1 | /* $NetBSD: auconv.c,v 1.25 2011/11/23 23:07:31 jmcneill Exp $ */ |
2 | |
3 | /* |
4 | * Copyright (c) 1996 The NetBSD Foundation, Inc. |
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 | * 3. All advertising materials mentioning features or use of this software |
16 | * must display the following acknowledgement: |
17 | * This product includes software developed by the Computer Systems |
18 | * Engineering Group at Lawrence Berkeley Laboratory. |
19 | * 4. Neither the name of the University nor of the Laboratory may be used |
20 | * to endorse or promote products derived from this software without |
21 | * specific prior written permission. |
22 | * |
23 | * THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND |
24 | * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE |
25 | * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE |
26 | * ARE DISCLAIMED. IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE |
27 | * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL |
28 | * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS |
29 | * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) |
30 | * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT |
31 | * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY |
32 | * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF |
33 | * SUCH DAMAGE. |
34 | * |
35 | */ |
36 | |
37 | #include <sys/cdefs.h> |
38 | __KERNEL_RCSID(0, "$NetBSD: auconv.c,v 1.25 2011/11/23 23:07:31 jmcneill Exp $" ); |
39 | |
40 | #include <sys/types.h> |
41 | #include <sys/audioio.h> |
42 | #include <sys/device.h> |
43 | #include <sys/errno.h> |
44 | #include <sys/malloc.h> |
45 | #include <sys/null.h> |
46 | #include <sys/systm.h> |
47 | #include <dev/audio_if.h> |
48 | #include <dev/auconv.h> |
49 | #include <dev/mulaw.h> |
50 | #include <machine/limits.h> |
51 | #ifndef _KERNEL |
52 | #include <stddef.h> |
53 | #include <stdio.h> |
54 | #include <stdlib.h> |
55 | #include <string.h> |
56 | #include <stdbool.h> |
57 | #endif |
58 | |
59 | #include <aurateconv.h> /* generated by config(8) */ |
60 | #include <mulaw.h> /* generated by config(8) */ |
61 | |
62 | /* #define AUCONV_DEBUG */ |
63 | #ifdef AUCONV_DEBUG |
64 | # define DPRINTF(x) printf x |
65 | #else |
66 | # define DPRINTF(x) |
67 | #endif |
68 | |
69 | #if NAURATECONV > 0 |
70 | static int auconv_rateconv_supportable(u_int, u_int, u_int); |
71 | static int auconv_rateconv_check_channels(const struct audio_format *, int, |
72 | int, const audio_params_t *, |
73 | stream_filter_list_t *); |
74 | static int auconv_rateconv_check_rates(const struct audio_format *, int, |
75 | int, const audio_params_t *, |
76 | audio_params_t *, |
77 | stream_filter_list_t *); |
78 | #endif |
79 | #ifdef AUCONV_DEBUG |
80 | static void auconv_dump_formats(const struct audio_format *, int); |
81 | #endif |
82 | static void auconv_dump_params(const audio_params_t *); |
83 | static int auconv_exact_match(const struct audio_format *, int, int, |
84 | const struct audio_params *); |
85 | static u_int auconv_normalize_encoding(u_int, u_int); |
86 | static int auconv_is_supported_rate(const struct audio_format *, u_int); |
87 | static int auconv_add_encoding(int, int, int, struct audio_encoding_set **, |
88 | int *); |
89 | |
90 | #ifdef _KERNEL |
91 | #define AUCONV_MALLOC(size) malloc(size, M_DEVBUF, M_NOWAIT) |
92 | #define AUCONV_REALLOC(p, size) realloc(p, size, M_DEVBUF, M_NOWAIT) |
93 | #define AUCONV_FREE(p) free(p, M_DEVBUF) |
94 | #else |
95 | #define AUCONV_MALLOC(size) malloc(size) |
96 | #define AUCONV_REALLOC(p, size) realloc(p, size) |
97 | #define AUCONV_FREE(p) free(p) |
98 | #endif |
99 | |
100 | struct audio_encoding_set { |
101 | int size; |
102 | audio_encoding_t items[1]; |
103 | }; |
104 | #define ENCODING_SET_SIZE(n) (offsetof(struct audio_encoding_set, items) \ |
105 | + sizeof(audio_encoding_t) * (n)) |
106 | |
107 | struct conv_table { |
108 | u_int encoding; |
109 | u_int validbits; |
110 | u_int precision; |
111 | stream_filter_factory_t *play_conv; |
112 | stream_filter_factory_t *rec_conv; |
113 | }; |
114 | /* |
115 | * SLINEAR-16 or SLINEAR-24 should precede in a table because |
116 | * aurateconv supports only SLINEAR. |
117 | */ |
118 | static const struct conv_table s8_table[] = { |
119 | {AUDIO_ENCODING_SLINEAR_LE, 16, 16, |
120 | linear8_to_linear16, linear16_to_linear8}, |
121 | {AUDIO_ENCODING_SLINEAR_BE, 16, 16, |
122 | linear8_to_linear16, linear16_to_linear8}, |
123 | {AUDIO_ENCODING_ULINEAR_LE, 8, 8, |
124 | change_sign8, change_sign8}, |
125 | {0, 0, 0, NULL, NULL}}; |
126 | static const struct conv_table u8_table[] = { |
127 | {AUDIO_ENCODING_SLINEAR_LE, 16, 16, |
128 | linear8_to_linear16, linear16_to_linear8}, |
129 | {AUDIO_ENCODING_SLINEAR_BE, 16, 16, |
130 | linear8_to_linear16, linear16_to_linear8}, |
131 | {AUDIO_ENCODING_SLINEAR_LE, 8, 8, |
132 | change_sign8, change_sign8}, |
133 | {AUDIO_ENCODING_ULINEAR_LE, 16, 16, |
134 | linear8_to_linear16, linear16_to_linear8}, |
135 | {AUDIO_ENCODING_ULINEAR_BE, 16, 16, |
