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/module.h> 30 #include <sound/soc.h> 31 #include <sound/initval.h> 32 #include <sound/tlv.h> 33 34 #define PW_MGMT1 0x00 35 #define PW_MGMT2 0x01 36 #define SG_SL1 0x02 37 #define SG_SL2 0x03 38 #define MD_CTL1 0x04 39 #define MD_CTL2 0x05 40 #define TIMER 0x06 41 #define ALC_CTL1 0x07 42 #define ALC_CTL2 0x08 43 #define L_IVC 0x09 44 #define L_DVC 0x0a 45 #define ALC_CTL3 0x0b 46 #define R_IVC 0x0c 47 #define R_DVC 0x0d 48 #define MD_CTL3 0x0e 49 #define MD_CTL4 0x0f 50 #define PW_MGMT3 0x10 51 #define DF_S 0x11 52 #define FIL3_0 0x12 53 #define FIL3_1 0x13 54 #define FIL3_2 0x14 55 #define FIL3_3 0x15 56 #define EQ_0 0x16 57 #define EQ_1 0x17 58 #define EQ_2 0x18 59 #define EQ_3 0x19 60 #define EQ_4 0x1a 61 #define EQ_5 0x1b 62 #define FIL1_0 0x1c 63 #define FIL1_1 0x1d 64 #define FIL1_2 0x1e 65 #define FIL1_3 0x1f 66 #define PW_MGMT4 0x20 67 #define MD_CTL5 0x21 68 #define LO_MS 0x22 69 #define HP_MS 0x23 70 #define SPK_MS 0x24 71 72 /* PW_MGMT1*/ 73 #define PMVCM (1 << 6) /* VCOM Power Management */ 74 #define PMMIN (1 << 5) /* MIN Input Power Management */ 75 #define PMDAC (1 << 2) /* DAC Power Management */ 76 #define PMADL (1 << 0) /* MIC Amp Lch and ADC Lch Power Management */ 77 78 /* PW_MGMT2 */ 79 #define HPMTN (1 << 6) 80 #define PMHPL (1 << 5) 81 #define PMHPR (1 << 4) 82 #define MS (1 << 3) /* master/slave select */ 83 #define MCKO (1 << 1) 84 #define PMPLL (1 << 0) 85 86 #define PMHP_MASK (PMHPL | PMHPR) 87 #define PMHP PMHP_MASK 88 89 /* PW_MGMT3 */ 90 #define PMADR (1 << 0) /* MIC L / ADC R Power Management */ 91 92 /* SG_SL1 */ 93 #define MINS (1 << 6) /* Switch from MIN to Speaker */ 94 #define DACL (1 << 4) /* Switch from DAC to Stereo or Receiver */ 95 #define PMMP (1 << 2) /* MPWR pin Power Management */ 96 #define MGAIN0 (1 << 0) /* MIC amp gain*/ 97 98 /* TIMER */ 99 #define ZTM(param) ((param & 0x3) << 4) /* ALC Zoro Crossing TimeOut */ 100 #define WTM(param) (((param & 0x4) << 4) | ((param & 0x3) << 2)) 101 102 /* ALC_CTL1 */ 103 #define ALC (1 << 5) /* ALC Enable */ 104 #define LMTH0 (1 << 0) /* ALC Limiter / Recovery Level */ 105 106 /* MD_CTL1 */ 107 #define PLL3 (1 << 7) 108 #define PLL2 (1 << 6) 109 #define PLL1 (1 << 5) 110 #define PLL0 (1 << 4) 111 #define PLL_MASK (PLL3 | PLL2 | PLL1 | PLL0) 112 113 #define BCKO_MASK (1 << 3) 114 #define BCKO_64 BCKO_MASK 115 116 #define DIF_MASK (3 << 0) 117 #define DSP (0 << 0) 118 #define RIGHT_J (1 << 0) 119 #define LEFT_J (2 << 0) 120 #define I2S (3 << 0) 121 122 /* MD_CTL2 */ 123 #define FS0 (1 << 0) 124 #define FS1 (1 << 1) 125 #define FS2 (1 << 2) 126 #define FS3 (1 << 5) 127 #define FS_MASK (FS0 | FS1 | FS2 | FS3) 128 129 /* MD_CTL3 */ 130 #define BST1 (1 << 3) 131 132 /* MD_CTL4 */ 133 #define DACH (1 << 0) 134 135 /* 136 * Playback Volume (table 39) 137 * 138 * max : 0x00 : +12.0 dB 139 * ( 0.5 dB step ) 140 * min : 0xFE : -115.0 dB 141 * mute: 0xFF 142 */ 143 static const