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dmasound_paula.c
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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/mm.h>
20 #include <linux/init.h>
21 #include <linux/ioport.h>
22 #include <linux/soundcard.h>
23 #include <linux/interrupt.h>
24 #include <linux/platform_device.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 #define custom amiga_custom
38  /*
39  * The minimum period for audio depends on htotal (for OCS/ECS/AGA)
40  * (Imported from arch/m68k/amiga/amisound.c)
41  */
42 
43 extern volatile u_short amiga_audio_min_period;
44 
45 
46  /*
47  * amiga_mksound() should be able to restore the period after beeping
48  * (Imported from arch/m68k/amiga/amisound.c)
49  */
50 
51 extern u_short amiga_audio_period;
52 
53 
54  /*
55  * Audio DMA masks
56  */
57 
58 #define AMI_AUDIO_OFF (DMAF_AUD0 | DMAF_AUD1 | DMAF_AUD2 | DMAF_AUD3)
59 #define AMI_AUDIO_8 (DMAF_SETCLR | DMAF_MASTER | DMAF_AUD0 | DMAF_AUD1)
60 #define AMI_AUDIO_14 (AMI_AUDIO_8 | DMAF_AUD2 | DMAF_AUD3)
61 
62 
63  /*
64  * Helper pointers for 16(14)-bit sound
65  */
66 
67 static int write_sq_block_size_half, write_sq_block_size_quarter;
68 
69 
70 /*** Low level stuff *********************************************************/
71 
72 
73 static void *AmiAlloc(unsigned int size, gfp_t flags);
74 static void AmiFree(void *obj, unsigned int size);
75 static int AmiIrqInit(void);
76 #ifdef MODULE
77 static void AmiIrqCleanUp(void);
78 #endif
79 static void AmiSilence(void);
80 static void AmiInit(void);
81 static int AmiSetFormat(int format);
82 static int AmiSetVolume(int volume);
83 static int AmiSetTreble(int treble);
84 static void AmiPlayNextFrame(int index);
85 static void AmiPlay(void);
86 static irqreturn_t AmiInterrupt(int irq, void *dummy);
87 
88 #ifdef CONFIG_HEARTBEAT
89 
90  /*
91  * Heartbeat interferes with sound since the 7 kHz low-pass filter and the
92  * power LED are controlled by the same line.
93  */
94 
95 static void (*saved_heartbeat)(int) = NULL;
96 
97 static inline void disable_heartbeat(void)
98 {
99  if (mach_heartbeat) {
100  saved_heartbeat = mach_heartbeat;
101  mach_heartbeat = NULL;
102  }
103  AmiSetTreble(dmasound.treble);
104 }
105 
106 static inline void enable_heartbeat(void)
107 {
108  if (saved_heartbeat)
109  mach_heartbeat = saved_heartbeat;
110 }
111 #else /* !CONFIG_HEARTBEAT */
112 #define disable_heartbeat() do { } while (0)
113 #define enable_heartbeat() do { } while (0)
114 #endif /* !CONFIG_HEARTBEAT */
115 
116 
117 /*** Mid level stuff *********************************************************/
118 
119 static void AmiMixerInit(void);
120 static int AmiMixerIoctl(u_int cmd, u_long arg);
121 static int AmiWriteSqSetup(void);
122 static int AmiStateInfo(char *buffer, size_t space);
123 
124 
125 /*** Translations ************************************************************/
126 
127 /* ++TeSche: radically changed for new expanding purposes...
128  *
129  * These two routines now deal with copying/expanding/translating the samples
130  * from user space into our buffer at the right frequency. They take care about
131  * how much data there's actually to read, how much buffer space there is and
132  * to convert samples into the right frequency/encoding. They will only work on
133  * complete samples so it may happen they leave some bytes in the input stream
134  * if the user didn't write a multiple of the current sample size. They both
135  * return the number of bytes they've used from both streams so you may detect
136  * such a situation. Luckily all programs should be able to cope with that.
137  *
138  * I think I've optimized anything as far as one can do in plain C, all
139  * variables should fit in registers and the loops are really short. There's
140  * one loop for every possible situation. Writing a more generalized and thus
141  * parameterized loop would only produce slower code. Feel free to optimize
142  * this in assembler if you like. :)
143  *
144  * I think these routines belong here because they're not yet really hardware
145  * independent, especially the fact that the Falcon can play 16bit samples
146  * only in stereo is hardcoded in both of them!
