xref: /netbsd-src/sys/dev/fdt/ausoc.c (revision 181254a7b1bdde6873432bffef2d2decc4b5c22f)
1 /* $NetBSD: ausoc.c,v 1.5 2019/11/16 12:47:47 jmcneill 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 1.5 2019/11/16 12:47:47 jmcneill 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 	const audio_params_t *params = (setmode & AUMODE_PLAY) != 0 ?
125 	    play : rec;
126 	int error;
127 
128 	if (link->link_mclk_fs) {
129 		const u_int rate = params->sample_rate * link->link_mclk_fs;
130 		error = audio_dai_set_sysclk(link->link_codec, rate,
131 		    AUDIO_DAI_CLOCK_IN);
132 		if (error)
133 			return error;
134 		error = audio_dai_set_sysclk(link->link_cpu, rate,
135 		    AUDIO_DAI_CLOCK_OUT);
136 		if (error)
137 			return error;
138 	}
139 
140 	error = audio_dai_mi_set_format(link->link_cpu, setmode,
141 	    play, rec, pfil, rfil);
142 	if (error)
143 		return error;
144 
145 	return audio_dai_mi_set_format(link->link_codec, setmode,
146 	    play, rec, pfil, rfil);
147 }
148 
149 static int
150 ausoc_set_port(void *priv, mixer_ctrl_t *mc)
151 {
152 	struct ausoc_link * const link = priv;
153 
154 	return audio_dai_set_port(link->link_codec, mc);
155 }
156 
157 static int
158 ausoc_get_port(void *priv, mixer_ctrl_t *mc)
159 {
160 	struct ausoc_link * const link = priv;
161 
162 	return audio_dai_get_port(link->link_codec, mc);
163 }
164 
165 static int
166 ausoc_query_devinfo(void *priv, mixer_devinfo_t *di)
167 {
168 	struct ausoc_link * const link = priv;
169 
170 	return audio_dai_query_devinfo(link->link_codec, di);
171 }
172 
173 static void *
174 ausoc_allocm(void *priv, int dir, size_t size)
175 {
176 	struct ausoc_link * const link = priv;
177 
178 	return audio_dai_allocm(link->link_cpu, dir, size);
179 }
180 
181 static void
182 ausoc_freem(void *priv, void *addr, size_t size)
183 {
184 	struct ausoc_link * const link = priv;
185 
186 	return audio_dai_freem(link->link_cpu, addr, size);
187 }
188 
189 static int
190 ausoc_getdev(void *priv, struct audio_device *adev)
191 {
192 	struct ausoc_link * const link = priv;
193 
194 	/* Defaults */
195 	snprintf(adev->name, sizeof(adev->name), "%s", link->link_name);
196 	snprintf(adev->version, sizeof(adev->version), "");
197 	snprintf(adev->config, sizeof(adev->config), "ausoc");
198 
199 	/* Codec can override */
200 	(void)audio_dai_getdev(link->link_codec, adev);
201 
202 	return 0;
203 }
204 
205 static int
206 ausoc_get_props(void *priv)
207 {
208 	struct ausoc_link * const link = priv;
209 
210 	return audio_dai_get_props(link->link_cpu);
211 }
212 
213 static int
214 ausoc_round_blocksize(void *priv, int bs, int mode,
215     const audio_params_t *params)
216 {
217 	struct ausoc_link * const link = priv;
218 
219 	return audio_dai_round_blocksize(link->link_cpu, bs, mode, params);
220 }
221 
222 static size_t
223 ausoc_round_buffersize(void *priv, int dir, size_t bufsize)
224 {
225 	struct ausoc_link * const link = priv;
226 
227 	return audio_dai_round_buffersize(link->link_cpu, dir, bufsize);
228 }
229 
230 static int
231 ausoc_halt_output(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_PLAY);
238 
239 	audio_dai_halt(link->link_codec, AUMODE_PLAY);
240 
241 	return audio_dai_halt(link->link_cpu, AUMODE_PLAY);
242 }
243 
244 static int
245 ausoc_halt_input(void *priv)
246 {
247 	struct ausoc_link * const link = priv;
248 	u_int n;
249 
250 	for (n = 0; n < link->link_naux; n++)
251 		audio_dai_halt(link->link_aux[n], AUMODE_RECORD);
252 
253 	audio_dai_halt(link->link_codec, AUMODE_RECORD);
254 
255 	return audio_dai_halt(link->link_cpu, AUMODE_RECORD);
256 }
257 
258 static int
259 ausoc_trigger_output(void *priv, void *start, void *end, int blksize,
