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13#include <linux/clk.h>
14#include <linux/gpio.h>
15#include <linux/gpio/consumer.h>
16#include <linux/module.h>
17#include <linux/of.h>
18#include <sound/pcm_params.h>
19#include <sound/soc.h>
20
21#include "i2s.h"
22#include "../codecs/wm5110.h"
23
24
25
26
27
28#define MCLK_RATE 24000000U
29
30#define TM2_DAI_AIF1 0
31#define TM2_DAI_AIF2 1
32
33struct tm2_machine_priv {
34 struct snd_soc_codec *codec;
35 unsigned int sysclk_rate;
36 struct gpio_desc *gpio_mic_bias;
37};
38
39static int tm2_start_sysclk(struct snd_soc_card *card)
40{
41 struct tm2_machine_priv *priv = snd_soc_card_get_drvdata(card);
42 struct snd_soc_codec *codec = priv->codec;
43 int ret;
44
45 ret = snd_soc_codec_set_pll(codec, WM5110_FLL1_REFCLK,
46 ARIZONA_FLL_SRC_MCLK1,
47 MCLK_RATE,
48 priv->sysclk_rate);
49 if (ret < 0) {
50 dev_err(codec->dev, "Failed to set FLL1 source: %d\n", ret);
51 return ret;
52 }
53
54 ret = snd_soc_codec_set_pll(codec, WM5110_FLL1,
55 ARIZONA_FLL_SRC_MCLK1,
56 MCLK_RATE,
57 priv->sysclk_rate);
58 if (ret < 0) {
59 dev_err(codec->dev, "Failed to start FLL1: %d\n", ret);
60 return ret;
61 }
62
63 ret = snd_soc_codec_set_sysclk(codec, ARIZONA_CLK_SYSCLK,
64 ARIZONA_CLK_SRC_FLL1,
65 priv->sysclk_rate,
66 SND_SOC_CLOCK_IN);
67 if (ret < 0) {
68 dev_err(codec->dev, "Failed to set SYSCLK source: %d\n", ret);
69 return ret;
70 }
71
72 return 0;
73}
74
75static int tm2_stop_sysclk(struct snd_soc_card *card)
76{
77 struct tm2_machine_priv *priv = snd_soc_card_get_drvdata(card);
78 struct snd_soc_codec *codec = priv->codec;
79 int ret;
80
81 ret = snd_soc_codec_set_pll(codec, WM5110_FLL1, 0, 0, 0);
82 if (ret < 0) {
83 dev_err(codec->dev, "Failed to stop FLL1: %d\n", ret);
84 return ret;
85 }
86
87 ret = snd_soc_codec_set_sysclk(codec, ARIZONA_CLK_SYSCLK,
88 ARIZONA_CLK_SRC_FLL1, 0, 0);
89 if (ret < 0) {
90 dev_err(codec->dev, "Failed to stop SYSCLK: %d\n", ret);
91 return ret;
92 }
93
94 return 0;
95}
96
97static int tm2_aif1_hw_params(struct snd_pcm_substream *substream,
98 struct snd_pcm_hw_params *params)
99{
100 struct snd_soc_pcm_runtime *rtd = substream->private_data;
101 struct snd_soc_codec *codec = rtd->codec;
102 struct tm2_machine_priv *priv = snd_soc_card_get_drvdata(rtd->card);
103
104 switch (params_rate(params)) {
105 case 4000:
106 case 8000:
107 case 12000:
108 case 16000:
109 case 24000:
110 case 32000:
111 case 48000:
112 case 96000:
113 case 192000:
114
115 priv->sysclk_rate = 147456000U;
116 break;
117 case 11025:
118 case 22050:
119 case 44100:
120 case 88200:
121 case 176400:
122
123 priv->sysclk_rate = 135475200U;
124 break;
125 default:
126 dev_err(codec->dev, "Not supported sample rate: %d\n",
127 params_rate(params));
128 return -EINVAL;
129 }
130
131 return tm2_start_sysclk(rtd->card);
132}
133
