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use existing sample size in frame instead of recalculating it (issue #5390)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6781 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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2
rtp.c
2
rtp.c
@@ -1231,7 +1231,7 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
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ms = calc_txstamp(rtp, &f->delivery);
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/* Default prediction */
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if (f->subclass < AST_FORMAT_MAX_AUDIO) {
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pred = rtp->lastts + ast_codec_get_samples(f);
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pred = rtp->lastts + f->samples;
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/* Re-calculate last TS */
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rtp->lastts = rtp->lastts + ms * 8;
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