1 // SPDX-License-Identifier: GPL-2.0 2 // 3 // Freescale Generic ASoC Sound Card driver with ASRC 4 // 5 // Copyright (C) 2014 Freescale Semiconductor, Inc. 6 // 7 // Author: Nicolin Chen <nicoleotsuka@gmail.com> 8 9 #include <linux/clk.h> 10 #include <linux/i2c.h> 11 #include <linux/module.h> 12 #include <linux/of_platform.h> 13 #if IS_ENABLED(CONFIG_SND_AC97_CODEC) 14 #include <sound/ac97_codec.h> 15 #endif 16 #include <sound/pcm_params.h> 17 #include <sound/soc.h> 18 #include <sound/jack.h> 19 #include <sound/simple_card_utils.h> 20 21 #include "fsl_esai.h" 22 #include "fsl_sai.h" 23 #include "imx-audmux.h" 24 25 #include "../codecs/sgtl5000.h" 26 #include "../codecs/wm8962.h" 27 #include "../codecs/wm8960.h" 28 #include "../codecs/wm8994.h" 29 #include "../codecs/tlv320aic31xx.h" 30 #include "../codecs/nau8822.h" 31 32 #define DRIVER_NAME "fsl-asoc-card" 33 34 #define CS427x_SYSCLK_MCLK 0 35 36 #define RX 0 37 #define TX 1 38 39 /* Default DAI format without Master and Slave flag */ 40 #define DAI_FMT_BASE (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF) 41 42 /** 43 * struct codec_priv - CODEC private data 44 * @mclk_freq: Clock rate of MCLK 45 * @free_freq: Clock rate of MCLK for hw_free() 46 * @mclk_id: MCLK (or main clock) id for set_sysclk() 47 * @fll_id: FLL (or secordary clock) id for set_sysclk() 48 * @pll_id: PLL id for set_pll() 49 */ 50 struct codec_priv { 51 struct clk *mclk; 52 unsigned long mclk_freq; 53 unsigned long free_freq; 54 u32 mclk_id; 55 u32 fll_id; 56 u32 pll_id; 57 }; 58 59 /** 60 * struct cpu_priv - CPU private data 61 * @sysclk_freq: SYSCLK rates for set_sysclk() 62 * @sysclk_dir: SYSCLK directions for set_sysclk() 63 * @sysclk_id: SYSCLK ids for set_sysclk() 64 * @slot_width: Slot width of each frame 65 * @slot_num: Number of slots of each frame 66 * 67 * Note: [1] for tx and [0] for rx 68 */ 69 struct cpu_priv { 70 unsigned long sysclk_freq[2]; 71 u32 sysclk_dir[2]; 72 u32 sysclk_id[2]; 73 u32 slot_width; 74 u32 slot_num; 75 }; 76 77 /** 78 * struct fsl_asoc_card_priv - Freescale Generic ASOC card private data 79 * @dai_link: DAI link structure including normal one and DPCM link 80 * @hp_jack: Headphone Jack structure 81 * @mic_jack: Microphone Jack structure 82 * @pdev: platform device pointer 83 * @codec_priv: CODEC private data 84 * @cpu_priv: CPU private data 85 * @card: ASoC card structure 86 * @streams: Mask of current active streams 87 * @sample_rate: Current sample rate 88 * @sample_format: Current sample format 89 * @asrc_rate: ASRC sample rate used by Back-Ends 90 * @asrc_format: ASRC sample format used by Back-Ends 91 * @dai_fmt: DAI format between CPU and CODEC 92 * @name: Card name 93 */ 94 95 struct fsl_asoc_card_priv { 96 struct snd_soc_dai_link dai_link[3]; 97 struct asoc_simple_jack hp_jack; 98 struct asoc_simple_jack mic_jack; 99 struct platform_device *pdev; 100 struct codec_priv codec_priv; 101 struct cpu_priv cpu_priv; 102 struct snd_soc_card card; 103 u8 streams; 104 u32 sample_rate; 105 snd_pcm_format_t sample_format; 106 u32 asrc_rate; 107 snd_pcm_format_t asrc_format; 108 u32 dai_fmt; 109 char name[32]; 110 }; 111 112 /* 113 * This dapm route map exists for DPCM link only. 114 * The other routes shall go through Device Tree. 115 * 116 * Note: keep all ASRC routes in the second half 117 * to drop them easily for non-ASRC cases. 