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>
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>
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
{
87 struct snd_pcm_substream
*substream
;
89 snd_pcm_uframes_t size
;
93 /* definition of the chip-specific record */
94 struct snd_sgio2audio
{
95 struct snd_card
*card
;
98 struct snd_ad1843 ad1843
;
99 spinlock_t ad1843_lock
;
102 struct snd_sgio2audio_chan channel
[3];
106 dma_addr_t ring_base_dma
;
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
;
122 spin_lock_irqsave(&chip
->ad1843_lock
, flags
);
124 writeq((reg
<< CODEC_CONTROL_ADDRESS_SHIFT
) |
125 CODEC_CONTROL_READ
, &mace
->perif
.audio
.codec_control
);
127 val
= readq(&mace
->perif
.audio
.codec_control
); /* flush bus */
130 val
= readq(&mace
->perif
.audio
.codec_read
);
132 spin_unlock_irqrestore(&chip
->ad1843_lock
, flags
);
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
;
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
);
151 val
= readq(&mace
->perif
.audio
.codec_control
); /* flush bus */
154 spin_unlock_irqrestore(&chip
->ad1843_lock
, flags
);
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
;
165 uinfo
->value
.integer
.min
= 0;
166 uinfo
->value
.integer
.max
= ad1843_get_gain_max(&chip
->ad1843
,
167 (int)kcontrol
->private_value
);
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
);
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;
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
);
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
,
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
;
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
]);
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
);
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
);
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",
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",
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",
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",
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
)
324 err
= snd_ctl_add(chip
->card
,
325 snd_ctl_new1(&sgio2audio_ctrl_pcm0
, chip
));
329 err
= snd_ctl_add(chip
->card
,
330 snd_ctl_new1(&sgio2audio_ctrl_pcm1
, chip
));
334 err
= snd_ctl_add(chip
->card
,
335 snd_ctl_new1(&sgio2audio_ctrl_reclevel
, chip
));
339 err
= snd_ctl_add(chip
->card
,
340 snd_ctl_new1(&sgio2audio_ctrl_recsource
, chip
));
343 err
= snd_ctl_add(chip
->card
,
344 snd_ctl_new1(&sgio2audio_ctrl_line
, chip
));
348 err
= snd_ctl_add(chip
->card
,
349 snd_ctl_new1(&sgio2audio_ctrl_cd
, chip
));
353 err
= snd_ctl_add(chip
->card
,
354 snd_ctl_new1(&sgio2audio_ctrl_mic
, chip
));
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
)
369 unsigned long src_base
, src_pos
, dst_mask
;
370 unsigned char *dst_base
;
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
;
392 src
= (u64
*)(src_base
+ src_pos
);
393 dst
= (s16
*)(dst_base
+ dst_pos
);
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
);
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
)
418 unsigned long dst_base
, dst_pos
, src_mask
;
419 unsigned char *src_base
;
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
;
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
);
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
;
467 /* reset DMA channel */
468 writeq(CHANNEL_CONTROL_RESET
, &mace
->perif
.audio
.chan
[ch
].control
);
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
);
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
);
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
;
497 substream
= chan
->substream
;
498 chip
= snd_pcm_substream_chip(substream
);
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
);
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
;
517 substream
= chan
->substream
;
518 chip
