[asterisk-dev] adding codec+format

[Beñat Urteaga]

>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?

Hi again! I know it's an odd audio format but it's the best quality we can get with our own devices.
However, I've been looking at the codec_g722.c, g722.h, g722_encode.c and g722_decode.c files and I'm still quite lost... :S
Should I leave all the values in coder-decoder parameters (q6[32], iln[32], ilp[32], wl[8]...) as they are? Where is the place where I can change the samplerate to 24 kHz?

Sorry but I'm not an expert programmer and there're some things that I can't get to understand...

Thank you!







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