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 #include "../codecs/wm8904.h"
32 
33 #define DRIVER_NAME "fsl-asoc-card"
34 
35 #define CS427x_SYSCLK_MCLK 0
36 
37 #define RX 0
38 #define TX 1
39 
40 /* Default DAI format without Master and Slave flag */
41 #define DAI_FMT_BASE (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF)
42 
43 /**
44  * struct codec_priv - CODEC private data
45  * @mclk: Main clock of the CODEC
46  * @mclk_freq: Clock rate of MCLK
47  * @free_freq: Clock rate of MCLK for hw_free()
48  * @mclk_id: MCLK (or main clock) id for set_sysclk()
49  * @fll_id: FLL (or secordary clock) id for set_sysclk()
50  * @pll_id: PLL id for set_pll()
51  */
52 struct codec_priv {
53 	struct clk *mclk;
54 	unsigned long mclk_freq;
55 	unsigned long free_freq;
56 	u32 mclk_id;
57 	int fll_id;
58 	int pll_id;
59 };
60 
61 /**
62  * struct cpu_priv - CPU private data
63  * @sysclk_freq: SYSCLK rates for set_sysclk()
64  * @sysclk_dir: SYSCLK directions for set_sysclk()
65  * @sysclk_id: SYSCLK ids for set_sysclk()
66  * @slot_width: Slot width of each frame
67  * @slot_num: Number of slots of each frame
68  *
69  * Note: [1] for tx and [0] for rx
70  */
71 struct cpu_priv {
72 	unsigned long sysclk_freq[2];
73 	u32 sysclk_dir[2];
74 	u32 sysclk_id[2];
75 	u32 slot_width;
76 	u32 slot_num;
77 };
78 
79 /**
80  * struct fsl_asoc_card_priv - Freescale Generic ASOC card private data
81  * @dai_link: DAI link structure including normal one and DPCM link
82  * @hp_jack: Headphone Jack structure
83  * @mic_jack: Microphone Jack structure
84  * @pdev: platform device pointer
85  * @codec_priv: CODEC private data
86  * @cpu_priv: CPU private data
87  * @card: ASoC card structure
88  * @streams: Mask of current active streams
89  * @sample_rate: Current sample rate
90  * @sample_format: Current sample format
91  * @asrc_rate: ASRC sample rate used by Back-Ends
92  * @asrc_format: ASRC sample format used by Back-Ends
93  * @dai_fmt: DAI format between CPU and CODEC
94  * @name: Card name
95  */
96 
97 struct fsl_asoc_card_priv {
98 	struct snd_soc_dai_link dai_link[3];
99 	struct simple_util_jack hp_jack;
100 	struct simple_util_jack mic_jack;
101 	struct platform_device *pdev;
102 	struct codec_priv codec_priv[2];
103 	struct cpu_priv cpu_priv;
104 	struct snd_soc_card card;
105 	u8 streams;
106 	u32 sample_rate;
107 	snd_pcm_format_t sample_format;
108 	u32 asrc_rate;
109 	snd_pcm_format_t asrc_format;
110 	u32 dai_fmt;
111 	char name[32];
112 };
113 
114 /*
115  * This dapm route map exists for DPCM link only.
116  * The other routes shall go through Device Tree.
117  *
118  * Note: keep all ASRC routes in the second half
119  *	 to drop them easily for non-ASRC cases.
120  */
121 static const struct snd_soc_dapm_route audio_map[] = {
122 	/* 1st half -- Normal DAPM routes */
123 	{"Playback",  NULL, "CPU-Playback"},
124 	{"CPU-Capture",  NULL, "Capture"},
125 	/* 2nd half -- ASRC DAPM routes */
126 	{"CPU-Playback",  NULL, "ASRC-Playback"},
127 	{"ASRC-Capture",  NULL, "CPU-Capture"},
128 };
129 
130 static const struct snd_soc_dapm_route audio_map_ac97[] = {
131 	/* 1st half -- Normal DAPM routes */
132 	{"AC97 Playback",  NULL, "CPU AC97 Playback"},
133 	{"CPU AC97 Capture",  NULL, "AC97 Capture"},
134 	/* 2nd half -- ASRC DAPM routes */
135 	{"CPU AC97 Playback",  NULL, "ASRC-Playback"},
136 	{"ASRC-Capture",  NULL, "CPU AC97 Capture"},
137 };
138 
139 static const struct snd_soc_dapm_route audio_map_tx[] = {
140 	/* 1st half -- Normal DAPM routes */
141 	{"Playback",  NULL, "CPU-Playback"},
142 	/* 2nd half -- ASRC DAPM routes */
143 	{"CPU-Playback",  NULL, "ASRC-Playback"},
144 };
145 
146 static const struct snd_soc_dapm_route audio_map_rx[] = {
147 	/* 1st half -- Normal DAPM routes */
148 	{"CPU-Capture",  NULL, "Capture"},
149 	/* 2nd half -- ASRC DAPM routes */
150 	{"ASRC-Capture",  NULL, "CPU-Capture"},
151 };
152 
153 /* Add all possible widgets into here without being redundant */
154 static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = {
155 	SND_SOC_DAPM_LINE("Line Out Jack", NULL),
156 	SND_SOC_DAPM_LINE("Line In Jack", NULL),
157 	SND_SOC_DAPM_HP("Headphone Jack", NULL),
158 	SND_SOC_DAPM_SPK("Ext Spk", NULL),
159 	SND_SOC_DAPM_MIC("Mic Jack", NULL),
160 	SND_SOC_DAPM_MIC("AMIC", NULL),
161 	SND_SOC_DAPM_MIC("DMIC", NULL),
162 };
163 
fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv * priv)164 static bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv)
165 {
166 	return priv->dai_fmt == SND_SOC_DAIFMT_AC97;
167 }
168 
fsl_asoc_card_hw_params(struct snd_pcm_substream * substream,struct snd_pcm_hw_params * params)169 static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
170 				   struct snd_pcm_hw_params *params)
171 {
172 	struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
173 	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
174 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
175 	struct codec_priv *codec_priv;
176 	struct snd_soc_dai *codec_dai;
177 	struct cpu_priv *cpu_priv = &priv->cpu_priv;
178 	struct device *dev = rtd->card->dev;
179 	unsigned int pll_out;
180 	int codec_idx;
181 	int ret;
182 
183 	priv->sample_rate = params_rate(params);
184 	priv->sample_format = params_format(params);
