?  search:
summary | log | graphiclog1 | graphiclog2 | commit | commitdiff | tree | refs | edit | fork
blame | history | raw | HEAD
Linux 2.6.17-rc6
[linux-2.6/next.git] / sound / arm / sa11xx-uda1341.c
blob13057d92f08afb0a8888f5fece7f5d9952790ad4
1 /*
2 * Driver for Philips UDA1341TS on Compaq iPAQ H3600 soundcard
3 * Copyright (C) 2002 Tomas Kasparek <tomas.kasparek@seznam.cz>
4 *
5 * This program is free software; you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License.
7 *
8 * History:
9 *
10 * 2002-03-13 Tomas Kasparek initial release - based on h3600-uda1341.c from OSS
11 * 2002-03-20 Tomas Kasparek playback over ALSA is working
12 * 2002-03-28 Tomas Kasparek playback over OSS emulation is working
13 * 2002-03-29 Tomas Kasparek basic capture is working (native ALSA)
14 * 2002-03-29 Tomas Kasparek capture is working (OSS emulation)
15 * 2002-04-04 Tomas Kasparek better rates handling (allow non-standard rates)
16 * 2003-02-14 Brian Avery fixed full duplex mode, other updates
17 * 2003-02-20 Tomas Kasparek merged updates by Brian (except HAL)
18 * 2003-04-19 Jaroslav Kysela recoded DMA stuff to follow 2.4.18rmk3-hh24 kernel
19 * working suspend and resume
20 * 2003-04-28 Tomas Kasparek updated work by Jaroslav to compile it under 2.5.x again
21 * merged HAL layer (patches from Brian)
22 */
23
24 /* $Id: sa11xx-uda1341.c,v 1.27 2005/12/07 09:13:42 cladisch Exp $ */
25
26 /***************************************************************************************************
27 *
28 * To understand what Alsa Drivers should be doing look at "Writing an Alsa Driver" by Takashi Iwai
29 * available in the Alsa doc section on the website
30 *
31 * A few notes to make things clearer. The UDA1341 is hooked up to Serial port 4 on the SA1100.
32 * We are using SSP mode to talk to the UDA1341. The UDA1341 bit & wordselect clocks are generated
33 * by this UART. Unfortunately, the clock only runs if the transmit buffer has something in it.
34 * So, if we are just recording, we feed the transmit DMA stream a bunch of 0x0000 so that the
35 * transmit buffer is full and the clock keeps going. The zeroes come from FLUSH_BASE_PHYS which
36 * is a mem loc that always decodes to 0's w/ no off chip access.
37 *
38 * Some alsa terminology:
39 * frame => num_channels * sample_size e.g stereo 16 bit is 2 * 16 = 32 bytes
40 * period => the least number of bytes that will generate an interrupt e.g. we have a 1024 byte
41 * buffer and 4 periods in the runtime structure this means we'll get an int every 256
42 * bytes or 4 times per buffer.
43 * A number of the sizes are in frames rather than bytes, use frames_to_bytes and
44 * bytes_to_frames to convert. The easiest way to tell the units is to look at the
45 * type i.e. runtime-> buffer_size is in frames and its type is snd_pcm_uframes_t
46 *
47 * Notes about the pointer fxn:
48 * The pointer fxn needs to return the offset into the dma buffer in frames.
49 * Interrupts must be blocked before calling the dma_get_pos fxn to avoid race with interrupts.
50 *
51 * Notes about pause/resume
52 * Implementing this would be complicated so it's skipped. The problem case is:
53 * A full duplex connection is going, then play is paused. At this point you need to start xmitting
54 * 0's to keep the record active which means you cant just freeze the dma and resume it later you'd
55 * need to save off the dma info, and restore it properly on a resume. Yeach!
56 *
57 * Notes about transfer methods:
58 * The async write calls fail. I probably need to implement something else to support them?
59 *
60 ***************************************************************************************************/
61
62 #include <linux/config.h>
63 #include <sound/driver.h>
64 #include <linux/module.h>
65 #include <linux/moduleparam.h>
66 #include <linux/init.h>
67 #include <linux/err.h>
68 #include <linux/platform_device.h>
69 #include <linux/errno.h>
70 #include <linux/ioctl.h>
71 #include <linux/delay.h>
72 #include <linux/slab.h>
73
74 #ifdef CONFIG_PM
75 #include <linux/pm.h>
76 #endif
77
78 #include <asm/hardware.h>
79 #include <asm/arch/h3600.h>
80 #include <asm/mach-types.h>
81 #include <asm/dma.h>
82
83 #ifdef CONFIG_H3600_HAL
84 #include <asm/semaphore.h>
85 #include <asm/uaccess.h>
86 #include <asm/arch/h3600_hal.h>
87 #endif
88
89 #include <sound/core.h>
90 #include <sound/pcm.h>
91 #include <sound/initval.h>
92
93 #include <linux/l3/l3.h>
94
95 #undef DEBUG_MODE
96 #undef DEBUG_FUNCTION_NAMES
97 #include <sound/uda1341.h>
98
99 /*
100 * FIXME: Is this enough as autodetection of 2.4.X-rmkY-hhZ kernels?
