/* arch/arm/mach-msm/qdsp5/audio_mp3.c
*
* mp3 audio output device
*
* Copyright (C) 2008 Google, Inc.
* Copyright (C) 2008 HTC Corporation
*
* This software is licensed under the terms of the GNU General Public
* License version 2, as published by the Free Software Foundation, and
* may be copied, distributed, and modified under those terms.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
*/
#include <linux/module.h>
#include <linux/fs.h>
#include <linux/miscdevice.h>
#include <linux/uaccess.h>
#include <linux/kthread.h>
#include <linux/wait.h>
#include <linux/dma-mapping.h>
#include <linux/delay.h>
#include <asm/atomic.h>
#include <asm/ioctls.h>
#include <mach/msm_adsp.h>
#include <linux/msm_audio.h>
#include "audmgr.h"
#include <mach/qdsp5/qdsp5audppcmdi.h>
#include <mach/qdsp5/qdsp5audppmsg.h>
#include <mach/qdsp5/qdsp5audplaycmdi.h>
#include <mach/qdsp5/qdsp5audplaymsg.h>
/* for queue ids - should be relative to module number*/
#include "adsp.h"
#ifdef DEBUG
#define dprintk(format, arg...) \
printk(KERN_DEBUG format, ## arg)
#else
#define dprintk(format, arg...) do {} while (0)
#endif
/* Size must be power of 2 */
#define BUFSZ_MAX 32768
#define BUFSZ_MIN 4096
#define DMASZ_MAX (BUFSZ_MAX * 2)
#define DMASZ_MIN (BUFSZ_MIN * 2)
#define AUDPLAY_INVALID_READ_PTR_OFFSET 0xFFFF
#define AUDDEC_DEC_MP3 2
#define PCM_BUFSZ_MIN 4800 /* Hold one stereo MP3 frame */
#define PCM_BUF_MAX_COUNT 5 /* DSP only accepts 5 buffers at most
but support 2 buffers currently */
#define ROUTING_MODE_FTRT 1
#define ROUTING_MODE_RT 2
/* Decoder status received from AUDPPTASK */
#define AUDPP_DEC_STATUS_SLEEP 0
#define AUDPP_DEC_STATUS_INIT 1
#define AUDPP_DEC_STATUS_CFG 2
#define AUDPP_DEC_STATUS_PLAY 3
struct buffer {
void *data;
unsigned size;
unsigned used; /* Input usage actual DSP produced PCM size */
unsigned addr;
};
struct audio {
struct buffer out[2];
spinlock_t dsp_lock;
uint8_t out_head;
uint8_t out_tail;
uint8_t out_needed; /* number of buffers the dsp is waiting for */
unsigned out_dma_sz;
atomic_t out_bytes;
struct mutex lock;
struct mutex write_lock;
wait_queue_head_t write_wait;
/* Host PCM section */
struct buffer in[PCM_BUF_MAX_COUNT];
struct mutex read_lock;
wait_queue_head_t read_wait; /* Wait queue for read */
char *read_data; /* pointer to reader buffer */
dma_addr_t read_phys; /* physical address of reader buffer */
uint8_t read_next; /* index to input buffers to be read next */
uint8_t fill_next; /* index to buffer that DSP should be filling */
uint8_t pcm_buf_count; /* number of pcm buffer allocated */
/* ---- End of Host PCM section */
struct msm_adsp_module *audplay;
/* configuration to use on next enable */
uint32_t out_sample_rate;
uint32_t out_channel_mode;
struct audmgr audmgr;
/* data allocated for various buffers */
char *data;
dma_addr_t phys;
int rflush; /* Read flush */
int wflush; /* Write flush */
int opened;
int enabled;
int running;
int stopped; /* set when stopped, cleared on flush */
int pcm_feedback;
int buf_refresh;
int reserved; /* A byte is being reserved */
char rsv_byte; /* Handle odd length user data */
unsigned volume;
uint16_t dec_id;
uint32_t read_ptr_offset;
};
static int auddec_dsp_config(struct audio *audio, int enable);
static void audpp_cmd_cfg_adec_params(struct audio *audio);
static void audpp_cmd_cfg_routing_mode(struct audio *audio);
