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/module.h>
63 #include <linux/moduleparam.h>
64 #include <linux/init.h>
65 #include <linux/err.h>
66 #include <linux/platform_device.h>
67 #include <linux/errno.h>
68 #include <linux/ioctl.h>
69 #include <linux/delay.h>
70 #include <linux/slab.h>
76 #include <asm/hardware.h>
77 #include <asm/arch/h3600.h>
78 #include <asm/mach-types.h>
81 #include <sound/core.h>
82 #include <sound/pcm.h>
83 #include <sound/initval.h>
85 #include <linux/l3/l3.h>
88 #undef DEBUG_FUNCTION_NAMES
89 #include <sound/uda1341.h>
92 * FIXME: Is this enough as autodetection of 2.4.X-rmkY-hhZ kernels?
93 * We use DMA stuff from 2.4.18-rmk3-hh24 here to be able to compile this
94 * module for Familiar 0.6.1
97 /* {{{ Type definitions */
99 MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek@seznam.cz>");
100 MODULE_LICENSE("GPL");
101 MODULE_DESCRIPTION("SA1100/SA1111 + UDA1341TS driver for ALSA");
102 MODULE_SUPPORTED_DEVICE("{{UDA1341,iPAQ H3600 UDA1341TS}}");
104 static char *id
; /* ID for this card */
106 module_param(id
, charp
, 0444);
107 MODULE_PARM_DESC(id
, "ID string for SA1100/SA1111 + UDA1341TS soundcard.");
109 struct audio_stream
{
110 char *id
; /* identification string */
111 int stream_id
; /* numeric identification */
112 dma_device_t dma_dev
; /* device identifier for DMA */
114 dmach_t dmach
; /* dma channel identification */
116 dma_regs_t
*dma_regs
; /* points to our DMA registers */
118 unsigned int active
:1; /* we are using this stream for transfer now */
119 int period
; /* current transfer period */
120 int periods
; /* current count of periods registerd in the DMA engine */
121 int tx_spin
; /* are we recoding - flag used to do DMA trans. for sync */
122 unsigned int old_offset
;
123 spinlock_t dma_lock
; /* for locking in DMA operations (see dma-sa1100.c in the kernel) */
124 struct snd_pcm_substream
*stream
;
127 struct sa11xx_uda1341
{
128 struct snd_card
*card
;
129 struct l3_client
*uda1341
;
132 struct audio_stream s
[2]; /* playback & capture */
135 static unsigned int rates
[] = {
136 8000, 10666, 10985, 14647,
137 16000, 21970, 22050, 24000,
138 29400, 32000, 44100, 48000,
141 static struct snd_pcm_hw_constraint_list hw_constraints_rates
= {
142 .count
= ARRAY_SIZE(rates
),
147 static struct platform_device
*device
;
151 /* {{{ Clock and sample rate stuff */
154 * Stop-gap solution until rest of hh.org HAL stuff is merged.
156 #define GPIO_H3600_CLK_SET0 GPIO_GPIO (12)
157 #define GPIO_H3600_CLK_SET1 GPIO_GPIO (13)
159 #ifdef CONFIG_SA1100_H3XXX
160 #define clr_sa11xx_uda1341_egpio(x) clr_h3600_egpio(x)
161 #define set_sa11xx_uda1341_egpio(x) set_h3600_egpio(x)
163 #error This driver could serve H3x00 handhelds only!
166 static void sa11xx_uda1341_set_audio_clock(long val
)
169 case 24000: case 32000: case 48000: /* 00: 12.288 MHz */
170 GPCR
= GPIO_H3600_CLK_SET0
| GPIO_H3600_CLK_SET1
;
173 case 22050: case 29400: case 44100: /* 01: 11.2896 MHz */
174 GPSR
= GPIO_H3600_CLK_SET0
;
175 GPCR
= GPIO_H3600_CLK_SET1
;
178 case 8000: case 10666: case 16000: /* 10: 4.096 MHz */
179 GPCR
= GPIO_H3600_CLK_SET0
;
180 GPSR
= GPIO_H3600_CLK_SET1
;
183 case 10985: case 14647: case 21970: /* 11: 5.6245 MHz */
184 GPSR
= GPIO_H3600_CLK_SET0
| GPIO_H3600_CLK_SET1
;
189 static void sa11xx_uda1341_set_samplerate(struct sa11xx_uda1341
*sa11xx_uda1341
, long rate
)
194 /* We don't want to mess with clocks when frames are in flight */
195 Ser4SSCR0
&= ~SSCR0_SSE
;
196 /* wait for any frame to complete */
200 * We have the following clock sources:
201 * 4.096 MHz, 5.6245 MHz, 11.2896 MHz, 12.288 MHz
202 * Those can be divided either by 256, 384 or 512.
