2 * Driver for Philips UDA1341TS on Compaq iPAQ H3600 soundcard
3 * Copyright (C) 2002 Tomas Kasparek <tomas.kasparek@seznam.cz>
5 * This program is free software; you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License.
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)
24 /***************************************************************************************************
26 * To understand what Alsa Drivers should be doing look at "Writing an Alsa Driver" by Takashi Iwai
27 * available in the Alsa doc section on the website
29 * A few notes to make things clearer. The UDA1341 is hooked up to Serial port 4 on the SA1100.
30 * We are using SSP mode to talk to the UDA1341. The UDA1341 bit & wordselect clocks are generated
31 * by this UART. Unfortunately, the clock only runs if the transmit buffer has something in it.
32 * So, if we are just recording, we feed the transmit DMA stream a bunch of 0x0000 so that the
33 * transmit buffer is full and the clock keeps going. The zeroes come from FLUSH_BASE_PHYS which
34 * is a mem loc that always decodes to 0's w/ no off chip access.
36 * Some alsa terminology:
37 * frame => num_channels * sample_size e.g stereo 16 bit is 2 * 16 = 32 bytes
38 * period => the least number of bytes that will generate an interrupt e.g. we have a 1024 byte
39 * buffer and 4 periods in the runtime structure this means we'll get an int every 256
40 * bytes or 4 times per buffer.
41 * A number of the sizes are in frames rather than bytes, use frames_to_bytes and
42 * bytes_to_frames to convert. The easiest way to tell the units is to look at the
43 * type i.e. runtime-> buffer_size is in frames and its type is snd_pcm_uframes_t
45 * Notes about the pointer fxn:
46 * The pointer fxn needs to return the offset into the dma buffer in frames.
47 * Interrupts must be blocked before calling the dma_get_pos fxn to avoid race with interrupts.
49 * Notes about pause/resume
50 * Implementing this would be complicated so it's skipped. The problem case is:
51 * A full duplex connection is going, then play is paused. At this point you need to start xmitting
52 * 0's to keep the record active which means you cant just freeze the dma and resume it later you'd
53 * need to save off the dma info, and restore it properly on a resume. Yeach!
55 * Notes about transfer methods:
56 * The async write calls fail. I probably need to implement something else to support them?
58 ***************************************************************************************************/
60 #include <linux/module.h>
61 #include <linux/moduleparam.h>
62 #include <linux/init.h>
63 #include <linux/err.h>
64 #include <linux/platform_device.h>
65 #include <linux/errno.h>
66 #include <linux/ioctl.h>
67 #include <linux/delay.h>
68 #include <linux/slab.h>
74 #include <mach/hardware.h>
75 #include <mach/h3600.h>
76 #include <asm/mach-types.h>
79 #include <sound/core.h>
80 #include <sound/pcm.h>
81 #include <sound/initval.h>
83 #include <linux/l3/l3.h>
86 #undef DEBUG_FUNCTION_NAMES
87 #include <sound/uda1341.h>
90 * FIXME: Is this enough as autodetection of 2.4.X-rmkY-hhZ kernels?
91 * We use DMA stuff from 2.4.18-rmk3-hh24 here to be able to compile this
92 * module for Familiar 0.6.1
95 /* {{{ Type definitions */
97 MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek@seznam.cz>");
98 MODULE_LICENSE("GPL");
99 MODULE_DESCRIPTION("SA1100/SA1111 + UDA1341TS driver for ALSA");
100 MODULE_SUPPORTED_DEVICE("{{UDA1341,iPAQ H3600 UDA1341TS}}");
102 static char *id; /* ID for this card */
104 module_param(id, charp, 0444);
105 MODULE_PARM_DESC(id, "ID string for SA1100/SA1111 + UDA1341TS soundcard.");
107 struct audio_stream {
108 char *id; /* identification string */
109 int stream_id; /* numeric identification */
110 dma_device_t dma_dev; /* device identifier for DMA */
112 dmach_t dmach; /* dma channel identification */
114 dma_regs_t *dma_regs; /* points to our DMA registers */
116 unsigned int active:1; /* we are using this stream for transfer now */
117 int period; /* current transfer period */
118 int periods; /* current count of periods registerd in the DMA engine */
119 int tx_spin; /* are we recoding - flag used to do DMA trans. for sync */
120 unsigned int old_offset;
121 spinlock_t dma_lock; /* for locking in DMA operations (see dma-sa1100.c in the kernel) */
122 struct snd_pcm_substream *stream;
125 struct sa11xx_uda1341 {
126 struct snd_card *card;
127 struct l3_client *uda1341;
130 struct audio_stream s[2]; /* playback & capture */
133 static unsigned int rates[] = {
134 8000, 10666, 10985, 14647,
135 16000, 21970, 22050, 24000,
136 29400, 32000, 44100, 48000,
139 static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
140 .count = ARRAY_SIZE(rates),
145 static struct platform_device *device;
149 /* {{{ Clock and sample rate stuff */
152 * Stop-gap solution until rest of hh.org HAL stuff is merged.
