| /* |
| * alc5623.c -- alc562[123] ALSA Soc Audio driver |
| * |
| * Copyright 2008 Realtek Microelectronics |
| * Author: flove <flove@realtek.com> Ethan <eku@marvell.com> |
| * |
| * Copyright 2010 Arnaud Patard <arnaud.patard@rtp-net.org> |
| * |
| * |
| * Based on WM8753.c |
| * |
| * This program is free software; you can redistribute it and/or modify |
| * it under the terms of the GNU General Public License version 2 as |
| * published by the Free Software Foundation. |
| * |
| */ |
| |
| #include <linux/module.h> |
| #include <linux/kernel.h> |
| #include <linux/init.h> |
| #include <linux/delay.h> |
| #include <linux/pm.h> |
| #include <linux/i2c.h> |
| #include <linux/slab.h> |
| #include <linux/platform_device.h> |
| #include <sound/core.h> |
| #include <sound/pcm.h> |
| #include <sound/pcm_params.h> |
| #include <sound/tlv.h> |
| #include <sound/soc.h> |
| #include <sound/initval.h> |
| #include <sound/alc5623.h> |
| |
| #include "alc5623.h" |
| |
| static int caps_charge = 2000; |
| module_param(caps_charge, int, 0); |
| MODULE_PARM_DESC(caps_charge, "ALC5623 cap charge time (msecs)"); |
| |
| /* codec private data */ |
| struct alc5623_priv { |
| enum snd_soc_control_type control_type; |
| void *control_data; |
| struct mutex mutex; |
| u8 id; |
| unsigned int sysclk; |
| u16 reg_cache[ALC5623_VENDOR_ID2+2]; |
| unsigned int add_ctrl; |
| unsigned int jack_det_ctrl; |
| }; |
| |
| static void alc5623_fill_cache(struct snd_soc_codec *codec) |
| { |
| int i, step = codec->driver->reg_cache_step; |
| u16 *cache = codec->reg_cache; |
| |
| /* not really efficient ... */ |
| for (i = 0 ; i < codec->driver->reg_cache_size ; i += step) |
| cache[i] = codec->hw_read(codec, i); |
| } |
| |
| static inline int alc5623_reset(struct snd_soc_codec *codec) |
| { |
| return snd_soc_write(codec, ALC5623_RESET, 0); |
| } |
| |
| static int amp_mixer_event(struct snd_soc_dapm_widget *w, |
| struct snd_kcontrol *kcontrol, int event) |
| { |
| /* to power-on/off class-d amp generators/speaker */ |
| /* need to write to 'index-46h' register : */ |
| /* so write index num (here 0x46) to reg 0x6a */ |
| /* and then 0xffff/0 to reg 0x6c */ |
| snd_soc_write(w->codec, ALC5623_HID_CTRL_INDEX, 0x46); |
| |
| switch (event) { |
| case SND_SOC_DAPM_PRE_PMU: |
| snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0xFFFF); |
| break; |
| case SND_SOC_DAPM_POST_PMD: |
| snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0); |
| break; |
| } |
| |
| return 0; |
| } |
| |
| /* |
| * ALC5623 Controls |
| */ |
| |
| static const DECLARE_TLV_DB_SCALE(vol_tlv, -3450, 150, 0); |
| static const DECLARE_TLV_DB_SCALE(hp_tlv, -4650, 150, 0); |
| static const DECLARE_TLV_DB_SCALE(adc_rec_tlv, -1650, 150, 0); |
| static const unsigned int boost_tlv[] = { |
| TLV_DB_RANGE_HEAD(3), |
| 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), |
| 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), |
| 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), |