static void audplay_send_data(struct audio *audio, unsigned needed);
static void audplay_config_hostpcm(struct audio *audio);
static void audplay_buffer_refresh(struct audio *audio);
static void audio_dsp_event(void *private, unsigned id, uint16_t *msg);
/* must be called with audio->lock held */
static int audio_enable(struct audio *audio)
{
struct audmgr_config cfg;
int rc;
pr_info("audio_enable()\n");
if (audio->enabled)
return 0;
audio->out_tail = 0;
audio->out_needed = 0;
cfg.tx_rate = RPC_AUD_DEF_SAMPLE_RATE_NONE;
cfg.rx_rate = RPC_AUD_DEF_SAMPLE_RATE_48000;
cfg.def_method = RPC_AUD_DEF_METHOD_PLAYBACK;
cfg.codec = RPC_AUD_DEF_CODEC_MP3;
cfg.snd_method = RPC_SND_METHOD_MIDI;
rc = audmgr_enable(&audio->audmgr, &cfg);
if (rc < 0)
return rc;
if (msm_adsp_enable(audio->audplay)) {
pr_err("audio: msm_adsp_enable(audplay) failed\n");
audmgr_disable(&audio->audmgr);
return -ENODEV;
}
if (audpp_enable(audio->dec_id, audio_dsp_event, audio)) {
pr_err("audio: audpp_enable() failed\n");
msm_adsp_disable(audio->audplay);
audmgr_disable(&audio->audmgr);
return -ENODEV;
}
audio->enabled = 1;
return 0;
}
/* must be called with audio->lock held */
static int audio_disable(struct audio *audio)
{
pr_info("audio_disable()\n");
if (audio->enabled) {
audio->enabled = 0;
auddec_dsp_config(audio, 0);
wake_up(&audio->write_wait);
wake_up(&audio->read_wait);
msm_adsp_disable(audio->audplay);
audpp_disable(audio->dec_id, audio);
audmgr_disable(&audio->audmgr);
audio->out_needed = 0;
}
return 0;
}
/* ------------------- dsp --------------------- */
static void audio_update_pcm_buf_entry(struct audio *audio, uint32_t *payload)
{
uint8_t index;
unsigned long flags;
if (audio->rflush) {
audio->buf_refresh = 1;
return;
}
spin_lock_irqsave(&audio->dsp_lock, flags);
for (index = 0; index < payload[1]; index++) {
if (audio->in[audio->fill_next].addr ==
payload[2 + index * 2]) {
pr_info("audio_update_pcm_buf_entry: in[%d] ready\n",
audio->fill_next);
audio->in[audio->fill_next].used =
payload[3 + index * 2];
if ((++audio->fill_next) == audio->pcm_buf_count)
audio->fill_next = 0;
} else {
pr_err
("audio_update_pcm_buf_entry: expected=%x ret=%x\n"
, audio->in[audio->fill_next].addr,
payload[1 + index * 2]);
break;
}
}
if (audio->in[audio->fill_next].used == 0) {
audplay_buffer_refresh(audio);
} else {
pr_info("audio_update_pcm_buf_entry: read cannot keep up\n");
audio->buf_refresh = 1;
}
wake_up(&audio->read_wait);
spin_unlock_irqrestore(&audio->dsp_lock, flags);
}
static void audplay_dsp_event(void *data, unsigned id, size_t len,
void (*getevent) (void *ptr, size_t len))
{
struct audio *audio = data;
uint32_t msg[28];
getevent(msg, sizeof(msg));
dprintk("audplay_dsp_event: msg_id=%x\n", id);
switch (id) {
case AUDPLAY_MSG_DEC_NEEDS_DATA:
audplay_send_data(audio, 1);
break;
case AUDPLAY_MSG_BUFFER_UPDATE:
audio_update_pcm_buf_entry(audio, msg);
break;
default:
pr_err("unexpected message from decoder \n");
break;
}
}
static void audio_dsp_event(void *private, unsigned id, uint16_t *msg)
{
struct audio *audio = private;
switch (id) {
case AUDPP_MSG_STATUS_MSG:{
unsigned status = msg[1];
switch (status) {
case AUDPP_DEC_STATUS_SLEEP:
pr_info("decoder status: sleep \n");
break;
case AUDPP_DEC_STATUS_INIT:
pr_info("decoder status: init \n");
audpp_cmd_cfg_routing_mode(audio);
break;
case AUDPP_DEC_STATUS_CFG:
