[asterisk-dev] downsampling slinear16 to ulaw (or alaw or g729)

Paul Albrecht albrecht at glccom.com
Mon Aug 30 14:29:41 CDT 2010


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.

> -- 
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> skype: kpfleming | jabber: kfleming at digium.com
> Check us out at www.digium.com & www.asterisk.org
> 
-- 
Paul Albrecht




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