xref: /openbmc/linux/sound/soc/fsl/fsl-asoc-card.c (revision 52451502)
1 // SPDX-License-Identifier: GPL-2.0
2 //
3 // Freescale Generic ASoC Sound Card driver with ASRC
4 //
5 // Copyright (C) 2014 Freescale Semiconductor, Inc.
6 //
7 // Author: Nicolin Chen <nicoleotsuka@gmail.com>
8 
9 #include <linux/clk.h>
10 #include <linux/i2c.h>
11 #include <linux/module.h>
12 #include <linux/of_platform.h>
13 #if IS_ENABLED(CONFIG_SND_AC97_CODEC)
14 #include <sound/ac97_codec.h>
15 #endif
16 #include <sound/pcm_params.h>
17 #include <sound/soc.h>
18 #include <sound/jack.h>
19 #include <sound/simple_card_utils.h>
20 
21 #include "fsl_esai.h"
22 #include "fsl_sai.h"
23 #include "imx-audmux.h"
24 
25 #include "../codecs/sgtl5000.h"
26 #include "../codecs/wm8962.h"
27 #include "../codecs/wm8960.h"
28 #include "../codecs/wm8994.h"
29 #include "../codecs/tlv320aic31xx.h"
30 #include "../codecs/nau8822.h"
31 
32 #define DRIVER_NAME "fsl-asoc-card"
33 
34 #define CS427x_SYSCLK_MCLK 0
35 
36 #define RX 0
37 #define TX 1
38 
39 /* Default DAI format without Master and Slave flag */
40 #define DAI_FMT_BASE (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF)
41 
42 /**
43  * struct codec_priv - CODEC private data
44  * @mclk_freq: Clock rate of MCLK
45  * @free_freq: Clock rate of MCLK for hw_free()
46  * @mclk_id: MCLK (or main clock) id for set_sysclk()
47  * @fll_id: FLL (or secordary clock) id for set_sysclk()
48  * @pll_id: PLL id for set_pll()
49  */
50 struct codec_priv {
51 	struct clk *mclk;
52 	unsigned long mclk_freq;
53 	unsigned long free_freq;
54 	u32 mclk_id;
55 	int fll_id;
56 	int pll_id;
57 };
58 
59 /**
60  * struct cpu_priv - CPU private data
61  * @sysclk_freq: SYSCLK rates for set_sysclk()
62  * @sysclk_dir: SYSCLK directions for set_sysclk()
63  * @sysclk_id: SYSCLK ids for set_sysclk()
64  * @slot_width: Slot width of each frame
65  * @slot_num: Number of slots of each frame
66  *
67  * Note: [1] for tx and [0] for rx
68  */
69 struct cpu_priv {
70 	unsigned long sysclk_freq[2];
71 	u32 sysclk_dir[2];
72 	u32 sysclk_id[2];
73 	u32 slot_width;
74 	u32 slot_num;
75 };
76 
77 /**
78  * struct fsl_asoc_card_priv - Freescale Generic ASOC card private data
79  * @dai_link: DAI link structure including normal one and DPCM link
80  * @hp_jack: Headphone Jack structure
81  * @mic_jack: Microphone Jack structure
82  * @pdev: platform device pointer
83  * @codec_priv: CODEC private data
84  * @cpu_priv: CPU private data
85  * @card: ASoC card structure
86  * @streams: Mask of current active streams
87  * @sample_rate: Current sample rate
88  * @sample_format: Current sample format
89  * @asrc_rate: ASRC sample rate used by Back-Ends
90  * @asrc_format: ASRC sample format used by Back-Ends
91  * @dai_fmt: DAI format between CPU and CODEC
92  * @name: Card name
93  */
94 
95 struct fsl_asoc_card_priv {
96 	struct snd_soc_dai_link dai_link[3];
97 	struct asoc_simple_jack hp_jack;
98 	struct asoc_simple_jack mic_jack;
99 	struct platform_device *pdev;
100 	struct codec_priv codec_priv;
101 	struct cpu_priv cpu_priv;
102 	struct snd_soc_card card;
103 	u8 streams;
104 	u32 sample_rate;
105 	snd_pcm_format_t sample_format;
106 	u32 asrc_rate;
107 	snd_pcm_format_t asrc_format;
108 	u32 dai_fmt;
109 	char name[32];
110 };
111 
112 /*
113  * This dapm route map exists for DPCM link only.
114  * The other routes shall go through Device Tree.
115  *
116  * Note: keep all ASRC routes in the second half
117  *	 to drop them easily for non-ASRC cases.
