1 /* 2 * Freescale Generic ASoC Sound Card driver with ASRC 3 * 4 * Copyright (C) 2014 Freescale Semiconductor, Inc. 5 * 6 * Author: Nicolin Chen <nicoleotsuka@gmail.com> 7 * 8 * This file is licensed under the terms of the GNU General Public License 9 * version 2. This program is licensed "as is" without any warranty of any 10 * kind, whether express or implied. 11 */ 12 13 #include <linux/clk.h> 14 #include <linux/i2c.h> 15 #include <linux/module.h> 16 #include <linux/of_platform.h> 17 #if IS_ENABLED(CONFIG_SND_AC97_CODEC) 18 #include <sound/ac97_codec.h> 19 #endif 20 #include <sound/pcm_params.h> 21 #include <sound/soc.h> 22 23 #include "fsl_esai.h" 24 #include "fsl_sai.h" 25 #include "imx-audmux.h" 26 27 #include "../codecs/sgtl5000.h" 28 #include "../codecs/wm8962.h" 29 #include "../codecs/wm8960.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 * CODEC private data 41 * 42 * @mclk_freq: Clock rate of MCLK 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 u32 mclk_id; 50 u32 fll_id; 51 u32 pll_id; 52 }; 53 54 /** 55 * CPU private data 56 * 57 * @sysclk_freq[2]: SYSCLK rates for set_sysclk() 58 * @sysclk_dir[2]: SYSCLK directions for set_sysclk() 59 * @sysclk_id[2]: 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 * Freescale Generic ASOC card private data 73 * 74 * @dai_link[3]: DAI link structure including normal one and DPCM link 75 * @pdev: platform device pointer 76 * @codec_priv: CODEC private data 77 * @cpu_priv: CPU private data 78 * @card: ASoC card structure 79 * @sample_rate: Current sample rate 80 * @sample_format: Current sample format 81 * @asrc_rate: ASRC sample rate used by Back-Ends 82 * @asrc_format: ASRC sample format used by Back-Ends 83 * @dai_fmt: DAI format between CPU and CODEC 84 * @name: Card name 85 */ 86 87 struct fsl_asoc_card_priv { 88 struct snd_soc_dai_link dai_link[3]; 89 struct platform_device *pdev; 90 struct codec_priv codec_priv; 91 struct cpu_priv cpu_priv; 92 struct snd_soc_card card; 93 u32 sample_rate; 94 snd_pcm_format_t sample_format; 95 u32 asrc_rate; 96 snd_pcm_format_t asrc_format; 97 u32 dai_fmt; 98 char name[32]; 99 }; 100 101 /** 102 * This dapm route map exsits for DPCM link only. 103 * The other routes shall go through Device Tree. 104 * 105 * Note: keep all ASRC routes in the second half 106 * to drop them easily for non-ASRC cases. 107 */ 108 static const struct snd_soc_dapm_route audio_map[] = { 109 /* 1st half -- Normal DAPM routes */ 110 {"Playback", NULL, "CPU-Playback"}, 111 {"CPU-Capture", NULL, "Capture"}, 112 /* 2nd half -- ASRC DAPM routes */ 113 {"CPU-Playback", NULL, "ASRC-Playback"}, 114 {"ASRC-Capture", NULL, "CPU-Capture"}, 115 }; 116 117 static const struct snd_soc_dapm_route audio_map_ac97[] = { 118 /* 1st half -- Normal DAPM routes */ 119 {"Playback", NULL, "AC97 Playback"}, 120 {"AC97 Capture", NULL, "Capture"}, 121 /* 2nd half -- ASRC DAPM routes */ 122 {"AC97 Playback", NULL, "ASRC-Playback"}, 123 {"ASRC-Capture", NULL, "AC97 Capture"}, 124 }; 125 126 /* Add all possible widgets into here without being redundant */ 127 static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = { 128 SND_SOC_DAPM_LINE("Line Out Jack", NULL), 129 SND_SOC_DAPM_LINE("Line In Jack", NULL), 130 SND_SOC_DAPM_HP("Headphone Jack", NULL), 131 SND_SOC_DAPM_SPK("Ext Spk", NULL), 132 SND_SOC_DAPM_MIC("Mic Jack", NULL), 133 SND_SOC_DAPM_MIC("AMIC", NULL), 134 SND_SOC_DAPM_MIC("DMIC", NULL), 135 }; 136 137 static bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv) 138 { 139 return priv->dai_fmt == SND_SOC_DAIFMT_AC97; 140 } 141 142 static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, 143 struct snd_pcm_hw_params *params) 144 { 145 struct snd_soc_pcm_runtime *rtd = substream->private_data; 146 struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); 147 bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; 148 struct cpu_priv *cpu_priv = &priv->cpu_priv; 149 struct device *dev = rtd->card->dev; 150 int ret; 151 152 priv->sample_rate = params_rate(params); 153 priv->sample_format = params_format(params); 154 155 /* 156 * If codec-dai is DAI Master and all configurations are already in the 157 * set_bias_level(), bypass the remaining settings in hw_params(). 