blob: 05bca03d6a9f57866c32842e7f189ad8c3d7e0f2 [file] [log] [blame]
/*
* aac audio decoder device
*
* Copyright (C) 2008 Google, Inc.
* Copyright (C) 2008 HTC Corporation
* Copyright (c) 2008-2012, Code Aurora Forum. All rights reserved.
*
* 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 <asm/atomic.h>
#include <asm/ioctls.h>
#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/debugfs.h>
#include <linux/delay.h>
#include <linux/list.h>
#include <linux/earlysuspend.h>
#include <linux/android_pmem.h>
#include <linux/slab.h>
#include <linux/msm_audio_aac.h>
#include <linux/memory_alloc.h>
#include <mach/msm_adsp.h>
#include <mach/iommu.h>
#include <mach/iommu_domains.h>
#include <mach/qdsp5v2/qdsp5audppmsg.h>
#include <mach/qdsp5v2/qdsp5audplaycmdi.h>
#include <mach/qdsp5v2/qdsp5audplaymsg.h>
#include <mach/qdsp5v2/audio_dev_ctl.h>
#include <mach/qdsp5v2/audpp.h>
#include <mach/qdsp5v2/audio_dev_ctl.h>
#include <mach/debug_mm.h>
#include <mach/msm_memtypes.h>
#define BUFSZ 32768
#define DMASZ (BUFSZ * 2)
#define BUFSZ_MIN 4096
#define DMASZ_MIN (BUFSZ_MIN * 2)
#define AUDPLAY_INVALID_READ_PTR_OFFSET 0xFFFF
#define AUDDEC_DEC_AAC 5
#define PCM_BUFSZ_MIN 9600 /* Hold one stereo AAC 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
#define AUDAAC_METAFIELD_MASK 0xFFFF0000
#define AUDAAC_EOS_FLG_OFFSET 0x0A /* Offset from beginning of buffer */
#define AUDAAC_EOS_FLG_MASK 0x01
#define AUDAAC_EOS_NONE 0x0 /* No EOS detected */
#define AUDAAC_EOS_SET 0x1 /* EOS set in meta field */
#define AUDAAC_EVENT_NUM 10 /* Default number of pre-allocated event packets */
#define BITSTREAM_ERROR_THRESHOLD_VALUE 0x1 /* DEFAULT THRESHOLD VALUE */
struct buffer {
void *data;
unsigned size;
unsigned used; /* Input usage actual DSP produced PCM size */
unsigned addr;
unsigned short mfield_sz; /*only useful for data has meta field */
};
#ifdef CONFIG_HAS_EARLYSUSPEND
struct audaac_suspend_ctl {
struct early_suspend node;
struct audio *audio;
};
#endif
struct audaac_event{
struct list_head list;
int event_type;
union msm_audio_event_payload payload;
};
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 */
int32_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 msm_audio_aac_config aac_config;
/* AV sync Info */
int avsync_flag; /* Flag to indicate feedback from DSP */
wait_queue_head_t avsync_wait;/* Wait queue for AV Sync Message */
/* 48 bits sample/bytes counter per channel */
uint16_t avsync[AUDPP_AVSYNC_CH_COUNT * AUDPP_AVSYNC_NUM_WORDS + 1];
/* data allocated for various buffers */
char *data;
int32_t phys; /* physical address of write buffer */
void *map_v_read;
void *map_v_write;
int mfield; /* meta field embedded in data */
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 teos; /* valid only if tunnel mode & no data left for decoder */
enum msm_aud_decoder_state dec_state; /* Represents decoder state */
int reserved; /* A byte is being reserved */
char rsv_byte; /* Handle odd length user data */
const char *module_name;
unsigned queue_id;
uint16_t dec_id;
uint32_t read_ptr_offset;
int16_t source;
#ifdef CONFIG_HAS_EARLYSUSPEND
struct audaac_suspend_ctl suspend_ctl;
#endif
#ifdef CONFIG_DEBUG_FS
struct dentry *dentry;
#endif
wait_queue_head_t wait;
struct list_head free_event_queue;
struct list_head event_queue;
wait_queue_head_t event_wait;
spinlock_t event_queue_lock;
struct mutex get_event_lock;
int event_abort;
uint32_t device_events;
struct msm_audio_bitstream_info stream_info;
struct msm_audio_bitstream_error_info bitstream_error_info;
uint32_t bitstream_error_threshold_value;
int eq_enable;
int eq_needs_commit;
struct audpp_cmd_cfg_object_params_eqalizer eq;
struct audpp_cmd_cfg_object_params_volume vol_pan;
};
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_error_threshold_config(struct audio *audio);
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);
static void audaac_post_event(struct audio *audio, int type,
union msm_audio_event_payload payload);
/* must be called with audio->lock held */
static int audio_enable(struct audio *audio)
{
MM_DBG("\n"); /* Macro prints the file name and function */
if (audio->enabled)
return 0;
audio->dec_state = MSM_AUD_DECODER_STATE_NONE;
audio->out_tail = 0;
audio->out_needed = 0;
if (msm_adsp_enable(audio->audplay)) {
MM_ERR("msm_adsp_enable(audplay) failed\n");
return -ENODEV;
}
if (audpp_enable(audio->dec_id, audio_dsp_event, audio)) {
MM_ERR("audpp_enable() failed\n");
msm_adsp_disable(audio->audplay);
return -ENODEV;
}
audio->enabled = 1;
return 0;
}
static void aac_listner(u32 evt_id, union auddev_evt_data *evt_payload,
void *private_data)
{
struct audio *audio = (struct audio *) private_data;
switch (evt_id) {
case AUDDEV_EVT_DEV_RDY:
MM_DBG(":AUDDEV_EVT_DEV_RDY\n");
audio->source |= (0x1 << evt_payload->routing_id);
if (audio->running == 1 && audio->enabled == 1)
audpp_route_stream(audio->dec_id, audio->source);
break;
case AUDDEV_EVT_DEV_RLS:
MM_DBG(":AUDDEV_EVT_DEV_RLS\n");
audio->source &= ~(0x1 << evt_payload->routing_id);
if (audio->running == 1 && audio->enabled == 1)
audpp_route_stream(audio->dec_id, audio->source);
break;
case AUDDEV_EVT_STREAM_VOL_CHG:
audio->vol_pan.volume = evt_payload->session_vol;
MM_DBG(":AUDDEV_EVT_STREAM_VOL_CHG, stream vol %d\n",
audio->vol_pan.volume);
if (audio->running)
audpp_dsp_set_vol_pan(audio->dec_id, &audio->vol_pan,
POPP);
break;
default:
MM_ERR(":ERROR:wrong event\n");
break;
}
}
/* must be called with audio->lock held */
static int audio_disable(struct audio *audio)
{
int rc = 0;
MM_DBG("\n"); /* Macro prints the file name and function */
