1 /* 2 * alc5623.c -- alc562[123] ALSA Soc Audio driver 3 * 4 * Copyright 2008 Realtek Microelectronics 5 * Author: flove <flove@realtek.com> Ethan <eku@marvell.com> 6 * 7 * Copyright 2010 Arnaud Patard <arnaud.patard@rtp-net.org> 8 * 9 * 10 * Based on WM8753.c 11 * 12 * This program is free software; you can redistribute it and/or modify 13 * it under the terms of the GNU General Public License version 2 as 14 * published by the Free Software Foundation. 15 * 16 */ 17 18 #include <linux/module.h> 19 #include <linux/kernel.h> 20 #include <linux/init.h> 21 #include <linux/delay.h> 22 #include <linux/pm.h> 23 #include <linux/i2c.h> 24 #include <linux/slab.h> 25 #include <sound/core.h> 26 #include <sound/pcm.h> 27 #include <sound/pcm_params.h> 28 #include <sound/tlv.h> 29 #include <sound/soc.h> 30 #include <sound/initval.h> 31 #include <sound/alc5623.h> 32 33 #include "alc5623.h" 34 35 static int caps_charge = 2000; 36 module_param(caps_charge, int, 0); 37 MODULE_PARM_DESC(caps_charge, "ALC5623 cap charge time (msecs)"); 38 39 /* codec private data */ 40 struct alc5623_priv { 41 enum snd_soc_control_type control_type; 42 u8 id; 43 unsigned int sysclk; 44 u16 reg_cache[ALC5623_VENDOR_ID2+2]; 45 unsigned int add_ctrl; 46 unsigned int jack_det_ctrl; 47 }; 48 49 static void alc5623_fill_cache(struct snd_soc_codec *codec) 50 { 51 int i, step = codec->driver->reg_cache_step; 52 u16 *cache = codec->reg_cache; 53 54 /* not really efficient ... */ 55 codec->cache_bypass = 1; 56 for (i = 0 ; i < codec->driver->reg_cache_size ; i += step) 57 cache[i] = snd_soc_read(codec, i); 58 codec->cache_bypass = 0; 59 } 60 61 static inline int alc5623_reset(struct snd_soc_codec *codec) 62 { 63 return snd_soc_write(codec, ALC5623_RESET, 0); 64 } 65 66 static int amp_mixer_event(struct snd_soc_dapm_widget *w, 67 struct snd_kcontrol *kcontrol, int event) 68 { 69 /* to power-on/off class-d amp generators/speaker */ 70 /* need to write to 'index-46h' register : */ 71 /* so write index num (here 0x46) to reg 0x6a */ 72 /* and then 0xffff/0 to reg 0x6c */ 73 snd_soc_write(w->codec, ALC5623_HID_CTRL_INDEX, 0x46); 74 75 switch (event) { 76 case SND_SOC_DAPM_PRE_PMU: 77 snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0xFFFF); 78 break; 79 case SND_SOC_DAPM_POST_PMD: 80 snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0); 81 break; 82 } 83 84 return 0; 85 } 86 87 /* 88 * ALC5623 Controls 89 */ 90 91 static const DECLARE_TLV_DB_SCALE(vol_tlv, -3450, 150, 0); 92 static const DECLARE_TLV_DB_SCALE(hp_tlv, -4650, 150, 0); 93 static const DECLARE_TLV_DB_SCALE(adc_rec_tlv, -1650, 150, 0); 94 static const unsigned int boost_tlv[] = { 95 TLV_DB_RANGE_HEAD(3), 96 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), 97 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), 98 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), 99 }; 100 static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0); 101 102 static const struct snd_kcontrol_new alc5621_vol_snd_controls[] = { 103 SOC_DOUBLE_TLV("Speaker Playback Volume", 104 ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), 105 SOC_DOUBLE("Speaker Playback Switch", 106 ALC5623_SPK_OUT_VOL, 15, 7, 1, 1), 107 SOC_DOUBLE_TLV("Headphone Playback Volume", 108 ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), 109 SOC_DOUBLE("Headphone Playback Switch", 110 ALC5623_HP_OUT_VOL, 15, 7, 1, 1), 111 }; 112 113 static const struct snd_kcontrol_new alc5622_vol_snd_controls[] = { 114 SOC_DOUBLE_TLV("Speaker Playback Volume", 115 ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), 116 SOC_DOUBLE("Speaker Playback Switch", 117 ALC5623_SPK_OUT_VOL, 15, 7, 1, 1), 118 SOC_DOUBLE_TLV("Line Playback Volume", 119 ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), 120 SOC_DOUBLE("Line Playback Switch", 121 ALC5623_HP_OUT_VOL, 15, 7, 1, 1), 122 }; 123 124 static const struct snd_kcontrol_new alc5623_vol_snd_controls[] = { 125 SOC_DOUBLE_TLV("Line Playback Volume", 126 ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), 127 SOC_DOUBLE("Line Playback Switch", 128 ALC5623_SPK_OUT_VOL, 15, 7, 1, 1), 129 SOC_DOUBLE_TLV("Headphone Playback Volume", 130 ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), 131 SOC_DOUBLE("Headphone Playback Switch", 132 ALC5623_HP_OUT_VOL, 15, 7, 1, 1), 133 }; 134 135 static const struct snd_kcontrol_new alc5623_snd_controls[] = { 136 SOC_DOUBLE_TLV("Auxout Playback Volume", 137 ALC5623_MONO_AUX_OUT_VOL, 8, 0, 31, 1, hp_tlv), 138 SOC_DOUBLE("Auxout Playback Switch", 139 ALC5623_MONO_AUX_OUT_VOL, 15, 7, 1, 1), 140 SOC_DOUBLE_TLV("PCM Playback Volume", 141 ALC5623_STEREO_DAC_VOL, 8, 0, 31, 1, vol_tlv), 142 SOC_DOUBLE_TLV("AuxI Capture Volume", 143 