260     void (*intr)(void *), void *intrarg, const audio_params_t *params)
261 {
262 	struct ausoc_link * const link = priv;
263 	int error;
264 	u_int n;
265 
266 	for (n = 0; n < link->link_naux; n++) {
267 		error = audio_dai_trigger(link->link_aux[n], start, end,
268 		    blksize, intr, intrarg, params, AUMODE_PLAY);
269 		if (error)
270 			goto failed;
271 	}
272 	error = audio_dai_trigger(link->link_codec, start, end, blksize,
273 	    intr, intrarg, params, AUMODE_PLAY);
274 	if (error)
275 		goto failed;
276 
277 	return audio_dai_trigger(link->link_cpu, start, end, blksize,
278 	    intr, intrarg, params, AUMODE_PLAY);
279 
280 failed:
281 	ausoc_halt_output(priv);
282 	return error;
283 }
284 
285 static int
286 ausoc_trigger_input(void *priv, void *start, void *end, int blksize,
287     void (*intr)(void *), void *intrarg, const audio_params_t *params)
288 {
289 	struct ausoc_link * const link = priv;
290 	int error;
291 	u_int n;
292 
293 	for (n = 0; n < link->link_naux; n++) {
294 		error = audio_dai_trigger(link->link_aux[n], start, end,
295 		    blksize, intr, intrarg, params, AUMODE_RECORD);
296 		if (error)
297 			goto failed;
298 	}
299 	error = audio_dai_trigger(link->link_codec, start, end, blksize,
300 	    intr, intrarg, params, AUMODE_RECORD);
301 	if (error)
302 		goto failed;
303 
304 	return audio_dai_trigger(link->link_cpu, start, end, blksize,
305 	    intr, intrarg, params, AUMODE_RECORD);
306 
307 failed:
308 	ausoc_halt_input(priv);
309 	return error;
310 }
311 
312 static void
313 ausoc_get_locks(void *priv, kmutex_t **intr, kmutex_t **thread)
314 {
315 	struct ausoc_link * const link = priv;
316 
317 	return audio_dai_get_locks(link->link_cpu, intr, thread);
318 }
319 
320 static const struct audio_hw_if ausoc_hw_if = {
321 	.open = ausoc_open,
322 	.close = ausoc_close,
323 	.query_format = ausoc_query_format,
324 	.set_format = ausoc_set_format,
325 	.allocm = ausoc_allocm,
326 	.freem = ausoc_freem,
327 	.getdev = ausoc_getdev,
328 	.set_port = ausoc_set_port,
329 	.get_port = ausoc_get_port,
330 	.query_devinfo = ausoc_query_devinfo,
331 	.get_props = ausoc_get_props,
332 	.round_blocksize = ausoc_round_blocksize,
333 	.round_buffersize = ausoc_round_buffersize,
334 	.trigger_output = ausoc_trigger_output,
335 	.trigger_input = ausoc_trigger_input,
336 	.halt_output = ausoc_halt_output,
337 	.halt_input = ausoc_halt_input,
338 	.get_locks = ausoc_get_locks,
339 };
340 
341 static int
342 ausoc_match(device_t parent, cfdata_t cf, void *aux)
343 {
344 	struct fdt_attach_args * const faa = aux;
345 
346 	return of_match_compatible(faa->faa_phandle, compatible);
347 }
348 
349 static struct {
350 	const char *name;
351 	u_int fmt;
352 } ausoc_dai_formats[] = {
353 	{ "i2s",	AUDIO_DAI_FORMAT_I2S },
354 	{ "right_j",	AUDIO_DAI_FORMAT_RJ },
355 	{ "left_j",	AUDIO_DAI_FORMAT_LJ },
356 	{ "dsp_a",	AUDIO_DAI_FORMAT_DSPA },
357 	{ "dsp_b",	AUDIO_DAI_FORMAT_DSPB },
358 	{ "ac97",	AUDIO_DAI_FORMAT_AC97 },
359 	{ "pdm",	AUDIO_DAI_FORMAT_PDM },
360 };
361 
362 static int
363 ausoc_link_format(struct ausoc_softc *sc, struct ausoc_link *link, int phandle,
364     int dai_phandle, bool single_link, u_int *format)
365 {
366 	const char *format_prop = single_link ?
367 	    "simple-audio-card,format" : "format";
368 	const char *frame_master_prop = single_link ?
369 	    "simple-audio-card,frame-master" : "frame-master";
370 	const char *bitclock_master_prop = single_link ?
371 	    "simple-audio-card,bitclock-master" : "bitclock-master";
372 	const char *bitclock_inversion_prop = single_link ?
373 	    "simple-audio-card,bitclock-inversion" : "bitclock-inversion";
374 	const char *frame_inversion_prop = single_link ?
