2 * linux/sound/oss/dmasound/dmasound_paula.c
4 * Amiga `Paula' DMA Sound Driver
6 * See linux/sound/oss/dmasound/dmasound_core.c for copyright and credits
9 * 28/01/2001 [0.1] Iain Sandoe
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
18 #include <linux/module.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>
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>
34 #define DMASOUND_PAULA_REVISION 0
35 #define DMASOUND_PAULA_EDITION 4
37 #define custom amiga_custom
39 * The minimum period for audio depends on htotal (for OCS/ECS/AGA)
40 * (Imported from arch/m68k/amiga/amisound.c)
43 extern volatile u_short amiga_audio_min_period;
47 * amiga_mksound() should be able to restore the period after beeping
48 * (Imported from arch/m68k/amiga/amisound.c)
51 extern u_short amiga_audio_period;
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)
64 * Helper pointers for 16(14)-bit sound
67 static int write_sq_block_size_half, write_sq_block_size_quarter;
70 /*** Low level stuff *********************************************************/
73 static void *AmiAlloc(unsigned int size, gfp_t flags);
74 static void AmiFree(void *obj, unsigned int size);
75 static int AmiIrqInit(void);
77 static void AmiIrqCleanUp(void);
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);
88 #ifdef CONFIG_HEARTBEAT
91 * Heartbeat interferes with sound since the 7 kHz low-pass filter and the
92 * power LED are controlled by the same line.
95 static void (*saved_heartbeat)(int) = NULL;
97 static inline void disable_heartbeat(void)
100 saved_heartbeat = mach_heartbeat;
101 mach_heartbeat = NULL;
103 AmiSetTreble(dmasound.treble);
106 static inline void enable_heartbeat(void)
109 mach_heartbeat = saved_heartbeat;
111 #else /* !CONFIG_HEARTBEAT */
112 #define disable_heartbeat() do { } while (0)
113 #define enable_heartbeat() do { } while (0)
114 #endif /* !CONFIG_HEARTBEAT */
117 /*** Mid level stuff *********************************************************/
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);
125 /*** Translations ************************************************************/
127 /* ++TeSche: radically changed for new expanding purposes...
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.
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. :)
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!
148 * ++geert: split in even more functions (one per format)
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)
161 if (!dmasound.soft.stereo) {
162 void *p = &frame[*frameUsed];
163 count = min_t(unsigned long, userCount, frameLeft) & ~1;
165 if (copy_from_user(p, userPtr, count))
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;
173 if (get_user(*left++, userPtr++)
174 || get_user(*right++, userPtr++))
185 * Copy and convert 8 bit data
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, \
193 ssize_t count, used; \
195 if (!dmasound.soft.stereo) { \
196 u_char *p = &frame[*frameUsed]; \
197 count = min_t(size_t, userCount, frameLeft) & ~1; \
199 while (count > 0) { \
201 if (get_user(data, userPtr++)) \
203 *p++ = convsample(data); \
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; \
211 while (count > 0) { \
213 if (get_user(data, userPtr++)) \
215 *left++ = convsample(data); \
216 if (get_user(data, userPtr++)) \
218 *right++ = convsample(data); \
222 *frameUsed += used; \
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)
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)
236 * Copy and convert 16 bit data
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, \
244 const u_short __user *ptr = (const u_short __user *)userPtr; \
245 ssize_t count, used; \
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; \
253 while (count > 0) { \
254 if (get_user(data, ptr++)) \
256 data = convsample(data); \
258 *low++ = (data>>2) & 0x3f; \
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; \
268 while (count > 0) { \
269 if (get_user(data, ptr++)) \
271 data = convsample(data); \
272 *lefth++ = data>>8; \
273 *leftl++ = (data>>2) & 0x3f; \
274 if (get_user(data, ptr++)) \
276 data = convsample(data); \
277 *righth++ = data>>8; \
278 *rightl++ = (data>>2) & 0x3f; \
282 *frameUsed += used; \
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)
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)
297 static TRANS transAmiga = {
298 .ct_ulaw = ami_ct_ulaw,
299 .ct_alaw = ami_ct_alaw,
302 .ct_s16be = ami_ct_s16be,
303 .ct_u16be = ami_ct_u16be,
304 .ct_s16le = ami_ct_s16le,
305 .ct_u16le = ami_ct_u16le,
308 /*** Low level stuff *********************************************************/
310 static inline void StopDMA(void)
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;
318 static void *AmiAlloc(unsigned int size, gfp_t flags)
320 return amiga_chip_alloc((long)size, "dmasound [Paula]");
323 static void AmiFree(void *obj, unsigned int size)
325 amiga_chip_free (obj);
328 static int __init AmiIrqInit(void)
330 /* turn off DMA for audio channels */
333 /* Register interrupt handler. */
334 if (request_irq(IRQ_AMIGA_AUD0, AmiInterrupt, 0, "DMA sound",
341 static void AmiIrqCleanUp(void)
343 /* turn off DMA for audio channels */
345 /* release the interrupt */
346 free_irq(IRQ_AMIGA_AUD0, AmiInterrupt);
350 static void AmiSilence(void)
352 /* turn off DMA for audio channels */
357 static void AmiInit(void)
363 if (dmasound.soft.speed)
364 period = amiga_colorclock/dmasound.soft.speed-1;
366 period = amiga_audio_min_period;
367 dmasound.hard = dmasound.soft;
368 dmasound.trans_write = &transAmiga;
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) {
376 dmasound.hard.speed = amiga_colorclock/(period+1);
378 for (i = 0; i < 4; i++)
379 custom.aud[i].audper = period;
380 amiga_audio_period = period;
384 static int AmiSetFormat(int format)
388 /* Amiga sound DMA supports 8bit and 16bit (pseudo 14 bit) modes */
392 return dmasound.soft.format;
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;
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)
426 static int AmiSetVolume(int volume)
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;
437 custom.aud[2].audvol = 0;
438 custom.aud[3].audvol = 0;
441 return VOLUME_AMI_TO_VOXWARE(dmasound.volume_left) |
442 (VOLUME_AMI_TO_VOXWARE(dmasound.volume_right) << 8);
445 static int AmiSetTreble(int treble)
447 dmasound.treble = treble;
456 #define AMI_PLAY_LOADED 1
457 #define AMI_PLAY_PLAYING 2
458 #define AMI_PLAY_MASK 3
461 static void AmiPlayNextFrame(int index)
463 u_char *start, *ch0, *ch1, *ch2, *ch3;
466 /* used by AmiPlay() if all doubts whether there really is something
467 * to be played are already wiped out.
