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/interrupt.h>
29 #include <linux/dma-mapping.h>
30 #include <linux/platform_device.h>
32 #include <linux/slab.h>
34 #include <asm/ip32/ip32_ints.h>
35 #include <asm/ip32/mace.h>
37 #include <sound/core.h>
38 #include <sound/control.h>
39 #include <sound/pcm.h>
41 #include <sound/initval.h>
42 #include <sound/ad1843.h>
45 MODULE_AUTHOR("Vivien Chappelier <vivien.chappelier@linux-mips.org>");
46 MODULE_DESCRIPTION("SGI O2 Audio");
47 MODULE_LICENSE("GPL");
48 MODULE_SUPPORTED_DEVICE("{{Silicon Graphics, O2 Audio}}");
50 static int index
= SNDRV_DEFAULT_IDX1
; /* Index 0-MAX */
51 static char *id
= SNDRV_DEFAULT_STR1
; /* ID for this card */
53 module_param(index
, int, 0444);
54 MODULE_PARM_DESC(index
, "Index value for SGI O2 soundcard.");
55 module_param(id
, charp
, 0444);
56 MODULE_PARM_DESC(id
, "ID string for SGI O2 soundcard.");
59 #define AUDIO_CONTROL_RESET BIT(0) /* 1: reset audio interface */
60 #define AUDIO_CONTROL_CODEC_PRESENT BIT(1) /* 1: codec detected */
62 #define CODEC_CONTROL_WORD_SHIFT 0
63 #define CODEC_CONTROL_READ BIT(16)
64 #define CODEC_CONTROL_ADDRESS_SHIFT 17
66 #define CHANNEL_CONTROL_RESET BIT(10) /* 1: reset channel */
67 #define CHANNEL_DMA_ENABLE BIT(9) /* 1: enable DMA transfer */
68 #define CHANNEL_INT_THRESHOLD_DISABLED (0 << 5) /* interrupt disabled */
69 #define CHANNEL_INT_THRESHOLD_25 (1 << 5) /* int on buffer >25% full */
70 #define CHANNEL_INT_THRESHOLD_50 (2 << 5) /* int on buffer >50% full */
71 #define CHANNEL_INT_THRESHOLD_75 (3 << 5) /* int on buffer >75% full */
72 #define CHANNEL_INT_THRESHOLD_EMPTY (4 << 5) /* int on buffer empty */
73 #define CHANNEL_INT_THRESHOLD_NOT_EMPTY (5 << 5) /* int on buffer !empty */
74 #define CHANNEL_INT_THRESHOLD_FULL (6 << 5) /* int on buffer empty */
75 #define CHANNEL_INT_THRESHOLD_NOT_FULL (7 << 5) /* int on buffer !empty */
77 #define CHANNEL_RING_SHIFT 12
78 #define CHANNEL_RING_SIZE (1 << CHANNEL_RING_SHIFT)
79 #define CHANNEL_RING_MASK (CHANNEL_RING_SIZE - 1)
81 #define CHANNEL_LEFT_SHIFT 40
82 #define CHANNEL_RIGHT_SHIFT 8
84 struct snd_sgio2audio_chan
{
86 struct snd_pcm_substream
*substream
;
88 snd_pcm_uframes_t size
;
92 /* definition of the chip-specific record */
93 struct snd_sgio2audio
{
94 struct snd_card
*card
;
97 struct snd_ad1843 ad1843
;
98 spinlock_t ad1843_lock
;
101 struct snd_sgio2audio_chan channel
[3];
105 dma_addr_t ring_base_dma
;
111 * read_ad1843_reg returns the current contents of a 16 bit AD1843 register.
113 * Returns unsigned register value on success, -errno on failure.
115 static int read_ad1843_reg(void *priv
, int reg
)
117 struct snd_sgio2audio
*chip
= priv
;
121 spin_lock_irqsave(&chip
->ad1843_lock
, flags
);
123 writeq((reg
<< CODEC_CONTROL_ADDRESS_SHIFT
) |
124 CODEC_CONTROL_READ
, &mace
->perif
.audio
.codec_control
);
126 val
= readq(&mace
->perif
.audio
.codec_control
); /* flush bus */
129 val
= readq(&mace
->perif
.audio
.codec_read
);
131 spin_unlock_irqrestore(&chip
->ad1843_lock
, flags
);
136 * write_ad1843_reg writes the specified value to a 16 bit AD1843 register.
