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 {"AC97 Playback", NULL, "CPU AC97 Playback"},
121 {"CPU AC97 Capture", NULL, "AC97 Capture"},
122 /* 2nd half -- ASRC DAPM routes */
123 {"CPU AC97 Playback", NULL, "ASRC-Playback"},
124 {"ASRC-Capture", NULL, "CPU 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
fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv * priv)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
fsl_asoc_card_hw_params(struct snd_pcm_substream * substream,struct snd_pcm_hw_params * params)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
fsl_asoc_card_hw_free(struct snd_pcm_substream * substream)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
be_hw_params_fixup(struct snd_soc_pcm_runtime * rtd,struct snd_pcm_hw_params * params)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
fsl_asoc_card_audmux_init(struct device_node * np,struct fsl_asoc_card_priv * priv)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
hp_jack_event(struct notifier_block * nb,unsigned long event,void * data)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
mic_jack_event(struct notifier_block * nb,unsigned long event,void * data)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
fsl_asoc_card_late_probe(struct snd_soc_card * card)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
fsl_asoc_card_probe(struct platform_device * pdev)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 priv->card.owner = THIS_MODULE;
724 ret = snd_soc_of_parse_card_name(&priv->card, "model");
725 if (ret) {
726 snprintf(priv->name, sizeof(priv->name), "%s-audio",
727 fsl_asoc_card_is_ac97(priv) ? "ac97" : codec_dev_name);
728 priv->card.name = priv->name;
729 }
730 priv->card.dai_link = priv->dai_link;
731 priv->card.late_probe = fsl_asoc_card_late_probe;
732 priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;
733 priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets);
734
735 /* Drop the second half of DAPM routes -- ASRC */
736 if (!asrc_pdev)
737 priv->card.num_dapm_routes /= 2;
738
739 if (of_property_read_bool(np, "audio-routing")) {
740 ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing");
741 if (ret) {
742 dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret);
743 goto asrc_fail;
744 }
745 }
746
747 /* Normal DAI Link */
748 priv->dai_link[0].cpus->of_node = cpu_np;
749 priv->dai_link[0].codecs->dai_name = codec_dai_name;
750
751 if (!fsl_asoc_card_is_ac97(priv))
752 priv->dai_link[0].codecs->of_node = codec_np;
753 else {
754 u32 idx;
755
756 ret = of_property_read_u32(cpu_np, "cell-index", &idx);
757 if (ret) {
758 dev_err(&pdev->dev,
759 "cannot get CPU index property\n");
760 goto asrc_fail;
761 }
762
763 priv->dai_link[0].codecs->name =
764 devm_kasprintf(&pdev->dev, GFP_KERNEL,
765 "ac97-codec.%u",
766 (unsigned int)idx);
767 if (!priv->dai_link[0].codecs->name) {
768 ret = -ENOMEM;
769 goto asrc_fail;
770 }
771 }
772
773 priv->dai_link[0].platforms->of_node = cpu_np;
774 priv->dai_link[0].dai_fmt = priv->dai_fmt;
775 priv->card.num_links = 1;
776
777 if (asrc_pdev) {
778 /* DPCM DAI Links only if ASRC exsits */
779 priv->dai_link[1].cpus->of_node = asrc_np;
780 priv->dai_link[1].platforms->of_node = asrc_np;
781 priv->dai_link[2].codecs->dai_name = codec_dai_name;
782 priv->dai_link[2].codecs->of_node = codec_np;
783 priv->dai_link[2].codecs->name =
784 priv->dai_link[0].codecs->name;
785 priv->dai_link[2].cpus->of_node = cpu_np;
786 priv->dai_link[2].dai_fmt = priv->dai_fmt;
787 priv->card.num_links = 3;
788
789 ret = of_property_read_u32(asrc_np, "fsl,asrc-rate",
790 &priv->asrc_rate);
791 if (ret) {
792 dev_err(&pdev->dev, "failed to get output rate\n");
793 ret = -EINVAL;
794 goto asrc_fail;
795 }
796
797 ret = of_property_read_u32(asrc_np, "fsl,asrc-format",
798 &priv->asrc_format);
799 if (ret) {
800 /* Fallback to old binding; translate to asrc_format */
801 ret = of_property_read_u32(asrc_np, "fsl,asrc-width",
802 &width);
803 if (ret) {
804 dev_err(&pdev->dev,
805 "failed to decide output format\n");
806 goto asrc_fail;
807 }
808
809 if (width == 24)
810 priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE;
811 else
812 priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE;
813 }
814 }
815
816 /* Finish card registering */
817 platform_set_drvdata(pdev, priv);
818 snd_soc_card_set_drvdata(&priv->card, priv);
819
820 ret = devm_snd_soc_register_card(&pdev->dev, &priv->card);
821 if (ret) {
822 if (ret != -EPROBE_DEFER)
823 dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
824 goto asrc_fail;
825 }
826
827 /*
828 * Properties "hp-det-gpio" and "mic-det-gpio" are optional, and
829 * asoc_simple_init_jack uses these properties for creating
830 * Headphone Jack and Microphone Jack.
831 *
832 * The notifier is initialized in snd_soc_card_jack_new(), then
833 * snd_soc_jack_notifier_register can be called.
834 */
835 if (of_property_read_bool(np, "hp-det-gpio")) {
836 ret = asoc_simple_init_jack(&priv->card, &priv->hp_jack,
837 1, NULL, "Headphone Jack");
838 if (ret)
839 goto asrc_fail;
840
841 snd_soc_jack_notifier_register(&priv->hp_jack.jack, &hp_jack_nb);
842 }
843
844 if (of_property_read_bool(np, "mic-det-gpio")) {
845 ret = asoc_simple_init_jack(&priv->card, &priv->mic_jack,
846 0, NULL, "Mic Jack");
847 if (ret)
848 goto asrc_fail;
849
850 snd_soc_jack_notifier_register(&priv->mic_jack.jack, &mic_jack_nb);
851 }
852
853 asrc_fail:
854 of_node_put(asrc_np);
855 of_node_put(codec_np);
856 put_device(&cpu_pdev->dev);
857 fail:
858 of_node_put(cpu_np);
859
860 return ret;
861 }
862
863 static const struct of_device_id fsl_asoc_card_dt_ids[] = {
864 { .compatible = "fsl,imx-audio-ac97", },
865 { .compatible = "fsl,imx-audio-cs42888", },
866 { .compatible = "fsl,imx-audio-cs427x", },
867 { .compatible = "fsl,imx-audio-tlv320aic32x4", },
868 { .compatible = "fsl,imx-audio-sgtl5000", },
869 { .compatible = "fsl,imx-audio-wm8962", },
870 { .compatible = "fsl,imx-audio-wm8960", },
871 { .compatible = "fsl,imx-audio-mqs", },
872 { .compatible = "fsl,imx-audio-wm8524", },
873 {}
874 };
875 MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids);
876
877 static struct platform_driver fsl_asoc_card_driver = {
878 .probe = fsl_asoc_card_probe,
879 .driver = {
880 .name = "fsl-asoc-card",
881 .pm = &snd_soc_pm_ops,
882 .of_match_table = fsl_asoc_card_dt_ids,
883 },
884 };
885 module_platform_driver(fsl_asoc_card_driver);
886
887 MODULE_DESCRIPTION("Freescale Generic ASoC Sound Card driver with ASRC");
888 MODULE_AUTHOR("Nicolin Chen <nicoleotsuka@gmail.com>");
889 MODULE_ALIAS("platform:fsl-asoc-card");
890 MODULE_LICENSE("GPL");
891