118 */ 119 static const struct snd_soc_dapm_route audio_map[] = { 120 /* 1st half -- Normal DAPM routes */ 121 {"Playback", NULL, "CPU-Playback"}, 122 {"CPU-Capture", NULL, "Capture"}, 123 /* 2nd half -- ASRC DAPM routes */ 124 {"CPU-Playback", NULL, "ASRC-Playback"}, 125 {"ASRC-Capture", NULL, "CPU-Capture"}, 126 }; 127 128 static const struct snd_soc_dapm_route audio_map_ac97[] = { 129 /* 1st half -- Normal DAPM routes */ 130 {"AC97 Playback", NULL, "CPU AC97 Playback"}, 131 {"CPU AC97 Capture", NULL, "AC97 Capture"}, 132 /* 2nd half -- ASRC DAPM routes */ 133 {"CPU AC97 Playback", NULL, "ASRC-Playback"}, 134 {"ASRC-Capture", NULL, "CPU AC97 Capture"}, 135 }; 136 137 static const struct snd_soc_dapm_route audio_map_tx[] = { 138 /* 1st half -- Normal DAPM routes */ 139 {"Playback", NULL, "CPU-Playback"}, 140 /* 2nd half -- ASRC DAPM routes */ 141 {"CPU-Playback", NULL, "ASRC-Playback"}, 142 }; 143 144 static const struct snd_soc_dapm_route audio_map_rx[] = { 145 /* 1st half -- Normal DAPM routes */ 146 {"CPU-Capture", NULL, "Capture"}, 147 /* 2nd half -- ASRC DAPM routes */ 148 {"ASRC-Capture", NULL, "CPU-Capture"}, 149 }; 150 151 /* Add all possible widgets into here without being redundant */ 152 static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = { 153 SND_SOC_DAPM_LINE("Line Out Jack", NULL), 154 SND_SOC_DAPM_LINE("Line In Jack", NULL), 155 SND_SOC_DAPM_HP("Headphone Jack", NULL), 156 SND_SOC_DAPM_SPK("Ext Spk", NULL), 157 SND_SOC_DAPM_MIC("Mic Jack", NULL), 158 SND_SOC_DAPM_MIC("AMIC", NULL), 159 SND_SOC_DAPM_MIC("DMIC", NULL), 160 }; 161 162 static bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv) 163 { 164 return priv->dai_fmt == SND_SOC_DAIFMT_AC97; 165 } 166 167 static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, 168 struct snd_pcm_hw_params *params) 169 { 170 struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); 171 struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); 172 bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; 173 struct codec_priv *codec_priv = &priv->codec_priv; 174 struct cpu_priv *cpu_priv = &priv->cpu_priv; 175 struct device *dev = rtd->card->dev; 176 unsigned int pll_out; 177 int ret; 178 179 priv->sample_rate = params_rate(params); 180 priv->sample_format = params_format(params); 181 priv->streams |= BIT(substream->stream); 182 183 if (fsl_asoc_card_is_ac97(priv)) 184 return 0; 185 186 /* Specific configurations of DAIs starts from here */ 187 ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), cpu_priv->sysclk_id[tx], 188 cpu_priv->sysclk_freq[tx], 189 cpu_priv->sysclk_dir[tx]); 190 if (ret && ret != -ENOTSUPP) { 191 dev_err(dev, "failed to set sysclk for cpu dai\n"); 192 goto fail; 193 } 194 195 if (cpu_priv->slot_width) { 196 if (!cpu_priv->slot_num) 197 cpu_priv->slot_num = 2; 198 199 ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 200 cpu_priv->slot_num, 201 cpu_priv->slot_width); 202 if (ret && ret != -ENOTSUPP) { 203 dev_err(dev, "failed to set TDM slot for cpu dai\n"); 204 goto fail; 205 } 206 } 207 208 /* Specific configuration for PLL */ 209 if (codec_priv->pll_id && codec_priv->fll_id) { 210 if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) 211 pll_out = priv->sample_rate * 384; 212 else 213 pll_out = priv->sample_rate * 256; 214 215 ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0), 216 codec_priv->pll_id, 217 codec_priv->mclk_id, 218 codec_priv->mclk_freq, pll_out); 219 if (ret) { 220 dev_err(dev, "failed to start FLL: %d\n", ret); 221 goto fail; 222 } 223 224 ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), 225 codec_priv->fll_id, 226 pll_out, SND_SOC_CLOCK_IN); 227 228 if (ret && ret != -ENOTSUPP) { 229 dev_err(dev, "failed to set SYSCLK: %d\n", ret); 230 goto fail; 231 } 232 } 233 234 return 0; 235 236 fail: 237 priv->streams &= ~BIT(substream->stream); 238 return ret; 239 } 240 241 static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream) 242 { 243 struct snd_soc_pcm_runtime *rtd = substream->private_data; 244 struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); 245 struct codec_priv *codec_priv = &priv->codec_priv; 246 struct device *dev = rtd->card->dev; 247 int ret; 248 249 priv->streams &= ~BIT(substream->stream); 250 251 if (!priv->streams && codec_priv->pll_id && codec_priv->fll_id) { 252 /* Force freq to be free_freq to avoid error message in codec */ 253 ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), 254 codec_priv->mclk_id, 