[svn-commits] russell: branch group/addons-merge r204398 - /team/group/addons-merge/addons/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Jun 30 09:47:30 CDT 2009
Author: russell
Date: Tue Jun 30 09:47:27 2009
New Revision: 204398
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=204398
Log:
actually add format_mp3.c, set svn:ignore on addons/
Added:
team/group/addons-merge/addons/format_mp3.c (with props)
Modified:
team/group/addons-merge/addons/ (props changed)
Propchange: team/group/addons-merge/addons/
------------------------------------------------------------------------------
--- svn:ignore (added)
+++ svn:ignore Tue Jun 30 09:47:27 2009
@@ -1,0 +1,11 @@
+*.a
+*.d
+*.eo
+*.eoo
+*.i
+*.makeopts
+*.moduleinfo
+*.s
+*.so
+modules.link
+
Added: team/group/addons-merge/addons/format_mp3.c
URL: http://svn.asterisk.org/svn-view/asterisk/team/group/addons-merge/addons/format_mp3.c?view=auto&rev=204398
==============================================================================
--- team/group/addons-merge/addons/format_mp3.c (added)
+++ team/group/addons-merge/addons/format_mp3.c Tue Jun 30 09:47:27 2009
@@ -1,0 +1,336 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Anthony Minessale <anthmct at yahoo.com>
+ *
+ * Derived from other asterisk sound formats by
+ * Mark Spencer <markster at linux-support.net>
+ *
+ * Thanks to mpglib from http://www.mpg123.org/
+ * and Chris Stenton [jacs at 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>
+ ***/
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "mp3/mpg123.h"
+#include "mp3/mpglib.h"
+
+#include "asterisk/module.h"
+#include "asterisk/mod_format.h"
+#include "asterisk/logger.h"
+
+#define MP3_BUFLEN 320
+#define MP3_SCACHE 16384
+#define MP3_DCACHE 8192
+
+struct mp3_private {
+ char waste[AST_FRIENDLY_OFFSET]; /* Buffer for sending frames, etc */
+ char empty; /* Empty character */
+ int lasttimeout;
+ int maxlen;
+ struct timeval last;
+ struct mpstr mp;
+ char sbuf[MP3_SCACHE];
+ char dbuf[MP3_DCACHE];
+ int buflen;
+ int sbuflen;
+ int dbuflen;
+ int dbufoffset;
+ int sbufoffset;
+ int lastseek;
+ 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);
+ p->dbuflen = 0;
+ s->fr.data.ptr = s->buf;
+ s->fr.frametype = AST_FRAME_VOICE;
+ s->fr.subclass = AST_FORMAT_SLINEAR;
+ /* datalen will vary for each frame */
+ s->fr.src = name;
+ s->fr.mallocd = 0;
+ p->offset = 0;
+ 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;
+
+ p->lastseek = ftell(s->f);
+ p->sbuflen = fread(p->sbuf, 1, MP3_SCACHE, s->f);
+ if(p->sbuflen < 0) {
+ ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", p->sbuflen, 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;
+
+ /* Send a frame from the file to the appropriate channel */
+
+ if(mp3_queue(s))
+ return NULL;
+
+ if(p->dbuflen) {
+ for(p->buflen=0; p->buflen < MP3_BUFLEN && p->buflen < p->dbuflen; p->buflen++) {
+ s->buf[p->buflen] = p->dbuf[p->buflen+p->dbufoffset];
+ p->sbufoffset++;
+ }
+ p->dbufoffset += p->buflen;
+ p->dbuflen -= p->buflen;
+
+ if(p->buflen < MP3_BUFLEN) {
+ if(mp3_queue(s))
+ return NULL;
+
+ for(save = p->buflen; p->buflen < MP3_BUFLEN; p->buflen++) {
+ s->buf[p->buflen] = p->dbuf[(p->buflen-save)+p->dbufoffset];
+ p->sbufoffset++;
+ }
+ p->dbufoffset += (MP3_BUFLEN - save);
+ p->dbuflen -= (MP3_BUFLEN - save);
+
+ }
+
+ }
+
+ p->offset += p->buflen;
+ delay = p->buflen/2;
+ s->fr.frametype = AST_FRAME_VOICE;
+ s->fr.subclass = AST_FORMAT_SLINEAR;
+ s->fr.offset = AST_FRIENDLY_OFFSET;
+ s->fr.datalen = p->buflen;
+ s->fr.data.ptr = s->buf;
+ s->fr.mallocd = 0;
+ 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 p->seek;
+
+}
+
+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 const struct ast_format mp3_f = {
+ .name = "mp3",
+ .exts = "mp3",
+ .format = AST_FORMAT_SLINEAR,
+ .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)
+{
+ InitMP3Constants();
+ return ast_format_register(&mp3_f);
+}
+
+static int unload_module(void)
+{
+ return ast_format_unregister(name);
+}
+
+AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "MP3 format [Any rate but 8000hz mono is optimal]");
Propchange: team/group/addons-merge/addons/format_mp3.c
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: team/group/addons-merge/addons/format_mp3.c
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: team/group/addons-merge/addons/format_mp3.c
------------------------------------------------------------------------------
svn:mime-type = text/plain
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