118  */
119 static const struct snd_soc_dapm_route audio_map[] = {
120 	/* 1st half -- Normal DAPM routes */
121 	{"Playback",  NULL, "CPU-Playback"},
122 	{"CPU-Capture",  NULL, "Capture"},
123 	/* 2nd half -- ASRC DAPM routes */
124 	{"CPU-Playback",  NULL, "ASRC-Playback"},
125 	{"ASRC-Capture",  NULL, "CPU-Capture"},
126 };
127 
128 static const struct snd_soc_dapm_route audio_map_ac97[] = {
129 	/* 1st half -- Normal DAPM routes */
130 	{"AC97 Playback",  NULL, "CPU AC97 Playback"},
131 	{"CPU AC97 Capture",  NULL, "AC97 Capture"},
132 	/* 2nd half -- ASRC DAPM routes */
133 	{"CPU AC97 Playback",  NULL, "ASRC-Playback"},
134 	{"ASRC-Capture",  NULL, "CPU AC97 Capture"},
135 };
136 
137 static const struct snd_soc_dapm_route audio_map_tx[] = {
138 	/* 1st half -- Normal DAPM routes */
139 	{"Playback",  NULL, "CPU-Playback"},
140 	/* 2nd half -- ASRC DAPM routes */
141 	{"CPU-Playback",  NULL, "ASRC-Playback"},
142 };
143 
144 static const struct snd_soc_dapm_route audio_map_rx[] = {
145 	/* 1st half -- Normal DAPM routes */
146 	{"CPU-Capture",  NULL, "Capture"},
147 	/* 2nd half -- ASRC DAPM routes */
148 	{"ASRC-Capture",  NULL, "CPU-Capture"},
149 };
150 
151 /* Add all possible widgets into here without being redundant */
152 static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = {
153 	SND_SOC_DAPM_LINE("Line Out Jack", NULL),
154 	SND_SOC_DAPM_LINE("Line In Jack", NULL),
155 	SND_SOC_DAPM_HP("Headphone Jack", NULL),
156 	SND_SOC_DAPM_SPK("Ext Spk", NULL),
157 	SND_SOC_DAPM_MIC("Mic Jack", NULL),
158 	SND_SOC_DAPM_MIC("AMIC", NULL),
159 	SND_SOC_DAPM_MIC("DMIC", NULL),
160 };
161 
162 static bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv)
163 {
164 	return priv->dai_fmt == SND_SOC_DAIFMT_AC97;
165 }
166 
167 static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
168 				   struct snd_pcm_hw_params *params)
169 {
170 	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
171 	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
172 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
173 	struct codec_priv *codec_priv = &priv->codec_priv;
174 	struct cpu_priv *cpu_priv = &priv->cpu_priv;
175 	struct device *dev = rtd->card->dev;
176 	unsigned int pll_out;
177 	int ret;
178 
179 	priv->sample_rate = params_rate(params);
180 	priv->sample_format = params_format(params);
181 	priv->streams |= BIT(substream->stream);
182 
183 	if (fsl_asoc_card_is_ac97(priv))
184 		return 0;
185 
186 	/* Specific configurations of DAIs starts from here */
187 	ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), cpu_priv->sysclk_id[tx],
188 				     cpu_priv->sysclk_freq[tx],
189 				     cpu_priv->sysclk_dir[tx]);
190 	if (ret && ret != -ENOTSUPP) {
191 		dev_err(dev, "failed to set sysclk for cpu dai\n");
192 		goto fail;
193 	}
194 
195 	if (cpu_priv->slot_width) {
196 		if (!cpu_priv->slot_num)
197 			cpu_priv->slot_num = 2;
198 
199 		ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3,
200 					       cpu_priv->slot_num,
201 					       cpu_priv->slot_width);
202 		if (ret && ret != -ENOTSUPP) {
203 			dev_err(dev, "failed to set TDM slot for cpu dai\n");
204 			goto fail;
205 		}
206 	}
207 
208 	/* Specific configuration for PLL */
209 	if (codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) {
210 		if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE)
211 			pll_out = priv->sample_rate * 384;
212 		else
213 			pll_out = priv->sample_rate * 256;
214 
215 		ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0),
216 					  codec_priv->pll_id,
217 					  codec_priv->mclk_id,
218 					  codec_priv->mclk_freq, pll_out);
219 		if (ret) {
220 			dev_err(dev, "failed to start FLL: %d\n", ret);
221 			goto fail;
222 		}
223 
224 		ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0),
225 					     codec_priv->fll_id,
226 					     pll_out, SND_SOC_CLOCK_IN);
227 
228 		if (ret && ret != -ENOTSUPP) {
229 			dev_err(dev, "failed to set SYSCLK: %d\n", ret);
230 			goto fail;
231 		}
232 	}
233 