101 * We use DMA stuff from 2.4.18-rmk3-hh24 here to be able to compile this
102 * module for Familiar 0.6.1
103 */
104 #ifdef CONFIG_H3600_HAL
105 #define HH_VERSION 1
106 #endif
107
108 /* {{{ Type definitions */
109
110 MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek@seznam.cz>");
111 MODULE_LICENSE("GPL");
112 MODULE_DESCRIPTION("SA1100/SA1111 + UDA1341TS driver for ALSA");
113 MODULE_SUPPORTED_DEVICE("{{UDA1341,iPAQ H3600 UDA1341TS}}");
114
115 static char *id = NULL; /* ID for this card */
116
117 module_param(id, charp, 0444);
118 MODULE_PARM_DESC(id, "ID string for SA1100/SA1111 + UDA1341TS soundcard.");
119
120 struct audio_stream {
121 char *id; /* identification string */
122 int stream_id; /* numeric identification */
123 dma_device_t dma_dev; /* device identifier for DMA */
124 #ifdef HH_VERSION
125 dmach_t dmach; /* dma channel identification */
126 #else
127 dma_regs_t *dma_regs; /* points to our DMA registers */
128 #endif
129 int active:1; /* we are using this stream for transfer now */
130 int period; /* current transfer period */
131 int periods; /* current count of periods registerd in the DMA engine */
132 int tx_spin; /* are we recoding - flag used to do DMA trans. for sync */
133 unsigned int old_offset;
134 spinlock_t dma_lock; /* for locking in DMA operations (see dma-sa1100.c in the kernel) */
135 struct snd_pcm_substream *stream;
136 };
137
138 struct sa11xx_uda1341 {
139 struct snd_card *card;
140 struct l3_client *uda1341;
141 struct snd_pcm *pcm;
142 long samplerate;
143 struct audio_stream s[2]; /* playback & capture */
144 };
145
146 static unsigned int rates[] = {
147 8000, 10666, 10985, 14647,
148 16000, 21970, 22050, 24000,
149 29400, 32000, 44100, 48000,
150 };
151
152 static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
153 .count = ARRAY_SIZE(rates),
154 .list = rates,
155 .mask = 0,
156 };
157
158 static struct platform_device *device;
159
160 /* }}} */
161
162 /* {{{ Clock and sample rate stuff */
163
164 /*
165 * Stop-gap solution until rest of hh.org HAL stuff is merged.
166 */
167 #define GPIO_H3600_CLK_SET0 GPIO_GPIO (12)
168 #define GPIO_H3600_CLK_SET1 GPIO_GPIO (13)
169
170 #ifdef CONFIG_SA1100_H3XXX
171 #define clr_sa11xx_uda1341_egpio(x) clr_h3600_egpio(x)
172 #define set_sa11xx_uda1341_egpio(x) set_h3600_egpio(x)
173 #else
174 #error This driver could serve H3x00 handhelds only!
175 #endif
176
177 static void sa11xx_uda1341_set_audio_clock(long val)
178 {
179 switch (val) {
180 case 24000: case 32000: case 48000: /* 00: 12.288 MHz */
181 GPCR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1;
182 break;
183
184 case 22050: case 29400: case 44100: /* 01: 11.2896 MHz */
185 GPSR = GPIO_H3600_CLK_SET0;
186 GPCR = GPIO_H3600_CLK_SET1;
187 break;
188
189 case 8000: case 10666: case 16000: /* 10: 4.096 MHz */
190 GPCR = GPIO_H3600_CLK_SET0;
191 GPSR = GPIO_H3600_CLK_SET1;
192 break;
193
194 case 10985: case 14647: case 21970: /* 11: 5.6245 MHz */
195 GPSR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1;
196 break;
197 }
198 }
199
200 static void sa11xx_uda1341_set_samplerate(struct sa11xx_uda1341 *sa11xx_uda1341, long rate)
201 {
202 int clk_div = 0;
203 int clk=0;
204
205 /* We don't want to mess with clocks when frames are in flight */
206 Ser4SSCR0 &= ~SSCR0_SSE;
207 /* wait for any frame to complete */
208 udelay(125);
209
210 /*
211 * We have the following clock sources:
212 * 4.096 MHz, 5.6245 MHz, 11.2896 MHz, 12.288 MHz
213 * Those can be divided either by 256, 384 or 512.
