-/*
- * sound/arm/omap-aic23.c
- *
- * Alsa Driver for AIC23 codec on OSK5912 platform board
- *
- * Copyright (C) 2005 Instituto Nokia de Tecnologia - INdT - Manaus Brazil
- * Written by Daniel Petrini, David Cohen, Anderson Briglia
- * {daniel.petrini, david.cohen, anderson.briglia}@indt.org.br
- *
- * Based on sa11xx-uda1341.c,
- * Copyright (C) 2002 Tomas Kasparek <tomas.kasparek@seznam.cz>
- *
- * This program is free software; you can redistribute it and/or modify it
- * under the terms of the GNU General Public License as published by the
- * Free Software Foundation; either version 2 of the License, or (at your
- * option) any later version.
- *
- * THIS SOFTWARE IS PROVIDED ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN
- * NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
- * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT
- * NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF
- * USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
- * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
- * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
- * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- *
- * You should have received a copy of the GNU General Public License along
- * with this program; if not, write to the Free Software Foundation, Inc.,
- * 675 Mass Ave, Cambridge, MA 02139, USA.
- *
- * History:
- *
- * 2005-07-29 INdT Kernel Team - Alsa driver for omap osk. Creation of new
- * file omap-aic23.c
- *
- * 2005-12-18 Dirk Behme - Added L/R Channel Interchange fix as proposed by Ajaya Babu
- */
-
-#include <linux/config.h>
-#include <sound/driver.h>
-#include <linux/module.h>
-#include <linux/platform_device.h>
-#include <linux/moduleparam.h>
-#include <linux/init.h>
-#include <linux/errno.h>
-#include <linux/ioctl.h>
-#include <linux/delay.h>
-#include <linux/slab.h>
-#include <linux/clk.h>
-
-#ifdef CONFIG_PM
-#include <linux/pm.h>
-#endif
-
-#include <asm/hardware.h>
-#include <asm/mach-types.h>
-#include <asm/arch/dma.h>
-#include <asm/arch/aic23.h>
-#include <asm/arch/mcbsp.h>
-#include <asm/arch/clock.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/initval.h>
-#include <sound/memalloc.h>
-
-#include "omap-alsa-dma.h"
-#include "omap-aic23.h"
-
-#undef DEBUG
-
-#ifdef DEBUG
-#define ADEBUG() printk("XXX Alsa debug f:%s, l:%d\n", __FUNCTION__, __LINE__)
-#else
-#define ADEBUG() /* nop */
-#endif
-
-/* Define to set the AIC23 as the master w.r.t McBSP */
-#define AIC23_MASTER
-
-/*
- * AUDIO related MACROS
- */
-#define DEFAULT_BITPERSAMPLE 16
-#define AUDIO_RATE_DEFAULT 44100
-#define AUDIO_MCBSP OMAP_MCBSP1
-#define NUMBER_SAMPLE_RATES_SUPPORTED 10
-
-
-MODULE_AUTHOR("Daniel Petrini, David Cohen, Anderson Briglia - INdT");
-MODULE_LICENSE("GPL");
-MODULE_DESCRIPTION("OMAP AIC23 driver for ALSA");
-MODULE_SUPPORTED_DEVICE("{{AIC23,OMAP AIC23}}");
-MODULE_ALIAS("omap_mcbsp.1");
-
-static char *id = NULL;
