xref: /openbmc/linux/sound/soc/fsl/fsl-asoc-card.c (revision 11a163f2)
1 // SPDX-License-Identifier: GPL-2.0
2 //
3 // Freescale Generic ASoC Sound Card driver with ASRC
4 //
5 // Copyright (C) 2014 Freescale Semiconductor, Inc.
6 //
7 // Author: Nicolin Chen <nicoleotsuka@gmail.com>
8 
9 #include <linux/clk.h>
10 #include <linux/i2c.h>
11 #include <linux/module.h>
12 #include <linux/of_platform.h>
13 #if IS_ENABLED(CONFIG_SND_AC97_CODEC)
14 #include <sound/ac97_codec.h>
15 #endif
16 #include <sound/pcm_params.h>
17 #include <sound/soc.h>
18 #include <sound/jack.h>
19 #include <sound/simple_card_utils.h>
20 
21 #include "fsl_esai.h"
22 #include "fsl_sai.h"
23 #include "imx-audmux.h"
24 
25 #include "../codecs/sgtl5000.h"
26 #include "../codecs/wm8962.h"
27 #include "../codecs/wm8960.h"
28 
29 #define CS427x_SYSCLK_MCLK 0
30 
31 #define RX 0
32 #define TX 1
33 
34 /* Default DAI format without Master and Slave flag */
35 #define DAI_FMT_BASE (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF)
36 
37 /**
38  * struct codec_priv - CODEC private data
39  * @mclk_freq: Clock rate of MCLK
40  * @mclk_id: MCLK (or main clock) id for set_sysclk()
41  * @fll_id: FLL (or secordary clock) id for set_sysclk()
42  * @pll_id: PLL id for set_pll()
43  */
44 struct codec_priv {
45 	unsigned long mclk_freq;
46 	u32 mclk_id;
47 	u32 fll_id;
48 	u32 pll_id;
49 };
50 
51 /**
52  * struct cpu_priv - CPU private data
53  * @sysclk_freq: SYSCLK rates for set_sysclk()
54  * @sysclk_dir: SYSCLK directions for set_sysclk()
55  * @sysclk_id: SYSCLK ids for set_sysclk()
56  * @slot_width: Slot width of each frame
57  *
58  * Note: [1] for tx and [0] for rx
59  */
60 struct cpu_priv {
61 	unsigned long sysclk_freq[2];
62 	u32 sysclk_dir[2];
63 	u32 sysclk_id[2];
64 	u32 slot_width;
65 };
66 
67 /**
68  * struct fsl_asoc_card_priv - Freescale Generic ASOC card private data
69  * @dai_link: DAI link structure including normal one and DPCM link
70  * @hp_jack: Headphone Jack structure
71  * @mic_jack: Microphone Jack structure
72  * @pdev: platform device pointer
73  * @codec_priv: CODEC private data
74  * @cpu_priv: CPU private data
75  * @card: ASoC card structure
76  * @streams: Mask of current active streams
77  * @sample_rate: Current sample rate
78  * @sample_format: Current sample format
79  * @asrc_rate: ASRC sample rate used by Back-Ends
80  * @asrc_format: ASRC sample format used by Back-Ends
81  * @dai_fmt: DAI format between CPU and CODEC
82  * @name: Card name
83  */
84 
85 struct fsl_asoc_card_priv {
86 	struct snd_soc_dai_link dai_link[3];
87 	struct asoc_simple_jack hp_jack;
88 	struct asoc_simple_jack mic_jack;
89 	struct platform_device *pdev;
90 	struct codec_priv codec_priv;
91 	struct cpu_priv cpu_priv;
92 	struct snd_soc_card card;
93 	u8 streams;
94 	u32 sample_rate;
95 	snd_pcm_format_t sample_format;
96 	u32 asrc_rate;
97 	snd_pcm_format_t asrc_format;
98 	u32 dai_fmt;
99 	char name[32];
100 };
101 
102 /*
103  * This dapm route map exists for DPCM link only.
104  * The other routes shall go through Device Tree.
105  *
106  * Note: keep all ASRC routes in the second half
107  *	 to drop them easily for non-ASRC cases.
