1 // SPDX-License-Identifier: GPL-2.0-only
2 /*
3 * linux/sound/oss/dmasound/dmasound_paula.c
4 *
5 * Amiga `Paula' DMA Sound Driver
6 *
7 * See linux/sound/oss/dmasound/dmasound_core.c for copyright and credits
8 * prior to 28/01/2001
9 *
10 * 28/01/2001 [0.1] Iain Sandoe
11 * - added versioning
12 * - put in and populated the hardware_afmts field.
13 * [0.2] - put in SNDCTL_DSP_GETCAPS value.
14 * [0.3] - put in constraint on state buffer usage.
15 * [0.4] - put in default hard/soft settings
16 */
17
18
19 #include <linux/module.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 #include <linux/platform_device.h>
26
27 #include <linux/uaccess.h>
28 #include <asm/setup.h>
29 #include <asm/amigahw.h>
30 #include <asm/amigaints.h>
31 #include <asm/machdep.h>
32
33 #include "dmasound.h"
34
35 #define DMASOUND_PAULA_REVISION 0
36 #define DMASOUND_PAULA_EDITION 4
37
38 #define custom amiga_custom
39 /*
40 * The minimum period for audio depends on htotal (for OCS/ECS/AGA)
41 * (Imported from arch/m68k/amiga/amisound.c)
42 */
43
44 extern volatile u_short amiga_audio_min_period;
45
46
47 /*
48 * amiga_mksound() should be able to restore the period after beeping
49 * (Imported from arch/m68k/amiga/amisound.c)
50 */
51
52 extern u_short amiga_audio_period;
53
54
55 /*
56 * Audio DMA masks
57 */
58
59 #define AMI_AUDIO_OFF (DMAF_AUD0 | DMAF_AUD1 | DMAF_AUD2 | DMAF_AUD3)
60 #define AMI_AUDIO_8 (DMAF_SETCLR | DMAF_MASTER | DMAF_AUD0 | DMAF_AUD1)
61 #define AMI_AUDIO_14 (AMI_AUDIO_8 | DMAF_AUD2 | DMAF_AUD3)
62
63
64 /*
65 * Helper pointers for 16(14)-bit sound
66 */
67
68 static int write_sq_block_size_half, write_sq_block_size_quarter;
69
70
71 /*** Low level stuff *********************************************************/
72
73
74 static void *AmiAlloc(unsigned int size, gfp_t flags);
75 static void AmiFree(void *obj, unsigned int size);
76 static int AmiIrqInit(void);
77 #ifdef MODULE
78 static void AmiIrqCleanUp(void);
79 #endif
80 static void AmiSilence(void);
81 static void AmiInit(void);
82 static int AmiSetFormat(int format);
83 static int AmiSetVolume(int volume);
84 static int AmiSetTreble(int treble);
85 static void AmiPlayNextFrame(int index);
86 static void AmiPlay(void);
87 static irqreturn_t AmiInterrupt(int irq, void *dummy);
88
89 #ifdef CONFIG_HEARTBEAT
90
91 /*
92 * Heartbeat interferes with sound since the 7 kHz low-pass filter and the
93 * power LED are controlled by the same line.
94 */
95
96 static void (*saved_heartbeat)(int) = NULL;
97
disable_heartbeat(void)98 static inline void disable_heartbeat(void)
99 {
100 if (mach_heartbeat) {
101 saved_heartbeat = mach_heartbeat;
102 mach_heartbeat = NULL;
103 }
104 AmiSetTreble(dmasound.treble);
105 }
106
enable_heartbeat(void)107 static inline void enable_heartbeat(void)
108 {
109 if (saved_heartbeat)
110 mach_heartbeat = saved_heartbeat;
111 }
112 #else /* !CONFIG_HEARTBEAT */
113 #define disable_heartbeat() do { } while (0)
114 #define enable_heartbeat() do { } while (0)
115 #endif /* !CONFIG_HEARTBEAT */
116
117
118 /*** Mid level stuff *********************************************************/
119
120 static void AmiMixerInit(void);
121 static int AmiMixerIoctl(u_int cmd, u_long arg);
122 static int AmiWriteSqSetup(void);
123 static int AmiStateInfo(char *buffer, size_t space);
124
125
126 /*** Translations ************************************************************/
127
128 /* ++TeSche: radically changed for new expanding purposes...