203 * This makes up 12 combinations for the following samplerates...
207 else if (rate
>= 44100)
209 else if (rate
>= 32000)
211 else if (rate
>= 29400)
213 else if (rate
>= 24000)
215 else if (rate
>= 22050)
217 else if (rate
>= 21970)
219 else if (rate
>= 16000)
221 else if (rate
>= 14647)
223 else if (rate
>= 10985)
225 else if (rate
>= 10666)
230 /* Set the external clock generator */
232 sa11xx_uda1341_set_audio_clock(rate
);
234 /* Select the clock divisor */
241 clk_div
= SSCR0_SerClkDiv(16);
248 clk_div
= SSCR0_SerClkDiv(8);
255 clk_div
= SSCR0_SerClkDiv(12);
259 /* FMT setting should be moved away when other FMTs are added (FIXME) */
260 l3_command(sa11xx_uda1341
->uda1341
, CMD_FORMAT
, (void *)LSB16
);
262 l3_command(sa11xx_uda1341
->uda1341
, CMD_FS
, (void *)clk
);
263 Ser4SSCR0
= (Ser4SSCR0
& ~0xff00) + clk_div
+ SSCR0_SSE
;
264 sa11xx_uda1341
->samplerate
= rate
;
269 /* {{{ HW init and shutdown */
271 static void sa11xx_uda1341_audio_init(struct sa11xx_uda1341
*sa11xx_uda1341
)
275 /* Setup DMA stuff */
276 sa11xx_uda1341
->s
[SNDRV_PCM_STREAM_PLAYBACK
].id
= "UDA1341 out";
277 sa11xx_uda1341
->s
[SNDRV_PCM_STREAM_PLAYBACK
].stream_id
= SNDRV_PCM_STREAM_PLAYBACK
;
278 sa11xx_uda1341
->s
[SNDRV_PCM_STREAM_PLAYBACK
].dma_dev
= DMA_Ser4SSPWr
;
280 sa11xx_uda1341
->s
[SNDRV_PCM_STREAM_CAPTURE
].id
= "UDA1341 in";
281 sa11xx_uda1341
->s
[SNDRV_PCM_STREAM_CAPTURE
].stream_id
= SNDRV_PCM_STREAM_CAPTURE
;
282 sa11xx_uda1341
->s
[SNDRV_PCM_STREAM_CAPTURE
].dma_dev
= DMA_Ser4SSPRd
;
284 /* Initialize the UDA1341 internal state */
286 /* Setup the uarts */
287 local_irq_save(flags
);
288 GAFR
|= (GPIO_SSP_CLK
);
289 GPDR
&= ~(GPIO_SSP_CLK
);
291 Ser4SSCR0
= SSCR0_DataSize(16) + SSCR0_TI
+ SSCR0_SerClkDiv(8);
292 Ser4SSCR1
= SSCR1_SClkIactL
+ SSCR1_SClk1P
+ SSCR1_ExtClk
;
293 Ser4SSCR0
|= SSCR0_SSE
;
294 local_irq_restore(flags
);
296 /* Enable the audio power */
298 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET
);
299 set_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON
);
300 set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE
);
302 /* Wait for the UDA1341 to wake up */
303 mdelay(1); //FIXME - was removed by Perex - Why?
