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/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.
165 extern int tlv320aic23_write_value(u8 reg, u16 value);
167 /* TLV320AIC23 is a write only device */
168 __inline__ void audio_aic23_write(u8 address, u16 data)
170 tlv320aic23_write_value(address, data);
174 * Sample rate changing
176 static void omap_aic23_set_samplerate(struct snd_card_omap_aic23
177 *omap_aic23, long rate)
182 /* Fix the rate if it has a wrong value */
185 else if (rate >= 88200)
187 else if (rate >= 48000)
189 else if (rate >= 44100)
191 else if (rate >= 32000)
193 else if (rate >= 24000)
195 else if (rate >= 22050)
197 else if (rate >= 16000)
199 else if (rate >= 8000)
204 /* Search for the right sample rate */
205 /* Verify what happens if the rate is not supported
206 * now it goes to 96Khz */
207 while ((rates[count] != rate) &&
208 (count < (NUMBER_SAMPLE_RATES_SUPPORTED - 1))) {
212 data = (rate_reg_info[count].divider << CLKIN_SHIFT) |
213 (rate_reg_info[count].control << BOSR_SHIFT) | USB_CLK_ON;
215 audio_aic23_write(SAMPLE_RATE_CONTROL_ADDR, data);
217 omap_aic23->samplerate = rate;
220 static inline void aic23_configure(void)
223 audio_aic23_write(RESET_CONTROL_ADDR, 0);
225 /* Initialize the AIC23 internal state */
227 /* Analog audio path control, DAC selected, delete INSEL_MIC for line in */
228 audio_aic23_write(ANALOG_AUDIO_CONTROL_ADDR, DEFAULT_ANALOG_AUDIO_CONTROL);
230 /* Digital audio path control, de-emphasis control 44.1kHz */
231 audio_aic23_write(DIGITAL_AUDIO_CONTROL_ADDR, DEEMP_44K);
233 /* Digital audio interface, master/slave mode, I2S, 16 bit */
235 audio_aic23_write(DIGITAL_AUDIO_FORMAT_ADDR,
236 MS_MASTER | IWL_16 | FOR_DSP);
238 audio_aic23_write(DIGITAL_AUDIO_FORMAT_ADDR, IWL_16 | FOR_DSP);
241 /* Enable digital interface */
242 audio_aic23_write(DIGITAL_INTERFACE_ACT_ADDR, ACT_ON);
246 static void omap_aic23_audio_init(struct snd_card_omap_aic23 *omap_aic23)
248 /* Setup DMA stuff */
249 omap_aic23->s[SNDRV_PCM_STREAM_PLAYBACK].id = "Alsa AIC23 out";
250 omap_aic23->s[SNDRV_PCM_STREAM_PLAYBACK].stream_id =
251 SNDRV_PCM_STREAM_PLAYBACK;
252 omap_aic23->s[SNDRV_PCM_STREAM_PLAYBACK].dma_dev =
255 omap_aic23->s[SNDRV_PCM_STREAM_CAPTURE].id = "Alsa AIC23 in";
256 omap_aic23->s[SNDRV_PCM_STREAM_CAPTURE].stream_id =
257 SNDRV_PCM_STREAM_CAPTURE;
258 omap_aic23->s[SNDRV_PCM_STREAM_CAPTURE].dma_dev =
261 /* configuring the McBSP */
262 omap_mcbsp_request(AUDIO_MCBSP);
264 /* if configured, then stop mcbsp */
265 omap_mcbsp_stop(AUDIO_MCBSP);
267 omap_mcbsp_config(AUDIO_MCBSP, &initial_config_mcbsp);
268 omap_mcbsp_start(AUDIO_MCBSP);
274 * Depends on omap-aic23-dma.c functions and (omap) dma.c
277 #define DMA_BUF_SIZE 1024 * 8
279 static int audio_dma_request(struct audio_stream *s,
280 void (*callback) (void *))
284 err = omap_request_sound_dma(s->dma_dev, s->id, s, &s->lch);
286 printk(KERN_ERR "unable to grab audio dma 0x%x\n",
291 static int audio_dma_free(struct audio_stream *s)
295 err = omap_free_sound_dma(s, &s->lch);
297 printk(KERN_ERR "Unable to free audio dma channels!\n");
302 * This function should calculate the current position of the dma in the
303 * buffer. It will help alsa middle layer to continue update the buffer.
