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