Linux 2.6.31.8
[linux/fpc-iii.git] / sound / mips / sgio2audio.c
blobe497525bc11bcf50612c071bdfc3c5ba186bacf1
1 /*
2 * Sound driver for Silicon Graphics O2 Workstations A/V board audio.
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>
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.
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.
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
25 #include <linux/init.h>
26 #include <linux/delay.h>
27 #include <linux/spinlock.h>
28 #include <linux/gfp.h>
29 #include <linux/vmalloc.h>
30 #include <linux/interrupt.h>
31 #include <linux/dma-mapping.h>
32 #include <linux/platform_device.h>
33 #include <linux/io.h>
35 #include <asm/ip32/ip32_ints.h>
36 #include <asm/ip32/mace.h>
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>
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}}");
51 static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */
52 static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */
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.");
60 #define AUDIO_CONTROL_RESET BIT(0) /* 1: reset audio interface */
61 #define AUDIO_CONTROL_CODEC_PRESENT BIT(1) /* 1: codec detected */
63 #define CODEC_CONTROL_WORD_SHIFT 0
64 #define CODEC_CONTROL_READ BIT(16)
65 #define CODEC_CONTROL_ADDRESS_SHIFT 17
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 */
78 #define CHANNEL_RING_SHIFT 12
79 #define CHANNEL_RING_SIZE (1 << CHANNEL_RING_SHIFT)
80 #define CHANNEL_RING_MASK (CHANNEL_RING_SIZE - 1)
82 #define CHANNEL_LEFT_SHIFT 40
83 #define CHANNEL_RIGHT_SHIFT 8
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;
93 /* definition of the chip-specific record */
94 struct snd_sgio2audio {
95 struct snd_card *card;
97 /* codec */
98 struct snd_ad1843 ad1843;
99 spinlock_t ad1843_lock;
101 /* channels */
102 struct snd_sgio2audio_chan channel[3];
104 /* resources */
105 void *ring_base;
106 dma_addr_t ring_base_dma;
109 /* AD1843 access */
112 * read_ad1843_reg returns the current contents of a 16 bit AD1843 register.
114 * Returns unsigned register value on success, -errno on failure.
116 static int read_ad1843_reg(void *priv, int reg)
118 struct snd_sgio2audio *chip = priv;
119 int val;
120 unsigned long flags;
122 spin_lock_irqsave(&chip->ad1843_lock, flags);
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);
130 val = readq(&mace->perif.audio.codec_read);
132 spin_unlock_irqrestore(&chip->ad1843_lock, flags);
133 return val;
137 * write_ad1843_reg writes the specified value to a 16 bit AD1843 register.
139 static int write_ad1843_reg(void *priv, int reg, int word)
141 struct snd_sgio2audio *chip = priv;
142 int val;
143 unsigned long flags;
145 spin_lock_irqsave(&chip->ad1843_lock, flags);
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);
154 spin_unlock_irqrestore(&chip->ad1843_lock, flags);
155 return 0;
158 static int sgio2audio_gain_info(struct snd_kcontrol *kcontrol,
159 struct snd_ctl_elem_info *uinfo)
161 struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
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;
171 static int sgio2audio_gain_get(struct snd_kcontrol *kcontrol,
172 struct snd_ctl_elem_value *ucontrol)
174 struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
175 int vol;
177 vol = ad1843_get_gain(&chip->ad1843, (int)kcontrol->private_value);