304 * Its correctness is crucial for good functioning.
306 static u_int audio_get_dma_pos(struct audio_stream *s)
308 snd_pcm_substream_t *substream = s->stream;
309 snd_pcm_runtime_t *runtime = substream->runtime;
315 /* this must be called w/ interrupts locked as requested in dma.c */
316 spin_lock_irqsave(&s->dma_lock, flags);
318 /* For the current period let's see where we are */
319 count = omap_get_dma_src_addr_counter(s->lch[s->dma_q_head]);
321 spin_unlock_irqrestore(&s->dma_lock, flags);
323 /* Now, the position related to the end of that period */
324 offset = bytes_to_frames(runtime, s->offset) - bytes_to_frames(runtime, count);
326 if (offset >= runtime->buffer_size || offset < 0)
333 * this stops the dma and clears the dma ptrs
335 static void audio_stop_dma(struct audio_stream *s)
340 spin_lock_irqsave(&s->dma_lock, flags);
345 /* this stops the dma channel and clears the buffer ptrs */
346 omap_audio_stop_dma(s);
348 omap_clear_sound_dma(s);
350 spin_unlock_irqrestore(&s->dma_lock, flags);
354 * Main dma routine, requests dma according where you are in main alsa buffer
356 static void audio_process_dma(struct audio_stream *s)
358 snd_pcm_substream_t *substream = s->stream;
359 snd_pcm_runtime_t *runtime;
360 unsigned int dma_size;
364 runtime = substream->runtime;
366 dma_size = frames_to_bytes(runtime, runtime->period_size);
367 offset = dma_size * s->period;
368 snd_assert(dma_size <= DMA_BUF_SIZE,);
370 omap_start_sound_dma(s,
371 (dma_addr_t) runtime->dma_area +
375 "audio_process_dma: cannot queue DMA buffer (%i)\n",
381 s->period %= runtime->periods;
388 * This is called when dma IRQ occurs at the end of each transmited block
390 void audio_dma_callback(void *data)
392 struct audio_stream *s = data;
395 * If we are getting a callback for an active stream then we inform
396 * the PCM middle layer we've finished a period
399 snd_pcm_period_elapsed(s->stream);
401 spin_lock(&s->dma_lock);
402 if (s->periods > 0) {
405 audio_process_dma(s);
406 spin_unlock(&s->dma_lock);
412 * PCM settings and callbacks
415 static int snd_omap_aic23_trigger(snd_pcm_substream_t * substream, int cmd)
417 struct snd_card_omap_aic23 *chip =
418 snd_pcm_substream_chip(substream);
419 int stream_id = substream->pstr->stream;
420 struct audio_stream *s = &chip->s[stream_id];
424 /* note local interrupts are already disabled in the midlevel code */
425 spin_lock(&s->dma_lock);
427 case SNDRV_PCM_TRIGGER_START:
428 /* requested stream startup */
430 audio_process_dma(s);
432 case SNDRV_PCM_TRIGGER_STOP:
433 /* requested stream shutdown */
440 spin_unlock(&s->dma_lock);
445 static int snd_omap_aic23_prepare(snd_pcm_substream_t * substream)
447 struct snd_card_omap_aic23 *chip =
448 snd_pcm_substream_chip(substream);
449 snd_pcm_runtime_t *runtime = substream->runtime;
450 struct audio_stream *s = &chip->s[substream->pstr->stream];
452 /* set requested samplerate */
453 omap_aic23_set_samplerate(chip, runtime->rate);
461 static snd_pcm_uframes_t snd_omap_aic23_pointer(snd_pcm_substream_t *
464 struct snd_card_omap_aic23 *chip =
465 snd_pcm_substream_chip(substream);
