]> git.proxmox.com Git - mirror_ubuntu-artful-kernel.git/blob - sound/oss/dmasound/dmasound_paula.c
Linux-2.6.12-rc2
[mirror_ubuntu-artful-kernel.git] / sound / oss / dmasound / dmasound_paula.c
1 /*
2 * linux/sound/oss/dmasound/dmasound_paula.c
3 *
4 * Amiga `Paula' DMA Sound Driver
5 *
6 * See linux/sound/oss/dmasound/dmasound_core.c for copyright and credits
7 * prior to 28/01/2001
8 *
9 * 28/01/2001 [0.1] Iain Sandoe
10 * - added versioning
11 * - put in and populated the hardware_afmts field.
12 * [0.2] - put in SNDCTL_DSP_GETCAPS value.
13 * [0.3] - put in constraint on state buffer usage.
14 * [0.4] - put in default hard/soft settings
15 */
16
17
18 #include <linux/module.h>
19 #include <linux/config.h>
20 #include <linux/mm.h>
21 #include <linux/init.h>
22 #include <linux/ioport.h>
23 #include <linux/soundcard.h>
24 #include <linux/interrupt.h>
25
26 #include <asm/uaccess.h>
27 #include <asm/setup.h>
28 #include <asm/amigahw.h>
29 #include <asm/amigaints.h>
30 #include <asm/machdep.h>
31
32 #include "dmasound.h"
33
34 #define DMASOUND_PAULA_REVISION 0
35 #define DMASOUND_PAULA_EDITION 4
36
37 /*
38 * The minimum period for audio depends on htotal (for OCS/ECS/AGA)
39 * (Imported from arch/m68k/amiga/amisound.c)
40 */
41
42 extern volatile u_short amiga_audio_min_period;
43
44
45 /*
46 * amiga_mksound() should be able to restore the period after beeping
47 * (Imported from arch/m68k/amiga/amisound.c)
48 */
49
50 extern u_short amiga_audio_period;
51
52
53 /*
54 * Audio DMA masks
55 */
56
57 #define AMI_AUDIO_OFF (DMAF_AUD0 | DMAF_AUD1 | DMAF_AUD2 | DMAF_AUD3)
58 #define AMI_AUDIO_8 (DMAF_SETCLR | DMAF_MASTER | DMAF_AUD0 | DMAF_AUD1)
59 #define AMI_AUDIO_14 (AMI_AUDIO_8 | DMAF_AUD2 | DMAF_AUD3)
60
61
62 /*
63 * Helper pointers for 16(14)-bit sound
64 */
65
66 static int write_sq_block_size_half, write_sq_block_size_quarter;
67
68
69 /*** Low level stuff *********************************************************/
70
71
72 static void *AmiAlloc(unsigned int size, int flags);
73 static void AmiFree(void *obj, unsigned int size);
74 static int AmiIrqInit(void);
75 #ifdef MODULE
76 static void AmiIrqCleanUp(void);
77 #endif
78 static void AmiSilence(void);
79 static void AmiInit(void);
80 static int AmiSetFormat(int format);
81 static int AmiSetVolume(int volume);
82 static int AmiSetTreble(int treble);
83 static void AmiPlayNextFrame(int index);
84 static void AmiPlay(void);
85 static irqreturn_t AmiInterrupt(int irq, void *dummy, struct pt_regs *fp);
86
87 #ifdef CONFIG_HEARTBEAT
88
89 /*
90 * Heartbeat interferes with sound since the 7 kHz low-pass filter and the
91 * power LED are controlled by the same line.
