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    21cb767d
    Merge changes from team/russell/codec_resample · 21cb767d
    Russell Bryant authored
    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
    21cb767d
    History
    Merge changes from team/russell/codec_resample
    Russell Bryant authored
    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