147  *
148  * ++geert: split in even more functions (one per format)
149  */
150 
151 
152  /*
153  * Native format
154  */
155 
156 static ssize_t ami_ct_s8(const u_char __user *userPtr, size_t userCount,
157  u_char frame[], ssize_t *frameUsed, ssize_t frameLeft)
158 {
159  ssize_t count, used;
160 
161  if (!dmasound.soft.stereo) {
162  void *p = &frame[*frameUsed];
163  count = min_t(unsigned long, userCount, frameLeft) & ~1;
164  used = count;
165  if (copy_from_user(p, userPtr, count))
166  return -EFAULT;
167  } else {
168  u_char *left = &frame[*frameUsed>>1];
169  u_char *right = left+write_sq_block_size_half;
170  count = min_t(unsigned long, userCount, frameLeft)>>1 & ~1;
171  used = count*2;
172  while (count > 0) {
173  if (get_user(*left++, userPtr++)
174  || get_user(*right++, userPtr++))
175  return -EFAULT;
176  count--;
177  }
178  }
179  *frameUsed += used;
180  return used;
181 }
182 
183 
184  /*
185  * Copy and convert 8 bit data
186  */
187 
188 #define GENERATE_AMI_CT8(funcname, convsample) \
189 static ssize_t funcname(const u_char __user *userPtr, size_t userCount, \
190  u_char frame[], ssize_t *frameUsed, \
191  ssize_t frameLeft) \
192 { \
193  ssize_t count, used; \
194  \
195  if (!dmasound.soft.stereo) { \
196  u_char *p = &frame[*frameUsed]; \
197  count = min_t(size_t, userCount, frameLeft) & ~1; \
198  used = count; \
199  while (count > 0) { \
200  u_char data; \
201  if (get_user(data, userPtr++)) \
202  return -EFAULT; \
203  *p++ = convsample(data); \
204  count--; \
205  } \
206  } else { \
207  u_char *left = &frame[*frameUsed>>1]; \
208  u_char *right = left+write_sq_block_size_half; \
209  count = min_t(size_t, userCount, frameLeft)>>1 & ~1; \
210  used = count*2; \
211  while (count > 0) { \
212  u_char data; \
213  if (get_user(data, userPtr++)) \
214  return -EFAULT; \
215  *left++ = convsample(data); \
216  if (get_user(data, userPtr++)) \
217  return -EFAULT; \
218  *right++ = convsample(data); \
219  count--; \
220  } \
221  } \
222  *frameUsed += used; \
223  return used; \
224 }
225 
226 #define AMI_CT_ULAW(x) (dmasound_ulaw2dma8[(x)])
227 #define AMI_CT_ALAW(x) (dmasound_alaw2dma8[(x)])
228 #define AMI_CT_U8(x) ((x) ^ 0x80)
229 
230 GENERATE_AMI_CT8(ami_ct_ulaw, AMI_CT_ULAW)
231 GENERATE_AMI_CT8(ami_ct_alaw, AMI_CT_ALAW)
232 GENERATE_AMI_CT8(ami_ct_u8, AMI_CT_U8)
233 
234 
235  /*
236  * Copy and convert 16 bit data
237  */
238 
239 #define GENERATE_AMI_CT_16(funcname, convsample) \
240 static ssize_t funcname(const u_char __user *userPtr, size_t userCount, \
241  u_char frame[], ssize_t *frameUsed, \
242  ssize_t frameLeft) \
243 { \
244  const u_short __user *ptr = (const u_short __user *)userPtr; \
245  ssize_t count, used; \
246  u_short data; \
247  \
248  if (!dmasound.soft.stereo) { \
249  u_char *high = &frame[*frameUsed>>1]; \
250  u_char *low = high+write_sq_block_size_half; \
251  count = min_t(size_t, userCount, frameLeft)>>1 & ~1; \
252  used = count*2; \
253  while (count > 0) { \
254  if (get_user(data, ptr++)) \
255  return -EFAULT; \
256  data = convsample(data); \
257  *high++ = data>>8; \
258  *low++ = (data>>2) & 0x3f; \
259  count--; \
260  } \
261  } else { \
262  u_char *lefth = &frame[*frameUsed>>2]; \
263  u_char *leftl = lefth+write_sq_block_size_quarter; \
264  u_char *righth = lefth+write_sq_block_size_half; \