136 | linear8_to_linear16, linear16_to_linear8}, |
137 | {0, 0, 0, NULL, NULL}}; |
138 | static const struct conv_table s16le_table[] = { |
139 | {AUDIO_ENCODING_SLINEAR_BE, 16, 16, |
140 | swap_bytes, swap_bytes}, |
141 | {AUDIO_ENCODING_ULINEAR_LE, 16, 16, |
142 | change_sign16, change_sign16}, |
143 | {AUDIO_ENCODING_ULINEAR_BE, 16, 16, |
144 | swap_bytes_change_sign16, swap_bytes_change_sign16}, |
145 | {0, 0, 0, NULL, NULL}}; |
146 | static const struct conv_table s16be_table[] = { |
147 | {AUDIO_ENCODING_SLINEAR_LE, 16, 16, |
148 | swap_bytes, swap_bytes}, |
149 | {AUDIO_ENCODING_ULINEAR_BE, 16, 16, |
150 | change_sign16, change_sign16}, |
151 | {AUDIO_ENCODING_ULINEAR_LE, 16, 16, |
152 | swap_bytes_change_sign16, swap_bytes_change_sign16}, |
153 | {0, 0, 0, NULL, NULL}}; |
154 | static const struct conv_table u16le_table[] = { |
155 | {AUDIO_ENCODING_SLINEAR_LE, 16, 16, |
156 | change_sign16, change_sign16}, |
157 | {AUDIO_ENCODING_ULINEAR_BE, 16, 16, |
158 | swap_bytes, swap_bytes}, |
159 | {AUDIO_ENCODING_SLINEAR_BE, 16, 16, |
160 | swap_bytes_change_sign16, swap_bytes_change_sign16}, |
161 | {0, 0, 0, NULL, NULL}}; |
162 | static const struct conv_table u16be_table[] = { |
163 | {AUDIO_ENCODING_SLINEAR_BE, 16, 16, |
164 | change_sign16, change_sign16}, |
165 | {AUDIO_ENCODING_ULINEAR_LE, 16, 16, |
166 | swap_bytes, swap_bytes}, |
167 | {AUDIO_ENCODING_SLINEAR_LE, 16, 16, |
168 | swap_bytes_change_sign16, swap_bytes_change_sign16}, |
169 | {0, 0, 0, NULL, NULL}}; |
170 | #if NMULAW > 0 |
171 | static const struct conv_table mulaw_table[] = { |
172 | {AUDIO_ENCODING_SLINEAR_LE, 16, 16, |
173 | mulaw_to_linear16, linear16_to_mulaw}, |
174 | {AUDIO_ENCODING_SLINEAR_BE, 16, 16, |
175 | mulaw_to_linear16, linear16_to_mulaw}, |
176 | {AUDIO_ENCODING_ULINEAR_LE, 16, 16, |
177 | mulaw_to_linear16, linear16_to_mulaw}, |
178 | {AUDIO_ENCODING_ULINEAR_BE, 16, 16, |
179 | mulaw_to_linear16, linear16_to_mulaw}, |
180 | {AUDIO_ENCODING_SLINEAR_LE, 8, 8, |
181 | mulaw_to_linear8, linear8_to_mulaw}, |
182 | {AUDIO_ENCODING_ULINEAR_LE, 8, 8, |
183 | mulaw_to_linear8, linear8_to_mulaw}, |
184 | {0, 0, 0, NULL, NULL}}; |
185 | static const struct conv_table alaw_table[] = { |
186 | {AUDIO_ENCODING_SLINEAR_LE, 16, 16, |
187 | alaw_to_linear16, linear16_to_alaw}, |
188 | {AUDIO_ENCODING_SLINEAR_BE, 16, 16, |
189 | alaw_to_linear16, linear16_to_alaw}, |
190 | {AUDIO_ENCODING_ULINEAR_LE, 16, 16, |
191 | alaw_to_linear16, linear16_to_alaw}, |
192 | {AUDIO_ENCODING_ULINEAR_BE, 16, 16, |
193 | alaw_to_linear16, linear16_to_alaw}, |
194 | {AUDIO_ENCODING_SLINEAR_LE, 8, 8, |
195 | alaw_to_linear8, linear8_to_alaw}, |
196 | {AUDIO_ENCODING_ULINEAR_LE, 8, 8, |
197 | alaw_to_linear8, linear8_to_alaw}, |
198 | {0, 0, 0, NULL, NULL}}; |
199 | #endif |
200 | #ifdef AUCONV_DEBUG |
201 | static const char *encoding_dbg_names[] = { |
202 | "none" , AudioEmulaw, AudioEalaw, "pcm16" , |
203 | "pcm8" , AudioEadpcm, AudioEslinear_le, AudioEslinear_be, |
204 | AudioEulinear_le, AudioEulinear_be, |
205 | AudioEslinear, AudioEulinear, |
206 | AudioEmpeg_l1_stream, AudioEmpeg_l1_packets, |
207 | AudioEmpeg_l1_system, AudioEmpeg_l2_stream, |
208 | AudioEmpeg_l2_packets, AudioEmpeg_l2_system, |
209 | AudioEac3 |
210 | }; |
211 | #endif |
212 | |
213 | void |
214 | stream_filter_set_fetcher(stream_filter_t *this, stream_fetcher_t *p) |
215 | { |
216 | this->prev = p; |
217 | } |
218 | |
219 | void |
220 | stream_filter_set_inputbuffer(stream_filter_t *this, audio_stream_t *stream) |
221 | { |
222 | this->src = stream; |
223 | } |
224 | |
225 | stream_filter_t * |
226 | auconv_nocontext_filter_factory( |
227 | int (*fetch_to)(struct audio_softc *, stream_fetcher_t *, |
228 | audio_stream_t *, int)) |
229 | { |
230 | stream_filter_t *this; |
231 | |
232 | this = AUCONV_MALLOC(sizeof(stream_filter_t)); |
233 | if (this == NULL) |
234 | return NULL; |
235 | this->base.fetch_to = fetch_to; |
236 | this->dtor = auconv_nocontext_filter_dtor; |
237 | this->set_fetcher = stream_filter_set_fetcher; |
238 | this->set_inputbuffer = stream_filter_set_inputbuffer; |
239 | this->prev = NULL; |
240 | this->src = NULL; |
241 | return this; |
242 | } |
243 | |
244 | void |
245 | auconv_nocontext_filter_dtor(struct stream_filter *this) |
246 | { |
247 | if (this != NULL) |
248 | AUCONV_FREE(this); |
249 | } |
250 | |
251 | #define DEFINE_FILTER(name) \ |
252 | static int \ |
253 | name##_fetch_to(struct audio_softc *, stream_fetcher_t *, audio_stream_t *, int); \ |
254 | stream_filter_t * \ |
255 | name(struct audio_softc *sc, const audio_params_t *from, \ |
256 | const audio_params_t *to) \ |
257 | { \ |
258 | return auconv_nocontext_filter_factory(name##_fetch_to); \ |
259 | } \ |
260 | static int \ |
261 | name##_fetch_to(struct audio_softc *sc, stream_fetcher_t *self, \ |
262 | audio_stream_t *dst, int max_used) |
263 | |
264 | DEFINE_FILTER(change_sign8) |
265 | { |
266 | stream_filter_t *this; |
267 | int m, err; |
268 | |
269 | this = (stream_filter_t *)self; |
