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authordvossel <dvossel@f38db490-d61c-443f-a65b-d21fe96a405b>2009-04-09 19:29:58 +0000
committerdvossel <dvossel@f38db490-d61c-443f-a65b-d21fe96a405b>2009-04-09 19:29:58 +0000
commit85d639b7bb9ddfdf9bfbc84ee27ffa6a42e93dfa (patch)
tree0b9510c71988a5b9fac894a4844fc28f07174c72
parentc27a9c0345d5f276498f302295e6a378b67547f3 (diff)
Merged revisions 186379 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r186379 | dvossel | 2009-04-03 11:29:47 -0500 (Fri, 03 Apr 2009) | 6 lines audio_audiohook_write_list() did not correctly update sample size after ast_translate. audio_audiohook_write_list() did not take into account that the sample size may change after translation depending on if the original frame is is 8khz or 16khz. the sample size is now updated after translating to reflect this possibility. This caused the audio on the receiving end to sound terrible. Thanks to jcolp and mmichelson for helping me work this out. (issue AST-197) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@187546 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r--main/audiohook.c1
1 files changed, 1 insertions, 0 deletions
diff --git a/main/audiohook.c b/main/audiohook.c
index 640ec2f5a..baf370dc7 100644
--- a/main/audiohook.c
+++ b/main/audiohook.c
@@ -576,6 +576,7 @@ 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 */