92 */
93
94 #ifdef CONFIG_APUS
95 #define mach_heartbeat ppc_md.heartbeat
96 #endif
97
98 static void (*saved_heartbeat)(int) = NULL;
99
100 static inline void disable_heartbeat(void)
101 {
102 if (mach_heartbeat) {
103 saved_heartbeat = mach_heartbeat;
104 mach_heartbeat = NULL;
105 }
106 AmiSetTreble(dmasound.treble);
107 }
108
109 static inline void enable_heartbeat(void)
110 {
111 if (saved_heartbeat)
112 mach_heartbeat = saved_heartbeat;
113 }
114 #else /* !CONFIG_HEARTBEAT */
115 #define disable_heartbeat() do { } while (0)
116 #define enable_heartbeat() do { } while (0)
117 #endif /* !CONFIG_HEARTBEAT */
118
119
120 /*** Mid level stuff *********************************************************/
121
122 static void AmiMixerInit(void);
123 static int AmiMixerIoctl(u_int cmd, u_long arg);
124 static int AmiWriteSqSetup(void);
125 static int AmiStateInfo(char *buffer, size_t space);
126
127
128 /*** Translations ************************************************************/
129
130 /* ++TeSche: radically changed for new expanding purposes...
131 *
132 * These two routines now deal with copying/expanding/translating the samples
133 * from user space into our buffer at the right frequency. They take care about
134 * how much data there's actually to read, how much buffer space there is and
135 * to convert samples into the right frequency/encoding. They will only work on
136 * complete samples so it may happen they leave some bytes in the input stream
137 * if the user didn't write a multiple of the current sample size. They both
138 * return the number of bytes they've used from both streams so you may detect
139 * such a situation. Luckily all programs should be able to cope with that.
140 *
141 * I think I've optimized anything as far as one can do in plain C, all
142 * variables should fit in registers and the loops are really short. There's
143 * one loop for every possible situation. Writing a more generalized and thus
144 * parameterized loop would only produce slower code. Feel free to optimize
145 * this in assembler if you like. :)
146 *
147 * I think these routines belong here because they're not yet really hardware
148 * independent, especially the fact that the Falcon can play 16bit samples
149 * only in stereo is hardcoded in both of them!
150 *
151 * ++geert: split in even more functions (one per format)
152 */
153
154
155 /*
156 * Native format
157 */
158
159 static ssize_t ami_ct_s8(const u_char *userPtr, size_t userCount,
160 u_char frame[], ssize_t *frameUsed, ssize_t frameLeft)
161 {
162 ssize_t count, used;
163
164 if (!dmasound.soft.stereo) {
165 void *p = &frame[*frameUsed];
166 count = min_t(unsigned long, userCount, frameLeft) & ~1;
167 used = count;
168 if (copy_from_user(p, userPtr, count))
169 return -EFAULT;
170 } else {
171 u_char *left = &frame[*frameUsed>>1];
172 u_char *right = left+write_sq_block_size_half;
173 count = min_t(unsigned long, userCount, frameLeft)>>1 & ~1;
174 used = count*2;
175 while (count > 0) {
176 if (get_user(*left++, userPtr++)
177 || get_user(*right++, userPtr++))
178 return -EFAULT;
179 count--;
180 }
181 }
182 *frameUsed += used;
183 return used;
184 }
185
186
187 /*
188 * Copy and convert 8 bit data
189 */
190
