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 <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>
78 #include <asm/hardware.h>
79 #include <asm/arch/h3600.h>
80 #include <asm/mach-types.h>
83 #ifdef CONFIG_H3600_HAL
84 #include <asm/semaphore.h>
85 #include <asm/uaccess.h>
86 #include <asm/arch/h3600_hal.h>
89 #include <sound/core.h>
90 #include <sound/pcm.h>
91 #include <sound/initval.h>
93 #include <linux/l3/l3.h>
96 #undef DEBUG_FUNCTION_NAMES
97 #include <sound/uda1341.h>
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
104 #ifdef CONFIG_H3600_HAL
108 /* {{{ Type definitions */
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}}");
115 static char *id; /* ID for this card */
117 module_param(id, charp, 0444);
118 MODULE_PARM_DESC(id, "ID string for SA1100/SA1111 + UDA1341TS soundcard.");
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 */
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 struct snd_pcm_substream *stream;
138 struct sa11xx_uda1341 {
139 struct snd_card *card;
140 struct l3_client *uda1341;
143 struct audio_stream s[2]; /* playback & capture */
146 static unsigned int rates[] = {
147 8000, 10666, 10985, 14647,
148 16000, 21970, 22050, 24000,
149 29400, 32000, 44100, 48000,
152 static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
153 .count = ARRAY_SIZE(rates),
158 static struct platform_device *device;
162 /* {{{ Clock and sample rate stuff */
165 * Stop-gap solution until rest of hh.org HAL stuff is merged.
167 #define GPIO_H3600_CLK_SET0 GPIO_GPIO (12)
168 #define GPIO_H3600_CLK_SET1 GPIO_GPIO (13)
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)
174 #error This driver could serve H3x00 handhelds only!
177 static void sa11xx_uda1341_set_audio_clock(long val)
180 case 24000: case 32000: case 48000: /* 00: 12.288 MHz */
181 GPCR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1;
184 case 22050: case 29400: case 44100: /* 01: 11.2896 MHz */
185 GPSR = GPIO_H3600_CLK_SET0;
186 GPCR = GPIO_H3600_CLK_SET1;
189 case 8000: case 10666: case 16000: /* 10: 4.096 MHz */
190 GPCR = GPIO_H3600_CLK_SET0;
191 GPSR = GPIO_H3600_CLK_SET1;
194 case 10985: case 14647: case 21970: /* 11: 5.6245 MHz */
195 GPSR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1;
200 static void sa11xx_uda1341_set_samplerate(struct sa11xx_uda1341 *sa11xx_uda1341, long rate)
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 */
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...
218 else if (rate >= 44100)
220 else if (rate >= 32000)
222 else if (rate >= 29400)
224 else if (rate >= 24000)
226 else if (rate >= 22050)
228 else if (rate >= 21970)
230 else if (rate >= 16000)
232 else if (rate >= 14647)
234 else if (rate >= 10985)
236 else if (rate >= 10666)
241 /* Set the external clock generator */
242 #ifdef CONFIG_H3600_HAL
243 h3600_audio_clock(rate);
245 sa11xx_uda1341_set_audio_clock(rate);
248 /* Select the clock divisor */
255 clk_div = SSCR0_SerClkDiv(16);
262 clk_div = SSCR0_SerClkDiv(8);
269 clk_div = SSCR0_SerClkDiv(12);
273 /* FMT setting should be moved away when other FMTs are added (FIXME) */
274 l3_command(sa11xx_uda1341->uda1341, CMD_FORMAT, (void *)LSB16);
276 l3_command(sa11xx_uda1341->uda1341, CMD_FS, (void *)clk);
277 Ser4SSCR0 = (Ser4SSCR0 & ~0xff00) + clk_div + SSCR0_SSE;
278 sa11xx_uda1341->samplerate = rate;
283 /* {{{ HW init and shutdown */
285 static void sa11xx_uda1341_audio_init(struct sa11xx_uda1341 *sa11xx_uda1341)
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;
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;
298 /* Initialize the UDA1341 internal state */
300 /* Setup the uarts */
301 local_irq_save(flags);
302 GAFR |= (GPIO_SSP_CLK);
303 GPDR &= ~(GPIO_SSP_CLK);
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);
310 /* Enable the audio power */
311 #ifdef CONFIG_H3600_HAL
312 h3600_audio_power(AUDIO_RATE_DEFAULT);
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);
319 /* Wait for the UDA1341 to wake up */
320 mdelay(1); //FIXME - was removed by Perex - Why?
