337 lines
7.0 KiB
C
337 lines
7.0 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Anthony Minessale <anthmct@yahoo.com>
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*
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* Derived from other asterisk sound formats by
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* Mark Spencer <markster@linux-support.net>
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*
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* Thanks to mpglib from http://www.mpg123.org/
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* and Chris Stenton [jacs@gnome.co.uk]
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* for coding the ability to play stereo and non-8khz files
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*!
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* \file
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* \brief MP3 Format Handler
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* \ingroup formats
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*/
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/*** MODULEINFO
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<defaultenabled>no</defaultenabled>
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***/
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include "mp3/mpg123.h"
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#include "mp3/mpglib.h"
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#include "asterisk/module.h"
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#include "asterisk/mod_format.h"
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#include "asterisk/logger.h"
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#define MP3_BUFLEN 320
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#define MP3_SCACHE 16384
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#define MP3_DCACHE 8192
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struct mp3_private {
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char waste[AST_FRIENDLY_OFFSET]; /* Buffer for sending frames, etc */
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char empty; /* Empty character */
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int lasttimeout;
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int maxlen;
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struct timeval last;
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struct mpstr mp;
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char sbuf[MP3_SCACHE];
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char dbuf[MP3_DCACHE];
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int buflen;
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int sbuflen;
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int dbuflen;
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int dbufoffset;
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int sbufoffset;
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int lastseek;
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int offset;
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long seek;
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};
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static const char name[] = "mp3";
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#define BLOCKSIZE 160
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#define OUTSCALE 4096
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#define GAIN -4 /* 2^GAIN is the multiple to increase the volume by */
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#if __BYTE_ORDER == __LITTLE_ENDIAN
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#define htoll(b) (b)
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#define htols(b) (b)
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#define ltohl(b) (b)
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#define ltohs(b) (b)
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#else
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#if __BYTE_ORDER == __BIG_ENDIAN
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#define htoll(b) \
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(((((b) ) & 0xFF) << 24) | \
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((((b) >> 8) & 0xFF) << 16) | \
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((((b) >> 16) & 0xFF) << 8) | \
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((((b) >> 24) & 0xFF) ))
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#define htols(b) \
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(((((b) ) & 0xFF) << 8) | \
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((((b) >> 8) & 0xFF) ))
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#define ltohl(b) htoll(b)
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#define ltohs(b) htols(b)
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#else
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#error "Endianess not defined"
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#endif
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#endif
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static int mp3_open(struct ast_filestream *s)
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{
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struct mp3_private *p = s->_private;
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InitMP3(&p->mp, OUTSCALE);
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p->dbuflen = 0;
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s->fr.data.ptr = s->buf;
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s->fr.frametype = AST_FRAME_VOICE;
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s->fr.subclass = AST_FORMAT_SLINEAR;
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/* datalen will vary for each frame */
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s->fr.src = name;
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s->fr.mallocd = 0;
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p->offset = 0;
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return 0;
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}
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static void mp3_close(struct ast_filestream *s)
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{
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struct mp3_private *p = s->_private;
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ExitMP3(&p->mp);
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return;
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}
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static int mp3_squeue(struct ast_filestream *s)
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{
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struct mp3_private *p = s->_private;
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int res=0;
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p->lastseek = ftell(s->f);
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p->sbuflen = fread(p->sbuf, 1, MP3_SCACHE, s->f);
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if(p->sbuflen < 0) {
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ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", p->sbuflen, strerror(errno));
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return -1;
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}
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res = decodeMP3(&p->mp,p->sbuf,p->sbuflen,p->dbuf,MP3_DCACHE,&p->dbuflen);
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if(res != MP3_OK)
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return -1;
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p->sbuflen -= p->dbuflen;
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p->dbufoffset = 0;
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return 0;
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}
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static int mp3_dqueue(struct ast_filestream *s)
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{
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struct mp3_private *p = s->_private;
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int res=0;
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if((res = decodeMP3(&p->mp,NULL,0,p->dbuf,MP3_DCACHE,&p->dbuflen)) == MP3_OK) {
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p->sbuflen -= p->dbuflen;
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p->dbufoffset = 0;
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}
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return res;
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}
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static int mp3_queue(struct ast_filestream *s)
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{
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struct mp3_private *p = s->_private;
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int res = 0, bytes = 0;
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if(p->seek) {
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ExitMP3(&p->mp);
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InitMP3(&p->mp, OUTSCALE);
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fseek(s->f, 0, SEEK_SET);
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p->sbuflen = p->dbuflen = p->offset = 0;
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while(p->offset < p->seek) {
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if(mp3_squeue(s))
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return -1;
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while(p->offset < p->seek && ((res = mp3_dqueue(s))) == MP3_OK) {
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for(bytes = 0 ; bytes < p->dbuflen ; bytes++) {
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p->dbufoffset++;
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p->offset++;
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if(p->offset >= p->seek)
