2 * sound/arm/omap-aic23.c
4 * Alsa Driver for AIC23 codec on OSK5912 platform board
6 * Copyright (C) 2005 Instituto Nokia de Tecnologia - INdT - Manaus Brazil
7 * Written by Daniel Petrini, David Cohen, Anderson Briglia
8 * {daniel.petrini, david.cohen, anderson.briglia}@indt.org.br
10 * Based on sa11xx-uda1341.c,
11 * Copyright (C) 2002 Tomas Kasparek <tomas.kasparek@seznam.cz>
13 * This program is free software; you can redistribute it and/or modify it
14 * under the terms of the GNU General Public License as published by the
15 * Free Software Foundation; either version 2 of the License, or (at your
16 * option) any later version.
18 * THIS SOFTWARE IS PROVIDED ``AS IS'' AND ANY EXPRESS OR IMPLIED
19 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
20 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN
21 * NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
22 * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT
23 * NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF
24 * USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
25 * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
26 * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
27 * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
29 * You should have received a copy of the GNU General Public License along
30 * with this program; if not, write to the Free Software Foundation, Inc.,
31 * 675 Mass Ave, Cambridge, MA 02139, USA.
35 * 2005-07-29 INdT Kernel Team - Alsa driver for omap osk. Creation of new
39 #include <linux/config.h>
40 #include <sound/driver.h>
41 #include <linux/module.h>
42 #include <linux/platform_device.h>
43 #include <linux/moduleparam.h>
44 #include <linux/init.h>
45 #include <linux/errno.h>
46 #include <linux/ioctl.h>
47 #include <linux/delay.h>
48 #include <linux/slab.h>
54 #include <asm/hardware.h>
55 #include <asm/mach-types.h>
56 #include <asm/arch/dma.h>
57 #include <asm/arch/aic23.h>
58 #include <asm/hardware/clock.h>
59 #include <asm/arch/mcbsp.h>
61 #include <sound/core.h>
62 #include <sound/pcm.h>
63 #include <sound/initval.h>
64 #include <sound/memalloc.h>
66 #include "omap-alsa-dma.h"
67 #include "omap-aic23.h"
72 #define ADEBUG() printk("XXX Alsa debug f:%s, l:%d\n", __FUNCTION__, __LINE__)
74 #define ADEBUG() /* nop */
77 /* Define to set the AIC23 as the master w.r.t McBSP */
81 * AUDIO related MACROS
83 #define DEFAULT_BITPERSAMPLE 16
84 #define AUDIO_RATE_DEFAULT 44100
85 #define AUDIO_MCBSP OMAP_MCBSP1
86 #define NUMBER_SAMPLE_RATES_SUPPORTED 10
89 MODULE_AUTHOR("Daniel Petrini, David Cohen, Anderson Briglia - INdT");
90 MODULE_LICENSE("GPL");
91 MODULE_DESCRIPTION("OMAP AIC23 driver for ALSA");
92 MODULE_SUPPORTED_DEVICE("{{AIC23,OMAP AIC23}}");
93 MODULE_ALIAS("omap_mcbsp.1");
95 static char *id = NULL;
96 MODULE_PARM_DESC(id, "OMAP OSK ALSA Driver for AIC23 chip.");
98 static struct snd_card_omap_aic23 *omap_aic23 = NULL;
100 static struct clk *aic23_mclk = 0;
102 struct sample_rate_rate_reg_info {
103 u8 control; /* SR3, SR2, SR1, SR0 and BOSR */
104 u8 divider; /* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */
108 * DAC USB-mode sampling rates (MCLK = 12 MHz)
109 * The rates and rate_reg_into MUST be in the same order
111 static unsigned int rates[] = {
112 4000, 8000, 16000, 22050,
116 static const struct sample_rate_rate_reg_info
117 rate_reg_info[NUMBER_SAMPLE_RATES_SUPPORTED] = {
118 {0x06, 1}, /* 4000 */
