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 /* $Id: sa11xx-uda1341.c,v 1.27 2005/12/07 09:13:42 cladisch Exp $ */
26 /***************************************************************************************************
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
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.
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
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.
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!
57 * Notes about transfer methods:
58 * The async write calls fail. I probably need to implement something else to support them?
60 ***************************************************************************************************/
62 #include <sound/driver.h>
63 #include <linux/module.h>
64 #include <linux/moduleparam.h>
65 #include <linux/init.h>
66 #include <linux/err.h>
67 #include <linux/platform_device.h>
68 #include <linux/errno.h>
69 #include <linux/ioctl.h>
70 #include <linux/delay.h>
71 #include <linux/slab.h>
77 #include <asm/hardware.h>
78 #include <asm/arch/h3600.h>
79 #include <asm/mach-types.h>
82 #ifdef CONFIG_H3600_HAL
83 #include <asm/semaphore.h>
84 #include <asm/uaccess.h>
85 #include <asm/arch/h3600_hal.h>
88 #include <sound/core.h>
89 #include <sound/pcm.h>
90 #include <sound/initval.h>
92 #include <linux/l3/l3.h>
95 #undef DEBUG_FUNCTION_NAMES
96 #include <sound/uda1341.h>
99 * FIXME: Is this enough as autodetection of 2.4.X-rmkY-hhZ kernels?
100 * We use DMA stuff from 2.4.18-rmk3-hh24 here to be able to compile this
101 * module for Familiar 0.6.1
103 #ifdef CONFIG_H3600_HAL
107 /* {{{ Type definitions */
109 MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek@seznam.cz>");
110 MODULE_LICENSE("GPL");
111 MODULE_DESCRIPTION("SA1100/SA1111 + UDA1341TS driver for ALSA");
112 MODULE_SUPPORTED_DEVICE("{{UDA1341,iPAQ H3600 UDA1341TS}}");
114 static char *id; /* ID for this card */
116 module_param(id, charp, 0444);
117 MODULE_PARM_DESC(id, "ID string for SA1100/SA1111 + UDA1341TS soundcard.");
119 struct audio_stream {
120 char *id; /* identification string */
121 int stream_id; /* numeric identification */
122 dma_device_t dma_dev; /* device identifier for DMA */
124 dmach_t dmach; /* dma channel identification */
126 dma_regs_t *dma_regs; /* points to our DMA registers */
128 unsigned int active:1; /* we are using this stream for transfer now */
129 int period; /* current transfer period */
130 int periods; /* current count of periods registerd in the DMA engine */
131 int tx_spin; /* are we recoding - flag used to do DMA trans. for sync */
132 unsigned int old_offset;
133 spinlock_t dma_lock; /* for locking in DMA operations (see dma-sa1100.c in the kernel) */
134 struct snd_pcm_substream *stream;
137 struct sa11xx_uda1341 {
138 struct snd_card *card;
139 struct l3_client *uda1341;
142 struct audio_stream s[2]; /* playback & capture */
145 static unsigned int rates[] = {
146 8000, 10666, 10985, 14647,
147 16000, 21970, 22050, 24000,
148 29400, 32000, 44100, 48000,
151 static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
152 .count = ARRAY_SIZE(rates),
157 static struct platform_device *device;
161 /* {{{ Clock and sample rate stuff */
164 * Stop-gap solution until rest of hh.org HAL stuff is merged.
166 #define GPIO_H3600_CLK_SET0 GPIO_GPIO (12)
167 #define GPIO_H3600_CLK_SET1 GPIO_GPIO (13)
169 #ifdef CONFIG_SA1100_H3XXX
170 #define clr_sa11xx_uda1341_egpio(x) clr_h3600_egpio(x)
171 #define set_sa11xx_uda1341_egpio(x) set_h3600_egpio(x)
173 #error This driver could serve H3x00 handhelds only!