191 #define GENERATE_AMI_CT8(funcname, convsample) \
192 static ssize_t funcname(const u_char *userPtr, size_t userCount, \
193 u_char frame[], ssize_t *frameUsed, \
194 ssize_t frameLeft) \
195 { \
196 ssize_t count, used; \
197 \
198 if (!dmasound.soft.stereo) { \
199 u_char *p = &frame[*frameUsed]; \
200 count = min_t(size_t, userCount, frameLeft) & ~1; \
201 used = count; \
202 while (count > 0) { \
203 u_char data; \
204 if (get_user(data, userPtr++)) \
205 return -EFAULT; \
206 *p++ = convsample(data); \
207 count--; \
208 } \
209 } else { \
210 u_char *left = &frame[*frameUsed>>1]; \
211 u_char *right = left+write_sq_block_size_half; \
212 count = min_t(size_t, userCount, frameLeft)>>1 & ~1; \
213 used = count*2; \
214 while (count > 0) { \
215 u_char data; \
216 if (get_user(data, userPtr++)) \
217 return -EFAULT; \
218 *left++ = convsample(data); \
219 if (get_user(data, userPtr++)) \
220 return -EFAULT; \
221 *right++ = convsample(data); \
222 count--; \
223 } \
224 } \
225 *frameUsed += used; \
226 return used; \
227 }
228
229 #define AMI_CT_ULAW(x) (dmasound_ulaw2dma8[(x)])
230 #define AMI_CT_ALAW(x) (dmasound_alaw2dma8[(x)])
231 #define AMI_CT_U8(x) ((x) ^ 0x80)
232
233 GENERATE_AMI_CT8(ami_ct_ulaw, AMI_CT_ULAW)
234 GENERATE_AMI_CT8(ami_ct_alaw, AMI_CT_ALAW)
235 GENERATE_AMI_CT8(ami_ct_u8, AMI_CT_U8)
236
237
238 /*
239 * Copy and convert 16 bit data
240 */
241
242 #define GENERATE_AMI_CT_16(funcname, convsample) \
243 static ssize_t funcname(const u_char *userPtr, size_t userCount, \
244 u_char frame[], ssize_t *frameUsed, \
245 ssize_t frameLeft) \
246 { \
247 ssize_t count, used; \
248 u_short data; \
249 \
250 if (!dmasound.soft.stereo) { \
251 u_char *high = &frame[*frameUsed>>1]; \
252 u_char *low = high+write_sq_block_size_half; \
253 count = min_t(size_t, userCount, frameLeft)>>1 & ~1; \
254 used = count*2; \
255 while (count > 0) { \
256 if (get_user(data, ((u_short *)userPtr)++)) \
257 return -EFAULT; \
258 data = convsample(data); \
259 *high++ = data>>8; \
260 *low++ = (data>>2) & 0x3f; \
261 count--; \
262 } \
263 } else { \
264 u_char *lefth = &frame[*frameUsed>>2]; \
265 u_char *leftl = lefth+write_sq_block_size_quarter; \
266 u_char *righth = lefth+write_sq_block_size_half; \
267 u_char *rightl = righth+write_sq_block_size_quarter; \
268 count = min_t(size_t, userCount, frameLeft)>>2 & ~1; \
269 used = count*4; \
270 while (count > 0) { \
271 if (get_user(data, ((u_short *)userPtr)++)) \
272 return -EFAULT; \
273 data = convsample(data); \
274 *lefth++ = data>>8; \
275 *leftl++ = (data>>2) & 0x3f; \
276 if (get_user(data, ((u_short *)userPtr)++)) \
277 return -EFAULT; \
278 data = convsample(data); \
279 *righth++ = data>>8; \
280 *rightl++ = (data>>2) & 0x3f; \
281 count--; \
282 } \
283 } \
284 *frameUsed += used; \
285 return used; \
286 }
287
288 #define AMI_CT_S16BE(x) (x)
289 #define AMI_CT_U16BE(x) ((x) ^ 0x8000)
290 #define AMI_CT_S16LE(x) (le2be16((x)))
291 #define AMI_CT_U16LE(x) (le2be16((x)) ^ 0x8000)
292
293 GENERATE_AMI_CT_16(ami_ct_s16be, AMI_CT_S16BE)
294 GENERATE_AMI_CT_16(ami_ct_u16be, AMI_CT_U16BE)
295 GENERATE_AMI_CT_16(ami_ct_s16le, AMI_CT_S16LE)
296 GENERATE_AMI_CT_16(ami_ct_u16le, AMI_CT_U16LE)
297
298
299 static TRANS transAmiga = {