119 {0x06, 0}, /* 8000 */
120 {0x0C, 1}, /* 16000 */
121 {0x11, 1}, /* 22050 */
122 {0x00, 1}, /* 24000 */
123 {0x0C, 0}, /* 32000 */
124 {0x11, 0}, /* 44100 */
125 {0x00, 0}, /* 48000 */
126 {0x1F, 0}, /* 88200 */
127 {0x0E, 0}, /* 96000 */
131 * mcbsp configuration structure
133 static struct omap_mcbsp_reg_cfg initial_config_mcbsp = {
134 .spcr2 = FREE | FRST | GRST | XRST | XINTM(3),
135 .spcr1 = RINTM(3) | RRST,
136 .rcr2 = RPHASE | RFRLEN2(OMAP_MCBSP_WORD_8) |
137 RWDLEN2(OMAP_MCBSP_WORD_16) | RDATDLY(0),
138 .rcr1 = RFRLEN1(OMAP_MCBSP_WORD_8) | RWDLEN1(OMAP_MCBSP_WORD_16),
139 .xcr2 = XPHASE | XFRLEN2(OMAP_MCBSP_WORD_8) |
140 XWDLEN2(OMAP_MCBSP_WORD_16) | XDATDLY(0) | XFIG,
141 .xcr1 = XFRLEN1(OMAP_MCBSP_WORD_8) | XWDLEN1(OMAP_MCBSP_WORD_16),
142 .srgr1 = FWID(DEFAULT_BITPERSAMPLE - 1),
143 .srgr2 = GSYNC | CLKSP | FSGM | FPER(DEFAULT_BITPERSAMPLE * 2 - 1),
145 /* configure McBSP to be the I2S master */
146 .pcr0 = FSXM | FSRM | CLKXM | CLKRM | CLKXP | CLKRP,
148 /* configure McBSP to be the I2S slave */
149 .pcr0 = CLKXP | CLKRP,
150 #endif /* AIC23_MASTER */
153 static snd_pcm_hw_constraint_list_t hw_constraints_rates = {
154 .count = ARRAY_SIZE(rates),
161 * Codec/mcbsp init and configuration section
162 * codec dependent code.
166 * Sample rate changing
168 static void omap_aic23_set_samplerate(struct snd_card_omap_aic23
169 *omap_aic23, long rate)
174 /* Fix the rate if it has a wrong value */
177 else if (rate >= 88200)
179 else if (rate >= 48000)
181 else if (rate >= 44100)
183 else if (rate >= 32000)
185 else if (rate >= 24000)
187 else if (rate >= 22050)
189 else if (rate >= 16000)
191 else if (rate >= 8000)
196 /* Search for the right sample rate */
197 /* Verify what happens if the rate is not supported
198 * now it goes to 96Khz */
199 while ((rates[count] != rate) &&
200 (count < (NUMBER_SAMPLE_RATES_SUPPORTED - 1))) {
204 data = (rate_reg_info[count].divider << CLKIN_SHIFT) |
205 (rate_reg_info[count].control << BOSR_SHIFT) | USB_CLK_ON;
207 audio_aic23_write(SAMPLE_RATE_CONTROL_ADDR, data);
209 omap_aic23->samplerate = rate;
212 static inline void aic23_configure(void)
215 audio_aic23_write(RESET_CONTROL_ADDR, 0);
217 /* Initialize the AIC23 internal state */
219 /* Analog audio path control, DAC selected, delete INSEL_MIC for line in */
220 audio_aic23_write(ANALOG_AUDIO_CONTROL_ADDR, DEFAULT_ANALOG_AUDIO_CONTROL);
222 /* Digital audio path control, de-emphasis control 44.1kHz */
223 audio_aic23_write(DIGITAL_AUDIO_CONTROL_ADDR, DEEMP_44K);
225 /* Digital audio interface, master/slave mode, I2S, 16 bit */
227 audio_aic23_write(DIGITAL_AUDIO_FORMAT_ADDR,
228 MS_MASTER | IWL_16 | FOR_DSP);
230 audio_aic23_write(DIGITAL_AUDIO_FORMAT_ADDR, IWL_16 | FOR_DSP);
233 /* Enable digital interface */
234 audio_aic23_write(DIGITAL_INTERFACE_ACT_ADDR, ACT_ON);
238 static void omap_aic23_audio_init(struct snd_card_omap_aic23 *omap_aic23)
240 /* Setup DMA stuff */
241 omap_aic23->s[SNDRV_PCM_STREAM_PLAYBACK].id = "Alsa AIC23 out";
242 omap_aic23->s[SNDRV_PCM_STREAM_PLAYBACK].stream_id =
243 SNDRV_PCM_STREAM_PLAYBACK;
244 omap_aic23->s[SNDRV_PCM_STREAM_PLAYBACK].dma_dev =
247 omap_aic23->s[SNDRV_PCM_STREAM_CAPTURE].id = "Alsa AIC23 in";
248 omap_aic23->s[SNDRV_PCM_STREAM_CAPTURE].stream_id =
249 SNDRV_PCM_STREAM_CAPTURE;
250 omap_aic23->s[SNDRV_PCM_STREAM_CAPTURE].dma_dev =
253 /* configuring the McBSP */
254 omap_mcbsp_request(AUDIO_MCBSP);
