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authordvossel <dvossel@f38db490-d61c-443f-a65b-d21fe96a405b>2009-04-14 15:02:04 +0000
committerdvossel <dvossel@f38db490-d61c-443f-a65b-d21fe96a405b>2009-04-14 15:02:04 +0000
commit3f0f4524766212d1f67370eb615b93596b70423b (patch)
tree7ad21c9a574ecc68f27602be89430f17cb6a7062 /main
parent07edcabcb5ee2788b5012892e2d73401ef78f29c (diff)
audio_audiohook_write_list() does not correctly update sample size after ast_translate.
audio_audiohook_write_list() does not take into account that the sample size may change after translation depending on if the original frame is is 8khz or 16khz. While no 16kz codecs are supported in 1.4 at the moment, this will save headaches in the future if they ever are. the sample size is now updated after translating to reflect this possibility. Thanks to jcolp and mmichelson for helping me work this out. (issue AST-197) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@188287 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'main')
-rw-r--r--main/audiohook.c1
1 files changed, 1 insertions, 0 deletions
diff --git a/main/audiohook.c b/main/audiohook.c
index fdd241db2..d055a73f1 100644
--- a/main/audiohook.c
+++ b/main/audiohook.c
@@ -588,6 +588,7 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
}
if (!(middle_frame = ast_translate(in_translate->trans_pvt, frame, 0)))
return frame;
+ samples = middle_frame->samples;
}
/* Queue up signed linear frame to each spy */