driver core: bus: use to_subsys_private and to_device_private_bus
[linux/fpc-iii.git] / sound / soc / fsl / fsl-asoc-card.c
blob562b3bd22d9ac1f86de0a7d08fe64d8145445db8
1 /*
2 * Freescale Generic ASoC Sound Card driver with ASRC
4 * Copyright (C) 2014 Freescale Semiconductor, Inc.
6 * Author: Nicolin Chen <nicoleotsuka@gmail.com>
8 * This file is licensed under the terms of the GNU General Public License
9 * version 2. This program is licensed "as is" without any warranty of any
10 * kind, whether express or implied.
13 #include <linux/clk.h>
14 #include <linux/i2c.h>
15 #include <linux/module.h>
16 #include <linux/of_platform.h>
17 #if IS_ENABLED(CONFIG_SND_AC97_CODEC)
18 #include <sound/ac97_codec.h>
19 #endif
20 #include <sound/pcm_params.h>
21 #include <sound/soc.h>
23 #include "fsl_esai.h"
24 #include "fsl_sai.h"
25 #include "imx-audmux.h"
27 #include "../codecs/sgtl5000.h"
28 #include "../codecs/wm8962.h"
29 #include "../codecs/wm8960.h"
31 #define RX 0
32 #define TX 1
34 /* Default DAI format without Master and Slave flag */
35 #define DAI_FMT_BASE (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF)
37 /**
38 * CODEC private data
40 * @mclk_freq: Clock rate of MCLK
41 * @mclk_id: MCLK (or main clock) id for set_sysclk()
42 * @fll_id: FLL (or secordary clock) id for set_sysclk()
43 * @pll_id: PLL id for set_pll()
45 struct codec_priv {
46 unsigned long mclk_freq;
47 u32 mclk_id;
48 u32 fll_id;
49 u32 pll_id;
52 /**
53 * CPU private data
55 * @sysclk_freq[2]: SYSCLK rates for set_sysclk()
56 * @sysclk_dir[2]: SYSCLK directions for set_sysclk()
57 * @sysclk_id[2]: SYSCLK ids for set_sysclk()
58 * @slot_width: Slot width of each frame
60 * Note: [1] for tx and [0] for rx
62 struct cpu_priv {
63 unsigned long sysclk_freq[2];
64 u32 sysclk_dir[2];
65 u32 sysclk_id[2];
66 u32 slot_width;
69 /**
70 * Freescale Generic ASOC card private data
72 * @dai_link[3]: DAI link structure including normal one and DPCM link
73 * @pdev: platform device pointer
74 * @codec_priv: CODEC private data
75 * @cpu_priv: CPU private data
76 * @card: ASoC card structure
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
85 struct fsl_asoc_card_priv {
86 struct snd_soc_dai_link dai_link[3];
87 struct platform_device *pdev;
88 struct codec_priv codec_priv;
89 struct cpu_priv cpu_priv;
90 struct snd_soc_card card;
91 u32 sample_rate;
92 u32 sample_format;
93 u32 asrc_rate;
94 u32 asrc_format;
95 u32 dai_fmt;
96 char name[32];
99 /**
100 * This dapm route map exsits for DPCM link only.
101 * The other routes shall go through Device Tree.