pr_info("decoder status: cfg \n");
break;
case AUDPP_DEC_STATUS_PLAY:
pr_info("decoder status: play \n");
if (audio->pcm_feedback) {
audplay_config_hostpcm(audio);
audplay_buffer_refresh(audio);
}
break;
default:
pr_err("unknown decoder status \n");
break;
}
break;
}
case AUDPP_MSG_CFG_MSG:
if (msg[0] == AUDPP_MSG_ENA_ENA) {
pr_info("audio_dsp_event: CFG_MSG ENABLE\n");
auddec_dsp_config(audio, 1);
audio->out_needed = 0;
audio->running = 1;
audpp_set_volume_and_pan(audio->dec_id, audio->volume,
0);
audpp_avsync(audio->dec_id, 22050);
} else if (msg[0] == AUDPP_MSG_ENA_DIS) {
pr_info("audio_dsp_event: CFG_MSG DISABLE\n");
audpp_avsync(audio->dec_id, 0);
audio->running = 0;
} else {
pr_err("audio_dsp_event: CFG_MSG %d?\n", msg[0]);
}
break;
case AUDPP_MSG_ROUTING_ACK:
pr_info("audio_dsp_event: ROUTING_ACK mode=%d\n", msg[1]);
audpp_cmd_cfg_adec_params(audio);
break;
case AUDPP_MSG_FLUSH_ACK:
dprintk("%s: FLUSH_ACK\n", __func__);
audio->wflush = 0;
audio->rflush = 0;
if (audio->pcm_feedback)
audplay_buffer_refresh(audio);
break;
default:
pr_err("audio_dsp_event: UNKNOWN (%d)\n", id);
}
}
struct msm_adsp_ops audplay_adsp_ops = {
.event = audplay_dsp_event,
};
#define audplay_send_queue0(audio, cmd, len) \
msm_adsp_write(audio->audplay, QDSP_uPAudPlay0BitStreamCtrlQueue, \
cmd, len)
static int auddec_dsp_config(struct audio *audio, int enable)
{
audpp_cmd_cfg_dec_type cmd;
memset(&cmd, 0, sizeof(cmd));
cmd.cmd_id = AUDPP_CMD_CFG_DEC_TYPE;
if (enable)
cmd.dec0_cfg = AUDPP_CMD_UPDATDE_CFG_DEC |
AUDPP_CMD_ENA_DEC_V |
AUDDEC_DEC_MP3;
else
cmd.dec0_cfg = AUDPP_CMD_UPDATDE_CFG_DEC |
AUDPP_CMD_DIS_DEC_V;
return audpp_send_queue1(&cmd, sizeof(cmd));
}
static void audpp_cmd_cfg_adec_params(struct audio *audio)
{
audpp_cmd_cfg_adec_params_mp3 cmd;
memset(&cmd, 0, sizeof(cmd));
cmd.common.cmd_id = AUDPP_CMD_CFG_ADEC_PARAMS;
cmd.common.length = AUDPP_CMD_CFG_ADEC_PARAMS_MP3_LEN;
cmd.common.dec_id = audio->dec_id;
cmd.common.input_sampling_frequency = audio->out_sample_rate;
audpp_send_queue2(&cmd, sizeof(cmd));
}
static void audpp_cmd_cfg_routing_mode(struct audio *audio)
{
struct audpp_cmd_routing_mode cmd;
pr_info("audpp_cmd_cfg_routing_mode()\n");
memset(&cmd, 0, sizeof(cmd));
cmd.cmd_id = AUDPP_CMD_ROUTING_MODE;
cmd.object_number = audio->dec_id;
if (audio->pcm_feedback)
cmd.routing_mode = ROUTING_MODE_FTRT;
else
cmd.routing_mode = ROUTING_MODE_RT;
audpp_send_queue1(&cmd, sizeof(cmd));
}
static int audplay_dsp_send_data_avail(struct audio *audio,
unsigned idx, unsigned len)
{
audplay_cmd_bitstream_data_avail cmd;
cmd.cmd_id = AUDPLAY_CMD_BITSTREAM_DATA_AVAIL;
cmd.decoder_id = audio->dec_id;
cmd.buf_ptr = audio->out[idx].addr;
cmd.buf_size = len/2;
cmd.partition_number = 0;
return audplay_send_queue0(audio, &cmd, sizeof(cmd));
}
static void audplay_buffer_refresh(struct audio *audio)
{
struct audplay_cmd_buffer_refresh refresh_cmd;
refresh_cmd.cmd_id = AUDPLAY_CMD_BUFFER_REFRESH;
refresh_cmd.num_buffers = 1;
refresh_cmd.buf0_address = audio->in[audio->fill_next].addr;
refresh_cmd.buf0_length = audio->in[audio->fill_next].size -
(audio->in[audio->fill_next].size % 576); /* Mp3 frame size */
refresh_cmd.buf_read_count = 0;
pr_info("audplay_buffer_fresh: buf0_addr=%x buf0_len=%d\n",
refresh_cmd.buf0_address, refresh_cmd.buf0_length);
(void)audplay_send_queue0(audio, &refresh_cmd, sizeof(refresh_cmd));
}