214 * This makes up 12 combinations for the following samplerates...
215 */
216 if (rate >= 48000)
217 rate = 48000;
218 else if (rate >= 44100)
219 rate = 44100;
220 else if (rate >= 32000)
221 rate = 32000;
222 else if (rate >= 29400)
223 rate = 29400;
224 else if (rate >= 24000)
225 rate = 24000;
226 else if (rate >= 22050)
227 rate = 22050;
228 else if (rate >= 21970)
229 rate = 21970;
230 else if (rate >= 16000)
231 rate = 16000;
232 else if (rate >= 14647)
233 rate = 14647;
234 else if (rate >= 10985)
235 rate = 10985;
236 else if (rate >= 10666)
237 rate = 10666;
238 else
239 rate = 8000;
240
241 /* Set the external clock generator */
242 #ifdef CONFIG_H3600_HAL
243 h3600_audio_clock(rate);
244 #else
245 sa11xx_uda1341_set_audio_clock(rate);
246 #endif
247
248 /* Select the clock divisor */
249 switch (rate) {
250 case 8000:
251 case 10985:
252 case 22050:
253 case 24000:
254 clk = F512;
255 clk_div = SSCR0_SerClkDiv(16);
256 break;
257 case 16000:
258 case 21970:
259 case 44100:
260 case 48000:
261 clk = F256;
262 clk_div = SSCR0_SerClkDiv(8);
263 break;
264 case 10666:
265 case 14647:
266 case 29400:
267 case 32000:
268 clk = F384;
269 clk_div = SSCR0_SerClkDiv(12);
270 break;
271 }
272
273 /* FMT setting should be moved away when other FMTs are added (FIXME) */
274 l3_command(sa11xx_uda1341->uda1341, CMD_FORMAT, (void *)LSB16);
275
276 l3_command(sa11xx_uda1341->uda1341, CMD_FS, (void *)clk);
277 Ser4SSCR0 = (Ser4SSCR0 & ~0xff00) + clk_div + SSCR0_SSE;
278 sa11xx_uda1341->samplerate = rate;
279 }
280
281 /* }}} */
282
283 /* {{{ HW init and shutdown */
284
285 static void sa11xx_uda1341_audio_init(struct sa11xx_uda1341 *sa11xx_uda1341)
286 {
287 unsigned long flags;
288
289 /* Setup DMA stuff */
290 sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].id = "UDA1341 out";
291 sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].stream_id = SNDRV_PCM_STREAM_PLAYBACK;
292 sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].dma_dev = DMA_Ser4SSPWr;
293
294 sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].id = "UDA1341 in";
295 sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].stream_id = SNDRV_PCM_STREAM_CAPTURE;
296 sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].dma_dev = DMA_Ser4SSPRd;
297
298 /* Initialize the UDA1341 internal state */
299
300 /* Setup the uarts */
301 local_irq_save(flags);
302 GAFR |= (GPIO_SSP_CLK);
303 GPDR &= ~(GPIO_SSP_CLK);
304 Ser4SSCR0 = 0;
305 Ser4SSCR0 = SSCR0_DataSize(16) + SSCR0_TI + SSCR0_SerClkDiv(8);
306 Ser4SSCR1 = SSCR1_SClkIactL + SSCR1_SClk1P + SSCR1_ExtClk;
307 Ser4SSCR0 |= SSCR0_SSE;
308 local_irq_restore(flags);
309
310 /* Enable the audio power */
311 #ifdef CONFIG_H3600_HAL
312 h3600_audio_power(AUDIO_RATE_DEFAULT);
313 #else
314 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
315 set_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
316 set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
317 #endif
318
319 /* Wait for the UDA1341 to wake up */
320 mdelay(1); //FIXME - was removed by Perex - Why?