103 static const struct snd_soc_dapm_route audio_map[] = {
104 {"CPU-Playback", NULL, "ASRC-Playback"},
105 {"Playback", NULL, "CPU-Playback"},
106 {"ASRC-Capture", NULL, "CPU-Capture"},
107 {"CPU-Capture", NULL, "Capture"},
110 static const struct snd_soc_dapm_route audio_map_ac97[] = {
111 {"AC97 Playback", NULL, "ASRC-Playback"},
112 {"Playback", NULL, "AC97 Playback"},
113 {"ASRC-Capture", NULL, "AC97 Capture"},
114 {"AC97 Capture", NULL, "Capture"},
117 /* Add all possible widgets into here without being redundant */
118 static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = {
119 SND_SOC_DAPM_LINE("Line Out Jack", NULL),
120 SND_SOC_DAPM_LINE("Line In Jack", NULL),
121 SND_SOC_DAPM_HP("Headphone Jack", NULL),
122 SND_SOC_DAPM_SPK("Ext Spk", NULL),
123 SND_SOC_DAPM_MIC("Mic Jack", NULL),
124 SND_SOC_DAPM_MIC("AMIC", NULL),
125 SND_SOC_DAPM_MIC("DMIC", NULL),
128 static bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv)
130 return priv->dai_fmt == SND_SOC_DAIFMT_AC97;
133 static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
134 struct snd_pcm_hw_params *params)
136 struct snd_soc_pcm_runtime *rtd = substream->private_data;
137 struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
138 bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
139 struct cpu_priv *cpu_priv = &priv->cpu_priv;
140 struct device *dev = rtd->card->dev;
141 int ret;
143 priv->sample_rate = params_rate(params);
144 priv->sample_format = params_format(params);
147 * If codec-dai is DAI Master and all configurations are already in the
148 * set_bias_level(), bypass the remaining settings in hw_params().
149 * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS.
151 if ((priv->card.set_bias_level &&
152 priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) ||
153 fsl_asoc_card_is_ac97(priv))
154 return 0;
156 /* Specific configurations of DAIs starts from here */
157 ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, cpu_priv->sysclk_id[tx],
158 cpu_priv->sysclk_freq[tx],
159 cpu_priv->sysclk_dir[tx]);
160 if (ret) {
161 dev_err(dev, "failed to set sysclk for cpu dai\n");
162 return ret;
165 if (cpu_priv->slot_width) {
166 ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2,
167 cpu_priv->slot_width);
168 if (ret) {
169 dev_err(dev, "failed to set TDM slot for cpu dai\n");
170 return ret;
174 return 0;
177 static struct snd_soc_ops fsl_asoc_card_ops = {
178 .hw_params = fsl_asoc_card_hw_params,
181 static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
182 struct snd_pcm_hw_params *params)
184 struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
185 struct snd_interval *rate;
186 struct snd_mask *mask;
188 rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
189 rate->max = rate->min = priv->asrc_rate;
191 mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
192 snd_mask_none(mask);
193 snd_mask_set(mask, priv->asrc_format);
195 return 0;
198 static struct snd_soc_dai_link fsl_asoc_card_dai[] = {
199 /* Default ASoC DAI Link*/
201 .name = "HiFi",
202 .stream_name = "HiFi",
203 .ops = &fsl_asoc_card_ops,
205 /* DPCM Link between Front-End and Back-End (Optional) */
207 .name = "HiFi-ASRC-FE",
208 .stream_name = "HiFi-ASRC-FE",
209 .codec_name = "snd-soc-dummy",
210 .codec_dai_name = "snd-soc-dummy-dai",
211 .dpcm_playback = 1,
212 .dpcm_capture = 1,
213 .dynamic = 1,
216 .name = "HiFi-ASRC-BE",
217 .stream_name = "HiFi-ASRC-BE",
218 .platform_name = "snd-soc-dummy",
219 .be_hw_params_fixup = be_hw_params_fixup,
220 .ops = &fsl_asoc_card_ops,
221 .dpcm_playback = 1,
222 .dpcm_capture = 1,
223 .no_pcm = 1,
227 static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card,
228 struct snd_soc_dapm_context *dapm,
229 enum snd_soc_bias_level level)