256 /* if configured, then stop mcbsp */
257 omap_mcbsp_stop(AUDIO_MCBSP);
259 omap_mcbsp_config(AUDIO_MCBSP, &initial_config_mcbsp);
260 omap_mcbsp_start(AUDIO_MCBSP);
266 * Depends on omap-aic23-dma.c functions and (omap) dma.c
269 #define DMA_BUF_SIZE 1024 * 8
271 static int audio_dma_request(struct audio_stream *s,
272 void (*callback) (void *))
276 err = omap_request_sound_dma(s->dma_dev, s->id, s, &s->lch);
278 printk(KERN_ERR "unable to grab audio dma 0x%x\n",
283 static int audio_dma_free(struct audio_stream *s)
287 err = omap_free_sound_dma(s, &s->lch);
289 printk(KERN_ERR "Unable to free audio dma channels!\n");
294 * This function should calculate the current position of the dma in the
295 * buffer. It will help alsa middle layer to continue update the buffer.
296 * Its correctness is crucial for good functioning.
298 static u_int audio_get_dma_pos(struct audio_stream *s)
300 snd_pcm_substream_t *substream = s->stream;
301 snd_pcm_runtime_t *runtime = substream->runtime;
307 /* this must be called w/ interrupts locked as requested in dma.c */
308 spin_lock_irqsave(&s->dma_lock, flags);
310 /* For the current period let's see where we are */
311 count = omap_get_dma_src_addr_counter(s->lch[s->dma_q_head]);
313 spin_unlock_irqrestore(&s->dma_lock, flags);
315 /* Now, the position related to the end of that period */
316 offset = bytes_to_frames(runtime, s->offset) - bytes_to_frames(runtime, count);
318 if (offset >= runtime->buffer_size || offset < 0)
325 * this stops the dma and clears the dma ptrs
327 static void audio_stop_dma(struct audio_stream *s)
332 spin_lock_irqsave(&s->dma_lock, flags);
337 /* this stops the dma channel and clears the buffer ptrs */
338 omap_audio_stop_dma(s);
340 omap_clear_sound_dma(s);
342 spin_unlock_irqrestore(&s->dma_lock, flags);
346 * Main dma routine, requests dma according where you are in main alsa buffer
348 static void audio_process_dma(struct audio_stream *s)
350 snd_pcm_substream_t *substream = s->stream;
351 snd_pcm_runtime_t *runtime;
352 unsigned int dma_size;
356 runtime = substream->runtime;
358 dma_size = frames_to_bytes(runtime, runtime->period_size);
359 offset = dma_size * s->period;
360 snd_assert(dma_size <= DMA_BUF_SIZE,);
362 omap_start_sound_dma(s,
363 (dma_addr_t) runtime->dma_area +
367 "audio_process_dma: cannot queue DMA buffer (%i)\n",
373 s->period %= runtime->periods;
380 * This is called when dma IRQ occurs at the end of each transmited block
382 void audio_dma_callback(void *data)
384 struct audio_stream *s = data;
387 * If we are getting a callback for an active stream then we inform
388 * the PCM middle layer we've finished a period
391 snd_pcm_period_elapsed(s->stream);
393 spin_lock(&s->dma_lock);
394 if (s->periods > 0) {
397 audio_process_dma(s);
398 spin_unlock(&s->dma_lock);
404 * PCM settings and callbacks
407 static int snd_omap_aic23_trigger(snd_pcm_substream_t * substream, int cmd)
409 struct snd_card_omap_aic23 *chip =
410 snd_pcm_substream_chip(substream);
411 int stream_id = substream->pstr->stream;
412 struct audio_stream *s = &chip->s[stream_id];
416 /* note local interrupts are already disabled in the midlevel code */
417 spin_lock(&s->dma_lock);
419 case SNDRV_PCM_TRIGGER_START:
420 /* requested stream startup */
422 audio_process_dma(s);
424 case SNDRV_PCM_TRIGGER_STOP:
425 /* requested stream shutdown */
432 spin_unlock(&s->dma_lock);