321
322 /* Initialize the UDA1341 internal state */
323 l3_open(sa11xx_uda1341->uda1341);
324
325 /* external clock configuration (after l3_open - regs must be initialized */
326 sa11xx_uda1341_set_samplerate(sa11xx_uda1341, sa11xx_uda1341->samplerate);
327
328 /* Wait for the UDA1341 to wake up */
329 set_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
330 mdelay(1);
331
332 /* make the left and right channels unswapped (flip the WS latch) */
333 Ser4SSDR = 0;
334
335 #ifdef CONFIG_H3600_HAL
336 h3600_audio_mute(0);
337 #else
338 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
339 #endif
340 }
341
342 static void sa11xx_uda1341_audio_shutdown(struct sa11xx_uda1341 *sa11xx_uda1341)
343 {
344 /* mute on */
345 #ifdef CONFIG_H3600_HAL
346 h3600_audio_mute(1);
347 #else
348 set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
349 #endif
350
351 /* disable the audio power and all signals leading to the audio chip */
352 l3_close(sa11xx_uda1341->uda1341);
353 Ser4SSCR0 = 0;
354 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
355
356 /* power off and mute off */
357 /* FIXME - is muting off necesary??? */
358 #ifdef CONFIG_H3600_HAL
359 h3600_audio_power(0);
360 h3600_audio_mute(0);
361 #else
362 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
363 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
364 #endif
365 }
366
367 /* }}} */
368
369 /* {{{ DMA staff */
370
371 /*
372 * these are the address and sizes used to fill the xmit buffer
373 * so we can get a clock in record only mode
374 */
375 #define FORCE_CLOCK_ADDR (dma_addr_t)FLUSH_BASE_PHYS
376 #define FORCE_CLOCK_SIZE 4096 // was 2048
377
378 // FIXME Why this value exactly - wrote comment
379 #define DMA_BUF_SIZE 8176 /* <= MAX_DMA_SIZE from asm/arch-sa1100/dma.h */
380
381 #ifdef HH_VERSION
382
383 static int audio_dma_request(struct audio_stream *s, void (*callback)(void *, int))
384 {
385 int ret;
386
387 ret = sa1100_request_dma(&s->dmach, s->id, s->dma_dev);
388 if (ret < 0) {
389 printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev);
390 return ret;
391 }
392 sa1100_dma_set_callback(s->dmach, callback);
393 return 0;
394 }
395
396 static inline void audio_dma_free(struct audio_stream *s)
397 {
398 sa1100_free_dma(s->dmach);
399 s->dmach = -1;
400 }
401
402 #else
403
404 static int audio_dma_request(struct audio_stream *s, void (*callback)(void *))
405 {
406 int ret;
407
408 ret = sa1100_request_dma(s->dma_dev, s->id, callback, s, &s->dma_regs);
409 if (ret < 0)
410 printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev);
411 return ret;
412 }
413
414 static void audio_dma_free(struct audio_stream *s)
415 {
416 sa1100_free_dma(s->dma_regs);
417 s->dma_regs = 0;
418 }
419
420 #endif
421
422 static u_int audio_get_dma_pos(struct audio_stream *s)
423 {
424 struct snd_pcm_substream *substream = s->stream;
425 struct snd_pcm_runtime *runtime = substream->runtime;
426 unsigned int offset;
427 unsigned long flags;
428 dma_addr_t addr;
429
430 // this must be called w/ interrupts locked out see dma-sa1100.c in the kernel
431 spin_lock_irqsave(&s->dma_lock, flags);
432 #ifdef HH_VERSION
433 sa1100_dma_get_current(s->dmach, NULL, &addr);
434 #else
435 addr = sa1100_get_dma_pos((s)->dma_regs);
436 #endif
437 offset = addr - runtime->dma_addr;
438 spin_unlock_irqrestore(&s->dma_lock, flags);
439
440 offset = bytes_to_frames(runtime,offset);
441 if (offset >= runtime->buffer_size)
442 offset = 0;
443
444 return offset;
445 }
446
447 /*
448 * this stops the dma and clears the dma ptrs
449 */
450 static void audio_stop_dma(struct audio_stream *s)
451 {
452 unsigned long flags;
453
454 spin_lock_irqsave(&s->dma_lock, flags);
455 s->active = 0;
456 s->period = 0;
457 /* this stops the dma channel and clears the buffer ptrs */
458 #ifdef HH_VERSION
459 sa1100_dma_flush_all(s->dmach);
460 #else
461 sa1100_clear_dma(s->dma_regs);
462 #endif
