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