if (audio->enabled) {
audio->enabled = 0;
audio->dec_state = MSM_AUD_DECODER_STATE_NONE;
auddec_dsp_config(audio, 0);
rc = wait_event_interruptible_timeout(audio->wait,
audio->dec_state != MSM_AUD_DECODER_STATE_NONE,
msecs_to_jiffies(MSM_AUD_DECODER_WAIT_MS));
if (rc == 0)
rc = -ETIMEDOUT;
else if (audio->dec_state != MSM_AUD_DECODER_STATE_CLOSE)
rc = -EFAULT;
else
rc = 0;
wake_up(&audio->write_wait);
wake_up(&audio->read_wait);
msm_adsp_disable(audio->audplay);
audpp_disable(audio->dec_id, audio);
audio->out_needed = 0;
}
return rc;
}
/* ------------------- dsp --------------------- */
static void audio_update_pcm_buf_entry(struct audio *audio, uint32_t *payload)
{
uint8_t index;
unsigned long flags;
if (audio->rflush)
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]) {
MM_DBG("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 {
MM_ERR("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 {
MM_DBG("read cannot keep up\n");
audio->buf_refresh = 1;
}
wake_up(&audio->read_wait);
spin_unlock_irqrestore(&audio->dsp_lock, flags);
}
static void audaac_bitstream_error_info(struct audio *audio, uint32_t *payload)
{
unsigned long flags;
union msm_audio_event_payload e_payload;
if (payload[0] != AUDDEC_DEC_AAC) {
MM_ERR("Unexpected bitstream error info from DSP:\
Invalid decoder\n");
return;
}
/* get stream info from DSP msg */
spin_lock_irqsave(&audio->dsp_lock, flags);
audio->bitstream_error_info.dec_id = payload[0];
audio->bitstream_error_info.err_msg_indicator = payload[1];
audio->bitstream_error_info.err_type = payload[2];
spin_unlock_irqrestore(&audio->dsp_lock, flags);
MM_ERR("bit_stream_error_type=%d error_count=%d\n",
audio->bitstream_error_info.err_type, (0x0000FFFF &
audio->bitstream_error_info.err_msg_indicator));
/* send event to ARM to notify error info coming */
e_payload.error_info = audio->bitstream_error_info;
audaac_post_event(audio, AUDIO_EVENT_BITSTREAM_ERROR_INFO, e_payload);
}
static void audaac_update_stream_info(struct audio *audio, uint32_t *payload)
{
unsigned long flags;
union msm_audio_event_payload e_payload;
/* get stream info from DSP msg */
spin_lock_irqsave(&audio->dsp_lock, flags);
audio->stream_info.codec_type = AUDIO_CODEC_TYPE_AAC;
audio->stream_info.chan_info = (0x0000FFFF & payload[1]);
audio->stream_info.sample_rate = (0x0000FFFF & payload[2]);
audio->stream_info.bit_stream_info = (0x0000FFFF & payload[3]);
audio->stream_info.bit_rate = payload[4];
spin_unlock_irqrestore(&audio->dsp_lock, flags);
MM_DBG("chan_info=%d, sample_rate=%d, bit_stream_info=%d\n",
audio->stream_info.chan_info,
audio->stream_info.sample_rate,
audio->stream_info.bit_stream_info);
/* send event to ARM to notify steam info coming */
e_payload.stream_info = audio->stream_info;
audaac_post_event(audio, AUDIO_EVENT_STREAM_INFO, e_payload);
}
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));
MM_DBG("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;
case AUDPLAY_UP_STREAM_INFO:
if ((msg[1] & AUDPLAY_STREAM_INFO_MSG_MASK) ==
AUDPLAY_STREAM_INFO_MSG_MASK) {
audaac_bitstream_error_info(audio, msg);
} else {
audaac_update_stream_info(audio, msg);
}
break;
case AUDPLAY_UP_OUTPORT_FLUSH_ACK:
MM_DBG("OUTPORT_FLUSH_ACK\n");
audio->rflush = 0;
wake_up(&audio->read_wait);
if (audio->pcm_feedback)
audplay_buffer_refresh(audio);
break;
case ADSP_MESSAGE_ID:
MM_DBG("Received ADSP event: module enable(audplaytask)\n");
break;
default:
MM_ERR("unexpected message from decoder \n");
}
}
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: {
uint16_t reason = msg[2];
MM_DBG("decoder status: sleep reason = \
0x%04x\n", reason);
if ((reason == AUDPP_MSG_REASON_MEM)
|| (reason ==
AUDPP_MSG_REASON_NODECODER)) {
audio->dec_state =
MSM_AUD_DECODER_STATE_FAILURE;
wake_up(&audio->wait);
} else if (reason == AUDPP_MSG_REASON_NONE) {
/* decoder is in disable state */
audio->dec_state =
MSM_AUD_DECODER_STATE_CLOSE;
wake_up(&audio->wait);
}
break;
}
case AUDPP_DEC_STATUS_INIT:
MM_DBG("decoder status: init \n");
if (audio->pcm_feedback)
audpp_cmd_cfg_routing_mode(audio);
else
audpp_cmd_cfg_adec_params(audio);
break;
case AUDPP_DEC_STATUS_CFG:
MM_DBG("decoder status: cfg \n");
break;
case AUDPP_DEC_STATUS_PLAY:
MM_DBG("decoder status: play \n");
/* send mixer command */
audpp_route_stream(audio->dec_id,
audio->source);
if (audio->pcm_feedback) {
audplay_error_threshold_config(audio);
audplay_config_hostpcm(audio);
audplay_buffer_refresh(audio);
}
audio->dec_state =
MSM_AUD_DECODER_STATE_SUCCESS;
wake_up(&audio->wait);
break;
default:
MM_ERR("unknown decoder status \n");
}
break;
}
case AUDPP_MSG_CFG_MSG:
if (msg[0] == AUDPP_MSG_ENA_ENA) {
MM_DBG("CFG_MSG ENABLE\n");
auddec_dsp_config(audio, 1);
audio->out_needed = 0;
audio->running = 1;
audpp_dsp_set_vol_pan(audio->dec_id, &audio->vol_pan,
POPP);
audpp_dsp_set_eq(audio->dec_id, audio->eq_enable,
&audio->eq, POPP);
} else if (msg[0] == AUDPP_MSG_ENA_DIS) {
MM_DBG("CFG_MSG DISABLE\n");
audio->running = 0;
} else {
MM_DBG("CFG_MSG %d?\n", msg[0]);
}
break;
case AUDPP_MSG_ROUTING_ACK:
MM_DBG("ROUTING_ACK mode=%d\n", msg[1]);
audpp_cmd_cfg_adec_params(audio);
break;
case AUDPP_MSG_FLUSH_ACK:
MM_DBG("FLUSH_ACK\n");
audio->wflush = 0;
audio->rflush = 0;
wake_up(&audio->write_wait);
if (audio->pcm_feedback)
audplay_buffer_refresh(audio);
break;
case AUDPP_MSG_PCMDMAMISSED:
MM_DBG("PCMDMAMISSED\n");
audio->teos = 1;
wake_up(&audio->write_wait);
break;
case AUDPP_MSG_AVSYNC_MSG:
MM_DBG("AUDPP_MSG_AVSYNC_MSG\n");
memcpy(&audio->avsync[0], msg, sizeof(audio->avsync));