463 spin_unlock_irqrestore(&s->dma_lock, flags);
464 }
465
466 static void audio_process_dma(struct audio_stream *s)
467 {
468 struct snd_pcm_substream *substream = s->stream;
469 struct snd_pcm_runtime *runtime;
470 unsigned int dma_size;
471 unsigned int offset;
472 int ret;
473
474 /* we are requested to process synchronization DMA transfer */
475 if (s->tx_spin) {
476 snd_assert(s->stream_id == SNDRV_PCM_STREAM_PLAYBACK, return);
477 /* fill the xmit dma buffers and return */
478 #ifdef HH_VERSION
479 sa1100_dma_set_spin(s->dmach, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE);
480 #else
481 while (1) {
482 ret = sa1100_start_dma(s->dma_regs, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE);
483 if (ret)
484 return;
485 }
486 #endif
487 return;
488 }
489
490 /* must be set here - only valid for running streams, not for forced_clock dma fills */
491 runtime = substream->runtime;
492 while (s->active && s->periods < runtime->periods) {
493 dma_size = frames_to_bytes(runtime, runtime->period_size);
494 if (s->old_offset) {
495 /* a little trick, we need resume from old position */
496 offset = frames_to_bytes(runtime, s->old_offset - 1);
497 s->old_offset = 0;
498 s->periods = 0;
499 s->period = offset / dma_size;
500 offset %= dma_size;
501 dma_size = dma_size - offset;
502 if (!dma_size)
503 continue; /* special case */
504 } else {
505 offset = dma_size * s->period;
506 snd_assert(dma_size <= DMA_BUF_SIZE, );
507 }
508 #ifdef HH_VERSION
509 ret = sa1100_dma_queue_buffer(s->dmach, s, runtime->dma_addr + offset, dma_size);
510 if (ret)
511 return; //FIXME
512 #else
513 ret = sa1100_start_dma((s)->dma_regs, runtime->dma_addr + offset, dma_size);
514 if (ret) {
515 printk(KERN_ERR "audio_process_dma: cannot queue DMA buffer (%i)\n", ret);
516 return;
517 }
518 #endif
519
520 s->period++;
521 s->period %= runtime->periods;
522 s->periods++;
523 }
524 }
525
526 #ifdef HH_VERSION
527 static void audio_dma_callback(void *data, int size)
528 #else
529 static void audio_dma_callback(void *data)
530 #endif
531 {
532 struct audio_stream *s = data;
533
534 /*
535 * If we are getting a callback for an active stream then we inform
536 * the PCM middle layer we've finished a period
537 */
538 if (s->active)
539 snd_pcm_period_elapsed(s->stream);
540
541 spin_lock(&s->dma_lock);
542 if (!s->tx_spin && s->periods > 0)
543 s->periods--;
544 audio_process_dma(s);
545 spin_unlock(&s->dma_lock);
546 }
547
548 /* }}} */
549
550 /* {{{ PCM setting */
551
552 /* {{{ trigger & timer */
553
554 static int snd_sa11xx_uda1341_trigger(struct snd_pcm_substream *substream, int cmd)
555 {
556 struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
557 int stream_id = substream->pstr->stream;
558 struct audio_stream *s = &chip->s[stream_id];
559 struct audio_stream *s1 = &chip->s[stream_id ^ 1];
560 int err = 0;
561
562 /* note local interrupts are already disabled in the midlevel code */
563 spin_lock(&s->dma_lock);
564 switch (cmd) {
565 case SNDRV_PCM_TRIGGER_START:
566 /* now we need to make sure a record only stream has a clock */
567 if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
568 /* we need to force fill the xmit DMA with zeros */
569 s1->tx_spin = 1;
570 audio_process_dma(s1);
571 }
572 /* this case is when you were recording then you turn on a
573 * playback stream so we stop (also clears it) the dma first,
574 * clear the sync flag and then we let it turned on
575 */
576 else {
577 s->tx_spin = 0;
578 }
579
580 /* requested stream startup */
581 s->active = 1;
582 audio_process_dma(s);
583 break;
584 case SNDRV_PCM_TRIGGER_STOP:
585 /* requested stream shutdown */
586 audio_stop_dma(s);
587
588 /*
589 * now we need to make sure a record only stream has a clock
590 * so if we're stopping a playback with an active capture
591 * we need to turn the 0 fill dma on for the xmit side
592 */
593 if (stream_id == SNDRV_PCM_STREAM_PLAYBACK && s1->active) {