265  u_char *rightl = righth+write_sq_block_size_quarter; \
266  count = min_t(size_t, userCount, frameLeft)>>2 & ~1; \
267  used = count*4; \
268  while (count > 0) { \
269  if (get_user(data, ptr++)) \
270  return -EFAULT; \
271  data = convsample(data); \
272  *lefth++ = data>>8; \
273  *leftl++ = (data>>2) & 0x3f; \
274  if (get_user(data, ptr++)) \
275  return -EFAULT; \
276  data = convsample(data); \
277  *righth++ = data>>8; \
278  *rightl++ = (data>>2) & 0x3f; \
279  count--; \
280  } \
281  } \
282  *frameUsed += used; \
283  return used; \
284 }
285 
286 #define AMI_CT_S16BE(x) (x)
287 #define AMI_CT_U16BE(x) ((x) ^ 0x8000)
288 #define AMI_CT_S16LE(x) (le2be16((x)))
289 #define AMI_CT_U16LE(x) (le2be16((x)) ^ 0x8000)
290 
291 GENERATE_AMI_CT_16(ami_ct_s16be, AMI_CT_S16BE)
292 GENERATE_AMI_CT_16(ami_ct_u16be, AMI_CT_U16BE)
293 GENERATE_AMI_CT_16(ami_ct_s16le, AMI_CT_S16LE)
294 GENERATE_AMI_CT_16(ami_ct_u16le, AMI_CT_U16LE)
295 
296 
297 static TRANS transAmiga = {
298  .ct_ulaw = ami_ct_ulaw,
299  .ct_alaw = ami_ct_alaw,
300  .ct_s8 = ami_ct_s8,
301  .ct_u8 = ami_ct_u8,
302  .ct_s16be = ami_ct_s16be,
303  .ct_u16be = ami_ct_u16be,
304  .ct_s16le = ami_ct_s16le,
305  .ct_u16le = ami_ct_u16le,
306 };
307 
308 /*** Low level stuff *********************************************************/
309 
310 static inline void StopDMA(void)
311 {
312  custom.aud[0].audvol = custom.aud[1].audvol = 0;
313  custom.aud[2].audvol = custom.aud[3].audvol = 0;
314  custom.dmacon = AMI_AUDIO_OFF;
315  enable_heartbeat();
316 }
317 
318 static void *AmiAlloc(unsigned int size, gfp_t flags)
319 {
320  return amiga_chip_alloc((long)size, "dmasound [Paula]");
321 }
322 
323 static void AmiFree(void *obj, unsigned int size)
324 {
325  amiga_chip_free (obj);
326 }
327 
328 static int __init AmiIrqInit(void)
329 {
330  /* turn off DMA for audio channels */
331  StopDMA();
332 
333  /* Register interrupt handler. */
334  if (request_irq(IRQ_AMIGA_AUD0, AmiInterrupt, 0, "DMA sound",
335  AmiInterrupt))
336  return 0;
337  return 1;
338 }
339 
340 #ifdef MODULE
341 static void AmiIrqCleanUp(void)
342 {
343  /* turn off DMA for audio channels */
344  StopDMA();
345  /* release the interrupt */
346  free_irq(IRQ_AMIGA_AUD0, AmiInterrupt);
347 }
348 #endif /* MODULE */
349 
350 static void AmiSilence(void)
351 {
352  /* turn off DMA for audio channels */
353  StopDMA();
354 }
355 
356 
357 static void AmiInit(void)
358 {
359  int period, i;
360 
361  AmiSilence();
362 
363  if (dmasound.soft.speed)
364  period = amiga_colorclock/dmasound.soft.speed-1;
365  else
366  period = amiga_audio_min_period;
367  dmasound.hard = dmasound.soft;
368  dmasound.trans_write = &transAmiga;
369 
370  if (period < amiga_audio_min_period) {
371  /* we would need to squeeze the sound, but we won't do that */
372  period = amiga_audio_min_period;
373  } else if (period > 65535) {
374  period = 65535;
375  }
376  dmasound.hard.speed = amiga_colorclock/(period+1);
377 
378  for (i = 0; i < 4; i++)
379  custom.aud[i].audper = period;
380  amiga_audio_period = period;
381 }
382 
383 
384 static int AmiSetFormat(int format)
385 {
386  int size;
387 
388  /* Amiga sound DMA supports 8bit and 16bit (pseudo 14 bit) modes */
389 
390  switch (format) {
391  case AFMT_QUERY:
392  return dmasound.soft.format;