158 * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS. 159 */ 160 if ((priv->card.set_bias_level && 161 priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) || 162 fsl_asoc_card_is_ac97(priv)) 163 return 0; 164 165 /* Specific configurations of DAIs starts from here */ 166 ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, cpu_priv->sysclk_id[tx], 167 cpu_priv->sysclk_freq[tx], 168 cpu_priv->sysclk_dir[tx]); 169 if (ret && ret != -ENOTSUPP) { 170 dev_err(dev, "failed to set sysclk for cpu dai\n"); 171 return ret; 172 } 173 174 if (cpu_priv->slot_width) { 175 ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 176 cpu_priv->slot_width); 177 if (ret && ret != -ENOTSUPP) { 178 dev_err(dev, "failed to set TDM slot for cpu dai\n"); 179 return ret; 180 } 181 } 182 183 return 0; 184 } 185 186 static const struct snd_soc_ops fsl_asoc_card_ops = { 187 .hw_params = fsl_asoc_card_hw_params, 188 }; 189 190 static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, 191 struct snd_pcm_hw_params *params) 192 { 193 struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); 194 struct snd_interval *rate; 195 struct snd_mask *mask; 196 197 rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); 198 rate->max = rate->min = priv->asrc_rate; 199 200 mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); 201 snd_mask_none(mask); 202 snd_mask_set(mask, (__force int)priv->asrc_format); 203 204 return 0; 205 } 206 207 static struct snd_soc_dai_link fsl_asoc_card_dai[] = { 208 /* Default ASoC DAI Link*/ 209 { 210 .name = "HiFi", 211 .stream_name = "HiFi", 212 .ops = &fsl_asoc_card_ops, 213 }, 214 /* DPCM Link between Front-End and Back-End (Optional) */ 215 { 216 .name = "HiFi-ASRC-FE", 217 .stream_name = "HiFi-ASRC-FE", 218 .codec_name = "snd-soc-dummy", 219 .codec_dai_name = "snd-soc-dummy-dai", 220 .dpcm_playback = 1, 221 .dpcm_capture = 1, 222 .dynamic = 1, 223 }, 224 { 225 .name = "HiFi-ASRC-BE", 226 .stream_name = "HiFi-ASRC-BE", 227 .platform_name = "snd-soc-dummy", 228 .be_hw_params_fixup = be_hw_params_fixup, 229 .ops = &fsl_asoc_card_ops, 230 .dpcm_playback = 1, 231 .dpcm_capture = 1, 232 .no_pcm = 1, 233 }, 234 }; 235 236 static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card, 237 struct snd_soc_dapm_context *dapm, 238 enum snd_soc_bias_level level) 239 { 240 struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); 241 struct snd_soc_pcm_runtime *rtd; 242 struct snd_soc_dai *codec_dai; 243 struct codec_priv *codec_priv = &priv->codec_priv; 244 struct device *dev = card->dev; 245 unsigned int pll_out; 246 int ret; 247 248 rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); 249 codec_dai = rtd->codec_dai; 250 if (dapm->dev != codec_dai->dev) 251 return 0; 252 253 switch (level) { 254 case SND_SOC_BIAS_PREPARE: 255 if (dapm->bias_level != SND_SOC_BIAS_STANDBY) 256 break; 257 258 if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) 259 pll_out = priv->sample_rate * 384; 260 else 261 pll_out = priv->sample_rate * 256; 262 263 ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, 264 codec_priv->mclk_id, 265 codec_priv->mclk_freq, pll_out); 266 if (ret) { 267 dev_err(dev, "failed to start FLL: %d\n", ret); 