185 	priv->streams |= BIT(substream->stream);
186 
187 	if (fsl_asoc_card_is_ac97(priv))
188 		return 0;
189 
190 	/* Specific configurations of DAIs starts from here */
191 	ret = snd_soc_dai_set_sysclk(snd_soc_rtd_to_cpu(rtd, 0), cpu_priv->sysclk_id[tx],
192 				     cpu_priv->sysclk_freq[tx],
193 				     cpu_priv->sysclk_dir[tx]);
194 	if (ret && ret != -ENOTSUPP) {
195 		dev_err(dev, "failed to set sysclk for cpu dai\n");
196 		goto fail;
197 	}
198 
199 	if (cpu_priv->slot_width) {
200 		if (!cpu_priv->slot_num)
201 			cpu_priv->slot_num = 2;
202 
203 		ret = snd_soc_dai_set_tdm_slot(snd_soc_rtd_to_cpu(rtd, 0), 0x3, 0x3,
204 					       cpu_priv->slot_num,
205 					       cpu_priv->slot_width);
206 		if (ret && ret != -ENOTSUPP) {
207 			dev_err(dev, "failed to set TDM slot for cpu dai\n");
208 			goto fail;
209 		}
210 	}
211 
212 	/* Specific configuration for PLL */
213 	for_each_rtd_codec_dais(rtd, codec_idx, codec_dai) {
214 		codec_priv = &priv->codec_priv[codec_idx];
215 
216 		if (codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) {
217 			if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE)
218 				pll_out = priv->sample_rate * 384;
219 			else
220 				pll_out = priv->sample_rate * 256;
221 
222 			ret = snd_soc_dai_set_pll(codec_dai,
223 						codec_priv->pll_id,
224 						codec_priv->mclk_id,
225 						codec_priv->mclk_freq, pll_out);
226 			if (ret) {
227 				dev_err(dev, "failed to start FLL: %d\n", ret);
228 				goto fail;
229 			}
230 
231 			ret = snd_soc_dai_set_sysclk(codec_dai,
232 						codec_priv->fll_id,
233 						pll_out, SND_SOC_CLOCK_IN);
234 
235 			if (ret && ret != -ENOTSUPP) {
236 				dev_err(dev, "failed to set SYSCLK: %d\n", ret);
237 				goto fail;
238 			}
239 		}
240 	}
241 
242 	return 0;
243 
244 fail:
245 	priv->streams &= ~BIT(substream->stream);
246 	return ret;
247 }
248 
fsl_asoc_card_hw_free(struct snd_pcm_substream * substream)249 static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream)
250 {
251 	struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
252 	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
253 	struct codec_priv *codec_priv;
254 	struct snd_soc_dai *codec_dai;
255 	struct device *dev = rtd->card->dev;
256 	int codec_idx;
257 	int ret;
258 
259 	priv->streams &= ~BIT(substream->stream);
260 
261 	for_each_rtd_codec_dais(rtd, codec_idx, codec_dai) {
262 		codec_priv = &priv->codec_priv[codec_idx];
263 
264 		if (!priv->streams && codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) {
265 			/* Force freq to be free_freq to avoid error message in codec */
266 			ret = snd_soc_dai_set_sysclk(codec_dai,
267 						codec_priv->mclk_id,
268 						codec_priv->free_freq,
269 						SND_SOC_CLOCK_IN);
270 			if (ret) {
271 				dev_err(dev, "failed to switch away from FLL: %d\n", ret);
272 				return ret;
273 			}
274 
275 			ret = snd_soc_dai_set_pll(codec_dai,
276 						codec_priv->pll_id, 0, 0, 0);
277 			if (ret && ret != -ENOTSUPP) {
278 				dev_err(dev, "failed to stop FLL: %d\n", ret);
279 				return ret;
280 			}
281 		}
282 	}
283 
284 	return 0;
285 }
286 
287 static const struct snd_soc_ops fsl_asoc_card_ops = {
288 	.hw_params = fsl_asoc_card_hw_params,
289 	.hw_free = fsl_asoc_card_hw_free,
290 };
291 
be_hw_params_fixup(struct snd_soc_pcm_runtime * rtd,struct snd_pcm_hw_params * params)292 static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
293 			      struct snd_pcm_hw_params *params)
294 {
295 	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
296 	struct snd_interval *rate;
297 	struct snd_mask *mask;
298 
299 	rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
300 	rate->max = rate->min = priv->asrc_rate;
301 
302 	mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
303 	snd_mask_none(mask);
304 	snd_mask_set_format(mask, priv->asrc_format);
305 
306 	return 0;
307 }
308 
309 static const struct snd_soc_dai_link fsl_asoc_card_dai[] = {
310 	/* Default ASoC DAI Link*/
311 	{
312 		.name = "HiFi",
313 		.stream_name = "HiFi",
314 		.ops = &fsl_asoc_card_ops,
315 	},
316 	/* DPCM Link between Front-End and Back-End (Optional) */
317 	{
318 		.name = "HiFi-ASRC-FE",
319 		.stream_name = "HiFi-ASRC-FE",
320 		.dpcm_playback = 1,
321 		.dpcm_capture = 1,
322 		.dynamic = 1,
323 	},
324 	{
325 		.name = "HiFi-ASRC-BE",
326 		.stream_name = "HiFi-ASRC-BE",
327 		.be_hw_params_fixup = be_hw_params_fixup,
328 		.ops = &fsl_asoc_card_ops,
329 		.dpcm_playback = 1,
330 		.dpcm_capture = 1,
331 		.no_pcm = 1,
332 	},
333 };
334 
fsl_asoc_card_audmux_init(struct device_node * np,struct fsl_asoc_card_priv * priv)335 static int fsl_asoc_card_audmux_init(struct device_node *np,
336 				     struct fsl_asoc_card_priv *priv)
337 {
338 	struct device *dev = &priv->pdev->dev;
339 	u32 int_ptcr = 0, ext_ptcr = 0;
340 	int int_port, ext_port;
341 	int ret;
342 
343 	ret = of_property_read_u32(np, "mux-int-port", &int_port);
344 	if (ret) {
345 		dev_err(dev, "mux-int-port missing or invalid\n");
346 		return ret;
347 	}
348 	ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
349 	if (ret) {
350 		dev_err(dev, "mux-ext-port missing or invalid\n");
351 		return ret;
352 	}
353 
354 	/*
355 	 * The port numbering in the hardware manual starts at 1, while
356 	 * the AUDMUX API expects it starts at 0.
