xref: /openbmc/linux/sound/soc/fsl/fsl-asoc-card.c (revision b830f94f)
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 
19 #include "fsl_esai.h"
20 #include "fsl_sai.h"
21 #include "imx-audmux.h"
22 
23 #include "../codecs/sgtl5000.h"
24 #include "../codecs/wm8962.h"
25 #include "../codecs/wm8960.h"
26 
27 #define CS427x_SYSCLK_MCLK 0
28 
29 #define RX 0
30 #define TX 1
31 
32 /* Default DAI format without Master and Slave flag */
33 #define DAI_FMT_BASE (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF)
34 
35 /**
36  * CODEC private data
37  *
38  * @mclk_freq: Clock rate of MCLK
39  * @mclk_id: MCLK (or main clock) id for set_sysclk()
40  * @fll_id: FLL (or secordary clock) id for set_sysclk()
41  * @pll_id: PLL id for set_pll()
42  */
43 struct codec_priv {
44 	unsigned long mclk_freq;
45 	u32 mclk_id;
46 	u32 fll_id;
47 	u32 pll_id;
48 };
49 
50 /**
51  * CPU private data
52  *
53  * @sysclk_freq[2]: SYSCLK rates for set_sysclk()
54  * @sysclk_dir[2]: SYSCLK directions for set_sysclk()
55  * @sysclk_id[2]: 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  * Freescale Generic ASOC card private data
69  *
70  * @dai_link[3]: DAI link structure including normal one and DPCM link
71  * @pdev: platform device pointer
72  * @codec_priv: CODEC private data
73  * @cpu_priv: CPU private data
74  * @card: ASoC card structure
75  * @sample_rate: Current sample rate
76  * @sample_format: Current sample format
77  * @asrc_rate: ASRC sample rate used by Back-Ends
78  * @asrc_format: ASRC sample format used by Back-Ends
79  * @dai_fmt: DAI format between CPU and CODEC
80  * @name: Card name
81  */
82 
83 struct fsl_asoc_card_priv {
84 	struct snd_soc_dai_link dai_link[3];
85 	struct platform_device *pdev;
86 	struct codec_priv codec_priv;
87 	struct cpu_priv cpu_priv;
88 	struct snd_soc_card card;
89 	u32 sample_rate;
90 	snd_pcm_format_t sample_format;
91 	u32 asrc_rate;
92 	snd_pcm_format_t asrc_format;
93 	u32 dai_fmt;
94 	char name[32];
95 };
96 
97 /**
98  * This dapm route map exsits for DPCM link only.
99  * The other routes shall go through Device Tree.
100  *
101  * Note: keep all ASRC routes in the second half
102  *	 to drop them easily for non-ASRC cases.
103  */
104 static const struct snd_soc_dapm_route audio_map[] = {
105 	/* 1st half -- Normal DAPM routes */
106 	{"Playback",  NULL, "CPU-Playback"},
107 	{"CPU-Capture",  NULL, "Capture"},
108 	/* 2nd half -- ASRC DAPM routes */
109 	{"CPU-Playback",  NULL, "ASRC-Playback"},
110 	{"ASRC-Capture",  NULL, "CPU-Capture"},
111 };
112 
113 static const struct snd_soc_dapm_route audio_map_ac97[] = {
114 	/* 1st half -- Normal DAPM routes */
115 	{"Playback",  NULL, "AC97 Playback"},
116 	{"AC97 Capture",  NULL, "Capture"},
117 	/* 2nd half -- ASRC DAPM routes */
118 	{"AC97 Playback",  NULL, "ASRC-Playback"},
119 	{"ASRC-Capture",  NULL, "AC97 Capture"},
120 };
121 
122 /* Add all possible widgets into here without being redundant */
123 static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = {
124 	SND_SOC_DAPM_LINE("Line Out Jack", NULL),
125 	SND_SOC_DAPM_LINE("Line In Jack", NULL),
126 	SND_SOC_DAPM_HP("Headphone Jack", NULL),
127 	SND_SOC_DAPM_SPK("Ext Spk", NULL),
128 	SND_SOC_DAPM_MIC("Mic Jack", NULL),
129 	SND_SOC_DAPM_MIC("AMIC", NULL),
130 	SND_SOC_DAPM_MIC("DMIC", NULL),
131 };
132 
133 static bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv)
134 {
135 	return priv->dai_fmt == SND_SOC_DAIFMT_AC97;
136 }
137 
138 static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
139 				   struct snd_pcm_hw_params *params)
140 {
141 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
142 	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
143 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
144 	struct cpu_priv *cpu_priv = &priv->cpu_priv;
145 	struct device *dev = rtd->card->dev;
146 	int ret;
147 
148 	priv->sample_rate = params_rate(params);
149 	priv->sample_format = params_format(params);
150 
151 	/*
152 	 * If codec-dai is DAI Master and all configurations are already in the
153 	 * set_bias_level(), bypass the remaining settings in hw_params().