179 ucontrol->value.integer.value[0] = (vol >> 8) & 0xFF;
180 ucontrol->value.integer.value[1] = vol & 0xFF;
182 return 0;
185 static int sgio2audio_gain_put(struct snd_kcontrol *kcontrol,
186 struct snd_ctl_elem_value *ucontrol)
188 struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
189 int newvol, oldvol;
191 oldvol = ad1843_get_gain(&chip->ad1843, kcontrol->private_value);
192 newvol = (ucontrol->value.integer.value[0] << 8) |
193 ucontrol->value.integer.value[1];
195 newvol = ad1843_set_gain(&chip->ad1843, kcontrol->private_value,
196 newvol);
198 return newvol != oldvol;
201 static int sgio2audio_source_info(struct snd_kcontrol *kcontrol,
202 struct snd_ctl_elem_info *uinfo)
204 static const char *texts[3] = {
205 "Cam Mic", "Mic", "Line"
207 uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
208 uinfo->count = 1;
209 uinfo->value.enumerated.items = 3;
210 if (uinfo->value.enumerated.item >= 3)
211 uinfo->value.enumerated.item = 1;
212 strcpy(uinfo->value.enumerated.name,
213 texts[uinfo->value.enumerated.item]);
214 return 0;
217 static int sgio2audio_source_get(struct snd_kcontrol *kcontrol,
218 struct snd_ctl_elem_value *ucontrol)
220 struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
222 ucontrol->value.enumerated.item[0] = ad1843_get_recsrc(&chip->ad1843);
223 return 0;
226 static int sgio2audio_source_put(struct snd_kcontrol *kcontrol,
227 struct snd_ctl_elem_value *ucontrol)
229 struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
230 int newsrc, oldsrc;
232 oldsrc = ad1843_get_recsrc(&chip->ad1843);
233 newsrc = ad1843_set_recsrc(&chip->ad1843,
234 ucontrol->value.enumerated.item[0]);
236 return newsrc != oldsrc;
239 /* dac1/pcm0 mixer control */
240 static struct snd_kcontrol_new sgio2audio_ctrl_pcm0 __devinitdata = {
241 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
242 .name = "PCM Playback Volume",
243 .index = 0,
244 .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
245 .private_value = AD1843_GAIN_PCM_0,
246 .info = sgio2audio_gain_info,
247 .get = sgio2audio_gain_get,
248 .put = sgio2audio_gain_put,
251 /* dac2/pcm1 mixer control */
252 static struct snd_kcontrol_new sgio2audio_ctrl_pcm1 __devinitdata = {
253 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
254 .name = "PCM Playback Volume",
255 .index = 1,
256 .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
257 .private_value = AD1843_GAIN_PCM_1,
258 .info = sgio2audio_gain_info,
259 .get = sgio2audio_gain_get,
260 .put = sgio2audio_gain_put,
263 /* record level mixer control */
264 static struct snd_kcontrol_new sgio2audio_ctrl_reclevel __devinitdata = {
265 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
266 .name = "Capture Volume",
267 .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
268 .private_value = AD1843_GAIN_RECLEV,
269 .info = sgio2audio_gain_info,
270 .get = sgio2audio_gain_get,
271 .put = sgio2audio_gain_put,
274 /* record level source control */
275 static struct snd_kcontrol_new sgio2audio_ctrl_recsource __devinitdata = {
276 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
277 .name = "Capture Source",
278 .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
279 .info = sgio2audio_source_info,
280 .get = sgio2audio_source_get,
281 .put = sgio2audio_source_put,
284 /* line mixer control */
285 static struct snd_kcontrol_new sgio2audio_ctrl_line __devinitdata = {