134static struct snd_soc_ops tm2_aif1_ops = {
135 .hw_params = tm2_aif1_hw_params,
136};
137
138static int tm2_aif2_hw_params(struct snd_pcm_substream *substream,
139 struct snd_pcm_hw_params *params)
140{
141 struct snd_soc_pcm_runtime *rtd = substream->private_data;
142 struct snd_soc_codec *codec = rtd->codec;
143 unsigned int asyncclk_rate;
144 int ret;
145
146 switch (params_rate(params)) {
147 case 8000:
148 case 12000:
149 case 16000:
150
151 asyncclk_rate = 49152000U;
152 break;
153 case 11025:
154
155 asyncclk_rate = 45158400U;
156 break;
157 default:
158 dev_err(codec->dev, "Not supported sample rate: %d\n",
159 params_rate(params));
160 return -EINVAL;
161 }
162
163 ret = snd_soc_codec_set_pll(codec, WM5110_FLL2_REFCLK,
164 ARIZONA_FLL_SRC_MCLK1,
165 MCLK_RATE,
166 asyncclk_rate);
167 if (ret < 0) {
168 dev_err(codec->dev, "Failed to set FLL2 source: %d\n", ret);
169 return ret;
170 }
171
172 ret = snd_soc_codec_set_pll(codec, WM5110_FLL2,
173 ARIZONA_FLL_SRC_MCLK1,
174 MCLK_RATE,
175 asyncclk_rate);
176 if (ret < 0) {
177 dev_err(codec->dev, "Failed to start FLL2: %d\n", ret);
178 return ret;
179 }
180
181 ret = snd_soc_codec_set_sysclk(codec, ARIZONA_CLK_ASYNCCLK,
182 ARIZONA_CLK_SRC_FLL2,
183 asyncclk_rate,
184 SND_SOC_CLOCK_IN);
185 if (ret < 0) {
186 dev_err(codec->dev, "Failed to set ASYNCCLK source: %d\n", ret);
187 return ret;
188 }
189
190 return 0;
191}
192
193static int tm2_aif2_hw_free(struct snd_pcm_substream *substream)
194{
195 struct snd_soc_pcm_runtime *rtd = substream->private_data;
196 struct snd_soc_codec *codec = rtd->codec;
197 int ret;
198
199
200 ret = snd_soc_codec_set_pll(codec, WM5110_FLL2, ARIZONA_FLL_SRC_MCLK1,
201 0, 0);
202 if (ret < 0)
203 dev_err(codec->dev, "Failed to stop FLL2: %d\n", ret);
204
205 return ret;
206}
207
208static struct snd_soc_ops tm2_aif2_ops = {
209 .hw_params = tm2_aif2_hw_params,
210 .hw_free = tm2_aif2_hw_free,
211};
212
213static int tm2_mic_bias(struct snd_soc_dapm_widget *w,
214 struct snd_kcontrol *kcontrol, int event)
215{
216 struct snd_soc_card *card = w->dapm->card;
217 struct tm2_machine_priv *priv = snd_soc_card_get_drvdata(card);
218
219 switch (event) {
220 case SND_SOC_DAPM_PRE_PMU:
221 gpiod_set_value_cansleep(priv->gpio_mic_bias, 1);
222 break;
223 case SND_SOC_DAPM_POST_PMD:
224 gpiod_set_value_cansleep(priv->gpio_mic_bias, 0);
225 break;
226 }
227
228 return 0;
229}
230
231static int tm2_set_bias_level(struct snd_soc_card *card,
232 struct snd_soc_dapm_context *dapm,
233 enum snd_soc_bias_level level)
234{
235 struct snd_soc_pcm_runtime *rtd;
236
237 rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name);
238
239 if (dapm->dev != rtd->codec_dai->dev)
240 return 0;
241
242 switch (level) {
243 case SND_SOC_BIAS_STANDBY:
244 if (card->dapm.bias_level == SND_SOC_BIAS_OFF)
245 tm2_start_sysclk(card);