176 static void sa11xx_uda1341_set_audio_clock(long val)
179 case 24000: case 32000: case 48000: /* 00: 12.288 MHz */
180 GPCR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1;
183 case 22050: case 29400: case 44100: /* 01: 11.2896 MHz */
184 GPSR = GPIO_H3600_CLK_SET0;
185 GPCR = GPIO_H3600_CLK_SET1;
188 case 8000: case 10666: case 16000: /* 10: 4.096 MHz */
189 GPCR = GPIO_H3600_CLK_SET0;
190 GPSR = GPIO_H3600_CLK_SET1;
193 case 10985: case 14647: case 21970: /* 11: 5.6245 MHz */
194 GPSR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1;
199 static void sa11xx_uda1341_set_samplerate(struct sa11xx_uda1341 *sa11xx_uda1341, long rate)
204 /* We don't want to mess with clocks when frames are in flight */
205 Ser4SSCR0 &= ~SSCR0_SSE;
206 /* wait for any frame to complete */
210 * We have the following clock sources:
211 * 4.096 MHz, 5.6245 MHz, 11.2896 MHz, 12.288 MHz
212 * Those can be divided either by 256, 384 or 512.
213 * This makes up 12 combinations for the following samplerates...
217 else if (rate >= 44100)
219 else if (rate >= 32000)
221 else if (rate >= 29400)
223 else if (rate >= 24000)
225 else if (rate >= 22050)
227 else if (rate >= 21970)
229 else if (rate >= 16000)
231 else if (rate >= 14647)
233 else if (rate >= 10985)
235 else if (rate >= 10666)
240 /* Set the external clock generator */
241 #ifdef CONFIG_H3600_HAL
242 h3600_audio_clock(rate);
244 sa11xx_uda1341_set_audio_clock(rate);
247 /* Select the clock divisor */
254 clk_div = SSCR0_SerClkDiv(16);
261 clk_div = SSCR0_SerClkDiv(8);
268 clk_div = SSCR0_SerClkDiv(12);
272 /* FMT setting should be moved away when other FMTs are added (FIXME) */
273 l3_command(sa11xx_uda1341->uda1341, CMD_FORMAT, (void *)LSB16);
275 l3_command(sa11xx_uda1341->uda1341, CMD_FS, (void *)clk);
276 Ser4SSCR0 = (Ser4SSCR0 & ~0xff00) + clk_div + SSCR0_SSE;
277 sa11xx_uda1341->samplerate = rate;
282 /* {{{ HW init and shutdown */
284 static void sa11xx_uda1341_audio_init(struct sa11xx_uda1341 *sa11xx_uda1341)
288 /* Setup DMA stuff */
289 sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].id = "UDA1341 out";
290 sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].stream_id = SNDRV_PCM_STREAM_PLAYBACK;
291 sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].dma_dev = DMA_Ser4SSPWr;
293 sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].id = "UDA1341 in";
294 sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].stream_id = SNDRV_PCM_STREAM_CAPTURE;
295 sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].dma_dev = DMA_Ser4SSPRd;
297 /* Initialize the UDA1341 internal state */
299 /* Setup the uarts */
300 local_irq_save(flags);
301 GAFR |= (GPIO_SSP_CLK);
302 GPDR &= ~(GPIO_SSP_CLK);
304 Ser4SSCR0 = SSCR0_DataSize(16) + SSCR0_TI + SSCR0_SerClkDiv(8);
305 Ser4SSCR1 = SSCR1_SClkIactL + SSCR1_SClk1P + SSCR1_ExtClk;
306 Ser4SSCR0 |= SSCR0_SSE;
307 local_irq_restore(flags);
309 /* Enable the audio power */
310 #ifdef CONFIG_H3600_HAL
311 h3600_audio_power(AUDIO_RATE_DEFAULT);
313 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
314 set_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
315 set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
318 /* Wait for the UDA1341 to wake up */
319 mdelay(1); //FIXME - was removed by Perex - Why?