305 /* Initialize the UDA1341 internal state */
306 l3_open(sa11xx_uda1341
->uda1341
);
308 /* external clock configuration (after l3_open - regs must be initialized */
309 sa11xx_uda1341_set_samplerate(sa11xx_uda1341
, sa11xx_uda1341
->samplerate
);
311 /* Wait for the UDA1341 to wake up */
312 set_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET
);
315 /* make the left and right channels unswapped (flip the WS latch) */
318 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE
);
321 static void sa11xx_uda1341_audio_shutdown(struct sa11xx_uda1341
*sa11xx_uda1341
)
324 set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE
);
326 /* disable the audio power and all signals leading to the audio chip */
327 l3_close(sa11xx_uda1341
->uda1341
);
329 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET
);
331 /* power off and mute off */
332 /* FIXME - is muting off necesary??? */
334 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON
);
335 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE
);
343 * these are the address and sizes used to fill the xmit buffer
344 * so we can get a clock in record only mode
346 #define FORCE_CLOCK_ADDR (dma_addr_t)FLUSH_BASE_PHYS
347 #define FORCE_CLOCK_SIZE 4096 // was 2048
349 // FIXME Why this value exactly - wrote comment
350 #define DMA_BUF_SIZE 8176 /* <= MAX_DMA_SIZE from asm/arch-sa1100/dma.h */
354 static int audio_dma_request(struct audio_stream
*s
, void (*callback
)(void *, int))
358 ret
= sa1100_request_dma(&s
->dmach
, s
->id
, s
->dma_dev
);
360 printk(KERN_ERR
"unable to grab audio dma 0x%x\n", s
->dma_dev
);
363 sa1100_dma_set_callback(s
->dmach
, callback
);
367 static inline void audio_dma_free(struct audio_stream
*s
)
369 sa1100_free_dma(s
->dmach
);
375 static int audio_dma_request(struct audio_stream
*s
, void (*callback
)(void *))
379 ret
= sa1100_request_dma(s
->dma_dev
, s
->id
, callback
, s
, &s
->dma_regs
);
381 printk(KERN_ERR
"unable to grab audio dma 0x%x\n", s
->dma_dev
);
385 static void audio_dma_free(struct audio_stream
*s
)
387 sa1100_free_dma(s
->dma_regs
);
393 static u_int
audio_get_dma_pos(struct audio_stream
*s
)
395 struct snd_pcm_substream
*substream
= s
->stream
;
396 struct snd_pcm_runtime
*runtime
= substream
->runtime
;
401 // this must be called w/ interrupts locked out see dma-sa1100.c in the kernel
402 spin_lock_irqsave(&s
->dma_lock
, flags
);
404 sa1100_dma_get_current(s
->dmach
, NULL
, &addr
);
406 addr
= sa1100_get_dma_pos((s
)->dma_regs
);
408 offset
= addr
- runtime
->dma_addr
;
409 spin_unlock_irqrestore(&s
->dma_lock
, flags
);
411 offset
= bytes_to_frames(runtime
,offset
);
412 if (offset
>= runtime
->buffer_size
)
419 * this stops the dma and clears the dma ptrs
421 static void audio_stop_dma(struct audio_stream
*s
)
425 spin_lock_irqsave(&s
->dma_lock
, flags
);
428 /* this stops the dma channel and clears the buffer ptrs */
430 sa1100_dma_flush_all(s
->dmach
);
432 sa1100_clear_dma(s
->dma_regs
);
434 spin_unlock_irqrestore(&s
->dma_lock
, flags
);
437 static void audio_process_dma(struct audio_stream
*s
)
439 struct snd_pcm_substream