467 return audio_get_dma_pos(&chip->s[substream->pstr->stream]);
470 /* Hardware capabilities */
472 static snd_pcm_hardware_t snd_omap_aic23_capture = {
473 .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
474 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID),
475 .formats = (SNDRV_PCM_FMTBIT_S16_LE),
476 .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
477 SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
478 SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
479 SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
480 SNDRV_PCM_RATE_KNOT),
485 .buffer_bytes_max = 128 * 1024,
486 .period_bytes_min = 32,
487 .period_bytes_max = 8 * 1024,
493 static snd_pcm_hardware_t snd_omap_aic23_playback = {
494 .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
495 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID),
496 .formats = (SNDRV_PCM_FMTBIT_S16_LE),
497 .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
498 SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
499 SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
500 SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
501 SNDRV_PCM_RATE_KNOT),
506 .buffer_bytes_max = 128 * 1024,
507 .period_bytes_min = 32,
508 .period_bytes_max = 8 * 1024,
514 static int snd_card_omap_aic23_open(snd_pcm_substream_t * substream)
516 struct snd_card_omap_aic23 *chip =
517 snd_pcm_substream_chip(substream);
518 snd_pcm_runtime_t *runtime = substream->runtime;
519 int stream_id = substream->pstr->stream;
523 chip->s[stream_id].stream = substream;
525 omap_aic23_clock_on();
527 if (stream_id == SNDRV_PCM_STREAM_PLAYBACK)
528 runtime->hw = snd_omap_aic23_playback;
530 runtime->hw = snd_omap_aic23_capture;
532 snd_pcm_hw_constraint_integer(runtime,
533 SNDRV_PCM_HW_PARAM_PERIODS)) <
537 snd_pcm_hw_constraint_list(runtime, 0,
538 SNDRV_PCM_HW_PARAM_RATE,
539 &hw_constraints_rates)) < 0)
545 static int snd_card_omap_aic23_close(snd_pcm_substream_t * substream)
547 struct snd_card_omap_aic23 *chip =
548 snd_pcm_substream_chip(substream);
551 omap_aic23_clock_off();
552 chip->s[substream->pstr->stream].stream = NULL;
557 /* HW params & free */
559 static int snd_omap_aic23_hw_params(snd_pcm_substream_t * substream,
560 snd_pcm_hw_params_t * hw_params)
562 return snd_pcm_lib_malloc_pages(substream,
563 params_buffer_bytes(hw_params));
566 static int snd_omap_aic23_hw_free(snd_pcm_substream_t * substream)
568 return snd_pcm_lib_free_pages(substream);
573 static snd_pcm_ops_t snd_card_omap_aic23_playback_ops = {
574 .open = snd_card_omap_aic23_open,
575 .close = snd_card_omap_aic23_close,
576 .ioctl = snd_pcm_lib_ioctl,
577 .hw_params = snd_omap_aic23_hw_params,
578 .hw_free = snd_omap_aic23_hw_free,
579 .prepare = snd_omap_aic23_prepare,
580 .trigger = snd_omap_aic23_trigger,
581 .pointer = snd_omap_aic23_pointer,
584 static snd_pcm_ops_t snd_card_omap_aic23_capture_ops = {
585 .open = snd_card_omap_aic23_open,
586 .close = snd_card_omap_aic23_close,
587 .ioctl = snd_pcm_lib_ioctl,
588 .hw_params = snd_omap_aic23_hw_params,
589 .hw_free = snd_omap_aic23_hw_free,
590 .prepare = snd_omap_aic23_prepare,
591 .trigger = snd_omap_aic23_trigger,