393  case AFMT_MU_LAW:
394  case AFMT_A_LAW:
395  case AFMT_U8:
396  case AFMT_S8:
397  size = 8;
398  break;
399  case AFMT_S16_BE:
400  case AFMT_U16_BE:
401  case AFMT_S16_LE:
402  case AFMT_U16_LE:
403  size = 16;
404  break;
405  default: /* :-) */
406  size = 8;
407  format = AFMT_S8;
408  }
409 
410  dmasound.soft.format = format;
411  dmasound.soft.size = size;
412  if (dmasound.minDev == SND_DEV_DSP) {
413  dmasound.dsp.format = format;
414  dmasound.dsp.size = dmasound.soft.size;
415  }
416  AmiInit();
417 
418  return format;
419 }
420 
421 
422 #define VOLUME_VOXWARE_TO_AMI(v) \
423  (((v) < 0) ? 0 : ((v) > 100) ? 64 : ((v) * 64)/100)
424 #define VOLUME_AMI_TO_VOXWARE(v) ((v)*100/64)
425 
426 static int AmiSetVolume(int volume)
427 {
428  dmasound.volume_left = VOLUME_VOXWARE_TO_AMI(volume & 0xff);
429  custom.aud[0].audvol = dmasound.volume_left;
430  dmasound.volume_right = VOLUME_VOXWARE_TO_AMI((volume & 0xff00) >> 8);
431  custom.aud[1].audvol = dmasound.volume_right;
432  if (dmasound.hard.size == 16) {
433  if (dmasound.volume_left == 64 && dmasound.volume_right == 64) {
434  custom.aud[2].audvol = 1;
435  custom.aud[3].audvol = 1;
436  } else {
437  custom.aud[2].audvol = 0;
438  custom.aud[3].audvol = 0;
439  }
440  }
441  return VOLUME_AMI_TO_VOXWARE(dmasound.volume_left) |
442  (VOLUME_AMI_TO_VOXWARE(dmasound.volume_right) << 8);
443 }
444 
445 static int AmiSetTreble(int treble)
446 {
447  dmasound.treble = treble;
448  if (treble < 50)
449  ciaa.pra &= ~0x02;
450  else
451  ciaa.pra |= 0x02;
452  return treble;
453 }
454 
455 
456 #define AMI_PLAY_LOADED 1
457 #define AMI_PLAY_PLAYING 2
458 #define AMI_PLAY_MASK 3
459 
460 
461 static void AmiPlayNextFrame(int index)
462 {
463  u_char *start, *ch0, *ch1, *ch2, *ch3;
464  u_long size;
465 
466  /* used by AmiPlay() if all doubts whether there really is something
467  * to be played are already wiped out.
468  */
469  start = write_sq.buffers[write_sq.front];
470  size = (write_sq.count == index ? write_sq.rear_size
471  : write_sq.block_size)>>1;
472 
473  if (dmasound.hard.stereo) {
474  ch0 = start;
475  ch1 = start+write_sq_block_size_half;
476  size >>= 1;
477  } else {
478  ch0 = start;
479  ch1 = start;
480  }
481 
482  disable_heartbeat();
483  custom.aud[0].audvol = dmasound.volume_left;
484  custom.aud[1].audvol = dmasound.volume_right;
485  if (dmasound.hard.size == 8) {
486  custom.aud[0].audlc = (u_short *)ZTWO_PADDR(ch0);
487  custom.aud[0].audlen = size;
488  custom.aud[1].audlc = (u_short *)ZTWO_PADDR(ch1);
489  custom.aud[1].audlen = size;
490  custom.dmacon = AMI_AUDIO_8;
491  } else {
492  size >>= 1;
493  custom.aud[0].audlc = (u_short *)ZTWO_PADDR(ch0);
494  custom.aud[0].audlen = size;
495  custom.aud[1].audlc = (u_short *)ZTWO_PADDR(ch1);
496  custom.aud[1].audlen = size;
497  if (dmasound.volume_left == 64 && dmasound.volume_right == 64) {
498  /* We can play pseudo 14-bit only with the maximum volume */
499  ch3 = ch0+write_sq_block_size_quarter;
500  ch2 = ch1+write_sq_block_size_quarter;
501  custom.aud[2].audvol = 1; /* we are being affected by the beeps */
502  custom.aud[3].audvol = 1; /* restoring volume here helps a bit */
503  custom.aud[2].audlc = (u_short *)ZTWO_PADDR(ch2);
504  custom.aud[2].audlen = size;
505  custom.aud[3].audlc = (u_short *)ZTWO_PADDR(ch3);