DECLARE_TLV_DB_SCALE(out_tlv, -11550, 50, 1); 144 145 static const struct snd_kcontrol_new ak4642_snd_controls[] = { 146 147 SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC, 148 0, 0xFF, 1, out_tlv), 149 }; 150 151 static const struct snd_kcontrol_new ak4642_headphone_control = 152 SOC_DAPM_SINGLE("Switch", PW_MGMT2, 6, 1, 0); 153 154 static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = { 155 SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0), 156 }; 157 158 static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { 159 160 /* Outputs */ 161 SND_SOC_DAPM_OUTPUT("HPOUTL"), 162 SND_SOC_DAPM_OUTPUT("HPOUTR"), 163 SND_SOC_DAPM_OUTPUT("LINEOUT"), 164 165 SND_SOC_DAPM_PGA("HPL Out", PW_MGMT2, 5, 0, NULL, 0), 166 SND_SOC_DAPM_PGA("HPR Out", PW_MGMT2, 4, 0, NULL, 0), 167 SND_SOC_DAPM_SWITCH("Headphone Enable", SND_SOC_NOPM, 0, 0, 168 &ak4642_headphone_control), 169 170 SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0), 171 172 SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0, 173 &ak4642_lout_mixer_controls[0], 174 ARRAY_SIZE(ak4642_lout_mixer_controls)), 175 176 /* DAC */ 177 SND_SOC_DAPM_DAC("DAC", "HiFi Playback", PW_MGMT1, 2, 0), 178 }; 179 180 static const struct snd_soc_dapm_route ak4642_intercon[] = { 181 182 /* Outputs */ 183 {"HPOUTL", NULL, "HPL Out"}, 184 {"HPOUTR", NULL, "HPR Out"}, 185 {"LINEOUT", NULL, "LINEOUT Mixer"}, 186 187 {"HPL Out", NULL, "Headphone Enable"}, 188 {"HPR Out", NULL, "Headphone Enable"}, 189 190 {"Headphone Enable", "Switch", "DACH"}, 191 192 {"DACH", NULL, "DAC"}, 193 194 {"LINEOUT Mixer", "DACL", "DAC"}, 195 }; 196 197 /* 198 * ak4642 register cache 199 */ 200 static const u8 ak4642_reg[] = { 201 0x00, 0x00, 0x01, 0x00, 202 0x02, 0x00, 0x00, 0x00, 203 0xe1, 0xe1, 0x18, 0x00, 204 0xe1, 0x18, 0x11, 0x08, 205 0x00, 0x00, 0x00, 0x00, 206 0x00, 0x00, 0x00, 0x00, 207 0x00, 0x00, 0x00, 0x00, 208 0x00, 0x00, 0x00, 0x00, 209 0x00, 0x00, 0x00, 0x00, 210 0x00, 211 }; 212 213 static const u8 ak4648_reg[] = { 214 0x00, 0x00, 0x01, 0x00, 215 0x02, 0x00, 0x00, 0x00, 216 0xe1, 0xe1, 0x18, 0x00, 217 0xe1, 0x18, 0x11, 0xb8, 218 0x00, 0x00, 0x00, 0x00, 219 0x00, 0x00, 0x00, 0x00, 220 0x00, 0x00, 0x00, 0x00, 221 0x00, 0x00, 0x00, 0x00, 222 0x00, 0x00, 0x00, 0x00, 223 0x00, 0x88, 0x88, 0x08, 224 }; 225 226 static int ak4642_dai_startup(struct snd_pcm_substream *substream, 227 struct snd_soc_dai *dai) 228 { 229 int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; 230 struct snd_soc_codec *codec = dai->codec; 231 232 if (is_play) { 233 /* 234 * start headphone output 235 * 236 * PLL, Master Mode 237 * Audio I/F Format :MSB justified (ADC & DAC) 238 * Bass Boost Level : Middle 239 * 240 * This operation came from example code of 241 * "ASAHI KASEI AK4642" (japanese) manual p97. 