ALC5623_AUXIN_VOL, 8, 0, 31, 1, vol_tlv), 144 SOC_DOUBLE_TLV("LineIn Capture Volume", 145 ALC5623_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv), 146 SOC_SINGLE_TLV("Mic1 Capture Volume", 147 ALC5623_MIC_VOL, 8, 31, 1, vol_tlv), 148 SOC_SINGLE_TLV("Mic2 Capture Volume", 149 ALC5623_MIC_VOL, 0, 31, 1, vol_tlv), 150 SOC_DOUBLE_TLV("Rec Capture Volume", 151 ALC5623_ADC_REC_GAIN, 7, 0, 31, 0, adc_rec_tlv), 152 SOC_SINGLE_TLV("Mic 1 Boost Volume", 153 ALC5623_MIC_CTRL, 10, 2, 0, boost_tlv), 154 SOC_SINGLE_TLV("Mic 2 Boost Volume", 155 ALC5623_MIC_CTRL, 8, 2, 0, boost_tlv), 156 SOC_SINGLE_TLV("Digital Boost Volume", 157 ALC5623_ADD_CTRL_REG, 4, 3, 0, dig_tlv), 158 }; 159 160 /* 161 * DAPM Controls 162 */ 163 static const struct snd_kcontrol_new alc5623_hp_mixer_controls[] = { 164 SOC_DAPM_SINGLE("LI2HP Playback Switch", ALC5623_LINE_IN_VOL, 15, 1, 1), 165 SOC_DAPM_SINGLE("AUXI2HP Playback Switch", ALC5623_AUXIN_VOL, 15, 1, 1), 166 SOC_DAPM_SINGLE("MIC12HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 15, 1, 1), 167 SOC_DAPM_SINGLE("MIC22HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 7, 1, 1), 168 SOC_DAPM_SINGLE("DAC2HP Playback Switch", ALC5623_STEREO_DAC_VOL, 15, 1, 1), 169 }; 170 171 static const struct snd_kcontrol_new alc5623_hpl_mixer_controls[] = { 172 SOC_DAPM_SINGLE("ADC2HP_L Playback Switch", ALC5623_ADC_REC_GAIN, 15, 1, 1), 173 }; 174 175 static const struct snd_kcontrol_new alc5623_hpr_mixer_controls[] = { 176 SOC_DAPM_SINGLE("ADC2HP_R Playback Switch", ALC5623_ADC_REC_GAIN, 14, 1, 1), 177 }; 178 179 static const struct snd_kcontrol_new alc5623_mono_mixer_controls[] = { 180 SOC_DAPM_SINGLE("ADC2MONO_L Playback Switch", ALC5623_ADC_REC_GAIN, 13, 1, 1), 181 SOC_DAPM_SINGLE("ADC2MONO_R Playback Switch", ALC5623_ADC_REC_GAIN, 12, 1, 1), 182 SOC_DAPM_SINGLE("LI2MONO Playback Switch", ALC5623_LINE_IN_VOL, 13, 1, 1), 183 SOC_DAPM_SINGLE("AUXI2MONO Playback Switch", ALC5623_AUXIN_VOL, 13, 1, 1), 184 SOC_DAPM_SINGLE("MIC12MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 13, 1, 1), 185 SOC_DAPM_SINGLE("MIC22MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 5, 1, 1), 186 SOC_DAPM_SINGLE("DAC2MONO Playback Switch", ALC5623_STEREO_DAC_VOL, 13, 1, 1), 187 }; 188 189 static const struct snd_kcontrol_new alc5623_speaker_mixer_controls[] = { 190 SOC_DAPM_SINGLE("LI2SPK Playback Switch", ALC5623_LINE_IN_VOL, 14, 1, 1), 191 SOC_DAPM_SINGLE("AUXI2SPK Playback Switch", ALC5623_AUXIN_VOL, 14, 1, 1), 192 SOC_DAPM_SINGLE("MIC12SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 14, 1, 1), 193 SOC_DAPM_SINGLE("MIC22SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 6, 1, 1), 194 SOC_DAPM_SINGLE("DAC2SPK Playback Switch", ALC5623_STEREO_DAC_VOL, 14, 1, 1), 195 }; 196 197 /* Left Record Mixer */ 198 static const struct snd_kcontrol_new alc5623_captureL_mixer_controls[] = { 199 SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 14, 1, 1), 200 SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 13, 1, 1), 201 SOC_DAPM_SINGLE("LineInL Capture Switch", ALC5623_ADC_REC_MIXER, 12, 1, 1), 202 SOC_DAPM_SINGLE("Left AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 11, 1, 1), 203 SOC_DAPM_SINGLE("HPMixerL Capture Switch", ALC5623_ADC_REC_MIXER, 10, 1, 1), 204 SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 9, 1, 1), 205 SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 8, 1, 1), 206 }; 207 208 /* Right Record Mixer */ 209 static const struct snd_kcontrol_new alc5623_captureR_mixer_controls[] = { 210 SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 6, 1, 1), 211 SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 5, 1, 1), 212 SOC_DAPM_SINGLE("LineInR Capture Switch", ALC5623_ADC_REC_MIXER, 4, 1, 1), 213 SOC_DAPM_SINGLE("Right AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 3, 1, 1), 214 SOC_DAPM_SINGLE("HPMixerR Capture Switch", ALC5623_ADC_REC_MIXER, 2, 1, 1), 215 SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 1, 1, 1), 216 SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 0, 1, 1), 217 }; 218 219 static const char *alc5623_spk_n_sour_sel[] = { 220 "RN/-R", "RP/+R", "LN/-R", "Vmid" }; 221 static const char *alc5623_hpl_out_input_sel[] = { 222 "Vmid", "HP Left Mix"}; 223 static const char *alc5623_hpr_out_input_sel[] = { 224 "Vmid", "HP Right Mix"}; 225 static const char *alc5623_spkout_input_sel[] = { 226 "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; 227 static const char *alc5623_aux_out_input_sel[] = { 228 "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; 229 230 /* auxout output mux */ 231 static const struct soc_enum alc5623_aux_out_input_enum = 232 SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 6, 