270 | if ((err = this->prev->fetch_to(sc, this->prev, this->src, max_used))) |
271 | return err; |
272 | m = dst->end - dst->start; |
273 | m = min(m, max_used); |
274 | FILTER_LOOP_PROLOGUE(this->src, 1, dst, 1, m) { |
275 | *d = *s ^ 0x80; |
276 | } FILTER_LOOP_EPILOGUE(this->src, dst); |
277 | return 0; |
278 | } |
279 | |
280 | DEFINE_FILTER(change_sign16) |
281 | { |
282 | stream_filter_t *this; |
283 | int m, err, enc; |
284 | |
285 | this = (stream_filter_t *)self; |
286 | max_used = (max_used + 1) & ~1; /* round up to even */ |
287 | if ((err = this->prev->fetch_to(sc, this->prev, this->src, max_used))) |
288 | return err; |
289 | m = (dst->end - dst->start) & ~1; |
290 | m = min(m, max_used); |
291 | enc = dst->param.encoding; |
292 | if (enc == AUDIO_ENCODING_SLINEAR_LE |
293 | || enc == AUDIO_ENCODING_ULINEAR_LE) { |
294 | FILTER_LOOP_PROLOGUE(this->src, 2, dst, 2, m) { |
295 | d[0] = s[0]; |
296 | d[1] = s[1] ^ 0x80; |
297 | } FILTER_LOOP_EPILOGUE(this->src, dst); |
298 | } else { |
299 | FILTER_LOOP_PROLOGUE(this->src, 2, dst, 2, m) { |
300 | d[0] = s[0] ^ 0x80; |
301 | d[1] = s[1]; |
302 | } FILTER_LOOP_EPILOGUE(this->src, dst); |
303 | } |
304 | return 0; |
305 | } |
306 | |
307 | DEFINE_FILTER(swap_bytes) |
308 | { |
309 | stream_filter_t *this; |
310 | int m, err; |
311 | |
312 | this = (stream_filter_t *)self; |
313 | max_used = (max_used + 1) & ~1; /* round up to even */ |
314 | if ((err = this->prev->fetch_to(sc, this->prev, this->src, max_used))) |
315 | return err; |
316 | m = (dst->end - dst->start) & ~1; |
317 | m = min(m, max_used); |
318 | FILTER_LOOP_PROLOGUE(this->src, 2, dst, 2, m) { |
319 | d[0] = s[1]; |
320 | d[1] = s[0]; |
321 | } FILTER_LOOP_EPILOGUE(this->src, dst); |
322 | return 0; |
323 | } |
324 | |
325 | DEFINE_FILTER(swap_bytes_change_sign16) |
326 | { |
327 | stream_filter_t *this; |
328 | int m, err, enc; |
329 | |
330 | this = (stream_filter_t *)self; |
331 | max_used = (max_used + 1) & ~1; /* round up to even */ |
332 | if ((err = this->prev->fetch_to(sc, this->prev, this->src, max_used))) |
333 | return err; |
334 | m = (dst->end - dst->start) & ~1; |
335 | m = min(m, max_used); |
336 | enc = dst->param.encoding; |
337 | if (enc == AUDIO_ENCODING_SLINEAR_LE |
338 | || enc == AUDIO_ENCODING_ULINEAR_LE) { |
339 | FILTER_LOOP_PROLOGUE(this->src, 2, dst, 2, m) { |
340 | d[0] = s[1]; |
341 | d[1] = s[0] ^ 0x80; |
342 | } FILTER_LOOP_EPILOGUE(this->src, dst); |
343 | } else { |
344 | FILTER_LOOP_PROLOGUE(this->src, 2, dst, 2, m) { |
345 | d[0] = s[1] ^ 0x80; |
346 | d[1] = s[0]; |
347 | } FILTER_LOOP_EPILOGUE(this->src, dst); |
348 | } |
349 | return 0; |
350 | } |
351 | |
352 | DEFINE_FILTER(linear8_to_linear16) |
353 | { |
354 | stream_filter_t *this; |
355 | int m, err, enc_dst, enc_src; |
356 | |
357 | this = (stream_filter_t *)self; |
358 | max_used = (max_used + 1) & ~1; /* round up to even */ |
359 | if ((err = this->prev->fetch_to(sc, this->prev, this->src, max_used / 2))) |
360 | return err; |
361 | m = (dst->end - dst->start) & ~1; |
362 | m = min(m, max_used); |
363 | enc_dst = dst->param.encoding; |
364 | enc_src = this->src->param.encoding; |
365 | if ((enc_src == AUDIO_ENCODING_SLINEAR_LE |
366 | && enc_dst == AUDIO_ENCODING_SLINEAR_LE) |
367 | || (enc_src == AUDIO_ENCODING_ULINEAR_LE |
368 | && enc_dst == AUDIO_ENCODING_ULINEAR_LE)) { |
369 | /* |
370 | * slinear8 -> slinear16_le |
371 | * ulinear8 -> ulinear16_le |
372 | */ |
373 | FILTER_LOOP_PROLOGUE(this->src, 1, dst, 2, m) { |
374 | d[0] = 0; |
375 | d[1] = s[0]; |
376 | } FILTER_LOOP_EPILOGUE(this->src, dst); |
377 | } else if ((enc_src == AUDIO_ENCODING_SLINEAR_LE |
378 | && enc_dst == AUDIO_ENCODING_SLINEAR_BE) |
379 | || (enc_src == AUDIO_ENCODING_ULINEAR_LE |
380 | && enc_dst == AUDIO_ENCODING_ULINEAR_BE)) { |
381 | /* |
382 | * slinear8 -> slinear16_be |
383 | * ulinear8 -> ulinear16_be |
384 | */ |
385 | FILTER_LOOP_PROLOGUE(this->src, 1, dst, 2, m) { |
386 | d[0] = s[0]; |
387 | d[1] = 0; |
388 | } FILTER_LOOP_EPILOGUE(this->src, dst); |
389 | } else if ((enc_src == AUDIO_ENCODING_SLINEAR_LE |
390 | && enc_dst == AUDIO_ENCODING_ULINEAR_LE) |
391 | || (enc_src == AUDIO_ENCODING_ULINEAR_LE |
392 | && enc_dst == AUDIO_ENCODING_SLINEAR_LE)) { |
393 | /* |
394 | * slinear8 -> ulinear16_le |
395 | * ulinear8 -> slinear16_le |
396 | */ |
397 | FILTER_LOOP_PROLOGUE(this->src, 1, dst, 2, m) { |
398 | d[0] = 0; |
399 | d[1] = s[0] ^ 0x80; |
400 | } FILTER_LOOP_EPILOGUE(this->src, dst); |
401 | } else { |
402 | /* |
403 | * slinear8 -> ulinear16_be |
404 | * ulinear8 -> slinear16_be |
405 | */ |
406 | FILTER_LOOP_PROLOGUE(this->src, 1, dst, 2, m) { |
407 | d[0] = s[0] ^ 0x80; |
408 | d[1] = 0; |
409 | } FILTER_LOOP_EPILOGUE(this->src, dst); |
410 | } |
411 | return 0; |
412 | } |
413 | |
414 | DEFINE_FILTER(linear16_to_linear8) |
415 | { |
416 | stream_filter_t *this; |
417 | int m, err, enc_src, enc_dst; |
418 | |
419 | this = (stream_filter_t *)self; |
420 | if ((err = this->prev->fetch_to(sc, this->prev, this->src, max_used * 2))) |