594 /* we need to force fill the xmit DMA with zeros */
595 s->tx_spin = 1;
596 audio_process_dma(s);
597 }
598 /*
599 * we killed a capture only stream, so we should also kill
600 * the zero fill transmit
601 */
602 else {
603 if (s1->tx_spin) {
604 s1->tx_spin = 0;
605 audio_stop_dma(s1);
606 }
607 }
608
609 break;
610 case SNDRV_PCM_TRIGGER_SUSPEND:
611 s->active = 0;
612 #ifdef HH_VERSION
613 sa1100_dma_stop(s->dmach);
614 #else
615 //FIXME - DMA API
616 #endif
617 s->old_offset = audio_get_dma_pos(s) + 1;
618 #ifdef HH_VERSION
619 sa1100_dma_flush_all(s->dmach);
620 #else
621 //FIXME - DMA API
622 #endif
623 s->periods = 0;
624 break;
625 case SNDRV_PCM_TRIGGER_RESUME:
626 s->active = 1;
627 s->tx_spin = 0;
628 audio_process_dma(s);
629 if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
630 s1->tx_spin = 1;
631 audio_process_dma(s1);
632 }
633 break;
634 case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
635 #ifdef HH_VERSION
636 sa1100_dma_stop(s->dmach);
637 #else
638 //FIXME - DMA API
639 #endif
640 s->active = 0;
641 if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) {
642 if (s1->active) {
643 s->tx_spin = 1;
644 s->old_offset = audio_get_dma_pos(s) + 1;
645 #ifdef HH_VERSION
646 sa1100_dma_flush_all(s->dmach);
647 #else
648 //FIXME - DMA API
649 #endif
650 audio_process_dma(s);
651 }
652 } else {
653 if (s1->tx_spin) {
654 s1->tx_spin = 0;
655 #ifdef HH_VERSION
656 sa1100_dma_flush_all(s1->dmach);
657 #else
658 //FIXME - DMA API
659 #endif
660 }
661 }
662 break;
663 case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
664 s->active = 1;
665 if (s->old_offset) {
666 s->tx_spin = 0;
667 audio_process_dma(s);
668 break;
669 }
670 if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
671 s1->tx_spin = 1;
672 audio_process_dma(s1);
673 }
674 #ifdef HH_VERSION
675 sa1100_dma_resume(s->dmach);
676 #else
677 //FIXME - DMA API
678 #endif
679 break;
680 default:
681 err = -EINVAL;
682 break;
683 }
684 spin_unlock(&s->dma_lock);
685 return err;
686 }
687
688 static int snd_sa11xx_uda1341_prepare(struct snd_pcm_substream *substream)
689 {
690 struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
691 struct snd_pcm_runtime *runtime = substream->runtime;
692 struct audio_stream *s = &chip->s[substream->pstr->stream];
693
694 /* set requested samplerate */
695 sa11xx_uda1341_set_samplerate(chip, runtime->rate);
696
697 /* set requestd format when available */
698 /* set FMT here !!! FIXME */
699
700 s->period = 0;
701 s->periods = 0;
702
703 return 0;
704 }
705
706 static snd_pcm_uframes_t snd_sa11xx_uda1341_pointer(struct snd_pcm_substream *substream)
707 {
708 struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
709 return audio_get_dma_pos(&chip->s[substream->pstr->stream]);
710 }
711
712 /* }}} */
713
714 static struct snd_pcm_hardware snd_sa11xx_uda1341_capture =
715 {
716 .info = (SNDRV_PCM_INFO_INTERLEAVED |
717 SNDRV_PCM_INFO_BLOCK_TRANSFER |
718 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
719 SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
720 .formats = SNDRV_PCM_FMTBIT_S16_LE,
721 .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\
722 SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\
723 SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
724 SNDRV_PCM_RATE_KNOT),
725 .rate_min = 8000,
726 .rate_max = 48000,
727 .channels_min = 2,
728 .channels_max = 2,
729 .buffer_bytes_max = 64*1024,
730 .period_bytes_min = 64,
731 .period_bytes_max = DMA_BUF_SIZE,
732 .periods_min = 2,
733 .periods_max = 255,
734 .fifo_size = 0,
735 };
736
737 static struct snd_pcm_hardware snd_sa11xx_uda1341_playback =
738 {
739 .info = (SNDRV_PCM_INFO_INTERLEAVED |
740 SNDRV_PCM_INFO_BLOCK_TRANSFER |
741 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
742 SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
743 .formats = SNDRV_PCM_FMTBIT_S16_LE,