108  */
109 static const struct snd_soc_dapm_route audio_map[] = {
110 	/* 1st half -- Normal DAPM routes */
111 	{"Playback",  NULL, "CPU-Playback"},
112 	{"CPU-Capture",  NULL, "Capture"},
113 	/* 2nd half -- ASRC DAPM routes */
114 	{"CPU-Playback",  NULL, "ASRC-Playback"},
115 	{"ASRC-Capture",  NULL, "CPU-Capture"},
116 };
117 
118 static const struct snd_soc_dapm_route audio_map_ac97[] = {
119 	/* 1st half -- Normal DAPM routes */
120 	{"Playback",  NULL, "AC97 Playback"},
121 	{"AC97 Capture",  NULL, "Capture"},
122 	/* 2nd half -- ASRC DAPM routes */
123 	{"AC97 Playback",  NULL, "ASRC-Playback"},
124 	{"ASRC-Capture",  NULL, "AC97 Capture"},
125 };
126 
127 static const struct snd_soc_dapm_route audio_map_tx[] = {
128 	/* 1st half -- Normal DAPM routes */
129 	{"Playback",  NULL, "CPU-Playback"},
130 	/* 2nd half -- ASRC DAPM routes */
131 	{"CPU-Playback",  NULL, "ASRC-Playback"},
132 };
133 
134 /* Add all possible widgets into here without being redundant */
135 static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = {
136 	SND_SOC_DAPM_LINE("Line Out Jack", NULL),
137 	SND_SOC_DAPM_LINE("Line In Jack", NULL),
138 	SND_SOC_DAPM_HP("Headphone Jack", NULL),
139 	SND_SOC_DAPM_SPK("Ext Spk", NULL),
140 	SND_SOC_DAPM_MIC("Mic Jack", NULL),
141 	SND_SOC_DAPM_MIC("AMIC", NULL),
142 	SND_SOC_DAPM_MIC("DMIC", NULL),
143 };
144 
145 static bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv)
146 {
147 	return priv->dai_fmt == SND_SOC_DAIFMT_AC97;
148 }
149 
150 static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
151 				   struct snd_pcm_hw_params *params)
152 {
153 	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
154 	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
155 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
156 	struct codec_priv *codec_priv = &priv->codec_priv;
157 	struct cpu_priv *cpu_priv = &priv->cpu_priv;
158 	struct device *dev = rtd->card->dev;
159 	unsigned int pll_out;
160 	int ret;
161 
162 	priv->sample_rate = params_rate(params);
163 	priv->sample_format = params_format(params);
164 	priv->streams |= BIT(substream->stream);
165 
166 	if (fsl_asoc_card_is_ac97(priv))
167 		return 0;
168 
169 	/* Specific configurations of DAIs starts from here */
170 	ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), cpu_priv->sysclk_id[tx],
171 				     cpu_priv->sysclk_freq[tx],
172 				     cpu_priv->sysclk_dir[tx]);
173 	if (ret && ret != -ENOTSUPP) {
174 		dev_err(dev, "failed to set sysclk for cpu dai\n");
175 		goto fail;
176 	}
177 
178 	if (cpu_priv->slot_width) {
179 		ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2,
180 					       cpu_priv->slot_width);
181 		if (ret && ret != -ENOTSUPP) {
182 			dev_err(dev, "failed to set TDM slot for cpu dai\n");
183 			goto fail;
184 		}
185 	}
186 
187 	/* Specific configuration for PLL */
188 	if (codec_priv->pll_id && codec_priv->fll_id) {
189 		if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE)
190 			pll_out = priv->sample_rate * 384;
191 		else
192 			pll_out = priv->sample_rate * 256;
193 
194 		ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0),
195 					  codec_priv->pll_id,
196 					  codec_priv->mclk_id,
197 					  codec_priv->mclk_freq, pll_out);
198 		if (ret) {
199 			dev_err(dev, "failed to start FLL: %d\n", ret);
200 			goto fail;
201 		}
202 
203 		ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0),
204 					     codec_priv->fll_id,
205 					     pll_out, SND_SOC_CLOCK_IN);
206 
207 		if (ret && ret != -ENOTSUPP) {
208 			dev_err(dev, "failed to set SYSCLK: %d\n", ret);
209 			goto fail;
210 		}
211 	}
212 
213 	return 0;
214 
215 fail:
