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 1.13 2004/03/02 15:32:35 perex 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 <linux/config.h>
63 #include <sound/driver.h>
64 #include <linux/module.h>
65 #include <linux/init.h>
66 #include <linux/errno.h>
67 #include <linux/ioctl.h>
68 #include <linux/delay.h>
69 #include <linux/slab.h>
75 #include <asm/hardware.h>
76 #include <asm/arch/h3600.h>
77 #include <asm/mach-types.h>
80 #ifdef CONFIG_H3600_HAL
81 #include <asm/semaphore.h>
82 #include <asm/uaccess.h>
83 #include <asm/arch/h3600_hal.h>
86 #include <sound/core.h>
87 #include <sound/pcm.h>
88 #include <sound/initval.h>
90 #include <linux/l3/l3.h>
93 #undef DEBUG_FUNCTION_NAMES
94 #include <sound/uda1341.h>
97 * FIXME: Is this enough as autodetection of 2.4.X-rmkY-hhZ kernels?
98 * We use DMA stuff from 2.4.18-rmk3-hh24 here to be able to compile this
99 * module for Familiar 0.6.1
101 #ifdef CONFIG_H3600_HAL
105 /* {{{ Type definitions */
107 MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek@seznam.cz>");
108 MODULE_LICENSE("GPL");
109 MODULE_DESCRIPTION("SA1100/SA1111 + UDA1341TS driver for ALSA");
110 MODULE_CLASSES("{sound}");
111 MODULE_DEVICES("{{UDA1341,iPAQ H3600 UDA1341TS}}");
113 static char *id = NULL; /* ID for this card */
115 MODULE_PARM(id, "s");
116 MODULE_PARM_DESC(id, "ID string for SA1100/SA1111 + UDA1341TS soundcard.");
118 #define chip_t sa11xx_uda1341_t
120 typedef struct audio_stream {
121 char *id; /* identification string */
122 int stream_id; /* numeric identification */
123 dma_device_t dma_dev; /* device identifier for DMA */
125 dmach_t dmach; /* dma channel identification */
127 dma_regs_t *dma_regs; /* points to our DMA registers */
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 snd_pcm_substream_t *stream;
138 typedef struct snd_card_sa11xx_uda1341 {
139 struct pm_dev *pm_dev;
141 struct l3_client *uda1341;
144 audio_stream_t s[2]; /* playback & capture */
147 static struct snd_card_sa11xx_uda1341 *sa11xx_uda1341 = NULL;
149 static unsigned int rates[] = {
150 8000, 10666, 10985, 14647,
151 16000, 21970, 22050, 24000,
152 29400, 32000, 44100, 48000,
155 #define RATES sizeof(rates) / sizeof(rates[0])
157 static snd_pcm_hw_constraint_list_t hw_constraints_rates = {
165 /* {{{ Clock and sample rate stuff */
168 * Stop-gap solution until rest of hh.org HAL stuff is merged.
170 #define GPIO_H3600_CLK_SET0 GPIO_GPIO (12)
171 #define GPIO_H3600_CLK_SET1 GPIO_GPIO (13)
173 #ifdef CONFIG_SA1100_H3XXX
174 #define clr_sa11xx_uda1341_egpio(x) clr_h3600_egpio(x)
175 #define set_sa11xx_uda1341_egpio(x) set_h3600_egpio(x)
177 #error This driver could serve H3x00 handhelds only!
180 static void sa11xx_uda1341_set_audio_clock(long val)
183 case 24000: case 32000: case 48000: /* 00: 12.288 MHz */
184 GPCR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1;
187 case 22050: case 29400: case 44100: /* 01: 11.2896 MHz */
188 GPSR = GPIO_H3600_CLK_SET0;
189 GPCR = GPIO_H3600_CLK_SET1;
192 case 8000: case 10666: case 16000: /* 10: 4.096 MHz */
193 GPCR = GPIO_H3600_CLK_SET0;
194 GPSR = GPIO_H3600_CLK_SET1;
197 case 10985: case 14647: case 21970: /* 11: 5.6245 MHz */
198 GPSR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1;
203 static void sa11xx_uda1341_set_samplerate(sa11xx_uda1341_t *sa11xx_uda1341, long rate)
208 /* We don't want to mess with clocks when frames are in flight */
209 Ser4SSCR0 &= ~SSCR0_SSE;
210 /* wait for any frame to complete */
214 * We have the following clock sources:
215 * 4.096 MHz, 5.6245 MHz, 11.2896 MHz, 12.288 MHz
216 * Those can be divided either by 256, 384 or 512.