246 break;
247 case SND_SOC_BIAS_OFF:
248 tm2_stop_sysclk(card);
249 break;
250 default:
251 break;
252 }
253
254 return 0;
255}
256
257static struct snd_soc_aux_dev tm2_speaker_amp_dev;
258
259static int tm2_late_probe(struct snd_soc_card *card)
260{
261 struct tm2_machine_priv *priv = snd_soc_card_get_drvdata(card);
262 struct snd_soc_dai_link_component dlc = { 0 };
263 unsigned int ch_map[] = { 0, 1 };
264 struct snd_soc_dai *amp_pdm_dai;
265 struct snd_soc_pcm_runtime *rtd;
266 struct snd_soc_dai *aif1_dai;
267 struct snd_soc_dai *aif2_dai;
268 int ret;
269
270 rtd = snd_soc_get_pcm_runtime(card, card->dai_link[TM2_DAI_AIF1].name);
271 aif1_dai = rtd->codec_dai;
272 priv->codec = rtd->codec;
273
274 ret = snd_soc_dai_set_sysclk(aif1_dai, ARIZONA_CLK_SYSCLK, 0, 0);
275 if (ret < 0) {
276 dev_err(aif1_dai->dev, "Failed to set SYSCLK: %d\n", ret);
277 return ret;
278 }
279
280 rtd = snd_soc_get_pcm_runtime(card, card->dai_link[TM2_DAI_AIF2].name);
281 aif2_dai = rtd->codec_dai;
282
283 ret = snd_soc_dai_set_sysclk(aif2_dai, ARIZONA_CLK_ASYNCCLK, 0, 0);
284 if (ret < 0) {
285 dev_err(aif2_dai->dev, "Failed to set ASYNCCLK: %d\n", ret);
286 return ret;
287 }
288
289 dlc.of_node = tm2_speaker_amp_dev.codec_of_node;
290 amp_pdm_dai = snd_soc_find_dai(&dlc);
291 if (!amp_pdm_dai)
292 return -ENODEV;
293
294
295 ret = snd_soc_dai_set_channel_map(amp_pdm_dai, ARRAY_SIZE(ch_map),
296 ch_map, 0, NULL);
297 if (ret < 0)
298 return ret;
299
300 ret = snd_soc_dai_set_tdm_slot(amp_pdm_dai, 0x3, 0x0, 2, 16);
301 if (ret < 0)
302 return ret;
303
304 return 0;
305}
306
307static const struct snd_kcontrol_new tm2_controls[] = {
308 SOC_DAPM_PIN_SWITCH("HP"),
309 SOC_DAPM_PIN_SWITCH("SPK"),
310 SOC_DAPM_PIN_SWITCH("RCV"),
311 SOC_DAPM_PIN_SWITCH("VPS"),
312 SOC_DAPM_PIN_SWITCH("HDMI"),
313
314 SOC_DAPM_PIN_SWITCH("Main Mic"),
315 SOC_DAPM_PIN_SWITCH("Sub Mic"),
316 SOC_DAPM_PIN_SWITCH("Third Mic"),
317
318 SOC_DAPM_PIN_SWITCH("Headset Mic"),
319};
320
321static const struct snd_soc_dapm_widget tm2_dapm_widgets[] = {
322 SND_SOC_DAPM_HP("HP", NULL),
323 SND_SOC_DAPM_SPK("SPK", NULL),
324 SND_SOC_DAPM_SPK("RCV", NULL),
325 SND_SOC_DAPM_LINE("VPS", NULL),
326 SND_SOC_DAPM_LINE("HDMI", NULL),
327
328 SND_SOC_DAPM_MIC("Main Mic", tm2_mic_bias),
329 SND_SOC_DAPM_MIC("Sub Mic", NULL),
330 SND_SOC_DAPM_MIC("Third Mic", NULL),
331
332 SND_SOC_DAPM_MIC("Headset Mic", NULL),
333};
334
335static const struct snd_soc_component_driver tm2_component = {
336 .name = "tm2-audio",
337};
338
339static struct snd_soc_dai_driver tm2_ext_dai[] = {
340 {
341 .name = "Voice call",
342 .playback = {
343 .channels_min = 1,
344 .channels_max = 4,
345 .rate_min = 8000,
346 .rate_max = 48000,
347 .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
348 SNDRV_PCM_RATE_48000),
349 .formats = SNDRV_PCM_FMTBIT_S16_LE,
350 },