129 *
130 * These two routines now deal with copying/expanding/translating the samples
131 * from user space into our buffer at the right frequency. They take care about
132 * how much data there's actually to read, how much buffer space there is and
133 * to convert samples into the right frequency/encoding. They will only work on
134 * complete samples so it may happen they leave some bytes in the input stream
135 * if the user didn't write a multiple of the current sample size. They both
136 * return the number of bytes they've used from both streams so you may detect
137 * such a situation. Luckily all programs should be able to cope with that.
138 *
139 * I think I've optimized anything as far as one can do in plain C, all
140 * variables should fit in registers and the loops are really short. There's
141 * one loop for every possible situation. Writing a more generalized and thus
142 * parameterized loop would only produce slower code. Feel free to optimize
143 * this in assembler if you like. :)
144 *
145 * I think these routines belong here because they're not yet really hardware
146 * independent, especially the fact that the Falcon can play 16bit samples
147 * only in stereo is hardcoded in both of them!
148 *
149 * ++geert: split in even more functions (one per format)
150 */
151
152
153 /*
154 * Native format
155 */
156
ami_ct_s8(const u_char __user * userPtr,size_t userCount,u_char frame[],ssize_t * frameUsed,ssize_t frameLeft)157 static ssize_t ami_ct_s8(const u_char __user *userPtr, size_t userCount,
158 u_char frame[], ssize_t *frameUsed, ssize_t frameLeft)
159 {
160 ssize_t count, used;
161
162 if (!dmasound.soft.stereo) {
163 void *p = &frame[*frameUsed];
164 count = min_t(unsigned long, userCount, frameLeft) & ~1;
165 used = count;
166 if (copy_from_user(p, userPtr, count))
167 return -EFAULT;
168 } else {
169 u_char *left = &frame[*frameUsed>>1];
170 u_char *right = left+write_sq_block_size_half;
171 count = min_t(unsigned long, userCount, frameLeft)>>1 & ~1;
172 used = count*2;
173 while (count > 0) {
174 if (get_user(*left++, userPtr++)
175 || get_user(*right++, userPtr++))
176 return -EFAULT;
177 count--;
178 }
179 }
180 *frameUsed += used;
181 return used;
182 }
183
184
185 /*
186 * Copy and convert 8 bit data
187 */
188
189 #define GENERATE_AMI_CT8(funcname, convsample) \
190 static ssize_t funcname(const u_char __user *userPtr, size_t userCount, \
191 u_char frame[], ssize_t *frameUsed, \
192 ssize_t frameLeft) \
193 { \
194 ssize_t count, used; \
195 \
196 if (!dmasound.soft.stereo) { \
197 u_char *p = &frame[*frameUsed]; \
198 count = min_t(size_t, userCount, frameLeft) & ~1; \
199 used = count; \
200 while (count > 0) { \
201 u_char data; \
202 if (get_user(data, userPtr++)) \
203 return -EFAULT; \
204 *p++ = convsample(data); \
205 count--; \
206 } \
207 } else { \
208 u_char *left = &frame[*frameUsed>>1]; \
209 u_char *right = left+write_sq_block_size_half; \
210 count = min_t(size_t, userCount, frameLeft)>>1 & ~1; \
211 used = count*2; \
212 while (count > 0) { \
213 u_char data; \
214 if (get_user(data, userPtr++)) \
215 return -EFAULT; \
216 *left++ = convsample(data); \
217 if (get_user(data, userPtr++)) \
218 return -EFAULT; \
219 *right++ = convsample(data); \
220 count--; \
221 } \
222 } \
223 *frameUsed += used; \
