?  search:
summary | log | graphiclog1 | graphiclog2 | commit | commitdiff | tree | refs | edit | fork
blame | history | raw | HEAD
Linux 4.16.11
[linux/fpc-iii.git] / sound / mips / sgio2audio.c
blob9fb68b35de5a9cddb9ed47f5f824f947c295c890
1 /*
2 * Sound driver for Silicon Graphics O2 Workstations A/V board audio.
3 *
4 * Copyright 2003 Vivien Chappelier <vivien.chappelier@linux-mips.org>
5 * Copyright 2008 Thomas Bogendoerfer <tsbogend@alpha.franken.de>
6 * Mxier part taken from mace_audio.c:
7 * Copyright 2007 Thorben Jändling <tj.trevelyan@gmail.com>
8 *
9 * This program is free software; you can redistribute it and/or modify
10 * it under the terms of the GNU General Public License as published by
11 * the Free Software Foundation; either version 2 of the License, or
12 * (at your option) any later version.
13 *
14 * This program is distributed in the hope that it will be useful,
15 * but WITHOUT ANY WARRANTY; without even the implied warranty of
16 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
17 * GNU General Public License for more details.
18 *
19 * You should have received a copy of the GNU General Public License
20 * along with this program; if not, write to the Free Software
21 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
22 *
23 */
24
25 #include <linux/init.h>
26 #include <linux/delay.h>
27 #include <linux/spinlock.h>
28 #include <linux/interrupt.h>
29 #include <linux/dma-mapping.h>
30 #include <linux/platform_device.h>
31 #include <linux/io.h>
32 #include <linux/slab.h>
33 #include <linux/module.h>
34
35 #include <asm/ip32/ip32_ints.h>
36 #include <asm/ip32/mace.h>
37
38 #include <sound/core.h>
39 #include <sound/control.h>
40 #include <sound/pcm.h>
41 #define SNDRV_GET_ID
42 #include <sound/initval.h>
43 #include <sound/ad1843.h>
44
45
46 MODULE_AUTHOR("Vivien Chappelier <vivien.chappelier@linux-mips.org>");
47 MODULE_DESCRIPTION("SGI O2 Audio");
48 MODULE_LICENSE("GPL");
49 MODULE_SUPPORTED_DEVICE("{{Silicon Graphics, O2 Audio}}");
50
51 static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */
52 static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */
53
54 module_param(index, int, 0444);
55 MODULE_PARM_DESC(index, "Index value for SGI O2 soundcard.");
56 module_param(id, charp, 0444);
57 MODULE_PARM_DESC(id, "ID string for SGI O2 soundcard.");
58
59
60 #define AUDIO_CONTROL_RESET BIT(0) /* 1: reset audio interface */
61 #define AUDIO_CONTROL_CODEC_PRESENT BIT(1) /* 1: codec detected */
62
63 #define CODEC_CONTROL_WORD_SHIFT 0
64 #define CODEC_CONTROL_READ BIT(16)
65 #define CODEC_CONTROL_ADDRESS_SHIFT 17
66
67 #define CHANNEL_CONTROL_RESET BIT(10) /* 1: reset channel */
68 #define CHANNEL_DMA_ENABLE BIT(9) /* 1: enable DMA transfer */
69 #define CHANNEL_INT_THRESHOLD_DISABLED (0 << 5) /* interrupt disabled */
70 #define CHANNEL_INT_THRESHOLD_25 (1 << 5) /* int on buffer >25% full */
71 #define CHANNEL_INT_THRESHOLD_50 (2 << 5) /* int on buffer >50% full */
72 #define CHANNEL_INT_THRESHOLD_75 (3 << 5) /* int on buffer >75% full */
73 #define CHANNEL_INT_THRESHOLD_EMPTY (4 << 5) /* int on buffer empty */
74 #define CHANNEL_INT_THRESHOLD_NOT_EMPTY (5 << 5) /* int on buffer !empty */
75 #define CHANNEL_INT_THRESHOLD_FULL (6 << 5) /* int on buffer empty */
76 #define CHANNEL_INT_THRESHOLD_NOT_FULL (7 << 5) /* int on buffer !empty */
77
78 #define CHANNEL_RING_SHIFT 12
79 #define CHANNEL_RING_SIZE (1 << CHANNEL_RING_SHIFT)
80 #define CHANNEL_RING_MASK (CHANNEL_RING_SIZE - 1)
81
82 #define CHANNEL_LEFT_SHIFT 40
83 #define CHANNEL_RIGHT_SHIFT 8
84
85 struct snd_sgio2audio_chan {
86 int idx;
87 struct snd_pcm_substream *substream;
88 int pos;
89 snd_pcm_uframes_t size;
90 spinlock_t lock;
91 };
92
93 /* definition of the chip-specific record */
94 struct snd_sgio2audio {
95 struct snd_card *card;
96
97 /* codec */
98 struct snd_ad1843 ad1843;
99 spinlock_t ad1843_lock;
100
101 /* channels */
102 struct snd_sgio2audio_chan channel[3];
103
104 /* resources */
105 void *ring_base;
106 dma_addr_t ring_base_dma;
107 };
108
109 /* AD1843 access */
110
111 /*
112 * read_ad1843_reg returns the current contents of a 16 bit AD1843 register.