351 .capture = {
352 .channels_min = 1,
353 .channels_max = 4,
354 .rate_min = 8000,
355 .rate_max = 48000,
356 .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
357 SNDRV_PCM_RATE_48000),
358 .formats = SNDRV_PCM_FMTBIT_S16_LE,
359 },
360 },
361 {
362 .name = "Bluetooth",
363 .playback = {
364 .channels_min = 1,
365 .channels_max = 4,
366 .rate_min = 8000,
367 .rate_max = 16000,
368 .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000),
369 .formats = SNDRV_PCM_FMTBIT_S16_LE,
370 },
371 .capture = {
372 .channels_min = 1,
373 .channels_max = 2,
374 .rate_min = 8000,
375 .rate_max = 16000,
376 .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000),
377 .formats = SNDRV_PCM_FMTBIT_S16_LE,
378 },
379 },
380};
381
382static struct snd_soc_dai_link tm2_dai_links[] = {
383 {
384 .name = "WM5110 AIF1",
385 .stream_name = "HiFi Primary",
386 .codec_dai_name = "wm5110-aif1",
387 .ops = &tm2_aif1_ops,
388 .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
389 SND_SOC_DAIFMT_CBM_CFM,
390 }, {
391 .name = "WM5110 Voice",
392 .stream_name = "Voice call",
393 .codec_dai_name = "wm5110-aif2",
394 .ops = &tm2_aif2_ops,
395 .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
396 SND_SOC_DAIFMT_CBM_CFM,
397 .ignore_suspend = 1,
398 }, {
399 .name = "WM5110 BT",
400 .stream_name = "Bluetooth",
401 .codec_dai_name = "wm5110-aif3",
402 .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
403 SND_SOC_DAIFMT_CBM_CFM,
404 .ignore_suspend = 1,
405 }
406};
407
408static struct snd_soc_card tm2_card = {
409 .owner = THIS_MODULE,
410
411 .dai_link = tm2_dai_links,
412 .num_links = ARRAY_SIZE(tm2_dai_links),
413 .controls = tm2_controls,
414 .num_controls = ARRAY_SIZE(tm2_controls),
415 .dapm_widgets = tm2_dapm_widgets,
416 .num_dapm_widgets = ARRAY_SIZE(tm2_dapm_widgets),
417 .aux_dev = &tm2_speaker_amp_dev,
418 .num_aux_devs = 1,
419
420 .late_probe = tm2_late_probe,
421 .set_bias_level = tm2_set_bias_level,
422};
423
424static int tm2_probe(struct platform_device *pdev)
425{
426 struct device *dev = &pdev->dev;
427 struct snd_soc_card *card = &tm2_card;
428 struct tm2_machine_priv *priv;
429 struct device_node *cpu_dai_node, *codec_dai_node;
430 int ret, i;
431
432 priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
433 if (!priv)
434 return -ENOMEM;
435
436 snd_soc_card_set_drvdata(card, priv);
437 card->dev = dev;
438
439 priv->gpio_mic_bias = devm_gpiod_get(dev, "mic-bias",
440 GPIOF_OUT_INIT_LOW);
441 if (IS_ERR(priv->gpio_mic_bias)) {
442 dev_err(dev, "Failed to get mic bias gpio\n");
443 return PTR_ERR(priv->gpio_mic_bias);
444 }
445
446 ret = snd_soc_of_parse_card_name(card, "model");
447 if (ret < 0) {
448 dev_err(dev, "Card name is not specified\n");
449 return ret;
450 }
451
452 ret = snd_soc_of_parse_audio_routing(card, "samsung,audio-routing");
453 if (ret < 0) {