154 #define GPIO_H3600_CLK_SET0 GPIO_GPIO (12)
155 #define GPIO_H3600_CLK_SET1 GPIO_GPIO (13)
157 #ifdef CONFIG_SA1100_H3XXX
158 #define clr_sa11xx_uda1341_egpio(x) clr_h3600_egpio(x)
159 #define set_sa11xx_uda1341_egpio(x) set_h3600_egpio(x)
161 #error This driver could serve H3x00 handhelds only!
164 static void sa11xx_uda1341_set_audio_clock(long val)
167 case 24000: case 32000: case 48000: /* 00: 12.288 MHz */
168 GPCR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1;
171 case 22050: case 29400: case 44100: /* 01: 11.2896 MHz */
172 GPSR = GPIO_H3600_CLK_SET0;
173 GPCR = GPIO_H3600_CLK_SET1;
176 case 8000: case 10666: case 16000: /* 10: 4.096 MHz */
177 GPCR = GPIO_H3600_CLK_SET0;
178 GPSR = GPIO_H3600_CLK_SET1;
181 case 10985: case 14647: case 21970: /* 11: 5.6245 MHz */
182 GPSR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1;
187 static void sa11xx_uda1341_set_samplerate(struct sa11xx_uda1341 *sa11xx_uda1341, long rate)
192 /* We don't want to mess with clocks when frames are in flight */
193 Ser4SSCR0 &= ~SSCR0_SSE;
194 /* wait for any frame to complete */
198 * We have the following clock sources:
199 * 4.096 MHz, 5.6245 MHz, 11.2896 MHz, 12.288 MHz
200 * Those can be divided either by 256, 384 or 512.
201 * This makes up 12 combinations for the following samplerates...
205 else if (rate >= 44100)
207 else if (rate >= 32000)
209 else if (rate >= 29400)
211 else if (rate >= 24000)
213 else if (rate >= 22050)
215 else if (rate >= 21970)
217 else if (rate >= 16000)
219 else if (rate >= 14647)
221 else if (rate >= 10985)
223 else if (rate >= 10666)
228 /* Set the external clock generator */
230 sa11xx_uda1341_set_audio_clock(rate);
232 /* Select the clock divisor */
239 clk_div = SSCR0_SerClkDiv(16);
246 clk_div = SSCR0_SerClkDiv(8);
253 clk_div = SSCR0_SerClkDiv(12);
257 /* FMT setting should be moved away when other FMTs are added (FIXME) */
258 l3_command(sa11xx_uda1341->uda1341, CMD_FORMAT, (void *)LSB16);
260 l3_command(sa11xx_uda1341->uda1341, CMD_FS, (void *)clk);
261 Ser4SSCR0 = (Ser4SSCR0 & ~0xff00) + clk_div + SSCR0_SSE;
262 sa11xx_uda1341->samplerate = rate;
267 /* {{{ HW init and shutdown */
269 static void sa11xx_uda1341_audio_init(struct sa11xx_uda1341 *sa11xx_uda1341)
273 /* Setup DMA stuff */
274 sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].id = "UDA1341 out";
275 sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].stream_id = SNDRV_PCM_STREAM_PLAYBACK;
276 sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].dma_dev = DMA_Ser4SSPWr;
278 sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].id = "UDA1341 in";
279 sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].stream_id = SNDRV_PCM_STREAM_CAPTURE;
280 sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].dma_dev = DMA_Ser4SSPRd;
282 /* Initialize the UDA1341 internal state */
284 /* Setup the uarts */
285 local_irq_save(flags);
286 GAFR |= (GPIO_SSP_CLK);
287 GPDR &= ~(GPIO_SSP_CLK);
289 Ser4SSCR0 = SSCR0_DataSize(16) + SSCR0_TI + SSCR0_SerClkDiv(8);
290 Ser4SSCR1 = SSCR1_SClkIactL + SSCR1_SClk1P + SSCR1_ExtClk;
291 Ser4SSCR0 |= SSCR0_SSE;
292 local_irq_restore(flags);
294 /* Enable the audio power */
296 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
297 set_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
298 set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
300 /* Wait for the UDA1341 to wake up */
301 mdelay(1); //FIXME - was removed by Perex - Why?
