1 | // SPDX-License-Identifier: GPL-2.0-or-later |
2 | /* |
3 | * Sound driver for Silicon Graphics O2 Workstations A/V board audio. |
4 | * |
5 | * Copyright 2003 Vivien Chappelier <vivien.chappelier@linux-mips.org> |
6 | * Copyright 2008 Thomas Bogendoerfer <tsbogend@alpha.franken.de> |
7 | * Mxier part taken from mace_audio.c: |
8 | * Copyright 2007 Thorben Jändling <tj.trevelyan@gmail.com> |
9 | */ |
10 | |
11 | #include <linux/init.h> |
12 | #include <linux/delay.h> |
13 | #include <linux/spinlock.h> |
14 | #include <linux/interrupt.h> |
15 | #include <linux/dma-mapping.h> |
16 | #include <linux/platform_device.h> |
17 | #include <linux/io.h> |
18 | #include <linux/slab.h> |
19 | #include <linux/module.h> |
20 | |
21 | #include <asm/ip32/ip32_ints.h> |
22 | #include <asm/ip32/mace.h> |
23 | |
24 | #include <sound/core.h> |
25 | #include <sound/control.h> |
26 | #include <sound/pcm.h> |
27 | #define SNDRV_GET_ID |
28 | #include <sound/initval.h> |
29 | #include <sound/ad1843.h> |
30 | |
31 | |
32 | MODULE_AUTHOR("Vivien Chappelier <vivien.chappelier@linux-mips.org>" ); |
33 | MODULE_DESCRIPTION("SGI O2 Audio" ); |
34 | MODULE_LICENSE("GPL" ); |
35 | |
36 | static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ |
37 | static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ |
38 | |
39 | module_param(index, int, 0444); |
40 | MODULE_PARM_DESC(index, "Index value for SGI O2 soundcard." ); |
41 | module_param(id, charp, 0444); |
42 | MODULE_PARM_DESC(id, "ID string for SGI O2 soundcard." ); |
43 | |
44 | |
45 | #define AUDIO_CONTROL_RESET BIT(0) /* 1: reset audio interface */ |
46 | #define AUDIO_CONTROL_CODEC_PRESENT BIT(1) /* 1: codec detected */ |
47 | |
48 | #define CODEC_CONTROL_WORD_SHIFT 0 |
49 | #define CODEC_CONTROL_READ BIT(16) |
50 | #define CODEC_CONTROL_ADDRESS_SHIFT 17 |
51 | |
52 | #define CHANNEL_CONTROL_RESET BIT(10) /* 1: reset channel */ |
53 | #define CHANNEL_DMA_ENABLE BIT(9) /* 1: enable DMA transfer */ |
54 | #define CHANNEL_INT_THRESHOLD_DISABLED (0 << 5) /* interrupt disabled */ |
55 | #define CHANNEL_INT_THRESHOLD_25 (1 << 5) /* int on buffer >25% full */ |
56 | #define CHANNEL_INT_THRESHOLD_50 (2 << 5) /* int on buffer >50% full */ |
57 | #define CHANNEL_INT_THRESHOLD_75 (3 << 5) /* int on buffer >75% full */ |
58 | #define CHANNEL_INT_THRESHOLD_EMPTY (4 << 5) /* int on buffer empty */ |
59 | #define CHANNEL_INT_THRESHOLD_NOT_EMPTY (5 << 5) /* int on buffer !empty */ |
60 | #define CHANNEL_INT_THRESHOLD_FULL (6 << 5) /* int on buffer empty */ |
61 | #define CHANNEL_INT_THRESHOLD_NOT_FULL (7 << 5) /* int on buffer !empty */ |
62 | |
63 | #define CHANNEL_RING_SHIFT 12 |
64 | #define CHANNEL_RING_SIZE (1 << CHANNEL_RING_SHIFT) |
65 | #define CHANNEL_RING_MASK (CHANNEL_RING_SIZE - 1) |
66 | |
67 | #define CHANNEL_LEFT_SHIFT 40 |
68 | #define CHANNEL_RIGHT_SHIFT 8 |
69 | |
70 | struct snd_sgio2audio_chan { |
71 | int idx; |
72 | struct snd_pcm_substream *substream; |
73 | int pos; |
74 | snd_pcm_uframes_t size; |
75 | spinlock_t lock; |
76 | }; |
77 | |
78 | /* definition of the chip-specific record */ |
79 | struct snd_sgio2audio { |
80 | struct snd_card *card; |
81 | |
82 | /* codec */ |
83 | struct snd_ad1843 ad1843; |
84 | spinlock_t ad1843_lock; |
85 | |
86 | /* channels */ |
87 | struct snd_sgio2audio_chan channel[3]; |
88 | |
89 | /* resources */ |
90 | void *ring_base; |
91 | dma_addr_t ring_base_dma; |
92 | }; |
93 | |
94 | /* AD1843 access */ |
95 | |
96 | /* |
97 | * read_ad1843_reg returns the current contents of a 16 bit AD1843 register. |
98 | * |
99 | * Returns unsigned register value on success, -errno on failure. |
100 | */ |
101 | static int read_ad1843_reg(void *priv, int reg) |
102 | { |
103 | struct snd_sgio2audio *chip = priv; |
104 | int val; |
105 | unsigned long flags; |
106 | |
107 | spin_lock_irqsave(&chip->ad1843_lock, flags); |
108 | |
109 | writeq(val: (reg << CODEC_CONTROL_ADDRESS_SHIFT) | |
110 | CODEC_CONTROL_READ, addr: &mace->perif.audio.codec_control); |
111 | wmb(); |
