Robin Getz wrote:
> On Tue 26 Jun 2007 08:44, Kevin P. Fleming pondered:
>
>> This is no longer the policy; now that we require
prompts in 48KHz
>> signed linear WAV format, in English, Spanish and
French, we (Digium)
>> are now responsible for getting prompts recorded
when necessary.
>>
>
> Why 48kHz? It is unlikely that anything could play it
out at this rate.
> Forcing unnecessary sample rate conversion is just a
needless waste of MIPs.
>
Using a 48KHz sampling rate for the source recording makes
perfect
sense: you want the best sound quality for your source
recordings as
possible.
48000Hz also has the nice property that it's evenly
divisible to reach 8
and 16KHz:
48000 / 6 = 8000 - standard TDM sampling rate
48000 / 3 = 16000 - wideband sampling rate
Using 44.1KHz would require interpolation to reach 8KHz and
16KHz,
resulting in loss of audio quality (not all that much, by
why lose any
if you don't have to?).
As for efficiency, I'd presume the downsampling would be
done before the
audio is handed over to Asterisk.
Cheers,
Nic.
--
Nic Bellamy,
Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.c
om--
asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-dev
|