| }; |
| static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0); |
| |
| static const struct snd_kcontrol_new rt5621_vol_snd_controls[] = { |
| SOC_DOUBLE_TLV("Speaker Playback Volume", |
| ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), |
| SOC_DOUBLE("Speaker Playback Switch", |
| ALC5623_SPK_OUT_VOL, 15, 7, 1, 1), |
| SOC_DOUBLE_TLV("Headphone Playback Volume", |
| ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), |
| SOC_DOUBLE("Headphone Playback Switch", |
| ALC5623_HP_OUT_VOL, 15, 7, 1, 1), |
| }; |
| |
| static const struct snd_kcontrol_new rt5622_vol_snd_controls[] = { |
| SOC_DOUBLE_TLV("Speaker Playback Volume", |
| ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), |
| SOC_DOUBLE("Speaker Playback Switch", |
| ALC5623_SPK_OUT_VOL, 15, 7, 1, 1), |
| SOC_DOUBLE_TLV("Line Playback Volume", |
| ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), |
| SOC_DOUBLE("Line Playback Switch", |
| ALC5623_HP_OUT_VOL, 15, 7, 1, 1), |
| }; |
| |
| static const struct snd_kcontrol_new alc5623_vol_snd_controls[] = { |
| SOC_DOUBLE_TLV("Line Playback Volume", |
| ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), |
| SOC_DOUBLE("Line Playback Switch", |
| ALC5623_SPK_OUT_VOL, 15, 7, 1, 1), |
| SOC_DOUBLE_TLV("Headphone Playback Volume", |
| ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), |
| SOC_DOUBLE("Headphone Playback Switch", |
| ALC5623_HP_OUT_VOL, 15, 7, 1, 1), |
| }; |
| |
| static const struct snd_kcontrol_new alc5623_snd_controls[] = { |
| SOC_DOUBLE_TLV("Auxout Playback Volume", |
| ALC5623_MONO_AUX_OUT_VOL, 8, 0, 31, 1, hp_tlv), |
| SOC_DOUBLE("Auxout Playback Switch", |
| ALC5623_MONO_AUX_OUT_VOL, 15, 7, 1, 1), |
| SOC_DOUBLE_TLV("PCM Playback Volume", |
| ALC5623_STEREO_DAC_VOL, 8, 0, 31, 1, vol_tlv), |
| SOC_DOUBLE_TLV("AuxI Capture Volume", |
| ALC5623_AUXIN_VOL, 8, 0, 31, 1, vol_tlv), |
| SOC_DOUBLE_TLV("LineIn Capture Volume", |
| ALC5623_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv), |
| SOC_SINGLE_TLV("Mic1 Capture Volume", |
| ALC5623_MIC_VOL, 8, 31, 1, vol_tlv), |
| SOC_SINGLE_TLV("Mic2 Capture Volume", |
| ALC5623_MIC_VOL, 0, 31, 1, vol_tlv), |
| SOC_DOUBLE_TLV("Rec Capture Volume", |
| ALC5623_ADC_REC_GAIN, 7, 0, 31, 0, adc_rec_tlv), |
| SOC_SINGLE_TLV("Mic 1 Boost Volume", |
| ALC5623_MIC_CTRL, 10, 2, 0, boost_tlv), |
| SOC_SINGLE_TLV("Mic 2 Boost Volume", |
| ALC5623_MIC_CTRL, 8, 2, 0, boost_tlv), |
| SOC_SINGLE_TLV("Digital Boost Volume", |
| ALC5623_ADD_CTRL_REG, 4, 3, 0, dig_tlv), |
| }; |
| |
| /* |
| * DAPM Controls |
| */ |
| static const struct snd_kcontrol_new alc5623_hp_mixer_controls[] = { |
| SOC_DAPM_SINGLE("LI2HP Playback Switch", ALC5623_LINE_IN_VOL, 15, 1, 1), |
| SOC_DAPM_SINGLE("AUXI2HP Playback Switch", ALC5623_AUXIN_VOL, 15, 1, 1), |
| SOC_DAPM_SINGLE("MIC12HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 15, 1, 1), |
| SOC_DAPM_SINGLE("MIC22HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 7, 1, 1), |
