[asterisk-dev] Thoughts on the messages dealing with temporary

[Nic Bellamy]

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/