112 | val = readq(addr: &mace->perif.audio.codec_control); /* flush bus */ |
113 | udelay(200); |
114 | |
115 | val = readq(addr: &mace->perif.audio.codec_read); |
116 | |
117 | spin_unlock_irqrestore(lock: &chip->ad1843_lock, flags); |
118 | return val; |
119 | } |
120 | |
121 | /* |
122 | * write_ad1843_reg writes the specified value to a 16 bit AD1843 register. |
123 | */ |
124 | static int write_ad1843_reg(void *priv, int reg, int word) |
125 | { |
126 | struct snd_sgio2audio *chip = priv; |
127 | int val; |
128 | unsigned long flags; |
129 | |
130 | spin_lock_irqsave(&chip->ad1843_lock, flags); |
131 | |
132 | writeq(val: (reg << CODEC_CONTROL_ADDRESS_SHIFT) | |
133 | (word << CODEC_CONTROL_WORD_SHIFT), |
134 | addr: &mace->perif.audio.codec_control); |
135 | wmb(); |
136 | val = readq(addr: &mace->perif.audio.codec_control); /* flush bus */ |
137 | udelay(200); |
138 | |
139 | spin_unlock_irqrestore(lock: &chip->ad1843_lock, flags); |
140 | return 0; |
141 | } |
142 | |
143 | static int sgio2audio_gain_info(struct snd_kcontrol *kcontrol, |
144 | struct snd_ctl_elem_info *uinfo) |
145 | { |
146 | struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol); |
147 | |
148 | uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; |
149 | uinfo->count = 2; |
150 | uinfo->value.integer.min = 0; |
151 | uinfo->value.integer.max = ad1843_get_gain_max(ad1843: &chip->ad1843, |
152 | id: (int)kcontrol->private_value); |
153 | return 0; |
154 | } |
155 | |
156 | static int sgio2audio_gain_get(struct snd_kcontrol *kcontrol, |
157 | struct snd_ctl_elem_value *ucontrol) |
158 | { |
159 | struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol); |
160 | int vol; |
161 | |
162 | vol = ad1843_get_gain(ad1843: &chip->ad1843, id: (int)kcontrol->private_value); |
163 | |
164 | ucontrol->value.integer.value[0] = (vol >> 8) & 0xFF; |
165 | ucontrol->value.integer.value[1] = vol & 0xFF; |
166 | |
167 | return 0; |
168 | } |
169 | |
170 | static int sgio2audio_gain_put(struct snd_kcontrol *kcontrol, |
171 | struct snd_ctl_elem_value *ucontrol) |
172 | { |
173 | struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol); |
174 | int newvol, oldvol; |
175 | |
176 | oldvol = ad1843_get_gain(ad1843: &chip->ad1843, id: kcontrol->private_value); |
177 | newvol = (ucontrol->value.integer.value[0] << 8) | |
178 | ucontrol->value.integer.value[1]; |
179 | |
180 | newvol = ad1843_set_gain(ad1843: &chip->ad1843, id: kcontrol->private_value, |
181 | newval: newvol); |
182 | |
183 | return newvol != oldvol; |
184 | } |
185 | |
186 | static int sgio2audio_source_info(struct snd_kcontrol *kcontrol, |
187 | struct snd_ctl_elem_info *uinfo) |
188 | { |
189 | static const char * const texts[3] = { |
190 | "Cam Mic" , "Mic" , "Line" |
191 | }; |
192 | return snd_ctl_enum_info(info: uinfo, channels: 1, items: 3, names: texts); |
193 | } |
194 | |
195 | static int sgio2audio_source_get(struct snd_kcontrol *kcontrol, |
196 | struct snd_ctl_elem_value *ucontrol) |
197 | { |
198 | struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol); |
199 | |
200 | ucontrol->value.enumerated.item[0] = ad1843_get_recsrc(ad1843: &chip->ad1843); |
201 | return 0; |
202 | } |
203 | |
204 | static int sgio2audio_source_put(struct snd_kcontrol *kcontrol, |
205 | struct snd_ctl_elem_value *ucontrol) |
206 | { |
207 | struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol); |
208 | int newsrc, oldsrc; |
209 | |
210 | oldsrc = ad1843_get_recsrc(ad1843: &chip->ad1843); |
211 | newsrc = ad1843_set_recsrc(ad1843: &chip->ad1843, |
212 | newsrc: ucontrol->value.enumerated.item[0]); |
213 | |
214 | return newsrc != oldsrc; |
215 | } |
216 | |
217 | /* dac1/pcm0 mixer control */ |
218 | static const struct snd_kcontrol_new sgio2audio_ctrl_pcm0 = { |
219 | .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
220 | .name = "PCM Playback Volume" , |
221 | .index = 0, |
222 | .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, |
223 | .private_value = AD1843_GAIN_PCM_0, |
224 | .info = sgio2audio_gain_info, |