224 return used; \
225 }
226
227 #define AMI_CT_ULAW(x) (dmasound_ulaw2dma8[(x)])
228 #define AMI_CT_ALAW(x) (dmasound_alaw2dma8[(x)])
229 #define AMI_CT_U8(x) ((x) ^ 0x80)
230
231 GENERATE_AMI_CT8(ami_ct_ulaw, AMI_CT_ULAW)
232 GENERATE_AMI_CT8(ami_ct_alaw, AMI_CT_ALAW)
233 GENERATE_AMI_CT8(ami_ct_u8, AMI_CT_U8)
234
235
236 /*
237 * Copy and convert 16 bit data
238 */
239
240 #define GENERATE_AMI_CT_16(funcname, convsample) \
241 static ssize_t funcname(const u_char __user *userPtr, size_t userCount, \
242 u_char frame[], ssize_t *frameUsed, \
243 ssize_t frameLeft) \
244 { \
245 const u_short __user *ptr = (const u_short __user *)userPtr; \
246 ssize_t count, used; \
247 u_short data; \
248 \
249 if (!dmasound.soft.stereo) { \
250 u_char *high = &frame[*frameUsed>>1]; \
251 u_char *low = high+write_sq_block_size_half; \
252 count = min_t(size_t, userCount, frameLeft)>>1 & ~1; \
253 used = count*2; \
254 while (count > 0) { \
255 if (get_user(data, ptr++)) \
256 return -EFAULT; \
257 data = convsample(data); \
258 *high++ = data>>8; \
259 *low++ = (data>>2) & 0x3f; \
260 count--; \
261 } \
262 } else { \
263 u_char *lefth = &frame[*frameUsed>>2]; \
264 u_char *leftl = lefth+write_sq_block_size_quarter; \
265 u_char *righth = lefth+write_sq_block_size_half; \
266 u_char *rightl = righth+write_sq_block_size_quarter; \
267 count = min_t(size_t, userCount, frameLeft)>>2 & ~1; \
268 used = count*4; \
269 while (count > 0) { \
270 if (get_user(data, ptr++)) \
271 return -EFAULT; \
272 data = convsample(data); \
273 *lefth++ = data>>8; \
274 *leftl++ = (data>>2) & 0x3f; \
275 if (get_user(data, ptr++)) \
276 return -EFAULT; \
277 data = convsample(data); \
278 *righth++ = data>>8; \
279 *rightl++ = (data>>2) & 0x3f; \
280 count--; \
281 } \
282 } \
283 *frameUsed += used; \
284 return used; \
285 }
286
287 #define AMI_CT_S16BE(x) (x)
288 #define AMI_CT_U16BE(x) ((x) ^ 0x8000)
289 #define AMI_CT_S16LE(x) (le2be16((x)))
290 #define AMI_CT_U16LE(x) (le2be16((x)) ^ 0x8000)
291
292 GENERATE_AMI_CT_16(ami_ct_s16be, AMI_CT_S16BE)
293 GENERATE_AMI_CT_16(ami_ct_u16be, AMI_CT_U16BE)
294 GENERATE_AMI_CT_16(ami_ct_s16le, AMI_CT_S16LE)
295 GENERATE_AMI_CT_16(ami_ct_u16le, AMI_CT_U16LE)
296
297
298 static TRANS transAmiga = {
299 .ct_ulaw = ami_ct_ulaw,
300 .ct_alaw = ami_ct_alaw,
301 .ct_s8 = ami_ct_s8,
302 .ct_u8 = ami_ct_u8,
303 .ct_s16be = ami_ct_s16be,
304 .ct_u16be = ami_ct_u16be,
305 .ct_s16le = ami_ct_s16le,
306 .ct_u16le = ami_ct_u16le,
307 };
308
309 /*** Low level stuff *********************************************************/
310
StopDMA(void)311 static inline void StopDMA(void)
312 {
313 custom.aud[0].audvol = custom.aud[1].audvol = 0;
314 custom.aud[2].audvol = custom.aud[3].audvol = 0;
315 custom.dmacon = AMI_AUDIO_OFF;
316 enable_heartbeat();
317 }
318
AmiAlloc(unsigned int size,gfp_t flags)319 static void *AmiAlloc(unsigned int size, gfp_t flags)
320 {
321 return amiga_chip_alloc((long)size, "dmasound [Paula]");
322 }
323
AmiFree(void * obj,unsigned int size)324 static void AmiFree(void *obj, unsigned int size)
325 {
326 amiga_chip_free (obj);
327 }
328
AmiIrqInit(void)329 static int __init AmiIrqInit(void)
330 {
331 /* turn off DMA for audio channels */
332 StopDMA();
333
334 /* Register interrupt handler. */