357 	 */
358 	int_port--;
359 	ext_port--;
360 
361 	/*
362 	 * Use asynchronous mode (6 wires) for all cases except AC97.
363 	 * If only 4 wires are needed, just set SSI into
364 	 * synchronous mode and enable 4 PADs in IOMUX.
365 	 */
366 	switch (priv->dai_fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) {
367 	case SND_SOC_DAIFMT_CBP_CFP:
368 		int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
369 			   IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
370 			   IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
371 			   IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
372 			   IMX_AUDMUX_V2_PTCR_RFSDIR |
373 			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
374 			   IMX_AUDMUX_V2_PTCR_TFSDIR |
375 			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
376 		break;
377 	case SND_SOC_DAIFMT_CBP_CFC:
378 		int_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
379 			   IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
380 			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
381 			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
382 		ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
383 			   IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
384 			   IMX_AUDMUX_V2_PTCR_RFSDIR |
385 			   IMX_AUDMUX_V2_PTCR_TFSDIR;
386 		break;
387 	case SND_SOC_DAIFMT_CBC_CFP:
388 		int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
389 			   IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
390 			   IMX_AUDMUX_V2_PTCR_RFSDIR |
391 			   IMX_AUDMUX_V2_PTCR_TFSDIR;
392 		ext_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
393 			   IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
394 			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
395 			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
396 		break;
397 	case SND_SOC_DAIFMT_CBC_CFC:
398 		ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
399 			   IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
400 			   IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
401 			   IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
402 			   IMX_AUDMUX_V2_PTCR_RFSDIR |
403 			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
404 			   IMX_AUDMUX_V2_PTCR_TFSDIR |
405 			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
406 		break;
407 	default:
408 		if (!fsl_asoc_card_is_ac97(priv))
409 			return -EINVAL;
410 	}
411 
412 	if (fsl_asoc_card_is_ac97(priv)) {
413 		int_ptcr = IMX_AUDMUX_V2_PTCR_SYN |
414 			   IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
415 			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
416 		ext_ptcr = IMX_AUDMUX_V2_PTCR_SYN |
417 			   IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
418 			   IMX_AUDMUX_V2_PTCR_TFSDIR;
419 	}
420 
421 	/* Asynchronous mode can not be set along with RCLKDIR */
422 	if (!fsl_asoc_card_is_ac97(priv)) {
423 		unsigned int pdcr =
424 				IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port);
425 
426 		ret = imx_audmux_v2_configure_port(int_port, 0,
427 						   pdcr);
428 		if (ret) {
429 			dev_err(dev, "audmux internal port setup failed\n");
430 			return ret;
431 		}
432 	}
433 
434 	ret = imx_audmux_v2_configure_port(int_port, int_ptcr,
435 					   IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
436 	if (ret) {
437 		dev_err(dev, "audmux internal port setup failed\n");
438 		return ret;
439 	}
440 
441 	if (!fsl_asoc_card_is_ac97(priv)) {
442 		unsigned int pdcr =
443 				IMX_AUDMUX_V2_PDCR_RXDSEL(int_port);
444 
445 		ret = imx_audmux_v2_configure_port(ext_port, 0,
446 						   pdcr);
447 		if (ret) {
448 			dev_err(dev, "audmux external port setup failed\n");
449 			return ret;
450 		}
451 	}
452 
453 	ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr,
454 					   IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
455 	if (ret) {
456 		dev_err(dev, "audmux external port setup failed\n");
457 		return ret;
458 	}
459 
460 	return 0;
461 }
462 
fsl_asoc_card_spdif_init(struct device_node * codec_np[],struct device_node * cpu_np,const char * codec_dai_name[],struct fsl_asoc_card_priv * priv)463 static int fsl_asoc_card_spdif_init(struct device_node *codec_np[],
464 				    struct device_node *cpu_np,
465 				    const char *codec_dai_name[],
466 				    struct fsl_asoc_card_priv *priv)
467 {
468 	struct device *dev = &priv->pdev->dev;
469 	struct device_node *np = dev->of_node;
470 
471 	if (!of_node_name_eq(cpu_np, "spdif")) {
472 		dev_err(dev, "CPU phandle invalid, should be an SPDIF device\n");
473 		return -EINVAL;
474 	}
475 
476 	priv->dai_link[0].playback_only = true;
477 	priv->dai_link[0].capture_only = true;
478 
479 	for (int i = 0; i < 2; i++) {
480 		if (!codec_np[i])
481 			break;
482 