audio->avsync_flag = 1;
wake_up(&audio->avsync_wait);
break;
default:
MM_ERR("UNKNOWN (%d)\n", id);
}
}
struct msm_adsp_ops audplay_adsp_ops_aac = {
.event = audplay_dsp_event,
};
#define audplay_send_queue0(audio, cmd, len) \
msm_adsp_write(audio->audplay, audio->queue_id,\
cmd, len)
static int auddec_dsp_config(struct audio *audio, int enable)
{
struct audpp_cmd_cfg_dec_type cfg_dec_cmd;
memset(&cfg_dec_cmd, 0, sizeof(cfg_dec_cmd));
cfg_dec_cmd.cmd_id = AUDPP_CMD_CFG_DEC_TYPE;
if (enable)
cfg_dec_cmd.dec_cfg = AUDPP_CMD_UPDATDE_CFG_DEC |
AUDPP_CMD_ENA_DEC_V | AUDDEC_DEC_AAC;
else
cfg_dec_cmd.dec_cfg = AUDPP_CMD_UPDATDE_CFG_DEC |
AUDPP_CMD_DIS_DEC_V;
cfg_dec_cmd.dm_mode = 0x0;
cfg_dec_cmd.stream_id = audio->dec_id;
return audpp_send_queue1(&cfg_dec_cmd, sizeof(cfg_dec_cmd));
}
static void audpp_cmd_cfg_adec_params(struct audio *audio)
{
struct audpp_cmd_cfg_adec_params_aac cmd;
memset(&cmd, 0, sizeof(cmd));
cmd.common.cmd_id = AUDPP_CMD_CFG_ADEC_PARAMS;
cmd.common.length = AUDPP_CMD_CFG_ADEC_PARAMS_AAC_LEN;
cmd.common.dec_id = audio->dec_id;
cmd.common.input_sampling_frequency = audio->out_sample_rate;
cmd.format = audio->aac_config.format;
cmd.audio_object = audio->aac_config.audio_object;
cmd.ep_config = audio->aac_config.ep_config;
cmd.aac_section_data_resilience_flag =
audio->aac_config.aac_section_data_resilience_flag;
cmd.aac_scalefactor_data_resilience_flag =
audio->aac_config.aac_scalefactor_data_resilience_flag;
cmd.aac_spectral_data_resilience_flag =
audio->aac_config.aac_spectral_data_resilience_flag;
cmd.sbr_on_flag = audio->aac_config.sbr_on_flag;
cmd.sbr_ps_on_flag = audio->aac_config.sbr_ps_on_flag;
cmd.channel_configuration = audio->aac_config.channel_configuration;
audpp_send_queue2(&cmd, sizeof(cmd));
}
static void audpp_cmd_cfg_routing_mode(struct audio *audio)
{
struct audpp_cmd_routing_mode cmd;
MM_DBG("\n"); /* Macro prints the file name and function */
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)
{
struct audplay_cmd_bitstream_data_avail_nt2 cmd;
cmd.cmd_id = AUDPLAY_CMD_BITSTREAM_DATA_AVAIL_NT2;
if (audio->mfield)
cmd.decoder_id = AUDAAC_METAFIELD_MASK |
(audio->out[idx].mfield_sz >> 1);
else
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;
/* AAC frame size */
refresh_cmd.buf0_length = audio->in[audio->fill_next].size -
(audio->in[audio->fill_next].size % 1024)
+ (audio->mfield ? 24 : 0);
refresh_cmd.buf_read_count = 0;
MM_DBG("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_outport_flush(struct audio *audio)
{
struct audplay_cmd_outport_flush op_flush_cmd;
MM_DBG("\n"); /* Macro prints the file name and function */
op_flush_cmd.cmd_id = AUDPLAY_CMD_OUTPORT_FLUSH;
(void)audplay_send_queue0(audio, &op_flush_cmd, sizeof(op_flush_cmd));
}
static void audplay_error_threshold_config(struct audio *audio)
{
union audplay_cmd_channel_info ch_cfg_cmd;
MM_DBG("\n"); /* Macro prints the file name and function */
ch_cfg_cmd.thr_update.cmd_id = AUDPLAY_CMD_CHANNEL_INFO;
ch_cfg_cmd.thr_update.threshold_update = AUDPLAY_ERROR_THRESHOLD_ENABLE;
ch_cfg_cmd.thr_update.threshold_value =
audio->bitstream_error_threshold_value;
(void)audplay_send_queue0(audio, &ch_cfg_cmd, sizeof(ch_cfg_cmd));
}
static void audplay_config_hostpcm(struct audio *audio)
{
struct audplay_cmd_hpcm_buf_cfg cfg_cmd;
MM_DBG("\n"); /* Macro prints the file name and function */
cfg_cmd.cmd_id = AUDPLAY_CMD_HPCM_BUF_CFG;
cfg_cmd.max_buffers = audio->pcm_buf_count;
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 (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) {
MM_DBG("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);
MM_DBG("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;
audio->out_needed = 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->buf_refresh = 0;
audio->read_next = 0;
audio->fill_next = 0;
}
static int audaac_validate_usr_config(struct msm_audio_aac_config *config)
{
int ret_val = -1;
if (config->format != AUDIO_AAC_FORMAT_ADTS &&
config->format != AUDIO_AAC_FORMAT_RAW &&
config->format != AUDIO_AAC_FORMAT_PSUEDO_RAW &&
config->format != AUDIO_AAC_FORMAT_LOAS)
goto done;
if (config->audio_object != AUDIO_AAC_OBJECT_LC &&
config->audio_object != AUDIO_AAC_OBJECT_LTP &&
config->audio_object != AUDIO_AAC_OBJECT_BSAC &&
config->audio_object != AUDIO_AAC_OBJECT_ERLC)
goto done;
if (config->audio_object == AUDIO_AAC_OBJECT_ERLC) {
if (config->ep_config > 3)
goto done;
if (config->aac_scalefactor_data_resilience_flag !=
AUDIO_AAC_SCA_DATA_RES_OFF &&
config->aac_scalefactor_data_resilience_flag !=
AUDIO_AAC_SCA_DATA_RES_ON)
goto done;
if (config->aac_section_data_resilience_flag !=
AUDIO_AAC_SEC_DATA_RES_OFF &&
config->aac_section_data_resilience_flag !=
AUDIO_AAC_SEC_DATA_RES_ON)
goto done;
if (config->aac_spectral_data_resilience_flag !=
AUDIO_AAC_SPEC_DATA_RES_OFF &&
config->aac_spectral_data_resilience_flag !=
AUDIO_AAC_SPEC_DATA_RES_ON)
goto done;
} else {
config->aac_section_data_resilience_flag =
AUDIO_AAC_SEC_DATA_RES_OFF;
config->aac_scalefactor_data_resilience_flag =
AUDIO_AAC_SCA_DATA_RES_OFF;
config->aac_spectral_data_resilience_flag =
AUDIO_AAC_SPEC_DATA_RES_OFF;
}
#ifndef CONFIG_AUDIO_AAC_PLUS
if (AUDIO_AAC_SBR_ON_FLAG_OFF != config->sbr_on_flag)
goto done;
#else
if (config->sbr_on_flag != AUDIO_AAC_SBR_ON_FLAG_OFF &&
config->sbr_on_flag != AUDIO_AAC_SBR_ON_FLAG_ON)
goto done;
#endif
#ifndef CONFIG_AUDIO_ENHANCED_AAC_PLUS
if (AUDIO_AAC_SBR_PS_ON_FLAG_OFF != config->sbr_ps_on_flag)
goto done;
#else
if (config->sbr_ps_on_flag != AUDIO_AAC_SBR_PS_ON_FLAG_OFF &&
config->sbr_ps_on_flag != AUDIO_AAC_SBR_PS_ON_FLAG_ON)
