1 /* 2 * ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver 3 * 4 * Copyright (C) 2009 Renesas Solutions Corp. 5 * Kuninori Morimoto <morimoto.kuninori@renesas.com> 6 * 7 * Based on wm8731.c by Richard Purdie 8 * Based on ak4535.c by Richard Purdie 9 * Based on wm8753.c by Liam Girdwood 10 * 11 * This program is free software; you can redistribute it and/or modify 12 * it under the terms of the GNU General Public License version 2 as 13 * published by the Free Software Foundation. 14 */ 15 16 /* ** CAUTION ** 17 * 18 * This is very simple driver. 19 * It can use headphone output / stereo input only 20 * 21 * AK4642 is tested. 22 * AK4643 is tested. 23 * AK4648 is tested. 24 */ 25 26 #include <linux/delay.h> 27 #include <linux/i2c.h> 28 #include <linux/slab.h> 29 #include <linux/of_device.h> 30 #include <linux/module.h> 31 #include <linux/regmap.h> 32 #include <sound/soc.h> 33 #include <sound/initval.h> 34 #include <sound/tlv.h> 35 36 #define PW_MGMT1 0x00 37 #define PW_MGMT2 0x01 38 #define SG_SL1 0x02 39 #define SG_SL2 0x03 40 #define MD_CTL1 0x04 41 #define MD_CTL2 0x05 42 #define TIMER 0x06 43 #define ALC_CTL1 0x07 44 #define ALC_CTL2 0x08 45 #define L_IVC 0x09 46 #define L_DVC 0x0a 47 #define ALC_CTL3 0x0b 48 #define R_IVC 0x0c 49 #define R_DVC 0x0d 50 #define MD_CTL3 0x0e 51 #define MD_CTL4 0x0f 52 #define PW_MGMT3 0x10 53 #define DF_S 0x11 54 #define FIL3_0 0x12 55 #define FIL3_1 0x13 56 #define FIL3_2 0x14 57 #define FIL3_3 0x15 58 #define EQ_0 0x16 59 #define EQ_1 0x17 60 #define EQ_2 0x18 61 #define EQ_3 0x19 62 #define EQ_4 0x1a 63 #define EQ_5 0x1b 64 #define FIL1_0 0x1c 65 #define FIL1_1 0x1d 66 #define FIL1_2 0x1e 67 #define FIL1_3 0x1f 68 #define PW_MGMT4 0x20 69 #define MD_CTL5 0x21 70 #define LO_MS 0x22 71 #define HP_MS 0x23 72 #define SPK_MS 0x24 73 74 /* PW_MGMT1*/ 75 #define PMVCM (1 << 6) /* VCOM Power Management */ 76 #define PMMIN (1 << 5) /* MIN Input Power Management */ 77 #define PMDAC (1 << 2) /* DAC Power Management */ 78 #define PMADL (1 << 0) /* MIC Amp Lch and ADC Lch Power Management */ 79 80 /* PW_MGMT2 */ 81 #define HPMTN (1 << 6) 82 #define PMHPL (1 << 5) 83 #define PMHPR (1 << 4) 84 #define MS (1 << 3) /* master/slave select */ 85 #define MCKO (1 << 1) 86 #define PMPLL (1 << 0) 87 88 #define PMHP_MASK (PMHPL | PMHPR) 89 #define PMHP PMHP_MASK 90 91 /* PW_MGMT3 */ 92 #define PMADR (1 << 0) /* MIC L / ADC R Power Management */ 93 94 /* SG_SL1 */ 95 #define MINS (1 << 6) /* Switch from MIN to Speaker */ 96 #define DACL (1 << 4) /* Switch from DAC to Stereo or Receiver */ 97 #define PMMP (1 << 2) /* MPWR pin Power Management */ 98 #define MGAIN0 (1 << 0) /* MIC amp gain*/ 99 100 /* TIMER */ 101 #define ZTM(param) ((param & 0x3) << 4) /* ALC Zoro Crossing TimeOut */ 102 #define WTM(param) (((param & 0x4) << 4) | ((param & 0x3) << 2)) 103 104 /* ALC_CTL1 */ 105 #define ALC (1 << 5) /* ALC Enable */ 106 #define LMTH0 (1 << 0) /* ALC Limiter / Recovery Level */ 107 108 /* MD_CTL1 */ 109 #define PLL3 (1 << 7) 110 #define PLL2 (1 << 6) 111 #define PLL1 (1 << 5) 112 #define PLL0 (1 << 4) 113 #define PLL_MASK (PLL3 | PLL2 | PLL1 | PLL0) 