454 dev_err(dev, "Audio routing is not specified or invalid\n");
455 return ret;
456 }
457
458 card->aux_dev[0].codec_of_node = of_parse_phandle(dev->of_node,
459 "audio-amplifier", 0);
460 if (!card->aux_dev[0].codec_of_node) {
461 dev_err(dev, "audio-amplifier property invalid or missing\n");
462 return -EINVAL;
463 }
464
465 cpu_dai_node = of_parse_phandle(dev->of_node, "i2s-controller", 0);
466 if (!cpu_dai_node) {
467 dev_err(dev, "i2s-controllers property invalid or missing\n");
468 ret = -EINVAL;
469 goto amp_node_put;
470 }
471
472 codec_dai_node = of_parse_phandle(dev->of_node, "audio-codec", 0);
473 if (!codec_dai_node) {
474 dev_err(dev, "audio-codec property invalid or missing\n");
475 ret = -EINVAL;
476 goto cpu_dai_node_put;
477 }
478
479 for (i = 0; i < card->num_links; i++) {
480 card->dai_link[i].cpu_dai_name = NULL;
481 card->dai_link[i].cpu_name = NULL;
482 card->dai_link[i].platform_name = NULL;
483 card->dai_link[i].codec_of_node = codec_dai_node;
484 card->dai_link[i].cpu_of_node = cpu_dai_node;
485 card->dai_link[i].platform_of_node = cpu_dai_node;
486 }
487
488 ret = devm_snd_soc_register_component(dev, &tm2_component,
489 tm2_ext_dai, ARRAY_SIZE(tm2_ext_dai));
490 if (ret < 0) {
491 dev_err(dev, "Failed to register component: %d\n", ret);
492 goto codec_dai_node_put;
493 }
494
495 ret = devm_snd_soc_register_card(dev, card);
496 if (ret < 0) {
497 dev_err(dev, "Failed to register card: %d\n", ret);
498 goto codec_dai_node_put;
499 }
500
501codec_dai_node_put:
502 of_node_put(codec_dai_node);
503cpu_dai_node_put:
504 of_node_put(cpu_dai_node);
505amp_node_put:
506 of_node_put(card->aux_dev[0].codec_of_node);
507 return ret;
508}
509
510static int tm2_pm_prepare(struct device *dev)
511{
512 struct snd_soc_card *card = dev_get_drvdata(dev);
513
514 return tm2_stop_sysclk(card);
515}
516
517static void tm2_pm_complete(struct device *dev)
518{
519 struct snd_soc_card *card = dev_get_drvdata(dev);
520
521 tm2_start_sysclk(card);
522}
523
524static const struct dev_pm_ops tm2_pm_ops = {
525 .prepare = tm2_pm_prepare,
526 .suspend = snd_soc_suspend,
527 .resume = snd_soc_resume,
528 .complete = tm2_pm_complete,
529 .freeze = snd_soc_suspend,
530 .thaw = snd_soc_resume,
531 .poweroff = snd_soc_poweroff,
532 .restore = snd_soc_resume,
533};
534
535static const struct of_device_id tm2_of_match[] = {
536 { .compatible = "samsung,tm2-audio" },
537 { },
538};
539MODULE_DEVICE_TABLE(of, tm2_of_match);
540
541static struct platform_driver tm2_driver = {
542 .driver = {
543 .name = "tm2-audio",
544 .pm = &tm2_pm_ops,
545 .of_match_table = tm2_of_match,
546 },
547 .probe = tm2_probe,
548};
549module_platform_driver(tm2_driver);
550
551MODULE_AUTHOR("Inha Song <ideal.song@samsung.com>");
552MODULE_DESCRIPTION("ALSA SoC Exynos TM2 Audio Support");
553MODULE_LICENSE("GPL v2");
554