322 /* Initialize the UDA1341 internal state */
323 l3_open(sa11xx_uda1341->uda1341);
325 /* external clock configuration (after l3_open - regs must be initialized */
326 sa11xx_uda1341_set_samplerate(sa11xx_uda1341, sa11xx_uda1341->samplerate);
328 /* Wait for the UDA1341 to wake up */
329 set_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
332 /* make the left and right channels unswapped (flip the WS latch) */
335 #ifdef CONFIG_H3600_HAL
338 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
342 static void sa11xx_uda1341_audio_shutdown(struct sa11xx_uda1341 *sa11xx_uda1341)
345 #ifdef CONFIG_H3600_HAL
348 set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
351 /* disable the audio power and all signals leading to the audio chip */
352 l3_close(sa11xx_uda1341->uda1341);
354 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
356 /* power off and mute off */
357 /* FIXME - is muting off necesary??? */
358 #ifdef CONFIG_H3600_HAL
359 h3600_audio_power(0);
362 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
363 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
372 * these are the address and sizes used to fill the xmit buffer
373 * so we can get a clock in record only mode
375 #define FORCE_CLOCK_ADDR (dma_addr_t)FLUSH_BASE_PHYS
376 #define FORCE_CLOCK_SIZE 4096 // was 2048
378 // FIXME Why this value exactly - wrote comment
379 #define DMA_BUF_SIZE 8176 /* <= MAX_DMA_SIZE from asm/arch-sa1100/dma.h */
383 static int audio_dma_request(struct audio_stream *s, void (*callback)(void *, int))
387 ret = sa1100_request_dma(&s->dmach, s->id, s->dma_dev);
389 printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev);
392 sa1100_dma_set_callback(s->dmach, callback);
396 static inline void audio_dma_free(struct audio_stream *s)
398 sa1100_free_dma(s->dmach);
404 static int audio_dma_request(struct audio_stream *s, void (*callback)(void *))
408 ret = sa1100_request_dma(s->dma_dev, s->id, callback, s, &s->dma_regs);
410 printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev);
414 static void audio_dma_free(struct audio_stream *s)
416 sa1100_free_dma(s->dma_regs);
422 static u_int audio_get_dma_pos(struct audio_stream *s)
424 struct snd_pcm_substream *substream = s->stream;
425 struct snd_pcm_runtime *runtime = substream->runtime;
430 // this must be called w/ interrupts locked out see dma-sa1100.c in the kernel
431 spin_lock_irqsave(&s->dma_lock, flags);
433 sa1100_dma_get_current(s->dmach, NULL, &addr);
435 addr = sa1100_get_dma_pos((s)->dma_regs);
437 offset = addr - runtime->dma_addr;
438 spin_unlock_irqrestore(&s->dma_lock, flags);
440 offset = bytes_to_frames(runtime,offset);
441 if (offset >= runtime->buffer_size)
448 * this stops the dma and clears the dma ptrs
450 static void audio_stop_dma(struct audio_stream *s)
454 spin_lock_irqsave(&s->dma_lock, flags);
457 /* this stops the dma channel and clears the buffer ptrs */
459 sa1100_dma_flush_all(s->dmach);
461 sa1100_clear_dma(s->dma_regs);
463 spin_unlock_irqrestore(&s->dma_lock, flags);
466 static void audio_process_dma(struct audio_stream *s)
468 struct snd_pcm_substream *substream = s->stream;
469 struct snd_pcm_runtime *runtime;
470 unsigned int dma_size;
474 /* we are requested to process synchronization DMA transfer */
476 snd_assert(s->stream_id == SNDRV_PCM_STREAM_PLAYBACK, return);
477 /* fill the xmit dma buffers and return */
479 sa1100_dma_set_spin(s->dmach, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE);
482 ret = sa1100_start_dma(s->dma_regs, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE);
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);
495 /* a little trick, we need resume from old position */
496 offset = frames_to_bytes(runtime, s->old_offset - 1);
499 s->period = offset / dma_size;
501 dma_size = dma_size - offset;
503 continue; /* special case */
505 offset = dma_size * s->period;
506 snd_assert(dma_size <= DMA_BUF_SIZE, );
509 ret = sa1100_dma_queue_buffer(s->dmach, s, runtime->dma_addr + offset, dma_size);
513 ret = sa1100_start_dma((s)->dma_regs, runtime->dma_addr + offset, dma_size);
