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44cb209d33
The control_data field is used to initialize the codec's control_data field, but since this is also done by the snd-soc-cache core, the redundant assignment can be removed and the field can be dropped. Signed-off-by: Axel Lin <axel.lin@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
1114 lines
34 KiB
C
1114 lines
34 KiB
C
/*
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* alc5623.c -- alc562[123] ALSA Soc Audio driver
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*
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* Copyright 2008 Realtek Microelectronics
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* Author: flove <flove@realtek.com> Ethan <eku@marvell.com>
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*
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* Copyright 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
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*
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*
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* Based on WM8753.c
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License version 2 as
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* published by the Free Software Foundation.
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*
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*/
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#include <linux/module.h>
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#include <linux/kernel.h>
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#include <linux/init.h>
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#include <linux/delay.h>
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#include <linux/pm.h>
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#include <linux/i2c.h>
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#include <linux/slab.h>
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#include <linux/platform_device.h>
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#include <sound/core.h>
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#include <sound/pcm.h>
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#include <sound/pcm_params.h>
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#include <sound/tlv.h>
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#include <sound/soc.h>
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#include <sound/initval.h>
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#include <sound/alc5623.h>
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#include "alc5623.h"
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static int caps_charge = 2000;
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module_param(caps_charge, int, 0);
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MODULE_PARM_DESC(caps_charge, "ALC5623 cap charge time (msecs)");
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/* codec private data */
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struct alc5623_priv {
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enum snd_soc_control_type control_type;
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u8 id;
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unsigned int sysclk;
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u16 reg_cache[ALC5623_VENDOR_ID2+2];
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unsigned int add_ctrl;
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unsigned int jack_det_ctrl;
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};
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static void alc5623_fill_cache(struct snd_soc_codec *codec)
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{
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int i, step = codec->driver->reg_cache_step;
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u16 *cache = codec->reg_cache;
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/* not really efficient ... */
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for (i = 0 ; i < codec->driver->reg_cache_size ; i += step)
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cache[i] = codec->hw_read(codec, i);
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}
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static inline int alc5623_reset(struct snd_soc_codec *codec)
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{
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return snd_soc_write(codec, ALC5623_RESET, 0);
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}
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static int amp_mixer_event(struct snd_soc_dapm_widget *w,
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struct snd_kcontrol *kcontrol, int event)
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{
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/* to power-on/off class-d amp generators/speaker */
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/* need to write to 'index-46h' register : */
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/* so write index num (here 0x46) to reg 0x6a */
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/* and then 0xffff/0 to reg 0x6c */
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snd_soc_write(w->codec, ALC5623_HID_CTRL_INDEX, 0x46);
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switch (event) {
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case SND_SOC_DAPM_PRE_PMU:
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snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0xFFFF);
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break;
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case SND_SOC_DAPM_POST_PMD:
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snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0);
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break;
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}
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return 0;
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}
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/*