255 codec_priv->free_freq, 256 SND_SOC_CLOCK_IN); 257 if (ret) { 258 dev_err(dev, "failed to switch away from FLL: %d\n", ret); 259 return ret; 260 } 261 262 ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0), 263 codec_priv->pll_id, 0, 0, 0); 264 if (ret && ret != -ENOTSUPP) { 265 dev_err(dev, "failed to stop FLL: %d\n", ret); 266 return ret; 267 } 268 } 269 270 return 0; 271 } 272 273 static const struct snd_soc_ops fsl_asoc_card_ops = { 274 .hw_params = fsl_asoc_card_hw_params, 275 .hw_free = fsl_asoc_card_hw_free, 276 }; 277 278 static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, 279 struct snd_pcm_hw_params *params) 280 { 281 struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); 282 struct snd_interval *rate; 283 struct snd_mask *mask; 284 285 rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); 286 rate->max = rate->min = priv->asrc_rate; 287 288 mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); 289 snd_mask_none(mask); 290 snd_mask_set_format(mask, priv->asrc_format); 291 292 return 0; 293 } 294 295 SND_SOC_DAILINK_DEFS(hifi, 296 DAILINK_COMP_ARRAY(COMP_EMPTY()), 297 DAILINK_COMP_ARRAY(COMP_EMPTY()), 298 DAILINK_COMP_ARRAY(COMP_EMPTY())); 299 300 SND_SOC_DAILINK_DEFS(hifi_fe, 301 DAILINK_COMP_ARRAY(COMP_EMPTY()), 302 DAILINK_COMP_ARRAY(COMP_DUMMY()), 303 DAILINK_COMP_ARRAY(COMP_EMPTY())); 304 305 SND_SOC_DAILINK_DEFS(hifi_be, 306 DAILINK_COMP_ARRAY(COMP_EMPTY()), 307 DAILINK_COMP_ARRAY(COMP_EMPTY()), 308 DAILINK_COMP_ARRAY(COMP_DUMMY())); 309 310 static const struct snd_soc_dai_link fsl_asoc_card_dai[] = { 311 /* Default ASoC DAI Link*/ 312 { 313 .name = "HiFi", 314 .stream_name = "HiFi", 315 .ops = &fsl_asoc_card_ops, 316 SND_SOC_DAILINK_REG(hifi), 317 }, 318 /* DPCM Link between Front-End and Back-End (Optional) */ 319 { 320 .name = "HiFi-ASRC-FE", 321 .stream_name = "HiFi-ASRC-FE", 322 .dpcm_playback = 1, 323 .dpcm_capture = 1, 324 .dynamic = 1, 325 SND_SOC_DAILINK_REG(hifi_fe), 326 }, 327 { 328 .name = "HiFi-ASRC-BE", 329 .stream_name = "HiFi-ASRC-BE", 330 .be_hw_params_fixup = be_hw_params_fixup, 331 .ops = &fsl_asoc_card_ops, 332 .dpcm_playback = 1, 333 .dpcm_capture = 1, 334 .no_pcm = 1, 335 SND_SOC_DAILINK_REG(hifi_be), 336 }, 337 }; 338 339 static int fsl_asoc_card_audmux_init(struct device_node *np, 340 struct fsl_asoc_card_priv *priv) 341 { 342 struct device *dev = &priv->pdev->dev; 343 u32 int_ptcr = 0, ext_ptcr = 0; 344 int int_port, ext_port; 345 int ret; 346 347 ret = of_property_read_u32(np, "mux-int-port", &int_port); 348 if (ret) { 349 dev_err(dev, "mux-int-port missing or invalid\n"); 350 return ret; 351 } 352 ret = of_property_read_u32(np, "mux-ext-port", &ext_port); 353 if (ret) { 354 dev_err(dev, "mux-ext-port missing or invalid\n"); 355 return ret; 356 } 357 358 /* 359 * The port numbering in the hardware manual starts at 1, while 360 * the AUDMUX API expects it starts at 0. 361 */ 362 int_port--; 363 ext_port--; 364 365 /* 366 * Use asynchronous mode (6 wires) for all cases except AC97. 367 * If only 4 wires are needed, just set SSI into 368 * synchronous mode and enable 4 PADs in IOMUX. 369 */ 370 switch (priv->dai_fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { 371 case SND_SOC_DAIFMT_CBP_CFP: 372 int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) | 373 IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) | 374 IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | 375 IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | 376 IMX_AUDMUX_V2_PTCR_RFSDIR | 377 IMX_AUDMUX_V2_PTCR_RCLKDIR | 378 IMX_AUDMUX_V2_PTCR_TFSDIR | 379 IMX_AUDMUX_V2_PTCR_TCLKDIR; 380 break; 381 case SND_SOC_DAIFMT_CBP_CFC: 382 int_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) | 383 IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | 384 IMX_AUDMUX_V2_PTCR_RCLKDIR | 385 IMX_AUDMUX_V2_PTCR_TCLKDIR; 386 ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) | 