300 .ct_ulaw = ami_ct_ulaw,
301 .ct_alaw = ami_ct_alaw,
302 .ct_s8 = ami_ct_s8,
303 .ct_u8 = ami_ct_u8,
304 .ct_s16be = ami_ct_s16be,
305 .ct_u16be = ami_ct_u16be,
306 .ct_s16le = ami_ct_s16le,
307 .ct_u16le = ami_ct_u16le,
308 };
309
310 /*** Low level stuff *********************************************************/
311
312 static inline void StopDMA(void)
313 {
314 custom.aud[0].audvol = custom.aud[1].audvol = 0;
315 custom.aud[2].audvol = custom.aud[3].audvol = 0;
316 custom.dmacon = AMI_AUDIO_OFF;
317 enable_heartbeat();
318 }
319
320 static void *AmiAlloc(unsigned int size, int flags)
321 {
322 return amiga_chip_alloc((long)size, "dmasound [Paula]");
323 }
324
325 static void AmiFree(void *obj, unsigned int size)
326 {
327 amiga_chip_free (obj);
328 }
329
330 static int __init AmiIrqInit(void)
331 {
332 /* turn off DMA for audio channels */
333 StopDMA();
334
335 /* Register interrupt handler. */
336 if (request_irq(IRQ_AMIGA_AUD0, AmiInterrupt, 0, "DMA sound",
337 AmiInterrupt))
338 return 0;
339 return 1;
340 }
341
342 #ifdef MODULE
343 static void AmiIrqCleanUp(void)
344 {
345 /* turn off DMA for audio channels */
346 StopDMA();
347 /* release the interrupt */
348 free_irq(IRQ_AMIGA_AUD0, AmiInterrupt);
349 }
350 #endif /* MODULE */
351
352 static void AmiSilence(void)
353 {
354 /* turn off DMA for audio channels */
355 StopDMA();
356 }
357
358
359 static void AmiInit(void)
360 {
361 int period, i;
362
363 AmiSilence();
364
365 if (dmasound.soft.speed)
366 period = amiga_colorclock/dmasound.soft.speed-1;
367 else
368 period = amiga_audio_min_period;
369 dmasound.hard = dmasound.soft;
370 dmasound.trans_write = &transAmiga;
371
372 if (period < amiga_audio_min_period) {
373 /* we would need to squeeze the sound, but we won't do that */
374 period = amiga_audio_min_period;
375 } else if (period > 65535) {
376 period = 65535;
377 }
378 dmasound.hard.speed = amiga_colorclock/(period+1);
379
380 for (i = 0; i < 4; i++)
381 custom.aud[i].audper = period;
382 amiga_audio_period = period;
383 }
384
385
386 static int AmiSetFormat(int format)
387 {
388 int size;
389
390 /* Amiga sound DMA supports 8bit and 16bit (pseudo 14 bit) modes */
391
392 switch (format) {
393 case AFMT_QUERY:
394 return dmasound.soft.format;
395 case AFMT_MU_LAW:
396 case AFMT_A_LAW:
397 case AFMT_U8:
398 case AFMT_S8:
399 size = 8;
400 break;
401 case AFMT_S16_BE:
402 case AFMT_U16_BE:
403 case AFMT_S16_LE:
404 case AFMT_U16_LE:
405 size = 16;
406 break;
407 default: /* :-) */
408 size = 8;
409 format = AFMT_S8;
410 }
411
412 dmasound.soft.format = format;
413 dmasound.soft.size = size;
414 if (dmasound.minDev == SND_DEV_DSP) {
415 dmasound.dsp.format = format;
416 dmasound.dsp.size = dmasound.soft.size;
417 }
418 AmiInit();
419
420 return format;
421 }
422
423
424 #define VOLUME_VOXWARE_TO_AMI(v) \
425 (((v) < 0) ? 0 : ((v) > 100) ? 64 : ((v) * 64)/100)
426 #define VOLUME_AMI_TO_VOXWARE(v) ((v)*100/64)
427
428 static int AmiSetVolume(int volume)
429 {
430 dmasound.volume_left = VOLUME_VOXWARE_TO_AMI(volume & 0xff);
431 custom.aud[0].audvol = dmasound.volume_left;
432 dmasound.volume_right = VOLUME_VOXWARE_TO_AMI((volume & 0xff00) >> 8);
433 custom.aud[1].audvol = dmasound.volume_right;
434 if (dmasound.hard.size == 16) {