286 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
287 .name = "Line Playback Volume",
288 .index = 0,
289 .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
290 .private_value = AD1843_GAIN_LINE,
291 .info = sgio2audio_gain_info,
292 .get = sgio2audio_gain_get,
293 .put = sgio2audio_gain_put,
296 /* cd mixer control */
297 static struct snd_kcontrol_new sgio2audio_ctrl_cd __devinitdata = {
298 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
299 .name = "Line Playback Volume",
300 .index = 1,
301 .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
302 .private_value = AD1843_GAIN_LINE_2,
303 .info = sgio2audio_gain_info,
304 .get = sgio2audio_gain_get,
305 .put = sgio2audio_gain_put,
308 /* mic mixer control */
309 static struct snd_kcontrol_new sgio2audio_ctrl_mic __devinitdata = {
310 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
311 .name = "Mic Playback Volume",
312 .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
313 .private_value = AD1843_GAIN_MIC,
314 .info = sgio2audio_gain_info,
315 .get = sgio2audio_gain_get,
316 .put = sgio2audio_gain_put,
320 static int __devinit snd_sgio2audio_new_mixer(struct snd_sgio2audio *chip)
322 int err;
324 err = snd_ctl_add(chip->card,
325 snd_ctl_new1(&sgio2audio_ctrl_pcm0, chip));
326 if (err < 0)
327 return err;
329 err = snd_ctl_add(chip->card,
330 snd_ctl_new1(&sgio2audio_ctrl_pcm1, chip));
331 if (err < 0)
332 return err;
334 err = snd_ctl_add(chip->card,
335 snd_ctl_new1(&sgio2audio_ctrl_reclevel, chip));
336 if (err < 0)
337 return err;
339 err = snd_ctl_add(chip->card,
340 snd_ctl_new1(&sgio2audio_ctrl_recsource, chip));
341 if (err < 0)
342 return err;
343 err = snd_ctl_add(chip->card,
344 snd_ctl_new1(&sgio2audio_ctrl_line, chip));
345 if (err < 0)
346 return err;
348 err = snd_ctl_add(chip->card,
349 snd_ctl_new1(&sgio2audio_ctrl_cd, chip));
350 if (err < 0)
351 return err;
353 err = snd_ctl_add(chip->card,
354 snd_ctl_new1(&sgio2audio_ctrl_mic, chip));
355 if (err < 0)
356 return err;
358 return 0;
361 /* low-level audio interface DMA */
363 /* get data out of bounce buffer, count must be a multiple of 32 */
364 /* returns 1 if a period has elapsed */
365 static int snd_sgio2audio_dma_pull_frag(struct snd_sgio2audio *chip,
366 unsigned int ch, unsigned int count)
368 int ret;
369 unsigned long src_base, src_pos, dst_mask;
370 unsigned char *dst_base;
371 int dst_pos;
372 u64 *src;
373 s16 *dst;
374 u64 x;
375 unsigned long flags;
376 struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime;
378 spin_lock_irqsave(&chip->channel[ch].lock, flags);
380 src_base = (unsigned long) chip->ring_base | (ch << CHANNEL_RING_SHIFT);
381 src_pos = readq(&mace->perif.audio.chan[ch].read_ptr);
382 dst_base = runtime->dma_area;
383 dst_pos = chip->channel[ch].pos;
384 dst_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1;
386 /* check if a period has elapsed */
387 chip->channel[ch].size += (count >> 3); /* in frames */
388 ret = chip->channel[ch].size >= runtime->period_size;
389 chip->channel[ch].size %= runtime->period_size;
391 while (count) {
392 src = (u64 *)(src_base + src_pos);
393 dst = (s16 *)(dst_base + dst_pos);
395 x = *src;
396 dst[0] = (x >> CHANNEL_LEFT_SHIFT) & 0xffff;
397 dst[1] = (x >> CHANNEL_RIGHT_SHIFT) & 0xffff;
399 src_pos = (src_pos + sizeof(u64)) & CHANNEL_RING_MASK;