321 /* Initialize the UDA1341 internal state */
322 l3_open(sa11xx_uda1341->uda1341);
324 /* external clock configuration (after l3_open - regs must be initialized */
325 sa11xx_uda1341_set_samplerate(sa11xx_uda1341, sa11xx_uda1341->samplerate);
327 /* Wait for the UDA1341 to wake up */
328 set_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
331 /* make the left and right channels unswapped (flip the WS latch) */
334 #ifdef CONFIG_H3600_HAL
337 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
341 static void sa11xx_uda1341_audio_shutdown(struct sa11xx_uda1341 *sa11xx_uda1341)
344 #ifdef CONFIG_H3600_HAL
347 set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
350 /* disable the audio power and all signals leading to the audio chip */
351 l3_close(sa11xx_uda1341->uda1341);
353 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
355 /* power off and mute off */
356 /* FIXME - is muting off necesary??? */
357 #ifdef CONFIG_H3600_HAL
358 h3600_audio_power(0);
361 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
362 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
371 * these are the address and sizes used to fill the xmit buffer
372 * so we can get a clock in record only mode
374 #define FORCE_CLOCK_ADDR (dma_addr_t)FLUSH_BASE_PHYS
375 #define FORCE_CLOCK_SIZE 4096 // was 2048
377 // FIXME Why this value exactly - wrote comment
378 #define DMA_BUF_SIZE 8176 /* <= MAX_DMA_SIZE from asm/arch-sa1100/dma.h */
382 static int audio_dma_request(struct audio_stream *s, void (*callback)(void *, int))
386 ret = sa1100_request_dma(&s->dmach, s->id, s->dma_dev);
388 printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev);
391 sa1100_dma_set_callback(s->dmach, callback);
395 static inline void audio_dma_free(struct audio_stream *s)
397 sa1100_free_dma(s->dmach);
403 static int audio_dma_request(struct audio_stream *s, void (*callback)(void *))
407 ret = sa1100_request_dma(s->dma_dev, s->id, callback, s, &s->dma_regs);
409 printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev);
413 static void audio_dma_free(struct audio_stream *s)
415 sa1100_free_dma(s->dma_regs);
421 static u_int audio_get_dma_pos(struct audio_stream *s)
423 struct snd_pcm_substream *substream = s->stream;
424 struct snd_pcm_runtime *runtime = substream->runtime;
429 // this must be called w/ interrupts locked out see dma-sa1100.c in the kernel
430 spin_lock_irqsave(&s->dma_lock, flags);
432 sa1100_dma_get_current(s->dmach, NULL, &addr);
434 addr = sa1100_get_dma_pos((s)->dma_regs);
436 offset = addr - runtime->dma_addr;
437 spin_unlock_irqrestore(&s->dma_lock, flags);
439 offset = bytes_to_frames(runtime,offset);
440 if (offset >= runtime->buffer_size)
447 * this stops the dma and clears the dma ptrs
449 static void audio_stop_dma(struct audio_stream *s)
453 spin_lock_irqsave(&s->dma_lock, flags);
456 /* this stops the dma channel and clears the buffer ptrs */
458 sa1100_dma_flush_all(s->dmach);
460 sa1100_clear_dma(s->dma_regs);
462 spin_unlock_irqrestore(&s->dma_lock, flags);
465 static void audio_process_dma(struct audio_stream *s)
467 struct snd_pcm_substream *substream = s->stream;
468 struct snd_pcm_runtime *runtime;
469 unsigned int dma_size;
473 /* we are requested to process synchronization DMA transfer */
475 snd_assert(s->stream_id == SNDRV_PCM_STREAM_PLAYBACK, return);
476 /* fill the xmit dma buffers and return */
478 sa1100_dma_set_spin(s->dmach, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE);
481 ret = sa1100_start_dma(s->dma_regs, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE);
489 /* must be set here - only valid for running streams, not for forced_clock dma fills */
490 runtime = substream->runtime;
