Update of /usr/cvsroot/asterisk/channels
In directory mongoose.digium.com:/tmp/cvs-serv32412/channels
Modified Files:
chan_iax2.c
Log Message:
Add remainder of rtp fixes, iax2 patch (bug #3961)
Index: chan_iax2.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_iax2.c,v
retrieving revision 1.279
retrieving revision 1.280
diff -u -d -r1.279 -r1.280
--- chan_iax2.c 12 May 2005 19:01:03 -0000 1.279
+++ chan_iax2.c 14 May 2005 23:57:44 -0000 1.280
@@ -859,48 +859,6 @@
return tmp;
}
-static int get_samples(struct ast_frame *f)
-{
- int samples=0;
- switch(f->subclass) {
- case AST_FORMAT_SPEEX:
- samples = 160; /* XXX Not necessarily true XXX */
- break;
- case AST_FORMAT_G723_1:
- samples = 240 /* XXX Not necessarily true XXX */;
- break;
- case AST_FORMAT_ILBC:
- samples = 240 * (f->datalen / 50);
- break;
- case AST_FORMAT_GSM:
- samples = 160 * (f->datalen / 33);
- break;
- case AST_FORMAT_G729A:
- samples = 160 * (f->datalen / 20);
- break;
- case AST_FORMAT_SLINEAR:
- samples = f->datalen / 2;
- break;
- case AST_FORMAT_LPC10:
- samples = 22 * 8;
- samples += (((char *)(f->data))[7] & 0x1) * 8;
- break;
- case AST_FORMAT_ULAW:
- samples = f->datalen;
- break;
- case AST_FORMAT_ALAW:
- samples = f->datalen;
- break;
- case AST_FORMAT_ADPCM:
- case AST_FORMAT_G726:
- samples = f->datalen *2;
- break;
- default:
- ast_log(LOG_WARNING, "Don't know how to calculate samples on %d
packets\n", f->subclass);
- }
- return samples;
-}
-
static struct iax_frame *iaxfrdup2(struct iax_frame *fr)
{
/* Malloc() a copy of a frame */
@@ -2347,7 +2305,7 @@
if(fr->af.frametype == AST_FRAME_VOICE) {
type = JB_TYPE_VOICE;
- len = get_samples(&fr->af)/8;
+ len = ast_codec_get_samples(&fr->af) / 8;
} else if(fr->af.frametype == AST_FRAME_CNG) {
type = JB_TYPE_SILENCE;
}
@@ -6240,7 +6198,7 @@
f.mallocd = 0;
f.offset = 0;
if
(f.datalen && (f.frametype == AST_FRAME_VOICE))
-
f.samples = get_samples(&f);
+
f.samples = ast_codec_get_samples(&f);
else
f.samples = 0;
fr.outoforder = 0;
@@ -7355,7 +7313,7 @@
f.mallocd = 0;
f.offset = 0;
if (f.datalen && (f.frametype == AST_FRAME_VOICE)) {
- f.samples = get_samples(&f);
+ f.samples = ast_codec_get_samples(&f);
/* We need to byteswap incoming slinear samples from network
byte order */
if (f.subclass == AST_FORMAT_SLINEAR)
ast_frame_byteswap_be(&f);
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