400 dst_pos = (dst_pos + 2 * sizeof(s16)) & dst_mask;
401 count -= sizeof(u64);
404 writeq(src_pos, &mace->perif.audio.chan[ch].read_ptr); /* in bytes */
405 chip->channel[ch].pos = dst_pos;
407 spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
408 return ret;
411 /* put some DMA data in bounce buffer, count must be a multiple of 32 */
412 /* returns 1 if a period has elapsed */
413 static int snd_sgio2audio_dma_push_frag(struct snd_sgio2audio *chip,
414 unsigned int ch, unsigned int count)
416 int ret;
417 s64 l, r;
418 unsigned long dst_base, dst_pos, src_mask;
419 unsigned char *src_base;
420 int src_pos;
421 u64 *dst;
422 s16 *src;
423 unsigned long flags;
424 struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime;
426 spin_lock_irqsave(&chip->channel[ch].lock, flags);
428 dst_base = (unsigned long)chip->ring_base | (ch << CHANNEL_RING_SHIFT);
429 dst_pos = readq(&mace->perif.audio.chan[ch].write_ptr);
430 src_base = runtime->dma_area;
431 src_pos = chip->channel[ch].pos;
432 src_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1;
434 /* check if a period has elapsed */
435 chip->channel[ch].size += (count >> 3); /* in frames */
436 ret = chip->channel[ch].size >= runtime->period_size;
437 chip->channel[ch].size %= runtime->period_size;
439 while (count) {
440 src = (s16 *)(src_base + src_pos);
441 dst = (u64 *)(dst_base + dst_pos);
443 l = src[0]; /* sign extend */
444 r = src[1]; /* sign extend */
446 *dst = ((l & 0x00ffffff) << CHANNEL_LEFT_SHIFT) |
447 ((r & 0x00ffffff) << CHANNEL_RIGHT_SHIFT);
449 dst_pos = (dst_pos + sizeof(u64)) & CHANNEL_RING_MASK;
450 src_pos = (src_pos + 2 * sizeof(s16)) & src_mask;
451 count -= sizeof(u64);
454 writeq(dst_pos, &mace->perif.audio.chan[ch].write_ptr); /* in bytes */
455 chip->channel[ch].pos = src_pos;
457 spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
458 return ret;
461 static int snd_sgio2audio_dma_start(struct snd_pcm_substream *substream)
463 struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
464 struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
465 int ch = chan->idx;
467 /* reset DMA channel */
468 writeq(CHANNEL_CONTROL_RESET, &mace->perif.audio.chan[ch].control);
469 udelay(10);
470 writeq(0, &mace->perif.audio.chan[ch].control);
472 if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
473 /* push a full buffer */
474 snd_sgio2audio_dma_push_frag(chip, ch, CHANNEL_RING_SIZE - 32);
476 /* set DMA to wake on 50% empty and enable interrupt */
477 writeq(CHANNEL_DMA_ENABLE | CHANNEL_INT_THRESHOLD_50,
478 &mace->perif.audio.chan[ch].control);
479 return 0;
482 static int snd_sgio2audio_dma_stop(struct snd_pcm_substream *substream)
484 struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
486 writeq(0, &mace->perif.audio.chan[chan->idx].control);
487 return 0;
490 static irqreturn_t snd_sgio2audio_dma_in_isr(int irq, void *dev_id)
492 struct snd_sgio2audio_chan *chan = dev_id;
493 struct snd_pcm_substream *substream;
494 struct snd_sgio2audio *chip;
495 int count, ch;
497 substream = chan->substream;
498 chip = snd_pcm_substream_chip(substream);
499 ch = chan->idx;
501 /* empty the ring */
502 count = CHANNEL_RING_SIZE -
503 readq(&mace->perif.audio.chan[ch].depth) - 32;
504 if (snd_sgio2audio_dma_pull_frag(chip, ch, count))
505 snd_pcm_period_elapsed(substream);
507 return IRQ_HANDLED;