4, alc5623_aux_out_input_sel); 233 static const struct snd_kcontrol_new alc5623_auxout_mux_controls = 234 SOC_DAPM_ENUM("Route", alc5623_aux_out_input_enum); 235 236 /* speaker output mux */ 237 static const struct soc_enum alc5623_spkout_input_enum = 238 SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 10, 4, alc5623_spkout_input_sel); 239 static const struct snd_kcontrol_new alc5623_spkout_mux_controls = 240 SOC_DAPM_ENUM("Route", alc5623_spkout_input_enum); 241 242 /* headphone left output mux */ 243 static const struct soc_enum alc5623_hpl_out_input_enum = 244 SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 9, 2, alc5623_hpl_out_input_sel); 245 static const struct snd_kcontrol_new alc5623_hpl_out_mux_controls = 246 SOC_DAPM_ENUM("Route", alc5623_hpl_out_input_enum); 247 248 /* headphone right output mux */ 249 static const struct soc_enum alc5623_hpr_out_input_enum = 250 SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 8, 2, alc5623_hpr_out_input_sel); 251 static const struct snd_kcontrol_new alc5623_hpr_out_mux_controls = 252 SOC_DAPM_ENUM("Route", alc5623_hpr_out_input_enum); 253 254 /* speaker output N select */ 255 static const struct soc_enum alc5623_spk_n_sour_enum = 256 SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 14, 4, alc5623_spk_n_sour_sel); 257 static const struct snd_kcontrol_new alc5623_spkoutn_mux_controls = 258 SOC_DAPM_ENUM("Route", alc5623_spk_n_sour_enum); 259 260 static const struct snd_soc_dapm_widget alc5623_dapm_widgets[] = { 261 /* Muxes */ 262 SND_SOC_DAPM_MUX("AuxOut Mux", SND_SOC_NOPM, 0, 0, 263 &alc5623_auxout_mux_controls), 264 SND_SOC_DAPM_MUX("SpeakerOut Mux", SND_SOC_NOPM, 0, 0, 265 &alc5623_spkout_mux_controls), 266 SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, 267 &alc5623_hpl_out_mux_controls), 268 SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, 269 &alc5623_hpr_out_mux_controls), 270 SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0, 271 &alc5623_spkoutn_mux_controls), 272 273 /* output mixers */ 274 SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0, 275 &alc5623_hp_mixer_controls[0], 276 ARRAY_SIZE(alc5623_hp_mixer_controls)), 277 SND_SOC_DAPM_MIXER("HPR Mix", ALC5623_PWR_MANAG_ADD2, 4, 0, 278 &alc5623_hpr_mixer_controls[0], 279 ARRAY_SIZE(alc5623_hpr_mixer_controls)), 280 SND_SOC_DAPM_MIXER("HPL Mix", ALC5623_PWR_MANAG_ADD2, 5, 0, 281 &alc5623_hpl_mixer_controls[0], 282 ARRAY_SIZE(alc5623_hpl_mixer_controls)), 283 SND_SOC_DAPM_MIXER("HPOut Mix", SND_SOC_NOPM, 0, 0, NULL, 0), 284 SND_SOC_DAPM_MIXER("Mono Mix", ALC5623_PWR_MANAG_ADD2, 2, 0, 285 &alc5623_mono_mixer_controls[0], 286 ARRAY_SIZE(alc5623_mono_mixer_controls)), 287 SND_SOC_DAPM_MIXER("Speaker Mix", ALC5623_PWR_MANAG_ADD2, 3, 0, 288 &alc5623_speaker_mixer_controls[0], 289 ARRAY_SIZE(alc5623_speaker_mixer_controls)), 290 291 /* input mixers */ 292 SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5623_PWR_MANAG_ADD2, 1, 0, 293 &alc5623_captureL_mixer_controls[0], 294 ARRAY_SIZE(alc5623_captureL_mixer_controls)), 295 SND_SOC_DAPM_MIXER("Right Capture Mix", ALC5623_PWR_MANAG_ADD2, 0, 0, 296 &alc5623_captureR_mixer_controls[0], 297 ARRAY_SIZE(alc5623_captureR_mixer_controls)), 298 299 SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", 300 ALC5623_PWR_MANAG_ADD2, 9, 0), 301 SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", 302 ALC5623_PWR_MANAG_ADD2, 8, 0), 303 SND_SOC_DAPM_MIXER("I2S Mix", ALC5623_PWR_MANAG_ADD1, 15, 0, NULL, 0), 304 SND_SOC_DAPM_MIXER("AuxI Mix", SND_SOC_NOPM, 0, 0, NULL, 0), 305 SND_SOC_DAPM_MIXER("Line Mix", SND_SOC_NOPM, 0, 0, NULL, 0), 306 SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", 307 ALC5623_PWR_MANAG_ADD2, 7, 0), 308 SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", 309 ALC5623_PWR_MANAG_ADD2, 6, 0), 310 SND_SOC_DAPM_PGA("Left Headphone", ALC5623_PWR_MANAG_ADD3, 10, 0, NULL, 0), 311 SND_SOC_DAPM_PGA("Right Headphone", ALC5623_PWR_MANAG_ADD3, 9, 0, NULL, 0), 312 SND_SOC_DAPM_PGA("SpeakerOut", ALC5623_PWR_MANAG_ADD3, 12, 0, NULL, 0), 313 SND_SOC_DAPM_PGA("Left AuxOut", ALC5623_PWR_MANAG_ADD3, 14, 0, NULL, 0), 314 SND_SOC_DAPM_PGA("Right AuxOut", ALC5623_PWR_MANAG_ADD3, 13, 0, NULL, 0), 315 SND_SOC_DAPM_PGA("Left LineIn", ALC5623_PWR_MANAG_ADD3, 7, 0, NULL, 0), 316 SND_SOC_DAPM_PGA("Right LineIn", ALC5623_PWR_MANAG_ADD3, 6, 0, NULL, 0), 317 SND_SOC_DAPM_PGA("Left AuxI", ALC5623_PWR_MANAG_ADD3, 5, 0, NULL, 0), 318 SND_SOC_DAPM_PGA("Right AuxI", ALC5623_PWR_MANAG_ADD3, 4, 0, NULL, 0), 319 SND_SOC_DAPM_PGA("MIC1 PGA", ALC5623_PWR_MANAG_ADD3, 3, 0, NULL, 0), 320 SND_SOC_DAPM_PGA("MIC2 PGA", ALC5623_PWR_MANAG_ADD3, 2, 0, NULL, 0), 321 SND_SOC_DAPM_PGA("MIC1 Pre Amp", ALC5623_PWR_MANAG_ADD3, 1, 0, NULL, 