216 	priv->streams &= ~BIT(substream->stream);
217 	return ret;
218 }
219 
220 static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream)
221 {
222 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
223 	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
224 	struct codec_priv *codec_priv = &priv->codec_priv;
225 	struct device *dev = rtd->card->dev;
226 	int ret;
227 
228 	priv->streams &= ~BIT(substream->stream);
229 
230 	if (!priv->streams && codec_priv->pll_id && codec_priv->fll_id) {
231 		/* Force freq to be 0 to avoid error message in codec */
232 		ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0),
233 					     codec_priv->mclk_id,
234 					     0,
235 					     SND_SOC_CLOCK_IN);
236 		if (ret) {
237 			dev_err(dev, "failed to switch away from FLL: %d\n", ret);
238 			return ret;
239 		}
240 
241 		ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0),
242 					  codec_priv->pll_id, 0, 0, 0);
243 		if (ret && ret != -ENOTSUPP) {
244 			dev_err(dev, "failed to stop FLL: %d\n", ret);
245 			return ret;
246 		}
247 	}
248 
249 	return 0;
250 }
251 
252 static const struct snd_soc_ops fsl_asoc_card_ops = {
253 	.hw_params = fsl_asoc_card_hw_params,
254 	.hw_free = fsl_asoc_card_hw_free,
255 };
256 
257 static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
258 			      struct snd_pcm_hw_params *params)
259 {
260 	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
261 	struct snd_interval *rate;
262 	struct snd_mask *mask;
263 
264 	rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
265 	rate->max = rate->min = priv->asrc_rate;
266 
267 	mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
268 	snd_mask_none(mask);
269 	snd_mask_set_format(mask, priv->asrc_format);
270 
271 	return 0;
272 }
273 
274 SND_SOC_DAILINK_DEFS(hifi,
275 	DAILINK_COMP_ARRAY(COMP_EMPTY()),
276 	DAILINK_COMP_ARRAY(COMP_EMPTY()),
277 	DAILINK_COMP_ARRAY(COMP_EMPTY()));
278 
279 SND_SOC_DAILINK_DEFS(hifi_fe,
280 	DAILINK_COMP_ARRAY(COMP_EMPTY()),
281 	DAILINK_COMP_ARRAY(COMP_DUMMY()),
282 	DAILINK_COMP_ARRAY(COMP_EMPTY()));
283 
284 SND_SOC_DAILINK_DEFS(hifi_be,
285 	DAILINK_COMP_ARRAY(COMP_EMPTY()),
286 	DAILINK_COMP_ARRAY(COMP_EMPTY()),
287 	DAILINK_COMP_ARRAY(COMP_DUMMY()));
288 
289 static struct snd_soc_dai_link fsl_asoc_card_dai[] = {
290 	/* Default ASoC DAI Link*/
291 	{
292 		.name = "HiFi",
293 		.stream_name = "HiFi",
294 		.ops = &fsl_asoc_card_ops,
295 		SND_SOC_DAILINK_REG(hifi),
296 	},
297 	/* DPCM Link between Front-End and Back-End (Optional) */
298 	{
299 		.name = "HiFi-ASRC-FE",
300 		.stream_name = "HiFi-ASRC-FE",
301 		.dpcm_playback = 1,
302 		.dpcm_capture = 1,
303 		.dynamic = 1,
304 		SND_SOC_DAILINK_REG(hifi_fe),
305 	},
306 	{
307 		.name = "HiFi-ASRC-BE",
308 		.stream_name = "HiFi-ASRC-BE",
309 		.be_hw_params_fixup = be_hw_params_fixup,
310 		.ops = &fsl_asoc_card_ops,
311 		.dpcm_playback = 1,
312 		.dpcm_capture = 1,
313 		.no_pcm = 1,
314 		SND_SOC_DAILINK_REG(hifi_be),
315 	},
316 };
317 
318 static int fsl_asoc_card_audmux_init(struct device_node *np,
319 				     struct fsl_asoc_card_priv *priv)
320 {
321 	struct device *dev = &priv->pdev->dev;
322 	u32 int_ptcr = 0, ext_ptcr = 0;
323 	int int_port, ext_port;
324 	int ret;
325 
326 	ret = of_property_read_u32(np, "mux-int-port", &int_port);
327 	if (ret) {
328 		dev_err(dev, "mux-int-port missing or invalid\n");
329 		return ret;
330 	}
331 	ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
332 	if (ret) {
333 		dev_err(dev, "mux-ext-port missing or invalid\n");
334 		return ret;
335 	}
336 
337 	/*
338 	 * The port numbering in the hardware manual starts at 1, while
339 	 * the AUDMUX API expects it starts at 0.
