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 if (snd_BUG_ON(s
->stream_id
!= SNDRV_PCM_STREAM_PLAYBACK
))
447 /* fill the xmit dma buffers and return */
449 sa1100_dma_set_spin(s
->dmach
, FORCE_CLOCK_ADDR
, FORCE_CLOCK_SIZE
);
452 ret
= sa1100_start_dma(s
->dma_regs
, FORCE_CLOCK_ADDR
, FORCE_CLOCK_SIZE
);
460 /* must be set here - only valid for running streams, not for forced_clock dma fills */
461 runtime
= substream
->runtime
;
462 while (s
->active
&& s
->periods
< runtime
->periods
) {
463 dma_size
= frames_to_bytes(runtime
, runtime
->period_size
);
465 /* a little trick, we need resume from old position */
466 offset
= frames_to_bytes(runtime
, s
->old_offset
- 1);
469 s
->period
= offset
/ dma_size
;
471 dma_size
= dma_size
- offset
;
473 continue; /* special case */
475 offset
= dma_size
* s
->period
;
476 snd_BUG_ON(dma_size
> DMA_BUF_SIZE
);
479 ret
= sa1100_dma_queue_buffer(s
->dmach
, s
, runtime
->dma_addr
+ offset
, dma_size
);
483 ret
= sa1100_start_dma((s
)->dma_regs
, runtime
->dma_addr
+ offset
, dma_size
);
485 printk(KERN_ERR
"audio_process_dma: cannot queue DMA buffer (%i)\n", ret
);
491 s
->period
%= runtime
->periods
;
497 static void audio_dma_callback(void *data
, int size
)
499 static void audio_dma_callback(void *data
)
502 struct audio_stream
*s
= data
;
505 * If we are getting a callback for an active stream then we inform
506 * the PCM middle layer we've finished a period
509 snd_pcm_period_elapsed(s
->stream
);
511 spin_lock(&s
->dma_lock
);
512 if (!s
->tx_spin
&& s
->periods
> 0)
514 audio_process_dma(s
);
515 spin_unlock(&s
->dma_lock
);
520 /* {{{ PCM setting */
522 /* {{{ trigger & timer */
524 static int snd_sa11xx_uda1341_trigger(struct snd_pcm_substream
*substream
, int cmd
)
526 struct sa11xx_uda1341
*chip
= snd_pcm_substream_chip(substream
);
527 int stream_id
= substream
->pstr
->stream
;
528 struct audio_stream
*s
= &chip
->s
[stream_id
];
529 struct audio_stream
*s1
= &chip
->s
[stream_id
^ 1];
532 /* note local interrupts are already disabled in the midlevel code */
533 spin_lock(&s
->dma_lock
);
535 case SNDRV_PCM_TRIGGER_START
:
536 /* now we need to make sure a record only stream has a clock */
537 if (stream_id
== SNDRV_PCM_STREAM_CAPTURE
&& !s1
->active
) {
538 /* we need to force fill the xmit DMA with zeros */
540 audio_process_dma(s1
);
542 /* this case is when you were recording then you turn on a
543 * playback stream so we stop (also clears it) the dma first,
544 * clear the sync flag and then we let it turned on
550 /* requested stream startup */
552 audio_process_dma(s
);
554 case SNDRV_PCM_TRIGGER_STOP
:
555 /* requested stream shutdown */
559 * now we need to make sure a record only stream has a clock
560 * so if we're stopping a playback with an active capture
561 * we need to turn the 0 fill dma on for the xmit side
563 if (stream_id
== SNDRV_PCM_STREAM_PLAYBACK
&& s1
->active
) {
564 /* we need to force fill the xmit DMA with zeros */
566 audio_process_dma(s
);
569 * we killed a capture only stream, so we should also kill
