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G.729 and G.723.1 codecs x86 (and x86_64) Linux and FreeBSD source code for Asterisk open source PBX

G.729 and G.723.1 codecs for Asterisk open source PBX.
http://asterisk.hosting.lv/
http://groups.google.com/group/asterisk-g729

To compile the codecs you need Intel IPP (Integrated Performance Primitives)
libraries installed. Tested against IPP 5.2 and 5.3.

Currently only Asterisk 1.4, 1.6 and TRUNK are supported. Use binaries from the
website for Asterisk 1.2 and Callweaver.

There are two ways to build the codecs:
1. Edit build.sh to select Asterisk version and preferred optimization parameters.
2. Use ./configure. Check the available options with ./configure --help.
   Specify --prefix in case Asterisk is installed in non-standard location.

G.723.1 send rate is configured in Asterisk codecs.conf file:
[g723]
; 6.3Kbps stream, default
sendrate=63
; 5.3Kbps
;sendrate=53
This option is for outgoing voice stream only. It does not affect incoming stream
that should be decoded automatically whatever the bitrate is.

There are also two Asterisk CLI commands "g723 debug" and "g729 debug" to print
statistics about received frames sizes. This can aid in debugging audio problems.
You need to bump Asterisk verbosity level to 3 to see the numbers.

Files:
- codec_g72x.c - GPL, code is based on code by Daniel Pocock at
  http://www.readytechnology.co.uk/open/ipp-codecs/
  and various Asterisk bundled codecs;
- build.sh - compile script;
- autotools files initially contributed by Michael E. Kromer
  michael.kromer at computergmbh dot de;
- g723_slin_ex.h, g729_slin_ex.h, slin_g72x_ex.h - sample speech data;
- all other C source files are copied from IPP samples, IPP license apply.

Before reporting any problem with the codecs, please read the website and make
sure you know what you're doing - compiling this codec is not a novice task.
Asking Asterisk G.729 Google group first is also good idea.

Author: Arkadi.Shishlov at gmail dot com