xref: /netbsd/sys/dev/fdt/ausoc.c (revision b950503f)
1 /* $NetBSD: ausoc.c,v 1.4 2019/05/08 13:40:18 isaki Exp $ */
2 
3 /*-
4  * Copyright (c) 2018 Jared McNeill <jmcneill@invisible.ca>
5  * All rights reserved.
6  *
7  * Redistribution and use in source and binary forms, with or without
8  * modification, are permitted provided that the following conditions
9  * are met:
10  * 1. Redistributions of source code must retain the above copyright
11  *    notice, this list of conditions and the following disclaimer.
12  * 2. Redistributions in binary form must reproduce the above copyright
13  *    notice, this list of conditions and the following disclaimer in the
14  *    documentation and/or other materials provided with the distribution.
15  *
16  * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
17  * IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
18  * OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED.
19  * IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
20  * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING,
21  * BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
22  * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED
23  * AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
24  * OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
25  * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
26  * SUCH DAMAGE.
27  */
28 
29 #include <sys/cdefs.h>
30 __KERNEL_RCSID(0, "$NetBSD: ausoc.c,v 1.4 2019/05/08 13:40:18 isaki Exp $");
31 
32 #include <sys/param.h>
33 #include <sys/bus.h>
34 #include <sys/cpu.h>
35 #include <sys/device.h>
36 #include <sys/kmem.h>
37 #include <sys/gpio.h>
38 
39 #include <sys/audioio.h>
40 #include <dev/audio/audio_if.h>
41 #include <dev/audio/audio_dai.h>
42 
43 #include <dev/fdt/fdtvar.h>
44 
45 static const char *compatible[] = { "simple-audio-card", NULL };
46 
47 struct ausoc_link {
48 	const char		*link_name;
49 
50 	audio_dai_tag_t		link_cpu;
51 	audio_dai_tag_t		link_codec;
52 	audio_dai_tag_t		*link_aux;
53 	u_int			link_naux;
54 
55 	u_int			link_mclk_fs;
56 
57 	kmutex_t		link_lock;
58 	kmutex_t		link_intr_lock;
59 };
60 
61 struct ausoc_softc {
62 	device_t		sc_dev;
63 	int			sc_phandle;
64 	const char		*sc_name;
65 
66 	struct ausoc_link	*sc_link;
67 	u_int			sc_nlink;
68 };
69 
70 static void
71 ausoc_close(void *priv)
72 {
73 	struct ausoc_link * const link = priv;
74 	u_int aux;
75 
76 	for (aux = 0; aux < link->link_naux; aux++)
77 		audio_dai_close(link->link_aux[aux]);
78 	audio_dai_close(link->link_codec);
79 	audio_dai_close(link->link_cpu);
80 }
81 
82 static int
83 ausoc_open(void *priv, int flags)
84 {
85 	struct ausoc_link * const link = priv;
86 	u_int aux;
87 	int error;
88 
89 	error = audio_dai_open(link->link_cpu, flags);
90 	if (error)
91 		goto failed;
92 
93 	error = audio_dai_open(link->link_codec, flags);
94 	if (error)
95 		goto failed;
96 
97 	for (aux = 0; aux < link->link_naux; aux++) {
98 		error = audio_dai_open(link->link_aux[aux], flags);
99 		if (error)
100 			goto failed;
101 	}
102 
103 	return 0;
104 
105 failed:
106 	ausoc_close(priv);
107 	return error;
108 }
109 
110 static int
111 ausoc_query_format(void *priv, audio_format_query_t *afp)
112 {
113 	struct ausoc_link * const link = priv;
114 
115 	return audio_dai_query_format(link->link_cpu, afp);
116 }
117 
118 static int
119 ausoc_set_format(void *priv, int setmode,
120     const audio_params_t *play, const audio_params_t *rec,
121     audio_filter_reg_t *pfil, audio_filter_reg_t *rfil)
122 {
123 	struct ausoc_link * const link = priv;
124 	int error;
125 
126 	error = audio_dai_mi_set_format(link->link_cpu, setmode,
127 	    play, rec, pfil, rfil);