375 	    "simple-audio-card,frame-inversion" : "frame-inversion";
376 
377 	u_int fmt, pol, clk;
378 	const char *s;
379 	u_int n;
380 
381 	s = fdtbus_get_string(phandle, format_prop);
382 	if (s) {
383 		for (n = 0; n < __arraycount(ausoc_dai_formats); n++) {
384 			if (strcmp(s, ausoc_dai_formats[n].name) == 0) {
385 				fmt = ausoc_dai_formats[n].fmt;
386 				break;
387 			}
388 		}
389 		if (n == __arraycount(ausoc_dai_formats))
390 			return EINVAL;
391 	} else {
392 		fmt = AUDIO_DAI_FORMAT_I2S;
393 	}
394 
395 	const bool frame_master =
396 	    dai_phandle == fdtbus_get_phandle(phandle, frame_master_prop);
397 	const bool bitclock_master =
398 	    dai_phandle == fdtbus_get_phandle(phandle, bitclock_master_prop);
399 	if (frame_master) {
400 		clk = bitclock_master ?
401 		    AUDIO_DAI_CLOCK_CBM_CFM : AUDIO_DAI_CLOCK_CBS_CFM;
402 	} else {
403 		clk = bitclock_master ?
404 		    AUDIO_DAI_CLOCK_CBM_CFS : AUDIO_DAI_CLOCK_CBS_CFS;
405 	}
406 
407 	const bool bitclock_inversion = of_hasprop(phandle, bitclock_inversion_prop);
408 	const bool frame_inversion = of_hasprop(phandle, frame_inversion_prop);
409 	if (bitclock_inversion) {
410 		pol = frame_inversion ?
411 		    AUDIO_DAI_POLARITY_IB_IF : AUDIO_DAI_POLARITY_IB_NF;
412 	} else {
413 		pol = frame_inversion ?
414 		    AUDIO_DAI_POLARITY_NB_IF : AUDIO_DAI_POLARITY_NB_NF;
415 	}
416 
417 	*format = __SHIFTIN(fmt, AUDIO_DAI_FORMAT_MASK) |
418 		  __SHIFTIN(pol, AUDIO_DAI_POLARITY_MASK) |
419 		  __SHIFTIN(clk, AUDIO_DAI_CLOCK_MASK);
420 
421 	return 0;
422 }
423 
424 static void
425 ausoc_attach_link(struct ausoc_softc *sc, struct ausoc_link *link,
426     int card_phandle, int link_phandle)
427 {
428 	const bool single_link = card_phandle == link_phandle;
429 	const char *cpu_prop = single_link ?
430 	    "simple-audio-card,cpu" : "cpu";
431 	const char *codec_prop = single_link ?
432 	    "simple-audio-card,codec" : "codec";
433 	const char *mclk_fs_prop = single_link ?
434 	    "simple-audio-card,mclk-fs" : "mclk-fs";
435 	const char *node_name = fdtbus_get_string(link_phandle, "name");
436 	u_int n, format;
437 
438 	const int cpu_phandle = of_find_firstchild_byname(link_phandle, cpu_prop);
439 	if (cpu_phandle <= 0) {
440 		aprint_error_dev(sc->sc_dev, "missing %s prop on %s node\n",
441 		    cpu_prop, node_name);
442 		return;
443 	}
444 
445 	link->link_cpu = fdtbus_dai_acquire(cpu_phandle, "sound-dai");
446 	if (!link->link_cpu) {
447 		aprint_error_dev(sc->sc_dev,
448 		    "couldn't acquire cpu dai on %s node\n", node_name);
449 		return;
450 	}
451 
452 	const int codec_phandle = of_find_firstchild_byname(link_phandle, codec_prop);
453 	if (codec_phandle <= 0) {
454 		aprint_error_dev(sc->sc_dev, "missing %s prop on %s node\n",
455 		    codec_prop, node_name);
456 		return;
457 	}
458 
459 	link->link_codec = fdtbus_dai_acquire(codec_phandle, "sound-dai");
460 	if (!link->link_codec) {
461 		aprint_error_dev(sc->sc_dev,
462 		    "couldn't acquire codec dai on %s node\n", node_name);
463 		return;
464 	}
465 
466 	for (;;) {
467 		if (fdtbus_dai_acquire_index(card_phandle,
468 		    "simple-audio-card,aux-devs", link->link_naux) == NULL)
469 			break;
470 		link->link_naux++;
471 	}
472 	if (link->link_naux) {
473 		link->link_aux = kmem_zalloc(sizeof(audio_dai_tag_t) * link->link_naux, KM_SLEEP);
474 		for (n = 0; n < link->link_naux; n++) {
475 			link->link_aux[n] = fdtbus_dai_acquire_index(card_phandle,
476 			    "simple-audio-card,aux-devs", n);
477 			KASSERT(link->link_aux[n] != NULL);