113 *
114 * Returns unsigned register value on success, -errno on failure.
115 */
116 static int read_ad1843_reg(void *priv, int reg)
117 {
118 struct snd_sgio2audio *chip = priv;
119 int val;
120 unsigned long flags;
121
122 spin_lock_irqsave(&chip->ad1843_lock, flags);
123
124 writeq((reg << CODEC_CONTROL_ADDRESS_SHIFT) |
125 CODEC_CONTROL_READ, &mace->perif.audio.codec_control);
126 wmb();
127 val = readq(&mace->perif.audio.codec_control); /* flush bus */
128 udelay(200);
129
130 val = readq(&mace->perif.audio.codec_read);
131
132 spin_unlock_irqrestore(&chip->ad1843_lock, flags);
133 return val;
134 }
135
136 /*
137 * write_ad1843_reg writes the specified value to a 16 bit AD1843 register.
138 */
139 static int write_ad1843_reg(void *priv, int reg, int word)
140 {
141 struct snd_sgio2audio *chip = priv;
142 int val;
143 unsigned long flags;
144
145 spin_lock_irqsave(&chip->ad1843_lock, flags);
146
147 writeq((reg << CODEC_CONTROL_ADDRESS_SHIFT) |
148 (word << CODEC_CONTROL_WORD_SHIFT),
149 &mace->perif.audio.codec_control);
150 wmb();
151 val = readq(&mace->perif.audio.codec_control); /* flush bus */
152 udelay(200);
153
154 spin_unlock_irqrestore(&chip->ad1843_lock, flags);
155 return 0;
156 }
157
158 static int sgio2audio_gain_info(struct snd_kcontrol *kcontrol,
159 struct snd_ctl_elem_info *uinfo)
160 {
161 struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
162
163 uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
164 uinfo->count = 2;
165 uinfo->value.integer.min = 0;
166 uinfo->value.integer.max = ad1843_get_gain_max(&chip->ad1843,
167 (int)kcontrol->private_value);
168 return 0;
169 }
170
171 static int sgio2audio_gain_get(struct snd_kcontrol *kcontrol,
172 struct snd_ctl_elem_value *ucontrol)
173 {
174 struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
175 int vol;
176
177 vol = ad1843_get_gain(&chip->ad1843, (int)kcontrol->private_value);
178
179 ucontrol->value.integer.value[0] = (vol >> 8) & 0xFF;
180 ucontrol->value.integer.value[1] = vol & 0xFF;
181
182 return 0;
183 }
184
185 static int sgio2audio_gain_put(struct snd_kcontrol *kcontrol,
186 struct snd_ctl_elem_value *ucontrol)
187 {
188 struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
189 int newvol, oldvol;
190
191 oldvol = ad1843_get_gain(&chip->ad1843, kcontrol->private_value);
192 newvol = (ucontrol->value.integer.value[0] << 8) |
193 ucontrol->value.integer.value[1];
194
195 newvol = ad1843_set_gain(&chip->ad1843, kcontrol->private_value,
196 newvol);
197
198 return newvol != oldvol;
199 }
200
201 static int sgio2audio_source_info(struct snd_kcontrol *kcontrol,
202 struct snd_ctl_elem_info *uinfo)
203 {
204 static const char * const texts[3] = {
205 "Cam Mic", "Mic", "Line"
206 };
207 return snd_ctl_enum_info(uinfo, 1, 3, texts);
208 }
209
210 static int sgio2audio_source_get(struct snd_kcontrol *kcontrol,
211 struct snd_ctl_elem_value *ucontrol)
212 {
213 struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
214
215 ucontrol->value.enumerated.item[0] = ad1843_get_recsrc(&chip->ad1843);
216 return 0;
217 }
218
219 static int sgio2audio_source_put(struct snd_kcontrol *kcontrol,
220 struct snd_ctl_elem_value *ucontrol)
221 {
222 struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
223 int newsrc, oldsrc;
224
225 oldsrc = ad1843_get_recsrc(&chip->ad1843);
226 newsrc = ad1843_set_recsrc(&chip->ad1843,
227 ucontrol->value.enumerated.item[0]);
228
229 return newsrc != oldsrc;
230 }
231
232 /* dac1/pcm0 mixer control */
233 static const struct snd_kcontrol_new sgio2audio_ctrl_pcm0 = {
234 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
235 .name = "PCM Playback Volume",
236 .index = 0,
237 .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
238 .private_value = AD1843_GAIN_PCM_0,