483 		if (of_device_is_compatible(codec_np[i], "linux,spdif-dit")) {
484 			priv->dai_link[0].capture_only = false;
485 			codec_dai_name[i] = "dit-hifi";
486 		} else if (of_device_is_compatible(codec_np[i], "linux,spdif-dir")) {
487 			priv->dai_link[0].playback_only = false;
488 			codec_dai_name[i] = "dir-hifi";
489 		}
490 	}
491 
492 	// Old SPDIF DT binding
493 	if (!codec_np[0]) {
494 		codec_dai_name[0] = snd_soc_dummy_dlc.dai_name;
495 		if (of_property_read_bool(np, "spdif-out"))
496 			priv->dai_link[0].capture_only = false;
497 		if (of_property_read_bool(np, "spdif-in"))
498 			priv->dai_link[0].playback_only = false;
499 	}
500 
501 	if (priv->dai_link[0].playback_only && priv->dai_link[0].capture_only) {
502 		dev_err(dev, "no enabled S/PDIF DAI link\n");
503 		return -EINVAL;
504 	}
505 
506 	if (priv->dai_link[0].playback_only) {
507 		priv->dai_link[1].dpcm_capture = false;
508 		priv->dai_link[2].dpcm_capture = false;
509 		priv->card.dapm_routes = audio_map_tx;
510 		priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx);
511 	} else if (priv->dai_link[0].capture_only) {
512 		priv->dai_link[1].dpcm_playback = false;
513 		priv->dai_link[2].dpcm_playback = false;
514 		priv->card.dapm_routes = audio_map_rx;
515 		priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_rx);
516 	}
517 
518 	// No DAPM routes with old bindings and dummy codec
519 	if (!codec_np[0]) {
520 		priv->card.dapm_routes = NULL;
521 		priv->card.num_dapm_routes = 0;
522 	}
523 
524 	if (codec_np[0] && codec_np[1]) {
525 		priv->dai_link[0].num_codecs = 2;
526 		priv->dai_link[2].num_codecs = 2;
527 	}
528 
529 	return 0;
530 }
531 
hp_jack_event(struct notifier_block * nb,unsigned long event,void * data)532 static int hp_jack_event(struct notifier_block *nb, unsigned long event,
533 			 void *data)
534 {
535 	struct snd_soc_jack *jack = (struct snd_soc_jack *)data;
536 	struct snd_soc_dapm_context *dapm = &jack->card->dapm;
537 
538 	if (event & SND_JACK_HEADPHONE)
539 		/* Disable speaker if headphone is plugged in */
540 		return snd_soc_dapm_disable_pin(dapm, "Ext Spk");
541 	else
542 		return snd_soc_dapm_enable_pin(dapm, "Ext Spk");
543 }
544 
545 static struct notifier_block hp_jack_nb = {
546 	.notifier_call = hp_jack_event,
547 };
548 
mic_jack_event(struct notifier_block * nb,unsigned long event,void * data)549 static int mic_jack_event(struct notifier_block *nb, unsigned long event,
550 			  void *data)
551 {
552 	struct snd_soc_jack *jack = (struct snd_soc_jack *)data;
553 	struct snd_soc_dapm_context *dapm = &jack->card->dapm;
554 
555 	if (event & SND_JACK_MICROPHONE)
556 		/* Disable dmic if microphone is plugged in */
557 		return snd_soc_dapm_disable_pin(dapm, "DMIC");
558 	else
559 		return snd_soc_dapm_enable_pin(dapm, "DMIC");
560 }
561 
562 static struct notifier_block mic_jack_nb = {
563 	.notifier_call = mic_jack_event,
564 };
565 
fsl_asoc_card_late_probe(struct snd_soc_card * card)566 static int fsl_asoc_card_late_probe(struct snd_soc_card *card)
567 {
568 	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
569 	struct snd_soc_pcm_runtime *rtd = list_first_entry(
570 			&card->rtd_list, struct snd_soc_pcm_runtime, list);
571 	struct snd_soc_dai *codec_dai;
572 	struct codec_priv *codec_priv;
573 	struct device *dev = card->dev;
574 	int codec_idx;
575 	int ret;
576 
577 	if (fsl_asoc_card_is_ac97(priv)) {
578 #if IS_ENABLED(CONFIG_SND_AC97_CODEC)
579 		struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component;
580 		struct snd_ac97 *ac97 = snd_soc_component_get_drvdata(component);
581 
582 		/*
583 		 * Use slots 3/4 for S/PDIF so SSI won't try to enable
584 		 * other slots and send some samples there
585 		 * due to SLOTREQ bits for S/PDIF received from codec
586 		 */
587 		snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS,
588 				     AC97_EA_SPSA_SLOT_MASK, AC97_EA_SPSA_3_4);
589 #endif
590 
591 		return 0;
592 	}
593 
594 	for_each_rtd_codec_dais(rtd, codec_idx, codec_dai) {
595 		codec_priv = &priv->codec_priv[codec_idx];
596 
597 		ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
598 					codec_priv->mclk_freq, SND_SOC_CLOCK_IN);
599 		if (ret && ret != -ENOTSUPP) {
600 			dev_err(dev, "failed to set sysclk in %s\n", __func__);
601 			return ret;
602 		}
603 
604 		if (!IS_ERR_OR_NULL(codec_priv->mclk))
605 			clk_prepare_enable(codec_priv->mclk);
606 	}
607 
608 	return 0;
609 }
610 
fsl_asoc_card_probe(struct platform_device * pdev)611 static int fsl_asoc_card_probe(struct platform_device *pdev)
612 {
613 	struct device_node *cpu_np, *asrc_np;
614 	struct snd_soc_dai_link_component *codec_comp;
615 	struct device_node *codec_np[2];
616 	struct device_node *np = pdev->dev.of_node;
617 	struct platform_device *asrc_pdev = NULL;