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break;
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}
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}
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if(res == MP3_ERR)
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return -1;
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}
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p->seek = 0;
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return 0;
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}
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if(p->dbuflen == 0) {
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if(p->sbuflen) {
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res = mp3_dqueue(s);
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if(res == MP3_ERR)
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return -1;
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}
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if(! p->sbuflen || res != MP3_OK) {
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if(mp3_squeue(s))
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return -1;
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}
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}
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return 0;
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}
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static struct ast_frame *mp3_read(struct ast_filestream *s, int *whennext)
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{
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struct mp3_private *p = s->_private;
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int delay =0;
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int save=0;
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/* Send a frame from the file to the appropriate channel */
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if(mp3_queue(s))
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return NULL;
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if(p->dbuflen) {
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for(p->buflen=0; p->buflen < MP3_BUFLEN && p->buflen < p->dbuflen; p->buflen++) {
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s->buf[p->buflen] = p->dbuf[p->buflen+p->dbufoffset];
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p->sbufoffset++;
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}
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p->dbufoffset += p->buflen;
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p->dbuflen -= p->buflen;
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if(p->buflen < MP3_BUFLEN) {
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if(mp3_queue(s))
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return NULL;
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for(save = p->buflen; p->buflen < MP3_BUFLEN; p->buflen++) {
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s->buf[p->buflen] = p->dbuf[(p->buflen-save)+p->dbufoffset];
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p->sbufoffset++;
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}
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p->dbufoffset += (MP3_BUFLEN - save);
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p->dbuflen -= (MP3_BUFLEN - save);
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}
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}
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p->offset += p->buflen;
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delay = p->buflen/2;
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s->fr.frametype = AST_FRAME_VOICE;
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s->fr.subclass = AST_FORMAT_SLINEAR;
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s->fr.offset = AST_FRIENDLY_OFFSET;
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s->fr.datalen = p->buflen;
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s->fr.data.ptr = s->buf;
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s->fr.mallocd = 0;
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s->fr.samples = delay;
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*whennext = delay;
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return &s->fr;
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}
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static int mp3_write(struct ast_filestream *fs, struct ast_frame *f)
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{
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ast_log(LOG_ERROR,"I Can't write MP3 only read them.\n");
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return -1;
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}
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static int mp3_seek(struct ast_filestream *s, off_t sample_offset, int whence)
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{
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struct mp3_private *p = s->_private;
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off_t min,max,cur;
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long offset=0,samples;
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samples = sample_offset * 2;
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min = 0;
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fseek(s->f, 0, SEEK_END);
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max = ftell(s->f) * 100;
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cur = p->offset;
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if (whence == SEEK_SET)
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offset = samples + min;
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else if (whence == SEEK_CUR || whence == SEEK_FORCECUR)
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offset = samples + cur;
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else if (whence == SEEK_END)
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offset = max - samples;
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if (whence != SEEK_FORCECUR) {
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offset = (offset > max)?max:offset;
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}
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p->seek = offset;
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return p->seek;
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}
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static int mp3_rewrite(struct ast_filestream *s, const char *comment)
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{
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ast_log(LOG_ERROR,"I Can't write MP3 only read them.\n");
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return -1;
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}
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static int mp3_trunc(struct ast_filestream *s)
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{
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ast_log(LOG_ERROR,"I Can't write MP3 only read them.\n");
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return -1;
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}
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static off_t mp3_tell(struct ast_filestream *s)
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{
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struct mp3_private *p = s->_private;
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return p->offset/2;
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}
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static char *mp3_getcomment(struct ast_filestream *s)
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{
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return NULL;
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}
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static const struct ast_format mp3_f = {
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.name = "mp3",
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.exts = "mp3",
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.format = AST_FORMAT_SLINEAR,
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.open = mp3_open,
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.write = mp3_write,
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.rewrite = mp3_rewrite,
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.seek = mp3_seek,
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.trunc = mp3_trunc,
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.tell = mp3_tell,
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.read = mp3_read,
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.close = mp3_close,
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.getcomment = mp3_getcomment,
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.buf_size = MP3_BUFLEN + AST_FRIENDLY_OFFSET,
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.desc_size = sizeof(struct mp3_private),
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};
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static int load_module(void)
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{
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InitMP3Constants();
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return ast_format_register(&mp3_f);
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}
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static int unload_module(void)
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{
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return ast_format_unregister(name);
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}
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AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "MP3 format [Any rate but 8000hz mono is optimal]");
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