231 struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
232 struct snd_soc_pcm_runtime *rtd;
233 struct snd_soc_dai *codec_dai;
234 struct codec_priv *codec_priv = &priv->codec_priv;
235 struct device *dev = card->dev;
236 unsigned int pll_out;
237 int ret;
239 rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name);
240 codec_dai = rtd->codec_dai;
241 if (dapm->dev != codec_dai->dev)
242 return 0;
244 switch (level) {
245 case SND_SOC_BIAS_PREPARE:
246 if (dapm->bias_level != SND_SOC_BIAS_STANDBY)
247 break;
249 if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE)
250 pll_out = priv->sample_rate * 384;
251 else
252 pll_out = priv->sample_rate * 256;
254 ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id,
255 codec_priv->mclk_id,
256 codec_priv->mclk_freq, pll_out);
257 if (ret) {
258 dev_err(dev, "failed to start FLL: %d\n", ret);
259 return ret;
262 ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id,
263 pll_out, SND_SOC_CLOCK_IN);
264 if (ret) {
265 dev_err(dev, "failed to set SYSCLK: %d\n", ret);
266 return ret;
268 break;
270 case SND_SOC_BIAS_STANDBY:
271 if (dapm->bias_level != SND_SOC_BIAS_PREPARE)
272 break;
274 ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
275 codec_priv->mclk_freq,
276 SND_SOC_CLOCK_IN);
277 if (ret) {
278 dev_err(dev, "failed to switch away from FLL: %d\n", ret);
279 return ret;
282 ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, 0, 0, 0);
283 if (ret) {
284 dev_err(dev, "failed to stop FLL: %d\n", ret);
285 return ret;
287 break;
289 default:
290 break;
293 return 0;
296 static int fsl_asoc_card_audmux_init(struct device_node *np,
297 struct fsl_asoc_card_priv *priv)
299 struct device *dev = &priv->pdev->dev;
300 u32 int_ptcr = 0, ext_ptcr = 0;
301 int int_port, ext_port;
302 int ret;
304 ret = of_property_read_u32(np, "mux-int-port", &int_port);
305 if (ret) {
306 dev_err(dev, "mux-int-port missing or invalid\n");
307 return ret;
309 ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
310 if (ret) {
311 dev_err(dev, "mux-ext-port missing or invalid\n");
312 return ret;
316 * The port numbering in the hardware manual starts at 1, while
317 * the AUDMUX API expects it starts at 0.
319 int_port--;
320 ext_port--;
323 * Use asynchronous mode (6 wires) for all cases except AC97.
324 * If only 4 wires are needed, just set SSI into
325 * synchronous mode and enable 4 PADs in IOMUX.
327 switch (priv->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) {
328 case SND_SOC_DAIFMT_CBM_CFM:
329 int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
330 IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
331 IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
332 IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
333 IMX_AUDMUX_V2_PTCR_RFSDIR |
334 IMX_AUDMUX_V2_PTCR_RCLKDIR |
335 IMX_AUDMUX_V2_PTCR_TFSDIR |
336 IMX_AUDMUX_V2_PTCR_TCLKDIR;
337 break;
338 case SND_SOC_DAIFMT_CBM_CFS:
339 int_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
340 IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
341 IMX_AUDMUX_V2_PTCR_RCLKDIR |
342 IMX_AUDMUX_V2_PTCR_TCLKDIR;
343 ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
344 IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
345 IMX_AUDMUX_V2_PTCR_RFSDIR |
346 IMX_AUDMUX_V2_PTCR_TFSDIR;
347 break;
348 case SND_SOC_DAIFMT_CBS_CFM:
349 int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
350 IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
351 IMX_AUDMUX_V2_PTCR_RFSDIR |
352 IMX_AUDMUX_V2_PTCR_TFSDIR;
353 ext_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
354 IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
355 IMX_AUDMUX_V2_PTCR_RCLKDIR |
356 IMX_AUDMUX_V2_PTCR_TCLKDIR;
357 break;
358 case SND_SOC_DAIFMT_CBS_CFS:
