[asterisk-bugs] [Asterisk 0010519]: add support for wideband speex (Openwengo variant)

noreply at bugs.digium.com noreply at bugs.digium.com
Tue Sep 4 18:02:00 CDT 2007


A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=10519 
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Reported By:                adamgundy
Assigned To:                
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Project:                    Asterisk
Issue ID:                   10519
Category:                   Codecs/codec_speex
Reproducibility:            always
Severity:                   feature
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.10.1  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             08-21-2007 17:08 CDT
Last Modified:              09-04-2007 18:02 CDT
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Summary:                    add support for wideband speex (Openwengo variant)
Description: 
the attached patch adds support for wideband speex to asterisk 1.4.10.1

it relies on the fact that the speex codec (in asterisk or the client)
will transparently handle conversion between wide and narrow band speex, so
there are no changes to the codec, we just register a new type. asterisk
will either (a) allow wideband packets to pass through client<->client
(sounds great!), or when actually talking to asterisk (voicemail, prompts
etc), it will send or decode as narrowband speex, so quality while talking
to asterisk itself is only narrowband (but of course this means minimal
changes since asterisk still deals with 8khz audio). this is all
transparent to the caller, apart from the lower quality while talking to
asterisk itself.

the only complex part of the patch is about teaching rtp.c and frame.c
about 16khz codecs, so that the RTP timestamps are set correctly.

openwengo, for some odd reason, insist on sending speex wideband as
'G726-64wb', with a different RTP type. this patch adds support for that.
it should be trivial to add support for 'speex/16000', for eg ekiga, but
that's another patch... and asterisk will have to 'transcode' between them,
because the RTP types don't match :-(
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---------------------------------------------------------------------- 
 qwell - 09-04-07 18:02  
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For the most part, this looks okay (except that ast_codec_get_rate returns
8000 for g722 :).

If you could figure out how to do this without taking up a new codec
number, that would be pretty awesome. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
09-04-07 18:02  qwell          Note Added: 0069943                          
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