435 if (dmasound.volume_left == 64 && dmasound.volume_right == 64) {
436 custom.aud[2].audvol = 1;
437 custom.aud[3].audvol = 1;
438 } else {
439 custom.aud[2].audvol = 0;
440 custom.aud[3].audvol = 0;
441 }
442 }
443 return VOLUME_AMI_TO_VOXWARE(dmasound.volume_left) |
444 (VOLUME_AMI_TO_VOXWARE(dmasound.volume_right) << 8);
445 }
446
447 static int AmiSetTreble(int treble)
448 {
449 dmasound.treble = treble;
450 if (treble < 50)
451 ciaa.pra &= ~0x02;
452 else
453 ciaa.pra |= 0x02;
454 return treble;
455 }
456
457
458 #define AMI_PLAY_LOADED 1
459 #define AMI_PLAY_PLAYING 2
460 #define AMI_PLAY_MASK 3
461
462
463 static void AmiPlayNextFrame(int index)
464 {
465 u_char *start, *ch0, *ch1, *ch2, *ch3;
466 u_long size;
467
468 /* used by AmiPlay() if all doubts whether there really is something
469 * to be played are already wiped out.
470 */
471 start = write_sq.buffers[write_sq.front];
472 size = (write_sq.count == index ? write_sq.rear_size
473 : write_sq.block_size)>>1;
474
475 if (dmasound.hard.stereo) {
476 ch0 = start;
477 ch1 = start+write_sq_block_size_half;
478 size >>= 1;
479 } else {
480 ch0 = start;
481 ch1 = start;
482 }
483
484 disable_heartbeat();
485 custom.aud[0].audvol = dmasound.volume_left;
486 custom.aud[1].audvol = dmasound.volume_right;
487 if (dmasound.hard.size == 8) {
488 custom.aud[0].audlc = (u_short *)ZTWO_PADDR(ch0);
489 custom.aud[0].audlen = size;
490 custom.aud[1].audlc = (u_short *)ZTWO_PADDR(ch1);
491 custom.aud[1].audlen = size;
492 custom.dmacon = AMI_AUDIO_8;
493 } else {
494 size >>= 1;
495 custom.aud[0].audlc = (u_short *)ZTWO_PADDR(ch0);
496 custom.aud[0].audlen = size;
497 custom.aud[1].audlc = (u_short *)ZTWO_PADDR(ch1);
498 custom.aud[1].audlen = size;
499 if (dmasound.volume_left == 64 && dmasound.volume_right == 64) {
500 /* We can play pseudo 14-bit only with the maximum volume */
501 ch3 = ch0+write_sq_block_size_quarter;
502 ch2 = ch1+write_sq_block_size_quarter;
503 custom.aud[2].audvol = 1; /* we are being affected by the beeps */
504 custom.aud[3].audvol = 1; /* restoring volume here helps a bit */
505 custom.aud[2].audlc = (u_short *)ZTWO_PADDR(ch2);
506 custom.aud[2].audlen = size;
507 custom.aud[3].audlc = (u_short *)ZTWO_PADDR(ch3);
508 custom.aud[3].audlen = size;
509 custom.dmacon = AMI_AUDIO_14;
510 } else {
511 custom.aud[2].audvol = 0;
512 custom.aud[3].audvol = 0;
513 custom.dmacon = AMI_AUDIO_8;
514 }
515 }
516 write_sq.front = (write_sq.front+1) % write_sq.max_count;
517 write_sq.active |= AMI_PLAY_LOADED;
518 }
519
520
521 static void AmiPlay(void)
522 {
523 int minframes = 1;
524
525 custom.intena = IF_AUD0;
526
527 if (write_sq.active & AMI_PLAY_LOADED) {
528 /* There's already a frame loaded */
529 custom.intena = IF_SETCLR | IF_AUD0;
530 return;
531 }
532
533 if (write_sq.active & AMI_PLAY_PLAYING)
534 /* Increase threshold: frame 1 is already being played */
535 minframes = 2;
536
537 if (write_sq.count < minframes) {
538 /* Nothing to do */
539 custom.intena = IF_SETCLR | IF_AUD0;
540 return;
541 }
542
543 if (write_sq.count <= minframes &&
544 write_sq.rear_size < write_sq.block_size && !write_sq.syncing) {
545 /* hmmm, the only existing frame is not
546 * yet filled and we're not syncing?