469 start = write_sq.buffers[write_sq.front];
470 size = (write_sq.count == index ? write_sq.rear_size
471 : write_sq.block_size)>>1;
473 if (dmasound.hard.stereo) {
475 ch1 = start+write_sq_block_size_half;
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;
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;
509 custom.aud[2].audvol = 0;
510 custom.aud[3].audvol = 0;
511 custom.dmacon = AMI_AUDIO_8;
514 write_sq.front = (write_sq.front+1) % write_sq.max_count;
515 write_sq.active |= AMI_PLAY_LOADED;
519 static void AmiPlay(void)
523 custom.intena = IF_AUD0;
525 if (write_sq.active & AMI_PLAY_LOADED) {
526 /* There's already a frame loaded */
527 custom.intena = IF_SETCLR | IF_AUD0;
531 if (write_sq.active & AMI_PLAY_PLAYING)
532 /* Increase threshold: frame 1 is already being played */
535 if (write_sq.count < minframes) {
537 custom.intena = IF_SETCLR | IF_AUD0;
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?
546 custom.intena = IF_SETCLR | IF_AUD0;
550 AmiPlayNextFrame(minframes);
552 custom.intena = IF_SETCLR | IF_AUD0;
556 static irqreturn_t AmiInterrupt(int irq, void *dummy)
560 custom.intena = IF_AUD0;
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.
566 WAKE_UP(write_sq.sync_queue);
570 if (write_sq.active & AMI_PLAY_PLAYING) {
571 /* We've just finished a frame */
573 WAKE_UP(write_sq.action_queue);
576 if (write_sq.active & AMI_PLAY_LOADED)
577 /* Increase threshold: frame 1 is already being played */
580 /* Shift the flags */
581 write_sq.active = (write_sq.active<<1) & AMI_PLAY_MASK;
583 if (!write_sq.active)
584 /* No frame is playing, disable audio DMA */
587 custom.intena = IF_SETCLR | IF_AUD0;
589 if (write_sq.count >= minframes)
590 /* Try to play the next frame */
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);
600 /*** Mid level stuff *********************************************************/
604 * /dev/mixer abstraction
607 static void __init AmiMixerInit(void)
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;
618 static int AmiMixerIoctl(u_int cmd, u_long arg)
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:
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:
639 return IOCTL_OUT(arg, dmasound_set_treble(data));
645 static int AmiWriteSqSetup(void)
647 write_sq_block_size_half = write_sq.block_size>>1;
648 write_sq_block_size_quarter = write_sq_block_size_half>>1;
653 static int AmiStateInfo(char *buffer, size_t space)
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);
661 printk(KERN_ERR "dmasound_paula: overflowed state buffer alloc.\n") ;
668 /*** Machine definitions *****************************************************/
670 static SETTINGS def_hard = {
677 static SETTINGS def_soft = {
684 static MACHINE machAmiga = {
687 .owner = THIS_MODULE,
688 .dma_alloc = AmiAlloc,
690 .irqinit = AmiIrqInit,
692 .irqcleanup = AmiIrqCleanUp,
695 .silence = AmiSilence,
696 .setFormat = AmiSetFormat,
697 .setVolume = AmiSetVolume,
698 .setTreble = AmiSetTreble,
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 */
711 /*** Config & Setup **********************************************************/
714 static int __init amiga_audio_probe(struct platform_device *pdev)
716 dmasound.mach = machAmiga;
717 dmasound.mach.default_hard = def_hard ;
718 dmasound.mach.default_soft = def_soft ;
719 return dmasound_init();
722 static int __exit amiga_audio_remove(struct platform_device *pdev)
728 static struct platform_driver amiga_audio_driver = {
729 .remove = __exit_p(amiga_audio_remove),
731 .name = "amiga-audio",
735 module_platform_driver_probe(amiga_audio_driver, amiga_audio_probe);
737 MODULE_LICENSE("GPL");
738 MODULE_ALIAS("platform:amiga-audio");