239 .info = sgio2audio_gain_info,
240 .get = sgio2audio_gain_get,
241 .put = sgio2audio_gain_put,
242 };
243
244 /* dac2/pcm1 mixer control */
245 static const struct snd_kcontrol_new sgio2audio_ctrl_pcm1 = {
246 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
247 .name = "PCM Playback Volume",
248 .index = 1,
249 .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
250 .private_value = AD1843_GAIN_PCM_1,
251 .info = sgio2audio_gain_info,
252 .get = sgio2audio_gain_get,
253 .put = sgio2audio_gain_put,
254 };
255
256 /* record level mixer control */
257 static const struct snd_kcontrol_new sgio2audio_ctrl_reclevel = {
258 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
259 .name = "Capture Volume",
260 .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
261 .private_value = AD1843_GAIN_RECLEV,
262 .info = sgio2audio_gain_info,
263 .get = sgio2audio_gain_get,
264 .put = sgio2audio_gain_put,
265 };
266
267 /* record level source control */
268 static const struct snd_kcontrol_new sgio2audio_ctrl_recsource = {
269 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
270 .name = "Capture Source",
271 .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
272 .info = sgio2audio_source_info,
273 .get = sgio2audio_source_get,
274 .put = sgio2audio_source_put,
275 };
276
277 /* line mixer control */
278 static const struct snd_kcontrol_new sgio2audio_ctrl_line = {
279 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
280 .name = "Line Playback Volume",
281 .index = 0,
282 .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
283 .private_value = AD1843_GAIN_LINE,
284 .info = sgio2audio_gain_info,
285 .get = sgio2audio_gain_get,
286 .put = sgio2audio_gain_put,
287 };
288
289 /* cd mixer control */
290 static const struct snd_kcontrol_new sgio2audio_ctrl_cd = {
291 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
292 .name = "Line Playback Volume",
293 .index = 1,
294 .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
295 .private_value = AD1843_GAIN_LINE_2,
296 .info = sgio2audio_gain_info,
297 .get = sgio2audio_gain_get,
298 .put = sgio2audio_gain_put,
299 };
300
301 /* mic mixer control */
302 static const struct snd_kcontrol_new sgio2audio_ctrl_mic = {
303 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
304 .name = "Mic Playback Volume",
305 .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
306 .private_value = AD1843_GAIN_MIC,
307 .info = sgio2audio_gain_info,
308 .get = sgio2audio_gain_get,
309 .put = sgio2audio_gain_put,
310 };
311
312
313 static int snd_sgio2audio_new_mixer(struct snd_sgio2audio *chip)
314 {
315 int err;
316
317 err = snd_ctl_add(chip->card,
318 snd_ctl_new1(&sgio2audio_ctrl_pcm0, chip));
319 if (err < 0)
320 return err;
321
322 err = snd_ctl_add(chip->card,
323 snd_ctl_new1(&sgio2audio_ctrl_pcm1, chip));
324 if (err < 0)
325 return err;
326
327 err = snd_ctl_add(chip->card,
328 snd_ctl_new1(&sgio2audio_ctrl_reclevel, chip));
329 if (err < 0)
330 return err;
331
332 err = snd_ctl_add(chip->card,
333 snd_ctl_new1(&sgio2audio_ctrl_recsource, chip));
334 if (err < 0)
335 return err;
336 err = snd_ctl_add(chip->card,
337 snd_ctl_new1(&sgio2audio_ctrl_line, chip));
338 if (err < 0)
339 return err;
340
341 err = snd_ctl_add(chip->card,
342 snd_ctl_new1(&sgio2audio_ctrl_cd, chip));
343 if (err < 0)
344 return err;
345
346 err = snd_ctl_add(chip->card,
347 snd_ctl_new1(&sgio2audio_ctrl_mic, chip));
348 if (err < 0)
349 return err;
350
351 return 0;
352 }
353
354 /* low-level audio interface DMA */
355
356 /* get data out of bounce buffer, count must be a multiple of 32 */
357 /* returns 1 if a period has elapsed */
358 static int snd_sgio2audio_dma_pull_frag(struct snd_sgio2audio *chip,
359 unsigned int ch, unsigned int count)
