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-rw-r--r--sound/arm/Kconfig18
-rw-r--r--sound/arm/Makefile8
-rw-r--r--sound/arm/sa11xx-uda1341.c973
3 files changed, 999 insertions, 0 deletions
diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig
new file mode 100644
index 000000000000..cdacf4d3a387
--- /dev/null
+++ b/sound/arm/Kconfig
@@ -0,0 +1,18 @@
1# ALSA ARM drivers
2
3menu "ALSA ARM devices"
4 depends on SND!=n && ARM
5
6config SND_SA11XX_UDA1341
7 tristate "SA11xx UDA1341TS driver (iPaq H3600)"
8 depends on ARCH_SA1100 && SND && L3
9 select SND_PCM
10 help
11 Say Y here if you have a Compaq iPaq H3x00 handheld computer
12 and want to use its Philips UDA 1341 audio chip.
13
14 To compile this driver as a module, choose M here: the module
15 will be called snd-sa11xx-uda1341.
16
17endmenu
18
diff --git a/sound/arm/Makefile b/sound/arm/Makefile
new file mode 100644
index 000000000000..d7e7dc0c3cdf
--- /dev/null
+++ b/sound/arm/Makefile
@@ -0,0 +1,8 @@
1#
2# Makefile for ALSA
3#
4
5snd-sa11xx-uda1341-objs := sa11xx-uda1341.o
6
7# Toplevel Module Dependency
8obj-$(CONFIG_SND_SA11XX_UDA1341) += snd-sa11xx-uda1341.o
diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c
new file mode 100644
index 000000000000..174bc032d1ad
--- /dev/null
+++ b/sound/arm/sa11xx-uda1341.c
@@ -0,0 +1,973 @@
1/*
2 * Driver for Philips UDA1341TS on Compaq iPAQ H3600 soundcard
3 * Copyright (C) 2002 Tomas Kasparek <tomas.kasparek@seznam.cz>
4 *
5 * This program is free software; you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License.
7 *
8 * History:
9 *
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)
22 */
23
24/* $Id: sa11xx-uda1341.c,v 1.21 2005/01/28 19:34:04 tiwai Exp $ */
25
26/***************************************************************************************************
27*
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
30*
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.
37*
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
46*
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.
50*
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!
56*
57* Notes about transfer methods:
58* The async write calls fail. I probably need to implement something else to support them?
59*
60***************************************************************************************************/
61
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/errno.h>
68#include <linux/ioctl.h>
69#include <linux/delay.h>
70#include <linux/slab.h>
71
72#ifdef CONFIG_PM
73#include <linux/pm.h>
74#endif
75
76#include <asm/hardware.h>
77#include <asm/arch/h3600.h>
78#include <asm/mach-types.h>
79#include <asm/dma.h>
80
81#ifdef CONFIG_H3600_HAL
82#include <asm/semaphore.h>
83#include <asm/uaccess.h>
84#include <asm/arch/h3600_hal.h>
85#endif
86
87#include <sound/core.h>
88#include <sound/pcm.h>
89#include <sound/initval.h>
90
91#include <linux/l3/l3.h>
92
93#undef DEBUG_MODE
94#undef DEBUG_FUNCTION_NAMES
95#include <sound/uda1341.h>
96
97/*
98 * FIXME: Is this enough as autodetection of 2.4.X-rmkY-hhZ kernels?