547 */
548 custom.intena = IF_SETCLR | IF_AUD0;
549 return;
550 }
551
552 AmiPlayNextFrame(minframes);
553
554 custom.intena = IF_SETCLR | IF_AUD0;
555 }
556
557
558 static irqreturn_t AmiInterrupt(int irq, void *dummy, struct pt_regs *fp)
559 {
560 int minframes = 1;
561
562 custom.intena = IF_AUD0;
563
564 if (!write_sq.active) {
565 /* Playing was interrupted and sq_reset() has already cleared
566 * the sq variables, so better don't do anything here.
567 */
568 WAKE_UP(write_sq.sync_queue);
569 return IRQ_HANDLED;
570 }
571
572 if (write_sq.active & AMI_PLAY_PLAYING) {
573 /* We've just finished a frame */
574 write_sq.count--;
575 WAKE_UP(write_sq.action_queue);
576 }
577
578 if (write_sq.active & AMI_PLAY_LOADED)
579 /* Increase threshold: frame 1 is already being played */
580 minframes = 2;
581
582 /* Shift the flags */
583 write_sq.active = (write_sq.active<<1) & AMI_PLAY_MASK;
584
585 if (!write_sq.active)
586 /* No frame is playing, disable audio DMA */
587 StopDMA();
588
589 custom.intena = IF_SETCLR | IF_AUD0;
590
591 if (write_sq.count >= minframes)
592 /* Try to play the next frame */
593 AmiPlay();
594
595 if (!write_sq.active)
596 /* Nothing to play anymore.
597 Wake up a process waiting for audio output to drain. */
598 WAKE_UP(write_sq.sync_queue);
599 return IRQ_HANDLED;
600 }
601
602 /*** Mid level stuff *********************************************************/
603
604
605 /*
606 * /dev/mixer abstraction
607 */
608
609 static void __init AmiMixerInit(void)
610 {
611 dmasound.volume_left = 64;
612 dmasound.volume_right = 64;
613 custom.aud[0].audvol = dmasound.volume_left;
614 custom.aud[3].audvol = 1; /* For pseudo 14bit */
615 custom.aud[1].audvol = dmasound.volume_right;
616 custom.aud[2].audvol = 1; /* For pseudo 14bit */
617 dmasound.treble = 50;
618 }
619
620 static int AmiMixerIoctl(u_int cmd, u_long arg)
621 {
622 int data;
623 switch (cmd) {
624 case SOUND_MIXER_READ_DEVMASK:
625 return IOCTL_OUT(arg, SOUND_MASK_VOLUME | SOUND_MASK_TREBLE);
626 case SOUND_MIXER_READ_RECMASK:
627 return IOCTL_OUT(arg, 0);
628 case SOUND_MIXER_READ_STEREODEVS:
629 return IOCTL_OUT(arg, SOUND_MASK_VOLUME);
630 case SOUND_MIXER_READ_VOLUME:
631 return IOCTL_OUT(arg,
632 VOLUME_AMI_TO_VOXWARE(dmasound.volume_left) |
633 VOLUME_AMI_TO_VOXWARE(dmasound.volume_right) << 8);
634 case SOUND_MIXER_WRITE_VOLUME:
635 IOCTL_IN(arg, data);
636 return IOCTL_OUT(arg, dmasound_set_volume(data));
637 case SOUND_MIXER_READ_TREBLE:
638 return IOCTL_OUT(arg, dmasound.treble);
639 case SOUND_MIXER_WRITE_TREBLE:
640 IOCTL_IN(arg, data);
641 return IOCTL_OUT(arg, dmasound_set_treble(data));
