format_mp3.c

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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
    <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]");