138 static int write_ad1843_reg(void *priv
, int reg
, int word
)
140 struct snd_sgio2audio
*chip
= priv
;
144 spin_lock_irqsave(&chip
->ad1843_lock
, flags
);
146 writeq((reg
<< CODEC_CONTROL_ADDRESS_SHIFT
) |
147 (word
<< CODEC_CONTROL_WORD_SHIFT
),
148 &mace
->perif
.audio
.codec_control
);
150 val
= readq(&mace
->perif
.audio
.codec_control
); /* flush bus */
153 spin_unlock_irqrestore(&chip
->ad1843_lock
, flags
);
157 static int sgio2audio_gain_info(struct snd_kcontrol
*kcontrol
,
158 struct snd_ctl_elem_info
*uinfo
)
160 struct snd_sgio2audio
*chip
= snd_kcontrol_chip(kcontrol
);
162 uinfo
->type
= SNDRV_CTL_ELEM_TYPE_INTEGER
;
164 uinfo
->value
.integer
.min
= 0;
165 uinfo
->value
.integer
.max
= ad1843_get_gain_max(&chip
->ad1843
,
166 (int)kcontrol
->private_value
);
170 static int sgio2audio_gain_get(struct snd_kcontrol
*kcontrol
,
171 struct snd_ctl_elem_value
*ucontrol
)
173 struct snd_sgio2audio
*chip
= snd_kcontrol_chip(kcontrol
);
176 vol
= ad1843_get_gain(&chip
->ad1843
, (int)kcontrol
->private_value
);
178 ucontrol
->value
.integer
.value
[0] = (vol
>> 8) & 0xFF;
179 ucontrol
->value
.integer
.value
[1] = vol
& 0xFF;
184 static int sgio2audio_gain_put(struct snd_kcontrol
*kcontrol
,
185 struct snd_ctl_elem_value
*ucontrol
)
187 struct snd_sgio2audio
*chip
= snd_kcontrol_chip(kcontrol
);
190 oldvol
= ad1843_get_gain(&chip
->ad1843
, kcontrol
->private_value
);
191 newvol
= (ucontrol
->value
.integer
.value
[0] << 8) |
192 ucontrol
->value
.integer
.value
[1];
194 newvol
= ad1843_set_gain(&chip
->ad1843
, kcontrol
->private_value
,
197 return newvol
!= oldvol
;
200 static int sgio2audio_source_info(struct snd_kcontrol
*kcontrol
,
201 struct snd_ctl_elem_info
*uinfo
)
203 static const char *texts
[3] = {
204 "Cam Mic", "Mic", "Line"
206 uinfo
->type
= SNDRV_CTL_ELEM_TYPE_ENUMERATED
;
208 uinfo
->value
.enumerated
.items
= 3;
209 if (uinfo
->value
.enumerated
.item
>= 3)
210 uinfo
->value
.enumerated
.item
= 1;
211 strcpy(uinfo
->value
.enumerated
.name
,
212 texts
[uinfo
->value
.enumerated
.item
]);
216 static int sgio2audio_source_get(struct snd_kcontrol
*kcontrol
,
217 struct snd_ctl_elem_value
*ucontrol
)
219 struct snd_sgio2audio
*chip
= snd_kcontrol_chip(kcontrol
);
221 ucontrol
->value
.enumerated
.item
[0] = ad1843_get_recsrc(&chip
->ad1843
);
225 static int sgio2audio_source_put(struct snd_kcontrol
*kcontrol
,
226 struct snd_ctl_elem_value
*ucontrol
)
228 struct snd_sgio2audio
*chip
= snd_kcontrol_chip(kcontrol
);
231 oldsrc
= ad1843_get_recsrc(&chip
->ad1843
);
232 newsrc
= ad1843_set_recsrc(&chip
->ad1843
,
233 ucontrol
->value
.enumerated
.item
[0]);
235 return newsrc
!= oldsrc
;
238 /* dac1/pcm0 mixer control */
239 static struct snd_kcontrol_new sgio2audio_ctrl_pcm0 __devinitdata
= {
240 .iface
= SNDRV_CTL_ELEM_IFACE_MIXER
,
241 .name
= "PCM Playback Volume",
243 .access
= SNDRV_CTL_ELEM_ACCESS_READWRITE
,
244 .private_value
= AD1843_GAIN_PCM_0
,
245 .info
= sgio2audio_gain_info
,
246 .get
= sgio2audio_gain_get
,
247 .put
= sgio2audio_gain_put
,
250 /* dac2/pcm1 mixer control */
251 static struct snd_kcontrol_new sgio2audio_ctrl_pcm1 __devinitdata
= {
252 .iface
= SNDRV_CTL_ELEM_IFACE_MIXER
,
253 .name
= "PCM Playback Volume",
255 .access
= SNDRV_CTL_ELEM_ACCESS_READWRITE