154 	 * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS.
155 	 */
156 	if ((priv->card.set_bias_level &&
157 	     priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) ||
158 	    fsl_asoc_card_is_ac97(priv))
159 		return 0;
160 
161 	/* Specific configurations of DAIs starts from here */
162 	ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, cpu_priv->sysclk_id[tx],
163 				     cpu_priv->sysclk_freq[tx],
164 				     cpu_priv->sysclk_dir[tx]);
165 	if (ret && ret != -ENOTSUPP) {
166 		dev_err(dev, "failed to set sysclk for cpu dai\n");
167 		return ret;
168 	}
169 
170 	if (cpu_priv->slot_width) {
171 		ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2,
172 					       cpu_priv->slot_width);
173 		if (ret && ret != -ENOTSUPP) {
174 			dev_err(dev, "failed to set TDM slot for cpu dai\n");
175 			return ret;
176 		}
177 	}
178 
179 	return 0;
180 }
181 
182 static const struct snd_soc_ops fsl_asoc_card_ops = {
183 	.hw_params = fsl_asoc_card_hw_params,
184 };
185 
186 static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
187 			      struct snd_pcm_hw_params *params)
188 {
189 	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
190 	struct snd_interval *rate;
191 	struct snd_mask *mask;
192 
193 	rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
194 	rate->max = rate->min = priv->asrc_rate;
195 
196 	mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
197 	snd_mask_none(mask);
198 	snd_mask_set_format(mask, priv->asrc_format);
199 
200 	return 0;
201 }
202 
203 SND_SOC_DAILINK_DEFS(hifi,
204 	DAILINK_COMP_ARRAY(COMP_EMPTY()),
205 	DAILINK_COMP_ARRAY(COMP_EMPTY()),
206 	DAILINK_COMP_ARRAY(COMP_EMPTY()));
207 
208 SND_SOC_DAILINK_DEFS(hifi_fe,
209 	DAILINK_COMP_ARRAY(COMP_EMPTY()),
210 	DAILINK_COMP_ARRAY(COMP_DUMMY()),
211 	DAILINK_COMP_ARRAY(COMP_EMPTY()));
212 
213 SND_SOC_DAILINK_DEFS(hifi_be,
214 	DAILINK_COMP_ARRAY(COMP_EMPTY()),
215 	DAILINK_COMP_ARRAY(COMP_EMPTY()),
216 	DAILINK_COMP_ARRAY(COMP_DUMMY()));
217 
218 static struct snd_soc_dai_link fsl_asoc_card_dai[] = {
219 	/* Default ASoC DAI Link*/
220 	{
221 		.name = "HiFi",
222 		.stream_name = "HiFi",
223 		.ops = &fsl_asoc_card_ops,
224 		SND_SOC_DAILINK_REG(hifi),
225 	},
226 	/* DPCM Link between Front-End and Back-End (Optional) */
227 	{
228 		.name = "HiFi-ASRC-FE",
229 		.stream_name = "HiFi-ASRC-FE",
230 		.dpcm_playback = 1,
231 		.dpcm_capture = 1,
232 		.dynamic = 1,
233 		SND_SOC_DAILINK_REG(hifi_fe),
234 	},
235 	{
236 		.name = "HiFi-ASRC-BE",
237 		.stream_name = "HiFi-ASRC-BE",
238 		.be_hw_params_fixup = be_hw_params_fixup,
239 		.ops = &fsl_asoc_card_ops,
240 		.dpcm_playback = 1,
241 		.dpcm_capture = 1,
242 		.no_pcm = 1,
243 		SND_SOC_DAILINK_REG(hifi_be),
244 	},
245 };
246 
247 static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card,
248 					struct snd_soc_dapm_context *dapm,
249 					enum snd_soc_bias_level level)
250 {
251 	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
252 	struct snd_soc_pcm_runtime *rtd;
253 	struct snd_soc_dai *codec_dai;
254 	struct codec_priv *codec_priv = &priv->codec_priv;
255 	struct device *dev = card->dev;
256 	unsigned int pll_out;
257 	int ret;
258 
259 	rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name);
260 	codec_dai = rtd->codec_dai;
261 	if (dapm->dev != codec_dai->dev)
262 		return 0;
263 
264 	switch (level) {
265 	case SND_SOC_BIAS_PREPARE:
266 		if (dapm->bias_level != SND_SOC_BIAS_STANDBY)
267 			break;
268 
269 		if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE)
270 			pll_out = priv->sample_rate * 384;
271 		else
272 			pll_out = priv->sample_rate * 256;
273 
274 		ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id,
275 					  codec_priv->mclk_id,
276 					  codec_priv->mclk_freq, pll_out);
277 		if (ret) {
278 			dev_err(dev, "failed to start FLL: %d\n", ret);
279 			return ret;
280 		}
281 
282 		ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id,
