Randy R wrote: > I am working with several SIP projects that use g722, or are trying to > do so, with pjsip library. > > According to pjsip team's interpretation of g722, it works with 14bits > PCM for input/output, so pjsip basically 'converts' the audio sample > from 16 bits to 14 when encoding and vice-versa. Some implementations > don't do 16<->14 bits conversion, so when pjmedia talks to one of > those the over-driven audio problems appear. > > What we need to know is what's the most used implementation: 14<->16 > bits conversion or not. > > Any pointers to help clear this up? We'd really like to see more > g722-capable SIP clients for our own conference on ZipDX.
As far as I am aware, for ITU-T compliance the codec only cares about 14 significant bits, but the reference source code needs those 14 bits in the *top* 14 bits of each 16-bit word that it supplies/produces. The Asterisk implementation does not do any bit-shifting or masking at all, and seems to interoperate with quite a few endpoints just fine, so presumably that means it's the correct implementation :-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: [email protected] Check us out at www.digium.com & www.asterisk.org _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