234 	return 0;
235 
236 fail:
237 	priv->streams &= ~BIT(substream->stream);
238 	return ret;
239 }
240 
241 static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream)
242 {
243 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
244 	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
245 	struct codec_priv *codec_priv = &priv->codec_priv;
246 	struct device *dev = rtd->card->dev;
247 	int ret;
248 
249 	priv->streams &= ~BIT(substream->stream);
250 
251 	if (!priv->streams && codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) {
252 		/* Force freq to be free_freq to avoid error message in codec */
253 		ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0),
254 					     codec_priv->mclk_id,
255 					     codec_priv->free_freq,
256 					     SND_SOC_CLOCK_IN);
257 		if (ret) {
258 			dev_err(dev, "failed to switch away from FLL: %d\n", ret);
259 			return ret;
260 		}
261 
262 		ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0),
263 					  codec_priv->pll_id, 0, 0, 0);
264 		if (ret && ret != -ENOTSUPP) {
265 			dev_err(dev, "failed to stop FLL: %d\n", ret);
266 			return ret;
267 		}
268 	}
269 
270 	return 0;
271 }
272 
273 static const struct snd_soc_ops fsl_asoc_card_ops = {
274 	.hw_params = fsl_asoc_card_hw_params,
275 	.hw_free = fsl_asoc_card_hw_free,
276 };
277 
278 static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
279 			      struct snd_pcm_hw_params *params)
280 {
281 	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
282 	struct snd_interval *rate;
283 	struct snd_mask *mask;
284 
285 	rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
286 	rate->max = rate->min = priv->asrc_rate;
287 
288 	mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
289 	snd_mask_none(mask);
290 	snd_mask_set_format(mask, priv->asrc_format);
291 
292 	return 0;
293 }
294 
295 SND_SOC_DAILINK_DEFS(hifi,
296 	DAILINK_COMP_ARRAY(COMP_EMPTY()),
297 	DAILINK_COMP_ARRAY(COMP_EMPTY()),
298 	DAILINK_COMP_ARRAY(COMP_EMPTY()));
299 
300 SND_SOC_DAILINK_DEFS(hifi_fe,
301 	DAILINK_COMP_ARRAY(COMP_EMPTY()),
302 	DAILINK_COMP_ARRAY(COMP_DUMMY()),
303 	DAILINK_COMP_ARRAY(COMP_EMPTY()));
304 
305 SND_SOC_DAILINK_DEFS(hifi_be,
306 	DAILINK_COMP_ARRAY(COMP_EMPTY()),
307 	DAILINK_COMP_ARRAY(COMP_EMPTY()),
308 	DAILINK_COMP_ARRAY(COMP_DUMMY()));
309 
310 static const struct snd_soc_dai_link fsl_asoc_card_dai[] = {
311 	/* Default ASoC DAI Link*/
312 	{
313 		.name = "HiFi",
314 		.stream_name = "HiFi",
315 		.ops = &fsl_asoc_card_ops,
316 		SND_SOC_DAILINK_REG(hifi),
317 	},
318 	/* DPCM Link between Front-End and Back-End (Optional) */
319 	{
320 		.name = "HiFi-ASRC-FE",
321 		.stream_name = "HiFi-ASRC-FE",
322 		.dpcm_playback = 1,
323 		.dpcm_capture = 1,
324 		.dynamic = 1,
325 		SND_SOC_DAILINK_REG(hifi_fe),
326 	},
327 	{
328 		.name = "HiFi-ASRC-BE",
329 		.stream_name = "HiFi-ASRC-BE",
330 		.be_hw_params_fixup = be_hw_params_fixup,
331 		.ops = &fsl_asoc_card_ops,
332 		.dpcm_playback = 1,
333 		.dpcm_capture = 1,
334 		.no_pcm = 1,
335 		SND_SOC_DAILINK_REG(hifi_be),
336 	},
337 };
338 
339 static int fsl_asoc_card_audmux_init(struct device_node *np,
340 				     struct fsl_asoc_card_priv *priv)
341 {
342 	struct device *dev = &priv->pdev->dev;
343 	u32 int_ptcr = 0, ext_ptcr = 0;
344 	int int_port, ext_port;
345 	int ret;
346 
347 	ret = of_property_read_u32(np, "mux-int-port", &int_port);
348 	if (ret) {
349 		dev_err(dev, "mux-int-port missing or invalid\n");
350 		return ret;
351 	}
352 	ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
353 	if (ret) {
354 		dev_err(dev, "mux-ext-port missing or invalid\n");
355 		return ret;
356 	}
357 
358 	/*
359 	 * The port numbering in the hardware manual starts at 1, while
360 	 * the AUDMUX API expects it starts at 0.
361 	 */
362 	int_port--;
363 	ext_port--;
364 
365 	/*
366 	 * Use asynchronous mode (6 wires) for all cases except AC97.
367 	 * If only 4 wires are needed, just set SSI into
368 	 * synchronous mode and enable 4 PADs in IOMUX.