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* ALC5623 Controls
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*/
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static const DECLARE_TLV_DB_SCALE(vol_tlv, -3450, 150, 0);
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static const DECLARE_TLV_DB_SCALE(hp_tlv, -4650, 150, 0);
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static const DECLARE_TLV_DB_SCALE(adc_rec_tlv, -1650, 150, 0);
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static const unsigned int boost_tlv[] = {
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TLV_DB_RANGE_HEAD(3),
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0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
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1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0),
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2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0),
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};
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static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0);
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static const struct snd_kcontrol_new rt5621_vol_snd_controls[] = {
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SOC_DOUBLE_TLV("Speaker Playback Volume",
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ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
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SOC_DOUBLE("Speaker Playback Switch",
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ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
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SOC_DOUBLE_TLV("Headphone Playback Volume",
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ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
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SOC_DOUBLE("Headphone Playback Switch",
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ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
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};
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static const struct snd_kcontrol_new rt5622_vol_snd_controls[] = {
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SOC_DOUBLE_TLV("Speaker Playback Volume",
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ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
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SOC_DOUBLE("Speaker Playback Switch",
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ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
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SOC_DOUBLE_TLV("Line Playback Volume",
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ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
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SOC_DOUBLE("Line Playback Switch",
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ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
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};
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static const struct snd_kcontrol_new alc5623_vol_snd_controls[] = {
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SOC_DOUBLE_TLV("Line Playback Volume",
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ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
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SOC_DOUBLE("Line Playback Switch",
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ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
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SOC_DOUBLE_TLV("Headphone Playback Volume",
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ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
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SOC_DOUBLE("Headphone Playback Switch",
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ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
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};
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static const struct snd_kcontrol_new alc5623_snd_controls[] = {
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SOC_DOUBLE_TLV("Auxout Playback Volume",
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ALC5623_MONO_AUX_OUT_VOL, 8, 0, 31, 1, hp_tlv),
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SOC_DOUBLE("Auxout Playback Switch",
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ALC5623_MONO_AUX_OUT_VOL, 15, 7, 1, 1),
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SOC_DOUBLE_TLV("PCM Playback Volume",
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ALC5623_STEREO_DAC_VOL, 8, 0, 31, 1, vol_tlv),
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SOC_DOUBLE_TLV("AuxI Capture Volume",
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ALC5623_AUXIN_VOL, 8, 0, 31, 1, vol_tlv),
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SOC_DOUBLE_TLV("LineIn Capture Volume",
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ALC5623_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv),
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SOC_SINGLE_TLV("Mic1 Capture Volume",
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ALC5623_MIC_VOL, 8, 31, 1, vol_tlv),
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SOC_SINGLE_TLV("Mic2 Capture Volume",
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ALC5623_MIC_VOL, 0, 31, 1, vol_tlv),
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SOC_DOUBLE_TLV("Rec Capture Volume",
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ALC5623_ADC_REC_GAIN, 7, 0, 31, 0, adc_rec_tlv),
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SOC_SINGLE_TLV("Mic 1 Boost Volume",
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ALC5623_MIC_CTRL, 10, 2, 0, boost_tlv),
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SOC_SINGLE_TLV("Mic 2 Boost Volume",
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ALC5623_MIC_CTRL, 8, 2, 0, boost_tlv),
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SOC_SINGLE_TLV("Digital Boost Volume",