359 ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
360 IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
361 IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
362 IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
363 IMX_AUDMUX_V2_PTCR_RFSDIR |
364 IMX_AUDMUX_V2_PTCR_RCLKDIR |
365 IMX_AUDMUX_V2_PTCR_TFSDIR |
366 IMX_AUDMUX_V2_PTCR_TCLKDIR;
367 break;
368 default:
369 if (!fsl_asoc_card_is_ac97(priv))
370 return -EINVAL;
373 if (fsl_asoc_card_is_ac97(priv)) {
374 int_ptcr = IMX_AUDMUX_V2_PTCR_SYN |
375 IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
376 IMX_AUDMUX_V2_PTCR_TCLKDIR;
377 ext_ptcr = IMX_AUDMUX_V2_PTCR_SYN |
378 IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
379 IMX_AUDMUX_V2_PTCR_TFSDIR;
382 /* Asynchronous mode can not be set along with RCLKDIR */
383 if (!fsl_asoc_card_is_ac97(priv)) {
384 unsigned int pdcr =
385 IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port);
387 ret = imx_audmux_v2_configure_port(int_port, 0,
388 pdcr);
389 if (ret) {
390 dev_err(dev, "audmux internal port setup failed\n");
391 return ret;
395 ret = imx_audmux_v2_configure_port(int_port, int_ptcr,
396 IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
397 if (ret) {
398 dev_err(dev, "audmux internal port setup failed\n");
399 return ret;
402 if (!fsl_asoc_card_is_ac97(priv)) {
403 unsigned int pdcr =
404 IMX_AUDMUX_V2_PDCR_RXDSEL(int_port);
406 ret = imx_audmux_v2_configure_port(ext_port, 0,
407 pdcr);
408 if (ret) {
409 dev_err(dev, "audmux external port setup failed\n");
410 return ret;
414 ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr,
415 IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
416 if (ret) {
417 dev_err(dev, "audmux external port setup failed\n");
418 return ret;
421 return 0;
424 static int fsl_asoc_card_late_probe(struct snd_soc_card *card)
426 struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
427 struct snd_soc_pcm_runtime *rtd = list_first_entry(
428 &card->rtd_list, struct snd_soc_pcm_runtime, list);
429 struct snd_soc_dai *codec_dai = rtd->codec_dai;
430 struct codec_priv *codec_priv = &priv->codec_priv;
431 struct device *dev = card->dev;
432 int ret;
434 if (fsl_asoc_card_is_ac97(priv)) {
435 #if IS_ENABLED(CONFIG_SND_AC97_CODEC)
436 struct snd_soc_codec *codec = rtd->codec;
437 struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
440 * Use slots 3/4 for S/PDIF so SSI won't try to enable
441 * other slots and send some samples there
442 * due to SLOTREQ bits for S/PDIF received from codec
444 snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS,
445 AC97_EA_SPSA_SLOT_MASK, AC97_EA_SPSA_3_4);
446 #endif
448 return 0;
451 ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
452 codec_priv->mclk_freq, SND_SOC_CLOCK_IN);
453 if (ret) {
454 dev_err(dev, "failed to set sysclk in %s\n", __func__);
455 return ret;
458 return 0;
461 static int fsl_asoc_card_probe(struct platform_device *pdev)
463 struct device_node *cpu_np, *codec_np, *asrc_np;
464 struct device_node *np = pdev->dev.of_node;
465 struct platform_device *asrc_pdev = NULL;
466 struct platform_device *cpu_pdev;
467 struct fsl_asoc_card_priv *priv;
468 struct i2c_client *codec_dev;
469 const char *codec_dai_name;
470 u32 width;
471 int ret;
473 priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
474 if (!priv)
475 return -ENOMEM;
477 cpu_np = of_parse_phandle(np, "audio-cpu", 0);
478 /* Give a chance to old DT binding */
479 if (!cpu_np)
480 cpu_np = of_parse_phandle(np, "ssi-controller", 0);
481 if (!cpu_np) {
482 dev_err(&pdev->dev, "CPU phandle missing or invalid\n");
483 ret = -EINVAL;
484 goto fail;
487 cpu_pdev = of_find_device_by_node(cpu_np);
488 if (!cpu_pdev) {
489 dev_err(&pdev->dev, "failed to find CPU DAI device\n");