340 	 */
341 	int_port--;
342 	ext_port--;
343 
344 	/*
345 	 * Use asynchronous mode (6 wires) for all cases except AC97.
346 	 * If only 4 wires are needed, just set SSI into
347 	 * synchronous mode and enable 4 PADs in IOMUX.
348 	 */
349 	switch (priv->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) {
350 	case SND_SOC_DAIFMT_CBM_CFM:
351 		int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
352 			   IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
353 			   IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
354 			   IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
355 			   IMX_AUDMUX_V2_PTCR_RFSDIR |
356 			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
357 			   IMX_AUDMUX_V2_PTCR_TFSDIR |
358 			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
359 		break;
360 	case SND_SOC_DAIFMT_CBM_CFS:
361 		int_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
362 			   IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
363 			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
364 			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
365 		ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
366 			   IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
367 			   IMX_AUDMUX_V2_PTCR_RFSDIR |
368 			   IMX_AUDMUX_V2_PTCR_TFSDIR;
369 		break;
370 	case SND_SOC_DAIFMT_CBS_CFM:
371 		int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
372 			   IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
373 			   IMX_AUDMUX_V2_PTCR_RFSDIR |
374 			   IMX_AUDMUX_V2_PTCR_TFSDIR;
375 		ext_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
376 			   IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
377 			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
378 			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
379 		break;
380 	case SND_SOC_DAIFMT_CBS_CFS:
381 		ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
382 			   IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
383 			   IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
384 			   IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
385 			   IMX_AUDMUX_V2_PTCR_RFSDIR |
386 			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
387 			   IMX_AUDMUX_V2_PTCR_TFSDIR |
388 			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
389 		break;
390 	default:
391 		if (!fsl_asoc_card_is_ac97(priv))
392 			return -EINVAL;
393 	}
394 
395 	if (fsl_asoc_card_is_ac97(priv)) {
396 		int_ptcr = IMX_AUDMUX_V2_PTCR_SYN |
397 			   IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
398 			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
399 		ext_ptcr = IMX_AUDMUX_V2_PTCR_SYN |
400 			   IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
401 			   IMX_AUDMUX_V2_PTCR_TFSDIR;
402 	}
403 
404 	/* Asynchronous mode can not be set along with RCLKDIR */
405 	if (!fsl_asoc_card_is_ac97(priv)) {
406 		unsigned int pdcr =
407 				IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port);
408 
409 		ret = imx_audmux_v2_configure_port(int_port, 0,
410 						   pdcr);
411 		if (ret) {
412 			dev_err(dev, "audmux internal port setup failed\n");
413 			return ret;
414 		}
415 	}
416 
417 	ret = imx_audmux_v2_configure_port(int_port, int_ptcr,
418 					   IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
419 	if (ret) {
420 		dev_err(dev, "audmux internal port setup failed\n");
421 		return ret;
422 	}
423 
424 	if (!fsl_asoc_card_is_ac97(priv)) {
425 		unsigned int pdcr =
426 				IMX_AUDMUX_V2_PDCR_RXDSEL(int_port);
427 
428 		ret = imx_audmux_v2_configure_port(ext_port, 0,
429 						   pdcr);
430 		if (ret) {
431 			dev_err(dev, "audmux external port setup failed\n");
432 			return ret;
433 		}
434 	}
435 
436 	ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr,
437 					   IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
438 	if (ret) {
439 		dev_err(dev, "audmux external port setup failed\n");
440 		return ret;
441 	}
442 
443 	return 0;
444 }
445 
446 static int hp_jack_event(struct notifier_block *nb, unsigned long event,
447 			 void *data)
448 {