-MODULE_PARM_DESC(id, "OMAP OSK ALSA Driver for AIC23 chip.");
-
-static struct snd_card_omap_codec *omap_aic23 = NULL;
-
-static struct clk *aic23_mclk = 0;
-
-struct sample_rate_rate_reg_info {
- u8 control; /* SR3, SR2, SR1, SR0 and BOSR */
- u8 divider; /* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */
-};
-
-/*
- * DAC USB-mode sampling rates (MCLK = 12 MHz)
- * The rates and rate_reg_into MUST be in the same order
- */
-static unsigned int rates[] = {
- 4000, 8000, 16000, 22050,
- 24000, 32000, 44100,
- 48000, 88200, 96000,
-};
-static const struct sample_rate_rate_reg_info
- rate_reg_info[NUMBER_SAMPLE_RATES_SUPPORTED] = {
- {0x06, 1}, /* 4000 */
- {0x06, 0}, /* 8000 */
- {0x0C, 1}, /* 16000 */
- {0x11, 1}, /* 22050 */
- {0x00, 1}, /* 24000 */
- {0x0C, 0}, /* 32000 */
- {0x11, 0}, /* 44100 */
- {0x00, 0}, /* 48000 */
- {0x1F, 0}, /* 88200 */
- {0x0E, 0}, /* 96000 */
-};
-
-/*
- * mcbsp configuration structure
- */
-static struct omap_mcbsp_reg_cfg initial_config_mcbsp = {
- .spcr2 = FREE | FRST | GRST | XRST | XINTM(3),
- .spcr1 = RINTM(3) | RRST,
- .rcr2 = RPHASE | RFRLEN2(OMAP_MCBSP_WORD_8) |
- RWDLEN2(OMAP_MCBSP_WORD_16) | RDATDLY(0),
- .rcr1 = RFRLEN1(OMAP_MCBSP_WORD_8) | RWDLEN1(OMAP_MCBSP_WORD_16),
- .xcr2 = XPHASE | XFRLEN2(OMAP_MCBSP_WORD_8) |
- XWDLEN2(OMAP_MCBSP_WORD_16) | XDATDLY(0) | XFIG,
- .xcr1 = XFRLEN1(OMAP_MCBSP_WORD_8) | XWDLEN1(OMAP_MCBSP_WORD_16),
- .srgr1 = FWID(DEFAULT_BITPERSAMPLE - 1),
- .srgr2 = GSYNC | CLKSP | FSGM | FPER(DEFAULT_BITPERSAMPLE * 2 - 1),
-#ifndef AIC23_MASTER
- /* configure McBSP to be the I2S master */
- .pcr0 = FSXM | FSRM | CLKXM | CLKRM | CLKXP | CLKRP,
-#else
- /* configure McBSP to be the I2S slave */
- .pcr0 = CLKXP | CLKRP,
-#endif /* AIC23_MASTER */
-};
-
-static snd_pcm_hw_constraint_list_t hw_constraints_rates = {
- .count = ARRAY_SIZE(rates),
- .list = rates,
- .mask = 0,
-};
-
-/*
- * HW interface start and stop helper functions
- */
-static int audio_ifc_start(void)
-{
- omap_mcbsp_start(AUDIO_MCBSP);
- return 0;
-}
-
-static int audio_ifc_stop(void)
-{
- omap_mcbsp_stop(AUDIO_MCBSP);
- return 0;
-}
-
-/*
- * Codec/mcbsp init and configuration section
- * codec dependent code.
- */
-
-/*
- * Sample rate changing
- */
-static void omap_aic23_set_samplerate(struct snd_card_omap_codec
- *omap_aic23, long rate)
-{
- u8 count = 0;
- u16 data = 0;
-
- /* Fix the rate if it has a wrong value */
- if (rate >= 96000)
- rate = 96000;
- else if (rate >= 88200)
- rate = 88200;
- else if (rate >= 48000)
- rate = 48000;
- else if (rate >= 44100)
- rate = 44100;
- else if (rate >= 32000)
- rate = 32000;
- else if (rate >= 24000)
- rate = 24000;
- else if (rate >= 22050)
- rate = 22050;
- else if (rate >= 16000)
- rate = 16000;
- else if (rate >= 8000)
- rate = 8000;
- else
- rate = 4000;
-
- /* Search for the right sample rate */
- /* Verify what happens if the rate is not supported
- * now it goes to 96Khz */
- while ((rates[count] != rate) &&