static void audplay_config_hostpcm(struct audio *audio)
{
struct audplay_cmd_hpcm_buf_cfg cfg_cmd;
pr_info("audplay_config_hostpcm()\n");
cfg_cmd.cmd_id = AUDPLAY_CMD_HPCM_BUF_CFG;
cfg_cmd.max_buffers = 1;
cfg_cmd.byte_swap = 0;
cfg_cmd.hostpcm_config = (0x8000) | (0x4000);
cfg_cmd.feedback_frequency = 1;
cfg_cmd.partition_number = 0;
(void)audplay_send_queue0(audio, &cfg_cmd, sizeof(cfg_cmd));
}
static void audplay_send_data(struct audio *audio, unsigned needed)
{
struct buffer *frame;
unsigned long flags;
spin_lock_irqsave(&audio->dsp_lock, flags);
if (!audio->running)
goto done;
if (audio->wflush) {
audio->out_needed = 1;
goto done;
}
if (needed && !audio->wflush) {
/* We were called from the callback because the DSP
* requested more data. Note that the DSP does want
* more data, and if a buffer was in-flight, mark it
* as available (since the DSP must now be done with
* it).
*/
audio->out_needed = 1;
frame = audio->out + audio->out_tail;
if (frame->used == 0xffffffff) {
dprintk("frame %d free\n", audio->out_tail);
frame->used = 0;
audio->out_tail ^= 1;
wake_up(&audio->write_wait);
}
}
if (audio->out_needed) {
/* If the DSP currently wants data and we have a
* buffer available, we will send it and reset
* the needed flag. We'll mark the buffer as in-flight
* so that it won't be recycled until the next buffer
* is requested
*/
frame = audio->out + audio->out_tail;
if (frame->used) {
BUG_ON(frame->used == 0xffffffff);
dprintk("frame %d busy\n", audio->out_tail);
audplay_dsp_send_data_avail(audio, audio->out_tail,
frame->used);
frame->used = 0xffffffff;
audio->out_needed = 0;
}
}
done:
spin_unlock_irqrestore(&audio->dsp_lock, flags);
}
/* ------------------- device --------------------- */
static void audio_flush(struct audio *audio)
{
audio->out[0].used = 0;
audio->out[1].used = 0;
audio->out_head = 0;
audio->out_tail = 0;
audio->reserved = 0;
atomic_set(&audio->out_bytes, 0);
}
static void audio_flush_pcm_buf(struct audio *audio)
{
uint8_t index;
for (index = 0; index < PCM_BUF_MAX_COUNT; index++)
audio->in[index].used = 0;
audio->read_next = 0;
audio->fill_next = 0;
}
static void audio_ioport_reset(struct audio *audio)
{
/* Make sure read/write thread are free from
* sleep and knowing that system is not able
* to process io request at the moment
*/
wake_up(&audio->write_wait);
mutex_lock(&audio->write_lock);
audio_flush(audio);
mutex_unlock(&audio->write_lock);
wake_up(&audio->read_wait);
mutex_lock(&audio->read_lock);
audio_flush_pcm_buf(audio);
mutex_unlock(&audio->read_lock);
}
static long audio_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
{
struct audio *audio = file->private_data;
int rc = 0;
pr_info("audio_ioctl() cmd = %d\n", cmd);
if (cmd == AUDIO_GET_STATS) {
struct msm_audio_stats stats;
stats.byte_count = audpp_avsync_byte_count(audio->dec_id);
stats.sample_count = audpp_avsync_sample_count(audio->dec_id);
if (copy_to_user((void *) arg, &stats, sizeof(stats)))
return -EFAULT;
return 0;
}
if (cmd == AUDIO_SET_VOLUME) {
unsigned long flags;
spin_lock_irqsave(&audio->dsp_lock, flags);
audio->volume = arg;
if (audio->running)
audpp_set_volume_and_pan(audio->dec_id, arg, 0);
spin_unlock_irqrestore(&audio->dsp_lock, flags);
return 0;
}
mutex_lock(&audio->lock);
switch (cmd) {
case AUDIO_START:
rc = audio_enable(audio);
break;
case AUDIO_STOP:
rc = audio_disable(audio);