303 /* Initialize the UDA1341 internal state */
304 l3_open(sa11xx_uda1341->uda1341);
306 /* external clock configuration (after l3_open - regs must be initialized */
307 sa11xx_uda1341_set_samplerate(sa11xx_uda1341, sa11xx_uda1341->samplerate);
309 /* Wait for the UDA1341 to wake up */
310 set_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
313 /* make the left and right channels unswapped (flip the WS latch) */
316 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
319 static void sa11xx_uda1341_audio_shutdown(struct sa11xx_uda1341 *sa11xx_uda1341)
322 set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
324 /* disable the audio power and all signals leading to the audio chip */
325 l3_close(sa11xx_uda1341->uda1341);
327 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
329 /* power off and mute off */
330 /* FIXME - is muting off necesary??? */
332 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
333 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
341 * these are the address and sizes used to fill the xmit buffer
342 * so we can get a clock in record only mode
344 #define FORCE_CLOCK_ADDR (dma_addr_t)FLUSH_BASE_PHYS
345 #define FORCE_CLOCK_SIZE 4096 // was 2048
347 // FIXME Why this value exactly - wrote comment
348 #define DMA_BUF_SIZE 8176 /* <= MAX_DMA_SIZE from asm/arch-sa1100/dma.h */
352 static int audio_dma_request(struct audio_stream *s, void (*callback)(void *, int))
356 ret = sa1100_request_dma(&s->dmach, s->id, s->dma_dev);
358 printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev);
361 sa1100_dma_set_callback(s->dmach, callback);
365 static inline void audio_dma_free(struct audio_stream *s)
367 sa1100_free_dma(s->dmach);
373 static int audio_dma_request(struct audio_stream *s, void (*callback)(void *))
377 ret = sa1100_request_dma(s->dma_dev, s->id, callback, s, &s->dma_regs);
379 printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev);
383 static void audio_dma_free(struct audio_stream *s)
385 sa1100_free_dma(s->dma_regs);
391 static u_int audio_get_dma_pos(struct audio_stream *s)
393 struct snd_pcm_substream *substream = s->stream;
394 struct snd_pcm_runtime *runtime = substream->runtime;
399 // this must be called w/ interrupts locked out see dma-sa1100.c in the kernel
400 spin_lock_irqsave(&s->dma_lock, flags);
402 sa1100_dma_get_current(s->dmach, NULL, &addr);
404 addr = sa1100_get_dma_pos((s)->dma_regs);
406 offset = addr - runtime->dma_addr;
407 spin_unlock_irqrestore(&s->dma_lock, flags);
409 offset = bytes_to_frames(runtime,offset);
410 if (offset >= runtime->buffer_size)
417 * this stops the dma and clears the dma ptrs
419 static void audio_stop_dma(struct audio_stream *s)
423 spin_lock_irqsave(&s->dma_lock, flags);
426 /* this stops the dma channel and clears the buffer ptrs */
428 sa1100_dma_flush_all(s->dmach);
430 sa1100_clear_dma(s->dma_regs);
432 spin_unlock_irqrestore(&s->dma_lock, flags);
435 static void audio_process_dma(struct audio_stream *s)
437 struct snd_pcm_substream *substream = s->stream;
438 struct snd_pcm_runtime *runtime;
439 unsigned int dma_size;
443 /* we are requested to process synchronization DMA transfer */
445 snd_assert(s->stream_id == SNDRV_PCM_STREAM_PLAYBACK, return);
446 /* fill the xmit dma buffers and return */
448 sa1100_dma_set_spin(s->dmach, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE);
451 ret = sa1100_start_dma(s->dma_regs, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE);
459 /* must be set here - only valid for running streams, not for forced_clock dma fills */
460 runtime = substream->runtime;
461 while (s->active && s->periods < runtime->periods) {
462 dma_size = frames_to_bytes(runtime, runtime->period_size);