225 | .get = sgio2audio_gain_get, |
226 | .put = sgio2audio_gain_put, |
227 | }; |
228 | |
229 | /* dac2/pcm1 mixer control */ |
230 | static const struct snd_kcontrol_new sgio2audio_ctrl_pcm1 = { |
231 | .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
232 | .name = "PCM Playback Volume" , |
233 | .index = 1, |
234 | .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, |
235 | .private_value = AD1843_GAIN_PCM_1, |
236 | .info = sgio2audio_gain_info, |
237 | .get = sgio2audio_gain_get, |
238 | .put = sgio2audio_gain_put, |
239 | }; |
240 | |
241 | /* record level mixer control */ |
242 | static const struct snd_kcontrol_new sgio2audio_ctrl_reclevel = { |
243 | .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
244 | .name = "Capture Volume" , |
245 | .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, |
246 | .private_value = AD1843_GAIN_RECLEV, |
247 | .info = sgio2audio_gain_info, |
248 | .get = sgio2audio_gain_get, |
249 | .put = sgio2audio_gain_put, |
250 | }; |
251 | |
252 | /* record level source control */ |
253 | static const struct snd_kcontrol_new sgio2audio_ctrl_recsource = { |
254 | .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
255 | .name = "Capture Source" , |
256 | .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, |
257 | .info = sgio2audio_source_info, |
258 | .get = sgio2audio_source_get, |
259 | .put = sgio2audio_source_put, |
260 | }; |
261 | |
262 | /* line mixer control */ |
263 | static const struct snd_kcontrol_new sgio2audio_ctrl_line = { |
264 | .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
265 | .name = "Line Playback Volume" , |
266 | .index = 0, |
267 | .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, |
268 | .private_value = AD1843_GAIN_LINE, |
269 | .info = sgio2audio_gain_info, |
270 | .get = sgio2audio_gain_get, |
271 | .put = sgio2audio_gain_put, |
272 | }; |
273 | |
274 | /* cd mixer control */ |
275 | static const struct snd_kcontrol_new sgio2audio_ctrl_cd = { |
276 | .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
277 | .name = "Line Playback Volume" , |
278 | .index = 1, |
279 | .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, |
280 | .private_value = AD1843_GAIN_LINE_2, |
281 | .info = sgio2audio_gain_info, |
282 | .get = sgio2audio_gain_get, |
283 | .put = sgio2audio_gain_put, |
284 | }; |
285 | |
286 | /* mic mixer control */ |
287 | static const struct snd_kcontrol_new sgio2audio_ctrl_mic = { |
288 | .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
289 | .name = "Mic Playback Volume" , |
290 | .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, |
291 | .private_value = AD1843_GAIN_MIC, |
292 | .info = sgio2audio_gain_info, |
293 | .get = sgio2audio_gain_get, |
294 | .put = sgio2audio_gain_put, |
295 | }; |
296 | |
297 | |
298 | static int snd_sgio2audio_new_mixer(struct snd_sgio2audio *chip) |
299 | { |
300 | int err; |
301 | |
302 | err = snd_ctl_add(card: chip->card, |
303 | kcontrol: snd_ctl_new1(kcontrolnew: &sgio2audio_ctrl_pcm0, private_data: chip)); |
304 | if (err < 0) |
305 | return err; |
306 | |
307 | err = snd_ctl_add(card: chip->card, |
308 | kcontrol: snd_ctl_new1(kcontrolnew: &sgio2audio_ctrl_pcm1, private_data: chip)); |
309 | if (err < 0) |
310 | return err; |
311 | |
312 | err = snd_ctl_add(card: chip->card, |
313 | kcontrol: snd_ctl_new1(kcontrolnew: &sgio2audio_ctrl_reclevel, private_data: chip)); |
314 | if (err < 0) |
315 | return err; |
316 | |
317 | err = snd_ctl_add(card: chip->card, |
318 | kcontrol: snd_ctl_new1(kcontrolnew: &sgio2audio_ctrl_recsource, private_data: chip)); |
319 | if (err < 0) |
320 | return err; |
321 | err = snd_ctl_add(card: chip->card, |
322 | kcontrol: snd_ctl_new1(kcontrolnew: &sgio2audio_ctrl_line, private_data: chip)); |
323 | if (err < 0) |
324 | return err; |
325 | |
326 | err = snd_ctl_add(card: chip->card, |
327 | kcontrol: snd_ctl_new1(kcontrolnew: &sgio2audio_ctrl_cd, private_data: chip)); |
328 | if (err < 0) |
329 | return err; |
330 | |
331 | err = snd_ctl_add(card: chip->card, |