= snd_pcm_substream_chip(substream
);
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
);
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
);
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
,
553 .buffer_bytes_max
= 65536,
554 .period_bytes_min
= 32768,
555 .period_bytes_max
= 65536,
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];
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];
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];
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
;
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
)
615 runtime
->dma_bytes
= size
;
619 /* hw_free callback */
620 static int snd_sgio2audio_pcm_hw_free(struct snd_pcm_substream
*substream
)
622 if (substream
->runtime
->dma_area
)
623 vfree(substream
->runtime
->dma_area
);
624 substream
->runtime
->dma_area
= NULL
;
628 /* prepare callback */
629 static int snd_sgio2audio_pcm_prepare(struct snd_pcm_substream
*substream
)
631 struct snd_sgio2audio
*chip
= snd_pcm_substream_chip(substream
);
632 struct snd_pcm_runtime
*runtime
= substream
->runtime
;
633 struct snd_sgio2audio_chan
*chan
= substream
->runtime
->private_data
;
637 spin_lock_irqsave(&chip
->channel
[ch
].lock
, flags
);
639 /* Setup the pseudo-dma transfer pointers. */
640 chip
->channel
[ch
].pos
= 0;
641 chip
->channel
[ch
].size
= 0;
642 chip
->channel
[ch
].substream
= substream
;
644 /* set AD1843 format */
645 /* hardware format is always S16_LE */
646 switch (substream
->stream
) {
647 case SNDRV_PCM_STREAM_PLAYBACK
:
648 ad1843_setup_dac(&chip
->ad1843
,
651 SNDRV_PCM_FORMAT_S16_LE
,
654 case SNDRV_PCM_STREAM_CAPTURE
:
655 ad1843_setup_adc(&chip
->ad1843
,
657 SNDRV_PCM_FORMAT_S16_LE
,
661 spin_unlock_irqrestore(&chip
->channel
[ch
].lock
, flags
);
665 /* trigger callback */
666 static int snd_sgio2audio_pcm_trigger(struct snd_pcm_substream
*substream
,
670 case SNDRV_PCM_TRIGGER_START
:
671 /* start the PCM engine */
672 snd_sgio2audio_dma_start(substream
);
674 case SNDRV_PCM_TRIGGER_STOP
:
675 /* stop the PCM engine */
676 snd_sgio2audio_dma_stop(substream
);
684 /* pointer callback */
685 static snd_pcm_uframes_t
686 snd_sgio2audio_pcm_pointer(struct snd_pcm_substream
*substream
)
688 struct snd_sgio2audio
*chip
= snd_pcm_substream_chip(substream
);
689 struct snd_sgio2audio_chan
*chan
= substream
->runtime
->private_data
;
691 /* get the current hardware pointer */
692 return bytes_to_frames(substream
->runtime
,
693 chip
->channel
[chan
->idx
].pos
);
696 /* get the physical page pointer on the given offset */
697 static struct page
*snd_sgio2audio_page(struct snd_pcm_substream
*substream
,
698 unsigned long offset
)
700 return vmalloc_to_page(substream
->runtime
->dma_area
+ offset
);
704 static struct snd_pcm_ops snd_sgio2audio_playback1_ops
= {
705 .open
= snd_sgio2audio_playback1_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_sgio2audio_page
,
716 static struct snd_pcm_ops snd_sgio2audio_playback2_ops
= {
717 .open
= snd_sgio2audio_playback2_open
,
718 .close
= snd_sgio2audio_pcm_close
,
719 .ioctl
= snd_pcm_lib_ioctl
,
720 .hw_params
= snd_sgio2audio_pcm_hw_params
,
721 .hw_free
= snd_sgio2audio_pcm_hw_free
,
722 .prepare
= snd_sgio2audio_pcm_prepare
,
723 .trigger
= snd_sgio2audio_pcm_trigger
,
724 .pointer
= snd_sgio2audio_pcm_pointer
,
725 .page
= snd_sgio2audio_page
,
728 static struct snd_pcm_ops snd_sgio2audio_capture_ops
= {
729 .open
= snd_sgio2audio_capture_open
,
730 .close
= snd_sgio2audio_pcm_close
,
731 .ioctl
= snd_pcm_lib_ioctl
,
732 .hw_params
= snd_sgio2audio_pcm_hw_params
,
733 .hw_free
= snd_sgio2audio_pcm_hw_free
,
734 .prepare
= snd_sgio2audio_pcm_prepare
,
735 .trigger
= snd_sgio2audio_pcm_trigger
,
736 .pointer
= snd_sgio2audio_pcm_pointer
,
737 .page
= snd_sgio2audio_page
,
741 * definitions of capture are omitted here...