128 	if (error)
129 		return error;
130 
131 	return audio_dai_mi_set_format(link->link_codec, setmode,
132 	    play, rec, pfil, rfil);
133 }
134 
135 static int
136 ausoc_set_port(void *priv, mixer_ctrl_t *mc)
137 {
138 	struct ausoc_link * const link = priv;
139 
140 	return audio_dai_set_port(link->link_codec, mc);
141 }
142 
143 static int
144 ausoc_get_port(void *priv, mixer_ctrl_t *mc)
145 {
146 	struct ausoc_link * const link = priv;
147 
148 	return audio_dai_get_port(link->link_codec, mc);
149 }
150 
151 static int
152 ausoc_query_devinfo(void *priv, mixer_devinfo_t *di)
153 {
154 	struct ausoc_link * const link = priv;
155 
156 	return audio_dai_query_devinfo(link->link_codec, di);
157 }
158 
159 static void *
160 ausoc_allocm(void *priv, int dir, size_t size)
161 {
162 	struct ausoc_link * const link = priv;
163 
164 	return audio_dai_allocm(link->link_cpu, dir, size);
165 }
166 
167 static void
168 ausoc_freem(void *priv, void *addr, size_t size)
169 {
170 	struct ausoc_link * const link = priv;
171 
172 	return audio_dai_freem(link->link_cpu, addr, size);
173 }
174 
175 static int
176 ausoc_getdev(void *priv, struct audio_device *adev)
177 {
178 	struct ausoc_link * const link = priv;
179 
180 	/* Defaults */
181 	snprintf(adev->name, sizeof(adev->name), "%s", link->link_name);
182 	snprintf(adev->version, sizeof(adev->version), "");
183 	snprintf(adev->config, sizeof(adev->config), "ausoc");
184 
185 	/* Codec can override */
186 	(void)audio_dai_getdev(link->link_codec, adev);
187 
188 	return 0;
189 }
190 
191 static int
192 ausoc_get_props(void *priv)
193 {
194 	struct ausoc_link * const link = priv;
195 
196 	return audio_dai_get_props(link->link_cpu);
197 }
198 
199 static int
200 ausoc_round_blocksize(void *priv, int bs, int mode,
201     const audio_params_t *params)
202 {
203 	struct ausoc_link * const link = priv;
204 
205 	return audio_dai_round_blocksize(link->link_cpu, bs, mode, params);
206 }
207 
208 static size_t
209 ausoc_round_buffersize(void *priv, int dir, size_t bufsize)
210 {
211 	struct ausoc_link * const link = priv;
212 
213 	return audio_dai_round_buffersize(link->link_cpu, dir, bufsize);
214 }
215 
216 static int
217 ausoc_halt_output(void *priv)
218 {
219 	struct ausoc_link * const link = priv;
220 	u_int n;
221 
222 	for (n = 0; n < link->link_naux; n++)
223 		audio_dai_halt(link->link_aux[n], AUMODE_PLAY);
224 
225 	audio_dai_halt(link->link_codec, AUMODE_PLAY);
226 
227 	return audio_dai_halt(link->link_cpu, AUMODE_PLAY);
228 }
229 
230 static int
231 ausoc_halt_input(void *priv)
232 {
233 	struct ausoc_link * const link = priv;
234 	u_int n;
235 
236 	for (n = 0; n < link->link_naux; n++)
237 		audio_dai_halt(link->link_aux[n], AUMODE_RECORD);
238 
239 	audio_dai_halt(link->link_codec, AUMODE_RECORD);
240 
241 	return audio_dai_halt(link->link_cpu, AUMODE_RECORD);
242 }
243 
244 static int
245 ausoc_trigger_output(void *priv, void *start, void *end, int blksize,
246     void (*intr)(void *), void *intrarg, const audio_params_t *params)
247 {
248 	struct ausoc_link * const link = priv;
249 	u_int n, rate;
250 	int error;
251 
252 	if (link->link_mclk_fs) {
253 		rate = params->sample_rate * link->link_mclk_fs;
254 		error = audio_dai_set_sysclk(link->link_codec, rate,
255 		    AUDIO_DAI_CLOCK_IN);
256 		if (error)
257 			goto failed;
258 		error = audio_dai_set_sysclk(link->link_cpu, rate,
259 		    AUDIO_DAI_CLOCK_OUT);
260 		if (error)
261 			goto failed;
262 	}
263 
264 	for (n = 0; n < link->link_naux; n++) {
265 		error = audio_dai_trigger(link->link_aux[n], start, end,
266 		    blksize, intr, intrarg, params, AUMODE_PLAY);
267 		if (error)
268 			goto failed;
269 	}