goto done;
#endif
if (config->dual_mono_mode > AUDIO_AAC_DUAL_MONO_PL_SR)
goto done;
if (config->channel_configuration > 2)
goto done;
ret_val = 0;
done:
return ret_val;
}
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);
audio->avsync_flag = 1;
wake_up(&audio->avsync_wait);
}
static int audaac_events_pending(struct audio *audio)
{
unsigned long flags;
int empty;
spin_lock_irqsave(&audio->event_queue_lock, flags);
empty = !list_empty(&audio->event_queue);
spin_unlock_irqrestore(&audio->event_queue_lock, flags);
return empty || audio->event_abort;
}
static void audaac_reset_event_queue(struct audio *audio)
{
unsigned long flags;
struct audaac_event *drv_evt;
struct list_head *ptr, *next;
spin_lock_irqsave(&audio->event_queue_lock, flags);
list_for_each_safe(ptr, next, &audio->event_queue) {
drv_evt = list_first_entry(&audio->event_queue,
struct audaac_event, list);
list_del(&drv_evt->list);
kfree(drv_evt);
}
list_for_each_safe(ptr, next, &audio->free_event_queue) {
drv_evt = list_first_entry(&audio->free_event_queue,
struct audaac_event, list);
list_del(&drv_evt->list);
kfree(drv_evt);
}
spin_unlock_irqrestore(&audio->event_queue_lock, flags);
return;
}
static long audaac_process_event_req(struct audio *audio, void __user *arg)
{
long rc;
struct msm_audio_event usr_evt;
struct audaac_event *drv_evt = NULL;
int timeout;
unsigned long flags;
if (copy_from_user(&usr_evt, arg, sizeof(struct msm_audio_event)))
return -EFAULT;
timeout = (int) usr_evt.timeout_ms;
if (timeout > 0) {
rc = wait_event_interruptible_timeout(
audio->event_wait, audaac_events_pending(audio),
msecs_to_jiffies(timeout));
if (rc == 0)
return -ETIMEDOUT;
} else {
rc = wait_event_interruptible(
audio->event_wait, audaac_events_pending(audio));
}
if (rc < 0)
return rc;
if (audio->event_abort) {
audio->event_abort = 0;
return -ENODEV;
}
rc = 0;
spin_lock_irqsave(&audio->event_queue_lock, flags);
if (!list_empty(&audio->event_queue)) {
drv_evt = list_first_entry(&audio->event_queue,
struct audaac_event, list);
list_del(&drv_evt->list);
}
if (drv_evt) {
usr_evt.event_type = drv_evt->event_type;
usr_evt.event_payload = drv_evt->payload;
list_add_tail(&drv_evt->list, &audio->free_event_queue);
} else
rc = -1;
spin_unlock_irqrestore(&audio->event_queue_lock, flags);
if (!rc && copy_to_user(arg, &usr_evt, sizeof(usr_evt)))
rc = -EFAULT;
return rc;
}
static int audio_enable_eq(struct audio *audio, int enable)
{
if (audio->eq_enable == enable && !audio->eq_needs_commit)
return 0;
audio->eq_enable = enable;
if (audio->running) {
audpp_dsp_set_eq(audio->dec_id, enable, &audio->eq, POPP);
audio->eq_needs_commit = 0;
}
return 0;
}
static int audio_get_avsync_data(struct audio *audio,
struct msm_audio_stats *stats)
{
int rc = -EINVAL;
unsigned long flags;
local_irq_save(flags);
if (audio->dec_id == audio->avsync[0] && audio->avsync_flag) {
/* av_sync sample count */
stats->sample_count = (audio->avsync[2] << 16) |
(audio->avsync[3]);
/* av_sync byte_count */
stats->byte_count = (audio->avsync[5] << 16) |
(audio->avsync[6]);
audio->avsync_flag = 0;
rc = 0;
}
local_irq_restore(flags);
return rc;
}
static long audio_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
{
struct audio *audio = file->private_data;
int rc = -EINVAL;
unsigned long flags = 0;
uint16_t enable_mask;
int enable;
int prev_state;
MM_DBG("cmd = %d\n", cmd);
if (cmd == AUDIO_GET_STATS) {
struct msm_audio_stats stats;
audio->avsync_flag = 0;
memset(&stats, 0, sizeof(stats));
if (audpp_query_avsync(audio->dec_id) < 0)
return rc;
rc = wait_event_interruptible_timeout(audio->avsync_wait,
(audio->avsync_flag == 1),
msecs_to_jiffies(AUDPP_AVSYNC_EVENT_TIMEOUT));
if (rc < 0)
return rc;
else if ((rc > 0) || ((rc == 0) && (audio->avsync_flag == 1))) {
if (audio_get_avsync_data(audio, &stats) < 0)
return rc;
if (copy_to_user((void *)arg, &stats, sizeof(stats)))
return -EFAULT;
return 0;
} else
return -EAGAIN;
}
switch (cmd) {
case AUDIO_ENABLE_AUDPP:
if (copy_from_user(&enable_mask, (void *) arg,
sizeof(enable_mask))) {
rc = -EFAULT;
break;
}
spin_lock_irqsave(&audio->dsp_lock, flags);
enable = (enable_mask & EQ_ENABLE) ? 1 : 0;
audio_enable_eq(audio, enable);
spin_unlock_irqrestore(&audio->dsp_lock, flags);
rc = 0;
break;
case AUDIO_SET_VOLUME:
spin_lock_irqsave(&audio->dsp_lock, flags);
audio->vol_pan.volume = arg;
if (audio->running)
audpp_dsp_set_vol_pan(audio->dec_id, &audio->vol_pan,
POPP);
spin_unlock_irqrestore(&audio->dsp_lock, flags);
rc = 0;
break;
case AUDIO_SET_PAN:
spin_lock_irqsave(&audio->dsp_lock, flags);
audio->vol_pan.pan = arg;
if (audio->running)
audpp_dsp_set_vol_pan(audio->dec_id, &audio->vol_pan,
POPP);
spin_unlock_irqrestore(&audio->dsp_lock, flags);
rc = 0;
break;
case AUDIO_SET_EQ:
prev_state = audio->eq_enable;
audio->eq_enable = 0;
if (copy_from_user(&audio->eq.num_bands, (void *) arg,
sizeof(audio->eq) -
(AUDPP_CMD_CFG_OBJECT_PARAMS_COMMON_LEN + 2))) {
rc = -EFAULT;
break;
}
audio->eq_enable = prev_state;
audio->eq_needs_commit = 1;
rc = 0;
break;
}
if (-EINVAL != rc)
return rc;
if (cmd == AUDIO_GET_EVENT) {
MM_DBG("AUDIO_GET_EVENT\n");
if (mutex_trylock(&audio->get_event_lock)) {
rc = audaac_process_event_req(audio,
(void __user *) arg);
mutex_unlock(&audio->get_event_lock);
} else
rc = -EBUSY;
return rc;
}
if (cmd == AUDIO_ABORT_GET_EVENT) {
audio->event_abort = 1;
wake_up(&audio->event_wait);
return 0;
}
mutex_lock(&audio->lock);
switch (cmd) {
case AUDIO_START:
MM_DBG("AUDIO_START\n");
rc = audio_enable(audio);
if (!rc) {
rc = wait_event_interruptible_timeout(audio->wait,
