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The functions needed doesn't exist in Speex 1.05 which is what a lot of distros use.
1.2 seems to have been in beta status for years, and does include the sexy functions needed for func_speex to work.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@227237 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009) | 5 lines
Pay attention to the return value of the manipulate function.
While this looks like an optimization, it prevents a crash from occurring
when used with certain audiohook callbacks (diagnosed with SVN trunk,
backported to 1.4 to keep the source consistent across versions).
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@224856 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211539 f38db490-d61c-443f-a65b-d21fe96a405b
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This branch adds additional methods to dialplan functions, whereby the result
buffers are now dynamic buffers, which can be expanded to the size of any
result. No longer are variable substitutions limited to 4095 bytes of data.
In addition, the common case of needing buffers much smaller than that will
enable substitution to only take up the amount of memory actually needed.
The existing variable substitution routines are still available, but users
of those API calls should transition to using the dynamic-buffer APIs.
Reviewboard: http://reviewboard.digium.com/r/174/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@191140 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153468 f38db490-d61c-443f-a65b-d21fe96a405b
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This commit introduces the first phase of an effort to manage documentation of the
interfaces in Asterisk in an XML format. Currently, a new format is available for
applications and dialplan functions. A good number of conversions to the new format
are also included.
For more information, see the following message to asterisk-dev:
http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153365 f38db490-d61c-443f-a65b-d21fe96a405b
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channel related, and add the ability to add/find/remove datastores to manager sessions
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135680 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117828 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115850 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115593 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r115327 | file | 2008-05-05 19:10:05 -0300 (Mon, 05 May 2008) | 2 lines
Make sure that either the main speex library contains preprocess functions or that speexdsp does. If both fail then speex stuff can not be built.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115328 f38db490-d61c-443f-a65b-d21fe96a405b
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func_speex.c is based on contributions from Switchvox.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114977 f38db490-d61c-443f-a65b-d21fe96a405b
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and denoising to a channel, AGC() and DENOISE(). Also included, is a change
to the audiohook API to add a new function (ast_audiohook_remove) that can
remove an audiohook from a channel before it is detached.
This code is based on a contribution from Switchvox.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114926 f38db490-d61c-443f-a65b-d21fe96a405b
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