515 printk(KERN_ERR "audio_process_dma: cannot queue DMA buffer (%i)\n", ret);
521 s->period %= runtime->periods;
527 static void audio_dma_callback(void *data, int size)
529 static void audio_dma_callback(void *data)
532 struct audio_stream *s = data;
535 * If we are getting a callback for an active stream then we inform
536 * the PCM middle layer we've finished a period
539 snd_pcm_period_elapsed(s->stream);
541 spin_lock(&s->dma_lock);
542 if (!s->tx_spin && s->periods > 0)
544 audio_process_dma(s);
545 spin_unlock(&s->dma_lock);
550 /* {{{ PCM setting */
552 /* {{{ trigger & timer */
554 static int snd_sa11xx_uda1341_trigger(struct snd_pcm_substream *substream, int cmd)
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];
562 /* note local interrupts are already disabled in the midlevel code */
563 spin_lock(&s->dma_lock);
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 */
570 audio_process_dma(s1);
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
580 /* requested stream startup */
582 audio_process_dma(s);
584 case SNDRV_PCM_TRIGGER_STOP:
585 /* requested stream shutdown */
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
593 if (stream_id == SNDRV_PCM_STREAM_PLAYBACK && s1->active) {
594 /* we need to force fill the xmit DMA with zeros */
596 audio_process_dma(s);
599 * we killed a capture only stream, so we should also kill
600 * the zero fill transmit
610 case SNDRV_PCM_TRIGGER_SUSPEND:
613 sa1100_dma_stop(s->dmach);
617 s->old_offset = audio_get_dma_pos(s) + 1;
619 sa1100_dma_flush_all(s->dmach);
625 case SNDRV_PCM_TRIGGER_RESUME:
628 audio_process_dma(s);
629 if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
631 audio_process_dma(s1);
634 case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
636 sa1100_dma_stop(s->dmach);
641 if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) {
644 s->old_offset = audio_get_dma_pos(s) + 1;
646 sa1100_dma_flush_all(s->dmach);
650 audio_process_dma(s);
656 sa1100_dma_flush_all(s1->dmach);
663 case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
667 audio_process_dma(s);
670 if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
672 audio_process_dma(s1);
675 sa1100_dma_resume(s->dmach);
684 spin_unlock(&s->dma_lock);
688 static int snd_sa11xx_uda1341_prepare(struct snd_pcm_substream *substream)
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];
694 /* set requested samplerate */
695 sa11xx_uda1341_set_samplerate(chip, runtime->rate);
697 /* set requestd format when available */
698 /* set FMT here !!! FIXME */
706 static snd_pcm_uframes_t snd_sa11xx_uda1341_pointer(struct snd_pcm_substream *substream)
708 struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
709 return audio_get_dma_pos(&chip->s[substream->pstr->stream]);
714 static struct snd_pcm_hardware snd_sa11xx_uda1341_capture =
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),
729 .buffer_bytes_max = 64*1024,
730 .period_bytes_min = 64,
731 .period_bytes_max = DMA_BUF_SIZE,
737 static struct snd_pcm_hardware snd_sa11xx_uda1341_playback =
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),
752 .buffer_bytes_max = 64*1024,
753 .period_bytes_min = 64,
754 .period_bytes_max = DMA_BUF_SIZE,
760 static int snd_card_sa11xx_uda1341_open(struct snd_pcm_substream *substream)
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;
767 chip->s[stream_id].stream = substream;
769 if (stream_id == SNDRV_PCM_STREAM_PLAYBACK)
770 runtime->hw = snd_sa11xx_uda1341_playback;
772 runtime->hw = snd_sa11xx_uda1341_capture;
773 if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
775 if ((err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates)) < 0)
781 static int snd_card_sa11xx_uda1341_close(struct snd_pcm_substream *substream)
783 struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
785 chip->s[substream->pstr->stream].stream = NULL;
789 /* {{{ HW params & free */
791 static int snd_sa11xx_uda1341_hw_params(struct snd_pcm_substream *substream,
792 struct snd_pcm_hw_params *hw_params)
795 return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