360 {
361 int ret;
362 unsigned long src_base, src_pos, dst_mask;
363 unsigned char *dst_base;
364 int dst_pos;
365 u64 *src;
366 s16 *dst;
367 u64 x;
368 unsigned long flags;
369 struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime;
370
371 spin_lock_irqsave(&chip->channel[ch].lock, flags);
372
373 src_base = (unsigned long) chip->ring_base | (ch << CHANNEL_RING_SHIFT);
374 src_pos = readq(&mace->perif.audio.chan[ch].read_ptr);
375 dst_base = runtime->dma_area;
376 dst_pos = chip->channel[ch].pos;
377 dst_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1;
378
379 /* check if a period has elapsed */
380 chip->channel[ch].size += (count >> 3); /* in frames */
381 ret = chip->channel[ch].size >= runtime->period_size;
382 chip->channel[ch].size %= runtime->period_size;
383
384 while (count) {
385 src = (u64 *)(src_base + src_pos);
386 dst = (s16 *)(dst_base + dst_pos);
387
388 x = *src;
389 dst[0] = (x >> CHANNEL_LEFT_SHIFT) & 0xffff;
390 dst[1] = (x >> CHANNEL_RIGHT_SHIFT) & 0xffff;
391
392 src_pos = (src_pos + sizeof(u64)) & CHANNEL_RING_MASK;
393 dst_pos = (dst_pos + 2 * sizeof(s16)) & dst_mask;
394 count -= sizeof(u64);
395 }
396
397 writeq(src_pos, &mace->perif.audio.chan[ch].read_ptr); /* in bytes */
398 chip->channel[ch].pos = dst_pos;
399
400 spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
401 return ret;
402 }
403
404 /* put some DMA data in bounce buffer, count must be a multiple of 32 */
405 /* returns 1 if a period has elapsed */
406 static int snd_sgio2audio_dma_push_frag(struct snd_sgio2audio *chip,
407 unsigned int ch, unsigned int count)
408 {
409 int ret;
410 s64 l, r;
411 unsigned long dst_base, dst_pos, src_mask;
412 unsigned char *src_base;
413 int src_pos;
414 u64 *dst;
415 s16 *src;
416 unsigned long flags;
417 struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime;
418
419 spin_lock_irqsave(&chip->channel[ch].lock, flags);
420
421 dst_base = (unsigned long)chip->ring_base | (ch << CHANNEL_RING_SHIFT);
422 dst_pos = readq(&mace->perif.audio.chan[ch].write_ptr);
423 src_base = runtime->dma_area;
424 src_pos = chip->channel[ch].pos;
425 src_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1;
426
427 /* check if a period has elapsed */
428 chip->channel[ch].size += (count >> 3); /* in frames */
429 ret = chip->channel[ch].size >= runtime->period_size;
430 chip->channel[ch].size %= runtime->period_size;
431
432 while (count) {
433 src = (s16 *)(src_base + src_pos);
434 dst = (u64 *)(dst_base + dst_pos);
435
436 l = src[0]; /* sign extend */
437 r = src[1]; /* sign extend */
438
439 *dst = ((l & 0x00ffffff) << CHANNEL_LEFT_SHIFT) |
440 ((r & 0x00ffffff) << CHANNEL_RIGHT_SHIFT);
441
442 dst_pos = (dst_pos + sizeof(u64)) & CHANNEL_RING_MASK;
443 src_pos = (src_pos + 2 * sizeof(s16)) & src_mask;
444 count -= sizeof(u64);
445 }
446
447 writeq(dst_pos, &mace->perif.audio.chan[ch].write_ptr); /* in bytes */
448 chip->channel[ch].pos = src_pos;
449
450 spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
451 return ret;
452 }
453
454 static int snd_sgio2audio_dma_start(struct snd_pcm_substream *substream)
455 {
456 struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
457 struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
458 int ch = chan->idx;
459
460 /* reset DMA channel */
461 writeq(CHANNEL_CONTROL_RESET, &mace->perif.audio.chan[ch].control);
462 udelay(10);
463 writeq(0, &mace->perif.audio.chan[ch].control);
464
465 if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
466 /* push a full buffer */
467 snd_sgio2audio_dma_push_frag(chip, ch, CHANNEL_RING_SIZE - 32);
468 }
469 /* set DMA to wake on 50% empty and enable interrupt */
470 writeq(CHANNEL_DMA_ENABLE | CHANNEL_INT_THRESHOLD_50,