332 | kcontrol: snd_ctl_new1(kcontrolnew: &sgio2audio_ctrl_mic, private_data: chip)); |
333 | if (err < 0) |
334 | return err; |
335 | |
336 | return 0; |
337 | } |
338 | |
339 | /* low-level audio interface DMA */ |
340 | |
341 | /* get data out of bounce buffer, count must be a multiple of 32 */ |
342 | /* returns 1 if a period has elapsed */ |
343 | static int snd_sgio2audio_dma_pull_frag(struct snd_sgio2audio *chip, |
344 | unsigned int ch, unsigned int count) |
345 | { |
346 | int ret; |
347 | unsigned long src_base, src_pos, dst_mask; |
348 | unsigned char *dst_base; |
349 | int dst_pos; |
350 | u64 *src; |
351 | s16 *dst; |
352 | u64 x; |
353 | unsigned long flags; |
354 | struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime; |
355 | |
356 | spin_lock_irqsave(&chip->channel[ch].lock, flags); |
357 | |
358 | src_base = (unsigned long) chip->ring_base | (ch << CHANNEL_RING_SHIFT); |
359 | src_pos = readq(&mace->perif.audio.chan[ch].read_ptr); |
360 | dst_base = runtime->dma_area; |
361 | dst_pos = chip->channel[ch].pos; |
362 | dst_mask = frames_to_bytes(runtime, size: runtime->buffer_size) - 1; |
363 | |
364 | /* check if a period has elapsed */ |
365 | chip->channel[ch].size += (count >> 3); /* in frames */ |
366 | ret = chip->channel[ch].size >= runtime->period_size; |
367 | chip->channel[ch].size %= runtime->period_size; |
368 | |
369 | while (count) { |
370 | src = (u64 *)(src_base + src_pos); |
371 | dst = (s16 *)(dst_base + dst_pos); |
372 | |
373 | x = *src; |
374 | dst[0] = (x >> CHANNEL_LEFT_SHIFT) & 0xffff; |
375 | dst[1] = (x >> CHANNEL_RIGHT_SHIFT) & 0xffff; |
376 | |
377 | src_pos = (src_pos + sizeof(u64)) & CHANNEL_RING_MASK; |
378 | dst_pos = (dst_pos + 2 * sizeof(s16)) & dst_mask; |
379 | count -= sizeof(u64); |
380 | } |
381 | |
382 | writeq(src_pos, &mace->perif.audio.chan[ch].read_ptr); /* in bytes */ |
383 | chip->channel[ch].pos = dst_pos; |
384 | |
385 | spin_unlock_irqrestore(lock: &chip->channel[ch].lock, flags); |
386 | return ret; |
387 | } |
388 | |
389 | /* put some DMA data in bounce buffer, count must be a multiple of 32 */ |
390 | /* returns 1 if a period has elapsed */ |
391 | static int snd_sgio2audio_dma_push_frag(struct snd_sgio2audio *chip, |
392 | unsigned int ch, unsigned int count) |
393 | { |
394 | int ret; |
395 | s64 l, r; |
396 | unsigned long dst_base, dst_pos, src_mask; |
397 | unsigned char *src_base; |
398 | int src_pos; |
399 | u64 *dst; |
400 | s16 *src; |
401 | unsigned long flags; |
402 | struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime; |
403 | |
404 | spin_lock_irqsave(&chip->channel[ch].lock, flags); |
405 | |
406 | dst_base = (unsigned long)chip->ring_base | (ch << CHANNEL_RING_SHIFT); |
407 | dst_pos = readq(&mace->perif.audio.chan[ch].write_ptr); |
408 | src_base = runtime->dma_area; |
409 | src_pos = chip->channel[ch].pos; |
410 | src_mask = frames_to_bytes(runtime, size: runtime->buffer_size) - 1; |
411 | |
412 | /* check if a period has elapsed */ |
413 | chip->channel[ch].size += (count >> 3); /* in frames */ |
414 | ret = chip->channel[ch].size >= runtime->period_size; |
415 | chip->channel[ch].size %= runtime->period_size; |
416 | |
417 | while (count) { |
418 | src = (s16 *)(src_base + src_pos); |
419 | dst = (u64 *)(dst_base + dst_pos); |
420 | |
421 | l = src[0]; /* sign extend */ |
422 | r = src[1]; /* sign extend */ |
423 | |
424 | *dst = ((l & 0x00ffffff) << CHANNEL_LEFT_SHIFT) | |
425 | ((r & 0x00ffffff) << CHANNEL_RIGHT_SHIFT); |
426 | |
427 | dst_pos = (dst_pos + sizeof(u64)) & CHANNEL_RING_MASK; |
428 | src_pos = (src_pos + 2 * sizeof(s16)) & src_mask; |
429 | count -= sizeof(u64); |
430 | } |
431 | |
432 | writeq(dst_pos, &mace->perif.audio.chan[ch].write_ptr); /* in bytes */ |
433 | chip->channel[ch].pos = src_pos; |
434 | |
435 | spin_unlock_irqrestore(lock: &chip->channel[ch].lock, flags); |
436 | return ret; |
437 | } |
438 | |
439 | static int snd_sgio2audio_dma_start(struct snd_pcm_substream *substream) |
440 | { |