744 /* create a pcm device */
745 static int __devinit
snd_sgio2audio_new_pcm(struct snd_sgio2audio
*chip
)
750 /* create first pcm device with one outputs and one input */
751 err
= snd_pcm_new(chip
->card
, "SGI O2 Audio", 0, 1, 1, &pcm
);
755 pcm
->private_data
= chip
;
756 strcpy(pcm
->name
, "SGI O2 DAC1");
759 snd_pcm_set_ops(pcm
, SNDRV_PCM_STREAM_PLAYBACK
,
760 &snd_sgio2audio_playback1_ops
);
761 snd_pcm_set_ops(pcm
, SNDRV_PCM_STREAM_CAPTURE
,
762 &snd_sgio2audio_capture_ops
);
764 /* create second pcm device with one outputs and no input */
765 err
= snd_pcm_new(chip
->card
, "SGI O2 Audio", 1, 1, 0, &pcm
);
769 pcm
->private_data
= chip
;
770 strcpy(pcm
->name
, "SGI O2 DAC2");
773 snd_pcm_set_ops(pcm
, SNDRV_PCM_STREAM_PLAYBACK
,
774 &snd_sgio2audio_playback2_ops
);
782 irqreturn_t (*isr
)(int, void *);
784 } snd_sgio2_isr_table
[] = {
787 .irq
= MACEISA_AUDIO1_DMAT_IRQ
,
788 .isr
= snd_sgio2audio_dma_in_isr
,
789 .desc
= "Capture DMA Channel 0"
792 .irq
= MACEISA_AUDIO1_OF_IRQ
,
793 .isr
= snd_sgio2audio_error_isr
,
794 .desc
= "Capture Overflow"
797 .irq
= MACEISA_AUDIO2_DMAT_IRQ
,
798 .isr
= snd_sgio2audio_dma_out_isr
,
799 .desc
= "Playback DMA Channel 1"
802 .irq
= MACEISA_AUDIO2_MERR_IRQ
,
803 .isr
= snd_sgio2audio_error_isr
,
804 .desc
= "Memory Error Channel 1"
807 .irq
= MACEISA_AUDIO3_DMAT_IRQ
,
808 .isr
= snd_sgio2audio_dma_out_isr
,
809 .desc
= "Playback DMA Channel 2"
812 .irq
= MACEISA_AUDIO3_MERR_IRQ
,
813 .isr
= snd_sgio2audio_error_isr
,
814 .desc
= "Memory Error Channel 2"
820 static int snd_sgio2audio_free(struct snd_sgio2audio
*chip
)
824 /* reset interface */
825 writeq(AUDIO_CONTROL_RESET
, &mace
->perif
.audio
.control
);
827 writeq(0, &mace
->perif
.audio
.control
);
830 for (i
= 0; i
< ARRAY_SIZE(snd_sgio2_isr_table
); i
++)
831 free_irq(snd_sgio2_isr_table
[i
].irq
,
832 &chip
->channel
[snd_sgio2_isr_table
[i
].idx
]);
834 dma_free_coherent(NULL
, MACEISA_RINGBUFFERS_SIZE
,
835 chip
->ring_base
, chip
->ring_base_dma
);
837 /* release card data */
842 static int snd_sgio2audio_dev_free(struct snd_device
*device
)
844 struct snd_sgio2audio
*chip
= device
->device_data
;
846 return snd_sgio2audio_free(chip
);
849 static struct snd_device_ops ops
= {
850 .dev_free
= snd_sgio2audio_dev_free
,
853 static int __devinit
snd_sgio2audio_create(struct snd_card
*card
,
854 struct snd_sgio2audio
**rchip
)
856 struct snd_sgio2audio
*chip
;
861 /* check if a codec is attached to the interface */
862 /* (Audio or Audio/Video board present) */
863 if (!(readq(&mace
->perif
.audio
.control
) & AUDIO_CONTROL_CODEC_PRESENT
))
866 chip
= kzalloc(sizeof(struct snd_sgio2audio
), GFP_KERNEL
);
872 chip
->ring_base