268 return ret; 269 } 270 271 ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id, 272 pll_out, SND_SOC_CLOCK_IN); 273 if (ret && ret != -ENOTSUPP) { 274 dev_err(dev, "failed to set SYSCLK: %d\n", ret); 275 return ret; 276 } 277 break; 278 279 case SND_SOC_BIAS_STANDBY: 280 if (dapm->bias_level != SND_SOC_BIAS_PREPARE) 281 break; 282 283 ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, 284 codec_priv->mclk_freq, 285 SND_SOC_CLOCK_IN); 286 if (ret && ret != -ENOTSUPP) { 287 dev_err(dev, "failed to switch away from FLL: %d\n", ret); 288 return ret; 289 } 290 291 ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, 0, 0, 0); 292 if (ret) { 293 dev_err(dev, "failed to stop FLL: %d\n", ret); 294 return ret; 295 } 296 break; 297 298 default: 299 break; 300 } 301 302 return 0; 303 } 304 305 static int fsl_asoc_card_audmux_init(struct device_node *np, 306 struct fsl_asoc_card_priv *priv) 307 { 308 struct device *dev = &priv->pdev->dev; 309 u32 int_ptcr = 0, ext_ptcr = 0; 310 int int_port, ext_port; 311 int ret; 312 313 ret = of_property_read_u32(np, "mux-int-port", &int_port); 314 if (ret) { 315 dev_err(dev, "mux-int-port missing or invalid\n"); 316 return ret; 317 } 318 ret = of_property_read_u32(np, "mux-ext-port", &ext_port); 319 if (ret) { 320 dev_err(dev, "mux-ext-port missing or invalid\n"); 321 return ret; 322 } 323 324 /* 325 * The port numbering in the hardware manual starts at 1, while 326 * the AUDMUX API expects it starts at 0. 327 */ 328 int_port--; 329 ext_port--; 330 331 /* 332 * Use asynchronous mode (6 wires) for all cases except AC97. 333 * If only 4 wires are needed, just set SSI into 334 * synchronous mode and enable 4 PADs in IOMUX. 335 */ 336 switch (priv->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) { 337 case SND_SOC_DAIFMT_CBM_CFM: 338 int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) | 339 IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) | 340 IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | 341 IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | 342 IMX_AUDMUX_V2_PTCR_RFSDIR | 343 IMX_AUDMUX_V2_PTCR_RCLKDIR | 344 IMX_AUDMUX_V2_PTCR_TFSDIR | 345 IMX_AUDMUX_V2_PTCR_TCLKDIR; 346 break; 347 case SND_SOC_DAIFMT_CBM_CFS: 348 int_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) | 349 IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | 350 IMX_AUDMUX_V2_PTCR_RCLKDIR | 351 IMX_AUDMUX_V2_PTCR_TCLKDIR; 352 ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) | 353 IMX_AUDMUX_V2_PTCR_TFSEL(int_port) | 354 IMX_AUDMUX_V2_PTCR_RFSDIR | 355 IMX_AUDMUX_V2_PTCR_TFSDIR; 356 break; 357 case SND_SOC_DAIFMT_CBS_CFM: 358 int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) | 359 IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | 360 IMX_AUDMUX_V2_PTCR_RFSDIR | 361 IMX_AUDMUX_V2_PTCR_TFSDIR; 362 ext_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) | 363 IMX_AUDMUX_V2_PTCR_TCSEL(int_port) | 364 IMX_AUDMUX_V2_PTCR_RCLKDIR | 365 IMX_AUDMUX_V2_PTCR_TCLKDIR; 366 break; 367 case SND_SOC_DAIFMT_CBS_CFS: 368 ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) | 369 IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) | 370 IMX_AUDMUX_V2_PTCR_TFSEL(int_port) | 371 IMX_AUDMUX_V2_PTCR_TCSEL(int_port) | 372 IMX_AUDMUX_V2_PTCR_RFSDIR | 373 IMX_AUDMUX_V2_PTCR_RCLKDIR | 374 IMX_AUDMUX_V2_PTCR_TFSDIR | 375 IMX_AUDMUX_V2_PTCR_TCLKDIR; 376 break; 377 default: 378 if (!fsl_asoc_card_is_ac97(priv)) 379 return -EINVAL; 380 } 381 382 if (fsl_asoc_card_is_ac97(priv)) { 383 int_ptcr = IMX_AUDMUX_V2_PTCR_SYN | 384 IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | 385 IMX_AUDMUX_V2_PTCR_TCLKDIR; 386 ext_ptcr = IMX_AUDMUX_V2_PTCR_SYN | 387 IMX_AUDMUX_V2_PTCR_TFSEL(int_port) | 