387 IMX_AUDMUX_V2_PTCR_TFSEL(int_port) | 388 IMX_AUDMUX_V2_PTCR_RFSDIR | 389 IMX_AUDMUX_V2_PTCR_TFSDIR; 390 break; 391 case SND_SOC_DAIFMT_CBC_CFP: 392 int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) | 393 IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | 394 IMX_AUDMUX_V2_PTCR_RFSDIR | 395 IMX_AUDMUX_V2_PTCR_TFSDIR; 396 ext_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) | 397 IMX_AUDMUX_V2_PTCR_TCSEL(int_port) | 398 IMX_AUDMUX_V2_PTCR_RCLKDIR | 399 IMX_AUDMUX_V2_PTCR_TCLKDIR; 400 break; 401 case SND_SOC_DAIFMT_CBC_CFC: 402 ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) | 403 IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) | 404 IMX_AUDMUX_V2_PTCR_TFSEL(int_port) | 405 IMX_AUDMUX_V2_PTCR_TCSEL(int_port) | 406 IMX_AUDMUX_V2_PTCR_RFSDIR | 407 IMX_AUDMUX_V2_PTCR_RCLKDIR | 408 IMX_AUDMUX_V2_PTCR_TFSDIR | 409 IMX_AUDMUX_V2_PTCR_TCLKDIR; 410 break; 411 default: 412 if (!fsl_asoc_card_is_ac97(priv)) 413 return -EINVAL; 414 } 415 416 if (fsl_asoc_card_is_ac97(priv)) { 417 int_ptcr = IMX_AUDMUX_V2_PTCR_SYN | 418 IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | 419 IMX_AUDMUX_V2_PTCR_TCLKDIR; 420 ext_ptcr = IMX_AUDMUX_V2_PTCR_SYN | 421 IMX_AUDMUX_V2_PTCR_TFSEL(int_port) | 422 IMX_AUDMUX_V2_PTCR_TFSDIR; 423 } 424 425 /* Asynchronous mode can not be set along with RCLKDIR */ 426 if (!fsl_asoc_card_is_ac97(priv)) { 427 unsigned int pdcr = 428 IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port); 429 430 ret = imx_audmux_v2_configure_port(int_port, 0, 431 pdcr); 432 if (ret) { 433 dev_err(dev, "audmux internal port setup failed\n"); 434 return ret; 435 } 436 } 437 438 ret = imx_audmux_v2_configure_port(int_port, int_ptcr, 439 IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); 440 if (ret) { 441 dev_err(dev, "audmux internal port setup failed\n"); 442 return ret; 443 } 444 445 if (!fsl_asoc_card_is_ac97(priv)) { 446 unsigned int pdcr = 447 IMX_AUDMUX_V2_PDCR_RXDSEL(int_port); 448 449 ret = imx_audmux_v2_configure_port(ext_port, 0, 450 pdcr); 451 if (ret) { 452 dev_err(dev, "audmux external port setup failed\n"); 453 return ret; 454 } 455 } 456 457 ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr, 458 IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); 459 if (ret) { 460 dev_err(dev, "audmux external port setup failed\n"); 461 return ret; 462 } 463 464 return 0; 465 } 466 467 static int hp_jack_event(struct notifier_block *nb, unsigned long event, 468 void *data) 469 { 470 struct snd_soc_jack *jack = (struct snd_soc_jack *)data; 471 struct snd_soc_dapm_context *dapm = &jack->card->dapm; 472 473 if (event & SND_JACK_HEADPHONE) 474 /* Disable speaker if headphone is plugged in */ 475 return snd_soc_dapm_disable_pin(dapm, "Ext Spk"); 476 else 477 return snd_soc_dapm_enable_pin(dapm, "Ext Spk"); 478 } 479 480 static struct notifier_block hp_jack_nb = { 481 .notifier_call = hp_jack_event, 482 }; 483 484 static int mic_jack_event(struct notifier_block *nb, unsigned long event, 485 void *data) 486 { 487 struct snd_soc_jack *jack = (struct snd_soc_jack *)data; 488 struct snd_soc_dapm_context *dapm = &jack->card->dapm; 489 490 if (event & SND_JACK_MICROPHONE) 491 /* Disable dmic if microphone is plugged in */ 492 return snd_soc_dapm_disable_pin(dapm, "DMIC"); 493 else 494 return snd_soc_dapm_enable_pin(dapm, "DMIC"); 495 } 496 497 static struct notifier_block mic_jack_nb = { 498 .notifier_call = mic_jack_event, 499 }; 500 501 static int fsl_asoc_card_late_probe(struct snd_soc_card *card) 502 { 503 struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); 504 struct snd_soc_pcm_runtime *rtd = list_first_entry( 505 &card->rtd_list, struct snd_soc_pcm_runtime, list); 506 struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); 507 struct codec_priv *codec_priv = &priv->codec_priv; 508 struct device *dev = card->dev; 509 int ret; 510 511 if (fsl_asoc_card_is_ac97(priv)) { 512 #if