464 /* a little trick, we need resume from old position */
465 offset = frames_to_bytes(runtime, s->old_offset - 1);
468 s->period = offset / dma_size;
470 dma_size = dma_size - offset;
472 continue; /* special case */
474 offset = dma_size * s->period;
475 snd_assert(dma_size <= DMA_BUF_SIZE, );
478 ret = sa1100_dma_queue_buffer(s->dmach, s, runtime->dma_addr + offset, dma_size);
482 ret = sa1100_start_dma((s)->dma_regs, runtime->dma_addr + offset, dma_size);
484 printk(KERN_ERR "audio_process_dma: cannot queue DMA buffer (%i)\n", ret);
490 s->period %= runtime->periods;
496 static void audio_dma_callback(void *data, int size)
498 static void audio_dma_callback(void *data)
501 struct audio_stream *s = data;
504 * If we are getting a callback for an active stream then we inform
505 * the PCM middle layer we've finished a period
508 snd_pcm_period_elapsed(s->stream);
510 spin_lock(&s->dma_lock);
511 if (!s->tx_spin && s->periods > 0)
513 audio_process_dma(s);
514 spin_unlock(&s->dma_lock);
519 /* {{{ PCM setting */
521 /* {{{ trigger & timer */
523 static int snd_sa11xx_uda1341_trigger(struct snd_pcm_substream *substream, int cmd)
525 struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
526 int stream_id = substream->pstr->stream;
527 struct audio_stream *s = &chip->s[stream_id];
528 struct audio_stream *s1 = &chip->s[stream_id ^ 1];
531 /* note local interrupts are already disabled in the midlevel code */
532 spin_lock(&s->dma_lock);
534 case SNDRV_PCM_TRIGGER_START:
535 /* now we need to make sure a record only stream has a clock */
536 if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
537 /* we need to force fill the xmit DMA with zeros */
539 audio_process_dma(s1);
541 /* this case is when you were recording then you turn on a
542 * playback stream so we stop (also clears it) the dma first,
543 * clear the sync flag and then we let it turned on
549 /* requested stream startup */
551 audio_process_dma(s);
553 case SNDRV_PCM_TRIGGER_STOP:
554 /* requested stream shutdown */
558 * now we need to make sure a record only stream has a clock
559 * so if we're stopping a playback with an active capture
560 * we need to turn the 0 fill dma on for the xmit side
562 if (stream_id == SNDRV_PCM_STREAM_PLAYBACK && s1->active) {
563 /* we need to force fill the xmit DMA with zeros */
565 audio_process_dma(s);
568 * we killed a capture only stream, so we should also kill
569 * the zero fill transmit
579 case SNDRV_PCM_TRIGGER_SUSPEND:
582 sa1100_dma_stop(s->dmach);
586 s->old_offset = audio_get_dma_pos(s) + 1;
588 sa1100_dma_flush_all(s->dmach);
594 case SNDRV_PCM_TRIGGER_RESUME:
597 audio_process_dma(s);
598 if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
600 audio_process_dma(s1);
603 case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
605 sa1100_dma_stop(s->dmach);
610 if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) {
613 s->old_offset = audio_get_dma_pos(s) + 1;
615 sa1100_dma_flush_all(s->dmach);
619 audio_process_dma(s);
625 sa1100_dma_flush_all(s1->dmach);
632 case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
636 audio_process_dma(s);
639 if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
641 audio_process_dma(s1);
644 sa1100_dma_resume(s->dmach);
653 spin_unlock(&s->dma_lock);
657 static int snd_sa11xx_uda1341_prepare(struct snd_pcm_substream *substream)
659 struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
660 struct snd_pcm_runtime *runtime = substream->runtime;
661 struct audio_stream *s = &chip->s[substream->pstr->stream];
663 /* set requested samplerate */
664 sa11xx_uda1341_set_samplerate(chip, runtime->rate);