441 | struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); |
442 | struct snd_sgio2audio_chan *chan = substream->runtime->private_data; |
443 | int ch = chan->idx; |
444 | |
445 | /* reset DMA channel */ |
446 | writeq(CHANNEL_CONTROL_RESET, &mace->perif.audio.chan[ch].control); |
447 | udelay(10); |
448 | writeq(0, &mace->perif.audio.chan[ch].control); |
449 | |
450 | if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { |
451 | /* push a full buffer */ |
452 | snd_sgio2audio_dma_push_frag(chip, ch, CHANNEL_RING_SIZE - 32); |
453 | } |
454 | /* set DMA to wake on 50% empty and enable interrupt */ |
455 | writeq(CHANNEL_DMA_ENABLE | CHANNEL_INT_THRESHOLD_50, |
456 | &mace->perif.audio.chan[ch].control); |
457 | return 0; |
458 | } |
459 | |
460 | static int snd_sgio2audio_dma_stop(struct snd_pcm_substream *substream) |
461 | { |
462 | struct snd_sgio2audio_chan *chan = substream->runtime->private_data; |
463 | |
464 | writeq(0, &mace->perif.audio.chan[chan->idx].control); |
465 | return 0; |
466 | } |
467 | |
468 | static irqreturn_t snd_sgio2audio_dma_in_isr(int irq, void *dev_id) |
469 | { |
470 | struct snd_sgio2audio_chan *chan = dev_id; |
471 | struct snd_pcm_substream *substream; |
472 | struct snd_sgio2audio *chip; |
473 | int count, ch; |
474 | |
475 | substream = chan->substream; |
476 | chip = snd_pcm_substream_chip(substream); |
477 | ch = chan->idx; |
478 | |
479 | /* empty the ring */ |
480 | count = CHANNEL_RING_SIZE - |
481 | readq(&mace->perif.audio.chan[ch].depth) - 32; |
482 | if (snd_sgio2audio_dma_pull_frag(chip, ch, count)) |
483 | snd_pcm_period_elapsed(substream); |
484 | |
485 | return IRQ_HANDLED; |
486 | } |
487 | |
488 | static irqreturn_t snd_sgio2audio_dma_out_isr(int irq, void *dev_id) |
489 | { |
490 | struct snd_sgio2audio_chan *chan = dev_id; |
491 | struct snd_pcm_substream *substream; |
492 | struct snd_sgio2audio *chip; |
493 | int count, ch; |
494 | |
495 | substream = chan->substream; |
496 | chip = snd_pcm_substream_chip(substream); |
497 | ch = chan->idx; |
498 | /* fill the ring */ |
499 | count = CHANNEL_RING_SIZE - |
500 | readq(&mace->perif.audio.chan[ch].depth) - 32; |
501 | if (snd_sgio2audio_dma_push_frag(chip, ch, count)) |
502 | snd_pcm_period_elapsed(substream); |
503 | |
504 | return IRQ_HANDLED; |
505 | } |
506 | |
507 | static irqreturn_t snd_sgio2audio_error_isr(int irq, void *dev_id) |
508 | { |
509 | struct snd_sgio2audio_chan *chan = dev_id; |
510 | struct snd_pcm_substream *substream; |
511 | |
512 | substream = chan->substream; |
513 | snd_sgio2audio_dma_stop(substream); |
514 | snd_sgio2audio_dma_start(substream); |
515 | return IRQ_HANDLED; |
516 | } |
517 | |
518 | /* PCM part */ |
519 | /* PCM hardware definition */ |
520 | static const struct snd_pcm_hardware snd_sgio2audio_pcm_hw = { |
521 | .info = (SNDRV_PCM_INFO_MMAP | |
522 | SNDRV_PCM_INFO_MMAP_VALID | |
523 | SNDRV_PCM_INFO_INTERLEAVED | |
524 | SNDRV_PCM_INFO_BLOCK_TRANSFER), |
525 | .formats = SNDRV_PCM_FMTBIT_S16_BE, |
526 | .rates = SNDRV_PCM_RATE_8000_48000, |
527 | .rate_min = 8000, |
528 | .rate_max = 48000, |
529 | .channels_min = 2, |
530 | .channels_max = 2, |
531 | .buffer_bytes_max = 65536, |
532 | .period_bytes_min = 32768, |
533 | .period_bytes_max = 65536, |
534 | .periods_min = 1, |
535 | .periods_max = 1024, |
536 | }; |
537 | |
538 | /* PCM playback open callback */ |
539 | static int snd_sgio2audio_playback1_open(struct snd_pcm_substream *substream) |
540 | { |
541 | struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); |
542 | struct snd_pcm_runtime *runtime = substream->runtime; |
543 | |
544 | runtime->hw = snd_sgio2audio_pcm_hw; |
545 | runtime->private_data = &chip->channel[1]; |
546 | return 0; |
547 | } |
548 | |
549 | static int snd_sgio2audio_playback2_open(struct snd_pcm_substream *substream) |
550 | { |
551 | struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); |
552 | struct snd_pcm_runtime *runtime = substream->runtime; |
553 | |
554 | runtime->hw = snd_sgio2audio_pcm_hw; |