506  custom.aud[3].audlen = size;
507  custom.dmacon = AMI_AUDIO_14;
508  } else {
509  custom.aud[2].audvol = 0;
510  custom.aud[3].audvol = 0;
511  custom.dmacon = AMI_AUDIO_8;
512  }
513  }
514  write_sq.front = (write_sq.front+1) % write_sq.max_count;
515  write_sq.active |= AMI_PLAY_LOADED;
516 }
517 
518 
519 static void AmiPlay(void)
520 {
521  int minframes = 1;
522 
523  custom.intena = IF_AUD0;
524 
525  if (write_sq.active & AMI_PLAY_LOADED) {
526  /* There's already a frame loaded */
527  custom.intena = IF_SETCLR | IF_AUD0;
528  return;
529  }
530 
531  if (write_sq.active & AMI_PLAY_PLAYING)
532  /* Increase threshold: frame 1 is already being played */
533  minframes = 2;
534 
535  if (write_sq.count < minframes) {
536  /* Nothing to do */
537  custom.intena = IF_SETCLR | IF_AUD0;
538  return;
539  }
540 
541  if (write_sq.count <= minframes &&
542  write_sq.rear_size < write_sq.block_size && !write_sq.syncing) {
543  /* hmmm, the only existing frame is not
544  * yet filled and we're not syncing?
545  */
546  custom.intena = IF_SETCLR | IF_AUD0;
547  return;
548  }
549 
550  AmiPlayNextFrame(minframes);
551 
552  custom.intena = IF_SETCLR | IF_AUD0;
553 }
554 
555 
556 static irqreturn_t AmiInterrupt(int irq, void *dummy)
557 {
558  int minframes = 1;
559 
560  custom.intena = IF_AUD0;
561 
562  if (!write_sq.active) {
563  /* Playing was interrupted and sq_reset() has already cleared
564  * the sq variables, so better don't do anything here.
565  */
566  WAKE_UP(write_sq.sync_queue);
567  return IRQ_HANDLED;
568  }
569 
570  if (write_sq.active & AMI_PLAY_PLAYING) {
571  /* We've just finished a frame */
572  write_sq.count--;
573  WAKE_UP(write_sq.action_queue);
574  }
575 
576  if (write_sq.active & AMI_PLAY_LOADED)
577  /* Increase threshold: frame 1 is already being played */
578  minframes = 2;
579 
580  /* Shift the flags */
581  write_sq.active = (write_sq.active<<1) & AMI_PLAY_MASK;
582 
583  if (!write_sq.active)
584  /* No frame is playing, disable audio DMA */
585  StopDMA();
586 
587  custom.intena = IF_SETCLR | IF_AUD0;
588 
589  if (write_sq.count >= minframes)
590  /* Try to play the next frame */
591  AmiPlay();
592 
593  if (!write_sq.active)
594  /* Nothing to play anymore.
595  Wake up a process waiting for audio output to drain. */
596  WAKE_UP(write_sq.sync_queue);
597  return IRQ_HANDLED;
598 }
599 
600 /*** Mid level stuff *********************************************************/
601 
602 
603 /*
604  * /dev/mixer abstraction
605  */
606 
607 static void __init AmiMixerInit(void)
608 {
609  dmasound.volume_left = 64;
610  dmasound.volume_right = 64;
611  custom.aud[0].audvol = dmasound.volume_left;
612  custom.aud[3].audvol = 1; /* For pseudo 14bit */
613  custom.aud[1].audvol = dmasound.volume_right;
614  custom.aud[2].audvol = 1; /* For pseudo 14bit */
615  dmasound.treble = 50;
616 }
617 
618 static int AmiMixerIoctl(u_int cmd, u_long arg)
619 {
620  int data;
621  switch (cmd) {
622  case SOUND_MIXER_READ_DEVMASK:
623  return IOCTL_OUT(arg, SOUND_MASK_VOLUME | SOUND_MASK_TREBLE);
624  case SOUND_MIXER_READ_RECMASK:
625  return IOCTL_OUT(arg, 0);
626  case SOUND_MIXER_READ_STEREODEVS:
627  return IOCTL_OUT(arg, SOUND_MASK_VOLUME);
628  case SOUND_MIXER_READ_VOLUME:
629  return IOCTL_OUT(arg,
630  VOLUME_AMI_TO_VOXWARE(dmasound.volume_left) |