471 &mace->perif.audio.chan[ch].control);
472 return 0;
473 }
474
475 static int snd_sgio2audio_dma_stop(struct snd_pcm_substream *substream)
476 {
477 struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
478
479 writeq(0, &mace->perif.audio.chan[chan->idx].control);
480 return 0;
481 }
482
483 static irqreturn_t snd_sgio2audio_dma_in_isr(int irq, void *dev_id)
484 {
485 struct snd_sgio2audio_chan *chan = dev_id;
486 struct snd_pcm_substream *substream;
487 struct snd_sgio2audio *chip;
488 int count, ch;
489
490 substream = chan->substream;
491 chip = snd_pcm_substream_chip(substream);
492 ch = chan->idx;
493
494 /* empty the ring */
495 count = CHANNEL_RING_SIZE -
496 readq(&mace->perif.audio.chan[ch].depth) - 32;
497 if (snd_sgio2audio_dma_pull_frag(chip, ch, count))
498 snd_pcm_period_elapsed(substream);
499
500 return IRQ_HANDLED;
501 }
502
503 static irqreturn_t snd_sgio2audio_dma_out_isr(int irq, void *dev_id)
504 {
505 struct snd_sgio2audio_chan *chan = dev_id;
506 struct snd_pcm_substream *substream;
507 struct snd_sgio2audio *chip;
508 int count, ch;
509
510 substream = chan->substream;
511 chip = snd_pcm_substream_chip(substream);
512 ch = chan->idx;
513 /* fill the ring */
514 count = CHANNEL_RING_SIZE -
515 readq(&mace->perif.audio.chan[ch].depth) - 32;
516 if (snd_sgio2audio_dma_push_frag(chip, ch, count))
517 snd_pcm_period_elapsed(substream);
518
519 return IRQ_HANDLED;
520 }
521
522 static irqreturn_t snd_sgio2audio_error_isr(int irq, void *dev_id)
523 {
524 struct snd_sgio2audio_chan *chan = dev_id;
525 struct snd_pcm_substream *substream;
526
527 substream = chan->substream;
528 snd_sgio2audio_dma_stop(substream);
529 snd_sgio2audio_dma_start(substream);
530 return IRQ_HANDLED;
531 }
532
533 /* PCM part */
534 /* PCM hardware definition */
535 static const struct snd_pcm_hardware snd_sgio2audio_pcm_hw = {
536 .info = (SNDRV_PCM_INFO_MMAP |
537 SNDRV_PCM_INFO_MMAP_VALID |
538 SNDRV_PCM_INFO_INTERLEAVED |
539 SNDRV_PCM_INFO_BLOCK_TRANSFER),
540 .formats = SNDRV_PCM_FMTBIT_S16_BE,
541 .rates = SNDRV_PCM_RATE_8000_48000,
542 .rate_min = 8000,
543 .rate_max = 48000,
544 .channels_min = 2,
545 .channels_max = 2,
546 .buffer_bytes_max = 65536,
547 .period_bytes_min = 32768,
548 .period_bytes_max = 65536,
549 .periods_min = 1,
550 .periods_max = 1024,
551 };
552
553 /* PCM playback open callback */
554 static int snd_sgio2audio_playback1_open(struct snd_pcm_substream *substream)
555 {
556 struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
557 struct snd_pcm_runtime *runtime = substream->runtime;
558
559 runtime->hw = snd_sgio2audio_pcm_hw;
560 runtime->private_data = &chip->channel[1];
561 return 0;
562 }
563
564 static int snd_sgio2audio_playback2_open(struct snd_pcm_substream *substream)
565 {
566 struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
567 struct snd_pcm_runtime *runtime = substream->runtime;
568
569 runtime->hw = snd_sgio2audio_pcm_hw;
570 runtime->private_data = &chip->channel[2];
571 return 0;
572 }
573
574 /* PCM capture open callback */
575 static int snd_sgio2audio_capture_open(struct snd_pcm_substream *substream)
576 {
577 struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
578 struct snd_pcm_runtime *runtime = substream->runtime;
579
580 runtime->hw = snd_sgio2audio_pcm_hw;
581 runtime->private_data = &chip->channel[0];
582 return 0;
583 }
584
585 /* PCM close callback */
586 static int snd_sgio2audio_pcm_close(struct snd_pcm_substream *substream)
587 {
588 struct snd_pcm_runtime *runtime = substream->runtime;
589
590 runtime->private_data = NULL;
591 return 0;
592 }
593
594
595 /* hw_params callback */
596 static int snd_sgio2audio_pcm_hw_params(struct snd_pcm_substream *substream,
597 struct snd_pcm_hw_params *hw_params)
598 {