335 if (request_irq(IRQ_AMIGA_AUD0, AmiInterrupt, 0, "DMA sound",
336 AmiInterrupt))
337 return 0;
338 return 1;
339 }
340
341 #ifdef MODULE
AmiIrqCleanUp(void)342 static void AmiIrqCleanUp(void)
343 {
344 /* turn off DMA for audio channels */
345 StopDMA();
346 /* release the interrupt */
347 free_irq(IRQ_AMIGA_AUD0, AmiInterrupt);
348 }
349 #endif /* MODULE */
350
AmiSilence(void)351 static void AmiSilence(void)
352 {
353 /* turn off DMA for audio channels */
354 StopDMA();
355 }
356
357
AmiInit(void)358 static void AmiInit(void)
359 {
360 int period, i;
361
362 AmiSilence();
363
364 if (dmasound.soft.speed)
365 period = amiga_colorclock/dmasound.soft.speed-1;
366 else
367 period = amiga_audio_min_period;
368 dmasound.hard = dmasound.soft;
369 dmasound.trans_write = &transAmiga;
370
371 if (period < amiga_audio_min_period) {
372 /* we would need to squeeze the sound, but we won't do that */
373 period = amiga_audio_min_period;
374 } else if (period > 65535) {
375 period = 65535;
376 }
377 dmasound.hard.speed = amiga_colorclock/(period+1);
378
379 for (i = 0; i < 4; i++)
380 custom.aud[i].audper = period;
381 amiga_audio_period = period;
382 }
383
384
AmiSetFormat(int format)385 static int AmiSetFormat(int format)
386 {
387 int size;
388
389 /* Amiga sound DMA supports 8bit and 16bit (pseudo 14 bit) modes */
390
391 switch (format) {
392 case AFMT_QUERY:
393 return dmasound.soft.format;
394 case AFMT_MU_LAW:
395 case AFMT_A_LAW:
396 case AFMT_U8:
397 case AFMT_S8:
398 size = 8;
399 break;
400 case AFMT_S16_BE:
401 case AFMT_U16_BE:
402 case AFMT_S16_LE:
403 case AFMT_U16_LE:
404 size = 16;
405 break;
406 default: /* :-) */
407 size = 8;
408 format = AFMT_S8;
409 }
410
411 dmasound.soft.format = format;
412 dmasound.soft.size = size;
413 if (dmasound.minDev == SND_DEV_DSP) {
414 dmasound.dsp.format = format;
415 dmasound.dsp.size = dmasound.soft.size;
416 }
417 AmiInit();
418
419 return format;
420 }
421
422
423 #define VOLUME_VOXWARE_TO_AMI(v) \
424 (((v) < 0) ? 0 : ((v) > 100) ? 64 : ((v) * 64)/100)
425 #define VOLUME_AMI_TO_VOXWARE(v) ((v)*100/64)
426
AmiSetVolume(int volume)427 static int AmiSetVolume(int volume)
428 {
429 dmasound.volume_left = VOLUME_VOXWARE_TO_AMI(volume & 0xff);
430 custom.aud[0].audvol = dmasound.volume_left;
431 dmasound.volume_right = VOLUME_VOXWARE_TO_AMI((volume & 0xff00) >> 8);
432 custom.aud[1].audvol = dmasound.volume_right;
433 if (dmasound.hard.size == 16) {
434 if (dmasound.volume_left == 64 && dmasound.volume_right == 64) {
435 custom.aud[2].audvol = 1;
436 custom.aud[3].audvol = 1;
437 } else {
438 custom.aud[2].audvol = 0;
439 custom.aud[3].audvol = 0;
440 }
441 }
442 return VOLUME_AMI_TO_VOXWARE(dmasound.volume_left) |
443 (VOLUME_AMI_TO_VOXWARE(dmasound.volume_right) << 8);
444 }
445
AmiSetTreble(int treble)446 static int AmiSetTreble(int treble)
447 {
448 dmasound.treble = treble;
449 if (treble < 50)
450 ciaa.pra &= ~0x02;
451 else
452 ciaa.pra |= 0x02;
453 return treble;
454 }
455
456
457 #define AMI_PLAY_LOADED 1
458 #define AMI_PLAY_PLAYING 2
459 #define AMI_PLAY_MASK 3
460
461
AmiPlayNextFrame(int index)462 static void AmiPlayNextFrame(int index)
463 {
464 u_char *start, *ch0, *ch1, *ch2, *ch3;
465 u_long size;
466
467 /* used by AmiPlay() if all doubts whether there really is something
468 * to be played are already wiped out.