217 * This makes up 12 combinations for the following samplerates...
221 else if (rate >= 44100)
223 else if (rate >= 32000)
225 else if (rate >= 29400)
227 else if (rate >= 24000)
229 else if (rate >= 22050)
231 else if (rate >= 21970)
233 else if (rate >= 16000)
235 else if (rate >= 14647)
237 else if (rate >= 10985)
239 else if (rate >= 10666)
244 /* Set the external clock generator */
245 #ifdef CONFIG_H3600_HAL
246 h3600_audio_clock(rate);
248 sa11xx_uda1341_set_audio_clock(rate);
251 /* Select the clock divisor */
258 clk_div = SSCR0_SerClkDiv(16);
265 clk_div = SSCR0_SerClkDiv(8);
272 clk_div = SSCR0_SerClkDiv(12);
276 /* FMT setting should be moved away when other FMTs are added (FIXME) */
277 l3_command(sa11xx_uda1341->uda1341, CMD_FORMAT, (void *)LSB16);
279 l3_command(sa11xx_uda1341->uda1341, CMD_FS, (void *)clk);
280 Ser4SSCR0 = (Ser4SSCR0 & ~0xff00) + clk_div + SSCR0_SSE;
281 sa11xx_uda1341->samplerate = rate;
286 /* {{{ HW init and shutdown */
288 static void sa11xx_uda1341_audio_init(sa11xx_uda1341_t *sa11xx_uda1341)
292 /* Setup DMA stuff */
293 sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].id = "UDA1341 out";
294 sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].stream_id = SNDRV_PCM_STREAM_PLAYBACK;
295 sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].dma_dev = DMA_Ser4SSPWr;
297 sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].id = "UDA1341 in";
298 sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].stream_id = SNDRV_PCM_STREAM_CAPTURE;
299 sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].dma_dev = DMA_Ser4SSPRd;
301 /* Initialize the UDA1341 internal state */
303 /* Setup the uarts */
304 local_irq_save(flags);
305 GAFR |= (GPIO_SSP_CLK);
306 GPDR &= ~(GPIO_SSP_CLK);
308 Ser4SSCR0 = SSCR0_DataSize(16) + SSCR0_TI + SSCR0_SerClkDiv(8);
309 Ser4SSCR1 = SSCR1_SClkIactL + SSCR1_SClk1P + SSCR1_ExtClk;
310 Ser4SSCR0 |= SSCR0_SSE;
311 local_irq_restore(flags);
313 /* Enable the audio power */
314 #ifdef CONFIG_H3600_HAL
315 h3600_audio_power(AUDIO_RATE_DEFAULT);
317 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
318 set_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
319 set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
322 /* Wait for the UDA1341 to wake up */
323 mdelay(1); //FIXME - was removed by Perex - Why?