0), 322 SND_SOC_DAPM_PGA("MIC2 Pre Amp", ALC5623_PWR_MANAG_ADD3, 0, 0, NULL, 0), 323 SND_SOC_DAPM_MICBIAS("Mic Bias1", ALC5623_PWR_MANAG_ADD1, 11, 0), 324 325 SND_SOC_DAPM_OUTPUT("AUXOUTL"), 326 SND_SOC_DAPM_OUTPUT("AUXOUTR"), 327 SND_SOC_DAPM_OUTPUT("HPL"), 328 SND_SOC_DAPM_OUTPUT("HPR"), 329 SND_SOC_DAPM_OUTPUT("SPKOUT"), 330 SND_SOC_DAPM_OUTPUT("SPKOUTN"), 331 SND_SOC_DAPM_INPUT("LINEINL"), 332 SND_SOC_DAPM_INPUT("LINEINR"), 333 SND_SOC_DAPM_INPUT("AUXINL"), 334 SND_SOC_DAPM_INPUT("AUXINR"), 335 SND_SOC_DAPM_INPUT("MIC1"), 336 SND_SOC_DAPM_INPUT("MIC2"), 337 SND_SOC_DAPM_VMID("Vmid"), 338 }; 339 340 static const char *alc5623_amp_names[] = {"AB Amp", "D Amp"}; 341 static const struct soc_enum alc5623_amp_enum = 342 SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 13, 2, alc5623_amp_names); 343 static const struct snd_kcontrol_new alc5623_amp_mux_controls = 344 SOC_DAPM_ENUM("Route", alc5623_amp_enum); 345 346 static const struct snd_soc_dapm_widget alc5623_dapm_amp_widgets[] = { 347 SND_SOC_DAPM_PGA_E("D Amp", ALC5623_PWR_MANAG_ADD2, 14, 0, NULL, 0, 348 amp_mixer_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), 349 SND_SOC_DAPM_PGA("AB Amp", ALC5623_PWR_MANAG_ADD2, 15, 0, NULL, 0), 350 SND_SOC_DAPM_MUX("AB-D Amp Mux", SND_SOC_NOPM, 0, 0, 351 &alc5623_amp_mux_controls), 352 }; 353 354 static const struct snd_soc_dapm_route intercon[] = { 355 /* virtual mixer - mixes left & right channels */ 356 {"I2S Mix", NULL, "Left DAC"}, 357 {"I2S Mix", NULL, "Right DAC"}, 358 {"Line Mix", NULL, "Right LineIn"}, 359 {"Line Mix", NULL, "Left LineIn"}, 360 {"AuxI Mix", NULL, "Left AuxI"}, 361 {"AuxI Mix", NULL, "Right AuxI"}, 362 {"AUXOUTL", NULL, "Left AuxOut"}, 363 {"AUXOUTR", NULL, "Right AuxOut"}, 364 365 /* HP mixer */ 366 {"HPL Mix", "ADC2HP_L Playback Switch", "Left Capture Mix"}, 367 {"HPL Mix", NULL, "HP Mix"}, 368 {"HPR Mix", "ADC2HP_R Playback Switch", "Right Capture Mix"}, 369 {"HPR Mix", NULL, "HP Mix"}, 370 {"HP Mix", "LI2HP Playback Switch", "Line Mix"}, 371 {"HP Mix", "AUXI2HP Playback Switch", "AuxI Mix"}, 372 {"HP Mix", "MIC12HP Playback Switch", "MIC1 PGA"}, 373 {"HP Mix", "MIC22HP Playback Switch", "MIC2 PGA"}, 374 {"HP Mix", "DAC2HP Playback Switch", "I2S Mix"}, 375 376 /* speaker mixer */ 377 {"Speaker Mix", "LI2SPK Playback Switch", "Line Mix"}, 378 {"Speaker Mix", "AUXI2SPK Playback Switch", "AuxI Mix"}, 379 {"Speaker Mix", "MIC12SPK Playback Switch", "MIC1 PGA"}, 380 {"Speaker Mix", "MIC22SPK Playback Switch", "MIC2 PGA"}, 381 {"Speaker Mix", "DAC2SPK Playback Switch", "I2S Mix"}, 382 383 /* mono mixer */ 384 {"Mono Mix", "ADC2MONO_L Playback Switch", "Left Capture Mix"}, 385 {"Mono Mix", "ADC2MONO_R Playback Switch", "Right Capture Mix"}, 386 {"Mono Mix", "LI2MONO Playback Switch", "Line Mix"}, 387 {"Mono Mix", "AUXI2MONO Playback Switch", "AuxI Mix"}, 388 {"Mono Mix", "MIC12MONO Playback Switch", "MIC1 PGA"}, 389 {"Mono Mix", "MIC22MONO Playback Switch", "MIC2 PGA"}, 390 {"Mono Mix", "DAC2MONO Playback Switch", "I2S Mix"}, 391 392 /* Left record mixer */ 393 {"Left Capture Mix", "LineInL Capture Switch", "LINEINL"}, 394 {"Left Capture Mix", "Left AuxI Capture Switch", "AUXINL"}, 395 {"Left Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"}, 396 {"Left Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"}, 397 {"Left Capture Mix", "HPMixerL Capture Switch", "HPL Mix"}, 398 {"Left Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"}, 399 {"Left Capture Mix", "MonoMixer Capture Switch", "Mono Mix"}, 400 401 /*Right record mixer */ 402 {"Right Capture Mix", "LineInR Capture Switch", "LINEINR"}, 403 {"Right Capture Mix", "Right AuxI Capture Switch", "AUXINR"}, 404 {"Right Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"}, 405 {"Right Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"}, 406 {"Right Capture Mix", "HPMixerR Capture Switch", "HPR Mix"}, 407 {"Right Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"}, 408 {"Right Capture Mix", "MonoMixer Capture Switch", "Mono Mix"}, 409 410 /* headphone left mux */ 411 {"Left Headphone Mux", "HP Left Mix", "HPL Mix"}, 412 {"Left Headphone Mux", "Vmid", "Vmid"}, 413 414 /* headphone right mux */ 415 {"Right Headphone Mux", "HP Right Mix", "HPR Mix"}, 416 {"Right Headphone Mux", "Vmid", "Vmid"}, 417 418 /* speaker out mux */ 419 {"SpeakerOut Mux", "Vmid", "Vmid"}, 420 {"SpeakerOut Mux", "HPOut Mix", "HPOut Mix"}, 421 {"SpeakerOut Mux", "Speaker Mix", "Speaker Mix"}, 422 {"SpeakerOut Mux", "Mono Mix", "Mono Mix"}, 423 424 /* Mono/Aux Out mux */ 425 {"AuxOut Mux", "Vmid", "Vmid"}, 426 {"AuxOut Mux", "HPOut Mix", "HPOut Mix"}, 427 {"AuxOut Mux", "Speaker Mix", "Speaker Mix"}, 428 {"AuxOut Mux", "Mono Mix", "Mono Mix"}, 