491 while (s->active && s->periods < runtime->periods) {
492 dma_size = frames_to_bytes(runtime, runtime->period_size);
494 /* a little trick, we need resume from old position */
495 offset = frames_to_bytes(runtime, s->old_offset - 1);
498 s->period = offset / dma_size;
500 dma_size = dma_size - offset;
502 continue; /* special case */
504 offset = dma_size * s->period;
505 snd_assert(dma_size <= DMA_BUF_SIZE, );
508 ret = sa1100_dma_queue_buffer(s->dmach, s, runtime->dma_addr + offset, dma_size);
512 ret = sa1100_start_dma((s)->dma_regs, runtime->dma_addr + offset, dma_size);
514 printk(KERN_ERR "audio_process_dma: cannot queue DMA buffer (%i)\n", ret);
520 s->period %= runtime->periods;
526 static void audio_dma_callback(void *data, int size)
528 static void audio_dma_callback(void *data)
531 struct audio_stream *s = data;
534 * If we are getting a callback for an active stream then we inform
535 * the PCM middle layer we've finished a period
538 snd_pcm_period_elapsed(s->stream);
540 spin_lock(&s->dma_lock);
541 if (!s->tx_spin && s->periods > 0)
543 audio_process_dma(s);
544 spin_unlock(&s->dma_lock);
549 /* {{{ PCM setting */
551 /* {{{ trigger & timer */
553 static int snd_sa11xx_uda1341_trigger(struct snd_pcm_substream *substream, int cmd)
555 struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
556 int stream_id = substream->pstr->stream;
557 struct audio_stream *s = &chip->s[stream_id];
558 struct audio_stream *s1 = &chip->s[stream_id ^ 1];
561 /* note local interrupts are already disabled in the midlevel code */
562 spin_lock(&s->dma_lock);
564 case SNDRV_PCM_TRIGGER_START:
565 /* now we need to make sure a record only stream has a clock */
566 if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
567 /* we need to force fill the xmit DMA with zeros */
569 audio_process_dma(s1);
571 /* this case is when you were recording then you turn on a
572 * playback stream so we stop (also clears it) the dma first,
573 * clear the sync flag and then we let it turned on
579 /* requested stream startup */
581 audio_process_dma(s);
583 case SNDRV_PCM_TRIGGER_STOP:
584 /* requested stream shutdown */
588 * now we need to make sure a record only stream has a clock
589 * so if we're stopping a playback with an active capture
590 * we need to turn the 0 fill dma on for the xmit side
592 if (stream_id == SNDRV_PCM_STREAM_PLAYBACK && s1->active) {
593 /* we need to force fill the xmit DMA with zeros */
595 audio_process_dma(s);
598 * we killed a capture only stream, so we should also kill
599 * the zero fill transmit
609 case SNDRV_PCM_TRIGGER_SUSPEND:
612 sa1100_dma_stop(s->dmach);
616 s->old_offset = audio_get_dma_pos(s) + 1;
618 sa1100_dma_flush_all(s->dmach);
624 case SNDRV_PCM_TRIGGER_RESUME:
627 audio_process_dma(s);
628 if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
630 audio_process_dma(s1);
633 case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
635 sa1100_dma_stop(s->dmach);
640 if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) {
643 s->old_offset = audio_get_dma_pos(s) + 1;
645 sa1100_dma_flush_all(s->dmach);
649 audio_process_dma(s);
655 sa1100_dma_flush_all(s1->dmach);
662 case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
666 audio_process_dma(s);
669 if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
671 audio_process_dma(s1);
674 sa1100_dma_resume(s->dmach);
683 spin_unlock(&s->dma_lock);
687 static int snd_sa11xx_uda1341_prepare(struct snd_pcm_substream *substream)
689 struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
690 struct snd_pcm_runtime *runtime = substream->runtime;
691 struct audio_stream *s = &chip->s[substream->pstr->stream];
693 /* set requested samplerate */
694 sa11xx_uda1341_set_samplerate(chip, runtime->rate);