*substream
= s
->stream
;
440 struct snd_pcm_runtime
*runtime
;
441 unsigned int dma_size
;
445 /* we are requested to process synchronization DMA transfer */
447 snd_assert(s
->stream_id
== SNDRV_PCM_STREAM_PLAYBACK
, return);
448 /* fill the xmit dma buffers and return */
450 sa1100_dma_set_spin(s
->dmach
, FORCE_CLOCK_ADDR
, FORCE_CLOCK_SIZE
);
453 ret
= sa1100_start_dma(s
->dma_regs
, FORCE_CLOCK_ADDR
, FORCE_CLOCK_SIZE
);
461 /* must be set here - only valid for running streams, not for forced_clock dma fills */
462 runtime
= substream
->runtime
;
463 while (s
->active
&& s
->periods
< runtime
->periods
) {
464 dma_size
= frames_to_bytes(runtime
, runtime
->period_size
);
466 /* a little trick, we need resume from old position */
467 offset
= frames_to_bytes(runtime
, s
->old_offset
- 1);
470 s
->period
= offset
/ dma_size
;
472 dma_size
= dma_size
- offset
;
474 continue; /* special case */
476 offset
= dma_size
* s
->period
;
477 snd_assert(dma_size
<= DMA_BUF_SIZE
, );
480 ret
= sa1100_dma_queue_buffer(s
->dmach
, s
, runtime
->dma_addr
+ offset
, dma_size
);
484 ret
= sa1100_start_dma((s
)->dma_regs
, runtime
->dma_addr
+ offset
, dma_size
);
486 printk(KERN_ERR
"audio_process_dma: cannot queue DMA buffer (%i)\n", ret
);
492 s
->period
%= runtime
->periods
;
498 static void audio_dma_callback(void *data
, int size
)
500 static void audio_dma_callback(void *data
)
503 struct audio_stream
*s
= data
;
506 * If we are getting a callback for an active stream then we inform
507 * the PCM middle layer we've finished a period
510 snd_pcm_period_elapsed(s
->stream
);
512 spin_lock(&s
->dma_lock
);
513 if (!s
->tx_spin
&& s
->periods
> 0)
515 audio_process_dma(s
);
516 spin_unlock(&s
->dma_lock
);
521 /* {{{ PCM setting */
523 /* {{{ trigger & timer */
525 static int snd_sa11xx_uda1341_trigger(struct snd_pcm_substream
*substream
, int cmd
)
527 struct sa11xx_uda1341
*chip
= snd_pcm_substream_chip(substream
);
528 int stream_id
= substream
->pstr
->stream
;
529 struct audio_stream
*s
= &chip
->s
[stream_id
];
530 struct audio_stream
*s1
= &chip
->s
[stream_id
^ 1];
533 /* note local interrupts are already disabled in the midlevel code */
534 spin_lock(&s
->dma_lock
);
536 case SNDRV_PCM_TRIGGER_START
:
537 /* now we need to make sure a record only stream has a clock */
538 if (stream_id
== SNDRV_PCM_STREAM_CAPTURE
&& !s1
->active
) {
539 /* we need to force fill the xmit DMA with zeros */
541 audio_process_dma(s1
);
543 /* this case is when you were recording then you turn on a
544 * playback stream so we stop (also clears it) the dma first,
545 * clear the sync flag and then we let it turned on
551 /* requested stream startup */
553 audio_process_dma(s
);
555 case SNDRV_PCM_TRIGGER_STOP
:
556 /* requested stream shutdown */
560 * now we need to make sure a record only stream has a clock
561 * so if we're stopping a playback with an active capture
562 * we need to turn the 0 fill dma on for the xmit side
564 if (stream_id
== SNDRV_PCM_STREAM_PLAYBACK
&& s1
->active
) {
565 /* we need to force fill the xmit DMA with zeros */