666 /* set requestd format when available */
667 /* set FMT here !!! FIXME */
675 static snd_pcm_uframes_t snd_sa11xx_uda1341_pointer(struct snd_pcm_substream *substream)
677 struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
678 return audio_get_dma_pos(&chip->s[substream->pstr->stream]);
683 static struct snd_pcm_hardware snd_sa11xx_uda1341_capture =
685 .info = (SNDRV_PCM_INFO_INTERLEAVED |
686 SNDRV_PCM_INFO_BLOCK_TRANSFER |
687 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
688 SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
689 .formats = SNDRV_PCM_FMTBIT_S16_LE,
690 .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\
691 SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\
692 SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
693 SNDRV_PCM_RATE_KNOT),
698 .buffer_bytes_max = 64*1024,
699 .period_bytes_min = 64,
700 .period_bytes_max = DMA_BUF_SIZE,
706 static struct snd_pcm_hardware snd_sa11xx_uda1341_playback =
708 .info = (SNDRV_PCM_INFO_INTERLEAVED |
709 SNDRV_PCM_INFO_BLOCK_TRANSFER |
710 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
711 SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
712 .formats = SNDRV_PCM_FMTBIT_S16_LE,
713 .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\
714 SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\
715 SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
716 SNDRV_PCM_RATE_KNOT),
721 .buffer_bytes_max = 64*1024,
722 .period_bytes_min = 64,
723 .period_bytes_max = DMA_BUF_SIZE,
729 static int snd_card_sa11xx_uda1341_open(struct snd_pcm_substream *substream)
731 struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
732 struct snd_pcm_runtime *runtime = substream->runtime;
733 int stream_id = substream->pstr->stream;
736 chip->s[stream_id].stream = substream;
738 if (stream_id == SNDRV_PCM_STREAM_PLAYBACK)
739 runtime->hw = snd_sa11xx_uda1341_playback;
741 runtime->hw = snd_sa11xx_uda1341_capture;
742 if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
744 if ((err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates)) < 0)
750 static int snd_card_sa11xx_uda1341_close(struct snd_pcm_substream *substream)
752 struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
754 chip->s[substream->pstr->stream].stream = NULL;
758 /* {{{ HW params & free */
760 static int snd_sa11xx_uda1341_hw_params(struct snd_pcm_substream *substream,
761 struct snd_pcm_hw_params *hw_params)
764 return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
767 static int snd_sa11xx_uda1341_hw_free(struct snd_pcm_substream *substream)
769 return snd_pcm_lib_free_pages(substream);
774 static struct snd_pcm_ops snd_card_sa11xx_uda1341_playback_ops = {
775 .open = snd_card_sa11xx_uda1341_open,
776 .close = snd_card_sa11xx_uda1341_close,
777 .ioctl = snd_pcm_lib_ioctl,
778 .hw_params = snd_sa11xx_uda1341_hw_params,
779 .hw_free = snd_sa11xx_uda1341_hw_free,
780 .prepare = snd_sa11xx_uda1341_prepare,
781 .trigger = snd_sa11xx_uda1341_trigger,
782 .pointer = snd_sa11xx_uda1341_pointer,
785 static struct snd_pcm_ops snd_card_sa11xx_uda1341_capture_ops = {
786 .open = snd_card_sa11xx_uda1341_open,
787 .close = snd_card_sa11xx_uda1341_close,
788 .ioctl = snd_pcm_lib_ioctl,
789 .hw_params = snd_sa11xx_uda1341_hw_params,
790 .hw_free = snd_sa11xx_uda1341_hw_free,
791 .prepare = snd_sa11xx_uda1341_prepare,
792 .trigger = snd_sa11xx_uda1341_trigger,
793 .pointer = snd_sa11xx_uda1341_pointer,
796 static int __init snd_card_sa11xx_uda1341_pcm(struct sa11xx_uda1341 *sa11xx_uda1341, int device)
801 if ((err = snd_pcm_new(sa11xx_uda1341->card, "UDA1341 PCM", device, 1, 1, &pcm)) < 0)
805 * this sets up our initial buffers and sets the dma_type to isa.