437 static int snd_omap_aic23_prepare(snd_pcm_substream_t * substream)
439 struct snd_card_omap_aic23 *chip =
440 snd_pcm_substream_chip(substream);
441 snd_pcm_runtime_t *runtime = substream->runtime;
442 struct audio_stream *s = &chip->s[substream->pstr->stream];
444 /* set requested samplerate */
445 omap_aic23_set_samplerate(chip, runtime->rate);
453 static snd_pcm_uframes_t snd_omap_aic23_pointer(snd_pcm_substream_t *
456 struct snd_card_omap_aic23 *chip =
457 snd_pcm_substream_chip(substream);
459 return audio_get_dma_pos(&chip->s[substream->pstr->stream]);
462 /* Hardware capabilities */
464 static snd_pcm_hardware_t snd_omap_aic23_capture = {
465 .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
466 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID),
467 .formats = (SNDRV_PCM_FMTBIT_S16_LE),
468 .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
469 SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
470 SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
471 SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
472 SNDRV_PCM_RATE_KNOT),
477 .buffer_bytes_max = 128 * 1024,
478 .period_bytes_min = 32,
479 .period_bytes_max = 8 * 1024,
485 static snd_pcm_hardware_t snd_omap_aic23_playback = {
486 .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
487 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID),
488 .formats = (SNDRV_PCM_FMTBIT_S16_LE),
489 .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
490 SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
491 SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
492 SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
493 SNDRV_PCM_RATE_KNOT),
498 .buffer_bytes_max = 128 * 1024,
499 .period_bytes_min = 32,
500 .period_bytes_max = 8 * 1024,
506 static int snd_card_omap_aic23_open(snd_pcm_substream_t * substream)
508 struct snd_card_omap_aic23 *chip =
509 snd_pcm_substream_chip(substream);
510 snd_pcm_runtime_t *runtime = substream->runtime;
511 int stream_id = substream->pstr->stream;
515 chip->s[stream_id].stream = substream;
517 omap_aic23_clock_on();
519 if (stream_id == SNDRV_PCM_STREAM_PLAYBACK)
520 runtime->hw = snd_omap_aic23_playback;
522 runtime->hw = snd_omap_aic23_capture;
524 snd_pcm_hw_constraint_integer(runtime,
525 SNDRV_PCM_HW_PARAM_PERIODS)) <
529 snd_pcm_hw_constraint_list(runtime, 0,
530 SNDRV_PCM_HW_PARAM_RATE,
531 &hw_constraints_rates)) < 0)
537 static int snd_card_omap_aic23_close(snd_pcm_substream_t * substream)
539 struct snd_card_omap_aic23 *chip =
540 snd_pcm_substream_chip(substream);
543 omap_aic23_clock_off();
544 chip->s[substream->pstr->stream].stream = NULL;
549 /* HW params & free */
551 static int snd_omap_aic23_hw_params(snd_pcm_substream_t * substream,
552 snd_pcm_hw_params_t * hw_params)
554 return snd_pcm_lib_malloc_pages(substream,
555 params_buffer_bytes(hw_params));
558 static int snd_omap_aic23_hw_free(snd_pcm_substream_t * substream)
560 return snd_pcm_lib_free_pages(substream);
565 static snd_pcm_ops_t snd_card_omap_aic23_playback_ops = {
566 .open = snd_card_omap_aic23_open,
567 .close = snd_card_omap_aic23_close,
568 .ioctl = snd_pcm_lib_ioctl,
569 .hw_params = snd_omap_aic23_hw_params,
570 .hw_free = snd_omap_aic23_hw_free,
571 .prepare = snd_omap_aic23_prepare,
572 .trigger = snd_omap_aic23_trigger,
573 .pointer = snd_omap_aic23_pointer,
576 static snd_pcm_ops_t snd_card_omap_aic23_capture_ops = {