IS_ENABLED(CONFIG_SND_AC97_CODEC) 513 struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; 514 struct snd_ac97 *ac97 = snd_soc_component_get_drvdata(component); 515 516 /* 517 * Use slots 3/4 for S/PDIF so SSI won't try to enable 518 * other slots and send some samples there 519 * due to SLOTREQ bits for S/PDIF received from codec 520 */ 521 snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS, 522 AC97_EA_SPSA_SLOT_MASK, AC97_EA_SPSA_3_4); 523 #endif 524 525 return 0; 526 } 527 528 ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, 529 codec_priv->mclk_freq, SND_SOC_CLOCK_IN); 530 if (ret && ret != -ENOTSUPP) { 531 dev_err(dev, "failed to set sysclk in %s\n", __func__); 532 return ret; 533 } 534 535 if (!IS_ERR_OR_NULL(codec_priv->mclk)) 536 clk_prepare_enable(codec_priv->mclk); 537 538 return 0; 539 } 540 541 static int fsl_asoc_card_probe(struct platform_device *pdev) 542 { 543 struct device_node *cpu_np, *codec_np, *asrc_np; 544 struct device_node *np = pdev->dev.of_node; 545 struct platform_device *asrc_pdev = NULL; 546 struct device_node *bitclkprovider = NULL; 547 struct device_node *frameprovider = NULL; 548 struct platform_device *cpu_pdev; 549 struct fsl_asoc_card_priv *priv; 550 struct device *codec_dev = NULL; 551 const char *codec_dai_name; 552 const char *codec_dev_name; 553 u32 asrc_fmt = 0; 554 u32 width; 555 int ret; 556 557 priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); 558 if (!priv) 559 return -ENOMEM; 560 561 cpu_np = of_parse_phandle(np, "audio-cpu", 0); 562 /* Give a chance to old DT binding */ 563 if (!cpu_np) 564 cpu_np = of_parse_phandle(np, "ssi-controller", 0); 565 if (!cpu_np) { 566 dev_err(&pdev->dev, "CPU phandle missing or invalid\n"); 567 ret = -EINVAL; 568 goto fail; 569 } 570 571 cpu_pdev = of_find_device_by_node(cpu_np); 572 if (!cpu_pdev) { 573 dev_err(&pdev->dev, "failed to find CPU DAI device\n"); 574 ret = -EINVAL; 575 goto fail; 576 } 577 578 codec_np = of_parse_phandle(np, "audio-codec", 0); 579 if (codec_np) { 580 struct platform_device *codec_pdev; 581 struct i2c_client *codec_i2c; 582 583 codec_i2c = of_find_i2c_device_by_node(codec_np); 584 if (codec_i2c) { 585 codec_dev = &codec_i2c->dev; 586 codec_dev_name = codec_i2c->name; 587 } 588 if (!codec_dev) { 589 codec_pdev = of_find_device_by_node(codec_np); 590 if (codec_pdev) { 591 codec_dev = &codec_pdev->dev; 592 codec_dev_name = codec_pdev->name; 593 } 594 } 595 } 596 597 asrc_np = of_parse_phandle(np, "audio-asrc", 0); 598 if (asrc_np) 599 asrc_pdev = of_find_device_by_node(asrc_np); 600 601 /* Get the MCLK rate only, and leave it controlled by CODEC drivers */ 602 if (codec_dev) { 603 struct clk *codec_clk = clk_get(codec_dev, NULL); 604 605 if (!IS_ERR(codec_clk)) { 606 priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); 607 clk_put(codec_clk); 608 } 609 } 610 611 /* Default sample rate and format, will be updated in hw_params() */ 612 priv->sample_rate = 44100; 613 priv->sample_format = SNDRV_PCM_FORMAT_S16_LE; 614 615 /* Assign a default DAI format, and allow each card to overwrite it */ 616 priv->dai_fmt = DAI_FMT_BASE; 617 618 memcpy(priv->dai_link, fsl_asoc_card_dai, 619 sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link)); 620 621 priv->card.dapm_routes = audio_map; 622 priv->card.num_dapm_routes = ARRAY_SIZE(audio_map); 623 priv->card.driver_name = DRIVER_NAME; 624 /* Diversify the card configurations */ 625 if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) { 626 codec_dai_name = "cs42888"; 627 priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq; 628 priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq; 629 priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT; 630 priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT; 631 priv->cpu_priv.slot_width = 32; 632 priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC; 633 } else if (of_device_is_compatible(np, "fsl,imx-audio-cs427x")) { 