449 	struct snd_soc_jack *jack = (struct snd_soc_jack *)data;
450 	struct snd_soc_dapm_context *dapm = &jack->card->dapm;
451 
452 	if (event & SND_JACK_HEADPHONE)
453 		/* Disable speaker if headphone is plugged in */
454 		snd_soc_dapm_disable_pin(dapm, "Ext Spk");
455 	else
456 		snd_soc_dapm_enable_pin(dapm, "Ext Spk");
457 
458 	return 0;
459 }
460 
461 static struct notifier_block hp_jack_nb = {
462 	.notifier_call = hp_jack_event,
463 };
464 
465 static int mic_jack_event(struct notifier_block *nb, unsigned long event,
466 			  void *data)
467 {
468 	struct snd_soc_jack *jack = (struct snd_soc_jack *)data;
469 	struct snd_soc_dapm_context *dapm = &jack->card->dapm;
470 
471 	if (event & SND_JACK_MICROPHONE)
472 		/* Disable dmic if microphone is plugged in */
473 		snd_soc_dapm_disable_pin(dapm, "DMIC");
474 	else
475 		snd_soc_dapm_enable_pin(dapm, "DMIC");
476 
477 	return 0;
478 }
479 
480 static struct notifier_block mic_jack_nb = {
481 	.notifier_call = mic_jack_event,
482 };
483 
484 static int fsl_asoc_card_late_probe(struct snd_soc_card *card)
485 {
486 	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
487 	struct snd_soc_pcm_runtime *rtd = list_first_entry(
488 			&card->rtd_list, struct snd_soc_pcm_runtime, list);
489 	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
490 	struct codec_priv *codec_priv = &priv->codec_priv;
491 	struct device *dev = card->dev;
492 	int ret;
493 
494 	if (fsl_asoc_card_is_ac97(priv)) {
495 #if IS_ENABLED(CONFIG_SND_AC97_CODEC)
496 		struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
497 		struct snd_ac97 *ac97 = snd_soc_component_get_drvdata(component);
498 
499 		/*
500 		 * Use slots 3/4 for S/PDIF so SSI won't try to enable
501 		 * other slots and send some samples there
502 		 * due to SLOTREQ bits for S/PDIF received from codec
503 		 */
504 		snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS,
505 				     AC97_EA_SPSA_SLOT_MASK, AC97_EA_SPSA_3_4);
506 #endif
507 
508 		return 0;
509 	}
510 
511 	ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
512 				     codec_priv->mclk_freq, SND_SOC_CLOCK_IN);
513 	if (ret && ret != -ENOTSUPP) {
514 		dev_err(dev, "failed to set sysclk in %s\n", __func__);
515 		return ret;
516 	}
517 
518 	return 0;
519 }
520 
521 static int fsl_asoc_card_probe(struct platform_device *pdev)
522 {
523 	struct device_node *cpu_np, *codec_np, *asrc_np;
524 	struct device_node *np = pdev->dev.of_node;
525 	struct platform_device *asrc_pdev = NULL;
526 	struct device_node *bitclkmaster = NULL;
527 	struct device_node *framemaster = NULL;
528 	struct platform_device *cpu_pdev;
529 	struct fsl_asoc_card_priv *priv;
530 	struct device *codec_dev = NULL;
531 	const char *codec_dai_name;
532 	const char *codec_dev_name;
533 	unsigned int daifmt;
534 	u32 width;
535 	int ret;
536 
537 	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
538 	if (!priv)
539 		return -ENOMEM;
540 
541 	cpu_np = of_parse_phandle(np, "audio-cpu", 0);
542 	/* Give a chance to old DT binding */
543 	if (!cpu_np)
544 		cpu_np = of_parse_phandle(np, "ssi-controller", 0);
545 	if (!cpu_np) {
546 		dev_err(&pdev->dev, "CPU phandle missing or invalid\n");
547 		ret = -EINVAL;
548 		goto fail;
549 	}
550 
551 	cpu_pdev = of_find_device_by_node(cpu_np);
552 	if (!cpu_pdev) {
553 		dev_err(&pdev->dev, "failed to find CPU DAI device\n");
554 		ret = -EINVAL;
555 		goto fail;
556 	}
557 
558 	codec_np = of_parse_phandle(np, "audio-codec", 0);
559 	if (codec_np) {
560 		struct platform_device *codec_pdev;
561 		struct i2c_client *codec_i2c;
562 
563 		codec_i2c = of_find_i2c_device_by_node(codec_np);
564 		if (codec_i2c) {
565 			codec_dev = &codec_i2c->dev;
566 			codec_dev_name = codec_i2c->name;
567 		}
568 		if (!codec_dev) {
569 			codec_pdev = of_find_device_by_node(codec_np);
570 			if (codec_pdev) {
571 				codec_dev = &codec_pdev->dev;
572 				codec_dev_name = codec_pdev->name;
573 			}
574 		}
575 	}
576 
577 	asrc_np = of_parse_phandle(np, "audio-asrc", 0);
578 	if (asrc_np)
579 		asrc_pdev = of_find_device_by_node(asrc_np);