325 /* Initialize the UDA1341 internal state */
326 l3_open(sa11xx_uda1341->uda1341);
328 /* external clock configuration (after l3_open - regs must be initialized */
329 sa11xx_uda1341_set_samplerate(sa11xx_uda1341, sa11xx_uda1341->samplerate);
331 /* Wait for the UDA1341 to wake up */
332 set_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
335 /* make the left and right channels unswapped (flip the WS latch) */
338 #ifdef CONFIG_H3600_HAL
341 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
345 static void sa11xx_uda1341_audio_shutdown(sa11xx_uda1341_t *sa11xx_uda1341)
348 #ifdef CONFIG_H3600_HAL
351 set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
354 /* disable the audio power and all signals leading to the audio chip */
355 l3_close(sa11xx_uda1341->uda1341);
357 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
359 /* power off and mute off */
360 /* FIXME - is muting off necesary??? */
361 #ifdef CONFIG_H3600_HAL
362 h3600_audio_power(0);
365 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
366 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
375 * these are the address and sizes used to fill the xmit buffer
376 * so we can get a clock in record only mode
378 #define FORCE_CLOCK_ADDR (dma_addr_t)FLUSH_BASE_PHYS
379 #define FORCE_CLOCK_SIZE 4096 // was 2048
381 // FIXME Why this value exactly - wrote comment
382 #define DMA_BUF_SIZE 8176 /* <= MAX_DMA_SIZE from asm/arch-sa1100/dma.h */
386 static int audio_dma_request(audio_stream_t *s, void (*callback)(void *, int))
390 ret = sa1100_request_dma(&s->dmach, s->id, s->dma_dev);
392 printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev);
395 sa1100_dma_set_callback(s->dmach, callback);
399 static inline void audio_dma_free(audio_stream_t *s)
401 sa1100_free_dma(s->dmach);
407 static int audio_dma_request(audio_stream_t *s, void (*callback)(void *))
411 ret = sa1100_request_dma(s->dma_dev, s->id, callback, s, &s->dma_regs);
413 printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev);
417 static void audio_dma_free(audio_stream_t *s)
419 sa1100_free_dma((s)->dma_regs);
425 static u_int audio_get_dma_pos(audio_stream_t *s)
427 snd_pcm_substream_t * substream = s->stream;
428 snd_pcm_runtime_t *runtime = substream->runtime;
433 // this must be called w/ interrupts locked out see dma-sa1100.c in the kernel
434 spin_lock_irqsave(&s->dma_lock, flags);
436 sa1100_dma_get_current(s->dmach, NULL, &addr);
438 addr = sa1100_get_dma_pos((s)->dma_regs);
440 offset = addr - runtime->dma_addr;
441 spin_unlock_irqrestore(&s->dma_lock, flags);
443 offset = bytes_to_frames(runtime,offset);
444 if (offset >= runtime->buffer_size)
451 * this stops the dma and clears the dma ptrs
453 static void audio_stop_dma(audio_stream_t *s)
457 spin_lock_irqsave(&s->dma_lock, flags);
460 /* this stops the dma channel and clears the buffer ptrs */
462 sa1100_dma_flush_all(s->dmach);
464 sa1100_clear_dma(s->dma_regs);
466 spin_unlock_irqrestore(&s->dma_lock, flags);
469 static void audio_process_dma(audio_stream_t *s)
471 snd_pcm_substream_t *substream = s->stream;
472 snd_pcm_runtime_t *runtime;
473 unsigned int dma_size;
477 /* we are requested to process synchronization DMA transfer */
479 snd_assert(s->stream_id == SNDRV_PCM_STREAM_PLAYBACK, return);
480 /* fill the xmit dma buffers and return */
482 sa1100_dma_set_spin(s->dmach, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE);
485 ret = sa1100_start_dma(s->dma_regs, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE);
493 /* must be set here - only valid for running streams, not for forced_clock dma fills */
494 runtime = substream->runtime;
495 while (s->active && s->periods < runtime->periods) {