270 	error = audio_dai_trigger(link->link_codec, start, end, blksize,
271 	    intr, intrarg, params, AUMODE_PLAY);
272 	if (error)
273 		goto failed;
274 
275 	return audio_dai_trigger(link->link_cpu, start, end, blksize,
276 	    intr, intrarg, params, AUMODE_PLAY);
277 
278 failed:
279 	ausoc_halt_output(priv);
280 	return error;
281 }
282 
283 static int
284 ausoc_trigger_input(void *priv, void *start, void *end, int blksize,
285     void (*intr)(void *), void *intrarg, const audio_params_t *params)
286 {
287 	struct ausoc_link * const link = priv;
288 	u_int n, rate;
289 	int error;
290 
291 	if (link->link_mclk_fs) {
292 		rate = params->sample_rate * link->link_mclk_fs;
293 		error = audio_dai_set_sysclk(link->link_codec, rate,
294 		    AUDIO_DAI_CLOCK_IN);
295 		if (error)
296 			goto failed;
297 		error = audio_dai_set_sysclk(link->link_cpu, rate,
298 		    AUDIO_DAI_CLOCK_OUT);
299 		if (error)
300 			goto failed;
301 	}
302 
303 	for (n = 0; n < link->link_naux; n++) {
304 		error = audio_dai_trigger(link->link_aux[n], start, end,
305 		    blksize, intr, intrarg, params, AUMODE_RECORD);
306 		if (error)
307 			goto failed;
308 	}
309 	error = audio_dai_trigger(link->link_codec, start, end, blksize,
310 	    intr, intrarg, params, AUMODE_RECORD);
311 	if (error)
312 		goto failed;
313 
314 	return audio_dai_trigger(link->link_cpu, start, end, blksize,
315 	    intr, intrarg, params, AUMODE_RECORD);
316 
317 failed:
318 	ausoc_halt_input(priv);
319 	return error;
320 }
321 
322 static void
323 ausoc_get_locks(void *priv, kmutex_t **intr, kmutex_t **thread)
324 {
325 	struct ausoc_link * const link = priv;
326 
327 	return audio_dai_get_locks(link->link_cpu, intr, thread);
328 }
329 
330 static const struct audio_hw_if ausoc_hw_if = {
331 	.open = ausoc_open,
332 	.close = ausoc_close,
333 	.query_format = ausoc_query_format,
334 	.set_format = ausoc_set_format,
335 	.allocm = ausoc_allocm,
336 	.freem = ausoc_freem,
337 	.getdev = ausoc_getdev,
338 	.set_port = ausoc_set_port,
339 	.get_port = ausoc_get_port,
340 	.query_devinfo = ausoc_query_devinfo,
341 	.get_props = ausoc_get_props,
342 	.round_blocksize = ausoc_round_blocksize,
343 	.round_buffersize = ausoc_round_buffersize,
344 	.trigger_output = ausoc_trigger_output,
345 	.trigger_input = ausoc_trigger_input,
346 	.halt_output = ausoc_halt_output,
347 	.halt_input = ausoc_halt_input,
348 	.get_locks = ausoc_get_locks,
349 };
350 
351 static int
352 ausoc_match(device_t parent, cfdata_t cf, void *aux)
353 {
354 	struct fdt_attach_args * const faa = aux;
355 
356 	return of_match_compatible(faa->faa_phandle, compatible);
357 }
358 
359 static struct {
360 	const char *name;
361 	u_int fmt;
362 } ausoc_dai_formats[] = {
363 	{ "i2s",	AUDIO_DAI_FORMAT_I2S },
364 	{ "right_j",	AUDIO_DAI_FORMAT_RJ },
365 	{ "left_j",	AUDIO_DAI_FORMAT_LJ },
366 	{ "dsp_a",	AUDIO_DAI_FORMAT_DSPA },
367 	{ "dsp_b",	AUDIO_DAI_FORMAT_DSPB },
368 	{ "ac97",	AUDIO_DAI_FORMAT_AC97 },
369 	{ "pdm",	AUDIO_DAI_FORMAT_PDM },
370 };
371 
372 static int
373 ausoc_link_format(struct ausoc_softc *sc, struct ausoc_link *link, int phandle,
374     int dai_phandle, bool single_link, u_int *format)
375 {
376 	const char *format_prop = single_link ?
377 	    "simple-audio-card,format" : "format";
378 	const char *frame_master_prop = single_link ?
379 	    "simple-audio-card,frame-master" : "frame-master";
380 	const char *bitclock_master_prop = single_link ?
381 	    "simple-audio-card,bitclock-master" : "bitclock-master";
382 	const char *bitclock_inversion_prop = single_link ?
383 	    "simple-audio-card,bitclock-inversion" : "bitclock-inversion";
384 	const char *frame_inversion_prop = single_link ?