,
256 .private_value
= AD1843_GAIN_PCM_1
,
257 .info
= sgio2audio_gain_info
,
258 .get
= sgio2audio_gain_get
,
259 .put
= sgio2audio_gain_put
,
262 /* record level mixer control */
263 static struct snd_kcontrol_new sgio2audio_ctrl_reclevel __devinitdata
= {
264 .iface
= SNDRV_CTL_ELEM_IFACE_MIXER
,
265 .name
= "Capture Volume",
266 .access
= SNDRV_CTL_ELEM_ACCESS_READWRITE
,
267 .private_value
= AD1843_GAIN_RECLEV
,
268 .info
= sgio2audio_gain_info
,
269 .get
= sgio2audio_gain_get
,
270 .put
= sgio2audio_gain_put
,
273 /* record level source control */
274 static struct snd_kcontrol_new sgio2audio_ctrl_recsource __devinitdata
= {
275 .iface
= SNDRV_CTL_ELEM_IFACE_MIXER
,
276 .name
= "Capture Source",
277 .access
= SNDRV_CTL_ELEM_ACCESS_READWRITE
,
278 .info
= sgio2audio_source_info
,
279 .get
= sgio2audio_source_get
,
280 .put
= sgio2audio_source_put
,
283 /* line mixer control */
284 static struct snd_kcontrol_new sgio2audio_ctrl_line __devinitdata
= {
285 .iface
= SNDRV_CTL_ELEM_IFACE_MIXER
,
286 .name
= "Line Playback Volume",
288 .access
= SNDRV_CTL_ELEM_ACCESS_READWRITE
,
289 .private_value
= AD1843_GAIN_LINE
,
290 .info
= sgio2audio_gain_info
,
291 .get
= sgio2audio_gain_get
,
292 .put
= sgio2audio_gain_put
,
295 /* cd mixer control */
296 static struct snd_kcontrol_new sgio2audio_ctrl_cd __devinitdata
= {
297 .iface
= SNDRV_CTL_ELEM_IFACE_MIXER
,
298 .name
= "Line Playback Volume",
300 .access
= SNDRV_CTL_ELEM_ACCESS_READWRITE
,
301 .private_value
= AD1843_GAIN_LINE_2
,
302 .info
= sgio2audio_gain_info
,
303 .get
= sgio2audio_gain_get
,
304 .put
= sgio2audio_gain_put
,
307 /* mic mixer control */
308 static struct snd_kcontrol_new sgio2audio_ctrl_mic __devinitdata
= {
309 .iface
= SNDRV_CTL_ELEM_IFACE_MIXER
,
310 .name
= "Mic Playback Volume",
311 .access
= SNDRV_CTL_ELEM_ACCESS_READWRITE
,
312 .private_value
= AD1843_GAIN_MIC
,
313 .info
= sgio2audio_gain_info
,
314 .get
= sgio2audio_gain_get
,
315 .put
= sgio2audio_gain_put
,
319 static int __devinit
snd_sgio2audio_new_mixer(struct snd_sgio2audio
*chip
)
323 err
= snd_ctl_add(chip
->card
,
324 snd_ctl_new1(&sgio2audio_ctrl_pcm0
, chip
));
328 err
= snd_ctl_add(chip
->card
,
329 snd_ctl_new1(&sgio2audio_ctrl_pcm1
, chip
));
333 err
= snd_ctl_add(chip
->card
,
334 snd_ctl_new1(&sgio2audio_ctrl_reclevel
, chip
));
338 err
= snd_ctl_add(chip
->card
,
339 snd_ctl_new1(&sgio2audio_ctrl_recsource
, chip
));
342 err
= snd_ctl_add(chip
->card
,
343 snd_ctl_new1(&sgio2audio_ctrl_line
, chip
));
347 err
= snd_ctl_add(chip
->card
,
348 snd_ctl_new1(&sgio2audio_ctrl_cd
, chip
));
352 err
= snd_ctl_add(chip
->card
,
353 snd_ctl_new1(&sgio2audio_ctrl_mic
, chip
));
360 /* low-level audio interface DMA */
362 /* get data out of bounce buffer, count must be a multiple of 32 */
363 /* returns 1 if a period has elapsed */
364 static int snd_sgio2audio_dma_pull_frag(struct snd_sgio2audio
*chip
,
365 unsigned int ch
, unsigned int count
)
368 unsigned long src_base
, src_pos
, dst_mask
;
369 unsigned char *dst_base
;
375 struct snd_pcm_runtime
*runtime
= chip
->channel
[ch
].substream
->runtime
;
377 spin_lock_irqsave(&chip
->channel
[ch
].lock
, flags
);
379 src_base
= (unsigned long) chip
->ring_base
| (ch
<< CHANNEL_RING_SHIFT
);
380 src_pos
= readq(&mace