510 static irqreturn_t snd_sgio2audio_dma_out_isr(int irq, void *dev_id)
512 struct snd_sgio2audio_chan *chan = dev_id;
513 struct snd_pcm_substream *substream;
514 struct snd_sgio2audio *chip;
515 int count, ch;
517 substream = chan->substream;
518 chip = snd_pcm_substream_chip(substream);
519 ch = chan->idx;
520 /* fill the ring */
521 count = CHANNEL_RING_SIZE -
522 readq(&mace->perif.audio.chan[ch].depth) - 32;
523 if (snd_sgio2audio_dma_push_frag(chip, ch, count))
524 snd_pcm_period_elapsed(substream);
526 return IRQ_HANDLED;
529 static irqreturn_t snd_sgio2audio_error_isr(int irq, void *dev_id)
531 struct snd_sgio2audio_chan *chan = dev_id;
532 struct snd_pcm_substream *substream;
534 substream = chan->substream;
535 snd_sgio2audio_dma_stop(substream);
536 snd_sgio2audio_dma_start(substream);
537 return IRQ_HANDLED;
540 /* PCM part */
541 /* PCM hardware definition */
542 static struct snd_pcm_hardware snd_sgio2audio_pcm_hw = {
543 .info = (SNDRV_PCM_INFO_MMAP |
544 SNDRV_PCM_INFO_MMAP_VALID |
545 SNDRV_PCM_INFO_INTERLEAVED |
546 SNDRV_PCM_INFO_BLOCK_TRANSFER),
547 .formats = SNDRV_PCM_FMTBIT_S16_BE,
548 .rates = SNDRV_PCM_RATE_8000_48000,
549 .rate_min = 8000,
550 .rate_max = 48000,
551 .channels_min = 2,
552 .channels_max = 2,
553 .buffer_bytes_max = 65536,
554 .period_bytes_min = 32768,
555 .period_bytes_max = 65536,
556 .periods_min = 1,
557 .periods_max = 1024,
560 /* PCM playback open callback */
561 static int snd_sgio2audio_playback1_open(struct snd_pcm_substream *substream)
563 struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
564 struct snd_pcm_runtime *runtime = substream->runtime;
566 runtime->hw = snd_sgio2audio_pcm_hw;
567 runtime->private_data = &chip->channel[1];
568 return 0;
571 static int snd_sgio2audio_playback2_open(struct snd_pcm_substream *substream)
573 struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
574 struct snd_pcm_runtime *runtime = substream->runtime;
576 runtime->hw = snd_sgio2audio_pcm_hw;
577 runtime->private_data = &chip->channel[2];
578 return 0;
581 /* PCM capture open callback */
582 static int snd_sgio2audio_capture_open(struct snd_pcm_substream *substream)
584 struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
585 struct snd_pcm_runtime *runtime = substream->runtime;
587 runtime->hw = snd_sgio2audio_pcm_hw;
588 runtime->private_data = &chip->channel[0];
589 return 0;
592 /* PCM close callback */
593 static int snd_sgio2audio_pcm_close(struct snd_pcm_substream *substream)
595 struct snd_pcm_runtime *runtime = substream->runtime;
597 runtime->private_data = NULL;
598 return 0;
602 /* hw_params callback */
603 static int snd_sgio2audio_pcm_hw_params(struct snd_pcm_substream *substream,
604 struct snd_pcm_hw_params *hw_params)
606 struct snd_pcm_runtime *runtime = substream->runtime;
607 int size = params_buffer_bytes(hw_params);
609 /* alloc virtual 'dma' area */
610 if (runtime->dma_area)
611 vfree(runtime->dma_area);
612 runtime->dma_area = vmalloc(size);
613 if (runtime->dma_area == NULL)
614 return -ENOMEM;
615 runtime->dma_bytes = size;
616 return 0;
619 /* hw_free callback */
620 static int snd_sgio2audio_pcm_hw_free(struct snd_pcm_substream *substream)
622 vfree(substream->runtime->dma_area);
623 substream->runtime->dma_area = NULL;
624 return 0;
627 /* prepare callback */
628 static int snd_sgio2audio_pcm_prepare(struct snd_pcm_substream *substream)