555 | runtime->private_data = &chip->channel[2]; |
556 | return 0; |
557 | } |
558 | |
559 | /* PCM capture open callback */ |
560 | static int snd_sgio2audio_capture_open(struct snd_pcm_substream *substream) |
561 | { |
562 | struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); |
563 | struct snd_pcm_runtime *runtime = substream->runtime; |
564 | |
565 | runtime->hw = snd_sgio2audio_pcm_hw; |
566 | runtime->private_data = &chip->channel[0]; |
567 | return 0; |
568 | } |
569 | |
570 | /* PCM close callback */ |
571 | static int snd_sgio2audio_pcm_close(struct snd_pcm_substream *substream) |
572 | { |
573 | struct snd_pcm_runtime *runtime = substream->runtime; |
574 | |
575 | runtime->private_data = NULL; |
576 | return 0; |
577 | } |
578 | |
579 | /* prepare callback */ |
580 | static int snd_sgio2audio_pcm_prepare(struct snd_pcm_substream *substream) |
581 | { |
582 | struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); |
583 | struct snd_pcm_runtime *runtime = substream->runtime; |
584 | struct snd_sgio2audio_chan *chan = substream->runtime->private_data; |
585 | int ch = chan->idx; |
586 | unsigned long flags; |
587 | |
588 | spin_lock_irqsave(&chip->channel[ch].lock, flags); |
589 | |
590 | /* Setup the pseudo-dma transfer pointers. */ |
591 | chip->channel[ch].pos = 0; |
592 | chip->channel[ch].size = 0; |
593 | chip->channel[ch].substream = substream; |
594 | |
595 | /* set AD1843 format */ |
596 | /* hardware format is always S16_LE */ |
597 | switch (substream->stream) { |
598 | case SNDRV_PCM_STREAM_PLAYBACK: |
599 | ad1843_setup_dac(ad1843: &chip->ad1843, |
600 | id: ch - 1, |
601 | framerate: runtime->rate, |
602 | SNDRV_PCM_FORMAT_S16_LE, |
603 | channels: runtime->channels); |
604 | break; |
605 | case SNDRV_PCM_STREAM_CAPTURE: |
606 | ad1843_setup_adc(ad1843: &chip->ad1843, |
607 | framerate: runtime->rate, |
608 | SNDRV_PCM_FORMAT_S16_LE, |
609 | channels: runtime->channels); |
610 | break; |
611 | } |
612 | spin_unlock_irqrestore(lock: &chip->channel[ch].lock, flags); |
613 | return 0; |
614 | } |
615 | |
616 | /* trigger callback */ |
617 | static int snd_sgio2audio_pcm_trigger(struct snd_pcm_substream *substream, |
618 | int cmd) |
619 | { |
620 | switch (cmd) { |
621 | case SNDRV_PCM_TRIGGER_START: |
622 | /* start the PCM engine */ |
623 | snd_sgio2audio_dma_start(substream); |
624 | break; |
625 | case SNDRV_PCM_TRIGGER_STOP: |
626 | /* stop the PCM engine */ |
627 | snd_sgio2audio_dma_stop(substream); |
628 | break; |
629 | default: |
630 | return -EINVAL; |
631 | } |
632 | return 0; |
633 | } |
634 | |
635 | /* pointer callback */ |
636 | static snd_pcm_uframes_t |
637 | snd_sgio2audio_pcm_pointer(struct snd_pcm_substream *substream) |
638 | { |
639 | struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); |
640 | struct snd_sgio2audio_chan *chan = substream->runtime->private_data; |
641 | |
642 | /* get the current hardware pointer */ |
643 | return bytes_to_frames(runtime: substream->runtime, |
644 | size: chip->channel[chan->idx].pos); |
645 | } |
646 | |
647 | /* operators */ |
648 | static const struct snd_pcm_ops snd_sgio2audio_playback1_ops = { |
649 | .open = snd_sgio2audio_playback1_open, |
650 | .close = snd_sgio2audio_pcm_close, |
651 | .prepare = snd_sgio2audio_pcm_prepare, |
652 | .trigger = snd_sgio2audio_pcm_trigger, |
653 | .pointer = snd_sgio2audio_pcm_pointer, |
654 | }; |
655 | |
656 | static const struct snd_pcm_ops snd_sgio2audio_playback2_ops = { |
657 | .open = snd_sgio2audio_playback2_open, |
658 | .close = snd_sgio2audio_pcm_close, |
659 | .prepare = snd_sgio2audio_pcm_prepare, |
660 | .trigger = snd_sgio2audio_pcm_trigger, |
661 | .pointer = snd_sgio2audio_pcm_pointer, |
662 | }; |
663 | |
664 | static const struct snd_pcm_ops snd_sgio2audio_capture_ops = { |
665 | .open = snd_sgio2audio_capture_open, |
666 | .close = snd_sgio2audio_pcm_close, |
667 | .prepare = snd_sgio2audio_pcm_prepare, |
668 | .trigger = snd_sgio2audio_pcm_trigger, |
669 | .pointer = snd_sgio2audio_pcm_pointer, |
670 | }; |
671 | |