469 */
470 start = write_sq.buffers[write_sq.front];
471 size = (write_sq.count == index ? write_sq.rear_size
472 : write_sq.block_size)>>1;
473
474 if (dmasound.hard.stereo) {
475 ch0 = start;
476 ch1 = start+write_sq_block_size_half;
477 size >>= 1;
478 } else {
479 ch0 = start;
480 ch1 = start;
481 }
482
483 disable_heartbeat();
484 custom.aud[0].audvol = dmasound.volume_left;
485 custom.aud[1].audvol = dmasound.volume_right;
486 if (dmasound.hard.size == 8) {
487 custom.aud[0].audlc = (u_short *)ZTWO_PADDR(ch0);
488 custom.aud[0].audlen = size;
489 custom.aud[1].audlc = (u_short *)ZTWO_PADDR(ch1);
490 custom.aud[1].audlen = size;
491 custom.dmacon = AMI_AUDIO_8;
492 } else {
493 size >>= 1;
494 custom.aud[0].audlc = (u_short *)ZTWO_PADDR(ch0);
495 custom.aud[0].audlen = size;
496 custom.aud[1].audlc = (u_short *)ZTWO_PADDR(ch1);
497 custom.aud[1].audlen = size;
498 if (dmasound.volume_left == 64 && dmasound.volume_right == 64) {
499 /* We can play pseudo 14-bit only with the maximum volume */
500 ch3 = ch0+write_sq_block_size_quarter;
501 ch2 = ch1+write_sq_block_size_quarter;
502 custom.aud[2].audvol = 1; /* we are being affected by the beeps */
503 custom.aud[3].audvol = 1; /* restoring volume here helps a bit */
504 custom.aud[2].audlc = (u_short *)ZTWO_PADDR(ch2);
505 custom.aud[2].audlen = size;
506 custom.aud[3].audlc = (u_short *)ZTWO_PADDR(ch3);
507 custom.aud[3].audlen = size;
508 custom.dmacon = AMI_AUDIO_14;
509 } else {
510 custom.aud[2].audvol = 0;
511 custom.aud[3].audvol = 0;
512 custom.dmacon = AMI_AUDIO_8;
513 }
514 }
515 write_sq.front = (write_sq.front+1) % write_sq.max_count;
516 write_sq.active |= AMI_PLAY_LOADED;
517 }
518
519
AmiPlay(void)520 static void AmiPlay(void)
521 {
522 int minframes = 1;
523
524 custom.intena = IF_AUD0;
525
526 if (write_sq.active & AMI_PLAY_LOADED) {
527 /* There's already a frame loaded */
528 custom.intena = IF_SETCLR | IF_AUD0;
529 return;
530 }
531
532 if (write_sq.active & AMI_PLAY_PLAYING)
533 /* Increase threshold: frame 1 is already being played */
534 minframes = 2;
535
536 if (write_sq.count < minframes) {
537 /* Nothing to do */
538 custom.intena = IF_SETCLR | IF_AUD0;
539 return;
540 }
541
542 if (write_sq.count <= minframes &&
543 write_sq.rear_size < write_sq.block_size && !write_sq.syncing) {
544 /* hmmm, the only existing frame is not
545 * yet filled and we're not syncing?
