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, -11500, 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 /* codec private data */ 198 struct ak4642_priv { 199 unsigned int sysclk; 200 enum snd_soc_control_type control_type; 201 }; 202 203 /* 204 * ak4642 register cache 205 */ 206 static const u8 ak4642_reg[] = { 207 0x00, 0x00, 0x01, 0x00, 208 0x02, 0x00, 0x00, 0x00, 209 0xe1, 0xe1, 0x18, 0x00, 210 0xe1, 0x18, 0x11, 0x08, 211 0x00, 0x00, 0x00, 0x00, 212 0x00, 0x00, 0x00, 0x00, 213 0x00, 0x00, 0x00, 0x00, 214 0x00, 0x00, 0x00, 0x00, 215 0x00, 0x00, 0x00, 0x00, 216 0x00, 217 }; 218 219 static const u8 ak4648_reg[] = { 220 0x00, 0x00, 0x01, 0x00, 221 0x02, 0x00, 0x00, 0x00, 222 0xe1, 0xe1, 0x18, 0x00, 223 0xe1, 0x18, 0x11, 0xb8, 224 0x00, 0x00, 0x00, 0x00, 225 0x00, 0x00, 0x00, 0x00, 226 0x00, 0x00, 0x00, 0x00, 227 0x00, 0x00, 0x00, 0x00, 228 0x00, 0x00, 0x00, 0x00, 229 0x00, 0x88, 0x88, 0x08, 230 }; 231 232 static int ak4642_dai_startup(struct snd_pcm_substream *substream, 233 struct snd_soc_dai *dai) 234 { 235 int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; 236 struct snd_soc_codec *codec = dai->codec; 237 238 if (is_play) { 239 /* 240 * start headphone output 241 * 242 * PLL, Master Mode 243 * Audio I/F Format :MSB justified (ADC & DAC) 244 * Bass Boost Level : Middle 245 * 246 * This operation came from example code of 247 * "ASAHI KASEI AK4642" (japanese) manual p97. 248 */ 249 snd_soc_write(codec, L_IVC, 0x91); /* volume */ 250 snd_soc_write(codec, R_IVC, 0x91); /* volume */ 251 } else { 252 /* 253 * start stereo input 254 * 255 * PLL Master Mode 256 * Audio I/F Format:MSB justified (ADC & DAC) 257 * Pre MIC AMP:+20dB 258 * MIC Power On 259 * ALC setting:Refer to Table 35 260 * ALC bit=“1” 261 * 262 * This operation came from example code of 263 * "ASAHI KASEI AK4642" (japanese) manual p94. 264 */ 265 snd_soc_write(codec, SG_SL1, PMMP | MGAIN0); 266 snd_soc_write(codec, TIMER, ZTM(0x3) | WTM(0x3)); 267 snd_soc_write(codec, ALC_CTL1, ALC | LMTH0); 268 snd_soc_update_bits(codec, PW_MGMT1, PMADL, PMADL); 269 snd_soc_update_bits(codec, PW_MGMT3, PMADR, PMADR); 270 } 271 272 return 0; 273 } 274 275 static void ak4642_dai_shutdown(struct snd_pcm_substream *substream, 276 struct snd_soc_dai *dai) 277 { 278 int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; 279 struct snd_soc_codec *codec = dai->codec; 280 281 if (is_play) { 282 } else { 283 /* stop stereo input */ 284 snd_soc_update_bits(codec, PW_MGMT1, PMADL, 0); 285 snd_soc_update_bits(codec, PW_MGMT3, PMADR, 0); 286 snd_soc_update_bits(codec, ALC_CTL1, ALC, 0); 287 } 288 } 289 290 static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai, 291 int clk_id, unsigned int freq, int dir) 292 { 293 struct snd_soc_codec *codec = codec_dai->codec; 294 u8 pll; 295 296 switch (freq) { 297 case 11289600: 298 pll = PLL2; 299 break; 300 case 12288000: 301 pll = PLL2 | PLL0; 302 break; 303 case 12000000: 304 pll = PLL2 | PLL1; 305 break; 306 case 24000000: 307 pll = PLL2 | PLL1 | PLL0; 308 break; 309 case 13500000: 310 pll = PLL3 | PLL2; 311 break; 312 case 27000000: 313 pll = PLL3 | PLL2 | PLL0; 314 break; 315 default: 316 return -EINVAL; 317 } 318 snd_soc_update_bits(codec, MD_CTL1, PLL_MASK, pll); 319 320 return 0; 321 } 322 323 static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) 324 { 325 struct snd_soc_codec *codec = dai->codec; 326 u8 data; 327 u8 bcko; 328 329 data = MCKO | PMPLL; /* use MCKO */ 330 bcko = 0; 331 332 /* set master/slave audio interface */ 333 switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { 334 case SND_SOC_DAIFMT_CBM_CFM: 335 data |= MS; 336 bcko = BCKO_64; 337 break; 338 case SND_SOC_DAIFMT_CBS_CFS: 339 break; 340 default: 341 return -EINVAL; 342 } 343 snd_soc_update_bits(codec, PW_MGMT2, MS | MCKO | PMPLL, data); 344 snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko); 345 346 /* format type */ 347 data = 0; 348 switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { 349 case SND_SOC_DAIFMT_LEFT_J: 350 data = LEFT_J; 351 break; 352 case SND_SOC_DAIFMT_I2S: 353 data = I2S; 354 break; 355 /* FIXME 356 * Please add RIGHT_J / DSP support here 357 */ 358 default: 359 return -EINVAL; 360 break; 361 } 362 snd_soc_update_bits(codec, MD_CTL1, DIF_MASK, data); 363 364 return 0; 365 } 366 367 static int ak4642_dai_hw_params(struct snd_pcm_substream *substream, 368 struct snd_pcm_hw_params *params, 369 struct snd_soc_dai *dai) 370 { 371 struct snd_soc_codec *codec = dai->codec; 372 u8 rate; 373 374 switch (params_rate(params)) { 375 case 7350: 376 rate = FS2; 377 break; 378 case 8000: 379 rate = 0; 380 break; 381 case 11025: 382 rate = FS2 | FS0; 383 break; 384 case 12000: 385 rate = FS0; 386 break; 387 case 14700: 388 rate = FS2 | FS1; 389 break; 390 case 16000: 391 rate = FS1; 392 break; 393 case 22050: 394 rate = FS2 | FS1 | FS0; 395 break; 396 case 24000: 397 rate = FS1 | FS0; 398 break; 399 case 29400: 400 rate = FS3 | FS2 | FS1; 401 break; 402 case 32000: 403 rate = FS3 | FS1; 404 break; 405 case 44100: 406 rate = FS3 | FS2 | FS1 | FS0; 407 break; 408 case 48000: 409 rate = FS3 | FS1 | FS0; 410 break; 411 default: 412 return -EINVAL; 413 break; 414 } 415 snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate); 416 417 return 0; 418 } 419 420 static int ak4642_set_bias_level(struct snd_soc_codec *codec, 421 enum snd_soc_bias_level level) 422 { 423 switch (level) { 424 case SND_SOC_BIAS_OFF: 425 snd_soc_write(codec, PW_MGMT1, 0x00); 426 break; 427 default: 428 snd_soc_update_bits(codec, PW_MGMT1, PMVCM, PMVCM); 429 break; 430 } 431 codec->dapm.bias_level = level; 432 433 return 0; 434 } 435 436 static const struct snd_soc_dai_ops ak4642_dai_ops = { 437 .startup = ak4642_dai_startup, 438 .shutdown = ak4642_dai_shutdown, 439 .set_sysclk = ak4642_dai_set_sysclk, 440 .set_fmt = ak4642_dai_set_fmt, 441 .hw_params = ak4642_dai_hw_params, 442 }; 443 444 static struct snd_soc_dai_driver ak4642_dai = { 445 .name = "ak4642-hifi", 446 .playback = { 447 .stream_name = "Playback", 448 .channels_min = 1, 449 .channels_max = 2, 450 .rates = SNDRV_PCM_RATE_8000_48000, 451 .formats = SNDRV_PCM_FMTBIT_S16_LE }, 452 .capture = { 453 .stream_name = "Capture", 454 .channels_min = 1, 455 .channels_max = 2, 456 .rates = SNDRV_PCM_RATE_8000_48000, 457 .formats = SNDRV_PCM_FMTBIT_S16_LE }, 458 .ops = &ak4642_dai_ops, 459 .symmetric_rates = 1, 460 }; 461 462 static int ak4642_resume(struct snd_soc_codec *codec) 463 { 464 snd_soc_cache_sync(codec); 465 return 0; 466 } 467 468 469 static int ak4642_probe(struct snd_soc_codec *codec) 470 { 471 struct ak4642_priv *ak4642 = snd_soc_codec_get_drvdata(codec); 472 int ret; 473 474 ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4642->control_type); 475 if (ret < 0) { 476 dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); 477 return ret; 478 } 479 480 snd_soc_add_controls(codec, ak4642_snd_controls, 481 ARRAY_SIZE(ak4642_snd_controls)); 482 483 ak4642_set_bias_level(codec, SND_SOC_BIAS_STANDBY); 484 485 return 0; 486 } 487 488 static int ak4642_remove(struct snd_soc_codec *codec) 489 { 490 ak4642_set_bias_level(codec, SND_SOC_BIAS_OFF); 491 return 0; 492 } 493 494 static struct snd_soc_codec_driver soc_codec_dev_ak4642 = { 495 .probe = ak4642_probe, 496 .remove = ak4642_remove, 497 .resume = ak4642_resume, 498 .set_bias_level = ak4642_set_bias_level, 499 .reg_cache_default = ak4642_reg, /* ak4642 reg */ 500 .reg_cache_size = ARRAY_SIZE(ak4642_reg), /* ak4642 reg */ 501 .reg_word_size = sizeof(u8), 502 .dapm_widgets = ak4642_dapm_widgets, 503 .num_dapm_widgets = ARRAY_SIZE(ak4642_dapm_widgets), 504 .dapm_routes = ak4642_intercon, 505 .num_dapm_routes = ARRAY_SIZE(ak4642_intercon), 506 }; 507 508 static struct snd_soc_codec_driver soc_codec_dev_ak4648 = { 509 .probe = ak4642_probe, 510 .remove = ak4642_remove, 511 .resume = ak4642_resume, 512 .set_bias_level = ak4642_set_bias_level, 513 .reg_cache_default = ak4648_reg, /* ak4648 reg */ 514 .reg_cache_size = ARRAY_SIZE(ak4648_reg), /* ak4648 reg */ 515 .reg_word_size = sizeof(u8), 516 .dapm_widgets = ak4642_dapm_widgets, 517 .num_dapm_widgets = ARRAY_SIZE(ak4642_dapm_widgets), 518 .dapm_routes = ak4642_intercon, 519 .num_dapm_routes = ARRAY_SIZE(ak4642_intercon), 520 }; 521 522 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) 523 static __devinit int ak4642_i2c_probe(struct i2c_client *i2c, 524 const struct i2c_device_id *id) 525 { 526 struct ak4642_priv *ak4642; 527 int ret; 528 529 ak4642 = devm_kzalloc(&i2c->dev, sizeof(struct ak4642_priv), 530 GFP_KERNEL); 531 if (!ak4642) 532 return -ENOMEM; 533 534 i2c_set_clientdata(i2c, ak4642); 535 ak4642->control_type = SND_SOC_I2C; 536 537 ret = snd_soc_register_codec(&i2c->dev, 538 (struct snd_soc_codec_driver *)id->driver_data, 539 &ak4642_dai, 1); 540 return ret; 541 } 542 543 static __devexit int ak4642_i2c_remove(struct i2c_client *client) 544 { 545 snd_soc_unregister_codec(&client->dev); 546 return 0; 547 } 548 549 static const struct i2c_device_id ak4642_i2c_id[] = { 550 { "ak4642", (kernel_ulong_t)&soc_codec_dev_ak4642 }, 551 { "ak4643", (kernel_ulong_t)&soc_codec_dev_ak4642 }, 552 { "ak4648", (kernel_ulong_t)&soc_codec_dev_ak4648 }, 553 { } 554 }; 555 MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id); 556 557 static struct i2c_driver ak4642_i2c_driver = { 558 .driver = { 559 .name = "ak4642-codec", 560 .owner = THIS_MODULE, 561 }, 562 .probe = ak4642_i2c_probe, 563 .remove = __devexit_p(ak4642_i2c_remove), 564 .id_table = ak4642_i2c_id, 565 }; 566 #endif 567 568 static int __init ak4642_modinit(void) 569 { 570 int ret = 0; 571 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) 572 ret = i2c_add_driver(&ak4642_i2c_driver); 573 #endif 574 return ret; 575 576 } 577 module_init(ak4642_modinit); 578 579 static void __exit ak4642_exit(void) 580 { 581 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) 582 i2c_del_driver(&ak4642_i2c_driver); 583 #endif 584 585 } 586 module_exit(ak4642_exit); 587 588 MODULE_DESCRIPTION("Soc AK4642 driver"); 589 MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>"); 590 MODULE_LICENSE("GPL"); 591