577 .open = snd_card_omap_aic23_open,
578 .close = snd_card_omap_aic23_close,
579 .ioctl = snd_pcm_lib_ioctl,
580 .hw_params = snd_omap_aic23_hw_params,
581 .hw_free = snd_omap_aic23_hw_free,
582 .prepare = snd_omap_aic23_prepare,
583 .trigger = snd_omap_aic23_trigger,
584 .pointer = snd_omap_aic23_pointer,
588 * Alsa init and exit section
590 * Inits pcm alsa structures, allocate the alsa buffer, suspend, resume
592 static int __init snd_card_omap_aic23_pcm(struct snd_card_omap_aic23
593 *omap_aic23, int device)
600 snd_pcm_new(omap_aic23->card, "AIC23 PCM", device, 1, 1,
604 /* sets up initial buffer with continuous allocation */
605 snd_pcm_lib_preallocate_pages_for_all(pcm,
606 SNDRV_DMA_TYPE_CONTINUOUS,
607 snd_dma_continuous_data
609 128 * 1024, 128 * 1024);
611 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
612 &snd_card_omap_aic23_playback_ops);
613 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
614 &snd_card_omap_aic23_capture_ops);
615 pcm->private_data = omap_aic23;
617 strcpy(pcm->name, "omap aic23 pcm");
619 omap_aic23_audio_init(omap_aic23);
621 /* setup DMA controller */
622 audio_dma_request(&omap_aic23->s[SNDRV_PCM_STREAM_PLAYBACK],
624 audio_dma_request(&omap_aic23->s[SNDRV_PCM_STREAM_CAPTURE],
627 omap_aic23->pcm = pcm;
635 static int snd_omap_aic23_suspend(snd_card_t * card, pm_message_t state)
637 struct snd_card_omap_aic23 *chip = card->pm_private_data;
640 if (chip->card->power_state != SNDRV_CTL_POWER_D3hot) {
641 snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot);
642 snd_pcm_suspend_all(chip->pcm);
643 /* Mutes and turn clock off */
644 omap_aic23_clock_off();
645 snd_omap_suspend_mixer();
652 * Prepare hardware for resume
654 static int snd_omap_aic23_resume(snd_card_t * card)
656 struct snd_card_omap_aic23 *chip = card->pm_private_data;
659 if (chip->card->power_state != SNDRV_CTL_POWER_D0) {
660 snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0);
661 omap_aic23_clock_on();
662 snd_omap_resume_mixer();
669 * Driver suspend/resume - calls alsa functions. Some hints from aaci.c
671 static int omap_aic23_suspend(struct device *dev, pm_message_t state)
673 snd_card_t *card = dev_get_drvdata(dev);
675 if (card->power_state != SNDRV_CTL_POWER_D3hot) {
676 snd_omap_aic23_suspend(card, PMSG_SUSPEND);
681 static int omap_aic23_resume(struct device *dev)
683 snd_card_t *card = dev_get_drvdata(dev);
685 if (card->power_state != SNDRV_CTL_POWER_D0) {
686 snd_omap_aic23_resume(card);
692 #define snd_omap_aic23_suspend NULL
693 #define snd_omap_aic23_resume NULL
694 #define omap_aic23_suspend NULL
695 #define omap_aic23_resume NULL
697 #endif /* CONFIG_PM */
701 void snd_omap_aic23_free(snd_card_t * card)
703 struct snd_card_omap_aic23 *chip = card->private_data;
707 * Turn off codec after it is done.
708 * Can't do it immediately, since it may still have
711 set_current_state(TASK_INTERRUPTIBLE);
714 omap_mcbsp_stop(AUDIO_MCBSP);
715 omap_mcbsp_free(AUDIO_MCBSP);
717 audio_aic23_write(RESET_CONTROL_ADDR, 0);
718 audio_aic23_write(POWER_DOWN_CONTROL_ADDR, 0xff);
720 audio_dma_free(&chip->s[SNDRV_PCM_STREAM_PLAYBACK]);
721 audio_dma_free(&chip->s[SNDRV_PCM_STREAM_CAPTURE]);
725 * Omap MCBSP clock configuration
727 * Here we have some functions that allows clock to be enabled and
728 * disabled only when needed. Besides doing clock configuration
729 * it allows turn on/turn off audio when necessary.