audio->stopped = 1;
audio_ioport_reset(audio);
audio->stopped = 0;
break;
case AUDIO_FLUSH:
dprintk("%s: AUDIO_FLUSH\n", __func__);
audio->rflush = 1;
audio->wflush = 1;
audio_ioport_reset(audio);
audio->rflush = 0;
audio->wflush = 0;
if (audio->buf_refresh) {
audio->buf_refresh = 0;
audplay_buffer_refresh(audio);
}
break;
case AUDIO_SET_CONFIG: {
struct msm_audio_config config;
if (copy_from_user(&config, (void *) arg, sizeof(config))) {
rc = -EFAULT;
break;
}
if (config.channel_count == 1) {
config.channel_count = AUDPP_CMD_PCM_INTF_MONO_V;
} else if (config.channel_count == 2) {
config.channel_count = AUDPP_CMD_PCM_INTF_STEREO_V;
} else {
rc = -EINVAL;
break;
}
audio->out_sample_rate = config.sample_rate;
audio->out_channel_mode = config.channel_count;
rc = 0;
break;
}
case AUDIO_GET_CONFIG: {
struct msm_audio_config config;
config.buffer_size = (audio->out_dma_sz >> 1);
config.buffer_count = 2;
config.sample_rate = audio->out_sample_rate;
if (audio->out_channel_mode == AUDPP_CMD_PCM_INTF_MONO_V) {
config.channel_count = 1;
} else {
config.channel_count = 2;
}
config.unused[0] = 0;
config.unused[1] = 0;
config.unused[2] = 0;
config.unused[3] = 0;
if (copy_to_user((void *) arg, &config, sizeof(config))) {
rc = -EFAULT;
} else {
rc = 0;
}
break;
}
case AUDIO_GET_PCM_CONFIG:{
struct msm_audio_pcm_config config;
config.pcm_feedback = 0;
config.buffer_count = PCM_BUF_MAX_COUNT;
config.buffer_size = PCM_BUFSZ_MIN;
if (copy_to_user((void *)arg, &config,
sizeof(config)))
rc = -EFAULT;
else
rc = 0;
break;
}
case AUDIO_SET_PCM_CONFIG:{
struct msm_audio_pcm_config config;
if (copy_from_user
(&config, (void *)arg, sizeof(config))) {
rc = -EFAULT;
break;
}
if ((config.buffer_count > PCM_BUF_MAX_COUNT) ||
(config.buffer_count == 1))
config.buffer_count = PCM_BUF_MAX_COUNT;
if (config.buffer_size < PCM_BUFSZ_MIN)
config.buffer_size = PCM_BUFSZ_MIN;
/* Check if pcm feedback is required */
if ((config.pcm_feedback) && (!audio->read_data)) {
pr_info("ioctl: allocate PCM buffer %d\n",
config.buffer_count *
config.buffer_size);
audio->read_data =
dma_alloc_coherent(NULL,
config.buffer_size *
config.buffer_count,
&audio->read_phys,
GFP_KERNEL);
if (!audio->read_data) {
pr_err("audio_mp3: malloc pcm \
buf failed\n");
rc = -1;
} else {
uint8_t index;
uint32_t offset = 0;
audio->pcm_feedback = 1;
audio->buf_refresh = 0;
audio->pcm_buf_count =
config.buffer_count;
audio->read_next = 0;
audio->fill_next = 0;
for (index = 0;
index < config.buffer_count;
index++) {
audio->in[index].data =
audio->read_data + offset;
audio->in[index].addr =
audio->read_phys + offset;
audio->in[index].size =
config.buffer_size;
audio->in[index].used = 0;
offset += config.buffer_size;
}
rc = 0;
}
} else {
rc = 0;
}
break;
}
case AUDIO_PAUSE:
dprintk("%s: AUDIO_PAUSE %ld\n", __func__, arg);
rc = audpp_pause(audio->dec_id, (int) arg);
break;
default:
rc = -EINVAL;
}
mutex_unlock(&audio->lock);
return rc;
}
static ssize_t audio_read(struct file *file, char __user *buf, size_t count,
loff_t *pos)
{
struct audio *audio = file->private_data;
const char __user *start = buf;
int rc = 0;
if (!audio->pcm_feedback)
return 0; /* PCM feedback disabled. Nothing to read */
mutex_lock(&audio->read_lock);
pr_info("audio_read() %d \n", count);
while (count > 0) {
rc = wait_event_interruptible(audio->read_wait,
(audio->in[audio->read_next].