634 codec_dai_name = "cs4271-hifi"; 635 priv->codec_priv.mclk_id = CS427x_SYSCLK_MCLK; 636 priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; 637 } else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) { 638 codec_dai_name = "sgtl5000"; 639 priv->codec_priv.mclk_id = SGTL5000_SYSCLK; 640 priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; 641 } else if (of_device_is_compatible(np, "fsl,imx-audio-tlv320aic32x4")) { 642 codec_dai_name = "tlv320aic32x4-hifi"; 643 priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; 644 } else if (of_device_is_compatible(np, "fsl,imx-audio-tlv320aic31xx")) { 645 codec_dai_name = "tlv320dac31xx-hifi"; 646 priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; 647 priv->dai_link[1].dpcm_capture = 0; 648 priv->dai_link[2].dpcm_capture = 0; 649 priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT; 650 priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT; 651 priv->card.dapm_routes = audio_map_tx; 652 priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx); 653 } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) { 654 codec_dai_name = "wm8962"; 655 priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK; 656 priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; 657 priv->codec_priv.pll_id = WM8962_FLL; 658 priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; 659 } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) { 660 codec_dai_name = "wm8960-hifi"; 661 priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; 662 priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; 663 priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; 664 } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { 665 codec_dai_name = "ac97-hifi"; 666 priv->dai_fmt = SND_SOC_DAIFMT_AC97; 667 priv->card.dapm_routes = audio_map_ac97; 668 priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_ac97); 669 } else if (of_device_is_compatible(np, "fsl,imx-audio-mqs")) { 670 codec_dai_name = "fsl-mqs-dai"; 671 priv->dai_fmt = SND_SOC_DAIFMT_LEFT_J | 672 SND_SOC_DAIFMT_CBC_CFC | 673 SND_SOC_DAIFMT_NB_NF; 674 priv->dai_link[1].dpcm_capture = 0; 675 priv->dai_link[2].dpcm_capture = 0; 676 priv->card.dapm_routes = audio_map_tx; 677 priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx); 678 } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8524")) { 679 codec_dai_name = "wm8524-hifi"; 680 priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC; 681 priv->dai_link[1].dpcm_capture = 0; 682 priv->dai_link[2].dpcm_capture = 0; 683 priv->cpu_priv.slot_width = 32; 684 priv->card.dapm_routes = audio_map_tx; 685 priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx); 686 } else if (of_device_is_compatible(np, "fsl,imx-audio-si476x")) { 687 codec_dai_name = "si476x-codec"; 688 priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC; 689 priv->card.dapm_routes = audio_map_rx; 690 priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_rx); 691 } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8958")) { 692 codec_dai_name = "wm8994-aif1"; 693 priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; 694 priv->codec_priv.mclk_id = WM8994_FLL_SRC_MCLK1; 695 priv->codec_priv.fll_id = WM8994_SYSCLK_FLL1; 696 priv->codec_priv.pll_id = WM8994_FLL1; 697 priv->codec_priv.free_freq = priv->codec_priv.mclk_freq; 698 priv->card.dapm_routes = NULL; 699 priv->card.num_dapm_routes = 0; 700 } else if (of_device_is_compatible(np, "fsl,imx-audio-nau8822")) { 701 codec_dai_name = "nau8822-hifi"; 702 priv->codec_priv.mclk_id = NAU8822_CLK_MCLK; 703 priv->codec_priv.fll_id = NAU8822_CLK_PLL; 704 priv->codec_priv.pll_id = NAU8822_CLK_PLL; 705 priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; 706 if (codec_dev) 707 priv->codec_priv.mclk = devm_clk_get(codec_dev, NULL); 708 } else { 709 dev_err(&pdev->dev, "unknown Device Tree compatible\n"); 710 ret = -EINVAL; 711 goto asrc_fail; 712 } 713 714 /* 715 * Allow setting mclk-id from the device-tree node. Otherwise, the 716 * default value for each card configuration is used. 717 */ 718 of_property_read_u32(np, "mclk-id", &priv->codec_priv.mclk_id); 719 720 /* Format info from DT is optional. */ 721 snd_soc_daifmt_parse_clock_provider_as_phandle(np, NULL, &bitclkprovider, &frameprovider); 722 if (bitclkprovider || frameprovider) { 723 unsigned int daifmt = snd_soc_daifmt_parse_format(np, NULL); 724 725 if (codec_np == bitclkprovider) 726 daifmt |= (codec_np == frameprovider) ? 