[Asterisk-Users] Sound file quality

[James Cloos]

>>>>> "David" == David Gurr <david.gurr at congruity.co.uk> writes:

David> As a result, I'd like to ensure that the voice prompts I'm
David> using have the best possible audio quality.

David> My callers will be coming in over PSTN to a VoIP gateway and
David> then to me by uLaw/aLaw ...

The optimal quality in the case where pstn is involved would be from:

    Using pro-quality (which these days does not necessarily mean
    pro $$$) recording equipment

    Recording in DAT quality (16 bit 48 kHz) 

        the recording equipment is more likely to support that
        than 16 kHz or 32 kHz

        32bit float rather than 16 bit int is ok, too

    Edit the files at this point for lead time, trail time,
    equal volume, et al.

    Use a high quality resampling algorithm (in sox use polyphase)
    to resample to signed-16bit 8 kHz.

        Optionally use a band-pass filter here to drop stuff
        outside of the PSTN frequenc range.  

    If you only do one of alaw/ulaw, you might as well convert
    the files to that, else leave them as signed-16bit

You can still get things like phase distortion if the path has jitter
and the receiver does not jitter-buffer.  You will also need to do
some experimenting to determine the optimal amplitude to avoid both
clipping and too-little use of the available u/a-law bandwidth.

-JimC
-- 
James H. Cloos, Jr. <cloos at jhcloos.com>