806 * isa works but I'm not sure why (or if) it's the right choice
807 * this may be too large, trying it for now
809 snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
813 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_card_sa11xx_uda1341_playback_ops);
814 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_card_sa11xx_uda1341_capture_ops);
815 pcm->private_data = sa11xx_uda1341;
817 strcpy(pcm->name, "UDA1341 PCM");
819 sa11xx_uda1341_audio_init(sa11xx_uda1341);
821 /* setup DMA controller */
822 audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK], audio_dma_callback);
823 audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE], audio_dma_callback);
825 sa11xx_uda1341->pcm = pcm;
832 /* {{{ module init & exit */
836 static int snd_sa11xx_uda1341_suspend(struct platform_device *devptr,
839 struct snd_card *card = platform_get_drvdata(devptr);
840 struct sa11xx_uda1341 *chip = card->private_data;
842 snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
843 snd_pcm_suspend_all(chip->pcm);
845 sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach);
846 sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach);
850 l3_command(chip->uda1341, CMD_SUSPEND, NULL);
851 sa11xx_uda1341_audio_shutdown(chip);
856 static int snd_sa11xx_uda1341_resume(struct platform_device *devptr)
858 struct snd_card *card = platform_get_drvdata(devptr);
859 struct sa11xx_uda1341 *chip = card->private_data;
861 sa11xx_uda1341_audio_init(chip);
862 l3_command(chip->uda1341, CMD_RESUME, NULL);
864 sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach);
865 sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach);
869 snd_power_change_state(card, SNDRV_CTL_POWER_D0);
872 #endif /* COMFIG_PM */
874 void snd_sa11xx_uda1341_free(struct snd_card *card)
876 struct sa11xx_uda1341 *chip = card->private_data;
878 audio_dma_free(&chip->s[SNDRV_PCM_STREAM_PLAYBACK]);
879 audio_dma_free(&chip->s[SNDRV_PCM_STREAM_CAPTURE]);
882 static int __init sa11xx_uda1341_probe(struct platform_device *devptr)
885 struct snd_card *card;
886 struct sa11xx_uda1341 *chip;
888 /* register the soundcard */
889 card = snd_card_new(-1, id, THIS_MODULE, sizeof(struct sa11xx_uda1341));
893 chip = card->private_data;
894 spin_lock_init(&chip->s[0].dma_lock);
895 spin_lock_init(&chip->s[1].dma_lock);
897 card->private_free = snd_sa11xx_uda1341_free;
899 chip->samplerate = AUDIO_RATE_DEFAULT;
902 if ((err = snd_chip_uda1341_mixer_new(card, &chip->uda1341)))
906 if ((err = snd_card_sa11xx_uda1341_pcm(chip, 0)) < 0)
909 strcpy(card->driver, "UDA1341");
910 strcpy(card->shortname, "H3600 UDA1341TS");
911 sprintf(card->longname, "Compaq iPAQ H3600 with Philips UDA1341TS");
913 snd_card_set_dev(card, &devptr->dev);
915 if ((err = snd_card_register(card)) == 0) {
916 printk( KERN_INFO "iPAQ audio support initialized\n" );
917 platform_set_drvdata(devptr, card);
926 static int __devexit sa11xx_uda1341_remove(struct platform_device *devptr)
928 snd_card_free(platform_get_drvdata(devptr));
929 platform_set_drvdata(devptr, NULL);
933 #define SA11XX_UDA1341_DRIVER "sa11xx_uda1341"
935 static struct platform_driver sa11xx_uda1341_driver = {
936 .probe = sa11xx_uda1341_probe,
937 .remove = __devexit_p(sa11xx_uda1341_remove),
939 .suspend = snd_sa11xx_uda1341_suspend,
940 .resume = snd_sa11xx_uda1341_resume,
943 .name = SA11XX_UDA1341_DRIVER,
947 static int __init sa11xx_uda1341_init(void)
951 if (!machine_is_h3xxx())
953 if ((err = platform_driver_register(&sa11xx_uda1341_driver)) < 0)
955 device = platform_device_register_simple(SA11XX_UDA1341_DRIVER, -1, NULL, 0);
956 if (!IS_ERR(device)) {
957 if (platform_get_drvdata(device))
959 platform_device_unregister(device);
962 err = PTR_ERR(device);
963 platform_driver_unregister(&sa11xx_uda1341_driver);
967 static void __exit sa11xx_uda1341_exit(void)
969 platform_device_unregister(device);
970 platform_driver_unregister(&sa11xx_uda1341_driver);
973 module_init(sa11xx_uda1341_init);
974 module_exit(sa11xx_uda1341_exit);
980 * indent-tabs-mode: t