99 * We use DMA stuff from 2.4.18-rmk3-hh24 here to be able to compile this
100 * module for Familiar 0.6.1
101 */
102#ifdef CONFIG_H3600_HAL
103#define HH_VERSION 1
104#endif
105
106/* {{{ Type definitions */
107
108MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek@seznam.cz>");
109MODULE_LICENSE("GPL");
110MODULE_DESCRIPTION("SA1100/SA1111 + UDA1341TS driver for ALSA");
111MODULE_SUPPORTED_DEVICE("{{UDA1341,iPAQ H3600 UDA1341TS}}");
112
113static char *id = NULL; /* ID for this card */
114
115module_param(id, charp, 0444);
116MODULE_PARM_DESC(id, "ID string for SA1100/SA1111 + UDA1341TS soundcard.");
117
118typedef struct audio_stream {
119 char *id; /* identification string */
120 int stream_id; /* numeric identification */
121 dma_device_t dma_dev; /* device identifier for DMA */
122#ifdef HH_VERSION
123 dmach_t dmach; /* dma channel identification */
124#else
125 dma_regs_t *dma_regs; /* points to our DMA registers */
126#endif
127 int active:1; /* we are using this stream for transfer now */
128 int period; /* current transfer period */
129 int periods; /* current count of periods registerd in the DMA engine */
130 int tx_spin; /* are we recoding - flag used to do DMA trans. for sync */
131 unsigned int old_offset;
132 spinlock_t dma_lock; /* for locking in DMA operations (see dma-sa1100.c in the kernel) */
133 snd_pcm_substream_t *stream;
134}audio_stream_t;
135
136typedef struct snd_card_sa11xx_uda1341 {
137 snd_card_t *card;
138 struct l3_client *uda1341;
139 snd_pcm_t *pcm;
140 long samplerate;
141 audio_stream_t s[2]; /* playback & capture */
142} sa11xx_uda1341_t;
143
144static struct snd_card_sa11xx_uda1341 *sa11xx_uda1341 = NULL;
145
146static unsigned int rates[] = {
147 8000, 10666, 10985, 14647,
148 16000, 21970, 22050, 24000,
149 29400, 32000, 44100, 48000,
150};
151
152static snd_pcm_hw_constraint_list_t hw_constraints_rates = {
153 .count = ARRAY_SIZE(rates),
154 .list = rates,
155 .mask = 0,
156};
157
158/* }}} */
159
160/* {{{ Clock and sample rate stuff */
161
162/*
163 * Stop-gap solution until rest of hh.org HAL stuff is merged.
164 */
165#define GPIO_H3600_CLK_SET0 GPIO_GPIO (12)
166#define GPIO_H3600_CLK_SET1 GPIO_GPIO (13)
167
168#ifdef CONFIG_SA1100_H3XXX
169#define clr_sa11xx_uda1341_egpio(x) clr_h3600_egpio(x)
170#define set_sa11xx_uda1341_egpio(x) set_h3600_egpio(x)
171#else
172#error This driver could serve H3x00 handhelds only!
173#endif
174
175static void sa11xx_uda1341_set_audio_clock(long val)
176{
177 switch (val) {
178 case 24000: case 32000: case 48000: /* 00: 12.288 MHz */
179 GPCR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1;
180 break;
181
182 case 22050: case 29400: case 44100: /* 01: 11.2896 MHz */
183 GPSR = GPIO_H3600_CLK_SET0;
184 GPCR = GPIO_H3600_CLK_SET1;
185 break;
186
187 case 8000: case 10666: case 16000: /* 10: 4.096 MHz */
188 GPCR = GPIO_H3600_CLK_SET0;
189 GPSR = GPIO_H3600_CLK_SET1;
190 break;
191
192 case 10985: case 14647: case 21970: /* 11: 5.6245 MHz */
193 GPSR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1;
194 break;
195 }
196}
197
198static void sa11xx_uda1341_set_samplerate(sa11xx_uda1341_t *sa11xx_uda1341, long rate)
199{
200 int clk_div = 0;
201 int clk=0;
202
203 /* We don't want to mess with clocks when frames are in flight */
204 Ser4SSCR0 &= ~SSCR0_SSE;
205 /* wait for any frame to complete */
206 udelay(125);
207
208 /*
209 * We have the following clock sources:
210 * 4.096 MHz, 5.6245 MHz, 11.2896 MHz, 12.288 MHz
211 * Those can be divided either by 256, 384 or 512.
212 * This makes up 12 combinations for the following samplerates...
