On 08/30/2010 02:29 PM, Paul Albrecht wrote: > On Mon, 2010-08-30 at 14:15 -0500, Kevin P. Fleming wrote: >> 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. >> > > I don't know what's causing the problem, but the translated gsm and > speex frames claim 160 samples which is what what I got when I used > AST_FORMAT_SLINEAR. The g729 was truncated in half, that is, only 80 > samples, which is much worse than ulaw/alaw truncation.
What audio are you feeding in to these translators? It is sampled audio, all zeroes, all ones, something else? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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