385 	    "simple-audio-card,frame-inversion" : "frame-inversion";
386 
387 	u_int fmt, pol, clk;
388 	const char *s;
389 	u_int n;
390 
391 	s = fdtbus_get_string(phandle, format_prop);
392 	if (s) {
393 		for (n = 0; n < __arraycount(ausoc_dai_formats); n++) {
394 			if (strcmp(s, ausoc_dai_formats[n].name) == 0) {
395 				fmt = ausoc_dai_formats[n].fmt;
396 				break;
397 			}
398 		}
399 		if (n == __arraycount(ausoc_dai_formats))
400 			return EINVAL;
401 	} else {
402 		fmt = AUDIO_DAI_FORMAT_I2S;
403 	}
404 
405 	const bool frame_master =
406 	    dai_phandle == fdtbus_get_phandle(phandle, frame_master_prop);
407 	const bool bitclock_master =
408 	    dai_phandle == fdtbus_get_phandle(phandle, bitclock_master_prop);
409 	if (frame_master) {
410 		clk = bitclock_master ?
411 		    AUDIO_DAI_CLOCK_CBM_CFM : AUDIO_DAI_CLOCK_CBS_CFM;
412 	} else {
413 		clk = bitclock_master ?
414 		    AUDIO_DAI_CLOCK_CBM_CFS : AUDIO_DAI_CLOCK_CBS_CFS;
415 	}
416 
417 	const bool bitclock_inversion = of_hasprop(phandle, bitclock_inversion_prop);
418 	const bool frame_inversion = of_hasprop(phandle, frame_inversion_prop);
419 	if (bitclock_inversion) {
420 		pol = frame_inversion ?
421 		    AUDIO_DAI_POLARITY_IB_IF : AUDIO_DAI_POLARITY_IB_NF;
422 	} else {
423 		pol = frame_inversion ?
424 		    AUDIO_DAI_POLARITY_NB_IF : AUDIO_DAI_POLARITY_NB_NF;
425 	}
426 
427 	*format = __SHIFTIN(fmt, AUDIO_DAI_FORMAT_MASK) |
428 		  __SHIFTIN(pol, AUDIO_DAI_POLARITY_MASK) |
429 		  __SHIFTIN(clk, AUDIO_DAI_CLOCK_MASK);
430 
431 	return 0;
432 }
433 
434 static void
435 ausoc_attach_link(struct ausoc_softc *sc, struct ausoc_link *link,
436     int card_phandle, int link_phandle)
437 {
438 	const bool single_link = card_phandle == link_phandle;
439 	const char *cpu_prop = single_link ?
440 	    "simple-audio-card,cpu" : "cpu";
441 	const char *codec_prop = single_link ?
442 	    "simple-audio-card,codec" : "codec";
443 	const char *mclk_fs_prop = single_link ?
444 	    "simple-audio-card,mclk-fs" : "mclk-fs";
445 	const char *node_name = fdtbus_get_string(link_phandle, "name");
446 	u_int n, format;
447 
448 	const int cpu_phandle = of_find_firstchild_byname(link_phandle, cpu_prop);
449 	if (cpu_phandle <= 0) {
450 		aprint_error_dev(sc->sc_dev, "missing %s prop on %s node\n",
451 		    cpu_prop, node_name);
452 		return;
453 	}
454 
455 	link->link_cpu = fdtbus_dai_acquire(cpu_phandle, "sound-dai");
456 	if (!link->link_cpu) {
457 		aprint_error_dev(sc->sc_dev,
458 		    "couldn't acquire cpu dai on %s node\n", node_name);
459 		return;
460 	}
461 
462 	const int codec_phandle = of_find_firstchild_byname(link_phandle, codec_prop);
463 	if (codec_phandle <= 0) {
464 		aprint_error_dev(sc->sc_dev, "missing %s prop on %s node\n",
465 		    codec_prop, node_name);
466 		return;
467 	}
468 
469 	link->link_codec = fdtbus_dai_acquire(codec_phandle, "sound-dai");
470 	if (!link->link_codec) {
471 		aprint_error_dev(sc->sc_dev,
472 		    "couldn't acquire codec dai on %s node\n", node_name);
473 		return;
474 	}
475 
476 	for (;;) {
477 		if (fdtbus_dai_acquire_index(card_phandle,
478 		    "simple-audio-card,aux-devs", link->link_naux) == NULL)
479 			break;
480 		link->link_naux++;
481 	}
482 	if (link->link_naux) {
483 		link->link_aux = kmem_zalloc(sizeof(audio_dai_tag_t) * link->link_naux, KM_SLEEP);
484 		for (n = 0; n < link->link_naux; n++) {
485 			link->link_aux[n] = fdtbus_dai_acquire_index(card_phandle,
486 			    "simple-audio-card,aux-devs", n);
487 			KASSERT(link->link_aux[n] != NULL);