used > 0) || (audio->stopped)
|| (audio->rflush));
if (rc < 0)
break;
if (audio->stopped || audio->rflush) {
rc = -EBUSY;
break;
}
if (count < audio->in[audio->read_next].used) {
/* Read must happen in frame boundary. Since
* driver does not know frame size, read count
* must be greater or equal
* to size of PCM samples
*/
pr_info("audio_read: no partial frame done reading\n");
break;
} else {
pr_info("audio_read: read from in[%d]\n",
audio->read_next);
if (copy_to_user
(buf, audio->in[audio->read_next].data,
audio->in[audio->read_next].used)) {
pr_err("audio_read: invalid addr %x \n",
(unsigned int)buf);
rc = -EFAULT;
break;
}
count -= audio->in[audio->read_next].used;
buf += audio->in[audio->read_next].used;
audio->in[audio->read_next].used = 0;
if ((++audio->read_next) == audio->pcm_buf_count)
audio->read_next = 0;
if (audio->in[audio->read_next].used == 0)
break; /* No data ready at this moment
* Exit while loop to prevent
* output thread sleep too long
*/
}
}
/* don't feed output buffer to HW decoder during flushing
* buffer refresh command will be sent once flush completes
* send buf refresh command here can confuse HW decoder
*/
if (audio->buf_refresh && !audio->rflush) {
audio->buf_refresh = 0;
pr_info("audio_read: kick start pcm feedback again\n");
audplay_buffer_refresh(audio);
}
mutex_unlock(&audio->read_lock);
if (buf > start)
rc = buf - start;
pr_info("audio_read: read %d bytes\n", rc);
return rc;
}
static ssize_t audio_write(struct file *file, const char __user *buf,
size_t count, loff_t *pos)
{
struct audio *audio = file->private_data;
const char __user *start = buf;
struct buffer *frame;
size_t xfer;
char *cpy_ptr;
int rc = 0;
unsigned dsize;
mutex_lock(&audio->write_lock);
while (count > 0) {
frame = audio->out + audio->out_head;
cpy_ptr = frame->data;
dsize = 0;
rc = wait_event_interruptible(audio->write_wait,
(frame->used == 0)
|| (audio->stopped)
|| (audio->wflush));
if (rc < 0)
break;
if (audio->stopped || audio->wflush) {
rc = -EBUSY;
break;
}
if (audio->reserved) {
dprintk("%s: append reserved byte %x\n",
__func__, audio->rsv_byte);
*cpy_ptr = audio->rsv_byte;
xfer = (count > (frame->size - 1)) ?