580 
581 	/* Get the MCLK rate only, and leave it controlled by CODEC drivers */
582 	if (codec_dev) {
583 		struct clk *codec_clk = clk_get(codec_dev, NULL);
584 
585 		if (!IS_ERR(codec_clk)) {
586 			priv->codec_priv.mclk_freq = clk_get_rate(codec_clk);
587 			clk_put(codec_clk);
588 		}
589 	}
590 
591 	/* Default sample rate and format, will be updated in hw_params() */
592 	priv->sample_rate = 44100;
593 	priv->sample_format = SNDRV_PCM_FORMAT_S16_LE;
594 
595 	/* Assign a default DAI format, and allow each card to overwrite it */
596 	priv->dai_fmt = DAI_FMT_BASE;
597 
598 	memcpy(priv->dai_link, fsl_asoc_card_dai,
599 	       sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
600 
601 	priv->card.dapm_routes = audio_map;
602 	priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
603 	/* Diversify the card configurations */
604 	if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
605 		codec_dai_name = "cs42888";
606 		priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq;
607 		priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq;
608 		priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT;
609 		priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT;
610 		priv->cpu_priv.slot_width = 32;
611 		priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
612 	} else if (of_device_is_compatible(np, "fsl,imx-audio-cs427x")) {
613 		codec_dai_name = "cs4271-hifi";
614 		priv->codec_priv.mclk_id = CS427x_SYSCLK_MCLK;
615 		priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
616 	} else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) {
617 		codec_dai_name = "sgtl5000";
618 		priv->codec_priv.mclk_id = SGTL5000_SYSCLK;
619 		priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
620 	} else if (of_device_is_compatible(np, "fsl,imx-audio-tlv320aic32x4")) {
621 		codec_dai_name = "tlv320aic32x4-hifi";
622 		priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
623 	} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) {
624 		codec_dai_name = "wm8962";
625 		priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK;
626 		priv->codec_priv.fll_id = WM8962_SYSCLK_FLL;
627 		priv->codec_priv.pll_id = WM8962_FLL;
628 		priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
629 	} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) {
630 		codec_dai_name = "wm8960-hifi";
631 		priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO;
632 		priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO;
633 		priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
634 	} else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) {
635 		codec_dai_name = "ac97-hifi";
636 		priv->dai_fmt = SND_SOC_DAIFMT_AC97;
637 		priv->card.dapm_routes = audio_map_ac97;
638 		priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_ac97);
639 	} else if (of_device_is_compatible(np, "fsl,imx-audio-mqs")) {
640 		codec_dai_name = "fsl-mqs-dai";
641 		priv->dai_fmt = SND_SOC_DAIFMT_LEFT_J |
642 				SND_SOC_DAIFMT_CBS_CFS |
643 				SND_SOC_DAIFMT_NB_NF;
644 		priv->dai_link[1].dpcm_capture = 0;
645 		priv->dai_link[2].dpcm_capture = 0;
646 		priv->card.dapm_routes = audio_map_tx;
647 		priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx);
648 	} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8524")) {
649 		codec_dai_name = "wm8524-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.slot_width = 32;
654 		priv->card.dapm_routes = audio_map_tx;
655 		priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx);
656 	} else {
657 		dev_err(&pdev->dev, "unknown Device Tree compatible\n");
658 		ret = -EINVAL;
659 		goto asrc_fail;
660 	}
661 
662 	/* Format info from DT is optional. */
663 	daifmt = snd_soc_of_parse_daifmt(np, NULL,
664 					 &bitclkmaster, &framemaster);
665 	daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK;
666 	if (bitclkmaster || framemaster) {
667 		if (codec_np == bitclkmaster)
668 			daifmt |= (codec_np == framemaster) ?
669 				SND_SOC_DAIFMT_CBM_CFM : SND_SOC_DAIFMT_CBM_CFS;
670 		else
671 			daifmt |= (codec_np == framemaster) ?
