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kernel-fxtec-pro1x/sound/soc/codecs/alc5632.c
Lars-Peter Clausen 85e7652d89 ASoC: Constify snd_soc_dai_ops structs 
Commit 1ee46ebd("ASoC: Make the DAI ops constant in the DAI structure")
introduced the possibility to have constant DAI ops structures, yet this is
barley used in both existing drivers and also new drivers being submitted,
although none of them modifies its DAI ops structure. The later is not
surprising since existing drivers are often used as templates for new drivers.
So this patch just constifies all existing snd_soc_dai_ops structs to eliminate
the issue altogether.

The patch was generated with the following coccinelle semantic patch:
// <smpl>
@@
identifier ops;
@@
-struct snd_soc_dai_ops ops =
+const struct snd_soc_dai_ops ops =
{ ... };
// </smpl>

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-11-23 10:40:46 +00:00

1159 lines
36 KiB
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/*
* 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 <linux/regmap.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 struct reg_default alc5632_reg_defaults[] = {
{ 2, 0x8080 }, /* R2 - Speaker Output Volume */
{ 4, 0x8080 }, /* R4 - Headphone Output Volume */
{ 6, 0x8080 }, /* R6 - AUXOUT Volume */
{ 8, 0xC800 }, /* R8 - Phone Input */
{ 10, 0xE808 }, /* R10 - LINE_IN Volume */
{ 12, 0x1010 }, /* R12 - STEREO DAC Input Volume */
{ 14, 0x0808 }, /* R14 - MIC Input Volume */
{ 16, 0xEE0F }, /* R16 - Stereo DAC and MIC Routing Control */
{ 18, 0xCBCB }, /* R18 - ADC Record Gain */
{ 20, 0x7F7F }, /* R20 - ADC Record Mixer Control */
{ 24, 0xE010 }, /* R24 - Voice DAC Volume */
{ 28, 0x8008 }, /* R28 - Output Mixer Control */
{ 34, 0x0000 }, /* R34 - Microphone Control */
{ 36, 0x00C0 }, /* R36 - Codec Digital MIC/Digital Boost
Control */
{ 46, 0x0000 }, /* R46 - Stereo DAC/Voice DAC/Stereo ADC
Function Select */
{ 52, 0x8000 }, /* R52 - Main Serial Data Port Control
(Stereo I2S) */
{ 54, 0x0000 }, /* R54 - Extend Serial Data Port Control
(VoDAC_I2S/PCM) */
{ 58, 0x0000 }, /* R58 - Power Management Addition 1 */
{ 60, 0x0000 }, /* R60 - Power Management Addition 2 */
{ 62, 0x8000 }, /* R62 - Power Management Addition 3 */
{ 64, 0x0C0A }, /* R64 - General Purpose Control Register 1 */
{ 66, 0x0000 }, /* R66 - General Purpose Control Register 2 */
{ 68, 0x0000 }, /* R68 - PLL1 Control */
{ 70, 0x0000 }, /* R70 - PLL2 Control */
{ 76, 0xBE3E }, /* R76 - GPIO Pin Configuration */
{ 78, 0xBE3E }, /* R78 - GPIO Pin Polarity */
{ 80, 0x0000 }, /* R80 - GPIO Pin Sticky */
{ 82, 0x0000 }, /* R82 - GPIO Pin Wake Up */
{ 86, 0x0000 }, /* R86 - Pin Sharing */
{ 90, 0x0009 }, /* R90 - Soft Volume Control Setting */
{ 92, 0x0000 }, /* R92 - GPIO_Output Pin Control */
{ 94, 0x3000 }, /* R94 - MISC Control */
{ 96, 0x3075 }, /* R96 - Stereo DAC Clock Control_1 */
{ 98, 0x1010 }, /* R98 - Stereo DAC Clock Control_2 */
{ 100, 0x3110 }, /* R100 - VoDAC_PCM Clock Control_1 */
{ 104, 0x0553 }, /* R104 - Pseudo Stereo and Spatial Effect
Block Control */
{ 106, 0x0000 }, /* R106 - Private Register Address */
};
/* codec private data */
struct alc5632_priv {
struct regmap *regmap;
u8 id;
unsigned int sysclk;
};
static bool alc5632_volatile_register(struct device *dev,
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:
case ALC5632_VENDOR_ID1:
case ALC5632_VENDOR_ID2:
return true;
default:
break;
}
return false;
}
static inline int alc5632_reset(struct regmap *map)
{
return regmap_write(map, ALC5632_RESET, 0x59B4);
}