618 	struct device_node *bitclkprovider = NULL;
619 	struct device_node *frameprovider = NULL;
620 	struct platform_device *cpu_pdev;
621 	struct fsl_asoc_card_priv *priv;
622 	struct device *codec_dev[2] = { NULL, NULL };
623 	struct snd_soc_dai_link_component *dlc;
624 	const char *codec_dai_name[2];
625 	const char *codec_dev_name[2];
626 	u32 asrc_fmt = 0;
627 	int codec_idx;
628 	u32 width;
629 	int ret;
630 
631 	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
632 	if (!priv)
633 		return -ENOMEM;
634 
635 	priv->pdev = pdev;
636 
637 	cpu_np = of_parse_phandle(np, "audio-cpu", 0);
638 	/* Give a chance to old DT bindings */
639 	if (!cpu_np)
640 		cpu_np = of_parse_phandle(np, "ssi-controller", 0);
641 	if (!cpu_np)
642 		cpu_np = of_parse_phandle(np, "spdif-controller", 0);
643 	if (!cpu_np) {
644 		dev_err(&pdev->dev, "CPU phandle missing or invalid\n");
645 		ret = -EINVAL;
646 		goto fail;
647 	}
648 
649 	cpu_pdev = of_find_device_by_node(cpu_np);
650 	if (!cpu_pdev) {
651 		dev_err(&pdev->dev, "failed to find CPU DAI device\n");
652 		ret = -EINVAL;
653 		goto fail;
654 	}
655 
656 	codec_np[0] = of_parse_phandle(np, "audio-codec", 0);
657 	codec_np[1] = of_parse_phandle(np, "audio-codec", 1);
658 
659 	for (codec_idx = 0; codec_idx < 2; codec_idx++) {
660 		if (codec_np[codec_idx]) {
661 			struct platform_device *codec_pdev;
662 			struct i2c_client *codec_i2c;
663 
664 			codec_i2c = of_find_i2c_device_by_node(codec_np[codec_idx]);
665 			if (codec_i2c) {
666 				codec_dev[codec_idx] = &codec_i2c->dev;
667 				codec_dev_name[codec_idx] = codec_i2c->name;
668 			}
669 			if (!codec_dev[codec_idx]) {
670 				codec_pdev = of_find_device_by_node(codec_np[codec_idx]);
671 				if (codec_pdev) {
672 					codec_dev[codec_idx] = &codec_pdev->dev;
673 					codec_dev_name[codec_idx] = codec_pdev->name;
674 				}
675 			}
676 		}
677 	}
678 
679 	asrc_np = of_parse_phandle(np, "audio-asrc", 0);
680 	if (asrc_np)
681 		asrc_pdev = of_find_device_by_node(asrc_np);
682 
683 	/* Get the MCLK rate only, and leave it controlled by CODEC drivers */
684 	for (codec_idx = 0; codec_idx < 2; codec_idx++) {
685 		if (codec_dev[codec_idx]) {
686 			struct clk *codec_clk = clk_get(codec_dev[codec_idx], NULL);
687 
688 			if (!IS_ERR(codec_clk)) {
689 				priv->codec_priv[codec_idx].mclk_freq = clk_get_rate(codec_clk);
690 				clk_put(codec_clk);
691 			}
692 		}
693 	}
694 
695 	/* Default sample rate and format, will be updated in hw_params() */
696 	priv->sample_rate = 44100;
697 	priv->sample_format = SNDRV_PCM_FORMAT_S16_LE;
698 
699 	/* Assign a default DAI format, and allow each card to overwrite it */
700 	priv->dai_fmt = DAI_FMT_BASE;
701 
702 	memcpy(priv->dai_link, fsl_asoc_card_dai,
703 	       sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
704 	/*
705 	 * "Default ASoC DAI Link": 1 cpus, 2 codecs, 1 platforms
706 	 * "DPCM Link Front-End":  1 cpus, 1 codecs (dummy), 1 platforms
707 	 * "DPCM Link Back-End": 1 cpus, 2 codecs
708 	 * totally 10 components
709 	 */
710 	dlc = devm_kcalloc(&pdev->dev, 10, sizeof(*dlc), GFP_KERNEL);
711 	if (!dlc) {
712 		ret = -ENOMEM;
713 		goto asrc_fail;
714 	}
715 
716 	priv->dai_link[0].cpus = &dlc[0];
717 	priv->dai_link[0].num_cpus = 1;
718 	priv->dai_link[0].codecs = &dlc[1];
719 	priv->dai_link[0].num_codecs = 1;
720 	priv->dai_link[0].platforms = &dlc[3];
721 	priv->dai_link[0].num_platforms = 1;
722 
723 	priv->dai_link[1].cpus = &dlc[4];
724 	priv->dai_link[1].num_cpus = 1;
725 	priv->dai_link[1].codecs = &dlc[5];
726 	priv->dai_link[1].num_codecs = 0; /* dummy */
727 	priv->dai_link[1].platforms = &dlc[6];
728 	priv->dai_link[1].num_platforms = 1;
729 
730 	priv->dai_link[2].cpus = &dlc[7];
731 	priv->dai_link[2].num_cpus = 1;
732 	priv->dai_link[2].codecs = &dlc[8];
733 	priv->dai_link[2].num_codecs = 1;
734 
735 	priv->card.dapm_routes = audio_map;
736 	priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
737 	priv->card.driver_name = DRIVER_NAME;
738 
739 	for (codec_idx = 0; codec_idx < 2; codec_idx++) {
740 		priv->codec_priv[codec_idx].fll_id = -1;
741 		priv->codec_priv[codec_idx].pll_id = -1;
742 	}
743 
744 	/* Diversify the card configurations */
745 	if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
746 		codec_dai_name[0] = "cs42888";
747 		priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv[0].mclk_freq;
748 		priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv[0].mclk_freq;
749 		priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT;
750 		priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT;
751 		priv->cpu_priv.slot_width = 32;
752 		priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC;
753 	} else if (of_device_is_compatible(np, "fsl,imx-audio-cs427x")) {
