This patch fixes a potential crash in mp3_read() by not assuming that dbuf has enough data to finish filling up the output buffer. The patch also makes sure that the dbuf state gets reset after we know we read everything out of it already. In passing, this patch includes some other cleanups of this module, including stripping trailing whitespace, formatting fixes based on coding guidelines, and removing a number of unused members from the private state struct. (closes issue ASTERISK-19761) Reported by: Chris Maciejewsk Tested by: Chris Maciejewsk ........ Merged revisions 366296 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366297 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366298 65c4cc65-6c06-0410-ace0-fbb531ad65f3
332 lines
7.4 KiB
C
332 lines
7.4 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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/*! state for the mp3 decoder */
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struct mpstr mp;
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/*! buffer to hold mp3 data after read from disk */
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char sbuf[MP3_SCACHE];
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/*! buffer for slinear audio after being decoded out of sbuf */
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char dbuf[MP3_DCACHE];
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/*! how much data has been written to the output buffer in the ast_filestream */
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int buflen;
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/*! how much data has been written to sbuf */
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int sbuflen;
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/*! how much data is left to be read out of dbuf, starting at dbufoffset */
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int dbuflen;
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/*! current offset for reading data out of dbuf */
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int dbufoffset;
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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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res = 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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/* Pre-populate the buffer that holds audio to be returned (dbuf) */
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if (mp3_queue(s)) {
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return NULL;
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}
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if (p->dbuflen) {
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/* Read out what's waiting in dbuf */
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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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}
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p->dbufoffset += p->buflen;
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p->dbuflen -= p->buflen;
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}
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if (p->buflen < MP3_BUFLEN) {
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/* dbuf didn't have enough, so reset dbuf, fill it back up and continue */
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p->dbuflen = p->dbufoffset = 0;
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if (mp3_queue(s)) {
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return NULL;
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}
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/* Make sure dbuf has enough to complete this read attempt */
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if (p->dbuflen >= (MP3_BUFLEN - p->buflen)) {
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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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}
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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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