727 SND_SOC_DAIFMT_CBP_CFP : SND_SOC_DAIFMT_CBP_CFC; 728 else 729 daifmt |= (codec_np == frameprovider) ? 730 SND_SOC_DAIFMT_CBC_CFP : SND_SOC_DAIFMT_CBC_CFC; 731 732 /* Override dai_fmt with value from DT */ 733 priv->dai_fmt = daifmt; 734 } 735 736 /* Change direction according to format */ 737 if (priv->dai_fmt & SND_SOC_DAIFMT_CBP_CFP) { 738 priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_IN; 739 priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_IN; 740 } 741 742 of_node_put(bitclkprovider); 743 of_node_put(frameprovider); 744 745 if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) { 746 dev_dbg(&pdev->dev, "failed to find codec device\n"); 747 ret = -EPROBE_DEFER; 748 goto asrc_fail; 749 } 750 751 /* Common settings for corresponding Freescale CPU DAI driver */ 752 if (of_node_name_eq(cpu_np, "ssi")) { 753 /* Only SSI needs to configure AUDMUX */ 754 ret = fsl_asoc_card_audmux_init(np, priv); 755 if (ret) { 756 dev_err(&pdev->dev, "failed to init audmux\n"); 757 goto asrc_fail; 758 } 759 } else if (of_node_name_eq(cpu_np, "esai")) { 760 struct clk *esai_clk = clk_get(&cpu_pdev->dev, "extal"); 761 762 if (!IS_ERR(esai_clk)) { 763 priv->cpu_priv.sysclk_freq[TX] = clk_get_rate(esai_clk); 764 priv->cpu_priv.sysclk_freq[RX] = clk_get_rate(esai_clk); 765 clk_put(esai_clk); 766 } else if (PTR_ERR(esai_clk) == -EPROBE_DEFER) { 767 ret = -EPROBE_DEFER; 768 goto asrc_fail; 769 } 770 771 priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL; 772 priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL; 773 } else if (of_node_name_eq(cpu_np, "sai")) { 774 priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1; 775 priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1; 776 } 777 778 /* Initialize sound card */ 779 priv->pdev = pdev; 780 priv->card.dev = &pdev->dev; 781 priv->card.owner = THIS_MODULE; 782 ret = snd_soc_of_parse_card_name(&priv->card, "model"); 783 if (ret) { 784 snprintf(priv->name, sizeof(priv->name), "%s-audio", 785 fsl_asoc_card_is_ac97(priv) ? "ac97" : codec_dev_name); 786 priv->card.name = priv->name; 787 } 788 priv->card.dai_link = priv->dai_link; 789 priv->card.late_probe = fsl_asoc_card_late_probe; 790 priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets; 791 priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets); 792 793 /* Drop the second half of DAPM routes -- ASRC */ 794 if (!asrc_pdev) 795 priv->card.num_dapm_routes /= 2; 796 797 if (of_property_read_bool(np, "audio-routing")) { 798 ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing"); 799 if (ret) { 800 dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret); 801 goto asrc_fail; 802 } 803 } 804 805 /* Normal DAI Link */ 806 priv->dai_link[0].cpus->of_node = cpu_np; 807 priv->dai_link[0].codecs->dai_name = codec_dai_name; 808 809 if (!fsl_asoc_card_is_ac97(priv)) 810 priv->dai_link[0].codecs->of_node = codec_np; 811 else { 812 u32 idx; 813 814 ret = of_property_read_u32(cpu_np, "cell-index", &idx); 815 if (ret) { 816 dev_err(&pdev->dev, 817 "cannot get CPU index property\n"); 818 goto asrc_fail; 819 } 820 821 priv->dai_link[0].codecs->name = 822 devm_kasprintf(&pdev->dev, GFP_KERNEL, 823 "ac97-codec.