- (count < (NUMBER_SAMPLE_RATES_SUPPORTED - 1))) {
- count++;
- }
-
- data = (rate_reg_info[count].divider << CLKIN_SHIFT) |
- (rate_reg_info[count].control << BOSR_SHIFT) | USB_CLK_ON;
-
- audio_aic23_write(SAMPLE_RATE_CONTROL_ADDR, data);
-
- omap_aic23->samplerate = rate;
-}
-
-static inline void aic23_configure(void)
-{
- /* Reset codec */
- audio_aic23_write(RESET_CONTROL_ADDR, 0);
-
- /* Initialize the AIC23 internal state */
-
- /* Analog audio path control, DAC selected, delete INSEL_MIC for line in */
- audio_aic23_write(ANALOG_AUDIO_CONTROL_ADDR, DEFAULT_ANALOG_AUDIO_CONTROL);
-
- /* Digital audio path control, de-emphasis control 44.1kHz */
- audio_aic23_write(DIGITAL_AUDIO_CONTROL_ADDR, DEEMP_44K);
-
- /* Digital audio interface, master/slave mode, I2S, 16 bit */
-#ifdef AIC23_MASTER
- audio_aic23_write(DIGITAL_AUDIO_FORMAT_ADDR,
- MS_MASTER | IWL_16 | FOR_DSP);
-#else
- audio_aic23_write(DIGITAL_AUDIO_FORMAT_ADDR, IWL_16 | FOR_DSP);
-#endif
-
- /* Enable digital interface */
- audio_aic23_write(DIGITAL_INTERFACE_ACT_ADDR, ACT_ON);
-
-}
-
-static void omap_aic23_audio_init(struct snd_card_omap_codec *omap_aic23)
-{
- /* Setup DMA stuff */
- omap_aic23->s[SNDRV_PCM_STREAM_PLAYBACK].id = "Alsa AIC23 out";
- omap_aic23->s[SNDRV_PCM_STREAM_PLAYBACK].stream_id =
- SNDRV_PCM_STREAM_PLAYBACK;
- omap_aic23->s[SNDRV_PCM_STREAM_PLAYBACK].dma_dev =
- OMAP_DMA_MCBSP1_TX;
- omap_aic23->s[SNDRV_PCM_STREAM_PLAYBACK].hw_start =
- audio_ifc_start;
- omap_aic23->s[SNDRV_PCM_STREAM_PLAYBACK].hw_stop =
- audio_ifc_stop;
-
- omap_aic23->s[SNDRV_PCM_STREAM_CAPTURE].id = "Alsa AIC23 in";
- omap_aic23->s[SNDRV_PCM_STREAM_CAPTURE].stream_id =
- SNDRV_PCM_STREAM_CAPTURE;
- omap_aic23->s[SNDRV_PCM_STREAM_CAPTURE].dma_dev =
- OMAP_DMA_MCBSP1_RX;
- omap_aic23->s[SNDRV_PCM_STREAM_CAPTURE].hw_start =
- audio_ifc_start;
- omap_aic23->s[SNDRV_PCM_STREAM_CAPTURE].hw_stop =
- audio_ifc_stop;
-
- /* configuring the McBSP */
- omap_mcbsp_request(AUDIO_MCBSP);
-
- /* if configured, then stop mcbsp */
- omap_mcbsp_stop(AUDIO_MCBSP);
-
- omap_mcbsp_config(AUDIO_MCBSP, &initial_config_mcbsp);
- omap_mcbsp_start(AUDIO_MCBSP);
- aic23_configure();
-}
-
-/*
- * DMA functions
- * Depends on omap-aic23-dma.c functions and (omap) dma.c
- *
- */
-#define DMA_BUF_SIZE 1024 * 8
-
-static int audio_dma_request(struct audio_stream *s,
- void (*callback) (void *))
-{
- int err;
-
- err = omap_request_alsa_sound_dma(s->dma_dev, s->id, s, &s->lch);
- if (err < 0)
- printk(KERN_ERR "unable to grab audio dma 0x%x\n",
- s->dma_dev);
- return err;
-}
-
-static int audio_dma_free(struct audio_stream *s)
-{
- int err = 0;
-
- err = omap_free_alsa_sound_dma(s, &s->lch);
- if (err < 0)
- printk(KERN_ERR "Unable to free audio dma channels!\n");
- return err;
-}
-
-/*
- * This function should calculate the current position of the dma in the
- * buffer. It will help alsa middle layer to continue update the buffer.
- * Its correctness is crucial for good functioning.