->perif
.audio
.chan
[ch
].read_ptr
);
381 dst_base
= runtime
->dma_area
;
382 dst_pos
= chip
->channel
[ch
].pos
;
383 dst_mask
= frames_to_bytes(runtime
, runtime
->buffer_size
) - 1;
385 /* check if a period has elapsed */
386 chip
->channel
[ch
].size
+= (count
>> 3); /* in frames */
387 ret
= chip
->channel
[ch
].size
>= runtime
->period_size
;
388 chip
->channel
[ch
].size
%= runtime
->period_size
;
391 src
= (u64
*)(src_base
+ src_pos
);
392 dst
= (s16
*)(dst_base
+ dst_pos
);
395 dst
[0] = (x
>> CHANNEL_LEFT_SHIFT
) & 0xffff;
396 dst
[1] = (x
>> CHANNEL_RIGHT_SHIFT
) & 0xffff;
398 src_pos
= (src_pos
+ sizeof(u64
)) & CHANNEL_RING_MASK
;
399 dst_pos
= (dst_pos
+ 2 * sizeof(s16
)) & dst_mask
;
400 count
-= sizeof(u64
);
403 writeq(src_pos
, &mace
->perif
.audio
.chan
[ch
].read_ptr
); /* in bytes */
404 chip
->channel
[ch
].pos
= dst_pos
;
406 spin_unlock_irqrestore(&chip
->channel
[ch
].lock
, flags
);
410 /* put some DMA data in bounce buffer, count must be a multiple of 32 */
411 /* returns 1 if a period has elapsed */
412 static int snd_sgio2audio_dma_push_frag(struct snd_sgio2audio
*chip
,
413 unsigned int ch
, unsigned int count
)
417 unsigned long dst_base
, dst_pos
, src_mask
;
418 unsigned char *src_base
;
423 struct snd_pcm_runtime
*runtime
= chip
->channel
[ch
].substream
->runtime
;
425 spin_lock_irqsave(&chip
->channel
[ch
].lock
, flags
);
427 dst_base
= (unsigned long)chip
->ring_base
| (ch
<< CHANNEL_RING_SHIFT
);
428 dst_pos
= readq(&mace
->perif
.audio
.chan
[ch
].write_ptr
);
429 src_base
= runtime
->dma_area
;
430 src_pos
= chip
->channel
[ch
].pos
;
431 src_mask
= frames_to_bytes(runtime
, runtime
->buffer_size
) - 1;
433 /* check if a period has elapsed */
434 chip
->channel
[ch
].size
+= (count
>> 3); /* in frames */
435 ret
= chip
->channel
[ch
].size
>= runtime
->period_size
;
436 chip
->channel
[ch
].size
%= runtime
->period_size
;
439 src
= (s16
*)(src_base
+ src_pos
);
440 dst
= (u64
*)(dst_base
+ dst_pos
);
442 l
= src
[0]; /* sign extend */
443 r
= src
[1]; /* sign extend */
445 *dst
= ((l
& 0x00ffffff) << CHANNEL_LEFT_SHIFT
) |
446 ((r
& 0x00ffffff) << CHANNEL_RIGHT_SHIFT
);
448 dst_pos
= (dst_pos
+ sizeof(u64
)) & CHANNEL_RING_MASK
;
449 src_pos
= (src_pos
+ 2 * sizeof(s16
)) & src_mask
;
450 count
-= sizeof(u64
);
453 writeq(dst_pos
, &mace
->perif
.audio
.chan
[ch
].write_ptr
); /* in bytes */
454 chip
->channel
[ch
].pos
= src_pos
;
456 spin_unlock_irqrestore(&chip
->channel
[ch
].lock
, flags
);
460 static int snd_sgio2audio_dma_start(struct snd_pcm_substream
*substream
)
462 struct snd_sgio2audio
*chip
= snd_pcm_substream_chip(substream
);
463 struct snd_sgio2audio_chan
*chan
= substream
->runtime
->private_data
;
466 /* reset DMA channel */
467 writeq(CHANNEL_CONTROL_RESET
, &mace
->perif
.audio
.chan
[ch
].control
);
469 writeq(0, &mace
->perif
.audio
.chan
[ch
].control
);
471 if (substream
->stream
== SNDRV_PCM_STREAM_PLAYBACK
) {
472 /* push a full buffer */
473 snd_sgio2audio_dma_push_frag(chip
, ch
, CHANNEL_RING_SIZE
- 32);
475 /* set DMA to wake on 50% empty and enable interrupt */
476 writeq(CHANNEL_DMA_ENABLE
| CHANNEL_INT_THRESHOLD_50
,
477 &mace
->perif
.audio
.chan
[ch
].control
);
481 static int snd_sgio2audio_dma_stop(struct snd_pcm_substream
