sound/soc/codecs/ak4642.c

   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");