- */
-static u_int audio_get_dma_pos(struct audio_stream *s)
-{
- snd_pcm_substream_t *substream = s->stream;
- snd_pcm_runtime_t *runtime = substream->runtime;
- unsigned int offset;
- unsigned long flags;
- dma_addr_t count;
- ADEBUG();
-
- /* this must be called w/ interrupts locked as requested in dma.c */
- spin_lock_irqsave(&s->dma_lock, flags);
-
- /* For the current period let's see where we are */
- count = omap_get_dma_src_addr_counter(s->lch[s->dma_q_head]);
-
- spin_unlock_irqrestore(&s->dma_lock, flags);
-
- /* Now, the position related to the end of that period */
- offset = bytes_to_frames(runtime, s->offset) - bytes_to_frames(runtime, count);
-
- if (offset >= runtime->buffer_size || offset < 0)
- offset = 0;
-
- return offset;
-}
-
-/*
- * this stops the dma and clears the dma ptrs
- */
-static void audio_stop_dma(struct audio_stream *s)
-{
- unsigned long flags;
- ADEBUG();
-
- spin_lock_irqsave(&s->dma_lock, flags);
- s->active = 0;
- s->period = 0;
- s->periods = 0;
-
- /* this stops the dma channel and clears the buffer ptrs */
- /* this stops the dma channel and clears the buffer ptrs */
- omap_stop_alsa_sound_dma(s);
-
- omap_clear_alsa_sound_dma(s);
-
- spin_unlock_irqrestore(&s->dma_lock, flags);
-}
-
-/*
- * Main dma routine, requests dma according where you are in main alsa buffer
- */
-static void audio_process_dma(struct audio_stream *s)
-{
- snd_pcm_substream_t *substream = s->stream;
- snd_pcm_runtime_t *runtime;
- unsigned int dma_size;
- unsigned int offset;
- int ret;
-
- runtime = substream->runtime;
- if (s->active) {
- dma_size = frames_to_bytes(runtime, runtime->period_size);
- offset = dma_size * s->period;
- snd_assert(dma_size <= DMA_BUF_SIZE,);
- ret = omap_start_alsa_sound_dma(s,
- (dma_addr_t) runtime->dma_area +
- offset, dma_size);
- if (ret) {
- printk(KERN_ERR
- "audio_process_dma: cannot queue DMA buffer (%i)\n",
- ret);
- return;
- }
-
- s->period++;
- s->period %= runtime->periods;
- s->periods++;
- s->offset = offset;
- }
-}
-
-/*
- * This is called when dma IRQ occurs at the end of each transmited block
- */
-void callback_omap_alsa_sound_dma(void *data)
-{
- struct audio_stream *s = data;
-
- /*
- * If we are getting a callback for an active stream then we inform
- * the PCM middle layer we've finished a period
- */
- if (s->active)
- snd_pcm_period_elapsed(s->stream);
-
- spin_lock(&s->dma_lock);
- if (s->periods > 0) {
- s->periods--;
- }
- audio_process_dma(s);
- spin_unlock(&s->dma_lock);
-}
-
-
-/*
- * Alsa section
- * PCM settings and callbacks
- */
-
-static int snd_omap_alsa_trigger(snd_pcm_substream_t * substream, int cmd)
-{
- struct snd_card_omap_codec *chip =
- snd_pcm_substream_chip(substream);
- int stream_id = substream->pstr->stream;
- struct audio_stream *s = &chip->s[stream_id];
- int err = 0;
- ADEBUG();
-
- /* note local interrupts are already disabled in the midlevel code */
- spin_lock(&s->dma_lock);
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_START:
- /* requested stream startup */
- s->active = 1;
- audio_process_dma(s);
- break;
- case SNDRV_PCM_TRIGGER_STOP:
- /* requested stream shutdown */
- audio_stop_dma(s);
- break;
- default:
- err = -EINVAL;
- break;
- }
- spin_unlock(&s->dma_lock);
-
- return err;
-}
-
-static int snd_omap_alsa_prepare(snd_pcm_substream_t * substream)
-{
- struct snd_card_omap_codec *chip =
- snd_pcm_substream_chip(substream);
- snd_pcm_runtime_t *runtime = substream->runtime;
- struct audio_stream *s = &chip->s[substream->pstr->stream];