213 */
214 if (rate >= 48000)
215 rate = 48000;
216 else if (rate >= 44100)
217 rate = 44100;
218 else if (rate >= 32000)
219 rate = 32000;
220 else if (rate >= 29400)
221 rate = 29400;
222 else if (rate >= 24000)
223 rate = 24000;
224 else if (rate >= 22050)
225 rate = 22050;
226 else if (rate >= 21970)
227 rate = 21970;
228 else if (rate >= 16000)
229 rate = 16000;
230 else if (rate >= 14647)
231 rate = 14647;
232 else if (rate >= 10985)
233 rate = 10985;
234 else if (rate >= 10666)
235 rate = 10666;
236 else
237 rate = 8000;
238
239 /* Set the external clock generator */
240#ifdef CONFIG_H3600_HAL
241 h3600_audio_clock(rate);
242#else
243 sa11xx_uda1341_set_audio_clock(rate);
244#endif
245
246 /* Select the clock divisor */
247 switch (rate) {
248 case 8000:
249 case 10985:
250 case 22050:
251 case 24000:
252 clk = F512;
253 clk_div = SSCR0_SerClkDiv(16);
254 break;
255 case 16000:
256 case 21970:
257 case 44100:
258 case 48000:
259 clk = F256;
260 clk_div = SSCR0_SerClkDiv(8);
261 break;
262 case 10666:
263 case 14647:
264 case 29400:
265 case 32000:
266 clk = F384;
267 clk_div = SSCR0_SerClkDiv(12);
268 break;
269 }
270
271 /* FMT setting should be moved away when other FMTs are added (FIXME) */
272 l3_command(sa11xx_uda1341->uda1341, CMD_FORMAT, (void *)LSB16);
273
274 l3_command(sa11xx_uda1341->uda1341, CMD_FS, (void *)clk);
275 Ser4SSCR0 = (Ser4SSCR0 & ~0xff00) + clk_div + SSCR0_SSE;
276 sa11xx_uda1341->samplerate = rate;
277}
278
279/* }}} */
280
281/* {{{ HW init and shutdown */
282
283static void sa11xx_uda1341_audio_init(sa11xx_uda1341_t *sa11xx_uda1341)
284{
285 unsigned long flags;
286
287 /* Setup DMA stuff */
288 sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].id = "UDA1341 out";
289 sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].stream_id = SNDRV_PCM_STREAM_PLAYBACK;
290 sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].dma_dev = DMA_Ser4SSPWr;
291
292 sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].id = "UDA1341 in";
293 sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].stream_id = SNDRV_PCM_STREAM_CAPTURE;
294 sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].dma_dev = DMA_Ser4SSPRd;
295
296 /* Initialize the UDA1341 internal state */
297
298 /* Setup the uarts */
299 local_irq_save(flags);
300 GAFR |= (GPIO_SSP_CLK);
301 GPDR &= ~(GPIO_SSP_CLK);
302 Ser4SSCR0 = 0;
303 Ser4SSCR0 = SSCR0_DataSize(16) + SSCR0_TI + SSCR0_SerClkDiv(8);
304 Ser4SSCR1 = SSCR1_SClkIactL + SSCR1_SClk1P + SSCR1_ExtClk;
305 Ser4SSCR0 |= SSCR0_SSE;
306 local_irq_restore(flags);
307
308 /* Enable the audio power */
309#ifdef CONFIG_H3600_HAL
310 h3600_audio_power(AUDIO_RATE_DEFAULT);
311#else
312 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
313 set_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
314 set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
315#endif
316
317 /* Wait for the UDA1341 to wake up */
318 mdelay(1); //FIXME - was removed by Perex - Why?
