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/*
* ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver
*
* Copyright (C) 2009 Renesas Solutions Corp.
* Kuninori Morimoto <morimoto.kuninori@renesas.com>
*
* Based on wm8731.c by Richard Purdie
* Based on ak4535.c by Richard Purdie
* Based on wm8753.c by Liam Girdwood
*
* 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.
*/
/* ** CAUTION **
*
* This is very simple driver.
* It can use headphone output / stereo input only
*
* AK4642 is tested.
* AK4643 is tested.
* AK4648 is tested.
*/
#include <linux/delay.h>
#include <linux/i2c.h>
#include <linux/slab.h>
#include <linux/of_device.h>
#include <linux/module.h>
#include <linux/regmap.h>
#include <sound/soc.h>
#include <sound/initval.h>
#include <sound/tlv.h>
#define PW_MGMT1 0x00
#define PW_MGMT2 0x01
#define SG_SL1 0x02
#define SG_SL2 0x03
#define MD_CTL1 0x04
#define MD_CTL2 0x05
#define TIMER 0x06
#define ALC_CTL1 0x07
#define ALC_CTL2 0x08
#define L_IVC 0x09
#define L_DVC 0x0a
#define ALC_CTL3 0x0b
#define R_IVC 0x0c
#define R_DVC 0x0d
#define MD_CTL3 0x0e
#define MD_CTL4 0x0f
#define PW_MGMT3 0x10
#define DF_S 0x11
#define FIL3_0 0x12
#define FIL3_1 0x13
#define FIL3_2 0x14
#define FIL3_3 0x15
#define EQ_0 0x16
#define EQ_1 0x17
#define EQ_2 0x18
#define EQ_3 0x19
#define EQ_4 0x1a
#define EQ_5 0x1b
#define FIL1_0 0x1c
#define FIL1_1 0x1d
#define FIL1_2 0x1e
#define FIL1_3 0x1f
#define PW_MGMT4 0x20
#define MD_CTL5 0x21
#define LO_MS 0x22
#define HP_MS 0x23
#define SPK_MS 0x24
/* PW_MGMT1*/
#define PMVCM (1 << 6) /* VCOM Power Management */
#define PMMIN (1 << 5) /* MIN Input Power Management */
#define PMDAC (1 << 2) /* DAC Power Management */
#define PMADL (1 << 0) /* MIC Amp Lch and ADC Lch Power Management */
/* PW_MGMT2 */
#define HPMTN (1 << 6)
#define PMHPL (1 << 5)
#define PMHPR (1 << 4)
#define MS (1 << 3) /* master/slave select */
#define MCKO (1 << 1)
#define PMPLL (1 << 0)
#define PMHP_MASK (PMHPL | PMHPR)
#define PMHP PMHP_MASK
/* PW_MGMT3 */
#define PMADR (1 << 0) /* MIC L / ADC R Power Management */
/* SG_SL1 */
#define MINS (1 << 6) /* Switch from MIN to Speaker */
#define DACL (1 << 4) /* Switch from DAC to Stereo or Receiver */
#define PMMP (1 << 2) /* MPWR pin Power Management */
#define MGAIN0 (1 << 0) /* MIC amp gain*/
/* TIMER */
#define ZTM(param) ((param & 0x3) << 4) /* ALC Zoro Crossing TimeOut */
#define WTM(param) (((param & 0x4) << 4) | ((param & 0x3) << 2))
/* ALC_CTL1 */
#define ALC (1 << 5) /* ALC Enable */
#define LMTH0 (1 << 0) /* ALC Limiter / Recovery Level */
/* MD_CTL1 */
#define PLL3 (1 << 7)
#define PLL2 (1 << 6)
#define PLL1 (1 << 5)
#define PLL0 (1 << 4)
#define PLL_MASK (PLL3 | PLL2 | PLL1 | PLL0)
#define BCKO_MASK (1 << 3)
#define BCKO_64 BCKO_MASK
#define DIF_MASK (3 << 0)
#define DSP (0 << 0)
#define RIGHT_J (1 << 0)
#define LEFT_J (2 << 0)
#define I2S (3 << 0)
/* MD_CTL2 */
#define FS0 (1 << 0)
#define FS1 (1 << 1)
#define FS2 (1 << 2)
#define FS3 (1 << 5)
