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/*
* Asterisk -- An open source telephony toolkit.
*
* Anthony Minessale <anthmct@yahoo.com>
*
* Derived from other asterisk sound formats by
* Mark Spencer <markster@linux-support.net>
*
* Thanks to mpglib from http://www.mpg123.org/
* and Chris Stenton [jacs@gnome.co.uk]
* for coding the ability to play stereo and non-8khz files
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*!
* \file
* \brief MP3 Format Handler
* \ingroup formats
*/
/*** MODULEINFO
<support_level>extended</support_level>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision: 366298 $")
#include "mp3/mpg123.h"
#include "mp3/mpglib.h"
#include "asterisk/module.h"
#include "asterisk/mod_format.h"
#include "asterisk/logger.h"
#define MP3_BUFLEN 320
#define MP3_SCACHE 16384
#define MP3_DCACHE 8192
struct mp3_private {
/*! state for the mp3 decoder */
struct mpstr mp;
/*! buffer to hold mp3 data after read from disk */
char sbuf[MP3_SCACHE];
/*! buffer for slinear audio after being decoded out of sbuf */
char dbuf[MP3_DCACHE];
/*! how much data has been written to the output buffer in the ast_filestream */
int buflen;
/*! how much data has been written to sbuf */
int sbuflen;
/*! how much data is left to be read out of dbuf, starting at dbufoffset */
int dbuflen;
/*! current offset for reading data out of dbuf */
int dbufoffset;
int offset;
long seek;
};
static const char name[] = "mp3";
#define BLOCKSIZE 160
#define OUTSCALE 4096
#define GAIN -4 /* 2^GAIN is the multiple to increase the volume by */
#if __BYTE_ORDER == __LITTLE_ENDIAN
#define htoll(b) (b)
#define htols(b) (b)
#define ltohl(b) (b)
#define ltohs(b) (b)
#else
#if __BYTE_ORDER == __BIG_ENDIAN
#define htoll(b) \
(((((b) ) & 0xFF) << 24) | \
((((b) >> 8) & 0xFF) << 16) | \
((((b) >> 16) & 0xFF) << 8) | \
((((b) >> 24) & 0xFF) ))
#define htols(b) \
(((((b) ) & 0xFF) << 8) | \
((((b) >> 8) & 0xFF) ))
#define ltohl(b) htoll(b)
#define ltohs(b) htols(b)
#else
#error "Endianess not defined"
#endif
#endif
static int mp3_open(struct ast_filestream *s)
{
struct mp3_private *p = s->_private;
InitMP3(&p->mp, OUTSCALE);
return 0;
}
static void mp3_close(struct ast_filestream *s)
{
struct mp3_private *p = s->_private;
ExitMP3(&p->mp);
return;
}
static int mp3_squeue(struct ast_filestream *s)
{
struct mp3_private *p = s->_private;
int res=0;
res = ftell(s->f);
p->sbuflen = fread(p->sbuf, 1, MP3_SCACHE, s->f);
if(p->sbuflen < 0) {
ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", p->sbuflen, strerror(errno));
return -1;
}
res = decodeMP3(&p->mp,p->sbuf,p->sbuflen,p->dbuf,MP3_DCACHE,&p->dbuflen);
if(res != MP3_OK)
return -1;
p->sbuflen -= p->dbuflen;
p->dbufoffset = 0;
return 0;
}
static int mp3_dqueue(struct ast_filestream *s)
{
struct mp3_private *p = s->_private;
int res=0;
if((res = decodeMP3(&p->mp,NULL,0,p->dbuf,MP3_DCACHE,&p->dbuflen)) == MP3_OK) {
p->sbuflen -= p->dbuflen;
p->dbufoffset = 0;
}
return res;
}
static int mp3_queue(struct ast_filestream *s)
{
struct mp3_private *p = s->_private;
int res = 0, bytes = 0;
if(p->seek) {
ExitMP3(&p->mp);
InitMP3(&p->mp, OUTSCALE);
fseek(s->f, 0, SEEK_SET);
p->sbuflen = p->dbuflen = p->offset = 0;
while(p->offset < p->seek) {
if(mp3_squeue(s))
return -1;
while(p->offset < p->seek && ((res = mp3_dqueue(s))) == MP3_OK) {
for(bytes = 0 ; bytes < p->dbuflen ; bytes++) {
p->dbufoffset++;
p->offset++;
if(p->offset >= p->seek)
break;
}
}