319
320 /* Initialize the UDA1341 internal state */
321 l3_open(sa11xx_uda1341->uda1341);
322
323 /* external clock configuration (after l3_open - regs must be initialized */
324 sa11xx_uda1341_set_samplerate(sa11xx_uda1341, sa11xx_uda1341->samplerate);
325
326 /* Wait for the UDA1341 to wake up */
327 set_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
328 mdelay(1);
329
330 /* make the left and right channels unswapped (flip the WS latch) */
331 Ser4SSDR = 0;
332
333#ifdef CONFIG_H3600_HAL
334 h3600_audio_mute(0);
335#else
336 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
337#endif
338}
339
340static void sa11xx_uda1341_audio_shutdown(sa11xx_uda1341_t *sa11xx_uda1341)
341{
342 /* mute on */
343#ifdef CONFIG_H3600_HAL
344 h3600_audio_mute(1);
345#else
346 set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
347#endif
348
349 /* disable the audio power and all signals leading to the audio chip */
350 l3_close(sa11xx_uda1341->uda1341);
351 Ser4SSCR0 = 0;
352 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
353
354 /* power off and mute off */
355 /* FIXME - is muting off necesary??? */
356#ifdef CONFIG_H3600_HAL
357 h3600_audio_power(0);
358 h3600_audio_mute(0);
359#else
360 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
361 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
362#endif
363}
364
365/* }}} */
366
367/* {{{ DMA staff */
368
369/*
370 * these are the address and sizes used to fill the xmit buffer
371 * so we can get a clock in record only mode
372 */
373#define FORCE_CLOCK_ADDR (dma_addr_t)FLUSH_BASE_PHYS
374#define FORCE_CLOCK_SIZE 4096 // was 2048
375
376// FIXME Why this value exactly - wrote comment
377#define DMA_BUF_SIZE 8176 /* <= MAX_DMA_SIZE from asm/arch-sa1100/dma.h */
378
379#ifdef HH_VERSION
380
381static int audio_dma_request(audio_stream_t *s, void (*callback)(void *, int))
382{
383 int ret;
384
385 ret = sa1100_request_dma(&s->dmach, s->id, s->dma_dev);
386 if (ret < 0) {
387 printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev);
388 return ret;
389 }
390 sa1100_dma_set_callback(s->dmach, callback);
391 return 0;
392}
393
394static inline void audio_dma_free(audio_stream_t *s)
395{
396 sa1100_free_dma(s->dmach);
397 s->dmach = -1;
398}
399
400#else
401
402static int audio_dma_request(audio_stream_t *s, void (*callback)(void *))
403{
404 int ret;
405
406 ret = sa1100_request_dma(s->dma_dev, s->id, callback, s, &s->dma_regs);
407 if (ret < 0)
408 printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev);
409 return ret;
410}
411
412static void audio_dma_free(audio_stream_t *s)
413{
414 sa1100_free_dma((s)->dma_regs);
415 (s)->dma_regs = 0;
416}
417
418#endif
419
420static u_int audio_get_dma_pos(audio_stream_t *s)
421{
422 snd_pcm_substream_t * substream = s->stream;
423 snd_pcm_runtime_t *runtime = substream->runtime;
424 unsigned int offset;
425 unsigned long flags;
426 dma_addr_t addr;
427
428 // this must be called w/ interrupts locked out see dma-sa1100.c in the kernel
429 spin_lock_irqsave(&s->dma_lock, flags);
430#ifdef HH_VERSION
431 sa1100_dma_get_current(s->dmach, NULL, &addr);
432#else
433 addr = sa1100_get_dma_pos((s)->dma_regs);
434#endif
435 offset = addr - runtime->dma_addr;
436 spin_unlock_irqrestore(&s->dma_lock, flags);
437
438 offset = bytes_to_frames(runtime,offset);
439 if (offset >= runtime->buffer_size)
440 offset = 0;
441
442 return offset;
443}
444
445/*
446 * this stops the dma and clears the dma ptrs
447 */
448static void audio_stop_dma(audio_stream_t *s)
449{
450 unsigned long flags;
451
452 spin_lock_irqsave(&s->dma_lock, flags);