| SOC_DAPM_SINGLE("DAC2HP Playback Switch", ALC5623_STEREO_DAC_VOL, 15, 1, 1), |
| }; |
| |
| static const struct snd_kcontrol_new alc5623_hpl_mixer_controls[] = { |
| SOC_DAPM_SINGLE("ADC2HP_L Playback Switch", ALC5623_ADC_REC_GAIN, 15, 1, 1), |
| }; |
| |
| static const struct snd_kcontrol_new alc5623_hpr_mixer_controls[] = { |
| SOC_DAPM_SINGLE("ADC2HP_R Playback Switch", ALC5623_ADC_REC_GAIN, 14, 1, 1), |
| }; |
| |
| static const struct snd_kcontrol_new alc5623_mono_mixer_controls[] = { |
| SOC_DAPM_SINGLE("ADC2MONO_L Playback Switch", ALC5623_ADC_REC_GAIN, 13, 1, 1), |
| SOC_DAPM_SINGLE("ADC2MONO_R Playback Switch", ALC5623_ADC_REC_GAIN, 12, 1, 1), |
| SOC_DAPM_SINGLE("LI2MONO Playback Switch", ALC5623_LINE_IN_VOL, 13, 1, 1), |
| SOC_DAPM_SINGLE("AUXI2MONO Playback Switch", ALC5623_AUXIN_VOL, 13, 1, 1), |
| SOC_DAPM_SINGLE("MIC12MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 13, 1, 1), |
| SOC_DAPM_SINGLE("MIC22MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 5, 1, 1), |
| SOC_DAPM_SINGLE("DAC2MONO Playback Switch", ALC5623_STEREO_DAC_VOL, 13, 1, 1), |
| }; |
| |
| static const struct snd_kcontrol_new alc5623_speaker_mixer_controls[] = { |
| SOC_DAPM_SINGLE("LI2SPK Playback Switch", ALC5623_LINE_IN_VOL, 14, 1, 1), |
| SOC_DAPM_SINGLE("AUXI2SPK Playback Switch", ALC5623_AUXIN_VOL, 14, 1, 1), |
| SOC_DAPM_SINGLE("MIC12SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 14, 1, 1), |
| SOC_DAPM_SINGLE("MIC22SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 6, 1, 1), |
| SOC_DAPM_SINGLE("DAC2SPK Playback Switch", ALC5623_STEREO_DAC_VOL, 14, 1, 1), |
| }; |
| |
| /* Left Record Mixer */ |
| static const struct snd_kcontrol_new alc5623_captureL_mixer_controls[] = { |
| SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 14, 1, 1), |
| SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 13, 1, 1), |
| SOC_DAPM_SINGLE("LineInL Capture Switch", ALC5623_ADC_REC_MIXER, 12, 1, 1), |
| SOC_DAPM_SINGLE("Left AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 11, 1, 1), |
| SOC_DAPM_SINGLE("HPMixerL Capture Switch", ALC5623_ADC_REC_MIXER, 10, 1, 1), |
| SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 9, 1, 1), |
| SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 8, 1, 1), |
| }; |
| |
| /* Right Record Mixer */ |
| static const struct snd_kcontrol_new alc5623_captureR_mixer_controls[] = { |
| SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 6, 1, 1), |
| SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 5, 1, 1), |
| SOC_DAPM_SINGLE("LineInR Capture Switch", ALC5623_ADC_REC_MIXER, 4, 1, 1), |
| SOC_DAPM_SINGLE("Right AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 3, 1, 1), |
| SOC_DAPM_SINGLE("HPMixerR Capture Switch", ALC5623_ADC_REC_MIXER, 2, 1, 1), |
| SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 1, 1, 1), |
| SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 0, 1, 1), |
| }; |
| |
| static const char *alc5623_spk_n_sour_sel[] = { |
| "RN/-R", "RP/+R", "LN/-R", "Vmid" }; |
