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Remove libresample from the Asterisk source tree. It is now available in its

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own repository, and must be installed like any other library for Asterisk to
use.  The two modules that require it are codec_resample and app_jack.

To install libresample:

$ svn co http://svn.digium.com/svn/libresample/trunk libresample
$ cd libresample
$ ./configure
$ make
$ sudo make install

This code is currently in our own repository because the build system did not
include the appropriate targets for building a dynamic library or for installing
the library.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132390 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Russell Bryant
2008-07-21 14:47:41 +00:00
parent 5de127e103
commit c87f901cfd
30 changed files with 331 additions and 10276 deletions
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build_tools/menuselect-deps.in
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@@ -30,6 +30,7 @@ PGSQL=@PBX_PGSQL@
POPT=@PBX_POPT@
PORTAUDIO=@PBX_PORTAUDIO@
PRI=@PBX_PRI@
RESAMPLE=@PBX_RESAMPLE@
AIS=@PBX_AIS@
RADIUS=@PBX_RADIUS@
SPANDSP=@PBX_SPANDSP@