audio->dec_state != MSM_AUD_DECODER_STATE_NONE,
msecs_to_jiffies(MSM_AUD_DECODER_WAIT_MS));
MM_INFO("dec_state %d rc = %d\n", audio->dec_state, rc);
if (audio->dec_state != MSM_AUD_DECODER_STATE_SUCCESS)
rc = -ENODEV;
else
rc = 0;
}
break;
case AUDIO_STOP:
MM_DBG("AUDIO_STOP\n");
rc = audio_disable(audio);
audio->stopped = 1;
audio_ioport_reset(audio);
audio->stopped = 0;
break;
case AUDIO_FLUSH:
MM_DBG("AUDIO_FLUSH running=%d\n", audio->running);
audio->rflush = 1;
audio->wflush = 1;
audio_ioport_reset(audio);
if (audio->running) {
audpp_flush(audio->dec_id);
rc = wait_event_interruptible(audio->write_wait,
!audio->wflush);
if (rc < 0) {
MM_ERR("AUDIO_FLUSH interrupted\n");
rc = -EINTR;
}
} else {
audio->rflush = 0;
audio->wflush = 0;
}
break;
case AUDIO_OUTPORT_FLUSH:
MM_DBG("AUDIO_OUTPORT_FLUSH\n");
audio->rflush = 1;
wake_up(&audio->read_wait);
mutex_lock(&audio->read_lock);
audio_flush_pcm_buf(audio);
mutex_unlock(&audio->read_lock);
audplay_outport_flush(audio);
rc = wait_event_interruptible(audio->read_wait,
!audio->rflush);
if (rc < 0) {
MM_ERR("AUDPLAY_OUTPORT_FLUSH interrupted\n");
rc = -EINTR;
}
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;
audio->mfield = config.meta_field;
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.meta_field = 0;
config.unused[0] = 0;
config.unused[1] = 0;
config.unused[2] = 0;
if (copy_to_user((void *)arg, &config,
sizeof(config)))
rc = -EFAULT;
else
rc = 0;
break;
}
case AUDIO_GET_AAC_CONFIG:{
if (copy_to_user((void *)arg, &audio->aac_config,
sizeof(audio->aac_config)))
rc = -EFAULT;
else
rc = 0;
break;
}
case AUDIO_SET_AAC_CONFIG:{
struct msm_audio_aac_config usr_config;
if (copy_from_user
(&usr_config, (void *)arg,
sizeof(usr_config))) {
rc = -EFAULT;
break;
}
if (audaac_validate_usr_config(&usr_config) == 0) {
audio->aac_config = usr_config;
rc = 0;
} else
rc = -EINVAL;
break;
}
case AUDIO_GET_PCM_CONFIG:{
struct msm_audio_pcm_config config;
config.pcm_feedback = audio->pcm_feedback;
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.pcm_feedback != audio->pcm_feedback) {
MM_ERR("Not sufficient permission to"
"change the playback mode\n");
rc = -EACCES;
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->buf_refresh = 0;
audio->read_next = 0;
audio->fill_next = 0;
}
rc = 0;
break;
}
case AUDIO_PAUSE:
MM_DBG("AUDIO_PAUSE %ld\n", arg);
rc = audpp_pause(audio->dec_id, (int) arg);
break;
case AUDIO_GET_STREAM_INFO:{
if (audio->stream_info.sample_rate == 0) {
/* haven't received DSP stream event,
the stream info is not updated */
rc = -EPERM;
break;
}
if (copy_to_user((void *)arg, &audio->stream_info,
sizeof(struct msm_audio_bitstream_info)))
rc = -EFAULT;
else
rc = 0;
break;
}
case AUDIO_GET_BITSTREAM_ERROR_INFO:{
if ((audio->bitstream_error_info.err_msg_indicator &
AUDPLAY_STREAM_INFO_MSG_MASK) ==
AUDPLAY_STREAM_INFO_MSG_MASK) {
/* haven't received bitstream error info event,
the bitstream error info is not updated */
rc = -EPERM;
break;
}
if (copy_to_user((void *)arg, &audio->bitstream_error_info,
sizeof(struct msm_audio_bitstream_error_info)))
rc = -EFAULT;
else
rc = 0;
break;
}
case AUDIO_GET_SESSION_ID:
if (copy_to_user((void *) arg, &audio->dec_id,
sizeof(unsigned short)))
rc = -EFAULT;
else
rc = 0;
break;
case AUDIO_SET_ERR_THRESHOLD_VALUE:
if (copy_from_user(&audio->bitstream_error_threshold_value,
(void *)arg, sizeof(uint32_t)))
rc = -EFAULT;
else
rc = 0;
break;
default:
rc = -EINVAL;
}
mutex_unlock(&audio->lock);
return rc;
}
/* Only useful in tunnel-mode */
static int audaac_fsync(struct file *file, loff_t ppos1, loff_t ppos2, int datasync)
{
struct audio *audio = file->private_data;
struct buffer *frame;
int rc = 0;
MM_DBG("\n"); /* Macro prints the file name and function */
if (!audio->running || audio->pcm_feedback) {
rc = -EINVAL;
goto done_nolock;
}
mutex_lock(&audio->write_lock);
rc = wait_event_interruptible(audio->write_wait,
(!audio->out[0].used &&
!audio->out[1].used &&
audio->out_needed) || audio->wflush);
if (rc < 0)
goto done;
else if (audio->wflush) {
rc = -EBUSY;
goto done;
}
if (audio->reserved) {
MM_DBG("send reserved byte\n");
frame = audio->out + audio->out_tail;
((char *) frame->data)[0] = audio->rsv_byte;
((char *) frame->data)[1] = 0;
frame->used = 2;
audplay_send_data(audio, 0);
rc = wait_event_interruptible(audio->write_wait,
(!audio->out[0].used &&
!audio->out[1].used &&
audio->out_needed) || audio->wflush);
if (rc < 0)
goto done;
else if (audio->wflush) {
rc = -EBUSY;
goto done;
}
}
/* pcm dmamiss message is sent continously
* when decoder is starved so no race
* condition concern
*/
audio->teos = 0;
rc = wait_event_interruptible(audio->write_wait,
audio->teos || audio->wflush);
if (audio->wflush)
rc = -EBUSY;
done:
mutex_unlock(&audio->write_lock);
done_nolock:
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 is not enabled. Nothing to read */
mutex_lock(&audio->read_lock);
MM_DBG("to read %d \n", count);
while (count > 0) {
rc = wait_event_interruptible_timeout(audio->read_wait,
(audio->in[audio->read_next].
used > 0) || (audio->stopped)
|| (audio->rflush),
msecs_to_jiffies(MSM_AUD_BUFFER_UPDATE_WAIT_MS));
if (rc == 0) {
rc = -ETIMEDOUT;
break;
} else 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 */
MM_DBG("no partial frame done reading\n");
break;
} else {
MM_DBG("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)) {
MM_ERR("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;
break;
/*
* Force to exit while loop
* to prevent output thread
* sleep too long if data is not
* ready at this moment.