369 	 */
370 	switch (priv->dai_fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) {
371 	case SND_SOC_DAIFMT_CBP_CFP:
372 		int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
373 			   IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
374 			   IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
375 			   IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
376 			   IMX_AUDMUX_V2_PTCR_RFSDIR |
377 			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
378 			   IMX_AUDMUX_V2_PTCR_TFSDIR |
379 			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
380 		break;
381 	case SND_SOC_DAIFMT_CBP_CFC:
382 		int_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
383 			   IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
384 			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
385 			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
386 		ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
387 			   IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
388 			   IMX_AUDMUX_V2_PTCR_RFSDIR |
389 			   IMX_AUDMUX_V2_PTCR_TFSDIR;
390 		break;
391 	case SND_SOC_DAIFMT_CBC_CFP:
392 		int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
393 			   IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
394 			   IMX_AUDMUX_V2_PTCR_RFSDIR |
395 			   IMX_AUDMUX_V2_PTCR_TFSDIR;
396 		ext_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
397 			   IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
398 			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
399 			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
400 		break;
401 	case SND_SOC_DAIFMT_CBC_CFC:
402 		ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
403 			   IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
404 			   IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
405 			   IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
406 			   IMX_AUDMUX_V2_PTCR_RFSDIR |
407 			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
408 			   IMX_AUDMUX_V2_PTCR_TFSDIR |
409 			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
410 		break;
411 	default:
412 		if (!fsl_asoc_card_is_ac97(priv))
413 			return -EINVAL;
414 	}
415 
416 	if (fsl_asoc_card_is_ac97(priv)) {
417 		int_ptcr = IMX_AUDMUX_V2_PTCR_SYN |
418 			   IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
419 			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
420 		ext_ptcr = IMX_AUDMUX_V2_PTCR_SYN |
421 			   IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
422 			   IMX_AUDMUX_V2_PTCR_TFSDIR;
423 	}
424 
425 	/* Asynchronous mode can not be set along with RCLKDIR */
426 	if (!fsl_asoc_card_is_ac97(priv)) {
427 		unsigned int pdcr =
428 				IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port);
429 
430 		ret = imx_audmux_v2_configure_port(int_port, 0,
431 						   pdcr);
432 		if (ret) {
433 			dev_err(dev, "audmux internal port setup failed\n");
434 			return ret;
435 		}
436 	}
437 
438 	ret = imx_audmux_v2_configure_port(int_port, int_ptcr,
439 					   IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
440 	if (ret) {
441 		dev_err(dev, "audmux internal port setup failed\n");
442 		return ret;
443 	}
444 
445 	if (!fsl_asoc_card_is_ac97(priv)) {
446 		unsigned int pdcr =
447 				IMX_AUDMUX_V2_PDCR_RXDSEL(int_port);
448 
449 		ret = imx_audmux_v2_configure_port(ext_port, 0,
450 						   pdcr);
451 		if (ret) {
452 			dev_err(dev, "audmux external port setup failed\n");
453 			return ret;
454 		}
455 	}
456 
457 	ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr,
458 					   IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
459 	if (ret) {
460 		dev_err(dev, "audmux external port setup failed\n");
461 		return ret;
462 	}
463 
464 	return 0;
465 }
466 
467 static int hp_jack_event(struct notifier_block *nb, unsigned long event,
468 			 void *data)
469 {
470 	struct snd_soc_jack *jack = (struct snd_soc_jack *)data;
471 	struct snd_soc_dapm_context *dapm = &jack->card->dapm;
472 
473 	if (event & SND_JACK_HEADPHONE)
474 		/* Disable speaker if headphone is plugged in */
475 		return snd_soc_dapm_disable_pin(dapm, "Ext Spk");
476 	else
477 		return snd_soc_dapm_enable_pin(dapm, "Ext Spk");
478 }
479 
480 static struct notifier_block hp_jack_nb = {
481 	.notifier_call = hp_jack_event,
482 };
483 
484 static int mic_jack_event(struct notifier_block *nb, unsigned long event,
485 			  void *data)
486 {
487 	struct snd_soc_jack *jack = (struct snd_soc_jack *)data;
488 	struct snd_soc_dapm_context *dapm = &jack->card->dapm;
489 
490 	if (event & SND_JACK_MICROPHONE)
491 		/* Disable dmic if microphone is plugged in */
492 		return snd_soc_dapm_disable_pin(dapm, "DMIC");
493 	else
494 		return snd_soc_dapm_enable_pin(dapm, "DMIC");
