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fixes logic error introduced by slin16 sip support

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
David Vossel
2010-06-21 20:33:41 +00:00
parent 63fd368411
commit 1a7e1aee5e
1 changed files with 2 additions and 1 deletions
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3
res/res_rtp_asterisk.c
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@@ -2230,8 +2230,9 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
if (rtp->f.subclass.codec & AST_FORMAT_AUDIO_MASK) {
rtp->f.samples = ast_codec_get_samples(&rtp->f);
if (rtp->f.subclass.codec == AST_FORMAT_SLINEAR || AST_FORMAT_SLINEAR16)
if ((rtp->f.subclass.codec == AST_FORMAT_SLINEAR) || (rtp->f.subclass.codec == AST_FORMAT_SLINEAR16)) {
ast_frame_byteswap_be(&rtp->f);
}
calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark);
/* Add timing data to let ast_generic_bridge() put the frame into a jitterbuf */
ast_set_flag(&rtp->f, AST_FRFLAG_HAS_TIMING_INFO);