114 115 #define BCKO_MASK (1 << 3) 116 #define BCKO_64 BCKO_MASK 117 118 #define DIF_MASK (3 << 0) 119 #define DSP (0 << 0) 120 #define RIGHT_J (1 << 0) 121 #define LEFT_J (2 << 0) 122 #define I2S (3 << 0) 123 124 /* MD_CTL2 */ 125 #define FS0 (1 << 0) 126 #define FS1 (1 << 1) 127 #define FS2 (1 << 2) 128 #define FS3 (1 << 5) 129 #define FS_MASK (FS0 | FS1 | FS2 | FS3) 130 131 /* MD_CTL3 */ 132 #define BST1 (1 << 3) 133 134 /* MD_CTL4 */ 135 #define DACH (1 << 0) 136 137 /* 138 * Playback Volume (table 39) 139 * 140 * max : 0x00 : +12.0 dB 141 * ( 0.5 dB step ) 142 * min : 0xFE : -115.0 dB 143 * mute: 0xFF 144 */ 145 static const DECLARE_TLV_DB_SCALE(out_tlv, -11550, 50, 1); 146 147 static const struct snd_kcontrol_new ak4642_snd_controls[] = { 148 149 SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC, 150 0, 0xFF, 1, out_tlv), 151 }; 152 153 static const struct snd_kcontrol_new ak4642_headphone_control = 154 SOC_DAPM_SINGLE("Switch", PW_MGMT2, 6, 1, 0); 155 156 static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = { 157 SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0), 158 }; 159 160 static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { 161 162 /* Outputs */ 163 SND_SOC_DAPM_OUTPUT("HPOUTL"), 164 SND_SOC_DAPM_OUTPUT("HPOUTR"), 165 SND_SOC_DAPM_OUTPUT("LINEOUT"), 166 167 SND_SOC_DAPM_PGA("HPL Out", PW_MGMT2, 5, 0, NULL, 0), 168 SND_SOC_DAPM_PGA("HPR Out", PW_MGMT2, 4, 0, NULL, 0), 169 SND_SOC_DAPM_SWITCH("Headphone Enable", SND_SOC_NOPM, 0, 0, 170 &ak4642_headphone_control), 171 172 SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0), 173 174 SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0, 175 &ak4642_lout_mixer_controls[0], 176 ARRAY_SIZE(ak4642_lout_mixer_controls)), 177 178 /* DAC */ 179 SND_SOC_DAPM_DAC("DAC", "HiFi Playback", PW_MGMT1, 2, 0), 180 }; 181 182 static const struct snd_soc_dapm_route ak4642_intercon[] = { 183 184 /* Outputs */ 185 {"HPOUTL", NULL, "HPL Out"}, 186 {"HPOUTR", NULL, "HPR Out"}, 187 {"LINEOUT", NULL, "LINEOUT Mixer"}, 188 189 {"HPL Out", NULL, "Headphone Enable"}, 190 {"HPR Out", NULL, "Headphone Enable"}, 191 192 {"Headphone Enable", "Switch", "DACH"}, 193 194 {"DACH", NULL, "DAC"}, 195 196 {"LINEOUT Mixer", "DACL", "DAC"}, 197 }; 198 199 /* 200 * ak4642 register cache 201 */ 202 static const struct reg_default ak4642_reg[] = { 203 { 0, 0x00 }, { 1, 0x00 }, { 2, 0x01 }, { 3, 0x00 }, 204 { 4, 0x02 }, { 5, 0x00 }, { 6, 0x00 }, { 7, 0x00 }, 205 { 8, 0xe1 }, { 9, 0xe1 }, { 10, 0x18 }, { 11, 0x00 }, 206 { 12, 0xe1 }, { 13, 0x18 }, { 14, 0x11 }, { 15, 0x08 }, 207 { 16, 0x00 }, { 17, 0x00 }, { 18, 0x00 }, { 19, 0x00 }, 208 { 20, 0x00 }, { 21, 0x00 }, { 22, 0x00 }, { 23, 0x00 }, 209 { 24, 0x00 }, { 25, 0x00 }, { 26, 0x00 }, { 27, 0x00 }, 210 { 28, 0x00 }, { 29, 0x00 }, { 30, 0x00 }, { 31, 0x00 }, 211 { 32, 0x00 }, { 33, 0x00 }, { 34, 0x00 }, { 35, 0x00 }, 212 { 36, 0x00 }, 213 }; 214 215 static const struct reg_default ak4648_reg[] = { 216 { 0, 0x00 }, { 1, 0x00 }, { 2, 0x01 }, { 3, 0x00 }, 217 { 4, 0x02 }, { 5, 0x00 }, { 6, 0x00 }, { 7, 0x00 }, 218 { 8, 0xe1 }, { 9, 0xe1 }, { 10, 0x18 }, { 11, 0x00 }, 