421 | return err; |
422 | m = dst->end - dst->start; |
423 | m = min(m, max_used); |
424 | enc_dst = dst->param.encoding; |
425 | enc_src = this->src->param.encoding; |
426 | if ((enc_src == AUDIO_ENCODING_SLINEAR_LE |
427 | && enc_dst == AUDIO_ENCODING_SLINEAR_LE) |
428 | || (enc_src == AUDIO_ENCODING_ULINEAR_LE |
429 | && enc_dst == AUDIO_ENCODING_ULINEAR_LE)) { |
430 | /* |
431 | * slinear16_le -> slinear8 |
432 | * ulinear16_le -> ulinear8 |
433 | */ |
434 | FILTER_LOOP_PROLOGUE(this->src, 2, dst, 1, m) { |
435 | d[0] = s[1]; |
436 | } FILTER_LOOP_EPILOGUE(this->src, dst); |
437 | } else if ((enc_src == AUDIO_ENCODING_SLINEAR_LE |
438 | && enc_dst == AUDIO_ENCODING_ULINEAR_LE) |
439 | || (enc_src == AUDIO_ENCODING_ULINEAR_LE |
440 | && enc_dst == AUDIO_ENCODING_SLINEAR_LE)) { |
441 | /* |
442 | * slinear16_le -> ulinear8 |
443 | * ulinear16_le -> slinear8 |
444 | */ |
445 | FILTER_LOOP_PROLOGUE(this->src, 2, dst, 1, m) { |
446 | d[0] = s[1] ^ 0x80; |
447 | } FILTER_LOOP_EPILOGUE(this->src, dst); |
448 | } else if ((enc_src == AUDIO_ENCODING_SLINEAR_BE |
449 | && enc_dst == AUDIO_ENCODING_SLINEAR_LE) |
450 | || (enc_src == AUDIO_ENCODING_ULINEAR_BE |
451 | && enc_dst == AUDIO_ENCODING_ULINEAR_LE)) { |
452 | /* |
453 | * slinear16_be -> slinear8 |
454 | * ulinear16_be -> ulinear8 |
455 | */ |
456 | FILTER_LOOP_PROLOGUE(this->src, 2, dst, 1, m) { |
457 | d[0] = s[0]; |
458 | } FILTER_LOOP_EPILOGUE(this->src, dst); |
459 | } else { |
460 | /* |
461 | * slinear16_be -> ulinear8 |
462 | * ulinear16_be -> slinear8 |
463 | */ |
464 | FILTER_LOOP_PROLOGUE(this->src, 2, dst, 1, m) { |
465 | d[0] = s[0] ^ 0x80; |
466 | } FILTER_LOOP_EPILOGUE(this->src, dst); |
467 | } |
468 | return 0; |
469 | } |
470 | |
471 | /** |
472 | * Set appropriate parameters in `param,' and return the index in |
473 | * the hardware capability array `formats.' |
474 | * |
475 | * @param formats [IN] An array of formats which a hardware can support. |
476 | * @param nformats [IN] The number of elements of the array. |
477 | * @param mode [IN] Either AUMODE_PLAY or AUMODE_RECORD. |
478 | * @param param [IN] Requested format. param->sw_code may be set. |
479 | * @param rateconv [IN] true if aurateconv may be used. |
480 | * @param list [OUT] stream_filters required for param. |
481 | * @return The index of selected audio_format entry. -1 if the device |
482 | * can not support the specified param. |
483 | */ |
484 | int |
485 | auconv_set_converter(const struct audio_format *formats, int nformats, |
486 | int mode, const audio_params_t *param, int rateconv, |
487 | stream_filter_list_t *list) |
488 | { |
489 | audio_params_t work; |
490 | const struct conv_table *table; |
491 | stream_filter_factory_t *conv; |
492 | int enc; |
493 | int i, j; |
494 | |
495 | #ifdef AUCONV_DEBUG |
496 | DPRINTF(("%s: ENTER rateconv=%d\n" , __func__, rateconv)); |
497 | auconv_dump_formats(formats, nformats); |
498 | #endif |
499 | enc = auconv_normalize_encoding(param->encoding, param->precision); |
500 | |
501 | /* check support by native format */ |
502 | i = auconv_exact_match(formats, nformats, mode, param); |
503 | if (i >= 0) { |
504 | DPRINTF(("%s: LEAVE with %d (exact)\n" , __func__, i)); |
505 | return i; |
506 | } |
507 | |
508 | #if NAURATECONV > 0 |
509 | /* native format with aurateconv */ |
510 | DPRINTF(("%s: native with aurateconv\n" , __func__)); |
511 | if (rateconv |
512 | && auconv_rateconv_supportable(enc, param->precision, |
513 | param->validbits)) { |
514 | i = auconv_rateconv_check_channels(formats, nformats, |
515 | mode, param, list); |
516 | if (i >= 0) { |
517 | DPRINTF(("%s: LEAVE with %d (aurateconv1)\n" , __func__, i)); |
518 | return i; |
519 | } |
520 | } |
521 | #endif |
522 | |
523 | /* check for emulation */ |
524 | DPRINTF(("%s: encoding emulation\n" , __func__)); |
525 | table = NULL; |
526 | switch (enc) { |
527 | case AUDIO_ENCODING_SLINEAR_LE: |
528 | if (param->precision == 8) |
529 | table = s8_table; |
530 | else if (param->precision == 16) |
531 | table = s16le_table; |
532 | break; |
533 | case AUDIO_ENCODING_SLINEAR_BE: |
534 | if (param->precision == 8) |
535 | table = s8_table; |
536 | else if (param->precision == 16) |
537 | table = s16be_table; |
538 | break; |
539 | case AUDIO_ENCODING_ULINEAR_LE: |
540 | if (param->precision == 8) |
541 | table = u8_table; |
542 | else if (param->precision == 16) |
543 | table = u16le_table; |
544 | break; |
545 | case AUDIO_ENCODING_ULINEAR_BE: |
546 | if (param->precision == 8) |
547 | table = u8_table; |
548 | else if (param->precision == 16) |
549 | table = u16be_table; |
550 | break; |
551 | #if NMULAW > 0 |
552 | case AUDIO_ENCODING_ULAW: |
553 | table = mulaw_table; |
554 | break; |
555 | case AUDIO_ENCODING_ALAW: |
556 | table = alaw_table; |
557 | break; |
558 | #endif |
559 | } |
560 | if (table == NULL) { |
561 | DPRINTF(("%s: LEAVE with -1 (no-emultable)\n" , __func__)); |
562 | return -1; |
563 | } |
564 | work = *param; |
565 | for (j = 0; table[j].precision != 0; j++) { |
