https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines Merged revisions 328209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines Introduce <support_level> tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
327 lines
6.9 KiB
C
327 lines
6.9 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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<support_level>extended</support_level>
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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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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 + AST_FRIENDLY_OFFSET] = 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 + AST_FRIENDLY_OFFSET] = 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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ast_format_set(&s->fr.subclass.format, AST_FORMAT_SLINEAR, 0);
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AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, p->buflen);
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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 fseek(s->f, offset, SEEK_SET);
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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 struct ast_format_def mp3_f = {
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.name = "mp3",
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.exts = "mp3",
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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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ast_format_set(&mp3_f.format, AST_FORMAT_SLINEAR, 0);
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InitMP3Constants();
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return ast_format_def_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_def_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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