453 s->active = 0;
454 s->period = 0;
455 /* this stops the dma channel and clears the buffer ptrs */
456#ifdef HH_VERSION
457 sa1100_dma_flush_all(s->dmach);
458#else
459 sa1100_clear_dma(s->dma_regs);
460#endif
461 spin_unlock_irqrestore(&s->dma_lock, flags);
462}
463
464static void audio_process_dma(audio_stream_t *s)
465{
466 snd_pcm_substream_t *substream = s->stream;
467 snd_pcm_runtime_t *runtime;
468 unsigned int dma_size;
469 unsigned int offset;
470 int ret;
471
472 /* we are requested to process synchronization DMA transfer */
473 if (s->tx_spin) {
474 snd_assert(s->stream_id == SNDRV_PCM_STREAM_PLAYBACK, return);
475 /* fill the xmit dma buffers and return */
476#ifdef HH_VERSION
477 sa1100_dma_set_spin(s->dmach, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE);
478#else
479 while (1) {
480 ret = sa1100_start_dma(s->dma_regs, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE);
481 if (ret)
482 return;
483 }
484#endif
485 return;
486 }
487
488 /* must be set here - only valid for running streams, not for forced_clock dma fills */
489 runtime = substream->runtime;
490 while (s->active && s->periods < runtime->periods) {
491 dma_size = frames_to_bytes(runtime, runtime->period_size);
492 if (s->old_offset) {
493 /* a little trick, we need resume from old position */
494 offset = frames_to_bytes(runtime, s->old_offset - 1);
495 s->old_offset = 0;
496 s->periods = 0;
497 s->period = offset / dma_size;
498 offset %= dma_size;
499 dma_size = dma_size - offset;
500 if (!dma_size)
501 continue; /* special case */
502 } else {
503 offset = dma_size * s->period;
504 snd_assert(dma_size <= DMA_BUF_SIZE, );
505 }
506#ifdef HH_VERSION
507 ret = sa1100_dma_queue_buffer(s->dmach, s, runtime->dma_addr + offset, dma_size);
508 if (ret)
509 return; //FIXME
510#else
511 ret = sa1100_start_dma((s)->dma_regs, runtime->dma_addr + offset, dma_size);
512 if (ret) {
513 printk(KERN_ERR "audio_process_dma: cannot queue DMA buffer (%i)\n", ret);
514 return;
515 }
516#endif
517
518 s->period++;
519 s->period %= runtime->periods;
520 s->periods++;
521 }
522}
523
524#ifdef HH_VERSION
525static void audio_dma_callback(void *data, int size)
526#else
527static void audio_dma_callback(void *data)
528#endif
529{
530 audio_stream_t *s = data;
531
532 /*
533 * If we are getting a callback for an active stream then we inform
534 * the PCM middle layer we've finished a period
535 */
536 if (s->active)
537 snd_pcm_period_elapsed(s->stream);
538
539 spin_lock(&s->dma_lock);
540 if (!s->tx_spin && s->periods > 0)
541 s->periods--;
542 audio_process_dma(s);
543 spin_unlock(&s->dma_lock);
544}
545
546/* }}} */
547
548/* {{{ PCM setting */
549
550/* {{{ trigger & timer */
551
552static int snd_sa11xx_uda1341_trigger(snd_pcm_substream_t * substream, int cmd)
553{
554 sa11xx_uda1341_t *chip = snd_pcm_substream_chip(substream);
555 int stream_id = substream->pstr->stream;
556 audio_stream_t *s = &chip->s[stream_id];
557 audio_stream_t *s1 = &chip->s[stream_id ^ 1];
558 int err = 0;
559
560 /* note local interrupts are already disabled in the midlevel code */
561 spin_lock(&s->dma_lock);
562 switch (cmd) {
563 case SNDRV_PCM_TRIGGER_START:
564 /* now we need to make sure a record only stream has a clock */
565 if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
566 /* we need to force fill the xmit DMA with zeros */
567 s1->tx_spin = 1;
568 audio_process_dma(s1);
569 }
570 /* this case is when you were recording then you turn on a
571 * playback stream so we stop (also clears it) the dma first,
572 * clear the sync flag and then we let it turned on
