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 <sound/soc.h> 32 #include <sound/initval.h> 33 #include <sound/tlv.h> 34 35 #define PW_MGMT1 0x00 36 #define PW_MGMT2 0x01 37 #define SG_SL1 0x02 38 #define SG_SL2 0x03 39 #define MD_CTL1 0x04 40 #define MD_CTL2 0x05 41 #define TIMER 0x06 42 #define ALC_CTL1 0x07 43 #define ALC_CTL2 0x08 44 #define L_IVC 0x09 45 #define L_DVC 0x0a 46 #define ALC_CTL3 0x0b 47 #define R_IVC 0x0c 48 #define R_DVC 0x0d 49 #define MD_CTL3 0x0e 50 #define MD_CTL4 0x0f 51 #define PW_MGMT3 0x10 52 #define DF_S 0x11 53 #define FIL3_0 0x12 54 #define FIL3_1 0x13 55 #define FIL3_2 0x14 56 #define FIL3_3 0x15 57 #define EQ_0 0x16 58 #define EQ_1 0x17 59 #define EQ_2 0x18 60 #define EQ_3 0x19 61 #define EQ_4 0x1a 62 #define EQ_5 0x1b 63 #define FIL1_0 0x1c 64 #define FIL1_1 0x1d 65 #define FIL1_2 0x1e 66 #define FIL1_3 0x1f 67 #define PW_MGMT4 0x20 68 #define MD_CTL5 0x21 69 #define LO_MS 0x22 70 #define HP_MS 0x23 71 #define SPK_MS 0x24 72 73 /* PW_MGMT1*/ 74 #define PMVCM (1 << 6) /* VCOM Power Management */ 75 #define PMMIN (1 << 5) /* MIN Input Power Management */ 76 #define PMDAC (1 << 2) /* DAC Power Management */ 77 #define PMADL (1 << 0) /* MIC Amp Lch and ADC Lch Power Management */ 78 79 /* PW_MGMT2 */ 80 #define HPMTN (1 << 6) 81 #define PMHPL (1 << 5) 82 #define PMHPR (1 << 4) 83 #define MS (1 << 3) /* master/slave select */ 84 #define MCKO (1 << 1) 85 #define PMPLL (1 << 0) 86 87 #define PMHP_MASK (PMHPL | PMHPR) 88 #define PMHP PMHP_MASK 89 90 /* PW_MGMT3 */ 91 #define PMADR (1 << 0) /* MIC L / ADC R Power Management */ 92 93 /* SG_SL1 */ 94 #define MINS (1 << 6) /* Switch from MIN to Speaker */ 95 #define DACL (1 << 4) /* Switch from DAC to Stereo or Receiver */ 96 #define PMMP (1 << 2) /* MPWR pin Power Management */ 97 #define MGAIN0 (1 << 0) /* MIC amp gain*/ 98 99 /* TIMER */ 100 #define ZTM(param) ((param & 0x3) << 4) /* ALC Zoro Crossing TimeOut */ 101 #define WTM(param) (((param & 0x4) << 4) | ((param & 0x3) << 2)) 102 103 /* ALC_CTL1 */ 104 #define ALC (1 << 5) /* ALC Enable */ 105 #define LMTH0 (1 << 0) /* ALC Limiter / Recovery Level */ 106 107 /* MD_CTL1 */ 108 #define PLL3 (1 << 7) 109 #define PLL2 (1 << 6) 110 #define PLL1 (1 << 5) 111 #define PLL0 (1 << 4) 112 #define PLL_MASK (PLL3 | PLL2 | PLL1 | PLL0) 113 114 #define BCKO_MASK (1 << 3) 115 #define BCKO_64 BCKO_MASK 116 117 #define DIF_MASK (3 << 0) 118 #define DSP (0 << 0) 119 #define RIGHT_J (1 << 0) 120 #define LEFT_J (2 << 0) 121 #define I2S (3 << 0) 122 123 /* MD_CTL2 */ 124 #define FS0 (1 << 0) 125 #define FS1 (1 << 1) 126 #define FS2 (1 << 2) 127 #define FS3 (1 << 5) 128 #define FS_MASK (FS0 | FS1 | FS2 | FS3) 129 130 /* MD_CTL3 */ 131 #define BST1 (1 << 3) 132 133 /* MD_CTL4 */ 134 #define DACH (1 << 0) 135 136 /* 137 * Playback Volume (table 39) 138 * 139 * max : 0x00 : +12.0 dB 140 * ( 0.5 dB step ) 141 * min : 0xFE : -115.0 dB 142 * mute: 0xFF 143 */ 144 static const DECLARE_TLV_DB_SCALE(out_tlv, -11550, 