546 */
547 custom.intena = IF_SETCLR | IF_AUD0;
548 return;
549 }
550
551 AmiPlayNextFrame(minframes);
552
553 custom.intena = IF_SETCLR | IF_AUD0;
554 }
555
556
AmiInterrupt(int irq,void * dummy)557 static irqreturn_t AmiInterrupt(int irq, void *dummy)
558 {
559 int minframes = 1;
560
561 custom.intena = IF_AUD0;
562
563 if (!write_sq.active) {
564 /* Playing was interrupted and sq_reset() has already cleared
565 * the sq variables, so better don't do anything here.
566 */
567 WAKE_UP(write_sq.sync_queue);
568 return IRQ_HANDLED;
569 }
570
571 if (write_sq.active & AMI_PLAY_PLAYING) {
572 /* We've just finished a frame */
573 write_sq.count--;
574 WAKE_UP(write_sq.action_queue);
575 }
576
577 if (write_sq.active & AMI_PLAY_LOADED)
578 /* Increase threshold: frame 1 is already being played */
579 minframes = 2;
580
581 /* Shift the flags */
582 write_sq.active = (write_sq.active<<1) & AMI_PLAY_MASK;
583
584 if (!write_sq.active)
585 /* No frame is playing, disable audio DMA */
586 StopDMA();
587
588 custom.intena = IF_SETCLR | IF_AUD0;
589
590 if (write_sq.count >= minframes)
591 /* Try to play the next frame */
592 AmiPlay();
593
594 if (!write_sq.active)
595 /* Nothing to play anymore.
596 Wake up a process waiting for audio output to drain. */
597 WAKE_UP(write_sq.sync_queue);
598 return IRQ_HANDLED;
599 }
600
601 /*** Mid level stuff *********************************************************/
602
603
604 /*
605 * /dev/mixer abstraction
606 */
607
AmiMixerInit(void)608 static void __init AmiMixerInit(void)
609 {
610 dmasound.volume_left = 64;
611 dmasound.volume_right = 64;
612 custom.aud[0].audvol = dmasound.volume_left;
613 custom.aud[3].audvol = 1; /* For pseudo 14bit */
614 custom.aud[1].audvol = dmasound.volume_right;
615 custom.aud[2].audvol = 1; /* For pseudo 14bit */
616 dmasound.treble = 50;
617 }
618
AmiMixerIoctl(u_int cmd,u_long arg)619 static int AmiMixerIoctl(u_int cmd, u_long arg)
620 {
621 int data;
622 switch (cmd) {
623 case SOUND_MIXER_READ_DEVMASK:
624 return IOCTL_OUT(arg, SOUND_MASK_VOLUME | SOUND_MASK_TREBLE);
625 case SOUND_MIXER_READ_RECMASK:
626 return IOCTL_OUT(arg, 0);
627 case SOUND_MIXER_READ_STEREODEVS:
628 return IOCTL_OUT(arg, SOUND_MASK_VOLUME);
629 case SOUND_MIXER_READ_VOLUME:
630 return IOCTL_OUT(arg,
631 VOLUME_AMI_TO_VOXWARE(dmasound.volume_left) |
632 VOLUME_AMI_TO_VOXWARE(dmasound.volume_right) << 8);
633 case SOUND_MIXER_WRITE_VOLUME:
634 IOCTL_IN(arg, data);
635 return IOCTL_OUT(arg, dmasound_set_volume(data));
636 case SOUND_MIXER_READ_TREBLE:
637 return IOCTL_OUT(arg, dmasound.treble);
638 case SOUND_MIXER_WRITE_TREBLE:
639 IOCTL_IN(arg, data);
640 return IOCTL_OUT(arg, dmasound_set_treble(data));
641 }
642 return -EINVAL;
643 }
644
645
AmiWriteSqSetup(void)646 static int AmiWriteSqSetup(void)
647 {
648 write_sq_block_size_half = write_sq.block_size>>1;