| static const char *alc5623_hpl_out_input_sel[] = { |
| "Vmid", "HP Left Mix"}; |
| static const char *alc5623_hpr_out_input_sel[] = { |
| "Vmid", "HP Right Mix"}; |
| static const char *alc5623_spkout_input_sel[] = { |
| "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; |
| static const char *alc5623_aux_out_input_sel[] = { |
| "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; |
| |
| /* auxout output mux */ |
| static const struct soc_enum alc5623_aux_out_input_enum = |
| SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 6, 4, alc5623_aux_out_input_sel); |
| static const struct snd_kcontrol_new alc5623_auxout_mux_controls = |
| SOC_DAPM_ENUM("Route", alc5623_aux_out_input_enum); |
| |
| /* speaker output mux */ |
| static const struct soc_enum alc5623_spkout_input_enum = |
| SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 10, 4, alc5623_spkout_input_sel); |
| static const struct snd_kcontrol_new alc5623_spkout_mux_controls = |
| SOC_DAPM_ENUM("Route", alc5623_spkout_input_enum); |
| |
| /* headphone left output mux */ |
| static const struct soc_enum alc5623_hpl_out_input_enum = |
| SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 9, 2, alc5623_hpl_out_input_sel); |
| static const struct snd_kcontrol_new alc5623_hpl_out_mux_controls = |
| SOC_DAPM_ENUM("Route", alc5623_hpl_out_input_enum); |
| |
| /* headphone right output mux */ |
| static const struct soc_enum alc5623_hpr_out_input_enum = |
| SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 8, 2, alc5623_hpr_out_input_sel); |
| static const struct snd_kcontrol_new alc5623_hpr_out_mux_controls = |
| SOC_DAPM_ENUM("Route", alc5623_hpr_out_input_enum); |
| |
| /* speaker output N select */ |
| static const struct soc_enum alc5623_spk_n_sour_enum = |
| SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 14, 4, alc5623_spk_n_sour_sel); |
| static const struct snd_kcontrol_new alc5623_spkoutn_mux_controls = |
| SOC_DAPM_ENUM("Route", alc5623_spk_n_sour_enum); |
| |
| static const struct snd_soc_dapm_widget alc5623_dapm_widgets[] = { |
| /* Muxes */ |
| SND_SOC_DAPM_MUX("AuxOut Mux", SND_SOC_NOPM, 0, 0, |
| &alc5623_auxout_mux_controls), |
| SND_SOC_DAPM_MUX("SpeakerOut Mux", SND_SOC_NOPM, 0, 0, |
| &alc5623_spkout_mux_controls), |
| SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, |
| &alc5623_hpl_out_mux_controls), |
| SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, |
| &alc5623_hpr_out_mux_controls), |
| SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0, |
| &alc5623_spkoutn_mux_controls), |
| |
| /* output mixers */ |
| SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0, |
| &alc5623_hp_mixer_controls[0], |
| ARRAY_SIZE(alc5623_hp_mixer_controls)), |
| SND_SOC_DAPM_MIXER("HPR Mix", ALC5623_PWR_MANAG_ADD2, 4, 0, |
| &alc5623_hpr_mixer_controls[0], |
| ARRAY_SIZE(alc5623_hpr_mixer_controls)), |
| SND_SOC_DAPM_MIXER("HPL Mix", ALC5623_PWR_MANAG_ADD2, 5, 0, |
| &alc5623_hpl_mixer_controls[0], |
| ARRAY_SIZE(alc5623_hpl_mixer_controls)), |