*substream
)
483 struct snd_sgio2audio_chan
*chan
= substream
->runtime
->private_data
;
485 writeq(0, &mace
->perif
.audio
.chan
[chan
->idx
].control
);
489 static irqreturn_t
snd_sgio2audio_dma_in_isr(int irq
, void *dev_id
)
491 struct snd_sgio2audio_chan
*chan
= dev_id
;
492 struct snd_pcm_substream
*substream
;
493 struct snd_sgio2audio
*chip
;
496 substream
= chan
->substream
;
497 chip
= snd_pcm_substream_chip(substream
);
501 count
= CHANNEL_RING_SIZE
-
502 readq(&mace
->perif
.audio
.chan
[ch
].depth
) - 32;
503 if (snd_sgio2audio_dma_pull_frag(chip
, ch
, count
))
504 snd_pcm_period_elapsed(substream
);
509 static irqreturn_t
snd_sgio2audio_dma_out_isr(int irq
, void *dev_id
)
511 struct snd_sgio2audio_chan
*chan
= dev_id
;
512 struct snd_pcm_substream
*substream
;
513 struct snd_sgio2audio
*chip
;
516 substream
= chan
->substream
;
517 chip
= snd_pcm_substream_chip(substream
);
520 count
= CHANNEL_RING_SIZE
-
521 readq(&mace
->perif
.audio
.chan
[ch
].depth
) - 32;
522 if (snd_sgio2audio_dma_push_frag(chip
, ch
, count
))
523 snd_pcm_period_elapsed(substream
);
528 static irqreturn_t
snd_sgio2audio_error_isr(int irq
, void *dev_id
)
530 struct snd_sgio2audio_chan
*chan
= dev_id
;
531 struct snd_pcm_substream
*substream
;
533 substream
= chan
->substream
;
534 snd_sgio2audio_dma_stop(substream
);
535 snd_sgio2audio_dma_start(substream
);
540 /* PCM hardware definition */
541 static struct snd_pcm_hardware snd_sgio2audio_pcm_hw
= {
542 .info
= (SNDRV_PCM_INFO_MMAP
|
543 SNDRV_PCM_INFO_MMAP_VALID
|
544 SNDRV_PCM_INFO_INTERLEAVED
|
545 SNDRV_PCM_INFO_BLOCK_TRANSFER
),
546 .formats
= SNDRV_PCM_FMTBIT_S16_BE
,
547 .rates
= SNDRV_PCM_RATE_8000_48000
,
552 .buffer_bytes_max
= 65536,
553 .period_bytes_min
= 32768,
554 .period_bytes_max
= 65536,
559 /* PCM playback open callback */
560 static int snd_sgio2audio_playback1_open(struct snd_pcm_substream
*substream
)
562 struct snd_sgio2audio
*chip
= snd_pcm_substream_chip(substream
);
563 struct snd_pcm_runtime
*runtime
= substream
->runtime
;
565 runtime
->hw
= snd_sgio2audio_pcm_hw
;
566 runtime
->private_data
= &chip
->channel
[1];
570 static int snd_sgio2audio_playback2_open(struct snd_pcm_substream
*substream
)
572 struct snd_sgio2audio
*chip
= snd_pcm_substream_chip(substream
);
573 struct snd_pcm_runtime
*runtime
= substream
->runtime
;
575 runtime
->hw
= snd_sgio2audio_pcm_hw
;
576 runtime
->private_data
= &chip
->channel
[2];
580 /* PCM capture open callback */
581 static int snd_sgio2audio_capture_open(struct snd_pcm_substream
*substream
)
583 struct snd_sgio2audio
*chip
= snd_pcm_substream_chip(substream
);
584 struct snd_pcm_runtime
*runtime
= substream
->runtime
;
586 runtime
->hw
= snd_sgio2audio_pcm_hw
;
587 runtime
->private_data
= &chip
->channel
[0];
591 /* PCM close callback */
592 static int snd_sgio2audio_pcm_close(struct snd_pcm_substream
*substream
)
594 struct snd_pcm_runtime
*runtime
= substream
->runtime
;
596 runtime
->private_data
= NULL
;
601 /* hw_params callback */
602 static int snd_sgio2audio_pcm_hw_params(struct snd_pcm_substream
*substream
,
603 struct snd_pcm_hw_params
*hw_params
)
605 return snd_pcm_lib_alloc_vmalloc_buffer(substream
,
606 params_buffer_bytes(hw_params
));
609 /* hw_free callback */