388 IMX_AUDMUX_V2_PTCR_TFSDIR; 389 } 390 391 /* Asynchronous mode can not be set along with RCLKDIR */ 392 if (!fsl_asoc_card_is_ac97(priv)) { 393 unsigned int pdcr = 394 IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port); 395 396 ret = imx_audmux_v2_configure_port(int_port, 0, 397 pdcr); 398 if (ret) { 399 dev_err(dev, "audmux internal port setup failed\n"); 400 return ret; 401 } 402 } 403 404 ret = imx_audmux_v2_configure_port(int_port, int_ptcr, 405 IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); 406 if (ret) { 407 dev_err(dev, "audmux internal port setup failed\n"); 408 return ret; 409 } 410 411 if (!fsl_asoc_card_is_ac97(priv)) { 412 unsigned int pdcr = 413 IMX_AUDMUX_V2_PDCR_RXDSEL(int_port); 414 415 ret = imx_audmux_v2_configure_port(ext_port, 0, 416 pdcr); 417 if (ret) { 418 dev_err(dev, "audmux external port setup failed\n"); 419 return ret; 420 } 421 } 422 423 ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr, 424 IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); 425 if (ret) { 426 dev_err(dev, "audmux external port setup failed\n"); 427 return ret; 428 } 429 430 return 0; 431 } 432 433 static int fsl_asoc_card_late_probe(struct snd_soc_card *card) 434 { 435 struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); 436 struct snd_soc_pcm_runtime *rtd = list_first_entry( 437 &card->rtd_list, struct snd_soc_pcm_runtime, list); 438 struct snd_soc_dai *codec_dai = rtd->codec_dai; 439 struct codec_priv *codec_priv = &priv->codec_priv; 440 struct device *dev = card->dev; 441 int ret; 442 443 if (fsl_asoc_card_is_ac97(priv)) { 444 #if IS_ENABLED(CONFIG_SND_AC97_CODEC) 445 struct snd_soc_component *component = rtd->codec_dai->component; 446 struct snd_ac97 *ac97 = snd_soc_component_get_drvdata(component); 447 448 /* 449 * Use slots 3/4 for S/PDIF so SSI won't try to enable 450 * other slots and send some samples there 451 * due to SLOTREQ bits for S/PDIF received from codec 452 */ 453 snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS, 454 AC97_EA_SPSA_SLOT_MASK, AC97_EA_SPSA_3_4); 455 #endif 456 457 return 0; 458 } 459 460 ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, 461 codec_priv->mclk_freq, SND_SOC_CLOCK_IN); 462 if (ret && ret != -ENOTSUPP) { 463 dev_err(dev, "failed to set sysclk in %s\n", __func__); 464 return ret; 465 } 466 467 return 0; 468 } 469 470 static int fsl_asoc_card_probe(struct platform_device *pdev) 471 { 472 struct device_node *cpu_np, *codec_np, *asrc_np; 473 struct device_node *np = pdev->dev.of_node; 474 struct platform_device *asrc_pdev = NULL; 475 struct platform_device *cpu_pdev; 476 struct fsl_asoc_card_priv *priv; 477 struct i2c_client *codec_dev; 478 const char *codec_dai_name; 479 u32 width; 480 int ret; 481 482 priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); 483 if (!priv) 484 return -ENOMEM; 485 486 cpu_np = of_parse_phandle(np, "audio-cpu", 0); 487 /* Give a chance to old DT binding */ 488 if (!cpu_np) 489 cpu_np = of_parse_phandle(np, "ssi-controller", 0); 490 if (!cpu_np) { 491 dev_err(&pdev->dev, "CPU phandle missing or invalid\n"); 492 ret = -EINVAL; 493 goto fail; 494 } 495 496 cpu_pdev = of_find_device_by_node(cpu_np); 497 if (!cpu_pdev) { 498 dev_err(&pdev->dev, "failed to find CPU DAI device\n"); 499 ret = -EINVAL; 500 goto fail; 501 } 502 503 codec_np = of_parse_phandle(np, "audio-codec", 0); 504 if (codec_np) 505 codec_dev = of_find_i2c_device_by_node(codec_np); 506 else 507 codec_dev = NULL; 508 509 asrc_np = of_parse_phandle(np, "audio-asrc", 0); 510 if (asrc_np) 511 asrc_pdev = of_find_device_by_node(asrc_np); 512 513 /* Get the MCLK rate only, and leave it controlled by CODEC drivers */ 514 if (codec_dev) { 515 struct clk *codec_clk = clk_get(&codec_dev->dev, NULL); 516 517 if (!IS_ERR(codec_clk)) { 518 priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); 519 clk_put(codec_clk); 520 } 521 } 522 523 /* Default sample rate and format, will be updated in hw_params() */ 524 priv->sample_rate = 44100; 525 priv->sample_format = SNDRV_PCM_FORMAT_S16_LE; 526 527 /* Assign a default DAI format, and allow each card to overwrite it */ 528 priv->dai_fmt = DAI_FMT_BASE; 529 530 /* Diversify the card configurations */ 531 if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) { 532 codec_dai_name = "cs42888"; 533 priv->card.set_bias_level = NULL; 534 priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq; 535 priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq; 536 priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT; 537 priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT; 538 priv->cpu_priv.slot_width = 32; 539 priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; 540 } else if (of_device_is_compatible(np, "fsl,imx-audio-cs427x")) { 541 codec_dai_name = "cs4271-hifi"; 542 priv->codec_priv.mclk_id = CS427x_SYSCLK_MCLK; 543 priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; 544 } else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) { 545 codec_dai_name = "sgtl5000"; 546 priv->codec_priv.mclk_id = SGTL5000_SYSCLK; 547 priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; 548 } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) { 549 codec_dai_name = "wm8962"; 550 priv->card.set_bias_level = fsl_asoc_card_set_bias_level; 551 priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK; 552 priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; 553 priv->codec_priv.pll_id = WM8962_FLL; 554 priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; 555 } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) { 556 codec_dai_name = "wm8960-hifi"; 557 priv->card.set_bias_level = fsl_asoc_card_set_bias_level; 558 priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; 559 priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; 560 priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; 561 } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { 562 codec_dai_name = "ac97-hifi"; 563 priv->card.set_bias_level = NULL; 564 priv->dai_fmt = SND_SOC_DAIFMT_AC97; 565 } else { 566 dev_err(&pdev->dev, "unknown Device Tree compatible\n"); 567 ret = -EINVAL; 568 goto asrc_fail; 569 } 570 571 if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) { 572 dev_err(&pdev->dev, "failed to find codec device\n"); 573 ret = -EINVAL; 574 goto asrc_fail; 575 } 576 577 /* Common settings for corresponding Freescale CPU DAI driver */ 578 if (strstr(cpu_np->name, "ssi")) { 579 /* Only SSI needs to configure AUDMUX */ 580 ret = fsl_asoc_card_audmux_init(np, priv); 581 if (ret) { 582 dev_err(&pdev->dev, "failed to init audmux\n"); 583 goto asrc_fail; 584 } 585 } else if (strstr(cpu_np->name, "esai")) { 586 priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL; 587 priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL; 588 } else if (strstr(cpu_np->name, "sai")) { 589 priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1; 590 priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1; 591 } 592 593 snprintf(priv->name, sizeof(priv->name), "%s-audio", 594 fsl_asoc_card_is_ac97(priv) ? "ac97" : 595 codec_dev->name); 596 597 /* Initialize sound card */ 598 priv->pdev = pdev; 599 priv->card.dev = &pdev->dev; 600 priv->card.name = priv->name; 601 priv->card.dai_link = priv->dai_link; 602 priv->card.dapm_routes = fsl_asoc_card_is_ac97(priv) ? 