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ALC5623_ADD_CTRL_REG, 4, 3, 0, dig_tlv),
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};
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/*
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* DAPM Controls
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*/
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static const struct snd_kcontrol_new alc5623_hp_mixer_controls[] = {
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SOC_DAPM_SINGLE("LI2HP Playback Switch", ALC5623_LINE_IN_VOL, 15, 1, 1),
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SOC_DAPM_SINGLE("AUXI2HP Playback Switch", ALC5623_AUXIN_VOL, 15, 1, 1),
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SOC_DAPM_SINGLE("MIC12HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 15, 1, 1),
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SOC_DAPM_SINGLE("MIC22HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 7, 1, 1),
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SOC_DAPM_SINGLE("DAC2HP Playback Switch", ALC5623_STEREO_DAC_VOL, 15, 1, 1),
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};
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static const struct snd_kcontrol_new alc5623_hpl_mixer_controls[] = {
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SOC_DAPM_SINGLE("ADC2HP_L Playback Switch", ALC5623_ADC_REC_GAIN, 15, 1, 1),
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};
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static const struct snd_kcontrol_new alc5623_hpr_mixer_controls[] = {
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SOC_DAPM_SINGLE("ADC2HP_R Playback Switch", ALC5623_ADC_REC_GAIN, 14, 1, 1),
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};
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static const struct snd_kcontrol_new alc5623_mono_mixer_controls[] = {
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SOC_DAPM_SINGLE("ADC2MONO_L Playback Switch", ALC5623_ADC_REC_GAIN, 13, 1, 1),
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SOC_DAPM_SINGLE("ADC2MONO_R Playback Switch", ALC5623_ADC_REC_GAIN, 12, 1, 1),
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SOC_DAPM_SINGLE("LI2MONO Playback Switch", ALC5623_LINE_IN_VOL, 13, 1, 1),
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SOC_DAPM_SINGLE("AUXI2MONO Playback Switch", ALC5623_AUXIN_VOL, 13, 1, 1),
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SOC_DAPM_SINGLE("MIC12MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 13, 1, 1),
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SOC_DAPM_SINGLE("MIC22MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 5, 1, 1),
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SOC_DAPM_SINGLE("DAC2MONO Playback Switch", ALC5623_STEREO_DAC_VOL, 13, 1, 1),
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};
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static const struct snd_kcontrol_new alc5623_speaker_mixer_controls[] = {
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SOC_DAPM_SINGLE("LI2SPK Playback Switch", ALC5623_LINE_IN_VOL, 14, 1, 1),
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SOC_DAPM_SINGLE("AUXI2SPK Playback Switch", ALC5623_AUXIN_VOL, 14, 1, 1),
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SOC_DAPM_SINGLE("MIC12SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 14, 1, 1),
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SOC_DAPM_SINGLE("MIC22SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 6, 1, 1),
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SOC_DAPM_SINGLE("DAC2SPK Playback Switch", ALC5623_STEREO_DAC_VOL, 14, 1, 1),
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};
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/* Left Record Mixer */
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static const struct snd_kcontrol_new alc5623_captureL_mixer_controls[] = {
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SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 14, 1, 1),
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SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 13, 1, 1),
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SOC_DAPM_SINGLE("LineInL Capture Switch", ALC5623_ADC_REC_MIXER, 12, 1, 1),
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SOC_DAPM_SINGLE("Left AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 11, 1, 1),
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SOC_DAPM_SINGLE("HPMixerL Capture Switch", ALC5623_ADC_REC_MIXER, 10, 1, 1),
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SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 9, 1, 1),
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SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 8, 1, 1),
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};
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/* Right Record Mixer */
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static const struct snd_kcontrol_new alc5623_captureR_mixer_controls[] = {
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SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 6, 1, 1),
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SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 5, 1, 1),
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SOC_DAPM_SINGLE("LineInR Capture Switch", ALC5623_ADC_REC_MIXER, 4, 1, 1),
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SOC_DAPM_SINGLE("Right AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 3, 1, 1),
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SOC_DAPM_SINGLE("HPMixerR Capture Switch", ALC5623_ADC_REC_MIXER, 2, 1, 1),
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SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 1, 1, 1),
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SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 0, 1, 1),
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};
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static const char *alc5623_spk_n_sour_sel[] = {
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"RN/-R", "RP/+R", "LN/-R", "Vmid" };
