERNEL);
+ if (alc5632 == NULL)
+ return -ENOMEM;
+
+ alc5632->id = vid2;
+ switch (alc5632->id) {
+ case 0x5c:
+ alc5632_dai.name = "alc5632-hifi";
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ i2c_set_clientdata(client, alc5632);
+ alc5632->control_data = client;
+ alc5632->control_type = SND_SOC_I2C;
+ mutex_init(&alc5632->mutex);
+
+ ret = snd_soc_register_codec(&client->dev,
+ &soc_codec_device_alc5632, &alc5632_dai, 1);
+ if (ret != 0)
+ dev_err(&client->dev, "Failed to register codec: %d\n", ret);
+
+ return ret;
+}
+
+static int alc5632_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+
+ return 0;
+}
+
+static const struct i2c_device_id alc5632_i2c_table[] = {
+ {"alc5632", 0x5c},
+ {}
+};
+MODULE_DEVICE_TABLE(i2c, alc5632_i2c_table);
+
+/* i2c codec control layer */
+static struct i2c_driver alc5632_i2c_driver = {
+ .driver = {
+ .name = "alc5632",
+ .owner = THIS_MODULE,
+ },
+ .probe = alc5632_i2c_probe,
+ .remove = __devexit_p(alc5632_i2c_remove),
+ .id_table = alc5632_i2c_table,
+};
+
+static int __init alc5632_modinit(void)
+{
+ int ret;
+
+ ret = i2c_add_driver(&alc5632_i2c_driver);
+ if (ret != 0) {
+ printk(KERN_ERR "%s: can't add i2c driver", __func__);
+ return ret;
+ }
+
+ return ret;
+}
+module_init(alc5632_modinit);
+
+static void __exit alc5632_modexit(void)
+{
+ i2c_del_driver(&alc5632_i2c_driver);
+}
+module_exit(alc5632_modexit);
+
+MODULE_DESCRIPTION("ASoC ALC5632 driver");
+MODULE_AUTHOR("Leon Romanovsky <leon@leon.nu>");
+MODULE_LICENSE("GPL");