495 }
496 
497 static struct notifier_block mic_jack_nb = {
498 	.notifier_call = mic_jack_event,
499 };
500 
501 static int fsl_asoc_card_late_probe(struct snd_soc_card *card)
502 {
503 	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
504 	struct snd_soc_pcm_runtime *rtd = list_first_entry(
505 			&card->rtd_list, struct snd_soc_pcm_runtime, list);
506 	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
507 	struct codec_priv *codec_priv = &priv->codec_priv;
508 	struct device *dev = card->dev;
509 	int ret;
510 
511 	if (fsl_asoc_card_is_ac97(priv)) {
512 #if IS_ENABLED(CONFIG_SND_AC97_CODEC)
513 		struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
514 		struct snd_ac97 *ac97 = snd_soc_component_get_drvdata(component);
515 
516 		/*
517 		 * Use slots 3/4 for S/PDIF so SSI won't try to enable
518 		 * other slots and send some samples there
519 		 * due to SLOTREQ bits for S/PDIF received from codec
520 		 */
521 		snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS,
522 				     AC97_EA_SPSA_SLOT_MASK, AC97_EA_SPSA_3_4);
523 #endif
524 
525 		return 0;
526 	}
527 
528 	ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
529 				     codec_priv->mclk_freq, SND_SOC_CLOCK_IN);
530 	if (ret && ret != -ENOTSUPP) {
531 		dev_err(dev, "failed to set sysclk in %s\n", __func__);
532 		return ret;
533 	}
534 
535 	if (!IS_ERR_OR_NULL(codec_priv->mclk))
536 		clk_prepare_enable(codec_priv->mclk);
537 
538 	return 0;
539 }
540 
541 static int fsl_asoc_card_probe(struct platform_device *pdev)
542 {
543 	struct device_node *cpu_np, *codec_np, *asrc_np;
544 	struct device_node *np = pdev->dev.of_node;
545 	struct platform_device *asrc_pdev = NULL;
546 	struct device_node *bitclkprovider = NULL;
547 	struct device_node *frameprovider = NULL;
548 	struct platform_device *cpu_pdev;
549 	struct fsl_asoc_card_priv *priv;
550 	struct device *codec_dev = NULL;
551 	const char *codec_dai_name;
552 	const char *codec_dev_name;
553 	u32 asrc_fmt = 0;
554 	u32 width;
555 	int ret;
556 
557 	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
558 	if (!priv)
559 		return -ENOMEM;
560 
561 	cpu_np = of_parse_phandle(np, "audio-cpu", 0);
562 	/* Give a chance to old DT binding */
563 	if (!cpu_np)
564 		cpu_np = of_parse_phandle(np, "ssi-controller", 0);
565 	if (!cpu_np) {
566 		dev_err(&pdev->dev, "CPU phandle missing or invalid\n");
567 		ret = -EINVAL;
568 		goto fail;
569 	}
570 
571 	cpu_pdev = of_find_device_by_node(cpu_np);
572 	if (!cpu_pdev) {
573 		dev_err(&pdev->dev, "failed to find CPU DAI device\n");
574 		ret = -EINVAL;
575 		goto fail;
576 	}
577 
578 	codec_np = of_parse_phandle(np, "audio-codec", 0);
579 	if (codec_np) {
580 		struct platform_device *codec_pdev;
581 		struct i2c_client *codec_i2c;
582 
583 		codec_i2c = of_find_i2c_device_by_node(codec_np);
584 		if (codec_i2c) {
585 			codec_dev = &codec_i2c->dev;
586 			codec_dev_name = codec_i2c->name;
587 		}
588 		if (!codec_dev) {
589 			codec_pdev = of_find_device_by_node(codec_np);
590 			if (codec_pdev) {
591 				codec_dev = &codec_pdev->dev;
592 				codec_dev_name = codec_pdev->name;
593 			}
594 		}
595 	}
596 
597 	asrc_np = of_parse_phandle(np, "audio-asrc", 0);
598 	if (asrc_np)
599 		asrc_pdev = of_find_device_by_node(asrc_np);
600 
601 	/* Get the MCLK rate only, and leave it controlled by CODEC drivers */
602 	if (codec_dev) {
603 		struct clk *codec_clk = clk_get(codec_dev, NULL);
604 
605 		if (!IS_ERR(codec_clk)) {
606 			priv->codec_priv.mclk_freq = clk_get_rate(codec_clk);
607 			clk_put(codec_clk);
608 		}
609 	}
610 
611 	/* Default sample rate and format, will be updated in hw_params() */
612 	priv->sample_rate = 44100;
613 	priv->sample_format = SNDRV_PCM_FORMAT_S16_LE;
614 
615 	/* Assign a default DAI format, and allow each card to overwrite it */
616 	priv->dai_fmt = DAI_FMT_BASE;
617 
618 	memcpy(priv->dai_link, fsl_asoc_card_dai,
619 	       sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
620 
621 	priv->card.dapm_routes = audio_map;
622 	priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
623 	priv->card.driver_name = DRIVER_NAME;
624 
625 	priv->codec_priv.fll_id = -1;
626 	priv->codec_priv.pll_id = -1;
627 
628 	/* Diversify the card configurations */
629 	if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
630 		codec_dai_name = "cs42888";
631 		priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq;
632 		priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq;
633 		priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT;
634 		priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT;
635 		priv->cpu_priv.slot_width = 32;
636 		priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC;