50, 1); 145 146 static const struct snd_kcontrol_new ak4642_snd_controls[] = { 147 148 SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC, 149 0, 0xFF, 1, out_tlv), 150 }; 151 152 static const struct snd_kcontrol_new ak4642_headphone_control = 153 SOC_DAPM_SINGLE("Switch", PW_MGMT2, 6, 1, 0); 154 155 static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = { 156 SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0), 157 }; 158 159 static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { 160 161 /* Outputs */ 162 SND_SOC_DAPM_OUTPUT("HPOUTL"), 163 SND_SOC_DAPM_OUTPUT("HPOUTR"), 164 SND_SOC_DAPM_OUTPUT("LINEOUT"), 165 166 SND_SOC_DAPM_PGA("HPL Out", PW_MGMT2, 5, 0, NULL, 0), 167 SND_SOC_DAPM_PGA("HPR Out", PW_MGMT2, 4, 0, NULL, 0), 168 SND_SOC_DAPM_SWITCH("Headphone Enable", SND_SOC_NOPM, 0, 0, 169 &ak4642_headphone_control), 170 171 SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0), 172 173 SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0, 174 &ak4642_lout_mixer_controls[0], 175 ARRAY_SIZE(ak4642_lout_mixer_controls)), 176 177 /* DAC */ 178 SND_SOC_DAPM_DAC("DAC", "HiFi Playback", PW_MGMT1, 2, 0), 179 }; 180 181 static const struct snd_soc_dapm_route ak4642_intercon[] = { 182 183 /* Outputs */ 184 {"HPOUTL", NULL, "HPL Out"}, 185 {"HPOUTR", NULL, "HPR Out"}, 186 {"LINEOUT", NULL, "LINEOUT Mixer"}, 187 188 {"HPL Out", NULL, "Headphone Enable"}, 189 {"HPR Out", NULL, "Headphone Enable"}, 190 191 {"Headphone Enable", "Switch", "DACH"}, 192 193 {"DACH", NULL, "DAC"}, 194 195 {"LINEOUT Mixer", "DACL", "DAC"}, 196 }; 197 198 /* 199 * ak4642 register cache 200 */ 201 static const u8 ak4642_reg[] = { 202 0x00, 0x00, 0x01, 0x00, 203 0x02, 0x00, 0x00, 0x00, 204 0xe1, 0xe1, 0x18, 0x00, 205 0xe1, 0x18, 0x11, 0x08, 206 0x00, 0x00, 0x00, 0x00, 207 0x00, 0x00, 0x00, 0x00, 208 0x00, 0x00, 0x00, 0x00, 209 0x00, 0x00, 0x00, 0x00, 210 0x00, 0x00, 0x00, 0x00, 211 0x00, 212 }; 213 214 static const u8 ak4648_reg[] = { 215 0x00, 0x00, 0x01, 0x00, 216 0x02, 0x00, 0x00, 0x00, 217 0xe1, 0xe1, 0x18, 0x00, 218 0xe1, 0x18, 0x11, 0xb8, 219 0x00, 0x00, 0x00, 0x00, 220 0x00, 0x00, 0x00, 0x00, 221 0x00, 0x00, 0x00, 0x00, 222 0x00, 0x00, 0x00, 0x00, 223 0x00, 0x00, 0x00, 0x00, 224 0x00, 0x88, 0x88, 0x08, 225 }; 226 227 static int ak4642_dai_startup(struct snd_pcm_substream *substream, 228 struct snd_soc_dai *dai) 229 { 230 int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; 231 struct snd_soc_codec *codec = dai->codec; 232 233 if (is_play) { 234 /* 235 * start headphone output 236 * 237 * PLL, Master Mode 238 * Audio I/F Format :MSB justified (ADC & DAC) 239 * Bass Boost Level : Middle 240 * 241 * This operation came from example code of 242 * "ASAHI KASEI AK4642" (japanese) manual p97. 243 */ 244 snd_soc_write(codec, L_IVC, 0x91); /* volume */ 245 snd_soc_write(codec, R_IVC, 0x91); /* volume */ 246 } else { 247 /* 248 * start stereo input 249 * 250 * PLL Master Mode 251 * Audio I/F Format:MSB justified (ADC & DAC) 252 * Pre MIC AMP:+20dB 253 * MIC Power On 254 * ALC setting:Refer to Table 35 255 * ALC bit=“1” 256 * 257 * This operation came from example code of 258 * "ASAHI KASEI AK4642" (japanese) manual p94. 