| SND_SOC_DAPM_MIXER("HPOut Mix", SND_SOC_NOPM, 0, 0, NULL, 0), |
| SND_SOC_DAPM_MIXER("Mono Mix", ALC5623_PWR_MANAG_ADD2, 2, 0, |
| &alc5623_mono_mixer_controls[0], |
| ARRAY_SIZE(alc5623_mono_mixer_controls)), |
| SND_SOC_DAPM_MIXER("Speaker Mix", ALC5623_PWR_MANAG_ADD2, 3, 0, |
| &alc5623_speaker_mixer_controls[0], |
| ARRAY_SIZE(alc5623_speaker_mixer_controls)), |
| |
| /* input mixers */ |
| SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5623_PWR_MANAG_ADD2, 1, 0, |
| &alc5623_captureL_mixer_controls[0], |
| ARRAY_SIZE(alc5623_captureL_mixer_controls)), |
| SND_SOC_DAPM_MIXER("Right Capture Mix", ALC5623_PWR_MANAG_ADD2, 0, 0, |
| &alc5623_captureR_mixer_controls[0], |
| ARRAY_SIZE(alc5623_captureR_mixer_controls)), |
| |
| SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", |
| ALC5623_PWR_MANAG_ADD2, 9, 0), |
| SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", |
| ALC5623_PWR_MANAG_ADD2, 8, 0), |
| SND_SOC_DAPM_MIXER("I2S Mix", ALC5623_PWR_MANAG_ADD1, 15, 0, NULL, 0), |
| SND_SOC_DAPM_MIXER("AuxI Mix", SND_SOC_NOPM, 0, 0, NULL, 0), |
| SND_SOC_DAPM_MIXER("Line Mix", SND_SOC_NOPM, 0, 0, NULL, 0), |
| SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", |
| ALC5623_PWR_MANAG_ADD2, 7, 0), |
| SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", |
| ALC5623_PWR_MANAG_ADD2, 6, 0), |
| SND_SOC_DAPM_PGA("Left Headphone", ALC5623_PWR_MANAG_ADD3, 10, 0, NULL, 0), |
| SND_SOC_DAPM_PGA("Right Headphone", ALC5623_PWR_MANAG_ADD3, 9, 0, NULL, 0), |
| SND_SOC_DAPM_PGA("SpeakerOut", ALC5623_PWR_MANAG_ADD3, 12, 0, NULL, 0), |
| SND_SOC_DAPM_PGA("Left AuxOut", ALC5623_PWR_MANAG_ADD3, 14, 0, NULL, 0), |
| SND_SOC_DAPM_PGA("Right AuxOut", ALC5623_PWR_MANAG_ADD3, 13, 0, NULL, 0), |
| SND_SOC_DAPM_PGA("Left LineIn", ALC5623_PWR_MANAG_ADD3, 7, 0, NULL, 0), |
| SND_SOC_DAPM_PGA("Right LineIn", ALC5623_PWR_MANAG_ADD3, 6, 0, NULL, 0), |
| SND_SOC_DAPM_PGA("Left AuxI", ALC5623_PWR_MANAG_ADD3, 5, 0, NULL, 0), |
| SND_SOC_DAPM_PGA("Right AuxI", ALC5623_PWR_MANAG_ADD3, 4, 0, NULL, 0), |
| SND_SOC_DAPM_PGA("MIC1 PGA", ALC5623_PWR_MANAG_ADD3, 3, 0, NULL, 0), |
| SND_SOC_DAPM_PGA("MIC2 PGA", ALC5623_PWR_MANAG_ADD3, 2, 0, NULL, 0), |
| SND_SOC_DAPM_PGA("MIC1 Pre Amp", ALC5623_PWR_MANAG_ADD3, 1, 0, NULL, 0), |
| SND_SOC_DAPM_PGA("MIC2 Pre Amp", ALC5623_PWR_MANAG_ADD3, 0, 0, NULL, 0), |
| SND_SOC_DAPM_MICBIAS("Mic Bias1", ALC5623_PWR_MANAG_ADD1, 11, 0), |
| |
| SND_SOC_DAPM_OUTPUT("AUXOUTL"), |
| SND_SOC_DAPM_OUTPUT("AUXOUTR"), |
| SND_SOC_DAPM_OUTPUT("HPL"), |
| SND_SOC_DAPM_OUTPUT("HPR"), |
| SND_SOC_DAPM_OUTPUT("SPKOUT"), |
| SND_SOC_DAPM_OUTPUT("SPKOUTN"), |
| SND_SOC_DAPM_INPUT("LINEINL"), |
| SND_SOC_DAPM_INPUT("LINEINR"), |
| SND_SOC_DAPM_INPUT("AUXINL"), |
| SND_SOC_DAPM_INPUT("AUXINR"), |
| SND_SOC_DAPM_INPUT("MIC1"), |
| SND_SOC_DAPM_INPUT("MIC2"), |
| SND_SOC_DAPM_VMID("Vmid"), |
| }; |
| |
| static const char *alc5623_amp_names[] = {"AB Amp", "D Amp"}; |
| static const struct soc_enum alc5623_amp_enum = |
| SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 13, 2, alc5623_amp_names); |
| static const struct snd_kcontrol_new alc5623_amp_mux_controls = |
| SOC_DAPM_ENUM("Route", alc5623_amp_enum); |
| |
| static const struct snd_soc_dapm_widget alc5623_dapm_amp_widgets[] = { |
| SND_SOC_DAPM_PGA_E("D Amp", ALC5623_PWR_MANAG_ADD2, 14, 0, NULL, 0, |
| amp_mixer_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), |
| SND_SOC_DAPM_PGA("AB Amp", ALC5623_PWR_MANAG_ADD2, 15, 0, NULL, 0), |
| SND_SOC_DAPM_MUX("AB-D Amp Mux", SND_SOC_NOPM, 0, 0, |
| &alc5623_amp_mux_controls), |
| }; |
| |
| static const struct snd_soc_dapm_route intercon[] = { |
| /* virtual mixer - mixes left & right channels */ |
| {"I2S Mix", NULL, "Left DAC"}, |
| {"I2S Mix", NULL, "Right DAC"}, |
| {"Line Mix", NULL, "Right LineIn"}, |
| {"Line Mix", NULL, "Left LineIn"}, |
| {"AuxI Mix", NULL, "Left AuxI"}, |
| {"AuxI Mix", NULL, "Right AuxI"}, |
| {"AUXOUTL", NULL, "Left AuxOut"}, |
| {"AUXOUTR", NULL, "Right AuxOut"}, |
| |
| /* HP mixer */ |
| {"HPL Mix", "ADC2HP_L Playback Switch", "Left Capture Mix"}, |
| {"HPL Mix", NULL, "HP Mix"}, |
| {"HPR Mix", "ADC2HP_R Playback Switch", "Right Capture Mix"}, |
| {"HPR Mix", NULL, "HP Mix"}, |
| {"HP Mix", "LI2HP Playback Switch", "Line Mix"}, |
| {"HP Mix", "AUXI2HP Playback Switch", "AuxI Mix"}, |
| {"HP Mix", "MIC12HP Playback Switch", "MIC1 PGA"}, |
| {"HP Mix", "MIC22HP Playback Switch", "MIC2 PGA"}, |
| {"HP Mix", "DAC2HP Playback Switch", "I2S Mix"}, |
| |
| /* speaker mixer */ |
| {"Speaker Mix", "LI2SPK Playback Switch", "Line Mix"}, |
| {"Speaker Mix", "AUXI2SPK Playback Switch", "AuxI Mix"}, |
| {"Speaker Mix", "MIC12SPK Playback Switch", "MIC1 PGA"}, |
| {"Speaker Mix", "MIC22SPK Playback Switch", "MIC2 PGA"}, |
| {"Speaker Mix", "DAC2SPK Playback Switch", "I2S Mix"}, |
| |
| /* mono mixer */ |
| {"Mono Mix", "ADC2MONO_L Playback Switch", "Left Capture Mix"}, |
| {"Mono Mix", "ADC2MONO_R Playback Switch", "Right Capture Mix"}, |
| {"Mono Mix", "LI2MONO Playback Switch", "Line Mix"}, |
| {"Mono Mix", "AUXI2MONO Playback Switch", "AuxI Mix"}, |
| {"Mono Mix", "MIC12MONO Playback Switch", "MIC1 PGA"}, |
| {"Mono Mix", "MIC22MONO Playback Switch", "MIC2 PGA"}, |
| {"Mono Mix", "DAC2MONO Playback Switch", "I2S Mix"}, |
| |
| /* Left record mixer */ |
| {"Left Capture Mix", "LineInL Capture Switch", "LINEINL"}, |
| {"Left Capture Mix", "Left AuxI Capture Switch", "AUXINL"}, |
| {"Left Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"}, |
| {"Left Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"}, |
| {"Left Capture Mix", "HPMixerL Capture Switch", "HPL Mix"}, |
| {"Left Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"}, |
| {"Left Capture Mix", "MonoMixer Capture Switch", "Mono Mix"}, |
| |
| /*Right record mixer */ |
| {"Right Capture Mix", "LineInR Capture Switch", "LINEINR"}, |
| {"Right Capture Mix", "Right AuxI Capture Switch", "AUXINR"}, |
| {"Right Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"}, |
