[Asterisk-Dev] Handling AST_FORMAT_SLINEAR at 48000Hz

[Carlos Antunes]

On 11/17/05, voipwala hindustani <voipwala at gmail.com> wrote:
>
> But inside asterisk AST_FORMAT_SLINEAR is assumed to be of 8000Hz sampled
> voice. But in my case it is 48000Hz sampled.



You need to do a thing called decimation (or downsampling). The easiest way
is to make the 48kHz signal go through a first order low-pass filter with
cutoff frequency at 4kHz and then collect every 6th sample. You can then
assemble the new stream as signed lineat at 8kHz.

Carlos



--
"We hold [...] that all men are created equal; that they are
endowed [...] with certain inalienable rights; that among
these are life, liberty, and the pursuit of happiness"
-- Thomas Jefferson
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-dev/attachments/20051117/4abf80d2/attachment.htm