631  VOLUME_AMI_TO_VOXWARE(dmasound.volume_right) << 8);
632  case SOUND_MIXER_WRITE_VOLUME:
633  IOCTL_IN(arg, data);
634  return IOCTL_OUT(arg, dmasound_set_volume(data));
635  case SOUND_MIXER_READ_TREBLE:
636  return IOCTL_OUT(arg, dmasound.treble);
637  case SOUND_MIXER_WRITE_TREBLE:
638  IOCTL_IN(arg, data);
639  return IOCTL_OUT(arg, dmasound_set_treble(data));
640  }
641  return -EINVAL;
642 }
643 
644 
645 static int AmiWriteSqSetup(void)
646 {
647  write_sq_block_size_half = write_sq.block_size>>1;
648  write_sq_block_size_quarter = write_sq_block_size_half>>1;
649  return 0;
650 }
651 
652 
653 static int AmiStateInfo(char *buffer, size_t space)
654 {
655  int len = 0;
656  len += sprintf(buffer+len, "\tsound.volume_left = %d [0...64]\n",
657  dmasound.volume_left);
658  len += sprintf(buffer+len, "\tsound.volume_right = %d [0...64]\n",
659  dmasound.volume_right);
660  if (len >= space) {
661  printk(KERN_ERR "dmasound_paula: overflowed state buffer alloc.\n") ;
662  len = space ;
663  }
664  return len;
665 }
666 
667 
668 /*** Machine definitions *****************************************************/
669 
670 static SETTINGS def_hard = {
671  .format = AFMT_S8,
672  .stereo = 0,
673  .size = 8,
674  .speed = 8000
675 } ;
676 
677 static SETTINGS def_soft = {
678  .format = AFMT_U8,
679  .stereo = 0,
680  .size = 8,
681  .speed = 8000
682 } ;
683 
684 static MACHINE machAmiga = {
685  .name = "Amiga",
686  .name2 = "AMIGA",
687  .owner = THIS_MODULE,
688  .dma_alloc = AmiAlloc,
689  .dma_free = AmiFree,
690  .irqinit = AmiIrqInit,
691 #ifdef MODULE
692  .irqcleanup = AmiIrqCleanUp,
693 #endif /* MODULE */
694  .init = AmiInit,
695  .silence = AmiSilence,
696  .setFormat = AmiSetFormat,
697  .setVolume = AmiSetVolume,
698  .setTreble = AmiSetTreble,
699  .play = AmiPlay,
700  .mixer_init = AmiMixerInit,
701  .mixer_ioctl = AmiMixerIoctl,
702  .write_sq_setup = AmiWriteSqSetup,
703  .state_info = AmiStateInfo,
704  .min_dsp_speed = 8000,
705  .version = ((DMASOUND_PAULA_REVISION<<8) | DMASOUND_PAULA_EDITION),
706  .hardware_afmts = (AFMT_S8 | AFMT_S16_BE), /* h'ware-supported formats *only* here */
707  .capabilities = DSP_CAP_BATCH /* As per SNDCTL_DSP_GETCAPS */
708 };
709 
710 
711 /*** Config & Setup **********************************************************/
712 
713 
714 static int __init amiga_audio_probe(struct platform_device *pdev)
715 {
716  dmasound.mach = machAmiga;
717  dmasound.mach.default_hard = def_hard ;
718  dmasound.mach.default_soft = def_soft ;
719  return dmasound_init();
720 }
721 
722 static int __exit amiga_audio_remove(struct platform_device *pdev)
723 {
724  dmasound_deinit();
725  return 0;
726 }
727 
728 static struct platform_driver amiga_audio_driver = {
729  .remove = __exit_p(amiga_audio_remove),
730  .driver = {
731  .name = "amiga-audio",
732  .owner = THIS_MODULE,
733  },
734 };
735 
736 static int __init amiga_audio_init(void)
737 {
738  return platform_driver_probe(&amiga_audio_driver, amiga_audio_probe);
739 }
740 
741 module_init(amiga_audio_init);
742 
743 static void __exit amiga_audio_exit(void)
744 {
745  platform_driver_unregister(&amiga_audio_driver);
746 }
747 
748 module_exit(amiga_audio_exit);
749 
750 MODULE_LICENSE("GPL");
751 MODULE_ALIAS("platform:amiga-audio");
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