731 #define CODEC_CLOCK 12000000
732 #define AUDIO_RATE_DEFAULT 44100
735 * Do clock framework mclk search
737 static __init void omap_aic23_clock_setup(void)
739 aic23_mclk = clk_get(0, "mclk");
743 * Do some sanity check, set clock rate, starts it and
744 * turn codec audio on
746 int omap_aic23_clock_on(void)
748 if (clk_get_usecount(aic23_mclk) > 0) {
749 /* MCLK is already in use */
751 "MCLK in use at %d Hz. We change it to %d Hz\n",
752 (uint) clk_get_rate(aic23_mclk),
756 if (clk_set_rate(aic23_mclk, CODEC_CLOCK)) {
758 "Cannot set MCLK for AIC23 CODEC\n");
765 "MCLK = %d [%d], usecount = %d\n",
766 (uint) clk_get_rate(aic23_mclk), CODEC_CLOCK,
767 clk_get_usecount(aic23_mclk));
769 /* Now turn the audio on */
770 audio_aic23_write(POWER_DOWN_CONTROL_ADDR,
771 ~DEVICE_POWER_OFF & ~OUT_OFF & ~DAC_OFF &
772 ~ADC_OFF & ~MIC_OFF & ~LINE_OFF);
777 * Do some sanity check, turn clock off and then turn
780 int omap_aic23_clock_off(void)
782 if (clk_get_usecount(aic23_mclk) > 0) {
783 if (clk_get_rate(aic23_mclk) != CODEC_CLOCK) {
785 "MCLK for audio should be %d Hz. But is %d Hz\n",
786 (uint) clk_get_rate(aic23_mclk),
790 clk_unuse(aic23_mclk);
793 audio_aic23_write(POWER_DOWN_CONTROL_ADDR,
794 DEVICE_POWER_OFF | OUT_OFF | DAC_OFF |
795 ADC_OFF | MIC_OFF | LINE_OFF);
799 /* module init & exit */
802 * Inits alsa soudcard structure
804 static int __init snd_omap_aic23_probe(struct device *dev)
810 /* gets clock from clock framework */
811 omap_aic23_clock_setup();
813 /* register the soundcard */
814 card = snd_card_new(-1, id, THIS_MODULE, sizeof(omap_aic23));
818 omap_aic23 = kcalloc(1, sizeof(*omap_aic23), GFP_KERNEL);
819 if (omap_aic23 == NULL)
822 card->private_data = (void *) omap_aic23;
823 card->private_free = snd_omap_aic23_free;
825 omap_aic23->card = card;
826 omap_aic23->samplerate = AUDIO_RATE_DEFAULT;
828 spin_lock_init(&omap_aic23->s[0].dma_lock);
829 spin_lock_init(&omap_aic23->s[1].dma_lock);
832 if ((err = snd_omap_mixer(omap_aic23)) < 0)
836 if ((err = snd_card_omap_aic23_pcm(omap_aic23, 0)) < 0)
839 snd_card_set_pm_callback(card, snd_omap_aic23_suspend,
840 snd_omap_aic23_resume, omap_aic23);
842 strcpy(card->driver, "AIC23");
843 strcpy(card->shortname, "OSK AIC23");
844 sprintf(card->longname, "OMAP OSK with AIC23");
846 snd_omap_init_mixer();
848 if ((err = snd_card_register(card)) == 0) {
849 printk(KERN_INFO "OSK audio support initialized\n");
850 dev_set_drvdata(dev, card);
855 snd_omap_aic23_free(card);
860 static int snd_omap_aic23_remove(struct device *dev)
862 snd_card_t *card = dev_get_drvdata(dev);
863 struct snd_card_omap_aic23 *chip = card->private_data;
868 card->private_data = NULL;
871 dev_set_drvdata(dev, NULL);
877 static struct device_driver omap_alsa_driver = {
878 .name = "omap_mcbsp",
879 .bus = &platform_bus_type,
880 .probe = snd_omap_aic23_probe,
881 .remove = snd_omap_aic23_remove,
882 .suspend = omap_aic23_suspend,
883 .resume = omap_aic23_resume,
886 static int __init omap_aic23_init(void)
891 err = driver_register(&omap_alsa_driver);
896 static void __exit omap_aic23_exit(void)
900 driver_unregister(&omap_alsa_driver);
903 module_init(omap_aic23_init);
904 module_exit(omap_aic23_exit);