-
- /* set requested samplerate */
- omap_aic23_set_samplerate(chip, runtime->rate);
-
- s->period = 0;
- s->periods = 0;
-
- return 0;
-}
-
-static snd_pcm_uframes_t snd_omap_alsa_pointer(snd_pcm_substream_t *
- substream)
-{
- struct snd_card_omap_codec *chip =
- snd_pcm_substream_chip(substream);
-
- return audio_get_dma_pos(&chip->s[substream->pstr->stream]);
-}
-
-/* Hardware capabilities */
-
-static snd_pcm_hardware_t snd_omap_alsa_capture = {
- .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
- SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID),
- .formats = (SNDRV_PCM_FMTBIT_S16_LE),
- .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
- SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
- SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
- SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
- SNDRV_PCM_RATE_KNOT),
- .rate_min = 8000,
- .rate_max = 96000,
- .channels_min = 2,
- .channels_max = 2,
- .buffer_bytes_max = 128 * 1024,
- .period_bytes_min = 32,
- .period_bytes_max = 8 * 1024,
- .periods_min = 16,
- .periods_max = 255,
- .fifo_size = 0,
-};
-
-static snd_pcm_hardware_t snd_omap_alsa_playback = {
- .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
- SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID),
- .formats = (SNDRV_PCM_FMTBIT_S16_LE),
- .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
- SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
- SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
- SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
- SNDRV_PCM_RATE_KNOT),
- .rate_min = 8000,
- .rate_max = 96000,
- .channels_min = 2,
- .channels_max = 2,
- .buffer_bytes_max = 128 * 1024,
- .period_bytes_min = 32,
- .period_bytes_max = 8 * 1024,
- .periods_min = 16,
- .periods_max = 255,
- .fifo_size = 0,
-};
-
-static int snd_card_omap_alsa_open(snd_pcm_substream_t * substream)
-{
- struct snd_card_omap_codec *chip =
- snd_pcm_substream_chip(substream);
- snd_pcm_runtime_t *runtime = substream->runtime;
- int stream_id = substream->pstr->stream;
- int err;
- ADEBUG();
-
- chip->s[stream_id].stream = substream;
-
- omap_aic23_clock_on();
-
- if (stream_id == SNDRV_PCM_STREAM_PLAYBACK)
- runtime->hw = snd_omap_alsa_playback;
- else
- runtime->hw = snd_omap_alsa_capture;
- if ((err =
- snd_pcm_hw_constraint_integer(runtime,
- SNDRV_PCM_HW_PARAM_PERIODS)) <
- 0)
- return err;
- if ((err =
- snd_pcm_hw_constraint_list(runtime, 0,
- SNDRV_PCM_HW_PARAM_RATE,
- &hw_constraints_rates)) < 0)
- return err;
-
- return 0;
-}
-
-static int snd_card_omap_alsa_close(snd_pcm_substream_t * substream)
-{
- struct snd_card_omap_codec *chip =
- snd_pcm_substream_chip(substream);
- ADEBUG();
-
- omap_aic23_clock_off();
- chip->s[substream->pstr->stream].stream = NULL;
-
- return 0;
-}
-
-/* HW params & free */
-
-static int snd_omap_alsa_hw_params(snd_pcm_substream_t * substream,
- snd_pcm_hw_params_t * hw_params)
-{
- return snd_pcm_lib_malloc_pages(substream,
- params_buffer_bytes(hw_params));
-}
-
-static int snd_omap_alsa_hw_free(snd_pcm_substream_t * substream)
-{
- return snd_pcm_lib_free_pages(substream);
-}
-
-/* pcm operations */
-static snd_pcm_ops_t snd_card_omap_alsa_playback_ops = {
- .open = snd_card_omap_alsa_open,
- .close = snd_card_omap_alsa_close,
- .ioctl = snd_pcm_lib_ioctl,
- .hw_params = snd_omap_alsa_hw_params,
- .hw_free = snd_omap_alsa_hw_free,
- .prepare = snd_omap_alsa_prepare,
- .trigger = snd_omap_alsa_trigger,