573 */
574 else {
575 s->tx_spin = 0;
576 }
577
578 /* requested stream startup */
579 s->active = 1;
580 audio_process_dma(s);
581 break;
582 case SNDRV_PCM_TRIGGER_STOP:
583 /* requested stream shutdown */
584 audio_stop_dma(s);
585
586 /*
587 * now we need to make sure a record only stream has a clock
588 * so if we're stopping a playback with an active capture
589 * we need to turn the 0 fill dma on for the xmit side
590 */
591 if (stream_id == SNDRV_PCM_STREAM_PLAYBACK && s1->active) {
592 /* we need to force fill the xmit DMA with zeros */
593 s->tx_spin = 1;
594 audio_process_dma(s);
595 }
596 /*
597 * we killed a capture only stream, so we should also kill
598 * the zero fill transmit
599 */
600 else {
601 if (s1->tx_spin) {
602 s1->tx_spin = 0;
603 audio_stop_dma(s1);
604 }
605 }
606
607 break;
608 case SNDRV_PCM_TRIGGER_SUSPEND:
609 s->active = 0;
610#ifdef HH_VERSION
611 sa1100_dma_stop(s->dmach);
612#else
613 //FIXME - DMA API
614#endif
615 s->old_offset = audio_get_dma_pos(s) + 1;
616#ifdef HH_VERSION
617 sa1100_dma_flush_all(s->dmach);
618#else
619 //FIXME - DMA API
620#endif
621 s->periods = 0;
622 break;
623 case SNDRV_PCM_TRIGGER_RESUME:
624 s->active = 1;
625 s->tx_spin = 0;
626 audio_process_dma(s);
627 if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
628 s1->tx_spin = 1;
629 audio_process_dma(s1);
630 }
631 break;
632 case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
633#ifdef HH_VERSION
634 sa1100_dma_stop(s->dmach);
635#else
636 //FIXME - DMA API
637#endif
638 s->active = 0;
639 if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) {
640 if (s1->active) {
641 s->tx_spin = 1;
642 s->old_offset = audio_get_dma_pos(s) + 1;
643#ifdef HH_VERSION
644 sa1100_dma_flush_all(s->dmach);
645#else
646 //FIXME - DMA API
647#endif
648 audio_process_dma(s);
649 }
650 } else {
651 if (s1->tx_spin) {
652 s1->tx_spin = 0;
653#ifdef HH_VERSION
654 sa1100_dma_flush_all(s1->dmach);
655#else
656 //FIXME - DMA API
657#endif
658 }
659 }
660 break;
661 case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
662 s->active = 1;
663 if (s->old_offset) {
664 s->tx_spin = 0;
665 audio_process_dma(s);
666 break;
667 }
668 if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
669 s1->tx_spin = 1;
670 audio_process_dma(s1);
671 }
672#ifdef HH_VERSION
673 sa1100_dma_resume(s->dmach);
674#else
675 //FIXME - DMA API
676#endif
677 break;
678 default:
679 err = -EINVAL;
680 break;
681 }
682 spin_unlock(&s->dma_lock);
683 return err;
684}
685
686static int snd_sa11xx_uda1341_prepare(snd_pcm_substream_t * substream)
687{
688 sa11xx_uda1341_t *chip = snd_pcm_substream_chip(substream);
689 snd_pcm_runtime_t *runtime = substream->runtime;
690 audio_stream_t *s = &chip->s[substream->pstr->stream];
691
692 /* set requested samplerate */
693 sa11xx_uda1341_set_samplerate(chip, runtime->rate);
694
695 /* set requestd format when available */
696 /* set FMT here !!! FIXME */
697
698 s->period = 0;
699 s->periods = 0;
700
701 return 0;
702}
703
704static snd_pcm_uframes_t snd_sa11xx_uda1341_pointer(snd_pcm_substream_t * substream)
705{
706 sa11xx_uda1341_t *chip = snd_pcm_substream_chip(substream);
707 return audio_get_dma_pos(&chip->s[substream->pstr->stream]);
708}
709
710/* }}} */
711
712static snd_pcm_hardware_t snd_sa11xx_uda1341_capture =
713{
714 .info = (SNDRV_PCM_INFO_INTERLEAVED |
715 SNDRV_PCM_INFO_BLOCK_TRANSFER |
716 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
717 SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
718 .formats = SNDRV_PCM_FMTBIT_S16_LE,
719 .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\
