Fix bug where the samples were not accurate when in G723 mode, which would

cause the timestamp field of the RTP header to be invalid.



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@139155 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Shaun Ruffell
2008-08-20 20:12:38 +00:00
parent 76be7d91a9
commit 5a59e03452

View File

@@ -82,11 +82,8 @@ static AST_LIST_HEAD_STATIC(translators, translator);
struct pvt {
int fd;
int fake;
#ifdef DEBUG_TRANSCODE
int totalms;
int lasttotalms;
#endif
struct dahdi_transcoder_formats fmts;
int samples;
};
static char *handle_cli_transcoder_show(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
@@ -159,7 +156,7 @@ static struct ast_frame *dahdi_frameout(struct ast_trans_pvt *pvt)
return NULL;
}
} else {
pvt->f.samples = res;
pvt->f.samples = dahdip->samples;
pvt->f.datalen = res;
pvt->datalen = 0;
pvt->f.frametype = AST_FRAME_VOICE;
@@ -248,10 +245,14 @@ static int dahdi_translate(struct ast_trans_pvt *pvt, int dest, int source)
switch (dahdip->fmts.dstfmt) {
case AST_FORMAT_G729A:
dahdip->samples = 160;
break;
case AST_FORMAT_G723_1:
dahdip->samples = 240;
ast_atomic_fetchadd_int(&channels.encoders, +1);
break;
default:
dahdip->samples = 160;
ast_atomic_fetchadd_int(&channels.decoders, +1);
break;
}