610 static int snd_sgio2audio_pcm_hw_free(struct snd_pcm_substream
*substream
)
612 return snd_pcm_lib_free_vmalloc_buffer(substream
);
615 /* prepare callback */
616 static int snd_sgio2audio_pcm_prepare(struct snd_pcm_substream
*substream
)
618 struct snd_sgio2audio
*chip
= snd_pcm_substream_chip(substream
);
619 struct snd_pcm_runtime
*runtime
= substream
->runtime
;
620 struct snd_sgio2audio_chan
*chan
= substream
->runtime
->private_data
;
624 spin_lock_irqsave(&chip
->channel
[ch
].lock
, flags
);
626 /* Setup the pseudo-dma transfer pointers. */
627 chip
->channel
[ch
].pos
= 0;
628 chip
->channel
[ch
].size
= 0;
629 chip
->channel
[ch
].substream
= substream
;
631 /* set AD1843 format */
632 /* hardware format is always S16_LE */
633 switch (substream
->stream
) {
634 case SNDRV_PCM_STREAM_PLAYBACK
:
635 ad1843_setup_dac(&chip
->ad1843
,
638 SNDRV_PCM_FORMAT_S16_LE
,
641 case SNDRV_PCM_STREAM_CAPTURE
:
642 ad1843_setup_adc(&chip
->ad1843
,
644 SNDRV_PCM_FORMAT_S16_LE
,
648 spin_unlock_irqrestore(&chip
->channel
[ch
].lock
, flags
);
652 /* trigger callback */
653 static int snd_sgio2audio_pcm_trigger(struct snd_pcm_substream
*substream
,
657 case SNDRV_PCM_TRIGGER_START
:
658 /* start the PCM engine */
659 snd_sgio2audio_dma_start(substream
);
661 case SNDRV_PCM_TRIGGER_STOP
:
662 /* stop the PCM engine */
663 snd_sgio2audio_dma_stop(substream
);
671 /* pointer callback */
672 static snd_pcm_uframes_t
673 snd_sgio2audio_pcm_pointer(struct snd_pcm_substream
*substream
)
675 struct snd_sgio2audio
*chip
= snd_pcm_substream_chip(substream
);
676 struct snd_sgio2audio_chan
*chan
= substream
->runtime
->private_data
;
678 /* get the current hardware pointer */
679 return bytes_to_frames(substream
->runtime
,
680 chip
->channel
[chan
->idx
].pos
);
684 static struct snd_pcm_ops snd_sgio2audio_playback1_ops
= {
685 .open
= snd_sgio2audio_playback1_open
,
686 .close
= snd_sgio2audio_pcm_close
,
687 .ioctl
= snd_pcm_lib_ioctl
,
688 .hw_params
= snd_sgio2audio_pcm_hw_params
,
689 .hw_free
= snd_sgio2audio_pcm_hw_free
,
690 .prepare
= snd_sgio2audio_pcm_prepare
,
691 .trigger
= snd_sgio2audio_pcm_trigger
,
692 .pointer
= snd_sgio2audio_pcm_pointer
,
693 .page
= snd_pcm_lib_get_vmalloc_page
,
694 .mmap
= snd_pcm_lib_mmap_vmalloc
,
697 static struct snd_pcm_ops snd_sgio2audio_playback2_ops
= {
698 .open
= snd_sgio2audio_playback2_open
,
699 .close
= snd_sgio2audio_pcm_close
,
700 .ioctl
= snd_pcm_lib_ioctl
,
701 .hw_params
= snd_sgio2audio_pcm_hw_params
,
702 .hw_free
= snd_sgio2audio_pcm_hw_free
,
703 .prepare
= snd_sgio2audio_pcm_prepare
,
704 .trigger
= snd_sgio2audio_pcm_trigger
,
705 .pointer
= snd_sgio2audio_pcm_pointer
,
706 .page
= snd_pcm_lib_get_vmalloc_page
,
707 .mmap
= snd_pcm_lib_mmap_vmalloc
,
710 static struct snd_pcm_ops snd_sgio2audio_capture_ops
= {
711 .open
= snd_sgio2audio_capture_open
,
712 .close
= snd_sgio2audio_pcm_close
,
713 .ioctl
= snd_pcm_lib_ioctl
,
714 .hw_params
= snd_sgio2audio_pcm_hw_params
,
715 .hw_free
= snd_sgio2audio_pcm_hw_free
,
716 .prepare
= snd_sgio2audio_pcm_prepare
,
717 .trigger
= snd_sgio2audio_pcm_trigger
,
718 .pointer
= snd_sgio2audio_pcm_pointer
,
719 .page
= snd_pcm_lib_get_vmalloc_page
,
720 .mmap
= snd_pcm_lib_mmap_vmalloc
,
724 * definitions of capture are omitted here...