630 struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
631 struct snd_pcm_runtime *runtime = substream->runtime;
632 struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
633 int ch = chan->idx;
634 unsigned long flags;
636 spin_lock_irqsave(&chip->channel[ch].lock, flags);
638 /* Setup the pseudo-dma transfer pointers. */
639 chip->channel[ch].pos = 0;
640 chip->channel[ch].size = 0;
641 chip->channel[ch].substream = substream;
643 /* set AD1843 format */
644 /* hardware format is always S16_LE */
645 switch (substream->stream) {
646 case SNDRV_PCM_STREAM_PLAYBACK:
647 ad1843_setup_dac(&chip->ad1843,
648 ch - 1,
649 runtime->rate,
650 SNDRV_PCM_FORMAT_S16_LE,
651 runtime->channels);
652 break;
653 case SNDRV_PCM_STREAM_CAPTURE:
654 ad1843_setup_adc(&chip->ad1843,
655 runtime->rate,
656 SNDRV_PCM_FORMAT_S16_LE,
657 runtime->channels);
658 break;
660 spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
661 return 0;
664 /* trigger callback */
665 static int snd_sgio2audio_pcm_trigger(struct snd_pcm_substream *substream,
666 int cmd)
668 switch (cmd) {
669 case SNDRV_PCM_TRIGGER_START:
670 /* start the PCM engine */
671 snd_sgio2audio_dma_start(substream);
672 break;
673 case SNDRV_PCM_TRIGGER_STOP:
674 /* stop the PCM engine */
675 snd_sgio2audio_dma_stop(substream);
676 break;
677 default:
678 return -EINVAL;
680 return 0;
683 /* pointer callback */
684 static snd_pcm_uframes_t
685 snd_sgio2audio_pcm_pointer(struct snd_pcm_substream *substream)
687 struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
688 struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
690 /* get the current hardware pointer */
691 return bytes_to_frames(substream->runtime,
692 chip->channel[chan->idx].pos);
695 /* get the physical page pointer on the given offset */
696 static struct page *snd_sgio2audio_page(struct snd_pcm_substream *substream,
697 unsigned long offset)
699 return vmalloc_to_page(substream->runtime->dma_area + offset);
702 /* operators */
703 static struct snd_pcm_ops snd_sgio2audio_playback1_ops = {
704 .open = snd_sgio2audio_playback1_open,
705 .close = snd_sgio2audio_pcm_close,
706 .ioctl = snd_pcm_lib_ioctl,
707 .hw_params = snd_sgio2audio_pcm_hw_params,
708 .hw_free = snd_sgio2audio_pcm_hw_free,
709 .prepare = snd_sgio2audio_pcm_prepare,
710 .trigger = snd_sgio2audio_pcm_trigger,
711 .pointer = snd_sgio2audio_pcm_pointer,
712 .page = snd_sgio2audio_page,
715 static struct snd_pcm_ops snd_sgio2audio_playback2_ops = {
716 .open = snd_sgio2audio_playback2_open,
717 .close = snd_sgio2audio_pcm_close,
718 .ioctl = snd_pcm_lib_ioctl,
719 .hw_params = snd_sgio2audio_pcm_hw_params,
720 .hw_free = snd_sgio2audio_pcm_hw_free,
721 .prepare = snd_sgio2audio_pcm_prepare,
722 .trigger = snd_sgio2audio_pcm_trigger,
723 .pointer = snd_sgio2audio_pcm_pointer,
724 .page = snd_sgio2audio_page,
727 static struct snd_pcm_ops snd_sgio2audio_capture_ops = {
728 .open = snd_sgio2audio_capture_open,
729 .close = snd_sgio2audio_pcm_close,
730 .ioctl = snd_pcm_lib_ioctl,
731 .hw_params = snd_sgio2audio_pcm_hw_params,
732 .hw_free = snd_sgio2audio_pcm_hw_free,
733 .prepare = snd_sgio2audio_pcm_prepare,
734 .trigger = snd_sgio2audio_pcm_trigger,
735 .pointer = snd_sgio2audio_pcm_pointer,
736 .page = snd_sgio2audio_page,
740 * definitions of capture are omitted here...