566 | work.encoding = table[j].encoding; |
567 | work.precision = table[j].precision; |
568 | work.validbits = table[j].validbits; |
569 | i = auconv_exact_match(formats, nformats, mode, &work); |
570 | if (i >= 0) { |
571 | conv = mode == AUMODE_PLAY |
572 | ? table[j].play_conv : table[j].rec_conv; |
573 | list->append(list, conv, &work); |
574 | DPRINTF(("%s: LEAVE with %d (emultable)\n" , __func__, i)); |
575 | return i; |
576 | } |
577 | } |
578 | /* not found */ |
579 | |
580 | #if NAURATECONV > 0 |
581 | /* emulation with aurateconv */ |
582 | DPRINTF(("%s: encoding emulation with aurateconv\n" , __func__)); |
583 | if (!rateconv) { |
584 | DPRINTF(("%s: LEAVE with -1 (no-rateconv)\n" , __func__)); |
585 | return -1; |
586 | } |
587 | work = *param; |
588 | for (j = 0; table[j].precision != 0; j++) { |
589 | if (!auconv_rateconv_supportable(table[j].encoding, |
590 | table[j].precision, |
591 | table[j].validbits)) |
592 | continue; |
593 | work.encoding = table[j].encoding; |
594 | work.precision = table[j].precision; |
595 | work.validbits = table[j].validbits; |
596 | i = auconv_rateconv_check_channels(formats, nformats, |
597 | mode, &work, list); |
598 | if (i >= 0) { |
599 | /* work<=>hw conversion is already registered */ |
600 | conv = mode == AUMODE_PLAY |
601 | ? table[j].play_conv : table[j].rec_conv; |
602 | /* register userland<=>work conversion */ |
603 | list->append(list, conv, &work); |
604 | DPRINTF(("%s: LEAVE with %d (rateconv2)\n" , __func__, i)); |
605 | return i; |
606 | } |
607 | } |
608 | |
609 | #endif |
610 | DPRINTF(("%s: LEAVE with -1 (bottom)\n" , __func__)); |
611 | return -1; |
612 | } |
613 | |
614 | #if NAURATECONV > 0 |
615 | static int |
616 | auconv_rateconv_supportable(u_int encoding, u_int precision, u_int validbits) |
617 | { |
618 | if (encoding != AUDIO_ENCODING_SLINEAR_LE |
619 | && encoding != AUDIO_ENCODING_SLINEAR_BE) |
620 | return false; |
621 | if (precision != 16 && precision != 24 && precision != 32) |
622 | return false; |
623 | if (precision < validbits) |
624 | return false; |
625 | return true; |
626 | } |
627 | |
628 | static int |
629 | auconv_rateconv_check_channels(const struct audio_format *formats, int nformats, |
630 | int mode, const audio_params_t *param, |
631 | stream_filter_list_t *list) |
632 | { |
633 | audio_params_t hw_param; |
634 | int ind, n; |
635 | |
636 | hw_param = *param; |
637 | /* check for the specified number of channels */ |
638 | ind = auconv_rateconv_check_rates(formats, nformats, mode, param, |
639 | &hw_param, list); |
640 | if (ind >= 0) |
641 | return ind; |
642 | |
643 | /* check for larger numbers */ |
644 | for (n = param->channels + 1; n <= AUDIO_MAX_CHANNELS; n++) { |
645 | hw_param.channels = n; |
646 | ind = auconv_rateconv_check_rates(formats, nformats, mode, |
647 | param, &hw_param, list); |
648 | if (ind >= 0) |
649 | return ind; |
650 | } |
651 | |
652 | /* check for stereo:monaural conversion */ |
653 | if (param->channels == 2) { |
654 | hw_param.channels = 1; |
655 | ind = auconv_rateconv_check_rates(formats, nformats, mode, |
656 | param, &hw_param, list); |
657 | if (ind >= 0) |
658 | return ind; |
659 | } |
660 | return -1; |
661 | } |
662 | |
663 | static int |
664 | auconv_rateconv_check_rates(const struct audio_format *formats, int nformats, |
665 | int mode, const audio_params_t *param, |
666 | audio_params_t *hw_param, stream_filter_list_t *list) |
667 | { |
668 | int ind, i, j, enc, f_enc; |
669 | u_int rate, minrate, maxrate, orig_rate; |
670 | |
671 | /* exact match */ |
672 | ind = auconv_exact_match(formats, nformats, mode, hw_param); |
673 | if (ind >= 0) |
674 | goto found; |
675 | |
676 | /* determine min/max of specified encoding/precision/channels */ |
677 | minrate = UINT_MAX; |
678 | maxrate = 0; |
679 | enc = auconv_normalize_encoding(param->encoding, |
680 | param->precision); |
681 | for (i = 0; i < nformats; i++) { |
682 | if (!AUFMT_IS_VALID(&formats[i])) |
683 | continue; |
684 | if ((formats[i].mode & mode) == 0) |
685 | continue; |
686 | f_enc = auconv_normalize_encoding(formats[i].encoding, |
687 | formats[i].precision); |
688 | if (f_enc != enc) |
689 | continue; |
690 | if (formats[i].validbits != hw_param->validbits) |
691 | continue; |
692 | if (formats[i].precision != hw_param->precision) |
693 | continue; |
694 | if (formats[i].channels != hw_param->channels) |
695 | continue; |
696 | if (formats[i].frequency_type == 0) { |
697 | if (formats[i].frequency[0] < minrate) |
698 | minrate = formats[i].frequency[0]; |
699 | if (formats[i].frequency[1] > maxrate) |
700 | maxrate = formats[i].frequency[1]; |
701 | } else { |
702 | for (j = 0; j < formats[i].frequency_type; j++) { |
703 | if (formats[i].frequency[j] < minrate) |
704 | minrate = formats[i].frequency[j]; |
705 | if (formats[i].frequency[j] > maxrate) |
706 | maxrate = formats[i].frequency[j]; |
707 | } |
708 | } |
709 | } |
710 | if (maxrate == 0) |
711 | return -1; |
712 | |
713 | /* try multiples of sample_rate */ |