672 | /* |
673 | * definitions of capture are omitted here... |
674 | */ |
675 | |
676 | /* create a pcm device */ |
677 | static int snd_sgio2audio_new_pcm(struct snd_sgio2audio *chip) |
678 | { |
679 | struct snd_pcm *pcm; |
680 | int err; |
681 | |
682 | /* create first pcm device with one outputs and one input */ |
683 | err = snd_pcm_new(card: chip->card, id: "SGI O2 Audio" , device: 0, playback_count: 1, capture_count: 1, rpcm: &pcm); |
684 | if (err < 0) |
685 | return err; |
686 | |
687 | pcm->private_data = chip; |
688 | strcpy(p: pcm->name, q: "SGI O2 DAC1" ); |
689 | |
690 | /* set operators */ |
691 | snd_pcm_set_ops(pcm, direction: SNDRV_PCM_STREAM_PLAYBACK, |
692 | ops: &snd_sgio2audio_playback1_ops); |
693 | snd_pcm_set_ops(pcm, direction: SNDRV_PCM_STREAM_CAPTURE, |
694 | ops: &snd_sgio2audio_capture_ops); |
695 | snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_VMALLOC, NULL, size: 0, max: 0); |
696 | |
697 | /* create second pcm device with one outputs and no input */ |
698 | err = snd_pcm_new(card: chip->card, id: "SGI O2 Audio" , device: 1, playback_count: 1, capture_count: 0, rpcm: &pcm); |
699 | if (err < 0) |
700 | return err; |
701 | |
702 | pcm->private_data = chip; |
703 | strcpy(p: pcm->name, q: "SGI O2 DAC2" ); |
704 | |
705 | /* set operators */ |
706 | snd_pcm_set_ops(pcm, direction: SNDRV_PCM_STREAM_PLAYBACK, |
707 | ops: &snd_sgio2audio_playback2_ops); |
708 | snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_VMALLOC, NULL, size: 0, max: 0); |
709 | |
710 | return 0; |
711 | } |
712 | |
713 | static struct { |
714 | int idx; |
715 | int irq; |
716 | irqreturn_t (*isr)(int, void *); |
717 | const char *desc; |
718 | } snd_sgio2_isr_table[] = { |
719 | { |
720 | .idx = 0, |
721 | .irq = MACEISA_AUDIO1_DMAT_IRQ, |
722 | .isr = snd_sgio2audio_dma_in_isr, |
723 | .desc = "Capture DMA Channel 0" |
724 | }, { |
725 | .idx = 0, |
726 | .irq = MACEISA_AUDIO1_OF_IRQ, |
727 | .isr = snd_sgio2audio_error_isr, |
728 | .desc = "Capture Overflow" |
729 | }, { |
730 | .idx = 1, |
731 | .irq = MACEISA_AUDIO2_DMAT_IRQ, |
732 | .isr = snd_sgio2audio_dma_out_isr, |
733 | .desc = "Playback DMA Channel 1" |
734 | }, { |
735 | .idx = 1, |
736 | .irq = MACEISA_AUDIO2_MERR_IRQ, |
737 | .isr = snd_sgio2audio_error_isr, |
738 | .desc = "Memory Error Channel 1" |
739 | }, { |
740 | .idx = 2, |
741 | .irq = MACEISA_AUDIO3_DMAT_IRQ, |
742 | .isr = snd_sgio2audio_dma_out_isr, |
743 | .desc = "Playback DMA Channel 2" |
744 | }, { |
745 | .idx = 2, |
746 | .irq = MACEISA_AUDIO3_MERR_IRQ, |
747 | .isr = snd_sgio2audio_error_isr, |
748 | .desc = "Memory Error Channel 2" |
749 | } |
750 | }; |
751 | |
752 | /* ALSA driver */ |
753 | |
754 | static int snd_sgio2audio_free(struct snd_sgio2audio *chip) |
755 | { |
756 | int i; |
757 | |
758 | /* reset interface */ |
759 | writeq(AUDIO_CONTROL_RESET, addr: &mace->perif.audio.control); |
760 | udelay(1); |
761 | writeq(val: 0, addr: &mace->perif.audio.control); |
762 | |
763 | /* release IRQ's */ |
764 | for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++) |
765 | free_irq(snd_sgio2_isr_table[i].irq, |
766 | &chip->channel[snd_sgio2_isr_table[i].idx]); |
767 | |
768 | dma_free_coherent(dev: chip->card->dev, size: MACEISA_RINGBUFFERS_SIZE, |
769 | cpu_addr: chip->ring_base, dma_handle: chip->ring_base_dma); |
770 | |
771 | /* release card data */ |
772 | kfree(objp: chip); |
773 | return 0; |
774 | } |
775 | |
776 | static int snd_sgio2audio_dev_free(struct snd_device *device) |
777 | { |
778 | struct snd_sgio2audio *chip = device->device_data; |
779 | |
780 | return snd_sgio2audio_free(chip); |
781 | } |
782 | |
783 | static const struct snd_device_ops ops = { |
784 | .dev_free = snd_sgio2audio_dev_free, |
785 | }; |
786 | |
787 | static int snd_sgio2audio_create(struct snd_card *card, |
788 | struct snd_sgio2audio **rchip) |
789 | { |
790 | struct snd_sgio2audio *chip; |
791 | int i, err; |
792 | |
793 | *rchip = NULL; |
794 | |
795 | /* check if a codec is attached to the interface */ |
