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| author | Russell Bryant <russell@russellbryant.com> | 2007-12-31 21:22:31 +0000 |
|---|---|---|
| committer | Russell Bryant <russell@russellbryant.com> | 2007-12-31 21:22:31 +0000 |
| commit | 21cb767db7d4854769d6a9d15e2008ab27c404a5 (patch) | |
| tree | 823393d4498b07581162c3e5266a494aa638a45c /CHANGES | |
| parent | f3e2f0bb0a0bed2960871087e3555b5acb239a78 (diff) | |
Merge changes from team/russell/codec_resample
This commit imports libresample for use in Asterisk. It also adds a new codec
module, codec_resample. This module uses libresample to re-sample signed linear
audio between 8 kHz and 16 kHz.
It also provides an alternative for converting between 16 kHz G.722 and 8 kHz
signed linear when using G.722, which will likely be useful as some people have
complained about volume issues when the current codec_g722 converts to 8 kHz
signed linear. But, to test this, you will have to disable the g722-to-slin and
g722-to-slin16 translators in codec_g722.c.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95501 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'CHANGES')
| -rw-r--r-- | CHANGES | 3 |
1 files changed, 3 insertions, 0 deletions
@@ -429,3 +429,6 @@ Miscellaneous * A new option when starting a remote asterisk (rasterisk, asterisk -r) for specifying which socket to use to connect to the running Asterisk daemon (-s) + * Added a new codec translation module, codec_resample, which re-samples + signed linear audio between 8 kHz and 16 kHz to help support wideband + codecs. |