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

[noreply at bugs.digium.com]

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:                     feedback
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:              11-06-2007 17:14 CST
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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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---------------------------------------------------------------------- 
 dimas - 11-06-07 17:14  
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To me it looks like significant part of your patch is not related to new
codec at all - I'm talking about introduction of AST_FORMAT_AUDIO_MASK and
use of it instead of AST_FORMAT_MAX_AUDIO. While I agree it is good thing
to do in general, I'm not sure it is a good idea to mix such a "code
cleanup" changes with actually new features like addition of new codec in
the same svn commit... 

Issue History 
Date Modified   Username       Field                    Change               
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11-06-07 17:14  dimas          Note Added: 0073256                          
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