[asterisk-dev] adding codec+format

[Kevin P. Fleming]

On 03/23/2012 07:23 AM, Beñat Urteaga wrote:
> Hi,
>
> I’m using asterisk 1.8.8.4 and I’d like to add an audio codec and format
> in order to be able to send audio to my end SIP devices at a 8 bit 24
> kHz. The devices of course support it. How could I get this? I guess I
> should add codec_whatever.c and format_whatever.c files to the source
> code directories, but: must I add anything else? For example, may be a
> SDP negotiation may need to be made, so I should add something else
> apart from the codec and format files?
>
> I think format_pcm.c and codec_ulaw.c could be a good starting point
> (example), but do you recommend something different?
>
> Any help would be much appreciated!!

Search the source code for references to G722 (not case sensitive) and 
that will give you an idea of all the places you'll need to look at and 
possibly modify to support a new format.

That sounds like a very odd audio format though. Is it really 8-bit PCM 
with a 24kHz sample rate?

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
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