727 /* create a pcm device */
728 static int __devinit
snd_sgio2audio_new_pcm(struct snd_sgio2audio
*chip
)
733 /* create first pcm device with one outputs and one input */
734 err
= snd_pcm_new(chip
->card
, "SGI O2 Audio", 0, 1, 1, &pcm
);
738 pcm
->private_data
= chip
;
739 strcpy(pcm
->name
, "SGI O2 DAC1");
742 snd_pcm_set_ops(pcm
, SNDRV_PCM_STREAM_PLAYBACK
,
743 &snd_sgio2audio_playback1_ops
);
744 snd_pcm_set_ops(pcm
, SNDRV_PCM_STREAM_CAPTURE
,
745 &snd_sgio2audio_capture_ops
);
747 /* create second pcm device with one outputs and no input */
748 err
= snd_pcm_new(chip
->card
, "SGI O2 Audio", 1, 1, 0, &pcm
);
752 pcm
->private_data
= chip
;
753 strcpy(pcm
->name
, "SGI O2 DAC2");
756 snd_pcm_set_ops(pcm
, SNDRV_PCM_STREAM_PLAYBACK
,
757 &snd_sgio2audio_playback2_ops
);
765 irqreturn_t (*isr
)(int, void *);
767 } snd_sgio2_isr_table
[] = {
770 .irq
= MACEISA_AUDIO1_DMAT_IRQ
,
771 .isr
= snd_sgio2audio_dma_in_isr
,
772 .desc
= "Capture DMA Channel 0"
775 .irq
= MACEISA_AUDIO1_OF_IRQ
,
776 .isr
= snd_sgio2audio_error_isr
,
777 .desc
= "Capture Overflow"
780 .irq
= MACEISA_AUDIO2_DMAT_IRQ
,
781 .isr
= snd_sgio2audio_dma_out_isr
,
782 .desc
= "Playback DMA Channel 1"
785 .irq
= MACEISA_AUDIO2_MERR_IRQ
,
786 .isr
= snd_sgio2audio_error_isr
,
787 .desc
= "Memory Error Channel 1"
790 .irq
= MACEISA_AUDIO3_DMAT_IRQ
,
791 .isr
= snd_sgio2audio_dma_out_isr
,
792 .desc
= "Playback DMA Channel 2"
795 .irq
= MACEISA_AUDIO3_MERR_IRQ
,
796 .isr
= snd_sgio2audio_error_isr
,
797 .desc
= "Memory Error Channel 2"
803 static int snd_sgio2audio_free(struct snd_sgio2audio
*chip
)
807 /* reset interface */
808 writeq(AUDIO_CONTROL_RESET
, &mace
->perif
.audio
.control
);
810 writeq(0, &mace
->perif
.audio
.control
);
813 for (i
= 0; i
< ARRAY_SIZE(snd_sgio2_isr_table
); i
++)
814 free_irq(snd_sgio2_isr_table
[i
].irq
,
815 &chip
->channel
[snd_sgio2_isr_table
[i
].idx
]);
817 dma_free_coherent(NULL
, MACEISA_RINGBUFFERS_SIZE
,
818 chip
->ring_base
, chip
->ring_base_dma
);
820 /* release card data */
825 static int snd_sgio2audio_dev_free(struct snd_device
*device
)
827 struct snd_sgio2audio
*chip
= device
->device_data
;
829 return snd_sgio2audio_free(chip
);
832 static struct snd_device_ops ops
= {
833 .dev_free
= snd_sgio2audio_dev_free
,
836 static int __devinit
snd_sgio2audio_create(struct snd_card
*card
,
837 struct snd_sgio2audio
**rchip
)
839 struct snd_sgio2audio
*chip
;
844 /* check if a codec is attached to the interface */
845 /* (Audio or Audio/Video board present) */
846 if (!(readq(&mace
->perif
.audio
.control
) & AUDIO_CONTROL_CODEC_PRESENT
))
849 chip
= kzalloc(sizeof(struct snd_sgio2audio
), GFP_KERNEL
);
855 chip
->ring_base