if(res == MP3_ERR)
return -1;
}
p->seek = 0;
return 0;
}
if(p->dbuflen == 0) {
if(p->sbuflen) {
res = mp3_dqueue(s);
if(res == MP3_ERR)
return -1;
}
if(! p->sbuflen || res != MP3_OK) {
if(mp3_squeue(s))
return -1;
}
}
return 0;
}
static struct ast_frame *mp3_read(struct ast_filestream *s, int *whennext)
{
struct mp3_private *p = s->_private;
int delay =0;
int save=0;
/* Pre-populate the buffer that holds audio to be returned (dbuf) */
if (mp3_queue(s)) {
return NULL;
}
if (p->dbuflen) {
/* Read out what's waiting in dbuf */
for (p->buflen = 0; p->buflen < MP3_BUFLEN && p->buflen < p->dbuflen; p->buflen++) {
s->buf[p->buflen + AST_FRIENDLY_OFFSET] = p->dbuf[p->buflen + p->dbufoffset];
}
p->dbufoffset += p->buflen;
p->dbuflen -= p->buflen;
}
if (p->buflen < MP3_BUFLEN) {
/* dbuf didn't have enough, so reset dbuf, fill it back up and continue */
p->dbuflen = p->dbufoffset = 0;
if (mp3_queue(s)) {
return NULL;
}
/* Make sure dbuf has enough to complete this read attempt */
if (p->dbuflen >= (MP3_BUFLEN - p->buflen)) {
for (save = p->buflen; p->buflen < MP3_BUFLEN; p->buflen++) {
s->buf[p->buflen + AST_FRIENDLY_OFFSET] = p->dbuf[(p->buflen - save) + p->dbufoffset];
}
p->dbufoffset += (MP3_BUFLEN - save);
p->dbuflen -= (MP3_BUFLEN - save);
}
}
p->offset += p->buflen;
delay = p->buflen / 2;
s->fr.frametype = AST_FRAME_VOICE;
ast_format_set(&s->fr.subclass.format, AST_FORMAT_SLINEAR, 0);
AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, p->buflen);
s->fr.mallocd = 0;
s->fr.samples = delay;
*whennext = delay;
return &s->fr;
}
static int mp3_write(struct ast_filestream *fs, struct ast_frame *f)
{
ast_log(LOG_ERROR,"I Can't write MP3 only read them.\n");
return -1;
}
static int mp3_seek(struct ast_filestream *s, off_t sample_offset, int whence)
{
struct mp3_private *p = s->_private;
off_t min,max,cur;
long offset=0,samples;
samples = sample_offset * 2;
min = 0;
fseek(s->f, 0, SEEK_END);
max = ftell(s->f) * 100;
cur = p->offset;
if (whence == SEEK_SET)
offset = samples + min;
else if (whence == SEEK_CUR || whence == SEEK_FORCECUR)
offset = samples + cur;
else if (whence == SEEK_END)
offset = max - samples;
if (whence != SEEK_FORCECUR) {
offset = (offset > max)?max:offset;
}
p->seek = offset;
return fseek(s->f, offset, SEEK_SET);
}
static int mp3_rewrite(struct ast_filestream *s, const char *comment)
{
ast_log(LOG_ERROR,"I Can't write MP3 only read them.\n");
return -1;
}
static int mp3_trunc(struct ast_filestream *s)
{
ast_log(LOG_ERROR,"I Can't write MP3 only read them.\n");
return -1;
}
static off_t mp3_tell(struct ast_filestream *s)
{
struct mp3_private *p = s->_private;
return p->offset/2;
}
static char *mp3_getcomment(struct ast_filestream *s)
{
return NULL;
}
static struct ast_format_def mp3_f = {
.name = "mp3",
.exts = "mp3",
.open = mp3_open,
.write = mp3_write,
.rewrite = mp3_rewrite,
.seek = mp3_seek,
.trunc = mp3_trunc,
.tell = mp3_tell,
.read = mp3_read,
.close = mp3_close,
.getcomment = mp3_getcomment,
.buf_size = MP3_BUFLEN + AST_FRIENDLY_OFFSET,
.desc_size = sizeof(struct mp3_private),
};
static int load_module(void)
{
ast_format_set(&mp3_f.format, AST_FORMAT_SLINEAR, 0);
InitMP3Constants();
return ast_format_def_register(&mp3_f);
}
static int unload_module(void)
{
return ast_format_def_unregister(name);
}
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "MP3 format [Any rate but 8000hz mono is optimal]");
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