798 static int snd_sa11xx_uda1341_hw_free(struct snd_pcm_substream *substream)
800 return snd_pcm_lib_free_pages(substream);
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,
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,
827 static int __init snd_card_sa11xx_uda1341_pcm(struct sa11xx_uda1341 *sa11xx_uda1341, int device)
832 if ((err = snd_pcm_new(sa11xx_uda1341->card, "UDA1341 PCM", device, 1, 1, &pcm)) < 0)
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
840 snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
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;
848 strcpy(pcm->name, "UDA1341 PCM");
850 sa11xx_uda1341_audio_init(sa11xx_uda1341);
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);
856 sa11xx_uda1341->pcm = pcm;
863 /* {{{ module init & exit */
867 static int snd_sa11xx_uda1341_suspend(struct platform_device *devptr,
870 struct snd_card *card = platform_get_drvdata(devptr);
871 struct sa11xx_uda1341 *chip = card->private_data;
873 snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
874 snd_pcm_suspend_all(chip->pcm);
876 sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach);
877 sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach);
881 l3_command(chip->uda1341, CMD_SUSPEND, NULL);
882 sa11xx_uda1341_audio_shutdown(chip);
887 static int snd_sa11xx_uda1341_resume(struct platform_device *devptr)
889 struct snd_card *card = platform_get_drvdata(devptr);
890 struct sa11xx_uda1341 *chip = card->private_data;
892 sa11xx_uda1341_audio_init(chip);
893 l3_command(chip->uda1341, CMD_RESUME, NULL);
895 sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach);
896 sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach);
900 snd_power_change_state(card, SNDRV_CTL_POWER_D0);
903 #endif /* COMFIG_PM */
905 void snd_sa11xx_uda1341_free(struct snd_card *card)
907 struct sa11xx_uda1341 *chip = card->private_data;
909 audio_dma_free(&chip->s[SNDRV_PCM_STREAM_PLAYBACK]);
910 audio_dma_free(&chip->s[SNDRV_PCM_STREAM_CAPTURE]);
913 static int __init sa11xx_uda1341_probe(struct platform_device *devptr)
916 struct snd_card *card;
917 struct sa11xx_uda1341 *chip;
919 /* register the soundcard */
920 card = snd_card_new(-1, id, THIS_MODULE, sizeof(struct sa11xx_uda1341));
924 chip = card->private_data;
925 spin_lock_init(&chip->s[0].dma_lock);
926 spin_lock_init(&chip->s[1].dma_lock);
928 card->private_free = snd_sa11xx_uda1341_free;
930 chip->samplerate = AUDIO_RATE_DEFAULT;
933 if ((err = snd_chip_uda1341_mixer_new(card, &chip->uda1341)))
937 if ((err = snd_card_sa11xx_uda1341_pcm(chip, 0)) < 0)
940 strcpy(card->driver, "UDA1341");
941 strcpy(card->shortname, "H3600 UDA1341TS");
942 sprintf(card->longname, "Compaq iPAQ H3600 with Philips UDA1341TS");
944 snd_card_set_dev(card, &devptr->dev);
946 if ((err = snd_card_register(card)) == 0) {
947 printk( KERN_INFO "iPAQ audio support initialized\n" );
948 platform_set_drvdata(devptr, card);
957 static int __devexit sa11xx_uda1341_remove(struct platform_device *devptr)
959 snd_card_free(platform_get_drvdata(devptr));
960 platform_set_drvdata(devptr, NULL);
964 #define SA11XX_UDA1341_DRIVER "sa11xx_uda1341"
966 static struct platform_driver sa11xx_uda1341_driver = {
967 .probe = sa11xx_uda1341_probe,
968 .remove = __devexit_p(sa11xx_uda1341_remove),
970 .suspend = snd_sa11xx_uda1341_suspend,
971 .resume = snd_sa11xx_uda1341_resume,
974 .name = SA11XX_UDA1341_DRIVER,
978 static int __init sa11xx_uda1341_init(void)
982 if (!machine_is_h3xxx())
984 if ((err = platform_driver_register(&sa11xx_uda1341_driver)) < 0)
986 device = platform_device_register_simple(SA11XX_UDA1341_DRIVER, -1, NULL, 0);
987 if (!IS_ERR(device)) {
988 if (platform_get_drvdata(device))
990 platform_device_unregister(device);
993 err = PTR_ERR(device);
994 platform_driver_unregister(&sa11xx_uda1341_driver);
998 static void __exit sa11xx_uda1341_exit(void)
1000 platform_device_unregister(device);
1001 platform_driver_unregister(&sa11xx_uda1341_driver);
1004 module_init(sa11xx_uda1341_init);
1005 module_exit(sa11xx_uda1341_exit);
1011 * indent-tabs-mode: t