1 files changed, 336 insertions, 0 deletions
diff --git a/addons/format_mp3.c b/addons/format_mp3.c
new file mode 100644
index 000000000..2c27243e2
--- /dev/null
+++ b/addons/format_mp3.c
@@ -0,0 +1,336 @@
+/*
+ * 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>
+ ***/
+
+#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]");