599 return snd_pcm_lib_alloc_vmalloc_buffer(substream,
600 params_buffer_bytes(hw_params));
601 }
602
603 /* hw_free callback */
604 static int snd_sgio2audio_pcm_hw_free(struct snd_pcm_substream *substream)
605 {
606 return snd_pcm_lib_free_vmalloc_buffer(substream);
607 }
608
609 /* prepare callback */
610 static int snd_sgio2audio_pcm_prepare(struct snd_pcm_substream *substream)
611 {
612 struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
613 struct snd_pcm_runtime *runtime = substream->runtime;
614 struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
615 int ch = chan->idx;
616 unsigned long flags;
617
618 spin_lock_irqsave(&chip->channel[ch].lock, flags);
619
620 /* Setup the pseudo-dma transfer pointers. */
621 chip->channel[ch].pos = 0;
622 chip->channel[ch].size = 0;
623 chip->channel[ch].substream = substream;
624
625 /* set AD1843 format */
626 /* hardware format is always S16_LE */
627 switch (substream->stream) {
628 case SNDRV_PCM_STREAM_PLAYBACK:
629 ad1843_setup_dac(&chip->ad1843,
630 ch - 1,
631 runtime->rate,
632 SNDRV_PCM_FORMAT_S16_LE,
633 runtime->channels);
634 break;
635 case SNDRV_PCM_STREAM_CAPTURE:
636 ad1843_setup_adc(&chip->ad1843,
637 runtime->rate,
638 SNDRV_PCM_FORMAT_S16_LE,
639 runtime->channels);
640 break;
641 }
642 spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
643 return 0;
644 }
645
646 /* trigger callback */
647 static int snd_sgio2audio_pcm_trigger(struct snd_pcm_substream *substream,
648 int cmd)
649 {
650 switch (cmd) {
651 case SNDRV_PCM_TRIGGER_START:
652 /* start the PCM engine */
653 snd_sgio2audio_dma_start(substream);
654 break;
655 case SNDRV_PCM_TRIGGER_STOP:
656 /* stop the PCM engine */
657 snd_sgio2audio_dma_stop(substream);
658 break;
659 default:
660 return -EINVAL;
661 }
662 return 0;
663 }
664
665 /* pointer callback */
666 static snd_pcm_uframes_t
667 snd_sgio2audio_pcm_pointer(struct snd_pcm_substream *substream)
668 {
669 struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
670 struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
671
672 /* get the current hardware pointer */
673 return bytes_to_frames(substream->runtime,
674 chip->channel[chan->idx].pos);
675 }
676
677 /* operators */
678 static const struct snd_pcm_ops snd_sgio2audio_playback1_ops = {
679 .open = snd_sgio2audio_playback1_open,
680 .close = snd_sgio2audio_pcm_close,
681 .ioctl = snd_pcm_lib_ioctl,
682 .hw_params = snd_sgio2audio_pcm_hw_params,
683 .hw_free = snd_sgio2audio_pcm_hw_free,
684 .prepare = snd_sgio2audio_pcm_prepare,
685 .trigger = snd_sgio2audio_pcm_trigger,
686 .pointer = snd_sgio2audio_pcm_pointer,
687 .page = snd_pcm_lib_get_vmalloc_page,
688 .mmap = snd_pcm_lib_mmap_vmalloc,
689 };
690
691 static const struct snd_pcm_ops snd_sgio2audio_playback2_ops = {
692 .open = snd_sgio2audio_playback2_open,
693 .close = snd_sgio2audio_pcm_close,
694 .ioctl = snd_pcm_lib_ioctl,
695 .hw_params = snd_sgio2audio_pcm_hw_params,
696 .hw_free = snd_sgio2audio_pcm_hw_free,
697 .prepare = snd_sgio2audio_pcm_prepare,
698 .trigger = snd_sgio2audio_pcm_trigger,
699 .pointer = snd_sgio2audio_pcm_pointer,
700 .page = snd_pcm_lib_get_vmalloc_page,
701 .mmap = snd_pcm_lib_mmap_vmalloc,
702 };
703
704 static const struct snd_pcm_ops snd_sgio2audio_capture_ops = {
705 .open = snd_sgio2audio_capture_open,
706 .close = snd_sgio2audio_pcm_close,
707 .ioctl = snd_pcm_lib_ioctl,
708 .hw_params = snd_sgio2audio_pcm_hw_params,
709 .hw_free = snd_sgio2audio_pcm_hw_free,
710 .prepare = snd_sgio2audio_pcm_prepare,
711 .trigger = snd_sgio2audio_pcm_trigger,
712 .pointer = snd_sgio2audio_pcm_pointer,
713 .page = snd_pcm_lib_get_vmalloc_page,
714 .mmap = snd_pcm_lib_mmap_vmalloc,
715 };
716
717 /*
718 * definitions of capture are omitted here...