496 dma_size = frames_to_bytes(runtime, runtime->period_size);
498 /* a little trick, we need resume from old position */
499 offset = frames_to_bytes(runtime, s->old_offset - 1);
502 s->period = offset / dma_size;
504 dma_size = dma_size - offset;
506 continue; /* special case */
508 offset = dma_size * s->period;
509 snd_assert(dma_size <= DMA_BUF_SIZE, );
512 ret = sa1100_dma_queue_buffer(s->dmach, s, runtime->dma_addr + offset, dma_size);
516 ret = sa1100_start_dma((s)->dma_regs, runtime->dma_addr + offset, dma_size);
518 printk(KERN_ERR "audio_process_dma: cannot queue DMA buffer (%i)\n", ret);
524 s->period %= runtime->periods;
530 static void audio_dma_callback(void *data, int size)
532 static void audio_dma_callback(void *data)
535 audio_stream_t *s = data;
538 * If we are getting a callback for an active stream then we inform
539 * the PCM middle layer we've finished a period
542 snd_pcm_period_elapsed(s->stream);
544 spin_lock(&s->dma_lock);
545 if (!s->tx_spin && s->periods > 0)
547 audio_process_dma(s);
548 spin_unlock(&s->dma_lock);
553 /* {{{ PCM setting */
555 /* {{{ trigger & timer */
557 static int snd_sa11xx_uda1341_trigger(snd_pcm_substream_t * substream, int cmd)
559 sa11xx_uda1341_t *chip = snd_pcm_substream_chip(substream);
560 int stream_id = substream->pstr->stream;
561 audio_stream_t *s = &chip->s[stream_id];
562 audio_stream_t *s1 = &chip->s[stream_id ^ 1];
565 /* note local interrupts are already disabled in the midlevel code */
566 spin_lock(&s->dma_lock);
568 case SNDRV_PCM_TRIGGER_START:
569 /* now we need to make sure a record only stream has a clock */
570 if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
571 /* we need to force fill the xmit DMA with zeros */
573 audio_process_dma(s1);
575 /* this case is when you were recording then you turn on a
576 * playback stream so we stop (also clears it) the dma first,
577 * clear the sync flag and then we let it turned on
583 /* requested stream startup */
585 audio_process_dma(s);
587 case SNDRV_PCM_TRIGGER_STOP:
588 /* requested stream shutdown */
592 * now we need to make sure a record only stream has a clock
593 * so if we're stopping a playback with an active capture
594 * we need to turn the 0 fill dma on for the xmit side
596 if (stream_id == SNDRV_PCM_STREAM_PLAYBACK && s1->active) {
597 /* we need to force fill the xmit DMA with zeros */
599 audio_process_dma(s);
602 * we killed a capture only stream, so we should also kill
603 * the zero fill transmit
613 case SNDRV_PCM_TRIGGER_SUSPEND:
616 sa1100_dma_stop(s->dmach);
620 s->old_offset = audio_get_dma_pos(s) + 1;
622 sa1100_dma_flush_all(s->dmach);
628 case SNDRV_PCM_TRIGGER_RESUME:
631 audio_process_dma(s);
632 if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
634 audio_process_dma(s1);
637 case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
639 sa1100_dma_stop(s->dmach);
644 if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) {
647 s->old_offset = audio_get_dma_pos(s) + 1;
649 sa1100_dma_flush_all(s->dmach);
653 audio_process_dma(s);
659 sa1100_dma_flush_all(s1->dmach);
666 case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
670 audio_process_dma(s);
673 if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
675 audio_process_dma(s1);
678 sa1100_dma_resume(s->dmach);
687 spin_unlock(&s->dma_lock);
691 static int snd_sa11xx_uda1341_prepare(snd_pcm_substream_t * substream)
693 sa11xx_uda1341_t *chip = snd_pcm_substream_chip(substream);
694 snd_pcm_runtime_t *runtime = substream->runtime;
695 audio_stream_t *s = &chip->s[substream->pstr->stream];
697 /* set requested samplerate */