592 .pointer = snd_omap_aic23_pointer,
596 * Alsa init and exit section
598 * Inits pcm alsa structures, allocate the alsa buffer, suspend, resume
600 static int __init snd_card_omap_aic23_pcm(struct snd_card_omap_aic23
601 *omap_aic23, int device)
608 snd_pcm_new(omap_aic23->card, "AIC23 PCM", device, 1, 1,
612 /* sets up initial buffer with continuous allocation */
613 snd_pcm_lib_preallocate_pages_for_all(pcm,
614 SNDRV_DMA_TYPE_CONTINUOUS,
615 snd_dma_continuous_data
617 128 * 1024, 128 * 1024);
619 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
620 &snd_card_omap_aic23_playback_ops);
621 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
622 &snd_card_omap_aic23_capture_ops);
623 pcm->private_data = omap_aic23;
625 strcpy(pcm->name, "omap aic23 pcm");
627 omap_aic23_audio_init(omap_aic23);
629 /* setup DMA controller */
630 audio_dma_request(&omap_aic23->s[SNDRV_PCM_STREAM_PLAYBACK],
632 audio_dma_request(&omap_aic23->s[SNDRV_PCM_STREAM_CAPTURE],
635 omap_aic23->pcm = pcm;
643 static int snd_omap_aic23_suspend(snd_card_t * card, pm_message_t state)
645 struct snd_card_omap_aic23 *chip = card->pm_private_data;
648 if (chip->card->power_state != SNDRV_CTL_POWER_D3hot) {
649 snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot);
650 snd_pcm_suspend_all(chip->pcm);
651 /* Mutes and turn clock off */
652 omap_aic23_clock_off();
653 snd_omap_suspend_mixer();
660 * Prepare hardware for resume
662 static int snd_omap_aic23_resume(snd_card_t * card)
664 struct snd_card_omap_aic23 *chip = card->pm_private_data;
667 if (chip->card->power_state != SNDRV_CTL_POWER_D0) {
668 snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0);
669 omap_aic23_clock_on();
670 snd_omap_resume_mixer();
677 * Driver suspend/resume - calls alsa functions. Some hints from aaci.c
679 static int omap_aic23_suspend(struct device *dev, pm_message_t state, u32 level)
681 snd_card_t *card = dev_get_drvdata(dev);
683 if (card->power_state != SNDRV_CTL_POWER_D3hot) {
684 snd_omap_aic23_suspend(card, 0);
689 static int omap_aic23_resume(struct device *dev, u32 level)
691 snd_card_t *card = dev_get_drvdata(dev);
693 if (card->power_state != SNDRV_CTL_POWER_D0) {
694 snd_omap_aic23_resume(card);
700 #define snd_omap_aic23_suspend NULL
701 #define snd_omap_aic23_resume NULL
702 #define omap_aic23_suspend NULL
703 #define omap_aic23_resume NULL
705 #endif /* CONFIG_PM */
709 void snd_omap_aic23_free(snd_card_t * card)
711 struct snd_card_omap_aic23 *chip = card->private_data;
715 * Turn off codec after it is done.
716 * Can't do it immediately, since it may still have
719 set_current_state(TASK_INTERRUPTIBLE);
722 omap_mcbsp_stop(AUDIO_MCBSP);
723 omap_mcbsp_free(AUDIO_MCBSP);
725 audio_aic23_write(RESET_CONTROL_ADDR, 0);
726 audio_aic23_write(POWER_DOWN_CONTROL_ADDR, 0xff);
728 audio_dma_free(&chip->s[SNDRV_PCM_STREAM_PLAYBACK]);
729 audio_dma_free(&chip->s[SNDRV_PCM_STREAM_CAPTURE]);
733 * Omap MCBSP clock configuration
735 * Here we have some functions that allows clock to be enabled and
736 * disabled only when needed. Besides doing clock configuration
737 * it allows turn on/turn off audio when necessary.