696 /* set requestd format when available */
697 /* set FMT here !!! FIXME */
705 static snd_pcm_uframes_t snd_sa11xx_uda1341_pointer(struct snd_pcm_substream *substream)
707 struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
708 return audio_get_dma_pos(&chip->s[substream->pstr->stream]);
713 static struct snd_pcm_hardware snd_sa11xx_uda1341_capture =
715 .info = (SNDRV_PCM_INFO_INTERLEAVED |
716 SNDRV_PCM_INFO_BLOCK_TRANSFER |
717 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
718 SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
719 .formats = SNDRV_PCM_FMTBIT_S16_LE,
720 .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\
721 SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\
722 SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
723 SNDRV_PCM_RATE_KNOT),
728 .buffer_bytes_max = 64*1024,
729 .period_bytes_min = 64,
730 .period_bytes_max = DMA_BUF_SIZE,
736 static struct snd_pcm_hardware snd_sa11xx_uda1341_playback =
738 .info = (SNDRV_PCM_INFO_INTERLEAVED |
739 SNDRV_PCM_INFO_BLOCK_TRANSFER |
740 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
741 SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
742 .formats = SNDRV_PCM_FMTBIT_S16_LE,
743 .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\
744 SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\
745 SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
746 SNDRV_PCM_RATE_KNOT),
751 .buffer_bytes_max = 64*1024,
752 .period_bytes_min = 64,
753 .period_bytes_max = DMA_BUF_SIZE,
759 static int snd_card_sa11xx_uda1341_open(struct snd_pcm_substream *substream)
761 struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
762 struct snd_pcm_runtime *runtime = substream->runtime;
763 int stream_id = substream->pstr->stream;
766 chip->s[stream_id].stream = substream;
768 if (stream_id == SNDRV_PCM_STREAM_PLAYBACK)
769 runtime->hw = snd_sa11xx_uda1341_playback;
771 runtime->hw = snd_sa11xx_uda1341_capture;
772 if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
774 if ((err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates)) < 0)
780 static int snd_card_sa11xx_uda1341_close(struct snd_pcm_substream *substream)
782 struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
784 chip->s[substream->pstr->stream].stream = NULL;
788 /* {{{ HW params & free */
790 static int snd_sa11xx_uda1341_hw_params(struct snd_pcm_substream *substream,
791 struct snd_pcm_hw_params *hw_params)
794 return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
797 static int snd_sa11xx_uda1341_hw_free(struct snd_pcm_substream *substream)
799 return snd_pcm_lib_free_pages(substream);
804 static struct snd_pcm_ops snd_card_sa11xx_uda1341_playback_ops = {
805 .open = snd_card_sa11xx_uda1341_open,
806 .close = snd_card_sa11xx_uda1341_close,
807 .ioctl = snd_pcm_lib_ioctl,
808 .hw_params = snd_sa11xx_uda1341_hw_params,
809 .hw_free = snd_sa11xx_uda1341_hw_free,
810 .prepare = snd_sa11xx_uda1341_prepare,
811 .trigger = snd_sa11xx_uda1341_trigger,
812 .pointer = snd_sa11xx_uda1341_pointer,
815 static struct snd_pcm_ops snd_card_sa11xx_uda1341_capture_ops = {
816 .open = snd_card_sa11xx_uda1341_open,
817 .close = snd_card_sa11xx_uda1341_close,
818 .ioctl = snd_pcm_lib_ioctl,
819 .hw_params = snd_sa11xx_uda1341_hw_params,
820 .hw_free = snd_sa11xx_uda1341_hw_free,
821 .prepare = snd_sa11xx_uda1341_prepare,
822 .trigger = snd_sa11xx_uda1341_trigger,
823 .pointer = snd_sa11xx_uda1341_pointer,
826 static int __init snd_card_sa11xx_uda1341_pcm(struct sa11xx_uda1341 *sa11xx_uda1341, int device)
831 if ((err = snd_pcm_new(sa11xx_uda1341->card, "UDA1341 PCM", device, 1, 1, &pcm)) < 0)
835 * this sets up our initial buffers and sets the dma_type to isa.
