On 08/30/2010 01:48 PM, Paul Albrecht wrote: > As for AST_FORMAT_SLINEAR16 to AST_FORMAT_SLINEAR translation, I get > truncation, that is, instead of the 160 samples I was expecting I get > 137 samples. I guess I don't know how to interpret these results, if > slinear16/slinear results in truncation that's a bug, right?
Yes. That particular transcoding step is just resampling, and it should produce exactly half as many samples as were input (unless an odd number were input, of course). > One more thing to mention, I have translated my silent frame to some > other codecs from wide slinear without truncation. They are gsm, speex, > and g722. Of course g722 is wide so that's not surprising, but I don't > think gsm is wide and it is not truncated. That's somewhat illogical; all paths to 8Khz codecs should go through the same resampling step first, then into the codec. If there are samples being dropped during resampling, it should occur for all of them. -- 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-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-security mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-security