698 sa11xx_uda1341_set_samplerate(chip, runtime->rate);
700 /* set requestd format when available */
701 /* set FMT here !!! FIXME */
709 static snd_pcm_uframes_t snd_sa11xx_uda1341_pointer(snd_pcm_substream_t * substream)
711 sa11xx_uda1341_t *chip = snd_pcm_substream_chip(substream);
712 return audio_get_dma_pos(&chip->s[substream->pstr->stream]);
717 static snd_pcm_hardware_t snd_sa11xx_uda1341_capture =
719 .info = (SNDRV_PCM_INFO_INTERLEAVED |
720 SNDRV_PCM_INFO_BLOCK_TRANSFER |
721 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
722 SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
723 .formats = SNDRV_PCM_FMTBIT_S16_LE,
724 .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\
725 SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\
726 SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
727 SNDRV_PCM_RATE_KNOT),
732 .buffer_bytes_max = 64*1024,
733 .period_bytes_min = 64,
734 .period_bytes_max = DMA_BUF_SIZE,
740 static snd_pcm_hardware_t snd_sa11xx_uda1341_playback =
742 .info = (SNDRV_PCM_INFO_INTERLEAVED |
743 SNDRV_PCM_INFO_BLOCK_TRANSFER |
744 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
745 SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
746 .formats = SNDRV_PCM_FMTBIT_S16_LE,
747 .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\
748 SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\
749 SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
750 SNDRV_PCM_RATE_KNOT),
755 .buffer_bytes_max = 64*1024,
756 .period_bytes_min = 64,
757 .period_bytes_max = DMA_BUF_SIZE,
763 static int snd_card_sa11xx_uda1341_open(snd_pcm_substream_t * substream)
765 sa11xx_uda1341_t *chip = snd_pcm_substream_chip(substream);
766 snd_pcm_runtime_t *runtime = substream->runtime;
767 int stream_id = substream->pstr->stream;
770 chip->s[stream_id].stream = substream;
772 if (stream_id == SNDRV_PCM_STREAM_PLAYBACK)
773 runtime->hw = snd_sa11xx_uda1341_playback;
775 runtime->hw = snd_sa11xx_uda1341_capture;
776 if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
778 if ((err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates)) < 0)
784 static int snd_card_sa11xx_uda1341_close(snd_pcm_substream_t * substream)
786 sa11xx_uda1341_t *chip = snd_pcm_substream_chip(substream);
788 chip->s[substream->pstr->stream].stream = NULL;
792 /* {{{ HW params & free */
794 static int snd_sa11xx_uda1341_hw_params(snd_pcm_substream_t * substream,
795 snd_pcm_hw_params_t * hw_params)
798 return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
801 static int snd_sa11xx_uda1341_hw_free(snd_pcm_substream_t * substream)
803 return snd_pcm_lib_free_pages(substream);
808 static snd_pcm_ops_t snd_card_sa11xx_uda1341_playback_ops = {
809 .open = snd_card_sa11xx_uda1341_open,
810 .close = snd_card_sa11xx_uda1341_close,
811 .ioctl = snd_pcm_lib_ioctl,
812 .hw_params = snd_sa11xx_uda1341_hw_params,
813 .hw_free = snd_sa11xx_uda1341_hw_free,
814 .prepare = snd_sa11xx_uda1341_prepare,
815 .trigger = snd_sa11xx_uda1341_trigger,
816 .pointer = snd_sa11xx_uda1341_pointer,
819 static snd_pcm_ops_t snd_card_sa11xx_uda1341_capture_ops = {
820 .open = snd_card_sa11xx_uda1341_open,
821 .close = snd_card_sa11xx_uda1341_close,
822 .ioctl = snd_pcm_lib_ioctl,
823 .hw_params = snd_sa11xx_uda1341_hw_params,
824 .hw_free = snd_sa11xx_uda1341_hw_free,
825 .prepare = snd_sa11xx_uda1341_prepare,
826 .trigger = snd_sa11xx_uda1341_trigger,
827 .pointer = snd_sa11xx_uda1341_pointer,
830 static int __init snd_card_sa11xx_uda1341_pcm(sa11xx_uda1341_t *sa11xx_uda1341, int device)
835 if ((err = snd_pcm_new(sa11xx_uda1341->card, "UDA1341 PCM", device, 1, 1, &pcm)) < 0)
839 * this sets up our initial buffers and sets the dma_type to isa.
