Merge branch 'asoc-5.5' into asoc-linus
[sfrench/cifs-2.6.git] / sound / oss / dmasound / dmasound_paula.c
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
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
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
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
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
319 static void *AmiAlloc(unsigned int size, gfp_t flags)
320 {
321         return amiga_chip_alloc((long)size, "dmasound [Paula]");
322 }
323
324 static void AmiFree(void *obj, unsigned int size)
325 {
326         amiga_chip_free (obj);
327 }
328
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
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
351 static void AmiSilence(void)
352 {
353         /* turn off DMA for audio channels */
354         StopDMA();
355 }
356
357
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
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
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
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
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
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
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
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
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
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
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
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
723 static int __exit amiga_audio_remove(struct platform_device *pdev)
724 {
725         dmasound_deinit();
726         return 0;
727 }
728
729 static struct platform_driver amiga_audio_driver = {
730         .remove = __exit_p(amiga_audio_remove),
731         .driver   = {
732                 .name   = "amiga-audio",
733         },
734 };
735
736 module_platform_driver_probe(amiga_audio_driver, amiga_audio_probe);
737
738 MODULE_LICENSE("GPL");
739 MODULE_ALIAS("platform:amiga-audio");