242 */ 243 snd_soc_write(codec, L_IVC, 0x91); /* volume */ 244 snd_soc_write(codec, R_IVC, 0x91); /* volume */ 245 } else { 246 /* 247 * start stereo input 248 * 249 * PLL Master Mode 250 * Audio I/F Format:MSB justified (ADC & DAC) 251 * Pre MIC AMP:+20dB 252 * MIC Power On 253 * ALC setting:Refer to Table 35 254 * ALC bit=“1” 255 * 256 * This operation came from example code of 257 * "ASAHI KASEI AK4642" (japanese) manual p94. 258 */ 259 snd_soc_write(codec, SG_SL1, PMMP | MGAIN0); 260 snd_soc_write(codec, TIMER, ZTM(0x3) | WTM(0x3)); 261 snd_soc_write(codec, ALC_CTL1, ALC | LMTH0); 262 snd_soc_update_bits(codec, PW_MGMT1, PMADL, PMADL); 263 snd_soc_update_bits(codec, PW_MGMT3, PMADR, PMADR); 264 } 265 266 return 0; 267 } 268 269 static void ak4642_dai_shutdown(struct snd_pcm_substream *substream, 270 struct snd_soc_dai *dai) 271 { 272 int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; 273 struct snd_soc_codec *codec = dai->codec; 274 275 if (is_play) { 276 } else { 277 /* stop stereo input */ 278 snd_soc_update_bits(codec, PW_MGMT1, PMADL, 0); 279 snd_soc_update_bits(codec, PW_MGMT3, PMADR, 0); 280 snd_soc_update_bits(codec, ALC_CTL1, ALC, 0); 281 } 282 } 283 284 static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai, 285 int clk_id, unsigned int freq, int dir) 286 { 287 struct snd_soc_codec *codec = codec_dai->codec; 288 u8 pll; 289 290 switch (freq) { 291 case 11289600: 292 pll = PLL2; 293 break; 294 case 12288000: 295 pll = PLL2 | PLL0; 296 break; 297 case 12000000: 298 pll = PLL2 | PLL1; 299 break; 300 case 24000000: 301 pll = PLL2 | PLL1 | PLL0; 302 break; 303 case 13500000: 304 pll = PLL3 | PLL2; 305 break; 306 case 27000000: 307 pll = PLL3 | PLL2 | PLL0; 308 break; 309 default: 310 return -EINVAL; 311 } 312 snd_soc_update_bits(codec, MD_CTL1, PLL_MASK, pll); 313 314 return 0; 315 } 316 317 static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) 318 { 319 struct snd_soc_codec *codec = dai->codec; 320 u8 data; 321 u8 bcko; 322 323 data = MCKO | PMPLL; /* use MCKO */ 324 bcko = 0; 325 326 /* set master/slave audio interface */ 327 switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { 328 case SND_SOC_DAIFMT_CBM_CFM: 329 data |= MS; 330 bcko = BCKO_64; 331 break; 332 case SND_SOC_DAIFMT_CBS_CFS: 333 break; 334 default: 335 return -EINVAL; 336 } 337 snd_soc_update_bits(codec, PW_MGMT2, MS | MCKO | PMPLL, data); 338 snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko); 339 340 /* format type */ 341 data = 0; 342 switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { 343 case SND_SOC_DAIFMT_LEFT_J: 344 data = LEFT_J; 345 break; 346 case SND_SOC_DAIFMT_I2S: 347 data = I2S; 348 break; 349 /* FIXME 350 * Please add RIGHT_J / DSP support here 351 */ 352 default: 353 return -EINVAL; 354 break; 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 break; 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 snd_soc_cache_sync(codec); 459 return 0; 460 } 461 462 463 static int ak4642_probe(struct snd_soc_codec *codec) 464 { 465 int ret; 466 467 ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); 468 if (ret < 0) { 469 dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); 470 return ret; 471 } 472 473 snd_soc_add_codec_controls(codec, ak4642_snd_controls, 474 ARRAY_SIZE(ak4642_snd_controls)); 475 476 ak4642_set_bias_level(codec, SND_SOC_BIAS_STANDBY); 477 478 return 0; 479 } 480 481 static int ak4642_remove(struct snd_soc_codec *codec) 482 { 483 ak4642_set_bias_level(codec, SND_SOC_BIAS_OFF); 484 return 0; 485 } 486 487 static struct snd_soc_codec_driver soc_codec_dev_ak4642 = { 488 .probe = ak4642_probe, 489 .remove = ak4642_remove, 490 .resume = ak4642_resume, 491 .set_bias_level = ak4642_set_bias_level, 492 .reg_cache_default = ak4642_reg, /* ak4642 reg */ 493 .reg_cache_size = ARRAY_SIZE(ak4642_reg), /* ak4642 reg */ 494 .reg_word_size = sizeof(u8), 495 .dapm_widgets = ak4642_dapm_widgets, 496 .num_dapm_widgets = ARRAY_SIZE(ak4642_dapm_widgets), 497 .dapm_routes = ak4642_intercon, 498 .num_dapm_routes = ARRAY_SIZE(ak4642_intercon), 499 }; 500 501 static struct snd_soc_codec_driver soc_codec_dev_ak4648 = { 502 .probe = ak4642_probe, 503 .remove = ak4642_remove, 504 .resume = ak4642_resume, 505 .set_bias_level = ak4642_set_bias_level, 506 .reg_cache_default = ak4648_reg, /* ak4648 reg */ 507 .reg_cache_size = ARRAY_SIZE(ak4648_reg), /* ak4648 reg */ 508 .reg_word_size = sizeof(u8), 509 .dapm_widgets = ak4642_dapm_widgets, 510 .num_dapm_widgets = ARRAY_SIZE(ak4642_dapm_widgets), 511 .dapm_routes = ak4642_intercon, 512 .num_dapm_routes = ARRAY_SIZE(ak4642_intercon), 513 }; 514 515 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) 516 static int ak4642_i2c_probe(struct i2c_client *i2c, 517 const struct i2c_device_id *id) 518 { 519 return snd_soc_register_codec(&i2c->dev, 520 (struct snd_soc_codec_driver *)id->driver_data, 521 &ak4642_dai, 1); 522 } 523 524 static int ak4642_i2c_remove(struct i2c_client *client) 525 { 526 snd_soc_unregister_codec(&client->dev); 527 return 0; 528 } 529 530 static const struct i2c_device_id ak4642_i2c_id[] = { 531 { "ak4642", (kernel_ulong_t)&soc_codec_dev_ak4642 }, 532 { "ak4643", (kernel_ulong_t)&soc_codec_dev_ak4642 }, 533 { "ak4648", (kernel_ulong_t)&soc_codec_dev_ak4648 }, 534 { } 535 }; 536 MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id); 537 538 static struct i2c_driver ak4642_i2c_driver = { 539 .driver = { 540 .name = "ak4642-codec", 541 .owner = THIS_MODULE, 542 }, 543 .probe = ak4642_i2c_probe, 544 .remove = ak4642_i2c_remove, 545 .id_table = ak4642_i2c_id, 546 }; 547 #endif 548 549 static int __init ak4642_modinit(void) 550 { 551 int ret = 0; 552 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) 553 ret = i2c_add_driver(&ak4642_i2c_driver); 554 #endif 555 return ret; 556 557 } 558 module_init(ak4642_modinit); 559 560 static void __exit ak4642_exit(void) 561 { 562 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) 563 i2c_del_driver(&ak4642_i2c_driver); 564 #endif 565 566 } 567 module_exit(ak4642_exit); 568 569 MODULE_DESCRIPTION("Soc AK4642 driver"); 570 MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>"); 571 MODULE_LICENSE("GPL"); 572