840 * isa works but I'm not sure why (or if) it's the right choice
841 * this may be too large, trying it for now
843 snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_ISA,
844 snd_pcm_dma_flags(0),
847 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_card_sa11xx_uda1341_playback_ops);
848 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_card_sa11xx_uda1341_capture_ops);
849 pcm->private_data = sa11xx_uda1341;
851 strcpy(pcm->name, "UDA1341 PCM");
853 sa11xx_uda1341_audio_init(sa11xx_uda1341);
855 /* setup DMA controller */
856 audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK], audio_dma_callback);
857 audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE], audio_dma_callback);
859 sa11xx_uda1341->pcm = pcm;
866 /* {{{ module init & exit */
870 static void snd_sa11xx_uda1341_suspend(sa11xx_uda1341_t *chip)
872 snd_card_t *card = chip->card;
874 if (card->power_state == SNDRV_CTL_POWER_D3hot)
876 snd_pcm_suspend_all(chip->pcm);
878 sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach);
879 sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach);
883 l3_command(chip->uda1341, CMD_SUSPEND, NULL);
884 sa11xx_uda1341_audio_shutdown(chip);
885 snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
888 static void snd_sa11xx_uda1341_resume(sa11xx_uda1341_t *chip)
890 snd_card_t *card = chip->card;
892 if (card->power_state == SNDRV_CTL_POWER_D0)
894 sa11xx_uda1341_audio_init(chip);
895 l3_command(chip->uda1341, CMD_RESUME, NULL);
897 sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach);
898 sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach);
902 snd_power_change_state(card, SNDRV_CTL_POWER_D0);
905 static int sa11xx_uda1341_pm_callback(struct pm_dev *pm_dev, pm_request_t req, void *data)
907 sa11xx_uda1341_t *chip = pm_dev->data;
910 case PM_SUSPEND: /* enter D1-D3 */
911 snd_sa11xx_uda1341_suspend(chip);
913 case PM_RESUME: /* enter D0 */
914 snd_sa11xx_uda1341_resume(chip);
920 static int sa11xx_uda1341_set_power_state(snd_card_t *card, unsigned int power_state)
922 sa11xx_uda1341_t *chip = snd_magic_cast(sa11xx_uda1341_t, card->power_state_private_data, return);
924 switch (power_state) {
925 case SNDRV_CTL_POWER_D0:
926 case SNDRV_CTL_POWER_D1:
927 case SNDRV_CTL_POWER_D2:
928 snd_sa11xx_uda1341_resume(chip);
930 case SNDRV_CTL_POWER_D3hot:
931 case SNDRV_CTL_POWER_D3cold:
932 snd_sa11xx_uda1341_suspend(chip);
940 #endif /* COMFIG_PM */
942 void snd_sa11xx_uda1341_free(snd_card_t *card)
944 sa11xx_uda1341_t *chip = snd_magic_cast(sa11xx_uda1341_t, card->private_data, return);
946 pm_unregister(chip->pm_dev);
947 audio_dma_free(&chip->s[SNDRV_PCM_STREAM_PLAYBACK]);
948 audio_dma_free(&chip->s[SNDRV_PCM_STREAM_CAPTURE]);
949 sa11xx_uda1341 = NULL;
950 card->private_data = NULL;
954 static int __init sa11xx_uda1341_init(void)
959 if (!machine_is_h3xxx())
962 /* register the soundcard */
963 card = snd_card_new(-1, id, THIS_MODULE, sizeof(sa11xx_uda1341_t));
967 sa11xx_uda1341 = snd_magic_kcalloc(sa11xx_uda1341_t, 0, GFP_KERNEL);
968 if (sa11xx_uda1341 == NULL)
970 spin_lock_init(&chip->s[0].dma_lock);
971 spin_lock_init(&chip->s[1].dma_lock);
973 card->private_data = (void *)sa11xx_uda1341;
974 card->private_free = snd_sa11xx_uda1341_free;
976 sa11xx_uda1341->card = card;
977 sa11xx_uda1341->samplerate = AUDIO_RATE_DEFAULT;
980 if ((err = snd_chip_uda1341_mixer_new(sa11xx_uda1341->card, &sa11xx_uda1341->uda1341)))
984 if ((err = snd_card_sa11xx_uda1341_pcm(sa11xx_uda1341, 0)) < 0)
989 card->power_state_private_data = sa11xx_uda1341;
990 card->set_power_state = sa11xx_uda1341_set_power_state;
991 sa11xx_uda1341->pm_dev = pm_register(PM_SYS_DEV, 0, sa11xx_uda1341_pm_callback);
992 if (sa11xx_uda1341->pm_dev)
993 sa11xx_uda1341->pm_dev->data = sa11xx_uda1341;
996 strcpy(card->driver, "UDA1341");
997 strcpy(card->shortname, "H3600 UDA1341TS");
998 sprintf(card->longname, "Compaq iPAQ H3600 with Philips UDA1341TS");
1000 if ((err = snd_card_register(card)) == 0) {
1001 printk( KERN_INFO "iPAQ audio support initialized\n" );
1006 snd_card_free(card);
1010 static void __exit sa11xx_uda1341_exit(void)
1012 snd_card_free(sa11xx_uda1341->card);
1015 module_init(sa11xx_uda1341_init);
1016 module_exit(sa11xx_uda1341_exit);
1022 * indent-tabs-mode: t