714 | orig_rate = hw_param->sample_rate; |
715 | for (i = 2; (rate = param->sample_rate * i) <= maxrate; i++) { |
716 | hw_param->sample_rate = rate; |
717 | ind = auconv_exact_match(formats, nformats, mode, hw_param); |
718 | if (ind >= 0) |
719 | goto found; |
720 | } |
721 | |
722 | hw_param->sample_rate = param->sample_rate >= minrate |
723 | ? maxrate : minrate; |
724 | ind = auconv_exact_match(formats, nformats, mode, hw_param); |
725 | if (ind >= 0) |
726 | goto found; |
727 | hw_param->sample_rate = orig_rate; |
728 | return -1; |
729 | |
730 | found: |
731 | list->append(list, aurateconv, hw_param); |
732 | return ind; |
733 | } |
734 | #endif /* NAURATECONV */ |
735 | |
736 | #ifdef AUCONV_DEBUG |
737 | static void |
738 | auconv_dump_formats(const struct audio_format *formats, int nformats) |
739 | { |
740 | const struct audio_format *f; |
741 | int i, j; |
742 | |
743 | for (i = 0; i < nformats; i++) { |
744 | f = &formats[i]; |
745 | printf("[%2d]: mode=" , i); |
746 | if (!AUFMT_IS_VALID(f)) { |
747 | printf("INVALID" ); |
748 | } else if (f->mode == AUMODE_PLAY) { |
749 | printf("PLAY" ); |
750 | } else if (f->mode == AUMODE_RECORD) { |
751 | printf("RECORD" ); |
752 | } else if (f->mode == (AUMODE_PLAY | AUMODE_RECORD)) { |
753 | printf("PLAY|RECORD" ); |
754 | } else { |
755 | printf("0x%x" , f->mode); |
756 | } |
757 | printf(" enc=%s" , encoding_dbg_names[f->encoding]); |
758 | printf(" %u/%ubit" , f->validbits, f->precision); |
759 | printf(" %uch" , f->channels); |
760 | |
761 | printf(" channel_mask=" ); |
762 | if (f->channel_mask == AUFMT_MONAURAL) { |
763 | printf("MONAURAL" ); |
764 | } else if (f->channel_mask == AUFMT_STEREO) { |
765 | printf("STEREO" ); |
766 | } else if (f->channel_mask == AUFMT_SURROUND4) { |
767 | printf("SURROUND4" ); |
768 | } else if (f->channel_mask == AUFMT_DOLBY_5_1) { |
769 | printf("DOLBY5.1" ); |
770 | } else { |
771 | printf("0x%x" , f->channel_mask); |
772 | } |
773 | |
774 | if (f->frequency_type == 0) { |
775 | printf(" %uHz-%uHz" , f->frequency[0], |
776 | f->frequency[1]); |
777 | } else { |
778 | printf(" %uHz" , f->frequency[0]); |
779 | for (j = 1; j < f->frequency_type; j++) |
780 | printf(",%uHz" , f->frequency[j]); |
781 | } |
782 | printf("\n" ); |
783 | } |
784 | } |
785 | |
786 | static void |
787 | auconv_dump_params(const audio_params_t *p) |
788 | { |
789 | printf("enc=%s" , encoding_dbg_names[p->encoding]); |
790 | printf(" %u/%ubit" , p->validbits, p->precision); |
791 | printf(" %uch" , p->channels); |
792 | printf(" %uHz" , p->sample_rate); |
793 | printf("\n" ); |
794 | } |
795 | #else |
796 | static void |
797 | auconv_dump_params(const audio_params_t *p) |
798 | { |
799 | } |
800 | #endif /* AUCONV_DEBUG */ |
801 | |
802 | /** |
803 | * a sub-routine for auconv_set_converter() |
804 | */ |
805 | static int |
806 | auconv_exact_match(const struct audio_format *formats, int nformats, |
807 | int mode, const audio_params_t *param) |
808 | { |
809 | int i, enc, f_enc; |
810 | |
811 | DPRINTF(("%s: ENTER: mode=0x%x target:" , __func__, mode)); |
812 | auconv_dump_params(param); |
813 | enc = auconv_normalize_encoding(param->encoding, |
814 | param->precision); |
815 | DPRINTF(("%s: target normalized: %s\n" , __func__, |
816 | encoding_dbg_names[enc])); |
817 | for (i = 0; i < nformats; i++) { |
818 | if (!AUFMT_IS_VALID(&formats[i])) |
819 | continue; |
820 | if ((formats[i].mode & mode) == 0) |
821 | continue; |
822 | f_enc = auconv_normalize_encoding(formats[i].encoding, |
823 | formats[i].precision); |
824 | DPRINTF(("%s: format[%d] normalized: %s\n" , |
825 | __func__, i, encoding_dbg_names[f_enc])); |
826 | if (f_enc != enc) |
827 | continue; |
828 | /** |
829 | * XXX we need encoding-dependent check. |
830 | * XXX Is to check precision/channels meaningful for |
831 | * MPEG encodings? |
832 | */ |
833 | if (enc != AUDIO_ENCODING_AC3) { |
834 | if (formats[i].validbits != param->validbits) |
835 | continue; |
836 | if (formats[i].precision != param->precision) |
837 | continue; |
838 | if (formats[i].channels != param->channels) |
839 | continue; |
840 | } |
841 | if (!auconv_is_supported_rate(&formats[i], |
842 | param->sample_rate)) |
843 | continue; |
844 | return i; |
845 | } |
846 | return -1; |
847 | } |
848 | |
849 | /** |
850 | * a sub-routine for auconv_set_converter() |
851 | * SLINEAR ==> SLINEAR_<host-endian> |
852 | * ULINEAR ==> ULINEAR_<host-endian> |
853 | * SLINEAR_BE 8bit ==> SLINEAR_LE 8bit |
854 | * ULINEAR_BE 8bit ==> ULINEAR_LE 8bit |
855 | * This should be the same rule as audio_check_params() |
856 | */ |
857 | static u_int |
858 | auconv_normalize_encoding(u_int encoding, u_int precision) |
859 | { |
860 | int enc; |
861 | |
862 | enc = encoding; |
863 | if (enc == AUDIO_ENCODING_SLINEAR_LE) |
864 | return enc; |
865 | if (enc == AUDIO_ENCODING_ULINEAR_LE) |
866 | return enc; |
867 | #if BYTE_ORDER == LITTLE_ENDIAN |
868 | if (enc == AUDIO_ENCODING_SLINEAR) |
869 | return AUDIO_ENCODING_SLINEAR_LE; |
870 | else if (enc == AUDIO_ENCODING_ULINEAR) |