*/
}
}
/* 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;
MM_DBG("kick start pcm feedback again\n");
audplay_buffer_refresh(audio);
}
mutex_unlock(&audio->read_lock);
if (buf > start)
rc = buf - start;
MM_DBG("read %d bytes\n", rc);
return rc;
}
static int audaac_process_eos(struct audio *audio,
const char __user *buf_start, unsigned short mfield_size)
{
struct buffer *frame;
char *buf_ptr;
int rc = 0;
MM_DBG("signal input EOS reserved=%d\n", audio->reserved);
if (audio->reserved) {
MM_DBG("Pass reserve byte\n");
frame = audio->out + audio->out_head;
buf_ptr = frame->data;
rc = wait_event_interruptible(audio->write_wait,
(frame->used == 0)
|| (audio->stopped)
|| (audio->wflush));
if (rc < 0)
goto done;
if (audio->stopped || audio->wflush) {
rc = -EBUSY;
goto done;
}
buf_ptr[0] = audio->rsv_byte;
buf_ptr[1] = 0;
audio->out_head ^= 1;
frame->mfield_sz = 0;
audio->reserved = 0;
frame->used = 2;
audplay_send_data(audio, 0);
}
MM_DBG("Now signal input EOS after reserved bytes %d %d %d\n",
audio->out[0].used, audio->out[1].used, audio->out_needed);
frame = audio->out + audio->out_head;
rc = wait_event_interruptible(audio->write_wait,
(audio->out_needed &&
audio->out[0].used == 0 &&
audio->out[1].used == 0)
|| (audio->stopped)
|| (audio->wflush));
if (rc < 0)
goto done;
if (audio->stopped || audio->wflush) {
rc = -EBUSY;
goto done;
}
if (copy_from_user(frame->data, buf_start, mfield_size)) {
rc = -EFAULT;
goto done;
}
frame->mfield_sz = mfield_size;
audio->out_head ^= 1;
frame->used = mfield_size;
audplay_send_data(audio, 0);
done:
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, eos_condition = AUDAAC_EOS_NONE;
unsigned dsize;
unsigned short mfield_size = 0;
MM_DBG("cnt=%d\n", count);
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->mfield) {
if (buf == start) {
/* Processing beginning of user buffer */
if (__get_user(mfield_size,
(unsigned short __user *) buf)) {
rc = -EFAULT;
break;
} else if (mfield_size > count) {
rc = -EINVAL;
break;
}
MM_DBG("mf offset_val %x\n", mfield_size);
if (copy_from_user(cpy_ptr, buf, mfield_size)) {
rc = -EFAULT;
break;
}
/* Check if EOS flag is set and buffer has
* contains just meta field
*/
if (cpy_ptr[AUDAAC_EOS_FLG_OFFSET] &
AUDAAC_EOS_FLG_MASK) {
MM_DBG("eos set\n");
eos_condition = AUDAAC_EOS_SET;
if (mfield_size == count) {
buf += mfield_size;
break;
} else
cpy_ptr[AUDAAC_EOS_FLG_OFFSET] &=
~AUDAAC_EOS_FLG_MASK;
}
/* Check EOS to see if */
cpy_ptr += mfield_size;
count -= mfield_size;
dsize += mfield_size;
buf += mfield_size;
} else {
mfield_size = 0;
MM_DBG("continuous buffer\n");
}
frame->mfield_sz = mfield_size;
}
if (audio->reserved) {
MM_DBG("append reserved byte %x\n",
audio->rsv_byte);
*cpy_ptr = audio->rsv_byte;
xfer = (count > ((frame->size - mfield_size) - 1)) ?
(frame->size - mfield_size) - 1 : count;
cpy_ptr++;
dsize += 1;
audio->reserved = 0;
} else
xfer = (count > (frame->size - mfield_size)) ?
(frame->size - mfield_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];
MM_DBG("odd length buf reserve last byte %x\n",
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);
}
}
MM_DBG("eos_condition %x buf[0x%x] start[0x%x]\n", eos_condition,
(int) buf, (int) start);
if (eos_condition == AUDAAC_EOS_SET)
rc = audaac_process_eos(audio, start, mfield_size);
mutex_unlock(&audio->write_lock);
if (!rc) {
if (buf > start)
return buf - start;
}
return rc;
}
static int audio_release(struct inode *inode, struct file *file)
{
struct audio *audio = file->private_data;
MM_INFO("audio instance 0x%08x freeing\n", (int)audio);
mutex_lock(&audio->lock);
auddev_unregister_evt_listner(AUDDEV_CLNT_DEC, audio->dec_id);
audio_disable(audio);
audio_flush(audio);
audio_flush_pcm_buf(audio);
msm_adsp_put(audio->audplay);
audpp_adec_free(audio->dec_id);
#ifdef CONFIG_HAS_EARLYSUSPEND
unregister_early_suspend(&audio->suspend_ctl.node);
#endif
audio->event_abort = 1;
wake_up(&audio->event_wait);
audaac_reset_event_queue(audio);
iounmap(audio->map_v_write);
free_contiguous_memory_by_paddr(audio->phys);
iounmap(audio->map_v_read);
free_contiguous_memory_by_paddr(audio->read_phys);
mutex_unlock(&audio->lock);
#ifdef CONFIG_DEBUG_FS
if (audio->dentry)
debugfs_remove(audio->dentry);
#endif
kfree(audio);
return 0;
}
static void audaac_post_event(struct audio *audio, int type,
union msm_audio_event_payload payload)
{
struct audaac_event *e_node = NULL;
unsigned long flags;
spin_lock_irqsave(&audio->event_queue_lock, flags);
if (!list_empty(&audio->free_event_queue)) {
e_node = list_first_entry(&audio->free_event_queue,
struct audaac_event, list);
list_del(&e_node->list);
} else {
e_node = kmalloc(sizeof(struct audaac_event), GFP_ATOMIC);
if (!e_node) {
MM_ERR("No mem to post event %d\n", type);
return;
}
}
e_node->event_type = type;
e_node->payload = payload;
list_add_tail(&e_node->list, &audio->event_queue);
spin_unlock_irqrestore(&audio->event_queue_lock, flags);
wake_up(&audio->event_wait);
}
#ifdef CONFIG_HAS_EARLYSUSPEND
static void audaac_suspend(struct early_suspend *h)
{
struct audaac_suspend_ctl *ctl =
container_of(h, struct audaac_suspend_ctl, node);
union msm_audio_event_payload payload;
MM_DBG("\n"); /* Macro prints the file name and function */