| {"Right Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"}, |
| {"Right Capture Mix", "HPMixerR Capture Switch", "HPR Mix"}, |
| {"Right Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"}, |
| {"Right Capture Mix", "MonoMixer Capture Switch", "Mono Mix"}, |
| |
| /* headphone left mux */ |
| {"Left Headphone Mux", "HP Left Mix", "HPL Mix"}, |
| {"Left Headphone Mux", "Vmid", "Vmid"}, |
| |
| /* headphone right mux */ |
| {"Right Headphone Mux", "HP Right Mix", "HPR Mix"}, |
| {"Right Headphone Mux", "Vmid", "Vmid"}, |
| |
| /* speaker out mux */ |
| {"SpeakerOut Mux", "Vmid", "Vmid"}, |
| {"SpeakerOut Mux", "HPOut Mix", "HPOut Mix"}, |
| {"SpeakerOut Mux", "Speaker Mix", "Speaker Mix"}, |
| {"SpeakerOut Mux", "Mono Mix", "Mono Mix"}, |
| |
| /* Mono/Aux Out mux */ |
| {"AuxOut Mux", "Vmid", "Vmid"}, |
| {"AuxOut Mux", "HPOut Mix", "HPOut Mix"}, |
| {"AuxOut Mux", "Speaker Mix", "Speaker Mix"}, |
| {"AuxOut Mux", "Mono Mix", "Mono Mix"}, |
| |
| /* output pga */ |
| {"HPL", NULL, "Left Headphone"}, |
| {"Left Headphone", NULL, "Left Headphone Mux"}, |
| {"HPR", NULL, "Right Headphone"}, |
| {"Right Headphone", NULL, "Right Headphone Mux"}, |
| {"Left AuxOut", NULL, "AuxOut Mux"}, |
| {"Right AuxOut", NULL, "AuxOut Mux"}, |
| |
| /* input pga */ |
| {"Left LineIn", NULL, "LINEINL"}, |
| {"Right LineIn", NULL, "LINEINR"}, |
| {"Left AuxI", NULL, "AUXINL"}, |
| {"Right AuxI", NULL, "AUXINR"}, |
| {"MIC1 Pre Amp", NULL, "MIC1"}, |
| {"MIC2 Pre Amp", NULL, "MIC2"}, |
| {"MIC1 PGA", NULL, "MIC1 Pre Amp"}, |
| {"MIC2 PGA", NULL, "MIC2 Pre Amp"}, |
| |
| /* left ADC */ |
| {"Left ADC", NULL, "Left Capture Mix"}, |
| |
| /* right ADC */ |
| {"Right ADC", NULL, "Right Capture Mix"}, |
| |
| {"SpeakerOut N Mux", "RN/-R", "SpeakerOut"}, |
| {"SpeakerOut N Mux", "RP/+R", "SpeakerOut"}, |
| {"SpeakerOut N Mux", "LN/-R", "SpeakerOut"}, |
| {"SpeakerOut N Mux", "Vmid", "Vmid"}, |
| |
| {"SPKOUT", NULL, "SpeakerOut"}, |
| {"SPKOUTN", NULL, "SpeakerOut N Mux"}, |
| }; |
| |
| static const struct snd_soc_dapm_route intercon_spk[] = { |
| {"SpeakerOut", NULL, "SpeakerOut Mux"}, |
| }; |
| |
| static const struct snd_soc_dapm_route intercon_amp_spk[] = { |
| {"AB Amp", NULL, "SpeakerOut Mux"}, |
| {"D Amp", NULL, "SpeakerOut Mux"}, |
| {"AB-D Amp Mux", "AB Amp", "AB Amp"}, |
| {"AB-D Amp Mux", "D Amp", "D Amp"}, |
| {"SpeakerOut", NULL, "AB-D Amp Mux"}, |
| }; |
| |
| /* PLL divisors */ |
| struct _pll_div { |
| u32 pll_in; |
| u32 pll_out; |
| u16 regvalue; |
| }; |
| |
| /* Note : pll code from original alc5623 driver. Not sure of how good it is */ |
| /* 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_data = client; |
| alc5623->control_type = SND_SOC_I2C; |
| mutex_init(&alc5623->mutex); |
| |
| ret = snd_soc_register_codec(&client->dev, |
| &soc_codec_device_alc5623, &alc5623_dai, 1); |
| if (ret != 0) { |
| dev_err(&client->dev, "Failed to register codec: %d\n", ret); |
| 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"); |