- .pointer = snd_omap_alsa_pointer,
-};
-
-static snd_pcm_ops_t snd_card_omap_alsa_capture_ops = {
- .open = snd_card_omap_alsa_open,
- .close = snd_card_omap_alsa_close,
- .ioctl = snd_pcm_lib_ioctl,
- .hw_params = snd_omap_alsa_hw_params,
- .hw_free = snd_omap_alsa_hw_free,
- .prepare = snd_omap_alsa_prepare,
- .trigger = snd_omap_alsa_trigger,
- .pointer = snd_omap_alsa_pointer,
-};
-
-/*
- * Alsa init and exit section
- *
- * Inits pcm alsa structures, allocate the alsa buffer, suspend, resume
- */
-static int __init snd_card_omap_alsa_pcm(struct snd_card_omap_codec *omap_alsa,
- int device)
-{
- snd_pcm_t *pcm;
- int err;
- ADEBUG();
-
- if ((err =
- snd_pcm_new(omap_aic23->card, "AIC23 PCM", device, 1, 1,
- &pcm)) < 0)
- return err;
-
- /* sets up initial buffer with continuous allocation */
- snd_pcm_lib_preallocate_pages_for_all(pcm,
- SNDRV_DMA_TYPE_CONTINUOUS,
- snd_dma_continuous_data
- (GFP_KERNEL),
- 128 * 1024, 128 * 1024);
-
- snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
- &snd_card_omap_alsa_playback_ops);
- snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
- &snd_card_omap_alsa_capture_ops);
- pcm->private_data = omap_aic23;
- pcm->info_flags = 0;
- strcpy(pcm->name, "omap aic23 pcm");
-
- omap_aic23_audio_init(omap_aic23);
-
- /* setup DMA controller */
- audio_dma_request(&omap_aic23->s[SNDRV_PCM_STREAM_PLAYBACK],
- callback_omap_alsa_sound_dma);
- audio_dma_request(&omap_aic23->s[SNDRV_PCM_STREAM_CAPTURE],
- callback_omap_alsa_sound_dma);
-
- omap_aic23->pcm = pcm;
-
- return 0;
-}
-
-#ifdef CONFIG_PM
-/*
- * Driver suspend/resume - calls alsa functions. Some hints from aaci.c
- */
-int snd_omap_alsa_suspend(struct platform_device *pdev, pm_message_t state)
-{
- struct snd_card_omap_codec *chip;
- snd_card_t *card = platform_get_drvdata(pdev);
-
- if (card->power_state != SNDRV_CTL_POWER_D3hot) {
- chip = card->private_data;
- if (chip->card->power_state != SNDRV_CTL_POWER_D3hot) {
- snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot);
- snd_pcm_suspend_all(chip->pcm);
- /* Mutes and turn clock off */
- omap_aic23_clock_off();
- snd_omap_suspend_mixer();
- }
- }
- return 0;
-}
-
-int snd_omap_alsa_resume(struct platform_device *pdev)
-{
- struct snd_card_omap_codec *chip;
- snd_card_t *card = platform_get_drvdata(pdev);
-
- if (card->power_state != SNDRV_CTL_POWER_D0) {
- chip = card->private_data;
- if (chip->card->power_state != SNDRV_CTL_POWER_D0) {
- snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0);
- omap_aic23_clock_on();
- snd_omap_resume_mixer();
- }
- }
- return 0;
-}
-
-#else
-#define snd_omap_alsa_suspend NULL
-#define snd_omap_alsa_resume NULL
-#endif /* CONFIG_PM */
-
-/*
- */
-void snd_omap_alsa_free(snd_card_t * card)
-{
- struct snd_card_omap_codec *chip = card->private_data;
- ADEBUG();
-
- /*
- * Turn off codec after it is done.
- * Can't do it immediately, since it may still have
- * buffered data.
- */
- set_current_state(TASK_INTERRUPTIBLE);
- schedule_timeout(2);
-
- omap_mcbsp_stop(AUDIO_MCBSP);
- omap_mcbsp_free(AUDIO_MCBSP);
-
- audio_aic23_write(RESET_CONTROL_ADDR, 0);
- audio_aic23_write(POWER_DOWN_CONTROL_ADDR, 0xff);
-
- audio_dma_free(&chip->s[SNDRV_PCM_STREAM_PLAYBACK]);
- audio_dma_free(&chip->s[SNDRV_PCM_STREAM_CAPTURE]);
-}
-
-/*
- * Omap MCBSP clock configuration
- *
- * Here we have some functions that allows clock to be enabled and
- * disabled only when needed. Besides doing clock configuration
- * it allows turn on/turn off audio when necessary.