719 */
720
721 /* create a pcm device */
722 static int snd_sgio2audio_new_pcm(struct snd_sgio2audio *chip)
723 {
724 struct snd_pcm *pcm;
725 int err;
726
727 /* create first pcm device with one outputs and one input */
728 err = snd_pcm_new(chip->card, "SGI O2 Audio", 0, 1, 1, &pcm);
729 if (err < 0)
730 return err;
731
732 pcm->private_data = chip;
733 strcpy(pcm->name, "SGI O2 DAC1");
734
735 /* set operators */
736 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
737 &snd_sgio2audio_playback1_ops);
738 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
739 &snd_sgio2audio_capture_ops);
740
741 /* create second pcm device with one outputs and no input */
742 err = snd_pcm_new(chip->card, "SGI O2 Audio", 1, 1, 0, &pcm);
743 if (err < 0)
744 return err;
745
746 pcm->private_data = chip;
747 strcpy(pcm->name, "SGI O2 DAC2");
748
749 /* set operators */
750 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
751 &snd_sgio2audio_playback2_ops);
752
753 return 0;
754 }
755
756 static struct {
757 int idx;
758 int irq;
759 irqreturn_t (*isr)(int, void *);
760 const char *desc;
761 } snd_sgio2_isr_table[] = {
762 {
763 .idx = 0,
764 .irq = MACEISA_AUDIO1_DMAT_IRQ,
765 .isr = snd_sgio2audio_dma_in_isr,
766 .desc = "Capture DMA Channel 0"
767 }, {
768 .idx = 0,
769 .irq = MACEISA_AUDIO1_OF_IRQ,
770 .isr = snd_sgio2audio_error_isr,
771 .desc = "Capture Overflow"
772 }, {
773 .idx = 1,
774 .irq = MACEISA_AUDIO2_DMAT_IRQ,
775 .isr = snd_sgio2audio_dma_out_isr,
776 .desc = "Playback DMA Channel 1"
777 }, {
778 .idx = 1,
779 .irq = MACEISA_AUDIO2_MERR_IRQ,
780 .isr = snd_sgio2audio_error_isr,
781 .desc = "Memory Error Channel 1"
782 }, {
783 .idx = 2,
784 .irq = MACEISA_AUDIO3_DMAT_IRQ,
785 .isr = snd_sgio2audio_dma_out_isr,
786 .desc = "Playback DMA Channel 2"
787 }, {
788 .idx = 2,
789 .irq = MACEISA_AUDIO3_MERR_IRQ,
790 .isr = snd_sgio2audio_error_isr,
791 .desc = "Memory Error Channel 2"
792 }
793 };
794
795 /* ALSA driver */
796
797 static int snd_sgio2audio_free(struct snd_sgio2audio *chip)
798 {
799 int i;
800
801 /* reset interface */
802 writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control);
803 udelay(1);
804 writeq(0, &mace->perif.audio.control);
805
806 /* release IRQ's */
807 for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++)
808 free_irq(snd_sgio2_isr_table[i].irq,
809 &chip->channel[snd_sgio2_isr_table[i].idx]);
810
811 dma_free_coherent(NULL, MACEISA_RINGBUFFERS_SIZE,
812 chip->ring_base, chip->ring_base_dma);
813
814 /* release card data */
815 kfree(chip);
816 return 0;
817 }
818
819 static int snd_sgio2audio_dev_free(struct snd_device *device)
820 {
821 struct snd_sgio2audio *chip = device->device_data;
822
823 return snd_sgio2audio_free(chip);
824 }
825
826 static struct snd_device_ops ops = {
827 .dev_free = snd_sgio2audio_dev_free,
828 };
829
830 static int snd_sgio2audio_create(struct snd_card *card,
831 struct snd_sgio2audio **rchip)
832 {
833 struct snd_sgio2audio *chip;
834 int i, err;
835
836 *rchip = NULL;
837
838 /* check if a codec is attached to the interface */
839 /* (Audio or Audio/Video board present) */
840 if (!(readq(&mace->perif.audio.control) & AUDIO_CONTROL_CODEC_PRESENT))
841 return -ENOENT;