490 ret = -EINVAL;
491 goto fail;
494 codec_np = of_parse_phandle(np, "audio-codec", 0);
495 if (codec_np)
496 codec_dev = of_find_i2c_device_by_node(codec_np);
497 else
498 codec_dev = NULL;
500 asrc_np = of_parse_phandle(np, "audio-asrc", 0);
501 if (asrc_np)
502 asrc_pdev = of_find_device_by_node(asrc_np);
504 /* Get the MCLK rate only, and leave it controlled by CODEC drivers */
505 if (codec_dev) {
506 struct clk *codec_clk = clk_get(&codec_dev->dev, NULL);
508 if (!IS_ERR(codec_clk)) {
509 priv->codec_priv.mclk_freq = clk_get_rate(codec_clk);
510 clk_put(codec_clk);
514 /* Default sample rate and format, will be updated in hw_params() */
515 priv->sample_rate = 44100;
516 priv->sample_format = SNDRV_PCM_FORMAT_S16_LE;
518 /* Assign a default DAI format, and allow each card to overwrite it */
519 priv->dai_fmt = DAI_FMT_BASE;
521 /* Diversify the card configurations */
522 if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
523 codec_dai_name = "cs42888";
524 priv->card.set_bias_level = NULL;
525 priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq;
526 priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq;
527 priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT;
528 priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT;
529 priv->cpu_priv.slot_width = 32;
530 priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
531 } else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) {
532 codec_dai_name = "sgtl5000";
533 priv->codec_priv.mclk_id = SGTL5000_SYSCLK;
534 priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
535 } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) {
536 codec_dai_name = "wm8962";
537 priv->card.set_bias_level = fsl_asoc_card_set_bias_level;
538 priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK;
539 priv->codec_priv.fll_id = WM8962_SYSCLK_FLL;
540 priv->codec_priv.pll_id = WM8962_FLL;
541 priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
542 } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) {
543 codec_dai_name = "wm8960-hifi";
544 priv->card.set_bias_level = fsl_asoc_card_set_bias_level;
545 priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO;
546 priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO;
547 priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
548 } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) {
549 codec_dai_name = "ac97-hifi";
550 priv->card.set_bias_level = NULL;
551 priv->dai_fmt = SND_SOC_DAIFMT_AC97;
552 } else {
553 dev_err(&pdev->dev, "unknown Device Tree compatible\n");
554 ret = -EINVAL;
555 goto asrc_fail;
558 if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) {
559 dev_err(&pdev->dev, "failed to find codec device\n");
560 ret = -EINVAL;
561 goto asrc_fail;
564 /* Common settings for corresponding Freescale CPU DAI driver */
565 if (strstr(cpu_np->name, "ssi")) {
566 /* Only SSI needs to configure AUDMUX */
567 ret = fsl_asoc_card_audmux_init(np, priv);
568 if (ret) {
569 dev_err(&pdev->dev, "failed to init audmux\n");
570 goto asrc_fail;
572 } else if (strstr(cpu_np->name, "esai")) {
573 priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL;
574 priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL;
575 } else if (strstr(cpu_np->name, "sai")) {
576 priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1;
577 priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1;
580 snprintf(priv->name, sizeof(priv->name), "%s-audio",
581 fsl_asoc_card_is_ac97(priv) ? "ac97" :
582 codec_dev->name);
584 /* Initialize sound card */
585 priv->pdev = pdev;
586 priv->card.dev = &pdev->dev;
587 priv->card.name = priv->name;
588 priv->card.dai_link = priv->dai_link;
589 priv->card.dapm_routes = fsl_asoc_card_is_ac97(priv) ?