488 
489 			/* Attach aux devices to codec */
490 			audio_dai_add_device(link->link_codec, link->link_aux[n]);
491 		}
492 	}
493 
494 	of_getprop_uint32(link_phandle, mclk_fs_prop, &link->link_mclk_fs);
495 	if (ausoc_link_format(sc, link, link_phandle, codec_phandle, single_link, &format) != 0) {
496 		aprint_error_dev(sc->sc_dev, "couldn't parse format properties\n");
497 		return;
498 	}
499 	if (audio_dai_set_format(link->link_cpu, format) != 0) {
500 		aprint_error_dev(sc->sc_dev, "couldn't set cpu format\n");
501 		return;
502 	}
503 	if (audio_dai_set_format(link->link_codec, format) != 0) {
504 		aprint_error_dev(sc->sc_dev, "couldn't set codec format\n");
505 		return;
506 	}
507 
508 	aprint_normal_dev(sc->sc_dev, "codec: %s, cpu: %s",
509 	    device_xname(audio_dai_device(link->link_codec)),
510 	    device_xname(audio_dai_device(link->link_cpu)));
511 	for (n = 0; n < link->link_naux; n++) {
512 		if (n == 0)
513 			aprint_normal(", aux:");
514 		aprint_normal(" %s",
515 		    device_xname(audio_dai_device(link->link_aux[n])));
516 	}
517 	aprint_normal("\n");
518 
519 	audio_attach_mi(&ausoc_hw_if, link, sc->sc_dev);
520 }
521 
522 static void
523 ausoc_attach_cb(device_t self)
524 {
525 	struct ausoc_softc * const sc = device_private(self);
526 	const int phandle = sc->sc_phandle;
527 	const char *name;
528 	int child, n;
529 	size_t len;
530 
531 	/*
532 	 * If the root node defines a cpu and codec, there is only one link. For
533 	 * cards with multiple links, there will be simple-audio-card,dai-link
534 	 * child nodes for each one.
535 	 */
536 	if (of_find_firstchild_byname(phandle, "simple-audio-card,cpu") > 0 &&
537 	    of_find_firstchild_byname(phandle, "simple-audio-card,codec") > 0) {
538 		sc->sc_nlink = 1;
539 		sc->sc_link = kmem_zalloc(sizeof(*sc->sc_link), KM_SLEEP);
540 		sc->sc_link[0].link_name = sc->sc_name;
541 		ausoc_attach_link(sc, &sc->sc_link[0], phandle, phandle);
542 	} else {
543 		for (child = OF_child(phandle); child; child = OF_peer(child)) {
544 			name = fdtbus_get_string(child, "name");
545 			len = strlen("simple-audio-card,dai-link");
546 			if (strncmp(name, "simple-audio-card,dai-link", len) != 0)
547 				continue;
548 			sc->sc_nlink++;
549 		}
550 		if (sc->sc_nlink == 0)
551 			return;
552 		sc->sc_link = kmem_zalloc(sizeof(*sc->sc_link) * sc->sc_nlink,
553 		    KM_SLEEP);
554 		for (child = OF_child(phandle), n = 0; child; child = OF_peer(child)) {
555 			name = fdtbus_get_string(child, "name");
556 			len = strlen("simple-audio-card,dai-link");
557 			if (strncmp(name, "simple-audio-card,dai-link", len) != 0)
558 				continue;
559 			sc->sc_link[n].link_name = sc->sc_name;
560 			ausoc_attach_link(sc, &sc->sc_link[n], phandle, child);
561 			n++;
562 		}
563 	}
564 }
565 
566 static void
567 ausoc_attach(device_t parent, device_t self, void *aux)
568 {
569 	struct ausoc_softc * const sc = device_private(self);
570 	struct fdt_attach_args * const faa = aux;
571 	const int phandle = faa->faa_phandle;
572 
573 	sc->sc_dev = self;
574 	sc->sc_phandle = phandle;
575 	sc->sc_name = fdtbus_get_string(phandle, "simple-audio-card,name");
576 	if (!sc->sc_name)
577 		sc->sc_name = "SoC Audio";
578 
579 	aprint_naive("\n");
580 	aprint_normal(": %s\n", sc->sc_name);
581 
582 	/*
583 	 * Defer attachment until all other drivers are ready.
584 	 */
585 	config_defer(self, ausoc_attach_cb);
586 }
587 
588 CFATTACH_DECL_NEW(ausoc, sizeof(struct ausoc_softc),
589     ausoc_match, ausoc_attach, NULL, NULL);
590