283 					     pll_out, SND_SOC_CLOCK_IN);
284 		if (ret && ret != -ENOTSUPP) {
285 			dev_err(dev, "failed to set SYSCLK: %d\n", ret);
286 			return ret;
287 		}
288 		break;
289 
290 	case SND_SOC_BIAS_STANDBY:
291 		if (dapm->bias_level != SND_SOC_BIAS_PREPARE)
292 			break;
293 
294 		ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
295 					     codec_priv->mclk_freq,
296 					     SND_SOC_CLOCK_IN);
297 		if (ret && ret != -ENOTSUPP) {
298 			dev_err(dev, "failed to switch away from FLL: %d\n", ret);
299 			return ret;
300 		}
301 
302 		ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, 0, 0, 0);
303 		if (ret) {
304 			dev_err(dev, "failed to stop FLL: %d\n", ret);
305 			return ret;
306 		}
307 		break;
308 
309 	default:
310 		break;
311 	}
312 
313 	return 0;
314 }
315 
316 static int fsl_asoc_card_audmux_init(struct device_node *np,
317 				     struct fsl_asoc_card_priv *priv)
318 {
319 	struct device *dev = &priv->pdev->dev;
320 	u32 int_ptcr = 0, ext_ptcr = 0;
321 	int int_port, ext_port;
322 	int ret;
323 
324 	ret = of_property_read_u32(np, "mux-int-port", &int_port);
325 	if (ret) {
326 		dev_err(dev, "mux-int-port missing or invalid\n");
327 		return ret;
328 	}
329 	ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
330 	if (ret) {
331 		dev_err(dev, "mux-ext-port missing or invalid\n");
332 		return ret;
333 	}
334 
335 	/*
336 	 * The port numbering in the hardware manual starts at 1, while
337 	 * the AUDMUX API expects it starts at 0.
338 	 */
339 	int_port--;
340 	ext_port--;
341 
342 	/*
343 	 * Use asynchronous mode (6 wires) for all cases except AC97.
344 	 * If only 4 wires are needed, just set SSI into
345 	 * synchronous mode and enable 4 PADs in IOMUX.
346 	 */
347 	switch (priv->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) {
348 	case SND_SOC_DAIFMT_CBM_CFM:
349 		int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
350 			   IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
351 			   IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
352 			   IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
353 			   IMX_AUDMUX_V2_PTCR_RFSDIR |
354 			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
355 			   IMX_AUDMUX_V2_PTCR_TFSDIR |
356 			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
357 		break;
358 	case SND_SOC_DAIFMT_CBM_CFS:
359 		int_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
360 			   IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
361 			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
362 			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
363 		ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
364 			   IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
365 			   IMX_AUDMUX_V2_PTCR_RFSDIR |
366 			   IMX_AUDMUX_V2_PTCR_TFSDIR;
367 		break;
368 	case SND_SOC_DAIFMT_CBS_CFM:
369 		int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
370 			   IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
371 			   IMX_AUDMUX_V2_PTCR_RFSDIR |
372 			   IMX_AUDMUX_V2_PTCR_TFSDIR;
373 		ext_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
374 			   IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
375 			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
376 			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
377 		break;
378 	case SND_SOC_DAIFMT_CBS_CFS:
379 		ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
380 			   IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
381 			   IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
382 			   IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
383 			   IMX_AUDMUX_V2_PTCR_RFSDIR |
384 			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
385 			   IMX_AUDMUX_V2_PTCR_TFSDIR |
386 			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
387 		break;
388 	default:
389 		if (!fsl_asoc_card_is_ac97(priv))
390 			return -EINVAL;
391 	}
392 
393 	if (fsl_asoc_card_is_ac97(priv)) {
394 		int_ptcr = IMX_AUDMUX_V2_PTCR_SYN |