720 SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\
721 SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
722 SNDRV_PCM_RATE_KNOT),
723 .rate_min = 8000,
724 .rate_max = 48000,
725 .channels_min = 2,
726 .channels_max = 2,
727 .buffer_bytes_max = 64*1024,
728 .period_bytes_min = 64,
729 .period_bytes_max = DMA_BUF_SIZE,
730 .periods_min = 2,
731 .periods_max = 255,
732 .fifo_size = 0,
733};
734
735static snd_pcm_hardware_t snd_sa11xx_uda1341_playback =
736{
737 .info = (SNDRV_PCM_INFO_INTERLEAVED |
738 SNDRV_PCM_INFO_BLOCK_TRANSFER |
739 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
740 SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
741 .formats = SNDRV_PCM_FMTBIT_S16_LE,
742 .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\
743 SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\
744 SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
745 SNDRV_PCM_RATE_KNOT),
746 .rate_min = 8000,
747 .rate_max = 48000,
748 .channels_min = 2,
749 .channels_max = 2,
750 .buffer_bytes_max = 64*1024,
751 .period_bytes_min = 64,
752 .period_bytes_max = DMA_BUF_SIZE,
753 .periods_min = 2,
754 .periods_max = 255,
755 .fifo_size = 0,
756};
757
758static int snd_card_sa11xx_uda1341_open(snd_pcm_substream_t * substream)
759{
760 sa11xx_uda1341_t *chip = snd_pcm_substream_chip(substream);
761 snd_pcm_runtime_t *runtime = substream->runtime;
762 int stream_id = substream->pstr->stream;
763 int err;
764
765 chip->s[stream_id].stream = substream;
766
767 if (stream_id == SNDRV_PCM_STREAM_PLAYBACK)
768 runtime->hw = snd_sa11xx_uda1341_playback;
769 else
770 runtime->hw = snd_sa11xx_uda1341_capture;
771 if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
772 return err;
773 if ((err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates)) < 0)
774 return err;
775
776 return 0;
777}
778
779static int snd_card_sa11xx_uda1341_close(snd_pcm_substream_t * substream)
780{
781 sa11xx_uda1341_t *chip = snd_pcm_substream_chip(substream);
782
783 chip->s[substream->pstr->stream].stream = NULL;
784 return 0;
785}
786
787/* {{{ HW params & free */
788
789static int snd_sa11xx_uda1341_hw_params(snd_pcm_substream_t * substream,
790 snd_pcm_hw_params_t * hw_params)
791{
792
793 return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
794}
795
796static int snd_sa11xx_uda1341_hw_free(snd_pcm_substream_t * substream)
797{
798 return snd_pcm_lib_free_pages(substream);
799}
800
801/* }}} */
802
803static snd_pcm_ops_t snd_card_sa11xx_uda1341_playback_ops = {
804 .open = snd_card_sa11xx_uda1341_open,
805 .close = snd_card_sa11xx_uda1341_close,
806 .ioctl = snd_pcm_lib_ioctl,
807 .hw_params = snd_sa11xx_uda1341_hw_params,
808 .hw_free = snd_sa11xx_uda1341_hw_free,
809 .prepare = snd_sa11xx_uda1341_prepare,
810 .trigger = snd_sa11xx_uda1341_trigger,
811 .pointer = snd_sa11xx_uda1341_pointer,
812};
813
814static snd_pcm_ops_t snd_card_sa11xx_uda1341_capture_ops = {
815 .open = snd_card_sa11xx_uda1341_open,
816 .close = snd_card_sa11xx_uda1341_close,
817 .ioctl = snd_pcm_lib_ioctl,
818 .hw_params = snd_sa11xx_uda1341_hw_params,
819 .hw_free = snd_sa11xx_uda1341_hw_free,
820 .prepare = snd_sa11xx_uda1341_prepare,
821 .trigger = snd_sa11xx_uda1341_trigger,
822 .pointer = snd_sa11xx_uda1341_pointer,
823};
824
825static int __init snd_card_sa11xx_uda1341_pcm(sa11xx_uda1341_t *sa11xx_uda1341, int device)
826{
827 snd_pcm_t *pcm;
828 int err;
829
830 if ((err = snd_pcm_new(sa11xx_uda1341->card, "UDA1341 PCM", device, 1, 1, &pcm)) < 0)
831 return err;
832
833 /*
834 * this sets up our initial buffers and sets the dma_type to isa.