219 { 12, 0xe1 }, { 13, 0x18 }, { 14, 0x11 }, { 15, 0xb8 }, 220 { 16, 0x00 }, { 17, 0x00 }, { 18, 0x00 }, { 19, 0x00 }, 221 { 20, 0x00 }, { 21, 0x00 }, { 22, 0x00 }, { 23, 0x00 }, 222 { 24, 0x00 }, { 25, 0x00 }, { 26, 0x00 }, { 27, 0x00 }, 223 { 28, 0x00 }, { 29, 0x00 }, { 30, 0x00 }, { 31, 0x00 }, 224 { 32, 0x00 }, { 33, 0x00 }, { 34, 0x00 }, { 35, 0x00 }, 225 { 36, 0x00 }, { 37, 0x88 }, { 38, 0x88 }, { 39, 0x08 }, 226 }; 227 228 static int ak4642_dai_startup(struct snd_pcm_substream *substream, 229 struct snd_soc_dai *dai) 230 { 231 int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; 232 struct snd_soc_codec *codec = dai->codec; 233 234 if (is_play) { 235 /* 236 * start headphone output 237 * 238 * PLL, Master Mode 239 * Audio I/F Format :MSB justified (ADC & DAC) 240 * Bass Boost Level : Middle 241 * 242 * This operation came from example code of 243 * "ASAHI KASEI AK4642" (japanese) manual p97. 244 */ 245 snd_soc_write(codec, L_IVC, 0x91); /* volume */ 246 snd_soc_write(codec, R_IVC, 0x91); /* volume */ 247 } else { 248 /* 249 * start stereo input 250 * 251 * PLL Master Mode 252 * Audio I/F Format:MSB justified (ADC & DAC) 253 * Pre MIC AMP:+20dB 254 * MIC Power On 255 * ALC setting:Refer to Table 35 256 * ALC bit=“1” 257 * 258 * This operation came from example code of 259 * "ASAHI KASEI AK4642" (japanese) manual p94. 260 */ 261 snd_soc_update_bits(codec, SG_SL1, PMMP | MGAIN0, PMMP | MGAIN0); 262 snd_soc_write(codec, TIMER, ZTM(0x3) | WTM(0x3)); 263 snd_soc_write(codec, ALC_CTL1, ALC | LMTH0); 264 snd_soc_update_bits(codec, PW_MGMT1, PMADL, PMADL); 265 snd_soc_update_bits(codec, PW_MGMT3, PMADR, PMADR); 266 } 267 268 return 0; 269 } 270 271 static void ak4642_dai_shutdown(struct snd_pcm_substream *substream, 272 struct snd_soc_dai *dai) 273 { 274 int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; 275 struct snd_soc_codec *codec = dai->codec; 276 277 if (is_play) { 278 } else { 279 /* stop stereo input */ 280 snd_soc_update_bits(codec, PW_MGMT1, PMADL, 0); 281 snd_soc_update_bits(codec, PW_MGMT3, PMADR, 0); 282 snd_soc_update_bits(codec, ALC_CTL1, ALC, 0); 283 } 284 } 285 286 static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai, 287 int clk_id, unsigned int freq, int dir) 288 { 289 struct snd_soc_codec *codec = codec_dai->codec; 290 u8 pll; 291 292 switch (freq) { 293 case 11289600: 294 pll = PLL2; 295 break; 296 case 12288000: 297 pll = PLL2 | PLL0; 298 break; 299 case 12000000: 300 pll = PLL2 | PLL1; 301 break; 302 case 24000000: 303 pll = PLL2 | PLL1 | PLL0; 304 break; 305 case 13500000: 306 pll = PLL3 | PLL2; 307 break; 308 case 27000000: 309 pll = PLL3 | PLL2 | PLL0; 310 break; 311 default: 312 return -EINVAL; 313 } 314 snd_soc_update_bits(codec, MD_CTL1, PLL_MASK, pll); 315 316 return 0; 317 } 318 319 static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) 320 { 321 struct snd_soc_codec *codec = dai->codec; 322 u8 data; 323 u8 bcko; 324 325 data = MCKO | PMPLL; /* use MCKO */ 326 bcko = 0; 327 328 /* set master/slave audio interface */ 329 switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { 330 case SND_SOC_DAIFMT_CBM_CFM: 