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("Speaker Playback Volume",
ALC5632_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
/* bit 15 mutes left, bit 7 right */
SOC_DOUBLE("Speaker Playback Switch",
ALC5632_SPK_OUT_VOL, 15, 7, 1, 1),
SOC_DOUBLE_TLV("Headphone Playback Volume",
ALC5632_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
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("Master Playback Volume",
ALC5632_STEREO_DAC_IN_VOL, 8, 0, 63, 1, vdac_tlv),
SOC_DOUBLE("Master Playback Switch",
ALC5632_STEREO_DAC_IN_VOL, 15, 7, 1, 1),
SOC_SINGLE_TLV("Mic1 Capture Volume",
ALC5632_MIC_VOL, 8, 31, 1, vol_tlv),
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 const 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,
};
#ifdef CONFIG_PM
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)
{
struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec);
regcache_sync(alc5632->regmap);
alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
return 0;
}
#else
#define alc5632_suspend NULL
#define alc5632_resume NULL
#endif
static int alc5632_probe(struct snd_soc_codec *codec)
{
struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec);
int ret;
codec->control_data = alc5632->regmap;
ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
if (ret != 0) {
dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
return ret;
}
/* 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,
.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),
};
static struct regmap_config alc5632_regmap = {
.reg_bits = 8,
.val_bits = 16,
.max_register = ALC5632_MAX_REGISTER,
.reg_defaults = alc5632_reg_defaults,
.num_reg_defaults = ARRAY_SIZE(alc5632_reg_defaults),
.volatile_reg = alc5632_volatile_register,
.cache_type = REGCACHE_RBTREE,
};
/*
* alc5632 2 wire address is determined by A1 pin
* state during powerup.
* low = 0x1a
* high = 0x1b
*/
static __devinit int alc5632_i2c_probe(struct i2c_client *client,
const struct i2c_device_id *id)
{
struct alc5632_priv *alc5632;
int ret, ret1, ret2;
unsigned int vid1, vid2;
alc5632 = devm_kzalloc(&client->dev,
sizeof(struct alc5632_priv), GFP_KERNEL);
if (alc5632 == NULL)
return -ENOMEM;
i2c_set_clientdata(client, alc5632);
alc5632->regmap = regmap_init_i2c(client, &alc5632_regmap);
if (IS_ERR(alc5632->regmap)) {
ret = PTR_ERR(alc5632->regmap);
dev_err(&client->dev, "regmap_init() failed: %d\n", ret);
return ret;
}
ret1 = regmap_read(alc5632->regmap, ALC5632_VENDOR_ID1, &vid1);
ret2 = regmap_read(alc5632->regmap, ALC5632_VENDOR_ID2, &vid2);
if (ret1 != 0 || ret2 != 0) {
dev_err(&client->dev,
"Failed to read chip ID: ret1=%d, ret2=%d\n", ret1, ret2);
regmap_exit(alc5632->regmap);
return -EIO;
}
vid2 >>= 8;
if ((vid1 != 0x10EC) || (vid2 != id->driver_data)) {
dev_err(&client->dev,
"Device is not a ALC5632: VID1=0x%x, VID2=0x%x\n", vid1, vid2);
regmap_exit(alc5632->regmap);
return -EINVAL;
}
ret = alc5632_reset(alc5632->regmap);
if (ret < 0) {
dev_err(&client->dev, "Failed to issue reset\n");
regmap_exit(alc5632->regmap);
return ret;
}
alc5632->id = vid2;
switch (alc5632->id) {
case 0x5c:
alc5632_dai.name = "alc5632-hifi";
break;
default:
return -EINVAL;
}
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);
regmap_exit(alc5632->regmap);
return ret;
}
return ret;
}
static int alc5632_i2c_remove(struct i2c_client *client)
{
struct alc5632_priv *alc5632 = i2c_get_clientdata(client);
snd_soc_unregister_codec(&client->dev);
regmap_exit(alc5632->regmap);
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");
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