871 | return AUDIO_ENCODING_ULINEAR_LE; |
872 | #else |
873 | if (enc == AUDIO_ENCODING_SLINEAR) |
874 | enc = AUDIO_ENCODING_SLINEAR_BE; |
875 | else if (enc == AUDIO_ENCODING_ULINEAR) |
876 | enc = AUDIO_ENCODING_ULINEAR_BE; |
877 | #endif |
878 | if (precision == 8 && enc == AUDIO_ENCODING_SLINEAR_BE) |
879 | return AUDIO_ENCODING_SLINEAR_LE; |
880 | if (precision == 8 && enc == AUDIO_ENCODING_ULINEAR_BE) |
881 | return AUDIO_ENCODING_ULINEAR_LE; |
882 | return enc; |
883 | } |
884 | |
885 | /** |
886 | * a sub-routine for auconv_set_converter() |
887 | */ |
888 | static int |
889 | auconv_is_supported_rate(const struct audio_format *format, u_int rate) |
890 | { |
891 | u_int i; |
892 | |
893 | if (format->frequency_type == 0) { |
894 | return format->frequency[0] <= rate |
895 | && rate <= format->frequency[1]; |
896 | } |
897 | for (i = 0; i < format->frequency_type; i++) { |
898 | if (format->frequency[i] == rate) |
899 | return true; |
900 | } |
901 | return false; |
902 | } |
903 | |
904 | /** |
905 | * Create an audio_encoding_set besed on hardware capability represented |
906 | * by audio_format. |
907 | * |
908 | * Usage: |
909 | * foo_attach(...) { |
910 | * : |
911 | * if (auconv_create_encodings(formats, nformats, |
912 | * &sc->sc_encodings) != 0) { |
913 | * // attach failure |
914 | * } |
915 | * |
916 | * @param formats [IN] An array of formats which a hardware can support. |
917 | * @param nformats [IN] The number of elements of the array. |
918 | * @param encodings [OUT] receives an address of an audio_encoding_set. |
919 | * @return errno; 0 for success. |
920 | */ |
921 | int |
922 | auconv_create_encodings(const struct audio_format *formats, int nformats, |
923 | struct audio_encoding_set **encodings) |
924 | { |
925 | struct audio_encoding_set *buf; |
926 | int capacity; |
927 | int i; |
928 | int err; |
929 | |
930 | #define ADD_ENCODING(enc, prec, flags) do { \ |
931 | err = auconv_add_encoding(enc, prec, flags, &buf, &capacity); \ |
932 | if (err != 0) goto err_exit; \ |
933 | } while (/*CONSTCOND*/0) |
934 | |
935 | capacity = 10; |
936 | buf = AUCONV_MALLOC(ENCODING_SET_SIZE(capacity)); |
937 | if (buf == NULL) { |
938 | err = ENOMEM; |
939 | goto err_exit; |
940 | } |
941 | buf->size = 0; |
942 | for (i = 0; i < nformats; i++) { |
943 | if (!AUFMT_IS_VALID(&formats[i])) |
944 | continue; |
945 | switch (formats[i].encoding) { |
946 | case AUDIO_ENCODING_SLINEAR_LE: |
947 | ADD_ENCODING(formats[i].encoding, |
948 | formats[i].precision, 0); |
949 | ADD_ENCODING(AUDIO_ENCODING_SLINEAR_BE, |
950 | formats[i].precision, |
951 | AUDIO_ENCODINGFLAG_EMULATED); |
952 | ADD_ENCODING(AUDIO_ENCODING_ULINEAR_LE, |
953 | formats[i].precision, |
954 | AUDIO_ENCODINGFLAG_EMULATED); |
955 | ADD_ENCODING(AUDIO_ENCODING_ULINEAR_BE, |
956 | formats[i].precision, |
957 | AUDIO_ENCODINGFLAG_EMULATED); |
958 | #if NMULAW > 0 |
959 | if (formats[i].precision == 8 |
960 | || formats[i].precision == 16) { |
961 | ADD_ENCODING(AUDIO_ENCODING_ULAW, 8, |
962 | AUDIO_ENCODINGFLAG_EMULATED); |
963 | ADD_ENCODING(AUDIO_ENCODING_ALAW, 8, |
964 | AUDIO_ENCODINGFLAG_EMULATED); |
965 | } |
966 | #endif |
967 | break; |
968 | case AUDIO_ENCODING_SLINEAR_BE: |
969 | ADD_ENCODING(formats[i].encoding, |
970 | formats[i].precision, 0); |
971 | ADD_ENCODING(AUDIO_ENCODING_SLINEAR_LE, |
972 | formats[i].precision, |
973 | AUDIO_ENCODINGFLAG_EMULATED); |
974 | ADD_ENCODING(AUDIO_ENCODING_ULINEAR_LE, |
975 | formats[i].precision, |
976 | AUDIO_ENCODINGFLAG_EMULATED); |
977 | ADD_ENCODING(AUDIO_ENCODING_ULINEAR_BE, |
978 | formats[i].precision, |
979 | AUDIO_ENCODINGFLAG_EMULATED); |
980 | #if NMULAW > 0 |
981 | if (formats[i].precision == 8 |
982 | || formats[i].precision == 16) { |
983 | ADD_ENCODING(AUDIO_ENCODING_ULAW, 8, |
984 | AUDIO_ENCODINGFLAG_EMULATED); |
985 | ADD_ENCODING(AUDIO_ENCODING_ALAW, 8, |
986 | AUDIO_ENCODINGFLAG_EMULATED); |
987 | } |
988 | #endif |
989 | break; |
990 | case AUDIO_ENCODING_ULINEAR_LE: |
991 | ADD_ENCODING(formats[i].encoding, |
992 | formats[i].precision, 0); |
993 | ADD_ENCODING(AUDIO_ENCODING_SLINEAR_BE, |
994 | formats[i].precision, |
995 | AUDIO_ENCODINGFLAG_EMULATED); |
996 | ADD_ENCODING(AUDIO_ENCODING_SLINEAR_LE, |
997 | formats[i].precision, |
998 | AUDIO_ENCODINGFLAG_EMULATED); |
999 | ADD_ENCODING(AUDIO_ENCODING_ULINEAR_BE, |
1000 | formats[i].precision, |
1001 | AUDIO_ENCODINGFLAG_EMULATED); |
1002 | #if NMULAW > 0 |
1003 | if (formats[i].precision == 8 |
1004 | || formats[i].precision == 16) { |
1005 | ADD_ENCODING(AUDIO_ENCODING_ULAW, 8, |
1006 | AUDIO_ENCODINGFLAG_EMULATED); |
1007 | ADD_ENCODING(AUDIO_ENCODING_ALAW, 8, |
1008 | AUDIO_ENCODINGFLAG_EMULATED); |
1009 | } |
1010 | #endif |
1011 | break; |
1012 | case AUDIO_ENCODING_ULINEAR_BE: |
1013 | ADD_ENCODING(formats[i].encoding, |
1014 | formats[i].precision, 0); |
1015 | ADD_ENCODING(AUDIO_ENCODING_SLINEAR_BE, |
1016 | formats[i].precision, |
1017 | AUDIO_ENCODINGFLAG_EMULATED); |
1018 | ADD_ENCODING(AUDIO_ENCODING_ULINEAR_LE, |
