format_mp3.c

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/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Anthony Minessale <[email protected]>
 *
 * Derived from other asterisk sound formats by
 * Mark Spencer <[email protected]>
 *
 * Thanks to mpglib from http://www.mpg123.org/
 * and Chris Stenton [[email protected]]
 * 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
 <defaultenabled>no</defaultenabled>
 <support_level>extended</support_level>
 ***/

#include "asterisk.h"

#include "mp3/mpg123.h"
#include "mp3/mpglib.h"

#include "asterisk/module.h"
#include "asterisk/mod_format.h"
#include "asterisk/logger.h"
#include "asterisk/format_cache.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 < MP3_SCACHE) {
 if (ferror(s->f)) {
 ast_log(LOG_WARNING, "Error while reading MP3 file: %s\n", 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;
 AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, p->buflen);
 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)
{
 mp3_f.format = ast_format_slin;
 InitMP3Constants();
 return ast_format_def_register(&mp3_f);
}

static int unload_module(void)
{
 return ast_format_def_unregister(name);
}

AST_MODULE_INFO_STANDARD_EXTENDED(ASTERISK_GPL_KEY, "MP3 format [Any rate but 8000hz mono is optimal]");