= dma_alloc_coherent(NULL
, MACEISA_RINGBUFFERS_SIZE
,
856 &chip
->ring_base_dma
, GFP_USER
);
857 if (chip
->ring_base
== NULL
) {
859 "sgio2audio: could not allocate ring buffers\n");
864 spin_lock_init(&chip
->ad1843_lock
);
866 /* initialize channels */
867 for (i
= 0; i
< 3; i
++) {
868 spin_lock_init(&chip
->channel
[i
].lock
);
869 chip
->channel
[i
].idx
= i
;
873 for (i
= 0; i
< ARRAY_SIZE(snd_sgio2_isr_table
); i
++) {
874 if (request_irq(snd_sgio2_isr_table
[i
].irq
,
875 snd_sgio2_isr_table
[i
].isr
,
877 snd_sgio2_isr_table
[i
].desc
,
878 &chip
->channel
[snd_sgio2_isr_table
[i
].idx
])) {
879 snd_sgio2audio_free(chip
);
880 printk(KERN_ERR
"sgio2audio: cannot allocate irq %d\n",
881 snd_sgio2_isr_table
[i
].irq
);
886 /* reset the interface */
887 writeq(AUDIO_CONTROL_RESET
, &mace
->perif
.audio
.control
);
889 writeq(0, &mace
->perif
.audio
.control
);
890 msleep_interruptible(1); /* give time to recover */
893 writeq(chip
->ring_base_dma
, &mace
->perif
.ctrl
.ringbase
);
895 /* attach the AD1843 codec */
896 chip
->ad1843
.read
= read_ad1843_reg
;
897 chip
->ad1843
.write
= write_ad1843_reg
;
898 chip
->ad1843
.chip
= chip
;
900 /* initialize the AD1843 codec */
901 err
= ad1843_init(&chip
->ad1843
);
903 snd_sgio2audio_free(chip
);
907 err
= snd_device_new(card
, SNDRV_DEV_LOWLEVEL
, chip
, &ops
);
909 snd_sgio2audio_free(chip
);
916 static int __devinit
snd_sgio2audio_probe(struct platform_device
*pdev
)
918 struct snd_card
*card
;
919 struct snd_sgio2audio
*chip
;
922 err
= snd_card_create(index
, id
, THIS_MODULE
, 0, &card
);
926 err
= snd_sgio2audio_create(card
, &chip
);
931 snd_card_set_dev(card
, &pdev
->dev
);
933 err
= snd_sgio2audio_new_pcm(chip
);
938 err
= snd_sgio2audio_new_mixer(chip
);
944 strcpy(card
->driver
, "SGI O2 Audio");
945 strcpy(card
->shortname
, "SGI O2 Audio");
946 sprintf(card
->longname
, "%s irq %i-%i",
948 MACEISA_AUDIO1_DMAT_IRQ
,
949 MACEISA_AUDIO3_MERR_IRQ
);
951 err
= snd_card_register(card
);
956 platform_set_drvdata(pdev
, card
);
960 static int __devexit
snd_sgio2audio_remove(struct platform_device
*pdev
)
962 struct snd_card
*card
= platform_get_drvdata(pdev
);
965 platform_set_drvdata(pdev
, NULL
);
969 static struct platform_driver sgio2audio_driver
= {
970 .probe
= snd_sgio2audio_probe
,
971 .remove
= __devexit_p(snd_sgio2audio_remove
),
973 .name
= "sgio2audio",
974 .owner
= THIS_MODULE
,
978 static int __init
alsa_card_sgio2audio_init(void)
980 return platform_driver_register(&sgio2audio_driver
);
983 static void __exit
alsa_card_sgio2audio_exit(void)
985 platform_driver_unregister(&sgio2audio_driver
);
988 module_init(alsa_card_sgio2audio_init
)
989 module_exit(alsa_card_sgio2audio_exit
)