744 .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\
745 SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\
746 SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
747 SNDRV_PCM_RATE_KNOT),
748 .rate_min = 8000,
749 .rate_max = 48000,
750 .channels_min = 2,
751 .channels_max = 2,
752 .buffer_bytes_max = 64*1024,
753 .period_bytes_min = 64,
754 .period_bytes_max = DMA_BUF_SIZE,
755 .periods_min = 2,
756 .periods_max = 255,
757 .fifo_size = 0,
758 };
759
760 static int snd_card_sa11xx_uda1341_open(struct snd_pcm_substream *substream)
761 {
762 struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
763 struct snd_pcm_runtime *runtime = substream->runtime;
764 int stream_id = substream->pstr->stream;
765 int err;
766
767 chip->s[stream_id].stream = substream;
768
769 if (stream_id == SNDRV_PCM_STREAM_PLAYBACK)
770 runtime->hw = snd_sa11xx_uda1341_playback;
771 else
772 runtime->hw = snd_sa11xx_uda1341_capture;
773 if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
774 return err;
775 if ((err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates)) < 0)
776 return err;
777
778 return 0;
779 }
780
781 static int snd_card_sa11xx_uda1341_close(struct snd_pcm_substream *substream)
782 {
783 struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
784
785 chip->s[substream->pstr->stream].stream = NULL;
786 return 0;
787 }
788
789 /* {{{ HW params & free */
790
791 static int snd_sa11xx_uda1341_hw_params(struct snd_pcm_substream *substream,
792 struct snd_pcm_hw_params *hw_params)
793 {
794
795 return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
796 }
797
798 static int snd_sa11xx_uda1341_hw_free(struct snd_pcm_substream *substream)
799 {
800 return snd_pcm_lib_free_pages(substream);
801 }
802
803 /* }}} */
804
805 static struct snd_pcm_ops snd_card_sa11xx_uda1341_playback_ops = {
806 .open = snd_card_sa11xx_uda1341_open,
807 .close = snd_card_sa11xx_uda1341_close,
808 .ioctl = snd_pcm_lib_ioctl,
809 .hw_params = snd_sa11xx_uda1341_hw_params,
810 .hw_free = snd_sa11xx_uda1341_hw_free,
811 .prepare = snd_sa11xx_uda1341_prepare,
812 .trigger = snd_sa11xx_uda1341_trigger,
813 .pointer = snd_sa11xx_uda1341_pointer,
814 };
815
816 static struct snd_pcm_ops snd_card_sa11xx_uda1341_capture_ops = {
817 .open = snd_card_sa11xx_uda1341_open,
818 .close = snd_card_sa11xx_uda1341_close,
819 .ioctl = snd_pcm_lib_ioctl,
820 .hw_params = snd_sa11xx_uda1341_hw_params,
821 .hw_free = snd_sa11xx_uda1341_hw_free,
822 .prepare = snd_sa11xx_uda1341_prepare,
823 .trigger = snd_sa11xx_uda1341_trigger,
824 .pointer = snd_sa11xx_uda1341_pointer,
825 };
826
827 static int __init snd_card_sa11xx_uda1341_pcm(struct sa11xx_uda1341 *sa11xx_uda1341, int device)
828 {
829 struct snd_pcm *pcm;
830 int err;
831
832 if ((err = snd_pcm_new(sa11xx_uda1341->card, "UDA1341 PCM", device, 1, 1, &pcm)) < 0)
833 return err;
834
835 /*
836 * this sets up our initial buffers and sets the dma_type to isa.
837 * isa works but I'm not sure why (or if) it's the right choice
838 * this may be too large, trying it for now
839 */
840 snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
841 snd_dma_isa_data(),
842 64*1024, 64*1024);
843
844 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_card_sa11xx_uda1341_playback_ops);
845 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_card_sa11xx_uda1341_capture_ops);
846 pcm->private_data = sa11xx_uda1341;
847 pcm->info_flags = 0;
848 strcpy(pcm->name, "UDA1341 PCM");
849
850 sa11xx_uda1341_audio_init(sa11xx_uda1341);
851
852 /* setup DMA controller */
853 audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK], audio_dma_callback);
854 audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE], audio_dma_callback);
855
856 sa11xx_uda1341->pcm = pcm;
857
858 return 0;
859 }
860
861 /* }}} */
862
863 /* {{{ module init & exit */
864
865 #ifdef CONFIG_PM
866
867 static int snd_sa11xx_uda1341_suspend(struct platform_device *devptr,