567 audio_process_dma(s
);
570 * we killed a capture only stream, so we should also kill
571 * the zero fill transmit
581 case SNDRV_PCM_TRIGGER_SUSPEND
:
584 sa1100_dma_stop(s
->dmach
);
588 s
->old_offset
= audio_get_dma_pos(s
) + 1;
590 sa1100_dma_flush_all(s
->dmach
);
596 case SNDRV_PCM_TRIGGER_RESUME
:
599 audio_process_dma(s
);
600 if (stream_id
== SNDRV_PCM_STREAM_CAPTURE
&& !s1
->active
) {
602 audio_process_dma(s1
);
605 case SNDRV_PCM_TRIGGER_PAUSE_PUSH
:
607 sa1100_dma_stop(s
->dmach
);
612 if (stream_id
== SNDRV_PCM_STREAM_PLAYBACK
) {
615 s
->old_offset
= audio_get_dma_pos(s
) + 1;
617 sa1100_dma_flush_all(s
->dmach
);
621 audio_process_dma(s
);
627 sa1100_dma_flush_all(s1
->dmach
);
634 case SNDRV_PCM_TRIGGER_PAUSE_RELEASE
:
638 audio_process_dma(s
);
641 if (stream_id
== SNDRV_PCM_STREAM_CAPTURE
&& !s1
->active
) {
643 audio_process_dma(s1
);
646 sa1100_dma_resume(s
->dmach
);
655 spin_unlock(&s
->dma_lock
);
659 static int snd_sa11xx_uda1341_prepare(struct snd_pcm_substream
*substream
)
661 struct sa11xx_uda1341
*chip
= snd_pcm_substream_chip(substream
);
662 struct snd_pcm_runtime
*runtime
= substream
->runtime
;
663 struct audio_stream
*s
= &chip
->s
[substream
->pstr
->stream
];
665 /* set requested samplerate */
666 sa11xx_uda1341_set_samplerate(chip
, runtime
->rate
);
668 /* set requestd format when available */
669 /* set FMT here !!! FIXME */
677 static snd_pcm_uframes_t
snd_sa11xx_uda1341_pointer(struct snd_pcm_substream
*substream
)
679 struct sa11xx_uda1341
*chip
= snd_pcm_substream_chip(substream
);
680 return audio_get_dma_pos(&chip
->s
[substream
->pstr
->stream
]);
685 static struct snd_pcm_hardware snd_sa11xx_uda1341_capture
=
687 .info
= (SNDRV_PCM_INFO_INTERLEAVED
|
688 SNDRV_PCM_INFO_BLOCK_TRANSFER
|
689 SNDRV_PCM_INFO_MMAP
| SNDRV_PCM_INFO_MMAP_VALID
|
690 SNDRV_PCM_INFO_PAUSE
| SNDRV_PCM_INFO_RESUME
),
691 .formats
= SNDRV_PCM_FMTBIT_S16_LE
,
692 .rates
= (SNDRV_PCM_RATE_8000
| SNDRV_PCM_RATE_16000
|\
693 SNDRV_PCM_RATE_22050
| SNDRV_PCM_RATE_32000
|\
694 SNDRV_PCM_RATE_44100
| SNDRV_PCM_RATE_48000
|\
695 SNDRV_PCM_RATE_KNOT
),
700 .buffer_bytes_max
= 64*1024,
701 .period_bytes_min
= 64,
702 .period_bytes_max
= DMA_BUF_SIZE
,
708 static struct snd_pcm_hardware snd_sa11xx_uda1341_playback
=
710 .info
= (SNDRV_PCM_INFO_INTERLEAVED
|
711 SNDRV_PCM_INFO_BLOCK_TRANSFER
|
712 SNDRV_PCM_INFO_MMAP
| SNDRV_PCM_INFO_MMAP_VALID
|
713 SNDRV_PCM_INFO_PAUSE
| SNDRV_PCM_INFO_RESUME
),
714 .formats
= SNDRV_PCM_FMTBIT_S16_LE
,
715 .rates
= (SNDRV_PCM_RATE_8000
| SNDRV_PCM_RATE_16000
|\
716 SNDRV_PCM_RATE_22050
| SNDRV_PCM_RATE_32000
|\
717 SNDRV_PCM_RATE_44100
| SNDRV_PCM_RATE_48000
|\
718 SNDRV_PCM_RATE_KNOT
),
723 .buffer_bytes_max
= 64*1024,
724 .period_bytes_min
= 64,
725 .period_bytes_max
= DMA_BUF_SIZE
,
731 static int snd_card_sa11xx_uda1341_open(struct snd_pcm_substream
*substream
)
733 struct sa11xx_uda1341
*chip
= snd_pcm_substream_chip(substream
);
734 struct snd_pcm_runtime
*runtime
= substream
->runtime
;
735 int stream_id
= substream
->pstr
->stream
;
738 chip
->s
[stream_id
].stream
= substream
;