637 	} else if (of_device_is_compatible(np, "fsl,imx-audio-cs427x")) {
638 		codec_dai_name = "cs4271-hifi";
639 		priv->codec_priv.mclk_id = CS427x_SYSCLK_MCLK;
640 		priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
641 	} else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) {
642 		codec_dai_name = "sgtl5000";
643 		priv->codec_priv.mclk_id = SGTL5000_SYSCLK;
644 		priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
645 	} else if (of_device_is_compatible(np, "fsl,imx-audio-tlv320aic32x4")) {
646 		codec_dai_name = "tlv320aic32x4-hifi";
647 		priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
648 	} else if (of_device_is_compatible(np, "fsl,imx-audio-tlv320aic31xx")) {
649 		codec_dai_name = "tlv320dac31xx-hifi";
650 		priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
651 		priv->dai_link[1].dpcm_capture = 0;
652 		priv->dai_link[2].dpcm_capture = 0;
653 		priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT;
654 		priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT;
655 		priv->card.dapm_routes = audio_map_tx;
656 		priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx);
657 	} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) {
658 		codec_dai_name = "wm8962";
659 		priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK;
660 		priv->codec_priv.fll_id = WM8962_SYSCLK_FLL;
661 		priv->codec_priv.pll_id = WM8962_FLL;
662 		priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
663 	} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) {
664 		codec_dai_name = "wm8960-hifi";
665 		priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO;
666 		priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO;
667 		priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
668 	} else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) {
669 		codec_dai_name = "ac97-hifi";
670 		priv->dai_fmt = SND_SOC_DAIFMT_AC97;
671 		priv->card.dapm_routes = audio_map_ac97;
672 		priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_ac97);
673 	} else if (of_device_is_compatible(np, "fsl,imx-audio-mqs")) {
674 		codec_dai_name = "fsl-mqs-dai";
675 		priv->dai_fmt = SND_SOC_DAIFMT_LEFT_J |
676 				SND_SOC_DAIFMT_CBC_CFC |
677 				SND_SOC_DAIFMT_NB_NF;
678 		priv->dai_link[1].dpcm_capture = 0;
679 		priv->dai_link[2].dpcm_capture = 0;
680 		priv->card.dapm_routes = audio_map_tx;
681 		priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx);
682 	} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8524")) {
683 		codec_dai_name = "wm8524-hifi";
684 		priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC;
685 		priv->dai_link[1].dpcm_capture = 0;
686 		priv->dai_link[2].dpcm_capture = 0;
687 		priv->cpu_priv.slot_width = 32;
688 		priv->card.dapm_routes = audio_map_tx;
689 		priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx);
690 	} else if (of_device_is_compatible(np, "fsl,imx-audio-si476x")) {
691 		codec_dai_name = "si476x-codec";
692 		priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC;
693 		priv->card.dapm_routes = audio_map_rx;
694 		priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_rx);
695 	} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8958")) {
696 		codec_dai_name = "wm8994-aif1";
697 		priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
698 		priv->codec_priv.mclk_id = WM8994_FLL_SRC_MCLK1;
699 		priv->codec_priv.fll_id = WM8994_SYSCLK_FLL1;
700 		priv->codec_priv.pll_id = WM8994_FLL1;
701 		priv->codec_priv.free_freq = priv->codec_priv.mclk_freq;
702 		priv->card.dapm_routes = NULL;
703 		priv->card.num_dapm_routes = 0;
704 	} else if (of_device_is_compatible(np, "fsl,imx-audio-nau8822")) {
705 		codec_dai_name = "nau8822-hifi";
706 		priv->codec_priv.mclk_id = NAU8822_CLK_MCLK;
707 		priv->codec_priv.fll_id = NAU8822_CLK_PLL;
708 		priv->codec_priv.pll_id = NAU8822_CLK_PLL;
709 		priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
710 		if (codec_dev)
711 			priv->codec_priv.mclk = devm_clk_get(codec_dev, NULL);
712 	} else {
713 		dev_err(&pdev->dev, "unknown Device Tree compatible\n");
714 		ret = -EINVAL;
715 		goto asrc_fail;
716 	}
717 
718 	/*
719 	 * Allow setting mclk-id from the device-tree node. Otherwise, the
720 	 * default value for each card configuration is used.
721 	 */
722 	of_property_read_u32(np, "mclk-id", &priv->codec_priv.mclk_id);
723 
724 	/* Format info from DT is optional. */
725 	snd_soc_daifmt_parse_clock_provider_as_phandle(np, NULL, &bitclkprovider, &frameprovider);
726 	if (bitclkprovider || frameprovider) {
727 		unsigned int daifmt = snd_soc_daifmt_parse_format(np, NULL);
728 
729 		if (codec_np == bitclkprovider)
730 			daifmt |= (codec_np == frameprovider) ?