672 				SND_SOC_DAIFMT_CBS_CFM : SND_SOC_DAIFMT_CBS_CFS;
673 
674 		/* Override dai_fmt with value from DT */
675 		priv->dai_fmt = daifmt;
676 	}
677 
678 	/* Change direction according to format */
679 	if (priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) {
680 		priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_IN;
681 		priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_IN;
682 	}
683 
684 	of_node_put(bitclkmaster);
685 	of_node_put(framemaster);
686 
687 	if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) {
688 		dev_err(&pdev->dev, "failed to find codec device\n");
689 		ret = -EPROBE_DEFER;
690 		goto asrc_fail;
691 	}
692 
693 	/* Common settings for corresponding Freescale CPU DAI driver */
694 	if (of_node_name_eq(cpu_np, "ssi")) {
695 		/* Only SSI needs to configure AUDMUX */
696 		ret = fsl_asoc_card_audmux_init(np, priv);
697 		if (ret) {
698 			dev_err(&pdev->dev, "failed to init audmux\n");
699 			goto asrc_fail;
700 		}
701 	} else if (of_node_name_eq(cpu_np, "esai")) {
702 		struct clk *esai_clk = clk_get(&cpu_pdev->dev, "extal");
703 
704 		if (!IS_ERR(esai_clk)) {
705 			priv->cpu_priv.sysclk_freq[TX] = clk_get_rate(esai_clk);
706 			priv->cpu_priv.sysclk_freq[RX] = clk_get_rate(esai_clk);
707 			clk_put(esai_clk);
708 		} else if (PTR_ERR(esai_clk) == -EPROBE_DEFER) {
709 			ret = -EPROBE_DEFER;
710 			goto asrc_fail;
711 		}
712 
713 		priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL;
714 		priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL;
715 	} else if (of_node_name_eq(cpu_np, "sai")) {
716 		priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1;
717 		priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1;
718 	}
719 
720 	/* Initialize sound card */
721 	priv->pdev = pdev;
722 	priv->card.dev = &pdev->dev;
723 	ret = snd_soc_of_parse_card_name(&priv->card, "model");
724 	if (ret) {
725 		snprintf(priv->name, sizeof(priv->name), "%s-audio",
726 			 fsl_asoc_card_is_ac97(priv) ? "ac97" : codec_dev_name);
727 		priv->card.name = priv->name;
728 	}
729 	priv->card.dai_link = priv->dai_link;
730 	priv->card.late_probe = fsl_asoc_card_late_probe;
731 	priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;
732 	priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets);
733 
734 	/* Drop the second half of DAPM routes -- ASRC */
735 	if (!asrc_pdev)
736 		priv->card.num_dapm_routes /= 2;
737 
738 	if (of_property_read_bool(np, "audio-routing")) {
739 		ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing");
740 		if (ret) {
741 			dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret);
742 			goto asrc_fail;
743 		}
744 	}
745 
746 	/* Normal DAI Link */
747 	priv->dai_link[0].cpus->of_node = cpu_np;
748 	priv->dai_link[0].codecs->dai_name = codec_dai_name;
749 
750 	if (!fsl_asoc_card_is_ac97(priv))
751 		priv->dai_link[0].codecs->of_node = codec_np;
752 	else {
753 		u32 idx;
754 
755 		ret = of_property_read_u32(cpu_np, "cell-index", &idx);
756 		if (ret) {
757 			dev_err(&pdev->dev,
758 				"cannot get CPU index property\n");
759 			goto asrc_fail;
760 		}
761 
762 		priv->dai_link[0].codecs->name =
763 				devm_kasprintf(&pdev->dev, GFP_KERNEL,
764 					       "ac97-codec.%u",
765 					       (unsigned int)idx);
766 		if (!priv->dai_link[0].codecs->name) {
767 			ret = -ENOMEM;
768 			goto asrc_fail;
769 		}
770 	}
771 
772 	priv->dai_link[0].platforms->of_node = cpu_np;
773 	priv->dai_link[0].dai_fmt = priv->dai_fmt;
774 	priv->card.num_links = 1;
775 
776 	if (asrc_pdev) {
777 		/* DPCM DAI Links only if ASRC exsits */
778 		priv->dai_link[1].cpus->of_node = asrc_np;