395 			   IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
396 			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
397 		ext_ptcr = IMX_AUDMUX_V2_PTCR_SYN |
398 			   IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
399 			   IMX_AUDMUX_V2_PTCR_TFSDIR;
400 	}
401 
402 	/* Asynchronous mode can not be set along with RCLKDIR */
403 	if (!fsl_asoc_card_is_ac97(priv)) {
404 		unsigned int pdcr =
405 				IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port);
406 
407 		ret = imx_audmux_v2_configure_port(int_port, 0,
408 						   pdcr);
409 		if (ret) {
410 			dev_err(dev, "audmux internal port setup failed\n");
411 			return ret;
412 		}
413 	}
414 
415 	ret = imx_audmux_v2_configure_port(int_port, int_ptcr,
416 					   IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
417 	if (ret) {
418 		dev_err(dev, "audmux internal port setup failed\n");
419 		return ret;
420 	}
421 
422 	if (!fsl_asoc_card_is_ac97(priv)) {
423 		unsigned int pdcr =
424 				IMX_AUDMUX_V2_PDCR_RXDSEL(int_port);
425 
426 		ret = imx_audmux_v2_configure_port(ext_port, 0,
427 						   pdcr);
428 		if (ret) {
429 			dev_err(dev, "audmux external port setup failed\n");
430 			return ret;
431 		}
432 	}
433 
434 	ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr,
435 					   IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
436 	if (ret) {
437 		dev_err(dev, "audmux external port setup failed\n");
438 		return ret;
439 	}
440 
441 	return 0;
442 }
443 
444 static int fsl_asoc_card_late_probe(struct snd_soc_card *card)
445 {
446 	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
447 	struct snd_soc_pcm_runtime *rtd = list_first_entry(
448 			&card->rtd_list, struct snd_soc_pcm_runtime, list);
449 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
450 	struct codec_priv *codec_priv = &priv->codec_priv;
451 	struct device *dev = card->dev;
452 	int ret;
453 
454 	if (fsl_asoc_card_is_ac97(priv)) {
455 #if IS_ENABLED(CONFIG_SND_AC97_CODEC)
456 		struct snd_soc_component *component = rtd->codec_dai->component;
457 		struct snd_ac97 *ac97 = snd_soc_component_get_drvdata(component);
458 
459 		/*
460 		 * Use slots 3/4 for S/PDIF so SSI won't try to enable
461 		 * other slots and send some samples there
462 		 * due to SLOTREQ bits for S/PDIF received from codec
463 		 */
464 		snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS,
465 				     AC97_EA_SPSA_SLOT_MASK, AC97_EA_SPSA_3_4);
466 #endif
467 
468 		return 0;
469 	}
470 
471 	ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
472 				     codec_priv->mclk_freq, SND_SOC_CLOCK_IN);
473 	if (ret && ret != -ENOTSUPP) {
474 		dev_err(dev, "failed to set sysclk in %s\n", __func__);
475 		return ret;
476 	}
477 
478 	return 0;
479 }
480 
481 static int fsl_asoc_card_probe(struct platform_device *pdev)
482 {
483 	struct device_node *cpu_np, *codec_np, *asrc_np;
484 	struct device_node *np = pdev->dev.of_node;
485 	struct platform_device *asrc_pdev = NULL;
486 	struct platform_device *cpu_pdev;
487 	struct fsl_asoc_card_priv *priv;
488 	struct i2c_client *codec_dev;
489 	const char *codec_dai_name;
490 	u32 width;
491 	int ret;
492 
493 	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
494 	if (!priv)
495 		return -ENOMEM;
496 
497 	cpu_np = of_parse_phandle(np, "audio-cpu", 0);
498 	/* Give a chance to old DT binding */
499 	if (!cpu_np)
500 		cpu_np = of_parse_phandle(np, "ssi-controller", 0);
501 	if (!cpu_np) {
502 		dev_err(&pdev->dev, "CPU phandle missing or invalid\n");
503 		ret = -EINVAL;
504 		goto fail;
505 	}
506 
507 	cpu_pdev = of_find_device_by_node(cpu_np);
508 	if (!cpu_pdev) {
509 		dev_err(&pdev->dev, "failed to find CPU DAI device\n");
510 		ret = -EINVAL;
511 		goto fail;
512 	}
513 
514 	codec_np = of_parse_phandle(np, "audio-codec", 0);
515 	if (codec_np)
516 		codec_dev = of_find_i2c_device_by_node(codec_np);
517 	else
518 		codec_dev = NULL;
519 
520 	asrc_np = of_parse_phandle(np, "audio-asrc", 0);