649 write_sq_block_size_quarter = write_sq_block_size_half>>1;
650 return 0;
651 }
652
653
AmiStateInfo(char * buffer,size_t space)654 static int AmiStateInfo(char *buffer, size_t space)
655 {
656 int len = 0;
657 len += sprintf(buffer+len, "\tsound.volume_left = %d [0...64]\n",
658 dmasound.volume_left);
659 len += sprintf(buffer+len, "\tsound.volume_right = %d [0...64]\n",
660 dmasound.volume_right);
661 if (len >= space) {
662 printk(KERN_ERR "dmasound_paula: overflowed state buffer alloc.\n") ;
663 len = space ;
664 }
665 return len;
666 }
667
668
669 /*** Machine definitions *****************************************************/
670
671 static SETTINGS def_hard = {
672 .format = AFMT_S8,
673 .stereo = 0,
674 .size = 8,
675 .speed = 8000
676 } ;
677
678 static SETTINGS def_soft = {
679 .format = AFMT_U8,
680 .stereo = 0,
681 .size = 8,
682 .speed = 8000
683 } ;
684
685 static MACHINE machAmiga = {
686 .name = "Amiga",
687 .name2 = "AMIGA",
688 .owner = THIS_MODULE,
689 .dma_alloc = AmiAlloc,
690 .dma_free = AmiFree,
691 .irqinit = AmiIrqInit,
692 #ifdef MODULE
693 .irqcleanup = AmiIrqCleanUp,
694 #endif /* MODULE */
695 .init = AmiInit,
696 .silence = AmiSilence,
697 .setFormat = AmiSetFormat,
698 .setVolume = AmiSetVolume,
699 .setTreble = AmiSetTreble,
700 .play = AmiPlay,
701 .mixer_init = AmiMixerInit,
702 .mixer_ioctl = AmiMixerIoctl,
703 .write_sq_setup = AmiWriteSqSetup,
704 .state_info = AmiStateInfo,
705 .min_dsp_speed = 8000,
706 .version = ((DMASOUND_PAULA_REVISION<<8) | DMASOUND_PAULA_EDITION),
707 .hardware_afmts = (AFMT_S8 | AFMT_S16_BE), /* h'ware-supported formats *only* here */
708 .capabilities = DSP_CAP_BATCH /* As per SNDCTL_DSP_GETCAPS */
709 };
710
711
712 /*** Config & Setup **********************************************************/
713
714
amiga_audio_probe(struct platform_device * pdev)715 static int __init amiga_audio_probe(struct platform_device *pdev)
716 {
717 dmasound.mach = machAmiga;
718 dmasound.mach.default_hard = def_hard ;
719 dmasound.mach.default_soft = def_soft ;
720 return dmasound_init();
721 }
722
amiga_audio_remove(struct platform_device * pdev)723 static void __exit amiga_audio_remove(struct platform_device *pdev)
724 {
725 dmasound_deinit();
726 }
727
728 /*
729 * amiga_audio_remove() lives in .exit.text. For drivers registered via
730 * module_platform_driver_probe() this is ok because they cannot get unbound at
731 * runtime. So mark the driver struct with __refdata to prevent modpost
732 * triggering a section mismatch warning.
733 */
734 static struct platform_driver amiga_audio_driver __refdata = {
735 .remove_new = __exit_p(amiga_audio_remove),
736 .driver = {
737 .name = "amiga-audio",
738 },
739 };
740
741 module_platform_driver_probe(amiga_audio_driver, amiga_audio_probe);
742
743 MODULE_DESCRIPTION("Amiga Paula DMA Sound Driver");
744 MODULE_LICENSE("GPL");
745 MODULE_ALIAS("platform:amiga-audio");
746