audaac_post_event(ctl->audio, AUDIO_EVENT_SUSPEND, payload);
}
static void audaac_resume(struct early_suspend *h)
{
struct audaac_suspend_ctl *ctl =
container_of(h, struct audaac_suspend_ctl, node);
union msm_audio_event_payload payload;
MM_DBG("\n"); /* Macro prints the file name and function */
audaac_post_event(ctl->audio, AUDIO_EVENT_RESUME, payload);
}
#endif
#ifdef CONFIG_DEBUG_FS
static ssize_t audaac_debug_open(struct inode *inode, struct file *file)
{
file->private_data = inode->i_private;
return 0;
}
static ssize_t audaac_debug_read(struct file *file, char __user *buf,
size_t count, loff_t *ppos)
{
const int debug_bufmax = 1024;
static char buffer[1024];
int n = 0, i;
struct audio *audio = file->private_data;
mutex_lock(&audio->lock);
n = scnprintf(buffer, debug_bufmax, "opened %d\n", audio->opened);
n += scnprintf(buffer + n, debug_bufmax - n,
"enabled %d\n", audio->enabled);
n += scnprintf(buffer + n, debug_bufmax - n,
"stopped %d\n", audio->stopped);
n += scnprintf(buffer + n, debug_bufmax - n,
"pcm_feedback %d\n", audio->pcm_feedback);
n += scnprintf(buffer + n, debug_bufmax - n,
"out_buf_sz %d\n", audio->out[0].size);
n += scnprintf(buffer + n, debug_bufmax - n,
"pcm_buf_count %d \n", audio->pcm_buf_count);
n += scnprintf(buffer + n, debug_bufmax - n,
"pcm_buf_sz %d \n", audio->in[0].size);
n += scnprintf(buffer + n, debug_bufmax - n,
"volume %x \n", audio->vol_pan.volume);
n += scnprintf(buffer + n, debug_bufmax - n,
"sample rate %d \n", audio->out_sample_rate);
n += scnprintf(buffer + n, debug_bufmax - n,
"channel mode %d \n", audio->out_channel_mode);
mutex_unlock(&audio->lock);
/* Following variables are only useful for debugging when
* when playback halts unexpectedly. Thus, no mutual exclusion
* enforced
*/
n += scnprintf(buffer + n, debug_bufmax - n,
"wflush %d\n", audio->wflush);
n += scnprintf(buffer + n, debug_bufmax - n,
"rflush %d\n", audio->rflush);
n += scnprintf(buffer + n, debug_bufmax - n,
"running %d \n", audio->running);
n += scnprintf(buffer + n, debug_bufmax - n,
"dec state %d \n", audio->dec_state);
n += scnprintf(buffer + n, debug_bufmax - n,
"out_needed %d \n", audio->out_needed);
n += scnprintf(buffer + n, debug_bufmax - n,
"out_head %d \n", audio->out_head);
n += scnprintf(buffer + n, debug_bufmax - n,
"out_tail %d \n", audio->out_tail);
n += scnprintf(buffer + n, debug_bufmax - n,
"out[0].used %d \n", audio->out[0].used);
n += scnprintf(buffer + n, debug_bufmax - n,
"out[1].used %d \n", audio->out[1].used);
n += scnprintf(buffer + n, debug_bufmax - n,
"buffer_refresh %d \n", audio->buf_refresh);
n += scnprintf(buffer + n, debug_bufmax - n,
"read_next %d \n", audio->read_next);
n += scnprintf(buffer + n, debug_bufmax - n,
"fill_next %d \n", audio->fill_next);
for (i = 0; i < audio->pcm_buf_count; i++)
n += scnprintf(buffer + n, debug_bufmax - n,
"in[%d].used %d \n", i, audio->in[i].used);
buffer[n] = 0;
return simple_read_from_buffer(buf, count, ppos, buffer, n);
}
static const struct file_operations audaac_debug_fops = {
.read = audaac_debug_read,
.open = audaac_debug_open,
};
#endif
static int audio_open(struct inode *inode, struct file *file)
{
struct audio *audio = NULL;
int rc, dec_attrb, decid, index, offset = 0;
unsigned pmem_sz = DMASZ;
struct audaac_event *e_node = NULL;
#ifdef CONFIG_DEBUG_FS
/* 4 bytes represents decoder number, 1 byte for terminate string */
char name[sizeof "msm_aac_" + 5];
#endif
/* Allocate audio instance, set to zero */
audio = kzalloc(sizeof(struct audio), GFP_KERNEL);
if (!audio) {
MM_ERR("no memory to allocate audio instance \n");
rc = -ENOMEM;
goto done;
}
MM_INFO("audio instance 0x%08x created\n", (int)audio);
/* Allocate the decoder */
dec_attrb = AUDDEC_DEC_AAC;
if ((file->f_mode & FMODE_WRITE) &&
(file->f_mode & FMODE_READ)) {
dec_attrb |= MSM_AUD_MODE_NONTUNNEL;
audio->pcm_feedback = NON_TUNNEL_MODE_PLAYBACK;
} else if ((file->f_mode & FMODE_WRITE) &&
!(file->f_mode & FMODE_READ)) {
dec_attrb |= MSM_AUD_MODE_TUNNEL;
audio->pcm_feedback = TUNNEL_MODE_PLAYBACK;
} else {
kfree(audio);
rc = -EACCES;
goto done;
}
decid = audpp_adec_alloc(dec_attrb, &audio->module_name,
&audio->queue_id);
if (decid < 0) {
MM_ERR("No free decoder available, freeing instance 0x%08x\n",
(int)audio);
rc = -ENODEV;
kfree(audio);
goto done;
}
audio->dec_id = decid & MSM_AUD_DECODER_MASK;
while (pmem_sz >= DMASZ_MIN) {
MM_DBG("pmemsz = %d\n", pmem_sz);
audio->phys = allocate_contiguous_ebi_nomap(pmem_sz, SZ_4K);
if (audio->phys) {
audio->map_v_write =
ioremap(audio->phys,
pmem_sz);
if (IS_ERR(audio->map_v_write)) {
MM_ERR("could not map write phys address, \
freeing instance 0x%08x\n",
(int)audio);
rc = -ENOMEM;
free_contiguous_memory_by_paddr(audio->phys);
audpp_adec_free(audio->dec_id);
kfree(audio);
goto done;
}
audio->data = (u8 *)audio->map_v_write;
MM_DBG("write buf: phy addr 0x%08x kernel addr \
0x%08x\n", audio->phys, (int)audio->data);
break;
} else if (pmem_sz == DMASZ_MIN) {
MM_ERR("could not allocate write buffers, freeing \
instance 0x%08x\n", (int)audio);
rc = -ENOMEM;
audpp_adec_free(audio->dec_id);
kfree(audio);
goto done;
} else
pmem_sz >>= 1;
}
audio->out_dma_sz = pmem_sz;
audio->read_phys = allocate_contiguous_ebi_nomap(PCM_BUFSZ_MIN
* PCM_BUF_MAX_COUNT, SZ_4K);