590 audio_map_ac97 : audio_map;
591 priv->card.late_probe = fsl_asoc_card_late_probe;
592 priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
593 priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;
594 priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets);
596 memcpy(priv->dai_link, fsl_asoc_card_dai,
597 sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
599 ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing");
600 if (ret) {
601 dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret);
602 goto asrc_fail;
605 /* Normal DAI Link */
606 priv->dai_link[0].cpu_of_node = cpu_np;
607 priv->dai_link[0].codec_dai_name = codec_dai_name;
609 if (!fsl_asoc_card_is_ac97(priv))
610 priv->dai_link[0].codec_of_node = codec_np;
611 else {
612 u32 idx;
614 ret = of_property_read_u32(cpu_np, "cell-index", &idx);
615 if (ret) {
616 dev_err(&pdev->dev,
617 "cannot get CPU index property\n");
618 goto asrc_fail;
621 priv->dai_link[0].codec_name =
622 devm_kasprintf(&pdev->dev, GFP_KERNEL,
623 "ac97-codec.%u",
624 (unsigned int)idx);
627 priv->dai_link[0].platform_of_node = cpu_np;
628 priv->dai_link[0].dai_fmt = priv->dai_fmt;
629 priv->card.num_links = 1;
631 if (asrc_pdev) {
632 /* DPCM DAI Links only if ASRC exsits */
633 priv->dai_link[1].cpu_of_node = asrc_np;
634 priv->dai_link[1].platform_of_node = asrc_np;
635 priv->dai_link[2].codec_dai_name = codec_dai_name;
636 priv->dai_link[2].codec_of_node = codec_np;
637 priv->dai_link[2].codec_name =
638 priv->dai_link[0].codec_name;
639 priv->dai_link[2].cpu_of_node = cpu_np;
640 priv->dai_link[2].dai_fmt = priv->dai_fmt;
641 priv->card.num_links = 3;
643 ret = of_property_read_u32(asrc_np, "fsl,asrc-rate",
644 &priv->asrc_rate);
645 if (ret) {
646 dev_err(&pdev->dev, "failed to get output rate\n");
647 ret = -EINVAL;
648 goto asrc_fail;
651 ret = of_property_read_u32(asrc_np, "fsl,asrc-width", &width);
652 if (ret) {
653 dev_err(&pdev->dev, "failed to get output rate\n");
654 ret = -EINVAL;
655 goto asrc_fail;
658 if (width == 24)
659 priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE;
660 else
661 priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE;
664 /* Finish card registering */
665 platform_set_drvdata(pdev, priv);
666 snd_soc_card_set_drvdata(&priv->card, priv);
668 ret = devm_snd_soc_register_card(&pdev->dev, &priv->card);
669 if (ret)
670 dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
672 asrc_fail:
673 of_node_put(asrc_np);
674 of_node_put(codec_np);
675 fail:
676 of_node_put(cpu_np);
678 return ret;
681 static const struct of_device_id fsl_asoc_card_dt_ids[] = {
682 { .compatible = "fsl,imx-audio-ac97", },
683 { .compatible = "fsl,imx-audio-cs42888", },
684 { .compatible = "fsl,imx-audio-sgtl5000", },
685 { .compatible = "fsl,imx-audio-wm8962", },
686 { .compatible = "fsl,imx-audio-wm8960", },
689 MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids);
691 static struct platform_driver fsl_asoc_card_driver = {
692 .probe = fsl_asoc_card_probe,
693 .driver = {
694 .name = "fsl-asoc-card",
695 .pm = &snd_soc_pm_ops,
696 .of_match_table = fsl_asoc_card_dt_ids,
699 module_platform_driver(fsl_asoc_card_driver);
701 MODULE_DESCRIPTION("Freescale Generic ASoC Sound Card driver with ASRC");
702 MODULE_AUTHOR("Nicolin Chen <nicoleotsuka@gmail.com>");
703 MODULE_ALIAS("platform:fsl-asoc-card");
704 MODULE_LICENSE("GPL");