429 430 /* output pga */ 431 {"HPL", NULL, "Left Headphone"}, 432 {"Left Headphone", NULL, "Left Headphone Mux"}, 433 {"HPR", NULL, "Right Headphone"}, 434 {"Right Headphone", NULL, "Right Headphone Mux"}, 435 {"Left AuxOut", NULL, "AuxOut Mux"}, 436 {"Right AuxOut", NULL, "AuxOut Mux"}, 437 438 /* input pga */ 439 {"Left LineIn", NULL, "LINEINL"}, 440 {"Right LineIn", NULL, "LINEINR"}, 441 {"Left AuxI", NULL, "AUXINL"}, 442 {"Right AuxI", NULL, "AUXINR"}, 443 {"MIC1 Pre Amp", NULL, "MIC1"}, 444 {"MIC2 Pre Amp", NULL, "MIC2"}, 445 {"MIC1 PGA", NULL, "MIC1 Pre Amp"}, 446 {"MIC2 PGA", NULL, "MIC2 Pre Amp"}, 447 448 /* left ADC */ 449 {"Left ADC", NULL, "Left Capture Mix"}, 450 451 /* right ADC */ 452 {"Right ADC", NULL, "Right Capture Mix"}, 453 454 {"SpeakerOut N Mux", "RN/-R", "SpeakerOut"}, 455 {"SpeakerOut N Mux", "RP/+R", "SpeakerOut"}, 456 {"SpeakerOut N Mux", "LN/-R", "SpeakerOut"}, 457 {"SpeakerOut N Mux", "Vmid", "Vmid"}, 458 459 {"SPKOUT", NULL, "SpeakerOut"}, 460 {"SPKOUTN", NULL, "SpeakerOut N Mux"}, 461 }; 462 463 static const struct snd_soc_dapm_route intercon_spk[] = { 464 {"SpeakerOut", NULL, "SpeakerOut Mux"}, 465 }; 466 467 static const struct snd_soc_dapm_route intercon_amp_spk[] = { 468 {"AB Amp", NULL, "SpeakerOut Mux"}, 469 {"D Amp", NULL, "SpeakerOut Mux"}, 470 {"AB-D Amp Mux", "AB Amp", "AB Amp"}, 471 {"AB-D Amp Mux", "D Amp", "D Amp"}, 472 {"SpeakerOut", NULL, "AB-D Amp Mux"}, 473 }; 474 475 /* PLL divisors */ 476 struct _pll_div { 477 u32 pll_in; 478 u32 pll_out; 479 u16 regvalue; 480 }; 481 482 /* Note : pll code from original alc5623 driver. Not sure of how good it is */ 483 /* useful only for master mode */ 484 static const struct _pll_div codec_master_pll_div[] = { 485 486 { 2048000, 8192000, 0x0ea0}, 487 { 3686400, 8192000, 0x4e27}, 488 { 12000000, 8192000, 0x456b}, 489 { 13000000, 8192000, 0x495f}, 490 { 13100000, 8192000, 0x0320}, 491 { 2048000, 11289600, 0xf637}, 492 { 3686400, 11289600, 0x2f22}, 493 { 12000000, 11289600, 0x3e2f}, 494 { 13000000, 11289600, 0x4d5b}, 495 { 13100000, 11289600, 0x363b}, 496 { 2048000, 16384000, 0x1ea0}, 497 { 3686400, 16384000, 0x9e27}, 498 { 12000000, 16384000, 0x452b}, 499 { 13000000, 16384000, 0x542f}, 500 { 13100000, 16384000, 0x03a0}, 501 { 2048000, 16934400, 0xe625}, 502 { 3686400, 16934400, 0x9126}, 503 { 12000000, 16934400, 0x4d2c}, 504 { 13000000, 16934400, 0x742f}, 505 { 13100000, 16934400, 0x3c27}, 506 { 2048000, 22579200, 0x2aa0}, 507 { 3686400, 22579200, 0x2f20}, 508 { 12000000, 22579200, 0x7e2f}, 509 { 13000000, 22579200, 0x742f}, 510 { 13100000, 22579200, 0x3c27}, 511 { 2048000, 24576000, 0x2ea0}, 512 { 3686400, 24576000, 0xee27}, 513 { 12000000, 24576000, 0x2915}, 514 { 13000000, 24576000, 0x772e}, 515 { 13100000, 24576000, 0x0d20}, 516 }; 517 518 static const struct _pll_div codec_slave_pll_div[] = { 519 520 { 1024000, 16384000, 0x3ea0}, 521 { 1411200, 22579200, 0x3ea0}, 522 { 1536000, 24576000, 0x3ea0}, 523 { 2048000, 16384000, 0x1ea0}, 524 { 2822400, 22579200, 0x1ea0}, 525 { 3072000, 24576000, 0x1ea0}, 526 527 }; 528 529 static int alc5623_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, 530 int source, unsigned int freq_in, unsigned int freq_out) 531 { 532 int i; 533 struct snd_soc_codec *codec = codec_dai->codec; 534 int gbl_clk = 0, pll_div = 0; 535 u16 reg; 536 537 if (pll_id < ALC5623_PLL_FR_MCLK || pll_id > ALC5623_PLL_FR_BCK) 538 return -ENODEV; 539 540 /* Disable PLL power */ 541 snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2, 542 ALC5623_PWR_ADD2_PLL, 543 0); 544 545 /* pll is not used in slave mode */ 546 reg = snd_soc_read(codec, ALC5623_DAI_CONTROL); 547 if (reg & ALC5623_DAI_SDP_SLAVE_MODE) 548 return 0; 549 550 if (!freq_in || !freq_out) 551 return 0; 552 553 switch (pll_id) { 554 case ALC5623_PLL_FR_MCLK: 555 for (i = 0; i < ARRAY_SIZE(codec_master_pll_div); i++) { 556 if (codec_master_pll_div[i].pll_in == freq_in 557 && codec_master_pll_div[i].pll_out == freq_out) { 558 /* PLL source from MCLK */ 559 pll_div = codec_master_pll_div[i].regvalue; 560 break; 561 } 562 } 563 break; 564 case ALC5623_PLL_FR_BCK: 565 for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) { 566 if (codec_slave_pll_div[i].pll_in == freq_in 567 && codec_slave_pll_div[i].pll_out == freq_out) { 568 /* PLL source from Bitclk */ 569 gbl_clk = ALC5623_GBL_CLK_PLL_SOUR_SEL_BITCLK; 570 pll_div = codec_slave_pll_div[i].regvalue; 571 break; 572 } 573 } 574 break; 575 default: 576 return -EINVAL; 577 } 578 579 if (!pll_div) 580 return -EINVAL; 581 582 snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk); 583 snd_soc_write(codec, ALC5623_PLL_CTRL, pll_div); 584 snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2, 