%u", 824 (unsigned int)idx); 825 if (!priv->dai_link[0].codecs->name) { 826 ret = -ENOMEM; 827 goto asrc_fail; 828 } 829 } 830 831 priv->dai_link[0].platforms->of_node = cpu_np; 832 priv->dai_link[0].dai_fmt = priv->dai_fmt; 833 priv->card.num_links = 1; 834 835 if (asrc_pdev) { 836 /* DPCM DAI Links only if ASRC exists */ 837 priv->dai_link[1].cpus->of_node = asrc_np; 838 priv->dai_link[1].platforms->of_node = asrc_np; 839 priv->dai_link[2].codecs->dai_name = codec_dai_name; 840 priv->dai_link[2].codecs->of_node = codec_np; 841 priv->dai_link[2].codecs->name = 842 priv->dai_link[0].codecs->name; 843 priv->dai_link[2].cpus->of_node = cpu_np; 844 priv->dai_link[2].dai_fmt = priv->dai_fmt; 845 priv->card.num_links = 3; 846 847 ret = of_property_read_u32(asrc_np, "fsl,asrc-rate", 848 &priv->asrc_rate); 849 if (ret) { 850 dev_err(&pdev->dev, "failed to get output rate\n"); 851 ret = -EINVAL; 852 goto asrc_fail; 853 } 854 855 ret = of_property_read_u32(asrc_np, "fsl,asrc-format", &asrc_fmt); 856 priv->asrc_format = (__force snd_pcm_format_t)asrc_fmt; 857 if (ret) { 858 /* Fallback to old binding; translate to asrc_format */ 859 ret = of_property_read_u32(asrc_np, "fsl,asrc-width", 860 &width); 861 if (ret) { 862 dev_err(&pdev->dev, 863 "failed to decide output format\n"); 864 goto asrc_fail; 865 } 866 867 if (width == 24) 868 priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE; 869 else 870 priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE; 871 } 872 } 873 874 /* Finish card registering */ 875 platform_set_drvdata(pdev, priv); 876 snd_soc_card_set_drvdata(&priv->card, priv); 877 878 ret = devm_snd_soc_register_card(&pdev->dev, &priv->card); 879 if (ret) { 880 dev_err_probe(&pdev->dev, ret, "snd_soc_register_card failed\n"); 881 goto asrc_fail; 882 } 883 884 /* 885 * Properties "hp-det-gpio" and "mic-det-gpio" are optional, and 886 * asoc_simple_init_jack uses these properties for creating 887 * Headphone Jack and Microphone Jack. 888 * 889 * The notifier is initialized in snd_soc_card_jack_new(), then 890 * snd_soc_jack_notifier_register can be called. 891 */ 892 if (of_property_read_bool(np, "hp-det-gpio")) { 893 ret = asoc_simple_init_jack(&priv->card, &priv->hp_jack, 894 1, NULL, "Headphone Jack"); 895 if (ret) 896 goto asrc_fail; 897 898 snd_soc_jack_notifier_register(&priv->hp_jack.jack, &hp_jack_nb); 899 } 900 901 if (of_property_read_bool(np, "mic-det-gpio")) { 902 ret = asoc_simple_init_jack(&priv->card, &priv->mic_jack, 903 0, NULL, "Mic Jack"); 904 if (ret) 905 goto asrc_fail; 906 907 snd_soc_jack_notifier_register(&priv->mic_jack.jack, &mic_jack_nb); 908 } 909 910 asrc_fail: 911 of_node_put(asrc_np); 912 of_node_put(codec_np); 913 put_device(&cpu_pdev->dev); 914 fail: 915 of_node_put(cpu_np); 916 917 return ret; 918 } 919 920 static const struct of_device_id fsl_asoc_card_dt_ids[] = { 921 { .compatible = "fsl,imx-audio-ac97", }, 922 { .compatible = "fsl,imx-audio-cs42888", }, 923 { .compatible = "fsl,imx-audio-cs427x", }, 924 { .compatible = "fsl,imx-audio-tlv320aic32x4", }, 925 { .compatible = "fsl,imx-audio-tlv320aic31xx", }, 926 { .compatible = "fsl,imx-audio-sgtl5000", }, 927 { .compatible = "fsl,imx-audio-wm8962", }, 928 { .compatible = "fsl,imx-audio-wm8960", }, 929 { .compatible = "fsl,imx-audio-mqs", }, 930 { .compatible = "fsl,imx-audio-wm8524", }, 931 { .compatible = "fsl,imx-audio-si476x", }, 932 { .compatible = "fsl,imx-audio-wm8958", }, 933 { .compatible = "fsl,imx-audio-nau8822", }, 934 {} 935 }; 936 MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids); 937 938 static struct platform_driver fsl_asoc_card_driver = { 939 .probe = fsl_asoc_card_probe, 940 .driver = { 941 .name = DRIVER_NAME, 942 .pm = &snd_soc_pm_ops, 943 .of_match_table = fsl_asoc_card_dt_ids, 944 }, 945 }; 946 module_platform_driver(fsl_asoc_card_driver); 947 948 MODULE_DESCRIPTION("Freescale Generic ASoC Sound Card driver with ASRC"); 949 MODULE_AUTHOR("Nicolin Chen <nicoleotsuka@gmail.com>"); 950 MODULE_ALIAS("platform:" DRIVER_NAME); 951 MODULE_LICENSE("GPL"); 952