836 * isa works but I'm not sure why (or if) it's the right choice
837 * this may be too large, trying it for now
839 snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
843 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_card_sa11xx_uda1341_playback_ops);
844 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_card_sa11xx_uda1341_capture_ops);
845 pcm->private_data = sa11xx_uda1341;
847 strcpy(pcm->name, "UDA1341 PCM");
849 sa11xx_uda1341_audio_init(sa11xx_uda1341);
851 /* setup DMA controller */
852 audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK], audio_dma_callback);
853 audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE], audio_dma_callback);
855 sa11xx_uda1341->pcm = pcm;
862 /* {{{ module init & exit */
866 static int snd_sa11xx_uda1341_suspend(struct platform_device *devptr,
869 struct snd_card *card = platform_get_drvdata(devptr);
870 struct sa11xx_uda1341 *chip = card->private_data;
872 snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
873 snd_pcm_suspend_all(chip->pcm);
875 sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach);
876 sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach);
880 l3_command(chip->uda1341, CMD_SUSPEND, NULL);
881 sa11xx_uda1341_audio_shutdown(chip);
886 static int snd_sa11xx_uda1341_resume(struct platform_device *devptr)
888 struct snd_card *card = platform_get_drvdata(devptr);
889 struct sa11xx_uda1341 *chip = card->private_data;
891 sa11xx_uda1341_audio_init(chip);
892 l3_command(chip->uda1341, CMD_RESUME, NULL);
894 sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach);
895 sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach);
899 snd_power_change_state(card, SNDRV_CTL_POWER_D0);
902 #endif /* COMFIG_PM */
904 void snd_sa11xx_uda1341_free(struct snd_card *card)
906 struct sa11xx_uda1341 *chip = card->private_data;
908 audio_dma_free(&chip->s[SNDRV_PCM_STREAM_PLAYBACK]);
909 audio_dma_free(&chip->s[SNDRV_PCM_STREAM_CAPTURE]);
912 static int __init sa11xx_uda1341_probe(struct platform_device *devptr)
915 struct snd_card *card;
916 struct sa11xx_uda1341 *chip;
918 /* register the soundcard */
919 card = snd_card_new(-1, id, THIS_MODULE, sizeof(struct sa11xx_uda1341));
923 chip = card->private_data;
924 spin_lock_init(&chip->s[0].dma_lock);
925 spin_lock_init(&chip->s[1].dma_lock);
927 card->private_free = snd_sa11xx_uda1341_free;
929 chip->samplerate = AUDIO_RATE_DEFAULT;
932 if ((err = snd_chip_uda1341_mixer_new(card, &chip->uda1341)))
936 if ((err = snd_card_sa11xx_uda1341_pcm(chip, 0)) < 0)
939 strcpy(card->driver, "UDA1341");
940 strcpy(card->shortname, "H3600 UDA1341TS");
941 sprintf(card->longname, "Compaq iPAQ H3600 with Philips UDA1341TS");
943 snd_card_set_dev(card, &devptr->dev);
945 if ((err = snd_card_register(card)) == 0) {
946 printk( KERN_INFO "iPAQ audio support initialized\n" );
947 platform_set_drvdata(devptr, card);
956 static int __devexit sa11xx_uda1341_remove(struct platform_device *devptr)
958 snd_card_free(platform_get_drvdata(devptr));
959 platform_set_drvdata(devptr, NULL);
963 #define SA11XX_UDA1341_DRIVER "sa11xx_uda1341"
965 static struct platform_driver sa11xx_uda1341_driver = {
966 .probe = sa11xx_uda1341_probe,
967 .remove = __devexit_p(sa11xx_uda1341_remove),
969 .suspend = snd_sa11xx_uda1341_suspend,
970 .resume = snd_sa11xx_uda1341_resume,
973 .name = SA11XX_UDA1341_DRIVER,
977 static int __init sa11xx_uda1341_init(void)
981 if (!machine_is_h3xxx())
983 if ((err = platform_driver_register(&sa11xx_uda1341_driver)) < 0)
985 device = platform_device_register_simple(SA11XX_UDA1341_DRIVER, -1, NULL, 0);
986 if (!IS_ERR(device)) {
987 if (platform_get_drvdata(device))
989 platform_device_unregister(device);
992 err = PTR_ERR(device);
993 platform_driver_unregister(&sa11xx_uda1341_driver);
997 static void __exit sa11xx_uda1341_exit(void)
999 platform_device_unregister(device);
1000 platform_driver_unregister(&sa11xx_uda1341_driver);
1003 module_init(sa11xx_uda1341_init);
1004 module_exit(sa11xx_uda1341_exit);
1010 * indent-tabs-mode: t