585 ALC5623_PWR_ADD2_PLL, 586 ALC5623_PWR_ADD2_PLL); 587 gbl_clk |= ALC5623_GBL_CLK_SYS_SOUR_SEL_PLL; 588 snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk); 589 590 return 0; 591 } 592 593 struct _coeff_div { 594 u16 fs; 595 u16 regvalue; 596 }; 597 598 /* codec hifi mclk (after PLL) clock divider coefficients */ 599 /* values inspired from column BCLK=32Fs of Appendix A table */ 600 static const struct _coeff_div coeff_div[] = { 601 {256*8, 0x3a69}, 602 {384*8, 0x3c6b}, 603 {256*4, 0x2a69}, 604 {384*4, 0x2c6b}, 605 {256*2, 0x1a69}, 606 {384*2, 0x1c6b}, 607 {256*1, 0x0a69}, 608 {384*1, 0x0c6b}, 609 }; 610 611 static int get_coeff(struct snd_soc_codec *codec, int rate) 612 { 613 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); 614 int i; 615 616 for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { 617 if (coeff_div[i].fs * rate == alc5623->sysclk) 618 return i; 619 } 620 return -EINVAL; 621 } 622 623 /* 624 * Clock after PLL and dividers 625 */ 626 static int alc5623_set_dai_sysclk(struct snd_soc_dai *codec_dai, 627 int clk_id, unsigned int freq, int dir) 628 { 629 struct snd_soc_codec *codec = codec_dai->codec; 630 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); 631 632 switch (freq) { 633 case 8192000: 634 case 11289600: 635 case 12288000: 636 case 16384000: 637 case 16934400: 638 case 18432000: 639 case 22579200: 640 case 24576000: 641 alc5623->sysclk = freq; 642 return 0; 643 } 644 return -EINVAL; 645 } 646 647 static int alc5623_set_dai_fmt(struct snd_soc_dai *codec_dai, 648 unsigned int fmt) 649 { 650 struct snd_soc_codec *codec = codec_dai->codec; 651 u16 iface = 0; 652 653 /* set master/slave audio interface */ 654 switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { 655 case SND_SOC_DAIFMT_CBM_CFM: 656 iface = ALC5623_DAI_SDP_MASTER_MODE; 657 break; 658 case SND_SOC_DAIFMT_CBS_CFS: 659 iface = ALC5623_DAI_SDP_SLAVE_MODE; 660 break; 661 default: 662 return -EINVAL; 663 } 664 665 /* interface format */ 666 switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { 667 case SND_SOC_DAIFMT_I2S: 668 iface |= ALC5623_DAI_I2S_DF_I2S; 669 break; 670 case SND_SOC_DAIFMT_RIGHT_J: 671 iface |= ALC5623_DAI_I2S_DF_RIGHT; 672 break; 673 case SND_SOC_DAIFMT_LEFT_J: 674 iface |= ALC5623_DAI_I2S_DF_LEFT; 675 break; 676 case SND_SOC_DAIFMT_DSP_A: 677 iface |= ALC5623_DAI_I2S_DF_PCM; 678 break; 679 case SND_SOC_DAIFMT_DSP_B: 680 iface |= ALC5623_DAI_I2S_DF_PCM | ALC5623_DAI_I2S_PCM_MODE; 681 break; 682 default: 683 return -EINVAL; 684 } 685 686 /* clock inversion */ 687 switch (fmt & SND_SOC_DAIFMT_INV_MASK) { 688 case SND_SOC_DAIFMT_NB_NF: 689 break; 690 case SND_SOC_DAIFMT_IB_IF: 691 iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL; 692 break; 693 case SND_SOC_DAIFMT_IB_NF: 694 iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL; 695 break; 696 case SND_SOC_DAIFMT_NB_IF: 697 break; 698 default: 699 return -EINVAL; 700 } 701 702 return snd_soc_write(codec, ALC5623_DAI_CONTROL, iface); 703 } 704 705 static int alc5623_pcm_hw_params(struct snd_pcm_substream *substream, 706 struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) 707 { 708 struct snd_soc_codec *codec = dai->codec; 709 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); 710 int coeff, rate; 711 u16 iface; 712 713 iface = snd_soc_read(codec, ALC5623_DAI_CONTROL); 714 iface &= ~ALC5623_DAI_I2S_DL_MASK; 715 716 /* bit size */ 717 switch (params_format(params)) { 718 case SNDRV_PCM_FORMAT_S16_LE: 719 iface |= ALC5623_DAI_I2S_DL_16; 720 break; 721 case SNDRV_PCM_FORMAT_S20_3LE: 722 iface |= ALC5623_DAI_I2S_DL_20; 723 break; 724 case SNDRV_PCM_FORMAT_S24_LE: 725 iface |= ALC5623_DAI_I2S_DL_24; 726 break; 727 case SNDRV_PCM_FORMAT_S32_LE: 728 iface |= ALC5623_DAI_I2S_DL_32; 729 break; 730 default: 731 return -EINVAL; 732 } 733 734 /* set iface & srate */ 735 snd_soc_write(codec, ALC5623_DAI_CONTROL, iface); 736 rate = params_rate(params); 737 coeff = get_coeff(codec, rate); 738 if (coeff < 0) 739 return -EINVAL; 740 741 coeff = coeff_div[coeff].regvalue; 742 dev_dbg(codec->dev, "%s: sysclk=%d,rate=%d,coeff=0x%04x\n", 743 __func__, alc5623->sysclk, rate, coeff); 744 snd_soc_write(codec, ALC5623_STEREO_AD_DA_CLK_CTRL, coeff); 745 746 return 0; 747 } 748 749 static int alc5623_mute(struct snd_soc_dai *dai, int mute) 750 { 751 struct snd_soc_codec *codec = dai->codec; 752 u16 hp_mute = ALC5623_MISC_M_DAC_L_INPUT | ALC5623_MISC_M_DAC_R_INPUT; 753 u16 mute_reg = snd_soc_read(codec, ALC5623_MISC_CTRL) & ~hp_mute; 754 755 if (mute) 756 mute_reg |= hp_mute; 757 758 return snd_soc_write(codec, ALC5623_MISC_CTRL, mute_reg); 759 } 760 761 #define ALC5623_ADD2_POWER_EN (ALC5623_PWR_ADD2_VREF \ 762 | ALC5623_PWR_ADD2_DAC_REF_CIR) 