835 * isa works but I'm not sure why (or if) it's the right choice
836 * this may be too large, trying it for now
837 */
838 snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_ISA,
839 snd_pcm_dma_flags(0),
840 64*1024, 64*1024);
841
842 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_card_sa11xx_uda1341_playback_ops);
843 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_card_sa11xx_uda1341_capture_ops);
844 pcm->private_data = sa11xx_uda1341;
845 pcm->info_flags = 0;
846 strcpy(pcm->name, "UDA1341 PCM");
847
848 sa11xx_uda1341_audio_init(sa11xx_uda1341);
849
850 /* setup DMA controller */
851 audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK], audio_dma_callback);
852 audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE], audio_dma_callback);
853
854 sa11xx_uda1341->pcm = pcm;
855
856 return 0;
857}
858
859/* }}} */
860
861/* {{{ module init & exit */
862
863#ifdef CONFIG_PM
864
865static int snd_sa11xx_uda1341_suspend(snd_card_t *card, pm_message_t state)
866{
867 sa11xx_uda1341_t *chip = card->pm_private_data;
868
869 snd_pcm_suspend_all(chip->pcm);
870#ifdef HH_VERSION
871 sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach);
872 sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach);
873#else
874 //FIXME
875#endif
876 l3_command(chip->uda1341, CMD_SUSPEND, NULL);
877 sa11xx_uda1341_audio_shutdown(chip);
878 return 0;
879}
880
881static int snd_sa11xx_uda1341_resume(snd_card_t *card)
882{
883 sa11xx_uda1341_t *chip = card->pm_private_data;
884
885 sa11xx_uda1341_audio_init(chip);
886 l3_command(chip->uda1341, CMD_RESUME, NULL);
887#ifdef HH_VERSION
888 sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach);
889 sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach);
890#else
891 //FIXME
892#endif
893 return 0;
894}
895#endif /* COMFIG_PM */
896
897void snd_sa11xx_uda1341_free(snd_card_t *card)
898{
899 sa11xx_uda1341_t *chip = card->private_data;
900
901 audio_dma_free(&chip->s[SNDRV_PCM_STREAM_PLAYBACK]);
902 audio_dma_free(&chip->s[SNDRV_PCM_STREAM_CAPTURE]);
903 sa11xx_uda1341 = NULL;
904 card->private_data = NULL;
905 kfree(chip);
906}
907
908static int __init sa11xx_uda1341_init(void)
909{
910 int err;
911 snd_card_t *card;
912
913 if (!machine_is_h3xxx())
914 return -ENODEV;
915
916 /* register the soundcard */
917 card = snd_card_new(-1, id, THIS_MODULE, sizeof(sa11xx_uda1341_t));
918 if (card == NULL)
919 return -ENOMEM;
920
921 sa11xx_uda1341 = kcalloc(1, sizeof(*sa11xx_uda1341), GFP_KERNEL);
922 if (sa11xx_uda1341 == NULL)
923 return -ENOMEM;
924 spin_lock_init(&chip->s[0].dma_lock);
925 spin_lock_init(&chip->s[1].dma_lock);
926
927 card->private_data = (void *)sa11xx_uda1341;
928 card->private_free = snd_sa11xx_uda1341_free;
929
930 sa11xx_uda1341->card = card;
931 sa11xx_uda1341->samplerate = AUDIO_RATE_DEFAULT;
932
933 // mixer
934 if ((err = snd_chip_uda1341_mixer_new(sa11xx_uda1341->card, &sa11xx_uda1341->uda1341)))
935 goto nodev;
936
937 // PCM
938 if ((err = snd_card_sa11xx_uda1341_pcm(sa11xx_uda1341, 0)) < 0)
939 goto nodev;
940
941 snd_card_set_generic_pm_callback(card,
942 snd_sa11xx_uda1341_suspend, snd_sa11_uda1341_resume,
943 sa11xx_uda1341);
944
945 strcpy(card->driver, "UDA1341");
946 strcpy(card->shortname, "H3600 UDA1341TS");
947 sprintf(card->longname, "Compaq iPAQ H3600 with Philips UDA1341TS");
948
949 if ((err = snd_card_register(card)) == 0) {
950 printk( KERN_INFO "iPAQ audio support initialized\n" );
951 return 0;
952 }
953
954 nodev:
955 snd_card_free(card);
956 return err;
957}
958
959static void __exit sa11xx_uda1341_exit(void)
960{
961 snd_card_free(sa11xx_uda1341->card);
962}
963
964module_init(sa11xx_uda1341_init);
965module_exit(sa11xx_uda1341_exit);
966
967/* }}} */
968
969/*
970 * Local variables:
971 * indent-tabs-mode: t
972 * End:
973 */
generated by cgit v1.2.2 (git 2.25.0) at 2024-11-21 10:06:35 -0500