743 /* create a pcm device */
744 static int __devinit snd_sgio2audio_new_pcm(struct snd_sgio2audio *chip)
746 struct snd_pcm *pcm;
747 int err;
749 /* create first pcm device with one outputs and one input */
750 err = snd_pcm_new(chip->card, "SGI O2 Audio", 0, 1, 1, &pcm);
751 if (err < 0)
752 return err;
754 pcm->private_data = chip;
755 strcpy(pcm->name, "SGI O2 DAC1");
757 /* set operators */
758 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
759 &snd_sgio2audio_playback1_ops);
760 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
761 &snd_sgio2audio_capture_ops);
763 /* create second pcm device with one outputs and no input */
764 err = snd_pcm_new(chip->card, "SGI O2 Audio", 1, 1, 0, &pcm);
765 if (err < 0)
766 return err;
768 pcm->private_data = chip;
769 strcpy(pcm->name, "SGI O2 DAC2");
771 /* set operators */
772 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
773 &snd_sgio2audio_playback2_ops);
775 return 0;
778 static struct {
779 int idx;
780 int irq;
781 irqreturn_t (*isr)(int, void *);
782 const char *desc;
783 } snd_sgio2_isr_table[] = {
785 .idx = 0,
786 .irq = MACEISA_AUDIO1_DMAT_IRQ,
787 .isr = snd_sgio2audio_dma_in_isr,
788 .desc = "Capture DMA Channel 0"
789 }, {
790 .idx = 0,
791 .irq = MACEISA_AUDIO1_OF_IRQ,
792 .isr = snd_sgio2audio_error_isr,
793 .desc = "Capture Overflow"
794 }, {
795 .idx = 1,
796 .irq = MACEISA_AUDIO2_DMAT_IRQ,
797 .isr = snd_sgio2audio_dma_out_isr,
798 .desc = "Playback DMA Channel 1"
799 }, {
800 .idx = 1,
801 .irq = MACEISA_AUDIO2_MERR_IRQ,
802 .isr = snd_sgio2audio_error_isr,
803 .desc = "Memory Error Channel 1"
804 }, {
805 .idx = 2,
806 .irq = MACEISA_AUDIO3_DMAT_IRQ,
807 .isr = snd_sgio2audio_dma_out_isr,
808 .desc = "Playback DMA Channel 2"
809 }, {
810 .idx = 2,
811 .irq = MACEISA_AUDIO3_MERR_IRQ,
812 .isr = snd_sgio2audio_error_isr,
813 .desc = "Memory Error Channel 2"
817 /* ALSA driver */
819 static int snd_sgio2audio_free(struct snd_sgio2audio *chip)
821 int i;
823 /* reset interface */
824 writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control);
825 udelay(1);
826 writeq(0, &mace->perif.audio.control);
828 /* release IRQ's */
829 for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++)
830 free_irq(snd_sgio2_isr_table[i].irq,
831 &chip->channel[snd_sgio2_isr_table[i].idx]);
833 dma_free_coherent(NULL, MACEISA_RINGBUFFERS_SIZE,
834 chip->ring_base, chip->ring_base_dma);
836 /* release card data */
837 kfree(chip);
838 return 0;
841 static int snd_sgio2audio_dev_free(struct snd_device *device)
843 struct snd_sgio2audio *chip = device->device_data;
845 return snd_sgio2audio_free(chip);
848 static struct snd_device_ops ops = {
849 .dev_free = snd_sgio2audio_dev_free,
852 static int __devinit snd_sgio2audio_create(struct snd_card *card,
853 struct snd_sgio2audio **rchip)
855 struct snd_sgio2audio *chip;
856 int i, err;
858 *rchip = NULL;
860 /* check if a codec is attached to the interface */
861 /* (Audio or Audio/Video board present) */
862 if (!(readq(&mace->perif.audio.control) & AUDIO_CONTROL_CODEC_PRESENT))
863 return -ENOENT;
865 chip = kzalloc(sizeof(struct snd_sgio2audio), GFP_KERNEL);
866 if (chip == NULL)
867 return -ENOMEM;
869 chip->card = card;