763 764 #define ALC5623_ADD3_POWER_EN (ALC5623_PWR_ADD3_MAIN_BIAS \ 765 | ALC5623_PWR_ADD3_MIC1_BOOST_AD) 766 767 #define ALC5623_ADD1_POWER_EN \ 768 (ALC5623_PWR_ADD1_SHORT_CURR_DET_EN | ALC5623_PWR_ADD1_SOFTGEN_EN \ 769 | ALC5623_PWR_ADD1_DEPOP_BUF_HP | ALC5623_PWR_ADD1_HP_OUT_AMP \ 770 | ALC5623_PWR_ADD1_HP_OUT_ENH_AMP) 771 772 #define ALC5623_ADD1_POWER_EN_5622 \ 773 (ALC5623_PWR_ADD1_SHORT_CURR_DET_EN \ 774 | ALC5623_PWR_ADD1_HP_OUT_AMP) 775 776 static void enable_power_depop(struct snd_soc_codec *codec) 777 { 778 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); 779 780 snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD1, 781 ALC5623_PWR_ADD1_SOFTGEN_EN, 782 ALC5623_PWR_ADD1_SOFTGEN_EN); 783 784 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, ALC5623_ADD3_POWER_EN); 785 786 snd_soc_update_bits(codec, ALC5623_MISC_CTRL, 787 ALC5623_MISC_HP_DEPOP_MODE2_EN, 788 ALC5623_MISC_HP_DEPOP_MODE2_EN); 789 790 msleep(500); 791 792 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, ALC5623_ADD2_POWER_EN); 793 794 /* avoid writing '1' into 5622 reserved bits */ 795 if (alc5623->id == 0x22) 796 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, 797 ALC5623_ADD1_POWER_EN_5622); 798 else 799 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, 800 ALC5623_ADD1_POWER_EN); 801 802 /* disable HP Depop2 */ 803 snd_soc_update_bits(codec, ALC5623_MISC_CTRL, 804 ALC5623_MISC_HP_DEPOP_MODE2_EN, 805 0); 806 807 } 808 809 static int alc5623_set_bias_level(struct snd_soc_codec *codec, 810 enum snd_soc_bias_level level) 811 { 812 switch (level) { 813 case SND_SOC_BIAS_ON: 814 enable_power_depop(codec); 815 break; 816 case SND_SOC_BIAS_PREPARE: 817 break; 818 case SND_SOC_BIAS_STANDBY: 819 /* everything off except vref/vmid, */ 820 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, 821 ALC5623_PWR_ADD2_VREF); 822 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, 823 ALC5623_PWR_ADD3_MAIN_BIAS); 824 break; 825 case SND_SOC_BIAS_OFF: 826 /* everything off, dac mute, inactive */ 827 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, 0); 828 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, 0); 829 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, 0); 830 break; 831 } 832 codec->dapm.bias_level = level; 833 return 0; 834 } 835 836 #define ALC5623_FORMATS (SNDRV_PCM_FMTBIT_S16_LE \ 837 | SNDRV_PCM_FMTBIT_S24_LE \ 838 | SNDRV_PCM_FMTBIT_S32_LE) 839 840 static const struct snd_soc_dai_ops alc5623_dai_ops = { 841 .hw_params = alc5623_pcm_hw_params, 842 .digital_mute = alc5623_mute, 843 .set_fmt = alc5623_set_dai_fmt, 844 .set_sysclk = alc5623_set_dai_sysclk, 845 .set_pll = alc5623_set_dai_pll, 846 }; 847 848 static struct snd_soc_dai_driver alc5623_dai = { 849 .name = "alc5623-hifi", 850 .playback = { 851 .stream_name = "Playback", 852 .channels_min = 1, 853 .channels_max = 2, 854 .rate_min = 8000, 855 .rate_max = 48000, 856 .rates = SNDRV_PCM_RATE_8000_48000, 857 .formats = ALC5623_FORMATS,}, 858 .capture = { 859 .stream_name = "Capture", 860 .channels_min = 1, 861 .channels_max = 2, 862 .rate_min = 8000, 863 .rate_max = 48000, 864 .rates = SNDRV_PCM_RATE_8000_48000, 865 .formats = ALC5623_FORMATS,}, 866 867 .ops = &alc5623_dai_ops, 868 }; 869 870 static int alc5623_suspend(struct snd_soc_codec *codec) 871 { 872 alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF); 873 return 0; 874 } 875 876 static int alc5623_resume(struct snd_soc_codec *codec) 877 { 878 int i, step = codec->driver->reg_cache_step; 879 u16 *cache = codec->reg_cache; 880 881 /* Sync reg_cache with the hardware */ 882 for (i = 2 ; i < codec->driver->reg_cache_size ; i += step) 883 snd_soc_write(codec, i, cache[i]); 884 885 alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); 886 887 /* charge alc5623 caps */ 888 if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) { 889 alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); 890 codec->dapm.bias_level = SND_SOC_BIAS_ON; 891 alc5623_set_bias_level(codec, codec->dapm.bias_level); 892 } 893 894 return 0; 895 } 896 897 static int alc5623_probe(struct snd_soc_codec *codec) 898 { 899 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); 900 struct snd_soc_dapm_context *dapm = &codec->dapm; 901 int ret; 902 903 ret = snd_soc_codec_set_cache_io(codec, 8, 16, alc5623->control_type); 904 if (ret < 0) { 905 dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); 906 return ret; 907 } 908 909 alc5623_reset(codec); 910 alc5623_fill_cache(codec); 911 912 /* power on device */ 913 alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); 914 915 if (alc5623->add_ctrl) { 916 snd_soc_write(codec, ALC5623_ADD_CTRL_REG, 917 alc5623->add_ctrl); 918 } 919 920 if (alc5623->jack_det_ctrl) { 921 snd_soc_write(codec, ALC5623_JACK_DET_CTRL, 922 alc5623->jack_det_ctrl); 923 } 924 925 switch (alc5623->id) { 926 case 0x21: 927 snd_soc_add_codec_controls(codec, alc5621_vol_snd_controls, 928 ARRAY_SIZE(alc5621_vol_snd_controls)); 929 break; 930 case 0x22: 931 snd_soc_add_codec_controls(codec, alc5622_vol_snd_controls, 932 ARRAY_SIZE(alc5622_vol_snd_controls)); 933 break; 934 case 0x23: 935 snd_soc_add_codec_controls(codec, alc5623_vol_snd_controls, 936 ARRAY_SIZE(alc5623_vol_snd_controls)); 937 break; 938 default: 939 return -EINVAL; 940 } 941 942 snd_soc_add_codec_controls(codec, alc5623_snd_controls, 943 ARRAY_SIZE(alc5623_snd_controls)); 944 945 snd_soc_dapm_new_controls(dapm, alc5623_dapm_widgets, 946 ARRAY_SIZE(alc5623_dapm_widgets)); 947 948 /* set up audio path interconnects */ 949 snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); 950 951 switch (alc5623->id) { 952 case 0x21: 953 case 0x22: 954 snd_soc_dapm_new_controls(dapm, alc5623_dapm_amp_widgets, 955 ARRAY_SIZE(alc5623_dapm_amp_widgets)); 956 snd_soc_dapm_add_routes(dapm, intercon_amp_spk, 957 ARRAY_SIZE(intercon_amp_spk)); 958 break; 959 case 0x23: 960 snd_soc_dapm_add_routes(dapm, intercon_spk, 961 ARRAY_SIZE(intercon_spk)); 962 break; 963 default: 964 return -EINVAL; 965 } 966 967 return ret; 968 } 969 970 /* power down chip */ 971 static int alc5623_remove(struct snd_soc_codec *codec) 972 { 973 alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF); 974 return 0; 975 } 976 977 static struct snd_soc_codec_driver soc_codec_device_alc5623 = { 978 .probe = alc5623_probe, 979 .remove = alc5623_remove, 980 .suspend = alc5623_suspend, 981 .resume = alc5623_resume, 982 .set_bias_level = alc5623_set_bias_level, 983 .reg_cache_size = ALC5623_VENDOR_ID2+2, 984 .reg_word_size = sizeof(u16), 985 .reg_cache_step = 2, 986 }; 987 988 /* 989 * ALC5623 2 wire address is determined by A1 pin 990 * state during powerup. 991 * low = 0x1a 992 * high = 0x1b 993 */ 994 static int alc5623_i2c_probe(struct i2c_client *client, 995 const struct i2c_device_id *id) 996 { 997 struct alc5623_platform_data *pdata; 998 struct alc5623_priv *alc5623; 999 int ret, vid1, vid2; 1000 1001 vid1 = i2c_smbus_read_word_data(client, ALC5623_VENDOR_ID1); 1002 if (vid1 < 0) { 1003 dev_err(&client->dev, "failed to read I2C\n"); 1004 return -EIO; 1005 } 1006 vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8); 1007 1008 vid2 = i2c_smbus_read_byte_data(client, ALC5623_VENDOR_ID2); 1009 if (vid2 < 0) { 1010 dev_err(&client->dev, "failed to read I2C\n"); 1011 return -EIO; 1012 } 1013 1014 if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) { 1015 dev_err(&client->dev, "unknown or wrong codec\n"); 1016 dev_err(&client->dev, "Expected %x:%lx, got %x:%x\n", 1017 0x10ec, id->driver_data, 1018 vid1, vid2); 1019 return -ENODEV; 1020 } 1021 1022 dev_dbg(&client->dev, "Found codec id : alc56%02x\n", vid2); 1023 1024 alc5623 = devm_kzalloc(&client->dev, sizeof(struct alc5623_priv), 1025 GFP_KERNEL); 1026 if (alc5623 == NULL) 1027 return -ENOMEM; 1028 1029 pdata = client->dev.platform_data; 1030 if (pdata) { 1031 alc5623->add_ctrl = pdata->add_ctrl; 1032 alc5623->jack_det_ctrl = pdata->jack_det_ctrl; 1033 } 1034 1035 alc5623->id = vid2; 1036 switch (alc5623->id) { 1037 case 0x21: 1038 alc5623_dai.name = "alc5621-hifi"; 1039 break; 1040 case 0x22: 1041 alc5623_dai.name = "alc5622-hifi"; 1042 break; 1043 case 0x23: 1044 alc5623_dai.name = "alc5623-hifi"; 1045 break; 1046 default: 1047 return -EINVAL; 1048 } 1049 1050 i2c_set_clientdata(client, alc5623); 1051 alc5623->control_type = SND_SOC_I2C; 1052 1053 ret = snd_soc_register_codec(&client->dev, 1054 &soc_codec_device_alc5623, &alc5623_dai, 1); 1055 if (ret != 0) 1056 dev_err(&client->dev, "Failed to register codec: %d\n", ret); 1057 1058 return ret; 1059 } 1060 1061 static int alc5623_i2c_remove(struct i2c_client *client) 1062 { 1063 snd_soc_unregister_codec(&client->dev); 1064 return 0; 1065 } 1066 1067 static const struct i2c_device_id alc5623_i2c_table[] = { 1068 {"alc5621", 0x21}, 1069 {"alc5622", 0x22}, 1070 {"alc5623", 0x23}, 1071 {} 1072 }; 1073 MODULE_DEVICE_TABLE(i2c, alc5623_i2c_table); 1074 1075 /* i2c codec control layer */ 1076 static struct i2c_driver alc5623_i2c_driver = { 1077 .driver = { 1078 .name = "alc562x-codec", 1079 .owner = THIS_MODULE, 1080 }, 1081 .probe = alc5623_i2c_probe, 1082 .remove = alc5623_i2c_remove, 1083 .id_table = alc5623_i2c_table, 1084 }; 1085 1086 module_i2c_driver(alc5623_i2c_driver); 1087 1088 MODULE_DESCRIPTION("ASoC alc5621/2/3 driver"); 1089 MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>"); 1090 MODULE_LICENSE("GPL"); 1091