if (!audio->read_phys) {
MM_ERR("could not allocate read buffers, freeing instance \
0x%08x\n", (int)audio);
rc = -ENOMEM;
iounmap(audio->map_v_write);
free_contiguous_memory_by_paddr(audio->phys);
audpp_adec_free(audio->dec_id);
kfree(audio);
goto done;
}
audio->map_v_read = ioremap(audio->read_phys,
PCM_BUFSZ_MIN * PCM_BUF_MAX_COUNT);
if (IS_ERR(audio->map_v_read)) {
MM_ERR("could not map read phys address, freeing instance \
0x%08x\n", (int)audio);
rc = -ENOMEM;
iounmap(audio->map_v_write);
free_contiguous_memory_by_paddr(audio->phys);
free_contiguous_memory_by_paddr(audio->read_phys);
audpp_adec_free(audio->dec_id);
kfree(audio);
goto done;
}
audio->read_data = audio->map_v_read;
MM_DBG("read buf: phy addr 0x%08x kernel addr 0x%08x\n",
audio->read_phys, (int)audio->read_data);
rc = msm_adsp_get(audio->module_name, &audio->audplay,
&audplay_adsp_ops_aac, audio);
if (rc) {
MM_ERR("failed to get %s module, freeing instance 0x%08x\n",
audio->module_name, (int)audio);
goto err;
}
mutex_init(&audio->lock);
mutex_init(&audio->write_lock);
mutex_init(&audio->read_lock);
mutex_init(&audio->get_event_lock);
spin_lock_init(&audio->dsp_lock);
spin_lock_init(&audio->event_queue_lock);
INIT_LIST_HEAD(&audio->free_event_queue);
INIT_LIST_HEAD(&audio->event_queue);
init_waitqueue_head(&audio->write_wait);
init_waitqueue_head(&audio->read_wait);
init_waitqueue_head(&audio->wait);
init_waitqueue_head(&audio->event_wait);
init_waitqueue_head(&audio->avsync_wait);
audio->out[0].data = audio->data + 0;
audio->out[0].addr = audio->phys + 0;
audio->out[0].size = audio->out_dma_sz >> 1;
audio->out[1].data = audio->data + audio->out[0].size;
audio->out[1].addr = audio->phys + audio->out[0].size;
audio->out[1].size = audio->out[0].size;
audio->pcm_buf_count = PCM_BUF_MAX_COUNT;
for (index = 0; index < PCM_BUF_MAX_COUNT; index++) {
audio->in[index].data = audio->read_data + offset;
audio->in[index].addr = audio->read_phys + offset;
audio->in[index].size = PCM_BUFSZ_MIN;
audio->in[index].used = 0;
offset += PCM_BUFSZ_MIN;
}
audio->out_sample_rate = 44100;
audio->out_channel_mode = AUDPP_CMD_PCM_INTF_STEREO_V;
audio->aac_config.format = AUDIO_AAC_FORMAT_ADTS;
audio->aac_config.audio_object = AUDIO_AAC_OBJECT_LC;
audio->aac_config.ep_config = 0;
audio->aac_config.aac_section_data_resilience_flag =
AUDIO_AAC_SEC_DATA_RES_OFF;
audio->aac_config.aac_scalefactor_data_resilience_flag =
AUDIO_AAC_SCA_DATA_RES_OFF;
audio->aac_config.aac_spectral_data_resilience_flag =
AUDIO_AAC_SPEC_DATA_RES_OFF;
#ifdef CONFIG_AUDIO_AAC_PLUS
audio->aac_config.sbr_on_flag = AUDIO_AAC_SBR_ON_FLAG_ON;
#else
audio->aac_config.sbr_on_flag = AUDIO_AAC_SBR_ON_FLAG_OFF;
#endif
#ifdef CONFIG_AUDIO_ENHANCED_AAC_PLUS
audio->aac_config.sbr_ps_on_flag = AUDIO_AAC_SBR_PS_ON_FLAG_ON;
#else
audio->aac_config.sbr_ps_on_flag = AUDIO_AAC_SBR_PS_ON_FLAG_OFF;
#endif
audio->aac_config.dual_mono_mode = AUDIO_AAC_DUAL_MONO_PL_SR;
audio->aac_config.channel_configuration = 2;
audio->vol_pan.volume = 0x2000;
audio->bitstream_error_threshold_value =
BITSTREAM_ERROR_THRESHOLD_VALUE;
audio_flush(audio);
file->private_data = audio;
audio->opened = 1;
audio->device_events = AUDDEV_EVT_DEV_RDY
|AUDDEV_EVT_DEV_RLS|
AUDDEV_EVT_STREAM_VOL_CHG;
rc = auddev_register_evt_listner(audio->device_events,
AUDDEV_CLNT_DEC,
audio->dec_id,
aac_listner,
(void *)audio);
if (rc) {
MM_ERR("%s: failed to register listner\n", __func__);
goto event_err;
}
#ifdef CONFIG_DEBUG_FS
snprintf(name, sizeof name, "msm_aac_%04x", audio->dec_id);
audio->dentry = debugfs_create_file(name, S_IFREG | S_IRUGO,
NULL, (void *) audio,
&audaac_debug_fops);
if (IS_ERR(audio->dentry))
MM_DBG("debugfs_create_file failed\n");
#endif
#ifdef CONFIG_HAS_EARLYSUSPEND
audio->suspend_ctl.node.level = EARLY_SUSPEND_LEVEL_DISABLE_FB;
audio->suspend_ctl.node.resume = audaac_resume;
audio->suspend_ctl.node.suspend = audaac_suspend;
audio->suspend_ctl.audio = audio;
register_early_suspend(&audio->suspend_ctl.node);
#endif
for (index = 0; index < AUDAAC_EVENT_NUM; index++) {
e_node = kmalloc(sizeof(struct audaac_event), GFP_KERNEL);
if (e_node)
list_add_tail(&e_node->list, &audio->free_event_queue);
else {
MM_ERR("event pkt alloc failed\n");
break;
}
}
memset(&audio->stream_info, 0, sizeof(struct msm_audio_bitstream_info));
memset(&audio->bitstream_error_info, 0,
sizeof(struct msm_audio_bitstream_info));
done:
return rc;
event_err:
msm_adsp_put(audio->audplay);
err:
iounmap(audio->map_v_write);
free_contiguous_memory_by_paddr(audio->phys);
iounmap(audio->map_v_read);
free_contiguous_memory_by_paddr(audio->read_phys);
audpp_adec_free(audio->dec_id);
kfree(audio);
return rc;
}
static const struct file_operations audio_aac_fops = {
.owner = THIS_MODULE,
.open = audio_open,
.release = audio_release,
.read = audio_read,
.write = audio_write,
.unlocked_ioctl = audio_ioctl,
.fsync = audaac_fsync
};
struct miscdevice audio_aac_misc = {
.minor = MISC_DYNAMIC_MINOR,
.name = "msm_aac",
.fops = &audio_aac_fops,
};
static int __init audio_init(void)
{
return misc_register(&audio_aac_misc);
}
static void __exit audio_exit(void)
{
misc_deregister(&audio_aac_misc);
}
module_init(audio_init);
module_exit(audio_exit);
MODULE_DESCRIPTION("MSM AAC driver");
MODULE_LICENSE("GPL v2");