731 				SND_SOC_DAIFMT_CBP_CFP : SND_SOC_DAIFMT_CBP_CFC;
732 		else
733 			daifmt |= (codec_np == frameprovider) ?
734 				SND_SOC_DAIFMT_CBC_CFP : SND_SOC_DAIFMT_CBC_CFC;
735 
736 		/* Override dai_fmt with value from DT */
737 		priv->dai_fmt = daifmt;
738 	}
739 
740 	/* Change direction according to format */
741 	if (priv->dai_fmt & SND_SOC_DAIFMT_CBP_CFP) {
742 		priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_IN;
743 		priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_IN;
744 	}
745 
746 	of_node_put(bitclkprovider);
747 	of_node_put(frameprovider);
748 
749 	if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) {
750 		dev_dbg(&pdev->dev, "failed to find codec device\n");
751 		ret = -EPROBE_DEFER;
752 		goto asrc_fail;
753 	}
754 
755 	/* Common settings for corresponding Freescale CPU DAI driver */
756 	if (of_node_name_eq(cpu_np, "ssi")) {
757 		/* Only SSI needs to configure AUDMUX */
758 		ret = fsl_asoc_card_audmux_init(np, priv);
759 		if (ret) {
760 			dev_err(&pdev->dev, "failed to init audmux\n");
761 			goto asrc_fail;
762 		}
763 	} else if (of_node_name_eq(cpu_np, "esai")) {
764 		struct clk *esai_clk = clk_get(&cpu_pdev->dev, "extal");
765 
766 		if (!IS_ERR(esai_clk)) {
767 			priv->cpu_priv.sysclk_freq[TX] = clk_get_rate(esai_clk);
768 			priv->cpu_priv.sysclk_freq[RX] = clk_get_rate(esai_clk);
769 			clk_put(esai_clk);
770 		} else if (PTR_ERR(esai_clk) == -EPROBE_DEFER) {
771 			ret = -EPROBE_DEFER;
772 			goto asrc_fail;
773 		}
774 
775 		priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL;
776 		priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL;
777 	} else if (of_node_name_eq(cpu_np, "sai")) {
778 		priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1;
779 		priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1;
780 	}
781 
782 	/* Initialize sound card */
783 	priv->pdev = pdev;
784 	priv->card.dev = &pdev->dev;
785 	priv->card.owner = THIS_MODULE;
786 	ret = snd_soc_of_parse_card_name(&priv->card, "model");
787 	if (ret) {
788 		snprintf(priv->name, sizeof(priv->name), "%s-audio",
789 			 fsl_asoc_card_is_ac97(priv) ? "ac97" : codec_dev_name);
790 		priv->card.name = priv->name;
791 	}
792 	priv->card.dai_link = priv->dai_link;
793 	priv->card.late_probe = fsl_asoc_card_late_probe;
794 	priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;
795 	priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets);
796 
797 	/* Drop the second half of DAPM routes -- ASRC */
798 	if (!asrc_pdev)
799 		priv->card.num_dapm_routes /= 2;
800 
801 	if (of_property_read_bool(np, "audio-routing")) {
802 		ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing");
803 		if (ret) {
804 			dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret);
805 			goto asrc_fail;
806 		}
807 	}
808 
809 	/* Normal DAI Link */
810 	priv->dai_link[0].cpus->of_node = cpu_np;
811 	priv->dai_link[0].codecs->dai_name = codec_dai_name;
812 
813 	if (!fsl_asoc_card_is_ac97(priv))
814 		priv->dai_link[0].codecs->of_node = codec_np;
815 	else {
816 		u32 idx;
817 
818 		ret = of_property_read_u32(cpu_np, "cell-index", &idx);
819 		if (ret) {
820 			dev_err(&pdev->dev,
821 				"cannot get CPU index property\n");
822 			goto asrc_fail;
823 		}
824 
825 		priv->dai_link[0].codecs->name =
826 				devm_kasprintf(&pdev->dev, GFP_KERNEL,
827 					       "ac97-codec.%u",
828 					       (unsigned int)idx);
829 		if (!priv->dai_link[0].codecs->name) {
830 			ret = -ENOMEM;
831 			goto asrc_fail;
832 		}
833 	}
834 
835 	priv->dai_link[0].platforms->of_node = cpu_np;
836 	priv->dai_link[0].dai_fmt = priv->dai_fmt;
837 	priv->card.num_links = 1;
838 
839 	if (asrc_pdev) {
840 		/* DPCM DAI Links only if ASRC exists */
841 		priv->dai_link[1].cpus->of_node = asrc_np;
842 		priv->dai_link[1].platforms->of_node = asrc_np;