842
843 chip = kzalloc(sizeof(*chip), GFP_KERNEL);
844 if (chip == NULL)
845 return -ENOMEM;
846
847 chip->card = card;
848
849 chip->ring_base = dma_alloc_coherent(NULL, MACEISA_RINGBUFFERS_SIZE,
850 &chip->ring_base_dma, GFP_USER);
851 if (chip->ring_base == NULL) {
852 printk(KERN_ERR
853 "sgio2audio: could not allocate ring buffers\n");
854 kfree(chip);
855 return -ENOMEM;
856 }
857
858 spin_lock_init(&chip->ad1843_lock);
859
860 /* initialize channels */
861 for (i = 0; i < 3; i++) {
862 spin_lock_init(&chip->channel[i].lock);
863 chip->channel[i].idx = i;
864 }
865
866 /* allocate IRQs */
867 for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++) {
868 if (request_irq(snd_sgio2_isr_table[i].irq,
869 snd_sgio2_isr_table[i].isr,
870 0,
871 snd_sgio2_isr_table[i].desc,
872 &chip->channel[snd_sgio2_isr_table[i].idx])) {
873 snd_sgio2audio_free(chip);
874 printk(KERN_ERR "sgio2audio: cannot allocate irq %d\n",
875 snd_sgio2_isr_table[i].irq);
876 return -EBUSY;
877 }
878 }
879
880 /* reset the interface */
881 writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control);
882 udelay(1);
883 writeq(0, &mace->perif.audio.control);
884 msleep_interruptible(1); /* give time to recover */
885
886 /* set ring base */
887 writeq(chip->ring_base_dma, &mace->perif.ctrl.ringbase);
888
889 /* attach the AD1843 codec */
890 chip->ad1843.read = read_ad1843_reg;
891 chip->ad1843.write = write_ad1843_reg;
892 chip->ad1843.chip = chip;
893
894 /* initialize the AD1843 codec */
895 err = ad1843_init(&chip->ad1843);
896 if (err < 0) {
897 snd_sgio2audio_free(chip);
898 return err;
899 }
900
901 err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
902 if (err < 0) {
903 snd_sgio2audio_free(chip);
904 return err;
905 }
906 *rchip = chip;
907 return 0;
908 }
909
910 static int snd_sgio2audio_probe(struct platform_device *pdev)
911 {
912 struct snd_card *card;
913 struct snd_sgio2audio *chip;
914 int err;
915
916 err = snd_card_new(&pdev->dev, index, id, THIS_MODULE, 0, &card);
917 if (err < 0)
918 return err;
919
920 err = snd_sgio2audio_create(card, &chip);
921 if (err < 0) {
922 snd_card_free(card);
923 return err;
924 }
925
926 err = snd_sgio2audio_new_pcm(chip);
927 if (err < 0) {
928 snd_card_free(card);
929 return err;
930 }
931 err = snd_sgio2audio_new_mixer(chip);
932 if (err < 0) {
933 snd_card_free(card);
934 return err;
935 }
936
937 strcpy(card->driver, "SGI O2 Audio");
938 strcpy(card->shortname, "SGI O2 Audio");
939 sprintf(card->longname, "%s irq %i-%i",
940 card->shortname,
941 MACEISA_AUDIO1_DMAT_IRQ,
942 MACEISA_AUDIO3_MERR_IRQ);
943
944 err = snd_card_register(card);
945 if (err < 0) {
946 snd_card_free(card);
947 return err;
948 }
949 platform_set_drvdata(pdev, card);
950 return 0;
951 }
952
953 static int snd_sgio2audio_remove(struct platform_device *pdev)
954 {
955 struct snd_card *card = platform_get_drvdata(pdev);
956
957 snd_card_free(card);
958 return 0;
959 }
960
961 static struct platform_driver sgio2audio_driver = {
962 .probe = snd_sgio2audio_probe,
963 .remove = snd_sgio2audio_remove,
964 .driver = {
965 .name = "sgio2audio",
966 }
967 };
968
969 module_platform_driver(sgio2audio_driver);
Linux with added FPC-III support