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static const char *alc5623_hpl_out_input_sel[] = {
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"Vmid", "HP Left Mix"};
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static const char *alc5623_hpr_out_input_sel[] = {
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"Vmid", "HP Right Mix"};
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static const char *alc5623_spkout_input_sel[] = {
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"Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
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static const char *alc5623_aux_out_input_sel[] = {
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"Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
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/* auxout output mux */
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static const struct soc_enum alc5623_aux_out_input_enum =
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SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 6, 4, alc5623_aux_out_input_sel);
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static const struct snd_kcontrol_new alc5623_auxout_mux_controls =
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SOC_DAPM_ENUM("Route", alc5623_aux_out_input_enum);
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/* speaker output mux */
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static const struct soc_enum alc5623_spkout_input_enum =
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SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 10, 4, alc5623_spkout_input_sel);
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static const struct snd_kcontrol_new alc5623_spkout_mux_controls =
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SOC_DAPM_ENUM("Route", alc5623_spkout_input_enum);
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/* headphone left output mux */
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static const struct soc_enum alc5623_hpl_out_input_enum =
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SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 9, 2, alc5623_hpl_out_input_sel);
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static const struct snd_kcontrol_new alc5623_hpl_out_mux_controls =
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SOC_DAPM_ENUM("Route", alc5623_hpl_out_input_enum);
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/* headphone right output mux */
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static const struct soc_enum alc5623_hpr_out_input_enum =
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SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 8, 2, alc5623_hpr_out_input_sel);
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static const struct snd_kcontrol_new alc5623_hpr_out_mux_controls =
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SOC_DAPM_ENUM("Route", alc5623_hpr_out_input_enum);
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/* speaker output N select */
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static const struct soc_enum alc5623_spk_n_sour_enum =
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SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 14, 4, alc5623_spk_n_sour_sel);
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static const struct snd_kcontrol_new alc5623_spkoutn_mux_controls =
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SOC_DAPM_ENUM("Route", alc5623_spk_n_sour_enum);
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static const struct snd_soc_dapm_widget alc5623_dapm_widgets[] = {
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/* Muxes */
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SND_SOC_DAPM_MUX("AuxOut Mux", SND_SOC_NOPM, 0, 0,
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&alc5623_auxout_mux_controls),
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SND_SOC_DAPM_MUX("SpeakerOut Mux", SND_SOC_NOPM, 0, 0,
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&alc5623_spkout_mux_controls),
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SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0,
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&alc5623_hpl_out_mux_controls),
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SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0,
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&alc5623_hpr_out_mux_controls),
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SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0,
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&alc5623_spkoutn_mux_controls),
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/* output mixers */
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SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0,
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&alc5623_hp_mixer_controls[0],
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ARRAY_SIZE(alc5623_hp_mixer_controls)),
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SND_SOC_DAPM_MIXER("HPR Mix", ALC5623_PWR_MANAG_ADD2, 4, 0,
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&alc5623_hpr_mixer_controls[0],
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ARRAY_SIZE(alc5623_hpr_mixer_controls)),
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SND_SOC_DAPM_MIXER("HPL Mix", ALC5623_PWR_MANAG_ADD2, 5, 0,
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&alc5623_hpl_mixer_controls[0],
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ARRAY_SIZE(alc5623_hpl_mixer_controls)),
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SND_SOC_DAPM_MIXER("HPOut Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
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SND_SOC_DAPM_MIXER("Mono Mix", ALC5623_PWR_MANAG_ADD2, 2, 0,
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&alc5623_mono_mixer_controls[0],
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ARRAY_SIZE(alc5623_mono_mixer_controls)),
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SND_SOC_DAPM_MIXER("Speaker Mix", ALC5623_PWR_MANAG_ADD2, 3, 0,