754 		codec_dai_name[0] = "cs4271-hifi";
755 		priv->codec_priv[0].mclk_id = CS427x_SYSCLK_MCLK;
756 		priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
757 	} else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) {
758 		codec_dai_name[0] = "sgtl5000";
759 		priv->codec_priv[0].mclk_id = SGTL5000_SYSCLK;
760 		priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
761 	} else if (of_device_is_compatible(np, "fsl,imx-audio-tlv320aic32x4")) {
762 		codec_dai_name[0] = "tlv320aic32x4-hifi";
763 		priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
764 	} else if (of_device_is_compatible(np, "fsl,imx-audio-tlv320aic31xx")) {
765 		codec_dai_name[0] = "tlv320dac31xx-hifi";
766 		priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
767 		priv->dai_link[1].dpcm_capture = 0;
768 		priv->dai_link[2].dpcm_capture = 0;
769 		priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT;
770 		priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT;
771 		priv->card.dapm_routes = audio_map_tx;
772 		priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx);
773 	} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) {
774 		codec_dai_name[0] = "wm8962";
775 		priv->codec_priv[0].mclk_id = WM8962_SYSCLK_MCLK;
776 		priv->codec_priv[0].fll_id = WM8962_SYSCLK_FLL;
777 		priv->codec_priv[0].pll_id = WM8962_FLL;
778 		priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
779 	} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) {
780 		codec_dai_name[0] = "wm8960-hifi";
781 		priv->codec_priv[0].fll_id = WM8960_SYSCLK_AUTO;
782 		priv->codec_priv[0].pll_id = WM8960_SYSCLK_AUTO;
783 		priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
784 	} else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) {
785 		codec_dai_name[0] = "ac97-hifi";
786 		priv->dai_fmt = SND_SOC_DAIFMT_AC97;
787 		priv->card.dapm_routes = audio_map_ac97;
788 		priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_ac97);
789 	} else if (of_device_is_compatible(np, "fsl,imx-audio-mqs")) {
790 		codec_dai_name[0] = "fsl-mqs-dai";
791 		priv->dai_fmt = SND_SOC_DAIFMT_LEFT_J |
792 				SND_SOC_DAIFMT_CBC_CFC |
793 				SND_SOC_DAIFMT_NB_NF;
794 		priv->dai_link[1].dpcm_capture = 0;
795 		priv->dai_link[2].dpcm_capture = 0;
796 		priv->card.dapm_routes = audio_map_tx;
797 		priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx);
798 	} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8524")) {
799 		codec_dai_name[0] = "wm8524-hifi";
800 		priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC;
801 		priv->dai_link[1].dpcm_capture = 0;
802 		priv->dai_link[2].dpcm_capture = 0;
803 		priv->cpu_priv.slot_width = 32;
804 		priv->card.dapm_routes = audio_map_tx;
805 		priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx);
806 	} else if (of_device_is_compatible(np, "fsl,imx-audio-si476x")) {
807 		codec_dai_name[0] = "si476x-codec";
808 		priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC;
809 		priv->card.dapm_routes = audio_map_rx;
810 		priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_rx);
811 	} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8958")) {
812 		codec_dai_name[0] = "wm8994-aif1";
813 		priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
814 		priv->codec_priv[0].mclk_id = WM8994_FLL_SRC_MCLK1;
815 		priv->codec_priv[0].fll_id = WM8994_SYSCLK_FLL1;
816 		priv->codec_priv[0].pll_id = WM8994_FLL1;
817 		priv->codec_priv[0].free_freq = priv->codec_priv[0].mclk_freq;
818 		priv->card.dapm_routes = NULL;
819 		priv->card.num_dapm_routes = 0;
820 	} else if (of_device_is_compatible(np, "fsl,imx-audio-nau8822")) {
821 		codec_dai_name[0] = "nau8822-hifi";
822 		priv->codec_priv[0].mclk_id = NAU8822_CLK_MCLK;
823 		priv->codec_priv[0].fll_id = NAU8822_CLK_PLL;
824 		priv->codec_priv[0].pll_id = NAU8822_CLK_PLL;
825 		priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
826 		if (codec_dev[0])
827 			priv->codec_priv[0].mclk = devm_clk_get(codec_dev[0], NULL);
828 	} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8904")) {
829 		codec_dai_name[0] = "wm8904-hifi";
830 		priv->codec_priv[0].mclk_id = WM8904_FLL_MCLK;
831 		priv->codec_priv[0].fll_id = WM8904_CLK_FLL;
832 		priv->codec_priv[0].pll_id = WM8904_FLL_MCLK;
833 		priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
834 	} else if (of_device_is_compatible(np, "fsl,imx-audio-spdif")) {
835 		ret = fsl_asoc_card_spdif_init(codec_np, cpu_np, codec_dai_name, priv);
836 		if (ret)
837 			goto asrc_fail;
838 	} else {
839 		dev_err(&pdev->dev, "unknown Device Tree compatible\n");
840 		ret = -EINVAL;
841 		goto asrc_fail;
842 	}
843 
844 	/*
845 	 * Allow setting mclk-id from the device-tree node. Otherwise, the
846 	 * default value for each card configuration is used.