#define FS_MASK (FS0 | FS1 | FS2 | FS3)
/* MD_CTL3 */
#define BST1 (1 << 3)
/* MD_CTL4 */
#define DACH (1 << 0)
/*
* Playback Volume (table 39)
*
* max : 0x00 : +12.0 dB
* ( 0.5 dB step )
* min : 0xFE : -115.0 dB
* mute: 0xFF
*/
static const DECLARE_TLV_DB_SCALE(out_tlv, -11550, 50, 1);
static const struct snd_kcontrol_new ak4642_snd_controls[] = {
SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC,
0, 0xFF, 1, out_tlv),
};
static const struct snd_kcontrol_new ak4642_headphone_control =
SOC_DAPM_SINGLE("Switch", PW_MGMT2, 6, 1, 0);
static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = {
SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0),
};
static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = {
/* Outputs */
SND_SOC_DAPM_OUTPUT("HPOUTL"),
SND_SOC_DAPM_OUTPUT("HPOUTR"),
SND_SOC_DAPM_OUTPUT("LINEOUT"),
SND_SOC_DAPM_PGA("HPL Out", PW_MGMT2, 5, 0, NULL, 0),
SND_SOC_DAPM_PGA("HPR Out", PW_MGMT2, 4, 0, NULL, 0),
SND_SOC_DAPM_SWITCH("Headphone Enable", SND_SOC_NOPM, 0, 0,
&ak4642_headphone_control),
SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0,
&ak4642_lout_mixer_controls[0],
ARRAY_SIZE(ak4642_lout_mixer_controls)),
/* DAC */
SND_SOC_DAPM_DAC("DAC", "HiFi Playback", PW_MGMT1, 2, 0),
};
static const struct snd_soc_dapm_route ak4642_intercon[] = {
/* Outputs */
{"HPOUTL", NULL, "HPL Out"},
{"HPOUTR", NULL, "HPR Out"},
{"LINEOUT", NULL, "LINEOUT Mixer"},
{"HPL Out", NULL, "Headphone Enable"},
{"HPR Out", NULL, "Headphone Enable"},
{"Headphone Enable", "Switch", "DACH"},
{"DACH", NULL, "DAC"},
{"LINEOUT Mixer", "DACL", "DAC"},
};
/*
* ak4642 register cache
*/
static const struct reg_default ak4642_reg[] = {
{ 0, 0x00 }, { 1, 0x00 }, { 2, 0x01 }, { 3, 0x00 },
{ 4, 0x02 }, { 5, 0x00 }, { 6, 0x00 }, { 7, 0x00 },
{ 8, 0xe1 }, { 9, 0xe1 }, { 10, 0x18 }, { 11, 0x00 },
{ 12, 0xe1 }, { 13, 0x18 }, { 14, 0x11 }, { 15, 0x08 },
{ 16, 0x00 }, { 17, 0x00 }, { 18, 0x00 }, { 19, 0x00 },
{ 20, 0x00 }, { 21, 0x00 }, { 22, 0x00 }, { 23, 0x00 },
{ 24, 0x00 }, { 25, 0x00 }, { 26, 0x00 }, { 27, 0x00 },
{ 28, 0x00 }, { 29, 0x00 }, { 30, 0x00 }, { 31, 0x00 },
{ 32, 0x00 }, { 33, 0x00 }, { 34, 0x00 }, { 35, 0x00 },
{ 36, 0x00 },
};
static const struct reg_default ak4648_reg[] = {
{ 0, 0x00 }, { 1, 0x00 }, { 2, 0x01 }, { 3, 0x00 },
{ 4, 0x02 }, { 5, 0x00 }, { 6, 0x00 }, { 7, 0x00 },
{ 8, 0xe1 }, { 9, 0xe1 }, { 10, 0x18 }, { 11, 0x00 },
{ 12, 0xe1 }, { 13, 0x18 }, { 14, 0x11 }, { 15, 0xb8 },
{ 16, 0x00 }, { 17, 0x00 }, { 18, 0x00 }, { 19, 0x00 },
{ 20, 0x00 }, { 21, 0x00 }, { 22, 0x00 }, { 23, 0x00 },
{ 24, 0x00 }, { 25, 0x00 }, { 26, 0x00 }, { 27, 0x00 },
{ 28, 0x00 }, { 29, 0x00 }, { 30, 0x00 }, { 31, 0x00 },
{ 32, 0x00 }, { 33, 0x00 }, { 34, 0x00 }, { 35, 0x00 },
{ 36, 0x00 }, { 37, 0x88 }, { 38, 0x88 }, { 39, 0x08 },
};
static int ak4642_dai_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
struct snd_soc_codec *codec = dai->codec;
if (is_play) {
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