frame->size - 1 : count;
cpy_ptr++;
dsize = 1;
audio->reserved = 0;
} else
xfer = (count > frame->size) ? frame->size : count;
if (copy_from_user(cpy_ptr, buf, xfer)) {
rc = -EFAULT;
break;
}
dsize += xfer;
if (dsize & 1) {
audio->rsv_byte = ((char *) frame->data)[dsize - 1];
dprintk("%s: odd length buf reserve last byte %x\n",
__func__, audio->rsv_byte);
audio->reserved = 1;
dsize--;
}
count -= xfer;
buf += xfer;
if (dsize > 0) {
audio->out_head ^= 1;
frame->used = dsize;
audplay_send_data(audio, 0);
}
}
mutex_unlock(&audio->write_lock);
if (buf > start)
return buf - start;
return rc;
}
static int audio_release(struct inode *inode, struct file *file)
{
struct audio *audio = file->private_data;
dprintk("audio_release()\n");
mutex_lock(&audio->lock);
audio_disable(audio);
audio_flush(audio);
audio_flush_pcm_buf(audio);
msm_adsp_put(audio->audplay);
audio->audplay = NULL;
audio->opened = 0;
audio->reserved = 0;
dma_free_coherent(NULL, audio->out_dma_sz, audio->data, audio->phys);
audio->data = NULL;
if (audio->read_data != NULL) {
dma_free_coherent(NULL,
audio->in[0].size * audio->pcm_buf_count,
audio->read_data, audio->read_phys);
audio->read_data = NULL;
}
audio->pcm_feedback = 0;
mutex_unlock(&audio->lock);
return 0;
}
static struct audio the_mp3_audio;
static int audio_open(struct inode *inode, struct file *file)
{
struct audio *audio = &the_mp3_audio;
int rc;
unsigned pmem_sz;
mutex_lock(&audio->lock);
if (audio->opened) {
pr_err("audio: busy\n");
rc = -EBUSY;
goto done;
}
pmem_sz = DMASZ_MAX;
while (pmem_sz >= DMASZ_MIN) {
audio->data = dma_alloc_coherent(NULL, pmem_sz,
&audio->phys, GFP_KERNEL);
if (audio->data)
break;
else if (pmem_sz == DMASZ_MIN) {
pr_err("audio: could not allocate DMA buffers\n");
rc = -ENOMEM;
goto done;
} else
pmem_sz >>= 1;
}
dprintk("%s: allocated %d bytes DMA buffer\n", __func__, pmem_sz);
rc = audmgr_open(&audio->audmgr);
if (rc) {
dma_free_coherent(NULL, pmem_sz,
audio->data, audio->phys);
goto done;
}
rc = msm_adsp_get("AUDPLAY0TASK", &audio->audplay, &audplay_adsp_ops,
audio);
if (rc) {
pr_err("audio: failed to get audplay0 dsp module\n");
dma_free_coherent(NULL, pmem_sz,
audio->data, audio->phys);
audmgr_close(&audio->audmgr);
goto done;
}
audio->out_dma_sz = pmem_sz;
pmem_sz >>= 1; /* Shift by 1 to get size of ping pong buffer */
audio->out_sample_rate = 44100;
audio->out_channel_mode = AUDPP_CMD_PCM_INTF_STEREO_V;
audio->dec_id = 0;
audio->out[0].data = audio->data + 0;
audio->out[0].addr = audio->phys + 0;
audio->out[0].size = pmem_sz;
audio->out[1].data = audio->data + pmem_sz;
audio->out[1].addr = audio->phys + pmem_sz;
audio->out[1].size = pmem_sz;
audio->volume = 0x2000; /* equal to Q13 number 1.0 Unit Gain */
audio_flush(audio);
file->private_data = audio;
audio->opened = 1;
rc = 0;
done:
mutex_unlock(&audio->lock);
return rc;
}
static struct file_operations audio_mp3_fops = {
.owner = THIS_MODULE,
.open = audio_open,
.release = audio_release,
.read = audio_read,
.write = audio_write,
.unlocked_ioctl = audio_ioctl,
};
struct miscdevice audio_mp3_misc = {
.minor = MISC_DYNAMIC_MINOR,
.name = "msm_mp3",
.fops = &audio_mp3_fops,
};
static int __init audio_init(void)
{
mutex_init(&the_mp3_audio.lock);
mutex_init(&the_mp3_audio.write_lock);
mutex_init(&the_mp3_audio.read_lock);
spin_lock_init(&the_mp3_audio.dsp_lock);
init_waitqueue_head(&the_mp3_audio.write_wait);
init_waitqueue_head(&the_mp3_audio.read_wait);
the_mp3_audio.read_data = NULL;
return misc_register(&audio_mp3_misc);
}
device_initcall(audio_init);