570 * the zero fill transmit
580 case SNDRV_PCM_TRIGGER_SUSPEND
:
583 sa1100_dma_stop(s
->dmach
);
587 s
->old_offset
= audio_get_dma_pos(s
) + 1;
589 sa1100_dma_flush_all(s
->dmach
);
595 case SNDRV_PCM_TRIGGER_RESUME
:
598 audio_process_dma(s
);
599 if (stream_id
== SNDRV_PCM_STREAM_CAPTURE
&& !s1
->active
) {
601 audio_process_dma(s1
);
604 case SNDRV_PCM_TRIGGER_PAUSE_PUSH
:
606 sa1100_dma_stop(s
->dmach
);
611 if (stream_id
== SNDRV_PCM_STREAM_PLAYBACK
) {
614 s
->old_offset
= audio_get_dma_pos(s
) + 1;
616 sa1100_dma_flush_all(s
->dmach
);
620 audio_process_dma(s
);
626 sa1100_dma_flush_all(s1
->dmach
);
633 case SNDRV_PCM_TRIGGER_PAUSE_RELEASE
:
637 audio_process_dma(s
);
640 if (stream_id
== SNDRV_PCM_STREAM_CAPTURE
&& !s1
->active
) {
642 audio_process_dma(s1
);
645 sa1100_dma_resume(s
->dmach
);
654 spin_unlock(&s
->dma_lock
);
658 static int snd_sa11xx_uda1341_prepare(struct snd_pcm_substream
*substream
)
660 struct sa11xx_uda1341
*chip
= snd_pcm_substream_chip(substream
);
661 struct snd_pcm_runtime
*runtime
= substream
->runtime
;
662 struct audio_stream
*s
= &chip
->s
[substream
->pstr
->stream
];
664 /* set requested samplerate */
665 sa11xx_uda1341_set_samplerate(chip
, runtime
->rate
);
667 /* set requestd format when available */
668 /* set FMT here !!! FIXME */
676 static snd_pcm_uframes_t
snd_sa11xx_uda1341_pointer(struct snd_pcm_substream
*substream
)
678 struct sa11xx_uda1341
*chip
= snd_pcm_substream_chip(substream
);
679 return audio_get_dma_pos(&chip
->s
[substream
->pstr
->stream
]);
684 static struct snd_pcm_hardware snd_sa11xx_uda1341_capture
=
686 .info
= (SNDRV_PCM_INFO_INTERLEAVED
|
687 SNDRV_PCM_INFO_BLOCK_TRANSFER
|
688 SNDRV_PCM_INFO_MMAP
| SNDRV_PCM_INFO_MMAP_VALID
|
689 SNDRV_PCM_INFO_PAUSE
| SNDRV_PCM_INFO_RESUME
),
690 .formats
= SNDRV_PCM_FMTBIT_S16_LE
,
691 .rates
= (SNDRV_PCM_RATE_8000
| SNDRV_PCM_RATE_16000
|\
692 SNDRV_PCM_RATE_22050
| SNDRV_PCM_RATE_32000
|\
693 SNDRV_PCM_RATE_44100
| SNDRV_PCM_RATE_48000
|\
694 SNDRV_PCM_RATE_KNOT
),
699 .buffer_bytes_max
= 64*1024,
700 .period_bytes_min
= 64,
701 .period_bytes_max
= DMA_BUF_SIZE
,
707 static struct snd_pcm_hardware snd_sa11xx_uda1341_playback
=
709 .info
= (SNDRV_PCM_INFO_INTERLEAVED
|
710 SNDRV_PCM_INFO_BLOCK_TRANSFER
|
711 SNDRV_PCM_INFO_MMAP
| SNDRV_PCM_INFO_MMAP_VALID
|
712 SNDRV_PCM_INFO_PAUSE
| SNDRV_PCM_INFO_RESUME
),
713 .formats
= SNDRV_PCM_FMTBIT_S16_LE
,
714 .rates
= (SNDRV_PCM_RATE_8000
| SNDRV_PCM_RATE_16000
|\
715 SNDRV_PCM_RATE_22050
| SNDRV_PCM_RATE_32000
|\
716 SNDRV_PCM_RATE_44100
| SNDRV_PCM_RATE_48000
|\
717 SNDRV_PCM_RATE_KNOT
),
722 .buffer_bytes_max
= 64*1024,
723 .period_bytes_min
= 64,
724 .period_bytes_max
= DMA_BUF_SIZE
,
730 static int snd_card_sa11xx_uda1341_open(struct snd_pcm_substream
*substream
)
732 struct sa11xx_uda1341
*chip
= snd_pcm_substream_chip(substream
);
733 struct snd_pcm_runtime
*runtime
= substream
->runtime
;
734 int stream_id
= substream
->pstr
->stream
;
737 chip
->s
[stream_id
].stream
= substream
;
739 if (stream_id
== SNDRV_PCM_STREAM_PLAYBACK
)
740 runtime
->hw