331 data |= MS; 332 bcko = BCKO_64; 333 break; 334 case SND_SOC_DAIFMT_CBS_CFS: 335 break; 336 default: 337 return -EINVAL; 338 } 339 snd_soc_update_bits(codec, PW_MGMT2, MS | MCKO | PMPLL, data); 340 snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko); 341 342 /* format type */ 343 data = 0; 344 switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { 345 case SND_SOC_DAIFMT_LEFT_J: 346 data = LEFT_J; 347 break; 348 case SND_SOC_DAIFMT_I2S: 349 data = I2S; 350 break; 351 /* FIXME 352 * Please add RIGHT_J / DSP support here 353 */ 354 default: 355 return -EINVAL; 356 } 357 snd_soc_update_bits(codec, MD_CTL1, DIF_MASK, data); 358 359 return 0; 360 } 361 362 static int ak4642_dai_hw_params(struct snd_pcm_substream *substream, 363 struct snd_pcm_hw_params *params, 364 struct snd_soc_dai *dai) 365 { 366 struct snd_soc_codec *codec = dai->codec; 367 u8 rate; 368 369 switch (params_rate(params)) { 370 case 7350: 371 rate = FS2; 372 break; 373 case 8000: 374 rate = 0; 375 break; 376 case 11025: 377 rate = FS2 | FS0; 378 break; 379 case 12000: 380 rate = FS0; 381 break; 382 case 14700: 383 rate = FS2 | FS1; 384 break; 385 case 16000: 386 rate = FS1; 387 break; 388 case 22050: 389 rate = FS2 | FS1 | FS0; 390 break; 391 case 24000: 392 rate = FS1 | FS0; 393 break; 394 case 29400: 395 rate = FS3 | FS2 | FS1; 396 break; 397 case 32000: 398 rate = FS3 | FS1; 399 break; 400 case 44100: 401 rate = FS3 | FS2 | FS1 | FS0; 402 break; 403 case 48000: 404 rate = FS3 | FS1 | FS0; 405 break; 406 default: 407 return -EINVAL; 408 } 409 snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate); 410 411 return 0; 412 } 413 414 static int ak4642_set_bias_level(struct snd_soc_codec *codec, 415 enum snd_soc_bias_level level) 416 { 417 switch (level) { 418 case SND_SOC_BIAS_OFF: 419 snd_soc_write(codec, PW_MGMT1, 0x00); 420 break; 421 default: 422 snd_soc_update_bits(codec, PW_MGMT1, PMVCM, PMVCM); 423 break; 424 } 425 codec->dapm.bias_level = level; 426 427 return 0; 428 } 429 430 static const struct snd_soc_dai_ops ak4642_dai_ops = { 431 .startup = ak4642_dai_startup, 432 .shutdown = ak4642_dai_shutdown, 433 .set_sysclk = ak4642_dai_set_sysclk, 434 .set_fmt = ak4642_dai_set_fmt, 435 .hw_params = ak4642_dai_hw_params, 436 }; 437 438 static struct snd_soc_dai_driver ak4642_dai = { 439 .name = "ak4642-hifi", 440 .playback = { 441 .stream_name = "Playback", 442 .channels_min = 1, 443 .channels_max = 2, 444 .rates = SNDRV_PCM_RATE_8000_48000, 445 .formats = SNDRV_PCM_FMTBIT_S16_LE }, 446 .capture = { 447 .stream_name = "Capture", 448 .channels_min = 1, 449 .channels_max = 2, 450 .rates = SNDRV_PCM_RATE_8000_48000, 451 .formats = SNDRV_PCM_FMTBIT_S16_LE }, 452 .ops = &ak4642_dai_ops, 453 .symmetric_rates = 1, 454 }; 455 456 static int ak4642_resume(struct snd_soc_codec *codec) 457 { 458 struct regmap *regmap = dev_get_regmap(codec->dev, NULL); 459 460 regcache_mark_dirty(regmap); 461 regcache_sync(regmap); 462 return 0; 463 } 464 465 466 static int ak4642_probe(struct snd_soc_codec *codec) 467 { 468 ak4642_set_bias_level(codec, SND_SOC_BIAS_STANDBY); 469 470 return 0; 471 } 472 473 static