259 */ 260 snd_soc_update_bits(codec, SG_SL1, PMMP | MGAIN0, PMMP | MGAIN0); 261 snd_soc_write(codec, TIMER, ZTM(0x3) | WTM(0x3)); 262 snd_soc_write(codec, ALC_CTL1, ALC | LMTH0); 263 snd_soc_update_bits(codec, PW_MGMT1, PMADL, PMADL); 264 snd_soc_update_bits(codec, PW_MGMT3, PMADR, PMADR); 265 } 266 267 return 0; 268 } 269 270 static void ak4642_dai_shutdown(struct snd_pcm_substream *substream, 271 struct snd_soc_dai *dai) 272 { 273 int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; 274 struct snd_soc_codec *codec = dai->codec; 275 276 if (is_play) { 277 } else { 278 /* stop stereo input */ 279 snd_soc_update_bits(codec, PW_MGMT1, PMADL, 0); 280 snd_soc_update_bits(codec, PW_MGMT3, PMADR, 0); 281 snd_soc_update_bits(codec, ALC_CTL1, ALC, 0); 282 } 283 } 284 285 static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai, 286 int clk_id, unsigned int freq, int dir) 287 { 288 struct snd_soc_codec *codec = codec_dai->codec; 289 u8 pll; 290 291 switch (freq) { 292 case 11289600: 293 pll = PLL2; 294 break; 295 case 12288000: 296 pll = PLL2 | PLL0; 297 break; 298 case 12000000: 299 pll = PLL2 | PLL1; 300 break; 301 case 24000000: 302 pll = PLL2 | PLL1 | PLL0; 303 break; 304 case 13500000: 305 pll = PLL3 | PLL2; 306 break; 307 case 27000000: 308 pll = PLL3 | PLL2 | PLL0; 309 break; 310 default: 311 return -EINVAL; 312 } 313 snd_soc_update_bits(codec, MD_CTL1, PLL_MASK, pll); 314 315 return 0; 316 } 317 318 static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) 319 { 320 struct snd_soc_codec *codec = dai->codec; 321 u8 data; 322 u8 bcko; 323 324 data = MCKO | PMPLL; /* use MCKO */ 325 bcko = 0; 326 327 /* set master/slave audio interface */ 328 switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { 329 case SND_SOC_DAIFMT_CBM_CFM: 330 data |= MS; 331 bcko = BCKO_64; 332 break; 333 case SND_SOC_DAIFMT_CBS_CFS: 334 break; 335 default: 336 return -EINVAL; 337 } 338 snd_soc_update_bits(codec, PW_MGMT2, MS | MCKO | PMPLL, data); 339 snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko); 340 341 /* format type */ 342 data = 0; 343 switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { 344 case SND_SOC_DAIFMT_LEFT_J: 345 data = LEFT_J; 346 break; 347 case SND_SOC_DAIFMT_I2S: 348 data = I2S; 349 break; 350 /* FIXME 351 * Please add RIGHT_J / DSP support here 352 */ 353 default: 354 return -EINVAL; 355 } 356 snd_soc_update_bits(codec, MD_CTL1, DIF_MASK, data); 357 358 return 0; 359 } 360 361 static int ak4642_dai_hw_params(struct snd_pcm_substream *substream, 362 struct snd_pcm_hw_params *params, 363 struct snd_soc_dai *dai) 364 { 365 struct snd_soc_codec *codec = dai->codec; 366 u8 rate; 367 368 switch (params_rate(params)) { 369 case 7350: 370 rate = FS2; 371 break; 372 case 8000: 373 rate = 0; 374 break; 375 case 11025: 376 rate = FS2 | FS0; 377 break; 378 case 12000: 379 rate = FS0; 380 break; 381 case 14700: 382 rate = FS2 | FS1; 383 break; 384 case 16000: 385 rate = FS1; 386 break; 387 case 22050: 388 rate = FS2 | FS1 | FS0; 389 break; 390 case 24000: 391 rate = FS1 | FS0; 392 break; 393 case 29400: 394 rate = FS3 | FS2 | FS1; 395 break; 396 case 32000: 397 rate = FS3 | FS1; 398 break; 399 case 44100: 400 rate = FS3 | FS2 | FS1 | FS0; 401 break; 402 case 48000: 403 rate = FS3 | FS1 | FS0; 404 break; 405 default: 406 return -EINVAL; 407 } 408 snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate); 409 410 return 0; 411 } 412 413 static int ak4642_set_bias_level(struct snd_soc_codec *codec, 414 enum snd_soc_bias_level level) 415 { 416 switch (level) { 417 case SND_SOC_BIAS_OFF: 418 snd_soc_write(codec, PW_MGMT1, 0x00); 419 break; 420 default: 421 snd_soc_update_bits(codec, PW_MGMT1, PMVCM, PMVCM); 422 break; 423 } 424 codec->dapm.bias_level = level; 425 426 return 0; 427 } 428 429 static const struct snd_soc_dai_ops ak4642_dai_ops = { 430 .startup = ak4642_dai_startup, 431 .shutdown = ak4642_dai_shutdown, 432 .set_sysclk = ak4642_dai_set_sysclk, 433 .set_fmt = ak4642_dai_set_fmt, 434 .hw_params = ak4642_dai_hw_params, 435 }; 436 437 static struct snd_soc_dai_driver ak4642_dai = { 438 .name = "ak4642-hifi", 439 .playback = { 440 .stream_name = "Playback", 441 .channels_min = 1, 442 .channels_max = 2, 443 .rates = SNDRV_PCM_RATE_8000_48000, 444 .formats = SNDRV_PCM_FMTBIT_S16_LE }, 445 .capture = { 446 .stream_name = "Capture", 447 .channels_min = 1, 448 .channels_max = 2, 449 .rates = SNDRV_PCM_RATE_8000_48000, 450 .formats = SNDRV_PCM_FMTBIT_S16_LE }, 451 .ops = &ak4642_dai_ops, 452 .symmetric_rates = 1, 453 }; 454 455 static int ak4642_resume(struct snd_soc_codec *codec) 456 { 457 snd_soc_cache_sync(codec); 458 return 0; 459 } 460 461 462 static int ak4642_probe(struct snd_soc_codec *codec) 463 { 464 int ret; 465 466 ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); 467 if (ret < 0) { 468 dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); 469 return ret; 470 } 471 472 snd_soc_add_codec_controls(codec, ak4642_snd_controls, 473 ARRAY_SIZE(ak4642_snd_controls)); 474 475 ak4642_set_bias_level(codec, SND_SOC_BIAS_STANDBY); 476 477 return 0; 478 } 479 480 static int ak4642_remove(struct snd_soc_codec *codec) 481 { 482 ak4642_set_bias_level(codec, SND_SOC_BIAS_OFF); 483 return 0; 484 } 485 486 static struct snd_soc_codec_driver soc_codec_dev_ak4642 = { 487 .probe = ak4642_probe, 488 .remove = ak4642_remove, 489 .resume = ak4642_resume, 490 .set_bias_level = ak4642_set_bias_level, 491 .reg_cache_default = ak4642_reg, /* ak4642 reg */ 492 .reg_cache_size = ARRAY_SIZE(ak4642_reg), /* ak4642 reg */ 493 .reg_word_size = sizeof(u8), 494 .dapm_widgets = ak4642_dapm_widgets, 495 .num_dapm_widgets = ARRAY_SIZE(ak4642_dapm_widgets), 496 .dapm_routes = ak4642_intercon, 497 .num_dapm_routes = ARRAY_SIZE(ak4642_intercon), 498 }; 499 500 static struct snd_soc_codec_driver soc_codec_dev_ak4648 = { 501 .probe = ak4642_probe, 502 .remove = ak4642_remove, 503 .resume = ak4642_resume, 504 .set_bias_level = ak4642_set_bias_level, 505 .reg_cache_default = ak4648_reg, /* ak4648 reg */ 506 .reg_cache_size = ARRAY_SIZE(ak4648_reg), /* ak4648 reg */ 507 .reg_word_size = sizeof(u8), 508 .dapm_widgets = ak4642_dapm_widgets, 509 .num_dapm_widgets = ARRAY_SIZE(ak4642_dapm_widgets), 510 .dapm_routes = ak4642_intercon, 511 .num_dapm_routes = ARRAY_SIZE(ak4642_intercon), 512 }; 513 514 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) 515 static struct of_device_id ak4642_of_match[]; 516 static int ak4642_i2c_probe(struct i2c_client *i2c, 517 const struct i2c_device_id *id) 518 { 519 struct device_node *np = i2c->dev.of_node; 520 const struct snd_soc_codec_driver *driver; 521 522 driver = NULL; 523 if (np) { 524 const struct of_device_id *of_id; 525 526 of_id = of_match_device(ak4642_of_match, &i2c->dev); 527 if (of_id) 528 driver = of_id->data; 529 } else { 530 driver = (struct snd_soc_codec_driver *)id->driver_data; 531 } 532 533 if (!driver) { 534 dev_err(&i2c->dev, "no driver\n"); 535 return -EINVAL; 536 } 537 538 return snd_soc_register_codec(&i2c->dev, 539 driver, &ak4642_dai, 1); 540 } 541 542 static int ak4642_i2c_remove(struct i2c_client *client) 543 { 544 snd_soc_unregister_codec(&client->dev); 545 return 0; 546 } 547 548 static struct of_device_id ak4642_of_match[] = { 549 { .compatible = "asahi-kasei,ak4642", .data = &soc_codec_dev_ak4642}, 550 { .compatible = "asahi-kasei,ak4643", .data = &soc_codec_dev_ak4642}, 551 { .compatible = "asahi-kasei,ak4648", .data = &soc_codec_dev_ak4648}, 552 {}, 553 }; 554 MODULE_DEVICE_TABLE(of, ak4642_of_match); 555 556 static const struct i2c_device_id ak4642_i2c_id[] = { 557 { "ak4642", (kernel_ulong_t)&soc_codec_dev_ak4642 }, 558 { "ak4643", (kernel_ulong_t)&soc_codec_dev_ak4642 }, 559 { "ak4648", (kernel_ulong_t)&soc_codec_dev_ak4648 }, 560 { } 561 }; 562 MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id); 563 564 static struct i2c_driver ak4642_i2c_driver = { 565 .driver = { 566 .name = "ak4642-codec", 567 .owner = THIS_MODULE, 568 .of_match_table = ak4642_of_match, 569 }, 570 .probe = ak4642_i2c_probe, 571 .remove = ak4642_i2c_remove, 572 .id_table = ak4642_i2c_id, 573 }; 574 #endif 575 576 static int __init ak4642_modinit(void) 577 { 578 int ret = 0; 579 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) 580 ret = i2c_add_driver(&ak4642_i2c_driver); 581 #endif 582 return ret; 583 584 } 585 module_init(ak4642_modinit); 586 587 static void __exit ak4642_exit(void) 588 { 589 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) 590 i2c_del_driver(&ak4642_i2c_driver); 591 #endif 592 593 } 594 module_exit(ak4642_exit); 595 596 MODULE_DESCRIPTION("Soc AK4642 driver"); 597 MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>"); 598 MODULE_LICENSE("GPL"); 599