= snd_sa11xx_uda1341_playback
;
742 runtime
->hw
= snd_sa11xx_uda1341_capture
;
743 if ((err
= snd_pcm_hw_constraint_integer(runtime
, SNDRV_PCM_HW_PARAM_PERIODS
)) < 0)
745 if ((err
= snd_pcm_hw_constraint_list(runtime
, 0, SNDRV_PCM_HW_PARAM_RATE
, &hw_constraints_rates
)) < 0)
751 static int snd_card_sa11xx_uda1341_close(struct snd_pcm_substream
*substream
)
753 struct sa11xx_uda1341
*chip
= snd_pcm_substream_chip(substream
);
755 chip
->s
[substream
->pstr
->stream
].stream
= NULL
;
759 /* {{{ HW params & free */
761 static int snd_sa11xx_uda1341_hw_params(struct snd_pcm_substream
*substream
,
762 struct snd_pcm_hw_params
*hw_params
)
765 return snd_pcm_lib_malloc_pages(substream
, params_buffer_bytes(hw_params
));
768 static int snd_sa11xx_uda1341_hw_free(struct snd_pcm_substream
*substream
)
770 return snd_pcm_lib_free_pages(substream
);
775 static struct snd_pcm_ops snd_card_sa11xx_uda1341_playback_ops
= {
776 .open
= snd_card_sa11xx_uda1341_open
,
777 .close
= snd_card_sa11xx_uda1341_close
,
778 .ioctl
= snd_pcm_lib_ioctl
,
779 .hw_params
= snd_sa11xx_uda1341_hw_params
,
780 .hw_free
= snd_sa11xx_uda1341_hw_free
,
781 .prepare
= snd_sa11xx_uda1341_prepare
,
782 .trigger
= snd_sa11xx_uda1341_trigger
,
783 .pointer
= snd_sa11xx_uda1341_pointer
,
786 static struct snd_pcm_ops snd_card_sa11xx_uda1341_capture_ops
= {
787 .open
= snd_card_sa11xx_uda1341_open
,
788 .close
= snd_card_sa11xx_uda1341_close
,
789 .ioctl
= snd_pcm_lib_ioctl
,
790 .hw_params
= snd_sa11xx_uda1341_hw_params
,
791 .hw_free
= snd_sa11xx_uda1341_hw_free
,
792 .prepare
= snd_sa11xx_uda1341_prepare
,
793 .trigger
= snd_sa11xx_uda1341_trigger
,
794 .pointer
= snd_sa11xx_uda1341_pointer
,
797 static int __init
snd_card_sa11xx_uda1341_pcm(struct sa11xx_uda1341
*sa11xx_uda1341
, int device
)
802 if ((err
= snd_pcm_new(sa11xx_uda1341
->card
, "UDA1341 PCM", device
, 1, 1, &pcm
)) < 0)
806 * this sets up our initial buffers and sets the dma_type to isa.
807 * isa works but I'm not sure why (or if) it's the right choice
808 * this may be too large, trying it for now
810 snd_pcm_lib_preallocate_pages_for_all(pcm
, SNDRV_DMA_TYPE_DEV
,
814 snd_pcm_set_ops(pcm
, SNDRV_PCM_STREAM_PLAYBACK
, &snd_card_sa11xx_uda1341_playback_ops
);
815 snd_pcm_set_ops(pcm
, SNDRV_PCM_STREAM_CAPTURE
, &snd_card_sa11xx_uda1341_capture_ops
);
816 pcm
->private_data
= sa11xx_uda1341
;
818 strcpy(pcm
->name
, "UDA1341 PCM");
820 sa11xx_uda1341_audio_init(sa11xx_uda1341
);
822 /* setup DMA controller */
823 audio_dma_request(&sa11xx_uda1341
->s
[SNDRV_PCM_STREAM_PLAYBACK
], audio_dma_callback
);
824 audio_dma_request(&sa11xx_uda1341
->s
[SNDRV_PCM_STREAM_CAPTURE
], audio_dma_callback
);
826 sa11xx_uda1341
->pcm
= pcm
;
833 /* {{{ module init & exit */
837 static int snd_sa11xx_uda1341_suspend(struct platform_device
*devptr
,
840 struct snd_card
*card
= platform_get_drvdata(devptr
);
841 struct sa11xx_uda1341
*chip
= card
->private_data
;
843 snd_power_change_state(card
, SNDRV_CTL_POWER_D3hot
);
844 snd_pcm_suspend_all(chip
->pcm
);
846 sa1100_dma_sleep(chip
->s
