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authordvossel <dvossel@f38db490-d61c-443f-a65b-d21fe96a405b>2009-04-03 16:36:44 +0000
committerdvossel <dvossel@f38db490-d61c-443f-a65b-d21fe96a405b>2009-04-03 16:36:44 +0000
commitdf807f1188cb267ad3a67fb482880892ad44214f (patch)
tree9bc6cb186a8cacfc0f398de6980b3669c1fa5106
parent11d70c6fb22fd77993e93a493b95aadd68704f34 (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) | 4 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/1.6.0@186380 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 184a29c34..13aacef2c 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 */