603 audio_map_ac97 : audio_map; 604 priv->card.late_probe = fsl_asoc_card_late_probe; 605 priv->card.num_dapm_routes = ARRAY_SIZE(audio_map); 606 priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets; 607 priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets); 608 609 /* Drop the second half of DAPM routes -- ASRC */ 610 if (!asrc_pdev) 611 priv->card.num_dapm_routes /= 2; 612 613 memcpy(priv->dai_link, fsl_asoc_card_dai, 614 sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link)); 615 616 ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing"); 617 if (ret) { 618 dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret); 619 goto asrc_fail; 620 } 621 622 /* Normal DAI Link */ 623 priv->dai_link[0].cpu_of_node = cpu_np; 624 priv->dai_link[0].codec_dai_name = codec_dai_name; 625 626 if (!fsl_asoc_card_is_ac97(priv)) 627 priv->dai_link[0].codec_of_node = codec_np; 628 else { 629 u32 idx; 630 631 ret = of_property_read_u32(cpu_np, "cell-index", &idx); 632 if (ret) { 633 dev_err(&pdev->dev, 634 "cannot get CPU index property\n"); 635 goto asrc_fail; 636 } 637 638 priv->dai_link[0].codec_name = 639 devm_kasprintf(&pdev->dev, GFP_KERNEL, 640 "ac97-codec.%u", 641 (unsigned int)idx); 642 if (!priv->dai_link[0].codec_name) { 643 ret = -ENOMEM; 644 goto asrc_fail; 645 } 646 } 647 648 priv->dai_link[0].platform_of_node = cpu_np; 649 priv->dai_link[0].dai_fmt = priv->dai_fmt; 650 priv->card.num_links = 1; 651 652 if (asrc_pdev) { 653 /* DPCM DAI Links only if ASRC exsits */ 654 priv->dai_link[1].cpu_of_node = asrc_np; 655 priv->dai_link[1].platform_of_node = asrc_np; 656 priv->dai_link[2].codec_dai_name = codec_dai_name; 657 priv->dai_link[2].codec_of_node = codec_np; 658 priv->dai_link[2].codec_name = 659 priv->dai_link[0].codec_name; 660 priv->dai_link[2].cpu_of_node = cpu_np; 661 priv->dai_link[2].dai_fmt = priv->dai_fmt; 662 priv->card.num_links = 3; 663 664 ret = of_property_read_u32(asrc_np, "fsl,asrc-rate", 665 &priv->asrc_rate); 666 if (ret) { 667 dev_err(&pdev->dev, "failed to get output rate\n"); 668 ret = -EINVAL; 669 goto asrc_fail; 670 } 671 672 ret = of_property_read_u32(asrc_np, "fsl,asrc-width", &width); 673 if (ret) { 674 dev_err(&pdev->dev, "failed to get output rate\n"); 675 ret = -EINVAL; 676 goto asrc_fail; 677 } 678 679 if (width == 24) 680 priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE; 681 else 682 priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE; 683 } 684 685 /* Finish card registering */ 686 platform_set_drvdata(pdev, priv); 687 snd_soc_card_set_drvdata(&priv->card, priv); 688 689 ret = devm_snd_soc_register_card(&pdev->dev, &priv->card); 690 if (ret && ret != -EPROBE_DEFER) 691 dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); 692 693 asrc_fail: 694 of_node_put(asrc_np); 695 of_node_put(codec_np); 696 fail: 697 of_node_put(cpu_np); 698 699 return ret; 700 } 701 702 static const struct of_device_id fsl_asoc_card_dt_ids[] = { 703 { .compatible = "fsl,imx-audio-ac97", }, 704 { .compatible = "fsl,imx-audio-cs42888", }, 705 { .compatible = "fsl,imx-audio-cs427x", }, 706 { .compatible = "fsl,imx-audio-sgtl5000", }, 707 { .compatible = "fsl,imx-audio-wm8962", }, 708 { .compatible = "fsl,imx-audio-wm8960", }, 709 {} 710 }; 711 MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids); 712 713 static struct platform_driver fsl_asoc_card_driver = { 714 .probe = fsl_asoc_card_probe, 715 .driver = { 716 .name = "fsl-asoc-card", 717 .pm = &snd_soc_pm_ops, 718 .of_match_table = fsl_asoc_card_dt_ids, 719 }, 720 }; 721 module_platform_driver(fsl_asoc_card_driver); 722 723 MODULE_DESCRIPTION("Freescale Generic ASoC Sound Card driver with ASRC"); 724 MODULE_AUTHOR("Nicolin Chen <nicoleotsuka@gmail.com>"); 725 MODULE_ALIAS("platform:fsl-asoc-card"); 726 MODULE_LICENSE("GPL"); 727