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