478 
479 			/* Attach aux devices to codec */
480 			audio_dai_add_device(link->link_codec, link->link_aux[n]);
481 		}
482 	}
483 
484 	of_getprop_uint32(link_phandle, mclk_fs_prop, &link->link_mclk_fs);
485 	if (ausoc_link_format(sc, link, link_phandle, codec_phandle, single_link, &format) != 0) {
486 		aprint_error_dev(sc->sc_dev, "couldn't parse format properties\n");
487 		return;
488 	}
489 	if (audio_dai_set_format(link->link_cpu, format) != 0) {
490 		aprint_error_dev(sc->sc_dev, "couldn't set cpu format\n");
491 		return;
492 	}
493 	if (audio_dai_set_format(link->link_codec, format) != 0) {
494 		aprint_error_dev(sc->sc_dev, "couldn't set codec format\n");
495 		return;
496 	}
497 
498 	aprint_normal_dev(sc->sc_dev, "codec: %s, cpu: %s",
499 	    device_xname(audio_dai_device(link->link_codec)),
500 	    device_xname(audio_dai_device(link->link_cpu)));
501 	for (n = 0; n < link->link_naux; n++) {
502 		if (n == 0)
503 			aprint_normal(", aux:");
504 		aprint_normal(" %s",
505 		    device_xname(audio_dai_device(link->link_aux[n])));
506 	}
507 	aprint_normal("\n");
508 
509 	audio_attach_mi(&ausoc_hw_if, link, sc->sc_dev);
510 }
511 
512 static void
513 ausoc_attach_cb(device_t self)
514 {
515 	struct ausoc_softc * const sc = device_private(self);
516 	const int phandle = sc->sc_phandle;
517 	const char *name;
518 	int child, n;
519 	size_t len;
520 
521 	/*
522 	 * If the root node defines a cpu and codec, there is only one link. For
523 	 * cards with multiple links, there will be simple-audio-card,dai-link
524 	 * child nodes for each one.
525 	 */
526 	if (of_find_firstchild_byname(phandle, "simple-audio-card,cpu") > 0 &&
527 	    of_find_firstchild_byname(phandle, "simple-audio-card,codec") > 0) {
528 		sc->sc_nlink = 1;
529 		sc->sc_link = kmem_zalloc(sizeof(*sc->sc_link), KM_SLEEP);
530 		sc->sc_link[0].link_name = sc->sc_name;
531 		ausoc_attach_link(sc, &sc->sc_link[0], phandle, phandle);
532 	} else {
533 		for (child = OF_child(phandle); child; child = OF_peer(child)) {
534 			name = fdtbus_get_string(child, "name");
535 			len = strlen("simple-audio-card,dai-link");
536 			if (strncmp(name, "simple-audio-card,dai-link", len) != 0)
537 				continue;
538 			sc->sc_nlink++;
539 		}
540 		if (sc->sc_nlink == 0)
541 			return;
542 		sc->sc_link = kmem_zalloc(sizeof(*sc->sc_link) * sc->sc_nlink,
543 		    KM_SLEEP);
544 		for (child = OF_child(phandle), n = 0; child; child = OF_peer(child)) {
545 			name = fdtbus_get_string(child, "name");
546 			len = strlen("simple-audio-card,dai-link");
547 			if (strncmp(name, "simple-audio-card,dai-link", len) != 0)
548 				continue;
549 			sc->sc_link[n].link_name = sc->sc_name;
550 			ausoc_attach_link(sc, &sc->sc_link[n], phandle, child);
551 			n++;
552 		}
553 	}
554 }
555 
556 static void
557 ausoc_attach(device_t parent, device_t self, void *aux)
558 {
559 	struct ausoc_softc * const sc = device_private(self);
560 	struct fdt_attach_args * const faa = aux;
561 	const int phandle = faa->faa_phandle;
562 
563 	sc->sc_dev = self;
564 	sc->sc_phandle = phandle;
565 	sc->sc_name = fdtbus_get_string(phandle, "simple-audio-card,name");
566 	if (!sc->sc_name)
567 		sc->sc_name = "SoC Audio";
568 
569 	aprint_naive("\n");
570 	aprint_normal(": %s\n", sc->sc_name);
571 
572 	/*
573 	 * Defer attachment until all other drivers are ready.
574 	 */
575 	config_defer(self, ausoc_attach_cb);
576 }
577 
578 CFATTACH_DECL_NEW(ausoc, sizeof(struct ausoc_softc),
579     ausoc_match, ausoc_attach, NULL, NULL);
580