1019 | formats[i].precision, |
1020 | AUDIO_ENCODINGFLAG_EMULATED); |
1021 | ADD_ENCODING(AUDIO_ENCODING_SLINEAR_LE, |
1022 | formats[i].precision, |
1023 | AUDIO_ENCODINGFLAG_EMULATED); |
1024 | #if NMULAW > 0 |
1025 | if (formats[i].precision == 8 |
1026 | || formats[i].precision == 16) { |
1027 | ADD_ENCODING(AUDIO_ENCODING_ULAW, 8, |
1028 | AUDIO_ENCODINGFLAG_EMULATED); |
1029 | ADD_ENCODING(AUDIO_ENCODING_ALAW, 8, |
1030 | AUDIO_ENCODINGFLAG_EMULATED); |
1031 | } |
1032 | #endif |
1033 | break; |
1034 | |
1035 | case AUDIO_ENCODING_ULAW: |
1036 | case AUDIO_ENCODING_ALAW: |
1037 | case AUDIO_ENCODING_ADPCM: |
1038 | case AUDIO_ENCODING_MPEG_L1_STREAM: |
1039 | case AUDIO_ENCODING_MPEG_L1_PACKETS: |
1040 | case AUDIO_ENCODING_MPEG_L1_SYSTEM: |
1041 | case AUDIO_ENCODING_MPEG_L2_STREAM: |
1042 | case AUDIO_ENCODING_MPEG_L2_PACKETS: |
1043 | case AUDIO_ENCODING_MPEG_L2_SYSTEM: |
1044 | case AUDIO_ENCODING_AC3: |
1045 | ADD_ENCODING(formats[i].encoding, |
1046 | formats[i].precision, 0); |
1047 | break; |
1048 | |
1049 | case AUDIO_ENCODING_SLINEAR: |
1050 | case AUDIO_ENCODING_ULINEAR: |
1051 | case AUDIO_ENCODING_LINEAR: |
1052 | case AUDIO_ENCODING_LINEAR8: |
1053 | default: |
1054 | printf("%s: invalid encoding value " |
1055 | "for audio_format: %d\n" , |
1056 | __func__, formats[i].encoding); |
1057 | break; |
1058 | } |
1059 | } |
1060 | *encodings = buf; |
1061 | return 0; |
1062 | |
1063 | err_exit: |
1064 | if (buf != NULL) |
1065 | AUCONV_FREE(buf); |
1066 | *encodings = NULL; |
1067 | return err; |
1068 | } |
1069 | |
1070 | /** |
1071 | * a sub-routine for auconv_create_encodings() |
1072 | */ |
1073 | static int |
1074 | auconv_add_encoding(int enc, int prec, int flags, |
1075 | struct audio_encoding_set **buf, int *capacity) |
1076 | { |
1077 | static const char *encoding_names[] = { |
1078 | NULL, AudioEmulaw, AudioEalaw, NULL, |
1079 | NULL, AudioEadpcm, AudioEslinear_le, AudioEslinear_be, |
1080 | AudioEulinear_le, AudioEulinear_be, |
1081 | AudioEslinear, AudioEulinear, |
1082 | AudioEmpeg_l1_stream, AudioEmpeg_l1_packets, |
1083 | AudioEmpeg_l1_system, AudioEmpeg_l2_stream, |
1084 | AudioEmpeg_l2_packets, AudioEmpeg_l2_system, |
1085 | AudioEac3 |
1086 | }; |
1087 | struct audio_encoding_set *set; |
1088 | struct audio_encoding_set *new_buf; |
1089 | audio_encoding_t *e; |
1090 | int i; |
1091 | |
1092 | set = *buf; |
1093 | /* already has the same encoding? */ |
1094 | e = set->items; |
1095 | for (i = 0; i < set->size; i++, e++) { |
1096 | if (e->encoding == enc && e->precision == prec) { |
1097 | /* overwrite EMULATED flag */ |
1098 | if ((e->flags & AUDIO_ENCODINGFLAG_EMULATED) |
1099 | && (flags & AUDIO_ENCODINGFLAG_EMULATED) == 0) { |
1100 | e->flags &= ~AUDIO_ENCODINGFLAG_EMULATED; |
1101 | } |
1102 | return 0; |
1103 | } |
1104 | } |
1105 | /* We don't have the specified one. */ |
1106 | |
1107 | if (set->size >= *capacity) { |
1108 | new_buf = AUCONV_REALLOC(set, |
1109 | ENCODING_SET_SIZE(*capacity + 10)); |
1110 | if (new_buf == NULL) |
1111 | return ENOMEM; |
1112 | *buf = new_buf; |
1113 | set = new_buf; |
1114 | *capacity += 10; |
1115 | } |
1116 | |
1117 | e = &set->items[set->size]; |
1118 | e->index = 0; |
1119 | strlcpy(e->name, encoding_names[enc], MAX_AUDIO_DEV_LEN); |
1120 | e->encoding = enc; |
1121 | e->precision = prec; |
1122 | e->flags = flags; |
1123 | set->size += 1; |
1124 | return 0; |
1125 | } |
1126 | |
1127 | /** |
1128 | * Delete an audio_encoding_set created by auconv_create_encodings(). |
1129 | * |
1130 | * Usage: |
1131 | * foo_detach(...) { |
1132 | * : |
1133 | * auconv_delete_encodings(sc->sc_encodings); |
1134 | * : |
1135 | * } |
1136 | * |
1137 | * @param encodings [IN] An audio_encoding_set which was created by |
1138 | * auconv_create_encodings(). |
1139 | * @return errno; 0 for success. |
1140 | */ |
1141 | int auconv_delete_encodings(struct audio_encoding_set *encodings) |
1142 | { |
1143 | if (encodings != NULL) |
1144 | AUCONV_FREE(encodings); |
1145 | return 0; |
1146 | } |
1147 | |
1148 | /** |
1149 | * Copy the element specified by aep->index. |
1150 | * |
1151 | * Usage: |
1152 | * int foo_query_encoding(void *v, audio_encoding_t *aep) { |
1153 | * struct foo_softc *sc = (struct foo_softc *)v; |
1154 | * return auconv_query_encoding(sc->sc_encodings, aep); |
1155 | * } |
1156 | * |
1157 | * @param encodings [IN] An audio_encoding_set created by |
1158 | * auconv_create_encodings(). |
1159 | * @param aep [IN/OUT] resultant audio_encoding_t. |
1160 | */ |
1161 | int |
1162 | auconv_query_encoding(const struct audio_encoding_set *encodings, |
1163 | audio_encoding_t *aep) |
1164 | { |
1165 | if (aep->index >= encodings->size) |
1166 | return EINVAL; |
1167 | strlcpy(aep->name, encodings->items[aep->index].name, |
1168 | MAX_AUDIO_DEV_LEN); |
1169 | aep->encoding = encodings->items[aep->index].encoding; |
1170 | aep->precision = encodings->items[aep->index].precision; |
1171 | aep->flags = encodings->items[aep->index].flags; |
1172 | return 0; |
1173 | } |
1174 | |