739 #define CODEC_CLOCK 12000000
740 #define AUDIO_RATE_DEFAULT 44100
743 * Do clock framework mclk search
745 static __init void omap_aic23_clock_setup(void)
747 aic23_mclk = clk_get(0, "mclk");
751 * Do some sanity check, set clock rate, starts it and
752 * turn codec audio on
754 int omap_aic23_clock_on(void)
756 if (clk_get_usecount(aic23_mclk) > 0) {
757 /* MCLK is already in use */
759 "MCLK in use at %d Hz. We change it to %d Hz\n",
760 (uint) clk_get_rate(aic23_mclk),
764 if (clk_set_rate(aic23_mclk, CODEC_CLOCK)) {
766 "Cannot set MCLK for AIC23 CODEC\n");
773 "MCLK = %d [%d], usecount = %d\n",
774 (uint) clk_get_rate(aic23_mclk), CODEC_CLOCK,
775 clk_get_usecount(aic23_mclk));
777 /* Now turn the audio on */
778 audio_aic23_write(POWER_DOWN_CONTROL_ADDR,
779 ~DEVICE_POWER_OFF & ~OUT_OFF & ~DAC_OFF &
780 ~ADC_OFF & ~MIC_OFF & ~LINE_OFF);
785 * Do some sanity check, turn clock off and then turn
788 int omap_aic23_clock_off(void)
790 if (clk_get_usecount(aic23_mclk) > 0) {
791 if (clk_get_rate(aic23_mclk) != CODEC_CLOCK) {
793 "MCLK for audio should be %d Hz. But is %d Hz\n",
794 (uint) clk_get_rate(aic23_mclk),
798 clk_unuse(aic23_mclk);
801 audio_aic23_write(POWER_DOWN_CONTROL_ADDR,
802 DEVICE_POWER_OFF | OUT_OFF | DAC_OFF |
803 ADC_OFF | MIC_OFF | LINE_OFF);
807 /* module init & exit */
810 * Inits alsa soudcard structure
812 static int __init snd_omap_aic23_probe(struct device *dev)
818 /* gets clock from clock framework */
819 omap_aic23_clock_setup();
821 /* register the soundcard */
822 card = snd_card_new(-1, id, THIS_MODULE, sizeof(omap_aic23));
826 omap_aic23 = kcalloc(1, sizeof(*omap_aic23), GFP_KERNEL);
827 if (omap_aic23 == NULL)
830 card->private_data = (void *) omap_aic23;
831 card->private_free = snd_omap_aic23_free;
833 omap_aic23->card = card;
834 omap_aic23->samplerate = AUDIO_RATE_DEFAULT;
836 spin_lock_init(&omap_aic23->s[0].dma_lock);
837 spin_lock_init(&omap_aic23->s[1].dma_lock);
840 if ((err = snd_omap_mixer(omap_aic23)) < 0)
844 if ((err = snd_card_omap_aic23_pcm(omap_aic23, 0)) < 0)
847 snd_card_set_pm_callback(card, snd_omap_aic23_suspend,
848 snd_omap_aic23_resume, omap_aic23);
850 strcpy(card->driver, "AIC23");
851 strcpy(card->shortname, "OSK AIC23");
852 sprintf(card->longname, "OMAP OSK with AIC23");
854 snd_omap_init_mixer();
856 if ((err = snd_card_register(card)) == 0) {
857 printk(KERN_INFO "OSK audio support initialized\n");
858 dev_set_drvdata(dev, card);
863 snd_omap_aic23_free(card);
868 static int snd_omap_aic23_remove(struct device *dev)
870 snd_card_t *card = dev_get_drvdata(dev);
871 struct snd_card_omap_aic23 *chip = card->private_data;
876 card->private_data = NULL;
879 dev_set_drvdata(dev, NULL);
885 static struct device_driver omap_alsa_driver = {
886 .name = "omap_mcbsp",
887 .bus = &platform_bus_type,
888 .probe = snd_omap_aic23_probe,
889 .remove = snd_omap_aic23_remove,
890 .suspend = omap_aic23_suspend,
891 .resume = omap_aic23_resume,
894 static int __init omap_aic23_init(void)
899 err = driver_register(&omap_alsa_driver);
904 static void __exit omap_aic23_exit(void)
908 driver_unregister(&omap_alsa_driver);
911 module_init(omap_aic23_init);
912 module_exit(omap_aic23_exit);