[asterisk-dev] g722 question

[Steve Underwood]

On 12/07/2009 11:59 PM, Saúl Ibarra wrote:
>> It looks like they have made no attempt to scale the signal properly.
>> The codec follows the G.722 spec in its handling of number scaling. That
>> means it takes in 14 bit linear samples, and output 14 bit linear
>> samples. A 16 bit signal needs to be shifted down 2 bits before
>> encoding, and shifted up 2 bits after decoding. 2 bits is 12 dB, which
>> is a pretty large change in volume.
>>
>>      
> They actually do that AFAIK, the problem then may be the other
> implementation, because it I don't shift those two bits audio is fine.
>    
Well, most phones are correct. Recent versions of Asterisk are correct. 
The trunk of Freeswitch is correct (although, due to a misunderstanding 
that was only fixed very recently). Anything which disagrees with those 
is wrong.

Steve