521 	if (asrc_np)
522 		asrc_pdev = of_find_device_by_node(asrc_np);
523 
524 	/* Get the MCLK rate only, and leave it controlled by CODEC drivers */
525 	if (codec_dev) {
526 		struct clk *codec_clk = clk_get(&codec_dev->dev, NULL);
527 
528 		if (!IS_ERR(codec_clk)) {
529 			priv->codec_priv.mclk_freq = clk_get_rate(codec_clk);
530 			clk_put(codec_clk);
531 		}
532 	}
533 
534 	/* Default sample rate and format, will be updated in hw_params() */
535 	priv->sample_rate = 44100;
536 	priv->sample_format = SNDRV_PCM_FORMAT_S16_LE;
537 
538 	/* Assign a default DAI format, and allow each card to overwrite it */
539 	priv->dai_fmt = DAI_FMT_BASE;
540 
541 	/* Diversify the card configurations */
542 	if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
543 		codec_dai_name = "cs42888";
544 		priv->card.set_bias_level = NULL;
545 		priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq;
546 		priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq;
547 		priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT;
548 		priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT;
549 		priv->cpu_priv.slot_width = 32;
550 		priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
551 	} else if (of_device_is_compatible(np, "fsl,imx-audio-cs427x")) {
552 		codec_dai_name = "cs4271-hifi";
553 		priv->codec_priv.mclk_id = CS427x_SYSCLK_MCLK;
554 		priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
555 	} else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) {
556 		codec_dai_name = "sgtl5000";
557 		priv->codec_priv.mclk_id = SGTL5000_SYSCLK;
558 		priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
559 	} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) {
560 		codec_dai_name = "wm8962";
561 		priv->card.set_bias_level = fsl_asoc_card_set_bias_level;
562 		priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK;
563 		priv->codec_priv.fll_id = WM8962_SYSCLK_FLL;
564 		priv->codec_priv.pll_id = WM8962_FLL;
565 		priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
566 	} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) {
567 		codec_dai_name = "wm8960-hifi";
568 		priv->card.set_bias_level = fsl_asoc_card_set_bias_level;
569 		priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO;
570 		priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO;
571 		priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
572 	} else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) {
573 		codec_dai_name = "ac97-hifi";
574 		priv->card.set_bias_level = NULL;
575 		priv->dai_fmt = SND_SOC_DAIFMT_AC97;
576 	} else {
577 		dev_err(&pdev->dev, "unknown Device Tree compatible\n");
578 		ret = -EINVAL;
579 		goto asrc_fail;
580 	}
581 
582 	if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) {
583 		dev_err(&pdev->dev, "failed to find codec device\n");
584 		ret = -EINVAL;
585 		goto asrc_fail;
586 	}
587 
588 	/* Common settings for corresponding Freescale CPU DAI driver */
589 	if (of_node_name_eq(cpu_np, "ssi")) {
590 		/* Only SSI needs to configure AUDMUX */
591 		ret = fsl_asoc_card_audmux_init(np, priv);
592 		if (ret) {
593 			dev_err(&pdev->dev, "failed to init audmux\n");
594 			goto asrc_fail;
595 		}
596 	} else if (of_node_name_eq(cpu_np, "esai")) {
597 		priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL;
598 		priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL;
599 	} else if (of_node_name_eq(cpu_np, "sai")) {
600 		priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1;
601 		priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1;
602 	}
603 
604 	snprintf(priv->name, sizeof(priv->name), "%s-audio",
605 		 fsl_asoc_card_is_ac97(priv) ? "ac97" :
606 		 codec_dev->name);
607 
608 	/* Initialize sound card */
609 	priv->pdev = pdev;
610 	priv->card.dev = &pdev->dev;
611 	priv->card.name = priv->name;
612 	priv->card.dai_link = priv->dai_link;
613 	priv->card.dapm_routes = fsl_asoc_card_is_ac97(priv) ?