847 	 */
848 	for_each_link_codecs((&(priv->dai_link[0])), codec_idx, codec_comp) {
849 		of_property_read_u32_index(np, "mclk-id", codec_idx,
850 					&priv->codec_priv[codec_idx].mclk_id);
851 	}
852 
853 	/* Format info from DT is optional. */
854 	snd_soc_daifmt_parse_clock_provider_as_phandle(np, NULL, &bitclkprovider, &frameprovider);
855 	if (bitclkprovider || frameprovider) {
856 		unsigned int daifmt = snd_soc_daifmt_parse_format(np, NULL);
857 		bool codec_bitclkprovider = false;
858 		bool codec_frameprovider = false;
859 
860 		for_each_link_codecs((&(priv->dai_link[0])), codec_idx, codec_comp) {
861 			if (bitclkprovider && codec_np[codec_idx] == bitclkprovider)
862 				codec_bitclkprovider = true;
863 			if (frameprovider && codec_np[codec_idx] == frameprovider)
864 				codec_frameprovider = true;
865 		}
866 
867 		if (codec_bitclkprovider)
868 			daifmt |= (codec_frameprovider) ?
869 				SND_SOC_DAIFMT_CBP_CFP : SND_SOC_DAIFMT_CBP_CFC;
870 		else
871 			daifmt |= (codec_frameprovider) ?
872 				SND_SOC_DAIFMT_CBC_CFP : SND_SOC_DAIFMT_CBC_CFC;
873 
874 		/* Override dai_fmt with value from DT */
875 		priv->dai_fmt = daifmt;
876 	}
877 
878 	/* Change direction according to format */
879 	if (priv->dai_fmt & SND_SOC_DAIFMT_CBP_CFP) {
880 		priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_IN;
881 		priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_IN;
882 	}
883 
884 	of_node_put(bitclkprovider);
885 	of_node_put(frameprovider);
886 
887 	if (!fsl_asoc_card_is_ac97(priv) && !codec_dev[0]
888 	    && codec_dai_name[0] != snd_soc_dummy_dlc.dai_name) {
889 		dev_dbg(&pdev->dev, "failed to find codec device\n");
890 		ret = -EPROBE_DEFER;
891 		goto asrc_fail;
892 	}
893 
894 	/* Common settings for corresponding Freescale CPU DAI driver */
895 	if (of_node_name_eq(cpu_np, "ssi")) {
896 		/* Only SSI needs to configure AUDMUX */
897 		ret = fsl_asoc_card_audmux_init(np, priv);
898 		if (ret) {
899 			dev_err(&pdev->dev, "failed to init audmux\n");
900 			goto asrc_fail;
901 		}
902 	} else if (of_node_name_eq(cpu_np, "esai")) {
903 		struct clk *esai_clk = clk_get(&cpu_pdev->dev, "extal");
904 
905 		if (!IS_ERR(esai_clk)) {
906 			priv->cpu_priv.sysclk_freq[TX] = clk_get_rate(esai_clk);
907 			priv->cpu_priv.sysclk_freq[RX] = clk_get_rate(esai_clk);
908 			clk_put(esai_clk);
909 		} else if (PTR_ERR(esai_clk) == -EPROBE_DEFER) {
910 			ret = -EPROBE_DEFER;
911 			goto asrc_fail;
912 		}
913 
914 		priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL;
915 		priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL;
916 	} else if (of_node_name_eq(cpu_np, "sai")) {
917 		priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1;
918 		priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1;
919 	}
920 
921 	/* Initialize sound card */
922 	priv->card.dev = &pdev->dev;
923 	priv->card.owner = THIS_MODULE;
924 	ret = snd_soc_of_parse_card_name(&priv->card, "model");
925 	if (ret) {
926 		snprintf(priv->name, sizeof(priv->name), "%s-audio",
927 			 fsl_asoc_card_is_ac97(priv) ? "ac97" : codec_dev_name[0]);
928 		priv->card.name = priv->name;
929 	}
930 	priv->card.dai_link = priv->dai_link;
931 	priv->card.late_probe = fsl_asoc_card_late_probe;
932 	priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;
933 	priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets);
934 
935 	/* Drop the second half of DAPM routes -- ASRC */
936 	if (!asrc_pdev)
937 		priv->card.num_dapm_routes /= 2;
938 
939 	if (of_property_read_bool(np, "audio-routing")) {
940 		ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing");
941 		if (ret) {
942 			dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret);
943 			goto asrc_fail;
944 		}
945 	}
946 
947 	/* Normal DAI Link */
948 	priv->dai_link[0].cpus->of_node = cpu_np;
949 	for_each_link_codecs((&(priv->dai_link[0])), codec_idx, codec_comp) {
950 		codec_comp->dai_name = codec_dai_name[codec_idx];
951 	}
952 
953 	// Old SPDIF DT binding support
954 	if (codec_dai_name[0] == snd_soc_dummy_dlc.dai_name)
955 		priv->dai_link[0].codecs[0].name = snd_soc_dummy_dlc.name;
956 
957 	if (!fsl_asoc_card_is_ac97(priv)) {
958 		for_each_link_codecs((&(priv->dai_link[0])), codec_idx, codec_comp) {
959 			codec_comp->of_node = codec_np[codec_idx];
960 		}
961 	} else {
962 		u32 idx;
963 
964 		ret = of_property_read_u32(cpu_np, "cell-index", &idx);
965 		if (ret) {
966 			dev_err(&pdev->dev,
967 				"cannot get CPU index property\n");
968 			goto asrc_fail;
969 		}
970 
971 		priv->dai_link[0].codecs[0].name =
972 				devm_kasprintf(&pdev->dev, GFP_KERNEL,
973 					       "ac97-codec.%u",