[SNDRV_PCM_STREAM_PLAYBACK
].dmach
);
847 sa1100_dma_sleep(chip
->s
[SNDRV_PCM_STREAM_CAPTURE
].dmach
);
851 l3_command(chip
->uda1341
, CMD_SUSPEND
, NULL
);
852 sa11xx_uda1341_audio_shutdown(chip
);
857 static int snd_sa11xx_uda1341_resume(struct platform_device
*devptr
)
859 struct snd_card
*card
= platform_get_drvdata(devptr
);
860 struct sa11xx_uda1341
*chip
= card
->private_data
;
862 sa11xx_uda1341_audio_init(chip
);
863 l3_command(chip
->uda1341
, CMD_RESUME
, NULL
);
865 sa1100_dma_wakeup(chip
->s
[SNDRV_PCM_STREAM_PLAYBACK
].dmach
);
866 sa1100_dma_wakeup(chip
->s
[SNDRV_PCM_STREAM_CAPTURE
].dmach
);
870 snd_power_change_state(card
, SNDRV_CTL_POWER_D0
);
873 #endif /* COMFIG_PM */
875 void snd_sa11xx_uda1341_free(struct snd_card
*card
)
877 struct sa11xx_uda1341
*chip
= card
->private_data
;
879 audio_dma_free(&chip
->s
[SNDRV_PCM_STREAM_PLAYBACK
]);
880 audio_dma_free(&chip
->s
[SNDRV_PCM_STREAM_CAPTURE
]);
883 static int __devinit
sa11xx_uda1341_probe(struct platform_device
*devptr
)
886 struct snd_card
*card
;
887 struct sa11xx_uda1341
*chip
;
889 /* register the soundcard */
890 card
= snd_card_new(-1, id
, THIS_MODULE
, sizeof(struct sa11xx_uda1341
));
894 chip
= card
->private_data
;
895 spin_lock_init(&chip
->s
[0].dma_lock
);
896 spin_lock_init(&chip
->s
[1].dma_lock
);
898 card
->private_free
= snd_sa11xx_uda1341_free
;
900 chip
->samplerate
= AUDIO_RATE_DEFAULT
;
903 if ((err
= snd_chip_uda1341_mixer_new(card
, &chip
->uda1341
)))
907 if ((err
= snd_card_sa11xx_uda1341_pcm(chip
, 0)) < 0)
910 strcpy(card
->driver
, "UDA1341");
911 strcpy(card
->shortname
, "H3600 UDA1341TS");
912 sprintf(card
->longname
, "Compaq iPAQ H3600 with Philips UDA1341TS");
914 snd_card_set_dev(card
, &devptr
->dev
);
916 if ((err
= snd_card_register(card
)) == 0) {
917 printk( KERN_INFO
"iPAQ audio support initialized\n" );
918 platform_set_drvdata(devptr
, card
);
927 static int __devexit
sa11xx_uda1341_remove(struct platform_device
*devptr
)
929 snd_card_free(platform_get_drvdata(devptr
));
930 platform_set_drvdata(devptr
, NULL
);
934 #define SA11XX_UDA1341_DRIVER "sa11xx_uda1341"
936 static struct platform_driver sa11xx_uda1341_driver
= {
937 .probe
= sa11xx_uda1341_probe
,
938 .remove
= __devexit_p(sa11xx_uda1341_remove
),
940 .suspend
= snd_sa11xx_uda1341_suspend
,
941 .resume
= snd_sa11xx_uda1341_resume
,
944 .name
= SA11XX_UDA1341_DRIVER
,
948 static int __init
sa11xx_uda1341_init(void)
952 if (!machine_is_h3xxx())
954 if ((err
= platform_driver_register(&sa11xx_uda1341_driver
)) < 0)
956 device
= platform_device_register_simple(SA11XX_UDA1341_DRIVER
, -1, NULL
, 0);
957 if (!IS_ERR(device
)) {
958 if (platform_get_drvdata(device
))
960 platform_device_unregister(device
);
963 err
= PTR_ERR(device
);
964 platform_driver_unregister(&sa11xx_uda1341_driver
);
968 static void __exit
sa11xx_uda1341_exit(void)
970 platform_device_unregister(device
);
971 platform_driver_unregister(&sa11xx_uda1341_driver
);
974 module_init(sa11xx_uda1341_init
);
975 module_exit(sa11xx_uda1341_exit
);
981 * indent-tabs-mode: t