614 				 audio_map_ac97 : audio_map;
615 	priv->card.late_probe = fsl_asoc_card_late_probe;
616 	priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
617 	priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;
618 	priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets);
619 
620 	/* Drop the second half of DAPM routes -- ASRC */
621 	if (!asrc_pdev)
622 		priv->card.num_dapm_routes /= 2;
623 
624 	memcpy(priv->dai_link, fsl_asoc_card_dai,
625 	       sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
626 
627 	ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing");
628 	if (ret) {
629 		dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret);
630 		goto asrc_fail;
631 	}
632 
633 	/* Normal DAI Link */
634 	priv->dai_link[0].cpus->of_node = cpu_np;
635 	priv->dai_link[0].codecs->dai_name = codec_dai_name;
636 
637 	if (!fsl_asoc_card_is_ac97(priv))
638 		priv->dai_link[0].codecs->of_node = codec_np;
639 	else {
640 		u32 idx;
641 
642 		ret = of_property_read_u32(cpu_np, "cell-index", &idx);
643 		if (ret) {
644 			dev_err(&pdev->dev,
645 				"cannot get CPU index property\n");
646 			goto asrc_fail;
647 		}
648 
649 		priv->dai_link[0].codecs->name =
650 				devm_kasprintf(&pdev->dev, GFP_KERNEL,
651 					       "ac97-codec.%u",
652 					       (unsigned int)idx);
653 		if (!priv->dai_link[0].codecs->name) {
654 			ret = -ENOMEM;
655 			goto asrc_fail;
656 		}
657 	}
658 
659 	priv->dai_link[0].platforms->of_node = cpu_np;
660 	priv->dai_link[0].dai_fmt = priv->dai_fmt;
661 	priv->card.num_links = 1;
662 
663 	if (asrc_pdev) {
664 		/* DPCM DAI Links only if ASRC exsits */
665 		priv->dai_link[1].cpus->of_node = asrc_np;
666 		priv->dai_link[1].platforms->of_node = asrc_np;
667 		priv->dai_link[2].codecs->dai_name = codec_dai_name;
668 		priv->dai_link[2].codecs->of_node = codec_np;
669 		priv->dai_link[2].codecs->name =
670 				priv->dai_link[0].codecs->name;
671 		priv->dai_link[2].cpus->of_node = cpu_np;
672 		priv->dai_link[2].dai_fmt = priv->dai_fmt;
673 		priv->card.num_links = 3;
674 
675 		ret = of_property_read_u32(asrc_np, "fsl,asrc-rate",
676 					   &priv->asrc_rate);
677 		if (ret) {
678 			dev_err(&pdev->dev, "failed to get output rate\n");
679 			ret = -EINVAL;
680 			goto asrc_fail;
681 		}
682 
683 		ret = of_property_read_u32(asrc_np, "fsl,asrc-width", &width);
684 		if (ret) {
685 			dev_err(&pdev->dev, "failed to get output rate\n");
686 			ret = -EINVAL;
687 			goto asrc_fail;
688 		}
689 
690 		if (width == 24)
691 			priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE;
692 		else
693 			priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE;
694 	}
695 
696 	/* Finish card registering */
697 	platform_set_drvdata(pdev, priv);
698 	snd_soc_card_set_drvdata(&priv->card, priv);
699 
700 	ret = devm_snd_soc_register_card(&pdev->dev, &priv->card);
701 	if (ret && ret != -EPROBE_DEFER)
702 		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
703 
704 asrc_fail:
705 	of_node_put(asrc_np);
706 	of_node_put(codec_np);
707 	put_device(&cpu_pdev->dev);
708 fail:
709 	of_node_put(cpu_np);
710 
711 	return ret;
712 }
713 
714 static const struct of_device_id fsl_asoc_card_dt_ids[] = {
715 	{ .compatible = "fsl,imx-audio-ac97", },
716 	{ .compatible = "fsl,imx-audio-cs42888", },
717 	{ .compatible = "fsl,imx-audio-cs427x", },
718 	{ .compatible = "fsl,imx-audio-sgtl5000", },
719 	{ .compatible = "fsl,imx-audio-wm8962", },
720 	{ .compatible = "fsl,imx-audio-wm8960", },
721 	{}
722 };
723 MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids);
724 
725 static struct platform_driver fsl_asoc_card_driver = {
726 	.probe = fsl_asoc_card_probe,
727 	.driver = {
728 		.name = "fsl-asoc-card",
729 		.pm = &snd_soc_pm_ops,
730 		.of_match_table = fsl_asoc_card_dt_ids,
731 	},
732 };
733 module_platform_driver(fsl_asoc_card_driver);
734 
735 MODULE_DESCRIPTION("Freescale Generic ASoC Sound Card driver with ASRC");
736 MODULE_AUTHOR("Nicolin Chen <nicoleotsuka@gmail.com>");
737 MODULE_ALIAS("platform:fsl-asoc-card");
738 MODULE_LICENSE("GPL");
739