974 					       (unsigned int)idx);
975 		if (!priv->dai_link[0].codecs[0].name) {
976 			ret = -ENOMEM;
977 			goto asrc_fail;
978 		}
979 	}
980 
981 	priv->dai_link[0].platforms->of_node = cpu_np;
982 	priv->dai_link[0].dai_fmt = priv->dai_fmt;
983 	priv->card.num_links = 1;
984 
985 	if (asrc_pdev) {
986 		/* DPCM DAI Links only if ASRC exists */
987 		priv->dai_link[1].cpus->of_node = asrc_np;
988 		priv->dai_link[1].platforms->of_node = asrc_np;
989 		for_each_link_codecs((&(priv->dai_link[2])), codec_idx, codec_comp) {
990 			codec_comp->dai_name = priv->dai_link[0].codecs[codec_idx].dai_name;
991 			codec_comp->of_node = priv->dai_link[0].codecs[codec_idx].of_node;
992 			codec_comp->name = priv->dai_link[0].codecs[codec_idx].name;
993 		}
994 		priv->dai_link[2].cpus->of_node = cpu_np;
995 		priv->dai_link[2].dai_fmt = priv->dai_fmt;
996 		priv->card.num_links = 3;
997 
998 		ret = of_property_read_u32(asrc_np, "fsl,asrc-rate",
999 					   &priv->asrc_rate);
1000 		if (ret) {
1001 			dev_err(&pdev->dev, "failed to get output rate\n");
1002 			ret = -EINVAL;
1003 			goto asrc_fail;
1004 		}
1005 
1006 		ret = of_property_read_u32(asrc_np, "fsl,asrc-format", &asrc_fmt);
1007 		priv->asrc_format = (__force snd_pcm_format_t)asrc_fmt;
1008 		if (ret) {
1009 			/* Fallback to old binding; translate to asrc_format */
1010 			ret = of_property_read_u32(asrc_np, "fsl,asrc-width",
1011 						   &width);
1012 			if (ret) {
1013 				dev_err(&pdev->dev,
1014 					"failed to decide output format\n");
1015 				goto asrc_fail;
1016 			}
1017 
1018 			if (width == 24)
1019 				priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE;
1020 			else
1021 				priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE;
1022 		}
1023 	}
1024 
1025 	/* Finish card registering */
1026 	platform_set_drvdata(pdev, priv);
1027 	snd_soc_card_set_drvdata(&priv->card, priv);
1028 
1029 	ret = devm_snd_soc_register_card(&pdev->dev, &priv->card);
1030 	if (ret) {
1031 		dev_err_probe(&pdev->dev, ret, "snd_soc_register_card failed\n");
1032 		goto asrc_fail;
1033 	}
1034 
1035 	/*
1036 	 * Properties "hp-det-gpio" and "mic-det-gpio" are optional, and
1037 	 * simple_util_init_jack() uses these properties for creating
1038 	 * Headphone Jack and Microphone Jack.
1039 	 *
1040 	 * The notifier is initialized in snd_soc_card_jack_new(), then
1041 	 * snd_soc_jack_notifier_register can be called.
1042 	 */
1043 	if (of_property_read_bool(np, "hp-det-gpio")) {
1044 		ret = simple_util_init_jack(&priv->card, &priv->hp_jack,
1045 					    1, NULL, "Headphone Jack");
1046 		if (ret)
1047 			goto asrc_fail;
1048 
1049 		snd_soc_jack_notifier_register(&priv->hp_jack.jack, &hp_jack_nb);
1050 	}
1051 
1052 	if (of_property_read_bool(np, "mic-det-gpio")) {
1053 		ret = simple_util_init_jack(&priv->card, &priv->mic_jack,
1054 					    0, NULL, "Mic Jack");
1055 		if (ret)
1056 			goto asrc_fail;
1057 
1058 		snd_soc_jack_notifier_register(&priv->mic_jack.jack, &mic_jack_nb);
1059 	}
1060 
1061 asrc_fail:
1062 	of_node_put(asrc_np);
1063 	of_node_put(codec_np[0]);
1064 	of_node_put(codec_np[1]);
1065 	put_device(&cpu_pdev->dev);
1066 fail:
1067 	of_node_put(cpu_np);
1068 
1069 	return ret;
1070 }
1071 
1072 static const struct of_device_id fsl_asoc_card_dt_ids[] = {
1073 	{ .compatible = "fsl,imx-audio-ac97", },
1074 	{ .compatible = "fsl,imx-audio-cs42888", },
1075 	{ .compatible = "fsl,imx-audio-cs427x", },
1076 	{ .compatible = "fsl,imx-audio-tlv320aic32x4", },
1077 	{ .compatible = "fsl,imx-audio-tlv320aic31xx", },
1078 	{ .compatible = "fsl,imx-audio-sgtl5000", },
1079 	{ .compatible = "fsl,imx-audio-wm8962", },
1080 	{ .compatible = "fsl,imx-audio-wm8960", },
1081 	{ .compatible = "fsl,imx-audio-mqs", },
1082 	{ .compatible = "fsl,imx-audio-wm8524", },
1083 	{ .compatible = "fsl,imx-audio-si476x", },
1084 	{ .compatible = "fsl,imx-audio-wm8958", },
1085 	{ .compatible = "fsl,imx-audio-nau8822", },
1086 	{ .compatible = "fsl,imx-audio-wm8904", },
1087 	{ .compatible = "fsl,imx-audio-spdif", },
1088 	{}
1089 };
1090 MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids);
1091 
1092 static struct platform_driver fsl_asoc_card_driver = {
1093 	.probe = fsl_asoc_card_probe,
1094 	.driver = {
1095 		.name = DRIVER_NAME,
1096 		.pm = &snd_soc_pm_ops,
1097 		.of_match_table = fsl_asoc_card_dt_ids,
1098 	},
1099 };
1100 module_platform_driver(fsl_asoc_card_driver);
1101 
1102 MODULE_DESCRIPTION("Freescale Generic ASoC Sound Card driver with ASRC");
1103 MODULE_AUTHOR("Nicolin Chen <nicoleotsuka@gmail.com>");
1104 MODULE_ALIAS("platform:" DRIVER_NAME);
1105 MODULE_LICENSE("GPL");
1106