642 }
643 return -EINVAL;
644 }
645
646
647 static int AmiWriteSqSetup(void)
648 {
649 write_sq_block_size_half = write_sq.block_size>>1;
650 write_sq_block_size_quarter = write_sq_block_size_half>>1;
651 return 0;
652 }
653
654
655 static int AmiStateInfo(char *buffer, size_t space)
656 {
657 int len = 0;
658 len += sprintf(buffer+len, "\tsound.volume_left = %d [0...64]\n",
659 dmasound.volume_left);
660 len += sprintf(buffer+len, "\tsound.volume_right = %d [0...64]\n",
661 dmasound.volume_right);
662 if (len >= space) {
663 printk(KERN_ERR "dmasound_paula: overlowed state buffer alloc.\n") ;
664 len = space ;
665 }
666 return len;
667 }
668
669
670 /*** Machine definitions *****************************************************/
671
672 static SETTINGS def_hard = {
673 .format = AFMT_S8,
674 .stereo = 0,
675 .size = 8,
676 .speed = 8000
677 } ;
678
679 static SETTINGS def_soft = {
680 .format = AFMT_U8,
681 .stereo = 0,
682 .size = 8,
683 .speed = 8000
684 } ;
685
686 static MACHINE machAmiga = {
687 .name = "Amiga",
688 .name2 = "AMIGA",
689 .owner = THIS_MODULE,
690 .dma_alloc = AmiAlloc,
691 .dma_free = AmiFree,
692 .irqinit = AmiIrqInit,
693 #ifdef MODULE
694 .irqcleanup = AmiIrqCleanUp,
695 #endif /* MODULE */
696 .init = AmiInit,
697 .silence = AmiSilence,
698 .setFormat = AmiSetFormat,
699 .setVolume = AmiSetVolume,
700 .setTreble = AmiSetTreble,
701 .play = AmiPlay,
702 .mixer_init = AmiMixerInit,
703 .mixer_ioctl = AmiMixerIoctl,
704 .write_sq_setup = AmiWriteSqSetup,
705 .state_info = AmiStateInfo,
706 .min_dsp_speed = 8000,
707 .version = ((DMASOUND_PAULA_REVISION<<8) | DMASOUND_PAULA_EDITION),
708 .hardware_afmts = (AFMT_S8 | AFMT_S16_BE), /* h'ware-supported formats *only* here */
709 .capabilities = DSP_CAP_BATCH /* As per SNDCTL_DSP_GETCAPS */
710 };
711
712
713 /*** Config & Setup **********************************************************/
714
715
716 int __init dmasound_paula_init(void)
717 {
718 int err;
719
720 if (MACH_IS_AMIGA && AMIGAHW_PRESENT(AMI_AUDIO)) {
721 if (!request_mem_region(CUSTOM_PHYSADDR+0xa0, 0x40,
722 "dmasound [Paula]"))
723 return -EBUSY;
724 dmasound.mach = machAmiga;
725 dmasound.mach.default_hard = def_hard ;
726 dmasound.mach.default_soft = def_soft ;
727 err = dmasound_init();
728 if (err)
729 release_mem_region(CUSTOM_PHYSADDR+0xa0, 0x40);
730 return err;
731 } else
732 return -ENODEV;
733 }
734
735 static void __exit dmasound_paula_cleanup(void)
736 {
737 dmasound_deinit();
738 release_mem_region(CUSTOM_PHYSADDR+0xa0, 0x40);
739 }
740
741 module_init(dmasound_paula_init);
742 module_exit(dmasound_paula_cleanup);
743 MODULE_LICENSE("GPL");