int ak4642_remove(struct snd_soc_codec *codec) 474 { 475 ak4642_set_bias_level(codec, SND_SOC_BIAS_OFF); 476 return 0; 477 } 478 479 static struct snd_soc_codec_driver soc_codec_dev_ak4642 = { 480 .probe = ak4642_probe, 481 .remove = ak4642_remove, 482 .resume = ak4642_resume, 483 .set_bias_level = ak4642_set_bias_level, 484 .controls = ak4642_snd_controls, 485 .num_controls = ARRAY_SIZE(ak4642_snd_controls), 486 .dapm_widgets = ak4642_dapm_widgets, 487 .num_dapm_widgets = ARRAY_SIZE(ak4642_dapm_widgets), 488 .dapm_routes = ak4642_intercon, 489 .num_dapm_routes = ARRAY_SIZE(ak4642_intercon), 490 }; 491 492 static const struct regmap_config ak4642_regmap = { 493 .reg_bits = 8, 494 .val_bits = 8, 495 .max_register = ARRAY_SIZE(ak4642_reg) + 1, 496 .reg_defaults = ak4642_reg, 497 .num_reg_defaults = ARRAY_SIZE(ak4642_reg), 498 }; 499 500 static const struct regmap_config ak4648_regmap = { 501 .reg_bits = 8, 502 .val_bits = 8, 503 .max_register = ARRAY_SIZE(ak4648_reg) + 1, 504 .reg_defaults = ak4648_reg, 505 .num_reg_defaults = ARRAY_SIZE(ak4648_reg), 506 }; 507 508 static struct of_device_id ak4642_of_match[]; 509 static int ak4642_i2c_probe(struct i2c_client *i2c, 510 const struct i2c_device_id *id) 511 { 512 struct device_node *np = i2c->dev.of_node; 513 const struct regmap_config *regmap_config = NULL; 514 struct regmap *regmap; 515 516 if (np) { 517 const struct of_device_id *of_id; 518 519 of_id = of_match_device(ak4642_of_match, &i2c->dev); 520 if (of_id) 521 regmap_config = of_id->data; 522 } else { 523 regmap_config = (const struct regmap_config *)id->driver_data; 524 } 525 526 if (!regmap_config) { 527 dev_err(&i2c->dev, "Unknown device type\n"); 528 return -EINVAL; 529 } 530 531 regmap = devm_regmap_init_i2c(i2c, regmap_config); 532 if (IS_ERR(regmap)) 533 return PTR_ERR(regmap); 534 535 return snd_soc_register_codec(&i2c->dev, 536 &soc_codec_dev_ak4642, &ak4642_dai, 1); 537 } 538 539 static int ak4642_i2c_remove(struct i2c_client *client) 540 { 541 snd_soc_unregister_codec(&client->dev); 542 return 0; 543 } 544 545 static struct of_device_id ak4642_of_match[] = { 546 { .compatible = "asahi-kasei,ak4642", .data = &ak4642_regmap}, 547 { .compatible = "asahi-kasei,ak4643", .data = &ak4642_regmap}, 548 { .compatible = "asahi-kasei,ak4648", .data = &ak4648_regmap}, 549 {}, 550 }; 551 MODULE_DEVICE_TABLE(of, ak4642_of_match); 552 553 static const struct i2c_device_id ak4642_i2c_id[] = { 554 { "ak4642", (kernel_ulong_t)&ak4642_regmap }, 555 { "ak4643", (kernel_ulong_t)&ak4642_regmap }, 556 { "ak4648", (kernel_ulong_t)&ak4648_regmap }, 557 { } 558 }; 559 MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id); 560 561 static struct i2c_driver ak4642_i2c_driver = { 562 .driver = { 563 .name = "ak4642-codec", 564 .owner = THIS_MODULE, 565 .of_match_table = ak4642_of_match, 566 }, 567 .probe = ak4642_i2c_probe, 568 .remove = ak4642_i2c_remove, 569 .id_table = ak4642_i2c_id, 570 }; 571 572 module_i2c_driver(ak4642_i2c_driver); 573 574 MODULE_DESCRIPTION("Soc AK4642 driver"); 575 MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>"); 576 MODULE_LICENSE("GPL"); 577