- */
-#define CODEC_CLOCK 12000000
-#define AUDIO_RATE_DEFAULT 44100
-
-/*
- * Do clock framework mclk search
- */
-static __init void omap_aic23_clock_setup(void)
-{
- aic23_mclk = clk_get(0, "mclk");
-}
-
-/*
- * Do some sanity check, set clock rate, starts it and
- * turn codec audio on
- */
-int omap_aic23_clock_on(void)
-{
- if (clk_get_usecount(aic23_mclk) > 0) {
- /* MCLK is already in use */
- printk(KERN_WARNING
- "MCLK in use at %d Hz. We change it to %d Hz\n",
- (uint) clk_get_rate(aic23_mclk),
- CODEC_CLOCK);
- }
-
- if (clk_set_rate(aic23_mclk, CODEC_CLOCK)) {
- printk(KERN_ERR
- "Cannot set MCLK for AIC23 CODEC\n");
- return -ECANCELED;
- }
-
- clk_enable(aic23_mclk);
-
- printk(KERN_DEBUG
- "MCLK = %d [%d], usecount = %d\n",
- (uint) clk_get_rate(aic23_mclk), CODEC_CLOCK,
- clk_get_usecount(aic23_mclk));
-
- /* Now turn the audio on */
- audio_aic23_write(POWER_DOWN_CONTROL_ADDR,
- ~DEVICE_POWER_OFF & ~OUT_OFF & ~DAC_OFF &
- ~ADC_OFF & ~MIC_OFF & ~LINE_OFF);
-
- return 0;
-}
-/*
- * Do some sanity check, turn clock off and then turn
- * codec audio off
- */
-int omap_aic23_clock_off(void)
-{
- if (clk_get_usecount(aic23_mclk) > 0) {
- if (clk_get_rate(aic23_mclk) != CODEC_CLOCK) {
- printk(KERN_WARNING
- "MCLK for audio should be %d Hz. But is %d Hz\n",
- (uint) clk_get_rate(aic23_mclk),
- CODEC_CLOCK);
- }
-
- clk_disable(aic23_mclk);
- }
-
- audio_aic23_write(POWER_DOWN_CONTROL_ADDR,
- DEVICE_POWER_OFF | OUT_OFF | DAC_OFF |
- ADC_OFF | MIC_OFF | LINE_OFF);
- return 0;
-}
-
-/* module init & exit */
-
-/*
- * Inits alsa soudcard structure
- */
-static int __init snd_omap_alsa_aic23_probe(struct platform_device *pdev)
-{
- int err = 0;
- snd_card_t *card;
- ADEBUG();
-
- /* gets clock from clock framework */
- omap_aic23_clock_setup();
-
- /* register the soundcard */
- card = snd_card_new(-1, id, THIS_MODULE, sizeof(omap_aic23));
- if (card == NULL)
- return -ENOMEM;
-
- omap_aic23 = kcalloc(1, sizeof(*omap_aic23), GFP_KERNEL);
- if (omap_aic23 == NULL)
- return -ENOMEM;
-
- card->private_data = (void *) omap_aic23;
- card->private_free = snd_omap_alsa_free;
-
- omap_aic23->card = card;
- omap_aic23->samplerate = AUDIO_RATE_DEFAULT;
-
- spin_lock_init(&omap_aic23->s[0].dma_lock);
- spin_lock_init(&omap_aic23->s[1].dma_lock);
-
- /* mixer */
- if ((err = snd_omap_mixer(omap_aic23)) < 0)
- goto nodev;
-
- /* PCM */
- if ((err = snd_card_omap_alsa_pcm(omap_aic23, 0)) < 0)
- goto nodev;
-
- strcpy(card->driver, "AIC23");
- strcpy(card->shortname, "OSK AIC23");
- sprintf(card->longname, "OMAP OSK with AIC23");
-
- snd_omap_init_mixer();
-
- snd_card_set_dev(card, &pdev->dev);
-
- if ((err = snd_card_register(card)) == 0) {
- printk(KERN_INFO "OSK audio support initialized\n");
- platform_set_drvdata(pdev, card);
- return 0;
- }
-
-nodev:
- snd_card_free(card);
-
- return err;
-}
-
-static int snd_omap_alsa_remove(struct platform_device *pdev)
-{
- snd_card_t *card = platform_get_drvdata(pdev);
- struct snd_card_omap_codec *chip = card->private_data;
-
- snd_card_free(card);
-
- omap_aic23 = NULL;
- card->private_data = NULL;
- kfree(chip);
-
- platform_set_drvdata(pdev, NULL);
-
- return 0;
-
-}
-
-static struct platform_driver omap_alsa_driver = {
- .probe = snd_omap_alsa_aic23_probe,
- .remove = snd_omap_alsa_remove,
- .suspend = snd_omap_alsa_suspend,
- .resume = snd_omap_alsa_resume,
- .driver = {
- .name = "omap_mcbsp",
- },
-};
-
-static int __init omap_alsa_aic23_init(void)
-{
- int err;
- ADEBUG();
-
- err = platform_driver_register(&omap_alsa_driver);
-
- return err;
-}
-
-static void __exit omap_alsa_aic23_exit(void)
-{
- ADEBUG();
-
- platform_driver_unregister(&omap_alsa_driver);
-}
-
-module_init(omap_alsa_aic23_init);
-module_exit(omap_alsa_aic23_exit);