740 if (stream_id
== SNDRV_PCM_STREAM_PLAYBACK
)
741 runtime
->hw
= snd_sa11xx_uda1341_playback
;
743 runtime
->hw
= snd_sa11xx_uda1341_capture
;
744 if ((err
= snd_pcm_hw_constraint_integer(runtime
, SNDRV_PCM_HW_PARAM_PERIODS
)) < 0)
746 if ((err
= snd_pcm_hw_constraint_list(runtime
, 0, SNDRV_PCM_HW_PARAM_RATE
, &hw_constraints_rates
)) < 0)
752 static int snd_card_sa11xx_uda1341_close(struct snd_pcm_substream
*substream
)
754 struct sa11xx_uda1341
*chip
= snd_pcm_substream_chip(substream
);
756 chip
->s
[substream
->pstr
->stream
].stream
= NULL
;
760 /* {{{ HW params & free */
762 static int snd_sa11xx_uda1341_hw_params(struct snd_pcm_substream
*substream
,
763 struct snd_pcm_hw_params
*hw_params
)
766 return snd_pcm_lib_malloc_pages(substream
, params_buffer_bytes(hw_params
));
769 static int snd_sa11xx_uda1341_hw_free(struct snd_pcm_substream
*substream
)
771 return snd_pcm_lib_free_pages(substream
);
776 static struct snd_pcm_ops snd_card_sa11xx_uda1341_playback_ops
= {
777 .open
= snd_card_sa11xx_uda1341_open
,
778 .close
= snd_card_sa11xx_uda1341_close
,
779 .ioctl
= snd_pcm_lib_ioctl
,
780 .hw_params
= snd_sa11xx_uda1341_hw_params
,
781 .hw_free
= snd_sa11xx_uda1341_hw_free
,
782 .prepare
= snd_sa11xx_uda1341_prepare
,
783 .trigger
= snd_sa11xx_uda1341_trigger
,
784 .pointer
= snd_sa11xx_uda1341_pointer
,
787 static struct snd_pcm_ops snd_card_sa11xx_uda1341_capture_ops
= {
788 .open
= snd_card_sa11xx_uda1341_open
,
789 .close
= snd_card_sa11xx_uda1341_close
,
790 .ioctl
= snd_pcm_lib_ioctl
,
791 .hw_params
= snd_sa11xx_uda1341_hw_params
,
792 .hw_free
= snd_sa11xx_uda1341_hw_free
,
793 .prepare
= snd_sa11xx_uda1341_prepare
,
794 .trigger
= snd_sa11xx_uda1341_trigger
,
795 .pointer
= snd_sa11xx_uda1341_pointer
,
798 static int __init
snd_card_sa11xx_uda1341_pcm(struct sa11xx_uda1341
*sa11xx_uda1341
, int device
)
803 if ((err
= snd_pcm_new(sa11xx_uda1341
->card
, "UDA1341 PCM", device
, 1, 1, &pcm
)) < 0)
807 * this sets up our initial buffers and sets the dma_type to isa.
808 * isa works but I'm not sure why (or if) it's the right choice
809 * this may be too large, trying it for now
811 snd_pcm_lib_preallocate_pages_for_all(pcm
, SNDRV_DMA_TYPE_DEV
,
815 snd_pcm_set_ops(pcm
, SNDRV_PCM_STREAM_PLAYBACK
, &snd_card_sa11xx_uda1341_playback_ops
);
816 snd_pcm_set_ops(pcm
, SNDRV_PCM_STREAM_CAPTURE
, &snd_card_sa11xx_uda1341_capture_ops
);
817 pcm
->private_data
= sa11xx_uda1341
;
819 strcpy(pcm
->name
, "UDA1341 PCM");
821 sa11xx_uda1341_audio_init(sa11xx_uda1341
);
823 /* setup DMA controller */
824 audio_dma_request(&sa11xx_uda1341
->s
[SNDRV_PCM_STREAM_PLAYBACK
], audio_dma_callback
);
825 audio_dma_request(&sa11xx_uda1341
->s
[SNDRV_PCM_STREAM_CAPTURE
], audio_dma_callback
);
827 sa11xx_uda1341
->pcm
= pcm
;
834 /* {{{ module init & exit */
838 static int snd_sa11xx_uda1341_suspend(struct platform_device
*devptr
,
841 struct snd_card
*card
= platform_get_drvdata(devptr
);
842 struct sa11xx_uda1341
*chip
= card
->private_data
;
844 snd_power_change_state(card
, SNDRV_CTL_POWER_D3hot
);
845 snd_pcm_suspend_all(chip
->pcm
);