843 		priv->dai_link[2].codecs->dai_name = codec_dai_name;
844 		priv->dai_link[2].codecs->of_node = codec_np;
845 		priv->dai_link[2].codecs->name =
846 				priv->dai_link[0].codecs->name;
847 		priv->dai_link[2].cpus->of_node = cpu_np;
848 		priv->dai_link[2].dai_fmt = priv->dai_fmt;
849 		priv->card.num_links = 3;
850 
851 		ret = of_property_read_u32(asrc_np, "fsl,asrc-rate",
852 					   &priv->asrc_rate);
853 		if (ret) {
854 			dev_err(&pdev->dev, "failed to get output rate\n");
855 			ret = -EINVAL;
856 			goto asrc_fail;
857 		}
858 
859 		ret = of_property_read_u32(asrc_np, "fsl,asrc-format", &asrc_fmt);
860 		priv->asrc_format = (__force snd_pcm_format_t)asrc_fmt;
861 		if (ret) {
862 			/* Fallback to old binding; translate to asrc_format */
863 			ret = of_property_read_u32(asrc_np, "fsl,asrc-width",
864 						   &width);
865 			if (ret) {
866 				dev_err(&pdev->dev,
867 					"failed to decide output format\n");
868 				goto asrc_fail;
869 			}
870 
871 			if (width == 24)
872 				priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE;
873 			else
874 				priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE;
875 		}
876 	}
877 
878 	/* Finish card registering */
879 	platform_set_drvdata(pdev, priv);
880 	snd_soc_card_set_drvdata(&priv->card, priv);
881 
882 	ret = devm_snd_soc_register_card(&pdev->dev, &priv->card);
883 	if (ret) {
884 		dev_err_probe(&pdev->dev, ret, "snd_soc_register_card failed\n");
885 		goto asrc_fail;
886 	}
887 
888 	/*
889 	 * Properties "hp-det-gpio" and "mic-det-gpio" are optional, and
890 	 * asoc_simple_init_jack uses these properties for creating
891 	 * Headphone Jack and Microphone Jack.
892 	 *
893 	 * The notifier is initialized in snd_soc_card_jack_new(), then
894 	 * snd_soc_jack_notifier_register can be called.
895 	 */
896 	if (of_property_read_bool(np, "hp-det-gpio")) {
897 		ret = asoc_simple_init_jack(&priv->card, &priv->hp_jack,
898 					    1, NULL, "Headphone Jack");
899 		if (ret)
900 			goto asrc_fail;
901 
902 		snd_soc_jack_notifier_register(&priv->hp_jack.jack, &hp_jack_nb);
903 	}
904 
905 	if (of_property_read_bool(np, "mic-det-gpio")) {
906 		ret = asoc_simple_init_jack(&priv->card, &priv->mic_jack,
907 					    0, NULL, "Mic Jack");
908 		if (ret)
909 			goto asrc_fail;
910 
911 		snd_soc_jack_notifier_register(&priv->mic_jack.jack, &mic_jack_nb);
912 	}
913 
914 asrc_fail:
915 	of_node_put(asrc_np);
916 	of_node_put(codec_np);
917 	put_device(&cpu_pdev->dev);
918 fail:
919 	of_node_put(cpu_np);
920 
921 	return ret;
922 }
923 
924 static const struct of_device_id fsl_asoc_card_dt_ids[] = {
925 	{ .compatible = "fsl,imx-audio-ac97", },
926 	{ .compatible = "fsl,imx-audio-cs42888", },
927 	{ .compatible = "fsl,imx-audio-cs427x", },
928 	{ .compatible = "fsl,imx-audio-tlv320aic32x4", },
929 	{ .compatible = "fsl,imx-audio-tlv320aic31xx", },
930 	{ .compatible = "fsl,imx-audio-sgtl5000", },
931 	{ .compatible = "fsl,imx-audio-wm8962", },
932 	{ .compatible = "fsl,imx-audio-wm8960", },
933 	{ .compatible = "fsl,imx-audio-mqs", },
934 	{ .compatible = "fsl,imx-audio-wm8524", },
935 	{ .compatible = "fsl,imx-audio-si476x", },
936 	{ .compatible = "fsl,imx-audio-wm8958", },
937 	{ .compatible = "fsl,imx-audio-nau8822", },
938 	{}
939 };
940 MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids);
941 
942 static struct platform_driver fsl_asoc_card_driver = {
943 	.probe = fsl_asoc_card_probe,
944 	.driver = {
945 		.name = DRIVER_NAME,
946 		.pm = &snd_soc_pm_ops,
947 		.of_match_table = fsl_asoc_card_dt_ids,
948 	},
949 };
950 module_platform_driver(fsl_asoc_card_driver);
951 
952 MODULE_DESCRIPTION("Freescale Generic ASoC Sound Card driver with ASRC");
953 MODULE_AUTHOR("Nicolin Chen <nicoleotsuka@gmail.com>");
954 MODULE_ALIAS("platform:" DRIVER_NAME);
955 MODULE_LICENSE("GPL");
956