871 chip->ring_base = dma_alloc_coherent(NULL, MACEISA_RINGBUFFERS_SIZE,
872 &chip->ring_base_dma, GFP_USER);
873 if (chip->ring_base == NULL) {
874 printk(KERN_ERR
875 "sgio2audio: could not allocate ring buffers\n");
876 kfree(chip);
877 return -ENOMEM;
880 spin_lock_init(&chip->ad1843_lock);
882 /* initialize channels */
883 for (i = 0; i < 3; i++) {
884 spin_lock_init(&chip->channel[i].lock);
885 chip->channel[i].idx = i;
888 /* allocate IRQs */
889 for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++) {
890 if (request_irq(snd_sgio2_isr_table[i].irq,
891 snd_sgio2_isr_table[i].isr,
893 snd_sgio2_isr_table[i].desc,
894 &chip->channel[snd_sgio2_isr_table[i].idx])) {
895 snd_sgio2audio_free(chip);
896 printk(KERN_ERR "sgio2audio: cannot allocate irq %d\n",
897 snd_sgio2_isr_table[i].irq);
898 return -EBUSY;
902 /* reset the interface */
903 writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control);
904 udelay(1);
905 writeq(0, &mace->perif.audio.control);
906 msleep_interruptible(1); /* give time to recover */
908 /* set ring base */
909 writeq(chip->ring_base_dma, &mace->perif.ctrl.ringbase);
911 /* attach the AD1843 codec */
912 chip->ad1843.read = read_ad1843_reg;
913 chip->ad1843.write = write_ad1843_reg;
914 chip->ad1843.chip = chip;
916 /* initialize the AD1843 codec */
917 err = ad1843_init(&chip->ad1843);
918 if (err < 0) {
919 snd_sgio2audio_free(chip);
920 return err;
923 err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
924 if (err < 0) {
925 snd_sgio2audio_free(chip);
926 return err;
928 *rchip = chip;
929 return 0;
932 static int __devinit snd_sgio2audio_probe(struct platform_device *pdev)
934 struct snd_card *card;
935 struct snd_sgio2audio *chip;
936 int err;
938 err = snd_card_create(index, id, THIS_MODULE, 0, &card);
939 if (err < 0)
940 return err;
942 err = snd_sgio2audio_create(card, &chip);
943 if (err < 0) {
944 snd_card_free(card);
945 return err;
947 snd_card_set_dev(card, &pdev->dev);
949 err = snd_sgio2audio_new_pcm(chip);
950 if (err < 0) {
951 snd_card_free(card);
952 return err;
954 err = snd_sgio2audio_new_mixer(chip);
955 if (err < 0) {
956 snd_card_free(card);
957 return err;
960 strcpy(card->driver, "SGI O2 Audio");
961 strcpy(card->shortname, "SGI O2 Audio");
962 sprintf(card->longname, "%s irq %i-%i",
963 card->shortname,
964 MACEISA_AUDIO1_DMAT_IRQ,
965 MACEISA_AUDIO3_MERR_IRQ);
967 err = snd_card_register(card);
968 if (err < 0) {
969 snd_card_free(card);
970 return err;
972 platform_set_drvdata(pdev, card);
973 return 0;
976 static int __exit snd_sgio2audio_remove(struct platform_device *pdev)
978 struct snd_card *card = platform_get_drvdata(pdev);
980 snd_card_free(card);
981 platform_set_drvdata(pdev, NULL);
982 return 0;
985 static struct platform_driver sgio2audio_driver = {
986 .probe = snd_sgio2audio_probe,
987 .remove = __devexit_p(snd_sgio2audio_remove),
988 .driver = {
989 .name = "sgio2audio",
990 .owner = THIS_MODULE,
994 static int __init alsa_card_sgio2audio_init(void)
996 return platform_driver_register(&sgio2audio_driver);
999 static void __exit alsa_card_sgio2audio_exit(void)
1001 platform_driver_unregister(&sgio2audio_driver);
1004 module_init(alsa_card_sgio2audio_init)
1005 module_exit(alsa_card_sgio2audio_exit)