779 		priv->dai_link[1].platforms->of_node = asrc_np;
780 		priv->dai_link[2].codecs->dai_name = codec_dai_name;
781 		priv->dai_link[2].codecs->of_node = codec_np;
782 		priv->dai_link[2].codecs->name =
783 				priv->dai_link[0].codecs->name;
784 		priv->dai_link[2].cpus->of_node = cpu_np;
785 		priv->dai_link[2].dai_fmt = priv->dai_fmt;
786 		priv->card.num_links = 3;
787 
788 		ret = of_property_read_u32(asrc_np, "fsl,asrc-rate",
789 					   &priv->asrc_rate);
790 		if (ret) {
791 			dev_err(&pdev->dev, "failed to get output rate\n");
792 			ret = -EINVAL;
793 			goto asrc_fail;
794 		}
795 
796 		ret = of_property_read_u32(asrc_np, "fsl,asrc-format",
797 					   &priv->asrc_format);
798 		if (ret) {
799 			/* Fallback to old binding; translate to asrc_format */
800 			ret = of_property_read_u32(asrc_np, "fsl,asrc-width",
801 						   &width);
802 			if (ret) {
803 				dev_err(&pdev->dev,
804 					"failed to decide output format\n");
805 				goto asrc_fail;
806 			}
807 
808 			if (width == 24)
809 				priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE;
810 			else
811 				priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE;
812 		}
813 	}
814 
815 	/* Finish card registering */
816 	platform_set_drvdata(pdev, priv);
817 	snd_soc_card_set_drvdata(&priv->card, priv);
818 
819 	ret = devm_snd_soc_register_card(&pdev->dev, &priv->card);
820 	if (ret) {
821 		if (ret != -EPROBE_DEFER)
822 			dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
823 		goto asrc_fail;
824 	}
825 
826 	/*
827 	 * Properties "hp-det-gpio" and "mic-det-gpio" are optional, and
828 	 * asoc_simple_init_jack uses these properties for creating
829 	 * Headphone Jack and Microphone Jack.
830 	 *
831 	 * The notifier is initialized in snd_soc_card_jack_new(), then
832 	 * snd_soc_jack_notifier_register can be called.
833 	 */
834 	if (of_property_read_bool(np, "hp-det-gpio")) {
835 		ret = asoc_simple_init_jack(&priv->card, &priv->hp_jack,
836 					    1, NULL, "Headphone Jack");
837 		if (ret)
838 			goto asrc_fail;
839 
840 		snd_soc_jack_notifier_register(&priv->hp_jack.jack, &hp_jack_nb);
841 	}
842 
843 	if (of_property_read_bool(np, "mic-det-gpio")) {
844 		ret = asoc_simple_init_jack(&priv->card, &priv->mic_jack,
845 					    0, NULL, "Mic Jack");
846 		if (ret)
847 			goto asrc_fail;
848 
849 		snd_soc_jack_notifier_register(&priv->mic_jack.jack, &mic_jack_nb);
850 	}
851 
852 asrc_fail:
853 	of_node_put(asrc_np);
854 	of_node_put(codec_np);
855 	put_device(&cpu_pdev->dev);
856 fail:
857 	of_node_put(cpu_np);
858 
859 	return ret;
860 }
861 
862 static const struct of_device_id fsl_asoc_card_dt_ids[] = {
863 	{ .compatible = "fsl,imx-audio-ac97", },
864 	{ .compatible = "fsl,imx-audio-cs42888", },
865 	{ .compatible = "fsl,imx-audio-cs427x", },
866 	{ .compatible = "fsl,imx-audio-tlv320aic32x4", },
867 	{ .compatible = "fsl,imx-audio-sgtl5000", },
868 	{ .compatible = "fsl,imx-audio-wm8962", },
869 	{ .compatible = "fsl,imx-audio-wm8960", },
870 	{ .compatible = "fsl,imx-audio-mqs", },
871 	{ .compatible = "fsl,imx-audio-wm8524", },
872 	{}
873 };
874 MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids);
875 
876 static struct platform_driver fsl_asoc_card_driver = {
877 	.probe = fsl_asoc_card_probe,
878 	.driver = {
879 		.name = "fsl-asoc-card",
880 		.pm = &snd_soc_pm_ops,
881 		.of_match_table = fsl_asoc_card_dt_ids,
882 	},
883 };
884 module_platform_driver(fsl_asoc_card_driver);
885 
886 MODULE_DESCRIPTION("Freescale Generic ASoC Sound Card driver with ASRC");
887 MODULE_AUTHOR("Nicolin Chen <nicoleotsuka@gmail.com>");
888 MODULE_ALIAS("platform:fsl-asoc-card");
889 MODULE_LICENSE("GPL");
890