796 | /* (Audio or Audio/Video board present) */ |
797 | if (!(readq(&mace->perif.audio.control) & AUDIO_CONTROL_CODEC_PRESENT)) |
798 | return -ENOENT; |
799 | |
800 | chip = kzalloc(size: sizeof(*chip), GFP_KERNEL); |
801 | if (chip == NULL) |
802 | return -ENOMEM; |
803 | |
804 | chip->card = card; |
805 | |
806 | chip->ring_base = dma_alloc_coherent(dev: card->dev, |
807 | size: MACEISA_RINGBUFFERS_SIZE, |
808 | dma_handle: &chip->ring_base_dma, GFP_KERNEL); |
809 | if (chip->ring_base == NULL) { |
810 | printk(KERN_ERR |
811 | "sgio2audio: could not allocate ring buffers\n" ); |
812 | kfree(objp: chip); |
813 | return -ENOMEM; |
814 | } |
815 | |
816 | spin_lock_init(&chip->ad1843_lock); |
817 | |
818 | /* initialize channels */ |
819 | for (i = 0; i < 3; i++) { |
820 | spin_lock_init(&chip->channel[i].lock); |
821 | chip->channel[i].idx = i; |
822 | } |
823 | |
824 | /* allocate IRQs */ |
825 | for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++) { |
826 | if (request_irq(snd_sgio2_isr_table[i].irq, |
827 | snd_sgio2_isr_table[i].isr, |
828 | 0, |
829 | snd_sgio2_isr_table[i].desc, |
830 | &chip->channel[snd_sgio2_isr_table[i].idx])) { |
831 | snd_sgio2audio_free(chip); |
832 | printk(KERN_ERR "sgio2audio: cannot allocate irq %d\n" , |
833 | snd_sgio2_isr_table[i].irq); |
834 | return -EBUSY; |
835 | } |
836 | } |
837 | |
838 | /* reset the interface */ |
839 | writeq(AUDIO_CONTROL_RESET, addr: &mace->perif.audio.control); |
840 | udelay(1); |
841 | writeq(val: 0, addr: &mace->perif.audio.control); |
842 | msleep_interruptible(msecs: 1); /* give time to recover */ |
843 | |
844 | /* set ring base */ |
845 | writeq(val: chip->ring_base_dma, addr: &mace->perif.ctrl.ringbase); |
846 | |
847 | /* attach the AD1843 codec */ |
848 | chip->ad1843.read = read_ad1843_reg; |
849 | chip->ad1843.write = write_ad1843_reg; |
850 | chip->ad1843.chip = chip; |
851 | |
852 | /* initialize the AD1843 codec */ |
853 | err = ad1843_init(ad1843: &chip->ad1843); |
854 | if (err < 0) { |
855 | snd_sgio2audio_free(chip); |
856 | return err; |
857 | } |
858 | |
859 | err = snd_device_new(card, type: SNDRV_DEV_LOWLEVEL, device_data: chip, ops: &ops); |
860 | if (err < 0) { |
861 | snd_sgio2audio_free(chip); |
862 | return err; |
863 | } |
864 | *rchip = chip; |
865 | return 0; |
866 | } |
867 | |
868 | static int snd_sgio2audio_probe(struct platform_device *pdev) |
869 | { |
870 | struct snd_card *card; |
871 | struct snd_sgio2audio *chip; |
872 | int err; |
873 | |
874 | err = snd_card_new(parent: &pdev->dev, idx: index, xid: id, THIS_MODULE, extra_size: 0, card_ret: &card); |
875 | if (err < 0) |
876 | return err; |
877 | |
878 | err = snd_sgio2audio_create(card, rchip: &chip); |
879 | if (err < 0) { |
880 | snd_card_free(card); |
881 | return err; |
882 | } |
883 | |
884 | err = snd_sgio2audio_new_pcm(chip); |
885 | if (err < 0) { |
886 | snd_card_free(card); |
887 | return err; |
888 | } |
889 | err = snd_sgio2audio_new_mixer(chip); |
890 | if (err < 0) { |
891 | snd_card_free(card); |
892 | return err; |
893 | } |
894 | |
895 | strcpy(p: card->driver, q: "SGI O2 Audio" ); |
896 | strcpy(p: card->shortname, q: "SGI O2 Audio" ); |
897 | sprintf(buf: card->longname, fmt: "%s irq %i-%i" , |
898 | card->shortname, |
899 | MACEISA_AUDIO1_DMAT_IRQ, |
900 | MACEISA_AUDIO3_MERR_IRQ); |
901 | |
902 | err = snd_card_register(card); |
903 | if (err < 0) { |
904 | snd_card_free(card); |
905 | return err; |
906 | } |
907 | platform_set_drvdata(pdev, data: card); |
908 | return 0; |
909 | } |
910 | |
911 | static void snd_sgio2audio_remove(struct platform_device *pdev) |
912 | { |
913 | struct snd_card *card = platform_get_drvdata(pdev); |
914 | |
915 | snd_card_free(card); |
916 | } |
917 | |
918 | static struct platform_driver sgio2audio_driver = { |
919 | .probe = snd_sgio2audio_probe, |
920 | .remove_new = snd_sgio2audio_remove, |
921 | .driver = { |
922 | .name = "sgio2audio" , |
923 | } |
924 | }; |
925 | |
926 | module_platform_driver(sgio2audio_driver); |
927 | |