847 sa1100_dma_sleep(chip
->s
[SNDRV_PCM_STREAM_PLAYBACK
].dmach
);
848 sa1100_dma_sleep(chip
->s
[SNDRV_PCM_STREAM_CAPTURE
].dmach
);
852 l3_command(chip
->uda1341
, CMD_SUSPEND
, NULL
);
853 sa11xx_uda1341_audio_shutdown(chip
);
858 static int snd_sa11xx_uda1341_resume(struct platform_device
*devptr
)
860 struct snd_card
*card
= platform_get_drvdata(devptr
);
861 struct sa11xx_uda1341
*chip
= card
->private_data
;
863 sa11xx_uda1341_audio_init(chip
);
864 l3_command(chip
->uda1341
, CMD_RESUME
, NULL
);
866 sa1100_dma_wakeup(chip
->s
[SNDRV_PCM_STREAM_PLAYBACK
].dmach
);
867 sa1100_dma_wakeup(chip
->s
[SNDRV_PCM_STREAM_CAPTURE
].dmach
);
871 snd_power_change_state(card
, SNDRV_CTL_POWER_D0
);
874 #endif /* COMFIG_PM */
876 void snd_sa11xx_uda1341_free(struct snd_card
*card
)
878 struct sa11xx_uda1341
*chip
= card
->private_data
;
880 audio_dma_free(&chip
->s
[SNDRV_PCM_STREAM_PLAYBACK
]);
881 audio_dma_free(&chip
->s
[SNDRV_PCM_STREAM_CAPTURE
]);
884 static int __init
sa11xx_uda1341_probe(struct platform_device
*devptr
)
887 struct snd_card
*card
;
888 struct sa11xx_uda1341
*chip
;
890 /* register the soundcard */
891 card
= snd_card_new(-1, id
, THIS_MODULE
, sizeof(struct sa11xx_uda1341
));
895 chip
= card
->private_data
;
896 spin_lock_init(&chip
->s
[0].dma_lock
);
897 spin_lock_init(&chip
->s
[1].dma_lock
);
899 card
->private_free
= snd_sa11xx_uda1341_free
;
901 chip
->samplerate
= AUDIO_RATE_DEFAULT
;
904 if ((err
= snd_chip_uda1341_mixer_new(card
, &chip
->uda1341
)))
908 if ((err
= snd_card_sa11xx_uda1341_pcm(chip
, 0)) < 0)
911 strcpy(card
->driver
, "UDA1341");
912 strcpy(card
->shortname
, "H3600 UDA1341TS");
913 sprintf(card
->longname
, "Compaq iPAQ H3600 with Philips UDA1341TS");
915 snd_card_set_dev(card
, &devptr
->dev
);
917 if ((err
= snd_card_register(card
)) == 0) {
918 printk( KERN_INFO
"iPAQ audio support initialized\n" );
919 platform_set_drvdata(devptr
, card
);
928 static int __devexit
sa11xx_uda1341_remove(struct platform_device
*devptr
)
930 snd_card_free(platform_get_drvdata(devptr
));
931 platform_set_drvdata(devptr
, NULL
);
935 #define SA11XX_UDA1341_DRIVER "sa11xx_uda1341"
937 static struct platform_driver sa11xx_uda1341_driver
= {
938 .probe
= sa11xx_uda1341_probe
,
939 .remove
= __devexit_p(sa11xx_uda1341_remove
),
941 .suspend
= snd_sa11xx_uda1341_suspend
,
942 .resume
= snd_sa11xx_uda1341_resume
,
945 .name
= SA11XX_UDA1341_DRIVER
,
949 static int __init
sa11xx_uda1341_init(void)
953 if (!machine_is_h3xxx())
955 if ((err
= platform_driver_register(&sa11xx_uda1341_driver
)) < 0)
957 device
= platform_device_register_simple(SA11XX_UDA1341_DRIVER
, -1, NULL
, 0);
958 if (!IS_ERR(device
)) {
959 if (platform_get_drvdata(device
))
961 platform_device_unregister(device
);
964 err
= PTR_ERR(device
);
965 platform_driver_unregister(&sa11xx_uda1341_driver
);
969 static void __exit
sa11xx_uda1341_exit(void)
971 platform_device_unregister(device
);
972 platform_driver_unregister(&sa11xx_uda1341_driver
);
975 module_init(sa11xx_uda1341_init
);
976 module_exit(sa11xx_uda1341_exit
);
982 * indent-tabs-mode: t