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&alc5623_speaker_mixer_controls[0],
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ARRAY_SIZE(alc5623_speaker_mixer_controls)),
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/* input mixers */
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SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5623_PWR_MANAG_ADD2, 1, 0,
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&alc5623_captureL_mixer_controls[0],
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ARRAY_SIZE(alc5623_captureL_mixer_controls)),
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SND_SOC_DAPM_MIXER("Right Capture Mix", ALC5623_PWR_MANAG_ADD2, 0, 0,
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&alc5623_captureR_mixer_controls[0],
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ARRAY_SIZE(alc5623_captureR_mixer_controls)),
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SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback",
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ALC5623_PWR_MANAG_ADD2, 9, 0),
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SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback",
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ALC5623_PWR_MANAG_ADD2, 8, 0),
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SND_SOC_DAPM_MIXER("I2S Mix", ALC5623_PWR_MANAG_ADD1, 15, 0, NULL, 0),
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SND_SOC_DAPM_MIXER("AuxI Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
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SND_SOC_DAPM_MIXER("Line Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
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SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture",
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ALC5623_PWR_MANAG_ADD2, 7, 0),
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SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture",
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ALC5623_PWR_MANAG_ADD2, 6, 0),
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SND_SOC_DAPM_PGA("Left Headphone", ALC5623_PWR_MANAG_ADD3, 10, 0, NULL, 0),
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SND_SOC_DAPM_PGA("Right Headphone", ALC5623_PWR_MANAG_ADD3, 9, 0, NULL, 0),
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SND_SOC_DAPM_PGA("SpeakerOut", ALC5623_PWR_MANAG_ADD3, 12, 0, NULL, 0),
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SND_SOC_DAPM_PGA("Left AuxOut", ALC5623_PWR_MANAG_ADD3, 14, 0, NULL, 0),
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SND_SOC_DAPM_PGA("Right AuxOut", ALC5623_PWR_MANAG_ADD3, 13, 0, NULL, 0),
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SND_SOC_DAPM_PGA("Left LineIn", ALC5623_PWR_MANAG_ADD3, 7, 0, NULL, 0),
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SND_SOC_DAPM_PGA("Right LineIn", ALC5623_PWR_MANAG_ADD3, 6, 0, NULL, 0),
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SND_SOC_DAPM_PGA("Left AuxI", ALC5623_PWR_MANAG_ADD3, 5, 0, NULL, 0),
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SND_SOC_DAPM_PGA("Right AuxI", ALC5623_PWR_MANAG_ADD3, 4, 0, NULL, 0),
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SND_SOC_DAPM_PGA("MIC1 PGA", ALC5623_PWR_MANAG_ADD3, 3, 0, NULL, 0),
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SND_SOC_DAPM_PGA("MIC2 PGA", ALC5623_PWR_MANAG_ADD3, 2, 0, NULL, 0),
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SND_SOC_DAPM_PGA("MIC1 Pre Amp", ALC5623_PWR_MANAG_ADD3, 1, 0, NULL, 0),
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SND_SOC_DAPM_PGA("MIC2 Pre Amp", ALC5623_PWR_MANAG_ADD3, 0, 0, NULL, 0),
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SND_SOC_DAPM_MICBIAS("Mic Bias1", ALC5623_PWR_MANAG_ADD1, 11, 0),
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SND_SOC_DAPM_OUTPUT("AUXOUTL"),
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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 */
|
|
/* useful 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->dapm.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 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->dapm.suspend_bias_level == SND_SOC_BIAS_ON) {
|
|
alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
|
|
codec->dapm.bias_level = SND_SOC_BIAS_ON;
|
|
alc5623_set_bias_level(codec, codec->dapm.bias_level);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int alc5623_probe(struct snd_soc_codec *codec)
|
|
{
|
|
struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
|
|
struct snd_soc_dapm_context *dapm = &codec->dapm;
|
|
int ret;
|
|
|
|
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) {
|
|
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;
|
|
default:
|
|
return -EINVAL;
|
|
}
|
|
|
|
snd_soc_add_controls(codec, alc5623_snd_controls,
|
|
ARRAY_SIZE(alc5623_snd_controls));
|
|
|
|
snd_soc_dapm_new_controls(dapm, alc5623_dapm_widgets,
|
|
ARRAY_SIZE(alc5623_dapm_widgets));
|
|
|
|
/* set up audio path interconnects */
|
|
snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
|
|
|
|
switch (alc5623->id) {
|
|
case 0x21:
|
|
case 0x22:
|
|
snd_soc_dapm_new_controls(dapm, alc5623_dapm_amp_widgets,
|
|
ARRAY_SIZE(alc5623_dapm_amp_widgets));
|
|
snd_soc_dapm_add_routes(dapm, intercon_amp_spk,
|
|
ARRAY_SIZE(intercon_amp_spk));
|
|
break;
|
|
case 0x23:
|
|
snd_soc_dapm_add_routes(dapm, intercon_spk,
|
|
ARRAY_SIZE(intercon_spk));
|
|
break;
|
|
default:
|
|
return -EINVAL;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
/* power down chip */
|
|
static int alc5623_remove(struct snd_soc_codec *codec)
|
|
{
|
|
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)
|
|
return -ENOMEM;
|
|
|
|
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;
|
|
case 0x23:
|
|
alc5623_dai.name = "alc5623-hifi";
|
|
break;
|
|
default:
|
|
kfree(alc5623);
|
|
return -EINVAL;
|
|
}
|
|
|
|
i2c_set_clientdata(client, alc5623);
|
|
alc5623->control_type = SND_SOC_I2C;
|
|
|
|
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);
|
|
kfree(alc5623);
|
|
}
|
|
|
|
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");
|