= dma_alloc_coherent(NULL
, MACEISA_RINGBUFFERS_SIZE
,
873 &chip
->ring_base_dma
, GFP_USER
);
874 if (chip
->ring_base
== NULL
) {
876 "sgio2audio: could not allocate ring buffers\n");
881 spin_lock_init(&chip
->ad1843_lock
);
883 /* initialize channels */
884 for (i
= 0; i
< 3; i
++) {
885 spin_lock_init(&chip
->channel
[i
].lock
);
886 chip
->channel
[i
].idx
= i
;
890 for (i
= 0; i
< ARRAY_SIZE(snd_sgio2_isr_table
); i
++) {
891 if (request_irq(snd_sgio2_isr_table
[i
].irq
,
892 snd_sgio2_isr_table
[i
].isr
,
894 snd_sgio2_isr_table
[i
].desc
,
895 &chip
->channel
[snd_sgio2_isr_table
[i
].idx
])) {
896 snd_sgio2audio_free(chip
);
897 printk(KERN_ERR
"sgio2audio: cannot allocate irq %d\n",
898 snd_sgio2_isr_table
[i
].irq
);
903 /* reset the interface */
904 writeq(AUDIO_CONTROL_RESET
, &mace
->perif
.audio
.control
);
906 writeq(0, &mace
->perif
.audio
.control
);
907 msleep_interruptible(1); /* give time to recover */
910 writeq(chip
->ring_base_dma
, &mace
->perif
.ctrl
.ringbase
);
912 /* attach the AD1843 codec */
913 chip
->ad1843
.read
= read_ad1843_reg
;
914 chip
->ad1843
.write
= write_ad1843_reg
;
915 chip
->ad1843
.chip
= chip
;
917 /* initialize the AD1843 codec */
918 err
= ad1843_init(&chip
->ad1843
);
920 snd_sgio2audio_free(chip
);
924 err
= snd_device_new(card
, SNDRV_DEV_LOWLEVEL
, chip
, &ops
);
926 snd_sgio2audio_free(chip
);
933 static int __devinit
snd_sgio2audio_probe(struct platform_device
*pdev
)
935 struct snd_card
*card
;
936 struct snd_sgio2audio
*chip
;
939 card
= snd_card_new(index
, id
, THIS_MODULE
, 0);
943 err
= snd_sgio2audio_create(card
, &chip
);
948 snd_card_set_dev(card
, &pdev
->dev
);
950 err
= snd_sgio2audio_new_pcm(chip
);
955 err
= snd_sgio2audio_new_mixer(chip
);
961 strcpy(card
->driver
, "SGI O2 Audio");
962 strcpy(card
->shortname
, "SGI O2 Audio");
963 sprintf(card
->longname
, "%s irq %i-%i",
965 MACEISA_AUDIO1_DMAT_IRQ
,
966 MACEISA_AUDIO3_MERR_IRQ
);
968 err
= snd_card_register(card
);
973 platform_set_drvdata(pdev
, card
);
977 static int __exit
snd_sgio2audio_remove(struct platform_device
*pdev
)
979 struct snd_card
*card
= platform_get_drvdata(pdev
);
982 platform_set_drvdata(pdev
, NULL
);
986 static struct platform_driver sgio2audio_driver
= {
987 .probe
= snd_sgio2audio_probe
,
988 .remove
= __devexit_p(snd_sgio2audio_remove
),
990 .name
= "sgio2audio",
991 .owner
= THIS_MODULE
,
995 static int __init
alsa_card_sgio2audio_init(void)
997 return platform_driver_register(&sgio2audio_driver
);
1000 static void __exit
alsa_card_sgio2audio_exit(void)
1002 platform_driver_unregister(&sgio2audio_driver
);
1005 module_init(alsa_card_sgio2audio_init
)
1006 module_exit(alsa_card_sgio2audio_exit
)