868 pm_message_t state)
869 {
870 struct snd_card *card = platform_get_drvdata(devptr);
871 struct sa11xx_uda1341 *chip = card->private_data;
872
873 snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
874 snd_pcm_suspend_all(chip->pcm);
875 #ifdef HH_VERSION
876 sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach);
877 sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach);
878 #else
879 //FIXME
880 #endif
881 l3_command(chip->uda1341, CMD_SUSPEND, NULL);
882 sa11xx_uda1341_audio_shutdown(chip);
883
884 return 0;
885 }
886
887 static int snd_sa11xx_uda1341_resume(struct platform_device *devptr)
888 {
889 struct snd_card *card = platform_get_drvdata(devptr);
890 struct sa11xx_uda1341 *chip = card->private_data;
891
892 sa11xx_uda1341_audio_init(chip);
893 l3_command(chip->uda1341, CMD_RESUME, NULL);
894 #ifdef HH_VERSION
895 sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach);
896 sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach);
897 #else
898 //FIXME
899 #endif
900 snd_power_change_state(card, SNDRV_CTL_POWER_D0);
901 return 0;
902 }
903 #endif /* COMFIG_PM */
904
905 void snd_sa11xx_uda1341_free(struct snd_card *card)
906 {
907 struct sa11xx_uda1341 *chip = card->private_data;
908
909 audio_dma_free(&chip->s[SNDRV_PCM_STREAM_PLAYBACK]);
910 audio_dma_free(&chip->s[SNDRV_PCM_STREAM_CAPTURE]);
911 }
912
913 static int __init sa11xx_uda1341_probe(struct platform_device *devptr)
914 {
915 int err;
916 struct snd_card *card;
917 struct sa11xx_uda1341 *chip;
918
919 /* register the soundcard */
920 card = snd_card_new(-1, id, THIS_MODULE, sizeof(struct sa11xx_uda1341));
921 if (card == NULL)
922 return -ENOMEM;
923
924 chip = card->private_data;
925 spin_lock_init(&chip->s[0].dma_lock);
926 spin_lock_init(&chip->s[1].dma_lock);
927
928 card->private_free = snd_sa11xx_uda1341_free;
929 chip->card = card;
930 chip->samplerate = AUDIO_RATE_DEFAULT;
931
932 // mixer
933 if ((err = snd_chip_uda1341_mixer_new(card, &chip->uda1341)))
934 goto nodev;
935
936 // PCM
937 if ((err = snd_card_sa11xx_uda1341_pcm(chip, 0)) < 0)
938 goto nodev;
939
940 strcpy(card->driver, "UDA1341");
941 strcpy(card->shortname, "H3600 UDA1341TS");
942 sprintf(card->longname, "Compaq iPAQ H3600 with Philips UDA1341TS");
943
944 snd_card_set_dev(card, &devptr->dev);
945
946 if ((err = snd_card_register(card)) == 0) {
947 printk( KERN_INFO "iPAQ audio support initialized\n" );
948 platform_set_drvdata(devptr, card);
949 return 0;
950 }
951
952 nodev:
953 snd_card_free(card);
954 return err;
955 }
956
957 static int __devexit sa11xx_uda1341_remove(struct platform_device *devptr)
958 {
959 snd_card_free(platform_get_drvdata(devptr));
960 platform_set_drvdata(devptr, NULL);
961 return 0;
962 }
963
964 #define SA11XX_UDA1341_DRIVER "sa11xx_uda1341"
965
966 static struct platform_driver sa11xx_uda1341_driver = {
967 .probe = sa11xx_uda1341_probe,
968 .remove = __devexit_p(sa11xx_uda1341_remove),
969 #ifdef CONFIG_PM
970 .suspend = snd_sa11xx_uda1341_suspend,
971 .resume = snd_sa11xx_uda1341_resume,
972 #endif
973 .driver = {
974 .name = SA11XX_UDA1341_DRIVER,
975 },
976 };
977
978 static int __init sa11xx_uda1341_init(void)
979 {
980 int err;
981
982 if (!machine_is_h3xxx())
983 return -ENODEV;
984 if ((err = platform_driver_register(&sa11xx_uda1341_driver)) < 0)
985 return err;
986 device = platform_device_register_simple(SA11XX_UDA1341_DRIVER, -1, NULL, 0);
987 if (IS_ERR(device)) {
988 platform_driver_unregister(&sa11xx_uda1341_driver);
989 return PTR_ERR(device);
990 }
991 return 0;
992 }
993
994 static void __exit sa11xx_uda1341_exit(void)
995 {
996 platform_device_unregister(device);
997 platform_driver_unregister(&sa11xx_uda1341_driver);
998 }
999
1000 module_init(sa11xx_uda1341_init);
1001 module_exit(sa11xx_uda1341_exit);
1002
1003 /* }}} */
1004
1005 /*
1006 * Local variables:
1007 * indent-tabs-mode: t
1008 * End:
1009 */
linux-next